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authorTakashi Iwai <tiwai@suse.de>2015-11-02 09:00:37 +0100
committerTakashi Iwai <tiwai@suse.de>2015-11-02 09:00:37 +0100
commitbc88c9e923819b02f8cfaec1bb80d7e5530c0ac5 (patch)
tree1731a322d153ee18017822a8fd66295a0ec7a4e2 /sound
parentcadd16ea33a938d49aee99edd4758cc76048b399 (diff)
parent2c8d08925272b9be383cd81f1973c6faf9fbf53a (diff)
downloadlinux-sh-bc88c9e923819b02f8cfaec1bb80d7e5530c0ac5.tar.gz
Merge branch 'for-next' into for-linus
Diffstat (limited to 'sound')
-rw-r--r--sound/arm/pxa2xx-ac97.c13
-rw-r--r--sound/arm/pxa2xx-pcm-lib.c201
-rw-r--r--sound/arm/pxa2xx-pcm.c12
-rw-r--r--sound/arm/pxa2xx-pcm.h2
-rw-r--r--sound/core/Kconfig13
-rw-r--r--sound/core/Makefile3
-rw-r--r--sound/core/oss/mixer_oss.c3
-rw-r--r--sound/core/pcm.c3
-rw-r--r--sound/core/pcm_lib.c24
-rw-r--r--sound/core/pcm_native.c42
-rw-r--r--sound/core/seq/oss/seq_oss_readq.c6
-rw-r--r--sound/core/seq/oss/seq_oss_writeq.c4
-rw-r--r--sound/firewire/Kconfig27
-rw-r--r--sound/firewire/Makefile5
-rw-r--r--sound/firewire/amdtp-am824.c465
-rw-r--r--sound/firewire/amdtp-am824.h52
-rw-r--r--sound/firewire/amdtp-stream.c (renamed from sound/firewire/amdtp.c)379
-rw-r--r--sound/firewire/amdtp-stream.h (renamed from sound/firewire/amdtp.h)116
-rw-r--r--sound/firewire/bebob/Makefile2
-rw-r--r--sound/firewire/bebob/bebob.c9
-rw-r--r--sound/firewire/bebob/bebob.h34
-rw-r--r--sound/firewire/bebob/bebob_focusrite.c26
-rw-r--r--sound/firewire/bebob/bebob_maudio.c34
-rw-r--r--sound/firewire/bebob/bebob_midi.c16
-rw-r--r--sound/firewire/bebob/bebob_pcm.c16
-rw-r--r--sound/firewire/bebob/bebob_proc.c6
-rw-r--r--sound/firewire/bebob/bebob_stream.c40
-rw-r--r--sound/firewire/bebob/bebob_terratec.c10
-rw-r--r--sound/firewire/bebob/bebob_yamaha.c6
-rw-r--r--sound/firewire/dice/Makefile2
-rw-r--r--sound/firewire/dice/dice-midi.c12
-rw-r--r--sound/firewire/dice/dice-pcm.c12
-rw-r--r--sound/firewire/dice/dice-stream.c34
-rw-r--r--sound/firewire/dice/dice.c3
-rw-r--r--sound/firewire/dice/dice.h2
-rw-r--r--sound/firewire/digi00x/Makefile4
-rw-r--r--sound/firewire/digi00x/amdtp-dot.c442
-rw-r--r--sound/firewire/digi00x/digi00x-hwdep.c200
-rw-r--r--sound/firewire/digi00x/digi00x-midi.c223
-rw-r--r--sound/firewire/digi00x/digi00x-pcm.c373
-rw-r--r--sound/firewire/digi00x/digi00x-proc.c99
-rw-r--r--sound/firewire/digi00x/digi00x-stream.c422
-rw-r--r--sound/firewire/digi00x/digi00x-transaction.c137
-rw-r--r--sound/firewire/digi00x/digi00x.c170
-rw-r--r--sound/firewire/digi00x/digi00x.h157
-rw-r--r--sound/firewire/fcp.c2
-rw-r--r--sound/firewire/fireworks/Makefile2
-rw-r--r--sound/firewire/fireworks/fireworks.c12
-rw-r--r--sound/firewire/fireworks/fireworks.h2
-rw-r--r--sound/firewire/fireworks/fireworks_command.c2
-rw-r--r--sound/firewire/fireworks/fireworks_midi.c12
-rw-r--r--sound/firewire/fireworks/fireworks_pcm.c12
-rw-r--r--sound/firewire/fireworks/fireworks_stream.c8
-rw-r--r--sound/firewire/lib.c142
-rw-r--r--sound/firewire/lib.h56
-rw-r--r--sound/firewire/oxfw/Makefile2
-rw-r--r--sound/firewire/oxfw/oxfw-midi.c40
-rw-r--r--sound/firewire/oxfw/oxfw-pcm.c10
-rw-r--r--sound/firewire/oxfw/oxfw-stream.c53
-rw-r--r--sound/firewire/oxfw/oxfw.c46
-rw-r--r--sound/firewire/oxfw/oxfw.h3
-rw-r--r--sound/firewire/tascam/Makefile4
-rw-r--r--sound/firewire/tascam/amdtp-tascam.c243
-rw-r--r--sound/firewire/tascam/tascam-hwdep.c201
-rw-r--r--sound/firewire/tascam/tascam-midi.c135
-rw-r--r--sound/firewire/tascam/tascam-pcm.c312
-rw-r--r--sound/firewire/tascam/tascam-proc.c88
-rw-r--r--sound/firewire/tascam/tascam-stream.c496
-rw-r--r--sound/firewire/tascam/tascam-transaction.c302
-rw-r--r--sound/firewire/tascam/tascam.c209
-rw-r--r--sound/firewire/tascam/tascam.h147
-rw-r--r--sound/hda/ext/hdac_ext_stream.c9
-rw-r--r--sound/hda/hda_bus_type.c13
-rw-r--r--sound/hda/hdac_bus.c14
-rw-r--r--sound/hda/hdac_device.c124
-rw-r--r--sound/hda/hdac_i915.c70
-rw-r--r--sound/hda/hdac_regmap.c10
-rw-r--r--sound/hda/hdac_stream.c3
-rw-r--r--sound/hda/hdac_sysfs.c8
-rw-r--r--sound/pci/cs46xx/cs46xx_lib.c9
-rw-r--r--sound/pci/hda/hda_bind.c69
-rw-r--r--sound/pci/hda/hda_codec.c44
-rw-r--r--sound/pci/hda/hda_codec.h44
-rw-r--r--sound/pci/hda/hda_controller.c1
-rw-r--r--sound/pci/hda/hda_generic.c7
-rw-r--r--sound/pci/hda/hda_local.h7
-rw-r--r--sound/pci/hda/hda_sysfs.c3
-rw-r--r--sound/pci/hda/patch_analog.c37
-rw-r--r--sound/pci/hda/patch_ca0110.c15
-rw-r--r--sound/pci/hda/patch_ca0132.c15
-rw-r--r--sound/pci/hda/patch_cirrus.c22
-rw-r--r--sound/pci/hda/patch_cmedia.c15
-rw-r--r--sound/pci/hda/patch_conexant.c118
-rw-r--r--sound/pci/hda/patch_hdmi.c220
-rw-r--r--sound/pci/hda/patch_realtek.c134
-rw-r--r--sound/pci/hda/patch_si3054.c39
-rw-r--r--sound/pci/hda/patch_sigmatel.c214
-rw-r--r--sound/pci/hda/patch_via.c171
-rw-r--r--sound/pci/korg1212/korg1212.c8
-rw-r--r--sound/pci/lx6464es/lx6464es.c4
-rw-r--r--sound/pci/maestro3.c25
-rw-r--r--sound/pci/rme32.c4
-rw-r--r--sound/pci/rme96.c8
-rw-r--r--sound/pci/rme9652/hdsp.c1
-rw-r--r--sound/pci/rme9652/hdspm.c9
-rw-r--r--sound/soc/Kconfig6
-rw-r--r--sound/soc/Makefile4
-rw-r--r--sound/soc/atmel/Kconfig9
-rw-r--r--sound/soc/atmel/Makefile2
-rw-r--r--sound/soc/atmel/atmel-classd.c679
-rw-r--r--sound/soc/atmel/atmel-classd.h120
-rw-r--r--sound/soc/atmel/atmel_wm8904.c1
-rw-r--r--sound/soc/au1x/db1000.c10
-rw-r--r--sound/soc/au1x/db1200.c10
-rw-r--r--sound/soc/blackfin/bf5xx-ad1836.c11
-rw-r--r--sound/soc/blackfin/bfin-eval-adau1373.c12
-rw-r--r--sound/soc/blackfin/bfin-eval-adau1701.c12
-rw-r--r--sound/soc/blackfin/bfin-eval-adav80x.c12
-rw-r--r--sound/soc/codecs/Kconfig23
-rw-r--r--sound/soc/codecs/Makefile10
-rw-r--r--sound/soc/codecs/ad193x-i2c.c8
-rw-r--r--sound/soc/codecs/ad193x-spi.c16
-rw-r--r--sound/soc/codecs/ad193x.c128
-rw-r--r--sound/soc/codecs/ad193x.h9
-rw-r--r--sound/soc/codecs/adav80x.c4
-rw-r--r--sound/soc/codecs/ak4613.c497
-rw-r--r--sound/soc/codecs/ak4642.c153
-rw-r--r--sound/soc/codecs/arizona.c49
-rw-r--r--sound/soc/codecs/arizona.h14
-rw-r--r--sound/soc/codecs/da7213.c190
-rw-r--r--sound/soc/codecs/da7213.h8
-rw-r--r--sound/soc/codecs/da7219-aad.c823
-rw-r--r--sound/soc/codecs/da7219-aad.h212
-rw-r--r--sound/soc/codecs/da7219.c1955
-rw-r--r--sound/soc/codecs/da7219.h820
-rw-r--r--sound/soc/codecs/es8328.c2
-rw-r--r--sound/soc/codecs/hdmi.c109
-rw-r--r--sound/soc/codecs/nau8825.c1309
-rw-r--r--sound/soc/codecs/nau8825.h341
-rw-r--r--sound/soc/codecs/rl6347a.c19
-rw-r--r--sound/soc/codecs/rl6347a.h2
-rw-r--r--sound/soc/codecs/rt286.c9
-rw-r--r--sound/soc/codecs/rt298.c11
-rw-r--r--sound/soc/codecs/rt5640.c22
-rw-r--r--sound/soc/codecs/rt5645.c172
-rw-r--r--sound/soc/codecs/rt5645.h30
-rw-r--r--sound/soc/codecs/ssm2518.c6
-rw-r--r--sound/soc/codecs/tlv320aic3x.c30
-rw-r--r--sound/soc/codecs/twl4030.c13
-rw-r--r--sound/soc/codecs/uda134x.c6
-rw-r--r--sound/soc/codecs/wl1273.c9
-rw-r--r--sound/soc/codecs/wm2000.c4
-rw-r--r--sound/soc/codecs/wm5110.c187
-rw-r--r--sound/soc/codecs/wm8731.c2
-rw-r--r--sound/soc/codecs/wm8903.c2
-rw-r--r--sound/soc/codecs/wm8904.c2
-rw-r--r--sound/soc/codecs/wm8955.c2
-rw-r--r--sound/soc/codecs/wm8960.c2
-rw-r--r--sound/soc/codecs/wm8998.c1430
-rw-r--r--sound/soc/codecs/wm8998.h23
-rw-r--r--sound/soc/davinci/davinci-mcasp.c305
-rw-r--r--sound/soc/dwc/designware_i2s.c123
-rw-r--r--sound/soc/fsl/fsl-asoc-card.c141
-rw-r--r--sound/soc/fsl/fsl_esai.c88
-rw-r--r--sound/soc/fsl/fsl_sai.c38
-rw-r--r--sound/soc/fsl/fsl_spdif.c73
-rw-r--r--sound/soc/fsl/fsl_ssi.c102
-rw-r--r--sound/soc/fsl/imx-spdif.c1
-rw-r--r--sound/soc/generic/simple-card.c8
-rw-r--r--sound/soc/intel/Kconfig15
-rw-r--r--sound/soc/intel/atom/sst-mfld-platform-pcm.c19
-rw-r--r--sound/soc/intel/boards/Makefile2
-rw-r--r--sound/soc/intel/boards/broadwell.c9
-rw-r--r--sound/soc/intel/boards/bytcr_rt5640.c14
-rw-r--r--sound/soc/intel/boards/cht_bsw_max98090_ti.c14
-rw-r--r--sound/soc/intel/boards/cht_bsw_rt5645.c14
-rw-r--r--sound/soc/intel/boards/cht_bsw_rt5672.c14
-rw-r--r--sound/soc/intel/boards/skl_rt286.c259
-rw-r--r--sound/soc/intel/common/Makefile6
-rw-r--r--sound/soc/intel/common/sst-dsp-priv.h1
-rw-r--r--sound/soc/intel/common/sst-dsp.c2
-rw-r--r--sound/soc/intel/common/sst-dsp.h2
-rw-r--r--sound/soc/intel/common/sst-firmware.c10
-rw-r--r--sound/soc/intel/skylake/Makefile3
-rw-r--r--sound/soc/intel/skylake/skl-messages.c44
-rw-r--r--sound/soc/intel/skylake/skl-nhlt.c10
-rw-r--r--sound/soc/intel/skylake/skl-pcm.c176
-rw-r--r--sound/soc/intel/skylake/skl-sst-dsp.c7
-rw-r--r--sound/soc/intel/skylake/skl-sst-ipc.c12
-rw-r--r--sound/soc/intel/skylake/skl-sst-ipc.h1
-rw-r--r--sound/soc/intel/skylake/skl-sst.c56
-rw-r--r--sound/soc/intel/skylake/skl-topology.c1252
-rw-r--r--sound/soc/intel/skylake/skl-topology.h36
-rw-r--r--sound/soc/intel/skylake/skl-tplg-interface.h83
-rw-r--r--sound/soc/intel/skylake/skl.c32
-rw-r--r--sound/soc/intel/skylake/skl.h17
-rw-r--r--sound/soc/jz4740/jz4740-i2s.c1
-rw-r--r--sound/soc/kirkwood/armada-370-db.c1
-rw-r--r--sound/soc/mediatek/mt8173-max98090.c11
-rw-r--r--sound/soc/mediatek/mt8173-rt5650-rt5676.c11
-rw-r--r--sound/soc/mxs/mxs-sgtl5000.c6
-rw-r--r--sound/soc/omap/n810.c3
-rw-r--r--sound/soc/omap/rx51.c5
-rw-r--r--sound/soc/pxa/brownstone.c9
-rw-r--r--sound/soc/pxa/corgi.c11
-rw-r--r--sound/soc/pxa/e740_wm9705.c5
-rw-r--r--sound/soc/pxa/e750_wm9705.c5
-rw-r--r--sound/soc/pxa/e800_wm9712.c5
-rw-r--r--sound/soc/pxa/hx4700.c4
-rw-r--r--sound/soc/pxa/imote2.c11
-rw-r--r--sound/soc/pxa/mioa701_wm9713.c11
-rw-r--r--sound/soc/pxa/palm27x.c9
-rw-r--r--sound/soc/pxa/poodle.c11
-rw-r--r--sound/soc/pxa/pxa-ssp.c1
-rw-r--r--sound/soc/pxa/pxa2xx-ac97.c49
-rw-r--r--sound/soc/pxa/pxa2xx-i2s.c3
-rw-r--r--sound/soc/pxa/pxa2xx-pcm.c33
-rw-r--r--sound/soc/pxa/spitz.c5
-rw-r--r--sound/soc/pxa/tosa.c5
-rw-r--r--sound/soc/pxa/ttc-dkb.c12
-rw-r--r--sound/soc/qcom/lpass-cpu.c3
-rw-r--r--sound/soc/rockchip/Kconfig12
-rw-r--r--sound/soc/rockchip/Makefile6
-rw-r--r--sound/soc/rockchip/rockchip_i2s.c48
-rw-r--r--sound/soc/rockchip/rockchip_i2s.h16
-rw-r--r--sound/soc/rockchip/rockchip_spdif.c405
-rw-r--r--sound/soc/rockchip/rockchip_spdif.h63
-rw-r--r--sound/soc/samsung/h1940_uda1380.c5
-rw-r--r--sound/soc/samsung/rx1950_uda1380.c5
-rw-r--r--sound/soc/sh/Kconfig3
-rw-r--r--sound/soc/sh/rcar/adg.c306
-rw-r--r--sound/soc/sh/rcar/core.c66
-rw-r--r--sound/soc/sh/rcar/ctu.c16
-rw-r--r--sound/soc/sh/rcar/dma.c2
-rw-r--r--sound/soc/sh/rcar/dvc.c16
-rw-r--r--sound/soc/sh/rcar/gen.c14
-rw-r--r--sound/soc/sh/rcar/mix.c16
-rw-r--r--sound/soc/sh/rcar/rcar_snd.h117
-rw-r--r--sound/soc/sh/rcar/rsnd.h29
-rw-r--r--sound/soc/sh/rcar/src.c27
-rw-r--r--sound/soc/sh/rcar/ssi.c113
-rw-r--r--sound/soc/sh/siu_dai.c85
-rw-r--r--sound/soc/soc-compress.c12
-rw-r--r--sound/soc/soc-core.c29
-rw-r--r--sound/soc/soc-dapm.c59
-rw-r--r--sound/soc/soc-ops.c8
-rw-r--r--sound/soc/soc-pcm.c59
-rw-r--r--sound/soc/soc-topology.c4
-rw-r--r--sound/soc/sunxi/Kconfig11
-rw-r--r--sound/soc/sunxi/Makefile2
-rw-r--r--sound/soc/sunxi/sun4i-codec.c712
-rw-r--r--sound/soc/ux500/mop500.c1
-rw-r--r--sound/soc/ux500/ux500_msp_dai.c5
-rw-r--r--sound/usb/card.h1
-rw-r--r--sound/usb/endpoint.c92
-rw-r--r--sound/usb/midi.c11
-rw-r--r--sound/usb/mixer_quirks.c2
-rw-r--r--sound/usb/pcm.c74
-rw-r--r--sound/usb/quirks-table.h23
-rw-r--r--sound/usb/quirks.c3
-rw-r--r--sound/usb/stream.c1
-rw-r--r--sound/usb/usbaudio.h1
262 files changed, 20305 insertions, 2841 deletions
diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c
index 38590b322c54..fbd5dad0c484 100644
--- a/sound/arm/pxa2xx-ac97.c
+++ b/sound/arm/pxa2xx-ac97.c
@@ -15,6 +15,7 @@
#include <linux/module.h>
#include <linux/platform_device.h>
#include <linux/dmaengine.h>
+#include <linux/dma/pxa-dma.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -43,7 +44,11 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = {
.reset = pxa2xx_ac97_reset,
};
-static unsigned long pxa2xx_ac97_pcm_out_req = 12;
+static struct pxad_param pxa2xx_ac97_pcm_out_req = {
+ .prio = PXAD_PRIO_LOWEST,
+ .drcmr = 12,
+};
+
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = {
.addr = __PREG(PCDR),
.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
@@ -51,7 +56,11 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_out = {
.filter_data = &pxa2xx_ac97_pcm_out_req,
};
-static unsigned long pxa2xx_ac97_pcm_in_req = 11;
+static struct pxad_param pxa2xx_ac97_pcm_in_req = {
+ .prio = PXAD_PRIO_LOWEST,
+ .drcmr = 11,
+};
+
static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_in = {
.addr = __PREG(PCDR),
.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES,
diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c
index 01f8fdc42b1b..e9b98af6b52c 100644
--- a/sound/arm/pxa2xx-pcm-lib.c
+++ b/sound/arm/pxa2xx-pcm-lib.c
@@ -8,6 +8,7 @@
#include <linux/module.h>
#include <linux/dma-mapping.h>
#include <linux/dmaengine.h>
+#include <linux/dma/pxa-dma.h>
#include <sound/core.h>
#include <sound/pcm.h>
@@ -15,8 +16,6 @@
#include <sound/pxa2xx-lib.h>
#include <sound/dmaengine_pcm.h>
-#include <mach/dma.h>
-
#include "pxa2xx-pcm.h"
static const struct snd_pcm_hardware pxa2xx_pcm_hardware = {
@@ -31,7 +30,7 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = {
.period_bytes_min = 32,
.period_bytes_max = 8192 - 32,
.periods_min = 1,
- .periods_max = PAGE_SIZE/sizeof(pxa_dma_desc),
+ .periods_max = 256,
.buffer_bytes_max = 128 * 1024,
.fifo_size = 32,
};
@@ -39,65 +38,29 @@ static const struct snd_pcm_hardware pxa2xx_pcm_hardware = {
int __pxa2xx_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct pxa2xx_runtime_data *rtd = runtime->private_data;
- size_t totsize = params_buffer_bytes(params);
- size_t period = params_period_bytes(params);
- pxa_dma_desc *dma_desc;
- dma_addr_t dma_buff_phys, next_desc_phys;
- u32 dcmd = DCMD_INCSRCADDR | DCMD_FLOWTRG;
+ struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream);
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_dmaengine_dai_dma_data *dma_params;
+ struct dma_slave_config config;
+ int ret;
- /* temporary transition hack */
- switch (rtd->params->addr_width) {
- case DMA_SLAVE_BUSWIDTH_1_BYTE:
- dcmd |= DCMD_WIDTH1;
- break;
- case DMA_SLAVE_BUSWIDTH_2_BYTES:
- dcmd |= DCMD_WIDTH2;
- break;
- case DMA_SLAVE_BUSWIDTH_4_BYTES:
- dcmd |= DCMD_WIDTH4;
- break;
- default:
- /* can't happen */
- break;
- }
+ dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ if (!dma_params)
+ return 0;
- switch (rtd->params->maxburst) {
- case 8:
- dcmd |= DCMD_BURST8;
- break;
- case 16:
- dcmd |= DCMD_BURST16;
- break;
- case 32:
- dcmd |= DCMD_BURST32;
- break;
- }
+ ret = snd_hwparams_to_dma_slave_config(substream, params, &config);
+ if (ret)
+ return ret;
- snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
- runtime->dma_bytes = totsize;
+ snd_dmaengine_pcm_set_config_from_dai_data(substream,
+ snd_soc_dai_get_dma_data(rtd->cpu_dai, substream),
+ &config);
- dma_desc = rtd->dma_desc_array;
- next_desc_phys = rtd->dma_desc_array_phys;
- dma_buff_phys = runtime->dma_addr;
- do {
- next_desc_phys += sizeof(pxa_dma_desc);
- dma_desc->ddadr = next_desc_phys;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
- dma_desc->dsadr = dma_buff_phys;
- dma_desc->dtadr = rtd->params->addr;
- } else {
- dma_desc->dsadr = rtd->params->addr;
- dma_desc->dtadr = dma_buff_phys;
- }
- if (period > totsize)
- period = totsize;
- dma_desc->dcmd = dcmd | period | DCMD_ENDIRQEN;
- dma_desc++;
- dma_buff_phys += period;
- } while (totsize -= period);
- dma_desc[-1].ddadr = rtd->dma_desc_array_phys;
+ ret = dmaengine_slave_config(chan, &config);
+ if (ret)
+ return ret;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
return 0;
}
@@ -105,13 +68,6 @@ EXPORT_SYMBOL(__pxa2xx_pcm_hw_params);
int __pxa2xx_pcm_hw_free(struct snd_pcm_substream *substream)
{
- struct pxa2xx_runtime_data *rtd = substream->runtime->private_data;
-
- if (rtd && rtd->params && rtd->params->filter_data) {
- unsigned long req = *(unsigned long *) rtd->params->filter_data;
- DRCMR(req) = 0;
- }
-
snd_pcm_set_runtime_buffer(substream, NULL);
return 0;
}
@@ -119,100 +75,36 @@ EXPORT_SYMBOL(__pxa2xx_pcm_hw_free);
int pxa2xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
{
- struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
- int ret = 0;
-
- switch (cmd) {
- case SNDRV_PCM_TRIGGER_START:
- DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys;
- DCSR(prtd->dma_ch) = DCSR_RUN;
- break;
-
- case SNDRV_PCM_TRIGGER_STOP:
- case SNDRV_PCM_TRIGGER_SUSPEND:
- case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
- DCSR(prtd->dma_ch) &= ~DCSR_RUN;
- break;
-
- case SNDRV_PCM_TRIGGER_RESUME:
- DCSR(prtd->dma_ch) |= DCSR_RUN;
- break;
- case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
- DDADR(prtd->dma_ch) = prtd->dma_desc_array_phys;
- DCSR(prtd->dma_ch) |= DCSR_RUN;
- break;
-
- default:
- ret = -EINVAL;
- }
-
- return ret;
+ return snd_dmaengine_pcm_trigger(substream, cmd);
}
EXPORT_SYMBOL(pxa2xx_pcm_trigger);
snd_pcm_uframes_t
pxa2xx_pcm_pointer(struct snd_pcm_substream *substream)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct pxa2xx_runtime_data *prtd = runtime->private_data;
-
- dma_addr_t ptr = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
- DSADR(prtd->dma_ch) : DTADR(prtd->dma_ch);
- snd_pcm_uframes_t x = bytes_to_frames(runtime, ptr - runtime->dma_addr);
-
- if (x == runtime->buffer_size)
- x = 0;
- return x;
+ return snd_dmaengine_pcm_pointer(substream);
}
EXPORT_SYMBOL(pxa2xx_pcm_pointer);
int __pxa2xx_pcm_prepare(struct snd_pcm_substream *substream)
{
- struct pxa2xx_runtime_data *prtd = substream->runtime->private_data;
- unsigned long req;
-
- if (!prtd || !prtd->params)
- return 0;
-
- if (prtd->dma_ch == -1)
- return -EINVAL;
-
- DCSR(prtd->dma_ch) &= ~DCSR_RUN;
- DCSR(prtd->dma_ch) = 0;
- DCMD(prtd->dma_ch) = 0;
- req = *(unsigned long *) prtd->params->filter_data;
- DRCMR(req) = prtd->dma_ch | DRCMR_MAPVLD;
-
return 0;
}
EXPORT_SYMBOL(__pxa2xx_pcm_prepare);
-void pxa2xx_pcm_dma_irq(int dma_ch, void *dev_id)
-{
- struct snd_pcm_substream *substream = dev_id;
- int dcsr;
-
- dcsr = DCSR(dma_ch);
- DCSR(dma_ch) = dcsr & ~DCSR_STOPIRQEN;
-
- if (dcsr & DCSR_ENDINTR) {
- snd_pcm_period_elapsed(substream);
- } else {
- printk(KERN_ERR "DMA error on channel %d (DCSR=%#x)\n",
- dma_ch, dcsr);
- snd_pcm_stop_xrun(substream);
- }
-}
-EXPORT_SYMBOL(pxa2xx_pcm_dma_irq);
-
int __pxa2xx_pcm_open(struct snd_pcm_substream *substream)
{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_pcm_runtime *runtime = substream->runtime;
- struct pxa2xx_runtime_data *rtd;
+ struct snd_dmaengine_dai_dma_data *dma_params;
int ret;
runtime->hw = pxa2xx_pcm_hardware;
+ dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+ if (!dma_params)
+ return 0;
+
/*
* For mysterious reasons (and despite what the manual says)
* playback samples are lost if the DMA count is not a multiple
@@ -221,48 +113,27 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream)
ret = snd_pcm_hw_constraint_step(runtime, 0,
SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
if (ret)
- goto out;
+ return ret;
ret = snd_pcm_hw_constraint_step(runtime, 0,
SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
if (ret)
- goto out;
+ return ret;
ret = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
if (ret < 0)
- goto out;
-
- ret = -ENOMEM;
- rtd = kzalloc(sizeof(*rtd), GFP_KERNEL);
- if (!rtd)
- goto out;
- rtd->dma_desc_array =
- dma_alloc_writecombine(substream->pcm->card->dev, PAGE_SIZE,
- &rtd->dma_desc_array_phys, GFP_KERNEL);
- if (!rtd->dma_desc_array)
- goto err1;
+ return ret;
- rtd->dma_ch = -1;
- runtime->private_data = rtd;
- return 0;
-
- err1:
- kfree(rtd);
- out:
- return ret;
+ return snd_dmaengine_pcm_open_request_chan(substream,
+ pxad_filter_fn,
+ dma_params->filter_data);
}
EXPORT_SYMBOL(__pxa2xx_pcm_open);
int __pxa2xx_pcm_close(struct snd_pcm_substream *substream)
{
- struct snd_pcm_runtime *runtime = substream->runtime;
- struct pxa2xx_runtime_data *rtd = runtime->private_data;
-
- dma_free_writecombine(substream->pcm->card->dev, PAGE_SIZE,
- rtd->dma_desc_array, rtd->dma_desc_array_phys);
- kfree(rtd);
- return 0;
+ return snd_dmaengine_pcm_close_release_chan(substream);
}
EXPORT_SYMBOL(__pxa2xx_pcm_close);
diff --git a/sound/arm/pxa2xx-pcm.c b/sound/arm/pxa2xx-pcm.c
index 83be8e3f095e..83fcfac97739 100644
--- a/sound/arm/pxa2xx-pcm.c
+++ b/sound/arm/pxa2xx-pcm.c
@@ -46,17 +46,13 @@ static int pxa2xx_pcm_open(struct snd_pcm_substream *substream)
rtd->params = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ?
client->playback_params : client->capture_params;
- ret = pxa_request_dma("dma", DMA_PRIO_LOW,
- pxa2xx_pcm_dma_irq, substream);
- if (ret < 0)
- goto err2;
- rtd->dma_ch = ret;
ret = client->startup(substream);
if (!ret)
- goto out;
+ goto err2;
+
+ return 0;
- pxa_free_dma(rtd->dma_ch);
err2:
__pxa2xx_pcm_close(substream);
out:
@@ -66,9 +62,7 @@ static int pxa2xx_pcm_open(struct snd_pcm_substream *substream)
static int pxa2xx_pcm_close(struct snd_pcm_substream *substream)
{
struct pxa2xx_pcm_client *client = substream->private_data;
- struct pxa2xx_runtime_data *rtd = substream->runtime->private_data;
- pxa_free_dma(rtd->dma_ch);
client->shutdown(substream);
return __pxa2xx_pcm_close(substream);
diff --git a/sound/arm/pxa2xx-pcm.h b/sound/arm/pxa2xx-pcm.h
index 00330985beec..8fa2b7c9e6b8 100644
--- a/sound/arm/pxa2xx-pcm.h
+++ b/sound/arm/pxa2xx-pcm.h
@@ -13,8 +13,6 @@
struct pxa2xx_runtime_data {
int dma_ch;
struct snd_dmaengine_dai_dma_data *params;
- struct pxa_dma_desc *dma_desc_array;
- dma_addr_t dma_desc_array_phys;
};
struct pxa2xx_pcm_client {
diff --git a/sound/core/Kconfig b/sound/core/Kconfig
index 6c96feeaf01e..e3e949126a56 100644
--- a/sound/core/Kconfig
+++ b/sound/core/Kconfig
@@ -4,7 +4,7 @@ config SND_TIMER
config SND_PCM
tristate
- select SND_TIMER
+ select SND_TIMER if SND_PCM_TIMER
config SND_PCM_ELD
bool
@@ -93,6 +93,17 @@ config SND_PCM_OSS_PLUGINS
support conversion of channels, formats and rates. It will
behave like most of new OSS/Free drivers in 2.4/2.6 kernels.
+config SND_PCM_TIMER
+ bool "PCM timer interface" if EXPERT
+ default y
+ help
+ If you disable this option, pcm timer will be inavailable, so
+ those stubs used pcm timer (e.g. dmix, dsnoop & co) may work
+ incorrectlly.
+
+ For some embedded device, we may disable it to reduce memory
+ footprint, about 20KB on x86_64 platform.
+
config SND_SEQUENCER_OSS
bool "OSS Sequencer API"
depends on SND_SEQUENCER
diff --git a/sound/core/Makefile b/sound/core/Makefile
index 3354f91e003a..48ab4b8f8279 100644
--- a/sound/core/Makefile
+++ b/sound/core/Makefile
@@ -13,8 +13,9 @@ snd-$(CONFIG_SND_OSSEMUL) += sound_oss.o
snd-$(CONFIG_SND_VMASTER) += vmaster.o
snd-$(CONFIG_SND_JACK) += ctljack.o jack.o
-snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_timer.o pcm_misc.o \
+snd-pcm-y := pcm.o pcm_native.o pcm_lib.o pcm_misc.o \
pcm_memory.o memalloc.o
+snd-pcm-$(CONFIG_SND_PCM_TIMER) += pcm_timer.o
snd-pcm-$(CONFIG_SND_DMA_SGBUF) += sgbuf.o
snd-pcm-$(CONFIG_SND_PCM_ELD) += pcm_drm_eld.o
snd-pcm-$(CONFIG_SND_PCM_IEC958) += pcm_iec958.o
diff --git a/sound/core/oss/mixer_oss.c b/sound/core/oss/mixer_oss.c
index a99f7200ff3f..7a8c79dd9734 100644
--- a/sound/core/oss/mixer_oss.c
+++ b/sound/core/oss/mixer_oss.c
@@ -1177,7 +1177,8 @@ static void snd_mixer_oss_proc_write(struct snd_info_entry *entry,
struct snd_mixer_oss *mixer = entry->private_data;
char line[128], str[32], idxstr[16];
const char *cptr;
- int ch, idx;
+ unsigned int idx;
+ int ch;
struct snd_mixer_oss_assign_table *tbl;
struct slot *slot;
diff --git a/sound/core/pcm.c b/sound/core/pcm.c
index 02bd96954dc4..308c9ecf73db 100644
--- a/sound/core/pcm.c
+++ b/sound/core/pcm.c
@@ -1014,9 +1014,6 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream)
snd_free_pages((void*)runtime->control,
PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control)));
kfree(runtime->hw_constraints.rules);
-#ifdef CONFIG_SND_PCM_XRUN_DEBUG
- kfree(runtime->hwptr_log);
-#endif
kfree(runtime);
substream->runtime = NULL;
put_pid(substream->pid);
diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c
index 7d45645f10ba..6b5a811e01a5 100644
--- a/sound/core/pcm_lib.c
+++ b/sound/core/pcm_lib.c
@@ -801,7 +801,7 @@ void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k,
* negative error code.
*/
int snd_interval_ratnum(struct snd_interval *i,
- unsigned int rats_count, struct snd_ratnum *rats,
+ unsigned int rats_count, const struct snd_ratnum *rats,
unsigned int *nump, unsigned int *denp)
{
unsigned int best_num, best_den;
@@ -920,7 +920,8 @@ EXPORT_SYMBOL(snd_interval_ratnum);
* negative error code.
*/
static int snd_interval_ratden(struct snd_interval *i,
- unsigned int rats_count, struct snd_ratden *rats,
+ unsigned int rats_count,
+ const struct snd_ratden *rats,
unsigned int *nump, unsigned int *denp)
{
unsigned int best_num, best_diff, best_den;
@@ -1339,7 +1340,7 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_ranges);
static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
- struct snd_pcm_hw_constraint_ratnums *r = rule->private;
+ const struct snd_pcm_hw_constraint_ratnums *r = rule->private;
unsigned int num = 0, den = 0;
int err;
err = snd_interval_ratnum(hw_param_interval(params, rule->var),
@@ -1363,10 +1364,10 @@ static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params,
int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime,
unsigned int cond,
snd_pcm_hw_param_t var,
- struct snd_pcm_hw_constraint_ratnums *r)
+ const struct snd_pcm_hw_constraint_ratnums *r)
{
return snd_pcm_hw_rule_add(runtime, cond, var,
- snd_pcm_hw_rule_ratnums, r,
+ snd_pcm_hw_rule_ratnums, (void *)r,
var, -1);
}
@@ -1375,7 +1376,7 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_ratnums);
static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params,
struct snd_pcm_hw_rule *rule)
{
- struct snd_pcm_hw_constraint_ratdens *r = rule->private;
+ const struct snd_pcm_hw_constraint_ratdens *r = rule->private;
unsigned int num = 0, den = 0;
int err = snd_interval_ratden(hw_param_interval(params, rule->var),
r->nrats, r->rats, &num, &den);
@@ -1398,10 +1399,10 @@ static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params,
int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime,
unsigned int cond,
snd_pcm_hw_param_t var,
- struct snd_pcm_hw_constraint_ratdens *r)
+ const struct snd_pcm_hw_constraint_ratdens *r)
{
return snd_pcm_hw_rule_add(runtime, cond, var,
- snd_pcm_hw_rule_ratdens, r,
+ snd_pcm_hw_rule_ratdens, (void *)r,
var, -1);
}
@@ -1875,20 +1876,17 @@ void snd_pcm_period_elapsed(struct snd_pcm_substream *substream)
return;
runtime = substream->runtime;
- if (runtime->transfer_ack_begin)
- runtime->transfer_ack_begin(substream);
-
snd_pcm_stream_lock_irqsave(substream, flags);
if (!snd_pcm_running(substream) ||
snd_pcm_update_hw_ptr0(substream, 1) < 0)
goto _end;
+#ifdef CONFIG_SND_PCM_TIMER
if (substream->timer_running)
snd_timer_interrupt(substream->timer, 1);
+#endif
_end:
snd_pcm_stream_unlock_irqrestore(substream, flags);
- if (runtime->transfer_ack_end)
- runtime->transfer_ack_end(substream);
kill_fasync(&runtime->fasync, SIGIO, POLL_IN);
}
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 75888dd38a7f..a8b27cdc2844 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -486,6 +486,16 @@ static void snd_pcm_set_state(struct snd_pcm_substream *substream, int state)
snd_pcm_stream_unlock_irq(substream);
}
+static inline void snd_pcm_timer_notify(struct snd_pcm_substream *substream,
+ int event)
+{
+#ifdef CONFIG_SND_PCM_TIMER
+ if (substream->timer)
+ snd_timer_notify(substream->timer, event,
+ &substream->runtime->trigger_tstamp);
+#endif
+}
+
static int snd_pcm_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -650,7 +660,8 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream,
}
snd_pcm_stream_unlock_irq(substream);
- if (params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST)
+ if (params->tstamp_mode < 0 ||
+ params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST)
return -EINVAL;
if (params->proto >= SNDRV_PROTOCOL_VERSION(2, 0, 12) &&
params->tstamp_type > SNDRV_PCM_TSTAMP_TYPE_LAST)
@@ -1042,9 +1053,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state)
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK &&
runtime->silence_size > 0)
snd_pcm_playback_silence(substream, ULONG_MAX);
- if (substream->timer)
- snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTART,
- &runtime->trigger_tstamp);
+ snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTART);
}
static struct action_ops snd_pcm_action_start = {
@@ -1092,9 +1101,7 @@ static void snd_pcm_post_stop(struct snd_pcm_substream *substream, int state)
if (runtime->status->state != state) {
snd_pcm_trigger_tstamp(substream);
runtime->status->state = state;
- if (substream->timer)
- snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSTOP,
- &runtime->trigger_tstamp);
+ snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSTOP);
}
wake_up(&runtime->sleep);
wake_up(&runtime->tsleep);
@@ -1208,18 +1215,12 @@ static void snd_pcm_post_pause(struct snd_pcm_substream *substream, int push)
snd_pcm_trigger_tstamp(substream);
if (push) {
runtime->status->state = SNDRV_PCM_STATE_PAUSED;
- if (substream->timer)
- snd_timer_notify(substream->timer,
- SNDRV_TIMER_EVENT_MPAUSE,
- &runtime->trigger_tstamp);
+ snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MPAUSE);
wake_up(&runtime->sleep);
wake_up(&runtime->tsleep);
} else {
runtime->status->state = SNDRV_PCM_STATE_RUNNING;
- if (substream->timer)
- snd_timer_notify(substream->timer,
- SNDRV_TIMER_EVENT_MCONTINUE,
- &runtime->trigger_tstamp);
+ snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MCONTINUE);
}
}
@@ -1267,9 +1268,7 @@ static void snd_pcm_post_suspend(struct snd_pcm_substream *substream, int state)
snd_pcm_trigger_tstamp(substream);
runtime->status->suspended_state = runtime->status->state;
runtime->status->state = SNDRV_PCM_STATE_SUSPENDED;
- if (substream->timer)
- snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MSUSPEND,
- &runtime->trigger_tstamp);
+ snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MSUSPEND);
wake_up(&runtime->sleep);
wake_up(&runtime->tsleep);
}
@@ -1373,9 +1372,7 @@ static void snd_pcm_post_resume(struct snd_pcm_substream *substream, int state)
struct snd_pcm_runtime *runtime = substream->runtime;
snd_pcm_trigger_tstamp(substream);
runtime->status->state = runtime->status->suspended_state;
- if (substream->timer)
- snd_timer_notify(substream->timer, SNDRV_TIMER_EVENT_MRESUME,
- &runtime->trigger_tstamp);
+ snd_pcm_timer_notify(substream, SNDRV_TIMER_EVENT_MRESUME);
}
static struct action_ops snd_pcm_action_resume = {
@@ -2226,7 +2223,8 @@ void snd_pcm_release_substream(struct snd_pcm_substream *substream)
snd_pcm_drop(substream);
if (substream->hw_opened) {
- if (substream->ops->hw_free != NULL)
+ if (substream->ops->hw_free &&
+ substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
substream->ops->hw_free(substream);
substream->ops->close(substream);
substream->hw_opened = 0;
diff --git a/sound/core/seq/oss/seq_oss_readq.c b/sound/core/seq/oss/seq_oss_readq.c
index ccd893566f1d..046cb586fb2f 100644
--- a/sound/core/seq/oss/seq_oss_readq.c
+++ b/sound/core/seq/oss/seq_oss_readq.c
@@ -91,8 +91,7 @@ snd_seq_oss_readq_clear(struct seq_oss_readq *q)
q->head = q->tail = 0;
}
/* if someone sleeping, wake'em up */
- if (waitqueue_active(&q->midi_sleep))
- wake_up(&q->midi_sleep);
+ wake_up(&q->midi_sleep);
q->input_time = (unsigned long)-1;
}
@@ -138,8 +137,7 @@ snd_seq_oss_readq_put_event(struct seq_oss_readq *q, union evrec *ev)
q->qlen++;
/* wake up sleeper */
- if (waitqueue_active(&q->midi_sleep))
- wake_up(&q->midi_sleep);
+ wake_up(&q->midi_sleep);
spin_unlock_irqrestore(&q->lock, flags);
diff --git a/sound/core/seq/oss/seq_oss_writeq.c b/sound/core/seq/oss/seq_oss_writeq.c
index d50338bbc21f..1f6788a18444 100644
--- a/sound/core/seq/oss/seq_oss_writeq.c
+++ b/sound/core/seq/oss/seq_oss_writeq.c
@@ -138,9 +138,7 @@ snd_seq_oss_writeq_wakeup(struct seq_oss_writeq *q, abstime_t time)
spin_lock_irqsave(&q->sync_lock, flags);
q->sync_time = time;
q->sync_event_put = 0;
- if (waitqueue_active(&q->sync_sleep)) {
- wake_up(&q->sync_sleep);
- }
+ wake_up(&q->sync_sleep);
spin_unlock_irqrestore(&q->sync_lock, flags);
}
diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig
index 8850b7de1d38..bee0e5f1a116 100644
--- a/sound/firewire/Kconfig
+++ b/sound/firewire/Kconfig
@@ -120,4 +120,31 @@ config SND_BEBOB
To compile this driver as a module, choose M here: the module
will be called snd-bebob.
+config SND_FIREWIRE_DIGI00X
+ tristate "Digidesign Digi 002/003 family support"
+ select SND_FIREWIRE_LIB
+ select SND_HWDEP
+ help
+ Say Y here to include support for Digidesign Digi 002/003 family.
+ * Digi 002 Console
+ * Digi 002 Rack
+ * Digi 003 Console
+ * Digi 003 Rack
+ * Digi 003 Rack+
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-firewire-digi00x.
+
+config SND_FIREWIRE_TASCAM
+ tristate "TASCAM FireWire series support"
+ select SND_FIREWIRE_LIB
+ select SND_HWDEP
+ help
+ Say Y here to include support for TASCAM.
+ * FW-1884
+ * FW-1082
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-firewire-tascam.
+
endif # SND_FIREWIRE
diff --git a/sound/firewire/Makefile b/sound/firewire/Makefile
index 8b37f084b2ab..f5fb62551c60 100644
--- a/sound/firewire/Makefile
+++ b/sound/firewire/Makefile
@@ -1,6 +1,5 @@
snd-firewire-lib-objs := lib.o iso-resources.o packets-buffer.o \
- fcp.o cmp.o amdtp.o
-snd-oxfw-objs := oxfw.o
+ fcp.o cmp.o amdtp-stream.o amdtp-am824.o
snd-isight-objs := isight.o
snd-scs1x-objs := scs1x.o
@@ -11,3 +10,5 @@ obj-$(CONFIG_SND_ISIGHT) += snd-isight.o
obj-$(CONFIG_SND_SCS1X) += snd-scs1x.o
obj-$(CONFIG_SND_FIREWORKS) += fireworks/
obj-$(CONFIG_SND_BEBOB) += bebob/
+obj-$(CONFIG_SND_FIREWIRE_DIGI00X) += digi00x/
+obj-$(CONFIG_SND_FIREWIRE_TASCAM) += tascam/
diff --git a/sound/firewire/amdtp-am824.c b/sound/firewire/amdtp-am824.c
new file mode 100644
index 000000000000..bebddc60fde8
--- /dev/null
+++ b/sound/firewire/amdtp-am824.c
@@ -0,0 +1,465 @@
+/*
+ * AM824 format in Audio and Music Data Transmission Protocol (IEC 61883-6)
+ *
+ * Copyright (c) Clemens Ladisch <clemens@ladisch.de>
+ * Copyright (c) 2015 Takashi Sakamoto <o-takashi@sakamocchi.jp>
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <linux/slab.h>
+
+#include "amdtp-am824.h"
+
+#define CIP_FMT_AM 0x10
+
+/* "Clock-based rate control mode" is just supported. */
+#define AMDTP_FDF_AM824 0x00
+
+/*
+ * Nominally 3125 bytes/second, but the MIDI port's clock might be
+ * 1% too slow, and the bus clock 100 ppm too fast.
+ */
+#define MIDI_BYTES_PER_SECOND 3093
+
+/*
+ * Several devices look only at the first eight data blocks.
+ * In any case, this is more than enough for the MIDI data rate.
+ */
+#define MAX_MIDI_RX_BLOCKS 8
+
+struct amdtp_am824 {
+ struct snd_rawmidi_substream *midi[AM824_MAX_CHANNELS_FOR_MIDI * 8];
+ int midi_fifo_limit;
+ int midi_fifo_used[AM824_MAX_CHANNELS_FOR_MIDI * 8];
+ unsigned int pcm_channels;
+ unsigned int midi_ports;
+
+ u8 pcm_positions[AM824_MAX_CHANNELS_FOR_PCM];
+ u8 midi_position;
+
+ void (*transfer_samples)(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames);
+
+ unsigned int frame_multiplier;
+};
+
+/**
+ * amdtp_am824_set_parameters - set stream parameters
+ * @s: the AMDTP stream to configure
+ * @rate: the sample rate
+ * @pcm_channels: the number of PCM samples in each data block, to be encoded
+ * as AM824 multi-bit linear audio
+ * @midi_ports: the number of MIDI ports (i.e., MPX-MIDI Data Channels)
+ * @double_pcm_frames: one data block transfers two PCM frames
+ *
+ * The parameters must be set before the stream is started, and must not be
+ * changed while the stream is running.
+ */
+int amdtp_am824_set_parameters(struct amdtp_stream *s, unsigned int rate,
+ unsigned int pcm_channels,
+ unsigned int midi_ports,
+ bool double_pcm_frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ unsigned int midi_channels;
+ unsigned int i;
+ int err;
+
+ if (amdtp_stream_running(s))
+ return -EINVAL;
+
+ if (pcm_channels > AM824_MAX_CHANNELS_FOR_PCM)
+ return -EINVAL;
+
+ midi_channels = DIV_ROUND_UP(midi_ports, 8);
+ if (midi_channels > AM824_MAX_CHANNELS_FOR_MIDI)
+ return -EINVAL;
+
+ if (WARN_ON(amdtp_stream_running(s)) ||
+ WARN_ON(pcm_channels > AM824_MAX_CHANNELS_FOR_PCM) ||
+ WARN_ON(midi_channels > AM824_MAX_CHANNELS_FOR_MIDI))
+ return -EINVAL;
+
+ err = amdtp_stream_set_parameters(s, rate,
+ pcm_channels + midi_channels);
+ if (err < 0)
+ return err;
+
+ s->fdf = AMDTP_FDF_AM824 | s->sfc;
+
+ p->pcm_channels = pcm_channels;
+ p->midi_ports = midi_ports;
+
+ /*
+ * In IEC 61883-6, one data block represents one event. In ALSA, one
+ * event equals to one PCM frame. But Dice has a quirk at higher
+ * sampling rate to transfer two PCM frames in one data block.
+ */
+ if (double_pcm_frames)
+ p->frame_multiplier = 2;
+ else
+ p->frame_multiplier = 1;
+
+ /* init the position map for PCM and MIDI channels */
+ for (i = 0; i < pcm_channels; i++)
+ p->pcm_positions[i] = i;
+ p->midi_position = p->pcm_channels;
+
+ /*
+ * We do not know the actual MIDI FIFO size of most devices. Just
+ * assume two bytes, i.e., one byte can be received over the bus while
+ * the previous one is transmitted over MIDI.
+ * (The value here is adjusted for midi_ratelimit_per_packet().)
+ */
+ p->midi_fifo_limit = rate - MIDI_BYTES_PER_SECOND * s->syt_interval + 1;
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_set_parameters);
+
+/**
+ * amdtp_am824_set_pcm_position - set an index of data channel for a channel
+ * of PCM frame
+ * @s: the AMDTP stream
+ * @index: the index of data channel in an data block
+ * @position: the channel of PCM frame
+ */
+void amdtp_am824_set_pcm_position(struct amdtp_stream *s, unsigned int index,
+ unsigned int position)
+{
+ struct amdtp_am824 *p = s->protocol;
+
+ if (index < p->pcm_channels)
+ p->pcm_positions[index] = position;
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_set_pcm_position);
+
+/**
+ * amdtp_am824_set_midi_position - set a index of data channel for MIDI
+ * conformant data channel
+ * @s: the AMDTP stream
+ * @position: the index of data channel in an data block
+ */
+void amdtp_am824_set_midi_position(struct amdtp_stream *s,
+ unsigned int position)
+{
+ struct amdtp_am824 *p = s->protocol;
+
+ p->midi_position = position;
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_set_midi_position);
+
+static void write_pcm_s32(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ const u32 *src;
+
+ channels = p->pcm_channels;
+ src = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ buffer[p->pcm_positions[c]] =
+ cpu_to_be32((*src >> 8) | 0x40000000);
+ src++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void write_pcm_s16(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ const u16 *src;
+
+ channels = p->pcm_channels;
+ src = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ buffer[p->pcm_positions[c]] =
+ cpu_to_be32((*src << 8) | 0x42000000);
+ src++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void read_pcm_s32(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ u32 *dst;
+
+ channels = p->pcm_channels;
+ dst = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ *dst = be32_to_cpu(buffer[p->pcm_positions[c]]) << 8;
+ dst++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ dst = (void *)runtime->dma_area;
+ }
+}
+
+static void write_pcm_silence(struct amdtp_stream *s,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ unsigned int i, c, channels = p->pcm_channels;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c)
+ buffer[p->pcm_positions[c]] = cpu_to_be32(0x40000000);
+ buffer += s->data_block_quadlets;
+ }
+}
+
+/**
+ * amdtp_am824_set_pcm_format - set the PCM format
+ * @s: the AMDTP stream to configure
+ * @format: the format of the ALSA PCM device
+ *
+ * The sample format must be set after the other parameters (rate/PCM channels/
+ * MIDI) and before the stream is started, and must not be changed while the
+ * stream is running.
+ */
+void amdtp_am824_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format)
+{
+ struct amdtp_am824 *p = s->protocol;
+
+ if (WARN_ON(amdtp_stream_pcm_running(s)))
+ return;
+
+ switch (format) {
+ default:
+ WARN_ON(1);
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S16:
+ if (s->direction == AMDTP_OUT_STREAM) {
+ p->transfer_samples = write_pcm_s16;
+ break;
+ }
+ WARN_ON(1);
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S32:
+ if (s->direction == AMDTP_OUT_STREAM)
+ p->transfer_samples = write_pcm_s32;
+ else
+ p->transfer_samples = read_pcm_s32;
+ break;
+ }
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_set_pcm_format);
+
+/**
+ * amdtp_am824_add_pcm_hw_constraints - add hw constraints for PCM substream
+ * @s: the AMDTP stream for AM824 data block, must be initialized.
+ * @runtime: the PCM substream runtime
+ *
+ */
+int amdtp_am824_add_pcm_hw_constraints(struct amdtp_stream *s,
+ struct snd_pcm_runtime *runtime)
+{
+ int err;
+
+ err = amdtp_stream_add_pcm_hw_constraints(s, runtime);
+ if (err < 0)
+ return err;
+
+ /* AM824 in IEC 61883-6 can deliver 24bit data. */
+ return snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_add_pcm_hw_constraints);
+
+/**
+ * amdtp_am824_midi_trigger - start/stop playback/capture with a MIDI device
+ * @s: the AMDTP stream
+ * @port: index of MIDI port
+ * @midi: the MIDI device to be started, or %NULL to stop the current device
+ *
+ * Call this function on a running isochronous stream to enable the actual
+ * transmission of MIDI data. This function should be called from the MIDI
+ * device's .trigger callback.
+ */
+void amdtp_am824_midi_trigger(struct amdtp_stream *s, unsigned int port,
+ struct snd_rawmidi_substream *midi)
+{
+ struct amdtp_am824 *p = s->protocol;
+
+ if (port < p->midi_ports)
+ ACCESS_ONCE(p->midi[port]) = midi;
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_midi_trigger);
+
+/*
+ * To avoid sending MIDI bytes at too high a rate, assume that the receiving
+ * device has a FIFO, and track how much it is filled. This values increases
+ * by one whenever we send one byte in a packet, but the FIFO empties at
+ * a constant rate independent of our packet rate. One packet has syt_interval
+ * samples, so the number of bytes that empty out of the FIFO, per packet(!),
+ * is MIDI_BYTES_PER_SECOND * syt_interval / sample_rate. To avoid storing
+ * fractional values, the values in midi_fifo_used[] are measured in bytes
+ * multiplied by the sample rate.
+ */
+static bool midi_ratelimit_per_packet(struct amdtp_stream *s, unsigned int port)
+{
+ struct amdtp_am824 *p = s->protocol;
+ int used;
+
+ used = p->midi_fifo_used[port];
+ if (used == 0) /* common shortcut */
+ return true;
+
+ used -= MIDI_BYTES_PER_SECOND * s->syt_interval;
+ used = max(used, 0);
+ p->midi_fifo_used[port] = used;
+
+ return used < p->midi_fifo_limit;
+}
+
+static void midi_rate_use_one_byte(struct amdtp_stream *s, unsigned int port)
+{
+ struct amdtp_am824 *p = s->protocol;
+
+ p->midi_fifo_used[port] += amdtp_rate_table[s->sfc];
+}
+
+static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ unsigned int f, port;
+ u8 *b;
+
+ for (f = 0; f < frames; f++) {
+ b = (u8 *)&buffer[p->midi_position];
+
+ port = (s->data_block_counter + f) % 8;
+ if (f < MAX_MIDI_RX_BLOCKS &&
+ midi_ratelimit_per_packet(s, port) &&
+ p->midi[port] != NULL &&
+ snd_rawmidi_transmit(p->midi[port], &b[1], 1) == 1) {
+ midi_rate_use_one_byte(s, port);
+ b[0] = 0x81;
+ } else {
+ b[0] = 0x80;
+ b[1] = 0;
+ }
+ b[2] = 0;
+ b[3] = 0;
+
+ buffer += s->data_block_quadlets;
+ }
+}
+
+static void read_midi_messages(struct amdtp_stream *s,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_am824 *p = s->protocol;
+ unsigned int f, port;
+ int len;
+ u8 *b;
+
+ for (f = 0; f < frames; f++) {
+ port = (s->data_block_counter + f) % 8;
+ b = (u8 *)&buffer[p->midi_position];
+
+ len = b[0] - 0x80;
+ if ((1 <= len) && (len <= 3) && (p->midi[port]))
+ snd_rawmidi_receive(p->midi[port], b + 1, len);
+
+ buffer += s->data_block_quadlets;
+ }
+}
+
+static unsigned int process_rx_data_blocks(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int data_blocks, unsigned int *syt)
+{
+ struct amdtp_am824 *p = s->protocol;
+ struct snd_pcm_substream *pcm = ACCESS_ONCE(s->pcm);
+ unsigned int pcm_frames;
+
+ if (pcm) {
+ p->transfer_samples(s, pcm, buffer, data_blocks);
+ pcm_frames = data_blocks * p->frame_multiplier;
+ } else {
+ write_pcm_silence(s, buffer, data_blocks);
+ pcm_frames = 0;
+ }
+
+ if (p->midi_ports)
+ write_midi_messages(s, buffer, data_blocks);
+
+ return pcm_frames;
+}
+
+static unsigned int process_tx_data_blocks(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int data_blocks, unsigned int *syt)
+{
+ struct amdtp_am824 *p = s->protocol;
+ struct snd_pcm_substream *pcm = ACCESS_ONCE(s->pcm);
+ unsigned int pcm_frames;
+
+ if (pcm) {
+ p->transfer_samples(s, pcm, buffer, data_blocks);
+ pcm_frames = data_blocks * p->frame_multiplier;
+ } else {
+ pcm_frames = 0;
+ }
+
+ if (p->midi_ports)
+ read_midi_messages(s, buffer, data_blocks);
+
+ return pcm_frames;
+}
+
+/**
+ * amdtp_am824_init - initialize an AMDTP stream structure to handle AM824
+ * data block
+ * @s: the AMDTP stream to initialize
+ * @unit: the target of the stream
+ * @dir: the direction of stream
+ * @flags: the packet transmission method to use
+ */
+int amdtp_am824_init(struct amdtp_stream *s, struct fw_unit *unit,
+ enum amdtp_stream_direction dir, enum cip_flags flags)
+{
+ amdtp_stream_process_data_blocks_t process_data_blocks;
+
+ if (dir == AMDTP_IN_STREAM)
+ process_data_blocks = process_tx_data_blocks;
+ else
+ process_data_blocks = process_rx_data_blocks;
+
+ return amdtp_stream_init(s, unit, dir, flags, CIP_FMT_AM,
+ process_data_blocks,
+ sizeof(struct amdtp_am824));
+}
+EXPORT_SYMBOL_GPL(amdtp_am824_init);
diff --git a/sound/firewire/amdtp-am824.h b/sound/firewire/amdtp-am824.h
new file mode 100644
index 000000000000..73b07b3109db
--- /dev/null
+++ b/sound/firewire/amdtp-am824.h
@@ -0,0 +1,52 @@
+#ifndef SOUND_FIREWIRE_AMDTP_AM824_H_INCLUDED
+#define SOUND_FIREWIRE_AMDTP_AM824_H_INCLUDED
+
+#include <sound/pcm.h>
+#include <sound/rawmidi.h>
+
+#include "amdtp-stream.h"
+
+#define AM824_IN_PCM_FORMAT_BITS SNDRV_PCM_FMTBIT_S32
+
+#define AM824_OUT_PCM_FORMAT_BITS (SNDRV_PCM_FMTBIT_S16 | \
+ SNDRV_PCM_FMTBIT_S32)
+
+/*
+ * This module supports maximum 64 PCM channels for one PCM stream
+ * This is for our convenience.
+ */
+#define AM824_MAX_CHANNELS_FOR_PCM 64
+
+/*
+ * AMDTP packet can include channels for MIDI conformant data.
+ * Each MIDI conformant data channel includes 8 MPX-MIDI data stream.
+ * Each MPX-MIDI data stream includes one data stream from/to MIDI ports.
+ *
+ * This module supports maximum 1 MIDI conformant data channels.
+ * Then this AMDTP packets can transfer maximum 8 MIDI data streams.
+ */
+#define AM824_MAX_CHANNELS_FOR_MIDI 1
+
+int amdtp_am824_set_parameters(struct amdtp_stream *s, unsigned int rate,
+ unsigned int pcm_channels,
+ unsigned int midi_ports,
+ bool double_pcm_frames);
+
+void amdtp_am824_set_pcm_position(struct amdtp_stream *s, unsigned int index,
+ unsigned int position);
+
+void amdtp_am824_set_midi_position(struct amdtp_stream *s,
+ unsigned int position);
+
+int amdtp_am824_add_pcm_hw_constraints(struct amdtp_stream *s,
+ struct snd_pcm_runtime *runtime);
+
+void amdtp_am824_set_pcm_format(struct amdtp_stream *s,
+ snd_pcm_format_t format);
+
+void amdtp_am824_midi_trigger(struct amdtp_stream *s, unsigned int port,
+ struct snd_rawmidi_substream *midi);
+
+int amdtp_am824_init(struct amdtp_stream *s, struct fw_unit *unit,
+ enum amdtp_stream_direction dir, enum cip_flags flags);
+#endif
diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp-stream.c
index 2a153d260836..ed2902609a4c 100644
--- a/sound/firewire/amdtp.c
+++ b/sound/firewire/amdtp-stream.c
@@ -11,28 +11,14 @@
#include <linux/firewire.h>
#include <linux/module.h>
#include <linux/slab.h>
-#include <linux/sched.h>
#include <sound/pcm.h>
#include <sound/pcm_params.h>
-#include <sound/rawmidi.h>
-#include "amdtp.h"
+#include "amdtp-stream.h"
#define TICKS_PER_CYCLE 3072
#define CYCLES_PER_SECOND 8000
#define TICKS_PER_SECOND (TICKS_PER_CYCLE * CYCLES_PER_SECOND)
-/*
- * Nominally 3125 bytes/second, but the MIDI port's clock might be
- * 1% too slow, and the bus clock 100 ppm too fast.
- */
-#define MIDI_BYTES_PER_SECOND 3093
-
-/*
- * Several devices look only at the first eight data blocks.
- * In any case, this is more than enough for the MIDI data rate.
- */
-#define MAX_MIDI_RX_BLOCKS 8
-
#define TRANSFER_DELAY_TICKS 0x2e00 /* 479.17 microseconds */
/* isochronous header parameters */
@@ -55,12 +41,8 @@
#define CIP_SYT_MASK 0x0000ffff
#define CIP_SYT_NO_INFO 0xffff
-/*
- * Audio and Music transfer protocol specific parameters
- * only "Clock-based rate control mode" is supported
- */
-#define CIP_FMT_AM (0x10 << CIP_FMT_SHIFT)
-#define AMDTP_FDF_AM824 (0 << (CIP_FDF_SHIFT + 3))
+/* Audio and Music transfer protocol specific parameters */
+#define CIP_FMT_AM 0x10
#define AMDTP_FDF_NO_DATA 0xff
/* TODO: make these configurable */
@@ -78,10 +60,23 @@ static void pcm_period_tasklet(unsigned long data);
* @unit: the target of the stream
* @dir: the direction of stream
* @flags: the packet transmission method to use
+ * @fmt: the value of fmt field in CIP header
+ * @process_data_blocks: callback handler to process data blocks
+ * @protocol_size: the size to allocate newly for protocol
*/
int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
- enum amdtp_stream_direction dir, enum cip_flags flags)
+ enum amdtp_stream_direction dir, enum cip_flags flags,
+ unsigned int fmt,
+ amdtp_stream_process_data_blocks_t process_data_blocks,
+ unsigned int protocol_size)
{
+ if (process_data_blocks == NULL)
+ return -EINVAL;
+
+ s->protocol = kzalloc(protocol_size, GFP_KERNEL);
+ if (!s->protocol)
+ return -ENOMEM;
+
s->unit = unit;
s->direction = dir;
s->flags = flags;
@@ -94,6 +89,9 @@ int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
s->callbacked = false;
s->sync_slave = NULL;
+ s->fmt = fmt;
+ s->process_data_blocks = process_data_blocks;
+
return 0;
}
EXPORT_SYMBOL(amdtp_stream_init);
@@ -105,6 +103,7 @@ EXPORT_SYMBOL(amdtp_stream_init);
void amdtp_stream_destroy(struct amdtp_stream *s)
{
WARN_ON(amdtp_stream_running(s));
+ kfree(s->protocol);
mutex_destroy(&s->mutex);
}
EXPORT_SYMBOL(amdtp_stream_destroy);
@@ -141,11 +140,6 @@ int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s,
{
int err;
- /* AM824 in IEC 61883-6 can deliver 24bit data */
- err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
- if (err < 0)
- goto end;
-
/*
* Currently firewire-lib processes 16 packets in one software
* interrupt callback. This equals to 2msec but actually the
@@ -190,39 +184,25 @@ EXPORT_SYMBOL(amdtp_stream_add_pcm_hw_constraints);
* amdtp_stream_set_parameters - set stream parameters
* @s: the AMDTP stream to configure
* @rate: the sample rate
- * @pcm_channels: the number of PCM samples in each data block, to be encoded
- * as AM824 multi-bit linear audio
- * @midi_ports: the number of MIDI ports (i.e., MPX-MIDI Data Channels)
+ * @data_block_quadlets: the size of a data block in quadlet unit
*
* The parameters must be set before the stream is started, and must not be
* changed while the stream is running.
*/
-void amdtp_stream_set_parameters(struct amdtp_stream *s,
- unsigned int rate,
- unsigned int pcm_channels,
- unsigned int midi_ports)
+int amdtp_stream_set_parameters(struct amdtp_stream *s, unsigned int rate,
+ unsigned int data_block_quadlets)
{
- unsigned int i, sfc, midi_channels;
+ unsigned int sfc;
- midi_channels = DIV_ROUND_UP(midi_ports, 8);
-
- if (WARN_ON(amdtp_stream_running(s)) |
- WARN_ON(pcm_channels > AMDTP_MAX_CHANNELS_FOR_PCM) |
- WARN_ON(midi_channels > AMDTP_MAX_CHANNELS_FOR_MIDI))
- return;
-
- for (sfc = 0; sfc < ARRAY_SIZE(amdtp_rate_table); ++sfc)
+ for (sfc = 0; sfc < ARRAY_SIZE(amdtp_rate_table); ++sfc) {
if (amdtp_rate_table[sfc] == rate)
- goto sfc_found;
- WARN_ON(1);
- return;
+ break;
+ }
+ if (sfc == ARRAY_SIZE(amdtp_rate_table))
+ return -EINVAL;
-sfc_found:
- s->pcm_channels = pcm_channels;
s->sfc = sfc;
- s->data_block_quadlets = s->pcm_channels + midi_channels;
- s->midi_ports = midi_ports;
-
+ s->data_block_quadlets = data_block_quadlets;
s->syt_interval = amdtp_syt_intervals[sfc];
/* default buffering in the device */
@@ -231,18 +211,7 @@ sfc_found:
/* additional buffering needed to adjust for no-data packets */
s->transfer_delay += TICKS_PER_SECOND * s->syt_interval / rate;
- /* init the position map for PCM and MIDI channels */
- for (i = 0; i < pcm_channels; i++)
- s->pcm_positions[i] = i;
- s->midi_position = s->pcm_channels;
-
- /*
- * We do not know the actual MIDI FIFO size of most devices. Just
- * assume two bytes, i.e., one byte can be received over the bus while
- * the previous one is transmitted over MIDI.
- * (The value here is adjusted for midi_ratelimit_per_packet().)
- */
- s->midi_fifo_limit = rate - MIDI_BYTES_PER_SECOND * s->syt_interval + 1;
+ return 0;
}
EXPORT_SYMBOL(amdtp_stream_set_parameters);
@@ -264,52 +233,6 @@ unsigned int amdtp_stream_get_max_payload(struct amdtp_stream *s)
}
EXPORT_SYMBOL(amdtp_stream_get_max_payload);
-static void write_pcm_s16(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames);
-static void write_pcm_s32(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames);
-static void read_pcm_s32(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames);
-
-/**
- * amdtp_stream_set_pcm_format - set the PCM format
- * @s: the AMDTP stream to configure
- * @format: the format of the ALSA PCM device
- *
- * The sample format must be set after the other parameters (rate/PCM channels/
- * MIDI) and before the stream is started, and must not be changed while the
- * stream is running.
- */
-void amdtp_stream_set_pcm_format(struct amdtp_stream *s,
- snd_pcm_format_t format)
-{
- if (WARN_ON(amdtp_stream_pcm_running(s)))
- return;
-
- switch (format) {
- default:
- WARN_ON(1);
- /* fall through */
- case SNDRV_PCM_FORMAT_S16:
- if (s->direction == AMDTP_OUT_STREAM) {
- s->transfer_samples = write_pcm_s16;
- break;
- }
- WARN_ON(1);
- /* fall through */
- case SNDRV_PCM_FORMAT_S32:
- if (s->direction == AMDTP_OUT_STREAM)
- s->transfer_samples = write_pcm_s32;
- else
- s->transfer_samples = read_pcm_s32;
- break;
- }
-}
-EXPORT_SYMBOL(amdtp_stream_set_pcm_format);
-
/**
* amdtp_stream_pcm_prepare - prepare PCM device for running
* @s: the AMDTP stream
@@ -412,182 +335,12 @@ static unsigned int calculate_syt(struct amdtp_stream *s,
}
}
-static void write_pcm_s32(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames)
-{
- struct snd_pcm_runtime *runtime = pcm->runtime;
- unsigned int channels, remaining_frames, i, c;
- const u32 *src;
-
- channels = s->pcm_channels;
- src = (void *)runtime->dma_area +
- frames_to_bytes(runtime, s->pcm_buffer_pointer);
- remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
-
- for (i = 0; i < frames; ++i) {
- for (c = 0; c < channels; ++c) {
- buffer[s->pcm_positions[c]] =
- cpu_to_be32((*src >> 8) | 0x40000000);
- src++;
- }
- buffer += s->data_block_quadlets;
- if (--remaining_frames == 0)
- src = (void *)runtime->dma_area;
- }
-}
-
-static void write_pcm_s16(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames)
-{
- struct snd_pcm_runtime *runtime = pcm->runtime;
- unsigned int channels, remaining_frames, i, c;
- const u16 *src;
-
- channels = s->pcm_channels;
- src = (void *)runtime->dma_area +
- frames_to_bytes(runtime, s->pcm_buffer_pointer);
- remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
-
- for (i = 0; i < frames; ++i) {
- for (c = 0; c < channels; ++c) {
- buffer[s->pcm_positions[c]] =
- cpu_to_be32((*src << 8) | 0x42000000);
- src++;
- }
- buffer += s->data_block_quadlets;
- if (--remaining_frames == 0)
- src = (void *)runtime->dma_area;
- }
-}
-
-static void read_pcm_s32(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames)
-{
- struct snd_pcm_runtime *runtime = pcm->runtime;
- unsigned int channels, remaining_frames, i, c;
- u32 *dst;
-
- channels = s->pcm_channels;
- dst = (void *)runtime->dma_area +
- frames_to_bytes(runtime, s->pcm_buffer_pointer);
- remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
-
- for (i = 0; i < frames; ++i) {
- for (c = 0; c < channels; ++c) {
- *dst = be32_to_cpu(buffer[s->pcm_positions[c]]) << 8;
- dst++;
- }
- buffer += s->data_block_quadlets;
- if (--remaining_frames == 0)
- dst = (void *)runtime->dma_area;
- }
-}
-
-static void write_pcm_silence(struct amdtp_stream *s,
- __be32 *buffer, unsigned int frames)
-{
- unsigned int i, c;
-
- for (i = 0; i < frames; ++i) {
- for (c = 0; c < s->pcm_channels; ++c)
- buffer[s->pcm_positions[c]] = cpu_to_be32(0x40000000);
- buffer += s->data_block_quadlets;
- }
-}
-
-/*
- * To avoid sending MIDI bytes at too high a rate, assume that the receiving
- * device has a FIFO, and track how much it is filled. This values increases
- * by one whenever we send one byte in a packet, but the FIFO empties at
- * a constant rate independent of our packet rate. One packet has syt_interval
- * samples, so the number of bytes that empty out of the FIFO, per packet(!),
- * is MIDI_BYTES_PER_SECOND * syt_interval / sample_rate. To avoid storing
- * fractional values, the values in midi_fifo_used[] are measured in bytes
- * multiplied by the sample rate.
- */
-static bool midi_ratelimit_per_packet(struct amdtp_stream *s, unsigned int port)
-{
- int used;
-
- used = s->midi_fifo_used[port];
- if (used == 0) /* common shortcut */
- return true;
-
- used -= MIDI_BYTES_PER_SECOND * s->syt_interval;
- used = max(used, 0);
- s->midi_fifo_used[port] = used;
-
- return used < s->midi_fifo_limit;
-}
-
-static void midi_rate_use_one_byte(struct amdtp_stream *s, unsigned int port)
-{
- s->midi_fifo_used[port] += amdtp_rate_table[s->sfc];
-}
-
-static void write_midi_messages(struct amdtp_stream *s,
- __be32 *buffer, unsigned int frames)
-{
- unsigned int f, port;
- u8 *b;
-
- for (f = 0; f < frames; f++) {
- b = (u8 *)&buffer[s->midi_position];
-
- port = (s->data_block_counter + f) % 8;
- if (f < MAX_MIDI_RX_BLOCKS &&
- midi_ratelimit_per_packet(s, port) &&
- s->midi[port] != NULL &&
- snd_rawmidi_transmit(s->midi[port], &b[1], 1) == 1) {
- midi_rate_use_one_byte(s, port);
- b[0] = 0x81;
- } else {
- b[0] = 0x80;
- b[1] = 0;
- }
- b[2] = 0;
- b[3] = 0;
-
- buffer += s->data_block_quadlets;
- }
-}
-
-static void read_midi_messages(struct amdtp_stream *s,
- __be32 *buffer, unsigned int frames)
-{
- unsigned int f, port;
- int len;
- u8 *b;
-
- for (f = 0; f < frames; f++) {
- port = (s->data_block_counter + f) % 8;
- b = (u8 *)&buffer[s->midi_position];
-
- len = b[0] - 0x80;
- if ((1 <= len) && (len <= 3) && (s->midi[port]))
- snd_rawmidi_receive(s->midi[port], b + 1, len);
-
- buffer += s->data_block_quadlets;
- }
-}
-
static void update_pcm_pointers(struct amdtp_stream *s,
struct snd_pcm_substream *pcm,
unsigned int frames)
{
unsigned int ptr;
- /*
- * In IEC 61883-6, one data block represents one event. In ALSA, one
- * event equals to one PCM frame. But Dice has a quirk to transfer
- * two PCM frames in one data block.
- */
- if (s->double_pcm_frames)
- frames *= 2;
-
ptr = s->pcm_buffer_pointer + frames;
if (ptr >= pcm->runtime->buffer_size)
ptr -= pcm->runtime->buffer_size;
@@ -656,23 +409,19 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks,
{
__be32 *buffer;
unsigned int payload_length;
+ unsigned int pcm_frames;
struct snd_pcm_substream *pcm;
buffer = s->buffer.packets[s->packet_index].buffer;
+ pcm_frames = s->process_data_blocks(s, buffer + 2, data_blocks, &syt);
+
buffer[0] = cpu_to_be32(ACCESS_ONCE(s->source_node_id_field) |
(s->data_block_quadlets << CIP_DBS_SHIFT) |
s->data_block_counter);
- buffer[1] = cpu_to_be32(CIP_EOH | CIP_FMT_AM | AMDTP_FDF_AM824 |
- (s->sfc << CIP_FDF_SHIFT) | syt);
- buffer += 2;
-
- pcm = ACCESS_ONCE(s->pcm);
- if (pcm)
- s->transfer_samples(s, pcm, buffer, data_blocks);
- else
- write_pcm_silence(s, buffer, data_blocks);
- if (s->midi_ports)
- write_midi_messages(s, buffer, data_blocks);
+ buffer[1] = cpu_to_be32(CIP_EOH |
+ ((s->fmt << CIP_FMT_SHIFT) & CIP_FMT_MASK) |
+ ((s->fdf << CIP_FDF_SHIFT) & CIP_FDF_MASK) |
+ (syt & CIP_SYT_MASK));
s->data_block_counter = (s->data_block_counter + data_blocks) & 0xff;
@@ -680,8 +429,9 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks,
if (queue_out_packet(s, payload_length, false) < 0)
return -EIO;
- if (pcm)
- update_pcm_pointers(s, pcm, data_blocks);
+ pcm = ACCESS_ONCE(s->pcm);
+ if (pcm && pcm_frames > 0)
+ update_pcm_pointers(s, pcm, pcm_frames);
/* No need to return the number of handled data blocks. */
return 0;
@@ -689,11 +439,13 @@ static int handle_out_packet(struct amdtp_stream *s, unsigned int data_blocks,
static int handle_in_packet(struct amdtp_stream *s,
unsigned int payload_quadlets, __be32 *buffer,
- unsigned int *data_blocks)
+ unsigned int *data_blocks, unsigned int syt)
{
u32 cip_header[2];
+ unsigned int fmt, fdf;
unsigned int data_block_quadlets, data_block_counter, dbc_interval;
- struct snd_pcm_substream *pcm = NULL;
+ struct snd_pcm_substream *pcm;
+ unsigned int pcm_frames;
bool lost;
cip_header[0] = be32_to_cpu(buffer[0]);
@@ -704,19 +456,30 @@ static int handle_in_packet(struct amdtp_stream *s,
* For convenience, also check FMT field is AM824 or not.
*/
if (((cip_header[0] & CIP_EOH_MASK) == CIP_EOH) ||
- ((cip_header[1] & CIP_EOH_MASK) != CIP_EOH) ||
- ((cip_header[1] & CIP_FMT_MASK) != CIP_FMT_AM)) {
+ ((cip_header[1] & CIP_EOH_MASK) != CIP_EOH)) {
dev_info_ratelimited(&s->unit->device,
"Invalid CIP header for AMDTP: %08X:%08X\n",
cip_header[0], cip_header[1]);
*data_blocks = 0;
+ pcm_frames = 0;
+ goto end;
+ }
+
+ /* Check valid protocol or not. */
+ fmt = (cip_header[1] & CIP_FMT_MASK) >> CIP_FMT_SHIFT;
+ if (fmt != s->fmt) {
+ dev_info_ratelimited(&s->unit->device,
+ "Detect unexpected protocol: %08x %08x\n",
+ cip_header[0], cip_header[1]);
+ *data_blocks = 0;
+ pcm_frames = 0;
goto end;
}
/* Calculate data blocks */
+ fdf = (cip_header[1] & CIP_FDF_MASK) >> CIP_FDF_SHIFT;
if (payload_quadlets < 3 ||
- ((cip_header[1] & CIP_FDF_MASK) ==
- (AMDTP_FDF_NO_DATA << CIP_FDF_SHIFT))) {
+ (fmt == CIP_FMT_AM && fdf == AMDTP_FDF_NO_DATA)) {
*data_blocks = 0;
} else {
data_block_quadlets =
@@ -763,16 +526,7 @@ static int handle_in_packet(struct amdtp_stream *s,
return -EIO;
}
- if (*data_blocks > 0) {
- buffer += 2;
-
- pcm = ACCESS_ONCE(s->pcm);
- if (pcm)
- s->transfer_samples(s, pcm, buffer, *data_blocks);
-
- if (s->midi_ports)
- read_midi_messages(s, buffer, *data_blocks);
- }
+ pcm_frames = s->process_data_blocks(s, buffer + 2, *data_blocks, &syt);
if (s->flags & CIP_DBC_IS_END_EVENT)
s->data_block_counter = data_block_counter;
@@ -783,8 +537,9 @@ end:
if (queue_in_packet(s) < 0)
return -EIO;
- if (pcm)
- update_pcm_pointers(s, pcm, *data_blocks);
+ pcm = ACCESS_ONCE(s->pcm);
+ if (pcm && pcm_frames > 0)
+ update_pcm_pointers(s, pcm, pcm_frames);
return 0;
}
@@ -854,15 +609,15 @@ static void in_stream_callback(struct fw_iso_context *context, u32 cycle,
break;
}
+ syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK;
if (handle_in_packet(s, payload_quadlets, buffer,
- &data_blocks) < 0) {
+ &data_blocks, syt) < 0) {
s->packet_index = -1;
break;
}
/* Process sync slave stream */
if (s->sync_slave && s->sync_slave->callbacked) {
- syt = be32_to_cpu(buffer[1]) & CIP_SYT_MASK;
if (handle_out_packet(s->sync_slave,
data_blocks, syt) < 0) {
s->packet_index = -1;
diff --git a/sound/firewire/amdtp.h b/sound/firewire/amdtp-stream.h
index b2cf9e75693b..8775704a3665 100644
--- a/sound/firewire/amdtp.h
+++ b/sound/firewire/amdtp-stream.h
@@ -4,6 +4,7 @@
#include <linux/err.h>
#include <linux/interrupt.h>
#include <linux/mutex.h>
+#include <linux/sched.h>
#include <sound/asound.h>
#include "packets-buffer.h"
@@ -80,100 +81,78 @@ enum cip_sfc {
CIP_SFC_COUNT
};
-#define AMDTP_IN_PCM_FORMAT_BITS SNDRV_PCM_FMTBIT_S32
-
-#define AMDTP_OUT_PCM_FORMAT_BITS (SNDRV_PCM_FMTBIT_S16 | \
- SNDRV_PCM_FMTBIT_S32)
-
-
-/*
- * This module supports maximum 64 PCM channels for one PCM stream
- * This is for our convenience.
