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[alsa-devel] [PATCH] ALSA: hda - add headset mic detect quirk for a Dell laptop
by Hui Wang 28 Apr '14

28 Apr '14

When we plug a 3-ring headset on the Dell machine (VID: 0x10ec0255, SID: 0x10280674), the headset mic can't be detected, after apply this patch, the headset mic can work well. BugLink: https://bugs.launchpad.net/bugs/1297581 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c643dfc..c1952c9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4621,6 +4621,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0667, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0668, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0669, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0674, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x067f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), -- 1.8.1.2

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[alsa-devel] [GIT PULL] ASoC updates for v3.15
by Mark Brown 28 Apr '14

28 Apr '14

The following changes since commit a798c10faf62a505d24e5f6213fbaf904a39623f: Linux 3.15-rc2 (2014-04-20 11:08:50 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.15-rc2 for you to fetch changes up to 98810a6dcf8af768d7b0b71e5075ff9c199f5196: Merge remote-tracking branches 'asoc/fix/intel', 'asoc/fix/jz4740', 'asoc/fix/rcar', 'asoc/fix/tlv320aic31xx' and 'asoc/fix/tlv320aic3x' into asoc-linus (2014-04-22 22:01:07 +0100) ---------------------------------------------------------------- ASoC: Fixes for v3.15 A smattering of driver-specific fixes here, nothing generic. The Cirrus CODEC conversions to devm_ are leak fixes - the conversion adds missing error handling code. ---------------------------------------------------------------- Alexandre Belloni (1): ASoC: tlv320aic31xx: document that the regulators are mandatory Andrew Lunn (1): ASoC: alc5623: Fix regmap endianness Axel Lin (2): ASoC: cs42l52: Convert to use devm_gpio_request_one ASoC: cs42l73: Convert to use devm_gpio_request_one Ben Dooks (1): ASoC: rsnd: fix clock prepare/unprepare Christian Engelmayer (1): ASoC: Intel: Fix incorrect sizeof() in sst_hsw_stream_get_volume() Dan Carpenter (1): ASoC: Intel: some incorrect sizeof() usages Jarkko Nikula (1): ASoC: dapm: Fix widget double free with auto-disable DAPM kcontrol Lars-Peter Clausen (1): ASoC: jz4740: Remove Makefile entry for removed file Mark Brown (3): Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branches 'asoc/fix/alc5623', 'asoc/fix/cs42l52', 'asoc/fix/cs42l73' and 'asoc/fix/fsl-spdif' into asoc-linus Merge remote-tracking branches 'asoc/fix/intel', 'asoc/fix/jz4740', 'asoc/fix/rcar', 'asoc/fix/tlv320aic31xx' and 'asoc/fix/tlv320aic3x' into asoc-linus Nicolin Chen (1): ASoC: fsl_spdif: Fix wrong OFFSET of STC_SYSCLK_DIV Sebastian Reichel (1): ASoC: tlv320aic3x: fix shared reset pin for DT Wenkai Du (1): ASoC: Intel: Fix audio crash due to negative address offset Documentation/devicetree/bindings/sound/tlv320aic31xx.txt | 6 +++--- sound/soc/codecs/alc5623.c | 2 +- sound/soc/codecs/cs42l52.c | 6 ++++-- sound/soc/codecs/cs42l73.c | 6 ++++-- sound/soc/codecs/tlv320aic3x.c | 9 +++++++-- sound/soc/fsl/fsl_spdif.h | 4 ++-- sound/soc/intel/sst-dsp-priv.h | 2 +- sound/soc/intel/sst-haswell-ipc.c | 7 ++++--- sound/soc/jz4740/Makefile | 2 -- sound/soc/sh/rcar/src.c | 4 ++-- sound/soc/sh/rcar/ssi.c | 4 ++-- sound/soc/soc-dapm.c | 1 - 12 files changed, 30 insertions(+), 23 deletions(-)

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[alsa-devel] [PATCH] SoC: rt5640: Add the rt5639 support to the OF match table
by Oder Chiou 28 Apr '14

28 Apr '14

The patch adds the rt5639 support to the OF match table. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 79635ee..2133fa0 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2157,6 +2157,7 @@ MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); #if defined(CONFIG_OF) static const struct of_device_id rt5640_of_match[] = { + { .compatible = "realtek,rt5639", }, { .compatible = "realtek,rt5640", }, {}, }; -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH v2] ASoC: rt5640: Add minimal support for RT5642
by bardliao＠realtek.com 28 Apr '14

28 Apr '14

From: Bard Liao <bardliao(a)realtek.com> We have been using rt5640.c codec driver with RT5642 codec chip before commit 022d21f004c1 ("ASoC: rt5640: add rt5639 support"). That commits starts using device ID reading in reset register for adding device specific controls and routes runtime. Now since device ID appears to be different between RT5640 and RT5642 the driver doesn't add those controls and routes that are valid also on RT5642. Fix this by adding a device ID found by debugging and minimal code for supporting RT5642. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- This patch add a mask for reading the device ID. This can remove the affect of other bits which is not related to device ID. --- sound/soc/codecs/rt5640.c | 8 +++++--- sound/soc/codecs/rt5640.h | 10 +++++++--- 2 files changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6674372..79635ee 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1997,8 +1997,9 @@ static int rt5640_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); - switch (snd_soc_read(codec, RT5640_RESET)) { - case RT5640_RESET_ID: + switch (snd_soc_read(codec, RT5640_RESET) & RT5640_ID_MASK) { + case RT5640_ID_5640: + case RT5640_ID_5642: snd_soc_add_codec_controls(codec, rt5640_specific_snd_controls, ARRAY_SIZE(rt5640_specific_snd_controls)); @@ -2009,7 +2010,7 @@ static int rt5640_probe(struct snd_soc_codec *codec) rt5640_specific_dapm_routes, ARRAY_SIZE(rt5640_specific_dapm_routes)); break; - case RT5639_RESET_ID: + case RT5640_ID_5639: snd_soc_dapm_new_controls(&codec->dapm, rt5639_specific_dapm_widgets, ARRAY_SIZE(rt5639_specific_dapm_widgets)); @@ -2149,6 +2150,7 @@ static const struct regmap_config rt5640_regmap = { static const struct i2c_device_id rt5640_i2c_id[] = { { "rt5640", 0 }, { "rt5639", 0 }, + { "rt5642", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 3b50459..ded2059 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -14,9 +14,6 @@ #include <sound/rt5640.h> -#define RT5639_RESET_ID 0x0008 -#define RT5640_RESET_ID 0x000c - /* Info */ #define RT5640_RESET 0x00 #define RT5640_VENDOR_ID 0xfd @@ -195,6 +192,13 @@ #define RT5640_R_VOL_MASK (0x3f) #define RT5640_R_VOL_SFT 0 +/* SW Reset & Device ID (0x00) */ +#define RT5640_ID_MASK (0x3 << 1) +#define RT5640_ID_5639 (0x0 << 1) +#define RT5640_ID_5640 (0x1 << 1) +#define RT5640_ID_5642 (0x3 << 1) + + /* IN1 and IN2 Control (0x0d) */ /* IN3 and IN4 Control (0x0e) */ #define RT5640_BST_SFT1 12 -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH] SoC: rt5640: Add the rt5639 support to the OF match table
by Oder Chiou 28 Apr '14

28 Apr '14

The patch adds the rt5639 support to the OF match table. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 79635ee..f2cc3ed 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2158,6 +2158,7 @@ MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); #if defined(CONFIG_OF) static const struct of_device_id rt5640_of_match[] = { { .compatible = "realtek,rt5640", }, + { .compatible = "realtek,rt5639", }, {}, }; MODULE_DEVICE_TABLE(of, rt5640_of_match); -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 0/5] Simple-audio-card support for HP t5325
by Andrew Lunn 28 Apr '14

28 Apr '14

This patchset allows the Kirkwood based HP t5325 device make use of simple-card to instantiate its audio hardware. Patch 1 Adds a DT binding to the ALC5623 codec Patch 2 Makes use of this new binding Patch 3 Add fixes up the kirkwood sound node in DT Patch 4 Extends simple-card to support the hw_params() needed when using MVEBU sound hardware. Patch 5 Swaps t5323 to using DT, removing all need for C code. Andrew Lunn (5): ASoC: alc5623: Add device tree binding ARM: Kirkwood: Use DT to instansiate codec ARM: Kirkwood: DT: Add missing #sound-dai-cells property ASoC: simple-card: Support setting mclk via a fixed factor ARM: Kirkwood: t5325: Use simple-card to instantiate audio .../devicetree/bindings/sound/alc5623.txt | 23 ++++++++++++ .../devicetree/bindings/sound/simple-card.txt | 2 ++ arch/arm/boot/dts/kirkwood-t5325.dts | 33 +++++++++++++++++ arch/arm/boot/dts/kirkwood.dtsi | 1 + arch/arm/mach-mvebu/board-t5325.c | 41 ---------------------- arch/arm/mach-mvebu/board.h | 6 ---- arch/arm/mach-mvebu/kirkwood.c | 3 -- sound/soc/codecs/alc5623.c | 13 +++++++ sound/soc/generic/simple-card.c | 28 +++++++++++++++ 9 files changed, 100 insertions(+), 50 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/alc5623.txt delete mode 100644 arch/arm/mach-mvebu/board-t5325.c -- 1.9.2

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[alsa-devel] [PATCH 0/2] Fix rxclk rate measurement
by Nicolin Chen 28 Apr '14

28 Apr '14

Add sysclk clock so as to calculate the rxclk rate correctly Nicolin Chen (2): ASoC: fsl_spdif: Fix clock source for rxclk rate measurement ASoC: fsl_spdif: Drop hard code in clk_get() for rxclk sound/soc/fsl/fsl_spdif.c | 17 ++++++++++++++--- sound/soc/fsl/fsl_spdif.h | 2 ++ 2 files changed, 16 insertions(+), 3 deletions(-) -- 1.8.4

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[alsa-devel] Bug? poll() timeout in alsa-jack plugin
by Sergey 27 Apr '14

27 Apr '14

Hello. I'm trying to debug mozilla code, which uses poll() when playing sound and works fine with regular alsa device, but plays no sound over alsa-jack plugin. I narrowed it down to a small sample. In that sample poll() returns 0 even when pcm can accept data. Please, help me to fix either a bug in that sample, or a bug in alsa-plugins. The problem is hardware-independent. To reproduce it you need: 1. install and start jack2 daemon: jackd -r -dalsa -dhw:0 -r44100 -p1024 -n2 2. add to .asoundrc: pcm.jackplug { type plug slave { pcm "jack" } } pcm.jack { type jack playback_ports { 0 system:playback_1 1 system:playback_2 } capture_ports { 0 system:capture_1 1 system:capture_2 } } 3. build and run the poll() code example gcc -o poll-example poll-example.c -lasound ./poll-example According to the common sense as long as PCM is ready to receive data it's expected that poll() returns 1 and writei() is called. Instead poll() returns 0 and the code plays nothing. ---------- poll() code example --------- #include <assert.h> #include <poll.h> #include <alsa/asoundlib.h> short buffer[96000]; int main() { snd_pcm_t *pcm; assert( snd_pcm_open(&pcm, "jackplug", SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK) == 0 ); assert( snd_pcm_nonblock(pcm, 1) == 0 ); assert( snd_pcm_set_params(pcm, SND_PCM_FORMAT_S16_LE, SND_PCM_ACCESS_RW_INTERLEAVED, 1/*channels*/, 48000/*rate*/, 1, 100000/*latency*/) == 0 ); nfds_t nfds = snd_pcm_poll_descriptors_count(pcm); assert( nfds > 0 ); struct pollfd *saved_fds = calloc(nfds, sizeof(struct pollfd)); assert( snd_pcm_poll_descriptors(pcm, saved_fds, nfds) == nfds ); // poll() and write(20) int timeout = 1000; int r = poll(saved_fds, nfds, timeout); fprintf(stderr, "poll=%d\n", r); if (r > 0) { unsigned short revents; r = snd_pcm_poll_descriptors_revents(pcm, saved_fds, nfds, &revents); fprintf(stderr, "snd_pcm_poll_descriptors_revents=%d\n", r); if (r == 0) fprintf(stderr, "snd_pcm_writei(20)=%d\n", snd_pcm_writei(pcm, buffer, 20)); } // poll() and write(50) r = poll(saved_fds, nfds, timeout); fprintf(stderr, "poll=%d\n", r); if (r > 0) { unsigned short revents; r = snd_pcm_poll_descriptors_revents(pcm, saved_fds, nfds, &revents); fprintf(stderr, "snd_pcm_poll_descriptors_revents=%d\n", r); if (r == 0) fprintf(stderr, "snd_pcm_writei(50)=%d\n", snd_pcm_writei(pcm, buffer, 50)); } // poll() and write(100) r = poll(saved_fds, nfds, timeout); fprintf(stderr, "poll=%d\n", r); if (r > 0) { unsigned short revents; r = snd_pcm_poll_descriptors_revents(pcm, saved_fds, nfds, &revents); fprintf(stderr, "snd_pcm_poll_descriptors_revents=%d\n", r); if (r == 0) fprintf(stderr, "snd_pcm_writei(100)=%d\n", snd_pcm_writei(pcm, buffer, 100)); } assert( snd_pcm_close(pcm) == 0 ); return 0; } ---------- end of code example --------- I tried to fix that in jack_pcm.c downloaded from http://git.alsa-project.org/?p=alsa-plugins.git;a=blob;f=jack/pcm_jack.c Making the first poll() return 1 was easy, I just added write(jack->fd, &i, 1); to the end of snd_pcm_jack_prepare() function. But that haven't solved the problem, because next call to snd_pcm_poll_descriptors_revents() exhausts the socket and next poll() still returns 0. Another guess was to define .transfer in snd_pcm_ioplug_callback_t and handle read()s and write()s to polling socket there, but it seems that .transfer is not called on writei() at all. What's the correct approach? Is there any official documentation about using poll() for ALSA? Or maybe there're some manuals with meaning of snd_pcm_ioplug_callback_t fields? --

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Re: [alsa-devel] [PATCH] ASoC: fsl-ssi: Allow to use driver with generic audio card
by Mark Brown 26 Apr '14

26 Apr '14

On Sat, Apr 26, 2014 at 09:02:06PM +0400, Alexander Shiyan wrote: > Sat, 26 Apr 2014 17:55:19 +0100 от Mark Brown <broonie(a)kernel.org>: > > On Sat, Apr 26, 2014 at 06:49:16PM +0400, Alexander Shiyan wrote: > > These are cases where we can do configuration but an invalid > > configuration has been provided and we have spotted an error. Ignoring > > that should cause runtime problems - I guess it isn't for you though so > > there may be something else going on? > > Especially for cases like DAI format we should probably enhance the > > framework so that these devices can set a variable in their struct > > saying what they do support but for now we don't do that. > Other solution to this problem is to check ret != EPROBE_DEFER > in the generic sound card. All other cases should be treated as > good for setup parameters. Is this better? I wouldn't expect to be getting -EPROBE_DEFER outside of the probe path? Can we take a step back here, please - what's the problem you're seeing?

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[alsa-devel] [PATCH] ASoC: Remove name_prefix unset during DAI link init hack again
by Lars-Peter Clausen 26 Apr '14

26 Apr '14

This was initially removed in commit 6479f15ad ("ASoC: Remove name_prefix unset during DAI link init hack"), but was brought back in commit 503ae5e0 ("ASoC: core: Add helpers for dai link and aux dev init") by accident. This patch removes it again. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-core.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f18112a..94c1668 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1272,14 +1272,10 @@ static int soc_aux_dev_init(struct snd_soc_card *card, { struct snd_soc_aux_dev *aux_dev = &card->aux_dev[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd_aux[num]; - const char *temp; int ret; rtd->card = card; - temp = codec->name_prefix; - codec->name_prefix = NULL; - /* do machine specific initialization */ if (aux_dev->init) { ret = aux_dev->init(&codec->dapm); @@ -1287,8 +1283,6 @@ static int soc_aux_dev_init(struct snd_soc_card *card, return ret; } - codec->name_prefix = temp; - rtd->codec = codec; return 0; @@ -1300,15 +1294,10 @@ static int soc_dai_link_init(struct snd_soc_card *card, { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - const char *temp; int ret; rtd->card = card; - /* machine controls, routes and widgets are not prefixed */ - temp = codec->name_prefix; - codec->name_prefix = NULL; - /* do machine specific initialization */ if (dai_link->init) { ret = dai_link->init(rtd); @@ -1316,8 +1305,6 @@ static int soc_dai_link_init(struct snd_soc_card *card, return ret; } - codec->name_prefix = temp; - rtd->codec = codec; return 0; -- 1.8.0

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[alsa-devel] [PATCH] ASoC: fsl-ssi: Allow to use driver with generic audio card
by Alexander Shiyan 26 Apr '14

26 Apr '14

Generic sound card checks return value of functions for setup format, clock and slot for -ENOTSUPP error. Add this compatibility to fsl-ssi driver. --- I'm really not sure that's a good solution, please correct me if I'm wrong. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/fsl_ssi.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 25865d8f..6891b8c 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -764,7 +764,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_SLAVE; break; default: - return -EINVAL; + return -ENOTSUPP; } /* Data on rising edge of bclk, frame low, 1clk before data */ @@ -789,7 +789,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) ssi_private->i2s_mode |= CCSR_SSI_SCR_I2S_MODE_NORMAL; break; default: - return -EINVAL; + return -ENOTSUPP; } scr |= ssi_private->i2s_mode; @@ -812,7 +812,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) strcr ^= CCSR_SSI_STCR_TFSI; break; default: - return -EINVAL; + return -ENOTSUPP; } /* DAI clock master masks */ @@ -825,7 +825,7 @@ static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; break; default: - return -EINVAL; + return -ENOTSUPP; } stcr |= strcr; @@ -897,7 +897,7 @@ static int fsl_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, /* Don't apply it to any non-baudclk circumstance */ if (IS_ERR(ssi_private->baudclk)) - return -EINVAL; + return -ENOTSUPP; /* It should be already enough to divide clock by setting pm alone */ psr = 0; @@ -986,7 +986,7 @@ static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, val = read_ssi(&ssi->scr) & (CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_NET); if (val && slots < 2) { dev_err(cpu_dai->dev, "slot number should be >= 2 in I2S or NET\n"); - return -EINVAL; + return -ENOTSUPP; } write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_DC_MASK, -- 1.8.3.2

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[alsa-devel] [PATCH RFC 00/17] Rework OMAP4+ HDMI audio support
by Jyri Sarha 26 Apr '14

26 Apr '14

This set of patches fixes OMAP4+ HDMI audio. The structure of the implementatin looks a bit different than before. Instead of creating a driver specific API for a separate ASoC component driver to connect to, this implementation integrates an the component driver into the HDMI driver. The idea is to use an existing ASoC component driver API instead of creating a new custom API for each HDMI IP and to avoid splitting the driver to half for separate video and audio parts connected with the API. The new implementation also uses simple-audio-card for a machine driver instead of having its own HW specific machine driver. The patches are based on 3.15-rc2 merged with git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next and git://git.kernel.org/pub/scm/linux/kernel/git/tomba/linux.git work/dss-dt-omap5 Everything is pushed here here: git://git.ti.com/~jyrisarha/ti-linux-kernel/jyrisarhas-audio-video-linux-fe… omap-hdmi-audio Best regards, Jyri Jyri Sarha (17): ASoC: omap-pcm: Move omap-pcm under include/sound OMAPDSS: hdmi-common: Add hdmi_dss_audio_from_hw_params() OMAPDSS: hdmi4: Integrated ASoC DAI component driver implementation OMAPDSS: Kconfig: Add depencies and help section to OMAP4_DSS_HDMI_AUDIO OMAPDSS: hdmi.h: Add HDMI_AUDIO_LAYOUT_6CH enum value OMAPDSS: hdmi5_core: Fix compilation with OMAP5_DSS_HDMI_AUDIO OMAPDSS: hdmi5: Integrated ASoC DAI component driver implementation OMAPDSS: Kconfig: Add depencies and help section to OMAP5_DSS_HDMI_AUDIO ASoC: omap: Remove obsolete HDMI audio code and Kconfig options OMAPDSS: Remove obsolete audio code ARM: omap4.dtsi: Add audio related parametes to hdmi node ARM: omap4-panda-common.dtsi: Add HDMI audio nodes ARM: omap5.dtsi: Add audio related parameters to hdmi node ARM: omap5-uevm.dts: Add hdmi audio related nodes ARM: omap2plus_defconfig: Build DSS HDMI support for OMAP4 in kernel ARM: omap2plus_defconfig: Enable OMAP5 HDMI support ARM: omap2plus_defconfig: Enable OMAP4+ HDMI audio support .../devicetree/bindings/video/ti,omap4-dss.txt | 4 + arch/arm/boot/dts/omap4-panda-common.dtsi | 21 +- arch/arm/boot/dts/omap4.dtsi | 2 + arch/arm/boot/dts/omap5-uevm.dts | 19 + arch/arm/boot/dts/omap5.dtsi | 2 + arch/arm/configs/omap2plus_defconfig | 20 +- .../fbdev/omap2/displays-new/connector-hdmi.c | 99 ------ .../fbdev/omap2/displays-new/encoder-tpd12s015.c | 56 --- drivers/video/fbdev/omap2/dss/Kconfig | 19 +- drivers/video/fbdev/omap2/dss/hdmi.h | 8 +- drivers/video/fbdev/omap2/dss/hdmi4.c | 216 ++++++++---- drivers/video/fbdev/omap2/dss/hdmi5.c | 216 ++++++++---- drivers/video/fbdev/omap2/dss/hdmi5_core.c | 2 +- drivers/video/fbdev/omap2/dss/hdmi_common.c | 117 ++++++- include/sound/omap-pcm.h | 30 ++ include/video/omapdss.h | 31 -- sound/soc/davinci/davinci-mcasp.c | 2 +- sound/soc/omap/Kconfig | 13 - sound/soc/omap/Makefile | 4 - sound/soc/omap/omap-dmic.c | 2 +- sound/soc/omap/omap-hdmi-card.c | 87 ----- sound/soc/omap/omap-hdmi.c | 364 -------------------- sound/soc/omap/omap-hdmi.h | 38 -- sound/soc/omap/omap-mcbsp.c | 2 +- sound/soc/omap/omap-mcpdm.c | 2 +- sound/soc/omap/omap-pcm.h | 30 -- 26 files changed, 536 insertions(+), 870 deletions(-) create mode 100644 include/sound/omap-pcm.h delete mode 100644 sound/soc/omap/omap-hdmi-card.c delete mode 100644 sound/soc/omap/omap-hdmi.c delete mode 100644 sound/soc/omap/omap-hdmi.h delete mode 100644 sound/soc/omap/omap-pcm.h -- 1.7.9.5

1 17

[alsa-devel] [PATCH 1/4] ASoC: tlv320dac3: Use devm_regulator_bulk_get()
by Fabio Estevam 25 Apr '14

25 Apr '14

From: Fabio Estevam <fabio.estevam(a)freescale.com> Using devm_regulator_bulk_get() can make the code cleaner and smaller as we do not need to call regulator_bulk_free() in the error and remove paths. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/tlv320dac33.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 517055a..df3a750 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -1540,7 +1540,7 @@ static int dac33_i2c_probe(struct i2c_client *client, for (i = 0; i < ARRAY_SIZE(dac33->supplies); i++) dac33->supplies[i].supply = dac33_supply_names[i]; - ret = regulator_bulk_get(&client->dev, ARRAY_SIZE(dac33->supplies), + ret = devm_regulator_bulk_get(&client->dev, ARRAY_SIZE(dac33->supplies), dac33->supplies); if (ret != 0) { @@ -1551,11 +1551,9 @@ static int dac33_i2c_probe(struct i2c_client *client, ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_tlv320dac33, &dac33_dai, 1); if (ret < 0) - goto err_register; + goto err_get; return ret; -err_register: - regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); err_get: if (dac33->power_gpio >= 0) gpio_free(dac33->power_gpio); @@ -1573,8 +1571,6 @@ static int dac33_i2c_remove(struct i2c_client *client) if (dac33->power_gpio >= 0) gpio_free(dac33->power_gpio); - regulator_bulk_free(ARRAY_SIZE(dac33->supplies), dac33->supplies); - snd_soc_unregister_codec(&client->dev); return 0; } -- 1.8.3.2

3 9

[alsa-devel] [PATCH 0/5] treewide: Convert logging message misuses of /n to \n
by Joe Perches 25 Apr '14

25 Apr '14

Joe Perches (5): qlge: Convert /n to \n iwlegacy: Convert /n to \n staging: rtl<foo>: Convert /n to \n appledisplay: Convert /n to \n sound: tlv320aic31xx: Convert /n to \n drivers/net/ethernet/qlogic/qlge/qlge_main.c | 2 +- drivers/net/wireless/iwlegacy/3945.c | 2 +- drivers/net/wireless/iwlegacy/4965-mac.c | 2 +- drivers/staging/rtl8188eu/hal/usb_ops_linux.c | 2 +- drivers/staging/rtl8188eu/os_dep/usb_intf.c | 2 +- drivers/staging/rtl8723au/hal/usb_ops_linux.c | 4 ++-- drivers/staging/rtl8723au/os_dep/usb_intf.c | 2 +- drivers/usb/misc/appledisplay.c | 2 +- sound/soc/codecs/tlv320aic31xx.c | 2 +- 9 files changed, 10 insertions(+), 10 deletions(-) -- 1.8.1.2.459.gbcd45b4.dirty

2 2

[alsa-devel] [PATCH] ASoC: sgtl5000: Use devm_regulator_bulk_get()
by Fabio Estevam 24 Apr '14

24 Apr '14

Using devm_regulator_bulk_get() can make the code cleaner and smaller as we do not need to call regulator_bulk_free() in the error and remove paths. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index b56caef..9626ee0 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1322,7 +1322,7 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) return ret; } - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), + ret = devm_regulator_bulk_get(codec->dev, ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); if (ret) goto err_ldo_remove; @@ -1330,16 +1330,13 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) ret = regulator_bulk_enable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); if (ret) - goto err_regulator_free; + goto err_ldo_remove; /* wait for all power rails bring up */ udelay(10); return 0; -err_regulator_free: - regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); err_ldo_remove: if (!external_vddd) ldo_regulator_remove(codec); @@ -1409,8 +1406,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) err: regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); - regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); ldo_regulator_remove(codec); return ret; @@ -1424,8 +1419,6 @@ static int sgtl5000_remove(struct snd_soc_codec *codec) regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), sgtl5000->supplies); - regulator_bulk_free(ARRAY_SIZE(sgtl5000->supplies), - sgtl5000->supplies); ldo_regulator_remove(codec); return 0; -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: simple-card: is_top_level_node parameter to simple_card_dai_link_of()
by Jyri Sarha 24 Apr '14

24 Apr '14

Restore correct parsing of dai-link subnodes with more explicit implementation for applying the "simple-audio-card,"-prefix to dai-link property and subnode names. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/generic/simple-card.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 98f97e5..06fe0e2 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -144,7 +144,8 @@ asoc_simple_card_sub_parse_of(struct device_node *np, static int simple_card_dai_link_of(struct device_node *node, struct device *dev, struct snd_soc_dai_link *dai_link, - struct simple_dai_props *dai_props) + struct simple_dai_props *dai_props, + bool is_top_level_node) { struct device_node *np = NULL; struct device_node *bitclkmaster = NULL; @@ -155,7 +156,8 @@ static int simple_card_dai_link_of(struct device_node *node, char *prefix = ""; int ret; - prefix = "simple-audio-card,"; + if (is_top_level_node) + prefix = "simple-audio-card,"; daifmt = snd_soc_of_parse_daifmt(node, prefix, &bitclkmaster, &framemaster); @@ -307,14 +309,15 @@ static int asoc_simple_card_parse_of(struct device_node *node, for (i = 0; (np = of_get_next_child(node, np)); i++) { dev_dbg(dev, "\tlink %d:\n", i); ret = simple_card_dai_link_of(np, dev, dai_link + i, - dai_props + i); + dai_props + i, false); if (ret < 0) { of_node_put(np); return ret; } } } else { - ret = simple_card_dai_link_of(node, dev, dai_link, dai_props); + ret = simple_card_dai_link_of(node, dev, dai_link, dai_props, + true); if (ret < 0) return ret; } -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH] ASoC: imx-sgtl5000.c: Add support for specifying PDCR_TXRXEN
by Markus Pargmann 24 Apr '14

24 Apr '14

Hi, On Wed, Apr 23, 2014 at 05:50:18PM +0200, David Jander wrote: > This enables effectively swapping RX/TX pins on the external audio port > > Signed-off-by: David Jander <david(a)protonic.nl> > --- > Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt | 4 ++++ > sound/soc/fsl/imx-sgtl5000.c | 5 ++++- > 2 files changed, 8 insertions(+), 1 deletion(-) > > diff --git a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt > index e4acdd8..3afea85 100644 > --- a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt > +++ b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt > @@ -29,6 +29,10 @@ Required properties: > - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) > - mux-ext-port : The external port of the i.MX audio muxer > > +Optional properties: > +- ext-port-txrxen : Flag. Set to swap RX/TX signals on external port (set the > + IMX_AUDMUX_V2_PDCR_TXRXEN bit in the PDC register of the external port) > + > Note: The AUDMUX port numbering should start at 1, which is consistent with > hardware manual. You could also try to use the simple sound card and use DT properties to setup audmux. This is a configuration example for the audmux: &audmux { status = "okay"; aud4 { fsl,audmux-port = <MX31_AUDMUX_PORT4_SSI_PINS_4>; fsl,port-config = < (IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSDIR | IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT1_SSI0) | IMX_AUDMUX_V2_PTCR_TCLKDIR | IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT1_SSI0)) IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0) >; }; }; Regards, Markus > > diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c > index 1cb22dd..b543fa6 100644 > --- a/sound/soc/fsl/imx-sgtl5000.c > +++ b/sound/soc/fsl/imx-sgtl5000.c > @@ -63,6 +63,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) > struct i2c_client *codec_dev; > struct imx_sgtl5000_data *data = NULL; > int int_port, ext_port; > + unsigned int ext_txrxen = 0; > int ret; > > ret = of_property_read_u32(np, "mux-int-port", &int_port); > @@ -75,6 +76,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) > dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); > return ret; > } > + if (of_property_read_bool(np, "ext-port-txrxen")) > + ext_txrxen = IMX_AUDMUX_V2_PDCR_TXRXEN; > > /* > * The port numbering in the hardware manual starts at 1, while > @@ -95,7 +98,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) > } > ret = imx_audmux_v2_configure_port(ext_port, > IMX_AUDMUX_V2_PTCR_SYN, > - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); > + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port) | ext_txrxen); > if (ret) { > dev_err(&pdev->dev, "audmux external port setup failed\n"); > return ret; > -- > 1.9.1 > > -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

2 3

[alsa-devel] [PATCH 0/3] Fix some minor issues in simple card driver
by Nicolin Chen 24 Apr '14

24 Apr '14

The series fixes one name limitation and code style problems. Nicolin Chen (3): ASoC: simple-card: Drop node->name checking ASoC: simple-card: Simplify error msg in simple_card_dai_link_of() ASoC: simple-card: Improve coding style sound/soc/generic/simple-card.c | 32 ++++++++++++++------------------ 1 file changed, 14 insertions(+), 18 deletions(-) -- 1.8.4

3 7

[alsa-devel] [PATCH] ASoC: fsl: Drop formats limitation for imx-pcm-dma.c
by Nicolin Chen 24 Apr '14

24 Apr '14

Now ASoC core is getting the intersection of supported formats not only from CPU and CODEC dai's but also from DMA's. However, there should be no specific width limitation from SDMA side. So drop it. Otherwise, we would only support S16_LE format for all i.MX platforms. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/imx-pcm-dma.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 2585ae4..0849b7b 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -40,7 +40,6 @@ static const struct snd_pcm_hardware imx_pcm_hardware = { SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE, .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, .period_bytes_min = 128, .period_bytes_max = 65535, /* Limited by SDMA engine */ -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: fsl_spdif: Add core clock control for DMA access
by Nicolin Chen 24 Apr '14

24 Apr '14

Regmap is able to enable/disable the core clock automatically each time it's going to access the registers. But for DMA cases during playback or recording, it's totally beyong control of regmap. So we have to open the clock manually. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index daa6198..e2836b3 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -80,6 +80,7 @@ struct fsl_spdif_priv { u8 rxclk_src; struct clk *txclk[SPDIF_TXRATE_MAX]; struct clk *rxclk; + struct clk *coreclk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -423,10 +424,16 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream, /* Reset module and interrupts only for first initialization */ if (!cpu_dai->active) { + ret = clk_prepare_enable(spdif_priv->coreclk); + if (ret) { + dev_err(&pdev->dev, "failed to enable core clock\n"); + return ret; + } + ret = spdif_softreset(spdif_priv); if (ret) { dev_err(&pdev->dev, "failed to soft reset\n"); - return ret; + goto err; } /* Disable all the interrupts */ @@ -454,6 +461,11 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream, regmap_update_bits(regmap, REG_SPDIF_SCR, SCR_LOW_POWER, 0); return 0; + +err: + clk_disable_unprepare(spdif_priv->coreclk); + + return ret; } static void fsl_spdif_shutdown(struct snd_pcm_substream *substream, @@ -484,6 +496,7 @@ static void fsl_spdif_shutdown(struct snd_pcm_substream *substream, spdif_intr_status_clear(spdif_priv); regmap_update_bits(regmap, REG_SPDIF_SCR, SCR_LOW_POWER, SCR_LOW_POWER); + clk_disable_unprepare(spdif_priv->coreclk); } } @@ -1134,6 +1147,13 @@ static int fsl_spdif_probe(struct platform_device *pdev) return ret; } + /* Get core clock for data register access via DMA */ + spdif_priv->coreclk = devm_clk_get(&pdev->dev, "core"); + if (IS_ERR(spdif_priv->coreclk)) { + dev_err(&pdev->dev, "no core clock in devicetree\n"); + return PTR_ERR(spdif_priv->coreclk); + } + /* Select clock source for rx/tx clock */ spdif_priv->rxclk = devm_clk_get(&pdev->dev, "rxtx1"); if (IS_ERR(spdif_priv->rxclk)) { -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: Intel: Cancel hsw_notification_work before freeing the stream
by Jarkko Nikula 24 Apr '14