- */
-#define AMDTP_MAX_CHANNELS_FOR_PCM 64
-
-/*
- * AMDTP packet can include channels for MIDI conformant data.
- * Each MIDI conformant data channel includes 8 MPX-MIDI data stream.
- * Each MPX-MIDI data stream includes one data stream from/to MIDI ports.
- *
- * This module supports maximum 1 MIDI conformant data channels.
- * Then this AMDTP packets can transfer maximum 8 MIDI data streams.
- */
-#define AMDTP_MAX_CHANNELS_FOR_MIDI 1
-
struct fw_unit;
struct fw_iso_context;
struct snd_pcm_substream;
struct snd_pcm_runtime;
-struct snd_rawmidi_substream;
enum amdtp_stream_direction {
AMDTP_OUT_STREAM = 0,
AMDTP_IN_STREAM
};
+struct amdtp_stream;
+typedef unsigned int (*amdtp_stream_process_data_blocks_t)(
+ struct amdtp_stream *s,
+ __be32 *buffer,
+ unsigned int data_blocks,
+ unsigned int *syt);
struct amdtp_stream {
struct fw_unit *unit;
enum cip_flags flags;
enum amdtp_stream_direction direction;
- struct fw_iso_context *context;
struct mutex mutex;
- enum cip_sfc sfc;
- unsigned int data_block_quadlets;
- unsigned int pcm_channels;
- unsigned int midi_ports;
- void (*transfer_samples)(struct amdtp_stream *s,
- struct snd_pcm_substream *pcm,
- __be32 *buffer, unsigned int frames);
- u8 pcm_positions[AMDTP_MAX_CHANNELS_FOR_PCM];
- u8 midi_position;
-
- unsigned int syt_interval;
- unsigned int transfer_delay;
- unsigned int source_node_id_field;
+ /* For packet processing. */
+ struct fw_iso_context *context;
struct iso_packets_buffer buffer;
-
- struct snd_pcm_substream *pcm;
- struct tasklet_struct period_tasklet;
-
int packet_index;
+
+ /* For CIP headers. */
+ unsigned int source_node_id_field;
+ unsigned int data_block_quadlets;
unsigned int data_block_counter;
+ unsigned int fmt;
+ unsigned int fdf;
+ /* quirk: fixed interval of dbc between previos/current packets. */
+ unsigned int tx_dbc_interval;
+ /* quirk: indicate the value of dbc field in a first packet. */
+ unsigned int tx_first_dbc;
+ /* Internal flags. */
+ enum cip_sfc sfc;
+ unsigned int syt_interval;
+ unsigned int transfer_delay;
unsigned int data_block_state;
-
unsigned int last_syt_offset;
unsigned int syt_offset_state;
+ /* For a PCM substream processing. */
+ struct snd_pcm_substream *pcm;
+ struct tasklet_struct period_tasklet;
unsigned int pcm_buffer_pointer;
unsigned int pcm_period_pointer;
bool pointer_flush;
- bool double_pcm_frames;
-
- struct snd_rawmidi_substream *midi[AMDTP_MAX_CHANNELS_FOR_MIDI * 8];
- int midi_fifo_limit;
- int midi_fifo_used[AMDTP_MAX_CHANNELS_FOR_MIDI * 8];
-
- /* quirk: fixed interval of dbc between previos/current packets. */
- unsigned int tx_dbc_interval;
- /* quirk: indicate the value of dbc field in a first packet. */
- unsigned int tx_first_dbc;
+ /* To wait for first packet. */
bool callbacked;
wait_queue_head_t callback_wait;
struct amdtp_stream *sync_slave;
+
+ /* For backends to process data blocks. */
+ void *protocol;
+ amdtp_stream_process_data_blocks_t process_data_blocks;
};
int amdtp_stream_init(struct amdtp_stream *s, struct fw_unit *unit,
- enum amdtp_stream_direction dir,
- enum cip_flags flags);
+ enum amdtp_stream_direction dir, enum cip_flags flags,
+ unsigned int fmt,
+ amdtp_stream_process_data_blocks_t process_data_blocks,
+ unsigned int protocol_size);
void amdtp_stream_destroy(struct amdtp_stream *s);
-void amdtp_stream_set_parameters(struct amdtp_stream *s,
- unsigned int rate,
- unsigned int pcm_channels,
- unsigned int midi_ports);
+int amdtp_stream_set_parameters(struct amdtp_stream *s, unsigned int rate,
+ unsigned int data_block_quadlets);
unsigned int amdtp_stream_get_max_payload(struct amdtp_stream *s);
int amdtp_stream_start(struct amdtp_stream *s, int channel, int speed);
@@ -182,8 +161,7 @@ void amdtp_stream_stop(struct amdtp_stream *s);
int amdtp_stream_add_pcm_hw_constraints(struct amdtp_stream *s,
struct snd_pcm_runtime *runtime);
-void amdtp_stream_set_pcm_format(struct amdtp_stream *s,
- snd_pcm_format_t format);
+
void amdtp_stream_pcm_prepare(struct amdtp_stream *s);
unsigned long amdtp_stream_pcm_pointer(struct amdtp_stream *s);
void amdtp_stream_pcm_abort(struct amdtp_stream *s);
@@ -240,24 +218,6 @@ static inline void amdtp_stream_pcm_trigger(struct amdtp_stream *s,
ACCESS_ONCE(s->pcm) = pcm;
}
-/**
- * amdtp_stream_midi_trigger - start/stop playback/capture with a MIDI device
- * @s: the AMDTP stream
- * @port: index of MIDI port
- * @midi: the MIDI device to be started, or %NULL to stop the current device
- *
- * Call this function on a running isochronous stream to enable the actual
- * transmission of MIDI data. This function should be called from the MIDI
- * device's .trigger callback.
- */
-static inline void amdtp_stream_midi_trigger(struct amdtp_stream *s,
- unsigned int port,
- struct snd_rawmidi_substream *midi)
-{
- if (port < s->midi_ports)
- ACCESS_ONCE(s->midi[port]) = midi;
-}
-
static inline bool cip_sfc_is_base_44100(enum cip_sfc sfc)
{
return sfc & 1;
diff --git a/sound/firewire/bebob/Makefile b/sound/firewire/bebob/Makefile
index 6cf470c80d1f..af7ed6643266 100644
--- a/sound/firewire/bebob/Makefile
+++ b/sound/firewire/bebob/Makefile
@@ -1,4 +1,4 @@
snd-bebob-objs := bebob_command.o bebob_stream.o bebob_proc.o bebob_midi.o \
bebob_pcm.o bebob_hwdep.o bebob_terratec.o bebob_yamaha.o \
bebob_focusrite.o bebob_maudio.o bebob.o
-obj-m += snd-bebob.o
+obj-$(CONFIG_SND_BEBOB) += snd-bebob.o
diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c
index 27a04ac8ffee..091290d1f3ea 100644
--- a/sound/firewire/bebob/bebob.c
+++ b/sound/firewire/bebob/bebob.c
@@ -41,7 +41,8 @@ static DECLARE_BITMAP(devices_used, SNDRV_CARDS);
#define VEN_EDIROL 0x000040ab
#define VEN_PRESONUS 0x00000a92
#define VEN_BRIDGECO 0x000007f5
-#define VEN_MACKIE 0x0000000f
+#define VEN_MACKIE1 0x0000000f
+#define VEN_MACKIE2 0x00000ff2
#define VEN_STANTON 0x00001260
#define VEN_TASCAM 0x0000022e
#define VEN_BEHRINGER 0x00001564
@@ -334,7 +335,7 @@ static void bebob_remove(struct fw_unit *unit)
snd_card_free_when_closed(bebob->card);
}
-static struct snd_bebob_rate_spec normal_rate_spec = {
+static const struct snd_bebob_rate_spec normal_rate_spec = {
.get = &snd_bebob_stream_get_rate,
.set = &snd_bebob_stream_set_rate
};
@@ -360,9 +361,9 @@ static const struct ieee1394_device_id bebob_id_table[] = {
/* BridgeCo, Audio5 */
SND_BEBOB_DEV_ENTRY(VEN_BRIDGECO, 0x00010049, &spec_normal),
/* Mackie, Onyx 1220/1620/1640 (Firewire I/O Card) */
- SND_BEBOB_DEV_ENTRY(VEN_MACKIE, 0x00010065, &spec_normal),
+ SND_BEBOB_DEV_ENTRY(VEN_MACKIE2, 0x00010065, &spec_normal),
/* Mackie, d.2 (Firewire Option) */
- SND_BEBOB_DEV_ENTRY(VEN_MACKIE, 0x00010067, &spec_normal),
+ SND_BEBOB_DEV_ENTRY(VEN_MACKIE1, 0x00010067, &spec_normal),
/* Stanton, ScratchAmp */
SND_BEBOB_DEV_ENTRY(VEN_STANTON, 0x00000001, &spec_normal),
/* Tascam, IF-FW DM */
diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h
index d23caca7f369..4d8fcc78e747 100644
--- a/sound/firewire/bebob/bebob.h
+++ b/sound/firewire/bebob/bebob.h
@@ -31,7 +31,7 @@
#include "../fcp.h"
#include "../packets-buffer.h"
#include "../iso-resources.h"
-#include "../amdtp.h"
+#include "../amdtp-am824.h"
#include "../cmp.h"
/* basic register addresses on DM1000/DM1100/DM1500 */
@@ -70,9 +70,9 @@ struct snd_bebob_meter_spec {
int (*get)(struct snd_bebob *bebob, u32 *target, unsigned int size);
};
struct snd_bebob_spec {
- struct snd_bebob_clock_spec *clock;
- struct snd_bebob_rate_spec *rate;
- struct snd_bebob_meter_spec *meter;
+ const struct snd_bebob_clock_spec *clock;
+ const struct snd_bebob_rate_spec *rate;
+ const struct snd_bebob_meter_spec *meter;
};
struct snd_bebob {
@@ -235,19 +235,19 @@ int snd_bebob_create_pcm_devices(struct snd_bebob *bebob);
int snd_bebob_create_hwdep_device(struct snd_bebob *bebob);
/* model specific operations */
-extern struct snd_bebob_spec phase88_rack_spec;
-extern struct snd_bebob_spec phase24_series_spec;
-extern struct snd_bebob_spec yamaha_go_spec;
-extern struct snd_bebob_spec saffirepro_26_spec;
-extern struct snd_bebob_spec saffirepro_10_spec;
-extern struct snd_bebob_spec saffire_le_spec;
-extern struct snd_bebob_spec saffire_spec;
-extern struct snd_bebob_spec maudio_fw410_spec;
-extern struct snd_bebob_spec maudio_audiophile_spec;
-extern struct snd_bebob_spec maudio_solo_spec;
-extern struct snd_bebob_spec maudio_ozonic_spec;
-extern struct snd_bebob_spec maudio_nrv10_spec;
-extern struct snd_bebob_spec maudio_special_spec;
+extern const struct snd_bebob_spec phase88_rack_spec;
+extern const struct snd_bebob_spec phase24_series_spec;
+extern const struct snd_bebob_spec yamaha_go_spec;
+extern const struct snd_bebob_spec saffirepro_26_spec;
+extern const struct snd_bebob_spec saffirepro_10_spec;
+extern const struct snd_bebob_spec saffire_le_spec;
+extern const struct snd_bebob_spec saffire_spec;
+extern const struct snd_bebob_spec maudio_fw410_spec;
+extern const struct snd_bebob_spec maudio_audiophile_spec;
+extern const struct snd_bebob_spec maudio_solo_spec;
+extern const struct snd_bebob_spec maudio_ozonic_spec;
+extern const struct snd_bebob_spec maudio_nrv10_spec;
+extern const struct snd_bebob_spec maudio_special_spec;
int snd_bebob_maudio_special_discover(struct snd_bebob *bebob, bool is1814);
int snd_bebob_maudio_load_firmware(struct fw_unit *unit);
diff --git a/sound/firewire/bebob/bebob_focusrite.c b/sound/firewire/bebob/bebob_focusrite.c
index a1a39494ea6c..f11090057949 100644
--- a/sound/firewire/bebob/bebob_focusrite.c
+++ b/sound/firewire/bebob/bebob_focusrite.c
@@ -200,7 +200,7 @@ end:
return err;
}
-struct snd_bebob_spec saffire_le_spec;
+const struct snd_bebob_spec saffire_le_spec;
static enum snd_bebob_clock_type saffire_both_clk_src_types[] = {
SND_BEBOB_CLOCK_TYPE_INTERNAL,
SND_BEBOB_CLOCK_TYPE_EXTERNAL,
@@ -229,7 +229,7 @@ static const char *const saffire_meter_labels[] = {
static int
saffire_meter_get(struct snd_bebob *bebob, u32 *buf, unsigned int size)
{
- struct snd_bebob_meter_spec *spec = bebob->spec->meter;
+ const struct snd_bebob_meter_spec *spec = bebob->spec->meter;
unsigned int channels;
u64 offset;
int err;
@@ -260,60 +260,60 @@ saffire_meter_get(struct snd_bebob *bebob, u32 *buf, unsigned int size)
return err;
}
-static struct snd_bebob_rate_spec saffirepro_both_rate_spec = {
+static const struct snd_bebob_rate_spec saffirepro_both_rate_spec = {
.get = &saffirepro_both_clk_freq_get,
.set = &saffirepro_both_clk_freq_set,
};
/* Saffire Pro 26 I/O */
-static struct snd_bebob_clock_spec saffirepro_26_clk_spec = {
+static const struct snd_bebob_clock_spec saffirepro_26_clk_spec = {
.num = ARRAY_SIZE(saffirepro_26_clk_src_types),
.types = saffirepro_26_clk_src_types,
.get = &saffirepro_both_clk_src_get,
};
-struct snd_bebob_spec saffirepro_26_spec = {
+const struct snd_bebob_spec saffirepro_26_spec = {
.clock = &saffirepro_26_clk_spec,
.rate = &saffirepro_both_rate_spec,
.meter = NULL
};
/* Saffire Pro 10 I/O */
-static struct snd_bebob_clock_spec saffirepro_10_clk_spec = {
+static const struct snd_bebob_clock_spec saffirepro_10_clk_spec = {
.num = ARRAY_SIZE(saffirepro_10_clk_src_types),
.types = saffirepro_10_clk_src_types,
.get = &saffirepro_both_clk_src_get,
};
-struct snd_bebob_spec saffirepro_10_spec = {
+const struct snd_bebob_spec saffirepro_10_spec = {
.clock = &saffirepro_10_clk_spec,
.rate = &saffirepro_both_rate_spec,
.meter = NULL
};
-static struct snd_bebob_rate_spec saffire_both_rate_spec = {
+static const struct snd_bebob_rate_spec saffire_both_rate_spec = {
.get = &snd_bebob_stream_get_rate,
.set = &snd_bebob_stream_set_rate,
};
-static struct snd_bebob_clock_spec saffire_both_clk_spec = {
+static const struct snd_bebob_clock_spec saffire_both_clk_spec = {
.num = ARRAY_SIZE(saffire_both_clk_src_types),
.types = saffire_both_clk_src_types,
.get = &saffire_both_clk_src_get,
};
/* Saffire LE */
-static struct snd_bebob_meter_spec saffire_le_meter_spec = {
+static const struct snd_bebob_meter_spec saffire_le_meter_spec = {
.num = ARRAY_SIZE(saffire_le_meter_labels),
.labels = saffire_le_meter_labels,
.get = &saffire_meter_get,
};
-struct snd_bebob_spec saffire_le_spec = {
+const struct snd_bebob_spec saffire_le_spec = {
.clock = &saffire_both_clk_spec,
.rate = &saffire_both_rate_spec,
.meter = &saffire_le_meter_spec
};
/* Saffire */
-static struct snd_bebob_meter_spec saffire_meter_spec = {
+static const struct snd_bebob_meter_spec saffire_meter_spec = {
.num = ARRAY_SIZE(saffire_meter_labels),
.labels = saffire_meter_labels,
.get = &saffire_meter_get,
};
-struct snd_bebob_spec saffire_spec = {
+const struct snd_bebob_spec saffire_spec = {
.clock = &saffire_both_clk_spec,
.rate = &saffire_both_rate_spec,
.meter = &saffire_meter_spec
diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c
index 057495d54ab0..07e5abdbceb5 100644
--- a/sound/firewire/bebob/bebob_maudio.c
+++ b/sound/firewire/bebob/bebob_maudio.c
@@ -628,7 +628,7 @@ static const char *const special_meter_labels[] = {
static int
special_meter_get(struct snd_bebob *bebob, u32 *target, unsigned int size)
{
- u16 *buf;
+ __be16 *buf;
unsigned int i, c, channels;
int err;
@@ -687,7 +687,7 @@ static const char *const nrv10_meter_labels[] = {
static int
normal_meter_get(struct snd_bebob *bebob, u32 *buf, unsigned int size)
{
- struct snd_bebob_meter_spec *spec = bebob->spec->meter;
+ const struct snd_bebob_meter_spec *spec = bebob->spec->meter;
unsigned int c, channels;
int err;
@@ -712,85 +712,85 @@ end:
}
/* for special customized devices */
-static struct snd_bebob_rate_spec special_rate_spec = {
+static const struct snd_bebob_rate_spec special_rate_spec = {
.get = &special_get_rate,
.set = &special_set_rate,
};
-static struct snd_bebob_clock_spec special_clk_spec = {
+static const struct snd_bebob_clock_spec special_clk_spec = {
.num = ARRAY_SIZE(special_clk_types),
.types = special_clk_types,
.get = &special_clk_get,
};
-static struct snd_bebob_meter_spec special_meter_spec = {
+static const struct snd_bebob_meter_spec special_meter_spec = {
.num = ARRAY_SIZE(special_meter_labels),
.labels = special_meter_labels,
.get = &special_meter_get
};
-struct snd_bebob_spec maudio_special_spec = {
+const struct snd_bebob_spec maudio_special_spec = {
.clock = &special_clk_spec,
.rate = &special_rate_spec,
.meter = &special_meter_spec
};
/* Firewire 410 specification */
-static struct snd_bebob_rate_spec usual_rate_spec = {
+static const struct snd_bebob_rate_spec usual_rate_spec = {
.get = &snd_bebob_stream_get_rate,
.set = &snd_bebob_stream_set_rate,
};
-static struct snd_bebob_meter_spec fw410_meter_spec = {
+static const struct snd_bebob_meter_spec fw410_meter_spec = {
.num = ARRAY_SIZE(fw410_meter_labels),
.labels = fw410_meter_labels,
.get = &normal_meter_get
};
-struct snd_bebob_spec maudio_fw410_spec = {
+const struct snd_bebob_spec maudio_fw410_spec = {
.clock = NULL,
.rate = &usual_rate_spec,
.meter = &fw410_meter_spec
};
/* Firewire Audiophile specification */
-static struct snd_bebob_meter_spec audiophile_meter_spec = {
+static const struct snd_bebob_meter_spec audiophile_meter_spec = {
.num = ARRAY_SIZE(audiophile_meter_labels),
.labels = audiophile_meter_labels,
.get = &normal_meter_get
};
-struct snd_bebob_spec maudio_audiophile_spec = {
+const struct snd_bebob_spec maudio_audiophile_spec = {
.clock = NULL,
.rate = &usual_rate_spec,
.meter = &audiophile_meter_spec
};
/* Firewire Solo specification */
-static struct snd_bebob_meter_spec solo_meter_spec = {
+static const struct snd_bebob_meter_spec solo_meter_spec = {
.num = ARRAY_SIZE(solo_meter_labels),
.labels = solo_meter_labels,
.get = &normal_meter_get
};
-struct snd_bebob_spec maudio_solo_spec = {
+const struct snd_bebob_spec maudio_solo_spec = {
.clock = NULL,
.rate = &usual_rate_spec,
.meter = &solo_meter_spec
};
/* Ozonic specification */
-static struct snd_bebob_meter_spec ozonic_meter_spec = {
+static const struct snd_bebob_meter_spec ozonic_meter_spec = {
.num = ARRAY_SIZE(ozonic_meter_labels),
.labels = ozonic_meter_labels,
.get = &normal_meter_get
};
-struct snd_bebob_spec maudio_ozonic_spec = {
+const struct snd_bebob_spec maudio_ozonic_spec = {
.clock = NULL,
.rate = &usual_rate_spec,
.meter = &ozonic_meter_spec
};
/* NRV10 specification */
-static struct snd_bebob_meter_spec nrv10_meter_spec = {
+static const struct snd_bebob_meter_spec nrv10_meter_spec = {
.num = ARRAY_SIZE(nrv10_meter_labels),
.labels = nrv10_meter_labels,
.get = &normal_meter_get
};
-struct snd_bebob_spec maudio_nrv10_spec = {
+const struct snd_bebob_spec maudio_nrv10_spec = {
.clock = NULL,
.rate = &usual_rate_spec,
.meter = &nrv10_meter_spec
diff --git a/sound/firewire/bebob/bebob_midi.c b/sound/firewire/bebob/bebob_midi.c
index 5681143925cd..90d95be499b0 100644
--- a/sound/firewire/bebob/bebob_midi.c
+++ b/sound/firewire/bebob/bebob_midi.c
@@ -72,11 +72,11 @@ static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up)
spin_lock_irqsave(&bebob->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&bebob->tx_stream,
- substrm->number, substrm);
+ amdtp_am824_midi_trigger(&bebob->tx_stream,
+ substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&bebob->tx_stream,
- substrm->number, NULL);
+ amdtp_am824_midi_trigger(&bebob->tx_stream,
+ substrm->number, NULL);
spin_unlock_irqrestore(&bebob->lock, flags);
}
@@ -89,11 +89,11 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
spin_lock_irqsave(&bebob->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&bebob->rx_stream,
- substrm->number, substrm);
+ amdtp_am824_midi_trigger(&bebob->rx_stream,
+ substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&bebob->rx_stream,
- substrm->number, NULL);
+ amdtp_am824_midi_trigger(&bebob->rx_stream,
+ substrm->number, NULL);
spin_unlock_irqrestore(&bebob->lock, flags);
}
diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c
index c0f018a61fdc..ef224d6f5c24 100644
--- a/sound/firewire/bebob/bebob_pcm.c
+++ b/sound/firewire/bebob/bebob_pcm.c
@@ -122,11 +122,11 @@ pcm_init_hw_params(struct snd_bebob *bebob,
SNDRV_PCM_INFO_MMAP_VALID;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- runtime->hw.formats = AMDTP_IN_PCM_FORMAT_BITS;
+ runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS;
s = &bebob->tx_stream;
formations = bebob->tx_stream_formations;
} else {
- runtime->hw.formats = AMDTP_OUT_PCM_FORMAT_BITS;
+ runtime->hw.formats = AM824_OUT_PCM_FORMAT_BITS;
s = &bebob->rx_stream;
formations = bebob->rx_stream_formations;
}
@@ -146,7 +146,7 @@ pcm_init_hw_params(struct snd_bebob *bebob,
if (err < 0)
goto end;
- err = amdtp_stream_add_pcm_hw_constraints(s, runtime);
+ err = amdtp_am824_add_pcm_hw_constraints(s, runtime);
end:
return err;
}
@@ -155,7 +155,7 @@ static int
pcm_open(struct snd_pcm_substream *substream)
{
struct snd_bebob *bebob = substream->private_data;
- struct snd_bebob_rate_spec *spec = bebob->spec->rate;
+ const struct snd_bebob_rate_spec *spec = bebob->spec->rate;
unsigned int sampling_rate;
enum snd_bebob_clock_type src;
int err;
@@ -220,8 +220,8 @@ pcm_capture_hw_params(struct snd_pcm_substream *substream,
if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN)
atomic_inc(&bebob->substreams_counter);
- amdtp_stream_set_pcm_format(&bebob->tx_stream,
- params_format(hw_params));
+
+ amdtp_am824_set_pcm_format(&bebob->tx_stream, params_format(hw_params));
return 0;
}
@@ -239,8 +239,8 @@ pcm_playback_hw_params(struct snd_pcm_substream *substream,
if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN)
atomic_inc(&bebob->substreams_counter);
- amdtp_stream_set_pcm_format(&bebob->rx_stream,
- params_format(hw_params));
+
+ amdtp_am824_set_pcm_format(&bebob->rx_stream, params_format(hw_params));
return 0;
}
diff --git a/sound/firewire/bebob/bebob_proc.c b/sound/firewire/bebob/bebob_proc.c
index 301cc6a93945..ec24f96794f5 100644
--- a/sound/firewire/bebob/bebob_proc.c
+++ b/sound/firewire/bebob/bebob_proc.c
@@ -73,7 +73,7 @@ proc_read_meters(struct snd_info_entry *entry,
struct snd_info_buffer *buffer)
{
struct snd_bebob *bebob = entry->private_data;
- struct snd_bebob_meter_spec *spec = bebob->spec->meter;
+ const struct snd_bebob_meter_spec *spec = bebob->spec->meter;
u32 *buf;
unsigned int i, c, channels, size;
@@ -138,8 +138,8 @@ proc_read_clock(struct snd_info_entry *entry,
"SYT-Match",
};
struct snd_bebob *bebob = entry->private_data;
- struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate;
- struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
+ const struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate;
+ const struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
enum snd_bebob_clock_type src;
unsigned int rate;
diff --git a/sound/firewire/bebob/bebob_stream.c b/sound/firewire/bebob/bebob_stream.c
index 5be5242e1ed8..926e5dcbb66a 100644
--- a/sound/firewire/bebob/bebob_stream.c
+++ b/sound/firewire/bebob/bebob_stream.c
@@ -119,7 +119,7 @@ end:
int snd_bebob_stream_get_clock_src(struct snd_bebob *bebob,
enum snd_bebob_clock_type *src)
{
- struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
+ const struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
u8 addr[AVC_BRIDGECO_ADDR_BYTES], input[7];
unsigned int id;
enum avc_bridgeco_plug_type type;
@@ -338,7 +338,7 @@ map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s)
err = -ENOSYS;
goto end;
}
- s->midi_position = stm_pos;
+ amdtp_am824_set_midi_position(s, stm_pos);
midi = stm_pos;
break;
/* for PCM data channel */
@@ -354,11 +354,12 @@ map_data_channels(struct snd_bebob *bebob, struct amdtp_stream *s)
case 0x09: /* Digital */
default:
location = pcm + sec_loc;
- if (location >= AMDTP_MAX_CHANNELS_FOR_PCM) {
+ if (location >= AM824_MAX_CHANNELS_FOR_PCM) {
err = -ENOSYS;
goto end;
}
- s->pcm_positions[location] = stm_pos;
+ amdtp_am824_set_pcm_position(s, location,
+ stm_pos);
break;
}
}
@@ -427,12 +428,19 @@ make_both_connections(struct snd_bebob *bebob, unsigned int rate)
index = get_formation_index(rate);
pcm_channels = bebob->tx_stream_formations[index].pcm;
midi_channels = bebob->tx_stream_formations[index].midi;
- amdtp_stream_set_parameters(&bebob->tx_stream,
- rate, pcm_channels, midi_channels * 8);
+ err = amdtp_am824_set_parameters(&bebob->tx_stream, rate,
+ pcm_channels, midi_channels * 8,
+ false);
+ if (err < 0)
+ goto end;
+
pcm_channels = bebob->rx_stream_formations[index].pcm;
midi_channels = bebob->rx_stream_formations[index].midi;
- amdtp_stream_set_parameters(&bebob->rx_stream,
- rate, pcm_channels, midi_channels * 8);
+ err = amdtp_am824_set_parameters(&bebob->rx_stream, rate,
+ pcm_channels, midi_channels * 8,
+ false);
+ if (err < 0)
+ goto end;
/* establish connections for both streams */
err = cmp_connection_establish(&bebob->out_conn,
@@ -530,8 +538,8 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob)
if (err < 0)
goto end;
- err = amdtp_stream_init(&bebob->tx_stream, bebob->unit,
- AMDTP_IN_STREAM, CIP_BLOCKING);
+ err = amdtp_am824_init(&bebob->tx_stream, bebob->unit,
+ AMDTP_IN_STREAM, CIP_BLOCKING);
if (err < 0) {
amdtp_stream_destroy(&bebob->tx_stream);
destroy_both_connections(bebob);
@@ -559,8 +567,8 @@ int snd_bebob_stream_init_duplex(struct snd_bebob *bebob)
if (bebob->maudio_special_quirk)
bebob->tx_stream.flags |= CIP_EMPTY_HAS_WRONG_DBC;
- err = amdtp_stream_init(&bebob->rx_stream, bebob->unit,
- AMDTP_OUT_STREAM, CIP_BLOCKING);
+ err = amdtp_am824_init(&bebob->rx_stream, bebob->unit,
+ AMDTP_OUT_STREAM, CIP_BLOCKING);
if (err < 0) {
amdtp_stream_destroy(&bebob->tx_stream);
amdtp_stream_destroy(&bebob->rx_stream);
@@ -572,7 +580,7 @@ end:
int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate)
{
- struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate;
+ const struct snd_bebob_rate_spec *rate_spec = bebob->spec->rate;
struct amdtp_stream *master, *slave;
enum cip_flags sync_mode;
unsigned int curr_rate;
@@ -864,8 +872,8 @@ parse_stream_formation(u8 *buf, unsigned int len,
}
}
- if (formation[i].pcm > AMDTP_MAX_CHANNELS_FOR_PCM ||
- formation[i].midi > AMDTP_MAX_CHANNELS_FOR_MIDI)
+ if (formation[i].pcm > AM824_MAX_CHANNELS_FOR_PCM ||
+ formation[i].midi > AM824_MAX_CHANNELS_FOR_MIDI)
return -ENOSYS;
return 0;
@@ -959,7 +967,7 @@ end:
int snd_bebob_stream_discover(struct snd_bebob *bebob)
{
- struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
+ const struct snd_bebob_clock_spec *clk_spec = bebob->spec->clock;
u8 plugs[AVC_PLUG_INFO_BUF_BYTES], addr[AVC_BRIDGECO_ADDR_BYTES];
enum avc_bridgeco_plug_type type;
unsigned int i;
diff --git a/sound/firewire/bebob/bebob_terratec.c b/sound/firewire/bebob/bebob_terratec.c
index 9242e33d2cf1..c38358b82ada 100644
--- a/sound/firewire/bebob/bebob_terratec.c
+++ b/sound/firewire/bebob/bebob_terratec.c
@@ -55,30 +55,30 @@ phase24_series_clk_src_get(struct snd_bebob *bebob, unsigned int *id)
return 0;
}
-static struct snd_bebob_rate_spec phase_series_rate_spec = {
+static const struct snd_bebob_rate_spec phase_series_rate_spec = {
.get = &snd_bebob_stream_get_rate,
.set = &snd_bebob_stream_set_rate,
};
/* PHASE 88 Rack FW */
-static struct snd_bebob_clock_spec phase88_rack_clk = {
+static const struct snd_bebob_clock_spec phase88_rack_clk = {
.num = ARRAY_SIZE(phase88_rack_clk_src_types),
.types = phase88_rack_clk_src_types,
.get = &phase88_rack_clk_src_get,
};
-struct snd_bebob_spec phase88_rack_spec = {
+const struct snd_bebob_spec phase88_rack_spec = {
.clock = &phase88_rack_clk,
.rate = &phase_series_rate_spec,
.meter = NULL
};
/* 'PHASE 24 FW' and 'PHASE X24 FW' */
-static struct snd_bebob_clock_spec phase24_series_clk = {
+static const struct snd_bebob_clock_spec phase24_series_clk = {
.num = ARRAY_SIZE(phase24_series_clk_src_types),
.types = phase24_series_clk_src_types,
.get = &phase24_series_clk_src_get,
};
-struct snd_bebob_spec phase24_series_spec = {
+const struct snd_bebob_spec phase24_series_spec = {
.clock = &phase24_series_clk,
.rate = &phase_series_rate_spec,
.meter = NULL
diff --git a/sound/firewire/bebob/bebob_yamaha.c b/sound/firewire/bebob/bebob_yamaha.c
index 58101702410b..90d4404f77ce 100644
--- a/sound/firewire/bebob/bebob_yamaha.c
+++ b/sound/firewire/bebob/bebob_yamaha.c
@@ -46,16 +46,16 @@ clk_src_get(struct snd_bebob *bebob, unsigned int *id)
return 0;
}
-static struct snd_bebob_clock_spec clock_spec = {
+static const struct snd_bebob_clock_spec clock_spec = {
.num = ARRAY_SIZE(clk_src_types),
.types = clk_src_types,
.get = &clk_src_get,
};
-static struct snd_bebob_rate_spec rate_spec = {
+static const struct snd_bebob_rate_spec rate_spec = {
.get = &snd_bebob_stream_get_rate,
.set = &snd_bebob_stream_set_rate,
};
-struct snd_bebob_spec yamaha_go_spec = {
+const struct snd_bebob_spec yamaha_go_spec = {
.clock = &clock_spec,
.rate = &rate_spec,
.meter = NULL
diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile
index 9ef228ef7baf..55b4be9b0034 100644
--- a/sound/firewire/dice/Makefile
+++ b/sound/firewire/dice/Makefile
@@ -1,3 +1,3 @@
snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \
dice-pcm.o dice-hwdep.o dice.o
-obj-m += snd-dice.o
+obj-$(CONFIG_SND_DICE) += snd-dice.o
diff --git a/sound/firewire/dice/dice-midi.c b/sound/firewire/dice/dice-midi.c
index fe43ce791f84..151b09f240f2 100644
--- a/sound/firewire/dice/dice-midi.c
+++ b/sound/firewire/dice/dice-midi.c
@@ -52,10 +52,10 @@ static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up)
spin_lock_irqsave(&dice->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&dice->tx_stream,
+ amdtp_am824_midi_trigger(&dice->tx_stream,
substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&dice->tx_stream,
+ amdtp_am824_midi_trigger(&dice->tx_stream,
substrm->number, NULL);
spin_unlock_irqrestore(&dice->lock, flags);
@@ -69,11 +69,11 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
spin_lock_irqsave(&dice->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&dice->rx_stream,
- substrm->number, substrm);
+ amdtp_am824_midi_trigger(&dice->rx_stream,
+ substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&dice->rx_stream,
- substrm->number, NULL);
+ amdtp_am824_midi_trigger(&dice->rx_stream,
+ substrm->number, NULL);
spin_unlock_irqrestore(&dice->lock, flags);
}
diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c
index 4e67b1da0fe6..9b3431999fc8 100644
--- a/sound/firewire/dice/dice-pcm.c
+++ b/sound/firewire/dice/dice-pcm.c
@@ -133,11 +133,11 @@ static int init_hw_info(struct snd_dice *dice,
SNDRV_PCM_INFO_BLOCK_TRANSFER;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- hw->formats = AMDTP_IN_PCM_FORMAT_BITS;
+ hw->formats = AM824_IN_PCM_FORMAT_BITS;
stream = &dice->tx_stream;
pcm_channels = dice->tx_channels;
} else {
- hw->formats = AMDTP_OUT_PCM_FORMAT_BITS;
+ hw->formats = AM824_OUT_PCM_FORMAT_BITS;
stream = &dice->rx_stream;
pcm_channels = dice->rx_channels;
}
@@ -156,7 +156,7 @@ static int init_hw_info(struct snd_dice *dice,
if (err < 0)
goto end;
- err = amdtp_stream_add_pcm_hw_constraints(stream, runtime);
+ err = amdtp_am824_add_pcm_hw_constraints(stream, runtime);
end:
return err;
}
@@ -243,8 +243,7 @@ static int capture_hw_params(struct snd_pcm_substream *substream,
mutex_unlock(&dice->mutex);
}
- amdtp_stream_set_pcm_format(&dice->tx_stream,
- params_format(hw_params));
+ amdtp_am824_set_pcm_format(&dice->tx_stream, params_format(hw_params));
return 0;
}
@@ -265,8 +264,7 @@ static int playback_hw_params(struct snd_pcm_substream *substream,
mutex_unlock(&dice->mutex);
}
- amdtp_stream_set_pcm_format(&dice->rx_stream,
- params_format(hw_params));
+ amdtp_am824_set_pcm_format(&dice->rx_stream, params_format(hw_params));
return 0;
}
diff --git a/sound/firewire/dice/dice-stream.c b/sound/firewire/dice/dice-stream.c
index 07dbd01d7a6b..a6a39f7ef58d 100644
--- a/sound/firewire/dice/dice-stream.c
+++ b/sound/firewire/dice/dice-stream.c
@@ -44,16 +44,16 @@ int snd_dice_stream_get_rate_mode(struct snd_dice *dice, unsigned int rate,
static void release_resources(struct snd_dice *dice,
struct fw_iso_resources *resources)
{
- unsigned int channel;
+ __be32 channel;
/* Reset channel number */
channel = cpu_to_be32((u32)-1);
if (resources == &dice->tx_resources)
snd_dice_transaction_write_tx(dice, TX_ISOCHRONOUS,
- &channel, 4);
+ &channel, sizeof(channel));
else
snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS,
- &channel, 4);
+ &channel, sizeof(channel));
fw_iso_resources_free(resources);
}
@@ -62,7 +62,7 @@ static int keep_resources(struct snd_dice *dice,
struct fw_iso_resources *resources,
unsigned int max_payload_bytes)
{
- unsigned int channel;
+ __be32 channel;
int err;
err = fw_iso_resources_allocate(resources, max_payload_bytes,
@@ -74,10 +74,10 @@ static int keep_resources(struct snd_dice *dice,
channel = cpu_to_be32(resources->channel);
if (resources == &dice->tx_resources)
err = snd_dice_transaction_write_tx(dice, TX_ISOCHRONOUS,
- &channel, 4);
+ &channel, sizeof(channel));
else
err = snd_dice_transaction_write_rx(dice, RX_ISOCHRONOUS,
- &channel, 4);
+ &channel, sizeof(channel));
if (err < 0)
release_resources(dice, resources);
end:
@@ -100,6 +100,7 @@ static int start_stream(struct snd_dice *dice, struct amdtp_stream *stream,
{
struct fw_iso_resources *resources;
unsigned int i, mode, pcm_chs, midi_ports;
+ bool double_pcm_frames;
int err;
err = snd_dice_stream_get_rate_mode(dice, rate, &mode);
@@ -125,21 +126,24 @@ static int start_stream(struct snd_dice *dice, struct amdtp_stream *stream,
* For this quirk, blocking mode is required and PCM buffer size should
* be aligned to SYT_INTERVAL.
*/
- if (mode > 1) {
+ double_pcm_frames = mode > 1;
+ if (double_pcm_frames) {
rate /= 2;
pcm_chs *= 2;
- stream->double_pcm_frames = true;
- } else {
- stream->double_pcm_frames = false;
}
- amdtp_stream_set_parameters(stream, rate, pcm_chs, midi_ports);
- if (mode > 1) {
+ err = amdtp_am824_set_parameters(stream, rate, pcm_chs, midi_ports,
+ double_pcm_frames);
+ if (err < 0)
+ goto end;
+
+ if (double_pcm_frames) {
pcm_chs /= 2;
for (i = 0; i < pcm_chs; i++) {
- stream->pcm_positions[i] = i * 2;
- stream->pcm_positions[i + pcm_chs] = i * 2 + 1;
+ amdtp_am824_set_pcm_position(stream, i, i * 2);
+ amdtp_am824_set_pcm_position(stream, i + pcm_chs,
+ i * 2 + 1);
}
}
@@ -302,7 +306,7 @@ static int init_stream(struct snd_dice *dice, struct amdtp_stream *stream)
goto end;
resources->channels_mask = 0x00000000ffffffffuLL;
- err = amdtp_stream_init(stream, dice->unit, dir, CIP_BLOCKING);
+ err = amdtp_am824_init(stream, dice->unit, dir, CIP_BLOCKING);
if (err < 0) {
amdtp_stream_destroy(stream);
fw_iso_resources_destroy(resources);
diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c
index 70a111d7f428..5d99436dfcae 100644
--- a/sound/firewire/dice/dice.c
+++ b/sound/firewire/dice/dice.c
@@ -29,7 +29,8 @@ static int dice_interface_check(struct fw_unit *unit)
struct fw_csr_iterator it;
int key, val, vendor = -1, model = -1, err;
unsigned int category, i;
- __be32 *pointers, value;
+ __be32 *pointers;
+ u32 value;
__be32 version;
pointers = kmalloc_array(ARRAY_SIZE(min_values), sizeof(__be32),
diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h
index ecf5dc862235..101550ac1a24 100644
--- a/sound/firewire/dice/dice.h
+++ b/sound/firewire/dice/dice.h
@@ -34,7 +34,7 @@
#include <sound/pcm_params.h>
#include <sound/rawmidi.h>
-#include "../amdtp.h"
+#include "../amdtp-am824.h"
#include "../iso-resources.h"
#include "../lib.h"
#include "dice-interface.h"
diff --git a/sound/firewire/digi00x/Makefile b/sound/firewire/digi00x/Makefile
new file mode 100644
index 000000000000..1123e68c8b28
--- /dev/null
+++ b/sound/firewire/digi00x/Makefile
@@ -0,0 +1,4 @@
+snd-firewire-digi00x-objs := amdtp-dot.o digi00x-stream.o digi00x-proc.o \
+ digi00x-pcm.o digi00x-hwdep.o \
+ digi00x-transaction.o digi00x-midi.o digi00x.o
+obj-$(CONFIG_SND_FIREWIRE_DIGI00X) += snd-firewire-digi00x.o
diff --git a/sound/firewire/digi00x/amdtp-dot.c b/sound/firewire/digi00x/amdtp-dot.c
new file mode 100644
index 000000000000..b02a5e8cad44
--- /dev/null
+++ b/sound/firewire/digi00x/amdtp-dot.c
@@ -0,0 +1,442 @@
+/*
+ * amdtp-dot.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ * Copyright (C) 2012 Robin Gareus <robin@gareus.org>
+ * Copyright (C) 2012 Damien Zammit <damien@zamaudio.com>
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <sound/pcm.h>
+#include "digi00x.h"
+
+#define CIP_FMT_AM 0x10
+
+/* 'Clock-based rate control mode' is just supported. */
+#define AMDTP_FDF_AM824 0x00
+
+/*
+ * Nominally 3125 bytes/second, but the MIDI port's clock might be
+ * 1% too slow, and the bus clock 100 ppm too fast.
+ */
+#define MIDI_BYTES_PER_SECOND 3093
+
+/*
+ * Several devices look only at the first eight data blocks.
+ * In any case, this is more than enough for the MIDI data rate.
+ */
+#define MAX_MIDI_RX_BLOCKS 8
+
+/*
+ * The double-oh-three algorithm was discovered by Robin Gareus and Damien
+ * Zammit in 2012, with reverse-engineering for Digi 003 Rack.
+ */
+struct dot_state {
+ u8 carry;
+ u8 idx;
+ unsigned int off;
+};
+
+struct amdtp_dot {
+ unsigned int pcm_channels;
+ struct dot_state state;
+
+ unsigned int midi_ports;
+ /* 2 = MAX(DOT_MIDI_IN_PORTS, DOT_MIDI_OUT_PORTS) */
+ struct snd_rawmidi_substream *midi[2];
+ int midi_fifo_used[2];
+ int midi_fifo_limit;
+
+ void (*transfer_samples)(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames);
+};
+
+/*
+ * double-oh-three look up table
+ *
+ * @param idx index byte (audio-sample data) 0x00..0xff
+ * @param off channel offset shift
+ * @return salt to XOR with given data
+ */
+#define BYTE_PER_SAMPLE (4)
+#define MAGIC_DOT_BYTE (2)
+#define MAGIC_BYTE_OFF(x) (((x) * BYTE_PER_SAMPLE) + MAGIC_DOT_BYTE)
+static const u8 dot_scrt(const u8 idx, const unsigned int off)
+{
+ /*
+ * the length of the added pattern only depends on the lower nibble
+ * of the last non-zero data
+ */
+ static const u8 len[16] = {0, 1, 3, 5, 7, 9, 11, 13, 14,
+ 12, 10, 8, 6, 4, 2, 0};
+
+ /*
+ * the lower nibble of the salt. Interleaved sequence.
+ * this is walked backwards according to len[]
+ */
+ static const u8 nib[15] = {0x8, 0x7, 0x9, 0x6, 0xa, 0x5, 0xb, 0x4,
+ 0xc, 0x3, 0xd, 0x2, 0xe, 0x1, 0xf};
+
+ /* circular list for the salt's hi nibble. */
+ static const u8 hir[15] = {0x0, 0x6, 0xf, 0x8, 0x7, 0x5, 0x3, 0x4,
+ 0xc, 0xd, 0xe, 0x1, 0x2, 0xb, 0xa};
+
+ /*
+ * start offset for upper nibble mapping.
+ * note: 9 is /special/. In the case where the high nibble == 0x9,
+ * hir[] is not used and - coincidentally - the salt's hi nibble is
+ * 0x09 regardless of the offset.
+ */
+ static const u8 hio[16] = {0, 11, 12, 6, 7, 5, 1, 4,
+ 3, 0x00, 14, 13, 8, 9, 10, 2};
+
+ const u8 ln = idx & 0xf;
+ const u8 hn = (idx >> 4) & 0xf;
+ const u8 hr = (hn == 0x9) ? 0x9 : hir[(hio[hn] + off) % 15];
+
+ if (len[ln] < off)
+ return 0x00;
+
+ return ((nib[14 + off - len[ln]]) | (hr << 4));
+}
+
+static void dot_encode_step(struct dot_state *state, __be32 *const buffer)
+{
+ u8 * const data = (u8 *) buffer;
+
+ if (data[MAGIC_DOT_BYTE] != 0x00) {
+ state->off = 0;
+ state->idx = data[MAGIC_DOT_BYTE] ^ state->carry;
+ }
+ data[MAGIC_DOT_BYTE] ^= state->carry;
+ state->carry = dot_scrt(state->idx, ++(state->off));
+}
+
+int amdtp_dot_set_parameters(struct amdtp_stream *s, unsigned int rate,
+ unsigned int pcm_channels)
+{
+ struct amdtp_dot *p = s->protocol;
+ int err;
+
+ if (amdtp_stream_running(s))
+ return -EBUSY;
+
+ /*
+ * A first data channel is for MIDI conformant data channel, the rest is
+ * Multi Bit Linear Audio data channel.
+ */
+ err = amdtp_stream_set_parameters(s, rate, pcm_channels + 1);
+ if (err < 0)
+ return err;
+
+ s->fdf = AMDTP_FDF_AM824 | s->sfc;
+
+ p->pcm_channels = pcm_channels;
+
+ if (s->direction == AMDTP_IN_STREAM)
+ p->midi_ports = DOT_MIDI_IN_PORTS;
+ else
+ p->midi_ports = DOT_MIDI_OUT_PORTS;
+
+ /*
+ * We do not know the actual MIDI FIFO size of most devices. Just
+ * assume two bytes, i.e., one byte can be received over the bus while
+ * the previous one is transmitted over MIDI.
+ * (The value here is adjusted for midi_ratelimit_per_packet().)
+ */
+ p->midi_fifo_limit = rate - MIDI_BYTES_PER_SECOND * s->syt_interval + 1;
+
+ return 0;
+}
+
+static void write_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_dot *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ const u32 *src;
+
+ channels = p->pcm_channels;
+ src = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ buffer++;
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ buffer[c] = cpu_to_be32((*src >> 8) | 0x40000000);
+ dot_encode_step(&p->state, &buffer[c]);
+ src++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void write_pcm_s16(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_dot *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ const u16 *src;
+
+ channels = p->pcm_channels;
+ src = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ buffer++;
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ buffer[c] = cpu_to_be32((*src << 8) | 0x40000000);
+ dot_encode_step(&p->state, &buffer[c]);
+ src++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void read_pcm_s32(struct amdtp_stream *s, struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_dot *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ u32 *dst;
+
+ channels = p->pcm_channels;
+ dst = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ buffer++;
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ *dst = be32_to_cpu(buffer[c]) << 8;
+ dst++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ dst = (void *)runtime->dma_area;
+ }
+}
+
+static void write_pcm_silence(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int data_blocks)
+{
+ struct amdtp_dot *p = s->protocol;
+ unsigned int channels, i, c;
+
+ channels = p->pcm_channels;
+
+ buffer++;
+ for (i = 0; i < data_blocks; ++i) {
+ for (c = 0; c < channels; ++c)
+ buffer[c] = cpu_to_be32(0x40000000);
+ buffer += s->data_block_quadlets;
+ }
+}
+
+static bool midi_ratelimit_per_packet(struct amdtp_stream *s, unsigned int port)
+{
+ struct amdtp_dot *p = s->protocol;
+ int used;
+
+ used = p->midi_fifo_used[port];
+ if (used == 0)
+ return true;
+
+ used -= MIDI_BYTES_PER_SECOND * s->syt_interval;
+ used = max(used, 0);
+ p->midi_fifo_used[port] = used;
+
+ return used < p->midi_fifo_limit;
+}
+
+static inline void midi_use_bytes(struct amdtp_stream *s,
+ unsigned int port, unsigned int count)
+{
+ struct amdtp_dot *p = s->protocol;
+
+ p->midi_fifo_used[port] += amdtp_rate_table[s->sfc] * count;
+}
+
+static void write_midi_messages(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int data_blocks)
+{
+ struct amdtp_dot *p = s->protocol;
+ unsigned int f, port;
+ int len;
+ u8 *b;
+
+ for (f = 0; f < data_blocks; f++) {
+ port = (s->data_block_counter + f) % 8;
+ b = (u8 *)&buffer[0];
+
+ len = 0;
+ if (port < p->midi_ports &&
+ midi_ratelimit_per_packet(s, port) &&
+ p->midi[port] != NULL)
+ len = snd_rawmidi_transmit(p->midi[port], b + 1, 2);
+
+ if (len > 0) {
+ b[3] = (0x10 << port) | len;
+ midi_use_bytes(s, port, len);
+ } else {
+ b[1] = 0;
+ b[2] = 0;
+ b[3] = 0;
+ }
+ b[0] = 0x80;
+
+ buffer += s->data_block_quadlets;
+ }
+}
+
+static void read_midi_messages(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int data_blocks)
+{
+ struct amdtp_dot *p = s->protocol;
+ unsigned int f, port, len;
+ u8 *b;
+
+ for (f = 0; f < data_blocks; f++) {
+ b = (u8 *)&buffer[0];
+ port = b[3] >> 4;
+ len = b[3] & 0x0f;
+
+ if (port < p->midi_ports && p->midi[port] && len > 0)
+ snd_rawmidi_receive(p->midi[port], b + 1, len);
+
+ buffer += s->data_block_quadlets;
+ }
+}
+
+int amdtp_dot_add_pcm_hw_constraints(struct amdtp_stream *s,
+ struct snd_pcm_runtime *runtime)
+{
+ int err;
+
+ /* This protocol delivers 24 bit data in 32bit data channel. */
+ err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ if (err < 0)
+ return err;
+
+ return amdtp_stream_add_pcm_hw_constraints(s, runtime);
+}
+
+void amdtp_dot_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format)
+{
+ struct amdtp_dot *p = s->protocol;
+
+ if (WARN_ON(amdtp_stream_pcm_running(s)))
+ return;
+
+ switch (format) {
+ default:
+ WARN_ON(1);
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S16:
+ if (s->direction == AMDTP_OUT_STREAM) {
+ p->transfer_samples = write_pcm_s16;
+ break;
+ }
+ WARN_ON(1);
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S32:
+ if (s->direction == AMDTP_OUT_STREAM)
+ p->transfer_samples = write_pcm_s32;
+ else
+ p->transfer_samples = read_pcm_s32;
+ break;
+ }
+}
+
+void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port,
+ struct snd_rawmidi_substream *midi)
+{
+ struct amdtp_dot *p = s->protocol;
+
+ if (port < p->midi_ports)
+ ACCESS_ONCE(p->midi[port]) = midi;
+}
+
+static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
+ __be32 *buffer,
+ unsigned int data_blocks,
+ unsigned int *syt)
+{
+ struct amdtp_dot *p = (struct amdtp_dot *)s->protocol;
+ struct snd_pcm_substream *pcm;
+ unsigned int pcm_frames;
+
+ pcm = ACCESS_ONCE(s->pcm);
+ if (pcm) {
+ p->transfer_samples(s, pcm, buffer, data_blocks);
+ pcm_frames = data_blocks;
+ } else {
+ pcm_frames = 0;
+ }
+
+ read_midi_messages(s, buffer, data_blocks);
+
+ return pcm_frames;
+}
+
+static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
+ __be32 *buffer,
+ unsigned int data_blocks,
+ unsigned int *syt)
+{
+ struct amdtp_dot *p = (struct amdtp_dot *)s->protocol;
+ struct snd_pcm_substream *pcm;
+ unsigned int pcm_frames;
+
+ pcm = ACCESS_ONCE(s->pcm);
+ if (pcm) {
+ p->transfer_samples(s, pcm, buffer, data_blocks);
+ pcm_frames = data_blocks;
+ } else {
+ write_pcm_silence(s, buffer, data_blocks);
+ pcm_frames = 0;
+ }
+
+ write_midi_messages(s, buffer, data_blocks);
+
+ return pcm_frames;
+}
+
+int amdtp_dot_init(struct amdtp_stream *s, struct fw_unit *unit,
+ enum amdtp_stream_direction dir)
+{
+ amdtp_stream_process_data_blocks_t process_data_blocks;
+ enum cip_flags flags;
+
+ /* Use different mode between incoming/outgoing. */
+ if (dir == AMDTP_IN_STREAM) {
+ flags = CIP_NONBLOCKING | CIP_SKIP_INIT_DBC_CHECK;
+ process_data_blocks = process_tx_data_blocks;
+ } else {
+ flags = CIP_BLOCKING;
+ process_data_blocks = process_rx_data_blocks;
+ }
+
+ return amdtp_stream_init(s, unit, dir, flags, CIP_FMT_AM,
+ process_data_blocks, sizeof(struct amdtp_dot));
+}
+
+void amdtp_dot_reset(struct amdtp_stream *s)
+{
+ struct amdtp_dot *p = s->protocol;
+
+ p->state.carry = 0x00;
+ p->state.idx = 0x00;
+ p->state.off = 0;
+}
diff --git a/sound/firewire/digi00x/digi00x-hwdep.c b/sound/firewire/digi00x/digi00x-hwdep.c
new file mode 100644
index 000000000000..f188e4758fd2
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-hwdep.c
@@ -0,0 +1,200 @@
+/*
+ * digi00x-hwdep.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+/*
+ * This codes give three functionality.
+ *
+ * 1.get firewire node information
+ * 2.get notification about starting/stopping stream
+ * 3.lock/unlock stream
+ * 4.get asynchronous messaging
+ */
+
+#include "digi00x.h"
+
+static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count,
+ loff_t *offset)
+{
+ struct snd_dg00x *dg00x = hwdep->private_data;
+ DEFINE_WAIT(wait);
+ union snd_firewire_event event;
+
+ spin_lock_irq(&dg00x->lock);
+
+ while (!dg00x->dev_lock_changed && dg00x->msg == 0) {
+ prepare_to_wait(&dg00x->hwdep_wait, &wait, TASK_INTERRUPTIBLE);
+ spin_unlock_irq(&dg00x->lock);
+ schedule();
+ finish_wait(&dg00x->hwdep_wait, &wait);
+ if (signal_pending(current))
+ return -ERESTARTSYS;
+ spin_lock_irq(&dg00x->lock);
+ }
+
+ memset(&event, 0, sizeof(event));
+ if (dg00x->dev_lock_changed) {
+ event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS;
+ event.lock_status.status = (dg00x->dev_lock_count > 0);
+ dg00x->dev_lock_changed = false;
+
+ count = min_t(long, count, sizeof(event.lock_status));
+ } else {
+ event.digi00x_message.type =
+ SNDRV_FIREWIRE_EVENT_DIGI00X_MESSAGE;
+ event.digi00x_message.message = dg00x->msg;
+ dg00x->msg = 0;
+
+ count = min_t(long, count, sizeof(event.digi00x_message));
+ }
+
+ spin_unlock_irq(&dg00x->lock);
+
+ if (copy_to_user(buf, &event, count))
+ return -EFAULT;
+
+ return count;
+}
+
+static unsigned int hwdep_poll(struct snd_hwdep *hwdep, struct file *file,
+ poll_table *wait)
+{
+ struct snd_dg00x *dg00x = hwdep->private_data;
+ unsigned int events;
+
+ poll_wait(file, &dg00x->hwdep_wait, wait);
+
+ spin_lock_irq(&dg00x->lock);
+ if (dg00x->dev_lock_changed || dg00x->msg)
+ events = POLLIN | POLLRDNORM;
+ else
+ events = 0;
+ spin_unlock_irq(&dg00x->lock);
+
+ return events;
+}
+
+static int hwdep_get_info(struct snd_dg00x *dg00x, void __user *arg)
+{
+ struct fw_device *dev = fw_parent_device(dg00x->unit);
+ struct snd_firewire_get_info info;
+
+ memset(&info, 0, sizeof(info));
+ info.type = SNDRV_FIREWIRE_TYPE_DIGI00X;
+ info.card = dev->card->index;
+ *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]);
+ *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]);
+ strlcpy(info.device_name, dev_name(&dev->device),
+ sizeof(info.device_name));
+
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+
+ return 0;
+}
+
+static int hwdep_lock(struct snd_dg00x *dg00x)
+{
+ int err;
+
+ spin_lock_irq(&dg00x->lock);
+
+ if (dg00x->dev_lock_count == 0) {
+ dg00x->dev_lock_count = -1;
+ err = 0;
+ } else {
+ err = -EBUSY;
+ }
+
+ spin_unlock_irq(&dg00x->lock);
+
+ return err;
+}
+
+static int hwdep_unlock(struct snd_dg00x *dg00x)
+{
+ int err;
+
+ spin_lock_irq(&dg00x->lock);
+
+ if (dg00x->dev_lock_count == -1) {
+ dg00x->dev_lock_count = 0;
+ err = 0;
+ } else {
+ err = -EBADFD;
+ }
+
+ spin_unlock_irq(&dg00x->lock);
+
+ return err;
+}
+
+static int hwdep_release(struct snd_hwdep *hwdep, struct file *file)
+{
+ struct snd_dg00x *dg00x = hwdep->private_data;
+
+ spin_lock_irq(&dg00x->lock);
+ if (dg00x->dev_lock_count == -1)
+ dg00x->dev_lock_count = 0;
+ spin_unlock_irq(&dg00x->lock);
+
+ return 0;
+}
+
+static int hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ struct snd_dg00x *dg00x = hwdep->private_data;
+
+ switch (cmd) {
+ case SNDRV_FIREWIRE_IOCTL_GET_INFO:
+ return hwdep_get_info(dg00x, (void __user *)arg);
+ case SNDRV_FIREWIRE_IOCTL_LOCK:
+ return hwdep_lock(dg00x);
+ case SNDRV_FIREWIRE_IOCTL_UNLOCK:
+ return hwdep_unlock(dg00x);
+ default:
+ return -ENOIOCTLCMD;
+ }
+}
+
+#ifdef CONFIG_COMPAT
+static int hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ return hwdep_ioctl(hwdep, file, cmd,
+ (unsigned long)compat_ptr(arg));
+}
+#else
+#define hwdep_compat_ioctl NULL
+#endif
+
+static const struct snd_hwdep_ops hwdep_ops = {
+ .read = hwdep_read,
+ .release = hwdep_release,
+ .poll = hwdep_poll,
+ .ioctl = hwdep_ioctl,
+ .ioctl_compat = hwdep_compat_ioctl,
+};
+
+int snd_dg00x_create_hwdep_device(struct snd_dg00x *dg00x)
+{
+ struct snd_hwdep *hwdep;
+ int err;
+
+ err = snd_hwdep_new(dg00x->card, "Digi00x", 0, &hwdep);
+ if (err < 0)
+ return err;
+
+ strcpy(hwdep->name, "Digi00x");
+ hwdep->iface = SNDRV_HWDEP_IFACE_FW_DIGI00X;
+ hwdep->ops = hwdep_ops;
+ hwdep->private_data = dg00x;
+ hwdep->exclusive = true;
+
+ return err;
+}
diff --git a/sound/firewire/digi00x/digi00x-midi.c b/sound/firewire/digi00x/digi00x-midi.c
new file mode 100644
index 000000000000..1a72a382b384
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-midi.c
@@ -0,0 +1,223 @@
+/*
+ * digi00x-midi.h - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+static int midi_phys_open(struct snd_rawmidi_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->rmidi->private_data;
+ int err;
+
+ err = snd_dg00x_stream_lock_try(dg00x);
+ if (err < 0)
+ return err;
+
+ mutex_lock(&dg00x->mutex);
+ dg00x->substreams_counter++;
+ err = snd_dg00x_stream_start_duplex(dg00x, 0);
+ mutex_unlock(&dg00x->mutex);
+ if (err < 0)
+ snd_dg00x_stream_lock_release(dg00x);
+
+ return err;
+}
+
+static int midi_phys_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->rmidi->private_data;
+
+ mutex_lock(&dg00x->mutex);
+ dg00x->substreams_counter--;
+ snd_dg00x_stream_stop_duplex(dg00x);
+ mutex_unlock(&dg00x->mutex);
+
+ snd_dg00x_stream_lock_release(dg00x);
+ return 0;
+}
+
+static void midi_phys_capture_trigger(struct snd_rawmidi_substream *substream,
+ int up)
+{
+ struct snd_dg00x *dg00x = substream->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&dg00x->lock, flags);
+
+ if (up)
+ amdtp_dot_midi_trigger(&dg00x->tx_stream, substream->number,
+ substream);
+ else
+ amdtp_dot_midi_trigger(&dg00x->tx_stream, substream->number,
+ NULL);
+
+ spin_unlock_irqrestore(&dg00x->lock, flags);
+}
+
+static void midi_phys_playback_trigger(struct snd_rawmidi_substream *substream,
+ int up)
+{
+ struct snd_dg00x *dg00x = substream->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&dg00x->lock, flags);
+
+ if (up)
+ amdtp_dot_midi_trigger(&dg00x->rx_stream, substream->number,
+ substream);
+ else
+ amdtp_dot_midi_trigger(&dg00x->rx_stream, substream->number,
+ NULL);
+
+ spin_unlock_irqrestore(&dg00x->lock, flags);
+}
+
+static struct snd_rawmidi_ops midi_phys_capture_ops = {
+ .open = midi_phys_open,
+ .close = midi_phys_close,
+ .trigger = midi_phys_capture_trigger,
+};
+
+static struct snd_rawmidi_ops midi_phys_playback_ops = {
+ .open = midi_phys_open,
+ .close = midi_phys_close,
+ .trigger = midi_phys_playback_trigger,
+};
+
+static int midi_ctl_open(struct snd_rawmidi_substream *substream)
+{
+ /* Do nothing. */
+ return 0;
+}
+
+static int midi_ctl_capture_close(struct snd_rawmidi_substream *substream)
+{
+ /* Do nothing. */
+ return 0;
+}
+
+static int midi_ctl_playback_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->rmidi->private_data;
+
+ snd_fw_async_midi_port_finish(&dg00x->out_control);
+
+ return 0;
+}
+
+static void midi_ctl_capture_trigger(struct snd_rawmidi_substream *substream,
+ int up)
+{
+ struct snd_dg00x *dg00x = substream->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&dg00x->lock, flags);
+
+ if (up)
+ dg00x->in_control = substream;
+ else
+ dg00x->in_control = NULL;
+
+ spin_unlock_irqrestore(&dg00x->lock, flags);
+}
+
+static void midi_ctl_playback_trigger(struct snd_rawmidi_substream *substream,
+ int up)
+{
+ struct snd_dg00x *dg00x = substream->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&dg00x->lock, flags);
+
+ if (up)
+ snd_fw_async_midi_port_run(&dg00x->out_control, substream);
+
+ spin_unlock_irqrestore(&dg00x->lock, flags);
+}
+
+static struct snd_rawmidi_ops midi_ctl_capture_ops = {
+ .open = midi_ctl_open,
+ .close = midi_ctl_capture_close,
+ .trigger = midi_ctl_capture_trigger,
+};
+
+static struct snd_rawmidi_ops midi_ctl_playback_ops = {
+ .open = midi_ctl_open,
+ .close = midi_ctl_playback_close,
+ .trigger = midi_ctl_playback_trigger,
+};
+
+static void set_midi_substream_names(struct snd_dg00x *dg00x,
+ struct snd_rawmidi_str *str,
+ bool is_ctl)
+{
+ struct snd_rawmidi_substream *subs;
+
+ list_for_each_entry(subs, &str->substreams, list) {
+ if (!is_ctl)
+ snprintf(subs->name, sizeof(subs->name),
+ "%s MIDI %d",
+ dg00x->card->shortname, subs->number + 1);
+ else
+ /* This port is for asynchronous transaction. */
+ snprintf(subs->name, sizeof(subs->name),
+ "%s control",
+ dg00x->card->shortname);
+ }
+}
+
+int snd_dg00x_create_midi_devices(struct snd_dg00x *dg00x)
+{
+ struct snd_rawmidi *rmidi[2];
+ struct snd_rawmidi_str *str;
+ unsigned int i;
+ int err;
+
+ /* Add physical midi ports. */
+ err = snd_rawmidi_new(dg00x->card, dg00x->card->driver, 0,
+ DOT_MIDI_OUT_PORTS, DOT_MIDI_IN_PORTS, &rmidi[0]);
+ if (err < 0)
+ return err;
+
+ snprintf(rmidi[0]->name, sizeof(rmidi[0]->name),
+ "%s MIDI", dg00x->card->shortname);
+
+ snd_rawmidi_set_ops(rmidi[0], SNDRV_RAWMIDI_STREAM_INPUT,
+ &midi_phys_capture_ops);
+ snd_rawmidi_set_ops(rmidi[0], SNDRV_RAWMIDI_STREAM_OUTPUT,
+ &midi_phys_playback_ops);
+
+ /* Add a pair of control midi ports. */
+ err = snd_rawmidi_new(dg00x->card, dg00x->card->driver, 1,
+ 1, 1, &rmidi[1]);
+ if (err < 0)
+ return err;
+
+ snprintf(rmidi[1]->name, sizeof(rmidi[1]->name),
+ "%s control", dg00x->card->shortname);
+
+ snd_rawmidi_set_ops(rmidi[1], SNDRV_RAWMIDI_STREAM_INPUT,
+ &midi_ctl_capture_ops);
+ snd_rawmidi_set_ops(rmidi[1], SNDRV_RAWMIDI_STREAM_OUTPUT,
+ &midi_ctl_playback_ops);
+
+ for (i = 0; i < ARRAY_SIZE(rmidi); i++) {
+ rmidi[i]->private_data = dg00x;
+
+ rmidi[i]->info_flags |= SNDRV_RAWMIDI_INFO_INPUT;
+ str = &rmidi[i]->streams[SNDRV_RAWMIDI_STREAM_INPUT];
+ set_midi_substream_names(dg00x, str, i);
+
+ rmidi[i]->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT;
+ str = &rmidi[i]->streams[SNDRV_RAWMIDI_STREAM_OUTPUT];
+ set_midi_substream_names(dg00x, str, i);
+
+ rmidi[i]->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX;
+ }
+
+ return 0;
+}
diff --git a/sound/firewire/digi00x/digi00x-pcm.c b/sound/firewire/digi00x/digi00x-pcm.c
new file mode 100644
index 000000000000..cac28f70aef7
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-pcm.c
@@ -0,0 +1,373 @@
+/*
+ * digi00x-pcm.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+static int hw_rule_rate(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *r =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+ const struct snd_interval *c =
+ hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_interval t = {
+ .min = UINT_MAX, .max = 0, .integer = 1,
+ };
+ unsigned int i;
+
+ for (i = 0; i < SND_DG00X_RATE_COUNT; i++) {
+ if (!snd_interval_test(c,
+ snd_dg00x_stream_pcm_channels[i]))
+ continue;
+
+ t.min = min(t.min, snd_dg00x_stream_rates[i]);
+ t.max = max(t.max, snd_dg00x_stream_rates[i]);
+ }
+
+ return snd_interval_refine(r, &t);
+}
+
+static int hw_rule_channels(struct snd_pcm_hw_params *params,
+ struct snd_pcm_hw_rule *rule)
+{
+ struct snd_interval *c =
+ hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ const struct snd_interval *r =
+ hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_RATE);
+ struct snd_interval t = {
+ .min = UINT_MAX, .max = 0, .integer = 1,
+ };
+ unsigned int i;
+
+ for (i = 0; i < SND_DG00X_RATE_COUNT; i++) {
+ if (!snd_interval_test(r, snd_dg00x_stream_rates[i]))
+ continue;
+
+ t.min = min(t.min, snd_dg00x_stream_pcm_channels[i]);
+ t.max = max(t.max, snd_dg00x_stream_pcm_channels[i]);
+ }
+
+ return snd_interval_refine(c, &t);
+}
+
+static int pcm_init_hw_params(struct snd_dg00x *dg00x,
+ struct snd_pcm_substream *substream)
+{
+ static const struct snd_pcm_hardware hardware = {
+ .info = SNDRV_PCM_INFO_BATCH |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_JOINT_DUPLEX |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID,
+ .rates = SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000,
+ .rate_min = 44100,
+ .rate_max = 96000,
+ .channels_min = 10,
+ .channels_max = 18,
+ .period_bytes_min = 4 * 18,
+ .period_bytes_max = 4 * 18 * 2048,
+ .buffer_bytes_max = 4 * 18 * 2048 * 2,
+ .periods_min = 2,
+ .periods_max = UINT_MAX,
+ };
+ struct amdtp_stream *s;
+ int err;
+
+ substream->runtime->hw = hardware;
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S32;
+ s = &dg00x->tx_stream;
+ } else {
+ substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S16 |
+ SNDRV_PCM_FMTBIT_S32;
+ s = &dg00x->rx_stream;
+ }
+
+ err = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_CHANNELS,
+ hw_rule_channels, NULL,
+ SNDRV_PCM_HW_PARAM_RATE, -1);
+ if (err < 0)
+ return err;
+
+ err = snd_pcm_hw_rule_add(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ hw_rule_rate, NULL,
+ SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+ if (err < 0)
+ return err;
+
+ return amdtp_dot_add_pcm_hw_constraints(s, substream->runtime);
+}
+
+static int pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+ enum snd_dg00x_clock clock;
+ bool detect;
+ unsigned int rate;
+ int err;
+
+ err = snd_dg00x_stream_lock_try(dg00x);
+ if (err < 0)
+ goto end;
+
+ err = pcm_init_hw_params(dg00x, substream);
+ if (err < 0)
+ goto err_locked;
+
+ /* Check current clock source. */
+ err = snd_dg00x_stream_get_clock(dg00x, &clock);
+ if (err < 0)
+ goto err_locked;
+ if (clock != SND_DG00X_CLOCK_INTERNAL) {
+ err = snd_dg00x_stream_check_external_clock(dg00x, &detect);
+ if (err < 0)
+ goto err_locked;
+ if (!detect) {
+ err = -EBUSY;
+ goto err_locked;
+ }
+ }
+
+ if ((clock != SND_DG00X_CLOCK_INTERNAL) ||
+ amdtp_stream_pcm_running(&dg00x->rx_stream) ||
+ amdtp_stream_pcm_running(&dg00x->tx_stream)) {
+ err = snd_dg00x_stream_get_external_rate(dg00x, &rate);
+ if (err < 0)
+ goto err_locked;
+ substream->runtime->hw.rate_min = rate;
+ substream->runtime->hw.rate_max = rate;
+ }
+
+ snd_pcm_set_sync(substream);
+end:
+ return err;
+err_locked:
+ snd_dg00x_stream_lock_release(dg00x);
+ return err;
+}
+
+static int pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+
+ snd_dg00x_stream_lock_release(dg00x);
+
+ return 0;
+}
+
+static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+ int err;
+
+ err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+
+ if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+ mutex_lock(&dg00x->mutex);
+ dg00x->substreams_counter++;
+ mutex_unlock(&dg00x->mutex);
+ }
+
+ amdtp_dot_set_pcm_format(&dg00x->tx_stream, params_format(hw_params));
+
+ return 0;
+}
+
+static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+ int err;
+
+ err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+
+ if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+ mutex_lock(&dg00x->mutex);
+ dg00x->substreams_counter++;
+ mutex_unlock(&dg00x->mutex);
+ }
+
+ amdtp_dot_set_pcm_format(&dg00x->rx_stream, params_format(hw_params));
+
+ return 0;
+}
+
+static int pcm_capture_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+
+ mutex_lock(&dg00x->mutex);
+
+ if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
+ dg00x->substreams_counter--;
+
+ snd_dg00x_stream_stop_duplex(dg00x);
+
+ mutex_unlock(&dg00x->mutex);
+
+ return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static int pcm_playback_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+
+ mutex_lock(&dg00x->mutex);
+
+ if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
+ dg00x->substreams_counter--;
+
+ snd_dg00x_stream_stop_duplex(dg00x);
+
+ mutex_unlock(&dg00x->mutex);
+
+ return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static int pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ mutex_lock(&dg00x->mutex);
+
+ err = snd_dg00x_stream_start_duplex(dg00x, runtime->rate);
+ if (err >= 0)
+ amdtp_stream_pcm_prepare(&dg00x->tx_stream);
+
+ mutex_unlock(&dg00x->mutex);
+
+ return err;
+}
+
+static int pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ mutex_lock(&dg00x->mutex);
+
+ err = snd_dg00x_stream_start_duplex(dg00x, runtime->rate);
+ if (err >= 0) {
+ amdtp_stream_pcm_prepare(&dg00x->rx_stream);
+ amdtp_dot_reset(&dg00x->rx_stream);
+ }
+
+ mutex_unlock(&dg00x->mutex);
+
+ return err;
+}
+
+static int pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ amdtp_stream_pcm_trigger(&dg00x->tx_stream, substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ amdtp_stream_pcm_trigger(&dg00x->tx_stream, NULL);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_dg00x *dg00x = substream->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ amdtp_stream_pcm_trigger(&dg00x->rx_stream, substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ amdtp_stream_pcm_trigger(&dg00x->rx_stream, NULL);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static snd_pcm_uframes_t pcm_capture_pointer(struct snd_pcm_substream *sbstrm)
+{
+ struct snd_dg00x *dg00x = sbstrm->private_data;
+
+ return amdtp_stream_pcm_pointer(&dg00x->tx_stream);
+}
+
+static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm)
+{
+ struct snd_dg00x *dg00x = sbstrm->private_data;
+
+ return amdtp_stream_pcm_pointer(&dg00x->rx_stream);
+}
+
+static struct snd_pcm_ops pcm_capture_ops = {
+ .open = pcm_open,
+ .close = pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pcm_capture_hw_params,
+ .hw_free = pcm_capture_hw_free,
+ .prepare = pcm_capture_prepare,
+ .trigger = pcm_capture_trigger,
+ .pointer = pcm_capture_pointer,
+ .page = snd_pcm_lib_get_vmalloc_page,
+};
+
+static struct snd_pcm_ops pcm_playback_ops = {
+ .open = pcm_open,
+ .close = pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pcm_playback_hw_params,
+ .hw_free = pcm_playback_hw_free,
+ .prepare = pcm_playback_prepare,
+ .trigger = pcm_playback_trigger,
+ .pointer = pcm_playback_pointer,
+ .page = snd_pcm_lib_get_vmalloc_page,
+ .mmap = snd_pcm_lib_mmap_vmalloc,
+};
+
+int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(dg00x->card, dg00x->card->driver, 0, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ pcm->private_data = dg00x;
+ snprintf(pcm->name, sizeof(pcm->name),
+ "%s PCM", dg00x->card->shortname);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_capture_ops);
+
+ return 0;
+}
diff --git a/sound/firewire/digi00x/digi00x-proc.c b/sound/firewire/digi00x/digi00x-proc.c
new file mode 100644
index 000000000000..a1d601f31165
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-proc.c
@@ -0,0 +1,99 @@
+/*
+ * digi00x-proc.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+static int get_optical_iface_mode(struct snd_dg00x *dg00x,
+ enum snd_dg00x_optical_mode *mode)
+{
+ __be32 data;
+ int err;
+
+ err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_OPT_IFACE_MODE,
+ &data, sizeof(data), 0);
+ if (err >= 0)
+ *mode = be32_to_cpu(data) & 0x01;
+
+ return err;
+}
+
+static void proc_read_clock(struct snd_info_entry *entry,
+ struct snd_info_buffer *buf)
+{
+ static const char *const source_name[] = {
+ [SND_DG00X_CLOCK_INTERNAL] = "internal",
+ [SND_DG00X_CLOCK_SPDIF] = "s/pdif",
+ [SND_DG00X_CLOCK_ADAT] = "adat",
+ [SND_DG00X_CLOCK_WORD] = "word clock",
+ };
+ static const char *const optical_name[] = {
+ [SND_DG00X_OPT_IFACE_MODE_ADAT] = "adat",
+ [SND_DG00X_OPT_IFACE_MODE_SPDIF] = "s/pdif",
+ };
+ struct snd_dg00x *dg00x = entry->private_data;
+ enum snd_dg00x_optical_mode mode;
+ unsigned int rate;
+ enum snd_dg00x_clock clock;
+ bool detect;
+
+ if (get_optical_iface_mode(dg00x, &mode) < 0)
+ return;
+ if (snd_dg00x_stream_get_local_rate(dg00x, &rate) < 0)
+ return;
+ if (snd_dg00x_stream_get_clock(dg00x, &clock) < 0)
+ return;
+
+ snd_iprintf(buf, "Optical mode: %s\n", optical_name[mode]);
+ snd_iprintf(buf, "Sampling Rate: %d\n", rate);
+ snd_iprintf(buf, "Clock Source: %s\n", source_name[clock]);
+
+ if (clock == SND_DG00X_CLOCK_INTERNAL)
+ return;
+
+ if (snd_dg00x_stream_check_external_clock(dg00x, &detect) < 0)
+ return;
+ snd_iprintf(buf, "External source: %s\n", detect ? "detected" : "not");
+ if (!detect)
+ return;
+
+ if (snd_dg00x_stream_get_external_rate(dg00x, &rate) >= 0)
+ snd_iprintf(buf, "External sampling rate: %d\n", rate);
+}
+
+void snd_dg00x_proc_init(struct snd_dg00x *dg00x)
+{
+ struct snd_info_entry *root, *entry;
+
+ /*
+ * All nodes are automatically removed at snd_card_disconnect(),
+ * by following to link list.
+ */
+ root = snd_info_create_card_entry(dg00x->card, "firewire",
+ dg00x->card->proc_root);
+ if (root == NULL)
+ return;
+
+ root->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+ if (snd_info_register(root) < 0) {
+ snd_info_free_entry(root);
+ return;
+ }
+
+ entry = snd_info_create_card_entry(dg00x->card, "clock", root);
+ if (entry == NULL) {
+ snd_info_free_entry(root);
+ return;
+ }
+
+ snd_info_set_text_ops(entry, dg00x, proc_read_clock);
+ if (snd_info_register(entry) < 0) {
+ snd_info_free_entry(entry);
+ snd_info_free_entry(root);
+ }
+}
diff --git a/sound/firewire/digi00x/digi00x-stream.c b/sound/firewire/digi00x/digi00x-stream.c
new file mode 100644
index 000000000000..4d3b4ebbdd49
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-stream.c
@@ -0,0 +1,422 @@
+/*
+ * digi00x-stream.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+#define CALLBACK_TIMEOUT 500
+
+const unsigned int snd_dg00x_stream_rates[SND_DG00X_RATE_COUNT] = {
+ [SND_DG00X_RATE_44100] = 44100,
+ [SND_DG00X_RATE_48000] = 48000,
+ [SND_DG00X_RATE_88200] = 88200,
+ [SND_DG00X_RATE_96000] = 96000,
+};
+
+/* Multi Bit Linear Audio data channels for each sampling transfer frequency. */
+const unsigned int
+snd_dg00x_stream_pcm_channels[SND_DG00X_RATE_COUNT] = {
+ /* Analog/ADAT/SPDIF */
+ [SND_DG00X_RATE_44100] = (8 + 8 + 2),
+ [SND_DG00X_RATE_48000] = (8 + 8 + 2),
+ /* Analog/SPDIF */
+ [SND_DG00X_RATE_88200] = (8 + 2),
+ [SND_DG00X_RATE_96000] = (8 + 2),
+};
+
+int snd_dg00x_stream_get_local_rate(struct snd_dg00x *dg00x, unsigned int *rate)
+{
+ u32 data;
+ __be32 reg;
+ int err;
+
+ err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_LOCAL_RATE,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ data = be32_to_cpu(reg) & 0x0f;
+ if (data < ARRAY_SIZE(snd_dg00x_stream_rates))
+ *rate = snd_dg00x_stream_rates[data];
+ else
+ err = -EIO;
+
+ return err;
+}
+
+int snd_dg00x_stream_set_local_rate(struct snd_dg00x *dg00x, unsigned int rate)
+{
+ __be32 reg;
+ unsigned int i;
+
+ for (i = 0; i < ARRAY_SIZE(snd_dg00x_stream_rates); i++) {
+ if (rate == snd_dg00x_stream_rates[i])
+ break;
+ }
+ if (i == ARRAY_SIZE(snd_dg00x_stream_rates))
+ return -EINVAL;
+
+ reg = cpu_to_be32(i);
+ return snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_LOCAL_RATE,
+ &reg, sizeof(reg), 0);
+}
+
+int snd_dg00x_stream_get_clock(struct snd_dg00x *dg00x,
+ enum snd_dg00x_clock *clock)
+{
+ __be32 reg;
+ int err;
+
+ err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_CLOCK_SOURCE,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ *clock = be32_to_cpu(reg) & 0x0f;
+ if (*clock >= SND_DG00X_CLOCK_COUNT)
+ err = -EIO;
+
+ return err;
+}
+
+int snd_dg00x_stream_check_external_clock(struct snd_dg00x *dg00x, bool *detect)
+{
+ __be32 reg;
+ int err;
+
+ err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_DETECT_EXTERNAL,
+ &reg, sizeof(reg), 0);
+ if (err >= 0)
+ *detect = be32_to_cpu(reg) > 0;
+
+ return err;
+}
+
+int snd_dg00x_stream_get_external_rate(struct snd_dg00x *dg00x,
+ unsigned int *rate)
+{
+ u32 data;
+ __be32 reg;
+ int err;
+
+ err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_EXTERNAL_RATE,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ data = be32_to_cpu(reg) & 0x0f;
+ if (data < ARRAY_SIZE(snd_dg00x_stream_rates))
+ *rate = snd_dg00x_stream_rates[data];
+ /* This means desync. */
+ else
+ err = -EBUSY;
+
+ return err;
+}
+
+static void finish_session(struct snd_dg00x *dg00x)
+{
+ __be32 data = cpu_to_be32(0x00000003);
+
+ snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_STREAMING_SET,
+ &data, sizeof(data), 0);
+}
+
+static int begin_session(struct snd_dg00x *dg00x)
+{
+ __be32 data;
+ u32 curr;
+ int err;
+
+ err = snd_fw_transaction(dg00x->unit, TCODE_READ_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_STREAMING_STATE,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ goto error;
+ curr = be32_to_cpu(data);
+
+ if (curr == 0)
+ curr = 2;
+
+ curr--;
+ while (curr > 0) {
+ data = cpu_to_be32(curr);
+ err = snd_fw_transaction(dg00x->unit,
+ TCODE_WRITE_QUADLET_REQUEST,
+ DG00X_ADDR_BASE +
+ DG00X_OFFSET_STREAMING_SET,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ goto error;
+
+ msleep(20);
+ curr--;
+ }
+
+ return 0;
+error:
+ finish_session(dg00x);
+ return err;
+}
+
+static void release_resources(struct snd_dg00x *dg00x)
+{
+ __be32 data = 0;
+
+ /* Unregister isochronous channels for both direction. */
+ snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_ISOC_CHANNELS,
+ &data, sizeof(data), 0);
+
+ /* Release isochronous resources. */
+ fw_iso_resources_free(&dg00x->tx_resources);
+ fw_iso_resources_free(&dg00x->rx_resources);
+}
+
+static int keep_resources(struct snd_dg00x *dg00x, unsigned int rate)
+{
+ unsigned int i;
+ __be32 data;
+ int err;
+
+ /* Check sampling rate. */
+ for (i = 0; i < SND_DG00X_RATE_COUNT; i++) {
+ if (snd_dg00x_stream_rates[i] == rate)
+ break;
+ }
+ if (i == SND_DG00X_RATE_COUNT)
+ return -EINVAL;
+
+ /* Keep resources for out-stream. */
+ err = amdtp_dot_set_parameters(&dg00x->rx_stream, rate,
+ snd_dg00x_stream_pcm_channels[i]);
+ if (err < 0)
+ return err;
+ err = fw_iso_resources_allocate(&dg00x->rx_resources,
+ amdtp_stream_get_max_payload(&dg00x->rx_stream),
+ fw_parent_device(dg00x->unit)->max_speed);
+ if (err < 0)
+ return err;
+
+ /* Keep resources for in-stream. */
+ err = amdtp_dot_set_parameters(&dg00x->tx_stream, rate,
+ snd_dg00x_stream_pcm_channels[i]);
+ if (err < 0)
+ return err;
+ err = fw_iso_resources_allocate(&dg00x->tx_resources,
+ amdtp_stream_get_max_payload(&dg00x->tx_stream),
+ fw_parent_device(dg00x->unit)->max_speed);
+ if (err < 0)
+ goto error;
+
+ /* Register isochronous channels for both direction. */
+ data = cpu_to_be32((dg00x->tx_resources.channel << 16) |
+ dg00x->rx_resources.channel);
+ err = snd_fw_transaction(dg00x->unit, TCODE_WRITE_QUADLET_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_ISOC_CHANNELS,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ goto error;
+
+ return 0;
+error:
+ release_resources(dg00x);
+ return err;
+}
+
+int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x)
+{
+ int err;
+
+ /* For out-stream. */
+ err = fw_iso_resources_init(&dg00x->rx_resources, dg00x->unit);
+ if (err < 0)
+ goto error;
+ err = amdtp_dot_init(&dg00x->rx_stream, dg00x->unit, AMDTP_OUT_STREAM);
+ if (err < 0)
+ goto error;
+
+ /* For in-stream. */
+ err = fw_iso_resources_init(&dg00x->tx_resources, dg00x->unit);
+ if (err < 0)
+ goto error;
+ err = amdtp_dot_init(&dg00x->tx_stream, dg00x->unit, AMDTP_IN_STREAM);
+ if (err < 0)
+ goto error;
+
+ return 0;
+error:
+ snd_dg00x_stream_destroy_duplex(dg00x);
+ return err;
+}
+
+/*
+ * This function should be called before starting streams or after stopping
+ * streams.
+ */
+void snd_dg00x_stream_destroy_duplex(struct snd_dg00x *dg00x)
+{
+ amdtp_stream_destroy(&dg00x->rx_stream);
+ fw_iso_resources_destroy(&dg00x->rx_resources);
+
+ amdtp_stream_destroy(&dg00x->tx_stream);
+ fw_iso_resources_destroy(&dg00x->tx_resources);
+}
+
+int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x, unsigned int rate)
+{
+ unsigned int curr_rate;
+ int err = 0;
+
+ if (dg00x->substreams_counter == 0)
+ goto end;
+
+ /* Check current sampling rate. */
+ err = snd_dg00x_stream_get_local_rate(dg00x, &curr_rate);
+ if (err < 0)
+ goto error;
+ if (rate == 0)
+ rate = curr_rate;
+ if (curr_rate != rate ||
+ amdtp_streaming_error(&dg00x->tx_stream) ||
+ amdtp_streaming_error(&dg00x->rx_stream)) {
+ finish_session(dg00x);
+
+ amdtp_stream_stop(&dg00x->tx_stream);
+ amdtp_stream_stop(&dg00x->rx_stream);
+ release_resources(dg00x);
+ }
+
+ /*
+ * No packets are transmitted without receiving packets, reagardless of
+ * which source of clock is used.
+ */
+ if (!amdtp_stream_running(&dg00x->rx_stream)) {
+ err = snd_dg00x_stream_set_local_rate(dg00x, rate);
+ if (err < 0)
+ goto error;
+
+ err = keep_resources(dg00x, rate);
+ if (err < 0)
+ goto error;
+
+ err = begin_session(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = amdtp_stream_start(&dg00x->rx_stream,
+ dg00x->rx_resources.channel,
+ fw_parent_device(dg00x->unit)->max_speed);
+ if (err < 0)
+ goto error;
+
+ if (!amdtp_stream_wait_callback(&dg00x->rx_stream,
+ CALLBACK_TIMEOUT)) {
+ err = -ETIMEDOUT;
+ goto error;
+ }
+ }
+
+ /*
+ * The value of SYT field in transmitted packets is always 0x0000. Thus,
+ * duplex streams with timestamp synchronization cannot be built.
+ */
+ if (!amdtp_stream_running(&dg00x->tx_stream)) {
+ err = amdtp_stream_start(&dg00x->tx_stream,
+ dg00x->tx_resources.channel,
+ fw_parent_device(dg00x->unit)->max_speed);
+ if (err < 0)
+ goto error;
+
+ if (!amdtp_stream_wait_callback(&dg00x->tx_stream,
+ CALLBACK_TIMEOUT)) {
+ err = -ETIMEDOUT;
+ goto error;
+ }
+ }
+end:
+ return err;
+error:
+ finish_session(dg00x);
+
+ amdtp_stream_stop(&dg00x->tx_stream);
+ amdtp_stream_stop(&dg00x->rx_stream);
+ release_resources(dg00x);
+
+ return err;
+}
+
+void snd_dg00x_stream_stop_duplex(struct snd_dg00x *dg00x)
+{
+ if (dg00x->substreams_counter > 0)
+ return;
+
+ amdtp_stream_stop(&dg00x->tx_stream);
+ amdtp_stream_stop(&dg00x->rx_stream);
+ finish_session(dg00x);
+ release_resources(dg00x);
+
+ /*
+ * Just after finishing the session, the device may lost transmitting
+ * functionality for a short time.
+ */
+ msleep(50);
+}
+
+void snd_dg00x_stream_update_duplex(struct snd_dg00x *dg00x)
+{
+ fw_iso_resources_update(&dg00x->tx_resources);
+ fw_iso_resources_update(&dg00x->rx_resources);
+
+ amdtp_stream_update(&dg00x->tx_stream);
+ amdtp_stream_update(&dg00x->rx_stream);
+}
+
+void snd_dg00x_stream_lock_changed(struct snd_dg00x *dg00x)
+{
+ dg00x->dev_lock_changed = true;
+ wake_up(&dg00x->hwdep_wait);
+}
+
+int snd_dg00x_stream_lock_try(struct snd_dg00x *dg00x)
+{
+ int err;
+
+ spin_lock_irq(&dg00x->lock);
+
+ /* user land lock this */
+ if (dg00x->dev_lock_count < 0) {
+ err = -EBUSY;
+ goto end;
+ }
+
+ /* this is the first time */
+ if (dg00x->dev_lock_count++ == 0)
+ snd_dg00x_stream_lock_changed(dg00x);
+ err = 0;
+end:
+ spin_unlock_irq(&dg00x->lock);
+ return err;
+}
+
+void snd_dg00x_stream_lock_release(struct snd_dg00x *dg00x)
+{
+ spin_lock_irq(&dg00x->lock);
+
+ if (WARN_ON(dg00x->dev_lock_count <= 0))
+ goto end;
+ if (--dg00x->dev_lock_count == 0)
+ snd_dg00x_stream_lock_changed(dg00x);
+end:
+ spin_unlock_irq(&dg00x->lock);
+}
diff --git a/sound/firewire/digi00x/digi00x-transaction.c b/sound/firewire/digi00x/digi00x-transaction.c
new file mode 100644
index 000000000000..554324d8c602
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x-transaction.c
@@ -0,0 +1,137 @@
+/*
+ * digi00x-transaction.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <sound/asound.h>
+#include "digi00x.h"
+
+static int fill_midi_message(struct snd_rawmidi_substream *substream, u8 *buf)
+{
+ int bytes;
+
+ buf[0] = 0x80;
+ bytes = snd_rawmidi_transmit_peek(substream, buf + 1, 2);
+ if (bytes >= 0)
+ buf[3] = 0xc0 | bytes;
+
+ return bytes;
+}
+
+static void handle_midi_control(struct snd_dg00x *dg00x, __be32 *buf,
+ unsigned int length)
+{
+ struct snd_rawmidi_substream *substream;
+ unsigned int i;
+ unsigned int len;
+ u8 *b;
+
+ substream = ACCESS_ONCE(dg00x->in_control);
+ if (substream == NULL)
+ return;
+
+ length /= 4;
+
+ for (i = 0; i < length; i++) {
+ b = (u8 *)&buf[i];
+ len = b[3] & 0xf;
+ if (len > 0)
+ snd_rawmidi_receive(dg00x->in_control, b + 1, len);
+ }
+}
+
+static void handle_unknown_message(struct snd_dg00x *dg00x,
+ unsigned long long offset, __be32 *buf)
+{
+ unsigned long flags;
+
+ spin_lock_irqsave(&dg00x->lock, flags);
+ dg00x->msg = be32_to_cpu(*buf);
+ spin_unlock_irqrestore(&dg00x->lock, flags);
+
+ wake_up(&dg00x->hwdep_wait);
+}
+
+static void handle_message(struct fw_card *card, struct fw_request *request,
+ int tcode, int destination, int source,
+ int generation, unsigned long long offset,
+ void *data, size_t length, void *callback_data)
+{
+ struct snd_dg00x *dg00x = callback_data;
+ __be32 *buf = (__be32 *)data;
+
+ if (offset == dg00x->async_handler.offset)
+ handle_unknown_message(dg00x, offset, buf);
+ else if (offset == dg00x->async_handler.offset + 4)
+ handle_midi_control(dg00x, buf, length);
+
+ fw_send_response(card, request, RCODE_COMPLETE);
+}
+
+int snd_dg00x_transaction_reregister(struct snd_dg00x *dg00x)
+{
+ struct fw_device *device = fw_parent_device(dg00x->unit);
+ __be32 data[2];
+ int err;
+
+ /* Unknown. 4bytes. */
+ data[0] = cpu_to_be32((device->card->node_id << 16) |
+ (dg00x->async_handler.offset >> 32));
+ data[1] = cpu_to_be32(dg00x->async_handler.offset);
+ err = snd_fw_transaction(dg00x->unit, TCODE_WRITE_BLOCK_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_MESSAGE_ADDR,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ return err;
+
+ /* Asynchronous transactions for MIDI control message. */
+ data[0] = cpu_to_be32((device->card->node_id << 16) |
+ (dg00x->async_handler.offset >> 32));
+ data[1] = cpu_to_be32(dg00x->async_handler.offset + 4);
+ return snd_fw_transaction(dg00x->unit, TCODE_WRITE_BLOCK_REQUEST,
+ DG00X_ADDR_BASE + DG00X_OFFSET_MIDI_CTL_ADDR,
+ &data, sizeof(data), 0);
+}
+
+int snd_dg00x_transaction_register(struct snd_dg00x *dg00x)
+{
+ static const struct fw_address_region resp_register_region = {
+ .start = 0xffffe0000000ull,
+ .end = 0xffffe000ffffull,
+ };
+ int err;
+
+ dg00x->async_handler.length = 12;
+ dg00x->async_handler.address_callback = handle_message;
+ dg00x->async_handler.callback_data = dg00x;
+
+ err = fw_core_add_address_handler(&dg00x->async_handler,
+ &resp_register_region);
+ if (err < 0)
+ return err;
+
+ err = snd_dg00x_transaction_reregister(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = snd_fw_async_midi_port_init(&dg00x->out_control, dg00x->unit,
+ DG00X_ADDR_BASE + DG00X_OFFSET_MMC,
+ 4, fill_midi_message);
+ if (err < 0)
+ goto error;
+
+ return err;
+error:
+ fw_core_remove_address_handler(&dg00x->async_handler);
+ dg00x->async_handler.address_callback = NULL;
+ return err;
+}
+
+void snd_dg00x_transaction_unregister(struct snd_dg00x *dg00x)
+{
+ snd_fw_async_midi_port_destroy(&dg00x->out_control);
+ fw_core_remove_address_handler(&dg00x->async_handler);
+}
diff --git a/sound/firewire/digi00x/digi00x.c b/sound/firewire/digi00x/digi00x.c
new file mode 100644
index 000000000000..1f33b7a1fca4
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x.c
@@ -0,0 +1,170 @@
+/*
+ * digi00x.c - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "digi00x.h"
+
+MODULE_DESCRIPTION("Digidesign Digi 002/003 family Driver");
+MODULE_AUTHOR("Takashi Sakamoto <o-takashi@sakamocchi.jp>");
+MODULE_LICENSE("GPL v2");
+
+#define VENDOR_DIGIDESIGN 0x00a07e
+#define MODEL_DIGI00X 0x000002
+
+static int name_card(struct snd_dg00x *dg00x)
+{
+ struct fw_device *fw_dev = fw_parent_device(dg00x->unit);
+ char name[32] = {0};
+ char *model;
+ int err;
+
+ err = fw_csr_string(dg00x->unit->directory, CSR_MODEL, name,
+ sizeof(name));
+ if (err < 0)
+ return err;
+
+ model = skip_spaces(name);
+
+ strcpy(dg00x->card->driver, "Digi00x");
+ strcpy(dg00x->card->shortname, model);
+ strcpy(dg00x->card->mixername, model);
+ snprintf(dg00x->card->longname, sizeof(dg00x->card->longname),
+ "Digidesign %s, GUID %08x%08x at %s, S%d", model,
+ fw_dev->config_rom[3], fw_dev->config_rom[4],
+ dev_name(&dg00x->unit->device), 100 << fw_dev->max_speed);
+
+ return 0;
+}
+
+static void dg00x_card_free(struct snd_card *card)
+{
+ struct snd_dg00x *dg00x = card->private_data;
+
+ snd_dg00x_stream_destroy_duplex(dg00x);
+ snd_dg00x_transaction_unregister(dg00x);
+
+ fw_unit_put(dg00x->unit);
+
+ mutex_destroy(&dg00x->mutex);
+}
+
+static int snd_dg00x_probe(struct fw_unit *unit,
+ const struct ieee1394_device_id *entry)
+{
+ struct snd_card *card;
+ struct snd_dg00x *dg00x;
+ int err;
+
+ /* create card */
+ err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE,
+ sizeof(struct snd_dg00x), &card);
+ if (err < 0)
+ return err;
+ card->private_free = dg00x_card_free;
+
+ /* initialize myself */
+ dg00x = card->private_data;
+ dg00x->card = card;
+ dg00x->unit = fw_unit_get(unit);
+
+ mutex_init(&dg00x->mutex);
+ spin_lock_init(&dg00x->lock);
+ init_waitqueue_head(&dg00x->hwdep_wait);
+
+ err = name_card(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = snd_dg00x_stream_init_duplex(dg00x);
+ if (err < 0)
+ goto error;
+
+ snd_dg00x_proc_init(dg00x);
+
+ err = snd_dg00x_create_pcm_devices(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = snd_dg00x_create_midi_devices(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = snd_dg00x_create_hwdep_device(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = snd_dg00x_transaction_register(dg00x);
+ if (err < 0)
+ goto error;
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto error;
+
+ dev_set_drvdata(&unit->device, dg00x);
+
+ return err;
+error:
+ snd_card_free(card);
+ return err;
+}
+
+static void snd_dg00x_update(struct fw_unit *unit)
+{
+ struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device);
+
+ snd_dg00x_transaction_reregister(dg00x);
+
+ mutex_lock(&dg00x->mutex);
+ snd_dg00x_stream_update_duplex(dg00x);
+ mutex_unlock(&dg00x->mutex);
+}
+
+static void snd_dg00x_remove(struct fw_unit *unit)
+{
+ struct snd_dg00x *dg00x = dev_get_drvdata(&unit->device);
+
+ /* No need to wait for releasing card object in this context. */
+ snd_card_free_when_closed(dg00x->card);
+}
+
+static const struct ieee1394_device_id snd_dg00x_id_table[] = {
+ /* Both of 002/003 use the same ID. */
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_MODEL_ID,
+ .vendor_id = VENDOR_DIGIDESIGN,
+ .model_id = MODEL_DIGI00X,
+ },
+ {}
+};
+MODULE_DEVICE_TABLE(ieee1394, snd_dg00x_id_table);
+
+static struct fw_driver dg00x_driver = {
+ .driver = {
+ .owner = THIS_MODULE,
+ .name = "snd-firewire-digi00x",
+ .bus = &fw_bus_type,
+ },
+ .probe = snd_dg00x_probe,
+ .update = snd_dg00x_update,
+ .remove = snd_dg00x_remove,
+ .id_table = snd_dg00x_id_table,
+};
+
+static int __init snd_dg00x_init(void)
+{
+ return driver_register(&dg00x_driver.driver);
+}
+
+static void __exit snd_dg00x_exit(void)
+{
+ driver_unregister(&dg00x_driver.driver);
+}
+
+module_init(snd_dg00x_init);
+module_exit(snd_dg00x_exit);
diff --git a/sound/firewire/digi00x/digi00x.h b/sound/firewire/digi00x/digi00x.h
new file mode 100644
index 000000000000..907e73993677
--- /dev/null
+++ b/sound/firewire/digi00x/digi00x.h
@@ -0,0 +1,157 @@
+/*
+ * digi00x.h - a part of driver for Digidesign Digi 002/003 family
+ *
+ * Copyright (c) 2014-2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#ifndef SOUND_DIGI00X_H_INCLUDED
+#define SOUND_DIGI00X_H_INCLUDED
+
+#include <linux/compat.h>
+#include <linux/device.h>
+#include <linux/firewire.h>
+#include <linux/module.h>
+#include <linux/mod_devicetable.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/info.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/firewire.h>
+#include <sound/hwdep.h>
+#include <sound/rawmidi.h>
+
+#include "../lib.h"
+#include "../iso-resources.h"
+#include "../amdtp-stream.h"
+
+struct snd_dg00x {
+ struct snd_card *card;
+ struct fw_unit *unit;
+
+ struct mutex mutex;
+ spinlock_t lock;
+
+ struct amdtp_stream tx_stream;
+ struct fw_iso_resources tx_resources;
+
+ struct amdtp_stream rx_stream;
+ struct fw_iso_resources rx_resources;
+
+ unsigned int substreams_counter;
+
+ /* for uapi */
+ int dev_lock_count;
+ bool dev_lock_changed;
+ wait_queue_head_t hwdep_wait;
+
+ /* For asynchronous messages. */
+ struct fw_address_handler async_handler;
+ u32 msg;
+
+ /* For asynchronous MIDI controls. */
+ struct snd_rawmidi_substream *in_control;
+ struct snd_fw_async_midi_port out_control;
+};
+
+#define DG00X_ADDR_BASE 0xffffe0000000ull
+
+#define DG00X_OFFSET_STREAMING_STATE 0x0000
+#define DG00X_OFFSET_STREAMING_SET 0x0004
+#define DG00X_OFFSET_MIDI_CTL_ADDR 0x0008
+/* For LSB of the address 0x000c */
+/* unknown 0x0010 */
+#define DG00X_OFFSET_MESSAGE_ADDR 0x0014
+/* For LSB of the address 0x0018 */
+/* unknown 0x001c */
+/* unknown 0x0020 */
+/* not used 0x0024--0x00ff */
+#define DG00X_OFFSET_ISOC_CHANNELS 0x0100
+/* unknown 0x0104 */
+/* unknown 0x0108 */
+/* unknown 0x010c */
+#define DG00X_OFFSET_LOCAL_RATE 0x0110
+#define DG00X_OFFSET_EXTERNAL_RATE 0x0114
+#define DG00X_OFFSET_CLOCK_SOURCE 0x0118
+#define DG00X_OFFSET_OPT_IFACE_MODE 0x011c
+/* unknown 0x0120 */
+/* Mixer control on/off 0x0124 */
+/* unknown 0x0128 */
+#define DG00X_OFFSET_DETECT_EXTERNAL 0x012c
+/* unknown 0x0138 */
+#define DG00X_OFFSET_MMC 0x0400
+
+enum snd_dg00x_rate {
+ SND_DG00X_RATE_44100 = 0,
+ SND_DG00X_RATE_48000,
+ SND_DG00X_RATE_88200,
+ SND_DG00X_RATE_96000,
+ SND_DG00X_RATE_COUNT,
+};
+
+enum snd_dg00x_clock {
+ SND_DG00X_CLOCK_INTERNAL = 0,
+ SND_DG00X_CLOCK_SPDIF,
+ SND_DG00X_CLOCK_ADAT,
+ SND_DG00X_CLOCK_WORD,
+ SND_DG00X_CLOCK_COUNT,
+};
+
+enum snd_dg00x_optical_mode {
+ SND_DG00X_OPT_IFACE_MODE_ADAT = 0,
+ SND_DG00X_OPT_IFACE_MODE_SPDIF,
+ SND_DG00X_OPT_IFACE_MODE_COUNT,
+};
+
+#define DOT_MIDI_IN_PORTS 1
+#define DOT_MIDI_OUT_PORTS 2
+
+int amdtp_dot_init(struct amdtp_stream *s, struct fw_unit *unit,
+ enum amdtp_stream_direction dir);
+int amdtp_dot_set_parameters(struct amdtp_stream *s, unsigned int rate,
+ unsigned int pcm_channels);
+void amdtp_dot_reset(struct amdtp_stream *s);
+int amdtp_dot_add_pcm_hw_constraints(struct amdtp_stream *s,
+ struct snd_pcm_runtime *runtime);
+void amdtp_dot_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format);
+void amdtp_dot_midi_trigger(struct amdtp_stream *s, unsigned int port,
+ struct snd_rawmidi_substream *midi);
+
+int snd_dg00x_transaction_register(struct snd_dg00x *dg00x);
+int snd_dg00x_transaction_reregister(struct snd_dg00x *dg00x);
+void snd_dg00x_transaction_unregister(struct snd_dg00x *dg00x);
+
+extern const unsigned int snd_dg00x_stream_rates[SND_DG00X_RATE_COUNT];
+extern const unsigned int snd_dg00x_stream_pcm_channels[SND_DG00X_RATE_COUNT];
+int snd_dg00x_stream_get_external_rate(struct snd_dg00x *dg00x,
+ unsigned int *rate);
+int snd_dg00x_stream_get_local_rate(struct snd_dg00x *dg00x,
+ unsigned int *rate);
+int snd_dg00x_stream_set_local_rate(struct snd_dg00x *dg00x, unsigned int rate);
+int snd_dg00x_stream_get_clock(struct snd_dg00x *dg00x,
+ enum snd_dg00x_clock *clock);
+int snd_dg00x_stream_check_external_clock(struct snd_dg00x *dg00x,
+ bool *detect);
+int snd_dg00x_stream_init_duplex(struct snd_dg00x *dg00x);
+int snd_dg00x_stream_start_duplex(struct snd_dg00x *dg00x, unsigned int rate);
+void snd_dg00x_stream_stop_duplex(struct snd_dg00x *dg00x);
+void snd_dg00x_stream_update_duplex(struct snd_dg00x *dg00x);
+void snd_dg00x_stream_destroy_duplex(struct snd_dg00x *dg00x);
+
+void snd_dg00x_stream_lock_changed(struct snd_dg00x *dg00x);
+int snd_dg00x_stream_lock_try(struct snd_dg00x *dg00x);
+void snd_dg00x_stream_lock_release(struct snd_dg00x *dg00x);
+
+void snd_dg00x_proc_init(struct snd_dg00x *dg00x);
+
+int snd_dg00x_create_pcm_devices(struct snd_dg00x *dg00x);
+
+int snd_dg00x_create_midi_devices(struct snd_dg00x *dg00x);
+
+int snd_dg00x_create_hwdep_device(struct snd_dg00x *dg00x);
+#endif
diff --git a/sound/firewire/fcp.c b/sound/firewire/fcp.c
index 0619597e3a3f..cce19768f43d 100644
--- a/sound/firewire/fcp.c
+++ b/sound/firewire/fcp.c
@@ -17,7 +17,7 @@
#include <linux/delay.h>
#include "fcp.h"
#include "lib.h"
-#include "amdtp.h"
+#include "amdtp-stream.h"
#define CTS_AVC 0x00
diff --git a/sound/firewire/fireworks/Makefile b/sound/firewire/fireworks/Makefile
index 0c7440826db8..15ef7f75a8ef 100644
--- a/sound/firewire/fireworks/Makefile
+++ b/sound/firewire/fireworks/Makefile
@@ -1,4 +1,4 @@
snd-fireworks-objs := fireworks_transaction.o fireworks_command.o \
fireworks_stream.o fireworks_proc.o fireworks_midi.o \
fireworks_pcm.o fireworks_hwdep.o fireworks.o
-obj-m += snd-fireworks.o
+obj-$(CONFIG_SND_FIREWORKS) += snd-fireworks.o
diff --git a/sound/firewire/fireworks/fireworks.c b/sound/firewire/fireworks/fireworks.c
index c94a432f7cc6..d5b19bc11e59 100644
--- a/sound/firewire/fireworks/fireworks.c
+++ b/sound/firewire/fireworks/fireworks.c
@@ -138,12 +138,12 @@ get_hardware_info(struct snd_efw *efw)
efw->midi_out_ports = hwinfo->midi_out_ports;
efw->midi_in_ports = hwinfo->midi_in_ports;
- if (hwinfo->amdtp_tx_pcm_channels > AMDTP_MAX_CHANNELS_FOR_PCM ||
- hwinfo->amdtp_tx_pcm_channels_2x > AMDTP_MAX_CHANNELS_FOR_PCM ||
- hwinfo->amdtp_tx_pcm_channels_4x > AMDTP_MAX_CHANNELS_FOR_PCM ||
- hwinfo->amdtp_rx_pcm_channels > AMDTP_MAX_CHANNELS_FOR_PCM ||
- hwinfo->amdtp_rx_pcm_channels_2x > AMDTP_MAX_CHANNELS_FOR_PCM ||
- hwinfo->amdtp_rx_pcm_channels_4x > AMDTP_MAX_CHANNELS_FOR_PCM) {
+ if (hwinfo->amdtp_tx_pcm_channels > AM824_MAX_CHANNELS_FOR_PCM ||
+ hwinfo->amdtp_tx_pcm_channels_2x > AM824_MAX_CHANNELS_FOR_PCM ||
+ hwinfo->amdtp_tx_pcm_channels_4x > AM824_MAX_CHANNELS_FOR_PCM ||
+ hwinfo->amdtp_rx_pcm_channels > AM824_MAX_CHANNELS_FOR_PCM ||
+ hwinfo->amdtp_rx_pcm_channels_2x > AM824_MAX_CHANNELS_FOR_PCM ||
+ hwinfo->amdtp_rx_pcm_channels_4x > AM824_MAX_CHANNELS_FOR_PCM) {
err = -ENOSYS;
goto end;
}
diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h
index 084d414b228c..c7cb7deafe48 100644
--- a/sound/firewire/fireworks/fireworks.h
+++ b/sound/firewire/fireworks/fireworks.h
@@ -29,7 +29,7 @@
#include "../packets-buffer.h"
#include "../iso-resources.h"
-#include "../amdtp.h"
+#include "../amdtp-am824.h"
#include "../cmp.h"
#include "../lib.h"
diff --git a/sound/firewire/fireworks/fireworks_command.c b/sound/firewire/fireworks/fireworks_command.c
index 166f80584c2a..94bab0476a65 100644
--- a/sound/firewire/fireworks/fireworks_command.c
+++ b/sound/firewire/fireworks/fireworks_command.c
@@ -257,7 +257,7 @@ int snd_efw_command_get_phys_meters(struct snd_efw *efw,
struct snd_efw_phys_meters *meters,
unsigned int len)
{
- __be32 *buf = (__be32 *)meters;
+ u32 *buf = (u32 *)meters;
unsigned int i;
int err;
diff --git a/sound/firewire/fireworks/fireworks_midi.c b/sound/firewire/fireworks/fireworks_midi.c
index cf9c65260439..fba01bbba456 100644
--- a/sound/firewire/fireworks/fireworks_midi.c
+++ b/sound/firewire/fireworks/fireworks_midi.c
@@ -73,10 +73,10 @@ static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up)
spin_lock_irqsave(&efw->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&efw->tx_stream,
+ amdtp_am824_midi_trigger(&efw->tx_stream,
substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&efw->tx_stream,
+ amdtp_am824_midi_trigger(&efw->tx_stream,
substrm->number, NULL);
spin_unlock_irqrestore(&efw->lock, flags);
@@ -90,11 +90,11 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
spin_lock_irqsave(&efw->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&efw->rx_stream,
- substrm->number, substrm);
+ amdtp_am824_midi_trigger(&efw->rx_stream,
+ substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&efw->rx_stream,
- substrm->number, NULL);
+ amdtp_am824_midi_trigger(&efw->rx_stream,
+ substrm->number, NULL);
spin_unlock_irqrestore(&efw->lock, flags);
}
diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c
index c30b2ffa8dfb..d27135bac513 100644
--- a/sound/firewire/fireworks/fireworks_pcm.c
+++ b/sound/firewire/fireworks/fireworks_pcm.c
@@ -159,11 +159,11 @@ pcm_init_hw_params(struct snd_efw *efw,
SNDRV_PCM_INFO_MMAP_VALID;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- runtime->hw.formats = AMDTP_IN_PCM_FORMAT_BITS;
+ runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS;
s = &efw->tx_stream;
pcm_channels = efw->pcm_capture_channels;
} else {
- runtime->hw.formats = AMDTP_OUT_PCM_FORMAT_BITS;
+ runtime->hw.formats = AM824_OUT_PCM_FORMAT_BITS;
s = &efw->rx_stream;
pcm_channels = efw->pcm_playback_channels;
}
@@ -187,7 +187,7 @@ pcm_init_hw_params(struct snd_efw *efw,
if (err < 0)
goto end;
- err = amdtp_stream_add_pcm_hw_constraints(s, runtime);
+ err = amdtp_am824_add_pcm_hw_constraints(s, runtime);
end:
return err;
}
@@ -253,7 +253,8 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN)
atomic_inc(&efw->capture_substreams);
- amdtp_stream_set_pcm_format(&efw->tx_stream, params_format(hw_params));
+
+ amdtp_am824_set_pcm_format(&efw->tx_stream, params_format(hw_params));
return 0;
}
@@ -270,7 +271,8 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN)
atomic_inc(&efw->playback_substreams);
- amdtp_stream_set_pcm_format(&efw->rx_stream, params_format(hw_params));
+
+ amdtp_am824_set_pcm_format(&efw->rx_stream, params_format(hw_params));
return 0;
}
diff --git a/sound/firewire/fireworks/fireworks_stream.c b/sound/firewire/fireworks/fireworks_stream.c
index 7e353f1f7bff..759f6e3ed44a 100644
--- a/sound/firewire/fireworks/fireworks_stream.c
+++ b/sound/firewire/fireworks/fireworks_stream.c
@@ -31,7 +31,7 @@ init_stream(struct snd_efw *efw, struct amdtp_stream *stream)
if (err < 0)
goto end;
- err = amdtp_stream_init(stream, efw->unit, s_dir, CIP_BLOCKING);
+ err = amdtp_am824_init(stream, efw->unit, s_dir, CIP_BLOCKING);
if (err < 0) {
amdtp_stream_destroy(stream);
cmp_connection_destroy(conn);
@@ -73,8 +73,10 @@ start_stream(struct snd_efw *efw, struct amdtp_stream *stream,
midi_ports = efw->midi_in_ports;
}
- amdtp_stream_set_parameters(stream, sampling_rate,
- pcm_channels, midi_ports);
+ err = amdtp_am824_set_parameters(stream, sampling_rate,
+ pcm_channels, midi_ports, false);
+ if (err < 0)
+ goto end;
/* establish connection via CMP */
err = cmp_connection_establish(conn,
diff --git a/sound/firewire/lib.c b/sound/firewire/lib.c
index 7409edba9f06..f80aafa44c89 100644
--- a/sound/firewire/lib.c
+++ b/sound/firewire/lib.c
@@ -9,6 +9,7 @@
#include <linux/device.h>
#include <linux/firewire.h>
#include <linux/module.h>
+#include <linux/slab.h>
#include "lib.h"
#define ERROR_RETRY_DELAY_MS 20
@@ -66,6 +67,147 @@ int snd_fw_transaction(struct fw_unit *unit, int tcode,
}
EXPORT_SYMBOL(snd_fw_transaction);
+static void async_midi_port_callback(struct fw_card *card, int rcode,
+ void *data, size_t length,
+ void *callback_data)
+{
+ struct snd_fw_async_midi_port *port = callback_data;
+ struct snd_rawmidi_substream *substream = ACCESS_ONCE(port->substream);
+
+ /* This port is closed. */
+ if (substream == NULL)
+ return;
+
+ if (rcode == RCODE_COMPLETE)
+ snd_rawmidi_transmit_ack(substream, port->consume_bytes);
+ else if (!rcode_is_permanent_error(rcode))
+ /* To start next transaction immediately for recovery. */
+ port->next_ktime = ktime_set(0, 0);
+ else
+ /* Don't continue processing. */
+ port->error = true;
+
+ port->idling = true;
+
+ if (!snd_rawmidi_transmit_empty(substream))
+ schedule_work(&port->work);
+}
+
+static void midi_port_work(struct work_struct *work)
+{
+ struct snd_fw_async_midi_port *port =
+ container_of(work, struct snd_fw_async_midi_port, work);
+ struct snd_rawmidi_substream *substream = ACCESS_ONCE(port->substream);
+ int generation;
+ int type;
+
+ /* Under transacting or error state. */
+ if (!port->idling || port->error)
+ return;
+
+ /* Nothing to do. */
+ if (substream == NULL || snd_rawmidi_transmit_empty(substream))
+ return;
+
+ /* Do it in next chance. */
+ if (ktime_after(port->next_ktime, ktime_get())) {
+ schedule_work(&port->work);
+ return;
+ }
+
+ /*
+ * Fill the buffer. The callee must use snd_rawmidi_transmit_peek().
+ * Later, snd_rawmidi_transmit_ack() is called.
+ */
+ memset(port->buf, 0, port->len);
+ port->consume_bytes = port->fill(substream, port->buf);
+ if (port->consume_bytes <= 0) {
+ /* Do it in next chance, immediately. */
+ if (port->consume_bytes == 0) {
+ port->next_ktime = ktime_set(0, 0);
+ schedule_work(&port->work);
+ } else {
+ /* Fatal error. */
+ port->error = true;
+ }
+ return;
+ }
+
+ /* Calculate type of transaction. */
+ if (port->len == 4)
+ type = TCODE_WRITE_QUADLET_REQUEST;
+ else
+ type = TCODE_WRITE_BLOCK_REQUEST;
+
+ /* Set interval to next transaction. */
+ port->next_ktime = ktime_add_ns(ktime_get(),
+ port->consume_bytes * 8 * NSEC_PER_SEC / 31250);
+
+ /* Start this transaction. */
+ port->idling = false;
+
+ /*
+ * In Linux FireWire core, when generation is updated with memory
+ * barrier, node id has already been updated. In this module, After
+ * this smp_rmb(), load/store instructions to memory are completed.
+ * Thus, both of generation and node id are available with recent
+ * values. This is a light-serialization solution to handle bus reset
+ * events on IEEE 1394 bus.
+ */
+ generation = port->parent->generation;
+ smp_rmb();
+
+ fw_send_request(port->parent->card, &port->transaction, type,
+ port->parent->node_id, generation,
+ port->parent->max_speed, port->addr,
+ port->buf, port->len, async_midi_port_callback,
+ port);
+}
+
+/**
+ * snd_fw_async_midi_port_init - initialize asynchronous MIDI port structure
+ * @port: the asynchronous MIDI port to initialize
+ * @unit: the target of the asynchronous transaction
+ * @addr: the address to which transactions are transferred
+ * @len: the length of transaction
+ * @fill: the callback function to fill given buffer, and returns the
+ * number of consumed bytes for MIDI message.
+ *
+ */
+int snd_fw_async_midi_port_init(struct snd_fw_async_midi_port *port,
+ struct fw_unit *unit, u64 addr, unsigned int len,
+ snd_fw_async_midi_port_fill fill)
+{
+ port->len = DIV_ROUND_UP(len, 4) * 4;
+ port->buf = kzalloc(port->len, GFP_KERNEL);
+ if (port->buf == NULL)
+ return -ENOMEM;
+
+ port->parent = fw_parent_device(unit);
+ port->addr = addr;
+ port->fill = fill;
+ port->idling = true;
+ port->next_ktime = ktime_set(0, 0);
+ port->error = false;
+
+ INIT_WORK(&port->work, midi_port_work);
+
+ return 0;
+}
+EXPORT_SYMBOL(snd_fw_async_midi_port_init);
+
+/**
+ * snd_fw_async_midi_port_destroy - free asynchronous MIDI port structure
+ * @port: the asynchronous MIDI port structure
+ */
+void snd_fw_async_midi_port_destroy(struct snd_fw_async_midi_port *port)
+{
+ snd_fw_async_midi_port_finish(port);
+ cancel_work_sync(&port->work);
+ kfree(port->buf);
+}
+EXPORT_SYMBOL(snd_fw_async_midi_port_destroy);
+
MODULE_DESCRIPTION("FireWire audio helper functions");
MODULE_AUTHOR("Clemens Ladisch <clemens@ladisch.de>");
MODULE_LICENSE("GPL v2");
diff --git a/sound/firewire/lib.h b/sound/firewire/lib.h
index 02cfabc9c3c4..f3f6f84c48d6 100644
--- a/sound/firewire/lib.h
+++ b/sound/firewire/lib.h
@@ -3,6 +3,8 @@
#include <linux/firewire-constants.h>
#include <linux/types.h>
+#include <linux/sched.h>
+#include <sound/rawmidi.h>
struct fw_unit;
@@ -20,4 +22,58 @@ static inline bool rcode_is_permanent_error(int rcode)
return rcode == RCODE_TYPE_ERROR || rcode == RCODE_ADDRESS_ERROR;
}
+struct snd_fw_async_midi_port;
+typedef int (*snd_fw_async_midi_port_fill)(
+ struct snd_rawmidi_substream *substream,
+ u8 *buf);
+
+struct snd_fw_async_midi_port {
+ struct fw_device *parent;
+ struct work_struct work;
+ bool idling;
+ ktime_t next_ktime;
+ bool error;
+
+ u64 addr;
+ struct fw_transaction transaction;
+
+ u8 *buf;
+ unsigned int len;
+
+ struct snd_rawmidi_substream *substream;
+ snd_fw_async_midi_port_fill fill;
+ unsigned int consume_bytes;
+};
+
+int snd_fw_async_midi_port_init(struct snd_fw_async_midi_port *port,
+ struct fw_unit *unit, u64 addr, unsigned int len,
+ snd_fw_async_midi_port_fill fill);
+void snd_fw_async_midi_port_destroy(struct snd_fw_async_midi_port *port);
+
+/**
+ * snd_fw_async_midi_port_run - run transactions for the async MIDI port
+ * @port: the asynchronous MIDI port
+ * @substream: the MIDI substream
+ */
+static inline void
+snd_fw_async_midi_port_run(struct snd_fw_async_midi_port *port,
+ struct snd_rawmidi_substream *substream)
+{
+ if (!port->error) {
+ port->substream = substream;
+ schedule_work(&port->work);
+ }
+}
+
+/**
+ * snd_fw_async_midi_port_finish - finish the asynchronous MIDI port
+ * @port: the asynchronous MIDI port
+ */
+static inline void
+snd_fw_async_midi_port_finish(struct snd_fw_async_midi_port *port)
+{
+ port->substream = NULL;
+ port->error = false;
+}
+
#endif
diff --git a/sound/firewire/oxfw/Makefile b/sound/firewire/oxfw/Makefile
index a926850864f6..06ff50f4e6c0 100644
--- a/sound/firewire/oxfw/Makefile
+++ b/sound/firewire/oxfw/Makefile
@@ -1,3 +1,3 @@
snd-oxfw-objs := oxfw-command.o oxfw-stream.o oxfw-control.o oxfw-pcm.o \
oxfw-proc.o oxfw-midi.o oxfw-hwdep.o oxfw.o
-obj-m += snd-oxfw.o
+obj-$(CONFIG_SND_OXFW) += snd-oxfw.o
diff --git a/sound/firewire/oxfw/oxfw-midi.c b/sound/firewire/oxfw/oxfw-midi.c
index 540a30338516..8665e1043d41 100644
--- a/sound/firewire/oxfw/oxfw-midi.c
+++ b/sound/firewire/oxfw/oxfw-midi.c
@@ -90,11 +90,11 @@ static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up)
spin_lock_irqsave(&oxfw->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&oxfw->tx_stream,
- substrm->number, substrm);
+ amdtp_am824_midi_trigger(&oxfw->tx_stream,
+ substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&oxfw->tx_stream,
- substrm->number, NULL);
+ amdtp_am824_midi_trigger(&oxfw->tx_stream,
+ substrm->number, NULL);
spin_unlock_irqrestore(&oxfw->lock, flags);
}
@@ -107,11 +107,11 @@ static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
spin_lock_irqsave(&oxfw->lock, flags);
if (up)
- amdtp_stream_midi_trigger(&oxfw->rx_stream,
- substrm->number, substrm);
+ amdtp_am824_midi_trigger(&oxfw->rx_stream,
+ substrm->number, substrm);
else
- amdtp_stream_midi_trigger(&oxfw->rx_stream,
- substrm->number, NULL);
+ amdtp_am824_midi_trigger(&oxfw->rx_stream,
+ substrm->number, NULL);
spin_unlock_irqrestore(&oxfw->lock, flags);
}
@@ -142,29 +142,11 @@ static void set_midi_substream_names(struct snd_oxfw *oxfw,
int snd_oxfw_create_midi(struct snd_oxfw *oxfw)
{
- struct snd_oxfw_stream_formation formation;
struct snd_rawmidi *rmidi;
struct snd_rawmidi_str *str;
- u8 *format;
- int i, err;
-
- /* If its stream has MIDI conformant data channel, add one MIDI port */
- for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
- format = oxfw->tx_stream_formats[i];
- if (format != NULL) {
- err = snd_oxfw_stream_parse_format(format, &formation);
- if (err >= 0 && formation.midi > 0)
- oxfw->midi_input_ports = 1;
- }
-
- format = oxfw->rx_stream_formats[i];
- if (format != NULL) {
- err = snd_oxfw_stream_parse_format(format, &formation);
- if (err >= 0 && formation.midi > 0)
- oxfw->midi_output_ports = 1;
- }
- }
- if ((oxfw->midi_input_ports == 0) && (oxfw->midi_output_ports == 0))
+ int err;
+
+ if (oxfw->midi_input_ports == 0 && oxfw->midi_output_ports == 0)
return 0;
/* create midi ports */
diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c
index 9c73930d0278..8d233417695d 100644
--- a/sound/firewire/oxfw/oxfw-pcm.c
+++ b/sound/firewire/oxfw/oxfw-pcm.c
@@ -134,11 +134,11 @@ static int init_hw_params(struct snd_oxfw *oxfw,
SNDRV_PCM_INFO_MMAP_VALID;
if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
- runtime->hw.formats = AMDTP_IN_PCM_FORMAT_BITS;
+ runtime->hw.formats = AM824_IN_PCM_FORMAT_BITS;
stream = &oxfw->tx_stream;
formats = oxfw->tx_stream_formats;
} else {
- runtime->hw.formats = AMDTP_OUT_PCM_FORMAT_BITS;
+ runtime->hw.formats = AM824_OUT_PCM_FORMAT_BITS;
stream = &oxfw->rx_stream;
formats = oxfw->rx_stream_formats;
}
@@ -158,7 +158,7 @@ static int init_hw_params(struct snd_oxfw *oxfw,
if (err < 0)
goto end;
- err = amdtp_stream_add_pcm_hw_constraints(stream, runtime);
+ err = amdtp_am824_add_pcm_hw_constraints(stream, runtime);
end:
return err;
}
@@ -244,7 +244,7 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
mutex_unlock(&oxfw->mutex);
}
- amdtp_stream_set_pcm_format(&oxfw->tx_stream, params_format(hw_params));
+ amdtp_am824_set_pcm_format(&oxfw->tx_stream, params_format(hw_params));
return 0;
}
@@ -265,7 +265,7 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
mutex_unlock(&oxfw->mutex);
}
- amdtp_stream_set_pcm_format(&oxfw->rx_stream, params_format(hw_params));
+ amdtp_am824_set_pcm_format(&oxfw->rx_stream, params_format(hw_params));
return 0;
}
diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index 77ad5b98e806..7cb5743c073b 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -148,14 +148,17 @@ static int start_stream(struct snd_oxfw *oxfw, struct amdtp_stream *stream,
}
pcm_channels = formation.pcm;
- midi_ports = DIV_ROUND_UP(formation.midi, 8);
+ midi_ports = formation.midi * 8;
/* The stream should have one pcm channels at least */
if (pcm_channels == 0) {
err = -EINVAL;
goto end;
}
- amdtp_stream_set_parameters(stream, rate, pcm_channels, midi_ports);
+ err = amdtp_am824_set_parameters(stream, rate, pcm_channels, midi_ports,
+ false);
+ if (err < 0)
+ goto end;
err = cmp_connection_establish(conn,
amdtp_stream_get_max_payload(stream));
@@ -225,7 +228,7 @@ int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw,
if (err < 0)
goto end;
- err = amdtp_stream_init(stream, oxfw->unit, s_dir, CIP_NONBLOCKING);
+ err = amdtp_am824_init(stream, oxfw->unit, s_dir, CIP_NONBLOCKING);
if (err < 0) {
amdtp_stream_destroy(stream);
cmp_connection_destroy(conn);
@@ -238,9 +241,12 @@ int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw,
* packets. As a result, next isochronous packet includes more data
* blocks than IEC 61883-6 defines.
*/
- if (stream == &oxfw->tx_stream)
+ if (stream == &oxfw->tx_stream) {
oxfw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK |
CIP_JUMBO_PAYLOAD;
+ if (oxfw->wrong_dbs)
+ oxfw->tx_stream.flags |= CIP_WRONG_DBS;
+ }
end:
return err;
}
@@ -480,8 +486,8 @@ int snd_oxfw_stream_parse_format(u8 *format,
}
}
- if (formation->pcm > AMDTP_MAX_CHANNELS_FOR_PCM ||
- formation->midi > AMDTP_MAX_CHANNELS_FOR_MIDI)
+ if (formation->pcm > AM824_MAX_CHANNELS_FOR_PCM ||
+ formation->midi > AM824_MAX_CHANNELS_FOR_MIDI)
return -ENOSYS;
return 0;
@@ -623,6 +629,9 @@ end:
int snd_oxfw_stream_discover(struct snd_oxfw *oxfw)
{
u8 plugs[AVC_PLUG_INFO_BUF_BYTES];
+ struct snd_oxfw_stream_formation formation;
+ u8 *format;
+ unsigned int i;
int err;
/* the number of plugs for isoc in/out, ext in/out */
@@ -642,12 +651,42 @@ int snd_oxfw_stream_discover(struct snd_oxfw *oxfw)
err = fill_stream_formats(oxfw, AVC_GENERAL_PLUG_DIR_OUT, 0);
if (err < 0)
goto end;
+
+ for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
+ format = oxfw->tx_stream_formats[i];
+ if (format == NULL)
+ continue;
+ err = snd_oxfw_stream_parse_format(format, &formation);
+ if (err < 0)
+ continue;
+
+ /* Add one MIDI port. */
+ if (formation.midi > 0)
+ oxfw->midi_input_ports = 1;
+ }
+
oxfw->has_output = true;
}
/* use iPCR[0] if exists */
- if (plugs[0] > 0)
+ if (plugs[0] > 0) {
err = fill_stream_formats(oxfw, AVC_GENERAL_PLUG_DIR_IN, 0);
+ if (err < 0)
+ goto end;
+
+ for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
+ format = oxfw->rx_stream_formats[i];
+ if (format == NULL)
+ continue;
+ err = snd_oxfw_stream_parse_format(format, &formation);
+ if (err < 0)
+ continue;
+
+ /* Add one MIDI port. */
+ if (formation.midi > 0)
+ oxfw->midi_output_ports = 1;
+ }
+ }
end:
return err;
}
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index 8c6ce019f437..588b93f20c2e 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -18,6 +18,9 @@
#define VENDOR_GRIFFIN 0x001292
#define VENDOR_BEHRINGER 0x001564
#define VENDOR_LACIE 0x00d04b
+#define VENDOR_TASCAM 0x00022e
+
+#define MODEL_SATELLITE 0x00200f
#define SPECIFIER_1394TA 0x00a02d
#define VERSION_AVC 0x010001
@@ -129,6 +132,40 @@ static void oxfw_card_free(struct snd_card *card)
mutex_destroy(&oxfw->mutex);
}
+static void detect_quirks(struct snd_oxfw *oxfw)
+{
+ struct fw_device *fw_dev = fw_parent_device(oxfw->unit);
+ struct fw_csr_iterator it;
+ int key, val;
+ int vendor, model;
+
+ /* Seek from Root Directory of Config ROM. */
+ vendor = model = 0;
+ fw_csr_iterator_init(&it, fw_dev->config_rom + 5);
+ while (fw_csr_iterator_next(&it, &key, &val)) {
+ if (key == CSR_VENDOR)
+ vendor = val;
+ else if (key == CSR_MODEL)
+ model = val;
+ }
+
+ /*
+ * Mackie Onyx Satellite with base station has a quirk to report a wrong
+ * value in 'dbs' field of CIP header against its format information.
+ */
+ if (vendor == VENDOR_LOUD && model == MODEL_SATELLITE)
+ oxfw->wrong_dbs = true;
+
+ /*
+ * TASCAM FireOne has physical control and requires a pair of additional
+ * MIDI ports.
+ */
+ if (vendor == VENDOR_TASCAM) {
+ oxfw->midi_input_ports++;
+ oxfw->midi_output_ports++;
+ }
+}
+
static int oxfw_probe(struct fw_unit *unit,
const struct ieee1394_device_id *id)
{
@@ -157,6 +194,8 @@ static int oxfw_probe(struct fw_unit *unit,
if (err < 0)
goto error;
+ detect_quirks(oxfw);
+
err = name_card(oxfw);
if (err < 0)
goto error;
@@ -294,6 +333,13 @@ static const struct ieee1394_device_id oxfw_id_table[] = {
.specifier_id = SPECIFIER_1394TA,
.version = VERSION_AVC,
},
+ /* TASCAM, FireOne */
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_MODEL_ID,
+ .vendor_id = VENDOR_TASCAM,
+ .model_id = 0x800007,
+ },
{ }
};
MODULE_DEVICE_TABLE(ieee1394, oxfw_id_table);
diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h
index cace5ad4fe76..8392c424ad1d 100644
--- a/sound/firewire/oxfw/oxfw.h
+++ b/sound/firewire/oxfw/oxfw.h
@@ -28,7 +28,7 @@
#include "../fcp.h"
#include "../packets-buffer.h"
#include "../iso-resources.h"
-#include "../amdtp.h"
+#include "../amdtp-am824.h"
#include "../cmp.h"
struct device_info {
@@ -49,6 +49,7 @@ struct snd_oxfw {
struct mutex mutex;
spinlock_t lock;
+ bool wrong_dbs;
bool has_output;
u8 *tx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES];
u8 *rx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES];
diff --git a/sound/firewire/tascam/Makefile b/sound/firewire/tascam/Makefile
new file mode 100644
index 000000000000..0fc955d5bd15
--- /dev/null
+++ b/sound/firewire/tascam/Makefile
@@ -0,0 +1,4 @@
+snd-firewire-tascam-objs := tascam-proc.o amdtp-tascam.o tascam-stream.o \
+ tascam-pcm.o tascam-hwdep.o tascam-transaction.o \
+ tascam-midi.o tascam.o
+obj-$(CONFIG_SND_FIREWIRE_TASCAM) += snd-firewire-tascam.o
diff --git a/sound/firewire/tascam/amdtp-tascam.c b/sound/firewire/tascam/amdtp-tascam.c
new file mode 100644
index 000000000000..9dd0fccd5ccc
--- /dev/null
+++ b/sound/firewire/tascam/amdtp-tascam.c
@@ -0,0 +1,243 @@
+/*
+ * amdtp-tascam.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <sound/pcm.h>
+#include "tascam.h"
+
+#define AMDTP_FMT_TSCM_TX 0x1e
+#define AMDTP_FMT_TSCM_RX 0x3e
+
+struct amdtp_tscm {
+ unsigned int pcm_channels;
+
+ void (*transfer_samples)(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames);
+};
+
+int amdtp_tscm_set_parameters(struct amdtp_stream *s, unsigned int rate)
+{
+ struct amdtp_tscm *p = s->protocol;
+ unsigned int data_channels;
+
+ if (amdtp_stream_running(s))
+ return -EBUSY;
+
+ data_channels = p->pcm_channels;
+
+ /* Packets in in-stream have extra 2 data channels. */
+ if (s->direction == AMDTP_IN_STREAM)
+ data_channels += 2;
+
+ return amdtp_stream_set_parameters(s, rate, data_channels);
+}
+
+static void write_pcm_s32(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_tscm *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ const u32 *src;
+
+ channels = p->pcm_channels;
+ src = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ buffer[c] = cpu_to_be32(*src);
+ src++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void write_pcm_s16(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_tscm *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ const u16 *src;
+
+ channels = p->pcm_channels;
+ src = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ buffer[c] = cpu_to_be32(*src << 16);
+ src++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ src = (void *)runtime->dma_area;
+ }
+}
+
+static void read_pcm_s32(struct amdtp_stream *s,
+ struct snd_pcm_substream *pcm,
+ __be32 *buffer, unsigned int frames)
+{
+ struct amdtp_tscm *p = s->protocol;
+ struct snd_pcm_runtime *runtime = pcm->runtime;
+ unsigned int channels, remaining_frames, i, c;
+ u32 *dst;
+
+ channels = p->pcm_channels;
+ dst = (void *)runtime->dma_area +
+ frames_to_bytes(runtime, s->pcm_buffer_pointer);
+ remaining_frames = runtime->buffer_size - s->pcm_buffer_pointer;
+
+ /* The first data channel is for event counter. */
+ buffer += 1;
+
+ for (i = 0; i < frames; ++i) {
+ for (c = 0; c < channels; ++c) {
+ *dst = be32_to_cpu(buffer[c]);
+ dst++;
+ }
+ buffer += s->data_block_quadlets;
+ if (--remaining_frames == 0)
+ dst = (void *)runtime->dma_area;
+ }
+}
+
+static void write_pcm_silence(struct amdtp_stream *s, __be32 *buffer,
+ unsigned int data_blocks)
+{
+ struct amdtp_tscm *p = s->protocol;
+ unsigned int channels, i, c;
+
+ channels = p->pcm_channels;
+
+ for (i = 0; i < data_blocks; ++i) {
+ for (c = 0; c < channels; ++c)
+ buffer[c] = 0x00000000;
+ buffer += s->data_block_quadlets;
+ }
+}
+
+int amdtp_tscm_add_pcm_hw_constraints(struct amdtp_stream *s,
+ struct snd_pcm_runtime *runtime)
+{
+ int err;
+
+ /*
+ * Our implementation allows this protocol to deliver 24 bit sample in
+ * 32bit data channel.
+ */
+ err = snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24);
+ if (err < 0)
+ return err;
+
+ return amdtp_stream_add_pcm_hw_constraints(s, runtime);
+}
+
+void amdtp_tscm_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format)
+{
+ struct amdtp_tscm *p = s->protocol;
+
+ if (WARN_ON(amdtp_stream_pcm_running(s)))
+ return;
+
+ switch (format) {
+ default:
+ WARN_ON(1);
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S16:
+ if (s->direction == AMDTP_OUT_STREAM) {
+ p->transfer_samples = write_pcm_s16;
+ break;
+ }
+ WARN_ON(1);
+ /* fall through */
+ case SNDRV_PCM_FORMAT_S32:
+ if (s->direction == AMDTP_OUT_STREAM)
+ p->transfer_samples = write_pcm_s32;
+ else
+ p->transfer_samples = read_pcm_s32;
+ break;
+ }
+}
+
+static unsigned int process_tx_data_blocks(struct amdtp_stream *s,
+ __be32 *buffer,
+ unsigned int data_blocks,
+ unsigned int *syt)
+{
+ struct amdtp_tscm *p = (struct amdtp_tscm *)s->protocol;
+ struct snd_pcm_substream *pcm;
+
+ pcm = ACCESS_ONCE(s->pcm);
+ if (data_blocks > 0 && pcm)
+ p->transfer_samples(s, pcm, buffer, data_blocks);
+
+ /* A place holder for control messages. */
+
+ return data_blocks;
+}
+
+static unsigned int process_rx_data_blocks(struct amdtp_stream *s,
+ __be32 *buffer,
+ unsigned int data_blocks,
+ unsigned int *syt)
+{
+ struct amdtp_tscm *p = (struct amdtp_tscm *)s->protocol;
+ struct snd_pcm_substream *pcm;
+
+ /* This field is not used. */
+ *syt = 0x0000;
+
+ pcm = ACCESS_ONCE(s->pcm);
+ if (pcm)
+ p->transfer_samples(s, pcm, buffer, data_blocks);
+ else
+ write_pcm_silence(s, buffer, data_blocks);
+
+ return data_blocks;
+}
+
+int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit,
+ enum amdtp_stream_direction dir, unsigned int pcm_channels)
+{
+ amdtp_stream_process_data_blocks_t process_data_blocks;
+ struct amdtp_tscm *p;
+ unsigned int fmt;
+ int err;
+
+ if (dir == AMDTP_IN_STREAM) {
+ fmt = AMDTP_FMT_TSCM_TX;
+ process_data_blocks = process_tx_data_blocks;
+ } else {
+ fmt = AMDTP_FMT_TSCM_RX;
+ process_data_blocks = process_rx_data_blocks;
+ }
+
+ err = amdtp_stream_init(s, unit, dir,
+ CIP_NONBLOCKING | CIP_SKIP_DBC_ZERO_CHECK, fmt,
+ process_data_blocks, sizeof(struct amdtp_tscm));
+ if (err < 0)
+ return 0;
+
+ /* Use fixed value for FDF field. */
+ s->fdf = 0x00;
+
+ /* This protocol uses fixed number of data channels for PCM samples. */
+ p = s->protocol;
+ p->pcm_channels = pcm_channels;
+
+ return 0;
+}
diff --git a/sound/firewire/tascam/tascam-hwdep.c b/sound/firewire/tascam/tascam-hwdep.c
new file mode 100644
index 000000000000..131267c3a042
--- /dev/null
+++ b/sound/firewire/tascam/tascam-hwdep.c
@@ -0,0 +1,201 @@
+/*
+ * tascam-hwdep.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+/*
+ * This codes give three functionality.
+ *
+ * 1.get firewire node information
+ * 2.get notification about starting/stopping stream
+ * 3.lock/unlock stream
+ */
+
+#include "tascam.h"
+
+static long hwdep_read_locked(struct snd_tscm *tscm, char __user *buf,
+ long count)
+{
+ union snd_firewire_event event;
+
+ memset(&event, 0, sizeof(event));
+
+ event.lock_status.type = SNDRV_FIREWIRE_EVENT_LOCK_STATUS;
+ event.lock_status.status = (tscm->dev_lock_count > 0);
+ tscm->dev_lock_changed = false;
+
+ count = min_t(long, count, sizeof(event.lock_status));
+
+ if (copy_to_user(buf, &event, count))
+ return -EFAULT;
+
+ return count;
+}
+
+static long hwdep_read(struct snd_hwdep *hwdep, char __user *buf, long count,
+ loff_t *offset)
+{
+ struct snd_tscm *tscm = hwdep->private_data;
+ DEFINE_WAIT(wait);
+ union snd_firewire_event event;
+
+ spin_lock_irq(&tscm->lock);
+
+ while (!tscm->dev_lock_changed) {
+ prepare_to_wait(&tscm->hwdep_wait, &wait, TASK_INTERRUPTIBLE);
+ spin_unlock_irq(&tscm->lock);
+ schedule();
+ finish_wait(&tscm->hwdep_wait, &wait);
+ if (signal_pending(current))
+ return -ERESTARTSYS;
+ spin_lock_irq(&tscm->lock);
+ }
+
+ memset(&event, 0, sizeof(event));
+ count = hwdep_read_locked(tscm, buf, count);
+ spin_unlock_irq(&tscm->lock);
+
+ return count;
+}
+
+static unsigned int hwdep_poll(struct snd_hwdep *hwdep, struct file *file,
+ poll_table *wait)
+{
+ struct snd_tscm *tscm = hwdep->private_data;
+ unsigned int events;
+
+ poll_wait(file, &tscm->hwdep_wait, wait);
+
+ spin_lock_irq(&tscm->lock);
+ if (tscm->dev_lock_changed)
+ events = POLLIN | POLLRDNORM;
+ else
+ events = 0;
+ spin_unlock_irq(&tscm->lock);
+
+ return events;
+}
+
+static int hwdep_get_info(struct snd_tscm *tscm, void __user *arg)
+{
+ struct fw_device *dev = fw_parent_device(tscm->unit);
+ struct snd_firewire_get_info info;
+
+ memset(&info, 0, sizeof(info));
+ info.type = SNDRV_FIREWIRE_TYPE_TASCAM;
+ info.card = dev->card->index;
+ *(__be32 *)&info.guid[0] = cpu_to_be32(dev->config_rom[3]);
+ *(__be32 *)&info.guid[4] = cpu_to_be32(dev->config_rom[4]);
+ strlcpy(info.device_name, dev_name(&dev->device),
+ sizeof(info.device_name));
+
+ if (copy_to_user(arg, &info, sizeof(info)))
+ return -EFAULT;
+
+ return 0;
+}
+
+static int hwdep_lock(struct snd_tscm *tscm)
+{
+ int err;
+
+ spin_lock_irq(&tscm->lock);
+
+ if (tscm->dev_lock_count == 0) {
+ tscm->dev_lock_count = -1;
+ err = 0;
+ } else {
+ err = -EBUSY;
+ }
+
+ spin_unlock_irq(&tscm->lock);
+
+ return err;
+}
+
+static int hwdep_unlock(struct snd_tscm *tscm)
+{
+ int err;
+
+ spin_lock_irq(&tscm->lock);
+
+ if (tscm->dev_lock_count == -1) {
+ tscm->dev_lock_count = 0;
+ err = 0;
+ } else {
+ err = -EBADFD;
+ }
+
+ spin_unlock_irq(&tscm->lock);
+
+ return err;
+}
+
+static int hwdep_release(struct snd_hwdep *hwdep, struct file *file)
+{
+ struct snd_tscm *tscm = hwdep->private_data;
+
+ spin_lock_irq(&tscm->lock);
+ if (tscm->dev_lock_count == -1)
+ tscm->dev_lock_count = 0;
+ spin_unlock_irq(&tscm->lock);
+
+ return 0;
+}
+
+static int hwdep_ioctl(struct snd_hwdep *hwdep, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ struct snd_tscm *tscm = hwdep->private_data;
+
+ switch (cmd) {
+ case SNDRV_FIREWIRE_IOCTL_GET_INFO:
+ return hwdep_get_info(tscm, (void __user *)arg);
+ case SNDRV_FIREWIRE_IOCTL_LOCK:
+ return hwdep_lock(tscm);
+ case SNDRV_FIREWIRE_IOCTL_UNLOCK:
+ return hwdep_unlock(tscm);
+ default:
+ return -ENOIOCTLCMD;
+ }
+}
+
+#ifdef CONFIG_COMPAT
+static int hwdep_compat_ioctl(struct snd_hwdep *hwdep, struct file *file,
+ unsigned int cmd, unsigned long arg)
+{
+ return hwdep_ioctl(hwdep, file, cmd,
+ (unsigned long)compat_ptr(arg));
+}
+#else
+#define hwdep_compat_ioctl NULL
+#endif
+
+static const struct snd_hwdep_ops hwdep_ops = {
+ .read = hwdep_read,
+ .release = hwdep_release,
+ .poll = hwdep_poll,
+ .ioctl = hwdep_ioctl,
+ .ioctl_compat = hwdep_compat_ioctl,
+};
+
+int snd_tscm_create_hwdep_device(struct snd_tscm *tscm)
+{
+ struct snd_hwdep *hwdep;
+ int err;
+
+ err = snd_hwdep_new(tscm->card, "Tascam", 0, &hwdep);
+ if (err < 0)
+ return err;
+
+ strcpy(hwdep->name, "Tascam");
+ hwdep->iface = SNDRV_HWDEP_IFACE_FW_TASCAM;
+ hwdep->ops = hwdep_ops;
+ hwdep->private_data = tscm;
+ hwdep->exclusive = true;
+
+ return err;
+}
diff --git a/sound/firewire/tascam/tascam-midi.c b/sound/firewire/tascam/tascam-midi.c
new file mode 100644
index 000000000000..41f842079d9d
--- /dev/null
+++ b/sound/firewire/tascam/tascam-midi.c
@@ -0,0 +1,135 @@
+/*
+ * tascam-midi.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "tascam.h"
+
+static int midi_capture_open(struct snd_rawmidi_substream *substream)
+{
+ /* Do nothing. */
+ return 0;
+}
+
+static int midi_playback_open(struct snd_rawmidi_substream *substream)
+{
+ struct snd_tscm *tscm = substream->rmidi->private_data;
+
+ /* Initialize internal status. */
+ tscm->running_status[substream->number] = 0;
+ tscm->on_sysex[substream->number] = 0;
+ return 0;
+}
+
+static int midi_capture_close(struct snd_rawmidi_substream *substream)
+{
+ /* Do nothing. */
+ return 0;
+}
+
+static int midi_playback_close(struct snd_rawmidi_substream *substream)
+{
+ struct snd_tscm *tscm = substream->rmidi->private_data;
+
+ snd_fw_async_midi_port_finish(&tscm->out_ports[substream->number]);
+
+ return 0;
+}
+
+static void midi_capture_trigger(struct snd_rawmidi_substream *substrm, int up)
+{
+ struct snd_tscm *tscm = substrm->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&tscm->lock, flags);
+
+ if (up)
+ tscm->tx_midi_substreams[substrm->number] = substrm;
+ else
+ tscm->tx_midi_substreams[substrm->number] = NULL;
+
+ spin_unlock_irqrestore(&tscm->lock, flags);
+}
+
+static void midi_playback_trigger(struct snd_rawmidi_substream *substrm, int up)
+{
+ struct snd_tscm *tscm = substrm->rmidi->private_data;
+ unsigned long flags;
+
+ spin_lock_irqsave(&tscm->lock, flags);
+
+ if (up)
+ snd_fw_async_midi_port_run(&tscm->out_ports[substrm->number],
+ substrm);
+
+ spin_unlock_irqrestore(&tscm->lock, flags);
+}
+
+static struct snd_rawmidi_ops midi_capture_ops = {
+ .open = midi_capture_open,
+ .close = midi_capture_close,
+ .trigger = midi_capture_trigger,
+};
+
+static struct snd_rawmidi_ops midi_playback_ops = {
+ .open = midi_playback_open,
+ .close = midi_playback_close,
+ .trigger = midi_playback_trigger,
+};
+
+int snd_tscm_create_midi_devices(struct snd_tscm *tscm)
+{
+ struct snd_rawmidi *rmidi;
+ struct snd_rawmidi_str *stream;
+ struct snd_rawmidi_substream *subs;
+ int err;
+
+ err = snd_rawmidi_new(tscm->card, tscm->card->driver, 0,
+ tscm->spec->midi_playback_ports,
+ tscm->spec->midi_capture_ports,
+ &rmidi);
+ if (err < 0)
+ return err;
+
+ snprintf(rmidi->name, sizeof(rmidi->name),
+ "%s MIDI", tscm->card->shortname);
+ rmidi->private_data = tscm;
+
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT;
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT,
+ &midi_capture_ops);
+ stream = &rmidi->streams[SNDRV_RAWMIDI_STREAM_INPUT];
+
+ /* Set port names for MIDI input. */
+ list_for_each_entry(subs, &stream->substreams, list) {
+ /* TODO: support virtual MIDI ports. */
+ if (subs->number < tscm->spec->midi_capture_ports) {
+ /* Hardware MIDI ports. */
+ snprintf(subs->name, sizeof(subs->name),
+ "%s MIDI %d",
+ tscm->card->shortname, subs->number + 1);
+ }
+ }
+
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT;
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT,
+ &midi_playback_ops);
+ stream = &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT];
+
+ /* Set port names for MIDI ourput. */
+ list_for_each_entry(subs, &stream->substreams, list) {
+ if (subs->number < tscm->spec->midi_playback_ports) {
+ /* Hardware MIDI ports only. */
+ snprintf(subs->name, sizeof(subs->name),
+ "%s MIDI %d",
+ tscm->card->shortname, subs->number + 1);
+ }
+ }
+
+ rmidi->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX;
+
+ return 0;
+}
diff --git a/sound/firewire/tascam/tascam-pcm.c b/sound/firewire/tascam/tascam-pcm.c
new file mode 100644
index 000000000000..380d3db969a5
--- /dev/null
+++ b/sound/firewire/tascam/tascam-pcm.c
@@ -0,0 +1,312 @@
+/*
+ * tascam-pcm.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "tascam.h"
+
+static void set_buffer_params(struct snd_pcm_hardware *hw)
+{
+ hw->period_bytes_min = 4 * hw->channels_min;
+ hw->period_bytes_max = hw->period_bytes_min * 2048;
+ hw->buffer_bytes_max = hw->period_bytes_max * 2;
+
+ hw->periods_min = 2;
+ hw->periods_max = UINT_MAX;
+}
+
+static int pcm_init_hw_params(struct snd_tscm *tscm,
+ struct snd_pcm_substream *substream)
+{
+ static const struct snd_pcm_hardware hardware = {
+ .info = SNDRV_PCM_INFO_BATCH |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_JOINT_DUPLEX |
+ SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID,
+ .rates = SNDRV_PCM_RATE_44100 |
+ SNDRV_PCM_RATE_48000 |
+ SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000,
+ .rate_min = 44100,
+ .rate_max = 96000,
+ .channels_min = 10,
+ .channels_max = 18,
+ };
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct amdtp_stream *stream;
+ unsigned int pcm_channels;
+
+ runtime->hw = hardware;
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ runtime->hw.formats = SNDRV_PCM_FMTBIT_S32;
+ stream = &tscm->tx_stream;
+ pcm_channels = tscm->spec->pcm_capture_analog_channels;
+ } else {
+ runtime->hw.formats =
+ SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_S32;
+ stream = &tscm->rx_stream;
+ pcm_channels = tscm->spec->pcm_playback_analog_channels;
+ }
+
+ if (tscm->spec->has_adat)
+ pcm_channels += 8;
+ if (tscm->spec->has_spdif)
+ pcm_channels += 2;
+ runtime->hw.channels_min = runtime->hw.channels_max = pcm_channels;
+
+ set_buffer_params(&runtime->hw);
+
+ return amdtp_tscm_add_pcm_hw_constraints(stream, runtime);
+}
+
+static int pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_tscm *tscm = substream->private_data;
+ enum snd_tscm_clock clock;
+ unsigned int rate;
+ int err;
+
+ err = snd_tscm_stream_lock_try(tscm);
+ if (err < 0)
+ goto end;
+
+ err = pcm_init_hw_params(tscm, substream);
+ if (err < 0)
+ goto err_locked;
+
+ err = snd_tscm_stream_get_clock(tscm, &clock);
+ if (clock != SND_TSCM_CLOCK_INTERNAL ||
+ amdtp_stream_pcm_running(&tscm->rx_stream) ||
+ amdtp_stream_pcm_running(&tscm->tx_stream)) {
+ err = snd_tscm_stream_get_rate(tscm, &rate);
+ if (err < 0)
+ goto err_locked;
+ substream->runtime->hw.rate_min = rate;
+ substream->runtime->hw.rate_max = rate;
+ }
+
+ snd_pcm_set_sync(substream);
+end:
+ return err;
+err_locked:
+ snd_tscm_stream_lock_release(tscm);
+ return err;
+}
+
+static int pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_tscm *tscm = substream->private_data;
+
+ snd_tscm_stream_lock_release(tscm);
+
+ return 0;
+}
+
+static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_tscm *tscm = substream->private_data;
+ int err;
+
+ err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+
+ if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+ mutex_lock(&tscm->mutex);
+ tscm->substreams_counter++;
+ mutex_unlock(&tscm->mutex);
+ }
+
+ amdtp_tscm_set_pcm_format(&tscm->tx_stream, params_format(hw_params));
+
+ return 0;
+}
+
+static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ struct snd_tscm *tscm = substream->private_data;
+ int err;
+
+ err = snd_pcm_lib_alloc_vmalloc_buffer(substream,
+ params_buffer_bytes(hw_params));
+ if (err < 0)
+ return err;
+
+ if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
+ mutex_lock(&tscm->mutex);
+ tscm->substreams_counter++;
+ mutex_unlock(&tscm->mutex);
+ }
+
+ amdtp_tscm_set_pcm_format(&tscm->rx_stream, params_format(hw_params));
+
+ return 0;
+}
+
+static int pcm_capture_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_tscm *tscm = substream->private_data;
+
+ mutex_lock(&tscm->mutex);
+
+ if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
+ tscm->substreams_counter--;
+
+ snd_tscm_stream_stop_duplex(tscm);
+
+ mutex_unlock(&tscm->mutex);
+
+ return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static int pcm_playback_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_tscm *tscm = substream->private_data;
+
+ mutex_lock(&tscm->mutex);
+
+ if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN)
+ tscm->substreams_counter--;
+
+ snd_tscm_stream_stop_duplex(tscm);
+
+ mutex_unlock(&tscm->mutex);
+
+ return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+static int pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_tscm *tscm = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ mutex_lock(&tscm->mutex);
+
+ err = snd_tscm_stream_start_duplex(tscm, runtime->rate);
+ if (err >= 0)
+ amdtp_stream_pcm_prepare(&tscm->tx_stream);
+
+ mutex_unlock(&tscm->mutex);
+
+ return err;
+}
+
+static int pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_tscm *tscm = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int err;
+
+ mutex_lock(&tscm->mutex);
+
+ err = snd_tscm_stream_start_duplex(tscm, runtime->rate);
+ if (err >= 0)
+ amdtp_stream_pcm_prepare(&tscm->rx_stream);
+
+ mutex_unlock(&tscm->mutex);
+
+ return err;
+}
+
+static int pcm_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_tscm *tscm = substream->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ amdtp_stream_pcm_trigger(&tscm->tx_stream, substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ amdtp_stream_pcm_trigger(&tscm->tx_stream, NULL);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int pcm_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ struct snd_tscm *tscm = substream->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ amdtp_stream_pcm_trigger(&tscm->rx_stream, substream);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ amdtp_stream_pcm_trigger(&tscm->rx_stream, NULL);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static snd_pcm_uframes_t pcm_capture_pointer(struct snd_pcm_substream *sbstrm)
+{
+ struct snd_tscm *tscm = sbstrm->private_data;
+
+ return amdtp_stream_pcm_pointer(&tscm->tx_stream);
+}
+
+static snd_pcm_uframes_t pcm_playback_pointer(struct snd_pcm_substream *sbstrm)
+{
+ struct snd_tscm *tscm = sbstrm->private_data;
+
+ return amdtp_stream_pcm_pointer(&tscm->rx_stream);
+}
+
+static struct snd_pcm_ops pcm_capture_ops = {
+ .open = pcm_open,
+ .close = pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pcm_capture_hw_params,
+ .hw_free = pcm_capture_hw_free,
+ .prepare = pcm_capture_prepare,
+ .trigger = pcm_capture_trigger,
+ .pointer = pcm_capture_pointer,
+ .page = snd_pcm_lib_get_vmalloc_page,
+};
+
+static struct snd_pcm_ops pcm_playback_ops = {
+ .open = pcm_open,
+ .close = pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = pcm_playback_hw_params,
+ .hw_free = pcm_playback_hw_free,
+ .prepare = pcm_playback_prepare,
+ .trigger = pcm_playback_trigger,
+ .pointer = pcm_playback_pointer,
+ .page = snd_pcm_lib_get_vmalloc_page,
+ .mmap = snd_pcm_lib_mmap_vmalloc,
+};
+
+int snd_tscm_create_pcm_devices(struct snd_tscm *tscm)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ err = snd_pcm_new(tscm->card, tscm->card->driver, 0, 1, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ pcm->private_data = tscm;
+ snprintf(pcm->name, sizeof(pcm->name),
+ "%s PCM", tscm->card->shortname);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &pcm_capture_ops);
+
+ return 0;
+}
diff --git a/sound/firewire/tascam/tascam-proc.c b/sound/firewire/tascam/tascam-proc.c
new file mode 100644
index 000000000000..bfd4a4c06914
--- /dev/null
+++ b/sound/firewire/tascam/tascam-proc.c
@@ -0,0 +1,88 @@
+/*
+ * tascam-proc.h - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "./tascam.h"
+
+static void proc_read_firmware(struct snd_info_entry *entry,
+ struct snd_info_buffer *buffer)
+{
+ struct snd_tscm *tscm = entry->private_data;
+ __be32 data;
+ unsigned int reg, fpga, arm, hw;
+ int err;
+
+ err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_REGISTER,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ return;
+ reg = be32_to_cpu(data);
+
+ err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_FPGA,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ return;
+ fpga = be32_to_cpu(data);
+
+ err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_ARM,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ return;
+ arm = be32_to_cpu(data);
+
+ err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_FIRMWARE_HW,
+ &data, sizeof(data), 0);
+ if (err < 0)
+ return;
+ hw = be32_to_cpu(data);
+
+ snd_iprintf(buffer, "Register: %d (0x%08x)\n", reg & 0xffff, reg);
+ snd_iprintf(buffer, "FPGA: %d (0x%08x)\n", fpga & 0xffff, fpga);
+ snd_iprintf(buffer, "ARM: %d (0x%08x)\n", arm & 0xffff, arm);
+ snd_iprintf(buffer, "Hardware: %d (0x%08x)\n", hw >> 16, hw);
+}
+
+static void add_node(struct snd_tscm *tscm, struct snd_info_entry *root,
+ const char *name,
+ void (*op)(struct snd_info_entry *e,
+ struct snd_info_buffer *b))
+{
+ struct snd_info_entry *entry;
+
+ entry = snd_info_create_card_entry(tscm->card, name, root);
+ if (entry == NULL)
+ return;
+
+ snd_info_set_text_ops(entry, tscm, op);
+ if (snd_info_register(entry) < 0)
+ snd_info_free_entry(entry);
+}
+
+void snd_tscm_proc_init(struct snd_tscm *tscm)
+{
+ struct snd_info_entry *root;
+
+ /*
+ * All nodes are automatically removed at snd_card_disconnect(),
+ * by following to link list.