24 Apr '14

I suppose there is a possibility that hsw_notification_work() may run after sst_hsw_stream_free() which can lead to a kernel crash since struct sst_hsw_stream is freed at that point and stream = container_of(work, struct sst_hsw_stream, notify_work) is not valid when hsw_notification_work() is run. Reported-by: Derek Basehore <dbasehore(a)chromium.org> Reported-by: Wenkai Du <wenkai.du(a)intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- Untested, I don't have at the moment Haswell to test this and Liam hasn't seen this patch yet so ack/tested-by from him might be good to have :-) --- sound/soc/intel/sst-haswell-ipc.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 6c0b4f247a86..5bcf5963a0ba 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1207,6 +1207,7 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream) trace_hsw_stream_free_req(stream, &stream->free_req); out: + cancel_work_sync(&stream->notify_work); spin_lock_irqsave(&sst->spinlock, flags); list_del(&stream->node); kfree(stream); -- 1.9.2

2 1

[alsa-devel] [PATCH] ASoC: imx-audmux: Fix section mismatch
by Lars-Peter Clausen 24 Apr '14

24 Apr '14

audmux_debugfs_init() is marked as __init, but is called from imx_audmux_probe() which is not marked as __init. This creates a section mismatch and a potential runtime crash (if imx_audmux_probe() is called after the .init section was dropped). This patch removes the __init annotation from audmux_debugfs_init(), which fixes the following warning: WARNING: sound/soc/built-in.o(.text+0x86960): Section mismatch in reference from the function imx_audmux_probe() to the function .init.text:audmux_debugfs_init() Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/fsl/imx-audmux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index ac86993..267717a 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -145,7 +145,7 @@ static const struct file_operations audmux_debugfs_fops = { .llseek = default_llseek, }; -static void __init audmux_debugfs_init(void) +static void audmux_debugfs_init(void) { int i; char buf[20]; -- 1.8.0

2 1

Re: [alsa-devel] [PATCH] ASoC: imx-sgtl5000.c: Add support for specifying PDCR_TXRXEN
by Shawn Guo 24 Apr '14

24 Apr '14

On Wed, Apr 23, 2014 at 05:50:18PM +0200, David Jander wrote: > This enables effectively swapping RX/TX pins on the external audio port > > Signed-off-by: David Jander <david(a)protonic.nl> You should copy Mark Brown <broonie(a)kernel.org> and devicetree(a)vger.kernel.org on this patch. Shawn > --- > Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt | 4 ++++ > sound/soc/fsl/imx-sgtl5000.c | 5 ++++- > 2 files changed, 8 insertions(+), 1 deletion(-) > > diff --git a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt > index e4acdd8..3afea85 100644 > --- a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt > +++ b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt > @@ -29,6 +29,10 @@ Required properties: > - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) > - mux-ext-port : The external port of the i.MX audio muxer > > +Optional properties: > +- ext-port-txrxen : Flag. Set to swap RX/TX signals on external port (set the > + IMX_AUDMUX_V2_PDCR_TXRXEN bit in the PDC register of the external port) > + > Note: The AUDMUX port numbering should start at 1, which is consistent with > hardware manual. > > diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c > index 1cb22dd..b543fa6 100644 > --- a/sound/soc/fsl/imx-sgtl5000.c > +++ b/sound/soc/fsl/imx-sgtl5000.c > @@ -63,6 +63,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) > struct i2c_client *codec_dev; > struct imx_sgtl5000_data *data = NULL; > int int_port, ext_port; > + unsigned int ext_txrxen = 0; > int ret; > > ret = of_property_read_u32(np, "mux-int-port", &int_port); > @@ -75,6 +76,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) > dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); > return ret; > } > + if (of_property_read_bool(np, "ext-port-txrxen")) > + ext_txrxen = IMX_AUDMUX_V2_PDCR_TXRXEN; > > /* > * The port numbering in the hardware manual starts at 1, while > @@ -95,7 +98,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) > } > ret = imx_audmux_v2_configure_port(ext_port, > IMX_AUDMUX_V2_PTCR_SYN, > - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); > + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port) | ext_txrxen); > if (ret) { > dev_err(&pdev->dev, "audmux external port setup failed\n"); > return ret; > -- > 1.9.1 >

1 0

[alsa-devel] [PATCH] ASoC: core: Don't break component searching if both id and num_dai are 0
by Nicolin Chen 23 Apr '14

23 Apr '14

The commit e41975ed (ASoC: core: Fix the DAI name getting) added a break within the "if (id < 0 || id >= pos->num_dai)" while the original design of the search didn't break the loop if that condition contented but only mark the ret error and let it go on to search the next component. In a case like dmaengine which's not a dai but as a component sharing an identical name with a dai, both the id and pos->num_dai here could be 0. If we break the search, we may never find the dai we want as it might be placed behind its dmaengine in the component list. So this patch fixes the issue above by following the original design to let the search carry on. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a17c047..82e9fdf 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4591,7 +4591,7 @@ int snd_soc_of_get_dai_name(struct device_node *of_node, if (id < 0 || id >= pos->num_dai) { ret = -EINVAL; - break; + continue; } ret = 0; -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: SAMSUNG: Don't clear clock setting during i2s_startup
by Tushar Behera 23 Apr '14

23 Apr '14

In exiting kernel, if DAIFMT flags are set in dai_link and I2S is set to run in master mode, the I2S clocks are not getting configured resulting in no output. Existing code clears the current I2S clock settings during i2s_startup and requires that the clocks are reconfigured. It then assumes that sound-card driver would call snd_soc_dai_{set_sysclk/set_fmt} to configure the root clock. 1. Since I2S clock settings remain fixed for a board, it would be better to set the clocks once during sound-card probe. 2. Also if the DAIFMT flags are set in dai_link, snd_soc_dai_set_fmt is called during DAI probe. If both these conditions are true, then I2S clock remains unconfigured during audio playback. Fix this by removing the code to clear rclk_srcrate in i2s_startup. Instead, reset this during DAI probe. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> --- The patch is based on v3.15-rc2. sound/soc/samsung/i2s.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 048ead9..6e61db7 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -724,9 +724,6 @@ static int i2s_startup(struct snd_pcm_substream *substream, else i2s->mode |= DAI_MANAGER; - /* Enforce set_sysclk in Master mode */ - i2s->rclk_srcrate = 0; - if (!any_active(i2s) && (i2s->quirks & QUIRK_NEED_RSTCLR)) writel(CON_RSTCLR, i2s->addr + I2SCON); @@ -984,6 +981,7 @@ probe_exit: /* Reset any constraint on RFS and BFS */ i2s->rfs = 0; i2s->bfs = 0; + i2s->rclk_srcrate = 0; i2s_txctrl(i2s, 0); i2s_rxctrl(i2s, 0); i2s_fifo(i2s, FIC_TXFLUSH); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: Intel: Fix audio crash due to race condition in stream deletion
by Jarkko Nikula 23 Apr '14

23 Apr '14

From: Wenkai Du <wenkai.du(a)intel.com> There is a race between sst_byt_stream_free() and sst_byt_get_stream() if sst_byt_get_stream() called from sst_byt_irq_thread() context is accessing the byt->stream_list while a stream is deleted from the list. A stream is added to byt->stream_list in sst_byt_stream_new() and deleted in sst_byt_stream_free(). sst_byt_get_stream() is always protected by sst->spinlock, but the stream addition and deletion are not protected. The patch adds spinlock to both stream addition and deletion. [Jarkko: Same fix added to sst-haswell-ipc.c too] Signed-off-by: Wenkai Du <wenkai.du(a)intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst-baytrail-ipc.c | 8 ++++++++ sound/soc/intel/sst-haswell-ipc.c | 8 ++++++++ 2 files changed, 16 insertions(+) diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index d0eaeee21be4..0d31dbbf4806 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -542,16 +542,20 @@ struct sst_byt_stream *sst_byt_stream_new(struct sst_byt *byt, int id, void *data) { struct sst_byt_stream *stream; + struct sst_dsp *sst = byt->dsp; + unsigned long flags; stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (stream == NULL) return NULL; + spin_lock_irqsave(&sst->spinlock, flags); list_add(&stream->node, &byt->stream_list); stream->notify_position = notify_position; stream->pdata = data; stream->byt = byt; stream->str_id = id; + spin_unlock_irqrestore(&sst->spinlock, flags); return stream; } @@ -630,6 +634,8 @@ int sst_byt_stream_free(struct sst_byt *byt, struct sst_byt_stream *stream) { u64 header; int ret = 0; + struct sst_dsp *sst = byt->dsp; + unsigned long flags; if (!stream->commited) goto out; @@ -644,8 +650,10 @@ int sst_byt_stream_free(struct sst_byt *byt, struct sst_byt_stream *stream) stream->commited = false; out: + spin_lock_irqsave(&sst->spinlock, flags); list_del(&stream->node); kfree(stream); + spin_unlock_irqrestore(&sst->spinlock, flags); return ret; } diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 50e4246d4b57..6c0b4f247a86 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1159,11 +1159,14 @@ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, void *data) { struct sst_hsw_stream *stream; + struct sst_dsp *sst = hsw->dsp; + unsigned long flags; stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (stream == NULL) return NULL; + spin_lock_irqsave(&sst->spinlock, flags); list_add(&stream->node, &hsw->stream_list); stream->notify_position = notify_position; stream->pdata = data; @@ -1172,6 +1175,7 @@ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, /* work to process notification messages */ INIT_WORK(&stream->notify_work, hsw_notification_work); + spin_unlock_irqrestore(&sst->spinlock, flags); return stream; } @@ -1180,6 +1184,8 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream) { u32 header; int ret = 0; + struct sst_dsp *sst = hsw->dsp; + unsigned long flags; /* dont free DSP streams that are not commited */ if (!stream->commited) @@ -1201,8 +1207,10 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream) trace_hsw_stream_free_req(stream, &stream->free_req); out: + spin_lock_irqsave(&sst->spinlock, flags); list_del(&stream->node); kfree(stream); + spin_unlock_irqrestore(&sst->spinlock, flags); return ret; } -- 1.9.2

2 1

[alsa-devel] [PATCH 1/6] ASoC: jz4740: Remove Makefile entry for removed file
by Lars-Peter Clausen 23 Apr '14

23 Apr '14

Commit 0406a40a0 ("ASoC: jz4740: Use the generic dmaengine PCM driver") jz4740-pcm.c file, but neglected to remove the Makefile entries. Fixes: 0406a40a0 ("ASoC: jz4740: Use the generic dmaengine PCM driver") Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Reported-by: Ralf Baechle <ralf(a)linux-mips.org> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/jz4740/Makefile | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/jz4740/Makefile b/sound/soc/jz4740/Makefile index be873c1..d32c540 100644 --- a/sound/soc/jz4740/Makefile +++ b/sound/soc/jz4740/Makefile @@ -1,10 +1,8 @@ # # Jz4740 Platform Support # -snd-soc-jz4740-objs := jz4740-pcm.o snd-soc-jz4740-i2s-objs := jz4740-i2s.o -obj-$(CONFIG_SND_JZ4740_SOC) += snd-soc-jz4740.o obj-$(CONFIG_SND_JZ4740_SOC_I2S) += snd-soc-jz4740-i2s.o # Jz4740 Machine Support -- 1.8.0

3 9

[alsa-devel] [PATCH] ASoC: rsnd: call rsnd_dai_pointer_update() from outside of lock
by Kuninori Morimoto 23 Apr '14

23 Apr '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd_soc_dai_trigger() will be called after rsnd_dai_pointer_update() function which is using rsnd_lock(). Thus, it should be called from outside of rsnd_lock(). Kernel will be hangup without this patch. Special thanks to Kataoka-san Reported-by: Ryo Kataoka <ryo.kataoka.wt(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1f6981a..964463d 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -197,13 +197,12 @@ static void rsnd_dma_complete(void *data) * rsnd_dai_pointer_update() will be called twice, * ant it will breaks io->byte_pos */ - - rsnd_dai_pointer_update(io, io->byte_per_period); - if (dma->submit_loop) rsnd_dma_continue(dma); rsnd_unlock(priv, flags); + + rsnd_dai_pointer_update(io, io->byte_per_period); } static void __rsnd_dma_start(struct rsnd_dma *dma) -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: max98095: Add an explicit of_match_table
by Tushar Behera 23 Apr '14

23 Apr '14

Create an explicit of_match_table entry for MAX98095 codec. Also add a binding Documentation for this compatible string. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> --- .../devicetree/bindings/sound/max98095.txt | 16 ++++++++++++++++ sound/soc/codecs/max98095.c | 7 +++++++ 2 files changed, 23 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98095.txt diff --git a/Documentation/devicetree/bindings/sound/max98095.txt b/Documentation/devicetree/bindings/sound/max98095.txt new file mode 100644 index 0000000..bacbeaa --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98095.txt @@ -0,0 +1,16 @@ +MAX98095 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "maxim,max98095". + +- reg : The I2C address of the device. + +Example: + +max98095: codec@11 { + compatible = "maxim,max98095"; + reg = <0x11>; +}; diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 03f0536..8e548af 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2399,10 +2399,17 @@ static const struct i2c_device_id max98095_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, max98095_i2c_id); +static const struct of_device_id max98095_of_match[] = { + { .compatible = "maxim,max98095", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98095_of_match); + static struct i2c_driver max98095_i2c_driver = { .driver = { .name = "max98095", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(max98095_of_match), }, .probe = max98095_i2c_probe, .remove = max98095_i2c_remove, -- 1.7.9.5

2 1

[alsa-devel] How to show hda trace events in a self-built kernel?
by Lin, Mengdong 23 Apr '14

23 Apr '14

Hi, How to show hda trace events in a self-built kernel? With a self-built kernel under Ubuntu 13.04 or 13.10, I enabled hda trace events by #cd /sys/kernel/debug/tracing/ #echo 1 > events/hda/enable #echo 1 > events/had_intel/enable Then I check the codec proc entry and run "cat trace". But I got no output of the hda trace events. But I switch to the original kernel in Ubuntu 13.04 or 13.10 and repeat the above steps, I can get the output of hda trace events. Thanks Mengdong

2 2

[alsa-devel] [PATCH] ASoC: SAMSUNG: Add sound card driver for Snow board
by Tushar Behera 23 Apr '14

23 Apr '14

Added machine driver to instantiate I2S based sound card on Snow board. It has MAX98095 audio codec on board. There are some other variants for Snow board which have MAX98090 audio codec. Hence support for MAX98090 is also added to this driver. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> --- Documentation/devicetree/bindings/sound/snow.txt | 17 ++ sound/soc/samsung/Kconfig | 11 ++ sound/soc/samsung/Makefile | 2 + sound/soc/samsung/snow.c | 190 ++++++++++++++++++++++ 4 files changed, 220 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/snow.txt create mode 100644 sound/soc/samsung/snow.c diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt new file mode 100644 index 0000000..6f3fd02 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/snow.txt @@ -0,0 +1,17 @@ +Audio Binding for Snow boards + +Required properties: +- compatible : Can be one of the following, + "google,snow-audio-max98090" or + "google,snow-audio-max98095" +- samsung,i2s-controller: The phandle of the Samsung I2S0 controller +- samsung,audio-codec: The phandle of the audio codec + +Example: + +sound { + compatible = "google,snow-audio-max98095"; + + samsung,i2s-controller = <&i2s0>; + samsung,audio-codec = <&max98095>; +}; diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index f2e2891..d07b04f 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -231,3 +231,13 @@ config SND_SOC_LITTLEMILL select SND_SAMSUNG_I2S select MFD_WM8994 select SND_SOC_WM8994 + +config SND_SOC_SNOW + tristate "Audio support for Google Snow boards" + depends on SND_SOC_SAMSUNG + select SND_SOC_MAX98090 + select SND_SOC_MAX98095 + select SND_SAMSUNG_I2S + help + Say Y if you want to add audio support for various Snow + boards based on Exynos5 series of SoCs. diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 86715d8..6d0212b 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -34,6 +34,7 @@ snd-soc-h1940-uda1380-objs := h1940_uda1380.o snd-soc-rx1950-uda1380-objs := rx1950_uda1380.o snd-soc-smdk-wm8580-objs := smdk_wm8580.o snd-soc-smdk-wm8994-objs := smdk_wm8994.o +snd-soc-snow-objs := snow.o snd-soc-smdk-wm9713-objs := smdk_wm9713.o snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o snd-soc-goni-wm8994-objs := goni_wm8994.o @@ -58,6 +59,7 @@ obj-$(CONFIG_SND_SOC_SAMSUNG_H1940_UDA1380) += snd-soc-h1940-uda1380.o obj-$(CONFIG_SND_SOC_SAMSUNG_RX1950_UDA1380) += snd-soc-rx1950-uda1380.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM8580) += snd-soc-smdk-wm8580.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM8994) += snd-soc-smdk-wm8994.o +obj-$(CONFIG_SND_SOC_SNOW) += snd-soc-snow.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_WM9713) += snd-soc-smdk-wm9713.o obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c new file mode 100644 index 0000000..e8eedd8 --- /dev/null +++ b/sound/soc/samsung/snow.c @@ -0,0 +1,190 @@ +/* + * ASoC machine driver for Snow boards + * + * 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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/of.h> +#include <linux/of_device.h> + +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> + +#include "i2s.h" + +#define FIN_PLL_RATE 24000000 + +static int snow_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int bfs, ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_U24: + case SNDRV_PCM_FORMAT_S24: + bfs = 48; + break; + case SNDRV_PCM_FORMAT_U16_LE: + case SNDRV_PCM_FORMAT_S16_LE: + bfs = 32; + break; + default: + return -EINVAL; + } + + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + FIN_PLL_RATE, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_CDCLK, + 0, SND_SOC_CLOCK_OUT); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, SAMSUNG_I2S_DIV_BCLK, bfs); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops snow_ops = { + .hw_params = snow_hw_params, +}; + +static int snow_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_sync(dapm); + + return 0; +} + +static struct snd_soc_dai_link snow_dai[] = { + { /* Playback DAI i/f */ + .name = "Playback", + .stream_name = "Playback", + .codec_dai_name = "HiFi", + .init = snow_init, + .ops = &snow_ops, + }, { /* Capture DAI i/f */ + .name = "Capture", + .stream_name = "Capture", + .codec_dai_name = "HiFi", + .ops = &snow_ops, + }, +}; + +static struct snd_soc_card snow_snd = { + .name = "Snow-I2S", + .dai_link = snow_dai, + .num_links = ARRAY_SIZE(snow_dai), +}; + +static int snow_driver_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &snow_snd; + struct device_node *i2s_node, *codec_node; + const char *name; + int i, ret; + + if (!pdev->dev.platform_data && !pdev->dev.of_node) { + dev_err(&pdev->dev, "No platform data supplied\n"); + return -EINVAL; + } + + name = of_get_property(pdev->dev.of_node, "card-name", NULL); + if (name) + card->name = name; + + i2s_node = of_parse_phandle(pdev->dev.of_node, + "samsung,i2s-controller", 0); + if (!i2s_node) { + dev_err(&pdev->dev, + "Property 'i2s-controller' missing or invalid\n"); + return -EINVAL; + } + + codec_node = of_parse_phandle(pdev->dev.of_node, + "samsung,audio-codec", 0); + if (!codec_node) { + dev_err(&pdev->dev, + "Property 'audio-codec' missing or invalid\n"); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(snow_dai); i++) { + snow_dai[i].codec_of_node = codec_node; + snow_dai[i].cpu_of_node = i2s_node; + snow_dai[i].platform_of_node = i2s_node; + } + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + return ret; + } + + return ret; +} + +static int snow_driver_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static const struct of_device_id snow_of_match[] = { + { .compatible = "google,snow-audio-max98090", }, + { .compatible = "google,snow-audio-max98095", }, + {}, +}; + +static struct platform_driver snow_driver = { + .driver = { + .name = "snow-audio", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + .of_match_table = snow_of_match, + }, + .probe = snow_driver_probe, + .remove = snow_driver_remove, +}; + +module_platform_driver(snow_driver); + +MODULE_DESCRIPTION("ALSA SoC Audio machine driver for Snow"); +MODULE_LICENSE("GPL"); -- 1.7.9.5

2 4

[alsa-devel] [PATCH] ASoC: Export devm_snd_soc_register_platform()
by Lars-Peter Clausen 23 Apr '14

23 Apr '14

devm_snd_soc_register_platform() is used in drivers which can be build as modules, so it needs to be exported to avoid linkers errors like: ERROR: "devm_snd_soc_register_platform" [sound/soc/omap/snd-soc-omap.ko] undefined! ERROR: "devm_snd_soc_register_platform" [sound/soc/davinci/snd-soc-davinci.ko] undefined! Fixes: 8931bf620 ("ASoC: Add resource managed snd_soc_register_platform()") Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/soc-devres.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index e94aa02..057e5ef 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -85,6 +85,7 @@ int devm_snd_soc_register_platform(struct device *dev, return ret; } +EXPORT_SYMBOL_GPL(devm_snd_soc_register_platform); static void devm_card_release(struct device *dev, void *res) { -- 1.8.0

2 1

[alsa-devel] [PATCH 0/3] ASoC: davinci: devm_snd_soc_register_platform and mcasp cleanup
by Peter Ujfalusi 22 Apr '14

22 Apr '14

Hi, Now that we have resource managed version of snd_soc_register_platform() convert the davinci ASoC code to use it. Clean the mcasp driver's init code while we are here so we can remove the platform driver's remove function since we now have managed resources only. Regards, Peter --- Peter Ujfalusi (3): ASoC: davinci-pcm: Convert to use devm_snd_soc_register_platform() ASoC: davinci-mcasp: Convert to use devm_snd_soc_register_component() ASoC: davinci-mcasp: Optimize pm_runtime usage and clean up the init code sound/soc/davinci/davinci-i2s.c | 1 - sound/soc/davinci/davinci-mcasp.c | 69 ++++++++++----------------------------- sound/soc/davinci/davinci-pcm.c | 8 +---- sound/soc/davinci/davinci-pcm.h | 4 --- sound/soc/davinci/davinci-vcif.c | 1 - 5 files changed, 19 insertions(+), 64 deletions(-) -- 1.9.2

2 8

[alsa-devel] [PATCH] ASoC: Intel: Fix audio crash due to negative address offset
by Jarkko Nikula 22 Apr '14

22 Apr '14

From: Wenkai Du <wenkai.du(a)intel.com> There were occasional ADSP crash during reboot testing: [ 11.883364] BUG: unable to handle kernel paging request at ffffc90121700000 [ 11.883380] IP: [<ffffffffc024d8bc>] sst_module_insert_fixed_block+0x24f/0x26d [snd_soc_sst_dsp] [ 11.883397] PGD 7800b067 PUD 0 [ 11.883405] Oops: 0002 [#1] SMP [ 11.886418] gsmi: Log Shutdown Reason 0x03 The virtual address, ffffc90121700000, was out of range. The virtual address is calculated by adding LPE base address with an offset: sst_memcpy32(dsp->addr.lpe + data->offset, data->data, data->size); The offset is calculated in sst_byt_parse_module, by subtraction of two virtual addresses dsp->addr.fw_ext and dsp->addr.lpe: block_data.offset = block->ram_offset + (dsp->addr.fw_ext - dsp->addr.lpe); These virtual addresses are assigned by kernel from ioremap: sst->addr.lpe = ioremap(pdata->lpe_base, pdata->lpe_size); sst->addr.fw_ext = ioremap(pdata->fw_base, pdata->fw_size); In current driver code, offset is defined as unsigned int32: struct sst_module_data { ... u32 offset; /* offset in FW file */ }; Most of the time kernel assigned virtual addresses with addr.fw_ext greater than addr.lpe. But sometimes it was the other way round. Fix the problem by declaring offset as signed int32_t. Signed-off-by: Wenkai Du <wenkai.du(a)intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst-dsp-priv.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-dsp-priv.h b/sound/soc/intel/sst-dsp-priv.h index fe8e81aad646..30ca14a6a835 100644 --- a/sound/soc/intel/sst-dsp-priv.h +++ b/sound/soc/intel/sst-dsp-priv.h @@ -136,7 +136,7 @@ struct sst_module_data { enum sst_data_type data_type; /* type of module data */ u32 size; /* size in bytes */ - u32 offset; /* offset in FW file */ + int32_t offset; /* offset in FW file */ u32 data_offset; /* offset in ADSP memory space */ void *data; /* module data */ }; -- 1.9.2

2 1

Re: [alsa-devel] For firewire drivers, reduce period time down to 2msec?
by Clemens Ladisch 22 Apr '14

22 Apr '14

Takashi Sakamoto wrote: > Currently I'm working for next patchsets and I'll post them in Friday. I should be healthy again by then. > The patchset also includes my idea to apply common PCM hw constraints to > dice/speakers(oxfw)/fireworks/bebob. Currently the constraints includes: > 1.Maximum bits for PCM sample is limited by 24. > 2.Minimum time for period is currently 2msec. > 3.In blocking mode, frames per period/buffer is aligned to SYT_INTERVAL > > I think you'll agree with 1 and 3. Yes. > So I want to discuss the second one beforehand. > > Currently dice/speakers driver apply 5msec for period time. This is just because of the fixed INTERRUPT_INTERVAL, plus lots of safety margin to reduce the relative jitter of the period interrupt. (If the interrupt interval were dynamic, it should be possible to reduce periods to about 0.25 ms.) > I want to reduce this to 2msec. The reason of this '2msec' is: > - Currently firewire-lib process 16 packets in one time. > - This equals to 2msec (= INTERRUPT_INTERVAL / 8000 * 1000) > - During this 2msec, when pcm_period_pointer passes through period > boundary, snd_pcm_period_elapsed() is scheduled with tasklet. > - But tasklet can execute one time even if the function is scheduled > several times before executing it. > - The packetizing process is enough fast so the function can be > scheduled before executing. > - As a result, snd_pcm_period_elapsed() can be called one time every 2msec. > > Actually my devices can work fine with period == 2msec or less than > 5msec, without any disadvantage. ALSA actually uses a fixed period _size_. When the period length is set to exactly 2 ms, but the 16 packets between two FireWire interrupts contain one sample less than needed, the ALSA period interupt is delayed until the next FireWire interrupt. In that case, Jack with two periods is likely to underrun. The minimum time should be increased so that at least one FireWire interrupt is guaranteed to happen for each period, even when there are rounding errors and clock differences. Regards, Clemens

2 1

[alsa-devel] Disabling DAPM from ASoC framework
by Santhosh N M 22 Apr '14

22 Apr '14

Hi, I'm a newbie to ALSA/ASoC framework. Lets say there is a requirement, where, the application (user space) takes the responsibility to switch ON/OFF the power components of the codec components (using ALSA Mixer Interface). In this case, is there a standard way to disable DAPM from ASoC? One way I see (and found to be working) is setting num_dapm_widgets=0 and num_dapm_routes=0 before invoking snd_soc_register_codec() Thanks in advance for your help. Best Regards, San

3 5

[alsa-devel] [PATCH] ASoC: omap-hdmi: Remove excess curly bracket
by Peter Ujfalusi 22 Apr '14

22 Apr '14

Fix the error added by commit: ASoC: omap-hdmi: Bind the platform driver to the dai driver when loading Reported-by: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/omap-hdmi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/omap/omap-hdmi.c b/sound/soc/omap/omap-hdmi.c index 32614b49653c..537a1ec8ad61 100644 --- a/sound/soc/omap/omap-hdmi.c +++ b/sound/soc/omap/omap-hdmi.c @@ -330,7 +330,6 @@ static int omap_hdmi_probe(struct platform_device *pdev) return omap_pcm_platform_register(&pdev->dev); } -} static int omap_hdmi_remove(struct platform_device *pdev) { -- 1.9.2

2 1

[alsa-devel] [PATCH v3 0/5] ASoC: Move IO and kcontrols to the component level
by Lars-Peter Clausen 22 Apr '14

22 Apr '14

The only change in v3 is the renaming of the component IO mutex from 'mutex' to 'io_mutex'. v2 cover letter: This series contains the remaining patches from the previous series that have not been applied yet. The series has been rebased ontop of asoc/for-next and depends on topic/sta350, topic/core, topic/dapm, topic/cache and topic/component. Some minor updates were made to the patches and three new patches: * One new user of snd_kcontrol_chip() showed up, the patch converts it to snd_soc_kcontrol_codec() * Change return type of snd_soc_write() from unsigned int to int * Remove ASoC level tracing Original cover letter: This series is the first step towards full componentisation of the ASoC core. It moves both the IO abstraction layers within ASoC as well as the standard set of kcontrols to the component level. This for example means we can get rid of constructs like if (w->codec) snd_soc_read(....) else if(w->platform) snd_soc_platform_read(...) Moving the kcontrols to the component level means we can use the same implementation also for other non-CODEC components. E.g. there seems to be an increasing amount of CPU components that have basic signal processing and things like volume controls etc. whose register layout is similar to those used in CODECs. Currently each CPU component driver re-implements these controls by hand. The first two patches introduce two new helper functions which hide the actual implementation on how the CODEC or platform struct that register a control can be obtained from the control. This means that when the actual implementation is changed only the two helper functions need to be updated and not every single driver. The patches that follow that are just cleanups removing unused IO stuff and move all IO functions to soc-io.c. The next step is to make platforms also components. And then finally first the IO abstraction layers in ASoC are unified at the component level and then on top of that the kcontrol helpers are moved to the component level. The series depends on quite a few topic branches related to changes to the core and cleanups for individual drivers. It is probably best to place it on top of asoc-v3.15-2. The patch that moves the kcontrols to the component level also has a runtime dependency on the not yet applied patches that move the ams-delta and mfld_machine controls to the card level. Lars-Peter Clausen (5): ASoC: Move IO abstraction to the component level ASoC: Move standard kcontrol helpers to the component level ASoC: Remove snd_soc_update_bits_locked() ASoC: dapm: Rename soc_widget_update_bits_locked() to soc_widget_update_bits() ASoC: Remove ASoC level IO tracing include/sound/soc-dapm.h | 1 + include/sound/soc.h | 51 +++++-- include/trace/events/asoc.h | 92 ------------- sound/soc/omap/omap-hdmi.c | 1 - sound/soc/soc-core.c | 285 +++++++++++++++++++++++++------------- sound/soc/soc-dapm.c | 91 ++---------- sound/soc/soc-io.c | 327 ++++++++++++++++++++++++++------------------ 7 files changed, 447 insertions(+), 401 deletions(-) -- 1.8.0

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[alsa-devel] [PATCH] ASoC: rt5645: Add codec driver
by Oder Chiou 22 Apr '14

22 Apr '14

This patch adds the Realtek ALC5645 codec driver. It is the base version that because the jack detect function is not implemented to it, the headphone and AMIC1 are not workable. We will fill up the further functions later. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5645.c | 2517 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5645.h | 2176 +++++++++++++++++++++++++++++++++++++++ 4 files changed, 4699 insertions(+) create mode 100644 sound/soc/codecs/rt5645.c create mode 100644 sound/soc/codecs/rt5645.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f0e8401..8113081 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -71,6 +71,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM512x_SPI if SPI_MASTER select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C + select SND_SOC_RT5645 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SIRF_AUDIO_CODEC @@ -396,6 +397,9 @@ config SND_SOC_RT5631 config SND_SOC_RT5640 tristate +config SND_SOC_RT5645 + tristate + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate "Freescale SGTL5000 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3c4d275..ce05f2a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -60,6 +60,7 @@ snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o +snd-soc-rt5645-objs := rt5645.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -211,6 +212,7 @@ obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o +obj-$(CONFIG_SND_SOC_RT5645) += snd-soc-rt5645.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c new file mode 100644 index 0000000..6948648 --- /dev/null +++ b/sound/soc/codecs/rt5645.c @@ -0,0 +1,2517 @@ +/* + * rt5645.c -- RT5645 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.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 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "rt5645.h" + +#define RT5645_DEVICE_ID 0x6308 + +#define RT5645_PR_RANGE_BASE (0xff + 1) +#define RT5645_PR_SPACING 0x100 + +#define RT5645_PR_BASE (RT5645_PR_RANGE_BASE + (0 * RT5645_PR_SPACING)) + +static const struct regmap_range_cfg rt5645_ranges[] = { + { + .name = "PR", + .range_min = RT5645_PR_BASE, + .range_max = RT5645_PR_BASE + 0xf8, + .selector_reg = RT5645_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT5645_PRIV_DATA, + .window_len = 0x1, + }, +}; + +static const struct reg_default init_list[] = { + {RT5645_PR_BASE + 0x3d, 0x3600}, +}; +#define RT5645_INIT_REG_LEN ARRAY_SIZE(init_list) + +static const struct reg_default rt5645_reg[] = { + { 0x00, 0x0000 }, + { 0x01, 0xc8c8 }, + { 0x02, 0xc8c8 }, + { 0x03, 0xc8c8 }, + { 0x0a, 0x0002 }, + { 0x0b, 0x2827 }, + { 0x0c, 0xe000 }, + { 0x0d, 0x0000 }, + { 0x0e, 0x0000 }, + { 0x0f, 0x0808 }, + { 0x14, 0x3333 }, + { 0x16, 0x4b00 }, + { 0x18, 0x018b }, + { 0x19, 0xafaf }, + { 0x1a, 0xafaf }, + { 0x1b, 0x0001 }, + { 0x1c, 0x2f2f }, + { 0x1d, 0x2f2f }, + { 0x1e, 0x0000 }, + { 0x20, 0x0000 }, + { 0x27, 0x7060 }, + { 0x28, 0x7070 }, + { 0x29, 0x8080 }, + { 0x2a, 0x5656 }, + { 0x2b, 0x5454 }, + { 0x2c, 0xaaa0 }, + { 0x2f, 0x1002 }, + { 0x31, 0x5000 }, + { 0x32, 0x0000 }, + { 0x33, 0x0000 }, + { 0x34, 0x0000 }, + { 0x35, 0x0000 }, + { 0x3b, 0x0000 }, + { 0x3c, 0x007f }, + { 0x3d, 0x0000 }, + { 0x3e, 0x007f }, + { 0x3f, 0x0000 }, + { 0x40, 0x001f }, + { 0x41, 0x0000 }, + { 0x42, 0x001f }, + { 0x45, 0x6000 }, + { 0x46, 0x003e }, + { 0x47, 0x003e }, + { 0x48, 0xf807 }, + { 0x4a, 0x0004 }, + { 0x4d, 0x0000 }, + { 0x4e, 0x0000 }, + { 0x4f, 0x01ff }, + { 0x50, 0x0000 }, + { 0x51, 0x0000 }, + { 0x52, 0x01ff }, + { 0x53, 0xf000 }, + { 0x56, 0x0111 }, + { 0x57, 0x0064 }, + { 0x58, 0xef0e }, + { 0x59, 0xf0f0 }, + { 0x5a, 0xef0e }, + { 0x5b, 0xf0f0 }, + { 0x5c, 0xef0e }, + { 0x5d, 0xf0f0 }, + { 0x5e, 0xf000 }, + { 0x5f, 0x0000 }, + { 0x61, 0x0300 }, + { 0x62, 0x0000 }, + { 0x63, 0x00c2 }, + { 0x64, 0x0000 }, + { 0x65, 0x0000 }, + { 0x66, 0x0000 }, + { 0x6a, 0x0000 }, + { 0x6c, 0x0aaa }, + { 0x70, 0x8000 }, + { 0x71, 0x8000 }, + { 0x72, 0x8000 }, + { 0x73, 0x7770 }, + { 0x74, 0x3e00 }, + { 0x75, 0x2409 }, + { 0x76, 0x000a }, + { 0x77, 0x0c00 }, + { 0x78, 0x0000 }, + { 0x80, 0x0000 }, + { 0x81, 0x0000 }, + { 0x82, 0x0000 }, + { 0x83, 0x0000 }, + { 0x84, 0x0000 }, + { 0x85, 0x0000 }, + { 0x8a, 0x0000 }, + { 0x8e, 0x0004 }, + { 0x8f, 0x1100 }, + { 0x90, 0x0646 }, + { 0x91, 0x0c06 }, + { 0x93, 0x0000 }, + { 0x94, 0x0200 }, + { 0x95, 0x0000 }, + { 0x9a, 0x2184 }, + { 0x9b, 0x010a }, + { 0x9c, 0x0aea }, + { 0x9d, 0x000c }, + { 0x9e, 0x0400 }, + { 0xa0, 0xa0a8 }, + { 0xa1, 0x0059 }, + { 0xa2, 0x0001 }, + { 0xae, 0x6000 }, + { 0xaf, 0x0000 }, + { 0xb0, 0x6000 }, + { 0xb1, 0x0000 }, + { 0xb2, 0x0000 }, + { 0xb3, 0x001f }, + { 0xb4, 0x020c }, + { 0xb5, 0x1f00 }, + { 0xb6, 0x0000 }, + { 0xbb, 0x0000 }, + { 0xbc, 0x0000 }, + { 0xbd, 0x0000 }, + { 0xbe, 0x0000 }, + { 0xbf, 0x3100 }, + { 0xc0, 0x0000 }, + { 0xc1, 0x0000 }, + { 0xc2, 0x0000 }, + { 0xc3, 0x2000 }, + { 0xcd, 0x0000 }, + { 0xce, 0x0000 }, + { 0xcf, 0x1813 }, + { 0xd0, 0x0690 }, + { 0xd1, 0x1c17 }, + { 0xd3, 0xb320 }, + { 0xd4, 0x0000 }, + { 0xd6, 0x0400 }, + { 0xd9, 0x0809 }, + { 0xda, 0x0000 }, + { 0xdb, 0x0003 }, + { 0xdc, 0x0049 }, + { 0xdd, 0x001b }, + { 0xe6, 0x8000 }, + { 0xe7, 0x0200 }, + { 0xec, 0xb300 }, + { 0xed, 0x0000 }, + { 0xf0, 0x001f }, + { 0xf1, 0x020c }, + { 0xf2, 0x1f00 }, + { 0xf3, 0x0000 }, + { 0xf4, 0x4000 }, + { 0xf8, 0x0000 }, + { 0xf9, 0x0000 }, + { 0xfa, 0x2060 }, + { 0xfb, 0x4040 }, + { 0xfc, 0x0000 }, + { 0xfd, 0x0002 }, + { 0xfe, 0x10ec }, + { 0xff, 0x6308 }, +}; + +static int rt5645_reset(struct snd_soc_codec *codec) +{ + return snd_soc_write(codec, RT5645_RESET, 0); +} + +static bool rt5645_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5645_ranges); i++) + if ((reg >= rt5645_ranges[i].window_start && + reg <= rt5645_ranges[i].window_start + + rt5645_ranges[i].window_len) || + (reg >= rt5645_ranges[i].range_min && + reg <= rt5645_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5645_RESET: + case RT5645_PRIV_DATA: + case RT5645_IN1_CTRL1: + case RT5645_IN1_CTRL2: + case RT5645_IN1_CTRL3: + case RT5645_A_JD_CTRL1: + case RT5645_ADC_EQ_CTRL1: + case RT5645_EQ_CTRL1: + case RT5645_ALC_CTRL_1: + case RT5645_IRQ_CTRL2: + case RT5645_IRQ_CTRL3: + case RT5645_INT_IRQ_ST: + case RT5645_IL_CMD: + case RT5645_VENDOR_ID: + case RT5645_VENDOR_ID1: + case RT5645_VENDOR_ID2: + return 1; + default: + return 0; + } +} + +static bool rt5645_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5645_ranges); i++) + if ((reg >= rt5645_ranges[i].window_start && + reg <= rt5645_ranges[i].window_start + + rt5645_ranges[i].window_len) || + (reg >= rt5645_ranges[i].range_min && + reg <= rt5645_ranges[i].range_max)) + return true; + + switch (reg) { + case RT5645_RESET: + case RT5645_SPK_VOL: + case RT5645_HP_VOL: + case RT5645_LOUT1: + case RT5645_IN1_CTRL1: + case RT5645_IN1_CTRL2: + case RT5645_IN1_CTRL3: + case RT5645_IN2_CTRL: + case RT5645_INL1_INR1_VOL: + case RT5645_SPK_FUNC_LIM: + case RT5645_ADJ_HPF_CTRL: + case RT5645_DAC1_DIG_VOL: + case RT5645_DAC2_DIG_VOL: + case RT5645_DAC_CTRL: + case RT5645_STO1_ADC_DIG_VOL: + case RT5645_MONO_ADC_DIG_VOL: + case RT5645_ADC_BST_VOL1: + case RT5645_ADC_BST_VOL2: + case RT5645_STO1_ADC_MIXER: + case RT5645_MONO_ADC_MIXER: + case RT5645_AD_DA_MIXER: + case RT5645_STO_DAC_MIXER: + case RT5645_MONO_DAC_MIXER: + case RT5645_DIG_MIXER: + case RT5645_DIG_INF1_DATA: + case RT5645_PDM_OUT_CTRL: + case RT5645_REC_L1_MIXER: + case RT5645_REC_L2_MIXER: + case RT5645_REC_R1_MIXER: + case RT5645_REC_R2_MIXER: + case RT5645_HPMIXL_CTRL: + case RT5645_HPOMIXL_CTRL: + case RT5645_HPMIXR_CTRL: + case RT5645_HPOMIXR_CTRL: + case RT5645_HPO_MIXER: + case RT5645_SPK_L_MIXER: + case RT5645_SPK_R_MIXER: + case RT5645_SPO_MIXER: + case RT5645_SPO_CLSD_RATIO: + case RT5645_OUT_L1_MIXER: + case RT5645_OUT_R1_MIXER: + case RT5645_OUT_L_GAIN1: + case RT5645_OUT_L_GAIN2: + case RT5645_OUT_R_GAIN1: + case RT5645_OUT_R_GAIN2: + case RT5645_LOUT_MIXER: + case RT5645_HAPTIC_CTRL1: + case RT5645_HAPTIC_CTRL2: + case RT5645_HAPTIC_CTRL3: + case RT5645_HAPTIC_CTRL4: + case RT5645_HAPTIC_CTRL5: + case RT5645_HAPTIC_CTRL6: + case RT5645_HAPTIC_CTRL7: + case RT5645_HAPTIC_CTRL8: + case RT5645_HAPTIC_CTRL9: + case RT5645_HAPTIC_CTRL10: + case RT5645_PWR_DIG1: + case RT5645_PWR_DIG2: + case RT5645_PWR_ANLG1: + case RT5645_PWR_ANLG2: + case RT5645_PWR_MIXER: + case RT5645_PWR_VOL: + case RT5645_PRIV_INDEX: + case RT5645_PRIV_DATA: + case RT5645_I2S1_SDP: + case RT5645_I2S2_SDP: + case RT5645_ADDA_CLK1: + case RT5645_ADDA_CLK2: + case RT5645_DMIC_CTRL1: + case RT5645_DMIC_CTRL2: + case RT5645_TDM_CTRL_1: + case RT5645_TDM_CTRL_2: + case RT5645_GLB_CLK: + case RT5645_PLL_CTRL1: + case RT5645_PLL_CTRL2: + case RT5645_ASRC_1: + case RT5645_ASRC_2: + case RT5645_ASRC_3: + case RT5645_ASRC_4: + case RT5645_DEPOP_M1: + case RT5645_DEPOP_M2: + case RT5645_DEPOP_M3: + case RT5645_MICBIAS: + case RT5645_A_JD_CTRL1: + case RT5645_VAD_CTRL4: + case RT5645_CLSD_OUT_CTRL: + case RT5645_ADC_EQ_CTRL1: + case RT5645_ADC_EQ_CTRL2: + case RT5645_EQ_CTRL1: + case RT5645_EQ_CTRL2: + case RT5645_ALC_CTRL_1: + case RT5645_ALC_CTRL_2: + case RT5645_ALC_CTRL_3: + case RT5645_ALC_CTRL_4: + case RT5645_ALC_CTRL_5: + case RT5645_JD_CTRL: + case RT5645_IRQ_CTRL1: + case RT5645_IRQ_CTRL2: + case RT5645_IRQ_CTRL3: + case RT5645_INT_IRQ_ST: + case RT5645_GPIO_CTRL1: + case RT5645_GPIO_CTRL2: + case RT5645_GPIO_CTRL3: + case RT5645_BASS_BACK: + case RT5645_MP3_PLUS1: + case RT5645_MP3_PLUS2: + case RT5645_ADJ_HPF1: + case RT5645_ADJ_HPF2: + case RT5645_HP_CALIB_AMP_DET: + case RT5645_SV_ZCD1: + case RT5645_SV_ZCD2: + case RT5645_IL_CMD: + case RT5645_IL_CMD2: + case RT5645_IL_CMD3: + case RT5645_DRC1_HL_CTRL1: + case RT5645_DRC2_HL_CTRL1: + case RT5645_ADC_MONO_HP_CTRL1: + case RT5645_ADC_MONO_HP_CTRL2: + case RT5645_DRC2_CTRL1: + case RT5645_DRC2_CTRL2: + case RT5645_DRC2_CTRL3: + case RT5645_DRC2_CTRL4: + case RT5645_DRC2_CTRL5: + case RT5645_JD_CTRL3: + case RT5645_JD_CTRL4: + case RT5645_GEN_CTRL1: + case RT5645_GEN_CTRL2: + case RT5645_GEN_CTRL3: + case RT5645_VENDOR_ID: + case RT5645_VENDOR_ID1: + case RT5645_VENDOR_ID2: + return 1; + default: + return 0; + } +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static unsigned int bst_tlv[] = { + TLV_DB_RANGE_HEAD(7), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), +}; + +/* IN1/IN2 Input Type */ +static const char * const rt5645_input_mode[] = { + "Single ended", "Differential" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_in2_mode_enum, RT5645_IN2_CTRL, + RT5645_IN_SFT2, rt5645_input_mode); + +static const char * const rt5645_tdm_data_swap_select[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_slot0_1_enum, + RT5645_TDM_CTRL_1, 6, rt5645_tdm_data_swap_select); + +static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_slot2_3_enum, + RT5645_TDM_CTRL_1, 4, rt5645_tdm_data_swap_select); + +static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_slot4_5_enum, + RT5645_TDM_CTRL_1, 2, rt5645_tdm_data_swap_select); + +static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_slot6_7_enum, + RT5645_TDM_CTRL_1, 0, rt5645_tdm_data_swap_select); + +static const char * const rt5645_tdm_adc_data_select[] = { + "1/2/R", "2/1/R", "R/1/2", "R/2/1" +}; + +static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_sel_enum, + RT5645_TDM_CTRL_1, 8, + rt5645_tdm_adc_data_select); + +static const struct snd_kcontrol_new rt5645_snd_controls[] = { + /* Speaker Output Volume */ + SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, + RT5645_VOL_L_SFT, RT5645_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("Speaker Playback Volume", RT5645_SPK_VOL, + RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv), + + /* Headphone Output Volume */ + SOC_DOUBLE("HP Channel Switch", RT5645_HP_VOL, + RT5645_VOL_L_SFT, RT5645_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("HP Playback Volume", RT5645_HP_VOL, + RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv), + + /* OUTPUT Control */ + SOC_DOUBLE("OUT Playback Switch", RT5645_LOUT1, + RT5645_L_MUTE_SFT, RT5645_R_MUTE_SFT, 1, 1), + SOC_DOUBLE("OUT Channel Switch", RT5645_LOUT1, + RT5645_VOL_L_SFT, RT5645_VOL_R_SFT, 1, 1), + SOC_DOUBLE_TLV("OUT Playback Volume", RT5645_LOUT1, + RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 39, 1, out_vol_tlv), + + /* DAC Digital Volume */ + SOC_DOUBLE("DAC2 Playback Switch", RT5645_DAC_CTRL, + RT5645_M_DAC_L2_VOL_SFT, RT5645_M_DAC_R2_VOL_SFT, 1, 1), + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5645_DAC1_DIG_VOL, + RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 175, 0, dac_vol_tlv), + SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5645_DAC2_DIG_VOL, + RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 175, 0, dac_vol_tlv), + + /* IN1/IN2 Control */ + SOC_SINGLE_TLV("IN1 Boost", RT5645_IN1_CTRL1, + RT5645_BST_SFT1, 8, 0, bst_tlv), + SOC_ENUM("IN2 Mode Control", rt5645_in2_mode_enum), + SOC_SINGLE_TLV("IN2 Boost", RT5645_IN2_CTRL, + RT5645_BST_SFT2, 8, 0, bst_tlv), + + /* INL/INR Volume Control */ + SOC_DOUBLE_TLV("IN Capture Volume", RT5645_INL1_INR1_VOL, + RT5645_INL_VOL_SFT, RT5645_INR_VOL_SFT, 31, 1, in_vol_tlv), + + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5645_STO1_ADC_DIG_VOL, + RT5645_L_MUTE_SFT, RT5645_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5645_STO1_ADC_DIG_VOL, + RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 127, 0, adc_vol_tlv), + SOC_DOUBLE("Mono ADC Capture Switch", RT5645_MONO_ADC_DIG_VOL, + RT5645_L_MUTE_SFT, RT5645_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5645_MONO_ADC_DIG_VOL, + RT5645_L_VOL_SFT, RT5645_R_VOL_SFT, 127, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain", RT5645_ADC_BST_VOL1, + RT5645_STO1_ADC_L_BST_SFT, RT5645_STO1_ADC_R_BST_SFT, 3, 0, + adc_bst_tlv), + SOC_DOUBLE_TLV("STO2 ADC Boost Gain", RT5645_ADC_BST_VOL1, + RT5645_STO2_ADC_L_BST_SFT, RT5645_STO2_ADC_R_BST_SFT, 3, 0, + adc_bst_tlv), + + /* I2S2 function select */ + SOC_SINGLE("I2S2 Func Switch", RT5645_GPIO_CTRL1, RT5645_I2S2_SEL_SFT, + 1, 1), + + /* TDM */ + SOC_ENUM("TDM Adc Slot0 1 Data", rt5645_tdm_adc_slot0_1_enum), + SOC_ENUM("TDM Adc Slot2 3 Data", rt5645_tdm_adc_slot2_3_enum), + SOC_ENUM("TDM Adc Slot4 5 Data", rt5645_tdm_adc_slot4_5_enum), + SOC_ENUM("TDM Adc Slot6 7 Data", rt5645_tdm_adc_slot6_7_enum), + SOC_ENUM("TDM IF1 ADC DATA Sel", rt5645_tdm_adc_sel_enum), + SOC_SINGLE("TDM IF1_DAC1_L Sel", RT5645_TDM_CTRL_3, 12, 7, 0), + SOC_SINGLE("TDM IF1_DAC1_R Sel", RT5645_TDM_CTRL_3, 8, 7, 0), + SOC_SINGLE("TDM IF1_DAC2_L Sel", RT5645_TDM_CTRL_3, 4, 7, 0), + SOC_SINGLE("TDM IF1_DAC2_R Sel", RT5645_TDM_CTRL_3, 0, 7, 0), +}; + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + int div[] = {2, 3, 4, 6, 8, 12}; + int idx = -EINVAL, i; + int rate, red, bound, temp; + + rate = rt5645->sysclk; + red = 3000000 * 12; + for (i = 0; i < ARRAY_SIZE(div); i++) { + bound = div[i] * 3000000; + if (rate > bound) + continue; + temp = bound - rate; + if (temp < red) { + red = temp; + idx = i; + } + } + + if (idx < 0) + dev_err(codec->dev, "Failed to set DMIC clock\n"); + else + snd_soc_update_bits(codec, RT5645_DMIC_CTRL1, + RT5645_DMIC_CLK_MASK, idx << RT5645_DMIC_CLK_SFT); + return idx; +} + +static int rt5645_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, RT5645_GPIO_CTRL1, + RT5645_GP2_PIN_MASK, RT5645_GP2_PIN_DMIC1_SCL); + snd_soc_update_bits(codec, RT5645_DMIC1, RT5645_DMIC_1_DP_MASK, + RT5645_DMIC_1_DP_IN2N); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5645_set_dmic2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_update_bits(codec, RT5645_GPIO_CTRL1, + RT5645_GP2_PIN_MASK, RT5645_GP2_PIN_DMIC1_SCL); + snd_soc_update_bits(codec, RT5645_DMIC1, RT5645_DMIC_2_DP_MASK, + RT5645_DMIC_2_DP_IN2P); + break; + + default: + return 0; + } + + return 0; +} + +static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + + val = snd_soc_read(source->codec, RT5645_GLB_CLK); + val &= RT5645_SCLK_SRC_MASK; + if (val == RT5645_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5645_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5645_STO1_ADC_MIXER, + RT5645_M_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5645_STO1_ADC_MIXER, + RT5645_M_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5645_STO1_ADC_MIXER, + RT5645_M_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5645_STO1_ADC_MIXER, + RT5645_M_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_mono_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5645_MONO_ADC_MIXER, + RT5645_M_MONO_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5645_MONO_ADC_MIXER, + RT5645_M_MONO_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_mono_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5645_MONO_ADC_MIXER, + RT5645_M_MONO_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5645_MONO_ADC_MIXER, + RT5645_M_MONO_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5645_AD_DA_MIXER, + RT5645_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5645_AD_DA_MIXER, + RT5645_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5645_AD_DA_MIXER, + RT5645_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5645_AD_DA_MIXER, + RT5645_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5645_STO_DAC_MIXER, + RT5645_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5645_STO_DAC_MIXER, + RT5645_M_DAC_L2_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5645_STO_DAC_MIXER, + RT5645_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5645_STO_DAC_MIXER, + RT5645_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5645_STO_DAC_MIXER, + RT5645_M_DAC_R2_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5645_STO_DAC_MIXER, + RT5645_M_DAC_L1_STO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_mono_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5645_MONO_DAC_MIXER, + RT5645_M_DAC_L1_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5645_MONO_DAC_MIXER, + RT5645_M_DAC_L2_MONO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5645_MONO_DAC_MIXER, + RT5645_M_DAC_R2_MONO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_mono_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5645_MONO_DAC_MIXER, + RT5645_M_DAC_R1_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5645_MONO_DAC_MIXER, + RT5645_M_DAC_R2_MONO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5645_MONO_DAC_MIXER, + RT5645_M_DAC_L2_MONO_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_dig_l_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix L Switch", RT5645_DIG_MIXER, + RT5645_M_STO_L_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5645_DIG_MIXER, + RT5645_M_DAC_L2_DAC_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5645_DIG_MIXER, + RT5645_M_DAC_R2_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_dig_r_mix[] = { + SOC_DAPM_SINGLE("Sto DAC Mix R Switch", RT5645_DIG_MIXER, + RT5645_M_STO_R_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5645_DIG_MIXER, + RT5645_M_DAC_R2_DAC_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5645_DIG_MIXER, + RT5645_M_DAC_L2_DAC_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5645_rec_l_mix[] = { + SOC_DAPM_SINGLE("HPOL Switch", RT5645_REC_L2_MIXER, + RT5645_M_HP_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5645_REC_L2_MIXER, + RT5645_M_IN_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5645_REC_L2_MIXER, + RT5645_M_BST2_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5645_REC_L2_MIXER, + RT5645_M_BST1_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXL Switch", RT5645_REC_L2_MIXER, + RT5645_M_OM_L_RM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_rec_r_mix[] = { + SOC_DAPM_SINGLE("HPOR Switch", RT5645_REC_R2_MIXER, + RT5645_M_HP_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5645_REC_R2_MIXER, + RT5645_M_IN_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5645_REC_R2_MIXER, + RT5645_M_BST2_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5645_REC_R2_MIXER, + RT5645_M_BST1_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("OUT MIXR Switch", RT5645_REC_R2_MIXER, + RT5645_M_OM_R_RM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_spk_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5645_SPK_L_MIXER, + RT5645_M_DAC_L1_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5645_SPK_L_MIXER, + RT5645_M_DAC_L2_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5645_SPK_L_MIXER, + RT5645_M_IN_L_SM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5645_SPK_L_MIXER, + RT5645_M_BST1_L_SM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_spk_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5645_SPK_R_MIXER, + RT5645_M_DAC_R1_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5645_SPK_R_MIXER, + RT5645_M_DAC_R2_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5645_SPK_R_MIXER, + RT5645_M_IN_R_SM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5645_SPK_R_MIXER, + RT5645_M_BST2_R_SM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_out_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", RT5645_OUT_L1_MIXER, + RT5645_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5645_OUT_L1_MIXER, + RT5645_M_IN_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5645_OUT_L1_MIXER, + RT5645_M_DAC_L2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5645_OUT_L1_MIXER, + RT5645_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5645_OUT_R1_MIXER, + RT5645_M_BST2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5645_OUT_R1_MIXER, + RT5645_M_IN_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5645_OUT_R1_MIXER, + RT5645_M_DAC_R2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5645_OUT_R1_MIXER, + RT5645_M_DAC_R1_OM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_spo_l_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5645_SPO_MIXER, + RT5645_M_DAC_R1_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5645_SPO_MIXER, + RT5645_M_DAC_L1_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL R Switch", RT5645_SPO_MIXER, + RT5645_M_SV_R_SPM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL L Switch", RT5645_SPO_MIXER, + RT5645_M_SV_L_SPM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_spo_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5645_SPO_MIXER, + RT5645_M_DAC_R1_SPM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("SPKVOL R Switch", RT5645_SPO_MIXER, + RT5645_M_SV_R_SPM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_hpo_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5645_HPO_MIXER, + RT5645_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPVOL Switch", RT5645_HPO_MIXER, + RT5645_M_HPVOL_HM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_hpvoll_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5645_HPOMIXL_CTRL, + RT5645_M_DAC1_HV_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC2 Switch", RT5645_HPOMIXL_CTRL, + RT5645_M_DAC2_HV_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5645_HPOMIXL_CTRL, + RT5645_M_IN_HV_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5645_HPOMIXL_CTRL, + RT5645_M_BST1_HV_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_hpvolr_mix[] = { + SOC_DAPM_SINGLE("DAC1 