+ */
+ root = snd_info_create_card_entry(tscm->card, "firewire",
+ tscm->card->proc_root);
+ if (root == NULL)
+ return;
+ root->mode = S_IFDIR | S_IRUGO | S_IXUGO;
+ if (snd_info_register(root) < 0) {
+ snd_info_free_entry(root);
+ return;
+ }
+
+ add_node(tscm, root, "firmware", proc_read_firmware);
+}
diff --git a/sound/firewire/tascam/tascam-stream.c b/sound/firewire/tascam/tascam-stream.c
new file mode 100644
index 000000000000..0e6dd5c61f53
--- /dev/null
+++ b/sound/firewire/tascam/tascam-stream.c
@@ -0,0 +1,496 @@
+/*
+ * tascam-stream.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include <linux/delay.h>
+#include "tascam.h"
+
+#define CALLBACK_TIMEOUT 500
+
+static int get_clock(struct snd_tscm *tscm, u32 *data)
+{
+ __be32 reg;
+ int err;
+
+ err = snd_fw_transaction(tscm->unit, TCODE_READ_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS,
+ &reg, sizeof(reg), 0);
+ if (err >= 0)
+ *data = be32_to_cpu(reg);
+
+ return err;
+}
+
+static int set_clock(struct snd_tscm *tscm, unsigned int rate,
+ enum snd_tscm_clock clock)
+{
+ u32 data;
+ __be32 reg;
+ int err;
+
+ err = get_clock(tscm, &data);
+ if (err < 0)
+ return err;
+ data &= 0x0000ffff;
+
+ if (rate > 0) {
+ data &= 0x000000ff;
+ /* Base rate. */
+ if ((rate % 44100) == 0) {
+ data |= 0x00000100;
+ /* Multiplier. */
+ if (rate / 44100 == 2)
+ data |= 0x00008000;
+ } else if ((rate % 48000) == 0) {
+ data |= 0x00000200;
+ /* Multiplier. */
+ if (rate / 48000 == 2)
+ data |= 0x00008000;
+ } else {
+ return -EAGAIN;
+ }
+ }
+
+ if (clock != INT_MAX) {
+ data &= 0x0000ff00;
+ data |= clock + 1;
+ }
+
+ reg = cpu_to_be32(data);
+
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_CLOCK_STATUS,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ if (data & 0x00008000)
+ reg = cpu_to_be32(0x0000001a);
+ else
+ reg = cpu_to_be32(0x0000000d);
+
+ return snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MULTIPLEX_MODE,
+ &reg, sizeof(reg), 0);
+}
+
+int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate)
+{
+ u32 data = 0x0;
+ unsigned int trials = 0;
+ int err;
+
+ while (data == 0x0 || trials++ < 5) {
+ err = get_clock(tscm, &data);
+ if (err < 0)
+ return err;
+
+ data = (data & 0xff000000) >> 24;
+ }
+
+ /* Check base rate. */
+ if ((data & 0x0f) == 0x01)
+ *rate = 44100;
+ else if ((data & 0x0f) == 0x02)
+ *rate = 48000;
+ else
+ return -EAGAIN;
+
+ /* Check multiplier. */
+ if ((data & 0xf0) == 0x80)
+ *rate *= 2;
+ else if ((data & 0xf0) != 0x00)
+ return -EAGAIN;
+
+ return err;
+}
+
+int snd_tscm_stream_get_clock(struct snd_tscm *tscm, enum snd_tscm_clock *clock)
+{
+ u32 data;
+ int err;
+
+ err = get_clock(tscm, &data);
+ if (err < 0)
+ return err;
+
+ *clock = ((data & 0x00ff0000) >> 16) - 1;
+ if (*clock < 0 || *clock > SND_TSCM_CLOCK_ADAT)
+ return -EIO;
+
+ return 0;
+}
+
+static int enable_data_channels(struct snd_tscm *tscm)
+{
+ __be32 reg;
+ u32 data;
+ unsigned int i;
+ int err;
+
+ data = 0;
+ for (i = 0; i < tscm->spec->pcm_capture_analog_channels; ++i)
+ data |= BIT(i);
+ if (tscm->spec->has_adat)
+ data |= 0x0000ff00;
+ if (tscm->spec->has_spdif)
+ data |= 0x00030000;
+
+ reg = cpu_to_be32(data);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_TX_PCM_CHANNELS,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ data = 0;
+ for (i = 0; i < tscm->spec->pcm_playback_analog_channels; ++i)
+ data |= BIT(i);
+ if (tscm->spec->has_adat)
+ data |= 0x0000ff00;
+ if (tscm->spec->has_spdif)
+ data |= 0x00030000;
+
+ reg = cpu_to_be32(data);
+ return snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_RX_PCM_CHANNELS,
+ &reg, sizeof(reg), 0);
+}
+
+static int set_stream_formats(struct snd_tscm *tscm, unsigned int rate)
+{
+ __be32 reg;
+ int err;
+
+ /* Set an option for unknown purpose. */
+ reg = cpu_to_be32(0x00200000);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_SET_OPTION,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ err = enable_data_channels(tscm);
+ if (err < 0)
+ return err;
+
+ return set_clock(tscm, rate, INT_MAX);
+}
+
+static void finish_session(struct snd_tscm *tscm)
+{
+ __be32 reg;
+
+ reg = 0;
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_START_STREAMING,
+ &reg, sizeof(reg), 0);
+
+ reg = 0;
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_ON,
+ &reg, sizeof(reg), 0);
+
+}
+
+static int begin_session(struct snd_tscm *tscm)
+{
+ __be32 reg;
+ int err;
+
+ reg = cpu_to_be32(0x00000001);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_START_STREAMING,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ reg = cpu_to_be32(0x00000001);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_ON,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ /* Set an option for unknown purpose. */
+ reg = cpu_to_be32(0x00002000);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_SET_OPTION,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ /* Start multiplexing PCM samples on packets. */
+ reg = cpu_to_be32(0x00000001);
+ return snd_fw_transaction(tscm->unit,
+ TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_TX_ON,
+ &reg, sizeof(reg), 0);
+}
+
+static void release_resources(struct snd_tscm *tscm)
+{
+ __be32 reg;
+
+ /* Unregister channels. */
+ reg = cpu_to_be32(0x00000000);
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_TX_CH,
+ &reg, sizeof(reg), 0);
+ reg = cpu_to_be32(0x00000000);
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_UNKNOWN,
+ &reg, sizeof(reg), 0);
+ reg = cpu_to_be32(0x00000000);
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_CH,
+ &reg, sizeof(reg), 0);
+
+ /* Release isochronous resources. */
+ fw_iso_resources_free(&tscm->tx_resources);
+ fw_iso_resources_free(&tscm->rx_resources);
+}
+
+static int keep_resources(struct snd_tscm *tscm, unsigned int rate)
+{
+ __be32 reg;
+ int err;
+
+ /* Keep resources for in-stream. */
+ err = amdtp_tscm_set_parameters(&tscm->tx_stream, rate);
+ if (err < 0)
+ return err;
+ err = fw_iso_resources_allocate(&tscm->tx_resources,
+ amdtp_stream_get_max_payload(&tscm->tx_stream),
+ fw_parent_device(tscm->unit)->max_speed);
+ if (err < 0)
+ goto error;
+
+ /* Keep resources for out-stream. */
+ err = amdtp_tscm_set_parameters(&tscm->rx_stream, rate);
+ if (err < 0)
+ return err;
+ err = fw_iso_resources_allocate(&tscm->rx_resources,
+ amdtp_stream_get_max_payload(&tscm->rx_stream),
+ fw_parent_device(tscm->unit)->max_speed);
+ if (err < 0)
+ return err;
+
+ /* Register the isochronous channel for transmitting stream. */
+ reg = cpu_to_be32(tscm->tx_resources.channel);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_TX_CH,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ goto error;
+
+ /* Unknown */
+ reg = cpu_to_be32(0x00000002);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_UNKNOWN,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ goto error;
+
+ /* Register the isochronous channel for receiving stream. */
+ reg = cpu_to_be32(tscm->rx_resources.channel);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_ISOC_RX_CH,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ goto error;
+
+ return 0;
+error:
+ release_resources(tscm);
+ return err;
+}
+
+int snd_tscm_stream_init_duplex(struct snd_tscm *tscm)
+{
+ unsigned int pcm_channels;
+ int err;
+
+ /* For out-stream. */
+ err = fw_iso_resources_init(&tscm->rx_resources, tscm->unit);
+ if (err < 0)
+ return err;
+ pcm_channels = tscm->spec->pcm_playback_analog_channels;
+ if (tscm->spec->has_adat)
+ pcm_channels += 8;
+ if (tscm->spec->has_spdif)
+ pcm_channels += 2;
+ err = amdtp_tscm_init(&tscm->rx_stream, tscm->unit, AMDTP_OUT_STREAM,
+ pcm_channels);
+ if (err < 0)
+ return err;
+
+ /* For in-stream. */
+ err = fw_iso_resources_init(&tscm->tx_resources, tscm->unit);
+ if (err < 0)
+ return err;
+ pcm_channels = tscm->spec->pcm_capture_analog_channels;
+ if (tscm->spec->has_adat)
+ pcm_channels += 8;
+ if (tscm->spec->has_spdif)
+ pcm_channels += 2;
+ err = amdtp_tscm_init(&tscm->tx_stream, tscm->unit, AMDTP_IN_STREAM,
+ pcm_channels);
+ if (err < 0)
+ amdtp_stream_destroy(&tscm->rx_stream);
+
+ return 0;
+}
+
+/* At bus reset, streaming is stopped and some registers are clear. */
+void snd_tscm_stream_update_duplex(struct snd_tscm *tscm)
+{
+ amdtp_stream_pcm_abort(&tscm->tx_stream);
+ amdtp_stream_stop(&tscm->tx_stream);
+
+ amdtp_stream_pcm_abort(&tscm->rx_stream);
+ amdtp_stream_stop(&tscm->rx_stream);
+}
+
+/*
+ * This function should be called before starting streams or after stopping
+ * streams.
+ */
+void snd_tscm_stream_destroy_duplex(struct snd_tscm *tscm)
+{
+ amdtp_stream_destroy(&tscm->rx_stream);
+ amdtp_stream_destroy(&tscm->tx_stream);
+
+ fw_iso_resources_destroy(&tscm->rx_resources);
+ fw_iso_resources_destroy(&tscm->tx_resources);
+}
+
+int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate)
+{
+ unsigned int curr_rate;
+ int err;
+
+ if (tscm->substreams_counter == 0)
+ return 0;
+
+ err = snd_tscm_stream_get_rate(tscm, &curr_rate);
+ if (err < 0)
+ return err;
+ if (curr_rate != rate ||
+ amdtp_streaming_error(&tscm->tx_stream) ||
+ amdtp_streaming_error(&tscm->rx_stream)) {
+ finish_session(tscm);
+
+ amdtp_stream_stop(&tscm->tx_stream);
+ amdtp_stream_stop(&tscm->rx_stream);
+
+ release_resources(tscm);
+ }
+
+ if (!amdtp_stream_running(&tscm->tx_stream)) {
+ amdtp_stream_set_sync(CIP_SYNC_TO_DEVICE,
+ &tscm->tx_stream, &tscm->rx_stream);
+ err = keep_resources(tscm, rate);
+ if (err < 0)
+ goto error;
+
+ err = set_stream_formats(tscm, rate);
+ if (err < 0)
+ goto error;
+
+ err = begin_session(tscm);
+ if (err < 0)
+ goto error;
+
+ err = amdtp_stream_start(&tscm->tx_stream,
+ tscm->tx_resources.channel,
+ fw_parent_device(tscm->unit)->max_speed);
+ if (err < 0)
+ goto error;
+
+ if (!amdtp_stream_wait_callback(&tscm->tx_stream,
+ CALLBACK_TIMEOUT)) {
+ err = -ETIMEDOUT;
+ goto error;
+ }
+ }
+
+ if (!amdtp_stream_running(&tscm->rx_stream)) {
+ err = amdtp_stream_start(&tscm->rx_stream,
+ tscm->rx_resources.channel,
+ fw_parent_device(tscm->unit)->max_speed);
+ if (err < 0)
+ goto error;
+
+ if (!amdtp_stream_wait_callback(&tscm->rx_stream,
+ CALLBACK_TIMEOUT)) {
+ err = -ETIMEDOUT;
+ goto error;
+ }
+ }
+
+ return 0;
+error:
+ amdtp_stream_stop(&tscm->tx_stream);
+ amdtp_stream_stop(&tscm->rx_stream);
+
+ finish_session(tscm);
+ release_resources(tscm);
+
+ return err;
+}
+
+void snd_tscm_stream_stop_duplex(struct snd_tscm *tscm)
+{
+ if (tscm->substreams_counter > 0)
+ return;
+
+ amdtp_stream_stop(&tscm->tx_stream);
+ amdtp_stream_stop(&tscm->rx_stream);
+
+ finish_session(tscm);
+ release_resources(tscm);
+}
+
+void snd_tscm_stream_lock_changed(struct snd_tscm *tscm)
+{
+ tscm->dev_lock_changed = true;
+ wake_up(&tscm->hwdep_wait);
+}
+
+int snd_tscm_stream_lock_try(struct snd_tscm *tscm)
+{
+ int err;
+
+ spin_lock_irq(&tscm->lock);
+
+ /* user land lock this */
+ if (tscm->dev_lock_count < 0) {
+ err = -EBUSY;
+ goto end;
+ }
+
+ /* this is the first time */
+ if (tscm->dev_lock_count++ == 0)
+ snd_tscm_stream_lock_changed(tscm);
+ err = 0;
+end:
+ spin_unlock_irq(&tscm->lock);
+ return err;
+}
+
+void snd_tscm_stream_lock_release(struct snd_tscm *tscm)
+{
+ spin_lock_irq(&tscm->lock);
+
+ if (WARN_ON(tscm->dev_lock_count <= 0))
+ goto end;
+ if (--tscm->dev_lock_count == 0)
+ snd_tscm_stream_lock_changed(tscm);
+end:
+ spin_unlock_irq(&tscm->lock);
+}
diff --git a/sound/firewire/tascam/tascam-transaction.c b/sound/firewire/tascam/tascam-transaction.c
new file mode 100644
index 000000000000..904ce0329fa1
--- /dev/null
+++ b/sound/firewire/tascam/tascam-transaction.c
@@ -0,0 +1,302 @@
+/*
+ * tascam-transaction.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "tascam.h"
+
+/*
+ * When return minus value, given argument is not MIDI status.
+ * When return 0, given argument is a beginning of system exclusive.
+ * When return the others, given argument is MIDI data.
+ */
+static inline int calculate_message_bytes(u8 status)
+{
+ switch (status) {
+ case 0xf6: /* Tune request. */
+ case 0xf8: /* Timing clock. */
+ case 0xfa: /* Start. */
+ case 0xfb: /* Continue. */
+ case 0xfc: /* Stop. */
+ case 0xfe: /* Active sensing. */
+ case 0xff: /* System reset. */
+ return 1;
+ case 0xf1: /* MIDI time code quarter frame. */
+ case 0xf3: /* Song select. */
+ return 2;
+ case 0xf2: /* Song position pointer. */
+ return 3;
+ case 0xf0: /* Exclusive. */
+ return 0;
+ case 0xf7: /* End of exclusive. */
+ break;
+ case 0xf4: /* Undefined. */
+ case 0xf5: /* Undefined. */
+ case 0xf9: /* Undefined. */
+ case 0xfd: /* Undefined. */
+ break;
+ default:
+ switch (status & 0xf0) {
+ case 0x80: /* Note on. */
+ case 0x90: /* Note off. */
+ case 0xa0: /* Polyphonic key pressure. */
+ case 0xb0: /* Control change and Mode change. */
+ case 0xe0: /* Pitch bend change. */
+ return 3;
+ case 0xc0: /* Program change. */
+ case 0xd0: /* Channel pressure. */
+ return 2;
+ default:
+ break;
+ }
+ break;
+ }
+
+ return -EINVAL;
+}
+
+static int fill_message(struct snd_rawmidi_substream *substream, u8 *buf)
+{
+ struct snd_tscm *tscm = substream->rmidi->private_data;
+ unsigned int port = substream->number;
+ int i, len, consume;
+ u8 *label, *msg;
+ u8 status;
+
+ /* The first byte is used for label, the rest for MIDI bytes. */
+ label = buf;
+ msg = buf + 1;
+
+ consume = snd_rawmidi_transmit_peek(substream, msg, 3);
+ if (consume == 0)
+ return 0;
+
+ /* On exclusive message. */
+ if (tscm->on_sysex[port]) {
+ /* Seek the end of exclusives. */
+ for (i = 0; i < consume; ++i) {
+ if (msg[i] == 0xf7) {
+ tscm->on_sysex[port] = false;
+ break;
+ }
+ }
+
+ /* At the end of exclusive message, use label 0x07. */
+ if (!tscm->on_sysex[port]) {
+ consume = i + 1;
+ *label = (port << 4) | 0x07;
+ /* During exclusive message, use label 0x04. */
+ } else if (consume == 3) {
+ *label = (port << 4) | 0x04;
+ /* We need to fill whole 3 bytes. Go to next change. */
+ } else {
+ return 0;
+ }
+
+ len = consume;
+ } else {
+ /* The beginning of exclusives. */
+ if (msg[0] == 0xf0) {
+ /* Transfer it in next chance in another condition. */
+ tscm->on_sysex[port] = true;
+ return 0;
+ } else {
+ /* On running-status. */
+ if ((msg[0] & 0x80) != 0x80)
+ status = tscm->running_status[port];
+ else
+ status = msg[0];
+
+ /* Calculate consume bytes. */
+ len = calculate_message_bytes(status);
+ if (len <= 0)
+ return 0;
+
+ /* On running-status. */
+ if ((msg[0] & 0x80) != 0x80) {
+ /* Enough MIDI bytes were not retrieved. */
+ if (consume < len - 1)
+ return 0;
+ consume = len - 1;
+
+ msg[2] = msg[1];
+ msg[1] = msg[0];
+ msg[0] = tscm->running_status[port];
+ } else {
+ /* Enough MIDI bytes were not retrieved. */
+ if (consume < len)
+ return 0;
+ consume = len;
+
+ tscm->running_status[port] = msg[0];
+ }
+ }
+
+ *label = (port << 4) | (msg[0] >> 4);
+ }
+
+ if (len > 0 && len < 3)
+ memset(msg + len, 0, 3 - len);
+
+ return consume;
+}
+
+static void handle_midi_tx(struct fw_card *card, struct fw_request *request,
+ int tcode, int destination, int source,
+ int generation, unsigned long long offset,
+ void *data, size_t length, void *callback_data)
+{
+ struct snd_tscm *tscm = callback_data;
+ u32 *buf = (u32 *)data;
+ unsigned int messages;
+ unsigned int i;
+ unsigned int port;
+ struct snd_rawmidi_substream *substream;
+ u8 *b;
+ int bytes;
+
+ if (offset != tscm->async_handler.offset)
+ goto end;
+
+ messages = length / 8;
+ for (i = 0; i < messages; i++) {
+ b = (u8 *)(buf + i * 2);
+
+ port = b[0] >> 4;
+ /* TODO: support virtual MIDI ports. */
+ if (port >= tscm->spec->midi_capture_ports)
+ goto end;
+
+ /* Assume the message length. */
+ bytes = calculate_message_bytes(b[1]);
+ /* On MIDI data or exclusives. */
+ if (bytes <= 0) {
+ /* Seek the end of exclusives. */
+ for (bytes = 1; bytes < 4; bytes++) {
+ if (b[bytes] == 0xf7)
+ break;
+ }
+ if (bytes == 4)
+ bytes = 3;
+ }
+
+ substream = ACCESS_ONCE(tscm->tx_midi_substreams[port]);
+ if (substream != NULL)
+ snd_rawmidi_receive(substream, b + 1, bytes);
+ }
+end:
+ fw_send_response(card, request, RCODE_COMPLETE);
+}
+
+int snd_tscm_transaction_register(struct snd_tscm *tscm)
+{
+ static const struct fw_address_region resp_register_region = {
+ .start = 0xffffe0000000ull,
+ .end = 0xffffe000ffffull,
+ };
+ unsigned int i;
+ int err;
+
+ /*
+ * Usually, two quadlets are transferred by one transaction. The first
+ * quadlet has MIDI messages, the rest includes timestamp.
+ * Sometimes, 8 set of the data is transferred by a block transaction.
+ */
+ tscm->async_handler.length = 8 * 8;
+ tscm->async_handler.address_callback = handle_midi_tx;
+ tscm->async_handler.callback_data = tscm;
+
+ err = fw_core_add_address_handler(&tscm->async_handler,
+ &resp_register_region);
+ if (err < 0)
+ return err;
+
+ err = snd_tscm_transaction_reregister(tscm);
+ if (err < 0)
+ goto error;
+
+ for (i = 0; i < TSCM_MIDI_OUT_PORT_MAX; i++) {
+ err = snd_fw_async_midi_port_init(
+ &tscm->out_ports[i], tscm->unit,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_RX_QUAD,
+ 4, fill_message);
+ if (err < 0)
+ goto error;
+ }
+
+ return err;
+error:
+ fw_core_remove_address_handler(&tscm->async_handler);
+ return err;
+}
+
+/* At bus reset, these registers are cleared. */
+int snd_tscm_transaction_reregister(struct snd_tscm *tscm)
+{
+ struct fw_device *device = fw_parent_device(tscm->unit);
+ __be32 reg;
+ int err;
+
+ /* Register messaging address. Block transaction is not allowed. */
+ reg = cpu_to_be32((device->card->node_id << 16) |
+ (tscm->async_handler.offset >> 32));
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_HI,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ reg = cpu_to_be32(tscm->async_handler.offset);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_LO,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ /* Turn on messaging. */
+ reg = cpu_to_be32(0x00000001);
+ err = snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ON,
+ &reg, sizeof(reg), 0);
+ if (err < 0)
+ return err;
+
+ /* Turn on FireWire LED. */
+ reg = cpu_to_be32(0x0001008e);
+ return snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_LED_POWER,
+ &reg, sizeof(reg), 0);
+}
+
+void snd_tscm_transaction_unregister(struct snd_tscm *tscm)
+{
+ __be32 reg;
+ unsigned int i;
+
+ /* Turn off FireWire LED. */
+ reg = cpu_to_be32(0x0000008e);
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_LED_POWER,
+ &reg, sizeof(reg), 0);
+
+ /* Turn off messaging. */
+ reg = cpu_to_be32(0x00000000);
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ON,
+ &reg, sizeof(reg), 0);
+
+ /* Unregister the address. */
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_HI,
+ &reg, sizeof(reg), 0);
+ snd_fw_transaction(tscm->unit, TCODE_WRITE_QUADLET_REQUEST,
+ TSCM_ADDR_BASE + TSCM_OFFSET_MIDI_TX_ADDR_LO,
+ &reg, sizeof(reg), 0);
+
+ fw_core_remove_address_handler(&tscm->async_handler);
+ for (i = 0; i < TSCM_MIDI_OUT_PORT_MAX; i++)
+ snd_fw_async_midi_port_destroy(&tscm->out_ports[i]);
+}
diff --git a/sound/firewire/tascam/tascam.c b/sound/firewire/tascam/tascam.c
new file mode 100644
index 000000000000..ee0bc1839508
--- /dev/null
+++ b/sound/firewire/tascam/tascam.c
@@ -0,0 +1,209 @@
+/*
+ * tascam.c - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#include "tascam.h"
+
+MODULE_DESCRIPTION("TASCAM FireWire series Driver");
+MODULE_AUTHOR("Takashi Sakamoto <o-takashi@sakamocchi.jp>");
+MODULE_LICENSE("GPL v2");
+
+static struct snd_tscm_spec model_specs[] = {
+ {
+ .name = "FW-1884",
+ .has_adat = true,
+ .has_spdif = true,
+ .pcm_capture_analog_channels = 8,
+ .pcm_playback_analog_channels = 8,
+ .midi_capture_ports = 4,
+ .midi_playback_ports = 4,
+ .is_controller = true,
+ },
+ {
+ .name = "FW-1082",
+ .has_adat = false,
+ .has_spdif = true,
+ .pcm_capture_analog_channels = 8,
+ .pcm_playback_analog_channels = 2,
+ .midi_capture_ports = 2,
+ .midi_playback_ports = 2,
+ .is_controller = true,
+ },
+ /* FW-1804 may be supported. */
+};
+
+static int identify_model(struct snd_tscm *tscm)
+{
+ struct fw_device *fw_dev = fw_parent_device(tscm->unit);
+ const u32 *config_rom = fw_dev->config_rom;
+ char model[9];
+ unsigned int i;
+ u8 c;
+
+ if (fw_dev->config_rom_length < 30) {
+ dev_err(&tscm->unit->device,
+ "Configuration ROM is too short.\n");
+ return -ENODEV;
+ }
+
+ /* Pick up model name from certain addresses. */
+ for (i = 0; i < 8; i++) {
+ c = config_rom[28 + i / 4] >> (24 - 8 * (i % 4));
+ if (c == '\0')
+ break;
+ model[i] = c;
+ }
+ model[i] = '\0';
+
+ for (i = 0; i < ARRAY_SIZE(model_specs); i++) {
+ if (strcmp(model, model_specs[i].name) == 0) {
+ tscm->spec = &model_specs[i];
+ break;
+ }
+ }
+ if (tscm->spec == NULL)
+ return -ENODEV;
+
+ strcpy(tscm->card->driver, "FW-TASCAM");
+ strcpy(tscm->card->shortname, model);
+ strcpy(tscm->card->mixername, model);
+ snprintf(tscm->card->longname, sizeof(tscm->card->longname),
+ "TASCAM %s, GUID %08x%08x at %s, S%d", model,
+ fw_dev->config_rom[3], fw_dev->config_rom[4],
+ dev_name(&tscm->unit->device), 100 << fw_dev->max_speed);
+
+ return 0;
+}
+
+static void tscm_card_free(struct snd_card *card)
+{
+ struct snd_tscm *tscm = card->private_data;
+
+ snd_tscm_transaction_unregister(tscm);
+ snd_tscm_stream_destroy_duplex(tscm);
+
+ fw_unit_put(tscm->unit);
+
+ mutex_destroy(&tscm->mutex);
+}
+
+static int snd_tscm_probe(struct fw_unit *unit,
+ const struct ieee1394_device_id *entry)
+{
+ struct snd_card *card;
+ struct snd_tscm *tscm;
+ int err;
+
+ /* create card */
+ err = snd_card_new(&unit->device, -1, NULL, THIS_MODULE,
+ sizeof(struct snd_tscm), &card);
+ if (err < 0)
+ return err;
+ card->private_free = tscm_card_free;
+
+ /* initialize myself */
+ tscm = card->private_data;
+ tscm->card = card;
+ tscm->unit = fw_unit_get(unit);
+
+ mutex_init(&tscm->mutex);
+ spin_lock_init(&tscm->lock);
+ init_waitqueue_head(&tscm->hwdep_wait);
+
+ err = identify_model(tscm);
+ if (err < 0)
+ goto error;
+
+ snd_tscm_proc_init(tscm);
+
+ err = snd_tscm_stream_init_duplex(tscm);
+ if (err < 0)
+ goto error;
+
+ err = snd_tscm_create_pcm_devices(tscm);
+ if (err < 0)
+ goto error;
+
+ err = snd_tscm_transaction_register(tscm);
+ if (err < 0)
+ goto error;
+
+ err = snd_tscm_create_midi_devices(tscm);
+ if (err < 0)
+ goto error;
+
+ err = snd_tscm_create_hwdep_device(tscm);
+ if (err < 0)
+ goto error;
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto error;
+
+ dev_set_drvdata(&unit->device, tscm);
+
+ return err;
+error:
+ snd_card_free(card);
+ return err;
+}
+
+static void snd_tscm_update(struct fw_unit *unit)
+{
+ struct snd_tscm *tscm = dev_get_drvdata(&unit->device);
+
+ snd_tscm_transaction_reregister(tscm);
+
+ mutex_lock(&tscm->mutex);
+ snd_tscm_stream_update_duplex(tscm);
+ mutex_unlock(&tscm->mutex);
+}
+
+static void snd_tscm_remove(struct fw_unit *unit)
+{
+ struct snd_tscm *tscm = dev_get_drvdata(&unit->device);
+
+ /* No need to wait for releasing card object in this context. */
+ snd_card_free_when_closed(tscm->card);
+}
+
+static const struct ieee1394_device_id snd_tscm_id_table[] = {
+ {
+ .match_flags = IEEE1394_MATCH_VENDOR_ID |
+ IEEE1394_MATCH_SPECIFIER_ID,
+ .vendor_id = 0x00022e,
+ .specifier_id = 0x00022e,
+ },
+ /* FE-08 requires reverse-engineering because it just has faders. */
+ {}
+};
+MODULE_DEVICE_TABLE(ieee1394, snd_tscm_id_table);
+
+static struct fw_driver tscm_driver = {
+ .driver = {
+ .owner = THIS_MODULE,
+ .name = "snd-firewire-tascam",
+ .bus = &fw_bus_type,
+ },
+ .probe = snd_tscm_probe,
+ .update = snd_tscm_update,
+ .remove = snd_tscm_remove,
+ .id_table = snd_tscm_id_table,
+};
+
+static int __init snd_tscm_init(void)
+{
+ return driver_register(&tscm_driver.driver);
+}
+
+static void __exit snd_tscm_exit(void)
+{
+ driver_unregister(&tscm_driver.driver);
+}
+
+module_init(snd_tscm_init);
+module_exit(snd_tscm_exit);
diff --git a/sound/firewire/tascam/tascam.h b/sound/firewire/tascam/tascam.h
new file mode 100644
index 000000000000..2d028d2bd3bd
--- /dev/null
+++ b/sound/firewire/tascam/tascam.h
@@ -0,0 +1,147 @@
+/*
+ * tascam.h - a part of driver for TASCAM FireWire series
+ *
+ * Copyright (c) 2015 Takashi Sakamoto
+ *
+ * Licensed under the terms of the GNU General Public License, version 2.
+ */
+
+#ifndef SOUND_TASCAM_H_INCLUDED
+#define SOUND_TASCAM_H_INCLUDED
+
+#include <linux/device.h>
+#include <linux/firewire.h>
+#include <linux/firewire-constants.h>
+#include <linux/module.h>
+#include <linux/mod_devicetable.h>
+#include <linux/mutex.h>
+#include <linux/slab.h>
+#include <linux/compat.h>
+
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/info.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/firewire.h>
+#include <sound/hwdep.h>
+#include <sound/rawmidi.h>
+
+#include "../lib.h"
+#include "../amdtp-stream.h"
+#include "../iso-resources.h"
+
+struct snd_tscm_spec {
+ const char *const name;
+ bool has_adat;
+ bool has_spdif;
+ unsigned int pcm_capture_analog_channels;
+ unsigned int pcm_playback_analog_channels;
+ unsigned int midi_capture_ports;
+ unsigned int midi_playback_ports;
+ bool is_controller;
+};
+
+#define TSCM_MIDI_IN_PORT_MAX 4
+#define TSCM_MIDI_OUT_PORT_MAX 4
+
+struct snd_tscm {
+ struct snd_card *card;
+ struct fw_unit *unit;
+
+ struct mutex mutex;
+ spinlock_t lock;
+
+ const struct snd_tscm_spec *spec;
+
+ struct fw_iso_resources tx_resources;
+ struct fw_iso_resources rx_resources;
+ struct amdtp_stream tx_stream;
+ struct amdtp_stream rx_stream;
+ unsigned int substreams_counter;
+
+ int dev_lock_count;
+ bool dev_lock_changed;
+ wait_queue_head_t hwdep_wait;
+
+ /* For MIDI message incoming transactions. */
+ struct fw_address_handler async_handler;
+ struct snd_rawmidi_substream *tx_midi_substreams[TSCM_MIDI_IN_PORT_MAX];
+
+ /* For MIDI message outgoing transactions. */
+ struct snd_fw_async_midi_port out_ports[TSCM_MIDI_OUT_PORT_MAX];
+ u8 running_status[TSCM_MIDI_OUT_PORT_MAX];
+ bool on_sysex[TSCM_MIDI_OUT_PORT_MAX];
+
+ /* For control messages. */
+ struct snd_firewire_tascam_status *status;
+};
+
+#define TSCM_ADDR_BASE 0xffff00000000ull
+
+#define TSCM_OFFSET_FIRMWARE_REGISTER 0x0000
+#define TSCM_OFFSET_FIRMWARE_FPGA 0x0004
+#define TSCM_OFFSET_FIRMWARE_ARM 0x0008
+#define TSCM_OFFSET_FIRMWARE_HW 0x000c
+
+#define TSCM_OFFSET_ISOC_TX_CH 0x0200
+#define TSCM_OFFSET_UNKNOWN 0x0204
+#define TSCM_OFFSET_START_STREAMING 0x0208
+#define TSCM_OFFSET_ISOC_RX_CH 0x020c
+#define TSCM_OFFSET_ISOC_RX_ON 0x0210 /* Little conviction. */
+#define TSCM_OFFSET_TX_PCM_CHANNELS 0x0214
+#define TSCM_OFFSET_RX_PCM_CHANNELS 0x0218
+#define TSCM_OFFSET_MULTIPLEX_MODE 0x021c
+#define TSCM_OFFSET_ISOC_TX_ON 0x0220
+/* Unknown 0x0224 */
+#define TSCM_OFFSET_CLOCK_STATUS 0x0228
+#define TSCM_OFFSET_SET_OPTION 0x022c
+
+#define TSCM_OFFSET_MIDI_TX_ON 0x0300
+#define TSCM_OFFSET_MIDI_TX_ADDR_HI 0x0304
+#define TSCM_OFFSET_MIDI_TX_ADDR_LO 0x0308
+
+#define TSCM_OFFSET_LED_POWER 0x0404
+
+#define TSCM_OFFSET_MIDI_RX_QUAD 0x4000
+
+enum snd_tscm_clock {
+ SND_TSCM_CLOCK_INTERNAL = 0,
+ SND_TSCM_CLOCK_WORD = 1,
+ SND_TSCM_CLOCK_SPDIF = 2,
+ SND_TSCM_CLOCK_ADAT = 3,
+};
+
+int amdtp_tscm_init(struct amdtp_stream *s, struct fw_unit *unit,
+ enum amdtp_stream_direction dir, unsigned int pcm_channels);
+int amdtp_tscm_set_parameters(struct amdtp_stream *s, unsigned int rate);
+int amdtp_tscm_add_pcm_hw_constraints(struct amdtp_stream *s,
+ struct snd_pcm_runtime *runtime);
+void amdtp_tscm_set_pcm_format(struct amdtp_stream *s, snd_pcm_format_t format);
+
+int snd_tscm_stream_get_rate(struct snd_tscm *tscm, unsigned int *rate);
+int snd_tscm_stream_get_clock(struct snd_tscm *tscm,
+ enum snd_tscm_clock *clock);
+int snd_tscm_stream_init_duplex(struct snd_tscm *tscm);
+void snd_tscm_stream_update_duplex(struct snd_tscm *tscm);
+void snd_tscm_stream_destroy_duplex(struct snd_tscm *tscm);
+int snd_tscm_stream_start_duplex(struct snd_tscm *tscm, unsigned int rate);
+void snd_tscm_stream_stop_duplex(struct snd_tscm *tscm);
+
+void snd_tscm_stream_lock_changed(struct snd_tscm *tscm);
+int snd_tscm_stream_lock_try(struct snd_tscm *tscm);
+void snd_tscm_stream_lock_release(struct snd_tscm *tscm);
+
+int snd_tscm_transaction_register(struct snd_tscm *tscm);
+int snd_tscm_transaction_reregister(struct snd_tscm *tscm);
+void snd_tscm_transaction_unregister(struct snd_tscm *tscm);
+
+void snd_tscm_proc_init(struct snd_tscm *tscm);
+
+int snd_tscm_create_pcm_devices(struct snd_tscm *tscm);
+
+int snd_tscm_create_midi_devices(struct snd_tscm *tscm);
+
+int snd_tscm_create_hwdep_device(struct snd_tscm *tscm);
+
+#endif
diff --git a/sound/hda/ext/hdac_ext_stream.c b/sound/hda/ext/hdac_ext_stream.c
index 33ba77dd32f2..cb89ec7c8147 100644
--- a/sound/hda/ext/hdac_ext_stream.c
+++ b/sound/hda/ext/hdac_ext_stream.c
@@ -227,7 +227,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_link_stream_setup);
void snd_hdac_ext_link_set_stream_id(struct hdac_ext_link *link,
int stream)
{
- snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, (1 << stream), 0);
+ snd_hdac_updatew(link->ml_addr, AZX_REG_ML_LOSIDV, (1 << stream), 1 << stream);
}
EXPORT_SYMBOL_GPL(snd_hdac_ext_link_set_stream_id);
@@ -385,14 +385,13 @@ void snd_hdac_ext_stream_release(struct hdac_ext_stream *stream, int type)
break;
case HDAC_EXT_STREAM_TYPE_HOST:
- if (stream->decoupled) {
+ if (stream->decoupled && !stream->link_locked)
snd_hdac_ext_stream_decouple(ebus, stream, false);
- snd_hdac_stream_release(&stream->hstream);
- }
+ snd_hdac_stream_release(&stream->hstream);
break;
case HDAC_EXT_STREAM_TYPE_LINK:
- if (stream->decoupled)
+ if (stream->decoupled && !stream->hstream.opened)
snd_hdac_ext_stream_decouple(ebus, stream, false);
spin_lock_irq(&bus->reg_lock);
stream->link_locked = 0;
diff --git a/sound/hda/hda_bus_type.c b/sound/hda/hda_bus_type.c
index 89c2711baaaf..3060e2aee36f 100644
--- a/sound/hda/hda_bus_type.c
+++ b/sound/hda/hda_bus_type.c
@@ -4,6 +4,7 @@
#include <linux/init.h>
#include <linux/device.h>
#include <linux/module.h>
+#include <linux/mod_devicetable.h>
#include <linux/export.h>
#include <sound/hdaudio.h>
@@ -63,9 +64,21 @@ static int hda_bus_match(struct device *dev, struct device_driver *drv)
return 1;
}
+static int hda_uevent(struct device *dev, struct kobj_uevent_env *env)
+{
+ char modalias[32];
+
+ snd_hdac_codec_modalias(dev_to_hdac_dev(dev), modalias,
+ sizeof(modalias));
+ if (add_uevent_var(env, "MODALIAS=%s", modalias))
+ return -ENOMEM;
+ return 0;
+}
+
struct bus_type snd_hda_bus_type = {
.name = "hdaudio",
.match = hda_bus_match,
+ .uevent = hda_uevent,
};
EXPORT_SYMBOL_GPL(snd_hda_bus_type);
diff --git a/sound/hda/hdac_bus.c b/sound/hda/hdac_bus.c
index 27c447e4fe5c..0e81ea89a596 100644
--- a/sound/hda/hdac_bus.c
+++ b/sound/hda/hdac_bus.c
@@ -172,6 +172,15 @@ static void process_unsol_events(struct work_struct *work)
}
}
+/**
+ * snd_hdac_bus_add_device - Add a codec to bus
+ * @bus: HDA core bus
+ * @codec: HDA core device to add
+ *
+ * Adds the given codec to the list in the bus. The caddr_tbl array
+ * and codec_powered bits are updated, as well.
+ * Returns zero if success, or a negative error code.
+ */
int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec)
{
if (bus->caddr_tbl[codec->addr]) {
@@ -188,6 +197,11 @@ int snd_hdac_bus_add_device(struct hdac_bus *bus, struct hdac_device *codec)
}
EXPORT_SYMBOL_GPL(snd_hdac_bus_add_device);
+/**
+ * snd_hdac_bus_remove_device - Remove a codec from bus
+ * @bus: HDA core bus
+ * @codec: HDA core device to remove
+ */
void snd_hdac_bus_remove_device(struct hdac_bus *bus,
struct hdac_device *codec)
{
diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c
index db96042a497f..e361024eabb6 100644
--- a/sound/hda/hdac_device.c
+++ b/sound/hda/hdac_device.c
@@ -164,6 +164,43 @@ void snd_hdac_device_unregister(struct hdac_device *codec)
EXPORT_SYMBOL_GPL(snd_hdac_device_unregister);
/**
+ * snd_hdac_device_set_chip_name - set/update the codec name
+ * @codec: the HDAC device
+ * @name: name string to set
+ *
+ * Returns 0 if the name is set or updated, or a negative error code.
+ */
+int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name)
+{
+ char *newname;
+
+ if (!name)
+ return 0;
+ newname = kstrdup(name, GFP_KERNEL);
+ if (!newname)
+ return -ENOMEM;
+ kfree(codec->chip_name);
+ codec->chip_name = newname;
+ return 0;
+}
+EXPORT_SYMBOL_GPL(snd_hdac_device_set_chip_name);
+
+/**
+ * snd_hdac_codec_modalias - give the module alias name
+ * @codec: HDAC device
+ * @buf: string buffer to store
+ * @size: string buffer size
+ *
+ * Returns the size of string, like snprintf(), or a negative error code.
+ */
+int snd_hdac_codec_modalias(struct hdac_device *codec, char *buf, size_t size)
+{
+ return snprintf(buf, size, "hdaudio:v%08Xr%08Xa%02X\n",
+ codec->vendor_id, codec->revision_id, codec->type);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_codec_modalias);
+
+/**
* snd_hdac_make_cmd - compose a 32bit command word to be sent to the
* HD-audio controller
* @codec: the codec object
@@ -592,8 +629,10 @@ int snd_hdac_power_down_pm(struct hdac_device *codec)
EXPORT_SYMBOL_GPL(snd_hdac_power_down_pm);
#endif
-/*
- * Enable/disable the link power for a codec.
+/**
+ * snd_hdac_link_power - Enable/disable the link power for a codec
+ * @codec: the codec object
+ * @bool: enable or disable the link power
*/
int snd_hdac_link_power(struct hdac_device *codec, bool enable)
{
@@ -952,3 +991,84 @@ bool snd_hdac_is_supported_format(struct hdac_device *codec, hda_nid_t nid,
return true;
}
EXPORT_SYMBOL_GPL(snd_hdac_is_supported_format);
+
+static unsigned int codec_read(struct hdac_device *hdac, hda_nid_t nid,
+ int flags, unsigned int verb, unsigned int parm)
+{
+ unsigned int cmd = snd_hdac_make_cmd(hdac, nid, verb, parm);
+ unsigned int res;
+
+ if (snd_hdac_exec_verb(hdac, cmd, flags, &res))
+ return -1;
+
+ return res;
+}
+
+static int codec_write(struct hdac_device *hdac, hda_nid_t nid,
+ int flags, unsigned int verb, unsigned int parm)
+{
+ unsigned int cmd = snd_hdac_make_cmd(hdac, nid, verb, parm);
+
+ return snd_hdac_exec_verb(hdac, cmd, flags, NULL);
+}
+
+/**
+ * snd_hdac_codec_read - send a command and get the response
+ * @hdac: the HDAC device
+ * @nid: NID to send the command
+ * @flags: optional bit flags
+ * @verb: the verb to send
+ * @parm: the parameter for the verb
+ *
+ * Send a single command and read the corresponding response.
+ *
+ * Returns the obtained response value, or -1 for an error.
+ */
+int snd_hdac_codec_read(struct hdac_device *hdac, hda_nid_t nid,
+ int flags, unsigned int verb, unsigned int parm)
+{
+ return codec_read(hdac, nid, flags, verb, parm);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_codec_read);
+
+/**
+ * snd_hdac_codec_write - send a single command without waiting for response
+ * @hdac: the HDAC device
+ * @nid: NID to send the command
+ * @flags: optional bit flags
+ * @verb: the verb to send
+ * @parm: the parameter for the verb
+ *
+ * Send a single command without waiting for response.
+ *
+ * Returns 0 if successful, or a negative error code.
+ */
+int snd_hdac_codec_write(struct hdac_device *hdac, hda_nid_t nid,
+ int flags, unsigned int verb, unsigned int parm)
+{
+ return codec_write(hdac, nid, flags, verb, parm);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_codec_write);
+
+/**
+ * snd_hdac_check_power_state - check whether the actual power state matches
+ * with the target state
+ *
+ * @hdac: the HDAC device
+ * @nid: NID to send the command
+ * @target_state: target state to check for
+ *
+ * Return true if state matches, false if not
+ */
+bool snd_hdac_check_power_state(struct hdac_device *hdac,
+ hda_nid_t nid, unsigned int target_state)
+{
+ unsigned int state = codec_read(hdac, nid, 0,
+ AC_VERB_GET_POWER_STATE, 0);
+
+ if (state & AC_PWRST_ERROR)
+ return true;
+ state = (state >> 4) & 0x0f;
+ return (state == target_state);
+}
+EXPORT_SYMBOL_GPL(snd_hdac_check_power_state);
diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c
index 55c3df4458f7..8fef1b8d1fd8 100644
--- a/sound/hda/hdac_i915.c
+++ b/sound/hda/hdac_i915.c
@@ -23,6 +23,19 @@
static struct i915_audio_component *hdac_acomp;
+/**
+ * snd_hdac_set_codec_wakeup - Enable / disable HDMI/DP codec wakeup
+ * @bus: HDA core bus
+ * @enable: enable or disable the wakeup
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with i915 graphics.
+ *
+ * This function should be called during the chip reset, also called at
+ * resume for updating STATESTS register read.
+ *
+ * Returns zero for success or a negative error code.
+ */
int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable)
{
struct i915_audio_component *acomp = bus->audio_component;
@@ -45,6 +58,19 @@ int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable)
}
EXPORT_SYMBOL_GPL(snd_hdac_set_codec_wakeup);
+/**
+ * snd_hdac_display_power - Power up / down the power refcount
+ * @bus: HDA core bus
+ * @enable: power up or down
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with i915 graphics.
+ *
+ * This function manages a refcount and calls the i915 get_power() and
+ * put_power() ops accordingly, toggling the codec wakeup, too.
+ *
+ * Returns zero for success or a negative error code.
+ */
int snd_hdac_display_power(struct hdac_bus *bus, bool enable)
{
struct i915_audio_component *acomp = bus->audio_component;
@@ -71,6 +97,16 @@ int snd_hdac_display_power(struct hdac_bus *bus, bool enable)
}
EXPORT_SYMBOL_GPL(snd_hdac_display_power);
+/**
+ * snd_hdac_get_display_clk - Get CDCLK in kHz
+ * @bus: HDA core bus
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with i915 graphics.
+ *
+ * This function queries CDCLK value in kHz from the graphics driver and
+ * returns the value. A negative code is returned in error.
+ */
int snd_hdac_get_display_clk(struct hdac_bus *bus)
{
struct i915_audio_component *acomp = bus->audio_component;
@@ -134,6 +170,17 @@ static int hdac_component_master_match(struct device *dev, void *data)
return !strcmp(dev->driver->name, "i915");
}
+/**
+ * snd_hdac_i915_register_notifier - Register i915 audio component ops
+ * @aops: i915 audio component ops
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with i915 graphics.
+ *
+ * This function sets the given ops to be called by the i915 graphics driver.
+ *
+ * Returns zero for success or a negative error code.
+ */
int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *aops)
{
if (WARN_ON(!hdac_acomp))
@@ -144,6 +191,18 @@ int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops
}
EXPORT_SYMBOL_GPL(snd_hdac_i915_register_notifier);
+/**
+ * snd_hdac_i915_init - Initialize i915 audio component
+ * @bus: HDA core bus
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with i915 graphics.
+ *
+ * This function initializes and sets up the audio component to communicate
+ * with i915 graphics driver.
+ *
+ * Returns zero for success or a negative error code.
+ */
int snd_hdac_i915_init(struct hdac_bus *bus)
{
struct component_match *match = NULL;
@@ -187,6 +246,17 @@ out_err:
}
EXPORT_SYMBOL_GPL(snd_hdac_i915_init);
+/**
+ * snd_hdac_i915_exit - Finalize i915 audio component
+ * @bus: HDA core bus
+ *
+ * This function is supposed to be used only by a HD-audio controller
+ * driver that needs the interaction with i915 graphics.
+ *
+ * This function releases the i915 audio component that has been used.
+ *
+ * Returns zero for success or a negative error code.
+ */
int snd_hdac_i915_exit(struct hdac_bus *bus)
{
struct device *dev = bus->dev;
diff --git a/sound/hda/hdac_regmap.c b/sound/hda/hdac_regmap.c
index b0ed870ffb88..eb8f7c30cb09 100644
--- a/sound/hda/hdac_regmap.c
+++ b/sound/hda/hdac_regmap.c
@@ -339,6 +339,12 @@ static const struct regmap_config hda_regmap_cfg = {
.use_single_rw = true,
};
+/**
+ * snd_hdac_regmap_init - Initialize regmap for HDA register accesses
+ * @codec: the codec object
+ *
+ * Returns zero for success or a negative error code.
+ */
int snd_hdac_regmap_init(struct hdac_device *codec)
{
struct regmap *regmap;
@@ -352,6 +358,10 @@ int snd_hdac_regmap_init(struct hdac_device *codec)
}
EXPORT_SYMBOL_GPL(snd_hdac_regmap_init);
+/**
+ * snd_hdac_regmap_init - Release the regmap from HDA codec
+ * @codec: the codec object
+ */
void snd_hdac_regmap_exit(struct hdac_device *codec)
{
if (codec->regmap) {
diff --git a/sound/hda/hdac_stream.c b/sound/hda/hdac_stream.c
index 8981159813ef..38990a77d7b7 100644
--- a/sound/hda/hdac_stream.c
+++ b/sound/hda/hdac_stream.c
@@ -426,7 +426,8 @@ int snd_hdac_stream_setup_periods(struct hdac_stream *azx_dev)
}
EXPORT_SYMBOL_GPL(snd_hdac_stream_setup_periods);
-/* snd_hdac_stream_set_params - set stream parameters
+/**
+ * snd_hdac_stream_set_params - set stream parameters
* @azx_dev: HD-audio core stream for which parameters are to be set
* @format_val: format value parameter
*
diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c
index c71142dea98a..42d61bf41969 100644
--- a/sound/hda/hdac_sysfs.c
+++ b/sound/hda/hdac_sysfs.c
@@ -45,6 +45,13 @@ CODEC_ATTR(mfg);
CODEC_ATTR_STR(vendor_name);
CODEC_ATTR_STR(chip_name);
+static ssize_t modalias_show(struct device *dev, struct device_attribute *attr,
+ char *buf)
+{
+ return snd_hdac_codec_modalias(dev_to_hdac_dev(dev), buf, 256);
+}
+static DEVICE_ATTR_RO(modalias);
+
static struct attribute *hdac_dev_attrs[] = {
&dev_attr_type.attr,
&dev_attr_vendor_id.attr,
@@ -54,6 +61,7 @@ static struct attribute *hdac_dev_attrs[] = {
&dev_attr_mfg.attr,
&dev_attr_vendor_name.attr,
&dev_attr_chip_name.attr,
+ &dev_attr_modalias.attr,
NULL
};
diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c
index 2a9f4a345171..2706f271a83b 100644
--- a/sound/pci/cs46xx/cs46xx_lib.c
+++ b/sound/pci/cs46xx/cs46xx_lib.c
@@ -1864,7 +1864,7 @@ int snd_cs46xx_pcm_iec958(struct snd_cs46xx *chip, int device)
/* global setup */
pcm->info_flags = 0;
strcpy(pcm->name, "CS46xx - IEC958");
- chip->pcm_rear = pcm;
+ chip->pcm_iec958 = pcm;
snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
snd_dma_pci_data(chip->pci), 64*1024, 256*1024);
@@ -2528,7 +2528,7 @@ int snd_cs46xx_mixer(struct snd_cs46xx *chip, int spdif_device)
#ifdef CONFIG_SND_CS46XX_NEW_DSP
if (chip->nr_ac97_codecs == 1) {
unsigned int id2 = chip->ac97[CS46XX_PRIMARY_CODEC_INDEX]->id & 0xffff;
- if (id2 == 0x592b || id2 == 0x592d) {
+ if ((id2 & 0xfff0) == 0x5920) { /* CS4294 and CS4298 */
err = snd_ctl_add(card, snd_ctl_new1(&snd_cs46xx_front_dup_ctl, chip));
if (err < 0)
return err;
@@ -3780,6 +3780,11 @@ static int snd_cs46xx_suspend(struct device *dev)
snd_power_change_state(card, SNDRV_CTL_POWER_D3hot);
chip->in_suspend = 1;
snd_pcm_suspend_all(chip->pcm);
+#ifdef CONFIG_SND_CS46XX_NEW_DSP
+ snd_pcm_suspend_all(chip->pcm_rear);
+ snd_pcm_suspend_all(chip->pcm_center_lfe);
+ snd_pcm_suspend_all(chip->pcm_iec958);
+#endif
// chip->ac97_powerdown = snd_cs46xx_codec_read(chip, AC97_POWER_CONTROL);
// chip->ac97_general_purpose = snd_cs46xx_codec_read(chip, BA0_AC97_GENERAL_PURPOSE);
diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c
index d5ac25cc7fee..70671ad65d24 100644
--- a/sound/pci/hda/hda_bind.c
+++ b/sound/pci/hda/hda_bind.c
@@ -15,21 +15,22 @@
#include "hda_local.h"
/*
- * find a matching codec preset
+ * find a matching codec id
*/
static int hda_codec_match(struct hdac_device *dev, struct hdac_driver *drv)
{
struct hda_codec *codec = container_of(dev, struct hda_codec, core);
struct hda_codec_driver *driver =
container_of(drv, struct hda_codec_driver, core);
- const struct hda_codec_preset *preset;
+ const struct hda_device_id *list;
/* check probe_id instead of vendor_id if set */
u32 id = codec->probe_id ? codec->probe_id : codec->core.vendor_id;
+ u32 rev_id = codec->core.revision_id;
- for (preset = driver->preset; preset->id; preset++) {
- if (preset->id == id &&
- (!preset->rev || preset->rev == codec->core.revision_id)) {
- codec->preset = preset;
+ for (list = driver->id; list->vendor_id; list++) {
+ if (list->vendor_id == id &&
+ (!list->rev_id || list->rev_id == rev_id)) {
+ codec->preset = list;
return 1;
}
}
@@ -45,26 +46,45 @@ static void hda_codec_unsol_event(struct hdac_device *dev, unsigned int ev)
codec->patch_ops.unsol_event(codec, ev);
}
-/* reset the codec name from the preset */
-static int codec_refresh_name(struct hda_codec *codec, const char *name)
+/**
+ * snd_hda_codec_set_name - set the codec name
+ * @codec: the HDA codec
+ * @name: name string to set
+ */
+int snd_hda_codec_set_name(struct hda_codec *codec, const char *name)
{
- if (name) {
- kfree(codec->core.chip_name);
- codec->core.chip_name = kstrdup(name, GFP_KERNEL);
+ int err;
+
+ if (!name)
+ return 0;
+ err = snd_hdac_device_set_chip_name(&codec->core, name);
+ if (err < 0)
+ return err;
+
+ /* update the mixer name */
+ if (!*codec->card->mixername ||
+ codec->bus->mixer_assigned >= codec->core.addr) {
+ snprintf(codec->card->mixername,
+ sizeof(codec->card->mixername), "%s %s",
+ codec->core.vendor_name, codec->core.chip_name);
+ codec->bus->mixer_assigned = codec->core.addr;
}
- return codec->core.chip_name ? 0 : -ENOMEM;
+
+ return 0;
}
+EXPORT_SYMBOL_GPL(snd_hda_codec_set_name);
static int hda_codec_driver_probe(struct device *dev)
{
struct hda_codec *codec = dev_to_hda_codec(dev);
struct module *owner = dev->driver->owner;
+ hda_codec_patch_t patch;
int err;
if (WARN_ON(!codec->preset))
return -EINVAL;
- err = codec_refresh_name(codec, codec->preset->name);
+ err = snd_hda_codec_set_name(codec, codec->preset->name);
if (err < 0)
goto error;
err = snd_hdac_regmap_init(&codec->core);
@@ -76,9 +96,12 @@ static int hda_codec_driver_probe(struct device *dev)
goto error;
}
- err = codec->preset->patch(codec);
- if (err < 0)
- goto error_module;
+ patch = (hda_codec_patch_t)codec->preset->driver_data;
+ if (patch) {
+ err = patch(codec);
+ if (err < 0)
+ goto error_module;
+ }
err = snd_hda_codec_build_pcms(codec);
if (err < 0)
@@ -155,11 +178,10 @@ static inline bool codec_probed(struct hda_codec *codec)
static void codec_bind_module(struct hda_codec *codec)
{
#ifdef MODULE
- request_module("snd-hda-codec-id:%08x", codec->core.vendor_id);
- if (codec_probed(codec))
- return;
- request_module("snd-hda-codec-id:%04x*",
- (codec->core.vendor_id >> 16) & 0xffff);
+ char modalias[32];
+
+ snd_hdac_codec_modalias(&codec->core, modalias, sizeof(modalias));
+ request_module(modalias);
if (codec_probed(codec))
return;
#endif
@@ -251,11 +273,6 @@ int snd_hda_codec_configure(struct hda_codec *codec)
}
}
- /* audio codec should override the mixer name */
- if (codec->core.afg || !*codec->card->mixername)
- snprintf(codec->card->mixername,
- sizeof(codec->card->mixername), "%s %s",
- codec->core.vendor_name, codec->core.chip_name);
return 0;
error:
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index a249d5486889..83741887faa1 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -91,50 +91,6 @@ static int codec_exec_verb(struct hdac_device *dev, unsigned int cmd,
}
/**
- * snd_hda_codec_read - send a command and get the response
- * @codec: the HDA codec
- * @nid: NID to send the command
- * @flags: optional bit flags
- * @verb: the verb to send
- * @parm: the parameter for the verb
- *
- * Send a single command and read the corresponding response.
- *
- * Returns the obtained response value, or -1 for an error.
- */
-unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
- int flags,
- unsigned int verb, unsigned int parm)
-{
- unsigned int cmd = snd_hdac_make_cmd(&codec->core, nid, verb, parm);
- unsigned int res;
- if (snd_hdac_exec_verb(&codec->core, cmd, flags, &res))
- return -1;
- return res;
-}
-EXPORT_SYMBOL_GPL(snd_hda_codec_read);
-
-/**
- * snd_hda_codec_write - send a single command without waiting for response
- * @codec: the HDA codec
- * @nid: NID to send the command
- * @flags: optional bit flags
- * @verb: the verb to send
- * @parm: the parameter for the verb
- *
- * Send a single command without waiting for response.
- *
- * Returns 0 if successful, or a negative error code.
- */
-int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags,
- unsigned int verb, unsigned int parm)
-{
- unsigned int cmd = snd_hdac_make_cmd(&codec->core, nid, verb, parm);
- return snd_hdac_exec_verb(&codec->core, cmd, flags, NULL);
-}
-EXPORT_SYMBOL_GPL(snd_hda_codec_write);
-
-/**
* snd_hda_sequence_write - sequence writes
* @codec: the HDA codec
* @seq: VERB array to send
diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h
index 2970413f18a0..373fcad840ea 100644
--- a/sound/pci/hda/hda_codec.h
+++ b/sound/pci/hda/hda_codec.h
@@ -22,6 +22,7 @@
#define __SOUND_HDA_CODEC_H
#include <linux/kref.h>
+#include <linux/mod_devicetable.h>
#include <sound/info.h>
#include <sound/control.h>
#include <sound/pcm.h>
@@ -69,6 +70,7 @@ struct hda_bus {
unsigned int no_response_fallback:1; /* don't fallback at RIRB error */
int primary_dig_out_type; /* primary digital out PCM type */
+ unsigned int mixer_assigned; /* codec addr for mixer name */
};
/* from hdac_bus to hda_bus */
@@ -80,19 +82,21 @@ struct hda_bus {
* Known codecs have the patch to build and set up the controls/PCMs
* better than the generic parser.