Switch", RT5645_HPOMIXR_CTRL, + RT5645_M_DAC1_HV_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC2 Switch", RT5645_HPOMIXR_CTRL, + RT5645_M_DAC2_HV_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5645_HPOMIXR_CTRL, + RT5645_M_IN_HV_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5645_HPOMIXR_CTRL, + RT5645_M_BST2_HV_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5645_lout_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5645_LOUT_MIXER, + RT5645_M_DAC_L1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5645_LOUT_MIXER, + RT5645_M_DAC_R1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX L Switch", RT5645_LOUT_MIXER, + RT5645_M_OV_L_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTMIX R Switch", RT5645_LOUT_MIXER, + RT5645_M_OV_R_LM_SFT, 1, 1), +}; + +/*DAC1 L/R source*/ /* MX-29 [9:8] [11:10] */ +static const char * const rt5645_dac1_src[] = { + "IF1 DAC", "IF2 DAC", "IF3 DAC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_dac1l_enum, RT5645_AD_DA_MIXER, + RT5645_DAC1_L_SEL_SFT, rt5645_dac1_src); + +static const struct snd_kcontrol_new rt5645_dac1l_mux = + SOC_DAPM_ENUM("DAC1 L source", rt5645_dac1l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5645_dac1r_enum, RT5645_AD_DA_MIXER, + RT5645_DAC1_R_SEL_SFT, rt5645_dac1_src); + +static const struct snd_kcontrol_new rt5645_dac1r_mux = + SOC_DAPM_ENUM("DAC1 R source", rt5645_dac1r_enum); + +/*DAC2 L/R source*/ /* MX-1B [6:4] [2:0] */ +static const char * const rt5645_dac12_src[] = { + "IF1 DAC", "IF2 DAC", "IF3 DAC", "Mono ADC", "VAD_ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_dac2l_enum, RT5645_DAC_CTRL, + RT5645_DAC2_L_SEL_SFT, rt5645_dac12_src); + +static const struct snd_kcontrol_new rt5645_dac_l2_mux = + SOC_DAPM_ENUM("DAC2 L source", rt5645_dac2l_enum); + +static const char * const rt5645_dacr2_src[] = { + "IF1 DAC", "IF2 DAC", "IF3 DAC", "Mono ADC", "Haptic" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_dac2r_enum, RT5645_DAC_CTRL, + RT5645_DAC2_R_SEL_SFT, rt5645_dacr2_src); + +static const struct snd_kcontrol_new rt5645_dac_r2_mux = + SOC_DAPM_ENUM("DAC2 R source", rt5645_dac2r_enum); + + +/* INL/R source */ +static const char * const rt5645_inl_src[] = { + "IN2P", "MonoP" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_inl_enum, RT5645_INL1_INR1_VOL, + RT5645_INL_SEL_SFT, rt5645_inl_src); + +static const struct snd_kcontrol_new rt5645_inl_mux = + SOC_DAPM_ENUM("INL source", rt5645_inl_enum); + +static const char * const rt5645_inr_src[] = { + "IN2N", "MonoN" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_inr_enum, RT5645_INL1_INR1_VOL, + RT5645_INR_SEL_SFT, rt5645_inr_src); + +static const struct snd_kcontrol_new rt5645_inr_mux = + SOC_DAPM_ENUM("INR source", rt5645_inr_enum); + +/* Stereo1 ADC source */ +/* MX-27 [12] */ +static const char * const rt5645_stereo_adc1_src[] = { + "DAC MIX", "ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_stereo1_adc1_enum, RT5645_STO1_ADC_MIXER, + RT5645_ADC_1_SRC_SFT, rt5645_stereo_adc1_src); + +static const struct snd_kcontrol_new rt5645_sto_adc1_mux = + SOC_DAPM_ENUM("Stereo1 ADC1 Mux", rt5645_stereo1_adc1_enum); + +/* MX-27 [11] */ +static const char * const rt5645_stereo_adc2_src[] = { + "DAC MIX", "DMIC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_stereo1_adc2_enum, RT5645_STO1_ADC_MIXER, + RT5645_ADC_2_SRC_SFT, rt5645_stereo_adc2_src); + +static const struct snd_kcontrol_new rt5645_sto_adc2_mux = + SOC_DAPM_ENUM("Stereo1 ADC2 Mux", rt5645_stereo1_adc2_enum); + +/* MX-27 [8] */ +static const char * const rt5645_stereo_dmic_src[] = { + "DMIC1", "DMIC2" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_stereo1_dmic_enum, RT5645_STO1_ADC_MIXER, + RT5645_DMIC_SRC_SFT, rt5645_stereo_dmic_src); + +static const struct snd_kcontrol_new rt5645_sto1_dmic_mux = + SOC_DAPM_ENUM("Stereo1 DMIC source", rt5645_stereo1_dmic_enum); + +/* Mono ADC source */ +/* MX-28 [12] */ +static const char * const rt5645_mono_adc_l1_src[] = { + "Mono DAC MIXL", "ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_mono_adc_l1_enum, RT5645_MONO_ADC_MIXER, + RT5645_MONO_ADC_L1_SRC_SFT, rt5645_mono_adc_l1_src); + +static const struct snd_kcontrol_new rt5645_mono_adc_l1_mux = + SOC_DAPM_ENUM("Mono ADC1 left source", rt5645_mono_adc_l1_enum); +/* MX-28 [11] */ +static const char * const rt5645_mono_adc_l2_src[] = { + "Mono DAC MIXL", "DMIC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_mono_adc_l2_enum, RT5645_MONO_ADC_MIXER, + RT5645_MONO_ADC_L2_SRC_SFT, rt5645_mono_adc_l2_src); + +static const struct snd_kcontrol_new rt5645_mono_adc_l2_mux = + SOC_DAPM_ENUM("Mono ADC2 left source", rt5645_mono_adc_l2_enum); + +/* MX-28 [8] */ +static const char * const rt5645_mono_dmic_src[] = { + "DMIC1", "DMIC2" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_mono_dmic_l_enum, RT5645_MONO_ADC_MIXER, + RT5645_MONO_DMIC_L_SRC_SFT, rt5645_mono_dmic_src); + +static const struct snd_kcontrol_new rt5645_mono_dmic_l_mux = + SOC_DAPM_ENUM("Mono DMIC left source", rt5645_mono_dmic_l_enum); +/* MX-28 [1:0] */ +static SOC_ENUM_SINGLE_DECL( + rt5645_mono_dmic_r_enum, RT5645_MONO_ADC_MIXER, + RT5645_MONO_DMIC_R_SRC_SFT, rt5645_mono_dmic_src); + +static const struct snd_kcontrol_new rt5645_mono_dmic_r_mux = + SOC_DAPM_ENUM("Mono DMIC Right source", rt5645_mono_dmic_r_enum); +/* MX-28 [4] */ +static const char * const rt5645_mono_adc_r1_src[] = { + "Mono DAC MIXR", "ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_mono_adc_r1_enum, RT5645_MONO_ADC_MIXER, + RT5645_MONO_ADC_R1_SRC_SFT, rt5645_mono_adc_r1_src); + +static const struct snd_kcontrol_new rt5645_mono_adc_r1_mux = + SOC_DAPM_ENUM("Mono ADC1 right source", rt5645_mono_adc_r1_enum); +/* MX-28 [3] */ +static const char * const rt5645_mono_adc_r2_src[] = { + "Mono DAC MIXR", "DMIC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_mono_adc_r2_enum, RT5645_MONO_ADC_MIXER, + RT5645_MONO_ADC_R2_SRC_SFT, rt5645_mono_adc_r2_src); + +static const struct snd_kcontrol_new rt5645_mono_adc_r2_mux = + SOC_DAPM_ENUM("Mono ADC2 right source", rt5645_mono_adc_r2_enum); + +/* MX-77 [9:8] */ +static const char * const rt5645_if1_adc_in_src[] = { + "IF_ADC1", "IF_ADC2", "VAD_ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_if1_adc_in_enum, RT5645_TDM_CTRL_1, + RT5645_IF1_ADC_IN_SFT, rt5645_if1_adc_in_src); + +static const struct snd_kcontrol_new rt5645_if1_adc_in_mux = + SOC_DAPM_ENUM("IF1 ADC IN source", rt5645_if1_adc_in_enum); + +/* MX-2F [13:12] */ +static const char * const rt5645_if2_adc_in_src[] = { + "IF_ADC1", "IF_ADC2", "VAD_ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_if2_adc_in_enum, RT5645_DIG_INF1_DATA, + RT5645_IF2_ADC_IN_SFT, rt5645_if2_adc_in_src); + +static const struct snd_kcontrol_new rt5645_if2_adc_in_mux = + SOC_DAPM_ENUM("IF2 ADC IN source", rt5645_if2_adc_in_enum); + +/* MX-2F [1:0] */ +static const char * const rt5645_if3_adc_in_src[] = { + "IF_ADC1", "IF_ADC2", "VAD_ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_if3_adc_in_enum, RT5645_DIG_INF1_DATA, + RT5645_IF3_ADC_IN_SFT, rt5645_if3_adc_in_src); + +static const struct snd_kcontrol_new rt5645_if3_adc_in_mux = + SOC_DAPM_ENUM("IF3 ADC IN source", rt5645_if3_adc_in_enum); + +/* MX-31 [15] [13] [11] [9] */ +static const char * const rt5645_pdm_src[] = { + "Mono DAC", "Stereo DAC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_pdm1_l_enum, RT5645_PDM_OUT_CTRL, + RT5645_PDM1_L_SFT, rt5645_pdm_src); + +static const struct snd_kcontrol_new rt5645_pdm1_l_mux = + SOC_DAPM_ENUM("PDM1 L source", rt5645_pdm1_l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5645_pdm1_r_enum, RT5645_PDM_OUT_CTRL, + RT5645_PDM1_R_SFT, rt5645_pdm_src); + +static const struct snd_kcontrol_new rt5645_pdm1_r_mux = + SOC_DAPM_ENUM("PDM1 R source", rt5645_pdm1_r_enum); + +/* MX-9D [9:8] */ +static const char * const rt5645_vad_adc_src[] = { + "Sto1 ADC L", "Mono ADC L", "Mono ADC R" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5645_vad_adc_enum, RT5645_VAD_CTRL4, + RT5645_VAD_SEL_SFT, rt5645_vad_adc_src); + +static const struct snd_kcontrol_new rt5645_vad_adc_mux = + SOC_DAPM_ENUM("VAD ADC source", rt5645_vad_adc_enum); + +static const struct snd_kcontrol_new spk_l_vol_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5645_SPK_VOL, + RT5645_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new spk_r_vol_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5645_SPK_VOL, + RT5645_R_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_l_vol_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5645_HP_VOL, + RT5645_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hp_r_vol_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5645_HP_VOL, + + RT5645_R_MUTE_SFT, 1, 1); +static void hp_amp_power(struct snd_soc_codec *codec, int on) +{ + static int hp_amp_power_count; + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + + if (on) { + if (hp_amp_power_count <= 0) { + /* depop parameters */ + snd_soc_update_bits(codec, RT5645_DEPOP_M2, + RT5645_DEPOP_MASK, RT5645_DEPOP_MAN); + snd_soc_write(codec, RT5645_DEPOP_M1, 0x000d); + regmap_write(rt5645->regmap, RT5645_PR_BASE + + RT5645_HP_DCC_INT1, 0x9f01); + mdelay(150); + /* headphone amp power on */ + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_FV1 | RT5645_PWR_FV2 , 0); + snd_soc_update_bits(codec, RT5645_PWR_VOL, + RT5645_PWR_HV_L | RT5645_PWR_HV_R, + RT5645_PWR_HV_L | RT5645_PWR_HV_R); + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_HP_L | RT5645_PWR_HP_R | + RT5645_PWR_HA, + RT5645_PWR_HP_L | RT5645_PWR_HP_R | + RT5645_PWR_HA); + mdelay(5); + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_FV1 | RT5645_PWR_FV2, + RT5645_PWR_FV1 | RT5645_PWR_FV2); + + snd_soc_update_bits(codec, RT5645_DEPOP_M1, + RT5645_HP_CO_MASK | RT5645_HP_SG_MASK, + RT5645_HP_CO_EN | RT5645_HP_SG_EN); + regmap_write(rt5645->regmap, RT5645_PR_BASE + + 0x14, 0x1aaa); + regmap_write(rt5645->regmap, RT5645_PR_BASE + + 0x24, 0x0430); + } + hp_amp_power_count++; + } else { + hp_amp_power_count--; + if (hp_amp_power_count <= 0) { + snd_soc_update_bits(codec, RT5645_DEPOP_M1, + RT5645_HP_SG_MASK | RT5645_HP_L_SMT_MASK | + RT5645_HP_R_SMT_MASK, RT5645_HP_SG_DIS | + RT5645_HP_L_SMT_DIS | RT5645_HP_R_SMT_DIS); + /* headphone amp power down */ + snd_soc_write(codec, RT5645_DEPOP_M1, 0x0000); + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_HP_L | RT5645_PWR_HP_R | + RT5645_PWR_HA, 0); + } + } +} + +static void rt5645_pmu_depop(struct snd_soc_codec *codec) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + + hp_amp_power(codec, 1); + /* headphone unmute sequence */ + snd_soc_update_bits(codec, RT5645_DEPOP_M3, + RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK | RT5645_CP_FQ3_MASK, + (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ1_SFT) | + (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | + (RT5645_CP_FQ_192_KHZ << RT5645_CP_FQ3_SFT)); + regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_MAMP_INT_REG2, + 0xfc00); + snd_soc_update_bits(codec, RT5645_DEPOP_M1, + RT5645_SMT_TRIG_MASK, RT5645_SMT_TRIG_EN); + snd_soc_update_bits(codec, RT5645_DEPOP_M1, + RT5645_RSTN_MASK, RT5645_RSTN_EN); + snd_soc_update_bits(codec, RT5645_DEPOP_M1, + RT5645_RSTN_MASK | RT5645_HP_L_SMT_MASK | RT5645_HP_R_SMT_MASK, + RT5645_RSTN_DIS | RT5645_HP_L_SMT_EN | RT5645_HP_R_SMT_EN); + msleep(40); + snd_soc_update_bits(codec, RT5645_DEPOP_M1, + RT5645_HP_SG_MASK | RT5645_HP_L_SMT_MASK | + RT5645_HP_R_SMT_MASK, RT5645_HP_SG_DIS | + RT5645_HP_L_SMT_DIS | RT5645_HP_R_SMT_DIS); + +} + +static void rt5645_pmd_depop(struct snd_soc_codec *codec) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + + /* headphone mute sequence */ + snd_soc_update_bits(codec, RT5645_DEPOP_M3, + RT5645_CP_FQ1_MASK | RT5645_CP_FQ2_MASK | RT5645_CP_FQ3_MASK, + (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ1_SFT) | + (RT5645_CP_FQ_12_KHZ << RT5645_CP_FQ2_SFT) | + (RT5645_CP_FQ_96_KHZ << RT5645_CP_FQ3_SFT)); + regmap_write(rt5645->regmap, RT5645_PR_BASE + RT5645_MAMP_INT_REG2, + 0xfc00); + snd_soc_update_bits(codec, RT5645_DEPOP_M1, + RT5645_HP_SG_MASK, RT5645_HP_SG_EN); + snd_soc_update_bits(codec, RT5645_DEPOP_M1, + RT5645_RSTP_MASK, RT5645_RSTP_EN); + snd_soc_update_bits(codec, RT5645_DEPOP_M1, + RT5645_RSTP_MASK | RT5645_HP_L_SMT_MASK | + RT5645_HP_R_SMT_MASK, RT5645_RSTP_DIS | + RT5645_HP_L_SMT_EN | RT5645_HP_R_SMT_EN); + msleep(30); + + hp_amp_power(codec, 0); +} + +static int rt5645_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt5645_pmu_depop(codec); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt5645_pmd_depop(codec); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5645_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + regmap_write(rt5645->regmap, RT5645_PR_BASE + 0x1c, 0xfd20); + regmap_write(rt5645->regmap, RT5645_PR_BASE + 0x20, 0x611f); + regmap_write(rt5645->regmap, RT5645_PR_BASE + 0x21, 0x4040); + regmap_write(rt5645->regmap, RT5645_PR_BASE + 0x23, 0x0004); + snd_soc_update_bits(codec, RT5645_PWR_DIG1, + RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | + RT5645_PWR_CLS_D_L, + RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | + RT5645_PWR_CLS_D_L); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5645_PWR_DIG1, + RT5645_PWR_CLS_D | RT5645_PWR_CLS_D_R | + RT5645_PWR_CLS_D_L, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5645_lout_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + hp_amp_power(codec, 1); + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_LM, RT5645_PWR_LM); + snd_soc_update_bits(codec, RT5645_LOUT1, + RT5645_L_MUTE | RT5645_R_MUTE, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5645_LOUT1, + RT5645_L_MUTE | RT5645_R_MUTE, + RT5645_L_MUTE | RT5645_R_MUTE); + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_LM, 0); + hp_amp_power(codec, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5645_bst2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5645_PWR_ANLG2, + RT5645_PWR_BST2_P, RT5645_PWR_BST2_P); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5645_PWR_ANLG2, + RT5645_PWR_BST2_P, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5645_pdm1_l_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5645_PDM_OUT_CTRL, + RT5645_M_PDM1_L, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5645_PDM_OUT_CTRL, + RT5645_M_PDM1_L, RT5645_M_PDM1_L); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5645_pdm1_r_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5645_PDM_OUT_CTRL, + RT5645_M_PDM1_R, 0); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5645_PDM_OUT_CTRL, + RT5645_M_PDM1_R, RT5645_M_PDM1_R); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO2", RT5645_PWR_MIXER, + RT5645_PWR_LDO2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL1", RT5645_PWR_ANLG2, + RT5645_PWR_PLL_BIT, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("JD Power", RT5645_PWR_ANLG2, + RT5645_PWR_JD1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5645_PWR_VOL, + RT5645_PWR_MIC_DET_BIT, 0, NULL, 0), + + /* Input Side */ + /* micbias */ + SND_SOC_DAPM_MICBIAS("micbias1", RT5645_PWR_ANLG2, + RT5645_PWR_MB1_BIT, 0), + SND_SOC_DAPM_MICBIAS("micbias2", RT5645_PWR_ANLG2, + RT5645_PWR_MB2_BIT, 0), + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + SND_SOC_DAPM_INPUT("DMIC L2"), + SND_SOC_DAPM_INPUT("DMIC R2"), + + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN1N"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + + SND_SOC_DAPM_INPUT("Haptic Generator"), + + SND_SOC_DAPM_PGA_E("DMIC1", SND_SOC_NOPM, 0, 0, NULL, 0, + rt5645_set_dmic1_event, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_PGA_E("DMIC2", SND_SOC_NOPM, 0, 0, NULL, 0, + rt5645_set_dmic2_event, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5645_DMIC_CTRL1, + RT5645_DMIC_1_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC2 Power", RT5645_DMIC_CTRL1, + RT5645_DMIC_2_EN_SFT, 0, NULL, 0), + /* Boost */ + SND_SOC_DAPM_PGA("BST1", RT5645_PWR_ANLG2, + RT5645_PWR_BST1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA_E("BST2", RT5645_PWR_ANLG2, + RT5645_PWR_BST2_BIT, 0, NULL, 0, rt5645_bst2_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + /* Input Volume */ + SND_SOC_DAPM_PGA("INL VOL", RT5645_PWR_VOL, + RT5645_PWR_IN_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR VOL", RT5645_PWR_VOL, + RT5645_PWR_IN_R_BIT, 0, NULL, 0), + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", RT5645_PWR_MIXER, RT5645_PWR_RM_L_BIT, + 0, rt5645_rec_l_mix, ARRAY_SIZE(rt5645_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", RT5645_PWR_MIXER, RT5645_PWR_RM_R_BIT, + 0, rt5645_rec_r_mix, ARRAY_SIZE(rt5645_rec_r_mix)), + /* ADCs */ + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_SUPPLY("ADC L power", RT5645_PWR_DIG1, + RT5645_PWR_ADC_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC R power", RT5645_PWR_DIG1, + RT5645_PWR_ADC_R_BIT, 0, NULL, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 DMIC Mux", SND_SOC_NOPM, 0, 0, + &rt5645_sto1_dmic_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5645_sto_adc2_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5645_sto_adc2_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5645_sto_adc1_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5645_sto_adc1_mux), + SND_SOC_DAPM_MUX("Mono DMIC L Mux", SND_SOC_NOPM, 0, 0, + &rt5645_mono_dmic_l_mux), + SND_SOC_DAPM_MUX("Mono DMIC R Mux", SND_SOC_NOPM, 0, 0, + &rt5645_mono_dmic_r_mux), + SND_SOC_DAPM_MUX("Mono ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5645_mono_adc_l2_mux), + SND_SOC_DAPM_MUX("Mono ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5645_mono_adc_l1_mux), + SND_SOC_DAPM_MUX("Mono ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5645_mono_adc_r1_mux), + SND_SOC_DAPM_MUX("Mono ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5645_mono_adc_r2_mux), + /* ADC Mixer */ + + SND_SOC_DAPM_SUPPLY_S("adc stereo1 filter", 1, RT5645_PWR_DIG2, + RT5645_PWR_ADC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("adc stereo2 filter", 1, RT5645_PWR_DIG2, + RT5645_PWR_ADC_S2F_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER_E("Sto1 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5645_sto1_adc_l_mix, ARRAY_SIZE(rt5645_sto1_adc_l_mix), + NULL, 0), + SND_SOC_DAPM_MIXER_E("Sto1 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5645_sto1_adc_r_mix, ARRAY_SIZE(rt5645_sto1_adc_r_mix), + NULL, 0), + SND_SOC_DAPM_SUPPLY_S("adc mono left filter", 1, RT5645_PWR_DIG2, + RT5645_PWR_ADC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER_E("Mono ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5645_mono_adc_l_mix, ARRAY_SIZE(rt5645_mono_adc_l_mix), + NULL, 0), + SND_SOC_DAPM_SUPPLY_S("adc mono right filter", 1, RT5645_PWR_DIG2, + RT5645_PWR_ADC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER_E("Mono ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5645_mono_adc_r_mix, ARRAY_SIZE(rt5645_mono_adc_r_mix), + NULL, 0), + + /* ADC PGA */ + SND_SOC_DAPM_PGA("Stereo1 ADC MIXL", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Stereo1 ADC MIXR", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Sto2 ADC LR MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("VAD_ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1_ADC4", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* IF1 2 Mux */ + SND_SOC_DAPM_MUX("IF1 ADC Mux", SND_SOC_NOPM, + 0, 0, &rt5645_if1_adc_in_mux), + SND_SOC_DAPM_MUX("IF2 ADC Mux", SND_SOC_NOPM, + 0, 0, &rt5645_if2_adc_in_mux), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5645_PWR_DIG1, + RT5645_PWR_I2S1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S2", RT5645_PWR_DIG1, + RT5645_PWR_I2S2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 ADC", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("VAD ADC Mux", SND_SOC_NOPM, + 0, 0, &rt5645_vad_adc_mux), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, + rt5645_dac_l_mix, ARRAY_SIZE(rt5645_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, + rt5645_dac_r_mix, ARRAY_SIZE(rt5645_dac_r_mix)), + + /* DAC2 channel Mux */ + SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, &rt5645_dac_l2_mux), + SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, &rt5645_dac_r2_mux), + SND_SOC_DAPM_PGA("DAC L2 Volume", RT5645_PWR_DIG1, + RT5645_PWR_DAC_L2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC R2 Volume", RT5645_PWR_DIG1, + RT5645_PWR_DAC_R2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DAC1 L Mux", SND_SOC_NOPM, 0, 0, &rt5645_dac1l_mux), + SND_SOC_DAPM_MUX("DAC1 R Mux", SND_SOC_NOPM, 0, 0, &rt5645_dac1r_mux), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY_S("dac stereo1 filter", 1, RT5645_PWR_DIG2, + RT5645_PWR_DAC_S1F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("dac mono left filter", 1, RT5645_PWR_DIG2, + RT5645_PWR_DAC_MF_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("dac mono right filter", 1, RT5645_PWR_DIG2, + RT5645_PWR_DAC_MF_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5645_sto_dac_l_mix, ARRAY_SIZE(rt5645_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5645_sto_dac_r_mix, ARRAY_SIZE(rt5645_sto_dac_r_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5645_mono_dac_l_mix, ARRAY_SIZE(rt5645_mono_dac_l_mix)), + SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5645_mono_dac_r_mix, ARRAY_SIZE(rt5645_mono_dac_r_mix)), + SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5645_dig_l_mix, ARRAY_SIZE(rt5645_dig_l_mix)), + SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5645_dig_r_mix, ARRAY_SIZE(rt5645_dig_r_mix)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5645_PWR_DIG1, RT5645_PWR_DAC_L1_BIT, + 0), + SND_SOC_DAPM_DAC("DAC L2", NULL, RT5645_PWR_DIG1, RT5645_PWR_DAC_L2_BIT, + 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5645_PWR_DIG1, RT5645_PWR_DAC_R1_BIT, + 0), + SND_SOC_DAPM_DAC("DAC R2", NULL, RT5645_PWR_DIG1, RT5645_PWR_DAC_R2_BIT, + 0), + /* OUT Mixer */ + SND_SOC_DAPM_MIXER("SPK MIXL", RT5645_PWR_MIXER, RT5645_PWR_SM_L_BIT, + 0, rt5645_spk_l_mix, ARRAY_SIZE(rt5645_spk_l_mix)), + SND_SOC_DAPM_MIXER("SPK MIXR", RT5645_PWR_MIXER, RT5645_PWR_SM_R_BIT, + 0, rt5645_spk_r_mix, ARRAY_SIZE(rt5645_spk_r_mix)), + SND_SOC_DAPM_MIXER("OUT MIXL", RT5645_PWR_MIXER, RT5645_PWR_OM_L_BIT, + 0, rt5645_out_l_mix, ARRAY_SIZE(rt5645_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5645_PWR_MIXER, RT5645_PWR_OM_R_BIT, + 0, rt5645_out_r_mix, ARRAY_SIZE(rt5645_out_r_mix)), + /* Ouput Volume */ + SND_SOC_DAPM_SWITCH("SPKVOL L", RT5645_PWR_VOL, RT5645_PWR_SV_L_BIT, 0, + &spk_l_vol_control), + SND_SOC_DAPM_SWITCH("SPKVOL R", RT5645_PWR_VOL, RT5645_PWR_SV_R_BIT, 0, + &spk_r_vol_control), + SND_SOC_DAPM_MIXER("HPOVOL MIXL", RT5645_PWR_VOL, RT5645_PWR_HV_L_BIT, + 0, rt5645_hpvoll_mix, ARRAY_SIZE(rt5645_hpvoll_mix)), + SND_SOC_DAPM_MIXER("HPOVOL MIXR", RT5645_PWR_VOL, RT5645_PWR_HV_R_BIT, + 0, rt5645_hpvolr_mix, ARRAY_SIZE(rt5645_hpvolr_mix)), + SND_SOC_DAPM_SUPPLY("HPOVOL MIXL Power", RT5645_PWR_MIXER, + RT5645_PWR_HM_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HPOVOL MIXR Power", RT5645_PWR_MIXER, + RT5645_PWR_HM_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC 1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC 2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HPOVOL", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SWITCH("HPOVOL