*/
-struct hda_codec_preset {
- unsigned int id;
- unsigned int rev;
- const char *name;
- int (*patch)(struct hda_codec *codec);
-};
+typedef int (*hda_codec_patch_t)(struct hda_codec *);
#define HDA_CODEC_ID_GENERIC_HDMI 0x00000101
#define HDA_CODEC_ID_GENERIC 0x00000201
+#define HDA_CODEC_REV_ENTRY(_vid, _rev, _name, _patch) \
+ { .vendor_id = (_vid), .rev_id = (_rev), .name = (_name), \
+ .api_version = HDA_DEV_LEGACY, \
+ .driver_data = (unsigned long)(_patch) }
+#define HDA_CODEC_ENTRY(_vid, _name, _patch) \
+ HDA_CODEC_REV_ENTRY(_vid, 0, _name, _patch)
+
struct hda_codec_driver {
struct hdac_driver core;
- const struct hda_codec_preset *preset;
+ const struct hda_device_id *id;
};
int __hda_codec_driver_register(struct hda_codec_driver *drv, const char *name,
@@ -183,7 +187,7 @@ struct hda_codec {
u32 probe_id; /* overridden id for probing */
/* detected preset */
- const struct hda_codec_preset *preset;
+ const struct hda_device_id *preset;
const char *modelname; /* model name for preset */
/* set by patch */
@@ -297,10 +301,6 @@ struct hda_codec {
/*
* constructors
*/
-int snd_hda_bus_new(struct snd_card *card,
- const struct hdac_bus_ops *ops,
- const struct hdac_io_ops *io_ops,
- struct hda_bus **busp);
int snd_hda_codec_new(struct hda_bus *bus, struct snd_card *card,
unsigned int codec_addr, struct hda_codec **codecp);
int snd_hda_codec_configure(struct hda_codec *codec);
@@ -309,11 +309,21 @@ int snd_hda_codec_update_widgets(struct hda_codec *codec);
/*
* low level functions
*/
-unsigned int snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
+static inline unsigned int
+snd_hda_codec_read(struct hda_codec *codec, hda_nid_t nid,
int flags,
- unsigned int verb, unsigned int parm);
-int snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags,
- unsigned int verb, unsigned int parm);
+ unsigned int verb, unsigned int parm)
+{
+ return snd_hdac_codec_read(&codec->core, nid, flags, verb, parm);
+}
+
+static inline int
+snd_hda_codec_write(struct hda_codec *codec, hda_nid_t nid, int flags,
+ unsigned int verb, unsigned int parm)
+{
+ return snd_hdac_codec_write(&codec->core, nid, flags, verb, parm);
+}
+
#define snd_hda_param_read(codec, nid, param) \
snd_hdac_read_parm(&(codec)->core, nid, param)
#define snd_hda_get_sub_nodes(codec, nid, start_nid) \
@@ -453,6 +463,8 @@ void snd_hda_unlock_devices(struct hda_bus *bus);
void snd_hda_bus_reset(struct hda_bus *bus);
void snd_hda_bus_reset_codecs(struct hda_bus *bus);
+int snd_hda_codec_set_name(struct hda_codec *codec, const char *name);
+
/*
* power management
*/
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 944455997fdc..d6b93a20361b 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -1045,6 +1045,7 @@ int azx_bus_init(struct azx *chip, const char *model,
mutex_init(&bus->prepare_mutex);
bus->pci = chip->pci;
bus->modelname = model;
+ bus->mixer_assigned = -1;
bus->core.snoop = azx_snoop(chip);
if (chip->get_position[0] != azx_get_pos_lpib ||
chip->get_position[1] != azx_get_pos_lpib)
diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c
index 24f91114a32c..c6e8a651cea1 100644
--- a/sound/pci/hda/hda_generic.c
+++ b/sound/pci/hda/hda_generic.c
@@ -5877,13 +5877,14 @@ error:
return err;
}
-static const struct hda_codec_preset snd_hda_preset_generic[] = {
- { .id = HDA_CODEC_ID_GENERIC, .patch = snd_hda_parse_generic_codec },
+static const struct hda_device_id snd_hda_id_generic[] = {
+ HDA_CODEC_ENTRY(HDA_CODEC_ID_GENERIC, "Generic", snd_hda_parse_generic_codec),
{} /* terminator */
};
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_generic);
static struct hda_codec_driver generic_driver = {
- .preset = snd_hda_preset_generic,
+ .id = snd_hda_id_generic,
};
module_hda_codec_driver(generic_driver);
diff --git a/sound/pci/hda/hda_local.h b/sound/pci/hda/hda_local.h
index 4a21c2199e02..d0e066e4c985 100644
--- a/sound/pci/hda/hda_local.h
+++ b/sound/pci/hda/hda_local.h
@@ -681,12 +681,7 @@ static inline bool
snd_hda_check_power_state(struct hda_codec *codec, hda_nid_t nid,
unsigned int target_state)
{
- unsigned int state = snd_hda_codec_read(codec, nid, 0,
- AC_VERB_GET_POWER_STATE, 0);
- if (state & AC_PWRST_ERROR)
- return true;
- state = (state >> 4) & 0x0f;
- return (state == target_state);
+ return snd_hdac_check_power_state(&codec->core, nid, target_state);
}
unsigned int snd_hda_codec_eapd_power_filter(struct hda_codec *codec,
diff --git a/sound/pci/hda/hda_sysfs.c b/sound/pci/hda/hda_sysfs.c
index a6e3d9b511ab..64e0d1d81ca5 100644
--- a/sound/pci/hda/hda_sysfs.c
+++ b/sound/pci/hda/hda_sysfs.c
@@ -595,8 +595,7 @@ static void parse_model_mode(char *buf, struct hda_bus *bus,
static void parse_chip_name_mode(char *buf, struct hda_bus *bus,
struct hda_codec **codecp)
{
- kfree((*codecp)->core.chip_name);
- (*codecp)->core.chip_name = kstrdup(buf, GFP_KERNEL);
+ snd_hda_codec_set_name(*codecp, buf);
}
#define DEFINE_PARSE_ID_MODE(name) \
diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c
index c033a4ee6547..e0fb8c6d1bc2 100644
--- a/sound/pci/hda/patch_analog.c
+++ b/sound/pci/hda/patch_analog.c
@@ -1165,32 +1165,31 @@ static int patch_ad1882(struct hda_codec *codec)
/*
* patch entries
*/
-static const struct hda_codec_preset snd_hda_preset_analog[] = {
- { .id = 0x11d4184a, .name = "AD1884A", .patch = patch_ad1884 },
- { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 },
- { .id = 0x11d41883, .name = "AD1883", .patch = patch_ad1884 },
- { .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 },
- { .id = 0x11d4194a, .name = "AD1984A", .patch = patch_ad1884 },
- { .id = 0x11d4194b, .name = "AD1984B", .patch = patch_ad1884 },
- { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 },
- { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 },
- { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1884 },
- { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a },
- { .id = 0x11d41988, .name = "AD1988", .patch = patch_ad1988 },
- { .id = 0x11d4198b, .name = "AD1988B", .patch = patch_ad1988 },
- { .id = 0x11d4882a, .name = "AD1882A", .patch = patch_ad1882 },
- { .id = 0x11d4989a, .name = "AD1989A", .patch = patch_ad1988 },
- { .id = 0x11d4989b, .name = "AD1989B", .patch = patch_ad1988 },
+static const struct hda_device_id snd_hda_id_analog[] = {
+ HDA_CODEC_ENTRY(0x11d4184a, "AD1884A", patch_ad1884),
+ HDA_CODEC_ENTRY(0x11d41882, "AD1882", patch_ad1882),
+ HDA_CODEC_ENTRY(0x11d41883, "AD1883", patch_ad1884),
+ HDA_CODEC_ENTRY(0x11d41884, "AD1884", patch_ad1884),
+ HDA_CODEC_ENTRY(0x11d4194a, "AD1984A", patch_ad1884),
+ HDA_CODEC_ENTRY(0x11d4194b, "AD1984B", patch_ad1884),
+ HDA_CODEC_ENTRY(0x11d41981, "AD1981", patch_ad1981),
+ HDA_CODEC_ENTRY(0x11d41983, "AD1983", patch_ad1983),
+ HDA_CODEC_ENTRY(0x11d41984, "AD1984", patch_ad1884),
+ HDA_CODEC_ENTRY(0x11d41986, "AD1986A", patch_ad1986a),
+ HDA_CODEC_ENTRY(0x11d41988, "AD1988", patch_ad1988),
+ HDA_CODEC_ENTRY(0x11d4198b, "AD1988B", patch_ad1988),
+ HDA_CODEC_ENTRY(0x11d4882a, "AD1882A", patch_ad1882),
+ HDA_CODEC_ENTRY(0x11d4989a, "AD1989A", patch_ad1988),
+ HDA_CODEC_ENTRY(0x11d4989b, "AD1989B", patch_ad1988),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:11d4*");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_analog);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Analog Devices HD-audio codec");
static struct hda_codec_driver analog_driver = {
- .preset = snd_hda_preset_analog,
+ .id = snd_hda_id_analog,
};
module_hda_codec_driver(analog_driver);
diff --git a/sound/pci/hda/patch_ca0110.c b/sound/pci/hda/patch_ca0110.c
index 484bbf4134cd..c2d9ee9cfdc0 100644
--- a/sound/pci/hda/patch_ca0110.c
+++ b/sound/pci/hda/patch_ca0110.c
@@ -83,22 +83,19 @@ static int patch_ca0110(struct hda_codec *codec)
/*
* patch entries
*/
-static const struct hda_codec_preset snd_hda_preset_ca0110[] = {
- { .id = 0x1102000a, .name = "CA0110-IBG", .patch = patch_ca0110 },
- { .id = 0x1102000b, .name = "CA0110-IBG", .patch = patch_ca0110 },
- { .id = 0x1102000d, .name = "SB0880 X-Fi", .patch = patch_ca0110 },
+static const struct hda_device_id snd_hda_id_ca0110[] = {
+ HDA_CODEC_ENTRY(0x1102000a, "CA0110-IBG", patch_ca0110),
+ HDA_CODEC_ENTRY(0x1102000b, "CA0110-IBG", patch_ca0110),
+ HDA_CODEC_ENTRY(0x1102000d, "SB0880 X-Fi", patch_ca0110),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:1102000a");
-MODULE_ALIAS("snd-hda-codec-id:1102000b");
-MODULE_ALIAS("snd-hda-codec-id:1102000d");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_ca0110);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Creative CA0110-IBG HD-audio codec");
static struct hda_codec_driver ca0110_driver = {
- .preset = snd_hda_preset_ca0110,
+ .id = snd_hda_id_ca0110,
};
module_hda_codec_driver(ca0110_driver);
diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 186792fe226e..f8a12ca477f1 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -2673,13 +2673,13 @@ static bool dspload_wait_loaded(struct hda_codec *codec)
do {
if (dspload_is_loaded(codec)) {
- pr_info("ca0132 DOWNLOAD OK :-) DSP IS RUNNING.\n");
+ codec_info(codec, "ca0132 DSP downloaded and running\n");
return true;
}
msleep(20);
} while (time_before(jiffies, timeout));
- pr_err("ca0132 DOWNLOAD FAILED!!! DSP IS NOT RUNNING.\n");
+ codec_err(codec, "ca0132 failed to download DSP\n");
return false;
}
@@ -4375,7 +4375,7 @@ static bool ca0132_download_dsp_images(struct hda_codec *codec)
dsp_os_image = (struct dsp_image_seg *)(fw_entry->data);
if (dspload_image(codec, dsp_os_image, 0, 0, true, 0)) {
- pr_err("ca0132 dspload_image failed.\n");
+ codec_err(codec, "ca0132 DSP load image failed\n");
goto exit_download;
}
@@ -4778,18 +4778,17 @@ static int patch_ca0132(struct hda_codec *codec)
/*
* patch entries
*/
-static struct hda_codec_preset snd_hda_preset_ca0132[] = {
- { .id = 0x11020011, .name = "CA0132", .patch = patch_ca0132 },
+static struct hda_device_id snd_hda_id_ca0132[] = {
+ HDA_CODEC_ENTRY(0x11020011, "CA0132", patch_ca0132),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:11020011");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_ca0132);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Creative Sound Core3D codec");
static struct hda_codec_driver ca0132_driver = {
- .preset = snd_hda_preset_ca0132,
+ .id = snd_hda_id_ca0132,
};
module_hda_codec_driver(ca0132_driver);
diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c
index 85813de26da8..a12ae8ac0914 100644
--- a/sound/pci/hda/patch_cirrus.c
+++ b/sound/pci/hda/patch_cirrus.c
@@ -570,6 +570,7 @@ static struct cs_spec *cs_alloc_spec(struct hda_codec *codec, int vendor_nid)
return NULL;
codec->spec = spec;
spec->vendor_nid = vendor_nid;
+ codec->power_save_node = 1;
snd_hda_gen_spec_init(&spec->gen);
return spec;
@@ -1200,26 +1201,21 @@ static int patch_cs4213(struct hda_codec *codec)
/*
* patch entries
*/
-static const struct hda_codec_preset snd_hda_preset_cirrus[] = {
- { .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x },
- { .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x },
- { .id = 0x10134208, .name = "CS4208", .patch = patch_cs4208 },
- { .id = 0x10134210, .name = "CS4210", .patch = patch_cs4210 },
- { .id = 0x10134213, .name = "CS4213", .patch = patch_cs4213 },
+static const struct hda_device_id snd_hda_id_cirrus[] = {
+ HDA_CODEC_ENTRY(0x10134206, "CS4206", patch_cs420x),
+ HDA_CODEC_ENTRY(0x10134207, "CS4207", patch_cs420x),
+ HDA_CODEC_ENTRY(0x10134208, "CS4208", patch_cs4208),
+ HDA_CODEC_ENTRY(0x10134210, "CS4210", patch_cs4210),
+ HDA_CODEC_ENTRY(0x10134213, "CS4213", patch_cs4213),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:10134206");
-MODULE_ALIAS("snd-hda-codec-id:10134207");
-MODULE_ALIAS("snd-hda-codec-id:10134208");
-MODULE_ALIAS("snd-hda-codec-id:10134210");
-MODULE_ALIAS("snd-hda-codec-id:10134213");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_cirrus);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Cirrus Logic HD-audio codec");
static struct hda_codec_driver cirrus_driver = {
- .preset = snd_hda_preset_cirrus,
+ .id = snd_hda_id_cirrus,
};
module_hda_codec_driver(cirrus_driver);
diff --git a/sound/pci/hda/patch_cmedia.c b/sound/pci/hda/patch_cmedia.c
index f5ed078710f8..1b2195dd2b26 100644
--- a/sound/pci/hda/patch_cmedia.c
+++ b/sound/pci/hda/patch_cmedia.c
@@ -123,22 +123,19 @@ static int patch_cmi8888(struct hda_codec *codec)
/*
* patch entries
*/
-static const struct hda_codec_preset snd_hda_preset_cmedia[] = {
- { .id = 0x13f68888, .name = "CMI8888", .patch = patch_cmi8888 },
- { .id = 0x13f69880, .name = "CMI9880", .patch = patch_cmi9880 },
- { .id = 0x434d4980, .name = "CMI9880", .patch = patch_cmi9880 },
+static const struct hda_device_id snd_hda_id_cmedia[] = {
+ HDA_CODEC_ENTRY(0x13f68888, "CMI8888", patch_cmi8888),
+ HDA_CODEC_ENTRY(0x13f69880, "CMI9880", patch_cmi9880),
+ HDA_CODEC_ENTRY(0x434d4980, "CMI9880", patch_cmi9880),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:13f68888");
-MODULE_ALIAS("snd-hda-codec-id:13f69880");
-MODULE_ALIAS("snd-hda-codec-id:434d4980");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_cmedia);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("C-Media HD-audio codec");
static struct hda_codec_driver cmedia_driver = {
- .preset = snd_hda_preset_cmedia,
+ .id = snd_hda_id_cmedia,
};
module_hda_codec_driver(cmedia_driver);
diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c
index 2f0ec7c45fc7..c8b8ef5246a6 100644
--- a/sound/pci/hda/patch_conexant.c
+++ b/sound/pci/hda/patch_conexant.c
@@ -954,100 +954,44 @@ static int patch_conexant_auto(struct hda_codec *codec)
/*
*/
-static const struct hda_codec_preset snd_hda_preset_conexant[] = {
- { .id = 0x14f15045, .name = "CX20549 (Venice)",
- .patch = patch_conexant_auto },
- { .id = 0x14f15047, .name = "CX20551 (Waikiki)",
- .patch = patch_conexant_auto },
- { .id = 0x14f15051, .name = "CX20561 (Hermosa)",
- .patch = patch_conexant_auto },
- { .id = 0x14f15066, .name = "CX20582 (Pebble)",
- .patch = patch_conexant_auto },
- { .id = 0x14f15067, .name = "CX20583 (Pebble HSF)",
- .patch = patch_conexant_auto },
- { .id = 0x14f15068, .name = "CX20584",
- .patch = patch_conexant_auto },
- { .id = 0x14f15069, .name = "CX20585",
- .patch = patch_conexant_auto },
- { .id = 0x14f1506c, .name = "CX20588",
- .patch = patch_conexant_auto },
- { .id = 0x14f1506e, .name = "CX20590",
- .patch = patch_conexant_auto },
- { .id = 0x14f15097, .name = "CX20631",
- .patch = patch_conexant_auto },
- { .id = 0x14f15098, .name = "CX20632",
- .patch = patch_conexant_auto },
- { .id = 0x14f150a1, .name = "CX20641",
- .patch = patch_conexant_auto },
- { .id = 0x14f150a2, .name = "CX20642",
- .patch = patch_conexant_auto },
- { .id = 0x14f150ab, .name = "CX20651",
- .patch = patch_conexant_auto },
- { .id = 0x14f150ac, .name = "CX20652",
- .patch = patch_conexant_auto },
- { .id = 0x14f150b8, .name = "CX20664",
- .patch = patch_conexant_auto },
- { .id = 0x14f150b9, .name = "CX20665",
- .patch = patch_conexant_auto },
- { .id = 0x14f150f1, .name = "CX20721",
- .patch = patch_conexant_auto },
- { .id = 0x14f150f2, .name = "CX20722",
- .patch = patch_conexant_auto },
- { .id = 0x14f150f3, .name = "CX20723",
- .patch = patch_conexant_auto },
- { .id = 0x14f150f4, .name = "CX20724",
- .patch = patch_conexant_auto },
- { .id = 0x14f1510f, .name = "CX20751/2",
- .patch = patch_conexant_auto },
- { .id = 0x14f15110, .name = "CX20751/2",
- .patch = patch_conexant_auto },
- { .id = 0x14f15111, .name = "CX20753/4",
- .patch = patch_conexant_auto },
- { .id = 0x14f15113, .name = "CX20755",
- .patch = patch_conexant_auto },
- { .id = 0x14f15114, .name = "CX20756",
- .patch = patch_conexant_auto },
- { .id = 0x14f15115, .name = "CX20757",
- .patch = patch_conexant_auto },
- { .id = 0x14f151d7, .name = "CX20952",
- .patch = patch_conexant_auto },
+static const struct hda_device_id snd_hda_id_conexant[] = {
+ HDA_CODEC_ENTRY(0x14f15045, "CX20549 (Venice)", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15047, "CX20551 (Waikiki)", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15051, "CX20561 (Hermosa)", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15066, "CX20582 (Pebble)", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15067, "CX20583 (Pebble HSF)", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15068, "CX20584", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15069, "CX20585", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f1506c, "CX20588", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f1506e, "CX20590", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15097, "CX20631", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15098, "CX20632", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150a1, "CX20641", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150a2, "CX20642", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150ab, "CX20651", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150ac, "CX20652", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150b8, "CX20664", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150b9, "CX20665", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150f1, "CX20721", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150f2, "CX20722", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150f3, "CX20723", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f150f4, "CX20724", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f1510f, "CX20751/2", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15110, "CX20751/2", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15111, "CX20753/4", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15113, "CX20755", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15114, "CX20756", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f15115, "CX20757", patch_conexant_auto),
+ HDA_CODEC_ENTRY(0x14f151d7, "CX20952", patch_conexant_auto),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:14f15045");
-MODULE_ALIAS("snd-hda-codec-id:14f15047");
-MODULE_ALIAS("snd-hda-codec-id:14f15051");
-MODULE_ALIAS("snd-hda-codec-id:14f15066");
-MODULE_ALIAS("snd-hda-codec-id:14f15067");
-MODULE_ALIAS("snd-hda-codec-id:14f15068");
-MODULE_ALIAS("snd-hda-codec-id:14f15069");
-MODULE_ALIAS("snd-hda-codec-id:14f1506c");
-MODULE_ALIAS("snd-hda-codec-id:14f1506e");
-MODULE_ALIAS("snd-hda-codec-id:14f15097");
-MODULE_ALIAS("snd-hda-codec-id:14f15098");
-MODULE_ALIAS("snd-hda-codec-id:14f150a1");
-MODULE_ALIAS("snd-hda-codec-id:14f150a2");
-MODULE_ALIAS("snd-hda-codec-id:14f150ab");
-MODULE_ALIAS("snd-hda-codec-id:14f150ac");
-MODULE_ALIAS("snd-hda-codec-id:14f150b8");
-MODULE_ALIAS("snd-hda-codec-id:14f150b9");
-MODULE_ALIAS("snd-hda-codec-id:14f150f1");
-MODULE_ALIAS("snd-hda-codec-id:14f150f2");
-MODULE_ALIAS("snd-hda-codec-id:14f150f3");
-MODULE_ALIAS("snd-hda-codec-id:14f150f4");
-MODULE_ALIAS("snd-hda-codec-id:14f1510f");
-MODULE_ALIAS("snd-hda-codec-id:14f15110");
-MODULE_ALIAS("snd-hda-codec-id:14f15111");
-MODULE_ALIAS("snd-hda-codec-id:14f15113");
-MODULE_ALIAS("snd-hda-codec-id:14f15114");
-MODULE_ALIAS("snd-hda-codec-id:14f15115");
-MODULE_ALIAS("snd-hda-codec-id:14f151d7");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_conexant);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Conexant HD-audio codec");
static struct hda_codec_driver conexant_driver = {
- .preset = snd_hda_preset_conexant,
+ .id = snd_hda_id_conexant,
};
module_hda_codec_driver(conexant_driver);
diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c
index acbfbe087ee8..f503a883bef3 100644
--- a/sound/pci/hda/patch_hdmi.c
+++ b/sound/pci/hda/patch_hdmi.c
@@ -1775,6 +1775,16 @@ static bool check_non_pcm_per_cvt(struct hda_codec *codec, hda_nid_t cvt_nid)
return non_pcm;
}
+/* There is a fixed mapping between audio pin node and display port
+ * on current Intel platforms:
+ * Pin Widget 5 - PORT B (port = 1 in i915 driver)
+ * Pin Widget 6 - PORT C (port = 2 in i915 driver)
+ * Pin Widget 7 - PORT D (port = 3 in i915 driver)
+ */
+static int intel_pin2port(hda_nid_t pin_nid)
+{
+ return pin_nid - 4;
+}
/*
* HDMI callbacks
@@ -1791,6 +1801,8 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
int pin_idx = hinfo_to_pin_index(codec, hinfo);
struct hdmi_spec_per_pin *per_pin = get_pin(spec, pin_idx);
hda_nid_t pin_nid = per_pin->pin_nid;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct i915_audio_component *acomp = codec->bus->core.audio_component;
bool non_pcm;
int pinctl;
@@ -1807,6 +1819,13 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo,
intel_not_share_assigned_cvt(codec, pin_nid, per_pin->mux_idx);
}
+ /* Call sync_audio_rate to set the N/CTS/M manually if necessary */
+ /* Todo: add DP1.2 MST audio support later */
+ if (acomp && acomp->ops && acomp->ops->sync_audio_rate)
+ acomp->ops->sync_audio_rate(acomp->dev,
+ intel_pin2port(pin_nid),
+ runtime->rate);
+
non_pcm = check_non_pcm_per_cvt(codec, cvt_nid);
mutex_lock(&per_pin->lock);
per_pin->channels = substream->runtime->channels;
@@ -2561,7 +2580,7 @@ static int simple_playback_pcm_open(struct hda_pcm_stream *hinfo,
struct hdmi_spec *spec = codec->spec;
struct snd_pcm_hw_constraint_list *hw_constraints_channels = NULL;
- switch (codec->preset->id) {
+ switch (codec->preset->vendor_id) {
case 0x10de0002:
case 0x10de0003:
case 0x10de0005:
@@ -2879,7 +2898,7 @@ static int nvhdmi_7x_8ch_build_controls(struct hda_codec *codec)
snd_pcm_alt_chmaps, 8, 0, &chmap);
if (err < 0)
return err;
- switch (codec->preset->id) {
+ switch (codec->preset->vendor_id) {
case 0x10de0002:
case 0x10de0003:
case 0x10de0005:
@@ -3487,138 +3506,77 @@ static int patch_via_hdmi(struct hda_codec *codec)
/*
* patch entries
*/
-static const struct hda_codec_preset snd_hda_preset_hdmi[] = {
-{ .id = 0x1002793c, .name = "RS600 HDMI", .patch = patch_atihdmi },
-{ .id = 0x10027919, .name = "RS600 HDMI", .patch = patch_atihdmi },
-{ .id = 0x1002791a, .name = "RS690/780 HDMI", .patch = patch_atihdmi },
-{ .id = 0x1002aa01, .name = "R6xx HDMI", .patch = patch_atihdmi },
-{ .id = 0x10951390, .name = "SiI1390 HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x10951392, .name = "SiI1392 HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x17e80047, .name = "Chrontel HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x10de0002, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x },
-{ .id = 0x10de0003, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x },
-{ .id = 0x10de0005, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x },
-{ .id = 0x10de0006, .name = "MCP77/78 HDMI", .patch = patch_nvhdmi_8ch_7x },
-{ .id = 0x10de0007, .name = "MCP79/7A HDMI", .patch = patch_nvhdmi_8ch_7x },
-{ .id = 0x10de000a, .name = "GPU 0a HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de000b, .name = "GPU 0b HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de000c, .name = "MCP89 HDMI", .patch = patch_nvhdmi },
-{ .id = 0x10de000d, .name = "GPU 0d HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0010, .name = "GPU 10 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0011, .name = "GPU 11 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0012, .name = "GPU 12 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0013, .name = "GPU 13 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0014, .name = "GPU 14 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0015, .name = "GPU 15 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0016, .name = "GPU 16 HDMI/DP", .patch = patch_nvhdmi },
+static const struct hda_device_id snd_hda_id_hdmi[] = {
+HDA_CODEC_ENTRY(0x1002793c, "RS600 HDMI", patch_atihdmi),
+HDA_CODEC_ENTRY(0x10027919, "RS600 HDMI", patch_atihdmi),
+HDA_CODEC_ENTRY(0x1002791a, "RS690/780 HDMI", patch_atihdmi),
+HDA_CODEC_ENTRY(0x1002aa01, "R6xx HDMI", patch_atihdmi),
+HDA_CODEC_ENTRY(0x10951390, "SiI1390 HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x10951392, "SiI1392 HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x17e80047, "Chrontel HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x10de0002, "MCP77/78 HDMI", patch_nvhdmi_8ch_7x),
+HDA_CODEC_ENTRY(0x10de0003, "MCP77/78 HDMI", patch_nvhdmi_8ch_7x),
+HDA_CODEC_ENTRY(0x10de0005, "MCP77/78 HDMI", patch_nvhdmi_8ch_7x),
+HDA_CODEC_ENTRY(0x10de0006, "MCP77/78 HDMI", patch_nvhdmi_8ch_7x),
+HDA_CODEC_ENTRY(0x10de0007, "MCP79/7A HDMI", patch_nvhdmi_8ch_7x),
+HDA_CODEC_ENTRY(0x10de000a, "GPU 0a HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de000b, "GPU 0b HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de000c, "MCP89 HDMI", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de000d, "GPU 0d HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0010, "GPU 10 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0011, "GPU 11 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0012, "GPU 12 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0013, "GPU 13 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0014, "GPU 14 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0015, "GPU 15 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0016, "GPU 16 HDMI/DP", patch_nvhdmi),
/* 17 is known to be absent */
-{ .id = 0x10de0018, .name = "GPU 18 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0019, .name = "GPU 19 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0020, .name = "Tegra30 HDMI", .patch = patch_tegra_hdmi },
-{ .id = 0x10de0022, .name = "Tegra114 HDMI", .patch = patch_tegra_hdmi },
-{ .id = 0x10de0028, .name = "Tegra124 HDMI", .patch = patch_tegra_hdmi },
-{ .id = 0x10de0029, .name = "Tegra210 HDMI/DP", .patch = patch_tegra_hdmi },
-{ .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0043, .name = "GPU 43 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0044, .name = "GPU 44 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0051, .name = "GPU 51 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0060, .name = "GPU 60 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0067, .name = "MCP67 HDMI", .patch = patch_nvhdmi_2ch },
-{ .id = 0x10de0070, .name = "GPU 70 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0071, .name = "GPU 71 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de0072, .name = "GPU 72 HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de007d, .name = "GPU 7d HDMI/DP", .patch = patch_nvhdmi },
-{ .id = 0x10de8001, .name = "MCP73 HDMI", .patch = patch_nvhdmi_2ch },
-{ .id = 0x11069f80, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi },
-{ .id = 0x11069f81, .name = "VX900 HDMI/DP", .patch = patch_via_hdmi },
-{ .id = 0x11069f84, .name = "VX11 HDMI/DP", .patch = patch_generic_hdmi },
-{ .id = 0x11069f85, .name = "VX11 HDMI/DP", .patch = patch_generic_hdmi },
-{ .id = 0x80860054, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862801, .name = "Bearlake HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862802, .name = "Cantiga HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862803, .name = "Eaglelake HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862804, .name = "IbexPeak HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862805, .name = "CougarPoint HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862806, .name = "PantherPoint HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862807, .name = "Haswell HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862808, .name = "Broadwell HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862809, .name = "Skylake HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x8086280a, .name = "Broxton HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862880, .name = "CedarTrail HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862882, .name = "Valleyview2 HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x80862883, .name = "Braswell HDMI", .patch = patch_generic_hdmi },
-{ .id = 0x808629fb, .name = "Crestline HDMI", .patch = patch_generic_hdmi },
+HDA_CODEC_ENTRY(0x10de0018, "GPU 18 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0019, "GPU 19 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de001a, "GPU 1a HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de001b, "GPU 1b HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de001c, "GPU 1c HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0020, "Tegra30 HDMI", patch_tegra_hdmi),
+HDA_CODEC_ENTRY(0x10de0022, "Tegra114 HDMI", patch_tegra_hdmi),
+HDA_CODEC_ENTRY(0x10de0028, "Tegra124 HDMI", patch_tegra_hdmi),
+HDA_CODEC_ENTRY(0x10de0029, "Tegra210 HDMI/DP", patch_tegra_hdmi),
+HDA_CODEC_ENTRY(0x10de0040, "GPU 40 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0041, "GPU 41 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0042, "GPU 42 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0043, "GPU 43 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0044, "GPU 44 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0051, "GPU 51 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0060, "GPU 60 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0067, "MCP67 HDMI", patch_nvhdmi_2ch),
+HDA_CODEC_ENTRY(0x10de0070, "GPU 70 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0071, "GPU 71 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de0072, "GPU 72 HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de007d, "GPU 7d HDMI/DP", patch_nvhdmi),
+HDA_CODEC_ENTRY(0x10de8001, "MCP73 HDMI", patch_nvhdmi_2ch),
+HDA_CODEC_ENTRY(0x11069f80, "VX900 HDMI/DP", patch_via_hdmi),
+HDA_CODEC_ENTRY(0x11069f81, "VX900 HDMI/DP", patch_via_hdmi),
+HDA_CODEC_ENTRY(0x11069f84, "VX11 HDMI/DP", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x11069f85, "VX11 HDMI/DP", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80860054, "IbexPeak HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862801, "Bearlake HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862802, "Cantiga HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862803, "Eaglelake HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862804, "IbexPeak HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862805, "CougarPoint HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862806, "PantherPoint HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862807, "Haswell HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862808, "Broadwell HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862809, "Skylake HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x8086280a, "Broxton HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_generic_hdmi),
+HDA_CODEC_ENTRY(0x808629fb, "Crestline HDMI", patch_generic_hdmi),
/* special ID for generic HDMI */
-{ .id = HDA_CODEC_ID_GENERIC_HDMI, .patch = patch_generic_hdmi },
+HDA_CODEC_ENTRY(HDA_CODEC_ID_GENERIC_HDMI, "Generic HDMI", patch_generic_hdmi),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:1002793c");
-MODULE_ALIAS("snd-hda-codec-id:10027919");
-MODULE_ALIAS("snd-hda-codec-id:1002791a");
-MODULE_ALIAS("snd-hda-codec-id:1002aa01");
-MODULE_ALIAS("snd-hda-codec-id:10951390");
-MODULE_ALIAS("snd-hda-codec-id:10951392");
-MODULE_ALIAS("snd-hda-codec-id:10de0002");
-MODULE_ALIAS("snd-hda-codec-id:10de0003");
-MODULE_ALIAS("snd-hda-codec-id:10de0005");
-MODULE_ALIAS("snd-hda-codec-id:10de0006");
-MODULE_ALIAS("snd-hda-codec-id:10de0007");
-MODULE_ALIAS("snd-hda-codec-id:10de000a");
-MODULE_ALIAS("snd-hda-codec-id:10de000b");
-MODULE_ALIAS("snd-hda-codec-id:10de000c");
-MODULE_ALIAS("snd-hda-codec-id:10de000d");
-MODULE_ALIAS("snd-hda-codec-id:10de0010");
-MODULE_ALIAS("snd-hda-codec-id:10de0011");
-MODULE_ALIAS("snd-hda-codec-id:10de0012");
-MODULE_ALIAS("snd-hda-codec-id:10de0013");
-MODULE_ALIAS("snd-hda-codec-id:10de0014");
-MODULE_ALIAS("snd-hda-codec-id:10de0015");
-MODULE_ALIAS("snd-hda-codec-id:10de0016");
-MODULE_ALIAS("snd-hda-codec-id:10de0018");
-MODULE_ALIAS("snd-hda-codec-id:10de0019");
-MODULE_ALIAS("snd-hda-codec-id:10de001a");
-MODULE_ALIAS("snd-hda-codec-id:10de001b");
-MODULE_ALIAS("snd-hda-codec-id:10de001c");
-MODULE_ALIAS("snd-hda-codec-id:10de0028");
-MODULE_ALIAS("snd-hda-codec-id:10de0040");
-MODULE_ALIAS("snd-hda-codec-id:10de0041");
-MODULE_ALIAS("snd-hda-codec-id:10de0042");
-MODULE_ALIAS("snd-hda-codec-id:10de0043");
-MODULE_ALIAS("snd-hda-codec-id:10de0044");
-MODULE_ALIAS("snd-hda-codec-id:10de0051");
-MODULE_ALIAS("snd-hda-codec-id:10de0060");
-MODULE_ALIAS("snd-hda-codec-id:10de0067");
-MODULE_ALIAS("snd-hda-codec-id:10de0070");
-MODULE_ALIAS("snd-hda-codec-id:10de0071");
-MODULE_ALIAS("snd-hda-codec-id:10de0072");
-MODULE_ALIAS("snd-hda-codec-id:10de007d");
-MODULE_ALIAS("snd-hda-codec-id:10de8001");
-MODULE_ALIAS("snd-hda-codec-id:11069f80");
-MODULE_ALIAS("snd-hda-codec-id:11069f81");
-MODULE_ALIAS("snd-hda-codec-id:11069f84");
-MODULE_ALIAS("snd-hda-codec-id:11069f85");
-MODULE_ALIAS("snd-hda-codec-id:17e80047");
-MODULE_ALIAS("snd-hda-codec-id:80860054");
-MODULE_ALIAS("snd-hda-codec-id:80862801");
-MODULE_ALIAS("snd-hda-codec-id:80862802");
-MODULE_ALIAS("snd-hda-codec-id:80862803");
-MODULE_ALIAS("snd-hda-codec-id:80862804");
-MODULE_ALIAS("snd-hda-codec-id:80862805");
-MODULE_ALIAS("snd-hda-codec-id:80862806");
-MODULE_ALIAS("snd-hda-codec-id:80862807");
-MODULE_ALIAS("snd-hda-codec-id:80862808");
-MODULE_ALIAS("snd-hda-codec-id:80862809");
-MODULE_ALIAS("snd-hda-codec-id:8086280a");
-MODULE_ALIAS("snd-hda-codec-id:80862880");
-MODULE_ALIAS("snd-hda-codec-id:80862882");
-MODULE_ALIAS("snd-hda-codec-id:80862883");
-MODULE_ALIAS("snd-hda-codec-id:808629fb");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_hdmi);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("HDMI HD-audio codec");
@@ -3627,7 +3585,7 @@ MODULE_ALIAS("snd-hda-codec-nvhdmi");
MODULE_ALIAS("snd-hda-codec-atihdmi");
static struct hda_codec_driver hdmi_driver = {
- .preset = snd_hda_preset_hdmi,
+ .id = snd_hda_id_hdmi,
};
module_hda_codec_driver(hdmi_driver);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 720a9fb32e20..2f7b065f9ac4 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -822,17 +822,7 @@ static const struct hda_codec_ops alc_patch_ops = {
};
-/* replace the codec chip_name with the given string */
-static int alc_codec_rename(struct hda_codec *codec, const char *name)
-{
- kfree(codec->core.chip_name);
- codec->core.chip_name = kstrdup(name, GFP_KERNEL);
- if (!codec->core.chip_name) {
- alc_free(codec);
- return -ENOMEM;
- }
- return 0;
-}
+#define alc_codec_rename(codec, name) snd_hda_codec_set_name(codec, name)
/*
* Rename codecs appropriately from COEF value or subvendor id
@@ -6640,78 +6630,70 @@ static int patch_alc680(struct hda_codec *codec)
/*
* patch entries
*/
-static const struct hda_codec_preset snd_hda_preset_realtek[] = {
- { .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 },
- { .id = 0x10ec0231, .name = "ALC231", .patch = patch_alc269 },
- { .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 },
- { .id = 0x10ec0235, .name = "ALC233", .patch = patch_alc269 },
- { .id = 0x10ec0255, .name = "ALC255", .patch = patch_alc269 },
- { .id = 0x10ec0256, .name = "ALC256", .patch = patch_alc269 },
- { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 },
- { .id = 0x10ec0262, .name = "ALC262", .patch = patch_alc262 },
- { .id = 0x10ec0267, .name = "ALC267", .patch = patch_alc268 },
- { .id = 0x10ec0268, .name = "ALC268", .patch = patch_alc268 },
- { .id = 0x10ec0269, .name = "ALC269", .patch = patch_alc269 },
- { .id = 0x10ec0270, .name = "ALC270", .patch = patch_alc269 },
- { .id = 0x10ec0272, .name = "ALC272", .patch = patch_alc662 },
- { .id = 0x10ec0275, .name = "ALC275", .patch = patch_alc269 },
- { .id = 0x10ec0276, .name = "ALC276", .patch = patch_alc269 },
- { .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 },
- { .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 },
- { .id = 0x10ec0283, .name = "ALC283", .patch = patch_alc269 },
- { .id = 0x10ec0284, .name = "ALC284", .patch = patch_alc269 },
- { .id = 0x10ec0285, .name = "ALC285", .patch = patch_alc269 },
- { .id = 0x10ec0286, .name = "ALC286", .patch = patch_alc269 },
- { .id = 0x10ec0288, .name = "ALC288", .patch = patch_alc269 },
- { .id = 0x10ec0290, .name = "ALC290", .patch = patch_alc269 },
- { .id = 0x10ec0292, .name = "ALC292", .patch = patch_alc269 },
- { .id = 0x10ec0293, .name = "ALC293", .patch = patch_alc269 },
- { .id = 0x10ec0298, .name = "ALC298", .patch = patch_alc269 },
- { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660",
- .patch = patch_alc861 },
- { .id = 0x10ec0660, .name = "ALC660-VD", .patch = patch_alc861vd },
- { .id = 0x10ec0861, .name = "ALC861", .patch = patch_alc861 },
- { .id = 0x10ec0862, .name = "ALC861-VD", .patch = patch_alc861vd },
- { .id = 0x10ec0662, .rev = 0x100002, .name = "ALC662 rev2",
- .patch = patch_alc882 },
- { .id = 0x10ec0662, .rev = 0x100101, .name = "ALC662 rev1",
- .patch = patch_alc662 },
- { .id = 0x10ec0662, .rev = 0x100300, .name = "ALC662 rev3",
- .patch = patch_alc662 },
- { .id = 0x10ec0663, .name = "ALC663", .patch = patch_alc662 },
- { .id = 0x10ec0665, .name = "ALC665", .patch = patch_alc662 },
- { .id = 0x10ec0667, .name = "ALC667", .patch = patch_alc662 },
- { .id = 0x10ec0668, .name = "ALC668", .patch = patch_alc662 },
- { .id = 0x10ec0670, .name = "ALC670", .patch = patch_alc662 },
- { .id = 0x10ec0671, .name = "ALC671", .patch = patch_alc662 },
- { .id = 0x10ec0680, .name = "ALC680", .patch = patch_alc680 },
- { .id = 0x10ec0867, .name = "ALC891", .patch = patch_alc882 },
- { .id = 0x10ec0880, .name = "ALC880", .patch = patch_alc880 },
- { .id = 0x10ec0882, .name = "ALC882", .patch = patch_alc882 },
- { .id = 0x10ec0883, .name = "ALC883", .patch = patch_alc882 },
- { .id = 0x10ec0885, .rev = 0x100101, .name = "ALC889A",
- .patch = patch_alc882 },
- { .id = 0x10ec0885, .rev = 0x100103, .name = "ALC889A",
- .patch = patch_alc882 },
- { .id = 0x10ec0885, .name = "ALC885", .patch = patch_alc882 },
- { .id = 0x10ec0887, .name = "ALC887", .patch = patch_alc882 },
- { .id = 0x10ec0888, .rev = 0x100101, .name = "ALC1200",
- .patch = patch_alc882 },
- { .id = 0x10ec0888, .name = "ALC888", .patch = patch_alc882 },
- { .id = 0x10ec0889, .name = "ALC889", .patch = patch_alc882 },
- { .id = 0x10ec0892, .name = "ALC892", .patch = patch_alc662 },
- { .id = 0x10ec0899, .name = "ALC898", .patch = patch_alc882 },
- { .id = 0x10ec0900, .name = "ALC1150", .patch = patch_alc882 },
+static const struct hda_device_id snd_hda_id_realtek[] = {
+ HDA_CODEC_ENTRY(0x10ec0221, "ALC221", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0231, "ALC231", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0233, "ALC233", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0235, "ALC233", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0255, "ALC255", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0256, "ALC256", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0260, "ALC260", patch_alc260),
+ HDA_CODEC_ENTRY(0x10ec0262, "ALC262", patch_alc262),
+ HDA_CODEC_ENTRY(0x10ec0267, "ALC267", patch_alc268),
+ HDA_CODEC_ENTRY(0x10ec0268, "ALC268", patch_alc268),
+ HDA_CODEC_ENTRY(0x10ec0269, "ALC269", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0270, "ALC270", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0272, "ALC272", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0275, "ALC275", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0276, "ALC276", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0280, "ALC280", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0282, "ALC282", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0283, "ALC283", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0284, "ALC284", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0285, "ALC285", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0286, "ALC286", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0288, "ALC288", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0290, "ALC290", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0292, "ALC292", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0293, "ALC293", patch_alc269),
+ HDA_CODEC_ENTRY(0x10ec0298, "ALC298", patch_alc269),
+ HDA_CODEC_REV_ENTRY(0x10ec0861, 0x100340, "ALC660", patch_alc861),
+ HDA_CODEC_ENTRY(0x10ec0660, "ALC660-VD", patch_alc861vd),
+ HDA_CODEC_ENTRY(0x10ec0861, "ALC861", patch_alc861),
+ HDA_CODEC_ENTRY(0x10ec0862, "ALC861-VD", patch_alc861vd),
+ HDA_CODEC_REV_ENTRY(0x10ec0662, 0x100002, "ALC662 rev2", patch_alc882),
+ HDA_CODEC_REV_ENTRY(0x10ec0662, 0x100101, "ALC662 rev1", patch_alc662),
+ HDA_CODEC_REV_ENTRY(0x10ec0662, 0x100300, "ALC662 rev3", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0663, "ALC663", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0665, "ALC665", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0667, "ALC667", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0668, "ALC668", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0670, "ALC670", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0671, "ALC671", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0680, "ALC680", patch_alc680),
+ HDA_CODEC_ENTRY(0x10ec0867, "ALC891", patch_alc882),
+ HDA_CODEC_ENTRY(0x10ec0880, "ALC880", patch_alc880),
+ HDA_CODEC_ENTRY(0x10ec0882, "ALC882", patch_alc882),
+ HDA_CODEC_ENTRY(0x10ec0883, "ALC883", patch_alc882),
+ HDA_CODEC_REV_ENTRY(0x10ec0885, 0x100101, "ALC889A", patch_alc882),
+ HDA_CODEC_REV_ENTRY(0x10ec0885, 0x100103, "ALC889A", patch_alc882),
+ HDA_CODEC_ENTRY(0x10ec0885, "ALC885", patch_alc882),
+ HDA_CODEC_ENTRY(0x10ec0887, "ALC887", patch_alc882),
+ HDA_CODEC_REV_ENTRY(0x10ec0888, 0x100101, "ALC1200", patch_alc882),
+ HDA_CODEC_ENTRY(0x10ec0888, "ALC888", patch_alc882),
+ HDA_CODEC_ENTRY(0x10ec0889, "ALC889", patch_alc882),
+ HDA_CODEC_ENTRY(0x10ec0892, "ALC892", patch_alc662),
+ HDA_CODEC_ENTRY(0x10ec0899, "ALC898", patch_alc882),
+ HDA_CODEC_ENTRY(0x10ec0900, "ALC1150", patch_alc882),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:10ec*");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_realtek);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Realtek HD-audio codec");
static struct hda_codec_driver realtek_driver = {
- .preset = snd_hda_preset_realtek,
+ .id = snd_hda_id_realtek,
};
module_hda_codec_driver(realtek_driver);
diff --git a/sound/pci/hda/patch_si3054.c b/sound/pci/hda/patch_si3054.c
index 5104bebb2286..ffda38c45509 100644
--- a/sound/pci/hda/patch_si3054.c
+++ b/sound/pci/hda/patch_si3054.c
@@ -289,41 +289,30 @@ static int patch_si3054(struct hda_codec *codec)
/*
* patch entries
*/
-static const struct hda_codec_preset snd_hda_preset_si3054[] = {
- { .id = 0x163c3055, .name = "Si3054", .patch = patch_si3054 },
- { .id = 0x163c3155, .name = "Si3054", .patch = patch_si3054 },
- { .id = 0x11c13026, .name = "Si3054", .patch = patch_si3054 },
- { .id = 0x11c13055, .name = "Si3054", .patch = patch_si3054 },
- { .id = 0x11c13155, .name = "Si3054", .patch = patch_si3054 },
- { .id = 0x10573055, .name = "Si3054", .patch = patch_si3054 },
- { .id = 0x10573057, .name = "Si3054", .patch = patch_si3054 },
- { .id = 0x10573155, .name = "Si3054", .patch = patch_si3054 },
+static const struct hda_device_id snd_hda_id_si3054[] = {
+ HDA_CODEC_ENTRY(0x163c3055, "Si3054", patch_si3054),
+ HDA_CODEC_ENTRY(0x163c3155, "Si3054", patch_si3054),
+ HDA_CODEC_ENTRY(0x11c13026, "Si3054", patch_si3054),
+ HDA_CODEC_ENTRY(0x11c13055, "Si3054", patch_si3054),
+ HDA_CODEC_ENTRY(0x11c13155, "Si3054", patch_si3054),
+ HDA_CODEC_ENTRY(0x10573055, "Si3054", patch_si3054),
+ HDA_CODEC_ENTRY(0x10573057, "Si3054", patch_si3054),
+ HDA_CODEC_ENTRY(0x10573155, "Si3054", patch_si3054),
/* VIA HDA on Clevo m540 */
- { .id = 0x11063288, .name = "Si3054", .patch = patch_si3054 },
+ HDA_CODEC_ENTRY(0x11063288, "Si3054", patch_si3054),
/* Asus A8J Modem (SM56) */
- { .id = 0x15433155, .name = "Si3054", .patch = patch_si3054 },
+ HDA_CODEC_ENTRY(0x15433155, "Si3054", patch_si3054),
/* LG LW20 modem */
- { .id = 0x18540018, .name = "Si3054", .patch = patch_si3054 },
+ HDA_CODEC_ENTRY(0x18540018, "Si3054", patch_si3054),
{}
};
-
-MODULE_ALIAS("snd-hda-codec-id:163c3055");
-MODULE_ALIAS("snd-hda-codec-id:163c3155");
-MODULE_ALIAS("snd-hda-codec-id:11c13026");
-MODULE_ALIAS("snd-hda-codec-id:11c13055");
-MODULE_ALIAS("snd-hda-codec-id:11c13155");
-MODULE_ALIAS("snd-hda-codec-id:10573055");
-MODULE_ALIAS("snd-hda-codec-id:10573057");
-MODULE_ALIAS("snd-hda-codec-id:10573155");
-MODULE_ALIAS("snd-hda-codec-id:11063288");
-MODULE_ALIAS("snd-hda-codec-id:15433155");
-MODULE_ALIAS("snd-hda-codec-id:18540018");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_si3054);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Si3054 HD-audio modem codec");
static struct hda_codec_driver si3054_driver = {
- .preset = snd_hda_preset_si3054,
+ .id = snd_hda_id_si3054,
};
module_hda_codec_driver(si3054_driver);
diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c
index def5cc8dff02..08a0f6a35cca 100644
--- a/sound/pci/hda/patch_sigmatel.c
+++ b/sound/pci/hda/patch_sigmatel.c
@@ -5012,121 +5012,119 @@ static int patch_stac9872(struct hda_codec *codec)
/*
* patch entries
*/
-static const struct hda_codec_preset snd_hda_preset_sigmatel[] = {
- { .id = 0x83847690, .name = "STAC9200", .patch = patch_stac9200 },
- { .id = 0x83847882, .name = "STAC9220 A1", .patch = patch_stac922x },
- { .id = 0x83847680, .name = "STAC9221 A1", .patch = patch_stac922x },
- { .id = 0x83847880, .name = "STAC9220 A2", .patch = patch_stac922x },
- { .id = 0x83847681, .name = "STAC9220D/9223D A2", .patch = patch_stac922x },
- { .id = 0x83847682, .name = "STAC9221 A2", .patch = patch_stac922x },
- { .id = 0x83847683, .name = "STAC9221D A2", .patch = patch_stac922x },
- { .id = 0x83847618, .name = "STAC9227", .patch = patch_stac927x },
- { .id = 0x83847619, .name = "STAC9227", .patch = patch_stac927x },
- { .id = 0x83847616, .name = "STAC9228", .patch = patch_stac927x },
- { .id = 0x83847617, .name = "STAC9228", .patch = patch_stac927x },
- { .id = 0x83847614, .name = "STAC9229", .patch = patch_stac927x },
- { .id = 0x83847615, .name = "STAC9229", .patch = patch_stac927x },
- { .id = 0x83847620, .name = "STAC9274", .patch = patch_stac927x },
- { .id = 0x83847621, .name = "STAC9274D", .patch = patch_stac927x },
- { .id = 0x83847622, .name = "STAC9273X", .patch = patch_stac927x },
- { .id = 0x83847623, .name = "STAC9273D", .patch = patch_stac927x },
- { .id = 0x83847624, .name = "STAC9272X", .patch = patch_stac927x },
- { .id = 0x83847625, .name = "STAC9272D", .patch = patch_stac927x },
- { .id = 0x83847626, .name = "STAC9271X", .patch = patch_stac927x },
- { .id = 0x83847627, .name = "STAC9271D", .patch = patch_stac927x },
- { .id = 0x83847628, .name = "STAC9274X5NH", .patch = patch_stac927x },
- { .id = 0x83847629, .name = "STAC9274D5NH", .patch = patch_stac927x },
- { .id = 0x83847632, .name = "STAC9202", .patch = patch_stac925x },
- { .id = 0x83847633, .name = "STAC9202D", .patch = patch_stac925x },
- { .id = 0x83847634, .name = "STAC9250", .patch = patch_stac925x },
- { .id = 0x83847635, .name = "STAC9250D", .patch = patch_stac925x },
- { .id = 0x83847636, .name = "STAC9251", .patch = patch_stac925x },
- { .id = 0x83847637, .name = "STAC9250D", .patch = patch_stac925x },
- { .id = 0x83847645, .name = "92HD206X", .patch = patch_stac927x },
- { .id = 0x83847646, .name = "92HD206D", .patch = patch_stac927x },
- /* The following does not take into account .id=0x83847661 when subsys =
- * 104D0C00 which is STAC9225s. Because of this, some SZ Notebooks are
- * currently not fully supported.
- */
- { .id = 0x83847661, .name = "CXD9872RD/K", .patch = patch_stac9872 },
- { .id = 0x83847662, .name = "STAC9872AK", .patch = patch_stac9872 },
- { .id = 0x83847664, .name = "CXD9872AKD", .patch = patch_stac9872 },
- { .id = 0x83847698, .name = "STAC9205", .patch = patch_stac9205 },
- { .id = 0x838476a0, .name = "STAC9205", .patch = patch_stac9205 },
- { .id = 0x838476a1, .name = "STAC9205D", .patch = patch_stac9205 },
- { .id = 0x838476a2, .name = "STAC9204", .patch = patch_stac9205 },
- { .id = 0x838476a3, .name = "STAC9204D", .patch = patch_stac9205 },
- { .id = 0x838476a4, .name = "STAC9255", .patch = patch_stac9205 },
- { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 },
- { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 },
- { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 },
- { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx},
- { .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76d4, .name = "92HD83C1C5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76d5, .name = "92HD81B1C5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76d1, .name = "92HD87B1/3", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76d9, .name = "92HD87B2/4", .patch = patch_stac92hd83xxx},
- { .id = 0x111d7666, .name = "92HD88B3", .patch = patch_stac92hd83xxx},
- { .id = 0x111d7667, .name = "92HD88B1", .patch = patch_stac92hd83xxx},
- { .id = 0x111d7668, .name = "92HD88B2", .patch = patch_stac92hd83xxx},
- { .id = 0x111d7669, .name = "92HD88B4", .patch = patch_stac92hd83xxx},
- { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx},
- { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx },
- { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx },
- { .id = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx },
- { .id = 0x111d7695, .name = "92HD95", .patch = patch_stac92hd95 },
- { .id = 0x111d76b0, .name = "92HD71B8X", .patch = patch_stac92hd71bxx },
- { .id = 0x111d76b1, .name = "92HD71B8X", .patch = patch_stac92hd71bxx },
- { .id = 0x111d76b2, .name = "92HD71B7X", .patch = patch_stac92hd71bxx },
- { .id = 0x111d76b3, .name = "92HD71B7X", .patch = patch_stac92hd71bxx },
- { .id = 0x111d76b4, .name = "92HD71B6X", .patch = patch_stac92hd71bxx },
- { .id = 0x111d76b5, .name = "92HD71B6X", .patch = patch_stac92hd71bxx },
- { .id = 0x111d76b6, .name = "92HD71B5X", .patch = patch_stac92hd71bxx },
- { .id = 0x111d76b7, .name = "92HD71B5X", .patch = patch_stac92hd71bxx },
- { .id = 0x111d76c0, .name = "92HD89C3", .patch = patch_stac92hd73xx },
- { .id = 0x111d76c1, .name = "92HD89C2", .patch = patch_stac92hd73xx },
- { .id = 0x111d76c2, .name = "92HD89C1", .patch = patch_stac92hd73xx },
- { .id = 0x111d76c3, .name = "92HD89B3", .patch = patch_stac92hd73xx },
- { .id = 0x111d76c4, .name = "92HD89B2", .patch = patch_stac92hd73xx },
- { .id = 0x111d76c5, .name = "92HD89B1", .patch = patch_stac92hd73xx },
- { .id = 0x111d76c6, .name = "92HD89E3", .patch = patch_stac92hd73xx },
- { .id = 0x111d76c7, .name = "92HD89E2", .patch = patch_stac92hd73xx },
- { .id = 0x111d76c8, .name = "92HD89E1", .patch = patch_stac92hd73xx },
- { .id = 0x111d76c9, .name = "92HD89D3", .patch = patch_stac92hd73xx },
- { .id = 0x111d76ca, .name = "92HD89D2", .patch = patch_stac92hd73xx },
- { .id = 0x111d76cb, .name = "92HD89D1", .patch = patch_stac92hd73xx },
- { .id = 0x111d76cc, .name = "92HD89F3", .patch = patch_stac92hd73xx },
- { .id = 0x111d76cd, .name = "92HD89F2", .patch = patch_stac92hd73xx },
- { .id = 0x111d76ce, .name = "92HD89F1", .patch = patch_stac92hd73xx },
- { .id = 0x111d76df, .name = "92HD93BXX", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76e0, .name = "92HD91BXX", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76e3, .name = "92HD98BXX", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76e5, .name = "92HD99BXX", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76e7, .name = "92HD90BXX", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76e8, .name = "92HD66B1X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76e9, .name = "92HD66B2X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76ea, .name = "92HD66B3X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76eb, .name = "92HD66C1X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76ec, .name = "92HD66C2X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76ed, .name = "92HD66C3X5", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76ee, .name = "92HD66B1X3", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76ef, .name = "92HD66B2X3", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76f0, .name = "92HD66B3X3", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76f1, .name = "92HD66C1X3", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76f2, .name = "92HD66C2X3", .patch = patch_stac92hd83xxx},
- { .id = 0x111d76f3, .name = "92HD66C3/65", .patch = patch_stac92hd83xxx},
+static const struct hda_device_id snd_hda_id_sigmatel[] = {
+ HDA_CODEC_ENTRY(0x83847690, "STAC9200", patch_stac9200),
+ HDA_CODEC_ENTRY(0x83847882, "STAC9220 A1", patch_stac922x),
+ HDA_CODEC_ENTRY(0x83847680, "STAC9221 A1", patch_stac922x),
+ HDA_CODEC_ENTRY(0x83847880, "STAC9220 A2", patch_stac922x),
+ HDA_CODEC_ENTRY(0x83847681, "STAC9220D/9223D A2", patch_stac922x),
+ HDA_CODEC_ENTRY(0x83847682, "STAC9221 A2", patch_stac922x),
+ HDA_CODEC_ENTRY(0x83847683, "STAC9221D A2", patch_stac922x),
+ HDA_CODEC_ENTRY(0x83847618, "STAC9227", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847619, "STAC9227", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847616, "STAC9228", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847617, "STAC9228", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847614, "STAC9229", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847615, "STAC9229", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847620, "STAC9274", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847621, "STAC9274D", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847622, "STAC9273X", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847623, "STAC9273D", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847624, "STAC9272X", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847625, "STAC9272D", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847626, "STAC9271X", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847627, "STAC9271D", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847628, "STAC9274X5NH", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847629, "STAC9274D5NH", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847632, "STAC9202", patch_stac925x),
+ HDA_CODEC_ENTRY(0x83847633, "STAC9202D", patch_stac925x),
+ HDA_CODEC_ENTRY(0x83847634, "STAC9250", patch_stac925x),
+ HDA_CODEC_ENTRY(0x83847635, "STAC9250D", patch_stac925x),
+ HDA_CODEC_ENTRY(0x83847636, "STAC9251", patch_stac925x),
+ HDA_CODEC_ENTRY(0x83847637, "STAC9250D", patch_stac925x),
+ HDA_CODEC_ENTRY(0x83847645, "92HD206X", patch_stac927x),
+ HDA_CODEC_ENTRY(0x83847646, "92HD206D", patch_stac927x),
+ /* The following does not take into account .id=0x83847661 when subsys =
+ * 104D0C00 which is STAC9225s. Because of this, some SZ Notebooks are
+ * currently not fully supported.
+ */
+ HDA_CODEC_ENTRY(0x83847661, "CXD9872RD/K", patch_stac9872),
+ HDA_CODEC_ENTRY(0x83847662, "STAC9872AK", patch_stac9872),
+ HDA_CODEC_ENTRY(0x83847664, "CXD9872AKD", patch_stac9872),
+ HDA_CODEC_ENTRY(0x83847698, "STAC9205", patch_stac9205),
+ HDA_CODEC_ENTRY(0x838476a0, "STAC9205", patch_stac9205),
+ HDA_CODEC_ENTRY(0x838476a1, "STAC9205D", patch_stac9205),
+ HDA_CODEC_ENTRY(0x838476a2, "STAC9204", patch_stac9205),
+ HDA_CODEC_ENTRY(0x838476a3, "STAC9204D", patch_stac9205),
+ HDA_CODEC_ENTRY(0x838476a4, "STAC9255", patch_stac9205),
+ HDA_CODEC_ENTRY(0x838476a5, "STAC9255D", patch_stac9205),
+ HDA_CODEC_ENTRY(0x838476a6, "STAC9254", patch_stac9205),
+ HDA_CODEC_ENTRY(0x838476a7, "STAC9254D", patch_stac9205),
+ HDA_CODEC_ENTRY(0x111d7603, "92HD75B3X5", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d7604, "92HD83C1X5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76d4, "92HD83C1C5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d7605, "92HD81B1X5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76d5, "92HD81B1C5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76d1, "92HD87B1/3", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76d9, "92HD87B2/4", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d7666, "92HD88B3", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d7667, "92HD88B1", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d7668, "92HD88B2", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d7669, "92HD88B4", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d7608, "92HD75B2X5", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d7674, "92HD73D1X5", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d7675, "92HD73C1X5", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d7676, "92HD73E1X5", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d7695, "92HD95", patch_stac92hd95),
+ HDA_CODEC_ENTRY(0x111d76b0, "92HD71B8X", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d76b1, "92HD71B8X", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d76b2, "92HD71B7X", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d76b3, "92HD71B7X", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d76b4, "92HD71B6X", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d76b5, "92HD71B6X", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d76b6, "92HD71B5X", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d76b7, "92HD71B5X", patch_stac92hd71bxx),
+ HDA_CODEC_ENTRY(0x111d76c0, "92HD89C3", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76c1, "92HD89C2", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76c2, "92HD89C1", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76c3, "92HD89B3", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76c4, "92HD89B2", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76c5, "92HD89B1", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76c6, "92HD89E3", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76c7, "92HD89E2", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76c8, "92HD89E1", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76c9, "92HD89D3", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76ca, "92HD89D2", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76cb, "92HD89D1", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76cc, "92HD89F3", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76cd, "92HD89F2", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76ce, "92HD89F1", patch_stac92hd73xx),
+ HDA_CODEC_ENTRY(0x111d76df, "92HD93BXX", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76e0, "92HD91BXX", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76e3, "92HD98BXX", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76e5, "92HD99BXX", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76e7, "92HD90BXX", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76e8, "92HD66B1X5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76e9, "92HD66B2X5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76ea, "92HD66B3X5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76eb, "92HD66C1X5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76ec, "92HD66C2X5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76ed, "92HD66C3X5", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76ee, "92HD66B1X3", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76ef, "92HD66B2X3", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76f0, "92HD66B3X3", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76f1, "92HD66C1X3", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76f2, "92HD66C2X3", patch_stac92hd83xxx),
+ HDA_CODEC_ENTRY(0x111d76f3, "92HD66C3/65", patch_stac92hd83xxx),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:8384*");
-MODULE_ALIAS("snd-hda-codec-id:111d*");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_sigmatel);
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("IDT/Sigmatel HD-audio codec");
static struct hda_codec_driver sigmatel_driver = {
- .preset = snd_hda_preset_sigmatel,
+ .id = snd_hda_id_sigmatel,
};
module_hda_codec_driver(sigmatel_driver);
diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c
index da5366405eda..fc30d1e8aa76 100644
--- a/sound/pci/hda/patch_via.c
+++ b/sound/pci/hda/patch_via.c
@@ -785,21 +785,11 @@ static int patch_vt1708S(struct hda_codec *codec)
override_mic_boost(codec, 0x1e, 0, 3, 40);
/* correct names for VT1708BCE */
- if (get_codec_type(codec) == VT1708BCE) {
- kfree(codec->core.chip_name);
- codec->core.chip_name = kstrdup("VT1708BCE", GFP_KERNEL);
- snprintf(codec->card->mixername,
- sizeof(codec->card->mixername),
- "%s %s", codec->core.vendor_name, codec->core.chip_name);
- }
+ if (get_codec_type(codec) == VT1708BCE)
+ snd_hda_codec_set_name(codec, "VT1708BCE");
/* correct names for VT1705 */
- if (codec->core.vendor_id == 0x11064397) {
- kfree(codec->core.chip_name);
- codec->core.chip_name = kstrdup("VT1705", GFP_KERNEL);
- snprintf(codec->card->mixername,
- sizeof(codec->card->mixername),
- "%s %s", codec->core.vendor_name, codec->core.chip_name);
- }
+ if (codec->core.vendor_id == 0x11064397)
+ snd_hda_codec_set_name(codec, "VT1705");
/* automatic parse from the BIOS config */
err = via_parse_auto_config(codec);
@@ -1210,109 +1200,64 @@ static int patch_vt3476(struct hda_codec *codec)
/*
* patch entries
*/
-static const struct hda_codec_preset snd_hda_preset_via[] = {
- { .id = 0x11061708, .name = "VT1708", .patch = patch_vt1708},
- { .id = 0x11061709, .name = "VT1708", .patch = patch_vt1708},
- { .id = 0x1106170a, .name = "VT1708", .patch = patch_vt1708},
- { .id = 0x1106170b, .name = "VT1708", .patch = patch_vt1708},
- { .id = 0x1106e710, .name = "VT1709 10-Ch",
- .patch = patch_vt1709},
- { .id = 0x1106e711, .name = "VT1709 10-Ch",
- .patch = patch_vt1709},
- { .id = 0x1106e712, .name = "VT1709 10-Ch",
- .patch = patch_vt1709},
- { .id = 0x1106e713, .name = "VT1709 10-Ch",
- .patch = patch_vt1709},
- { .id = 0x1106e714, .name = "VT1709 6-Ch",
- .patch = patch_vt1709},
- { .id = 0x1106e715, .name = "VT1709 6-Ch",
- .patch = patch_vt1709},
- { .id = 0x1106e716, .name = "VT1709 6-Ch",
- .patch = patch_vt1709},
- { .id = 0x1106e717, .name = "VT1709 6-Ch",
- .patch = patch_vt1709},
- { .id = 0x1106e720, .name = "VT1708B 8-Ch",
- .patch = patch_vt1708B},
- { .id = 0x1106e721, .name = "VT1708B 8-Ch",
- .patch = patch_vt1708B},
- { .id = 0x1106e722, .name = "VT1708B 8-Ch",
- .patch = patch_vt1708B},
- { .id = 0x1106e723, .name = "VT1708B 8-Ch",
- .patch = patch_vt1708B},
- { .id = 0x1106e724, .name = "VT1708B 4-Ch",
- .patch = patch_vt1708B},
- { .id = 0x1106e725, .name = "VT1708B 4-Ch",
- .patch = patch_vt1708B},
- { .id = 0x1106e726, .name = "VT1708B 4-Ch",
- .patch = patch_vt1708B},
- { .id = 0x1106e727, .name = "VT1708B 4-Ch",
- .patch = patch_vt1708B},
- { .id = 0x11060397, .name = "VT1708S",
- .patch = patch_vt1708S},
- { .id = 0x11061397, .name = "VT1708S",
- .patch = patch_vt1708S},
- { .id = 0x11062397, .name = "VT1708S",
- .patch = patch_vt1708S},
- { .id = 0x11063397, .name = "VT1708S",
- .patch = patch_vt1708S},
- { .id = 0x11064397, .name = "VT1705",
- .patch = patch_vt1708S},
- { .id = 0x11065397, .name = "VT1708S",
- .patch = patch_vt1708S},
- { .id = 0x11066397, .name = "VT1708S",
- .patch = patch_vt1708S},
- { .id = 0x11067397, .name = "VT1708S",
- .patch = patch_vt1708S},
- { .id = 0x11060398, .name = "VT1702",
- .patch = patch_vt1702},
- { .id = 0x11061398, .name = "VT1702",
- .patch = patch_vt1702},
- { .id = 0x11062398, .name = "VT1702",
- .patch = patch_vt1702},
- { .id = 0x11063398, .name = "VT1702",
- .patch = patch_vt1702},
- { .id = 0x11064398, .name = "VT1702",
- .patch = patch_vt1702},
- { .id = 0x11065398, .name = "VT1702",
- .patch = patch_vt1702},
- { .id = 0x11066398, .name = "VT1702",
- .patch = patch_vt1702},
- { .id = 0x11067398, .name = "VT1702",
- .patch = patch_vt1702},
- { .id = 0x11060428, .name = "VT1718S",
- .patch = patch_vt1718S},
- { .id = 0x11064428, .name = "VT1718S",
- .patch = patch_vt1718S},
- { .id = 0x11060441, .name = "VT2020",
- .patch = patch_vt1718S},
- { .id = 0x11064441, .name = "VT1828S",
- .patch = patch_vt1718S},
- { .id = 0x11060433, .name = "VT1716S",
- .patch = patch_vt1716S},
- { .id = 0x1106a721, .name = "VT1716S",
- .patch = patch_vt1716S},
- { .id = 0x11060438, .name = "VT2002P", .patch = patch_vt2002P},
- { .id = 0x11064438, .name = "VT2002P", .patch = patch_vt2002P},
- { .id = 0x11060448, .name = "VT1812", .patch = patch_vt1812},
- { .id = 0x11060440, .name = "VT1818S",
- .patch = patch_vt1708S},
- { .id = 0x11060446, .name = "VT1802",
- .patch = patch_vt2002P},
- { .id = 0x11068446, .name = "VT1802",
- .patch = patch_vt2002P},
- { .id = 0x11064760, .name = "VT1705CF",
- .patch = patch_vt3476},
- { .id = 0x11064761, .name = "VT1708SCE",
- .patch = patch_vt3476},
- { .id = 0x11064762, .name = "VT1808",
- .patch = patch_vt3476},
+static const struct hda_device_id snd_hda_id_via[] = {
+ HDA_CODEC_ENTRY(0x11061708, "VT1708", patch_vt1708),
+ HDA_CODEC_ENTRY(0x11061709, "VT1708", patch_vt1708),
+ HDA_CODEC_ENTRY(0x1106170a, "VT1708", patch_vt1708),
+ HDA_CODEC_ENTRY(0x1106170b, "VT1708", patch_vt1708),
+ HDA_CODEC_ENTRY(0x1106e710, "VT1709 10-Ch", patch_vt1709),
+ HDA_CODEC_ENTRY(0x1106e711, "VT1709 10-Ch", patch_vt1709),
+ HDA_CODEC_ENTRY(0x1106e712, "VT1709 10-Ch", patch_vt1709),
+ HDA_CODEC_ENTRY(0x1106e713, "VT1709 10-Ch", patch_vt1709),
+ HDA_CODEC_ENTRY(0x1106e714, "VT1709 6-Ch", patch_vt1709),
+ HDA_CODEC_ENTRY(0x1106e715, "VT1709 6-Ch", patch_vt1709),
+ HDA_CODEC_ENTRY(0x1106e716, "VT1709 6-Ch", patch_vt1709),
+ HDA_CODEC_ENTRY(0x1106e717, "VT1709 6-Ch", patch_vt1709),
+ HDA_CODEC_ENTRY(0x1106e720, "VT1708B 8-Ch", patch_vt1708B),
+ HDA_CODEC_ENTRY(0x1106e721, "VT1708B 8-Ch", patch_vt1708B),
+ HDA_CODEC_ENTRY(0x1106e722, "VT1708B 8-Ch", patch_vt1708B),
+ HDA_CODEC_ENTRY(0x1106e723, "VT1708B 8-Ch", patch_vt1708B),
+ HDA_CODEC_ENTRY(0x1106e724, "VT1708B 4-Ch", patch_vt1708B),
+ HDA_CODEC_ENTRY(0x1106e725, "VT1708B 4-Ch", patch_vt1708B),
+ HDA_CODEC_ENTRY(0x1106e726, "VT1708B 4-Ch", patch_vt1708B),
+ HDA_CODEC_ENTRY(0x1106e727, "VT1708B 4-Ch", patch_vt1708B),
+ HDA_CODEC_ENTRY(0x11060397, "VT1708S", patch_vt1708S),
+ HDA_CODEC_ENTRY(0x11061397, "VT1708S", patch_vt1708S),
+ HDA_CODEC_ENTRY(0x11062397, "VT1708S", patch_vt1708S),
+ HDA_CODEC_ENTRY(0x11063397, "VT1708S", patch_vt1708S),
+ HDA_CODEC_ENTRY(0x11064397, "VT1705", patch_vt1708S),
+ HDA_CODEC_ENTRY(0x11065397, "VT1708S", patch_vt1708S),
+ HDA_CODEC_ENTRY(0x11066397, "VT1708S", patch_vt1708S),
+ HDA_CODEC_ENTRY(0x11067397, "VT1708S", patch_vt1708S),
+ HDA_CODEC_ENTRY(0x11060398, "VT1702", patch_vt1702),
+ HDA_CODEC_ENTRY(0x11061398, "VT1702", patch_vt1702),
+ HDA_CODEC_ENTRY(0x11062398, "VT1702", patch_vt1702),
+ HDA_CODEC_ENTRY(0x11063398, "VT1702", patch_vt1702),
+ HDA_CODEC_ENTRY(0x11064398, "VT1702", patch_vt1702),
+ HDA_CODEC_ENTRY(0x11065398, "VT1702", patch_vt1702),
+ HDA_CODEC_ENTRY(0x11066398, "VT1702", patch_vt1702),
+ HDA_CODEC_ENTRY(0x11067398, "VT1702", patch_vt1702),
+ HDA_CODEC_ENTRY(0x11060428, "VT1718S", patch_vt1718S),
+ HDA_CODEC_ENTRY(0x11064428, "VT1718S", patch_vt1718S),
+ HDA_CODEC_ENTRY(0x11060441, "VT2020", patch_vt1718S),
+ HDA_CODEC_ENTRY(0x11064441, "VT1828S", patch_vt1718S),
+ HDA_CODEC_ENTRY(0x11060433, "VT1716S", patch_vt1716S),
+ HDA_CODEC_ENTRY(0x1106a721, "VT1716S", patch_vt1716S),
+ HDA_CODEC_ENTRY(0x11060438, "VT2002P", patch_vt2002P),
+ HDA_CODEC_ENTRY(0x11064438, "VT2002P", patch_vt2002P),
+ HDA_CODEC_ENTRY(0x11060448, "VT1812", patch_vt1812),
+ HDA_CODEC_ENTRY(0x11060440, "VT1818S", patch_vt1708S),
+ HDA_CODEC_ENTRY(0x11060446, "VT1802", patch_vt2002P),
+ HDA_CODEC_ENTRY(0x11068446, "VT1802", patch_vt2002P),
+ HDA_CODEC_ENTRY(0x11064760, "VT1705CF", patch_vt3476),
+ HDA_CODEC_ENTRY(0x11064761, "VT1708SCE", patch_vt3476),
+ HDA_CODEC_ENTRY(0x11064762, "VT1808", patch_vt3476),
{} /* terminator */
};
-
-MODULE_ALIAS("snd-hda-codec-id:1106*");
+MODULE_DEVICE_TABLE(hdaudio, snd_hda_id_via);
static struct hda_codec_driver via_driver = {
- .preset = snd_hda_preset_via,
+ .id = snd_hda_id_via,
};
MODULE_LICENSE("GPL");
diff --git a/sound/pci/korg1212/korg1212.c b/sound/pci/korg1212/korg1212.c
index 7acbc21d642a..9e1ad119a3ce 100644
--- a/sound/pci/korg1212/korg1212.c
+++ b/sound/pci/korg1212/korg1212.c
@@ -1394,7 +1394,9 @@ static int snd_korg1212_playback_open(struct snd_pcm_substream *substream)
spin_unlock_irqrestore(&korg1212->lock, flags);
- snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, kPlayBufferFrames, kPlayBufferFrames);
+ snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ kPlayBufferFrames);
+
return 0;
}
@@ -1422,8 +1424,8 @@ static int snd_korg1212_capture_open(struct snd_pcm_substream *substream)
spin_unlock_irqrestore(&korg1212->lock, flags);
- snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
- kPlayBufferFrames, kPlayBufferFrames);
+ snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+ kPlayBufferFrames);
return 0;
}
diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c
index cba89beb2b38..8b8e2e54fba3 100644
--- a/sound/pci/lx6464es/lx6464es.c
+++ b/sound/pci/lx6464es/lx6464es.c
@@ -234,8 +234,8 @@ static int lx_pcm_open(struct snd_pcm_substream *substream)
/* the clock rate cannot be changed */
board_rate = chip->board_sample_rate;
- err = snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE,
- board_rate, board_rate);
+ err = snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_RATE,
+ board_rate);
if (err < 0) {
dev_warn(chip->card->dev, "could not constrain periods\n");
diff --git a/sound/pci/maestro3.c b/sound/pci/maestro3.c
index 72e89cedc52d..17ae92613de4 100644
--- a/sound/pci/maestro3.c
+++ b/sound/pci/maestro3.c
@@ -1929,15 +1929,32 @@ snd_m3_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val)
return;
snd_m3_outw(chip, val, CODEC_DATA);
snd_m3_outb(chip, reg & 0x7f, CODEC_COMMAND);
+ /*
+ * Workaround for buggy ES1988 integrated AC'97 codec. It remains silent
+ * until the MASTER volume or mute is touched (alsactl restore does not
+ * work).
+ */
+ if (ac97->id == 0x45838308 && reg == AC97_MASTER) {
+ snd_m3_ac97_wait(chip);
+ snd_m3_outw(chip, val, CODEC_DATA);
+ snd_m3_outb(chip, reg & 0x7f, CODEC_COMMAND);
+ }
}
-static void snd_m3_remote_codec_config(int io, int isremote)
+static void snd_m3_remote_codec_config(struct snd_m3 *chip, int isremote)
{
+ int io = chip->iobase;
+ u16 tmp;
+
isremote = isremote ? 1 : 0;
- outw((inw(io + RING_BUS_CTRL_B) & ~SECOND_CODEC_ID_MASK) | isremote,
- io + RING_BUS_CTRL_B);
+ tmp = inw(io + RING_BUS_CTRL_B) & ~SECOND_CODEC_ID_MASK;
+ /* enable dock on Dell Latitude C810 */
+ if (chip->pci->subsystem_vendor == 0x1028 &&
+ chip->pci->subsystem_device == 0x00e5)
+ tmp |= M3I_DOCK_ENABLE;
+ outw(tmp | isremote, io + RING_BUS_CTRL_B);
outw((inw(io + SDO_OUT_DEST_CTRL) & ~COMMAND_ADDR_OUT) | isremote,
io + SDO_OUT_DEST_CTRL);
outw((inw(io + SDO_IN_DEST_CTRL) & ~STATUS_ADDR_IN) | isremote,
@@ -1989,7 +2006,7 @@ static void snd_m3_ac97_reset(struct snd_m3 *chip)
if (!chip->irda_workaround)
dir |= 0x10; /* assuming pci bus master? */
- snd_m3_remote_codec_config(io, 0);
+ snd_m3_remote_codec_config(chip, 0);
outw(IO_SRAM_ENABLE, io + RING_BUS_CTRL_A);
udelay(20);
diff --git a/sound/pci/rme32.c b/sound/pci/rme32.c
index 23d7f5d30c41..cd94ac548ba3 100644
--- a/sound/pci/rme32.c
+++ b/sound/pci/rme32.c
@@ -831,9 +831,9 @@ static struct snd_pcm_hw_constraint_list hw_constraints_period_bytes = {
static void snd_rme32_set_buffer_constraint(struct rme32 *rme32, struct snd_pcm_runtime *runtime)
{
if (! rme32->fullduplex_mode) {
- snd_pcm_hw_constraint_minmax(runtime,
+ snd_pcm_hw_constraint_single(runtime,
SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
- RME32_BUFFER_SIZE, RME32_BUFFER_SIZE);
+ RME32_BUFFER_SIZE);
snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
&hw_constraints_period_bytes);
diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c
index 2306ccf7281e..714df906249e 100644
--- a/sound/pci/rme96.c
+++ b/sound/pci/rme96.c
@@ -1152,13 +1152,13 @@ rme96_set_buffer_size_constraint(struct rme96 *rme96,
{
unsigned int size;
- snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
- RME96_BUFFER_SIZE, RME96_BUFFER_SIZE);
+ snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ RME96_BUFFER_SIZE);
if ((size = rme96->playback_periodsize) != 0 ||
(size = rme96->capture_periodsize) != 0)
- snd_pcm_hw_constraint_minmax(runtime,
+ snd_pcm_hw_constraint_single(runtime,
SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
- size, size);
+ size);
else
snd_pcm_hw_constraint_list(runtime, 0,
SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c
index 9bba275b4c9b..2875b4f6d8c9 100644
--- a/sound/pci/rme9652/hdsp.c
+++ b/sound/pci/rme9652/hdsp.c
@@ -5112,6 +5112,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp)
dev_err(hdsp->card->dev,
"too short firmware size %d (expected %d)\n",
(int)fw->size, HDSP_FIRMWARE_SIZE);
+ release_firmware(fw);
return -EINVAL;
}
diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c
index cb666c73712d..8bc8016c173d 100644
--- a/sound/pci/rme9652/hdspm.c
+++ b/sound/pci/rme9652/hdspm.c
@@ -6080,18 +6080,17 @@ static int snd_hdspm_open(struct snd_pcm_substream *substream)
SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
32, 4096);
/* RayDAT & AIO have a fixed buffer of 16384 samples per channel */
- snd_pcm_hw_constraint_minmax(runtime,
+ snd_pcm_hw_constraint_single(runtime,
SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
- 16384, 16384);
+ 16384);
break;
default:
snd_pcm_hw_constraint_minmax(runtime,
SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
64, 8192);
- snd_pcm_hw_constraint_minmax(runtime,
- SNDRV_PCM_HW_PARAM_PERIODS,
- 2, 2);
+ snd_pcm_hw_constraint_single(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS, 2);
break;
}
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 225bfda414e9..7ff7d88e46dd 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -9,7 +9,6 @@ menuconfig SND_SOC
select SND_JACK if INPUT=y || INPUT=SND
select REGMAP_I2C if I2C
select REGMAP_SPI if SPI_MASTER
- select SND_COMPRESS_OFFLOAD
---help---
If you want ASoC support, you should say Y here and also to the
@@ -30,6 +29,10 @@ config SND_SOC_GENERIC_DMAENGINE_PCM
bool
select SND_DMAENGINE_PCM
+config SND_SOC_COMPRESS
+ bool
+ select SND_COMPRESS_OFFLOAD
+
config SND_SOC_TOPOLOGY
bool
@@ -58,6 +61,7 @@ source "sound/soc/sh/Kconfig"
source "sound/soc/sirf/Kconfig"
source "sound/soc/spear/Kconfig"
source "sound/soc/sti/Kconfig"
+source "sound/soc/sunxi/Kconfig"
source "sound/soc/tegra/Kconfig"
source "sound/soc/txx9/Kconfig"
source "sound/soc/ux500/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 134aca150a50..8eb06db32fa0 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -1,5 +1,6 @@
snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o
-snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o
+snd-soc-core-objs += soc-pcm.o soc-io.o soc-devres.o soc-ops.o
+snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o
ifneq ($(CONFIG_SND_SOC_TOPOLOGY),)
snd-soc-core-objs += soc-topology.o
@@ -40,6 +41,7 @@ obj-$(CONFIG_SND_SOC) += sh/
obj-$(CONFIG_SND_SOC) += sirf/
obj-$(CONFIG_SND_SOC) += spear/
obj-$(CONFIG_SND_SOC) += sti/
+obj-$(CONFIG_SND_SOC) += sunxi/
obj-$(CONFIG_SND_SOC) += tegra/
obj-$(CONFIG_SND_SOC) += txx9/
obj-$(CONFIG_SND_SOC) += ux500/
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index 1489cd461aec..2d30464b81ce 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -59,4 +59,13 @@ config SND_AT91_SOC_SAM9X5_WM8731
help
Say Y if you want to add support for audio SoC on an
at91sam9x5 based board that is using WM8731 codec.
+
+config SND_ATMEL_SOC_CLASSD
+ tristate "Atmel ASoC driver for boards using CLASSD"
+ depends on ARCH_AT91 || COMPILE_TEST
+ select SND_ATMEL_SOC_DMA
+ select REGMAP_MMIO
+ help
+ Say Y if you want to add support for Atmel ASoC driver for boards using
+ CLASSD.
endif
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index b327e5cc8de3..f6f7db428216 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -11,7 +11,9 @@ obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o
snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o
snd-atmel-soc-wm8904-objs := atmel_wm8904.o
snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o
+snd-atmel-soc-classd-objs := atmel-classd.o
obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o
obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o
+obj-$(CONFIG_SND_ATMEL_SOC_CLASSD) += snd-atmel-soc-classd.o
diff --git a/sound/soc/atmel/atmel-classd.c b/sound/soc/atmel/atmel-classd.c
new file mode 100644
index 000000000000..8276675730ef
--- /dev/null
+++ b/sound/soc/atmel/atmel-classd.c
@@ -0,0 +1,679 @@
+/* Atmel ALSA SoC Audio Class D Amplifier (CLASSD) driver
+ *
+ * Copyright (C) 2015 Atmel
+ *
+ * Author: Songjun Wu <songjun.wu@atmel.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 or later
+ * as published by the Free Software Foundation.
+ */
+
+#include <linux/of.h>
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/regmap.h>
+#include <sound/core.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+#include "atmel-classd.h"
+
+struct atmel_classd_pdata {
+ bool non_overlap_enable;
+ int non_overlap_time;
+ int pwm_type;
+ const char *card_name;
+};
+
+struct atmel_classd {
+ dma_addr_t phy_base;
+ struct regmap *regmap;
+ struct clk *pclk;
+ struct clk *gclk;
+ struct clk *aclk;
+ int irq;
+ const struct atmel_classd_pdata *pdata;
+};
+
+#ifdef CONFIG_OF
+static const struct of_device_id atmel_classd_of_match[] = {
+ {
+ .compatible = "atmel,sama5d2-classd",
+ }, {
+ /* sentinel */
+ }
+};
+MODULE_DEVICE_TABLE(of, atmel_classd_of_match);
+
+static struct atmel_classd_pdata *atmel_classd_dt_init(struct device *dev)
+{
+ struct device_node *np = dev->of_node;
+ struct atmel_classd_pdata *pdata;
+ const char *pwm_type;
+ int ret;
+
+ if (!np) {
+ dev_err(dev, "device node not found\n");
+ return ERR_PTR(-EINVAL);
+ }
+
+ pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL);
+ if (!pdata)
+ return ERR_PTR(-ENOMEM);
+
+ ret = of_property_read_string(np, "atmel,pwm-type", &pwm_type);
+ if ((ret == 0) && (strcmp(pwm_type, "diff") == 0))
+ pdata->pwm_type = CLASSD_MR_PWMTYP_DIFF;
+ else
+ pdata->pwm_type = CLASSD_MR_PWMTYP_SINGLE;
+
+ ret = of_property_read_u32(np,
+ "atmel,non-overlap-time", &pdata->non_overlap_time);
+ if (ret)
+ pdata->non_overlap_enable = false;
+ else
+ pdata->non_overlap_enable = true;
+
+ ret = of_property_read_string(np, "atmel,model", &pdata->card_name);
+ if (ret)
+ pdata->card_name = "CLASSD";
+
+ return pdata;
+}
+#else
+static inline struct atmel_classd_pdata *
+atmel_classd_dt_init(struct device *dev)
+{
+ return ERR_PTR(-EINVAL);
+}
+#endif
+
+#define ATMEL_CLASSD_RATES (SNDRV_PCM_RATE_8000 \
+ | SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 \
+ | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 \
+ | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 \
+ | SNDRV_PCM_RATE_96000)
+
+static const struct snd_pcm_hardware atmel_classd_hw = {
+ .info = SNDRV_PCM_INFO_MMAP
+ | SNDRV_PCM_INFO_MMAP_VALID
+ | SNDRV_PCM_INFO_INTERLEAVED
+ | SNDRV_PCM_INFO_RESUME
+ | SNDRV_PCM_INFO_PAUSE,
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE),
+ .rates = ATMEL_CLASSD_RATES,
+ .rate_min = 8000,
+ .rate_max = 96000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 64 * 1024,
+ .period_bytes_min = 256,
+ .period_bytes_max = 32 * 1024,
+ .periods_min = 2,
+ .periods_max = 256,
+};
+
+#define ATMEL_CLASSD_PREALLOC_BUF_SIZE (64 * 1024)
+
+/* cpu dai component */
+static int atmel_classd_cpu_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
+
+ regmap_write(dd->regmap, CLASSD_THR, 0x0);
+
+ return clk_prepare_enable(dd->pclk);
+}
+
+static void atmel_classd_cpu_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *cpu_dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
+
+ clk_disable_unprepare(dd->pclk);
+}
+
+static const struct snd_soc_dai_ops atmel_classd_cpu_dai_ops = {
+ .startup = atmel_classd_cpu_dai_startup,
+ .shutdown = atmel_classd_cpu_dai_shutdown,
+};
+
+static struct snd_soc_dai_driver atmel_classd_cpu_dai = {
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = ATMEL_CLASSD_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = &atmel_classd_cpu_dai_ops,
+};
+
+static const struct snd_soc_component_driver atmel_classd_cpu_dai_component = {
+ .name = "atmel-classd",
+};
+
+/* platform */
+static int
+atmel_classd_platform_configure_dma(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct dma_slave_config *slave_config)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
+
+ if (params_physical_width(params) != 16) {
+ dev_err(rtd->platform->dev,
+ "only supports 16-bit audio data\n");
+ return -EINVAL;
+ }
+
+ slave_config->direction = DMA_MEM_TO_DEV;
+ slave_config->dst_addr = dd->phy_base + CLASSD_THR;
+ slave_config->dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+ slave_config->dst_maxburst = 1;
+ slave_config->src_maxburst = 1;
+ slave_config->device_fc = false;
+
+ return 0;
+}
+
+static const struct snd_dmaengine_pcm_config
+atmel_classd_dmaengine_pcm_config = {
+ .prepare_slave_config = atmel_classd_platform_configure_dma,
+ .pcm_hardware = &atmel_classd_hw,
+ .prealloc_buffer_size = ATMEL_CLASSD_PREALLOC_BUF_SIZE,
+};
+
+/* codec */
+static const char * const mono_mode_text[] = {
+ "mix", "sat", "left", "right"
+};
+
+static SOC_ENUM_SINGLE_DECL(classd_mono_mode_enum,
+ CLASSD_INTPMR, CLASSD_INTPMR_MONO_MODE_SHIFT,
+ mono_mode_text);
+
+static const char * const eqcfg_text[] = {
+ "Treble-12dB", "Treble-6dB",
+ "Medium-8dB", "Medium-3dB",
+ "Bass-12dB", "Bass-6dB",
+ "0 dB",
+ "Bass+6dB", "Bass+12dB",
+ "Medium+3dB", "Medium+8dB",
+ "Treble+6dB", "Treble+12dB",
+};
+
+static const unsigned int eqcfg_value[] = {
+ CLASSD_INTPMR_EQCFG_T_CUT_12, CLASSD_INTPMR_EQCFG_T_CUT_6,
+ CLASSD_INTPMR_EQCFG_M_CUT_8, CLASSD_INTPMR_EQCFG_M_CUT_3,
+ CLASSD_INTPMR_EQCFG_B_CUT_12, CLASSD_INTPMR_EQCFG_B_CUT_6,
+ CLASSD_INTPMR_EQCFG_FLAT,
+ CLASSD_INTPMR_EQCFG_B_BOOST_6, CLASSD_INTPMR_EQCFG_B_BOOST_12,
+ CLASSD_INTPMR_EQCFG_M_BOOST_3, CLASSD_INTPMR_EQCFG_M_BOOST_8,
+ CLASSD_INTPMR_EQCFG_T_BOOST_6, CLASSD_INTPMR_EQCFG_T_BOOST_12,
+};
+
+static SOC_VALUE_ENUM_SINGLE_DECL(classd_eqcfg_enum,
+ CLASSD_INTPMR, CLASSD_INTPMR_EQCFG_SHIFT, 0xf,
+ eqcfg_text, eqcfg_value);
+
+static const DECLARE_TLV_DB_SCALE(classd_digital_tlv, -7800, 100, 1);
+
+static const struct snd_kcontrol_new atmel_classd_snd_controls[] = {
+SOC_DOUBLE_TLV("Playback Volume", CLASSD_INTPMR,
+ CLASSD_INTPMR_ATTL_SHIFT, CLASSD_INTPMR_ATTR_SHIFT,
+ 78, 1, classd_digital_tlv),
+
+SOC_SINGLE("Deemphasis Switch", CLASSD_INTPMR,
+ CLASSD_INTPMR_DEEMP_SHIFT, 1, 0),
+
+SOC_SINGLE("Mono Switch", CLASSD_INTPMR, CLASSD_INTPMR_MONO_SHIFT, 1, 0),
+
+SOC_SINGLE("Swap Switch", CLASSD_INTPMR, CLASSD_INTPMR_SWAP_SHIFT, 1, 0),
+
+SOC_ENUM("Mono Mode", classd_mono_mode_enum),
+
+SOC_ENUM("EQ", classd_eqcfg_enum),
+};
+
+static const char * const pwm_type[] = {
+ "Single ended", "Differential"
+};
+
+static int atmel_classd_codec_probe(struct snd_soc_codec *codec)
+{
+ struct snd_soc_card *card = snd_soc_codec_get_drvdata(codec);
+ struct atmel_classd *dd = snd_soc_card_get_drvdata(card);
+ const struct atmel_classd_pdata *pdata = dd->pdata;
+ u32 mask, val;
+
+ mask = CLASSD_MR_PWMTYP_MASK;
+ val = pdata->pwm_type << CLASSD_MR_PWMTYP_SHIFT;
+
+ mask |= CLASSD_MR_NON_OVERLAP_MASK;
+ if (pdata->non_overlap_enable) {
+ val |= (CLASSD_MR_NON_OVERLAP_EN
+ << CLASSD_MR_NON_OVERLAP_SHIFT);
+
+ mask |= CLASSD_MR_NOVR_VAL_MASK;
+ switch (pdata->non_overlap_time) {
+ case 5:
+ val |= (CLASSD_MR_NOVR_VAL_5NS
+ << CLASSD_MR_NOVR_VAL_SHIFT);
+ break;
+ case 10:
+ val |= (CLASSD_MR_NOVR_VAL_10NS
+ << CLASSD_MR_NOVR_VAL_SHIFT);
+ break;
+ case 15:
+ val |= (CLASSD_MR_NOVR_VAL_15NS
+ << CLASSD_MR_NOVR_VAL_SHIFT);
+ break;
+ case 20:
+ val |= (CLASSD_MR_NOVR_VAL_20NS
+ << CLASSD_MR_NOVR_VAL_SHIFT);
+ break;
+ default:
+ val |= (CLASSD_MR_NOVR_VAL_10NS
+ << CLASSD_MR_NOVR_VAL_SHIFT);
+ dev_warn(codec->dev,
+ "non-overlapping value %d is invalid, the default value 10 is specified\n",
+ pdata->non_overlap_time);
+ break;
+ }
+ }
+
+ snd_soc_update_bits(codec, CLASSD_MR, mask, val);
+
+ dev_info(codec->dev,
+ "PWM modulation type is %s, non-overlapping is %s\n",
+ pwm_type[pdata->pwm_type],
+ pdata->non_overlap_enable?"enabled":"disabled");
+
+ return 0;
+}
+
+static struct regmap *atmel_classd_codec_get_remap(struct device *dev)
+{
+ return dev_get_regmap(dev, NULL);
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_classd = {
+ .probe = atmel_classd_codec_probe,
+ .controls = atmel_classd_snd_controls,
+ .num_controls = ARRAY_SIZE(atmel_classd_snd_controls),
+ .get_regmap = atmel_classd_codec_get_remap,
+};
+
+/* codec dai component */
+static int atmel_classd_codec_dai_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *codec_dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
+ int ret;
+
+ ret = clk_prepare_enable(dd->aclk);
+ if (ret)
+ return ret;
+
+ return clk_prepare_enable(dd->gclk);
+}
+
+static int atmel_classd_codec_dai_digital_mute(struct snd_soc_dai *codec_dai,
+ int mute)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u32 mask, val;
+
+ mask = CLASSD_MR_LMUTE_MASK | CLASSD_MR_RMUTE_MASK;
+
+ if (mute)
+ val = mask;
+ else
+ val = 0;
+
+ snd_soc_update_bits(codec, CLASSD_MR, mask, val);
+
+ return 0;
+}
+
+#define CLASSD_ACLK_RATE_11M2896_MPY_8 (112896 * 100 * 8)
+#define CLASSD_ACLK_RATE_12M288_MPY_8 (12228 * 1000 * 8)
+
+static struct {
+ int rate;
+ int sample_rate;
+ int dsp_clk;
+ unsigned long aclk_rate;
+} const sample_rates[] = {
+ { 8000, CLASSD_INTPMR_FRAME_8K,
+ CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 },
+ { 16000, CLASSD_INTPMR_FRAME_16K,
+ CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 },
+ { 32000, CLASSD_INTPMR_FRAME_32K,
+ CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 },
+ { 48000, CLASSD_INTPMR_FRAME_48K,
+ CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 },
+ { 96000, CLASSD_INTPMR_FRAME_96K,
+ CLASSD_INTPMR_DSP_CLK_FREQ_12M288, CLASSD_ACLK_RATE_12M288_MPY_8 },
+ { 22050, CLASSD_INTPMR_FRAME_22K,
+ CLASSD_INTPMR_DSP_CLK_FREQ_11M2896, CLASSD_ACLK_RATE_11M2896_MPY_8 },
+ { 44100, CLASSD_INTPMR_FRAME_44K,
+ CLASSD_INTPMR_DSP_CLK_FREQ_11M2896, CLASSD_ACLK_RATE_11M2896_MPY_8 },
+ { 88200, CLASSD_INTPMR_FRAME_88K,
+ CLASSD_INTPMR_DSP_CLK_FREQ_11M2896, CLASSD_ACLK_RATE_11M2896_MPY_8 },
+};
+
+static int
+atmel_classd_codec_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *codec_dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
+ struct snd_soc_codec *codec = codec_dai->codec;
+ int fs;
+ int i, best, best_val, cur_val, ret;
+ u32 mask, val;
+
+ fs = params_rate(params);
+
+ best = 0;
+ best_val = abs(fs - sample_rates[0].rate);
+ for (i = 1; i < ARRAY_SIZE(sample_rates); i++) {
+ /* Closest match */
+ cur_val = abs(fs - sample_rates[i].rate);
+ if (cur_val < best_val) {
+ best = i;
+ best_val = cur_val;
+ }
+ }
+
+ dev_dbg(codec->dev,
+ "Selected SAMPLE_RATE of %dHz, ACLK_RATE of %ldHz\n",
+ sample_rates[best].rate, sample_rates[best].aclk_rate);
+
+ clk_disable_unprepare(dd->gclk);
+ clk_disable_unprepare(dd->aclk);
+
+ ret = clk_set_rate(dd->aclk, sample_rates[best].aclk_rate);
+ if (ret)
+ return ret;
+
+ mask = CLASSD_INTPMR_DSP_CLK_FREQ_MASK | CLASSD_INTPMR_FRAME_MASK;
+ val = (sample_rates[best].dsp_clk << CLASSD_INTPMR_DSP_CLK_FREQ_SHIFT)
+ | (sample_rates[best].sample_rate << CLASSD_INTPMR_FRAME_SHIFT);
+
+ snd_soc_update_bits(codec, CLASSD_INTPMR, mask, val);
+
+ ret = clk_prepare_enable(dd->aclk);
+ if (ret)
+ return ret;
+
+ return clk_prepare_enable(dd->gclk);
+}
+
+static void
+atmel_classd_codec_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *codec_dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct atmel_classd *dd = snd_soc_card_get_drvdata(rtd->card);
+
+ clk_disable_unprepare(dd->gclk);
+ clk_disable_unprepare(dd->aclk);
+}
+
+static int atmel_classd_codec_dai_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *codec_dai)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+
+ snd_soc_update_bits(codec, CLASSD_MR,
+ CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK,
+ (CLASSD_MR_LEN_DIS << CLASSD_MR_LEN_SHIFT)
+ |(CLASSD_MR_REN_DIS << CLASSD_MR_REN_SHIFT));
+
+ return 0;
+}
+
+static int atmel_classd_codec_dai_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *codec_dai)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ u32 mask, val;
+
+ mask = CLASSD_MR_LEN_MASK | CLASSD_MR_REN_MASK;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ val = mask;
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ val = (CLASSD_MR_LEN_DIS << CLASSD_MR_LEN_SHIFT)
+ | (CLASSD_MR_REN_DIS << CLASSD_MR_REN_SHIFT);
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, CLASSD_MR, mask, val);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops atmel_classd_codec_dai_ops = {
+ .digital_mute = atmel_classd_codec_dai_digital_mute,
+ .startup = atmel_classd_codec_dai_startup,
+ .shutdown = atmel_classd_codec_dai_shutdown,
+ .hw_params = atmel_classd_codec_dai_hw_params,
+ .prepare = atmel_classd_codec_dai_prepare,
+ .trigger = atmel_classd_codec_dai_trigger,
+};
+
+#define ATMEL_CLASSD_CODEC_DAI_NAME "atmel-classd-hifi"
+
+static struct snd_soc_dai_driver atmel_classd_codec_dai = {
+ .name = ATMEL_CLASSD_CODEC_DAI_NAME,
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = ATMEL_CLASSD_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ },
+ .ops = &atmel_classd_codec_dai_ops,
+};
+
+/* ASoC sound card */
+static int atmel_classd_asoc_card_init(struct device *dev,
+ struct snd_soc_card *card)
+{
+ struct snd_soc_dai_link *dai_link;
+ struct atmel_classd *dd = snd_soc_card_get_drvdata(card);
+
+ dai_link = devm_kzalloc(dev, sizeof(*dai_link), GFP_KERNEL);
+ if (!dai_link)
+ return -ENOMEM;
+
+ dai_link->name = "CLASSD";
+ dai_link->stream_name = "CLASSD PCM";
+ dai_link->codec_dai_name = ATMEL_CLASSD_CODEC_DAI_NAME;
+ dai_link->cpu_dai_name = dev_name(dev);
+ dai_link->codec_name = dev_name(dev);
+ dai_link->platform_name = dev_name(dev);
+
+ card->dai_link = dai_link;
+ card->num_links = 1;
+ card->name = dd->pdata->card_name;
+ card->dev = dev;
+
+ return 0;
+};
+
+/* regmap configuration */
+static const struct reg_default atmel_classd_reg_defaults[] = {
+ { CLASSD_INTPMR, 0x00301212 },
+};
+
+#define ATMEL_CLASSD_REG_MAX 0xE4
+static const struct regmap_config atmel_classd_regmap_config = {
+ .reg_bits = 32,
+ .reg_stride = 4,
+ .val_bits = 32,
+ .max_register = ATMEL_CLASSD_REG_MAX,
+
+ .cache_type = REGCACHE_FLAT,
+ .reg_defaults = atmel_classd_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(atmel_classd_reg_defaults),
+};
+
+static int atmel_classd_probe(struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ struct atmel_classd *dd;
+ struct resource *res;
+ void __iomem *io_base;
+ const struct atmel_classd_pdata *pdata;
+ struct snd_soc_card *card;
+ int ret;
+
+ pdata = dev_get_platdata(dev);
+ if (!pdata) {
+ pdata = atmel_classd_dt_init(dev);
+ if (IS_ERR(pdata))
+ return PTR_ERR(pdata);
+ }
+
+ dd = devm_kzalloc(dev, sizeof(*dd), GFP_KERNEL);
+ if (!dd)
+ return -ENOMEM;
+
+ dd->pdata = pdata;
+
+ dd->irq = platform_get_irq(pdev, 0);
+ if (dd->irq < 0) {
+ ret = dd->irq;
+ dev_err(dev, "failed to could not get irq: %d\n", ret);
+ return ret;
+ }
+
+ dd->pclk = devm_clk_get(dev, "pclk");
+ if (IS_ERR(dd->pclk)) {
+ ret = PTR_ERR(dd->pclk);
+ dev_err(dev, "failed to get peripheral clock: %d\n", ret);
+ return ret;
+ }
+
+ dd->gclk = devm_clk_get(dev, "gclk");
+ if (IS_ERR(dd->gclk)) {
+ ret = PTR_ERR(dd->gclk);
+ dev_err(dev, "failed to get GCK clock: %d\n", ret);
+ return ret;
+ }
+
+ dd->aclk = devm_clk_get(dev, "aclk");
+ if (IS_ERR(dd->aclk)) {
+ ret = PTR_ERR(dd->aclk);
+ dev_err(dev, "failed to get audio clock: %d\n", ret);
+ return ret;
+ }
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!res) {
+ dev_err(dev, "no memory resource\n");
+ return -ENXIO;
+ }
+
+ io_base = devm_ioremap_resource(dev, res);
+ if (IS_ERR(io_base)) {
+ ret = PTR_ERR(io_base);
+ dev_err(dev, "failed to remap register memory: %d\n", ret);
+ return ret;
+ }
+
+ dd->phy_base = res->start;
+
+ dd->regmap = devm_regmap_init_mmio(dev, io_base,
+ &atmel_classd_regmap_config);
+ if (IS_ERR(dd->regmap)) {
+ ret = PTR_ERR(dd->regmap);
+ dev_err(dev, "failed to init register map: %d\n", ret);
+ return ret;
+ }
+
+ ret = devm_snd_soc_register_component(dev,
+ &atmel_classd_cpu_dai_component,
+ &atmel_classd_cpu_dai, 1);
+ if (ret) {
+ dev_err(dev, "could not register CPU DAI: %d\n", ret);
+ return ret;
+ }
+
+ ret = devm_snd_dmaengine_pcm_register(dev,
+ &atmel_classd_dmaengine_pcm_config,
+ 0);
+ if (ret) {
+ dev_err(dev, "could not register platform: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_register_codec(dev, &soc_codec_dev_classd,
+ &atmel_classd_codec_dai, 1);
+ if (ret) {
+ dev_err(dev, "could not register codec: %d\n", ret);
+ return ret;
+ }
+
+ /* register sound card */
+ card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL);
+ if (!card)
+ return -ENOMEM;
+
+ snd_soc_card_set_drvdata(card, dd);
+ platform_set_drvdata(pdev, card);
+
+ ret = atmel_classd_asoc_card_init(dev, card);
+ if (ret) {
+ dev_err(dev, "failed to init sound card\n");
+ return ret;
+ }
+
+ ret = devm_snd_soc_register_card(dev, card);
+ if (ret) {
+ dev_err(dev, "failed to register sound card: %d\n", ret);
+ return ret;
+ }
+
+ return 0;
+}
+
+static int atmel_classd_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+static struct platform_driver atmel_classd_driver = {
+ .driver = {
+ .name = "atmel-classd",
+ .of_match_table = of_match_ptr(atmel_classd_of_match),
+ .pm = &snd_soc_pm_ops,
+ },
+ .probe = atmel_classd_probe,
+ .remove = atmel_classd_remove,
+};
+module_platform_driver(atmel_classd_driver);
+
+MODULE_DESCRIPTION("Atmel ClassD driver under ALSA SoC architecture");
+MODULE_AUTHOR("Songjun Wu <songjun.wu@atmel.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/atmel/atmel-classd.h b/sound/soc/atmel/atmel-classd.h
new file mode 100644
index 000000000000..73f8fdd1ca83
--- /dev/null
+++ b/sound/soc/atmel/atmel-classd.h
@@ -0,0 +1,120 @@
+#ifndef __ATMEL_CLASSD_H_
+#define __ATMEL_CLASSD_H_
+
+#define CLASSD_CR 0x00000000
+#define CLASSD_CR_RESET 0x1
+
+#define CLASSD_MR 0x00000004
+
+#define CLASSD_MR_LEN_DIS 0x0
+#define CLASSD_MR_LEN_EN 0x1
+#define CLASSD_MR_LEN_MASK (0x1 << 0)
+#define CLASSD_MR_LEN_SHIFT (0)
+
+#define CLASSD_MR_LMUTE_DIS 0x0
+#define CLASSD_MR_LMUTE_EN 0x1
+#define CLASSD_MR_LMUTE_SHIFT (0x1)
+#define CLASSD_MR_LMUTE_MASK (0x1 << 1)
+
+#define CLASSD_MR_REN_DIS 0x0
+#define CLASSD_MR_REN_EN 0x1
+#define CLASSD_MR_REN_MASK (0x1 << 4)
+#define CLASSD_MR_REN_SHIFT (4)
+
+#define CLASSD_MR_RMUTE_DIS 0x0
+#define CLASSD_MR_RMUTE_EN 0x1
+#define CLASSD_MR_RMUTE_SHIFT (0x5)
+#define CLASSD_MR_RMUTE_MASK (0x1 << 5)
+
+#define CLASSD_MR_PWMTYP_SINGLE 0x0
+#define CLASSD_MR_PWMTYP_DIFF 0x1
+#define CLASSD_MR_PWMTYP_MASK (0x1 << 8)
+#define CLASSD_MR_PWMTYP_SHIFT (8)
+
+#define CLASSD_MR_NON_OVERLAP_DIS 0x0
+#define CLASSD_MR_NON_OVERLAP_EN 0x1
+#define CLASSD_MR_NON_OVERLAP_MASK (0x1 << 16)
+#define CLASSD_MR_NON_OVERLAP_SHIFT (16)
+
+#define CLASSD_MR_NOVR_VAL_5NS 0x0
+#define CLASSD_MR_NOVR_VAL_10NS 0x1
+#define CLASSD_MR_NOVR_VAL_15NS 0x2
+#define CLASSD_MR_NOVR_VAL_20NS 0x3
+#define CLASSD_MR_NOVR_VAL_MASK (0x3 << 20)
+#define CLASSD_MR_NOVR_VAL_SHIFT (20)
+
+#define CLASSD_INTPMR 0x00000008
+
+#define CLASSD_INTPMR_ATTL_MASK (0x3f << 0)
+#define CLASSD_INTPMR_ATTL_SHIFT (0)
+#define CLASSD_INTPMR_ATTR_MASK (0x3f << 8)
+#define CLASSD_INTPMR_ATTR_SHIFT (8)
+
+#define CLASSD_INTPMR_DSP_CLK_FREQ_12M288 0x0
+#define CLASSD_INTPMR_DSP_CLK_FREQ_11M2896 0x1
+#define CLASSD_INTPMR_DSP_CLK_FREQ_MASK (0x1 << 16)
+#define CLASSD_INTPMR_DSP_CLK_FREQ_SHIFT (16)
+
+#define CLASSD_INTPMR_DEEMP_DIS 0x0
+#define CLASSD_INTPMR_DEEMP_EN 0x1
+#define CLASSD_INTPMR_DEEMP_MASK (0x1 << 18)
+#define CLASSD_INTPMR_DEEMP_SHIFT (18)
+
+#define CLASSD_INTPMR_SWAP_LEFT_ON_LSB 0x0
+#define CLASSD_INTPMR_SWAP_RIGHT_ON_LSB 0x1
+#define CLASSD_INTPMR_SWAP_MASK (0x1 << 19)
+#define CLASSD_INTPMR_SWAP_SHIFT (19)
+
+#define CLASSD_INTPMR_FRAME_8K 0x0
+#define CLASSD_INTPMR_FRAME_16K 0x1
+#define CLASSD_INTPMR_FRAME_32K 0x2
+#define CLASSD_INTPMR_FRAME_48K 0x3
+#define CLASSD_INTPMR_FRAME_96K 0x4
+#define CLASSD_INTPMR_FRAME_22K 0x5
+#define CLASSD_INTPMR_FRAME_44K 0x6
+#define CLASSD_INTPMR_FRAME_88K 0x7
+#define CLASSD_INTPMR_FRAME_MASK (0x7 << 20)
+#define CLASSD_INTPMR_FRAME_SHIFT (20)
+
+#define CLASSD_INTPMR_EQCFG_FLAT 0x0
+#define CLASSD_INTPMR_EQCFG_B_BOOST_12 0x1
+#define CLASSD_INTPMR_EQCFG_B_BOOST_6 0x2
+#define CLASSD_INTPMR_EQCFG_B_CUT_12 0x3
+#define CLASSD_INTPMR_EQCFG_B_CUT_6 0x4
+#define CLASSD_INTPMR_EQCFG_M_BOOST_3 0x5
+#define CLASSD_INTPMR_EQCFG_M_BOOST_8 0x6
+#define CLASSD_INTPMR_EQCFG_M_CUT_3 0x7
+#define CLASSD_INTPMR_EQCFG_M_CUT_8 0x8
+#define CLASSD_INTPMR_EQCFG_T_BOOST_12 0x9
+#define CLASSD_INTPMR_EQCFG_T_BOOST_6 0xa
+#define CLASSD_INTPMR_EQCFG_T_CUT_12 0xb
+#define CLASSD_INTPMR_EQCFG_T_CUT_6 0xc
+#define CLASSD_INTPMR_EQCFG_SHIFT (24)
+
+#define CLASSD_INTPMR_MONO_DIS 0x0
+#define CLASSD_INTPMR_MONO_EN 0x1
+#define CLASSD_INTPMR_MONO_MASK (0x1 << 28)
+#define CLASSD_INTPMR_MONO_SHIFT (28)
+
+#define CLASSD_INTPMR_MONO_MODE_MIX 0x0
+#define CLASSD_INTPMR_MONO_MODE_SAT 0x1
+#define CLASSD_INTPMR_MONO_MODE_LEFT 0x2
+#define CLASSD_INTPMR_MONO_MODE_RIGHT 0x3
+#define CLASSD_INTPMR_MONO_MODE_MASK (0x3 << 29)
+#define CLASSD_INTPMR_MONO_MODE_SHIFT (29)
+
+#define CLASSD_INTSR 0x0000000c
+
+#define CLASSD_THR 0x00000010
+
+#define CLASSD_IER 0x00000014
+
+#define CLASSD_IDR 0x00000018
+
+#define CLASSD_IMR 0x0000001c
+
+#define CLASSD_ISR 0x00000020
+
+#define CLASSD_WPMR 0x000000e4
+
+#endif
diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c
index aa354e1c6ff7..1933bcd46cca 100644
--- a/sound/soc/atmel/atmel_wm8904.c
+++ b/sound/soc/atmel/atmel_wm8904.c
@@ -176,6 +176,7 @@ static const struct of_device_id atmel_asoc_wm8904_dt_ids[] = {
{ .compatible = "atmel,asoc-wm8904", },
{ }
};
+MODULE_DEVICE_TABLE(of, atmel_asoc_wm8904_dt_ids);
#endif
static struct platform_driver atmel_asoc_wm8904_driver = {
diff --git a/sound/soc/au1x/db1000.c b/sound/soc/au1x/db1000.c
index 452f404abfd2..e97c32798e98 100644
--- a/sound/soc/au1x/db1000.c
+++ b/sound/soc/au1x/db1000.c
@@ -38,14 +38,7 @@ static int db1000_audio_probe(struct platform_device *pdev)
{
struct snd_soc_card *card = &db1000_ac97;
card->dev = &pdev->dev;
- return snd_soc_register_card(card);
-}
-
-static int db1000_audio_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
- snd_soc_unregister_card(card);
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, card);
}
static struct platform_driver db1000_audio_driver = {
@@ -54,7 +47,6 @@ static struct platform_driver db1000_audio_driver = {
.pm = &snd_soc_pm_ops,
},
.probe = db1000_audio_probe,
- .remove = db1000_audio_remove,
};
module_platform_driver(db1000_audio_driver);
diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c
index 8c907ebea189..5c73061d912a 100644
--- a/sound/soc/au1x/db1200.c
+++ b/sound/soc/au1x/db1200.c
@@ -178,14 +178,7 @@ static int db1200_audio_probe(struct platform_device *pdev)
card = db1200_cards[pid->driver_data];
card->dev = &pdev->dev;
- return snd_soc_register_card(card);
-}
-
-static int db1200_audio_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
- snd_soc_unregister_card(card);
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, card);
}
static struct platform_driver db1200_audio_driver = {
@@ -195,7 +188,6 @@ static struct platform_driver db1200_audio_driver = {
},
.id_table = db1200_pids,
.probe = db1200_audio_probe,
- .remove = db1200_audio_remove,
};
module_platform_driver(db1200_audio_driver);
diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c
index 5bf1501e5e3c..864df2616e10 100644
--- a/sound/soc/blackfin/bf5xx-ad1836.c
+++ b/sound/soc/blackfin/bf5xx-ad1836.c
@@ -87,27 +87,18 @@ static int bf5xx_ad1836_driver_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
platform_set_drvdata(pdev, card);
- ret = snd_soc_register_card(card);
+ ret = devm_snd_soc_register_card(&pdev->dev, card);
if (ret)
dev_err(&pdev->dev, "Failed to register card\n");
return ret;
}
-static int bf5xx_ad1836_driver_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
- return 0;
-}
-
static struct platform_driver bf5xx_ad1836_driver = {
.driver = {
.name = "bfin-snd-ad1836",
.pm = &snd_soc_pm_ops,
},
.probe = bf5xx_ad1836_driver_probe,
- .remove = bf5xx_ad1836_driver_remove,
};
module_platform_driver(bf5xx_ad1836_driver);
diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c
index 523baf5820d7..72ac78988426 100644
--- a/sound/soc/blackfin/bfin-eval-adau1373.c
+++ b/sound/soc/blackfin/bfin-eval-adau1373.c
@@ -154,16 +154,7 @@ static int bfin_eval_adau1373_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- return snd_soc_register_card(&bfin_eval_adau1373);
-}
-
-static int bfin_eval_adau1373_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1373);
}
static struct platform_driver bfin_eval_adau1373_driver = {
@@ -172,7 +163,6 @@ static struct platform_driver bfin_eval_adau1373_driver = {
.pm = &snd_soc_pm_ops,
},
.probe = bfin_eval_adau1373_probe,
- .remove = bfin_eval_adau1373_remove,
};
module_platform_driver(bfin_eval_adau1373_driver);
diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c
index f9e926dfd4ef..5c67f72cf9a9 100644
--- a/sound/soc/blackfin/bfin-eval-adau1701.c
+++ b/sound/soc/blackfin/bfin-eval-adau1701.c
@@ -94,16 +94,7 @@ static int bfin_eval_adau1701_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- return snd_soc_register_card(&bfin_eval_adau1701);
-}
-
-static int bfin_eval_adau1701_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1701);
}
static struct platform_driver bfin_eval_adau1701_driver = {
@@ -112,7 +103,6 @@ static struct platform_driver bfin_eval_adau1701_driver = {
.pm = &snd_soc_pm_ops,
},
.probe = bfin_eval_adau1701_probe,
- .remove = bfin_eval_adau1701_remove,
};
module_platform_driver(bfin_eval_adau1701_driver);
diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c
index 27eee66afdb2..1037477d10b2 100644
--- a/sound/soc/blackfin/bfin-eval-adav80x.c
+++ b/sound/soc/blackfin/bfin-eval-adav80x.c
@@ -119,16 +119,7 @@ static int bfin_eval_adav80x_probe(struct platform_device *pdev)
card->dev = &pdev->dev;
- return snd_soc_register_card(&bfin_eval_adav80x);
-}
-
-static int bfin_eval_adav80x_remove(struct platform_device *pdev)
-{
- struct snd_soc_card *card = platform_get_drvdata(pdev);
-
- snd_soc_unregister_card(card);
-
- return 0;
+ return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adav80x);
}
static const struct platform_device_id bfin_eval_adav80x_ids[] = {
@@ -144,7 +135,6 @@ static struct platform_driver bfin_eval_adav80x_driver = {
.pm = &snd_soc_pm_ops,
},
.probe = bfin_eval_adav80x_probe,
- .remove = bfin_eval_adav80x_remove,
.id_table = bfin_eval_adav80x_ids,
};
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 0c9733ecd17f..cfdafc4c11ea 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_AK4104 if SPI_MASTER
select SND_SOC_AK4535 if I2C
select SND_SOC_AK4554
+ select SND_SOC_AK4613 if I2C
select SND_SOC_AK4641 if I2C
select SND_SOC_AK4642 if I2C
select SND_SOC_AK4671 if I2C
@@ -57,6 +58,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_CX20442 if TTY
select SND_SOC_DA7210 if SND_SOC_I2C_AND_SPI
select SND_SOC_DA7213 if I2C
+ select SND_SOC_DA7219 if I2C
select SND_SOC_DA732X if I2C
select SND_SOC_DA9055 if I2C
select SND_SOC_DMIC
@@ -79,7 +81,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_MAX9877 if I2C
select SND_SOC_MC13783 if MFD_MC13XXX
select SND_SOC_ML26124 if I2C
- select SND_SOC_HDMI_CODEC
+ select SND_SOC_NAU8825 if I2C
select SND_SOC_PCM1681 if I2C
select SND_SOC_PCM1792A if SPI_MASTER
select SND_SOC_PCM3008
@@ -171,6 +173,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_WM8995 if SND_SOC_I2C_AND_SPI
select SND_SOC_WM8996 if I2C
select SND_SOC_WM8997 if MFD_WM8997
+ select SND_SOC_WM8998 if MFD_WM8998
select SND_SOC_WM9081 if I2C
select SND_SOC_WM9090 if I2C
select SND_SOC_WM9705 if SND_SOC_AC97_BUS
@@ -195,9 +198,11 @@ config SND_SOC_ARIZONA
default y if SND_SOC_WM5102=y
default y if SND_SOC_WM5110=y
default y if SND_SOC_WM8997=y
+ default y if SND_SOC_WM8998=y
default m if SND_SOC_WM5102=m
default m if SND_SOC_WM5110=m
default m if SND_SOC_WM8997=m
+ default m if SND_SOC_WM8998=m
config SND_SOC_WM_HUBS
tristate
@@ -319,6 +324,10 @@ config SND_SOC_AK4535
config SND_SOC_AK4554
tristate "AKM AK4554 CODEC"
+config SND_SOC_AK4613
+ tristate "AKM AK4613 CODEC"
+ depends on I2C
+
config SND_SOC_AK4641
tristate
@@ -430,6 +439,9 @@ config SND_SOC_DA7210
config SND_SOC_DA7213
tristate
+config SND_SOC_DA7219
+ tristate
+
config SND_SOC_DA732X
tristate
@@ -442,9 +454,6 @@ config SND_SOC_BT_SCO
config SND_SOC_DMIC
tristate
-config SND_SOC_HDMI_CODEC
- tristate "HDMI stub CODEC"
-
config SND_SOC_ES8328
tristate "Everest Semi ES8328 CODEC"
@@ -865,6 +874,9 @@ config SND_SOC_WM8996
config SND_SOC_WM8997
tristate
+config SND_SOC_WM8998
+ tristate
+
config SND_SOC_WM9081
tristate
@@ -896,6 +908,9 @@ config SND_SOC_MC13783
config SND_SOC_ML26124
tristate
+config SND_SOC_NAU8825
+ tristate
+
config SND_SOC_TPA6130A2
tristate "Texas Instruments TPA6130A2 headphone amplifier"
depends on I2C
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 4a32077954ae..f632fc42f59f 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -26,6 +26,7 @@ snd-soc-ads117x-objs := ads117x.o
snd-soc-ak4104-objs := ak4104.o
snd-soc-ak4535-objs := ak4535.o
snd-soc-ak4554-objs := ak4554.o
+snd-soc-ak4613-objs := ak4613.o
snd-soc-ak4641-objs := ak4641.o
snd-soc-ak4642-objs := ak4642.o
snd-soc-ak4671-objs := ak4671.o
@@ -49,6 +50,7 @@ snd-soc-cs4349-objs := cs4349.o
snd-soc-cx20442-objs := cx20442.o
snd-soc-da7210-objs := da7210.o
snd-soc-da7213-objs := da7213.o
+snd-soc-da7219-objs := da7219.o da7219-aad.o
snd-soc-da732x-objs := da732x.o
snd-soc-da9055-objs := da9055.o
snd-soc-bt-sco-objs := bt-sco.o
@@ -72,7 +74,7 @@ snd-soc-max98925-objs := max98925.o
snd-soc-max9850-objs := max9850.o
snd-soc-mc13783-objs := mc13783.o
snd-soc-ml26124-objs := ml26124.o
-snd-soc-hdmi-codec-objs := hdmi.o
+snd-soc-nau8825-objs := nau8825.o
snd-soc-pcm1681-objs := pcm1681.o
snd-soc-pcm1792a-codec-objs := pcm1792a.o
snd-soc-pcm3008-objs := pcm3008.o
@@ -176,6 +178,7 @@ snd-soc-wm8993-objs := wm8993.o
snd-soc-wm8994-objs := wm8994.o wm8958-dsp2.o
snd-soc-wm8995-objs := wm8995.o
snd-soc-wm8997-objs := wm8997.o
+snd-soc-wm8998-objs := wm8998.o
snd-soc-wm9081-objs := wm9081.o
snd-soc-wm9090-objs := wm9090.o
snd-soc-wm9705-objs := wm9705.o
@@ -216,6 +219,7 @@ obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o
obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o
obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o
obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o
+obj-$(CONFIG_SND_SOC_AK4613) += snd-soc-ak4613.o
obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o
obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o
obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o
@@ -241,6 +245,7 @@ obj-$(CONFIG_SND_SOC_CS4349) += snd-soc-cs4349.o
obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o
obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o
obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o
+obj-$(CONFIG_SND_SOC_DA7219) += snd-soc-da7219.o
obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o
obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o
obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o
@@ -264,7 +269,7 @@ obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o
obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o
obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o
obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o
-obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o
+obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o
obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o
obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
@@ -364,6 +369,7 @@ obj-$(CONFIG_SND_SOC_WM8993) += snd-soc-wm8993.o
obj-$(CONFIG_SND_SOC_WM8994) += snd-soc-wm8994.o
obj-$(CONFIG_SND_SOC_WM8995) += snd-soc-wm8995.o
obj-$(CONFIG_SND_SOC_WM8997) += snd-soc-wm8997.o
+obj-$(CONFIG_SND_SOC_WM8998) += snd-soc-wm8998.o
obj-$(CONFIG_SND_SOC_WM9081) += snd-soc-wm9081.o
obj-$(CONFIG_SND_SOC_WM9090) += snd-soc-wm9090.o
obj-$(CONFIG_SND_SOC_WM9705) += snd-soc-wm9705.o
diff --git a/sound/soc/codecs/ad193x-i2c.c b/sound/soc/codecs/ad193x-i2c.c
index df3a1a415825..171313664bc8 100644
--- a/sound/soc/codecs/ad193x-i2c.c
+++ b/sound/soc/codecs/ad193x-i2c.c
@@ -15,8 +15,8 @@
#include "ad193x.h"
static const struct i2c_device_id ad193x_id[] = {
- { "ad1936", 0 },
- { "ad1937", 0 },
+ { "ad1936", AD193X },
+ { "ad1937", AD193X },
{ }
};
MODULE_DEVICE_TABLE(i2c, ad193x_id);
@@ -30,7 +30,9 @@ static int ad193x_i2c_probe(struct i2c_client *client,
config.val_bits = 8;
config.reg_bits = 8;
- return ad193x_probe(&client->dev, devm_regmap_init_i2c(client, &config));
+ return ad193x_probe(&client->dev,
+ devm_regmap_init_i2c(client, &config),
+ (enum ad193x_type)id->driver_data);
}
static int ad193x_i2c_remove(struct i2c_client *client)
diff --git a/sound/soc/codecs/ad193x-spi.c b/sound/soc/codecs/ad193x-spi.c
index 390cef9b9dc2..431f95da1de1 100644
--- a/sound/soc/codecs/ad193x-spi.c
+++ b/sound/soc/codecs/ad193x-spi.c
@@ -16,6 +16,7 @@
static int ad193x_spi_probe(struct spi_device *spi)
{
+ const struct spi_device_id *id = spi_get_device_id(spi);
struct regmap_config config;
config = ad193x_regmap_config;
@@ -24,7 +25,8 @@ static int ad193x_spi_probe(struct spi_device *spi)
config.read_flag_mask = 0x09;
config.write_flag_mask = 0x08;
- return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config));
+ return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config),
+ (enum ad193x_type)id->driver_data);
}
static int ad193x_spi_remove(struct spi_device *spi)
@@ -33,6 +35,17 @@ static int ad193x_spi_remove(struct spi_device *spi)
return 0;
}
+static const struct spi_device_id ad193x_spi_id[] = {
+ { "ad193x", AD193X },
+ { "ad1933", AD1933 },
+ { "ad1934", AD1934 },
+ { "ad1938", AD193X },
+ { "ad1939", AD193X },
+ { "adau1328", AD193X },
+ { }
+};
+MODULE_DEVICE_TABLE(spi, ad193x_spi_id);
+
static struct spi_driver ad193x_spi_driver = {
.driver = {
.name = "ad193x",
@@ -40,6 +53,7 @@ static struct spi_driver ad193x_spi_driver = {
},
.probe = ad193x_spi_probe,
.remove = ad193x_spi_remove,
+ .id_table = ad193x_spi_id,
};
module_spi_driver(ad193x_spi_driver);
diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c
index 17c953595660..3a3f3f2343d7 100644
--- a/sound/soc/codecs/ad193x.c
+++ b/sound/soc/codecs/ad193x.c
@@ -23,6 +23,7 @@
/* codec private data */
struct ad193x_priv {
struct regmap *regmap;
+ enum ad193x_type type;
int sysclk;
};
@@ -47,12 +48,6 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = {
SOC_DOUBLE_R_TLV("DAC4 Volume", AD193X_DAC_L4_VOL,
AD193X_DAC_R4_VOL, 0, 0xFF, 1, adau193x_tlv),
- /* ADC switch control */
- SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE,
- AD193X_ADCR1_MUTE, 1, 1),
- SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE,
- AD193X_ADCR2_MUTE, 1, 1),
-
/* DAC switch control */
SOC_DOUBLE("DAC1 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL1_MUTE,
AD193X_DACR1_MUTE, 1, 1),
@@ -63,26 +58,37 @@ static const struct snd_kcontrol_new ad193x_snd_controls[] = {
SOC_DOUBLE("DAC4 Switch", AD193X_DAC_CHNL_MUTE, AD193X_DACL4_MUTE,
AD193X_DACR4_MUTE, 1, 1),
+ /* DAC de-emphasis */
+ SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum),
+};
+
+static const struct snd_kcontrol_new ad193x_adc_snd_controls[] = {
+ /* ADC switch control */
+ SOC_DOUBLE("ADC1 Switch", AD193X_ADC_CTRL0, AD193X_ADCL1_MUTE,
+ AD193X_ADCR1_MUTE, 1, 1),
+ SOC_DOUBLE("ADC2 Switch", AD193X_ADC_CTRL0, AD193X_ADCL2_MUTE,
+ AD193X_ADCR2_MUTE, 1, 1),
+
/* ADC high-pass filter */
SOC_SINGLE("ADC High Pass Filter Switch", AD193X_ADC_CTRL0,
AD193X_ADC_HIGHPASS_FILTER, 1, 0),
-
- /* DAC de-emphasis */
- SOC_ENUM("Playback Deemphasis", ad193x_deemp_enum),
};
static const struct snd_soc_dapm_widget ad193x_dapm_widgets[] = {
SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_PGA("DAC Output", AD193X_DAC_CTRL0, 0, 1, NULL, 0),
- SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
SND_SOC_DAPM_SUPPLY("PLL_PWR", AD193X_PLL_CLK_CTRL0, 0, 1, NULL, 0),
- SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0),
SND_SOC_DAPM_SUPPLY("SYSCLK", AD193X_PLL_CLK_CTRL0, 7, 0, NULL, 0),
SND_SOC_DAPM_VMID("VMID"),
SND_SOC_DAPM_OUTPUT("DAC1OUT"),
SND_SOC_DAPM_OUTPUT("DAC2OUT"),
SND_SOC_DAPM_OUTPUT("DAC3OUT"),
SND_SOC_DAPM_OUTPUT("DAC4OUT"),
+};
+
+static const struct snd_soc_dapm_widget ad193x_adc_widgets[] = {
+ SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_SUPPLY("ADC_PWR", AD193X_ADC_CTRL0, 0, 1, NULL, 0),
SND_SOC_DAPM_INPUT("ADC1IN"),
SND_SOC_DAPM_INPUT("ADC2IN"),
};
@@ -91,18 +97,33 @@ static const struct snd_soc_dapm_route audio_paths[] = {
{ "DAC", NULL, "SYSCLK" },
{ "DAC Output", NULL, "DAC" },
{ "DAC Output", NULL, "VMID" },
- { "ADC", NULL, "SYSCLK" },
- { "DAC", NULL, "ADC_PWR" },
- { "ADC", NULL, "ADC_PWR" },
{ "DAC1OUT", NULL, "DAC Output" },
{ "DAC2OUT", NULL, "DAC Output" },
{ "DAC3OUT", NULL, "DAC Output" },
{ "DAC4OUT", NULL, "DAC Output" },
+ { "SYSCLK", NULL, "PLL_PWR" },
+};
+
+static const struct snd_soc_dapm_route ad193x_adc_audio_paths[] = {
+ { "ADC", NULL, "SYSCLK" },
+ { "ADC", NULL, "ADC_PWR" },
{ "ADC", NULL, "ADC1IN" },
{ "ADC", NULL, "ADC2IN" },
- { "SYSCLK", NULL, "PLL_PWR" },
};
+static inline bool ad193x_has_adc(const struct ad193x_priv *ad193x)
+{
+ switch (ad193x->type) {
+ case AD1933:
+ case AD1934:
+ return false;
+ default:
+ break;
+ }
+
+ return true;
+}
+
/*
* DAI ops entries
*/
@@ -147,8 +168,10 @@ static int ad193x_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1,
AD193X_DAC_CHAN_MASK, channels << AD193X_DAC_CHAN_SHFT);
- regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2,
- AD193X_ADC_CHAN_MASK, channels << AD193X_ADC_CHAN_SHFT);
+ if (ad193x_has_adc(ad193x))
+ regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2,
+ AD193X_ADC_CHAN_MASK,
+ channels << AD193X_ADC_CHAN_SHFT);
return 0;
}
@@ -172,7 +195,9 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai,
adc_serfmt |= AD193X_ADC_SERFMT_AUX;
break;
default:
- return -EINVAL;
+ if (ad193x_has_adc(ad193x))
+ return -EINVAL;
+ break;
}
switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
@@ -217,10 +242,12 @@ static int ad193x_set_dai_fmt(struct snd_soc_dai *codec_dai,
return -EINVAL;
}
- regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1,
- AD193X_ADC_SERFMT_MASK, adc_serfmt);
- regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2,
- AD193X_ADC_FMT_MASK, adc_fmt);
+ if (ad193x_has_adc(ad193x)) {
+ regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1,
+ AD193X_ADC_SERFMT_MASK, adc_serfmt);
+ regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL2,
+ AD193X_ADC_FMT_MASK, adc_fmt);
+ }
regmap_update_bits(ad193x->regmap, AD193X_DAC_CTRL1,
AD193X_DAC_FMT_MASK, dac_fmt);
@@ -287,8 +314,9 @@ static int ad193x_hw_params(struct snd_pcm_substream *substream,
AD193X_DAC_WORD_LEN_MASK,
word_len << AD193X_DAC_WORD_LEN_SHFT);
- regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1,
- AD193X_ADC_WORD_LEN_MASK, word_len);
+ if (ad193x_has_adc(ad193x))
+ regmap_update_bits(ad193x->regmap, AD193X_ADC_CTRL1,
+ AD193X_ADC_WORD_LEN_MASK, word_len);
return 0;
}
@@ -326,6 +354,8 @@ static struct snd_soc_dai_driver ad193x_dai = {
static int ad193x_codec_probe(struct snd_soc_codec *codec)
{
struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec);
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ int num, ret;
/* default setting for ad193x */
@@ -335,14 +365,46 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec)
regmap_write(ad193x->regmap, AD193X_DAC_CTRL2, 0x1A);
/* dac in tdm mode */
regmap_write(ad193x->regmap, AD193X_DAC_CTRL0, 0x40);
- /* high-pass filter enable */
- regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3);
- /* sata delay=1, adc aux mode */
- regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43);
+
+ /* adc only */
+ if (ad193x_has_adc(ad193x)) {
+ /* high-pass filter enable */
+ regmap_write(ad193x->regmap, AD193X_ADC_CTRL0, 0x3);
+ /* sata delay=1, adc aux mode */
+ regmap_write(ad193x->regmap, AD193X_ADC_CTRL1, 0x43);
+ }
+
/* pll input: mclki/xi */
regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */
regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04);
+ /* adc only */
+ if (ad193x_has_adc(ad193x)) {
+ /* add adc controls */
+ num = ARRAY_SIZE(ad193x_adc_snd_controls);
+ ret = snd_soc_add_codec_controls(codec,
+ ad193x_adc_snd_controls,
+ num);
+ if (ret)
+ return ret;
+
+ /* add adc widgets */
+ num = ARRAY_SIZE(ad193x_adc_widgets);
+ ret = snd_soc_dapm_new_controls(dapm,
+ ad193x_adc_widgets,
+ num);
+ if (ret)
+ return ret;
+
+ /* add adc routes */
+ num = ARRAY_SIZE(ad193x_adc_audio_paths);
+ ret = snd_soc_dapm_add_routes(dapm,
+ ad193x_adc_audio_paths,
+ num);
+ if (ret)
+ return ret;
+ }
+
return 0;
}
@@ -356,18 +418,13 @@ static struct snd_soc_codec_driver soc_codec_dev_ad193x = {
.num_dapm_routes = ARRAY_SIZE(audio_paths),
};
-static bool adau193x_reg_volatile(struct device *dev, unsigned int reg)
-{
- return false;
-}
-
const struct regmap_config ad193x_regmap_config = {
.max_register = AD193X_NUM_REGS - 1,
- .volatile_reg = adau193x_reg_volatile,
};
EXPORT_SYMBOL_GPL(ad193x_regmap_config);
-int ad193x_probe(struct device *dev, struct regmap *regmap)
+int ad193x_probe(struct device *dev, struct regmap *regmap,
+ enum ad193x_type type)
{
struct ad193x_priv *ad193x;
@@ -379,6 +436,7 @@ int ad193x_probe(struct device *dev, struct regmap *regmap)
return -ENOMEM;
ad193x->regmap = regmap;
+ ad193x->type = type;
dev_set_drvdata(dev, ad193x);
diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h
index ab9a998f15be..8b1e65f928d2 100644
--- a/sound/soc/codecs/ad193x.h
+++ b/sound/soc/codecs/ad193x.h
@@ -13,8 +13,15 @@
struct device;
+enum ad193x_type {
+ AD193X,
+ AD1933,
+ AD1934,
+};
+
extern const struct regmap_config ad193x_regmap_config;
-int ad193x_probe(struct device *dev, struct regmap *regmap);
+int ad193x_probe(struct device *dev, struct regmap *regmap,
+ enum ad193x_type type);
#define AD193X_PLL_CLK_CTRL0 0x00
#define AD193X_PLL_POWERDOWN 0x01
diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c
index 198c924551b7..acff8d62059c 100644
--- a/sound/soc/codecs/adav80x.c
+++ b/sound/soc/codecs/adav80x.c
@@ -728,8 +728,8 @@ static int adav80x_dai_startup(struct snd_pcm_substream *substream,
if (!snd_soc_codec_is_active(codec) || !adav80x->rate)
return 0;
- return snd_pcm_hw_constraint_minmax(substream->runtime,
- SNDRV_PCM_HW_PARAM_RATE, adav80x->rate, adav80x->rate);
+ return snd_pcm_hw_constraint_single(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE, adav80x->rate);
}
static void adav80x_dai_shutdown(struct snd_pcm_substream *substream,
diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c
new file mode 100644
index 000000000000..07a266460ec3
--- /dev/null
+++ b/sound/soc/codecs/ak4613.c
@@ -0,0 +1,497 @@
+/*
+ * ak4613.c -- Asahi Kasei ALSA Soc Audio driver
+ *
+ * Copyright (C) 2015 Renesas Electronics Corporation
+ * Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
+ *
+ * Based on ak4642.c by Kuninori Morimoto
+ * Based on wm8731.c by Richard Purdie
+ * Based on ak4535.c by Richard Purdie
+ * Based on wm8753.c by Liam Girdwood
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <linux/slab.h>
+#include <linux/of_device.h>
+#include <linux/module.h>
+#include <linux/regmap.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+#include <sound/tlv.h>
+
+#define PW_MGMT1 0x00 /* Power Management 1 */
+#define PW_MGMT2 0x01 /* Power Management 2 */
+#define PW_MGMT3 0x02 /* Power Management 3 */
+#define CTRL1 0x03 /* Control 1 */
+#define CTRL2 0x04 /* Control 2 */
+#define DEMP1 0x05 /* De-emphasis1 */
+#define DEMP2 0x06 /* De-emphasis2 */
+#define OFD 0x07 /* Overflow Detect */
+#define ZRD 0x08 /* Zero Detect */
+#define ICTRL 0x09 /* Input Control */
+#define OCTRL 0x0a /* Output Control */
+#define LOUT1 0x0b /* LOUT1 Volume Control */
+#define ROUT1 0x0c /* ROUT1 Volume Control */
+#define LOUT2 0x0d /* LOUT2 Volume Control */
+#define ROUT2 0x0e /* ROUT2 Volume Control */
+#define LOUT3 0x0f /* LOUT3 Volume Control */
+#define ROUT3 0x10 /* ROUT3 Volume Control */
+#define LOUT4 0x11 /* LOUT4 Volume Control */
+#define ROUT4 0x12 /* ROUT4 Volume Control */
+#define LOUT5 0x13 /* LOUT5 Volume Control */
+#define ROUT5 0x14 /* ROUT5 Volume Control */
+#define LOUT6 0x15 /* LOUT6 Volume Control */
+#define ROUT6 0x16 /* ROUT6 Volume Control */
+
+/* PW_MGMT1 */
+#define RSTN BIT(0)
+#define PMDAC BIT(1)
+#define PMADC BIT(2)
+#define PMVR BIT(3)
+
+/* PW_MGMT2 */
+#define PMAD_ALL 0x7
+
+/* PW_MGMT3 */
+#define PMDA_ALL 0x3f
+
+/* CTRL1 */
+#define DIF0 BIT(3)
+#define DIF1 BIT(4)
+#define DIF2 BIT(5)
+#define TDM0 BIT(6)
+#define TDM1 BIT(7)
+#define NO_FMT (0xff)
+#define FMT_MASK (0xf8)
+
+/* CTRL2 */
+#define DFS_NORMAL_SPEED (0 << 2)
+#define DFS_DOUBLE_SPEED (1 << 2)
+#define DFS_QUAD_SPEED (2 << 2)
+
+struct ak4613_priv {
+ struct mutex lock;
+
+ unsigned int fmt;
+ u8 fmt_ctrl;
+ int cnt;
+};
+
+struct ak4613_formats {
+ unsigned int width;
+ unsigned int fmt;
+};
+
+struct ak4613_interface {
+ struct ak4613_formats capture;
+ struct ak4613_formats playback;
+};
+
+/*
+ * Playback Volume
+ *
+ * max : 0x00 : 0 dB
+ * ( 0.5 dB step )
+ * min : 0xFE : -127.0 dB
+ * mute: 0xFF
+ */
+static const DECLARE_TLV_DB_SCALE(out_tlv, -12750, 50, 1);
+
+static const struct snd_kcontrol_new ak4613_snd_controls[] = {
+ SOC_DOUBLE_R_TLV("Digital Playback Volume1", LOUT1, ROUT1,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume2", LOUT2, ROUT2,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume3", LOUT3, ROUT3,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume4", LOUT4, ROUT4,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume5", LOUT5, ROUT5,
+ 0, 0xFF, 1, out_tlv),
+ SOC_DOUBLE_R_TLV("Digital Playback Volume6", LOUT6, ROUT6,
+ 0, 0xFF, 1, out_tlv),
+};
+
+static const struct reg_default ak4613_reg[] = {
+ { 0x0, 0x0f }, { 0x1, 0x07 }, { 0x2, 0x3f }, { 0x3, 0x20 },
+ { 0x4, 0x20 }, { 0x5, 0x55 }, { 0x6, 0x05 }, { 0x7, 0x07 },
+ { 0x8, 0x0f }, { 0x9, 0x07 }, { 0xa, 0x3f }, { 0xb, 0x00 },
+ { 0xc, 0x00 }, { 0xd, 0x00 }, { 0xe, 0x00 }, { 0xf, 0x00 },
+ { 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, { 0x13, 0x00 },
+ { 0x14, 0x00 }, { 0x15, 0x00 }, { 0x16, 0x00 },
+};
+
+#define AUDIO_IFACE_IDX_TO_VAL(i) (i << 3)
+#define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt }
+static const struct ak4613_interface ak4613_iface[] = {
+ /* capture */ /* playback */
+ [0] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(16, RIGHT_J) },
+ [1] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(20, RIGHT_J) },
+ [2] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, RIGHT_J) },
+ [3] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, LEFT_J) },
+ [4] = { AUDIO_IFACE(24, I2S), AUDIO_IFACE(24, I2S) },
+};
+
+static const struct regmap_config ak4613_regmap_cfg = {
+ .reg_bits = 8,
+ .val_bits = 8,
+ .max_register = 0x16,
+ .reg_defaults = ak4613_reg,
+ .num_reg_defaults = ARRAY_SIZE(ak4613_reg),
+};
+
+static const struct of_device_id ak4613_of_match[] = {
+ { .compatible = "asahi-kasei,ak4613", .data = &ak4613_regmap_cfg },
+ {},
+};
+MODULE_DEVICE_TABLE(of, ak4613_of_match);
+
+static const struct i2c_device_id ak4613_i2c_id[] = {
+ { "ak4613", (kernel_ulong_t)&ak4613_regmap_cfg },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, ak4613_i2c_id);
+
+static const struct snd_soc_dapm_widget ak4613_dapm_widgets[] = {
+
+ /* Outputs */
+ SND_SOC_DAPM_OUTPUT("LOUT1"),
+ SND_SOC_DAPM_OUTPUT("LOUT2"),
+ SND_SOC_DAPM_OUTPUT("LOUT3"),
+ SND_SOC_DAPM_OUTPUT("LOUT4"),
+ SND_SOC_DAPM_OUTPUT("LOUT5"),
+ SND_SOC_DAPM_OUTPUT("LOUT6"),
+
+ SND_SOC_DAPM_OUTPUT("ROUT1"),
+ SND_SOC_DAPM_OUTPUT("ROUT2"),
+ SND_SOC_DAPM_OUTPUT("ROUT3"),
+ SND_SOC_DAPM_OUTPUT("ROUT4"),
+ SND_SOC_DAPM_OUTPUT("ROUT5"),
+ SND_SOC_DAPM_OUTPUT("ROUT6"),
+
+ /* Inputs */
+ SND_SOC_DAPM_INPUT("LIN1"),
+ SND_SOC_DAPM_INPUT("LIN2"),
+
+ SND_SOC_DAPM_INPUT("RIN1"),
+ SND_SOC_DAPM_INPUT("RIN2"),
+
+ /* DAC */
+ SND_SOC_DAPM_DAC("DAC1", NULL, PW_MGMT3, 0, 0),
+ SND_SOC_DAPM_DAC("DAC2", NULL, PW_MGMT3, 1, 0),
+ SND_SOC_DAPM_DAC("DAC3", NULL, PW_MGMT3, 2, 0),
+ SND_SOC_DAPM_DAC("DAC4", NULL, PW_MGMT3, 3, 0),
+ SND_SOC_DAPM_DAC("DAC5", NULL, PW_MGMT3, 4, 0),
+ SND_SOC_DAPM_DAC("DAC6", NULL, PW_MGMT3, 5, 0),
+
+ /* ADC */
+ SND_SOC_DAPM_ADC("ADC1", NULL, PW_MGMT2, 0, 0),
+ SND_SOC_DAPM_ADC("ADC2", NULL, PW_MGMT2, 1, 0),
+};
+
+static const struct snd_soc_dapm_route ak4613_intercon[] = {
+ {"LOUT1", NULL, "DAC1"},
+ {"LOUT2", NULL, "DAC2"},
+ {"LOUT3", NULL, "DAC3"},
+ {"LOUT4", NULL, "DAC4"},
+ {"LOUT5", NULL, "DAC5"},
+ {"LOUT6", NULL, "DAC6"},
+
+ {"ROUT1", NULL, "DAC1"},
+ {"ROUT2", NULL, "DAC2"},
+ {"ROUT3", NULL, "DAC3"},
+ {"ROUT4", NULL, "DAC4"},
+ {"ROUT5", NULL, "DAC5"},
+ {"ROUT6", NULL, "DAC6"},
+
+ {"DAC1", NULL, "Playback"},
+ {"DAC2", NULL, "Playback"},
+ {"DAC3", NULL, "Playback"},
+ {"DAC4", NULL, "Playback"},
+ {"DAC5", NULL, "Playback"},
+ {"DAC6", NULL, "Playback"},
+
+ {"Capture", NULL, "ADC1"},
+ {"Capture", NULL, "ADC2"},
+
+ {"ADC1", NULL, "LIN1"},
+ {"ADC2", NULL, "LIN2"},
+
+ {"ADC1", NULL, "RIN1"},
+ {"ADC2", NULL, "RIN2"},
+};
+
+static void ak4613_dai_shutdown(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec);
+ struct device *dev = codec->dev;
+
+ mutex_lock(&priv->lock);
+ priv->cnt--;
+ if (priv->cnt < 0) {
+ dev_err(dev, "unexpected counter error\n");
+ priv->cnt = 0;
+ }
+ if (!priv->cnt)
+ priv->fmt_ctrl = NO_FMT;
+ mutex_unlock(&priv->lock);
+}
+
+static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ fmt &= SND_SOC_DAIFMT_FORMAT_MASK;
+
+ switch (fmt) {
+ case SND_SOC_DAIFMT_RIGHT_J:
+ case SND_SOC_DAIFMT_LEFT_J:
+ case SND_SOC_DAIFMT_I2S:
+ priv->fmt = fmt;
+
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int ak4613_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec);
+ const struct ak4613_formats *fmts;
+ struct device *dev = codec->dev;
+ unsigned int width = params_width(params);
+ unsigned int fmt = priv->fmt;
+ unsigned int rate;
+ int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK;
+ int i, ret;
+ u8 fmt_ctrl, ctrl2;
+
+ rate = params_rate(params);
+ switch (rate) {
+ case 32000:
+ case 44100:
+ case 48000:
+ ctrl2 = DFS_NORMAL_SPEED;
+ break;
+ case 88200:
+ case 96000:
+ ctrl2 = DFS_DOUBLE_SPEED;
+ break;
+ case 176400:
+ case 192000:
+ ctrl2 = DFS_QUAD_SPEED;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /*
+ * FIXME
+ *
+ * It doesn't support TDM at this point
+ */
+ fmt_ctrl = NO_FMT;
+ for (i = 0; i < ARRAY_SIZE(ak4613_iface); i++) {
+ fmts = (is_play) ? &ak4613_iface[i].playback :
+ &ak4613_iface[i].capture;
+
+ if (fmts->fmt != fmt)
+ continue;
+
+ if (fmt == SND_SOC_DAIFMT_RIGHT_J) {
+ if (fmts->width != width)
+ continue;
+ } else {
+ if (fmts->width < width)
+ continue;
+ }
+
+ fmt_ctrl = AUDIO_IFACE_IDX_TO_VAL(i);
+ break;
+ }
+
+ ret = -EINVAL;
+ if (fmt_ctrl == NO_FMT)
+ goto hw_params_end;
+
+ mutex_lock(&priv->lock);
+ if ((priv->fmt_ctrl == NO_FMT) ||
+ (priv->fmt_ctrl == fmt_ctrl)) {
+ priv->fmt_ctrl = fmt_ctrl;
+ priv->cnt++;
+ ret = 0;
+ }
+ mutex_unlock(&priv->lock);
+
+ if (ret < 0)
+ goto hw_params_end;
+
+ snd_soc_update_bits(codec, CTRL1, FMT_MASK, fmt_ctrl);
+ snd_soc_write(codec, CTRL2, ctrl2);
+
+hw_params_end:
+ if (ret < 0)
+ dev_warn(dev, "unsupported data width/format combination\n");
+
+ return ret;
+}
+
+static int ak4613_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ u8 mgmt1 = 0;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ mgmt1 |= RSTN;
+ /* fall through */
+ case SND_SOC_BIAS_PREPARE:
+ mgmt1 |= PMADC | PMDAC;
+ /* fall through */
+ case SND_SOC_BIAS_STANDBY:
+ mgmt1 |= PMVR;
+ /* fall through */
+ case SND_SOC_BIAS_OFF:
+ default:
+ break;
+ }
+
+ snd_soc_write(codec, PW_MGMT1, mgmt1);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops ak4613_dai_ops = {
+ .shutdown = ak4613_dai_shutdown,
+ .set_fmt = ak4613_dai_set_fmt,
+ .hw_params = ak4613_dai_hw_params,
+};
+
+#define AK4613_PCM_RATE (SNDRV_PCM_RATE_32000 |\
+ SNDRV_PCM_RATE_44100 |\
+ SNDRV_PCM_RATE_48000 |\
+ SNDRV_PCM_RATE_64000 |\
+ SNDRV_PCM_RATE_88200 |\
+ SNDRV_PCM_RATE_96000 |\
+ SNDRV_PCM_RATE_176400 |\
+ SNDRV_PCM_RATE_192000)
+#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S16_LE |\
+ SNDRV_PCM_FMTBIT_S24_LE)
+
+static struct snd_soc_dai_driver ak4613_dai = {
+ .name = "ak4613-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = AK4613_PCM_RATE,
+ .formats = AK4613_PCM_FMTBIT,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = AK4613_PCM_RATE,
+ .formats = AK4613_PCM_FMTBIT,
+ },
+ .ops = &ak4613_dai_ops,
+ .symmetric_rates = 1,
+};
+
+static int ak4613_resume(struct snd_soc_codec *codec)
+{
+ struct regmap *regmap = dev_get_regmap(codec->dev, NULL);
+
+ regcache_mark_dirty(regmap);
+ return regcache_sync(regmap);
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_ak4613 = {
+ .resume = ak4613_resume,
+ .set_bias_level = ak4613_set_bias_level,
+ .controls = ak4613_snd_controls,
+ .num_controls = ARRAY_SIZE(ak4613_snd_controls),
+ .dapm_widgets = ak4613_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(ak4613_dapm_widgets),
+ .dapm_routes = ak4613_intercon,
+ .num_dapm_routes = ARRAY_SIZE(ak4613_intercon),
+};
+
+static int ak4613_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct device *dev = &i2c->dev;
+ struct device_node *np = dev->of_node;
+ const struct regmap_config *regmap_cfg;
+ struct regmap *regmap;
+ struct ak4613_priv *priv;
+
+ regmap_cfg = NULL;
+ if (np) {
+ const struct of_device_id *of_id;
+
+ of_id = of_match_device(ak4613_of_match, dev);
+ if (of_id)
+ regmap_cfg = of_id->data;
+ } else {
+ regmap_cfg = (const struct regmap_config *)id->driver_data;
+ }
+
+ if (!regmap_cfg)
+ return -EINVAL;
+
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
+ if (!priv)
+ return -ENOMEM;
+
+ priv->fmt_ctrl = NO_FMT;
+ priv->cnt = 0;
+
+ mutex_init(&priv->lock);
+
+ i2c_set_clientdata(i2c, priv);
+
+ regmap = devm_regmap_init_i2c(i2c, regmap_cfg);
+ if (IS_ERR(regmap))
+ return PTR_ERR(regmap);
+
+ return snd_soc_register_codec(dev, &soc_codec_dev_ak4613,
+ &ak4613_dai, 1);
+}
+
+static int ak4613_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static struct i2c_driver ak4613_i2c_driver = {
+ .driver = {
+ .name = "ak4613-codec",
+ .owner = THIS_MODULE,
+ .of_match_table = ak4613_of_match,
+ },
+ .probe = ak4613_i2c_probe,
+ .remove = ak4613_i2c_remove,
+ .id_table = ak4613_i2c_id,
+};
+
+module_i2c_driver(ak4613_i2c_driver);
+
+MODULE_DESCRIPTION("Soc AK4613 driver");
+MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 4a90143d0e90..cda27c22812a 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -23,6 +23,8 @@
* AK4648 is tested.
*/
+#include <linux/clk.h>
+#include <linux/clk-provider.h>
#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/slab.h>
@@ -128,11 +130,8 @@
#define I2S (3 << 0)
/* MD_CTL2 */
-#define FS0 (1 << 0)
-#define FS1 (1 << 1)
-#define FS2 (1 << 2)
-#define FS3 (1 << 5)
-#define FS_MASK (FS0 | FS1 | FS2 | FS3)
+#define FSs(val) (((val & 0x7) << 0) | ((val & 0x8) << 2))
+#define PSs(val) ((val & 0x3) << 6)
/* MD_CTL3 */
#define BST1 (1 << 3)
@@ -147,6 +146,7 @@ struct ak4642_drvdata {
struct ak4642_priv {
const struct ak4642_drvdata *drvdata;
+ struct clk *mcko;
};
/*
@@ -430,56 +430,56 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
return 0;
}
+static int ak4642_set_mcko(struct snd_soc_codec *codec,
+ u32 frequency)
+{
+ u32 fs_list[] = {
+ [0] = 8000,
+ [1] = 12000,
+ [2] = 16000,
+ [3] = 24000,
+ [4] = 7350,
+ [5] = 11025,
+ [6] = 14700,
+ [7] = 22050,
+ [10] = 32000,
+ [11] = 48000,
+ [14] = 29400,
+ [15] = 44100,
+ };
+ u32 ps_list[] = {
+ [0] = 256,
+ [1] = 128,
+ [2] = 64,
+ [3] = 32
+ };
+ int ps, fs;
+
+ for (ps = 0; ps < ARRAY_SIZE(ps_list); ps++) {
+ for (fs = 0; fs < ARRAY_SIZE(fs_list); fs++) {
+ if (frequency == ps_list[ps] * fs_list[fs]) {
+ snd_soc_write(codec, MD_CTL2,
+ PSs(ps) | FSs(fs));
+ return 0;
+ }
+ }
+ }
+
+ return 0;
+}
+
static int ak4642_dai_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params,
struct snd_soc_dai *dai)
{
struct snd_soc_codec *codec = dai->codec;
- u8 rate;
+ struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec);
+ u32 rate = clk_get_rate(priv->mcko);
- switch (params_rate(params)) {
- case 7350:
- rate = FS2;
- break;
- case 8000:
- rate = 0;
- break;
- case 11025:
- rate = FS2 | FS0;
- break;
- case 12000:
- rate = FS0;
- break;
- case 14700:
- rate = FS2 | FS1;
- break;
- case 16000:
- rate = FS1;
- break;
- case 22050:
- rate = FS2 | FS1 | FS0;
- break;
- case 24000:
- rate = FS1 | FS0;
- break;
- case 29400:
- rate = FS3 | FS2 | FS1;
- break;
- case 32000:
- rate = FS3 | FS1;
- break;
- case 44100:
- rate = FS3 | FS2 | FS1 | FS0;
- break;
- case 48000:
- rate = FS3 | FS1 | FS0;
- break;
- default:
- return -EINVAL;
- }
- snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate);
+ if (!rate)
+ rate = params_rate(params) * 256;
- return 0;
+ return ak4642_set_mcko(codec, rate);
}
static int ak4642_set_bias_level(struct snd_soc_codec *codec,
@@ -532,7 +532,18 @@ static int ak4642_resume(struct snd_soc_codec *codec)
return 0;
}
+static int ak4642_probe(struct snd_soc_codec *codec)
+{
+ struct ak4642_priv *priv = snd_soc_codec_get_drvdata(codec);
+
+ if (priv->mcko)
+ ak4642_set_mcko(codec, clk_get_rate(priv->mcko));
+
+ return 0;
+}
+
static struct snd_soc_codec_driver soc_codec_dev_ak4642 = {
+ .probe = ak4642_probe,
.resume = ak4642_resume,
.set_bias_level = ak4642_set_bias_level,
.controls = ak4642_snd_controls,
@@ -580,19 +591,54 @@ static const struct ak4642_drvdata ak4648_drvdata = {
.extended_frequencies = 1,
};
+#ifdef CONFIG_COMMON_CLK
+static struct clk *ak4642_of_parse_mcko(struct device *dev)
+{
+ struct device_node *np = dev->of_node;
+ struct clk *clk;
+ const char *clk_name = np->name;
+ const char *parent_clk_name = NULL;
+ u32 rate;
+
+ if (of_property_read_u32(np, "clock-frequency", &rate))
+ return NULL;
+
+ if (of_property_read_bool(np, "clocks"))
+ parent_clk_name = of_clk_get_parent_name(np, 0);
+
+ of_property_read_string(np, "clock-output-names", &clk_name);
+
+ clk = clk_register_fixed_rate(dev, clk_name, parent_clk_name,
+ (parent_clk_name) ? 0 : CLK_IS_ROOT,
+ rate);
+ if (!IS_ERR(clk))
+ of_clk_add_provider(np, of_clk_src_simple_get, clk);
+
+ return clk;
+}
+#else
+#define ak4642_of_parse_mcko(d) 0
+#endif
+
static const struct of_device_id ak4642_of_match[];
static int ak4642_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
- struct device_node *np = i2c->dev.of_node;
+ struct device *dev = &i2c->dev;
+ struct device_node *np = dev->of_node;
const struct ak4642_drvdata *drvdata = NULL;
struct regmap *regmap;
struct ak4642_priv *priv;
+ struct clk *mcko = NULL;
if (np) {
const struct of_device_id *of_id;
- of_id = of_match_device(ak4642_of_match, &i2c->dev);
+ mcko = ak4642_of_parse_mcko(dev);
+ if (IS_ERR(mcko))
+ mcko = NULL;
+
+ of_id = of_match_device(ak4642_of_match, dev);
if (of_id)
drvdata = of_id->data;
} else {
@@ -600,15 +646,16 @@ static int ak4642_i2c_probe(struct i2c_client *i2c,
}
if (!drvdata) {
- dev_err(&i2c->dev, "Unknown device type\n");
+ dev_err(dev, "Unknown device type\n");
return -EINVAL;
}
- priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL);
+ priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL);
if (!priv)
return -ENOMEM;
priv->drvdata = drvdata;
+ priv->mcko = mcko;
i2c_set_clientdata(i2c, priv);
@@ -616,7 +663,7 @@ static int ak4642_i2c_probe(struct i2c_client *i2c,
if (IS_ERR(regmap))
return PTR_ERR(regmap);
- return snd_soc_register_codec(&i2c->dev,
+ return snd_soc_register_codec(dev,
&soc_codec_dev_ak4642, &ak4642_dai, 1);
}
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c
index 8a2221ab3d10..9929efc6b9aa 100644
--- a/sound/soc/codecs/arizona.c
+++ b/sound/soc/codecs/arizona.c
@@ -147,6 +147,8 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w,
0x4f5, 0x0da);
}
break;
+ default:
+ break;
}
return 0;
@@ -314,6 +316,7 @@ const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = {
"Tone Generator 2",
"Haptics",
"AEC",
+ "AEC2",
"Mic Mute Mixer",
"Noise Generator",
"IN1L",
@@ -421,6 +424,7 @@ int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = {
0x05,
0x06, /* Haptics */
0x08, /* AEC */
+ 0x09, /* AEC2 */
0x0c, /* Noise mixer */
0x0d, /* Comfort noise */
0x10, /* IN1L */
@@ -525,6 +529,32 @@ EXPORT_SYMBOL_GPL(arizona_mixer_values);
const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0);
EXPORT_SYMBOL_GPL(arizona_mixer_tlv);
+const char * const arizona_sample_rate_text[ARIZONA_SAMPLE_RATE_ENUM_SIZE] = {
+ "12kHz", "24kHz", "48kHz", "96kHz", "192kHz",
+ "11.025kHz", "22.05kHz", "44.1kHz", "88.2kHz", "176.4kHz",
+ "4kHz", "8kHz", "16kHz", "32kHz",
+};
+EXPORT_SYMBOL_GPL(arizona_sample_rate_text);
+
+const unsigned int arizona_sample_rate_val[ARIZONA_SAMPLE_RATE_ENUM_SIZE] = {
+ 0x01, 0x02, 0x03, 0x04, 0x05, 0x09, 0x0A, 0x0B, 0x0C, 0x0D,
+ 0x10, 0x11, 0x12, 0x13,
+};
+EXPORT_SYMBOL_GPL(arizona_sample_rate_val);
+
+const char *arizona_sample_rate_val_to_name(unsigned int rate_val)
+{
+ int i;
+
+ for (i = 0; i < ARRAY_SIZE(arizona_sample_rate_val); ++i) {
+ if (arizona_sample_rate_val[i] == rate_val)
+ return arizona_sample_rate_text[i];
+ }
+
+ return "Illegal";
+}
+EXPORT_SYMBOL_GPL(arizona_sample_rate_val_to_name);
+
const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE] = {
"SYNCCLK rate", "8kHz", "16kHz", "ASYNCCLK rate",
};
@@ -689,6 +719,15 @@ static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena)
ARIZONA_IN_VU, val);
}
+bool arizona_input_analog(struct snd_soc_codec *codec, int shift)
+{
+ unsigned int reg = ARIZONA_IN1L_CONTROL + ((shift / 2) * 8);
+ unsigned int val = snd_soc_read(codec, reg);
+
+ return !(val & ARIZONA_IN1_MODE_MASK);
+}
+EXPORT_SYMBOL_GPL(arizona_input_analog);
+
int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
int event)
{
@@ -725,6 +764,9 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol,
reg = snd_soc_read(codec, ARIZONA_INPUT_ENABLES);
if (reg == 0)
arizona_in_set_vu(codec, 0);
+ break;
+ default:
+ break;
}
return 0;
@@ -806,6 +848,8 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w,
break;
}
break;
+ default:
+ break;
}
return 0;
@@ -1868,6 +1912,11 @@ static int arizona_calc_fratio(struct arizona_fll *fll,
if (fll->arizona->rev < 3 || sync)
return init_ratio;
break;
+ case WM8998:
+ case WM1814:
+ if (sync)
+ return init_ratio;
+ break;
default:
return init_ratio;
}
diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h
index ada0a418ff4b..fea8b8ae8e1a 100644
--- a/sound/soc/codecs/arizona.h
+++ b/sound/soc/codecs/arizona.h
@@ -93,12 +93,17 @@ struct arizona_priv {
bool dvfs_cached;
};
-#define ARIZONA_NUM_MIXER_INPUTS 103
+#define ARIZONA_NUM_MIXER_INPUTS 104
extern const unsigned int arizona_mixer_tlv[];
extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS];
extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS];
+#define ARIZONA_GAINMUX_CONTROLS(name, base) \
+ SOC_SINGLE_RANGE_TLV(name " Input Volume", base + 1, \
+ ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \
+ arizona_mixer_tlv)
+
#define ARIZONA_MIXER_CONTROLS(name, base) \
SOC_SINGLE_RANGE_TLV(name " Input 1 Volume", base + 1, \
ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \
@@ -209,8 +214,12 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS];
.num_regs = 1 }) }
#define ARIZONA_RATE_ENUM_SIZE 4
+#define ARIZONA_SAMPLE_RATE_ENUM_SIZE 14
+
extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE];
extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE];
+extern const char * const arizona_sample_rate_text[ARIZONA_SAMPLE_RATE_ENUM_SIZE];
+extern const unsigned int arizona_sample_rate_val[ARIZONA_SAMPLE_RATE_ENUM_SIZE];
extern const struct soc_enum arizona_isrc_fsl[];
extern const struct soc_enum arizona_isrc_fsh[];
@@ -294,4 +303,7 @@ extern int arizona_init_dai(struct arizona_priv *priv, int dai);
int arizona_set_output_mode(struct snd_soc_codec *codec, int output,
bool diff);
+extern bool arizona_input_analog(struct snd_soc_codec *codec, int shift);
+
+extern const char *arizona_sample_rate_val_to_name(unsigned int rate_val);
#endif
diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c
index a9c86efb3187..7278f93460c1 100644
--- a/sound/soc/codecs/da7213.c
+++ b/sound/soc/codecs/da7213.c
@@ -12,6 +12,7 @@
* option) any later version.
*/
+#include <linux/clk.h>
#include <linux/delay.h>
#include <linux/i2c.h>
#include <linux/regmap.h>
@@ -1222,23 +1223,44 @@ static int da7213_set_dai_sysclk(struct snd_soc_dai *codec_dai,
{
struct snd_soc_codec *codec = codec_dai->codec;
struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec);
+ int ret = 0;
+
+ if ((da7213->clk_src == clk_id) && (da7213->mclk_rate == freq))
+ return 0;
+
+ if (((freq < 5000000) && (freq != 32768)) || (freq > 54000000)) {
+ dev_err(codec_dai->dev, "Unsupported MCLK value %d\n",
+ freq);
+ return -EINVAL;
+ }
switch (clk_id) {
case DA7213_CLKSRC_MCLK:
- if ((freq == 32768) ||
- ((freq >= 5000000) && (freq <= 54000000))) {
- da7213->mclk_rate = freq;
- return 0;
- } else {
- dev_err(codec_dai->dev, "Unsupported MCLK value %d\n",
- freq);
- return -EINVAL;
- }
+ da7213->mclk_squarer_en = false;
+ break;
+ case DA7213_CLKSRC_MCLK_SQR:
+ da7213->mclk_squarer_en = true;
break;
default:
dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id);
return -EINVAL;
}
+
+ da7213->clk_src = clk_id;
+
+ if (da7213->mclk) {
+ freq = clk_round_rate(da7213->mclk, freq);
+ ret = clk_set_rate(da7213->mclk, freq);
+ if (ret) {
+ dev_err(codec_dai->dev, "Failed to set clock rate %d\n",
+ freq);
+ return ret;
+ }
+ }
+
+ da7213->mclk_rate = freq;
+
+ return 0;
}
/* Supported PLL input frequencies are 5MHz - 54MHz. */
@@ -1366,12 +1388,25 @@ static struct snd_soc_dai_driver da7213_dai = {
static int da7213_set_bias_level(struct snd_soc_codec *codec,
enum snd_soc_bias_level level)
{
+ struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
switch (level) {
case SND_SOC_BIAS_ON:
case SND_SOC_BIAS_PREPARE:
break;
case SND_SOC_BIAS_STANDBY:
if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
+ /* MCLK */
+ if (da7213->mclk) {
+ ret = clk_prepare_enable(da7213->mclk);
+ if (ret) {
+ dev_err(codec->dev,
+ "Failed to enable mclk\n");
+ return ret;
+ }
+ }
+
/* Enable VMID reference & master bias */
snd_soc_update_bits(codec, DA7213_REFERENCES,
DA7213_VMID_EN | DA7213_BIAS_EN,
@@ -1382,15 +1417,127 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec,
/* Disable VMID reference & master bias */
snd_soc_update_bits(codec, DA7213_REFERENCES,
DA7213_VMID_EN | DA7213_BIAS_EN, 0);
+
+ /* MCLK */
+ if (da7213->mclk)
+ clk_disable_unprepare(da7213->mclk);
break;
}
return 0;
}
+/* DT */
+static const struct of_device_id da7213_of_match[] = {
+ { .compatible = "dlg,da7213", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, da7213_of_match);
+
+static enum da7213_micbias_voltage
+ da7213_of_micbias_lvl(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 1600:
+ return DA7213_MICBIAS_1_6V;
+ case 2200:
+ return DA7213_MICBIAS_2_2V;
+ case 2500:
+ return DA7213_MICBIAS_2_5V;
+ case 3000:
+ return DA7213_MICBIAS_3_0V;
+ default:
+ dev_warn(codec->dev, "Invalid micbias level\n");
+ return DA7213_MICBIAS_2_2V;
+ }
+}
+
+static enum da7213_dmic_data_sel
+ da7213_of_dmic_data_sel(struct snd_soc_codec *codec, const char *str)
+{
+ if (!strcmp(str, "lrise_rfall")) {
+ return DA7213_DMIC_DATA_LRISE_RFALL;
+ } else if (!strcmp(str, "lfall_rrise")) {
+ return DA7213_DMIC_DATA_LFALL_RRISE;
+ } else {
+ dev_warn(codec->dev, "Invalid DMIC data select type\n");
+ return DA7213_DMIC_DATA_LRISE_RFALL;
+ }
+}
+
+static enum da7213_dmic_samplephase
+ da7213_of_dmic_samplephase(struct snd_soc_codec *codec, const char *str)
+{
+ if (!strcmp(str, "on_clkedge")) {
+ return DA7213_DMIC_SAMPLE_ON_CLKEDGE;
+ } else if (!strcmp(str, "between_clkedge")) {
+ return DA7213_DMIC_SAMPLE_BETWEEN_CLKEDGE;
+ } else {
+ dev_warn(codec->dev, "Invalid DMIC sample phase\n");
+ return DA7213_DMIC_SAMPLE_ON_CLKEDGE;
+ }
+}
+
+static enum da7213_dmic_clk_rate
+ da7213_of_dmic_clkrate(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 1500000:
+ return DA7213_DMIC_CLK_1_5MHZ;
+ case 3000000:
+ return DA7213_DMIC_CLK_3_0MHZ;
+ default:
+ dev_warn(codec->dev, "Invalid DMIC clock rate\n");
+ return DA7213_DMIC_CLK_1_5MHZ;
+ }
+}
+
+static struct da7213_platform_data
+ *da7213_of_to_pdata(struct snd_soc_codec *codec)
+{
+ struct device_node *np = codec->dev->of_node;
+ struct da7213_platform_data *pdata;
+ const char *of_str;
+ u32 of_val32;
+
+ pdata = devm_kzalloc(codec->dev, sizeof(*pdata), GFP_KERNEL);
+ if (!pdata) {
+ dev_warn(codec->dev, "Failed to allocate memory for pdata\n");
+ return NULL;
+ }
+
+ if (of_property_read_u32(np, "dlg,micbias1-lvl", &of_val32) >= 0)
+ pdata->micbias1_lvl = da7213_of_micbias_lvl(codec, of_val32);
+ else
+ pdata->micbias1_lvl = DA7213_MICBIAS_2_2V;
+
+ if (of_property_read_u32(np, "dlg,micbias2-lvl", &of_val32) >= 0)
+ pdata->micbias2_lvl = da7213_of_micbias_lvl(codec, of_val32);
+ else
+ pdata->micbias2_lvl = DA7213_MICBIAS_2_2V;
+
+ if (!of_property_read_string(np, "dlg,dmic-data-sel", &of_str))
+ pdata->dmic_data_sel = da7213_of_dmic_data_sel(codec, of_str);
+ else
+ pdata->dmic_data_sel = DA7213_DMIC_DATA_LRISE_RFALL;
+
+ if (!of_property_read_string(np, "dlg,dmic-samplephase", &of_str))
+ pdata->dmic_samplephase =
+ da7213_of_dmic_samplephase(codec, of_str);
+ else
+ pdata->dmic_samplephase = DA7213_DMIC_SAMPLE_ON_CLKEDGE;
+
+ if (of_property_read_u32(np, "dlg,dmic-clkrate", &of_val32) >= 0)
+ pdata->dmic_clk_rate = da7213_of_dmic_clkrate(codec, of_val32);
+ else
+ pdata->dmic_clk_rate = DA7213_DMIC_CLK_3_0MHZ;
+
+ return pdata;
+}
+
+
static int da7213_probe(struct snd_soc_codec *codec)
{
struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec);
- struct da7213_platform_data *pdata = da7213->pdata;
/* Default to using ALC auto offset calibration mode. */
snd_soc_update_bits(codec, DA7213_ALC_CTRL1,
@@ -1450,8 +1597,15 @@ static int da7213_probe(struct snd_soc_codec *codec)
snd_soc_update_bits(codec, DA7213_LINE_CTRL,
DA7213_LINE_AMP_OE, DA7213_LINE_AMP_OE);
+ /* Handle DT/Platform data */
+ if (codec->dev->of_node)
+ da7213->pdata = da7213_of_to_pdata(codec);
+ else
+ da7213->pdata = dev_get_platdata(codec->dev);
+
/* Set platform data values */
if (da7213->pdata) {
+ struct da7213_platform_data *pdata = da7213->pdata;
u8 micbias_lvl = 0, dmic_cfg = 0;
/* Set Mic Bias voltages */
@@ -1503,10 +1657,17 @@ static int da7213_probe(struct snd_soc_codec *codec)
DA7213_DMIC_DATA_SEL_MASK |
DA7213_DMIC_SAMPLEPHASE_MASK |
DA7213_DMIC_CLK_RATE_MASK, dmic_cfg);
+ }
- /* Set MCLK squaring */
- da7213->mclk_squarer_en = pdata->mclk_squaring;
+ /* Check if MCLK provided */
+ da7213->mclk = devm_clk_get(codec->dev, "mclk");
+ if (IS_ERR(da7213->mclk)) {
+ if (PTR_ERR(da7213->mclk) != -ENOENT)
+ return PTR_ERR(da7213->mclk);
+ else
+ da7213->mclk = NULL;
}
+
return 0;
}
@@ -1537,7 +1698,6 @@ static int da7213_i2c_probe(struct i2c_client *i2c,
const struct i2c_device_id *id)
{
struct da7213_priv *da7213;
- struct da7213_platform_data *pdata = dev_get_platdata(&i2c->dev);
int ret;
da7213 = devm_kzalloc(&i2c->dev, sizeof(struct da7213_priv),
@@ -1545,9 +1705,6 @@ static int da7213_i2c_probe(struct i2c_client *i2c,
if (!da7213)
return -ENOMEM;
- if (pdata)
- da7213->pdata = pdata;
-
i2c_set_clientdata(i2c, da7213);
da7213->regmap = devm_regmap_init_i2c(i2c, &da7213_regmap_config);
@@ -1582,6 +1739,7 @@ MODULE_DEVICE_TABLE(i2c, da7213_i2c_id);
static struct i2c_driver da7213_i2c_driver = {
.driver = {
.name = "da7213",
+ .of_match_table = of_match_ptr(da7213_of_match),
},
.probe = da7213_i2c_probe,
.remove = da7213_remove,
diff --git a/sound/soc/codecs/da7213.h b/sound/soc/codecs/da7213.h
index 9cb9ddd01282..030fd691b076 100644
--- a/sound/soc/codecs/da7213.h
+++ b/sound/soc/codecs/da7213.h
@@ -13,6 +13,7 @@
#ifndef _DA7213_H
#define _DA7213_H
+#include <linux/clk.h>
#include <linux/regmap.h>
#include <sound/da7213.h>
@@ -504,14 +505,17 @@
#define DA7213_PLL_INDIV_20_40_MHZ_VAL 8
#define DA7213_PLL_INDIV_40_54_MHZ_VAL 16
-enum clk_src {
- DA7213_CLKSRC_MCLK
+enum da7213_clk_src {
+ DA7213_CLKSRC_MCLK = 0,
+ DA7213_CLKSRC_MCLK_SQR,
};
/* Codec private data */
struct da7213_priv {
struct regmap *regmap;
+ struct clk *mclk;
unsigned int mclk_rate;
+ int clk_src;
bool master;
bool mclk_squarer_en;
bool srm_en;
diff --git a/sound/soc/codecs/da7219-aad.c b/sound/soc/codecs/da7219-aad.c
new file mode 100644
index 000000000000..9459593eef13
--- /dev/null
+++ b/sound/soc/codecs/da7219-aad.c
@@ -0,0 +1,823 @@
+/*
+ * da7219-aad.c - Dialog DA7219 ALSA SoC AAD Driver
+ *
+ * Copyright (c) 2015 Dialog Semiconductor Ltd.
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/of_device.h>
+#include <linux/of_irq.h>
+#include <linux/pm_wakeirq.h>
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <linux/workqueue.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/da7219.h>
+
+#include "da7219.h"
+#include "da7219-aad.h"
+
+
+/*
+ * Detection control
+ */
+
+void da7219_aad_jack_det(struct snd_soc_codec *codec, struct snd_soc_jack *jack)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+ da7219->aad->jack = jack;
+ da7219->aad->jack_inserted = false;
+
+ /* Send an initial empty report */
+ snd_soc_jack_report(jack, 0, DA7219_AAD_REPORT_ALL_MASK);
+
+ /* Enable/Disable jack detection */
+ snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1,
+ DA7219_ACCDET_EN_MASK,
+ (jack ? DA7219_ACCDET_EN_MASK : 0));
+}
+EXPORT_SYMBOL_GPL(da7219_aad_jack_det);
+
+/*
+ * Button/HPTest work
+ */
+
+static void da7219_aad_btn_det_work(struct work_struct *work)
+{
+ struct da7219_aad_priv *da7219_aad =
+ container_of(work, struct da7219_aad_priv, btn_det_work);
+ struct snd_soc_codec *codec = da7219_aad->codec;
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ u8 statusa, micbias_ctrl;
+ bool micbias_up = false;
+ int retries = 0;
+
+ /* Drive headphones/lineout */
+ snd_soc_update_bits(codec, DA7219_HP_L_CTRL,
+ DA7219_HP_L_AMP_OE_MASK,
+ DA7219_HP_L_AMP_OE_MASK);
+ snd_soc_update_bits(codec, DA7219_HP_R_CTRL,
+ DA7219_HP_R_AMP_OE_MASK,
+ DA7219_HP_R_AMP_OE_MASK);
+
+ /* Make sure mic bias is up */
+ snd_soc_dapm_force_enable_pin(dapm, "Mic Bias");
+ snd_soc_dapm_sync(dapm);
+
+ do {
+ statusa = snd_soc_read(codec, DA7219_ACCDET_STATUS_A);
+ if (statusa & DA7219_MICBIAS_UP_STS_MASK)
+ micbias_up = true;
+ else if (retries++ < DA7219_AAD_MICBIAS_CHK_RETRIES)
+ msleep(DA7219_AAD_MICBIAS_CHK_DELAY);
+ } while ((!micbias_up) && (retries < DA7219_AAD_MICBIAS_CHK_RETRIES));
+
+ if (retries >= DA7219_AAD_MICBIAS_CHK_RETRIES)
+ dev_warn(codec->dev, "Mic bias status check timed out");
+
+ /*
+ * Mic bias pulse required to enable mic, must be done before enabling
+ * button detection to prevent erroneous button readings.