L", SND_SOC_NOPM, 0, 0, &hp_l_vol_control), + SND_SOC_DAPM_SWITCH("HPOVOL R", SND_SOC_NOPM, 0, 0, &hp_r_vol_control), + + /* HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("SPOL MIX", SND_SOC_NOPM, 0, 0, rt5645_spo_l_mix, + ARRAY_SIZE(rt5645_spo_l_mix)), + SND_SOC_DAPM_MIXER("SPOR MIX", SND_SOC_NOPM, 0, 0, rt5645_spo_r_mix, + ARRAY_SIZE(rt5645_spo_r_mix)), + SND_SOC_DAPM_MIXER("HPO MIX", SND_SOC_NOPM, 0, 0, rt5645_hpo_mix, + ARRAY_SIZE(rt5645_hpo_mix)), + SND_SOC_DAPM_MIXER("LOUT MIX", SND_SOC_NOPM, 0, 0, rt5645_lout_mix, + ARRAY_SIZE(rt5645_lout_mix)), + + SND_SOC_DAPM_PGA_S("HP amp", 1, SND_SOC_NOPM, 0, 0, rt5645_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("LOUT amp", 1, SND_SOC_NOPM, 0, 0, rt5645_lout_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("SPK amp", 2, SND_SOC_NOPM, 0, 0, rt5645_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* PDM */ + SND_SOC_DAPM_SUPPLY("PDM1 Power", RT5645_PWR_DIG2, RT5645_PWR_PDM1_BIT, + 0, NULL, 0), + SND_SOC_DAPM_MUX_E("PDM1 L Mux", SND_SOC_NOPM, 0, 0, &rt5645_pdm1_l_mux, + rt5645_pdm1_l_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MUX_E("PDM1 R Mux", SND_SOC_NOPM, 0, 0, &rt5645_pdm1_r_mux, + rt5645_pdm1_r_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("LOUTL"), + SND_SOC_DAPM_OUTPUT("LOUTR"), + SND_SOC_DAPM_OUTPUT("PDM1L"), + SND_SOC_DAPM_OUTPUT("PDM1R"), + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), +}; + +static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { + { "IN1P", NULL, "LDO2" }, + { "IN2P", NULL, "LDO2" }, + + { "DMIC1", NULL, "DMIC L1" }, + { "DMIC1", NULL, "DMIC R1" }, + { "DMIC2", NULL, "DMIC L2" }, + { "DMIC2", NULL, "DMIC R2" }, + + { "BST1", NULL, "IN1P" }, + { "BST1", NULL, "IN1N" }, + { "BST1", NULL, "JD Power" }, + { "BST1", NULL, "Mic Det Power" }, + { "BST2", NULL, "IN2P" }, + { "BST2", NULL, "IN2N" }, + + { "INL VOL", NULL, "IN2P" }, + { "INR VOL", NULL, "IN2N" }, + + { "RECMIXL", "HPOL Switch", "HPOL" }, + { "RECMIXL", "INL Switch", "INL VOL" }, + { "RECMIXL", "BST2 Switch", "BST2" }, + { "RECMIXL", "BST1 Switch", "BST1" }, + { "RECMIXL", "OUT MIXL Switch", "OUT MIXL" }, + + { "RECMIXR", "HPOR Switch", "HPOR" }, + { "RECMIXR", "INR Switch", "INR VOL" }, + { "RECMIXR", "BST2 Switch", "BST2" }, + { "RECMIXR", "BST1 Switch", "BST1" }, + { "RECMIXR", "OUT MIXR Switch", "OUT MIXR" }, + + { "ADC L", NULL, "RECMIXL" }, + { "ADC L", NULL, "ADC L power" }, + { "ADC R", NULL, "RECMIXR" }, + { "ADC R", NULL, "ADC R power" }, + + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC L1", NULL, "DMIC1 Power"}, + {"DMIC R1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC1 Power"}, + {"DMIC L2", NULL, "DMIC CLK"}, + {"DMIC L2", NULL, "DMIC2 Power"}, + {"DMIC R2", NULL, "DMIC CLK"}, + {"DMIC R2", NULL, "DMIC2 Power"}, + + { "Stereo1 DMIC Mux", "DMIC1", "DMIC1" }, + { "Stereo1 DMIC Mux", "DMIC2", "DMIC2" }, + + { "Mono DMIC L Mux", "DMIC1", "DMIC L1" }, + { "Mono DMIC L Mux", "DMIC2", "DMIC L2" }, + + { "Mono DMIC R Mux", "DMIC1", "DMIC R1" }, + { "Mono DMIC R Mux", "DMIC2", "DMIC R2" }, + + { "Stereo1 ADC L2 Mux", "DMIC", "Stereo1 DMIC Mux" }, + { "Stereo1 ADC L2 Mux", "DAC MIX", "DAC MIXL" }, + { "Stereo1 ADC L1 Mux", "ADC", "ADC L" }, + { "Stereo1 ADC L1 Mux", "DAC MIX", "DAC MIXL" }, + + { "Stereo1 ADC R1 Mux", "ADC", "ADC R" }, + { "Stereo1 ADC R1 Mux", "DAC MIX", "DAC MIXR" }, + { "Stereo1 ADC R2 Mux", "DMIC", "Stereo1 DMIC Mux" }, + { "Stereo1 ADC R2 Mux", "DAC MIX", "DAC MIXR" }, + + { "Mono ADC L2 Mux", "DMIC", "Mono DMIC L Mux" }, + { "Mono ADC L2 Mux", "Mono DAC MIXL", "Mono DAC MIXL" }, + { "Mono ADC L1 Mux", "Mono DAC MIXL", "Mono DAC MIXL" }, + { "Mono ADC L1 Mux", "ADC", "ADC L" }, + + { "Mono ADC R1 Mux", "Mono DAC MIXR", "Mono DAC MIXR" }, + { "Mono ADC R1 Mux", "ADC", "ADC R" }, + { "Mono ADC R2 Mux", "DMIC", "Mono DMIC R Mux" }, + { "Mono ADC R2 Mux", "Mono DAC MIXR", "Mono DAC MIXR" }, + + { "Sto1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux" }, + { "Sto1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux" }, + { "Sto1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux" }, + { "Sto1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux" }, + + { "Stereo1 ADC MIXL", NULL, "Sto1 ADC MIXL" }, + { "Stereo1 ADC MIXL", NULL, "adc stereo1 filter" }, + { "adc stereo1 filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Stereo1 ADC MIXR", NULL, "Sto1 ADC MIXR" }, + { "Stereo1 ADC MIXR", NULL, "adc stereo1 filter" }, + { "adc stereo1 filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Mono ADC MIXL", "ADC1 Switch", "Mono ADC L1 Mux" }, + { "Mono ADC MIXL", "ADC2 Switch", "Mono ADC L2 Mux" }, + { "Mono ADC MIXL", NULL, "adc mono left filter" }, + { "adc mono left filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "Mono ADC MIXR", "ADC1 Switch", "Mono ADC R1 Mux" }, + { "Mono ADC MIXR", "ADC2 Switch", "Mono ADC R2 Mux" }, + { "Mono ADC MIXR", NULL, "adc mono right filter" }, + { "adc mono right filter", NULL, "PLL1", is_sys_clk_from_pll }, + + { "VAD ADC Mux", "Sto1 ADC L", "Stereo1 ADC MIXL" }, + { "VAD ADC Mux", "Mono ADC L", "Mono ADC MIXL" }, + { "VAD ADC Mux", "Mono ADC R", "Mono ADC MIXR" }, + + { "IF_ADC1", NULL, "Stereo1 ADC MIXL" }, + { "IF_ADC1", NULL, "Stereo1 ADC MIXR" }, + { "IF_ADC2", NULL, "Mono ADC MIXL" }, + { "IF_ADC2", NULL, "Mono ADC MIXR" }, + { "VAD_ADC", NULL, "VAD ADC Mux" }, + + { "IF1 ADC Mux", "IF_ADC1", "IF_ADC1" }, + { "IF1 ADC Mux", "IF_ADC2", "IF_ADC2" }, + { "IF1 ADC Mux", "VAD_ADC", "VAD_ADC" }, + + { "IF2 ADC Mux", "IF_ADC1", "IF_ADC1" }, + { "IF2 ADC Mux", "IF_ADC2", "IF_ADC2" }, + { "IF2 ADC Mux", "VAD_ADC", "VAD_ADC" }, + + { "IF1 ADC", NULL, "I2S1" }, + { "IF1 ADC", NULL, "IF1 ADC Mux" }, + { "IF2 ADC", NULL, "I2S2" }, + { "IF2 ADC", NULL, "IF2 ADC Mux" }, + + { "AIF1TX", NULL, "IF1 ADC" }, + { "AIF1TX", NULL, "IF2 ADC" }, + { "AIF2TX", NULL, "IF2 ADC" }, + + { "IF1 DAC1", NULL, "AIF1RX" }, + { "IF1 DAC2", NULL, "AIF1RX" }, + { "IF2 DAC", NULL, "AIF2RX" }, + + { "IF1 DAC1", NULL, "I2S1" }, + { "IF1 DAC2", NULL, "I2S1" }, + { "IF2 DAC", NULL, "I2S2" }, + + { "IF1 DAC2 L", NULL, "IF1 DAC2" }, + { "IF1 DAC2 R", NULL, "IF1 DAC2" }, + { "IF1 DAC1 L", NULL, "IF1 DAC1" }, + { "IF1 DAC1 R", NULL, "IF1 DAC1" }, + { "IF2 DAC L", NULL, "IF2 DAC" }, + { "IF2 DAC R", NULL, "IF2 DAC" }, + + { "DAC1 L Mux", "IF1 DAC", "IF1 DAC1 L" }, + { "DAC1 L Mux", "IF2 DAC", "IF2 DAC L" }, + + { "DAC1 R Mux", "IF1 DAC", "IF1 DAC1 R" }, + { "DAC1 R Mux", "IF2 DAC", "IF2 DAC R" }, + + { "DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL" }, + { "DAC1 MIXL", "DAC1 Switch", "DAC1 L Mux" }, + { "DAC1 MIXL", NULL, "dac stereo1 filter" }, + { "DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR" }, + { "DAC1 MIXR", "DAC1 Switch", "DAC1 R Mux" }, + { "DAC1 MIXR", NULL, "dac stereo1 filter" }, + + { "DAC L2 Mux", "IF1 DAC", "IF1 DAC2 L" }, + { "DAC L2 Mux", "IF2 DAC", "IF2 DAC L" }, + { "DAC L2 Mux", "Mono ADC", "Mono ADC MIXL" }, + { "DAC L2 Mux", "VAD_ADC", "VAD_ADC" }, + { "DAC L2 Volume", NULL, "DAC L2 Mux" }, + { "DAC L2 Volume", NULL, "dac mono left filter" }, + + { "DAC R2 Mux", "IF1 DAC", "IF1 DAC2 R" }, + { "DAC R2 Mux", "IF2 DAC", "IF2 DAC R" }, + { "DAC R2 Mux", "Mono ADC", "Mono ADC MIXR" }, + { "DAC R2 Mux", "Haptic", "Haptic Generator" }, + { "DAC R2 Volume", NULL, "DAC R2 Mux" }, + { "DAC R2 Volume", NULL, "dac mono right filter" }, + + { "Stereo DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXL", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "Stereo DAC MIXL", NULL, "dac stereo1 filter" }, + { "Stereo DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" }, + { "Stereo DAC MIXR", "DAC L1 Switch", "DAC1 MIXL" }, + { "Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "Stereo DAC MIXR", NULL, "dac stereo1 filter" }, + + { "Mono DAC MIXL", "DAC L1 Switch", "DAC1 MIXL" }, + { "Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Volume" }, + { "Mono DAC MIXL", NULL, "dac mono left filter" }, + { "Mono DAC MIXR", "DAC R1 Switch", "DAC1 MIXR" }, + { "Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" }, + { "Mono DAC MIXR", NULL, "dac mono right filter" }, + + { "DAC MIXL", "Sto DAC Mix L Switch", "Stereo DAC MIXL" }, + { "DAC MIXL", "DAC L2 Switch", "DAC L2 Volume" }, + { "DAC MIXL", "DAC R2 Switch", "DAC R2 Volume" }, + { "DAC MIXR", "Sto DAC Mix R Switch", "Stereo DAC MIXR" }, + { "DAC MIXR", "DAC R2 Switch", "DAC R2 Volume" }, + { "DAC MIXR", "DAC L2 Switch", "DAC L2 Volume" }, + + { "DAC L1", NULL, "Stereo DAC MIXL" }, + { "DAC L1", NULL, "PLL1", is_sys_clk_from_pll }, + { "DAC R1", NULL, "Stereo DAC MIXR" }, + { "DAC R1", NULL, "PLL1", is_sys_clk_from_pll }, + { "DAC L2", NULL, "Mono DAC MIXL" }, + { "DAC L2", NULL, "PLL1", is_sys_clk_from_pll }, + { "DAC R2", NULL, "Mono DAC MIXR" }, + { "DAC R2", NULL, "PLL1", is_sys_clk_from_pll }, + + { "SPK MIXL", "BST1 Switch", "BST1" }, + { "SPK MIXL", "INL Switch", "INL VOL" }, + { "SPK MIXL", "DAC L1 Switch", "DAC L1" }, + { "SPK MIXL", "DAC L2 Switch", "DAC L2" }, + { "SPK MIXR", "BST2 Switch", "BST2" }, + { "SPK MIXR", "INR Switch", "INR VOL" }, + { "SPK MIXR", "DAC R1 Switch", "DAC R1" }, + { "SPK MIXR", "DAC R2 Switch", "DAC R2" }, + + { "OUT MIXL", "BST1 Switch", "BST1" }, + { "OUT MIXL", "INL Switch", "INL VOL" }, + { "OUT MIXL", "DAC L2 Switch", "DAC L2" }, + { "OUT MIXL", "DAC L1 Switch", "DAC L1" }, + + { "OUT MIXR", "BST2 Switch", "BST2" }, + { "OUT MIXR", "INR Switch", "INR VOL" }, + { "OUT MIXR", "DAC R2 Switch", "DAC R2" }, + { "OUT MIXR", "DAC R1 Switch", "DAC R1" }, + + { "HPOVOL MIXL", "DAC1 Switch", "DAC L1" }, + { "HPOVOL MIXL", "DAC2 Switch", "DAC L2" }, + { "HPOVOL MIXL", "INL Switch", "INL VOL" }, + { "HPOVOL MIXL", "BST1 Switch", "BST1" }, + { "HPOVOL MIXL", NULL, "HPOVOL MIXL Power" }, + { "HPOVOL MIXR", "DAC1 Switch", "DAC R1" }, + { "HPOVOL MIXR", "DAC2 Switch", "DAC R2" }, + { "HPOVOL MIXR", "INR Switch", "INR VOL" }, + { "HPOVOL MIXR", "BST2 Switch", "BST2" }, + { "HPOVOL MIXR", NULL, "HPOVOL MIXR Power" }, + + { "DAC 2", NULL, "DAC L2" }, + { "DAC 2", NULL, "DAC R2" }, + { "DAC 1", NULL, "DAC L1" }, + { "DAC 1", NULL, "DAC R1" }, + { "HPOVOL L", "Switch", "HPOVOL MIXL" }, + { "HPOVOL R", "Switch", "HPOVOL MIXR" }, + { "HPOVOL", NULL, "HPOVOL L" }, + { "HPOVOL", NULL, "HPOVOL R" }, + { "HPO MIX", "DAC1 Switch", "DAC 1" }, + { "HPO MIX", "HPVOL Switch", "HPOVOL" }, + + { "SPKVOL L", "Switch", "SPK MIXL" }, + { "SPKVOL R", "Switch", "SPK MIXR" }, + + { "SPOL MIX", "DAC R1 Switch", "DAC R1" }, + { "SPOL MIX", "DAC L1 Switch", "DAC L1" }, + { "SPOL MIX", "SPKVOL R Switch", "SPKVOL R" }, + { "SPOL MIX", "SPKVOL L Switch", "SPKVOL L" }, + { "SPOR MIX", "DAC R1 Switch", "DAC R1" }, + { "SPOR MIX", "SPKVOL R Switch", "SPKVOL R" }, + + { "LOUT MIX", "DAC L1 Switch", "DAC L1" }, + { "LOUT MIX", "DAC R1 Switch", "DAC R1" }, + { "LOUT MIX", "OUTMIX L Switch", "OUT MIXL" }, + { "LOUT MIX", "OUTMIX R Switch", "OUT MIXR" }, + + { "PDM1 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, + { "PDM1 L Mux", "Mono DAC", "Mono DAC MIXL" }, + { "PDM1 L Mux", NULL, "PDM1 Power" }, + { "PDM1 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, + { "PDM1 R Mux", "Mono DAC", "Mono DAC MIXR" }, + { "PDM1 R Mux", NULL, "PDM1 Power" }, + + { "HP amp", NULL, "HPO MIX" }, + { "HP amp", NULL, "JD Power" }, + { "HP amp", NULL, "Mic Det Power" }, + { "HP amp", NULL, "LDO2" }, + { "HPOL", NULL, "HP amp" }, + { "HPOR", NULL, "HP amp" }, + + { "LOUT amp", NULL, "LOUT MIX" }, + { "LOUTL", NULL, "LOUT amp" }, + { "LOUTR", NULL, "LOUT amp" }, + + { "PDM1L", NULL, "PDM1 L Mux" }, + { "PDM1R", NULL, "PDM1 R Mux" }, + + { "SPK amp", NULL, "SPOL MIX" }, + { "SPK amp", NULL, "SPOR MIX" }, + { "SPOL", NULL, "SPK amp" }, + { "SPOR", NULL, "SPK amp" }, +}; + +static int get_clk_info(int sclk, int rate) +{ + int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + + if (sclk <= 0 || rate <= 0) + return -EINVAL; + + rate = rate << 8; + for (i = 0; i < ARRAY_SIZE(pd); i++) + if (sclk == rate * pd[i]) + return i; + + return -EINVAL; +} + +static int rt5645_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 rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk; + int pre_div, bclk_ms, frame_size; + + rt5645->lrck[dai->id] = params_rate(params); + pre_div = get_clk_info(rt5645->sysclk, rt5645->lrck[dai->id]); + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting\n"); + return -EINVAL; + } + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return -EINVAL; + } + bclk_ms = frame_size > 32 ? 1 : 0; + rt5645->bclk[dai->id] = rt5645->lrck[dai->id] * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5645->bclk[dai->id], rt5645->lrck[dai->id]); + dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len |= RT5645_I2S_DL_20; + break; + case 24: + val_len |= RT5645_I2S_DL_24; + break; + case 8: + val_len |= RT5645_I2S_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5645_AIF1: + mask_clk = RT5645_I2S_BCLK_MS1_MASK | RT5645_I2S_PD1_MASK; + val_clk = bclk_ms << RT5645_I2S_BCLK_MS1_SFT | + pre_div << RT5645_I2S_PD1_SFT; + snd_soc_update_bits(codec, RT5645_I2S1_SDP, + RT5645_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); + break; + case RT5645_AIF2: + mask_clk = RT5645_I2S_BCLK_MS2_MASK | RT5645_I2S_PD2_MASK; + val_clk = bclk_ms << RT5645_I2S_BCLK_MS2_SFT | + pre_div << RT5645_I2S_PD2_SFT; + snd_soc_update_bits(codec, RT5645_I2S2_SDP, + RT5645_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5645_ADDA_CLK1, mask_clk, val_clk); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5645->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5645_I2S_MS_S; + rt5645->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5645_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5645_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5645_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5645_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + switch (dai->id) { + case RT5645_AIF1: + snd_soc_update_bits(codec, RT5645_I2S1_SDP, + RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_DF_MASK, reg_val); + break; + case RT5645_AIF2: + snd_soc_update_bits(codec, RT5645_I2S1_SDP, + RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | + RT5645_I2S_DF_MASK, reg_val); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt5645_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5645->sysclk && clk_id == rt5645->sysclk_src) + return 0; + + switch (clk_id) { + case RT5645_SCLK_S_MCLK: + reg_val |= RT5645_SCLK_SRC_MCLK; + break; + case RT5645_SCLK_S_PLL1: + reg_val |= RT5645_SCLK_SRC_PLL1; + break; + case RT5645_SCLK_S_RCCLK: + reg_val |= RT5645_SCLK_SRC_RCCLK; + break; + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5645_GLB_CLK, + RT5645_SCLK_SRC_MASK, reg_val); + rt5645->sysclk = freq; + rt5645->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + + return 0; +} + +/** + * rt5645_pll_calc - Calcualte PLL M/N/K code. + * @freq_in: external clock provided to codec. + * @freq_out: target clock which codec works on. + * @pll_code: Pointer to structure with M, N, K and bypass flag. + * + * Calcualte M/N/K code to configure PLL for codec. And K is assigned to 2 + * which make calculation more efficiently. + * + * Returns 0 for success or negative error code. + */ +static int rt5645_pll_calc(const unsigned int freq_in, + const unsigned int freq_out, struct rt5645_pll_code *pll_code) +{ + int max_n = RT5645_PLL_N_MAX, max_m = RT5645_PLL_M_MAX; + int k, n = 0, m = 0, red, n_t, m_t, pll_out, in_t, out_t; + int red_t = abs(freq_out - freq_in); + bool bypass = false; + + if (RT5645_PLL_INP_MAX < freq_in || RT5645_PLL_INP_MIN > freq_in) + return -EINVAL; + + k = 100000000 / freq_out - 2; + if (k > RT5645_PLL_K_MAX) + k = RT5645_PLL_K_MAX; + for (n_t = 0; n_t <= max_n; n_t++) { + in_t = freq_in / (k + 2); + pll_out = freq_out / (n_t + 2); + if (in_t < 0) + continue; + if (in_t == pll_out) { + bypass = true; + n = n_t; + goto code_find; + } + red = abs(in_t - pll_out); + if (red < red_t) { + bypass = true; + n = n_t; + m = m_t; + if (red == 0) + goto code_find; + red_t = red; + } + for (m_t = 0; m_t <= max_m; m_t++) { + out_t = in_t / (m_t + 2); + red = abs(out_t - pll_out); + if (red < red_t) { + bypass = false; + n = n_t; + m = m_t; + if (red == 0) + goto code_find; + red_t = red; + } + } + } + pr_debug("Only get approximation about PLL\n"); + +code_find: + + pll_code->m_bp = bypass; + pll_code->m_code = m; + pll_code->n_code = n; + pll_code->k_code = k; + return 0; +} + +static int rt5645_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + struct rt5645_pll_code pll_code; + int ret; + + if (source == rt5645->pll_src && freq_in == rt5645->pll_in && + freq_out == rt5645->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5645->pll_in = 0; + rt5645->pll_out = 0; + snd_soc_update_bits(codec, RT5645_GLB_CLK, + RT5645_SCLK_SRC_MASK, RT5645_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5645_PLL1_S_MCLK: + snd_soc_update_bits(codec, RT5645_GLB_CLK, + RT5645_PLL1_SRC_MASK, RT5645_PLL1_SRC_MCLK); + break; + case RT5645_PLL1_S_BCLK1: + case RT5645_PLL1_S_BCLK2: + switch (dai->id) { + case RT5645_AIF1: + snd_soc_update_bits(codec, RT5645_GLB_CLK, + RT5645_PLL1_SRC_MASK, RT5645_PLL1_SRC_BCLK1); + break; + case RT5645_AIF2: + snd_soc_update_bits(codec, RT5645_GLB_CLK, + RT5645_PLL1_SRC_MASK, RT5645_PLL1_SRC_BCLK2); + break; + default: + dev_err(codec->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + break; + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rt5645_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_write(codec, RT5645_PLL_CTRL1, + pll_code.n_code << RT5645_PLL_N_SFT | pll_code.k_code); + snd_soc_write(codec, RT5645_PLL_CTRL2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5645_PLL_M_SFT | + pll_code.m_bp << RT5645_PLL_M_BP_SFT); + + rt5645->pll_in = freq_in; + rt5645->pll_out = freq_out; + rt5645->pll_src = source; + + return 0; +} + +static int rt5645_set_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; + unsigned int val = 0; + + if (rx_mask || tx_mask) + val |= (1 << 14); + + switch (slots) { + case 4: + val |= (1 << 12); + break; + case 6: + val |= (2 << 12); + break; + case 8: + val |= (3 << 12); + break; + case 2: + default: + break; + } + + switch (slot_width) { + case 20: + val |= (1 << 10); + break; + case 24: + val |= (2 << 10); + break; + case 32: + val |= (3 << 10); + break; + case 16: + default: + break; + } + + snd_soc_update_bits(codec, RT5645_TDM_CTRL_1, 0x7c00, val); + + return 0; +} + +static int rt5645_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_STANDBY: + if (SND_SOC_BIAS_OFF == codec->dapm.bias_level) { + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_VREF1 | RT5645_PWR_MB | + RT5645_PWR_BG | RT5645_PWR_VREF2, + RT5645_PWR_VREF1 | RT5645_PWR_MB | + RT5645_PWR_BG | RT5645_PWR_VREF2); + mdelay(10); + snd_soc_update_bits(codec, RT5645_PWR_ANLG1, + RT5645_PWR_FV1 | RT5645_PWR_FV2, + RT5645_PWR_FV1 | RT5645_PWR_FV2); + snd_soc_update_bits(codec, RT5645_GEN_CTRL1, + RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); + } + break; + + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, RT5645_DEPOP_M2, 0x1100); + snd_soc_write(codec, RT5645_GEN_CTRL1, 0x0128); + snd_soc_write(codec, RT5645_PWR_DIG1, 0x0000); + snd_soc_write(codec, RT5645_PWR_DIG2, 0x0000); + snd_soc_write(codec, RT5645_PWR_VOL, 0x0000); + snd_soc_write(codec, RT5645_PWR_MIXER, 0x0000); + snd_soc_write(codec, RT5645_PWR_ANLG1, 0x0000); + snd_soc_write(codec, RT5645_PWR_ANLG2, 0x0000); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt5645_probe(struct snd_soc_codec *codec) +{ + struct rt5645_priv *rt5645 = snd_soc_codec_get_drvdata(codec); + + rt5645->codec = codec; + + rt5645_set_bias_level(codec, SND_SOC_BIAS_OFF); + + snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200); + + return 0; +} + +static int rt5645_remove(struct snd_soc_codec *codec) +{ + rt5645_reset(codec); + return 0; +} + +#ifdef CONFIG_PM +static int rt5645_suspend(struct snd_soc_codec *codec) +{ + rt5645_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int rt5645_resume(struct snd_soc_codec *codec) +{ + return 0; +} +#else +#define rt5645_suspend NULL +#define rt5645_resume NULL +#endif + +#define RT5645_STEREO_RATES SNDRV_PCM_RATE_8000_96000 +#define RT5645_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_ops rt5645_aif_dai_ops = { + .hw_params = rt5645_hw_params, + .set_fmt = rt5645_set_dai_fmt, + .set_sysclk = rt5645_set_dai_sysclk, + .set_tdm_slot = rt5645_set_tdm_slot, + .set_pll = rt5645_set_dai_pll, +}; + +struct snd_soc_dai_driver rt5645_dai[] = { + { + .name = "rt5645-aif1", + .id = RT5645_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5645_STEREO_RATES, + .formats = RT5645_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5645_STEREO_RATES, + .formats = RT5645_FORMATS, + }, + .ops = &rt5645_aif_dai_ops, + }, + { + .name = "rt5645-aif2", + .id = RT5645_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5645_STEREO_RATES, + .formats = RT5645_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5645_STEREO_RATES, + .formats = RT5645_FORMATS, + }, + .ops = &rt5645_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5645 = { + .probe = rt5645_probe, + .remove = rt5645_remove, + .suspend = rt5645_suspend, + .resume = rt5645_resume, + .set_bias_level = rt5645_set_bias_level, + .idle_bias_off = true, + .controls = rt5645_snd_controls, + .num_controls = ARRAY_SIZE(rt5645_snd_controls), + .dapm_widgets = rt5645_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5645_dapm_widgets), + .dapm_routes = rt5645_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5645_dapm_routes), +}; + +static const struct regmap_config rt5645_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = RT5645_VENDOR_ID2 + 1 + (ARRAY_SIZE(rt5645_ranges) * + RT5645_PR_SPACING), + .volatile_reg = rt5645_volatile_register, + .readable_reg = rt5645_readable_register, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5645_reg, + .num_reg_defaults = ARRAY_SIZE(rt5645_reg), + .ranges = rt5645_ranges, + .num_ranges = ARRAY_SIZE(rt5645_ranges), +}; + +static const struct i2c_device_id rt5645_i2c_id[] = { + { "rt5645", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5645_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id rt5645_of_match[] = { + { .compatible = "realtek,rt5645", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5645_of_match); +#endif + +static int rt5645_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5645_priv *rt5645; + int ret; + unsigned int val; + + rt5645 = devm_kzalloc(&i2c->dev, sizeof(struct rt5645_priv), + GFP_KERNEL); + if (rt5645 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5645); + + rt5645->regmap = devm_regmap_init_i2c(i2c, &rt5645_regmap); + if (IS_ERR(rt5645->regmap)) { + ret = PTR_ERR(rt5645->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt5645->regmap, RT5645_VENDOR_ID2, &val); + if (val != RT5645_DEVICE_ID) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt5645\n", val); + return -ENODEV; + } + + regmap_write(rt5645->regmap, RT5645_RESET, 0); + + ret = regmap_register_patch(rt5645->regmap, init_list, + ARRAY_SIZE(init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5645, + rt5645_dai, ARRAY_SIZE(rt5645_dai)); + if (ret < 0) + goto err; + + return 0; +err: + return ret; +} + +static int rt5645_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + kfree(i2c_get_clientdata(i2c)); + return 0; +} + +void rt5645_i2c_shutdown(struct i2c_client *client) +{ + struct rt5645_priv *rt5645 = i2c_get_clientdata(client); + struct snd_soc_codec *codec = rt5645->codec; + + if (codec != NULL) + rt5645_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +struct i2c_driver rt5645_i2c_driver = { + .driver = { + .name = "rt5645", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(rt5645_of_match), + }, + .probe = rt5645_i2c_probe, + .remove = rt5645_i2c_remove, + .shutdown = rt5645_i2c_shutdown, + .id_table = rt5645_i2c_id, +}; +module_i2c_driver(rt5645_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5645 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h new file mode 100644 index 0000000..d683ea2 --- /dev/null +++ b/sound/soc/codecs/rt5645.h @@ -0,0 +1,2176 @@ +/* + * rt5645.h -- RT5645 ALSA SoC audio driver + * + * Copyright 2013 Realtek Microelectronics + * Author: Bard Liao <bardliao(a)realtek.