+ */
+ if (da7219_aad->micbias_pulse_lvl && da7219_aad->micbias_pulse_time) {
+ /* Pulse higher level voltage */
+ micbias_ctrl = snd_soc_read(codec, DA7219_MICBIAS_CTRL);
+ snd_soc_update_bits(codec, DA7219_MICBIAS_CTRL,
+ DA7219_MICBIAS1_LEVEL_MASK,
+ da7219_aad->micbias_pulse_lvl);
+ msleep(da7219_aad->micbias_pulse_time);
+ snd_soc_write(codec, DA7219_MICBIAS_CTRL, micbias_ctrl);
+
+ }
+
+ snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1,
+ DA7219_BUTTON_CONFIG_MASK,
+ da7219_aad->btn_cfg);
+}
+
+static void da7219_aad_hptest_work(struct work_struct *work)
+{
+ struct da7219_aad_priv *da7219_aad =
+ container_of(work, struct da7219_aad_priv, hptest_work);
+ struct snd_soc_codec *codec = da7219_aad->codec;
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+ u16 tonegen_freq_hptest;
+ u8 accdet_cfg8;
+ int report = 0;
+
+ /* Lock DAPM and any Kcontrols that are affected by this test */
+ snd_soc_dapm_mutex_lock(dapm);
+ mutex_lock(&da7219->lock);
+
+ /* Bypass cache so it saves current settings */
+ regcache_cache_bypass(da7219->regmap, true);
+
+ /* Make sure Tone Generator is disabled */
+ snd_soc_write(codec, DA7219_TONE_GEN_CFG1, 0);
+
+ /* Enable HPTest block, 1KOhms check */
+ snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_8,
+ DA7219_HPTEST_EN_MASK | DA7219_HPTEST_RES_SEL_MASK,
+ DA7219_HPTEST_EN_MASK |
+ DA7219_HPTEST_RES_SEL_1KOHMS);
+
+ /* Set gains to 0db */
+ snd_soc_write(codec, DA7219_DAC_L_GAIN, DA7219_DAC_DIGITAL_GAIN_0DB);
+ snd_soc_write(codec, DA7219_DAC_R_GAIN, DA7219_DAC_DIGITAL_GAIN_0DB);
+ snd_soc_write(codec, DA7219_HP_L_GAIN, DA7219_HP_AMP_GAIN_0DB);
+ snd_soc_write(codec, DA7219_HP_R_GAIN, DA7219_HP_AMP_GAIN_0DB);
+
+ /* Disable DAC filters, EQs and soft mute */
+ snd_soc_update_bits(codec, DA7219_DAC_FILTERS1, DA7219_HPF_MODE_MASK,
+ 0);
+ snd_soc_update_bits(codec, DA7219_DAC_FILTERS4, DA7219_DAC_EQ_EN_MASK,
+ 0);
+ snd_soc_update_bits(codec, DA7219_DAC_FILTERS5,
+ DA7219_DAC_SOFTMUTE_EN_MASK, 0);
+
+ /* Enable HP left & right paths */
+ snd_soc_update_bits(codec, DA7219_CP_CTRL, DA7219_CP_EN_MASK,
+ DA7219_CP_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_DIG_ROUTING_DAC,
+ DA7219_DAC_L_SRC_MASK | DA7219_DAC_R_SRC_MASK,
+ DA7219_DAC_L_SRC_TONEGEN |
+ DA7219_DAC_R_SRC_TONEGEN);
+ snd_soc_update_bits(codec, DA7219_DAC_L_CTRL,
+ DA7219_DAC_L_EN_MASK | DA7219_DAC_L_MUTE_EN_MASK,
+ DA7219_DAC_L_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_DAC_R_CTRL,
+ DA7219_DAC_R_EN_MASK | DA7219_DAC_R_MUTE_EN_MASK,
+ DA7219_DAC_R_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_MIXOUT_L_SELECT,
+ DA7219_MIXOUT_L_MIX_SELECT_MASK,
+ DA7219_MIXOUT_L_MIX_SELECT_MASK);
+ snd_soc_update_bits(codec, DA7219_MIXOUT_R_SELECT,
+ DA7219_MIXOUT_R_MIX_SELECT_MASK,
+ DA7219_MIXOUT_R_MIX_SELECT_MASK);
+ snd_soc_update_bits(codec, DA7219_DROUTING_ST_OUTFILT_1L,
+ DA7219_OUTFILT_ST_1L_SRC_MASK,
+ DA7219_DMIX_ST_SRC_OUTFILT1L);
+ snd_soc_update_bits(codec, DA7219_DROUTING_ST_OUTFILT_1R,
+ DA7219_OUTFILT_ST_1R_SRC_MASK,
+ DA7219_DMIX_ST_SRC_OUTFILT1R);
+ snd_soc_update_bits(codec, DA7219_MIXOUT_L_CTRL,
+ DA7219_MIXOUT_L_AMP_EN_MASK,
+ DA7219_MIXOUT_L_AMP_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_MIXOUT_R_CTRL,
+ DA7219_MIXOUT_R_AMP_EN_MASK,
+ DA7219_MIXOUT_R_AMP_EN_MASK);
+ snd_soc_write(codec, DA7219_HP_L_CTRL,
+ DA7219_HP_L_AMP_OE_MASK | DA7219_HP_L_AMP_EN_MASK);
+ snd_soc_write(codec, DA7219_HP_R_CTRL,
+ DA7219_HP_R_AMP_OE_MASK | DA7219_HP_R_AMP_EN_MASK);
+
+ /* Configure & start Tone Generator */
+ snd_soc_write(codec, DA7219_TONE_GEN_ON_PER, DA7219_BEEP_ON_PER_MASK);
+ tonegen_freq_hptest = cpu_to_le16(DA7219_AAD_HPTEST_RAMP_FREQ);
+ regmap_raw_write(da7219->regmap, DA7219_TONE_GEN_FREQ1_L,
+ &tonegen_freq_hptest, sizeof(tonegen_freq_hptest));
+ snd_soc_update_bits(codec, DA7219_TONE_GEN_CFG2,
+ DA7219_SWG_SEL_MASK | DA7219_TONE_GEN_GAIN_MASK,
+ DA7219_SWG_SEL_SRAMP |
+ DA7219_TONE_GEN_GAIN_MINUS_15DB);
+ snd_soc_write(codec, DA7219_TONE_GEN_CFG1, DA7219_START_STOPN_MASK);
+
+ msleep(DA7219_AAD_HPTEST_PERIOD);
+
+ /* Grab comparator reading */
+ accdet_cfg8 = snd_soc_read(codec, DA7219_ACCDET_CONFIG_8);
+ if (accdet_cfg8 & DA7219_HPTEST_COMP_MASK)
+ report |= SND_JACK_HEADPHONE;
+ else
+ report |= SND_JACK_LINEOUT;
+
+ /* Stop tone generator */
+ snd_soc_write(codec, DA7219_TONE_GEN_CFG1, 0);
+
+ msleep(DA7219_AAD_HPTEST_PERIOD);
+
+ /* Restore original settings from cache */
+ regcache_mark_dirty(da7219->regmap);
+ regcache_sync_region(da7219->regmap, DA7219_HP_L_CTRL,
+ DA7219_HP_R_CTRL);
+ regcache_sync_region(da7219->regmap, DA7219_MIXOUT_L_CTRL,
+ DA7219_MIXOUT_R_CTRL);
+ regcache_sync_region(da7219->regmap, DA7219_DROUTING_ST_OUTFILT_1L,
+ DA7219_DROUTING_ST_OUTFILT_1R);
+ regcache_sync_region(da7219->regmap, DA7219_MIXOUT_L_SELECT,
+ DA7219_MIXOUT_R_SELECT);
+ regcache_sync_region(da7219->regmap, DA7219_DAC_L_CTRL,
+ DA7219_DAC_R_CTRL);
+ regcache_sync_region(da7219->regmap, DA7219_DIG_ROUTING_DAC,
+ DA7219_DIG_ROUTING_DAC);
+ regcache_sync_region(da7219->regmap, DA7219_CP_CTRL, DA7219_CP_CTRL);
+ regcache_sync_region(da7219->regmap, DA7219_DAC_FILTERS5,
+ DA7219_DAC_FILTERS5);
+ regcache_sync_region(da7219->regmap, DA7219_DAC_FILTERS4,
+ DA7219_DAC_FILTERS1);
+ regcache_sync_region(da7219->regmap, DA7219_HP_L_GAIN,
+ DA7219_HP_R_GAIN);
+ regcache_sync_region(da7219->regmap, DA7219_DAC_L_GAIN,
+ DA7219_DAC_R_GAIN);
+ regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_ON_PER,
+ DA7219_TONE_GEN_ON_PER);
+ regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_FREQ1_L,
+ DA7219_TONE_GEN_FREQ1_U);
+ regcache_sync_region(da7219->regmap, DA7219_TONE_GEN_CFG1,
+ DA7219_TONE_GEN_CFG2);
+
+ regcache_cache_bypass(da7219->regmap, false);
+
+ /* Disable HPTest block */
+ snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_8,
+ DA7219_HPTEST_EN_MASK, 0);
+
+ /* Drive Headphones/lineout */
+ snd_soc_update_bits(codec, DA7219_HP_L_CTRL, DA7219_HP_L_AMP_OE_MASK,
+ DA7219_HP_L_AMP_OE_MASK);
+ snd_soc_update_bits(codec, DA7219_HP_R_CTRL, DA7219_HP_R_AMP_OE_MASK,
+ DA7219_HP_R_AMP_OE_MASK);
+
+ mutex_unlock(&da7219->lock);
+ snd_soc_dapm_mutex_unlock(dapm);
+
+ /*
+ * Only send report if jack hasn't been removed during process,
+ * otherwise it's invalid and we drop it.
+ */
+ if (da7219_aad->jack_inserted)
+ snd_soc_jack_report(da7219_aad->jack, report,
+ SND_JACK_HEADSET | SND_JACK_LINEOUT);
+}
+
+
+/*
+ * IRQ
+ */
+
+static irqreturn_t da7219_aad_irq_thread(int irq, void *data)
+{
+ struct da7219_aad_priv *da7219_aad = data;
+ struct snd_soc_codec *codec = da7219_aad->codec;
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ u8 events[DA7219_AAD_IRQ_REG_MAX];
+ u8 statusa;
+ int i, report = 0, mask = 0;
+
+ /* Read current IRQ events */
+ regmap_bulk_read(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A,
+ events, DA7219_AAD_IRQ_REG_MAX);
+
+ if (!events[DA7219_AAD_IRQ_REG_A] && !events[DA7219_AAD_IRQ_REG_B])
+ return IRQ_NONE;
+
+ /* Read status register for jack insertion & type status */
+ statusa = snd_soc_read(codec, DA7219_ACCDET_STATUS_A);
+
+ /* Clear events */
+ regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_EVENT_A,
+ events, DA7219_AAD_IRQ_REG_MAX);
+
+ dev_dbg(codec->dev, "IRQ events = 0x%x|0x%x, status = 0x%x\n",
+ events[DA7219_AAD_IRQ_REG_A], events[DA7219_AAD_IRQ_REG_B],
+ statusa);
+
+ if (statusa & DA7219_JACK_INSERTION_STS_MASK) {
+ /* Jack Insertion */
+ if (events[DA7219_AAD_IRQ_REG_A] &
+ DA7219_E_JACK_INSERTED_MASK) {
+ report |= SND_JACK_MECHANICAL;
+ mask |= SND_JACK_MECHANICAL;
+ da7219_aad->jack_inserted = true;
+ }
+
+ /* Jack type detection */
+ if (events[DA7219_AAD_IRQ_REG_A] &
+ DA7219_E_JACK_DETECT_COMPLETE_MASK) {
+ /*
+ * If 4-pole, then enable button detection, else perform
+ * HP impedance test to determine output type to report.
+ *
+ * We schedule work here as the tasks themselves can
+ * take time to complete, and in particular for hptest
+ * we want to be able to check if the jack was removed
+ * during the procedure as this will invalidate the
+ * result. By doing this as work, the IRQ thread can
+ * handle a removal, and we can check at the end of
+ * hptest if we have a valid result or not.
+ */
+ if (statusa & DA7219_JACK_TYPE_STS_MASK) {
+ report |= SND_JACK_HEADSET;
+ mask |= SND_JACK_HEADSET | SND_JACK_LINEOUT;
+ schedule_work(&da7219_aad->btn_det_work);
+ } else {
+ schedule_work(&da7219_aad->hptest_work);
+ }
+ }
+
+ /* Button support for 4-pole jack */
+ if (statusa & DA7219_JACK_TYPE_STS_MASK) {
+ for (i = 0; i < DA7219_AAD_MAX_BUTTONS; ++i) {
+ /* Button Press */
+ if (events[DA7219_AAD_IRQ_REG_B] &
+ (DA7219_E_BUTTON_A_PRESSED_MASK << i)) {
+ report |= SND_JACK_BTN_0 >> i;
+ mask |= SND_JACK_BTN_0 >> i;
+ }
+ }
+ snd_soc_jack_report(da7219_aad->jack, report, mask);
+
+ for (i = 0; i < DA7219_AAD_MAX_BUTTONS; ++i) {
+ /* Button Release */
+ if (events[DA7219_AAD_IRQ_REG_B] &
+ (DA7219_E_BUTTON_A_RELEASED_MASK >> i)) {
+ report &= ~(SND_JACK_BTN_0 >> i);
+ mask |= SND_JACK_BTN_0 >> i;
+ }
+ }
+ }
+ } else {
+ /* Jack removal */
+ if (events[DA7219_AAD_IRQ_REG_A] & DA7219_E_JACK_REMOVED_MASK) {
+ report = 0;
+ mask |= DA7219_AAD_REPORT_ALL_MASK;
+ da7219_aad->jack_inserted = false;
+
+ /* Un-drive headphones/lineout */
+ snd_soc_update_bits(codec, DA7219_HP_R_CTRL,
+ DA7219_HP_R_AMP_OE_MASK, 0);
+ snd_soc_update_bits(codec, DA7219_HP_L_CTRL,
+ DA7219_HP_L_AMP_OE_MASK, 0);
+
+ /* Ensure button detection disabled */
+ snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1,
+ DA7219_BUTTON_CONFIG_MASK, 0);
+
+ /* Disable mic bias */
+ snd_soc_dapm_disable_pin(dapm, "Mic Bias");
+ snd_soc_dapm_sync(dapm);
+
+ /* Cancel any pending work */
+ cancel_work_sync(&da7219_aad->btn_det_work);
+ cancel_work_sync(&da7219_aad->hptest_work);
+ }
+ }
+
+ snd_soc_jack_report(da7219_aad->jack, report, mask);
+
+ return IRQ_HANDLED;
+}
+
+/*
+ * DT to pdata conversion
+ */
+
+static enum da7219_aad_micbias_pulse_lvl
+ da7219_aad_of_micbias_pulse_lvl(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 2800:
+ return DA7219_AAD_MICBIAS_PULSE_LVL_2_8V;
+ case 2900:
+ return DA7219_AAD_MICBIAS_PULSE_LVL_2_9V;
+ default:
+ dev_warn(codec->dev, "Invalid micbias pulse level");
+ return DA7219_AAD_MICBIAS_PULSE_LVL_OFF;
+ }
+}
+
+static enum da7219_aad_btn_cfg
+ da7219_aad_of_btn_cfg(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 2:
+ return DA7219_AAD_BTN_CFG_2MS;
+ case 5:
+ return DA7219_AAD_BTN_CFG_5MS;
+ case 10:
+ return DA7219_AAD_BTN_CFG_10MS;
+ case 50:
+ return DA7219_AAD_BTN_CFG_50MS;
+ case 100:
+ return DA7219_AAD_BTN_CFG_100MS;
+ case 200:
+ return DA7219_AAD_BTN_CFG_200MS;
+ case 500:
+ return DA7219_AAD_BTN_CFG_500MS;
+ default:
+ dev_warn(codec->dev, "Invalid button config");
+ return DA7219_AAD_BTN_CFG_10MS;
+ }
+}
+
+static enum da7219_aad_mic_det_thr
+ da7219_aad_of_mic_det_thr(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 200:
+ return DA7219_AAD_MIC_DET_THR_200_OHMS;
+ case 500:
+ return DA7219_AAD_MIC_DET_THR_500_OHMS;
+ case 750:
+ return DA7219_AAD_MIC_DET_THR_750_OHMS;
+ case 1000:
+ return DA7219_AAD_MIC_DET_THR_1000_OHMS;
+ default:
+ dev_warn(codec->dev, "Invalid mic detect threshold");
+ return DA7219_AAD_MIC_DET_THR_500_OHMS;
+ }
+}
+
+static enum da7219_aad_jack_ins_deb
+ da7219_aad_of_jack_ins_deb(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 5:
+ return DA7219_AAD_JACK_INS_DEB_5MS;
+ case 10:
+ return DA7219_AAD_JACK_INS_DEB_10MS;
+ case 20:
+ return DA7219_AAD_JACK_INS_DEB_20MS;
+ case 50:
+ return DA7219_AAD_JACK_INS_DEB_50MS;
+ case 100:
+ return DA7219_AAD_JACK_INS_DEB_100MS;
+ case 200:
+ return DA7219_AAD_JACK_INS_DEB_200MS;
+ case 500:
+ return DA7219_AAD_JACK_INS_DEB_500MS;
+ case 1000:
+ return DA7219_AAD_JACK_INS_DEB_1S;
+ default:
+ dev_warn(codec->dev, "Invalid jack insert debounce");
+ return DA7219_AAD_JACK_INS_DEB_20MS;
+ }
+}
+
+static enum da7219_aad_jack_det_rate
+ da7219_aad_of_jack_det_rate(struct snd_soc_codec *codec, const char *str)
+{
+ if (!strcmp(str, "32ms_64ms")) {
+ return DA7219_AAD_JACK_DET_RATE_32_64MS;
+ } else if (!strcmp(str, "64ms_128ms")) {
+ return DA7219_AAD_JACK_DET_RATE_64_128MS;
+ } else if (!strcmp(str, "128ms_256ms")) {
+ return DA7219_AAD_JACK_DET_RATE_128_256MS;
+ } else if (!strcmp(str, "256ms_512ms")) {
+ return DA7219_AAD_JACK_DET_RATE_256_512MS;
+ } else {
+ dev_warn(codec->dev, "Invalid jack detect rate");
+ return DA7219_AAD_JACK_DET_RATE_256_512MS;
+ }
+}
+
+static enum da7219_aad_jack_rem_deb
+ da7219_aad_of_jack_rem_deb(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 1:
+ return DA7219_AAD_JACK_REM_DEB_1MS;
+ case 5:
+ return DA7219_AAD_JACK_REM_DEB_5MS;
+ case 10:
+ return DA7219_AAD_JACK_REM_DEB_10MS;
+ case 20:
+ return DA7219_AAD_JACK_REM_DEB_20MS;
+ default:
+ dev_warn(codec->dev, "Invalid jack removal debounce");
+ return DA7219_AAD_JACK_REM_DEB_1MS;
+ }
+}
+
+static enum da7219_aad_btn_avg
+ da7219_aad_of_btn_avg(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 1:
+ return DA7219_AAD_BTN_AVG_1;
+ case 2:
+ return DA7219_AAD_BTN_AVG_2;
+ case 4:
+ return DA7219_AAD_BTN_AVG_4;
+ case 8:
+ return DA7219_AAD_BTN_AVG_8;
+ default:
+ dev_warn(codec->dev, "Invalid button average value");
+ return DA7219_AAD_BTN_AVG_2;
+ }
+}
+
+static enum da7219_aad_adc_1bit_rpt
+ da7219_aad_of_adc_1bit_rpt(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 1:
+ return DA7219_AAD_ADC_1BIT_RPT_1;
+ case 2:
+ return DA7219_AAD_ADC_1BIT_RPT_2;
+ case 4:
+ return DA7219_AAD_ADC_1BIT_RPT_4;
+ case 8:
+ return DA7219_AAD_ADC_1BIT_RPT_8;
+ default:
+ dev_warn(codec->dev, "Invalid ADC 1-bit repeat value");
+ return DA7219_AAD_ADC_1BIT_RPT_1;
+ }
+}
+
+static struct da7219_aad_pdata *da7219_aad_of_to_pdata(struct snd_soc_codec *codec)
+{
+ struct device_node *np = codec->dev->of_node;
+ struct device_node *aad_np = of_find_node_by_name(np, "da7219_aad");
+ struct da7219_aad_pdata *aad_pdata;
+ const char *of_str;
+ u32 of_val32;
+
+ if (!aad_np)
+ return NULL;
+
+ aad_pdata = devm_kzalloc(codec->dev, sizeof(*aad_pdata), GFP_KERNEL);
+ if (!aad_pdata)
+ goto out;
+
+ aad_pdata->irq = irq_of_parse_and_map(np, 0);
+
+ if (of_property_read_u32(aad_np, "dlg,micbias-pulse-lvl",
+ &of_val32) >= 0)
+ aad_pdata->micbias_pulse_lvl =
+ da7219_aad_of_micbias_pulse_lvl(codec, of_val32);
+ else
+ aad_pdata->micbias_pulse_lvl = DA7219_AAD_MICBIAS_PULSE_LVL_OFF;
+
+ if (of_property_read_u32(aad_np, "dlg,micbias-pulse-time",
+ &of_val32) >= 0)
+ aad_pdata->micbias_pulse_time = of_val32;
+
+ if (of_property_read_u32(aad_np, "dlg,btn-cfg", &of_val32) >= 0)
+ aad_pdata->btn_cfg = da7219_aad_of_btn_cfg(codec, of_val32);
+ else
+ aad_pdata->btn_cfg = DA7219_AAD_BTN_CFG_10MS;
+
+ if (of_property_read_u32(aad_np, "dlg,mic-det-thr", &of_val32) >= 0)
+ aad_pdata->mic_det_thr =
+ da7219_aad_of_mic_det_thr(codec, of_val32);
+ else
+ aad_pdata->mic_det_thr = DA7219_AAD_MIC_DET_THR_500_OHMS;
+
+ if (of_property_read_u32(aad_np, "dlg,jack-ins-deb", &of_val32) >= 0)
+ aad_pdata->jack_ins_deb =
+ da7219_aad_of_jack_ins_deb(codec, of_val32);
+ else
+ aad_pdata->jack_ins_deb = DA7219_AAD_JACK_INS_DEB_20MS;
+
+ if (!of_property_read_string(aad_np, "dlg,jack-det-rate", &of_str))
+ aad_pdata->jack_det_rate =
+ da7219_aad_of_jack_det_rate(codec, of_str);
+ else
+ aad_pdata->jack_det_rate = DA7219_AAD_JACK_DET_RATE_256_512MS;
+
+ if (of_property_read_u32(aad_np, "dlg,jack-rem-deb", &of_val32) >= 0)
+ aad_pdata->jack_rem_deb =
+ da7219_aad_of_jack_rem_deb(codec, of_val32);
+ else
+ aad_pdata->jack_rem_deb = DA7219_AAD_JACK_REM_DEB_1MS;
+
+ if (of_property_read_u32(aad_np, "dlg,a-d-btn-thr", &of_val32) >= 0)
+ aad_pdata->a_d_btn_thr = (u8) of_val32;
+ else
+ aad_pdata->a_d_btn_thr = 0xA;
+
+ if (of_property_read_u32(aad_np, "dlg,d-b-btn-thr", &of_val32) >= 0)
+ aad_pdata->d_b_btn_thr = (u8) of_val32;
+ else
+ aad_pdata->d_b_btn_thr = 0x16;
+
+ if (of_property_read_u32(aad_np, "dlg,b-c-btn-thr", &of_val32) >= 0)
+ aad_pdata->b_c_btn_thr = (u8) of_val32;
+ else
+ aad_pdata->b_c_btn_thr = 0x21;
+
+ if (of_property_read_u32(aad_np, "dlg,c-mic-btn-thr", &of_val32) >= 0)
+ aad_pdata->c_mic_btn_thr = (u8) of_val32;
+ else
+ aad_pdata->c_mic_btn_thr = 0x3E;
+
+ if (of_property_read_u32(aad_np, "dlg,btn-avg", &of_val32) >= 0)
+ aad_pdata->btn_avg = da7219_aad_of_btn_avg(codec, of_val32);
+ else
+ aad_pdata->btn_avg = DA7219_AAD_BTN_AVG_2;
+
+ if (of_property_read_u32(aad_np, "dlg,adc-1bit-rpt", &of_val32) >= 0)
+ aad_pdata->adc_1bit_rpt =
+ da7219_aad_of_adc_1bit_rpt(codec, of_val32);
+ else
+ aad_pdata->adc_1bit_rpt = DA7219_AAD_ADC_1BIT_RPT_1;
+
+out:
+ of_node_put(aad_np);
+
+ return aad_pdata;
+}
+
+static void da7219_aad_handle_pdata(struct snd_soc_codec *codec)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ struct da7219_aad_priv *da7219_aad = da7219->aad;
+ struct da7219_pdata *pdata = da7219->pdata;
+
+ if ((pdata) && (pdata->aad_pdata)) {
+ struct da7219_aad_pdata *aad_pdata = pdata->aad_pdata;
+ u8 cfg, mask;
+
+ da7219_aad->irq = aad_pdata->irq;
+
+ switch (aad_pdata->micbias_pulse_lvl) {
+ case DA7219_AAD_MICBIAS_PULSE_LVL_2_8V:
+ case DA7219_AAD_MICBIAS_PULSE_LVL_2_9V:
+ da7219_aad->micbias_pulse_lvl =
+ (aad_pdata->micbias_pulse_lvl <<
+ DA7219_MICBIAS1_LEVEL_SHIFT);
+ break;
+ default:
+ break;
+ }
+
+ da7219_aad->micbias_pulse_time = aad_pdata->micbias_pulse_time;
+
+ switch (aad_pdata->btn_cfg) {
+ case DA7219_AAD_BTN_CFG_2MS:
+ case DA7219_AAD_BTN_CFG_5MS:
+ case DA7219_AAD_BTN_CFG_10MS:
+ case DA7219_AAD_BTN_CFG_50MS:
+ case DA7219_AAD_BTN_CFG_100MS:
+ case DA7219_AAD_BTN_CFG_200MS:
+ case DA7219_AAD_BTN_CFG_500MS:
+ da7219_aad->btn_cfg = (aad_pdata->btn_cfg <<
+ DA7219_BUTTON_CONFIG_SHIFT);
+ }
+
+ cfg = 0;
+ mask = 0;
+ switch (aad_pdata->mic_det_thr) {
+ case DA7219_AAD_MIC_DET_THR_200_OHMS:
+ case DA7219_AAD_MIC_DET_THR_500_OHMS:
+ case DA7219_AAD_MIC_DET_THR_750_OHMS:
+ case DA7219_AAD_MIC_DET_THR_1000_OHMS:
+ cfg |= (aad_pdata->mic_det_thr <<
+ DA7219_MIC_DET_THRESH_SHIFT);
+ mask |= DA7219_MIC_DET_THRESH_MASK;
+ }
+ snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1, mask, cfg);
+
+ cfg = 0;
+ mask = 0;
+ switch (aad_pdata->jack_ins_deb) {
+ case DA7219_AAD_JACK_INS_DEB_5MS:
+ case DA7219_AAD_JACK_INS_DEB_10MS:
+ case DA7219_AAD_JACK_INS_DEB_20MS:
+ case DA7219_AAD_JACK_INS_DEB_50MS:
+ case DA7219_AAD_JACK_INS_DEB_100MS:
+ case DA7219_AAD_JACK_INS_DEB_200MS:
+ case DA7219_AAD_JACK_INS_DEB_500MS:
+ case DA7219_AAD_JACK_INS_DEB_1S:
+ cfg |= (aad_pdata->jack_ins_deb <<
+ DA7219_JACKDET_DEBOUNCE_SHIFT);
+ mask |= DA7219_JACKDET_DEBOUNCE_MASK;
+ }
+ switch (aad_pdata->jack_det_rate) {
+ case DA7219_AAD_JACK_DET_RATE_32_64MS:
+ case DA7219_AAD_JACK_DET_RATE_64_128MS:
+ case DA7219_AAD_JACK_DET_RATE_128_256MS:
+ case DA7219_AAD_JACK_DET_RATE_256_512MS:
+ cfg |= (aad_pdata->jack_det_rate <<
+ DA7219_JACK_DETECT_RATE_SHIFT);
+ mask |= DA7219_JACK_DETECT_RATE_MASK;
+ }
+ switch (aad_pdata->jack_rem_deb) {
+ case DA7219_AAD_JACK_REM_DEB_1MS:
+ case DA7219_AAD_JACK_REM_DEB_5MS:
+ case DA7219_AAD_JACK_REM_DEB_10MS:
+ case DA7219_AAD_JACK_REM_DEB_20MS:
+ cfg |= (aad_pdata->jack_rem_deb <<
+ DA7219_JACKDET_REM_DEB_SHIFT);
+ mask |= DA7219_JACKDET_REM_DEB_MASK;
+ }
+ snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_2, mask, cfg);
+
+ snd_soc_write(codec, DA7219_ACCDET_CONFIG_3,
+ aad_pdata->a_d_btn_thr);
+ snd_soc_write(codec, DA7219_ACCDET_CONFIG_4,
+ aad_pdata->d_b_btn_thr);
+ snd_soc_write(codec, DA7219_ACCDET_CONFIG_5,
+ aad_pdata->b_c_btn_thr);
+ snd_soc_write(codec, DA7219_ACCDET_CONFIG_6,
+ aad_pdata->c_mic_btn_thr);
+
+ cfg = 0;
+ mask = 0;
+ switch (aad_pdata->btn_avg) {
+ case DA7219_AAD_BTN_AVG_1:
+ case DA7219_AAD_BTN_AVG_2:
+ case DA7219_AAD_BTN_AVG_4:
+ case DA7219_AAD_BTN_AVG_8:
+ cfg |= (aad_pdata->btn_avg <<
+ DA7219_BUTTON_AVERAGE_SHIFT);
+ mask |= DA7219_BUTTON_AVERAGE_MASK;
+ }
+ switch (aad_pdata->adc_1bit_rpt) {
+ case DA7219_AAD_ADC_1BIT_RPT_1:
+ case DA7219_AAD_ADC_1BIT_RPT_2:
+ case DA7219_AAD_ADC_1BIT_RPT_4:
+ case DA7219_AAD_ADC_1BIT_RPT_8:
+ cfg |= (aad_pdata->adc_1bit_rpt <<
+ DA7219_ADC_1_BIT_REPEAT_SHIFT);
+ mask |= DA7219_ADC_1_BIT_REPEAT_MASK;
+ }
+ snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_7, mask, cfg);
+ }
+}
+
+
+/*
+ * Init/Exit
+ */
+
+int da7219_aad_init(struct snd_soc_codec *codec)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ struct da7219_aad_priv *da7219_aad;
+ u8 mask[DA7219_AAD_IRQ_REG_MAX];
+ int ret;
+
+ da7219_aad = devm_kzalloc(codec->dev, sizeof(*da7219_aad), GFP_KERNEL);
+ if (!da7219_aad)
+ return -ENOMEM;
+
+ da7219->aad = da7219_aad;
+ da7219_aad->codec = codec;
+
+ /* Handle any DT/platform data */
+ if ((codec->dev->of_node) && (da7219->pdata))
+ da7219->pdata->aad_pdata = da7219_aad_of_to_pdata(codec);
+
+ da7219_aad_handle_pdata(codec);
+
+ /* Disable button detection */
+ snd_soc_update_bits(codec, DA7219_ACCDET_CONFIG_1,
+ DA7219_BUTTON_CONFIG_MASK, 0);
+
+ INIT_WORK(&da7219_aad->btn_det_work, da7219_aad_btn_det_work);
+ INIT_WORK(&da7219_aad->hptest_work, da7219_aad_hptest_work);
+
+ ret = request_threaded_irq(da7219_aad->irq, NULL,
+ da7219_aad_irq_thread,
+ IRQF_TRIGGER_LOW | IRQF_ONESHOT,
+ "da7219-aad", da7219_aad);
+ if (ret) {
+ dev_err(codec->dev, "Failed to request IRQ: %d\n", ret);
+ return ret;
+ }
+
+ /* Unmask AAD IRQs */
+ memset(mask, 0, DA7219_AAD_IRQ_REG_MAX);
+ regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_MASK_A,
+ &mask, DA7219_AAD_IRQ_REG_MAX);
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(da7219_aad_init);
+
+void da7219_aad_exit(struct snd_soc_codec *codec)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ struct da7219_aad_priv *da7219_aad = da7219->aad;
+ u8 mask[DA7219_AAD_IRQ_REG_MAX];
+
+ /* Mask off AAD IRQs */
+ memset(mask, DA7219_BYTE_MASK, DA7219_AAD_IRQ_REG_MAX);
+ regmap_bulk_write(da7219->regmap, DA7219_ACCDET_IRQ_MASK_A,
+ mask, DA7219_AAD_IRQ_REG_MAX);
+
+ free_irq(da7219_aad->irq, da7219_aad);
+
+ cancel_work_sync(&da7219_aad->btn_det_work);
+ cancel_work_sync(&da7219_aad->hptest_work);
+}
+EXPORT_SYMBOL_GPL(da7219_aad_exit);
+
+MODULE_DESCRIPTION("ASoC DA7219 AAD Driver");
+MODULE_AUTHOR("Adam Thomson <Adam.Thomson.Opensource@diasemi.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/da7219-aad.h b/sound/soc/codecs/da7219-aad.h
new file mode 100644
index 000000000000..4fccf677cd06
--- /dev/null
+++ b/sound/soc/codecs/da7219-aad.h
@@ -0,0 +1,212 @@
+/*
+ * da7219-aad.h - DA7322 ASoC AAD Driver
+ *
+ * Copyright (c) 2015 Dialog Semiconductor Ltd.
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#ifndef __DA7219_AAD_H
+#define __DA7219_AAD_H
+
+#include <linux/timer.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+#include <sound/da7219-aad.h>
+
+/*
+ * Registers
+ */
+
+#define DA7219_ACCDET_STATUS_A 0xC0
+#define DA7219_ACCDET_STATUS_B 0xC1
+#define DA7219_ACCDET_IRQ_EVENT_A 0xC2
+#define DA7219_ACCDET_IRQ_EVENT_B 0xC3
+#define DA7219_ACCDET_IRQ_MASK_A 0xC4
+#define DA7219_ACCDET_IRQ_MASK_B 0xC5
+#define DA7219_ACCDET_CONFIG_1 0xC6
+#define DA7219_ACCDET_CONFIG_2 0xC7
+#define DA7219_ACCDET_CONFIG_3 0xC8
+#define DA7219_ACCDET_CONFIG_4 0xC9
+#define DA7219_ACCDET_CONFIG_5 0xCA
+#define DA7219_ACCDET_CONFIG_6 0xCB
+#define DA7219_ACCDET_CONFIG_7 0xCC
+#define DA7219_ACCDET_CONFIG_8 0xCD
+
+
+/*
+ * Bit Fields
+ */
+
+/* DA7219_ACCDET_STATUS_A = 0xC0 */
+#define DA7219_JACK_INSERTION_STS_SHIFT 0
+#define DA7219_JACK_INSERTION_STS_MASK (0x1 << 0)
+#define DA7219_JACK_TYPE_STS_SHIFT 1
+#define DA7219_JACK_TYPE_STS_MASK (0x1 << 1)
+#define DA7219_JACK_PIN_ORDER_STS_SHIFT 2
+#define DA7219_JACK_PIN_ORDER_STS_MASK (0x1 << 2)
+#define DA7219_MICBIAS_UP_STS_SHIFT 3
+#define DA7219_MICBIAS_UP_STS_MASK (0x1 << 3)
+
+/* DA7219_ACCDET_STATUS_B = 0xC1 */
+#define DA7219_BUTTON_TYPE_STS_SHIFT 0
+#define DA7219_BUTTON_TYPE_STS_MASK (0xFF << 0)
+
+/* DA7219_ACCDET_IRQ_EVENT_A = 0xC2 */
+#define DA7219_E_JACK_INSERTED_SHIFT 0
+#define DA7219_E_JACK_INSERTED_MASK (0x1 << 0)
+#define DA7219_E_JACK_REMOVED_SHIFT 1
+#define DA7219_E_JACK_REMOVED_MASK (0x1 << 1)
+#define DA7219_E_JACK_DETECT_COMPLETE_SHIFT 2
+#define DA7219_E_JACK_DETECT_COMPLETE_MASK (0x1 << 2)
+
+/* DA7219_ACCDET_IRQ_EVENT_B = 0xC3 */
+#define DA7219_E_BUTTON_A_PRESSED_SHIFT 0
+#define DA7219_E_BUTTON_A_PRESSED_MASK (0x1 << 0)
+#define DA7219_E_BUTTON_B_PRESSED_SHIFT 1
+#define DA7219_E_BUTTON_B_PRESSED_MASK (0x1 << 1)
+#define DA7219_E_BUTTON_C_PRESSED_SHIFT 2
+#define DA7219_E_BUTTON_C_PRESSED_MASK (0x1 << 2)
+#define DA7219_E_BUTTON_D_PRESSED_SHIFT 3
+#define DA7219_E_BUTTON_D_PRESSED_MASK (0x1 << 3)
+#define DA7219_E_BUTTON_D_RELEASED_SHIFT 4
+#define DA7219_E_BUTTON_D_RELEASED_MASK (0x1 << 4)
+#define DA7219_E_BUTTON_C_RELEASED_SHIFT 5
+#define DA7219_E_BUTTON_C_RELEASED_MASK (0x1 << 5)
+#define DA7219_E_BUTTON_B_RELEASED_SHIFT 6
+#define DA7219_E_BUTTON_B_RELEASED_MASK (0x1 << 6)
+#define DA7219_E_BUTTON_A_RELEASED_SHIFT 7
+#define DA7219_E_BUTTON_A_RELEASED_MASK (0x1 << 7)
+
+/* DA7219_ACCDET_IRQ_MASK_A = 0xC4 */
+#define DA7219_M_JACK_INSERTED_SHIFT 0
+#define DA7219_M_JACK_INSERTED_MASK (0x1 << 0)
+#define DA7219_M_JACK_REMOVED_SHIFT 1
+#define DA7219_M_JACK_REMOVED_MASK (0x1 << 1)
+#define DA7219_M_JACK_DETECT_COMPLETE_SHIFT 2
+#define DA7219_M_JACK_DETECT_COMPLETE_MASK (0x1 << 2)
+
+/* DA7219_ACCDET_IRQ_MASK_B = 0xC5 */
+#define DA7219_M_BUTTON_A_PRESSED_SHIFT 0
+#define DA7219_M_BUTTON_A_PRESSED_MASK (0x1 << 0)
+#define DA7219_M_BUTTON_B_PRESSED_SHIFT 1
+#define DA7219_M_BUTTON_B_PRESSED_MASK (0x1 << 1)
+#define DA7219_M_BUTTON_C_PRESSED_SHIFT 2
+#define DA7219_M_BUTTON_C_PRESSED_MASK (0x1 << 2)
+#define DA7219_M_BUTTON_D_PRESSED_SHIFT 3
+#define DA7219_M_BUTTON_D_PRESSED_MASK (0x1 << 3)
+#define DA7219_M_BUTTON_D_RELEASED_SHIFT 4
+#define DA7219_M_BUTTON_D_RELEASED_MASK (0x1 << 4)
+#define DA7219_M_BUTTON_C_RELEASED_SHIFT 5
+#define DA7219_M_BUTTON_C_RELEASED_MASK (0x1 << 5)
+#define DA7219_M_BUTTON_B_RELEASED_SHIFT 6
+#define DA7219_M_BUTTON_B_RELEASED_MASK (0x1 << 6)
+#define DA7219_M_BUTTON_A_RELEASED_SHIFT 7
+#define DA7219_M_BUTTON_A_RELEASED_MASK (0x1 << 7)
+
+/* DA7219_ACCDET_CONFIG_1 = 0xC6 */
+#define DA7219_ACCDET_EN_SHIFT 0
+#define DA7219_ACCDET_EN_MASK (0x1 << 0)
+#define DA7219_BUTTON_CONFIG_SHIFT 1
+#define DA7219_BUTTON_CONFIG_MASK (0x7 << 1)
+#define DA7219_MIC_DET_THRESH_SHIFT 4
+#define DA7219_MIC_DET_THRESH_MASK (0x3 << 4)
+#define DA7219_JACK_TYPE_DET_EN_SHIFT 6
+#define DA7219_JACK_TYPE_DET_EN_MASK (0x1 << 6)
+#define DA7219_PIN_ORDER_DET_EN_SHIFT 7
+#define DA7219_PIN_ORDER_DET_EN_MASK (0x1 << 7)
+
+/* DA7219_ACCDET_CONFIG_2 = 0xC7 */
+#define DA7219_ACCDET_PAUSE_SHIFT 0
+#define DA7219_ACCDET_PAUSE_MASK (0x1 << 0)
+#define DA7219_JACKDET_DEBOUNCE_SHIFT 1
+#define DA7219_JACKDET_DEBOUNCE_MASK (0x7 << 1)
+#define DA7219_JACK_DETECT_RATE_SHIFT 4
+#define DA7219_JACK_DETECT_RATE_MASK (0x3 << 4)
+#define DA7219_JACKDET_REM_DEB_SHIFT 6
+#define DA7219_JACKDET_REM_DEB_MASK (0x3 << 6)
+
+/* DA7219_ACCDET_CONFIG_3 = 0xC8 */
+#define DA7219_A_D_BUTTON_THRESH_SHIFT 0
+#define DA7219_A_D_BUTTON_THRESH_MASK (0xFF << 0)
+
+/* DA7219_ACCDET_CONFIG_4 = 0xC9 */
+#define DA7219_D_B_BUTTON_THRESH_SHIFT 0
+#define DA7219_D_B_BUTTON_THRESH_MASK (0xFF << 0)
+
+/* DA7219_ACCDET_CONFIG_5 = 0xCA */
+#define DA7219_B_C_BUTTON_THRESH_SHIFT 0
+#define DA7219_B_C_BUTTON_THRESH_MASK (0xFF << 0)
+
+/* DA7219_ACCDET_CONFIG_6 = 0xCB */
+#define DA7219_C_MIC_BUTTON_THRESH_SHIFT 0
+#define DA7219_C_MIC_BUTTON_THRESH_MASK (0xFF << 0)
+
+/* DA7219_ACCDET_CONFIG_7 = 0xCC */
+#define DA7219_BUTTON_AVERAGE_SHIFT 0
+#define DA7219_BUTTON_AVERAGE_MASK (0x3 << 0)
+#define DA7219_ADC_1_BIT_REPEAT_SHIFT 2
+#define DA7219_ADC_1_BIT_REPEAT_MASK (0x3 << 2)
+#define DA7219_PIN_ORDER_FORCE_SHIFT 4
+#define DA7219_PIN_ORDER_FORCE_MASK (0x1 << 4)
+#define DA7219_JACK_TYPE_FORCE_SHIFT 5
+#define DA7219_JACK_TYPE_FORCE_MASK (0x1 << 5)
+
+/* DA7219_ACCDET_CONFIG_8 = 0xCD */
+#define DA7219_HPTEST_EN_SHIFT 0
+#define DA7219_HPTEST_EN_MASK (0x1 << 0)
+#define DA7219_HPTEST_RES_SEL_SHIFT 1
+#define DA7219_HPTEST_RES_SEL_MASK (0x3 << 1)
+#define DA7219_HPTEST_RES_SEL_1KOHMS (0x0 << 1)
+#define DA7219_HPTEST_COMP_SHIFT 4
+#define DA7219_HPTEST_COMP_MASK (0x1 << 4)
+
+
+#define DA7219_AAD_MAX_BUTTONS 4
+#define DA7219_AAD_REPORT_ALL_MASK (SND_JACK_MECHANICAL | \
+ SND_JACK_HEADSET | SND_JACK_LINEOUT | \
+ SND_JACK_BTN_0 | SND_JACK_BTN_1 | \
+ SND_JACK_BTN_2 | SND_JACK_BTN_3)
+
+#define DA7219_AAD_MICBIAS_CHK_DELAY 10
+#define DA7219_AAD_MICBIAS_CHK_RETRIES 5
+
+#define DA7219_AAD_HPTEST_RAMP_FREQ 0x28
+#define DA7219_AAD_HPTEST_PERIOD 65
+
+enum da7219_aad_event_regs {
+ DA7219_AAD_IRQ_REG_A = 0,
+ DA7219_AAD_IRQ_REG_B,
+ DA7219_AAD_IRQ_REG_MAX,
+};
+
+/* Private data */
+struct da7219_aad_priv {
+ struct snd_soc_codec *codec;
+ int irq;
+
+ u8 micbias_pulse_lvl;
+ u32 micbias_pulse_time;
+
+ u8 btn_cfg;
+
+ struct work_struct btn_det_work;
+ struct work_struct hptest_work;
+
+ struct snd_soc_jack *jack;
+ bool jack_inserted;
+};
+
+/* AAD control */
+void da7219_aad_jack_det(struct snd_soc_codec *codec, struct snd_soc_jack *jack);
+
+/* Init/Exit */
+int da7219_aad_init(struct snd_soc_codec *codec);
+void da7219_aad_exit(struct snd_soc_codec *codec);
+
+#endif /* __DA7219_AAD_H */
diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c
new file mode 100644
index 000000000000..f238c1e8a69c
--- /dev/null
+++ b/sound/soc/codecs/da7219.c
@@ -0,0 +1,1955 @@
+/*
+ * da7219.c - DA7219 ALSA SoC Codec Driver
+ *
+ * Copyright (c) 2015 Dialog Semiconductor
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#include <linux/clk.h>
+#include <linux/i2c.h>
+#include <linux/of_device.h>
+#include <linux/regmap.h>
+#include <linux/slab.h>
+#include <linux/pm.h>
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/regulator/consumer.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <asm/div64.h>
+
+#include <sound/da7219.h>
+#include "da7219.h"
+#include "da7219-aad.h"
+
+
+/*
+ * TLVs and Enums
+ */
+
+/* Input TLVs */
+static const DECLARE_TLV_DB_SCALE(da7219_mic_gain_tlv, -600, 600, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_mixin_gain_tlv, -450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_adc_dig_gain_tlv, -8325, 75, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_alc_threshold_tlv, -9450, 150, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_alc_gain_tlv, 0, 600, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_alc_ana_gain_tlv, 0, 600, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_sidetone_gain_tlv, -4200, 300, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_tonegen_gain_tlv, -4500, 300, 0);
+
+/* Output TLVs */
+static const DECLARE_TLV_DB_SCALE(da7219_dac_eq_band_tlv, -1050, 150, 0);
+
+static const DECLARE_TLV_DB_RANGE(da7219_dac_dig_gain_tlv,
+ 0x0, 0x07, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1),
+ /* -77.25dB to 12dB */
+ 0x08, 0x7f, TLV_DB_SCALE_ITEM(-7725, 75, 0)
+);
+
+static const DECLARE_TLV_DB_SCALE(da7219_dac_ng_threshold_tlv, -10200, 600, 0);
+static const DECLARE_TLV_DB_SCALE(da7219_hp_gain_tlv, -5700, 100, 0);
+
+/* Input Enums */
+static const char * const da7219_alc_attack_rate_txt[] = {
+ "7.33/fs", "14.66/fs", "29.32/fs", "58.64/fs", "117.3/fs", "234.6/fs",
+ "469.1/fs", "938.2/fs", "1876/fs", "3753/fs", "7506/fs", "15012/fs",
+ "30024/fs"
+};
+
+static const struct soc_enum da7219_alc_attack_rate =
+ SOC_ENUM_SINGLE(DA7219_ALC_CTRL2, DA7219_ALC_ATTACK_SHIFT,
+ DA7219_ALC_ATTACK_MAX, da7219_alc_attack_rate_txt);
+
+static const char * const da7219_alc_release_rate_txt[] = {
+ "28.66/fs", "57.33/fs", "114.6/fs", "229.3/fs", "458.6/fs", "917.1/fs",
+ "1834/fs", "3668/fs", "7337/fs", "14674/fs", "29348/fs"
+};
+
+static const struct soc_enum da7219_alc_release_rate =
+ SOC_ENUM_SINGLE(DA7219_ALC_CTRL2, DA7219_ALC_RELEASE_SHIFT,
+ DA7219_ALC_RELEASE_MAX, da7219_alc_release_rate_txt);
+
+static const char * const da7219_alc_hold_time_txt[] = {
+ "62/fs", "124/fs", "248/fs", "496/fs", "992/fs", "1984/fs", "3968/fs",
+ "7936/fs", "15872/fs", "31744/fs", "63488/fs", "126976/fs",
+ "253952/fs", "507904/fs", "1015808/fs", "2031616/fs"
+};
+
+static const struct soc_enum da7219_alc_hold_time =
+ SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_HOLD_SHIFT,
+ DA7219_ALC_HOLD_MAX, da7219_alc_hold_time_txt);
+
+static const char * const da7219_alc_env_rate_txt[] = {
+ "1/4", "1/16", "1/256", "1/65536"
+};
+
+static const struct soc_enum da7219_alc_env_attack_rate =
+ SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_INTEG_ATTACK_SHIFT,
+ DA7219_ALC_INTEG_MAX, da7219_alc_env_rate_txt);
+
+static const struct soc_enum da7219_alc_env_release_rate =
+ SOC_ENUM_SINGLE(DA7219_ALC_CTRL3, DA7219_ALC_INTEG_RELEASE_SHIFT,
+ DA7219_ALC_INTEG_MAX, da7219_alc_env_rate_txt);
+
+static const char * const da7219_alc_anticlip_step_txt[] = {
+ "0.034dB/fs", "0.068dB/fs", "0.136dB/fs", "0.272dB/fs"
+};
+
+static const struct soc_enum da7219_alc_anticlip_step =
+ SOC_ENUM_SINGLE(DA7219_ALC_ANTICLIP_CTRL,
+ DA7219_ALC_ANTICLIP_STEP_SHIFT,
+ DA7219_ALC_ANTICLIP_STEP_MAX,
+ da7219_alc_anticlip_step_txt);
+
+/* Input/Output Enums */
+static const char * const da7219_gain_ramp_rate_txt[] = {
+ "Nominal Rate * 8", "Nominal Rate", "Nominal Rate / 8",
+ "Nominal Rate / 16"
+};
+
+static const struct soc_enum da7219_gain_ramp_rate =
+ SOC_ENUM_SINGLE(DA7219_GAIN_RAMP_CTRL, DA7219_GAIN_RAMP_RATE_SHIFT,
+ DA7219_GAIN_RAMP_RATE_MAX, da7219_gain_ramp_rate_txt);
+
+static const char * const da7219_hpf_mode_txt[] = {
+ "Disabled", "Audio", "Voice"
+};
+
+static const unsigned int da7219_hpf_mode_val[] = {
+ DA7219_HPF_DISABLED, DA7219_HPF_AUDIO_EN, DA7219_HPF_VOICE_EN,
+};
+
+static const struct soc_enum da7219_adc_hpf_mode =
+ SOC_VALUE_ENUM_SINGLE(DA7219_ADC_FILTERS1, DA7219_HPF_MODE_SHIFT,
+ DA7219_HPF_MODE_MASK, DA7219_HPF_MODE_MAX,
+ da7219_hpf_mode_txt, da7219_hpf_mode_val);
+
+static const struct soc_enum da7219_dac_hpf_mode =
+ SOC_VALUE_ENUM_SINGLE(DA7219_DAC_FILTERS1, DA7219_HPF_MODE_SHIFT,
+ DA7219_HPF_MODE_MASK, DA7219_HPF_MODE_MAX,
+ da7219_hpf_mode_txt, da7219_hpf_mode_val);
+
+static const char * const da7219_audio_hpf_corner_txt[] = {
+ "2Hz", "4Hz", "8Hz", "16Hz"
+};
+
+static const struct soc_enum da7219_adc_audio_hpf_corner =
+ SOC_ENUM_SINGLE(DA7219_ADC_FILTERS1,
+ DA7219_ADC_AUDIO_HPF_CORNER_SHIFT,
+ DA7219_AUDIO_HPF_CORNER_MAX,
+ da7219_audio_hpf_corner_txt);
+
+static const struct soc_enum da7219_dac_audio_hpf_corner =
+ SOC_ENUM_SINGLE(DA7219_DAC_FILTERS1,
+ DA7219_DAC_AUDIO_HPF_CORNER_SHIFT,
+ DA7219_AUDIO_HPF_CORNER_MAX,
+ da7219_audio_hpf_corner_txt);
+
+static const char * const da7219_voice_hpf_corner_txt[] = {
+ "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz"
+};
+
+static const struct soc_enum da7219_adc_voice_hpf_corner =
+ SOC_ENUM_SINGLE(DA7219_ADC_FILTERS1,
+ DA7219_ADC_VOICE_HPF_CORNER_SHIFT,
+ DA7219_VOICE_HPF_CORNER_MAX,
+ da7219_voice_hpf_corner_txt);
+
+static const struct soc_enum da7219_dac_voice_hpf_corner =
+ SOC_ENUM_SINGLE(DA7219_DAC_FILTERS1,
+ DA7219_DAC_VOICE_HPF_CORNER_SHIFT,
+ DA7219_VOICE_HPF_CORNER_MAX,
+ da7219_voice_hpf_corner_txt);
+
+static const char * const da7219_tonegen_dtmf_key_txt[] = {
+ "0", "1", "2", "3", "4", "5", "6", "7", "8", "9", "A", "B", "C", "D",
+ "*", "#"
+};
+
+static const struct soc_enum da7219_tonegen_dtmf_key =
+ SOC_ENUM_SINGLE(DA7219_TONE_GEN_CFG1, DA7219_DTMF_REG_SHIFT,
+ DA7219_DTMF_REG_MAX, da7219_tonegen_dtmf_key_txt);
+
+static const char * const da7219_tonegen_swg_sel_txt[] = {
+ "Sum", "SWG1", "SWG2", "SWG1_1-Cos"
+};
+
+static const struct soc_enum da7219_tonegen_swg_sel =
+ SOC_ENUM_SINGLE(DA7219_TONE_GEN_CFG2, DA7219_SWG_SEL_SHIFT,
+ DA7219_SWG_SEL_MAX, da7219_tonegen_swg_sel_txt);
+
+/* Output Enums */
+static const char * const da7219_dac_softmute_rate_txt[] = {
+ "1 Sample", "2 Samples", "4 Samples", "8 Samples", "16 Samples",
+ "32 Samples", "64 Samples"
+};
+
+static const struct soc_enum da7219_dac_softmute_rate =
+ SOC_ENUM_SINGLE(DA7219_DAC_FILTERS5, DA7219_DAC_SOFTMUTE_RATE_SHIFT,
+ DA7219_DAC_SOFTMUTE_RATE_MAX,
+ da7219_dac_softmute_rate_txt);
+
+static const char * const da7219_dac_ng_setup_time_txt[] = {
+ "256 Samples", "512 Samples", "1024 Samples", "2048 Samples"
+};
+
+static const struct soc_enum da7219_dac_ng_setup_time =
+ SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME,
+ DA7219_DAC_NG_SETUP_TIME_SHIFT,
+ DA7219_DAC_NG_SETUP_TIME_MAX,
+ da7219_dac_ng_setup_time_txt);
+
+static const char * const da7219_dac_ng_rampup_txt[] = {
+ "0.22ms/dB", "0.0138ms/dB"
+};
+
+static const struct soc_enum da7219_dac_ng_rampup_rate =
+ SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME,
+ DA7219_DAC_NG_RAMPUP_RATE_SHIFT,
+ DA7219_DAC_NG_RAMP_RATE_MAX,
+ da7219_dac_ng_rampup_txt);
+
+static const char * const da7219_dac_ng_rampdown_txt[] = {
+ "0.88ms/dB", "14.08ms/dB"
+};
+
+static const struct soc_enum da7219_dac_ng_rampdown_rate =
+ SOC_ENUM_SINGLE(DA7219_DAC_NG_SETUP_TIME,
+ DA7219_DAC_NG_RAMPDN_RATE_SHIFT,
+ DA7219_DAC_NG_RAMP_RATE_MAX,
+ da7219_dac_ng_rampdown_txt);
+
+
+static const char * const da7219_cp_track_mode_txt[] = {
+ "Largest Volume", "DAC Volume", "Signal Magnitude"
+};
+
+static const unsigned int da7219_cp_track_mode_val[] = {
+ DA7219_CP_MCHANGE_LARGEST_VOL, DA7219_CP_MCHANGE_DAC_VOL,
+ DA7219_CP_MCHANGE_SIG_MAG
+};
+
+static const struct soc_enum da7219_cp_track_mode =
+ SOC_VALUE_ENUM_SINGLE(DA7219_CP_CTRL, DA7219_CP_MCHANGE_SHIFT,
+ DA7219_CP_MCHANGE_REL_MASK, DA7219_CP_MCHANGE_MAX,
+ da7219_cp_track_mode_txt,
+ da7219_cp_track_mode_val);
+
+
+/*
+ * Control Functions
+ */
+
+/* Locked Kcontrol calls */
+static int da7219_volsw_locked_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ mutex_lock(&da7219->lock);
+ ret = snd_soc_get_volsw(kcontrol, ucontrol);
+ mutex_unlock(&da7219->lock);
+
+ return ret;
+}
+
+static int da7219_volsw_locked_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ mutex_lock(&da7219->lock);
+ ret = snd_soc_put_volsw(kcontrol, ucontrol);
+ mutex_unlock(&da7219->lock);
+
+ return ret;
+}
+
+static int da7219_enum_locked_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ mutex_lock(&da7219->lock);
+ ret = snd_soc_get_enum_double(kcontrol, ucontrol);
+ mutex_unlock(&da7219->lock);
+
+ return ret;
+}
+
+static int da7219_enum_locked_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ mutex_lock(&da7219->lock);
+ ret = snd_soc_put_enum_double(kcontrol, ucontrol);
+ mutex_unlock(&da7219->lock);
+
+ return ret;
+}
+
+/* ALC */
+static void da7219_alc_calib(struct snd_soc_codec *codec)
+{
+ u8 mic_ctrl, mixin_ctrl, adc_ctrl, calib_ctrl;
+
+ /* Save current state of mic control register */
+ mic_ctrl = snd_soc_read(codec, DA7219_MIC_1_CTRL);
+
+ /* Save current state of input mixer control register */
+ mixin_ctrl = snd_soc_read(codec, DA7219_MIXIN_L_CTRL);
+
+ /* Save current state of input ADC control register */
+ adc_ctrl = snd_soc_read(codec, DA7219_ADC_L_CTRL);
+
+ /* Enable then Mute MIC PGAs */
+ snd_soc_update_bits(codec, DA7219_MIC_1_CTRL, DA7219_MIC_1_AMP_EN_MASK,
+ DA7219_MIC_1_AMP_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_MIC_1_CTRL,
+ DA7219_MIC_1_AMP_MUTE_EN_MASK,
+ DA7219_MIC_1_AMP_MUTE_EN_MASK);
+
+ /* Enable input mixers unmuted */
+ snd_soc_update_bits(codec, DA7219_MIXIN_L_CTRL,
+ DA7219_MIXIN_L_AMP_EN_MASK |
+ DA7219_MIXIN_L_AMP_MUTE_EN_MASK,
+ DA7219_MIXIN_L_AMP_EN_MASK);
+
+ /* Enable input filters unmuted */
+ snd_soc_update_bits(codec, DA7219_ADC_L_CTRL,
+ DA7219_ADC_L_MUTE_EN_MASK | DA7219_ADC_L_EN_MASK,
+ DA7219_ADC_L_EN_MASK);
+
+ /* Perform auto calibration */
+ snd_soc_update_bits(codec, DA7219_ALC_CTRL1,
+ DA7219_ALC_AUTO_CALIB_EN_MASK,
+ DA7219_ALC_AUTO_CALIB_EN_MASK);
+ do {
+ calib_ctrl = snd_soc_read(codec, DA7219_ALC_CTRL1);
+ } while (calib_ctrl & DA7219_ALC_AUTO_CALIB_EN_MASK);
+
+ /* If auto calibration fails, disable DC offset, hybrid ALC */
+ if (calib_ctrl & DA7219_ALC_CALIB_OVERFLOW_MASK) {
+ dev_warn(codec->dev,
+ "ALC auto calibration failed with overflow\n");
+ snd_soc_update_bits(codec, DA7219_ALC_CTRL1,
+ DA7219_ALC_OFFSET_EN_MASK |
+ DA7219_ALC_SYNC_MODE_MASK, 0);
+ } else {
+ /* Enable DC offset cancellation, hybrid mode */
+ snd_soc_update_bits(codec, DA7219_ALC_CTRL1,
+ DA7219_ALC_OFFSET_EN_MASK |
+ DA7219_ALC_SYNC_MODE_MASK,
+ DA7219_ALC_OFFSET_EN_MASK |
+ DA7219_ALC_SYNC_MODE_MASK);
+ }
+
+ /* Restore input filter control register to original state */
+ snd_soc_write(codec, DA7219_ADC_L_CTRL, adc_ctrl);
+
+ /* Restore input mixer control registers to original state */
+ snd_soc_write(codec, DA7219_MIXIN_L_CTRL, mixin_ctrl);
+
+ /* Restore MIC control registers to original states */
+ snd_soc_write(codec, DA7219_MIC_1_CTRL, mic_ctrl);
+}
+
+static int da7219_mixin_gain_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ ret = snd_soc_put_volsw(kcontrol, ucontrol);
+
+ /*
+ * If ALC in operation and value of control has been updated,
+ * make sure calibrated offsets are updated.
+ */
+ if ((ret == 1) && (da7219->alc_en))
+ da7219_alc_calib(codec);
+
+ return ret;
+}
+
+static int da7219_alc_sw_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+
+ /* Force ALC offset calibration if enabling ALC */
+ if ((ucontrol->value.integer.value[0]) && (!da7219->alc_en)) {
+ da7219_alc_calib(codec);
+ da7219->alc_en = true;
+ } else {
+ da7219->alc_en = false;
+ }
+
+ return snd_soc_put_volsw(kcontrol, ucontrol);
+}
+
+/* ToneGen */
+static int da7219_tonegen_freq_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ struct soc_mixer_control *mixer_ctrl =
+ (struct soc_mixer_control *) kcontrol->private_value;
+ unsigned int reg = mixer_ctrl->reg;
+ u16 val;
+ int ret;
+
+ mutex_lock(&da7219->lock);
+ ret = regmap_raw_read(da7219->regmap, reg, &val, sizeof(val));
+ mutex_unlock(&da7219->lock);
+
+ if (ret)
+ return ret;
+
+ /*
+ * Frequency value spans two 8-bit registers, lower then upper byte.
+ * Therefore we need to convert to host endianness here.
+ */
+ ucontrol->value.integer.value[0] = le16_to_cpu(val);
+
+ return 0;
+}
+
+static int da7219_tonegen_freq_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ struct soc_mixer_control *mixer_ctrl =
+ (struct soc_mixer_control *) kcontrol->private_value;
+ unsigned int reg = mixer_ctrl->reg;
+ u16 val;
+ int ret;
+
+ /*
+ * Frequency value spans two 8-bit registers, lower then upper byte.
+ * Therefore we need to convert to little endian here to align with
+ * HW registers.