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 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5645_H__ +#define __RT5645_H__ + +/* Info */ +#define RT5645_RESET 0x00 +#define RT5645_VENDOR_ID 0xfd +#define RT5645_VENDOR_ID1 0xfe +#define RT5645_VENDOR_ID2 0xff +/* I/O - Output */ +#define RT5645_SPK_VOL 0x01 +#define RT5645_HP_VOL 0x02 +#define RT5645_LOUT1 0x03 +#define RT5645_LOUT_CTRL 0x05 +/* I/O - Input */ +#define RT5645_IN1_CTRL1 0x0a +#define RT5645_IN1_CTRL2 0x0b +#define RT5645_IN1_CTRL3 0x0c +#define RT5645_IN2_CTRL 0x0d +#define RT5645_INL1_INR1_VOL 0x0f +#define RT5645_SPK_FUNC_LIM 0x14 +#define RT5645_ADJ_HPF_CTRL 0x16 +/* I/O - ADC/DAC/DMIC */ +#define RT5645_DAC1_DIG_VOL 0x19 +#define RT5645_DAC2_DIG_VOL 0x1a +#define RT5645_DAC_CTRL 0x1b +#define RT5645_STO1_ADC_DIG_VOL 0x1c +#define RT5645_MONO_ADC_DIG_VOL 0x1d +#define RT5645_ADC_BST_VOL1 0x1e +/* Mixer - D-D */ +#define RT5645_ADC_BST_VOL2 0x20 +#define RT5645_STO1_ADC_MIXER 0x27 +#define RT5645_MONO_ADC_MIXER 0x28 +#define RT5645_AD_DA_MIXER 0x29 +#define RT5645_STO_DAC_MIXER 0x2a +#define RT5645_MONO_DAC_MIXER 0x2b +#define RT5645_DIG_MIXER 0x2c +#define RT5645_DIG_INF1_DATA 0x2f +/* Mixer - PDM */ +#define RT5645_PDM_OUT_CTRL 0x31 +/* Mixer - ADC */ +#define RT5645_REC_L1_MIXER 0x3b +#define RT5645_REC_L2_MIXER 0x3c +#define RT5645_REC_R1_MIXER 0x3d +#define RT5645_REC_R2_MIXER 0x3e +/* Mixer - DAC */ +#define RT5645_HPMIXL_CTRL 0x3f +#define RT5645_HPOMIXL_CTRL 0x40 +#define RT5645_HPMIXR_CTRL 0x41 +#define RT5645_HPOMIXR_CTRL 0x42 +#define RT5645_HPO_MIXER 0x45 +#define RT5645_SPK_L_MIXER 0x46 +#define RT5645_SPK_R_MIXER 0x47 +#define RT5645_SPO_MIXER 0x48 +#define RT5645_SPO_CLSD_RATIO 0x4a +#define RT5645_OUT_L_GAIN1 0x4d +#define RT5645_OUT_L_GAIN2 0x4e +#define RT5645_OUT_L1_MIXER 0x4f +#define RT5645_OUT_R_GAIN1 0x50 +#define RT5645_OUT_R_GAIN2 0x51 +#define RT5645_OUT_R1_MIXER 0x52 +#define RT5645_LOUT_MIXER 0x53 +/* Haptic */ +#define RT5645_HAPTIC_CTRL1 0x56 +#define RT5645_HAPTIC_CTRL2 0x57 +#define RT5645_HAPTIC_CTRL3 0x58 +#define RT5645_HAPTIC_CTRL4 0x59 +#define RT5645_HAPTIC_CTRL5 0x5a +#define RT5645_HAPTIC_CTRL6 0x5b +#define RT5645_HAPTIC_CTRL7 0x5c +#define RT5645_HAPTIC_CTRL8 0x5d +#define RT5645_HAPTIC_CTRL9 0x5e +#define RT5645_HAPTIC_CTRL10 0x5f +/* Power */ +#define RT5645_PWR_DIG1 0x61 +#define RT5645_PWR_DIG2 0x62 +#define RT5645_PWR_ANLG1 0x63 +#define RT5645_PWR_ANLG2 0x64 +#define RT5645_PWR_MIXER 0x65 +#define RT5645_PWR_VOL 0x66 +/* Private Register Control */ +#define RT5645_PRIV_INDEX 0x6a +#define RT5645_PRIV_DATA 0x6c +/* Format - ADC/DAC */ +#define RT5645_I2S1_SDP 0x70 +#define RT5645_I2S2_SDP 0x71 +#define RT5645_ADDA_CLK1 0x73 +#define RT5645_ADDA_CLK2 0x74 +#define RT5645_DMIC_CTRL1 0x75 +#define RT5645_DMIC_CTRL2 0x76 +/* Format - TDM Control */ +#define RT5645_TDM_CTRL_1 0x77 +#define RT5645_TDM_CTRL_2 0x78 +#define RT5645_TDM_CTRL_3 0x79 + +/* Function - Analog */ +#define RT5645_GLB_CLK 0x80 +#define RT5645_PLL_CTRL1 0x81 +#define RT5645_PLL_CTRL2 0x82 +#define RT5645_ASRC_1 0x83 +#define RT5645_ASRC_2 0x84 +#define RT5645_ASRC_3 0x85 +#define RT5645_ASRC_4 0x8a +#define RT5645_DEPOP_M1 0x8e +#define RT5645_DEPOP_M2 0x8f +#define RT5645_DEPOP_M3 0x90 +#define RT5645_CHARGE_PUMP 0x91 +#define RT5645_MICBIAS 0x93 +#define RT5645_A_JD_CTRL1 0x94 +#define RT5645_VAD_CTRL4 0x9d +#define RT5645_CLSD_OUT_CTRL 0xa0 +/* Function - Digital */ +#define RT5645_ADC_EQ_CTRL1 0xae +#define RT5645_ADC_EQ_CTRL2 0xaf +#define RT5645_EQ_CTRL1 0xb0 +#define RT5645_EQ_CTRL2 0xb1 +#define RT5645_ALC_CTRL_1 0xb3 +#define RT5645_ALC_CTRL_2 0xb4 +#define RT5645_ALC_CTRL_3 0xb5 +#define RT5645_ALC_CTRL_4 0xb6 +#define RT5645_ALC_CTRL_5 0xb7 +#define RT5645_JD_CTRL 0xbb +#define RT5645_IRQ_CTRL1 0xbc +#define RT5645_IRQ_CTRL2 0xbd +#define RT5645_IRQ_CTRL3 0xbe +#define RT5645_INT_IRQ_ST 0xbf +#define RT5645_GPIO_CTRL1 0xc0 +#define RT5645_GPIO_CTRL2 0xc1 +#define RT5645_GPIO_CTRL3 0xc2 +#define RT5645_BASS_BACK 0xcf +#define RT5645_MP3_PLUS1 0xd0 +#define RT5645_MP3_PLUS2 0xd1 +#define RT5645_ADJ_HPF1 0xd3 +#define RT5645_ADJ_HPF2 0xd4 +#define RT5645_HP_CALIB_AMP_DET 0xd6 +#define RT5645_SV_ZCD1 0xd9 +#define RT5645_SV_ZCD2 0xda +#define RT5645_IL_CMD 0xdb +#define RT5645_IL_CMD2 0xdc +#define RT5645_IL_CMD3 0xdd +#define RT5645_DRC1_HL_CTRL1 0xe7 +#define RT5645_DRC2_HL_CTRL1 0xe9 +#define RT5645_MUTI_DRC_CTRL1 0xea +#define RT5645_ADC_MONO_HP_CTRL1 0xec +#define RT5645_ADC_MONO_HP_CTRL2 0xed +#define RT5645_DRC2_CTRL1 0xf0 +#define RT5645_DRC2_CTRL2 0xf1 +#define RT5645_DRC2_CTRL3 0xf2 +#define RT5645_DRC2_CTRL4 0xf3 +#define RT5645_DRC2_CTRL5 0xf4 +#define RT5645_JD_CTRL3 0xf8 +#define RT5645_JD_CTRL4 0xf9 +/* General Control */ +#define RT5645_GEN_CTRL1 0xfa +#define RT5645_GEN_CTRL2 0xfb +#define RT5645_GEN_CTRL3 0xfc + + +/* Index of Codec Private Register definition */ +#define RT5645_DIG_VOL 0x00 +#define RT5645_PR_ALC_CTRL_1 0x01 +#define RT5645_PR_ALC_CTRL_2 0x02 +#define RT5645_PR_ALC_CTRL_3 0x03 +#define RT5645_PR_ALC_CTRL_4 0x04 +#define RT5645_PR_ALC_CTRL_5 0x05 +#define RT5645_PR_ALC_CTRL_6 0x06 +#define RT5645_BIAS_CUR1 0x12 +#define RT5645_BIAS_CUR3 0x14 +#define RT5645_CLSD_INT_REG1 0x1c +#define RT5645_MAMP_INT_REG2 0x37 +#define RT5645_CHOP_DAC_ADC 0x3d +#define RT5645_MIXER_INT_REG 0x3f +#define RT5645_3D_SPK 0x63 +#define RT5645_WND_1 0x6c +#define RT5645_WND_2 0x6d +#define RT5645_WND_3 0x6e +#define RT5645_WND_4 0x6f +#define RT5645_WND_5 0x70 +#define RT5645_WND_8 0x73 +#define RT5645_DIP_SPK_INF 0x75 +#define RT5645_HP_DCC_INT1 0x77 +#define RT5645_EQ_BW_LOP 0xa0 +#define RT5645_EQ_GN_LOP 0xa1 +#define RT5645_EQ_FC_BP1 0xa2 +#define RT5645_EQ_BW_BP1 0xa3 +#define RT5645_EQ_GN_BP1 0xa4 +#define RT5645_EQ_FC_BP2 0xa5 +#define RT5645_EQ_BW_BP2 0xa6 +#define RT5645_EQ_GN_BP2 0xa7 +#define RT5645_EQ_FC_BP3 0xa8 +#define RT5645_EQ_BW_BP3 0xa9 +#define RT5645_EQ_GN_BP3 0xaa +#define RT5645_EQ_FC_BP4 0xab +#define RT5645_EQ_BW_BP4 0xac +#define RT5645_EQ_GN_BP4 0xad +#define RT5645_EQ_FC_HIP1 0xae +#define RT5645_EQ_GN_HIP1 0xaf +#define RT5645_EQ_FC_HIP2 0xb0 +#define RT5645_EQ_BW_HIP2 0xb1 +#define RT5645_EQ_GN_HIP2 0xb2 +#define RT5645_EQ_PRE_VOL 0xb3 +#define RT5645_EQ_PST_VOL 0xb4 + + +/* global definition */ +#define RT5645_L_MUTE (0x1 << 15) +#define RT5645_L_MUTE_SFT 15 +#define RT5645_VOL_L_MUTE (0x1 << 14) +#define RT5645_VOL_L_SFT 14 +#define RT5645_R_MUTE (0x1 << 7) +#define RT5645_R_MUTE_SFT 7 +#define RT5645_VOL_R_MUTE (0x1 << 6) +#define RT5645_VOL_R_SFT 6 +#define RT5645_L_VOL_MASK (0x3f << 8) +#define RT5645_L_VOL_SFT 8 +#define RT5645_R_VOL_MASK (0x3f) +#define RT5645_R_VOL_SFT 0 + +/* IN1 Control 1 (0x0a) */ +#define RT5645_CBJ_BST1_MASK (0xf << 12) +#define RT5645_CBJ_BST1_SFT (12) +#define RT5645_CBJ_JD_HP_EN (0x1 << 9) +#define RT5645_CBJ_JD_MIC_EN (0x1 << 8) +#define RT5645_CBJ_JD_MIC_SW_EN (0x1 << 7) +#define RT5645_CBJ_MIC_SEL_R (0x1 << 6) +#define RT5645_CBJ_MIC_SEL_L (0x1 << 5) +#define RT5645_CBJ_MIC_SW (0x1 << 4) +#define RT5645_CBJ_BST1_EN (0x1 << 2) + +/* IN1 Control 2 (0x0b) */ +#define RT5645_CBJ_MN_JD (0x1 << 12) +#define RT5645_CAPLESS_EN (0x1 << 11) +#define RT5645_CBJ_DET_MODE (0x1 << 7) + +/* IN1 Control 3 (0x0c) */ +#define RT5645_CBJ_TIE_G_L (0x1 << 15) +#define RT5645_CBJ_TIE_G_R (0x1 << 14) + +/* IN2 Control (0x0d) */ +#define RT5645_BST_MASK1 (0xf<<12) +#define RT5645_BST_SFT1 12 +#define RT5645_BST_MASK2 (0xf<<8) +#define RT5645_BST_SFT2 8 +#define RT5645_IN_DF2 (0x1 << 6) +#define RT5645_IN_SFT2 6 + +/* INL and INR Volume Control (0x0f) */ +#define RT5645_INL_SEL_MASK (0x1 << 15) +#define RT5645_INL_SEL_SFT 15 +#define RT5645_INL_SEL_IN4P (0x0 << 15) +#define RT5645_INL_SEL_MONOP (0x1 << 15) +#define RT5645_INL_VOL_MASK (0x1f << 8) +#define RT5645_INL_VOL_SFT 8 +#define RT5645_INR_SEL_MASK (0x1 << 7) +#define RT5645_INR_SEL_SFT 7 +#define RT5645_INR_SEL_IN4N (0x0 << 7) +#define RT5645_INR_SEL_MONON (0x1 << 7) +#define RT5645_INR_VOL_MASK (0x1f) +#define RT5645_INR_VOL_SFT 0 + +/* DAC1 Digital Volume (0x19) */ +#define RT5645_DAC_L1_VOL_MASK (0xff << 8) +#define RT5645_DAC_L1_VOL_SFT 8 +#define RT5645_DAC_R1_VOL_MASK (0xff) +#define RT5645_DAC_R1_VOL_SFT 0 + +/* DAC2 Digital Volume (0x1a) */ +#define RT5645_DAC_L2_VOL_MASK (0xff << 8) +#define RT5645_DAC_L2_VOL_SFT 8 +#define RT5645_DAC_R2_VOL_MASK (0xff) +#define RT5645_DAC_R2_VOL_SFT 0 + +/* DAC2 Control (0x1b) */ +#define RT5645_M_DAC_L2_VOL (0x1 << 13) +#define RT5645_M_DAC_L2_VOL_SFT 13 +#define RT5645_M_DAC_R2_VOL (0x1 << 12) +#define RT5645_M_DAC_R2_VOL_SFT 12 +#define RT5645_DAC2_L_SEL_MASK (0x7 << 4) +#define RT5645_DAC2_L_SEL_SFT 4 +#define RT5645_DAC2_R_SEL_MASK (0x7 << 0) +#define RT5645_DAC2_R_SEL_SFT 0 + +/* ADC Digital Volume Control (0x1c) */ +#define RT5645_ADC_L_VOL_MASK (0x7f << 8) +#define RT5645_ADC_L_VOL_SFT 8 +#define RT5645_ADC_R_VOL_MASK (0x7f) +#define RT5645_ADC_R_VOL_SFT 0 + +/* Mono ADC Digital Volume Control (0x1d) */ +#define RT5645_MONO_ADC_L_VOL_MASK (0x7f << 8) +#define RT5645_MONO_ADC_L_VOL_SFT 8 +#define RT5645_MONO_ADC_R_VOL_MASK (0x7f) +#define RT5645_MONO_ADC_R_VOL_SFT 0 + +/* ADC Boost Volume Control (0x1e) */ +#define RT5645_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5645_STO1_ADC_L_BST_SFT 14 +#define RT5645_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5645_STO1_ADC_R_BST_SFT 12 +#define RT5645_STO1_ADC_COMP_MASK (0x3 << 10) +#define RT5645_STO1_ADC_COMP_SFT 10 +#define RT5645_STO2_ADC_L_BST_MASK (0x3 << 8) +#define RT5645_STO2_ADC_L_BST_SFT 8 +#define RT5645_STO2_ADC_R_BST_MASK (0x3 << 6) +#define RT5645_STO2_ADC_R_BST_SFT 6 +#define RT5645_STO2_ADC_COMP_MASK (0x3 << 4) +#define RT5645_STO2_ADC_COMP_SFT 4 + +/* Stereo2 ADC Mixer Control (0x26) */ +#define RT5645_STO2_ADC_SRC_MASK (0x1 << 15) +#define RT5645_STO2_ADC_SRC_SFT 15 + +/* Stereo ADC Mixer Control (0x27) */ +#define RT5645_M_ADC_L1 (0x1 << 14) +#define RT5645_M_ADC_L1_SFT 14 +#define RT5645_M_ADC_L2 (0x1 << 13) +#define RT5645_M_ADC_L2_SFT 13 +#define RT5645_ADC_1_SRC_MASK (0x1 << 12) +#define RT5645_ADC_1_SRC_SFT 12 +#define RT5645_ADC_1_SRC_ADC (0x1 << 12) +#define RT5645_ADC_1_SRC_DACMIX (0x0 << 12) +#define RT5645_ADC_2_SRC_MASK (0x1 << 11) +#define RT5645_ADC_2_SRC_SFT 11 +#define RT5645_DMIC_SRC_MASK (0x1 << 8) +#define RT5645_DMIC_SRC_SFT 8 +#define RT5645_M_ADC_R1 (0x1 << 6) +#define RT5645_M_ADC_R1_SFT 6 +#define RT5645_M_ADC_R2 (0x1 << 5) +#define RT5645_M_ADC_R2_SFT 5 +#define RT5645_DMIC3_SRC_MASK (0x1 << 1) +#define RT5645_DMIC3_SRC_SFT 0 + +/* Mono ADC Mixer Control (0x28) */ +#define RT5645_M_MONO_ADC_L1 (0x1 << 14) +#define RT5645_M_MONO_ADC_L1_SFT 14 +#define RT5645_M_MONO_ADC_L2 (0x1 << 13) +#define RT5645_M_MONO_ADC_L2_SFT 13 +#define RT5645_MONO_ADC_L1_SRC_MASK (0x1 << 12) +#define RT5645_MONO_ADC_L1_SRC_SFT 12 +#define RT5645_MONO_ADC_L1_SRC_DACMIXL (0x0 << 12) +#define RT5645_MONO_ADC_L1_SRC_ADCL (0x1 << 12) +#define RT5645_MONO_ADC_L2_SRC_MASK (0x1 << 11) +#define RT5645_MONO_ADC_L2_SRC_SFT 11 +#define RT5645_MONO_DMIC_L_SRC_MASK (0x1 << 8) +#define RT5645_MONO_DMIC_L_SRC_SFT 8 +#define RT5645_M_MONO_ADC_R1 (0x1 << 6) +#define RT5645_M_MONO_ADC_R1_SFT 6 +#define RT5645_M_MONO_ADC_R2 (0x1 << 5) +#define RT5645_M_MONO_ADC_R2_SFT 5 +#define RT5645_MONO_ADC_R1_SRC_MASK (0x1 << 4) +#define RT5645_MONO_ADC_R1_SRC_SFT 4 +#define RT5645_MONO_ADC_R1_SRC_ADCR (0x1 << 4) +#define RT5645_MONO_ADC_R1_SRC_DACMIXR (0x0 << 4) +#define RT5645_MONO_ADC_R2_SRC_MASK (0x1 << 3) +#define RT5645_MONO_ADC_R2_SRC_SFT 3 +#define RT5645_MONO_DMIC_R_SRC_MASK (0x3) +#define RT5645_MONO_DMIC_R_SRC_SFT 0 + +/* ADC Mixer to DAC Mixer Control (0x29) */ +#define RT5645_M_ADCMIX_L (0x1 << 15) +#define RT5645_M_ADCMIX_L_SFT 15 +#define RT5645_M_DAC1_L (0x1 << 14) +#define RT5645_M_DAC1_L_SFT 14 +#define RT5645_DAC1_R_SEL_MASK (0x3 << 10) +#define RT5645_DAC1_R_SEL_SFT 10 +#define RT5645_DAC1_R_SEL_IF1 (0x0 << 10) +#define RT5645_DAC1_R_SEL_IF2 (0x1 << 10) +#define RT5645_DAC1_R_SEL_IF3 (0x2 << 10) +#define RT5645_DAC1_R_SEL_IF4 (0x3 << 10) +#define RT5645_DAC1_L_SEL_MASK (0x3 << 8) +#define RT5645_DAC1_L_SEL_SFT 8 +#define RT5645_DAC1_L_SEL_IF1 (0x0 << 8) +#define RT5645_DAC1_L_SEL_IF2 (0x1 << 8) +#define RT5645_DAC1_L_SEL_IF3 (0x2 << 8) +#define RT5645_DAC1_L_SEL_IF4 (0x3 << 8) +#define RT5645_M_ADCMIX_R (0x1 << 7) +#define RT5645_M_ADCMIX_R_SFT 7 +#define RT5645_M_DAC1_R (0x1 << 6) +#define RT5645_M_DAC1_R_SFT 6 + +/* Stereo DAC Mixer Control (0x2a) */ +#define RT5645_M_DAC_L1 (0x1 << 14) +#define RT5645_M_DAC_L1_SFT 14 +#define RT5645_DAC_L1_STO_L_VOL_MASK (0x1 << 13) +#define RT5645_DAC_L1_STO_L_VOL_SFT 13 +#define RT5645_M_DAC_L2 (0x1 << 12) +#define RT5645_M_DAC_L2_SFT 12 +#define RT5645_DAC_L2_STO_L_VOL_MASK (0x1 << 11) +#define RT5645_DAC_L2_STO_L_VOL_SFT 11 +#define RT5645_M_ANC_DAC_L (0x1 << 10) +#define RT5645_M_ANC_DAC_L_SFT 10 +#define RT5645_M_DAC_R1_STO_L (0x1 << 9) +#define RT5645_M_DAC_R1_STO_L_SFT 9 +#define RT5645_DAC_R1_STO_L_VOL_MASK (0x1 << 8) +#define RT5645_DAC_R1_STO_L_VOL_SFT 8 +#define RT5645_M_DAC_R1 (0x1 << 6) +#define RT5645_M_DAC_R1_SFT 6 +#define RT5645_DAC_R1_STO_R_VOL_MASK (0x1 << 5) +#define RT5645_DAC_R1_STO_R_VOL_SFT 5 +#define RT5645_M_DAC_R2 (0x1 << 4) +#define RT5645_M_DAC_R2_SFT 4 +#define RT5645_DAC_R2_STO_R_VOL_MASK (0x1 << 3) +#define RT5645_DAC_R2_STO_R_VOL_SFT 3 +#define RT5645_M_ANC_DAC_R (0x1 << 2) +#define RT5645_M_ANC_DAC_R_SFT 2 +#define RT5645_M_DAC_L1_STO_R (0x1 << 1) +#define RT5645_M_DAC_L1_STO_R_SFT 1 +#define RT5645_DAC_L1_STO_R_VOL_MASK (0x1) +#define RT5645_DAC_L1_STO_R_VOL_SFT 0 + +/* Mono DAC Mixer Control (0x2b) */ +#define RT5645_M_DAC_L1_MONO_L (0x1 << 14) +#define RT5645_M_DAC_L1_MONO_L_SFT 14 +#define RT5645_DAC_L1_MONO_L_VOL_MASK (0x1 << 13) +#define RT5645_DAC_L1_MONO_L_VOL_SFT 13 +#define RT5645_M_DAC_L2_MONO_L (0x1 << 12) +#define RT5645_M_DAC_L2_MONO_L_SFT 12 +#define RT5645_DAC_L2_MONO_L_VOL_MASK (0x1 << 11) +#define RT5645_DAC_L2_MONO_L_VOL_SFT 11 +#define RT5645_M_DAC_R2_MONO_L (0x1 << 10) +#define RT5645_M_DAC_R2_MONO_L_SFT 10 +#define RT5645_DAC_R2_MONO_L_VOL_MASK (0x1 << 9) +#define RT5645_DAC_R2_MONO_L_VOL_SFT 9 +#define RT5645_M_DAC_R1_MONO_R (0x1 << 6) +#define RT5645_M_DAC_R1_MONO_R_SFT 6 +#define RT5645_DAC_R1_MONO_R_VOL_MASK (0x1 << 5) +#define RT5645_DAC_R1_MONO_R_VOL_SFT 5 +#define RT5645_M_DAC_R2_MONO_R (0x1 << 4) +#define RT5645_M_DAC_R2_MONO_R_SFT 4 +#define RT5645_DAC_R2_MONO_R_VOL_MASK (0x1 << 3) +#define RT5645_DAC_R2_MONO_R_VOL_SFT 3 +#define RT5645_M_DAC_L2_MONO_R (0x1 << 2) +#define RT5645_M_DAC_L2_MONO_R_SFT 2 +#define RT5645_DAC_L2_MONO_R_VOL_MASK (0x1 << 1) +#define RT5645_DAC_L2_MONO_R_VOL_SFT 1 + +/* Digital Mixer Control (0x2c) */ +#define RT5645_M_STO_L_DAC_L (0x1 << 15) +#define RT5645_M_STO_L_DAC_L_SFT 15 +#define RT5645_STO_L_DAC_L_VOL_MASK (0x1 << 14) +#define RT5645_STO_L_DAC_L_VOL_SFT 14 +#define RT5645_M_DAC_L2_DAC_L (0x1 << 13) +#define RT5645_M_DAC_L2_DAC_L_SFT 13 +#define RT5645_DAC_L2_DAC_L_VOL_MASK (0x1 << 12) +#define RT5645_DAC_L2_DAC_L_VOL_SFT 12 +#define RT5645_M_STO_R_DAC_R (0x1 << 11) +#define RT5645_M_STO_R_DAC_R_SFT 11 +#define RT5645_STO_R_DAC_R_VOL_MASK (0x1 << 10) +#define RT5645_STO_R_DAC_R_VOL_SFT 10 +#define RT5645_M_DAC_R2_DAC_R (0x1 << 9) +#define RT5645_M_DAC_R2_DAC_R_SFT 9 +#define RT5645_DAC_R2_DAC_R_VOL_MASK (0x1 << 8) +#define RT5645_DAC_R2_DAC_R_VOL_SFT 8 +#define RT5645_M_DAC_R2_DAC_L (0x1 << 7) +#define RT5645_M_DAC_R2_DAC_L_SFT 7 +#define RT5645_DAC_R2_DAC_L_VOL_MASK (0x1 << 6) +#define RT5645_DAC_R2_DAC_L_VOL_SFT 6 +#define RT5645_M_DAC_L2_DAC_R (0x1 << 5) +#define RT5645_M_DAC_L2_DAC_R_SFT 5 +#define RT5645_DAC_L2_DAC_R_VOL_MASK (0x1 << 4) +#define RT5645_DAC_L2_DAC_R_VOL_SFT 4 + +/* Digital Interface Data Control (0x2f) */ +#define RT5645_IF1_ADC2_IN_SEL (0x1 << 15) +#define RT5645_IF1_ADC2_IN_SFT 15 +#define RT5645_IF2_ADC_IN_MASK (0x7 << 12) +#define RT5645_IF2_ADC_IN_SFT 12 +#define RT5645_IF2_DAC_SEL_MASK (0x3 << 10) +#define RT5645_IF2_DAC_SEL_SFT 10 +#define RT5645_IF2_ADC_SEL_MASK (0x3 << 8) +#define RT5645_IF2_ADC_SEL_SFT 8 +#define RT5645_IF3_DAC_SEL_MASK (0x3 << 6) +#define RT5645_IF3_DAC_SEL_SFT 6 +#define RT5645_IF3_ADC_SEL_MASK (0x3 << 4) +#define RT5645_IF3_ADC_SEL_SFT 4 +#define RT5645_IF3_ADC_IN_MASK (0x7) +#define RT5645_IF3_ADC_IN_SFT 0 + +/* PDM Output Control (0x31) */ +#define RT5645_PDM1_L_MASK (0x1 << 15) +#define RT5645_PDM1_L_SFT 15 +#define RT5645_M_PDM1_L (0x1 << 14) +#define RT5645_M_PDM1_L_SFT 14 +#define RT5645_PDM1_R_MASK (0x1 << 13) +#define RT5645_PDM1_R_SFT 13 +#define RT5645_M_PDM1_R (0x1 << 12) +#define RT5645_M_PDM1_R_SFT 12 +#define RT5645_PDM2_L_MASK (0x1 << 11) +#define RT5645_PDM2_L_SFT 11 +#define RT5645_M_PDM2_L (0x1 << 10) +#define RT5645_M_PDM2_L_SFT 10 +#define RT5645_PDM2_R_MASK (0x1 << 9) +#define RT5645_PDM2_R_SFT 9 +#define RT5645_M_PDM2_R (0x1 << 8) +#define RT5645_M_PDM2_R_SFT 8 +#define RT5645_PDM2_BUSY (0x1 << 7) +#define RT5645_PDM1_BUSY (0x1 << 6) +#define RT5645_PDM_PATTERN (0x1 << 5) +#define RT5645_PDM_GAIN (0x1 << 4) +#define RT5645_PDM_DIV_MASK (0x3) + +/* REC Left Mixer Control 1 (0x3b) */ +#define RT5645_G_HP_L_RM_L_MASK (0x7 << 13) +#define RT5645_G_HP_L_RM_L_SFT 13 +#define RT5645_G_IN_L_RM_L_MASK (0x7 << 10) +#define RT5645_G_IN_L_RM_L_SFT 10 +#define RT5645_G_BST4_RM_L_MASK (0x7 << 7) +#define RT5645_G_BST4_RM_L_SFT 7 +#define RT5645_G_BST3_RM_L_MASK (0x7 << 4) +#define RT5645_G_BST3_RM_L_SFT 4 +#define RT5645_G_BST2_RM_L_MASK (0x7 << 1) +#define RT5645_G_BST2_RM_L_SFT 1 + +/* REC Left Mixer Control 2 (0x3c) */ +#define RT5645_G_BST1_RM_L_MASK (0x7 << 13) +#define RT5645_G_BST1_RM_L_SFT 13 +#define RT5645_G_OM_L_RM_L_MASK (0x7 << 10) +#define RT5645_G_OM_L_RM_L_SFT 10 +#define RT5645_M_MM_L_RM_L (0x1 << 6) +#define RT5645_M_MM_L_RM_L_SFT 6 +#define RT5645_M_IN_L_RM_L (0x1 << 5) +#define RT5645_M_IN_L_RM_L_SFT 5 +#define RT5645_M_HP_L_RM_L (0x1 << 4) +#define RT5645_M_HP_L_RM_L_SFT 4 +#define RT5645_M_BST3_RM_L (0x1 << 3) +#define RT5645_M_BST3_RM_L_SFT 3 +#define RT5645_M_BST2_RM_L (0x1 << 2) +#define RT5645_M_BST2_RM_L_SFT 2 +#define RT5645_M_BST1_RM_L (0x1 << 1) +#define RT5645_M_BST1_RM_L_SFT 1 +#define RT5645_M_OM_L_RM_L (0x1) +#define RT5645_M_OM_L_RM_L_SFT 0 + +/* REC Right Mixer Control 1 (0x3d) */ +#define RT5645_G_HP_R_RM_R_MASK (0x7 << 13) +#define RT5645_G_HP_R_RM_R_SFT 13 +#define RT5645_G_IN_R_RM_R_MASK (0x7 << 10) +#define RT5645_G_IN_R_RM_R_SFT 10 +#define RT5645_G_BST4_RM_R_MASK (0x7 << 7) +#define RT5645_G_BST4_RM_R_SFT 7 +#define RT5645_G_BST3_RM_R_MASK (0x7 << 4) +#define RT5645_G_BST3_RM_R_SFT 4 +#define RT5645_G_BST2_RM_R_MASK (0x7 << 1) +#define RT5645_G_BST2_RM_R_SFT 1 + +/* REC Right Mixer Control 2 (0x3e) */ +#define RT5645_G_BST1_RM_R_MASK (0x7 << 13) +#define RT5645_G_BST1_RM_R_SFT 13 +#define RT5645_G_OM_R_RM_R_MASK (0x7 << 10) +#define RT5645_G_OM_R_RM_R_SFT 10 +#define RT5645_M_MM_R_RM_R (0x1 << 6) +#define RT5645_M_MM_R_RM_R_SFT 6 +#define RT5645_M_IN_R_RM_R (0x1 << 5) +#define RT5645_M_IN_R_RM_R_SFT 5 +#define RT5645_M_HP_R_RM_R (0x1 << 4) +#define RT5645_M_HP_R_RM_R_SFT 4 +#define RT5645_M_BST3_RM_R (0x1 << 3) +#define RT5645_M_BST3_RM_R_SFT 3 +#define RT5645_M_BST2_RM_R (0x1 << 2) +#define RT5645_M_BST2_RM_R_SFT 2 +#define RT5645_M_BST1_RM_R (0x1 << 1) +#define RT5645_M_BST1_RM_R_SFT 1 +#define RT5645_M_OM_R_RM_R (0x1) +#define RT5645_M_OM_R_RM_R_SFT 0 + +/* HPOMIX Control (0x40) (0x42) */ +#define RT5645_M_BST1_HV (0x1 << 4) +#define RT5645_M_BST1_HV_SFT 4 +#define RT5645_M_BST2_HV (0x1 << 4) +#define RT5645_M_BST2_HV_SFT 4 +#define RT5645_M_BST3_HV (0x1 << 3) +#define RT5645_M_BST3_HV_SFT 3 +#define RT5645_M_IN_HV (0x1 << 2) +#define RT5645_M_IN_HV_SFT 2 +#define RT5645_M_DAC2_HV (0x1 << 1) +#define RT5645_M_DAC2_HV_SFT 1 +#define RT5645_M_DAC1_HV (0x1 << 0) +#define RT5645_M_DAC1_HV_SFT 0 + +/* HPMIX Control (0x45) */ +#define RT5645_M_DAC1_HM (0x1 << 14) +#define RT5645_M_DAC1_HM_SFT 14 +#define RT5645_M_HPVOL_HM (0x1 << 13) +#define RT5645_M_HPVOL_HM_SFT 13 + +/* SPK Left Mixer Control (0x46) */ +#define RT5645_G_RM_L_SM_L_MASK (0x3 << 14) +#define RT5645_G_RM_L_SM_L_SFT 14 +#define RT5645_G_IN_L_SM_L_MASK (0x3 << 12) +#define RT5645_G_IN_L_SM_L_SFT 12 +#define RT5645_G_DAC_L1_SM_L_MASK (0x3 << 10) +#define RT5645_G_DAC_L1_SM_L_SFT 10 +#define RT5645_G_DAC_L2_SM_L_MASK (0x3 << 8) +#define RT5645_G_DAC_L2_SM_L_SFT 8 +#define RT5645_G_OM_L_SM_L_MASK (0x3 << 6) +#define RT5645_G_OM_L_SM_L_SFT 6 +#define RT5645_M_BST1_L_SM_L (0x1 << 5) +#define RT5645_M_BST1_L_SM_L_SFT 5 +#define RT5645_M_IN_L_SM_L (0x1 << 3) +#define RT5645_M_IN_L_SM_L_SFT 3 +#define RT5645_M_DAC_L1_SM_L (0x1 << 1) +#define RT5645_M_DAC_L1_SM_L_SFT 1 +#define RT5645_M_DAC_L2_SM_L (0x1 << 2) +#define RT5645_M_DAC_L2_SM_L_SFT 2 +#define RT5645_M_BST3_L_SM_L (0x1 << 4) +#define RT5645_M_BST3_L_SM_L_SFT 4 + +/* SPK Right Mixer Control (0x47) */ +#define RT5645_G_RM_R_SM_R_MASK (0x3 << 14) +#define RT5645_G_RM_R_SM_R_SFT 14 +#define RT5645_G_IN_R_SM_R_MASK (0x3 << 12) +#define RT5645_G_IN_R_SM_R_SFT 12 +#define RT5645_G_DAC_R1_SM_R_MASK (0x3 << 10) +#define RT5645_G_DAC_R1_SM_R_SFT 10 +#define RT5645_G_DAC_R2_SM_R_MASK (0x3 << 8) +#define RT5645_G_DAC_R2_SM_R_SFT 8 +#define RT5645_G_OM_R_SM_R_MASK (0x3 << 6) +#define RT5645_G_OM_R_SM_R_SFT 6 +#define RT5645_M_BST2_R_SM_R (0x1 << 5) +#define RT5645_M_BST2_R_SM_R_SFT 5 +#define RT5645_M_IN_R_SM_R (0x1 << 3) +#define RT5645_M_IN_R_SM_R_SFT 3 +#define RT5645_M_DAC_R1_SM_R (0x1 << 1) +#define RT5645_M_DAC_R1_SM_R_SFT 1 +#define RT5645_M_DAC_R2_SM_R (0x1 << 2) +#define RT5645_M_DAC_R2_SM_R_SFT 2 +#define RT5645_M_BST3_R_SM_R (0x1 << 4) +#define RT5645_M_BST3_R_SM_R_SFT 4 + +/* SPOLMIX Control (0x48) */ +#define RT5645_M_DAC_L1_SPM_L (0x1 << 15) +#define RT5645_M_DAC_L1_SPM_L_SFT 15 +#define RT5645_M_DAC_R1_SPM_L (0x1 << 14) +#define RT5645_M_DAC_R1_SPM_L_SFT 14 +#define RT5645_M_SV_L_SPM_L (0x1 << 13) +#define RT5645_M_SV_L_SPM_L_SFT 13 +#define RT5645_M_SV_R_SPM_L (0x1 << 12) +#define RT5645_M_SV_R_SPM_L_SFT 12 +#define RT5645_M_BST3_SPM_L (0x1 << 11) +#define RT5645_M_BST3_SPM_L_SFT 