+ */
+ val = cpu_to_le16(ucontrol->value.integer.value[0]);
+
+ mutex_lock(&da7219->lock);
+ ret = regmap_raw_write(da7219->regmap, reg, &val, sizeof(val));
+ mutex_unlock(&da7219->lock);
+
+ return ret;
+}
+
+
+/*
+ * KControls
+ */
+
+static const struct snd_kcontrol_new da7219_snd_controls[] = {
+ /* Mics */
+ SOC_SINGLE_TLV("Mic Volume", DA7219_MIC_1_GAIN,
+ DA7219_MIC_1_AMP_GAIN_SHIFT, DA7219_MIC_1_AMP_GAIN_MAX,
+ DA7219_NO_INVERT, da7219_mic_gain_tlv),
+ SOC_SINGLE("Mic Switch", DA7219_MIC_1_CTRL,
+ DA7219_MIC_1_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_INVERT),
+
+ /* Mixer Input */
+ SOC_SINGLE_EXT_TLV("Mixin Volume", DA7219_MIXIN_L_GAIN,
+ DA7219_MIXIN_L_AMP_GAIN_SHIFT,
+ DA7219_MIXIN_L_AMP_GAIN_MAX, DA7219_NO_INVERT,
+ snd_soc_get_volsw, da7219_mixin_gain_put,
+ da7219_mixin_gain_tlv),
+ SOC_SINGLE("Mixin Switch", DA7219_MIXIN_L_CTRL,
+ DA7219_MIXIN_L_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_INVERT),
+ SOC_SINGLE("Mixin Gain Ramp Switch", DA7219_MIXIN_L_CTRL,
+ DA7219_MIXIN_L_AMP_RAMP_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_NO_INVERT),
+ SOC_SINGLE("Mixin ZC Gain Switch", DA7219_MIXIN_L_CTRL,
+ DA7219_MIXIN_L_AMP_ZC_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_NO_INVERT),
+
+ /* ADC */
+ SOC_SINGLE_TLV("Capture Digital Volume", DA7219_ADC_L_GAIN,
+ DA7219_ADC_L_DIGITAL_GAIN_SHIFT,
+ DA7219_ADC_L_DIGITAL_GAIN_MAX, DA7219_NO_INVERT,
+ da7219_adc_dig_gain_tlv),
+ SOC_SINGLE("Capture Digital Switch", DA7219_ADC_L_CTRL,
+ DA7219_ADC_L_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_INVERT),
+ SOC_SINGLE("Capture Digital Gain Ramp Switch", DA7219_ADC_L_CTRL,
+ DA7219_ADC_L_RAMP_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_NO_INVERT),
+
+ /* ALC */
+ SOC_ENUM("ALC Attack Rate", da7219_alc_attack_rate),
+ SOC_ENUM("ALC Release Rate", da7219_alc_release_rate),
+ SOC_ENUM("ALC Hold Time", da7219_alc_hold_time),
+ SOC_ENUM("ALC Envelope Attack Rate", da7219_alc_env_attack_rate),
+ SOC_ENUM("ALC Envelope Release Rate", da7219_alc_env_release_rate),
+ SOC_SINGLE_TLV("ALC Noise Threshold", DA7219_ALC_NOISE,
+ DA7219_ALC_NOISE_SHIFT, DA7219_ALC_THRESHOLD_MAX,
+ DA7219_INVERT, da7219_alc_threshold_tlv),
+ SOC_SINGLE_TLV("ALC Min Threshold", DA7219_ALC_TARGET_MIN,
+ DA7219_ALC_THRESHOLD_MIN_SHIFT, DA7219_ALC_THRESHOLD_MAX,
+ DA7219_INVERT, da7219_alc_threshold_tlv),
+ SOC_SINGLE_TLV("ALC Max Threshold", DA7219_ALC_TARGET_MAX,
+ DA7219_ALC_THRESHOLD_MAX_SHIFT, DA7219_ALC_THRESHOLD_MAX,
+ DA7219_INVERT, da7219_alc_threshold_tlv),
+ SOC_SINGLE_TLV("ALC Max Attenuation", DA7219_ALC_GAIN_LIMITS,
+ DA7219_ALC_ATTEN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX,
+ DA7219_NO_INVERT, da7219_alc_gain_tlv),
+ SOC_SINGLE_TLV("ALC Max Volume", DA7219_ALC_GAIN_LIMITS,
+ DA7219_ALC_GAIN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX,
+ DA7219_NO_INVERT, da7219_alc_gain_tlv),
+ SOC_SINGLE_RANGE_TLV("ALC Min Analog Volume", DA7219_ALC_ANA_GAIN_LIMITS,
+ DA7219_ALC_ANA_GAIN_MIN_SHIFT,
+ DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX,
+ DA7219_NO_INVERT, da7219_alc_ana_gain_tlv),
+ SOC_SINGLE_RANGE_TLV("ALC Max Analog Volume", DA7219_ALC_ANA_GAIN_LIMITS,
+ DA7219_ALC_ANA_GAIN_MAX_SHIFT,
+ DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX,
+ DA7219_NO_INVERT, da7219_alc_ana_gain_tlv),
+ SOC_ENUM("ALC Anticlip Step", da7219_alc_anticlip_step),
+ SOC_SINGLE("ALC Anticlip Switch", DA7219_ALC_ANTICLIP_CTRL,
+ DA7219_ALC_ANTIPCLIP_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_NO_INVERT),
+ SOC_SINGLE_EXT("ALC Switch", DA7219_ALC_CTRL1, DA7219_ALC_EN_SHIFT,
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT,
+ snd_soc_get_volsw, da7219_alc_sw_put),
+
+ /* Input High-Pass Filters */
+ SOC_ENUM("ADC HPF Mode", da7219_adc_hpf_mode),
+ SOC_ENUM("ADC HPF Corner Audio", da7219_adc_audio_hpf_corner),
+ SOC_ENUM("ADC HPF Corner Voice", da7219_adc_voice_hpf_corner),
+
+ /* Sidetone Filter */
+ SOC_SINGLE_TLV("Sidetone Volume", DA7219_SIDETONE_GAIN,
+ DA7219_SIDETONE_GAIN_SHIFT, DA7219_SIDETONE_GAIN_MAX,
+ DA7219_NO_INVERT, da7219_sidetone_gain_tlv),
+ SOC_SINGLE("Sidetone Switch", DA7219_SIDETONE_CTRL,
+ DA7219_SIDETONE_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_INVERT),
+
+ /* Tone Generator */
+ SOC_SINGLE_EXT_TLV("ToneGen Volume", DA7219_TONE_GEN_CFG2,
+ DA7219_TONE_GEN_GAIN_SHIFT, DA7219_TONE_GEN_GAIN_MAX,
+ DA7219_NO_INVERT, da7219_volsw_locked_get,
+ da7219_volsw_locked_put, da7219_tonegen_gain_tlv),
+ SOC_ENUM_EXT("ToneGen DTMF Key", da7219_tonegen_dtmf_key,
+ da7219_enum_locked_get, da7219_enum_locked_put),
+ SOC_SINGLE_EXT("ToneGen DTMF Switch", DA7219_TONE_GEN_CFG1,
+ DA7219_DTMF_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_NO_INVERT, da7219_volsw_locked_get,
+ da7219_volsw_locked_put),
+ SOC_ENUM_EXT("ToneGen Sinewave Gen Type", da7219_tonegen_swg_sel,
+ da7219_enum_locked_get, da7219_enum_locked_put),
+ SOC_SINGLE_EXT("ToneGen Sinewave1 Freq", DA7219_TONE_GEN_FREQ1_L,
+ DA7219_FREQ1_L_SHIFT, DA7219_FREQ_MAX, DA7219_NO_INVERT,
+ da7219_tonegen_freq_get, da7219_tonegen_freq_put),
+ SOC_SINGLE_EXT("ToneGen Sinewave2 Freq", DA7219_TONE_GEN_FREQ2_L,
+ DA7219_FREQ2_L_SHIFT, DA7219_FREQ_MAX, DA7219_NO_INVERT,
+ da7219_tonegen_freq_get, da7219_tonegen_freq_put),
+ SOC_SINGLE_EXT("ToneGen On Time", DA7219_TONE_GEN_ON_PER,
+ DA7219_BEEP_ON_PER_SHIFT, DA7219_BEEP_ON_OFF_MAX,
+ DA7219_NO_INVERT, da7219_volsw_locked_get,
+ da7219_volsw_locked_put),
+ SOC_SINGLE("ToneGen Off Time", DA7219_TONE_GEN_OFF_PER,
+ DA7219_BEEP_OFF_PER_SHIFT, DA7219_BEEP_ON_OFF_MAX,
+ DA7219_NO_INVERT),
+
+ /* Gain ramping */
+ SOC_ENUM("Gain Ramp Rate", da7219_gain_ramp_rate),
+
+ /* DAC High-Pass Filter */
+ SOC_ENUM_EXT("DAC HPF Mode", da7219_dac_hpf_mode,
+ da7219_enum_locked_get, da7219_enum_locked_put),
+ SOC_ENUM("DAC HPF Corner Audio", da7219_dac_audio_hpf_corner),
+ SOC_ENUM("DAC HPF Corner Voice", da7219_dac_voice_hpf_corner),
+
+ /* DAC 5-Band Equaliser */
+ SOC_SINGLE_TLV("DAC EQ Band1 Volume", DA7219_DAC_FILTERS2,
+ DA7219_DAC_EQ_BAND1_SHIFT, DA7219_DAC_EQ_BAND_MAX,
+ DA7219_NO_INVERT, da7219_dac_eq_band_tlv),
+ SOC_SINGLE_TLV("DAC EQ Band2 Volume", DA7219_DAC_FILTERS2,
+ DA7219_DAC_EQ_BAND2_SHIFT, DA7219_DAC_EQ_BAND_MAX,
+ DA7219_NO_INVERT, da7219_dac_eq_band_tlv),
+ SOC_SINGLE_TLV("DAC EQ Band3 Volume", DA7219_DAC_FILTERS3,
+ DA7219_DAC_EQ_BAND3_SHIFT, DA7219_DAC_EQ_BAND_MAX,
+ DA7219_NO_INVERT, da7219_dac_eq_band_tlv),
+ SOC_SINGLE_TLV("DAC EQ Band4 Volume", DA7219_DAC_FILTERS3,
+ DA7219_DAC_EQ_BAND4_SHIFT, DA7219_DAC_EQ_BAND_MAX,
+ DA7219_NO_INVERT, da7219_dac_eq_band_tlv),
+ SOC_SINGLE_TLV("DAC EQ Band5 Volume", DA7219_DAC_FILTERS4,
+ DA7219_DAC_EQ_BAND5_SHIFT, DA7219_DAC_EQ_BAND_MAX,
+ DA7219_NO_INVERT, da7219_dac_eq_band_tlv),
+ SOC_SINGLE_EXT("DAC EQ Switch", DA7219_DAC_FILTERS4,
+ DA7219_DAC_EQ_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_NO_INVERT, da7219_volsw_locked_get,
+ da7219_volsw_locked_put),
+
+ /* DAC Softmute */
+ SOC_ENUM("DAC Soft Mute Rate", da7219_dac_softmute_rate),
+ SOC_SINGLE_EXT("DAC Soft Mute Switch", DA7219_DAC_FILTERS5,
+ DA7219_DAC_SOFTMUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_NO_INVERT, da7219_volsw_locked_get,
+ da7219_volsw_locked_put),
+
+ /* DAC Noise Gate */
+ SOC_ENUM("DAC NG Setup Time", da7219_dac_ng_setup_time),
+ SOC_ENUM("DAC NG Rampup Rate", da7219_dac_ng_rampup_rate),
+ SOC_ENUM("DAC NG Rampdown Rate", da7219_dac_ng_rampdown_rate),
+ SOC_SINGLE_TLV("DAC NG Off Threshold", DA7219_DAC_NG_OFF_THRESH,
+ DA7219_DAC_NG_OFF_THRESHOLD_SHIFT,
+ DA7219_DAC_NG_THRESHOLD_MAX, DA7219_NO_INVERT,
+ da7219_dac_ng_threshold_tlv),
+ SOC_SINGLE_TLV("DAC NG On Threshold", DA7219_DAC_NG_ON_THRESH,
+ DA7219_DAC_NG_ON_THRESHOLD_SHIFT,
+ DA7219_DAC_NG_THRESHOLD_MAX, DA7219_NO_INVERT,
+ da7219_dac_ng_threshold_tlv),
+ SOC_SINGLE("DAC NG Switch", DA7219_DAC_NG_CTRL, DA7219_DAC_NG_EN_SHIFT,
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+
+ /* DACs */
+ SOC_DOUBLE_R_EXT_TLV("Playback Digital Volume", DA7219_DAC_L_GAIN,
+ DA7219_DAC_R_GAIN, DA7219_DAC_L_DIGITAL_GAIN_SHIFT,
+ DA7219_DAC_DIGITAL_GAIN_MAX, DA7219_NO_INVERT,
+ da7219_volsw_locked_get, da7219_volsw_locked_put,
+ da7219_dac_dig_gain_tlv),
+ SOC_DOUBLE_R_EXT("Playback Digital Switch", DA7219_DAC_L_CTRL,
+ DA7219_DAC_R_CTRL, DA7219_DAC_L_MUTE_EN_SHIFT,
+ DA7219_SWITCH_EN_MAX, DA7219_INVERT,
+ da7219_volsw_locked_get, da7219_volsw_locked_put),
+ SOC_DOUBLE_R("Playback Digital Gain Ramp Switch", DA7219_DAC_L_CTRL,
+ DA7219_DAC_R_CTRL, DA7219_DAC_L_RAMP_EN_SHIFT,
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+
+ /* CP */
+ SOC_ENUM("Charge Pump Track Mode", da7219_cp_track_mode),
+ SOC_SINGLE("Charge Pump Threshold", DA7219_CP_VOL_THRESHOLD1,
+ DA7219_CP_THRESH_VDD2_SHIFT, DA7219_CP_THRESH_VDD2_MAX,
+ DA7219_NO_INVERT),
+
+ /* Headphones */
+ SOC_DOUBLE_R_EXT_TLV("Headphone Volume", DA7219_HP_L_GAIN,
+ DA7219_HP_R_GAIN, DA7219_HP_L_AMP_GAIN_SHIFT,
+ DA7219_HP_AMP_GAIN_MAX, DA7219_NO_INVERT,
+ da7219_volsw_locked_get, da7219_volsw_locked_put,
+ da7219_hp_gain_tlv),
+ SOC_DOUBLE_R_EXT("Headphone Switch", DA7219_HP_L_CTRL, DA7219_HP_R_CTRL,
+ DA7219_HP_L_AMP_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX,
+ DA7219_INVERT, da7219_volsw_locked_get,
+ da7219_volsw_locked_put),
+ SOC_DOUBLE_R("Headphone Gain Ramp Switch", DA7219_HP_L_CTRL,
+ DA7219_HP_R_CTRL, DA7219_HP_L_AMP_RAMP_EN_SHIFT,
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+ SOC_DOUBLE_R("Headphone ZC Gain Switch", DA7219_HP_L_CTRL,
+ DA7219_HP_R_CTRL, DA7219_HP_L_AMP_ZC_EN_SHIFT,
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+};
+
+
+/*
+ * DAPM Mux Controls
+ */
+
+static const char * const da7219_out_sel_txt[] = {
+ "ADC", "Tone Generator", "DAIL", "DAIR"
+};
+
+static const struct soc_enum da7219_out_dail_sel =
+ SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAI,
+ DA7219_DAI_L_SRC_SHIFT,
+ DA7219_OUT_SRC_MAX,
+ da7219_out_sel_txt);
+
+static const struct snd_kcontrol_new da7219_out_dail_sel_mux =
+ SOC_DAPM_ENUM("Out DAIL Mux", da7219_out_dail_sel);
+
+static const struct soc_enum da7219_out_dair_sel =
+ SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAI,
+ DA7219_DAI_R_SRC_SHIFT,
+ DA7219_OUT_SRC_MAX,
+ da7219_out_sel_txt);
+
+static const struct snd_kcontrol_new da7219_out_dair_sel_mux =
+ SOC_DAPM_ENUM("Out DAIR Mux", da7219_out_dair_sel);
+
+static const struct soc_enum da7219_out_dacl_sel =
+ SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAC,
+ DA7219_DAC_L_SRC_SHIFT,
+ DA7219_OUT_SRC_MAX,
+ da7219_out_sel_txt);
+
+static const struct snd_kcontrol_new da7219_out_dacl_sel_mux =
+ SOC_DAPM_ENUM("Out DACL Mux", da7219_out_dacl_sel);
+
+static const struct soc_enum da7219_out_dacr_sel =
+ SOC_ENUM_SINGLE(DA7219_DIG_ROUTING_DAC,
+ DA7219_DAC_R_SRC_SHIFT,
+ DA7219_OUT_SRC_MAX,
+ da7219_out_sel_txt);
+
+static const struct snd_kcontrol_new da7219_out_dacr_sel_mux =
+ SOC_DAPM_ENUM("Out DACR Mux", da7219_out_dacr_sel);
+
+
+/*
+ * DAPM Mixer Controls
+ */
+
+static const struct snd_kcontrol_new da7219_mixin_controls[] = {
+ SOC_DAPM_SINGLE("Mic Switch", DA7219_MIXIN_L_SELECT,
+ DA7219_MIXIN_L_MIX_SELECT_SHIFT,
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+};
+
+static const struct snd_kcontrol_new da7219_mixout_l_controls[] = {
+ SOC_DAPM_SINGLE("DACL Switch", DA7219_MIXOUT_L_SELECT,
+ DA7219_MIXOUT_L_MIX_SELECT_SHIFT,
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+};
+
+static const struct snd_kcontrol_new da7219_mixout_r_controls[] = {
+ SOC_DAPM_SINGLE("DACR Switch", DA7219_MIXOUT_R_SELECT,
+ DA7219_MIXOUT_R_MIX_SELECT_SHIFT,
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT),
+};
+
+#define DA7219_DMIX_ST_CTRLS(reg) \
+ SOC_DAPM_SINGLE("Out FilterL Switch", reg, \
+ DA7219_DMIX_ST_SRC_OUTFILT1L_SHIFT, \
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), \
+ SOC_DAPM_SINGLE("Out FilterR Switch", reg, \
+ DA7219_DMIX_ST_SRC_OUTFILT1R_SHIFT, \
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), \
+ SOC_DAPM_SINGLE("Sidetone Switch", reg, \
+ DA7219_DMIX_ST_SRC_SIDETONE_SHIFT, \
+ DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT) \
+
+static const struct snd_kcontrol_new da7219_st_out_filtl_mix_controls[] = {
+ DA7219_DMIX_ST_CTRLS(DA7219_DROUTING_ST_OUTFILT_1L),
+};
+
+static const struct snd_kcontrol_new da7219_st_out_filtr_mix_controls[] = {
+ DA7219_DMIX_ST_CTRLS(DA7219_DROUTING_ST_OUTFILT_1R),
+};
+
+
+/*
+ * DAPM Events
+ */
+
+static int da7219_dai_event(struct snd_soc_dapm_widget *w,
+ struct snd_kcontrol *kcontrol, int event)
+{
+ struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ u8 pll_ctrl, pll_status;
+ int i = 0;
+ bool srm_lock = false;
+
+ switch (event) {
+ case SND_SOC_DAPM_PRE_PMU:
+ if (da7219->master)
+ /* Enable DAI clks for master mode */
+ snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE,
+ DA7219_DAI_CLK_EN_MASK,
+ DA7219_DAI_CLK_EN_MASK);
+
+ /* PC synchronised to DAI */
+ snd_soc_update_bits(codec, DA7219_PC_COUNT,
+ DA7219_PC_FREERUN_MASK, 0);
+
+ /* Slave mode, if SRM not enabled no need for status checks */
+ pll_ctrl = snd_soc_read(codec, DA7219_PLL_CTRL);
+ if ((pll_ctrl & DA7219_PLL_MODE_MASK) != DA7219_PLL_MODE_SRM)
+ return 0;
+
+ /* Check SRM has locked */
+ do {
+ pll_status = snd_soc_read(codec, DA7219_PLL_SRM_STS);
+ if (pll_status & DA7219_PLL_SRM_STS_SRM_LOCK) {
+ srm_lock = true;
+ } else {
+ ++i;
+ msleep(50);
+ }
+ } while ((i < DA7219_SRM_CHECK_RETRIES) & (!srm_lock));
+
+ if (!srm_lock)
+ dev_warn(codec->dev, "SRM failed to lock\n");
+
+ return 0;
+ case SND_SOC_DAPM_POST_PMD:
+ /* PC free-running */
+ snd_soc_update_bits(codec, DA7219_PC_COUNT,
+ DA7219_PC_FREERUN_MASK,
+ DA7219_PC_FREERUN_MASK);
+
+ /* Disable DAI clks if in master mode */
+ if (da7219->master)
+ snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE,
+ DA7219_DAI_CLK_EN_MASK, 0);
+ return 0;
+ default:
+ return -EINVAL;
+ }
+}
+
+
+/*
+ * DAPM Widgets
+ */
+
+static const struct snd_soc_dapm_widget da7219_dapm_widgets[] = {
+ /* Input Supplies */
+ SND_SOC_DAPM_SUPPLY("Mic Bias", DA7219_MICBIAS_CTRL,
+ DA7219_MICBIAS1_EN_SHIFT, DA7219_NO_INVERT,
+ NULL, 0),
+
+ /* Inputs */
+ SND_SOC_DAPM_INPUT("MIC"),
+
+ /* Input PGAs */
+ SND_SOC_DAPM_PGA("Mic PGA", DA7219_MIC_1_CTRL,
+ DA7219_MIC_1_AMP_EN_SHIFT, DA7219_NO_INVERT,
+ NULL, 0),
+ SND_SOC_DAPM_PGA("Mixin PGA", DA7219_MIXIN_L_CTRL,
+ DA7219_MIXIN_L_AMP_EN_SHIFT, DA7219_NO_INVERT,
+ NULL, 0),
+
+ /* Input Filters */
+ SND_SOC_DAPM_ADC("ADC", NULL, DA7219_ADC_L_CTRL, DA7219_ADC_L_EN_SHIFT,
+ DA7219_NO_INVERT),
+
+ /* Tone Generator */
+ SND_SOC_DAPM_SIGGEN("TONE"),
+ SND_SOC_DAPM_PGA("Tone Generator", DA7219_TONE_GEN_CFG1,
+ DA7219_START_STOPN_SHIFT, DA7219_NO_INVERT, NULL, 0),
+
+ /* Sidetone Input */
+ SND_SOC_DAPM_ADC("Sidetone Filter", NULL, DA7219_SIDETONE_CTRL,
+ DA7219_SIDETONE_EN_SHIFT, DA7219_NO_INVERT),
+
+ /* Input Mixer Supply */
+ SND_SOC_DAPM_SUPPLY("Mixer In Supply", DA7219_MIXIN_L_CTRL,
+ DA7219_MIXIN_L_MIX_EN_SHIFT, DA7219_NO_INVERT,
+ NULL, 0),
+
+ /* Input Mixer */
+ SND_SOC_DAPM_MIXER("Mixer In", SND_SOC_NOPM, 0, 0,
+ da7219_mixin_controls,
+ ARRAY_SIZE(da7219_mixin_controls)),
+
+ /* Input Muxes */
+ SND_SOC_DAPM_MUX("Out DAIL Mux", SND_SOC_NOPM, 0, 0,
+ &da7219_out_dail_sel_mux),
+ SND_SOC_DAPM_MUX("Out DAIR Mux", SND_SOC_NOPM, 0, 0,
+ &da7219_out_dair_sel_mux),
+
+ /* DAI Supply */
+ SND_SOC_DAPM_SUPPLY("DAI", DA7219_DAI_CTRL, DA7219_DAI_EN_SHIFT,
+ DA7219_NO_INVERT, da7219_dai_event,
+ SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD),
+
+ /* DAI */
+ SND_SOC_DAPM_AIF_OUT("DAIOUT", "Capture", 0, SND_SOC_NOPM, 0, 0),
+ SND_SOC_DAPM_AIF_IN("DAIIN", "Playback", 0, SND_SOC_NOPM, 0, 0),
+
+ /* Output Muxes */
+ SND_SOC_DAPM_MUX("Out DACL Mux", SND_SOC_NOPM, 0, 0,
+ &da7219_out_dacl_sel_mux),
+ SND_SOC_DAPM_MUX("Out DACR Mux", SND_SOC_NOPM, 0, 0,
+ &da7219_out_dacr_sel_mux),
+
+ /* Output Mixers */
+ SND_SOC_DAPM_MIXER("Mixer Out FilterL", SND_SOC_NOPM, 0, 0,
+ da7219_mixout_l_controls,
+ ARRAY_SIZE(da7219_mixout_l_controls)),
+ SND_SOC_DAPM_MIXER("Mixer Out FilterR", SND_SOC_NOPM, 0, 0,
+ da7219_mixout_r_controls,
+ ARRAY_SIZE(da7219_mixout_r_controls)),
+
+ /* Sidetone Mixers */
+ SND_SOC_DAPM_MIXER("ST Mixer Out FilterL", SND_SOC_NOPM, 0, 0,
+ da7219_st_out_filtl_mix_controls,
+ ARRAY_SIZE(da7219_st_out_filtl_mix_controls)),
+ SND_SOC_DAPM_MIXER("ST Mixer Out FilterR", SND_SOC_NOPM, 0,
+ 0, da7219_st_out_filtr_mix_controls,
+ ARRAY_SIZE(da7219_st_out_filtr_mix_controls)),
+
+ /* DACs */
+ SND_SOC_DAPM_DAC("DACL", NULL, DA7219_DAC_L_CTRL, DA7219_DAC_L_EN_SHIFT,
+ DA7219_NO_INVERT),
+ SND_SOC_DAPM_DAC("DACR", NULL, DA7219_DAC_R_CTRL, DA7219_DAC_R_EN_SHIFT,
+ DA7219_NO_INVERT),
+
+ /* Output PGAs */
+ SND_SOC_DAPM_PGA("Mixout Left PGA", DA7219_MIXOUT_L_CTRL,
+ DA7219_MIXOUT_L_AMP_EN_SHIFT, DA7219_NO_INVERT,
+ NULL, 0),
+ SND_SOC_DAPM_PGA("Mixout Right PGA", DA7219_MIXOUT_R_CTRL,
+ DA7219_MIXOUT_R_AMP_EN_SHIFT, DA7219_NO_INVERT,
+ NULL, 0),
+ SND_SOC_DAPM_PGA("Headphone Left PGA", DA7219_HP_L_CTRL,
+ DA7219_HP_L_AMP_EN_SHIFT, DA7219_NO_INVERT, NULL, 0),
+ SND_SOC_DAPM_PGA("Headphone Right PGA", DA7219_HP_R_CTRL,
+ DA7219_HP_R_AMP_EN_SHIFT, DA7219_NO_INVERT, NULL, 0),
+
+ /* Output Supplies */
+ SND_SOC_DAPM_SUPPLY("Charge Pump", DA7219_CP_CTRL, DA7219_CP_EN_SHIFT,
+ DA7219_NO_INVERT, NULL, 0),
+
+ /* Outputs */
+ SND_SOC_DAPM_OUTPUT("HPL"),
+ SND_SOC_DAPM_OUTPUT("HPR"),
+};
+
+
+/*
+ * DAPM Mux Routes
+ */
+
+#define DA7219_OUT_DAI_MUX_ROUTES(name) \
+ {name, "ADC", "Mixer In"}, \
+ {name, "Tone Generator", "Tone Generator"}, \
+ {name, "DAIL", "DAIOUT"}, \
+ {name, "DAIR", "DAIOUT"}
+
+#define DA7219_OUT_DAC_MUX_ROUTES(name) \
+ {name, "ADC", "Mixer In"}, \
+ {name, "Tone Generator", "Tone Generator"}, \
+ {name, "DAIL", "DAIIN"}, \
+ {name, "DAIR", "DAIIN"}
+
+/*
+ * DAPM Mixer Routes
+ */
+
+#define DA7219_DMIX_ST_ROUTES(name) \
+ {name, "Out FilterL Switch", "Mixer Out FilterL"}, \
+ {name, "Out FilterR Switch", "Mixer Out FilterR"}, \
+ {name, "Sidetone Switch", "Sidetone Filter"}
+
+
+/*
+ * DAPM audio route definition
+ */
+
+static const struct snd_soc_dapm_route da7219_audio_map[] = {
+ /* Input paths */
+ {"MIC", NULL, "Mic Bias"},
+ {"Mic PGA", NULL, "MIC"},
+ {"Mixin PGA", NULL, "Mic PGA"},
+ {"ADC", NULL, "Mixin PGA"},
+
+ {"Sidetone Filter", NULL, "ADC"},
+ {"Mixer In", NULL, "Mixer In Supply"},
+ {"Mixer In", "Mic Switch", "ADC"},
+
+ {"Tone Generator", NULL, "TONE"},
+
+ DA7219_OUT_DAI_MUX_ROUTES("Out DAIL Mux"),
+ DA7219_OUT_DAI_MUX_ROUTES("Out DAIR Mux"),
+
+ {"DAIOUT", NULL, "Out DAIL Mux"},
+ {"DAIOUT", NULL, "Out DAIR Mux"},
+ {"DAIOUT", NULL, "DAI"},
+
+ /* Output paths */
+ {"DAIIN", NULL, "DAI"},
+
+ DA7219_OUT_DAC_MUX_ROUTES("Out DACL Mux"),
+ DA7219_OUT_DAC_MUX_ROUTES("Out DACR Mux"),
+
+ {"Mixer Out FilterL", "DACL Switch", "Out DACL Mux"},
+ {"Mixer Out FilterR", "DACR Switch", "Out DACR Mux"},
+
+ DA7219_DMIX_ST_ROUTES("ST Mixer Out FilterL"),
+ DA7219_DMIX_ST_ROUTES("ST Mixer Out FilterR"),
+
+ {"DACL", NULL, "ST Mixer Out FilterL"},
+ {"DACR", NULL, "ST Mixer Out FilterR"},
+
+ {"Mixout Left PGA", NULL, "DACL"},
+ {"Mixout Right PGA", NULL, "DACR"},
+
+ {"Headphone Left PGA", NULL, "Mixout Left PGA"},
+ {"Headphone Right PGA", NULL, "Mixout Right PGA"},
+
+ {"HPL", NULL, "Headphone Left PGA"},
+ {"HPR", NULL, "Headphone Right PGA"},
+
+ {"HPL", NULL, "Charge Pump"},
+ {"HPR", NULL, "Charge Pump"},
+};
+
+
+/*
+ * DAI operations
+ */
+
+static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ int ret = 0;
+
+ if ((da7219->clk_src == clk_id) && (da7219->mclk_rate == freq))
+ return 0;
+
+ if (((freq < 2000000) && (freq != 32768)) || (freq > 54000000)) {
+ dev_err(codec_dai->dev, "Unsupported MCLK value %d\n",
+ freq);
+ return -EINVAL;
+ }
+
+ switch (clk_id) {
+ case DA7219_CLKSRC_MCLK_SQR:
+ snd_soc_update_bits(codec, DA7219_PLL_CTRL,
+ DA7219_PLL_MCLK_SQR_EN_MASK,
+ DA7219_PLL_MCLK_SQR_EN_MASK);
+ break;
+ case DA7219_CLKSRC_MCLK:
+ snd_soc_update_bits(codec, DA7219_PLL_CTRL,
+ DA7219_PLL_MCLK_SQR_EN_MASK, 0);
+ break;
+ default:
+ dev_err(codec_dai->dev, "Unknown clock source %d\n", clk_id);
+ return -EINVAL;
+ }
+
+ da7219->clk_src = clk_id;
+
+ if (da7219->mclk) {
+ freq = clk_round_rate(da7219->mclk, freq);
+ ret = clk_set_rate(da7219->mclk, freq);
+ if (ret) {
+ dev_err(codec_dai->dev, "Failed to set clock rate %d\n",
+ freq);
+ return ret;
+ }
+ }
+
+ da7219->mclk_rate = freq;
+
+ return 0;
+}
+
+static int da7219_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id,
+ int source, unsigned int fref, unsigned int fout)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+ u8 pll_ctrl, indiv_bits, indiv;
+ u8 pll_frac_top, pll_frac_bot, pll_integer;
+ u32 freq_ref;
+ u64 frac_div;
+
+ /* Verify 32KHz, 2MHz - 54MHz MCLK provided, and set input divider */
+ if (da7219->mclk_rate == 32768) {
+ indiv_bits = DA7219_PLL_INDIV_2_5_MHZ;
+ indiv = DA7219_PLL_INDIV_2_5_MHZ_VAL;
+ } else if (da7219->mclk_rate < 2000000) {
+ dev_err(codec->dev, "PLL input clock %d below valid range\n",
+ da7219->mclk_rate);
+ return -EINVAL;
+ } else if (da7219->mclk_rate <= 5000000) {
+ indiv_bits = DA7219_PLL_INDIV_2_5_MHZ;
+ indiv = DA7219_PLL_INDIV_2_5_MHZ_VAL;
+ } else if (da7219->mclk_rate <= 10000000) {
+ indiv_bits = DA7219_PLL_INDIV_5_10_MHZ;
+ indiv = DA7219_PLL_INDIV_5_10_MHZ_VAL;
+ } else if (da7219->mclk_rate <= 20000000) {
+ indiv_bits = DA7219_PLL_INDIV_10_20_MHZ;
+ indiv = DA7219_PLL_INDIV_10_20_MHZ_VAL;
+ } else if (da7219->mclk_rate <= 40000000) {
+ indiv_bits = DA7219_PLL_INDIV_20_40_MHZ;
+ indiv = DA7219_PLL_INDIV_20_40_MHZ_VAL;
+ } else if (da7219->mclk_rate <= 54000000) {
+ indiv_bits = DA7219_PLL_INDIV_40_54_MHZ;
+ indiv = DA7219_PLL_INDIV_40_54_MHZ_VAL;
+ } else {
+ dev_err(codec->dev, "PLL input clock %d above valid range\n",
+ da7219->mclk_rate);
+ return -EINVAL;
+ }
+ freq_ref = (da7219->mclk_rate / indiv);
+ pll_ctrl = indiv_bits;
+
+ /* Configure PLL */
+ switch (source) {
+ case DA7219_SYSCLK_MCLK:
+ pll_ctrl |= DA7219_PLL_MODE_BYPASS;
+ snd_soc_update_bits(codec, DA7219_PLL_CTRL,
+ DA7219_PLL_INDIV_MASK |
+ DA7219_PLL_MODE_MASK, pll_ctrl);
+ return 0;
+ case DA7219_SYSCLK_PLL:
+ pll_ctrl |= DA7219_PLL_MODE_NORMAL;
+ break;
+ case DA7219_SYSCLK_PLL_SRM:
+ pll_ctrl |= DA7219_PLL_MODE_SRM;
+ break;
+ case DA7219_SYSCLK_PLL_32KHZ:
+ pll_ctrl |= DA7219_PLL_MODE_32KHZ;
+ break;
+ default:
+ dev_err(codec->dev, "Invalid PLL config\n");
+ return -EINVAL;
+ }
+
+ /* Calculate dividers for PLL */
+ pll_integer = fout / freq_ref;
+ frac_div = (u64)(fout % freq_ref) * 8192ULL;
+ do_div(frac_div, freq_ref);
+ pll_frac_top = (frac_div >> DA7219_BYTE_SHIFT) & DA7219_BYTE_MASK;
+ pll_frac_bot = (frac_div) & DA7219_BYTE_MASK;
+
+ /* Write PLL config & dividers */
+ snd_soc_write(codec, DA7219_PLL_FRAC_TOP, pll_frac_top);
+ snd_soc_write(codec, DA7219_PLL_FRAC_BOT, pll_frac_bot);
+ snd_soc_write(codec, DA7219_PLL_INTEGER, pll_integer);
+ snd_soc_update_bits(codec, DA7219_PLL_CTRL,
+ DA7219_PLL_INDIV_MASK | DA7219_PLL_MODE_MASK,
+ pll_ctrl);
+
+ return 0;
+}
+
+static int da7219_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt)
+{
+ struct snd_soc_codec *codec = codec_dai->codec;
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ u8 dai_clk_mode = 0, dai_ctrl = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ da7219->master = true;
+ break;
+ case SND_SOC_DAIFMT_CBS_CFS:
+ da7219->master = false;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+ case SND_SOC_DAIFMT_NB_NF:
+ break;
+ case SND_SOC_DAIFMT_NB_IF:
+ dai_clk_mode |= DA7219_DAI_WCLK_POL_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_NF:
+ dai_clk_mode |= DA7219_DAI_CLK_POL_INV;
+ break;
+ case SND_SOC_DAIFMT_IB_IF:
+ dai_clk_mode |= DA7219_DAI_WCLK_POL_INV |
+ DA7219_DAI_CLK_POL_INV;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ dai_ctrl |= DA7219_DAI_FORMAT_I2S;
+ break;
+ case SND_SOC_DAIFMT_LEFT_J:
+ dai_ctrl |= DA7219_DAI_FORMAT_LEFT_J;
+ break;
+ case SND_SOC_DAIFMT_RIGHT_J:
+ dai_ctrl |= DA7219_DAI_FORMAT_RIGHT_J;
+ break;
+ case SND_SOC_DAIFMT_DSP_B:
+ dai_ctrl |= DA7219_DAI_FORMAT_DSP;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ /* By default 64 BCLKs per WCLK is supported */
+ dai_clk_mode |= DA7219_DAI_BCLKS_PER_WCLK_64;
+
+ snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE,
+ DA7219_DAI_BCLKS_PER_WCLK_MASK |
+ DA7219_DAI_CLK_POL_MASK | DA7219_DAI_WCLK_POL_MASK,
+ dai_clk_mode);
+ snd_soc_update_bits(codec, DA7219_DAI_CTRL, DA7219_DAI_FORMAT_MASK,
+ dai_ctrl);
+
+ return 0;
+}
+
+static int da7219_set_dai_tdm_slot(struct snd_soc_dai *dai,
+ unsigned int tx_mask, unsigned int rx_mask,
+ int slots, int slot_width)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ u8 dai_bclks_per_wclk;
+ u16 offset;
+ u32 frame_size;
+
+ /* No channels enabled so disable TDM, revert to 64-bit frames */
+ if (!tx_mask) {
+ snd_soc_update_bits(codec, DA7219_DAI_TDM_CTRL,
+ DA7219_DAI_TDM_CH_EN_MASK |
+ DA7219_DAI_TDM_MODE_EN_MASK, 0);
+ snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE,
+ DA7219_DAI_BCLKS_PER_WCLK_MASK,
+ DA7219_DAI_BCLKS_PER_WCLK_64);
+ return 0;
+ }
+
+ /* Check we have valid slots */
+ if (fls(tx_mask) > DA7219_DAI_TDM_MAX_SLOTS) {
+ dev_err(codec->dev, "Invalid number of slots, max = %d\n",
+ DA7219_DAI_TDM_MAX_SLOTS);
+ return -EINVAL;
+ }
+
+ /* Check we have a valid offset given */
+ if (rx_mask > DA7219_DAI_OFFSET_MAX) {
+ dev_err(codec->dev, "Invalid slot offset, max = %d\n",
+ DA7219_DAI_OFFSET_MAX);
+ return -EINVAL;
+ }
+
+ /* Calculate & validate frame size based on slot info provided. */
+ frame_size = slots * slot_width;
+ switch (frame_size) {
+ case 32:
+ dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_32;
+ break;
+ case 64:
+ dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_64;
+ break;
+ case 128:
+ dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_128;
+ break;
+ case 256:
+ dai_bclks_per_wclk = DA7219_DAI_BCLKS_PER_WCLK_256;
+ break;
+ default:
+ dev_err(codec->dev, "Invalid frame size %d\n", frame_size);
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, DA7219_DAI_CLK_MODE,
+ DA7219_DAI_BCLKS_PER_WCLK_MASK,
+ dai_bclks_per_wclk);
+
+ offset = cpu_to_le16(rx_mask);
+ regmap_bulk_write(da7219->regmap, DA7219_DAI_OFFSET_LOWER,
+ &offset, sizeof(offset));
+
+ snd_soc_update_bits(codec, DA7219_DAI_TDM_CTRL,
+ DA7219_DAI_TDM_CH_EN_MASK |
+ DA7219_DAI_TDM_MODE_EN_MASK,
+ (tx_mask << DA7219_DAI_TDM_CH_EN_SHIFT) |
+ DA7219_DAI_TDM_MODE_EN_MASK);
+
+ return 0;
+}
+
+static int da7219_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 dai_ctrl = 0, fs;
+ unsigned int channels;
+
+ switch (params_width(params)) {
+ case 16:
+ dai_ctrl |= DA7219_DAI_WORD_LENGTH_S16_LE;
+ break;
+ case 20:
+ dai_ctrl |= DA7219_DAI_WORD_LENGTH_S20_LE;
+ break;
+ case 24:
+ dai_ctrl |= DA7219_DAI_WORD_LENGTH_S24_LE;
+ break;
+ case 32:
+ dai_ctrl |= DA7219_DAI_WORD_LENGTH_S32_LE;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ channels = params_channels(params);
+ if ((channels < 1) | (channels > DA7219_DAI_CH_NUM_MAX)) {
+ dev_err(codec->dev,
+ "Invalid number of channels, only 1 to %d supported\n",
+ DA7219_DAI_CH_NUM_MAX);
+ return -EINVAL;
+ }
+ dai_ctrl |= channels << DA7219_DAI_CH_NUM_SHIFT;
+
+ switch (params_rate(params)) {
+ case 8000:
+ fs = DA7219_SR_8000;
+ break;
+ case 11025:
+ fs = DA7219_SR_11025;
+ break;
+ case 12000:
+ fs = DA7219_SR_12000;
+ break;
+ case 16000:
+ fs = DA7219_SR_16000;
+ break;
+ case 22050:
+ fs = DA7219_SR_22050;
+ break;
+ case 24000:
+ fs = DA7219_SR_24000;
+ break;
+ case 32000:
+ fs = DA7219_SR_32000;
+ break;
+ case 44100:
+ fs = DA7219_SR_44100;
+ break;
+ case 48000:
+ fs = DA7219_SR_48000;
+ break;
+ case 88200:
+ fs = DA7219_SR_88200;
+ break;
+ case 96000:
+ fs = DA7219_SR_96000;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ snd_soc_update_bits(codec, DA7219_DAI_CTRL,
+ DA7219_DAI_WORD_LENGTH_MASK |
+ DA7219_DAI_CH_NUM_MASK,
+ dai_ctrl);
+ snd_soc_write(codec, DA7219_SR, fs);
+
+ return 0;
+}
+
+static const struct snd_soc_dai_ops da7219_dai_ops = {
+ .hw_params = da7219_hw_params,
+ .set_sysclk = da7219_set_dai_sysclk,
+ .set_pll = da7219_set_dai_pll,
+ .set_fmt = da7219_set_dai_fmt,
+ .set_tdm_slot = da7219_set_dai_tdm_slot,
+};
+
+#define DA7219_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
+ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
+
+static struct snd_soc_dai_driver da7219_dai = {
+ .name = "da7219-hifi",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = DA7219_DAI_CH_NUM_MAX,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = DA7219_FORMATS,
+ },
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = DA7219_DAI_CH_NUM_MAX,
+ .rates = SNDRV_PCM_RATE_8000_96000,
+ .formats = DA7219_FORMATS,
+ },
+ .ops = &da7219_dai_ops,
+ .symmetric_rates = 1,
+ .symmetric_channels = 1,
+ .symmetric_samplebits = 1,
+};
+
+
+/*
+ * DT
+ */
+
+static const struct of_device_id da7219_of_match[] = {
+ { .compatible = "dlg,da7219", },
+ { }
+};
+MODULE_DEVICE_TABLE(of, da7219_of_match);
+
+static enum da7219_ldo_lvl_sel da7219_of_ldo_lvl(struct snd_soc_codec *codec,
+ u32 val)
+{
+ switch (val) {
+ case 1050:
+ return DA7219_LDO_LVL_SEL_1_05V;
+ case 1100:
+ return DA7219_LDO_LVL_SEL_1_10V;
+ case 1200:
+ return DA7219_LDO_LVL_SEL_1_20V;
+ case 1400:
+ return DA7219_LDO_LVL_SEL_1_40V;
+ default:
+ dev_warn(codec->dev, "Invalid LDO level");
+ return DA7219_LDO_LVL_SEL_1_05V;
+ }
+}
+
+static enum da7219_micbias_voltage
+ da7219_of_micbias_lvl(struct snd_soc_codec *codec, u32 val)
+{
+ switch (val) {
+ case 1800:
+ return DA7219_MICBIAS_1_8V;
+ case 2000:
+ return DA7219_MICBIAS_2_0V;
+ case 2200:
+ return DA7219_MICBIAS_2_2V;
+ case 2400:
+ return DA7219_MICBIAS_2_4V;
+ case 2600:
+ return DA7219_MICBIAS_2_6V;
+ default:
+ dev_warn(codec->dev, "Invalid micbias level");
+ return DA7219_MICBIAS_2_2V;
+ }
+}
+
+static enum da7219_mic_amp_in_sel
+ da7219_of_mic_amp_in_sel(struct snd_soc_codec *codec, const char *str)
+{
+ if (!strcmp(str, "diff")) {
+ return DA7219_MIC_AMP_IN_SEL_DIFF;
+ } else if (!strcmp(str, "se_p")) {
+ return DA7219_MIC_AMP_IN_SEL_SE_P;
+ } else if (!strcmp(str, "se_n")) {
+ return DA7219_MIC_AMP_IN_SEL_SE_N;
+ } else {
+ dev_warn(codec->dev, "Invalid mic input type selection");
+ return DA7219_MIC_AMP_IN_SEL_DIFF;
+ }
+}
+
+static struct da7219_pdata *da7219_of_to_pdata(struct snd_soc_codec *codec)
+{
+ struct device_node *np = codec->dev->of_node;
+ struct da7219_pdata *pdata;
+ const char *of_str;
+ u32 of_val32;
+
+ pdata = devm_kzalloc(codec->dev, sizeof(*pdata), GFP_KERNEL);
+ if (!pdata)
+ return NULL;
+
+ if (of_property_read_u32(np, "dlg,ldo-lvl", &of_val32) >= 0)
+ pdata->ldo_lvl_sel = da7219_of_ldo_lvl(codec, of_val32);
+
+ if (of_property_read_u32(np, "dlg,micbias-lvl", &of_val32) >= 0)
+ pdata->micbias_lvl = da7219_of_micbias_lvl(codec, of_val32);
+ else
+ pdata->micbias_lvl = DA7219_MICBIAS_2_2V;
+
+ if (!of_property_read_string(np, "dlg,mic-amp-in-sel", &of_str))
+ pdata->mic_amp_in_sel = da7219_of_mic_amp_in_sel(codec, of_str);
+ else
+ pdata->mic_amp_in_sel = DA7219_MIC_AMP_IN_SEL_DIFF;
+
+ return pdata;
+}
+
+
+/*
+ * Codec driver functions
+ */
+
+static int da7219_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ switch (level) {
+ case SND_SOC_BIAS_ON:
+ case SND_SOC_BIAS_PREPARE:
+ break;
+ case SND_SOC_BIAS_STANDBY:
+ if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) {
+ /* MCLK */
+ if (da7219->mclk) {
+ ret = clk_prepare_enable(da7219->mclk);
+ if (ret) {
+ dev_err(codec->dev,
+ "Failed to enable mclk\n");
+ return ret;
+ }
+ }
+
+ /* Master bias */
+ snd_soc_update_bits(codec, DA7219_REFERENCES,
+ DA7219_BIAS_EN_MASK,
+ DA7219_BIAS_EN_MASK);
+
+ /* Enable Internal Digital LDO */
+ snd_soc_update_bits(codec, DA7219_LDO_CTRL,
+ DA7219_LDO_EN_MASK,
+ DA7219_LDO_EN_MASK);
+ }
+ break;
+ case SND_SOC_BIAS_OFF:
+ /* Only disable if jack detection not active */
+ if (!da7219->aad->jack) {
+ /* Bypass Internal Digital LDO */
+ snd_soc_update_bits(codec, DA7219_LDO_CTRL,
+ DA7219_LDO_EN_MASK, 0);
+
+ /* Master bias */
+ snd_soc_update_bits(codec, DA7219_REFERENCES,
+ DA7219_BIAS_EN_MASK, 0);
+ }
+
+ /* MCLK */
+ if (da7219->mclk)
+ clk_disable_unprepare(da7219->mclk);
+ break;
+ }
+
+ return 0;
+}
+
+static const char *da7219_supply_names[DA7219_NUM_SUPPLIES] = {
+ [DA7219_SUPPLY_VDD] = "VDD",
+ [DA7219_SUPPLY_VDDMIC] = "VDDMIC",
+ [DA7219_SUPPLY_VDDIO] = "VDDIO",
+};
+
+static int da7219_handle_supplies(struct snd_soc_codec *codec)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ struct regulator *vddio;
+ u8 io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_2_5V_3_6V;
+ int i, ret;
+
+ /* Get required supplies */
+ for (i = 0; i < DA7219_NUM_SUPPLIES; ++i)
+ da7219->supplies[i].supply = da7219_supply_names[i];
+
+ ret = devm_regulator_bulk_get(codec->dev, DA7219_NUM_SUPPLIES,
+ da7219->supplies);
+ if (ret) {
+ dev_err(codec->dev, "Failed to get supplies");
+ return ret;
+ }
+
+ /* Determine VDDIO voltage provided */
+ vddio = da7219->supplies[DA7219_SUPPLY_VDDIO].consumer;
+ ret = regulator_get_voltage(vddio);
+ if (ret < 1200000)
+ dev_warn(codec->dev, "Invalid VDDIO voltage\n");
+ else if (ret < 2800000)
+ io_voltage_lvl = DA7219_IO_VOLTAGE_LEVEL_1_2V_2_8V;
+
+ /* Enable main supplies */
+ ret = regulator_bulk_enable(DA7219_NUM_SUPPLIES, da7219->supplies);
+ if (ret) {
+ dev_err(codec->dev, "Failed to enable supplies");
+ return ret;
+ }
+
+ /* Ensure device in active mode */
+ snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, DA7219_SYSTEM_ACTIVE_MASK);
+
+ /* Update IO voltage level range */
+ snd_soc_write(codec, DA7219_IO_CTRL, io_voltage_lvl);
+
+ return 0;
+}
+
+static void da7219_handle_pdata(struct snd_soc_codec *codec)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ struct da7219_pdata *pdata = da7219->pdata;
+
+ if (pdata) {
+ u8 micbias_lvl = 0;
+
+ /* Internal LDO */
+ switch (pdata->ldo_lvl_sel) {
+ case DA7219_LDO_LVL_SEL_1_05V:
+ case DA7219_LDO_LVL_SEL_1_10V:
+ case DA7219_LDO_LVL_SEL_1_20V:
+ case DA7219_LDO_LVL_SEL_1_40V:
+ snd_soc_update_bits(codec, DA7219_LDO_CTRL,
+ DA7219_LDO_LEVEL_SELECT_MASK,
+ (pdata->ldo_lvl_sel <<
+ DA7219_LDO_LEVEL_SELECT_SHIFT));
+ break;
+ }
+
+ /* Mic Bias voltages */
+ switch (pdata->micbias_lvl) {
+ case DA7219_MICBIAS_1_8V:
+ case DA7219_MICBIAS_2_0V:
+ case DA7219_MICBIAS_2_2V:
+ case DA7219_MICBIAS_2_4V:
+ case DA7219_MICBIAS_2_6V:
+ micbias_lvl |= (pdata->micbias_lvl <<
+ DA7219_MICBIAS1_LEVEL_SHIFT);
+ break;
+ }
+
+ snd_soc_write(codec, DA7219_MICBIAS_CTRL, micbias_lvl);
+
+ /* Mic */
+ switch (pdata->mic_amp_in_sel) {
+ case DA7219_MIC_AMP_IN_SEL_DIFF:
+ case DA7219_MIC_AMP_IN_SEL_SE_P:
+ case DA7219_MIC_AMP_IN_SEL_SE_N:
+ snd_soc_write(codec, DA7219_MIC_1_SELECT,
+ pdata->mic_amp_in_sel);
+ break;
+ }
+ }
+}
+
+static int da7219_probe(struct snd_soc_codec *codec)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+ int ret;
+
+ mutex_init(&da7219->lock);
+
+ /* Regulator configuration */
+ ret = da7219_handle_supplies(codec);
+ if (ret)
+ return ret;
+
+ /* Handle DT/Platform data */
+ if (codec->dev->of_node)
+ da7219->pdata = da7219_of_to_pdata(codec);
+ else
+ da7219->pdata = dev_get_platdata(codec->dev);
+
+ da7219_handle_pdata(codec);
+
+ /* Check if MCLK provided */
+ da7219->mclk = devm_clk_get(codec->dev, "mclk");
+ if (IS_ERR(da7219->mclk)) {
+ if (PTR_ERR(da7219->mclk) != -ENOENT)
+ return PTR_ERR(da7219->mclk);
+ else
+ da7219->mclk = NULL;
+ }
+
+ /* Default PC counter to free-running */
+ snd_soc_update_bits(codec, DA7219_PC_COUNT, DA7219_PC_FREERUN_MASK,
+ DA7219_PC_FREERUN_MASK);
+
+ /* Default gain ramping */
+ snd_soc_update_bits(codec, DA7219_MIXIN_L_CTRL,
+ DA7219_MIXIN_L_AMP_RAMP_EN_MASK,
+ DA7219_MIXIN_L_AMP_RAMP_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_ADC_L_CTRL, DA7219_ADC_L_RAMP_EN_MASK,
+ DA7219_ADC_L_RAMP_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_DAC_L_CTRL, DA7219_DAC_L_RAMP_EN_MASK,
+ DA7219_DAC_L_RAMP_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_DAC_R_CTRL, DA7219_DAC_R_RAMP_EN_MASK,
+ DA7219_DAC_R_RAMP_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_HP_L_CTRL,
+ DA7219_HP_L_AMP_RAMP_EN_MASK,
+ DA7219_HP_L_AMP_RAMP_EN_MASK);
+ snd_soc_update_bits(codec, DA7219_HP_R_CTRL,
+ DA7219_HP_R_AMP_RAMP_EN_MASK,
+ DA7219_HP_R_AMP_RAMP_EN_MASK);
+
+ /* Default infinite tone gen, start/stop by Kcontrol */
+ snd_soc_write(codec, DA7219_TONE_GEN_CYCLES, DA7219_BEEP_CYCLES_MASK);
+
+ /* Initialise AAD block */
+ return da7219_aad_init(codec);
+}
+
+static int da7219_remove(struct snd_soc_codec *codec)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+ da7219_aad_exit(codec);
+
+ /* Supplies */
+ return regulator_bulk_disable(DA7219_NUM_SUPPLIES, da7219->supplies);
+}
+
+#ifdef CONFIG_PM
+static int da7219_suspend(struct snd_soc_codec *codec)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+ snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_OFF);
+
+ /* Put device into standby mode if jack detection disabled */
+ if (!da7219->aad->jack)
+ snd_soc_write(codec, DA7219_SYSTEM_ACTIVE, 0);
+
+ return 0;
+}
+
+static int da7219_resume(struct snd_soc_codec *codec)
+{
+ struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec);
+
+ /* Put device into active mode if previously pushed to standby */
+ if (!da7219->aad->jack)
+ snd_soc_write(codec, DA7219_SYSTEM_ACTIVE,
+ DA7219_SYSTEM_ACTIVE_MASK);
+
+ snd_soc_codec_force_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ return 0;
+}
+#else
+#define da7219_suspend NULL
+#define da7219_resume NULL
+#endif
+
+static struct snd_soc_codec_driver soc_codec_dev_da7219 = {
+ .probe = da7219_probe,
+ .remove = da7219_remove,
+ .suspend = da7219_suspend,
+ .resume = da7219_resume,
+ .set_bias_level = da7219_set_bias_level,
+
+ .controls = da7219_snd_controls,
+ .num_controls = ARRAY_SIZE(da7219_snd_controls),
+
+ .dapm_widgets = da7219_dapm_widgets,
+ .num_dapm_widgets = ARRAY_SIZE(da7219_dapm_widgets),
+ .dapm_routes = da7219_audio_map,
+ .num_dapm_routes = ARRAY_SIZE(da7219_audio_map),
+};
+
+
+/*
+ * Regmap configs
+ */
+
+static struct reg_default da7219_reg_defaults[] = {
+ { DA7219_MIC_1_SELECT, 0x00 },
+ { DA7219_CIF_TIMEOUT_CTRL, 0x01 },
+ { DA7219_SR_24_48, 0x00 },
+ { DA7219_SR, 0x0A },
+ { DA7219_CIF_I2C_ADDR_CFG, 0x02 },
+ { DA7219_PLL_CTRL, 0x10 },
+ { DA7219_PLL_FRAC_TOP, 0x00 },
+ { DA7219_PLL_FRAC_BOT, 0x00 },
+ { DA7219_PLL_INTEGER, 0x20 },
+ { DA7219_DIG_ROUTING_DAI, 0x10 },
+ { DA7219_DAI_CLK_MODE, 0x01 },
+ { DA7219_DAI_CTRL, 0x28 },
+ { DA7219_DAI_TDM_CTRL, 0x40 },
+ { DA7219_DIG_ROUTING_DAC, 0x32 },
+ { DA7219_DAI_OFFSET_LOWER, 0x00 },
+ { DA7219_DAI_OFFSET_UPPER, 0x00 },
+ { DA7219_REFERENCES, 0x00 },
+ { DA7219_MIXIN_L_SELECT, 0x00 },
+ { DA7219_MIXIN_L_GAIN, 0x03 },
+ { DA7219_ADC_L_GAIN, 0x6F },
+ { DA7219_ADC_FILTERS1, 0x80 },
+ { DA7219_MIC_1_GAIN, 0x01 },
+ { DA7219_SIDETONE_CTRL, 0x40 },
+ { DA7219_SIDETONE_GAIN, 0x0E },
+ { DA7219_DROUTING_ST_OUTFILT_1L, 0x01 },
+ { DA7219_DROUTING_ST_OUTFILT_1R, 0x02 },
+ { DA7219_DAC_FILTERS5, 0x00 },
+ { DA7219_DAC_FILTERS2, 0x88 },
+ { DA7219_DAC_FILTERS3, 0x88 },
+ { DA7219_DAC_FILTERS4, 0x08 },
+ { DA7219_DAC_FILTERS1, 0x80 },
+ { DA7219_DAC_L_GAIN, 0x6F },
+ { DA7219_DAC_R_GAIN, 0x6F },
+ { DA7219_CP_CTRL, 0x20 },
+ { DA7219_HP_L_GAIN, 0x39 },
+ { DA7219_HP_R_GAIN, 0x39 },
+ { DA7219_MIXOUT_L_SELECT, 0x00 },
+ { DA7219_MIXOUT_R_SELECT, 0x00 },
+ { DA7219_MICBIAS_CTRL, 0x03 },
+ { DA7219_MIC_1_CTRL, 0x40 },
+ { DA7219_MIXIN_L_CTRL, 0x40 },
+ { DA7219_ADC_L_CTRL, 0x40 },
+ { DA7219_DAC_L_CTRL, 0x40 },
+ { DA7219_DAC_R_CTRL, 0x40 },
+ { DA7219_HP_L_CTRL, 0x40 },
+ { DA7219_HP_R_CTRL, 0x40 },
+ { DA7219_MIXOUT_L_CTRL, 0x10 },
+ { DA7219_MIXOUT_R_CTRL, 0x10 },
+ { DA7219_CHIP_ID1, 0x23 },
+ { DA7219_CHIP_ID2, 0x93 },
+ { DA7219_CHIP_REVISION, 0x00 },
+ { DA7219_LDO_CTRL, 0x00 },
+ { DA7219_IO_CTRL, 0x00 },
+ { DA7219_GAIN_RAMP_CTRL, 0x00 },
+ { DA7219_PC_COUNT, 0x02 },
+ { DA7219_CP_VOL_THRESHOLD1, 0x0E },
+ { DA7219_DIG_CTRL, 0x00 },
+ { DA7219_ALC_CTRL2, 0x00 },
+ { DA7219_ALC_CTRL3, 0x00 },
+ { DA7219_ALC_NOISE, 0x3F },
+ { DA7219_ALC_TARGET_MIN, 0x3F },
+ { DA7219_ALC_TARGET_MAX, 0x00 },
+ { DA7219_ALC_GAIN_LIMITS, 0xFF },
+ { DA7219_ALC_ANA_GAIN_LIMITS, 0x71 },
+ { DA7219_ALC_ANTICLIP_CTRL, 0x00 },
+ { DA7219_ALC_ANTICLIP_LEVEL, 0x00 },
+ { DA7219_DAC_NG_SETUP_TIME, 0x00 },
+ { DA7219_DAC_NG_OFF_THRESH, 0x00 },
+ { DA7219_DAC_NG_ON_THRESH, 0x00 },
+ { DA7219_DAC_NG_CTRL, 0x00 },
+ { DA7219_TONE_GEN_CFG1, 0x00 },
+ { DA7219_TONE_GEN_CFG2, 0x00 },
+ { DA7219_TONE_GEN_CYCLES, 0x00 },
+ { DA7219_TONE_GEN_FREQ1_L, 0x55 },
+ { DA7219_TONE_GEN_FREQ1_U, 0x15 },
+ { DA7219_TONE_GEN_FREQ2_L, 0x00 },
+ { DA7219_TONE_GEN_FREQ2_U, 0x40 },
+ { DA7219_TONE_GEN_ON_PER, 0x02 },
+ { DA7219_TONE_GEN_OFF_PER, 0x01 },
+ { DA7219_ACCDET_IRQ_MASK_A, 0x00 },
+ { DA7219_ACCDET_IRQ_MASK_B, 0x00 },
+ { DA7219_ACCDET_CONFIG_1, 0xD6 },
+ { DA7219_ACCDET_CONFIG_2, 0x34 },
+ { DA7219_ACCDET_CONFIG_3, 0x0A },
+ { DA7219_ACCDET_CONFIG_4, 0x16 },
+ { DA7219_ACCDET_CONFIG_5, 0x21 },
+ { DA7219_ACCDET_CONFIG_6, 0x3E },
+ { DA7219_ACCDET_CONFIG_7, 0x01 },
+ { DA7219_SYSTEM_ACTIVE, 0x00 },
+};
+
+static bool da7219_volatile_register(struct device *dev, unsigned int reg)
+{
+ switch (reg) {
+ case DA7219_MIC_1_GAIN_STATUS:
+ case DA7219_MIXIN_L_GAIN_STATUS:
+ case DA7219_ADC_L_GAIN_STATUS:
+ case DA7219_DAC_L_GAIN_STATUS:
+ case DA7219_DAC_R_GAIN_STATUS:
+ case DA7219_HP_L_GAIN_STATUS:
+ case DA7219_HP_R_GAIN_STATUS:
+ case DA7219_CIF_CTRL:
+ case DA7219_PLL_SRM_STS:
+ case DA7219_ALC_CTRL1:
+ case DA7219_SYSTEM_MODES_INPUT:
+ case DA7219_SYSTEM_MODES_OUTPUT:
+ case DA7219_ALC_OFFSET_AUTO_M_L:
+ case DA7219_ALC_OFFSET_AUTO_U_L:
+ case DA7219_TONE_GEN_CFG1:
+ case DA7219_ACCDET_STATUS_A:
+ case DA7219_ACCDET_STATUS_B:
+ case DA7219_ACCDET_IRQ_EVENT_A:
+ case DA7219_ACCDET_IRQ_EVENT_B:
+ case DA7219_ACCDET_CONFIG_8:
+ case DA7219_SYSTEM_STATUS:
+ return 1;
+ default:
+ return 0;
+ }
+}
+
+static const struct regmap_config da7219_regmap_config = {
+ .reg_bits = 8,
+ .val_bits = 8,
+
+ .max_register = DA7219_SYSTEM_ACTIVE,
+ .reg_defaults = da7219_reg_defaults,
+ .num_reg_defaults = ARRAY_SIZE(da7219_reg_defaults),
+ .volatile_reg = da7219_volatile_register,
+ .cache_type = REGCACHE_RBTREE,
+};
+
+
+/*
+ * I2C layer
+ */
+
+static int da7219_i2c_probe(struct i2c_client *i2c,
+ const struct i2c_device_id *id)
+{
+ struct da7219_priv *da7219;
+ int ret;
+
+ da7219 = devm_kzalloc(&i2c->dev, sizeof(struct da7219_priv),
+ GFP_KERNEL);
+ if (!da7219)
+ return -ENOMEM;
+
+ i2c_set_clientdata(i2c, da7219);
+
+ da7219->regmap = devm_regmap_init_i2c(i2c, &da7219_regmap_config);
+ if (IS_ERR(da7219->regmap)) {
+ ret = PTR_ERR(da7219->regmap);
+ dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret);
+ return ret;
+ }
+
+ ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da7219,
+ &da7219_dai, 1);
+ if (ret < 0) {
+ dev_err(&i2c->dev, "Failed to register da7219 codec: %d\n",
+ ret);
+ }
+ return ret;
+}
+
+static int da7219_i2c_remove(struct i2c_client *client)
+{
+ snd_soc_unregister_codec(&client->dev);
+ return 0;
+}
+
+static const struct i2c_device_id da7219_i2c_id[] = {
+ { "da7219", },
+ { }
+};
+MODULE_DEVICE_TABLE(i2c, da7219_i2c_id);
+
+static struct i2c_driver da7219_i2c_driver = {
+ .driver = {
+ .name = "da7219",
+ .of_match_table = of_match_ptr(da7219_of_match),
+ },
+ .probe = da7219_i2c_probe,
+ .remove = da7219_i2c_remove,
+ .id_table = da7219_i2c_id,
+};
+
+module_i2c_driver(da7219_i2c_driver);
+
+MODULE_DESCRIPTION("ASoC DA7219 Codec Driver");
+MODULE_AUTHOR("Adam Thomson <Adam.Thomson.Opensource@diasemi.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h
new file mode 100644
index 000000000000..b514268c6c56
--- /dev/null
+++ b/sound/soc/codecs/da7219.h
@@ -0,0 +1,820 @@
+/*
+ * da7219.h - DA7219 ALSA SoC Codec Driver
+ *
+ * Copyright (c) 2015 Dialog Semiconductor
+ *
+ * Author: Adam Thomson <Adam.Thomson.Opensource@diasemi.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#ifndef __DA7219_H
+#define __DA7219_H
+
+#include <linux/regmap.h>
+#include <linux/regulator/consumer.h>
+#include <sound/da7219.h>
+
+/*
+ * Registers
+ */
+
+#define DA7219_MIC_1_GAIN_STATUS 0x6
+#define DA7219_MIXIN_L_GAIN_STATUS 0x8
+#define DA7219_ADC_L_GAIN_STATUS 0xA
+#define DA7219_DAC_L_GAIN_STATUS 0xC
+#define DA7219_DAC_R_GAIN_STATUS 0xD
+#define DA7219_HP_L_GAIN_STATUS 0xE
+#define DA7219_HP_R_GAIN_STATUS 0xF
+#define DA7219_MIC_1_SELECT 0x10
+#define DA7219_CIF_TIMEOUT_CTRL 0x12
+#define DA7219_CIF_CTRL 0x13
+#define DA7219_SR_24_48 0x16
+#define DA7219_SR 0x17
+#define DA7219_CIF_I2C_ADDR_CFG 0x1B
+#define DA7219_PLL_CTRL 0x20
+#define DA7219_PLL_FRAC_TOP 0x22
+#define DA7219_PLL_FRAC_BOT 0x23
+#define DA7219_PLL_INTEGER 0x24
+#define DA7219_PLL_SRM_STS 0x25
+#define DA7219_DIG_ROUTING_DAI 0x2A
+#define DA7219_DAI_CLK_MODE 0x2B
+#define DA7219_DAI_CTRL 0x2C
+#define DA7219_DAI_TDM_CTRL 0x2D
+#define DA7219_DIG_ROUTING_DAC 0x2E
+#define DA7219_ALC_CTRL1 0x2F
+#define DA7219_DAI_OFFSET_LOWER 0x30
+#define DA7219_DAI_OFFSET_UPPER 0x31
+#define DA7219_REFERENCES 0x32
+#define DA7219_MIXIN_L_SELECT 0x33
+#define DA7219_MIXIN_L_GAIN 0x34
+#define DA7219_ADC_L_GAIN 0x36
+#define DA7219_ADC_FILTERS1 0x38
+#define DA7219_MIC_1_GAIN 0x39
+#define DA7219_SIDETONE_CTRL 0x3A
+#define DA7219_SIDETONE_GAIN 0x3B
+#define DA7219_DROUTING_ST_OUTFILT_1L 0x3C
+#define DA7219_DROUTING_ST_OUTFILT_1R 0x3D
+#define DA7219_DAC_FILTERS5 0x40
+#define DA7219_DAC_FILTERS2 0x41
+#define DA7219_DAC_FILTERS3 0x42
+#define DA7219_DAC_FILTERS4 0x43
+#define DA7219_DAC_FILTERS1 0x44
+#define DA7219_DAC_L_GAIN 0x45
+#define DA7219_DAC_R_GAIN 0x46
+#define DA7219_CP_CTRL 0x47
+#define DA7219_HP_L_GAIN 0x48
+#define DA7219_HP_R_GAIN 0x49
+#define DA7219_MIXOUT_L_SELECT 0x4B
+#define DA7219_MIXOUT_R_SELECT 0x4C
+#define DA7219_SYSTEM_MODES_INPUT 0x50
+#define DA7219_SYSTEM_MODES_OUTPUT 0x51
+#define DA7219_MICBIAS_CTRL 0x62
+#define DA7219_MIC_1_CTRL 0x63
+#define DA7219_MIXIN_L_CTRL 0x65
+#define DA7219_ADC_L_CTRL 0x67
+#define DA7219_DAC_L_CTRL 0x69
+#define DA7219_DAC_R_CTRL 0x6A
+#define DA7219_HP_L_CTRL 0x6B
+#define DA7219_HP_R_CTRL 0x6C
+#define DA7219_MIXOUT_L_CTRL 0x6E
+#define DA7219_MIXOUT_R_CTRL 0x6F
+#define DA7219_CHIP_ID1 0x81
+#define DA7219_CHIP_ID2 0x82
+#define DA7219_CHIP_REVISION 0x83
+#define DA7219_LDO_CTRL 0x90
+#define DA7219_IO_CTRL 0x91
+#define DA7219_GAIN_RAMP_CTRL 0x92
+#define DA7219_PC_COUNT 0x94
+#define DA7219_CP_VOL_THRESHOLD1 0x95
+#define DA7219_CP_DELAY 0x96
+#define DA7219_DIG_CTRL 0x99
+#define DA7219_ALC_CTRL2 0x9A
+#define DA7219_ALC_CTRL3 0x9B
+#define DA7219_ALC_NOISE 0x9C
+#define DA7219_ALC_TARGET_MIN 0x9D
+#define DA7219_ALC_TARGET_MAX 0x9E
+#define DA7219_ALC_GAIN_LIMITS 0x9F
+#define DA7219_ALC_ANA_GAIN_LIMITS 0xA0
+#define DA7219_ALC_ANTICLIP_CTRL 0xA1
+#define DA7219_ALC_ANTICLIP_LEVEL 0xA2
+#define DA7219_ALC_OFFSET_AUTO_M_L 0xA3
+#define DA7219_ALC_OFFSET_AUTO_U_L 0xA4
+#define DA7219_DAC_NG_SETUP_TIME 0xAF
+#define DA7219_DAC_NG_OFF_THRESH 0xB0
+#define DA7219_DAC_NG_ON_THRESH 0xB1
+#define DA7219_DAC_NG_CTRL 0xB2
+#define DA7219_TONE_GEN_CFG1 0xB4
+#define DA7219_TONE_GEN_CFG2 0xB5
+#define DA7219_TONE_GEN_CYCLES 0xB6
+#define DA7219_TONE_GEN_FREQ1_L 0xB7
+#define DA7219_TONE_GEN_FREQ1_U 0xB8
+#define DA7219_TONE_GEN_FREQ2_L 0xB9
+#define DA7219_TONE_GEN_FREQ2_U 0xBA
+#define DA7219_TONE_GEN_ON_PER 0xBB
+#define DA7219_TONE_GEN_OFF_PER 0xBC
+#define DA7219_SYSTEM_STATUS 0xE0
+#define DA7219_SYSTEM_ACTIVE 0xFD
+
+
+/*
+ * Bit Fields
+ */
+
+#define DA7219_SWITCH_EN_MAX 0x1
+
+/* DA7219_MIC_1_GAIN_STATUS = 0x6 */
+#define DA7219_MIC_1_AMP_GAIN_STATUS_SHIFT 0
+#define DA7219_MIC_1_AMP_GAIN_STATUS_MASK (0x7 << 0)
+#define DA7219_MIC_1_AMP_GAIN_MAX 0x7
+
+/* DA7219_MIXIN_L_GAIN_STATUS = 0x8 */
+#define DA7219_MIXIN_L_AMP_GAIN_STATUS_SHIFT 0
+#define DA7219_MIXIN_L_AMP_GAIN_STATUS_MASK (0xF << 0)
+
+/* DA7219_ADC_L_GAIN_STATUS = 0xA */
+#define DA7219_ADC_L_DIGITAL_GAIN_STATUS_SHIFT 0
+#define DA7219_ADC_L_DIGITAL_GAIN_STATUS_MASK (0x7F << 0)
+
+/* DA7219_DAC_L_GAIN_STATUS = 0xC */
+#define DA7219_DAC_L_DIGITAL_GAIN_STATUS_SHIFT 0
+#define DA7219_DAC_L_DIGITAL_GAIN_STATUS_MASK (0x7F << 0)
+
+/* DA7219_DAC_R_GAIN_STATUS = 0xD */
+#define DA7219_DAC_R_DIGITAL_GAIN_STATUS_SHIFT 0
+#define DA7219_DAC_R_DIGITAL_GAIN_STATUS_MASK (0x7F << 0)
+
+/* DA7219_HP_L_GAIN_STATUS = 0xE */
+#define DA7219_HP_L_AMP_GAIN_STATUS_SHIFT 0
+#define DA7219_HP_L_AMP_GAIN_STATUS_MASK (0x3F << 0)
+
+/* DA7219_HP_R_GAIN_STATUS = 0xF */
+#define DA7219_HP_R_AMP_GAIN_STATUS_SHIFT 0
+#define DA7219_HP_R_AMP_GAIN_STATUS_MASK (0x3F << 0)
+
+/* DA7219_MIC_1_SELECT = 0x10 */
+#define DA7219_MIC_1_AMP_IN_SEL_SHIFT 0
+#define DA7219_MIC_1_AMP_IN_SEL_MASK (0x3 << 0)
+
+/* DA7219_CIF_TIMEOUT_CTRL = 0x12 */
+#define DA7219_I2C_TIMEOUT_EN_SHIFT 0
+#define DA7219_I2C_TIMEOUT_EN_MASK (0x1 << 0)
+
+/* DA7219_CIF_CTRL = 0x13 */
+#define DA7219_CIF_I2C_WRITE_MODE_SHIFT 0
+#define DA7219_CIF_I2C_WRITE_MODE_MASK (0x1 << 0)
+#define DA7219_CIF_REG_SOFT_RESET_SHIFT 7
+#define DA7219_CIF_REG_SOFT_RESET_MASK (0x1 << 7)
+
+/* DA7219_SR_24_48 = 0x16 */
+#define DA7219_SR_24_48_SHIFT 0
+#define DA7219_SR_24_48_MASK (0x1 << 0)
+
+/* DA7219_SR = 0x17 */
+#define DA7219_SR_SHIFT 0
+#define DA7219_SR_MASK (0xF << 0)
+#define DA7219_SR_8000 (0x01 << 0)
+#define DA7219_SR_11025 (0x02 << 0)
+#define DA7219_SR_12000 (0x03 << 0)
+#define DA7219_SR_16000 (0x05 << 0)
+#define DA7219_SR_22050 (0x06 << 0)
+#define DA7219_SR_24000 (0x07 << 0)
+#define DA7219_SR_32000 (0x09 << 0)
+#define DA7219_SR_44100 (0x0A << 0)
+#define DA7219_SR_48000 (0x0B << 0)
+#define DA7219_SR_88200 (0x0E << 0)
+#define DA7219_SR_96000 (0x0F << 0)
+
+/* DA7219_CIF_I2C_ADDR_CFG = 0x1B */
+#define DA7219_CIF_I2C_ADDR_CFG_SHIFT 0
+#define DA7219_CIF_I2C_ADDR_CFG_MASK (0x3 << 0)
+
+/* DA7219_PLL_CTRL = 0x20 */
+#define DA7219_PLL_INDIV_SHIFT 2
+#define DA7219_PLL_INDIV_MASK (0x7 << 2)
+#define DA7219_PLL_INDIV_2_5_MHZ (0x0 << 2)
+#define DA7219_PLL_INDIV_5_10_MHZ (0x1 << 2)
+#define DA7219_PLL_INDIV_10_20_MHZ (0x2 << 2)
+#define DA7219_PLL_INDIV_20_40_MHZ (0x3 << 2)
+#define DA7219_PLL_INDIV_40_54_MHZ (0x4 << 2)
+#define DA7219_PLL_MCLK_SQR_EN_SHIFT 5
+#define DA7219_PLL_MCLK_SQR_EN_MASK (0x1 << 5)
+#define DA7219_PLL_MODE_SHIFT 6
+#define DA7219_PLL_MODE_MASK (0x3 << 6)
+#define DA7219_PLL_MODE_BYPASS (0x0 << 6)
+#define DA7219_PLL_MODE_NORMAL (0x1 << 6)
+#define DA7219_PLL_MODE_SRM (0x2 << 6)
+#define DA7219_PLL_MODE_32KHZ (0x3 << 6)
+
+/* DA7219_PLL_FRAC_TOP = 0x22 */
+#define DA7219_PLL_FBDIV_FRAC_TOP_SHIFT 0
+#define DA7219_PLL_FBDIV_FRAC_TOP_MASK (0x1F << 0)
+
+/* DA7219_PLL_FRAC_BOT = 0x23 */
+#define DA7219_PLL_FBDIV_FRAC_BOT_SHIFT 0
+#define DA7219_PLL_FBDIV_FRAC_BOT_MASK (0xFF << 0)
+
+/* DA7219_PLL_INTEGER = 0x24 */
+#define DA7219_PLL_FBDIV_INTEGER_SHIFT 0
+#define DA7219_PLL_FBDIV_INTEGER_MASK (0x7F << 0)
+
+/* DA7219_PLL_SRM_STS = 0x25 */
+#define DA7219_PLL_SRM_STATE_SHIFT 0
+#define DA7219_PLL_SRM_STATE_MASK (0xF << 0)
+#define DA7219_PLL_SRM_STATUS_SHIFT 4
+#define DA7219_PLL_SRM_STATUS_MASK (0xF << 4)
+#define DA7219_PLL_SRM_STS_SRM_LOCK (0x1 << 7)
+
+/* DA7219_DIG_ROUTING_DAI = 0x2A */
+#define DA7219_DAI_L_SRC_SHIFT 0
+#define DA7219_DAI_L_SRC_MASK (0x3 << 0)
+#define DA7219_DAI_R_SRC_SHIFT 4
+#define DA7219_DAI_R_SRC_MASK (0x3 << 4)
+#define DA7219_OUT_SRC_MAX 4
+
+/* DA7219_DAI_CLK_MODE = 0x2B */
+#define DA7219_DAI_BCLKS_PER_WCLK_SHIFT 0
+#define DA7219_DAI_BCLKS_PER_WCLK_MASK (0x3 << 0)
+#define DA7219_DAI_BCLKS_PER_WCLK_32 (0x0 << 0)
+#define DA7219_DAI_BCLKS_PER_WCLK_64 (0x1 << 0)
+#define DA7219_DAI_BCLKS_PER_WCLK_128 (0x2 << 0)
+#define DA7219_DAI_BCLKS_PER_WCLK_256 (0x3 << 0)
+#define DA7219_DAI_CLK_POL_SHIFT 2
+#define DA7219_DAI_CLK_POL_MASK (0x1 << 2)
+#define DA7219_DAI_CLK_POL_INV (0x1 << 2)
+#define DA7219_DAI_WCLK_POL_SHIFT 3
+#define DA7219_DAI_WCLK_POL_MASK (0x1 << 3)
+#define DA7219_DAI_WCLK_POL_INV (0x1 << 3)
+#define DA7219_DAI_WCLK_TRI_STATE_SHIFT 4
+#define DA7219_DAI_WCLK_TRI_STATE_MASK (0x1 << 4)
+#define DA7219_DAI_CLK_EN_SHIFT 7
+#define DA7219_DAI_CLK_EN_MASK (0x1 << 7)
+
+/* DA7219_DAI_CTRL = 0x2C */
+#define DA7219_DAI_FORMAT_SHIFT 0
+#define DA7219_DAI_FORMAT_MASK (0x3 << 0)
+#define DA7219_DAI_FORMAT_I2S (0x0 << 0)
+#define DA7219_DAI_FORMAT_LEFT_J (0x1 << 0)
+#define DA7219_DAI_FORMAT_RIGHT_J (0x2 << 0)
+#define DA7219_DAI_FORMAT_DSP (0x3 << 0)
+#define DA7219_DAI_WORD_LENGTH_SHIFT 2
+#define DA7219_DAI_WORD_LENGTH_MASK (0x3 << 2)
+#define DA7219_DAI_WORD_LENGTH_S16_LE (0x0 << 2)
+#define DA7219_DAI_WORD_LENGTH_S20_LE (0x1 << 2)
+#define DA7219_DAI_WORD_LENGTH_S24_LE (0x2 << 2)
+#define DA7219_DAI_WORD_LENGTH_S32_LE (0x3 << 2)
+#define DA7219_DAI_CH_NUM_SHIFT 4
+#define DA7219_DAI_CH_NUM_MASK (0x3 << 4)
+#define DA7219_DAI_CH_NUM_MAX 2
+#define DA7219_DAI_EN_SHIFT 7
+#define DA7219_DAI_EN_MASK (0x1 << 7)
+
+/* DA7219_DAI_TDM_CTRL = 0x2D */
+#define DA7219_DAI_TDM_CH_EN_SHIFT 0
+#define DA7219_DAI_TDM_CH_EN_MASK (0x3 << 0)
+#define DA7219_DAI_OE_SHIFT 6
+#define DA7219_DAI_OE_MASK (0x1 << 6)
+#define DA7219_DAI_TDM_MODE_EN_SHIFT 7
+#define DA7219_DAI_TDM_MODE_EN_MASK (0x1 << 7)
+#define DA7219_DAI_TDM_MAX_SLOTS 2
+
+/* DA7219_DIG_ROUTING_DAC = 0x2E */
+#define DA7219_DAC_L_SRC_SHIFT 0
+#define DA7219_DAC_L_SRC_MASK (0x3 << 0)
+#define DA7219_DAC_L_SRC_TONEGEN (0x1 << 0)
+#define DA7219_DAC_L_MONO_SHIFT 3
+#define DA7219_DAC_L_MONO_MASK (0x1 << 3)
+#define DA7219_DAC_R_SRC_SHIFT 4
+#define DA7219_DAC_R_SRC_MASK (0x3 << 4)
+#define DA7219_DAC_R_SRC_TONEGEN (0x1 << 4)
+#define DA7219_DAC_R_MONO_SHIFT 7
+#define DA7219_DAC_R_MONO_MASK (0x1 << 7)
+
+/* DA7219_ALC_CTRL1 = 0x2F */
+#define DA7219_ALC_OFFSET_EN_SHIFT 0
+#define DA7219_ALC_OFFSET_EN_MASK (0x1 << 0)
+#define DA7219_ALC_SYNC_MODE_SHIFT 1
+#define DA7219_ALC_SYNC_MODE_MASK (0x1 << 1)
+#define DA7219_ALC_EN_SHIFT 3
+#define DA7219_ALC_EN_MASK (0x1 << 3)
+#define DA7219_ALC_AUTO_CALIB_EN_SHIFT 4
+#define DA7219_ALC_AUTO_CALIB_EN_MASK (0x1 << 4)
+#define DA7219_ALC_CALIB_OVERFLOW_SHIFT 5
+#define DA7219_ALC_CALIB_OVERFLOW_MASK (0x1 << 5)
+
+/* DA7219_DAI_OFFSET_LOWER = 0x30 */
+#define DA7219_DAI_OFFSET_LOWER_SHIFT 0
+#define DA7219_DAI_OFFSET_LOWER_MASK (0xFF << 0)
+
+/* DA7219_DAI_OFFSET_UPPER = 0x31 */
+#define DA7219_DAI_OFFSET_UPPER_SHIFT 0
+#define DA7219_DAI_OFFSET_UPPER_MASK (0x7 << 0)
+#define DA7219_DAI_OFFSET_MAX 0x2FF
+
+/* DA7219_REFERENCES = 0x32 */
+#define DA7219_BIAS_EN_SHIFT 3
+#define DA7219_BIAS_EN_MASK (0x1 << 3)
+#define DA7219_VMID_FAST_CHARGE_SHIFT 4
+#define DA7219_VMID_FAST_CHARGE_MASK (0x1 << 4)
+
+/* DA7219_MIXIN_L_SELECT = 0x33 */
+#define DA7219_MIXIN_L_MIX_SELECT_SHIFT 0
+#define DA7219_MIXIN_L_MIX_SELECT_MASK (0x1 << 0)
+
+/* DA7219_MIXIN_L_GAIN = 0x34 */
+#define DA7219_MIXIN_L_AMP_GAIN_SHIFT 0
+#define DA7219_MIXIN_L_AMP_GAIN_MASK (0xF << 0)
+#define DA7219_MIXIN_L_AMP_GAIN_MAX 0xF
+
+/* DA7219_ADC_L_GAIN = 0x36 */
+#define DA7219_ADC_L_DIGITAL_GAIN_SHIFT 0
+#define DA7219_ADC_L_DIGITAL_GAIN_MASK (0x7F << 0)
+#define DA7219_ADC_L_DIGITAL_GAIN_MAX 0x7F
+
+/* DA7219_ADC_FILTERS1 = 0x38 */
+#define DA7219_ADC_VOICE_HPF_CORNER_SHIFT 0
+#define DA7219_ADC_VOICE_HPF_CORNER_MASK (0x7 << 0)
+#define DA7219_VOICE_HPF_CORNER_MAX 8
+#define DA7219_ADC_VOICE_EN_SHIFT 3
+#define DA7219_ADC_VOICE_EN_MASK (0x1 << 3)
+#define DA7219_ADC_AUDIO_HPF_CORNER_SHIFT 4
+#define DA7219_ADC_AUDIO_HPF_CORNER_MASK (0x3 << 4)
+#define DA7219_AUDIO_HPF_CORNER_MAX 4
+#define DA7219_ADC_HPF_EN_SHIFT 7
+#define DA7219_ADC_HPF_EN_MASK (0x1 << 7)
+#define DA7219_HPF_MODE_SHIFT 0
+#define DA7219_HPF_DISABLED ((0x0 << 3) | (0x0 << 7))
+#define DA7219_HPF_AUDIO_EN ((0x0 << 3) | (0x1 << 7))
+#define DA7219_HPF_VOICE_EN ((0x1 << 3) | (0x1 << 7))
+#define DA7219_HPF_MODE_MASK ((0x1 << 3) | (0x1 << 7))
+#define DA7219_HPF_MODE_MAX 3
+
+/* DA7219_MIC_1_GAIN = 0x39 */
+#define DA7219_MIC_1_AMP_GAIN_SHIFT 0
+#define DA7219_MIC_1_AMP_GAIN_MASK (0x7 << 0)
+
+/* DA7219_SIDETONE_CTRL = 0x3A */
+#define DA7219_SIDETONE_MUTE_EN_SHIFT 6
+#define DA7219_SIDETONE_MUTE_EN_MASK (0x1 << 6)
+#define DA7219_SIDETONE_EN_SHIFT 7
+#define DA7219_SIDETONE_EN_MASK (0x1 << 7)
+
+/* DA7219_SIDETONE_GAIN = 0x3B */
+#define DA7219_SIDETONE_GAIN_SHIFT 0
+#define DA7219_SIDETONE_GAIN_MASK (0xF << 0)
+#define DA7219_SIDETONE_GAIN_MAX 0xE
+
+/* DA7219_DROUTING_ST_OUTFILT_1L = 0x3C */
+#define DA7219_OUTFILT_ST_1L_SRC_SHIFT 0
+#define DA7219_OUTFILT_ST_1L_SRC_MASK (0x7 << 0)
+#define DA7219_DMIX_ST_SRC_OUTFILT1L_SHIFT 0
+#define DA7219_DMIX_ST_SRC_OUTFILT1R_SHIFT 1
+#define DA7219_DMIX_ST_SRC_SIDETONE_SHIFT 2
+#define DA7219_DMIX_ST_SRC_OUTFILT1L (0x1 << 0)
+#define DA7219_DMIX_ST_SRC_OUTFILT1R (0x1 << 1)
+
+/* DA7219_DROUTING_ST_OUTFILT_1R = 0x3D */
+#define DA7219_OUTFILT_ST_1R_SRC_SHIFT 0
+#define DA7219_OUTFILT_ST_1R_SRC_MASK (0x7 << 0)
+
+/* DA7219_DAC_FILTERS5 = 0x40 */
+#define DA7219_DAC_SOFTMUTE_RATE_SHIFT 4
+#define DA7219_DAC_SOFTMUTE_RATE_MASK (0x7 << 4)
+#define DA7219_DAC_SOFTMUTE_RATE_MAX 7
+#define DA7219_DAC_SOFTMUTE_EN_SHIFT 7
+#define DA7219_DAC_SOFTMUTE_EN_MASK (0x1 << 7)
+
+/* DA7219_DAC_FILTERS2 = 0x41 */
+#define DA7219_DAC_EQ_BAND1_SHIFT 0
+#define DA7219_DAC_EQ_BAND1_MASK (0xF << 0)
+#define DA7219_DAC_EQ_BAND2_SHIFT 4
+#define DA7219_DAC_EQ_BAND2_MASK (0xF << 4)
+#define DA7219_DAC_EQ_BAND_MAX 0xF
+
+/* DA7219_DAC_FILTERS3 = 0x42 */
+#define DA7219_DAC_EQ_BAND3_SHIFT 0
+#define DA7219_DAC_EQ_BAND3_MASK (0xF << 0)
+#define DA7219_DAC_EQ_BAND4_SHIFT 4
+#define DA7219_DAC_EQ_BAND4_MASK (0xF << 4)
+
+/* DA7219_DAC_FILTERS4 = 0x43 */
+#define DA7219_DAC_EQ_BAND5_SHIFT 0
+#define DA7219_DAC_EQ_BAND5_MASK (0xF << 0)
+#define DA7219_DAC_EQ_EN_SHIFT 7
+#define DA7219_DAC_EQ_EN_MASK (0x1 << 7)
+
+/* DA7219_DAC_FILTERS1 = 0x44 */
+#define DA7219_DAC_VOICE_HPF_CORNER_SHIFT 0
+#define DA7219_DAC_VOICE_HPF_CORNER_MASK (0x7 << 0)
+#define DA7219_DAC_VOICE_EN_SHIFT 3
+#define DA7219_DAC_VOICE_EN_MASK (0x1 << 3)
+#define DA7219_DAC_AUDIO_HPF_CORNER_SHIFT 4
+#define DA7219_DAC_AUDIO_HPF_CORNER_MASK (0x3 << 4)
+#define DA7219_DAC_HPF_EN_SHIFT 7
+#define DA7219_DAC_HPF_EN_MASK (0x1 << 7)
+
+/* DA7219_DAC_L_GAIN = 0x45 */
+#define DA7219_DAC_L_DIGITAL_GAIN_SHIFT 0
+#define DA7219_DAC_L_DIGITAL_GAIN_MASK (0x7F << 0)
+#define DA7219_DAC_DIGITAL_GAIN_MAX 0x7F
+#define DA7219_DAC_DIGITAL_GAIN_0DB (0x6F << 0)
+
+/* DA7219_DAC_R_GAIN = 0x46 */
+#define DA7219_DAC_R_DIGITAL_GAIN_SHIFT 0
+#define DA7219_DAC_R_DIGITAL_GAIN_MASK (0x7F << 0)
+
+/* DA7219_CP_CTRL = 0x47 */
+#define DA7219_CP_MCHANGE_SHIFT 4
+#define DA7219_CP_MCHANGE_MASK (0x3 << 4)
+#define DA7219_CP_MCHANGE_REL_MASK 0x3
+#define DA7219_CP_MCHANGE_MAX 3
+#define DA7219_CP_MCHANGE_LARGEST_VOL 0x1
+#define DA7219_CP_MCHANGE_DAC_VOL 0x2
+#define DA7219_CP_MCHANGE_SIG_MAG 0x3
+#define DA7219_CP_EN_SHIFT 7
+#define DA7219_CP_EN_MASK (0x1 << 7)
+
+/* DA7219_HP_L_GAIN = 0x48 */
+#define DA7219_HP_L_AMP_GAIN_SHIFT 0
+#define DA7219_HP_L_AMP_GAIN_MASK (0x3F << 0)
+#define DA7219_HP_AMP_GAIN_MAX 0x3F
+#define DA7219_HP_AMP_GAIN_0DB (0x39 << 0)
+
+/* DA7219_HP_R_GAIN = 0x49 */
+#define DA7219_HP_R_AMP_GAIN_SHIFT 0
+#define DA7219_HP_R_AMP_GAIN_MASK (0x3F << 0)
+
+/* DA7219_MIXOUT_L_SELECT = 0x4B */
+#define DA7219_MIXOUT_L_MIX_SELECT_SHIFT 0
+#define DA7219_MIXOUT_L_MIX_SELECT_MASK (0x1 << 0)
+
+/* DA7219_MIXOUT_R_SELECT = 0x4C */
+#define DA7219_MIXOUT_R_MIX_SELECT_SHIFT 0
+#define DA7219_MIXOUT_R_MIX_SELECT_MASK (0x1 << 0)
+
+/* DA7219_SYSTEM_MODES_INPUT = 0x50 */
+#define DA7219_MODE_SUBMIT_SHIFT 0
+#define DA7219_MODE_SUBMIT_MASK (0x1 << 0)
+#define DA7219_ADC_MODE_SHIFT 1
+#define DA7219_ADC_MODE_MASK (0x7F << 1)
+
+/* DA7219_SYSTEM_MODES_OUTPUT = 0x51 */
+#define DA7219_MODE_SUBMIT_SHIFT 0
+#define DA7219_MODE_SUBMIT_MASK (0x1 << 0)
+#define DA7219_DAC_MODE_SHIFT 1
+#define DA7219_DAC_MODE_MASK (0x7F << 1)
+
+/* DA7219_MICBIAS_CTRL = 0x62 */
+#define DA7219_MICBIAS1_LEVEL_SHIFT 0
+#define DA7219_MICBIAS1_LEVEL_MASK (0x7 << 0)
+#define DA7219_MICBIAS1_EN_SHIFT 3
+#define DA7219_MICBIAS1_EN_MASK (0x1 << 3)
+
+/* DA7219_MIC_1_CTRL = 0x63 */
+#define DA7219_MIC_1_AMP_RAMP_EN_SHIFT 5
+#define DA7219_MIC_1_AMP_RAMP_EN_MASK (0x1 << 5)
+#define DA7219_MIC_1_AMP_MUTE_EN_SHIFT 6
+#define DA7219_MIC_1_AMP_MUTE_EN_MASK (0x1 << 6)
+#define DA7219_MIC_1_AMP_EN_SHIFT 7
+#define DA7219_MIC_1_AMP_EN_MASK (0x1 << 7)
+
+/* DA7219_MIXIN_L_CTRL = 0x65 */
+#define DA7219_MIXIN_L_MIX_EN_SHIFT 3
+#define DA7219_MIXIN_L_MIX_EN_MASK (0x1 << 3)
+#define DA7219_MIXIN_L_AMP_ZC_EN_SHIFT 4
+#define DA7219_MIXIN_L_AMP_ZC_EN_MASK (0x1 << 4)
+#define DA7219_MIXIN_L_AMP_RAMP_EN_SHIFT 5
+#define DA7219_MIXIN_L_AMP_RAMP_EN_MASK (0x1 << 5)
+#define DA7219_MIXIN_L_AMP_MUTE_EN_SHIFT 6
+#define DA7219_MIXIN_L_AMP_MUTE_EN_MASK (0x1 << 6)
+#define DA7219_MIXIN_L_AMP_EN_SHIFT 7
+#define DA7219_MIXIN_L_AMP_EN_MASK (0x1 << 7)
+
+/* DA7219_ADC_L_CTRL = 0x67 */
+#define DA7219_ADC_L_BIAS_SHIFT 0
+#define DA7219_ADC_L_BIAS_MASK (0x3 << 0)
+#define DA7219_ADC_L_RAMP_EN_SHIFT 5
+#define DA7219_ADC_L_RAMP_EN_MASK (0x1 << 5)
+#define DA7219_ADC_L_MUTE_EN_SHIFT 6
+#define DA7219_ADC_L_MUTE_EN_MASK (0x1 << 6)
+#define DA7219_ADC_L_EN_SHIFT 7
+#define DA7219_ADC_L_EN_MASK (0x1 << 7)
+
+/* DA7219_DAC_L_CTRL = 0x69 */
+#define DA7219_DAC_L_RAMP_EN_SHIFT 5
+#define DA7219_DAC_L_RAMP_EN_MASK (0x1 << 5)
+#define DA7219_DAC_L_MUTE_EN_SHIFT 6
+#define DA7219_DAC_L_MUTE_EN_MASK (0x1 << 6)
+#define DA7219_DAC_L_EN_SHIFT 7
+#define DA7219_DAC_L_EN_MASK (0x1 << 7)
+
+/* DA7219_DAC_R_CTRL = 0x6A */
+#define DA7219_DAC_R_RAMP_EN_SHIFT 5
+#define DA7219_DAC_R_RAMP_EN_MASK (0x1 << 5)
+#define DA7219_DAC_R_MUTE_EN_SHIFT 6
+#define DA7219_DAC_R_MUTE_EN_MASK (0x1 << 6)
+#define DA7219_DAC_R_EN_SHIFT 7
+#define DA7219_DAC_R_EN_MASK (0x1 << 7)
+
+/* DA7219_HP_L_CTRL = 0x6B */
+#define DA7219_HP_L_AMP_MIN_GAIN_EN_SHIFT 2
+#define DA7219_HP_L_AMP_MIN_GAIN_EN_MASK (0x1 << 2)
+#define DA7219_HP_L_AMP_OE_SHIFT 3
+#define DA7219_HP_L_AMP_OE_MASK (0x1 << 3)
+#define DA7219_HP_L_AMP_ZC_EN_SHIFT 4
+#define DA7219_HP_L_AMP_ZC_EN_MASK (0x1 << 4)
+#define DA7219_HP_L_AMP_RAMP_EN_SHIFT 5
+#define DA7219_HP_L_AMP_RAMP_EN_MASK (0x1 << 5)
+#define DA7219_HP_L_AMP_MUTE_EN_SHIFT 6
+#define DA7219_HP_L_AMP_MUTE_EN_MASK (0x1 << 6)
+#define DA7219_HP_L_AMP_EN_SHIFT 7
+#define DA7219_HP_L_AMP_EN_MASK (0x1 << 7)
+
+/* DA7219_HP_R_CTRL = 0x6C */
+#define DA7219_HP_R_AMP_MIN_GAIN_EN_SHIFT 2
+#define DA7219_HP_R_AMP_MIN_GAIN_EN_MASK (0x1 << 2)
+#define DA7219_HP_R_AMP_OE_SHIFT 3
+#define DA7219_HP_R_AMP_OE_MASK (0x1 << 3)
+#define DA7219_HP_R_AMP_ZC_EN_SHIFT 4
+#define DA7219_HP_R_AMP_ZC_EN_MASK (0x1 << 4)
+#define DA7219_HP_R_AMP_RAMP_EN_SHIFT 5
+#define DA7219_HP_R_AMP_RAMP_EN_MASK (0x1 << 5)
+#define DA7219_HP_R_AMP_MUTE_EN_SHIFT 6
+#define DA7219_HP_R_AMP_MUTE_EN_MASK (0x1 << 6)
+#define DA7219_HP_R_AMP_EN_SHIFT 7
+#define DA7219_HP_R_AMP_EN_MASK (0x1 << 7)
+
+/* DA7219_MIXOUT_L_CTRL = 0x6E */
+#define DA7219_MIXOUT_L_AMP_EN_SHIFT 7
+#define DA7219_MIXOUT_L_AMP_EN_MASK (0x1 << 7)
+
+/* DA7219_MIXOUT_R_CTRL = 0x6F */
+#define DA7219_MIXOUT_R_AMP_EN_SHIFT 7
+#define DA7219_MIXOUT_R_AMP_EN_MASK (0x1 << 7)
+
+/* DA7219_CHIP_ID1 = 0x81 */
+#define DA7219_CHIP_ID1_SHIFT 0
+#define DA7219_CHIP_ID1_MASK (0xFF << 0)
+
+/* DA7219_CHIP_ID2 = 0x82 */
+#define DA7219_CHIP_ID2_SHIFT 0
+#define DA7219_CHIP_ID2_MASK (0xFF << 0)
+
+/* DA7219_CHIP_REVISION = 0x83 */
+#define DA7219_CHIP_MINOR_SHIFT 0
+#define DA7219_CHIP_MINOR_MASK (0xF << 0)
+#define DA7219_CHIP_MAJOR_SHIFT 4
+#define DA7219_CHIP_MAJOR