11 +#define RT5645_M_DAC_R1_SPM_R (0x1 << 2) +#define RT5645_M_DAC_R1_SPM_R_SFT 2 +#define RT5645_M_BST3_SPM_R (0x1 << 1) +#define RT5645_M_BST3_SPM_R_SFT 1 +#define RT5645_M_SV_R_SPM_R (0x1 << 0) +#define RT5645_M_SV_R_SPM_R_SFT 0 + +/* Mono Output Mixer Control (0x4c) */ +#define RT5645_M_OV_L_MM (0x1 << 9) +#define RT5645_M_OV_L_MM_SFT 9 +#define RT5645_M_DAC_L2_MA (0x1 << 8) +#define RT5645_M_DAC_L2_MA_SFT 8 +#define RT5645_G_MONOMIX_MASK (0x1 << 10) +#define RT5645_G_MONOMIX_SFT 10 +#define RT5645_M_BST2_MM (0x1 << 4) +#define RT5645_M_BST2_MM_SFT 4 +#define RT5645_M_DAC_R1_MM (0x1 << 3) +#define RT5645_M_DAC_R1_MM_SFT 3 +#define RT5645_M_DAC_R2_MM (0x1 << 2) +#define RT5645_M_DAC_R2_MM_SFT 2 +#define RT5645_M_DAC_L2_MM (0x1 << 1) +#define RT5645_M_DAC_L2_MM_SFT 1 +#define RT5645_M_BST3_MM (0x1 << 0) +#define RT5645_M_BST3_MM_SFT 0 + +/* Output Left Mixer Control 1 (0x4d) */ +#define RT5645_G_BST3_OM_L_MASK (0x7 << 13) +#define RT5645_G_BST3_OM_L_SFT 13 +#define RT5645_G_BST2_OM_L_MASK (0x7 << 10) +#define RT5645_G_BST2_OM_L_SFT 10 +#define RT5645_G_BST1_OM_L_MASK (0x7 << 7) +#define RT5645_G_BST1_OM_L_SFT 7 +#define RT5645_G_IN_L_OM_L_MASK (0x7 << 4) +#define RT5645_G_IN_L_OM_L_SFT 4 +#define RT5645_G_RM_L_OM_L_MASK (0x7 << 1) +#define RT5645_G_RM_L_OM_L_SFT 1 + +/* Output Left Mixer Control 2 (0x4e) */ +#define RT5645_G_DAC_R2_OM_L_MASK (0x7 << 13) +#define RT5645_G_DAC_R2_OM_L_SFT 13 +#define RT5645_G_DAC_L2_OM_L_MASK (0x7 << 10) +#define RT5645_G_DAC_L2_OM_L_SFT 10 +#define RT5645_G_DAC_L1_OM_L_MASK (0x7 << 7) +#define RT5645_G_DAC_L1_OM_L_SFT 7 + +/* Output Left Mixer Control 3 (0x4f) */ +#define RT5645_M_BST3_OM_L (0x1 << 4) +#define RT5645_M_BST3_OM_L_SFT 4 +#define RT5645_M_BST1_OM_L (0x1 << 3) +#define RT5645_M_BST1_OM_L_SFT 3 +#define RT5645_M_IN_L_OM_L (0x1 << 2) +#define RT5645_M_IN_L_OM_L_SFT 2 +#define RT5645_M_DAC_L2_OM_L (0x1 << 1) +#define RT5645_M_DAC_L2_OM_L_SFT 1 +#define RT5645_M_DAC_L1_OM_L (0x1) +#define RT5645_M_DAC_L1_OM_L_SFT 0 + +/* Output Right Mixer Control 1 (0x50) */ +#define RT5645_G_BST4_OM_R_MASK (0x7 << 13) +#define RT5645_G_BST4_OM_R_SFT 13 +#define RT5645_G_BST2_OM_R_MASK (0x7 << 10) +#define RT5645_G_BST2_OM_R_SFT 10 +#define RT5645_G_BST1_OM_R_MASK (0x7 << 7) +#define RT5645_G_BST1_OM_R_SFT 7 +#define RT5645_G_IN_R_OM_R_MASK (0x7 << 4) +#define RT5645_G_IN_R_OM_R_SFT 4 +#define RT5645_G_RM_R_OM_R_MASK (0x7 << 1) +#define RT5645_G_RM_R_OM_R_SFT 1 + +/* Output Right Mixer Control 2 (0x51) */ +#define RT5645_G_DAC_L2_OM_R_MASK (0x7 << 13) +#define RT5645_G_DAC_L2_OM_R_SFT 13 +#define RT5645_G_DAC_R2_OM_R_MASK (0x7 << 10) +#define RT5645_G_DAC_R2_OM_R_SFT 10 +#define RT5645_G_DAC_R1_OM_R_MASK (0x7 << 7) +#define RT5645_G_DAC_R1_OM_R_SFT 7 + +/* Output Right Mixer Control 3 (0x52) */ +#define RT5645_M_BST3_OM_R (0x1 << 4) +#define RT5645_M_BST3_OM_R_SFT 4 +#define RT5645_M_BST2_OM_R (0x1 << 3) +#define RT5645_M_BST2_OM_R_SFT 3 +#define RT5645_M_IN_R_OM_R (0x1 << 2) +#define RT5645_M_IN_R_OM_R_SFT 2 +#define RT5645_M_DAC_R2_OM_R (0x1 << 1) +#define RT5645_M_DAC_R2_OM_R_SFT 1 +#define RT5645_M_DAC_R1_OM_R (0x1) +#define RT5645_M_DAC_R1_OM_R_SFT 0 + +/* LOUT Mixer Control (0x53) */ +#define RT5645_M_DAC_L1_LM (0x1 << 15) +#define RT5645_M_DAC_L1_LM_SFT 15 +#define RT5645_M_DAC_R1_LM (0x1 << 14) +#define RT5645_M_DAC_R1_LM_SFT 14 +#define RT5645_M_OV_L_LM (0x1 << 13) +#define RT5645_M_OV_L_LM_SFT 13 +#define RT5645_M_OV_R_LM (0x1 << 12) +#define RT5645_M_OV_R_LM_SFT 12 +#define RT5645_G_LOUTMIX_MASK (0x1 << 11) +#define RT5645_G_LOUTMIX_SFT 11 + +/* Power Management for Digital 1 (0x61) */ +#define RT5645_PWR_I2S1 (0x1 << 15) +#define RT5645_PWR_I2S1_BIT 15 +#define RT5645_PWR_I2S2 (0x1 << 14) +#define RT5645_PWR_I2S2_BIT 14 +#define RT5645_PWR_I2S3 (0x1 << 13) +#define RT5645_PWR_I2S3_BIT 13 +#define RT5645_PWR_DAC_L1 (0x1 << 12) +#define RT5645_PWR_DAC_L1_BIT 12 +#define RT5645_PWR_DAC_R1 (0x1 << 11) +#define RT5645_PWR_DAC_R1_BIT 11 +#define RT5645_PWR_CLS_D_R (0x1 << 9) +#define RT5645_PWR_CLS_D_R_BIT 9 +#define RT5645_PWR_CLS_D_L (0x1 << 8) +#define RT5645_PWR_CLS_D_L_BIT 8 +#define RT5645_PWR_ADC_R (0x1 << 1) +#define RT5645_PWR_ADC_R_BIT 1 +#define RT5645_PWR_DAC_L2 (0x1 << 7) +#define RT5645_PWR_DAC_L2_BIT 7 +#define RT5645_PWR_DAC_R2 (0x1 << 6) +#define RT5645_PWR_DAC_R2_BIT 6 +#define RT5645_PWR_ADC_L (0x1 << 2) +#define RT5645_PWR_ADC_L_BIT 2 +#define RT5645_PWR_ADC_R (0x1 << 1) +#define RT5645_PWR_ADC_R_BIT 1 +#define RT5645_PWR_CLS_D (0x1) +#define RT5645_PWR_CLS_D_BIT 0 + +/* Power Management for Digital 2 (0x62) */ +#define RT5645_PWR_ADC_S1F (0x1 << 15) +#define RT5645_PWR_ADC_S1F_BIT 15 +#define RT5645_PWR_ADC_MF_L (0x1 << 14) +#define RT5645_PWR_ADC_MF_L_BIT 14 +#define RT5645_PWR_ADC_MF_R (0x1 << 13) +#define RT5645_PWR_ADC_MF_R_BIT 13 +#define RT5645_PWR_I2S_DSP (0x1 << 12) +#define RT5645_PWR_I2S_DSP_BIT 12 +#define RT5645_PWR_DAC_S1F (0x1 << 11) +#define RT5645_PWR_DAC_S1F_BIT 11 +#define RT5645_PWR_DAC_MF_L (0x1 << 10) +#define RT5645_PWR_DAC_MF_L_BIT 10 +#define RT5645_PWR_DAC_MF_R (0x1 << 9) +#define RT5645_PWR_DAC_MF_R_BIT 9 +#define RT5645_PWR_ADC_S2F (0x1 << 8) +#define RT5645_PWR_ADC_S2F_BIT 8 +#define RT5645_PWR_PDM1 (0x1 << 7) +#define RT5645_PWR_PDM1_BIT 7 +#define RT5645_PWR_PDM2 (0x1 << 6) +#define RT5645_PWR_PDM2_BIT 6 +#define RT5645_PWR_IPTV (0x1 << 1) +#define RT5645_PWR_IPTV_BIT 1 +#define RT5645_PWR_PAD (0x1) +#define RT5645_PWR_PAD_BIT 0 + +/* Power Management for Analog 1 (0x63) */ +#define RT5645_PWR_VREF1 (0x1 << 15) +#define RT5645_PWR_VREF1_BIT 15 +#define RT5645_PWR_FV1 (0x1 << 14) +#define RT5645_PWR_FV1_BIT 14 +#define RT5645_PWR_MB (0x1 << 13) +#define RT5645_PWR_MB_BIT 13 +#define RT5645_PWR_LM (0x1 << 12) +#define RT5645_PWR_LM_BIT 12 +#define RT5645_PWR_BG (0x1 << 11) +#define RT5645_PWR_BG_BIT 11 +#define RT5645_PWR_MA (0x1 << 10) +#define RT5645_PWR_MA_BIT 10 +#define RT5645_PWR_HP_L (0x1 << 7) +#define RT5645_PWR_HP_L_BIT 7 +#define RT5645_PWR_HP_R (0x1 << 6) +#define RT5645_PWR_HP_R_BIT 6 +#define RT5645_PWR_HA (0x1 << 5) +#define RT5645_PWR_HA_BIT 5 +#define RT5645_PWR_VREF2 (0x1 << 4) +#define RT5645_PWR_VREF2_BIT 4 +#define RT5645_PWR_FV2 (0x1 << 3) +#define RT5645_PWR_FV2_BIT 3 +#define RT5645_LDO_SEL_MASK (0x3) +#define RT5645_LDO_SEL_SFT 0 + +/* Power Management for Analog 2 (0x64) */ +#define RT5645_PWR_BST1 (0x1 << 15) +#define RT5645_PWR_BST1_BIT 15 +#define RT5645_PWR_BST2 (0x1 << 14) +#define RT5645_PWR_BST2_BIT 14 +#define RT5645_PWR_BST3 (0x1 << 13) +#define RT5645_PWR_BST3_BIT 13 +#define RT5645_PWR_BST4 (0x1 << 12) +#define RT5645_PWR_BST4_BIT 12 +#define RT5645_PWR_MB1 (0x1 << 11) +#define RT5645_PWR_MB1_BIT 11 +#define RT5645_PWR_MB2 (0x1 << 10) +#define RT5645_PWR_MB2_BIT 10 +#define RT5645_PWR_PLL (0x1 << 9) +#define RT5645_PWR_PLL_BIT 9 +#define RT5645_PWR_BST2_P (0x1 << 5) +#define RT5645_PWR_BST2_P_BIT 5 +#define RT5645_PWR_BST3_P (0x1 << 4) +#define RT5645_PWR_BST3_P_BIT 4 +#define RT5645_PWR_BST4_P (0x1 << 3) +#define RT5645_PWR_BST4_P_BIT 3 +#define RT5645_PWR_JD1 (0x1 << 2) +#define RT5645_PWR_JD1_BIT 2 +#define RT5645_PWR_JD (0x1 << 1) +#define RT5645_PWR_JD_BIT 1 + +/* Power Management for Mixer (0x65) */ +#define RT5645_PWR_OM_L (0x1 << 15) +#define RT5645_PWR_OM_L_BIT 15 +#define RT5645_PWR_OM_R (0x1 << 14) +#define RT5645_PWR_OM_R_BIT 14 +#define RT5645_PWR_SM_L (0x1 << 13) +#define RT5645_PWR_SM_L_BIT 13 +#define RT5645_PWR_SM_R (0x1 << 12) +#define RT5645_PWR_SM_R_BIT 12 +#define RT5645_PWR_RM_L (0x1 << 11) +#define RT5645_PWR_RM_L_BIT 11 +#define RT5645_PWR_RM_R (0x1 << 10) +#define RT5645_PWR_RM_R_BIT 10 +#define RT5645_PWR_MM (0x1 << 8) +#define RT5645_PWR_MM_BIT 8 +#define RT5645_PWR_HM_L (0x1 << 7) +#define RT5645_PWR_HM_L_BIT 7 +#define RT5645_PWR_HM_R (0x1 << 6) +#define RT5645_PWR_HM_R_BIT 6 +#define RT5645_PWR_LDO2 (0x1 << 1) +#define RT5645_PWR_LDO2_BIT 1 + +/* Power Management for Volume (0x66) */ +#define RT5645_PWR_SV_L (0x1 << 15) +#define RT5645_PWR_SV_L_BIT 15 +#define RT5645_PWR_SV_R (0x1 << 14) +#define RT5645_PWR_SV_R_BIT 14 +#define RT5645_PWR_HV_L (0x1 << 11) +#define RT5645_PWR_HV_L_BIT 11 +#define RT5645_PWR_HV_R (0x1 << 10) +#define RT5645_PWR_HV_R_BIT 10 +#define RT5645_PWR_IN_L (0x1 << 9) +#define RT5645_PWR_IN_L_BIT 9 +#define RT5645_PWR_IN_R (0x1 << 8) +#define RT5645_PWR_IN_R_BIT 8 +#define RT5645_PWR_MIC_DET (0x1 << 5) +#define RT5645_PWR_MIC_DET_BIT 5 + +/* I2S1/2 Audio Serial Data Port Control (0x70 0x71) */ +#define RT5645_I2S_MS_MASK (0x1 << 15) +#define RT5645_I2S_MS_SFT 15 +#define RT5645_I2S_MS_M (0x0 << 15) +#define RT5645_I2S_MS_S (0x1 << 15) +#define RT5645_I2S_O_CP_MASK (0x3 << 10) +#define RT5645_I2S_O_CP_SFT 10 +#define RT5645_I2S_O_CP_OFF (0x0 << 10) +#define RT5645_I2S_O_CP_U_LAW (0x1 << 10) +#define RT5645_I2S_O_CP_A_LAW (0x2 << 10) +#define RT5645_I2S_I_CP_MASK (0x3 << 8) +#define RT5645_I2S_I_CP_SFT 8 +#define RT5645_I2S_I_CP_OFF (0x0 << 8) +#define RT5645_I2S_I_CP_U_LAW (0x1 << 8) +#define RT5645_I2S_I_CP_A_LAW (0x2 << 8) +#define RT5645_I2S_BP_MASK (0x1 << 7) +#define RT5645_I2S_BP_SFT 7 +#define RT5645_I2S_BP_NOR (0x0 << 7) +#define RT5645_I2S_BP_INV (0x1 << 7) +#define RT5645_I2S_DL_MASK (0x3 << 2) +#define RT5645_I2S_DL_SFT 2 +#define RT5645_I2S_DL_16 (0x0 << 2) +#define RT5645_I2S_DL_20 (0x1 << 2) +#define RT5645_I2S_DL_24 (0x2 << 2) +#define RT5645_I2S_DL_8 (0x3 << 2) +#define RT5645_I2S_DF_MASK (0x3) +#define RT5645_I2S_DF_SFT 0 +#define RT5645_I2S_DF_I2S (0x0) +#define RT5645_I2S_DF_LEFT (0x1) +#define RT5645_I2S_DF_PCM_A (0x2) +#define RT5645_I2S_DF_PCM_B (0x3) + +/* I2S2 Audio Serial Data Port Control (0x71) */ +#define RT5645_I2S2_SDI_MASK (0x1 << 6) +#define RT5645_I2S2_SDI_SFT 6 +#define RT5645_I2S2_SDI_I2S1 (0x0 << 6) +#define RT5645_I2S2_SDI_I2S2 (0x1 << 6) + +/* ADC/DAC Clock Control 1 (0x73) */ +#define RT5645_I2S_BCLK_MS1_MASK (0x1 << 15) +#define RT5645_I2S_BCLK_MS1_SFT 15 +#define RT5645_I2S_BCLK_MS1_32 (0x0 << 15) +#define RT5645_I2S_BCLK_MS1_64 (0x1 << 15) +#define RT5645_I2S_PD1_MASK (0x7 << 12) +#define RT5645_I2S_PD1_SFT 12 +#define RT5645_I2S_PD1_1 (0x0 << 12) +#define RT5645_I2S_PD1_2 (0x1 << 12) +#define RT5645_I2S_PD1_3 (0x2 << 12) +#define RT5645_I2S_PD1_4 (0x3 << 12) +#define RT5645_I2S_PD1_6 (0x4 << 12) +#define RT5645_I2S_PD1_8 (0x5 << 12) +#define RT5645_I2S_PD1_12 (0x6 << 12) +#define RT5645_I2S_PD1_16 (0x7 << 12) +#define RT5645_I2S_BCLK_MS2_MASK (0x1 << 11) +#define RT5645_I2S_BCLK_MS2_SFT 11 +#define RT5645_I2S_BCLK_MS2_32 (0x0 << 11) +#define RT5645_I2S_BCLK_MS2_64 (0x1 << 11) +#define RT5645_I2S_PD2_MASK (0x7 << 8) +#define RT5645_I2S_PD2_SFT 8 +#define RT5645_I2S_PD2_1 (0x0 << 8) +#define RT5645_I2S_PD2_2 (0x1 << 8) +#define RT5645_I2S_PD2_3 (0x2 << 8) +#define RT5645_I2S_PD2_4 (0x3 << 8) +#define RT5645_I2S_PD2_6 (0x4 << 8) +#define RT5645_I2S_PD2_8 (0x5 << 8) +#define RT5645_I2S_PD2_12 (0x6 << 8) +#define RT5645_I2S_PD2_16 (0x7 << 8) +#define RT5645_I2S_BCLK_MS3_MASK (0x1 << 7) +#define RT5645_I2S_BCLK_MS3_SFT 7 +#define RT5645_I2S_BCLK_MS3_32 (0x0 << 7) +#define RT5645_I2S_BCLK_MS3_64 (0x1 << 7) +#define RT5645_I2S_PD3_MASK (0x7 << 4) +#define RT5645_I2S_PD3_SFT 4 +#define RT5645_I2S_PD3_1 (0x0 << 4) +#define RT5645_I2S_PD3_2 (0x1 << 4) +#define RT5645_I2S_PD3_3 (0x2 << 4) +#define RT5645_I2S_PD3_4 (0x3 << 4) +#define RT5645_I2S_PD3_6 (0x4 << 4) +#define RT5645_I2S_PD3_8 (0x5 << 4) +#define RT5645_I2S_PD3_12 (0x6 << 4) +#define RT5645_I2S_PD3_16 (0x7 << 4) +#define RT5645_DAC_OSR_MASK (0x3 << 2) +#define RT5645_DAC_OSR_SFT 2 +#define RT5645_DAC_OSR_128 (0x0 << 2) +#define RT5645_DAC_OSR_64 (0x1 << 2) +#define RT5645_DAC_OSR_32 (0x2 << 2) +#define RT5645_DAC_OSR_16 (0x3 << 2) +#define RT5645_ADC_OSR_MASK (0x3) +#define RT5645_ADC_OSR_SFT 0 +#define RT5645_ADC_OSR_128 (0x0) +#define RT5645_ADC_OSR_64 (0x1) +#define RT5645_ADC_OSR_32 (0x2) +#define RT5645_ADC_OSR_16 (0x3) + +/* ADC/DAC Clock Control 2 (0x74) */ +#define RT5645_DAC_L_OSR_MASK (0x3 << 14) +#define RT5645_DAC_L_OSR_SFT 14 +#define RT5645_DAC_L_OSR_128 (0x0 << 14) +#define RT5645_DAC_L_OSR_64 (0x1 << 14) +#define RT5645_DAC_L_OSR_32 (0x2 << 14) +#define RT5645_DAC_L_OSR_16 (0x3 << 14) +#define RT5645_ADC_R_OSR_MASK (0x3 << 12) +#define RT5645_ADC_R_OSR_SFT 12 +#define RT5645_ADC_R_OSR_128 (0x0 << 12) +#define RT5645_ADC_R_OSR_64 (0x1 << 12) +#define RT5645_ADC_R_OSR_32 (0x2 << 12) +#define RT5645_ADC_R_OSR_16 (0x3 << 12) +#define RT5645_DAHPF_EN (0x1 << 11) +#define RT5645_DAHPF_EN_SFT 11 +#define RT5645_ADHPF_EN (0x1 << 10) +#define RT5645_ADHPF_EN_SFT 10 + +/* Digital Microphone Control (0x75) */ +#define RT5645_DMIC_1_EN_MASK (0x1 << 15) +#define RT5645_DMIC_1_EN_SFT 15 +#define RT5645_DMIC_1_DIS (0x0 << 15) +#define RT5645_DMIC_1_EN (0x1 << 15) +#define RT5645_DMIC_2_EN_MASK (0x1 << 14) +#define RT5645_DMIC_2_EN_SFT 14 +#define RT5645_DMIC_2_DIS (0x0 << 14) +#define RT5645_DMIC_2_EN (0x1 << 14) +#define RT5645_DMIC_1L_LH_MASK (0x1 << 13) +#define RT5645_DMIC_1L_LH_SFT 13 +#define RT5645_DMIC_1L_LH_FALLING (0x0 << 13) +#define RT5645_DMIC_1L_LH_RISING (0x1 << 13) +#define RT5645_DMIC_1R_LH_MASK (0x1 << 12) +#define RT5645_DMIC_1R_LH_SFT 12 +#define RT5645_DMIC_1R_LH_FALLING (0x0 << 12) +#define RT5645_DMIC_1R_LH_RISING (0x1 << 12) +#define RT5645_DMIC_2_DP_MASK (0x3 << 10) +#define RT5645_DMIC_2_DP_SFT 10 +#define RT5645_DMIC_2_DP_GPIO6 (0x0 << 10) +#define RT5645_DMIC_2_DP_GPIO10 (0x1 << 10) +#define RT5645_DMIC_2_DP_GPIO12 (0x2 << 10) +#define RT5645_DMIC_2_DP_IN2P (0x3 << 10) +#define RT5645_DMIC_2L_LH_MASK (0x1 << 9) +#define RT5645_DMIC_2L_LH_SFT 9 +#define RT5645_DMIC_2L_LH_FALLING (0x0 << 9) +#define RT5645_DMIC_2L_LH_RISING (0x1 << 9) +#define RT5645_DMIC_2R_LH_MASK (0x1 << 8) +#define RT5645_DMIC_2R_LH_SFT 8 +#define RT5645_DMIC_2R_LH_FALLING (0x0 << 8) +#define RT5645_DMIC_2R_LH_RISING (0x1 << 8) +#define RT5645_DMIC_CLK_MASK (0x7 << 5) +#define RT5645_DMIC_CLK_SFT 5 +#define RT5645_DMIC_3_EN_MASK (0x1 << 4) +#define RT5645_DMIC_3_EN_SFT 4 +#define RT5645_DMIC_3_DIS (0x0 << 4) +#define RT5645_DMIC_3_EN (0x1 << 4) +#define RT5645_DMIC_1_DP_MASK (0x3 << 0) +#define RT5645_DMIC_1_DP_SFT 0 +#define RT5645_DMIC_1_DP_GPIO5 (0x0 << 0) +#define RT5645_DMIC_1_DP_IN2N (0x1 << 0) +#define RT5645_DMIC_1_DP_GPIO11 (0x2 << 0) + +/* TDM Control 1 (0x77) */ +#define RT5645_IF1_ADC_IN_MASK (0x3 << 8) +#define RT5645_IF1_ADC_IN_SFT 8 + +/* Global Clock Control (0x80) */ +#define RT5645_SCLK_SRC_MASK (0x3 << 14) +#define RT5645_SCLK_SRC_SFT 14 +#define RT5645_SCLK_SRC_MCLK (0x0 << 14) +#define RT5645_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5645_SCLK_SRC_RCCLK (0x2 << 14) /* 15MHz */ +#define RT5645_PLL1_SRC_MASK (0x3 << 12) +#define RT5645_PLL1_SRC_SFT 12 +#define RT5645_PLL1_SRC_MCLK (0x0 << 12) +#define RT5645_PLL1_SRC_BCLK1 (0x1 << 12) +#define RT5645_PLL1_SRC_BCLK2 (0x2 << 12) +#define RT5645_PLL1_SRC_BCLK3 (0x3 << 12) +#define RT5645_PLL1_PD_MASK (0x1 << 3) +#define RT5645_PLL1_PD_SFT 3 +#define RT5645_PLL1_PD_1 (0x0 << 3) +#define RT5645_PLL1_PD_2 (0x1 << 3) + +#define RT5645_PLL_INP_MAX 40000000 +#define RT5645_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x81) */ +#define RT5645_PLL_N_MAX 0x1ff +#define RT5645_PLL_N_MASK (RT5645_PLL_N_MAX << 7) +#define RT5645_PLL_N_SFT 7 +#define RT5645_PLL_K_MAX 0x1f +#define RT5645_PLL_K_MASK (RT5645_PLL_K_MAX) +#define RT5645_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x82) */ +#define RT5645_PLL_M_MAX 0xf +#define RT5645_PLL_M_MASK (RT5645_PLL_M_MAX << 12) +#define RT5645_PLL_M_SFT 12 +#define RT5645_PLL_M_BP (0x1 << 11) +#define RT5645_PLL_M_BP_SFT 11 + +/* ASRC Control 1 (0x83) */ +#define RT5645_STO_T_MASK (0x1 << 15) +#define RT5645_STO_T_SFT 15 +#define RT5645_STO_T_SCLK (0x0 << 15) +#define RT5645_STO_T_LRCK1 (0x1 << 15) +#define RT5645_M1_T_MASK (0x1 << 14) +#define RT5645_M1_T_SFT 14 +#define RT5645_M1_T_I2S2 (0x0 << 14) +#define RT5645_M1_T_I2S2_D3 (0x1 << 14) +#define RT5645_I2S2_F_MASK (0x1 << 12) +#define RT5645_I2S2_F_SFT 12 +#define RT5645_I2S2_F_I2S2_D2 (0x0 << 12) +#define RT5645_I2S2_F_I2S1_TCLK (0x1 << 12) +#define RT5645_DMIC_1_M_MASK (0x1 << 9) +#define RT5645_DMIC_1_M_SFT 9 +#define RT5645_DMIC_1_M_NOR (0x0 << 9) +#define RT5645_DMIC_1_M_ASYN (0x1 << 9) +#define RT5645_DMIC_2_M_MASK (0x1 << 8) +#define RT5645_DMIC_2_M_SFT 8 +#define RT5645_DMIC_2_M_NOR (0x0 << 8) +#define RT5645_DMIC_2_M_ASYN (0x1 << 8) + +/* ASRC Control 2 (0x84) */ +#define RT5645_MDA_L_M_MASK (0x1 << 15) +#define RT5645_MDA_L_M_SFT 15 +#define RT5645_MDA_L_M_NOR (0x0 << 15) +#define RT5645_MDA_L_M_ASYN (0x1 << 15) +#define RT5645_MDA_R_M_MASK (0x1 << 14) +#define RT5645_MDA_R_M_SFT 14 +#define RT5645_MDA_R_M_NOR (0x0 << 14) +#define RT5645_MDA_R_M_ASYN (0x1 << 14) +#define RT5645_MAD_L_M_MASK (0x1 << 13) +#define RT5645_MAD_L_M_SFT 13 +#define RT5645_MAD_L_M_NOR (0x0 << 13) +#define RT5645_MAD_L_M_ASYN (0x1 << 13) +#define RT5645_MAD_R_M_MASK (0x1 << 12) +#define RT5645_MAD_R_M_SFT 12 +#define RT5645_MAD_R_M_NOR (0x0 << 12) +#define RT5645_MAD_R_M_ASYN (0x1 << 12) +#define RT5645_ADC_M_MASK (0x1 << 11) +#define RT5645_ADC_M_SFT 11 +#define RT5645_ADC_M_NOR (0x0 << 11) +#define RT5645_ADC_M_ASYN (0x1 << 11) +#define RT5645_STO_DAC_M_MASK (0x1 << 5) +#define RT5645_STO_DAC_M_SFT 5 +#define RT5645_STO_DAC_M_NOR (0x0 << 5) +#define RT5645_STO_DAC_M_ASYN (0x1 << 5) +#define RT5645_I2S1_R_D_MASK (0x1 << 4) +#define RT5645_I2S1_R_D_SFT 4 +#define RT5645_I2S1_R_D_DIS (0x0 << 4) +#define RT5645_I2S1_R_D_EN (0x1 << 4) +#define RT5645_I2S2_R_D_MASK (0x1 << 3) +#define RT5645_I2S2_R_D_SFT 3 +#define RT5645_I2S2_R_D_DIS (0x0 << 3) +#define RT5645_I2S2_R_D_EN (0x1 << 3) +#define RT5645_PRE_SCLK_MASK (0x3) +#define RT5645_PRE_SCLK_SFT 0 +#define RT5645_PRE_SCLK_512 (0x0) +#define RT5645_PRE_SCLK_1024 (0x1) +#define RT5645_PRE_SCLK_2048 (0x2) + +/* ASRC Control 3 (0x85) */ +#define RT5645_I2S1_RATE_MASK (0xf << 12) +#define RT5645_I2S1_RATE_SFT 12 +#define RT5645_I2S2_RATE_MASK (0xf << 8) +#define RT5645_I2S2_RATE_SFT 8 + +/* ASRC Control 4 (0x89) */ +#define RT5645_I2S1_PD_MASK (0x7 << 12) +#define RT5645_I2S1_PD_SFT 12 +#define RT5645_I2S2_PD_MASK (0x7 << 8) +#define RT5645_I2S2_PD_SFT 8 + +/* HPOUT Over Current Detection (0x8b) */ +#define RT5645_HP_OVCD_MASK (0x1 << 10) +#define RT5645_HP_OVCD_SFT 10 +#define RT5645_HP_OVCD_DIS (0x0 << 10) +#define RT5645_HP_OVCD_EN (0x1 << 10) +#define RT5645_HP_OC_TH_MASK (0x3 << 8) +#define RT5645_HP_OC_TH_SFT 8 +#define RT5645_HP_OC_TH_90 (0x0 << 8) +#define RT5645_HP_OC_TH_105 (0x1 << 8) +#define RT5645_HP_OC_TH_120 (0x2 << 8) +#define RT5645_HP_OC_TH_135 (0x3 << 8) + +/* Class D Over Current Control (0x8c) */ +#define RT5645_CLSD_OC_MASK (0x1 << 9) +#define RT5645_CLSD_OC_SFT 9 +#define RT5645_CLSD_OC_PU (0x0 << 9) +#define RT5645_CLSD_OC_PD (0x1 << 9) +#define RT5645_AUTO_PD_MASK (0x1 << 8) +#define RT5645_AUTO_PD_SFT 8 +#define RT5645_AUTO_PD_DIS (0x0 << 8) +#define RT5645_AUTO_PD_EN (0x1 << 8) +#define RT5645_CLSD_OC_TH_MASK (0x3f) +#define RT5645_CLSD_OC_TH_SFT 0 + +/* Class D Output Control (0x8d) */ +#define RT5645_CLSD_RATIO_MASK (0xf << 12) +#define RT5645_CLSD_RATIO_SFT 12 +#define RT5645_CLSD_OM_MASK (0x1 << 11) +#define RT5645_CLSD_OM_SFT 11 +#define RT5645_CLSD_OM_MONO (0x0 << 11) +#define RT5645_CLSD_OM_STO (0x1 << 11) +#define RT5645_CLSD_SCH_MASK (0x1 << 10) +#define RT5645_CLSD_SCH_SFT 10 +#define RT5645_CLSD_SCH_L (0x0 << 10) +#define RT5645_CLSD_SCH_S (0x1 << 10) + +/* Depop Mode Control 1 (0x8e) */ +#define RT5645_SMT_TRIG_MASK (0x1 << 15) +#define RT5645_SMT_TRIG_SFT 15 +#define RT5645_SMT_TRIG_DIS (0x0 << 15) +#define RT5645_SMT_TRIG_EN (0x1 << 15) +#define RT5645_HP_L_SMT_MASK (0x1 << 9) +#define RT5645_HP_L_SMT_SFT 9 +#define RT5645_HP_L_SMT_DIS (0x0 << 9) +#define RT5645_HP_L_SMT_EN (0x1 << 9) +#define RT5645_HP_R_SMT_MASK (0x1 << 8) +#define RT5645_HP_R_SMT_SFT 8 +#define RT5645_HP_R_SMT_DIS (0x0 << 8) +#define RT5645_HP_R_SMT_EN (0x1 << 8) +#define RT5645_HP_CD_PD_MASK (0x1 << 7) +#define RT5645_HP_CD_PD_SFT 7 +#define RT5645_HP_CD_PD_DIS (0x0 << 7) +#define RT5645_HP_CD_PD_EN (0x1 << 7) +#define RT5645_RSTN_MASK (0x1 << 6) +#define RT5645_RSTN_SFT 6 +#define RT5645_RSTN_DIS (0x0 << 6) +#define RT5645_RSTN_EN (0x1 << 6) +#define RT5645_RSTP_MASK (0x1 << 5) +#define RT5645_RSTP_SFT 5 +#define RT5645_RSTP_DIS (0x0 << 5) +#define RT5645_RSTP_EN (0x1 << 5) +#define RT5645_HP_CO_MASK (0x1 << 4) +#define RT5645_HP_CO_SFT 4 +#define RT5645_HP_CO_DIS (0x0 << 4) +#define RT5645_HP_CO_EN (0x1 << 4) +#define RT5645_HP_CP_MASK (0x1 << 3) +#define RT5645_HP_CP_SFT 3 +#define RT5645_HP_CP_PD (0x0 << 3) +#define RT5645_HP_CP_PU (0x1 << 3) +#define RT5645_HP_SG_MASK (0x1 << 2) +#define RT5645_HP_SG_SFT 2 +#define RT5645_HP_SG_DIS (0x0 << 2) +#define RT5645_HP_SG_EN (0x1 << 2) +#define RT5645_HP_DP_MASK (0x1 << 1) +#define RT5645_HP_DP_SFT 1 +#define RT5645_HP_DP_PD (0x0 << 1) +#define RT5645_HP_DP_PU (0x1 << 1) +#define RT5645_HP_CB_MASK (0x1) +#define RT5645_HP_CB_SFT 0 +#define RT5645_HP_CB_PD (0x0) +#define RT5645_HP_CB_PU (0x1) + +/* Depop Mode Control 2 (0x8f) */ +#define RT5645_DEPOP_MASK (0x1 << 13) +#define RT5645_DEPOP_SFT 13 +#define RT5645_DEPOP_AUTO (0x0 << 13) +#define RT5645_DEPOP_MAN (0x1 << 13) +#define RT5645_RAMP_MASK (0x1 << 12) +#define RT5645_RAMP_SFT 12 +#define RT5645_RAMP_DIS (0x0 << 12) +#define RT5645_RAMP_EN (0x1 << 12) +#define RT5645_BPS_MASK (0x1 << 11) +#define RT5645_BPS_SFT 11 +#define RT5645_BPS_DIS (0x0 << 11) +#define RT5645_BPS_EN (0x1 << 11) +#define RT5645_FAST_UPDN_MASK (0x1 << 10) +#define RT5645_FAST_UPDN_SFT 10 +#define RT5645_FAST_UPDN_DIS (0x0 << 10) +#define RT5645_FAST_UPDN_EN (0x1 << 10) +#define RT5645_MRES_MASK (0x3 << 8) +#define RT5645_MRES_SFT 8 +#define RT5645_MRES_15MO (0x0 << 8) +#define RT5645_MRES_25MO (0x1 << 8) +#define RT5645_MRES_35MO (0x2 << 8) +#define RT5645_MRES_45MO (0x3 << 8) +#define RT5645_VLO_MASK (0x1 << 7) +#define RT5645_VLO_SFT 7 +#define RT5645_VLO_3V (0x0 << 7) +#define RT5645_VLO_32V (0x1 << 7) +#define RT5645_DIG_DP_MASK (0x1 << 6) +#define RT5645_DIG_DP_SFT 6 +#define RT5645_DIG_DP_DIS (0x0 << 6) +#define RT5645_DIG_DP_EN (0x1 << 6) +#define RT5645_DP_TH_MASK (0x3 << 4) +#define RT5645_DP_TH_SFT 4 + +/* Depop Mode Control 3 (0x90) */ +#define RT5645_CP_SYS_MASK (0x7 << 12) +#define RT5645_CP_SYS_SFT 12 +#define RT5645_CP_FQ1_MASK (0x7 << 8) +#define RT5645_CP_FQ1_SFT 8 +#define RT5645_CP_FQ2_MASK (0x7 << 4) +#define RT5645_CP_FQ2_SFT 4 +#define RT5645_CP_FQ3_MASK (0x7) +#define RT5645_CP_FQ3_SFT 0 +#define RT5645_CP_FQ_1_5_KHZ 0 +#define RT5645_CP_FQ_3_KHZ 1 +#define RT5645_CP_FQ_6_KHZ 2 +#define RT5645_CP_FQ_12_KHZ 3 +#define RT5645_CP_FQ_24_KHZ 4 +#define RT5645_CP_FQ_48_KHZ 5 +#define RT5645_CP_FQ_96_KHZ 6 +#define RT5645_CP_FQ_192_KHZ 7 + +/* PV detection and SPK gain control (0x92) */ +#define RT5645_PVDD_DET_MASK (0x1 << 15) +#define RT5645_PVDD_DET_SFT 15 +#define RT5645_PVDD_DET_DIS (0x0 << 15) +#define RT5645_PVDD_DET_EN (0x1 << 15) +#define RT5645_SPK_AG_MASK (0x1 << 14) +#define RT5645_SPK_AG_SFT 14 +#define RT5645_SPK_AG_DIS (0x0 << 14) +#define RT5645_SPK_AG_EN (0x1 << 14) + +/* Micbias Control (0x93) */ +#define RT5645_MIC1_BS_MASK (0x1 << 15) +#define RT5645_MIC1_BS_SFT 15 +#define RT5645_MIC1_BS_9AV (0x0 << 15) +#define RT5645_MIC1_BS_75AV (0x1 << 15) +#define RT5645_MIC2_BS_MASK (0x1 << 14) +#define RT5645_MIC2_BS_SFT 14 +#define RT5645_MIC2_BS_9AV (0x0 << 14) +#define RT5645_MIC2_BS_75AV (0x1 << 14) +#define RT5645_MIC1_CLK_MASK (0x1 << 13) +#define RT5645_MIC1_CLK_SFT 13 +#define RT5645_MIC1_CLK_DIS (0x0 << 13) +#define RT5645_MIC1_CLK_EN (0x1 << 13) +#define RT5645_MIC2_CLK_MASK (0x1 << 12) +#define RT5645_MIC2_CLK_SFT 12 +#define RT5645_MIC2_CLK_DIS (0x0 << 12) +#define RT5645_MIC2_CLK_EN (0x1 << 12) +#define RT5645_MIC1_OVCD_MASK (0x1 << 11) +#define RT5645_MIC1_OVCD_SFT 11 +#define RT5645_MIC1_OVCD_DIS (0x0 << 11) +#define RT5645_MIC1_OVCD_EN (0x1 << 11) +#define RT5645_MIC1_OVTH_MASK (0x3 << 9) +#define RT5645_MIC1_OVTH_SFT 9 +#define RT5645_MIC1_OVTH_600UA (0x0 << 9) +#define RT5645_MIC1_OVTH_1500UA (0x1 << 9) +#define RT5645_MIC1_OVTH_2000UA (0x2 << 9) +#define RT5645_MIC2_OVCD_MASK (0x1 << 8) +#define RT5645_MIC2_OVCD_SFT 8 +#define RT5645_MIC2_OVCD_DIS (0x0 << 8) +#define RT5645_MIC2_OVCD_EN (0x1 << 8) +#define RT5645_MIC2_OVTH_MASK (0x3 << 6) +#define RT5645_MIC2_OVTH_SFT 6 +#define RT5645_MIC2_OVTH_600UA (0x0 << 6) +#define RT5645_MIC2_OVTH_1500UA (0x1 << 6) +#define RT5645_MIC2_OVTH_2000UA (0x2 << 6) +#define RT5645_PWR_MB_MASK (0x1 << 5) +#define RT5645_PWR_MB_SFT 5 +#define RT5645_PWR_MB_PD (0x0 << 5) +#define RT5645_PWR_MB_PU (0x1 << 5) +#define RT5645_PWR_CLK25M_MASK (0x1 << 4) +#define RT5645_PWR_CLK25M_SFT 4 +#define RT5645_PWR_CLK25M_PD (0x0 << 4) +#define RT5645_PWR_CLK25M_PU (0x1 << 4) + +/* VAD Control 4 (0x9d) */ +#define RT5645_VAD_SEL_MASK (0x3 << 8) +#define RT5645_VAD_SEL_SFT 8 + +/* EQ Control 1 (0xb0) */ +#define RT5645_EQ_SRC_MASK (0x1 << 15) +#define RT5645_EQ_SRC_SFT 15 +#define RT5645_EQ_SRC_DAC (0x0 << 15) +#define RT5645_EQ_SRC_ADC (0x1 << 15) +#define RT5645_EQ_UPD (0x1 << 14) +#define RT5645_EQ_UPD_BIT 14 +#define RT5645_EQ_CD_MASK (0x1 << 13) +#define RT5645_EQ_CD_SFT 13 +#define RT5645_EQ_CD_DIS (0x0 << 13) +#define RT5645_EQ_CD_EN (0x1 << 13) +#define RT5645_EQ_DITH_MASK (0x3 << 8) +#define RT5645_EQ_DITH_SFT 8 +#define RT5645_EQ_DITH_NOR (0x0 << 8) +#define RT5645_EQ_DITH_LSB (0x1 << 8) +#define RT5645_EQ_DITH_LSB_1 (0x2 << 8) +#define RT5645_EQ_DITH_LSB_2 (0x3 << 8) + +/* EQ Control 2 (0xb1) */ +#define RT5645_EQ_HPF1_M_MASK (0x1 << 8) +#define RT5645_EQ_HPF1_M_SFT 8 +#define RT5645_EQ_HPF1_M_HI (0x0 << 8) +#define RT5645_EQ_HPF1_M_1ST (0x1 << 8) +#define RT5645_EQ_LPF1_M_MASK (0x1 << 7) +#define RT5645_EQ_LPF1_M_SFT 7 +#define RT5645_EQ_LPF1_M_LO (0x0 << 7) +#define RT5645_EQ_LPF1_M_1ST (0x1 << 7) +#define RT5645_EQ_HPF2_MASK (0x1 << 6) +#define RT5645_EQ_HPF2_SFT 6 +#define RT5645_EQ_HPF2_DIS (0x0 << 6) +#define RT5645_EQ_HPF2_EN (0x1 << 6) +#define RT5645_EQ_HPF1_MASK (0x1 << 5) +#define RT5645_EQ_HPF1_SFT 5 +#define RT5645_EQ_HPF1_DIS (0x0 << 5) +#define RT5645_EQ_HPF1_EN (0x1 << 5) +#define RT5645_EQ_BPF4_MASK (0x1 << 4) +#define RT5645_EQ_BPF4_SFT 4 +#define RT5645_EQ_BPF4_DIS (0x0 << 4) +#define RT5645_EQ_BPF4_EN (0x1 << 4) +#define RT5645_EQ_BPF3_MASK (0x1 << 3) +#define RT5645_EQ_BPF3_SFT 3 +#define RT5645_EQ_BPF3_DIS (0x0 << 3) +#define RT5645_EQ_BPF3_EN (0x1 << 3) +#define RT5645_EQ_BPF2_MASK (0x1 << 2) +#define RT5645_EQ_BPF2_SFT 2 +#define RT5645_EQ_BPF2_DIS (0x0 << 2) +#define RT5645_EQ_BPF2_EN (0x1 << 2) +#define RT5645_EQ_BPF1_MASK (0x1 << 1) +#define RT5645_EQ_BPF1_SFT 1 +#define RT5645_EQ_BPF1_DIS (0x0 << 1) +#define RT5645_EQ_BPF1_EN (0x1 << 1) +#define RT5645_EQ_LPF_MASK (0x1) +#define RT5645_EQ_LPF_SFT 0 +#define RT5645_EQ_LPF_DIS (0x0) +#define RT5645_EQ_LPF_EN (0x1) +#define RT5645_EQ_CTRL_MASK (0x7f) + +/* Memory Test (0xb2) */ +#define RT5645_MT_MASK (0x1 << 15) +#define RT5645_MT_SFT 15 +#define RT5645_MT_DIS (0x0 << 15) +#define RT5645_MT_EN (0x1 << 15) + +/* DRC/AGC Control 1 (0xb4) */ +#define RT5645_DRC_AGC_P_MASK (0x1 << 15) +#define RT5645_DRC_AGC_P_SFT 15 +#define RT5645_DRC_AGC_P_DAC (0x0 << 15) +#define RT5645_DRC_AGC_P_ADC (0x1 << 15) +#define RT5645_DRC_AGC_MASK (0x1 << 14) +#define RT5645_DRC_AGC_SFT 14 +#define RT5645_DRC_AGC_DIS (0x0 << 14) +#define RT5645_DRC_AGC_EN (0x1 << 14) +#define RT5645_DRC_AGC_UPD (0x1 << 13) +#define RT5645_DRC_AGC_UPD_BIT 13 +#define RT5645_DRC_AGC_AR_MASK (0x1f << 8) +#define RT5645_DRC_AGC_AR_SFT 8 +#define RT5645_DRC_AGC_R_MASK (0x7 << 5) +#define RT5645_DRC_AGC_R_SFT 5 +#define RT5645_DRC_AGC_R_48K (0x1 << 5) +#define RT5645_DRC_AGC_R_96K (0x2 << 5) +#define RT5645_DRC_AGC_R_192K (0x3 << 5) +#define RT5645_DRC_AGC_R_441K (0x5 << 5) +#define RT5645_DRC_AGC_R_882K (0x6 << 5) +#define RT5645_DRC_AGC_R_1764K (0x7 << 5) +#define RT5645_DRC_AGC_RC_MASK (0x1f) +#define RT5645_DRC_AGC_RC_SFT 0 + +/* DRC/AGC Control 2 (0xb5) */ +#define RT5645_DRC_AGC_POB_MASK (0x3f << 8) +#define RT5645_DRC_AGC_POB_SFT 8 +#define RT5645_DRC_AGC_CP_MASK (0x1 << 7) +#define RT5645_DRC_AGC_CP_SFT 7 +#define RT5645_DRC_AGC_CP_DIS (0x0 << 7) +#define RT5645_DRC_AGC_CP_EN (0x1 << 7) +#define RT5645_DRC_AGC_CPR_MASK (0x3 << 5) +#define RT5645_DRC_AGC_CPR_SFT 5 +#define RT5645_DRC_AGC_CPR_1_1 (0x0 << 5) +#define RT5645_DRC_AGC_CPR_1_2 (0x1 << 5) +#define RT5645_DRC_AGC_CPR_1_3 (0x2 << 5) +#define RT5645_DRC_AGC_CPR_1_4 (0x3 << 5) +#define RT5645_DRC_AGC_PRB_MASK (0x1f) +#define RT5645_DRC_AGC_PRB_SFT 0 + +/* DRC/AGC Control 3 (0xb6) */ +#define RT5645_DRC_AGC_NGB_MASK (0xf << 12) +#define RT5645_DRC_AGC_NGB_SFT 12 +#define RT5645_DRC_AGC_TAR_MASK (0x1f << 7) +#define RT5645_DRC_AGC_TAR_SFT 7 +#define RT5645_DRC_AGC_NG_MASK (0x1 << 6) +#define RT5645_DRC_AGC_NG_SFT 6 +#define RT5645_DRC_AGC_NG_DIS (0x0 << 6) +#define RT5645_DRC_AGC_NG_EN (0x1 << 6) +#define RT5645_DRC_AGC_NGH_MASK (0x1 << 5) +#define RT5645_DRC_AGC_NGH_SFT 5 +#define RT5645_DRC_AGC_NGH_DIS (0x0 << 5) +#define RT5645_DRC_AGC_NGH_EN (0x1 << 5) +#define RT5645_DRC_AGC_NGT_MASK (0x1f) +#define RT5645_DRC_AGC_NGT_SFT 0 + +/* ANC Control 1 (0xb8) */ +#define RT5645_ANC_M_MASK (0x1 << 15) +#define RT5645_ANC_M_SFT 15 +#define RT5645_ANC_M_NOR (0x0 << 15) +#define RT5645_ANC_M_REV (0x1 << 15) +#define RT5645_ANC_MASK (0x1 << 14) +#define RT5645_ANC_SFT 14 +#define RT5645_ANC_DIS (0x0 << 14) +#define RT5645_ANC_EN (0x1 << 14) +#define RT5645_ANC_MD_MASK (0x3 << 12) +#define RT5645_ANC_MD_SFT 12 +#define RT5645_ANC_MD_DIS (0x0 << 12) +#define RT5645_ANC_MD_67MS (0x1 << 12) +#define RT5645_ANC_MD_267MS (0x2 << 12) +#define RT5645_ANC_MD_1067MS (0x3 << 12) +#define RT5645_ANC_SN_MASK (0x1 << 11) +#define RT5645_ANC_SN_SFT 11 +#define RT5645_ANC_SN_DIS (0x0 << 11) +#define RT5645_ANC_SN_EN (0x1 << 11) +#define RT5645_ANC_CLK_MASK (0x1 << 10) +#define RT5645_ANC_CLK_SFT 10 +#define RT5645_ANC_CLK_ANC (0x0 << 10) +#define RT5645_ANC_CLK_REG (0x1 << 10) +#define RT5645_ANC_ZCD_MASK (0x3 << 8) +#define RT5645_ANC_ZCD_SFT 8 +#define RT5645_ANC_ZCD_DIS (0x0 << 8) +#define RT5645_ANC_ZCD_T1 (0x1 << 8) +#define RT5645_ANC_ZCD_T2 (0x2 << 8) +#define RT5645_ANC_ZCD_WT (0x3 << 8) +#define RT5645_ANC_CS_MASK (0x1 << 7) +#define RT5645_ANC_CS_SFT 7 +#define RT5645_ANC_CS_DIS (0x0 << 7) +#define RT5645_ANC_CS_EN (0x1 << 7) +#define RT5645_ANC_SW_MASK (0x1 << 6) +#define RT5645_ANC_SW_SFT 6 +#define RT5645_ANC_SW_NOR (0x0 << 6) +#define RT5645_ANC_SW_AUTO (0x1 << 6) +#define RT5645_ANC_CO_L_MASK (0x3f) +#define RT5645_ANC_CO_L_SFT 0 + +/* ANC Control 2 (0xb6) */ +#define RT5645_ANC_FG_R_MASK (0xf << 12) +#define RT5645_ANC_FG_R_SFT 12 +#define RT5645_ANC_FG_L_MASK (0xf << 8) +#define RT5645_ANC_FG_L_SFT 8 +#define RT5645_ANC_CG_R_MASK (0xf << 4) +#define RT5645_ANC_CG_R_SFT 4 +#define RT5645_ANC_CG_L_MASK (0xf) +#define RT5645_ANC_CG_L_SFT 0 + +/* ANC Control 3 (0xb6) */ +#define RT5645_ANC_CD_MASK (0x1 << 6) +#define RT5645_ANC_CD_SFT 6 +#define RT5645_ANC_CD_BOTH (0x0 << 6) +#define RT5645_ANC_CD_IND (0x1 << 6) +#define RT5645_ANC_CO_R_MASK (0x3f) +#define RT5645_ANC_CO_R_SFT 0 + +/* Jack Detect Control (0xbb) */ +#define RT5645_JD_MASK (0x7 << 13) +#define RT5645_JD_SFT 13 +#define RT5645_JD_DIS (0x0 << 13) +#define RT5645_JD_GPIO1 (0x1 << 13) +#define RT5645_JD_JD1_IN4P (0x2 << 13) +#define RT5645_JD_JD2_IN4N (0x3 << 13) +#define RT5645_JD_GPIO2 (0x4 << 13) +#define RT5645_JD_GPIO3 (0x5 << 13) +#define RT5645_JD_GPIO4 (0x6 << 13) +#define RT5645_JD_HP_MASK (0x1 << 11) +#define RT5645_JD_HP_SFT 11 +#define RT5645_JD_HP_DIS (0x0 << 11) +#define RT5645_JD_HP_EN (0x1 << 11) +#define RT5645_JD_HP_TRG_MASK (0x1 << 10) +#define RT5645_JD_HP_TRG_SFT 10 +#define RT5645_JD_HP_TRG_LO (0x0 << 10) +#define RT5645_JD_HP_TRG_HI (0x1 << 10) +#define RT5645_JD_SPL_MASK (0x1 << 9) +#define RT5645_JD_SPL_SFT 9 +#define RT5645_JD_SPL_DIS (0x0 << 9) +#define RT5645_JD_SPL_EN (0x1 << 9) +#define RT5645_JD_SPL_TRG_MASK (0x1 << 8) +#define RT5645_JD_SPL_TRG_SFT 8 +#define RT5645_JD_SPL_TRG_LO (0x0 << 8) +#define RT5645_JD_SPL_TRG_HI (0x1 << 8) +#define RT5645_JD_SPR_MASK (0x1 << 7) +#define RT5645_JD_SPR_SFT 7 +#define RT5645_JD_SPR_DIS (0x0 << 7) +#define RT5645_JD_SPR_EN (0x1 << 7) +#define RT5645_JD_SPR_TRG_MASK (0x1 << 6) +#define RT5645_JD_SPR_TRG_SFT 6 +#define RT5645_JD_SPR_TRG_LO (0x0 << 6) +#define RT5645_JD_SPR_TRG_HI (0x1 << 6) +#define RT5645_JD_MO_MASK (0x1 << 5) +#define RT5645_JD_MO_SFT 5 +#define RT5645_JD_MO_DIS (0x0 << 5) +#define RT5645_JD_MO_EN (0x1 << 5) +#define RT5645_JD_MO_TRG_MASK (0x1 << 4) +#define RT5645_JD_MO_TRG_SFT 4 +#define RT5645_JD_MO_TRG_LO (0x0 << 4) +#define RT5645_JD_MO_TRG_HI (0x1 << 4) +#define RT5645_JD_LO_MASK (0x1 << 3) +#define RT5645_JD_LO_SFT 3 +#define RT5645_JD_LO_DIS (0x0 << 3) +#define RT5645_JD_LO_EN (0x1 << 3) +#define RT5645_JD_LO_TRG_MASK (0x1 << 2) +#define RT5645_JD_LO_TRG_SFT 2 +#define RT5645_JD_LO_TRG_LO (0x0 << 2) +#define RT5645_JD_LO_TRG_HI (0x1 << 2) +#define RT5645_JD1_IN4P_MASK (0x1 << 1) +#define RT5645_JD1_IN4P_SFT 1 +#define RT5645_JD1_IN4P_DIS (0x0 << 1) +#define RT5645_JD1_IN4P_EN (0x1 << 1) +#define RT5645_JD2_IN4N_MASK (0x1) +#define RT5645_JD2_IN4N_SFT 0 +#define RT5645_JD2_IN4N_DIS (0x0) +#define RT5645_JD2_IN4N_EN (0x1) + +/* Jack detect for ANC (0xbc) */ +#define RT5645_ANC_DET_MASK (0x3 << 4) +#define RT5645_ANC_DET_SFT 4 +#define RT5645_ANC_DET_DIS (0x0 << 4) +#define RT5645_ANC_DET_MB1 (0x1 << 4) +#define RT5645_ANC_DET_MB2 (0x2 << 4) +#define RT5645_ANC_DET_JD (0x3 << 4) +#define RT5645_AD_TRG_MASK (0x1 << 3) +#define RT5645_AD_TRG_SFT 3 +#define RT5645_AD_TRG_LO (0x0 << 3) +#define RT5645_AD_TRG_HI (0x1 << 3) +#define RT5645_ANCM_DET_MASK (0x3 << 4) +#define RT5645_ANCM_DET_SFT 4 +#define RT5645_ANCM_DET_DIS (0x0 << 4) +#define RT5645_ANCM_DET_MB1 (0x1 << 4) +#define RT5645_ANCM_DET_MB2 (0x2 << 4) +#define RT5645_ANCM_DET_JD (0x3 << 4) +#define RT5645_AMD_TRG_MASK (0x1 << 3) +#define RT5645_AMD_TRG_SFT 3 +#define RT5645_AMD_TRG_LO (0x0 << 3) +#define RT5645_AMD_TRG_HI (0x1 << 3) + +/* IRQ Control 1 (0xbd) */ +#define RT5645_IRQ_JD_MASK (0x1 << 15) +#define RT5645_IRQ_JD_SFT 15 +#define RT5645_IRQ_JD_BP (0x0 << 15) +#define RT5645_IRQ_JD_NOR (0x1 << 15) +#define RT5645_IRQ_OT_MASK (0x1 << 14) +#define RT5645_IRQ_OT_SFT 14 +#define RT5645_IRQ_OT_BP (0x0 << 14) +#define RT5645_IRQ_OT_NOR (0x1 << 14) +#define RT5645_JD_STKY_MASK (0x1 << 13) +#define RT5645_JD_STKY_SFT 13 +#define RT5645_JD_STKY_DIS (0x0 << 13) +#define RT5645_JD_STKY_EN (0x1 << 13) +#define RT5645_OT_STKY_MASK (0x1 << 12) +#define RT5645_OT_STKY_SFT 12 +#define RT5645_OT_STKY_DIS (0x0 << 12) +#define RT5645_OT_STKY_EN (0x1 << 12) +#define RT5645_JD_P_MASK (0x1 << 11) +#define RT5645_JD_P_SFT 11 +#define RT5645_JD_P_NOR (0x0 << 11) +#define RT5645_JD_P_INV (0x1 << 11) +#define RT5645_OT_P_MASK (0x1 << 10) +#define RT5645_OT_P_SFT 10 +#define RT5645_OT_P_NOR (0x0 << 10) +#define RT5645_OT_P_INV (0x1 << 10) + +/* IRQ Control 2 (0xbe) */ +#define RT5645_IRQ_MB1_OC_MASK (0x1 << 15) +#define RT5645_IRQ_MB1_OC_SFT 15 +#define RT5645_IRQ_MB1_OC_BP (0x0 << 15) +#define RT5645_IRQ_MB1_OC_NOR (0x1 << 15) +#define RT5645_IRQ_MB2_OC_MASK (0x1 << 14) +#define RT5645_IRQ_MB2_OC_SFT 14 +#define RT5645_IRQ_MB2_OC_BP (0x0 << 14) +#define RT5645_IRQ_MB2_OC_NOR (0x1 << 14) +#define RT5645_MB1_OC_STKY_MASK (0x1 << 13) +#define RT5645_MB1_OC_STKY_SFT 13 +#define RT5645_MB1_OC_STKY_DIS (0x0 << 13) +#define RT5645_MB1_OC_STKY_EN (0x1 << 13) +#define RT5645_MB2_OC_STKY_MASK (0x1 << 12) +#define RT5645_MB2_OC_STKY_SFT 12 +#define RT5645_MB2_OC_STKY_DIS (0x0 << 12) +#define RT5645_MB2_OC_STKY_EN (0x1 << 12) +#define RT5645_MB1_OC_P_MASK (0x1 << 7) +#define RT5645_MB1_OC_P_SFT 7 +#define RT5645_MB1_OC_P_NOR (0x0 << 7) +#define RT5645_MB1_OC_P_INV (0x1 << 7) +#define RT5645_MB2_OC_P_MASK (0x1 << 6) +#define RT5645_MB2_OC_P_SFT 6 +#define RT5645_MB2_OC_P_NOR (0x0 << 6) +#define RT5645_MB2_OC_P_INV (0x1 << 6) +#define RT5645_MB1_OC_CLR (0x1 << 3) +#define RT5645_MB1_OC_CLR_SFT 3 +#define RT5645_MB2_OC_CLR (0x1 << 2) +#define RT5645_MB2_OC_CLR_SFT 2 + +/* GPIO Control 1 (0xc0) */ +#define RT5645_GP1_PIN_MASK (0x1 << 15) +#define RT5645_GP1_PIN_SFT 15 +#define RT5645_GP1_PIN_GPIO1 (0x0 << 15) +#define RT5645_GP1_PIN_IRQ (0x1 << 15) +#define RT5645_GP2_PIN_MASK (0x1 << 14) +#define RT5645_GP2_PIN_SFT 14 +#define RT5645_GP2_PIN_GPIO2 (0x0 << 14) +#define RT5645_GP2_PIN_DMIC1_SCL (0x1 << 14) +#define RT5645_GP3_PIN_MASK (0x3 << 12) +#define RT5645_GP3_PIN_SFT 12 +#define RT5645_GP3_PIN_GPIO3 (0x0 << 12) +#define RT5645_GP3_PIN_DMIC1_SDA (0x1 << 12) +#define RT5645_GP3_PIN_IRQ (0x2 << 12) +#define RT5645_GP4_PIN_MASK (0x1 << 11) +#define RT5645_GP4_PIN_SFT 11 +#define RT5645_GP4_PIN_GPIO4 (0x0 << 11) +#define RT5645_GP4_PIN_DMIC2_SDA (0x1 << 11) +#define RT5645_DP_SIG_MASK (0x1 << 10) +#define RT5645_DP_SIG_SFT 10 +#define RT5645_DP_SIG_TEST (0x0 << 10) +#define RT5645_DP_SIG_AP (0x1 << 10) +#define RT5645_GPIO_M_MASK (0x1 << 9) +#define RT5645_GPIO_M_SFT 9 +#define RT5645_GPIO_M_FLT (0x0 << 9) +#define RT5645_GPIO_M_PH (0x1 << 9) +#define RT5645_I2S2_SEL (0x1 << 8) +#define RT5645_I2S2_SEL_SFT 8 +#define RT5645_GP5_PIN_MASK (0x1 << 7) +#define RT5645_GP5_PIN_SFT 7 +#define RT5645_GP5_PIN_GPIO5 (0x0 << 7) +#define RT5645_GP5_PIN_DMIC3_SCL (0x1 << 7) +#define RT5645_GP7_PIN_MASK (0x3 << 4) +#define RT5645_GP7_PIN_SFT 4 +#define RT5645_GP7_PIN_GPIO7 (0x0 << 4) +#define RT5645_GP7_PIN_DMIC1_SDA (0x1 << 4) +#define RT5645_GP7_PIN_PDM_SCL2 (0x2 << 4) +#define RT5645_GP8_PIN_MASK (0x1 << 3) +#define RT5645_GP8_PIN_SFT 3 +#define RT5645_GP8_PIN_GPIO8 (0x0 << 3) +#define RT5645_GP8_PIN_DMIC2_SDA (0x1 << 3) +#define RT5645_GP9_PIN_MASK (0x1 << 2) +#define RT5645_GP9_PIN_SFT 2 +#define RT5645_GP9_PIN_GPIO9 (0x0 << 2) +#define RT5645_GP9_PIN_DMIC3_SDA (0x1 << 2) +#define RT5645_GP10_PIN_MASK (0x3) +#define RT5645_GP10_PIN_SFT 0 +#define RT5645_GP10_PIN_GPIO9 (0x0) +#define RT5645_GP10_PIN_DMIC3_SDA (0x1) +#define RT5645_GP10_PIN_PDM_ADT2 (0x2) + +/* GPIO Control 3 (0xc2) */ +#define RT5645_GP4_PF_MASK (0x1 << 11) +#define RT5645_GP4_PF_SFT 11 +#define RT5645_GP4_PF_IN (0x0 << 11) +#define RT5645_GP4_PF_OUT (0x1 << 11) +#define RT5645_GP4_OUT_MASK (0x1 << 10) +#define RT5645_GP4_OUT_SFT 10 +#define RT5645_GP4_OUT_LO (0x0 << 10) +#define RT5645_GP4_OUT_HI (0x1 << 10) +#define RT5645_GP4_P_MASK (0x1 << 9) +#define RT5645_GP4_P_SFT 9 +#define RT5645_GP4_P_NOR (0x0 << 9) +#define RT5645_GP4_P_INV (0x1 << 9) +#define RT5645_GP3_PF_MASK (0x1 << 8) +#define RT5645_GP3_PF_SFT 8 +#define RT5645_GP3_PF_IN (0x0 << 8) +#define RT5645_GP3_PF_OUT (0x1 << 8) +#define RT5645_GP3_OUT_MASK (0x1 << 7) +#define RT5645_GP3_OUT_SFT 7 +#define RT5645_GP3_OUT_LO (0x0 << 7) +#define RT5645_GP3_OUT_HI (0x1 << 7) +#define RT5645_GP3_P_MASK (0x1 << 6) +#define RT5645_GP3_P_SFT 6 +#define RT5645_GP3_P_NOR (0x0 << 6) +#define RT5645_GP3_P_INV (0x1 << 6) +#define RT5645_GP2_PF_MASK (0x1 << 5) +#define RT5645_GP2_PF_SFT 5 +#define RT5645_GP2_PF_IN (0x0 << 5) +#define RT5645_GP2_PF_OUT (0x1 << 5) +#define RT5645_GP2_OUT_MASK (0x1 << 4) +#define RT5645_GP2_OUT_SFT 4 +#define RT5645_GP2_OUT_LO (0x0 << 4) +#define RT5645_GP2_OUT_HI (0x1 << 4) +#define RT5645_GP2_P_MASK (0x1 << 3) +#define RT5645_GP2_P_SFT 3 +#define RT5645_GP2_P_NOR (0x0 << 3) +#define RT5645_GP2_P_INV (0x1 << 3) +#define RT5645_GP1_PF_MASK (0x1 << 2) +#define RT5645_GP1_PF_SFT 2 +#define RT5645_GP1_PF_IN (0x0 << 2) +#define RT5645_GP1_PF_OUT (0x1 << 2) +#define RT5645_GP1_OUT_MASK (0x1 << 1) +#define RT5645_GP1_OUT_SFT 1 +#define RT5645_GP1_OUT_LO (0x0 << 1) +#define RT5645_GP1_OUT_HI (0x1 << 1) +#define RT5645_GP1_P_MASK (0x1) +#define RT5645_GP1_P_SFT 0 +#define RT5645_GP1_P_NOR (0x0) +#define RT5645_GP1_P_INV (0x1) + +/* Programmable Register Array Control 1 (0xc8) */ +#define RT5645_REG_SEQ_MASK (0xf << 12) +#define RT5645_REG_SEQ_SFT 12 +#define RT5645_SEQ1_ST_MASK (0x1 << 11) /*RO*/ +#define RT5645_SEQ1_ST_SFT 11 +#define RT5645_SEQ1_ST_RUN (0x0 << 11) +#define RT5645_SEQ1_ST_FIN (0x1 << 11) +#define RT5645_SEQ2_ST_MASK (0x1 << 10) /*RO*/ +#define RT5645_SEQ2_ST_SFT 10 +#define RT5645_SEQ2_ST_RUN (0x0 << 10) +#define RT5645_SEQ2_ST_FIN (0x1 << 10) +#define RT5645_REG_LV_MASK (0x1 << 9) +#define RT5645_REG_LV_SFT 9 +#define RT5645_REG_LV_MX (0x0 << 9) +#define RT5645_REG_LV_PR (0x1 << 9) +#define RT5645_SEQ_2_PT_MASK (0x1 << 8) +#define RT5645_SEQ_2_PT_BIT 8 +#define RT5645_REG_IDX_MASK (0xff) +#define RT5645_REG_IDX_SFT 0 + +/* Programmable Register Array Control 2 (0xc9) */ +#define RT5645_REG_DAT_MASK (0xffff) +#define RT5645_REG_DAT_SFT 0 + +/* Programmable Register Array Control 3 (0xca) */ +#define RT5645_SEQ_DLY_MASK (0xff << 8) +#define RT5645_SEQ_DLY_SFT 8 +#define RT5645_PROG_MASK (0x1 << 7) +#define RT5645_PROG_SFT 7 +#define RT5645_PROG_DIS (0x0 << 7) +#define RT5645_PROG_EN (0x1 << 7) +#define RT5645_SEQ1_PT_RUN (0x1 << 6) +#define RT5645_SEQ1_PT_RUN_BIT 6 +#define RT5645_SEQ2_PT_RUN (0x1 << 5) +#define RT5645_SEQ2_PT_RUN_BIT 5 + +/* Programmable Register Array Control 4 (0xcb) */ +#define RT5645_SEQ1_START_MASK (0xf << 8) +#define RT5645_SEQ1_START_SFT 8 +#define RT5645_SEQ1_END_MASK (0xf) +#define RT5645_SEQ1_END_SFT 0 + +/* Programmable Register Array Control 5 (0xcc) */ +#define RT5645_SEQ2_START_MASK (0xf << 8) +#define RT5645_SEQ2_START_SFT 8 +#define RT5645_SEQ2_END_MASK (0xf) +#define RT5645_SEQ2_END_SFT 0 + +/* Scramble Function (0xcd) */ +#define RT5645_SCB_KEY_MASK (0xff) +#define RT5645_SCB_KEY_SFT 0 + +/* Scramble Control (0xce) */ +#define RT5645_SCB_SWAP_MASK (0x1 << 15) +#define RT5645_SCB_SWAP_SFT 15 +#define RT5645_SCB_SWAP_DIS (0x0 << 15) +#define RT5645_SCB_SWAP_EN (0x1 << 15) +#define RT5645_SCB_MASK (0x1 << 14) +#define RT5645_SCB_SFT 14 +#define RT5645_SCB_DIS (0x0 << 14) +#define RT5645_SCB_EN (0x1 << 14) + +/* Baseback Control (0xcf) */ +#define RT5645_BB_MASK (0x1 << 15) +#define RT5645_BB_SFT 15 +#define RT5645_BB_DIS (0x0 << 15) +#define RT5645_BB_EN (0x1 << 15) +#define RT5645_BB_CT_MASK (0x7 << 12) +#define RT5645_BB_CT_SFT 12 +#define RT5645_BB_CT_A (0x0 << 12) +#define RT5645_BB_CT_B (0x1 << 12) +#define RT5645_BB_CT_C (0x2 << 12) +#define RT5645_BB_CT_D (0x3 << 12) +#define RT5645_M_BB_L_MASK (0x1 << 9) +#define RT5645_M_BB_L_SFT 9 +#define RT5645_M_BB_R_MASK (0x1 << 8) +#define RT5645_M_BB_R_SFT 8 +#define RT5645_M_BB_HPF_L_MASK (0x1 << 7) +#define RT5645_M_BB_HPF_L_SFT 7 +#define RT5645_M_BB_HPF_R_MASK (0x1 << 6) +#define RT5645_M_BB_HPF_R_SFT 6 +#define RT5645_G_BB_BST_MASK (0x3f) +#define RT5645_G_BB_BST_SFT 0 + +/* MP3 Plus Control 1 (0xd0) */ +#define RT5645_M_MP3_L_MASK (0x1 << 15) +#define RT5645_M_MP3_L_SFT 15 +#define RT5645_M_MP3_R_MASK (0x1 << 14) +#define RT5645_M_MP3_R_SFT 14 +#define RT5645_M_MP3_MASK (0x1 << 13) +#define RT5645_M_MP3_SFT 13 +#define RT5645_M_MP3_DIS (0x0 << 13) +#define RT5645_M_MP3_EN (0x1 << 13) +#define RT5645_EG_MP3_MASK (0x1f << 8) +#define RT5645_EG_MP3_SFT 8 +#define RT5645_MP3_HLP_MASK (0x1 << 7) +#define RT5645_MP3_HLP_SFT 7 +#define RT5645_MP3_HLP_DIS (0x0 << 7) +#define RT5645_MP3_HLP_EN (0x1 << 7) +#define RT5645_M_MP3_ORG_L_MASK (0x1 << 6) +#define RT5645_M_MP3_ORG_L_SFT 6 +#define RT5645_M_MP3_ORG_R_MASK (0x1 << 5) +#define RT5645_M_MP3_ORG_R_SFT 5 + +/* MP3 Plus Control 2 (0xd1) */ +#define RT5645_MP3_WT_MASK (0x1 << 13) +#define RT5645_MP3_WT_SFT 13 +#define RT5645_MP3_WT_1_4 (0x0 << 13) +#define RT5645_MP3_WT_1_2 (0x1 << 13) +#define RT5645_OG_MP3_MASK (0x1f << 8) +#define RT5645_OG_MP3_SFT 8 +#define RT5645_HG_MP3_MASK (0x3f) +#define RT5645_HG_MP3_SFT 0 + +/* 3D HP Control 1 (0xd2) */ +#define RT5645_3D_CF_MASK (0x1 << 15) +#define RT5645_3D_CF_SFT 15 +#define RT5645_3D_CF_DIS (0x0 << 15) +#define RT5645_3D_CF_EN (0x1 << 15) +#define RT5645_3D_HP_MASK (0x1 << 14) +#define RT5645_3D_HP_SFT 14 +#define RT5645_3D_HP_DIS (0x0 << 14) +#define RT5645_3D_HP_EN (0x1 << 14) +#define RT5645_3D_BT_MASK (0x1 << 13) +#define RT5645_3D_BT_SFT 13 +#define RT5645_3D_BT_DIS (0x0 << 13) +#define RT5645_3D_BT_EN (0x1 << 13) +#define RT5645_3D_1F_MIX_MASK (0x3 << 11) +#define RT5645_3D_1F_MIX_SFT 11 +#define RT5645_3D_HP_M_MASK (0x1 << 10) +#define RT5645_3D_HP_M_SFT 10 +#define RT5645_3D_HP_M_SUR (0x0 << 10) +#define RT5645_3D_HP_M_FRO (0x1 << 10) +#define RT5645_M_3D_HRTF_MASK (0x1 << 9) +#define RT5645_M_3D_HRTF_SFT 9 +#define RT5645_M_3D_D2H_MASK (0x1 << 8) +#define RT5645_M_3D_D2H_SFT 8 +#define RT5645_M_3D_D2R_MASK (0x1 << 7) +#define RT5645_M_3D_D2R_SFT 7 +#define RT5645_M_3D_REVB_MASK (0x1 << 6) +#define RT5645_M_3D_REVB_SFT 6 + +/* Adjustable high pass filter control 1 (0xd3) */ +#define RT5645_2ND_HPF_MASK (0x1 << 15) +#define RT5645_2ND_HPF_SFT 15 +#define RT5645_2ND_HPF_DIS (0x0 << 15) +#define RT5645_2ND_HPF_EN (0x1 << 15) +#define RT5645_HPF_CF_L_MASK (0x7 << 12) +#define RT5645_HPF_CF_L_SFT 12 +#define RT5645_1ST_HPF_MASK (0x1 << 11) +#define RT5645_1ST_HPF_SFT 11 +#define RT5645_1ST_HPF_DIS (0x0 << 11) +#define RT5645_1ST_HPF_EN (0x1 << 11) +#define RT5645_HPF_CF_R_MASK (0x7 << 8) +#define RT5645_HPF_CF_R_SFT 8 +#define RT5645_ZD_T_MASK (0x3 << 6) +#define RT5645_ZD_T_SFT 6 +#define RT5645_ZD_F_MASK (0x3 << 4) +#define RT5645_ZD_F_SFT 4 +#define RT5645_ZD_F_IM (0x0 << 4) +#define RT5645_ZD_F_ZC_IM (0x1 << 4) +#define RT5645_ZD_F_ZC_IOD (0x2 << 4) +#define RT5645_ZD_F_UN (0x3 << 4) + +/* HP calibration control and Amp detection (0xd6) */ +#define RT5645_SI_DAC_MASK (0x1 << 11) +#define RT5645_SI_DAC_SFT 11 +#define RT5645_SI_DAC_AUTO (0x0 << 11) +#define RT5645_SI_DAC_TEST (0x1 << 11) +#define RT5645_DC_CAL_M_MASK (0x1 << 10) +#define RT5645_DC_CAL_M_SFT 10 +#define RT5645_DC_CAL_M_CAL (0x0 << 10) +#define RT5645_DC_CAL_M_NOR (0x1 << 10) +#define RT5645_DC_CAL_MASK (0x1 << 9) +#define RT5645_DC_CAL_SFT 9 +#define RT5645_DC_CAL_DIS (0x0 << 9) +#define RT5645_DC_CAL_EN (0x1 << 9) +#define RT5645_HPD_RCV_MASK (0x7 << 6) +#define RT5645_HPD_RCV_SFT 6 +#define RT5645_HPD_PS_MASK (0x1 << 5) +#define RT5645_HPD_PS_SFT 5 +#define RT5645_HPD_PS_DIS (0x0 << 5) +#define RT5645_HPD_PS_EN (0x1 << 5) +#define RT5645_CAL_M_MASK (0x1 << 4) +#define RT5645_CAL_M_SFT 4 +#define RT5645_CAL_M_DEP (0x0 << 4) +#define RT5645_CAL_M_CAL (0x1 << 4) +#define RT5645_CAL_MASK (0x1 << 3) +#define RT5645_CAL_SFT 3 +#define RT5645_CAL_DIS (0x0 << 3) +#define RT5645_CAL_EN (0x1 << 3) +#define RT5645_CAL_TEST_MASK (0x1 << 2) +#define RT5645_CAL_TEST_SFT 2 +#define RT5645_CAL_TEST_DIS (0x0 << 2) +#define RT5645_CAL_TEST_EN (0x1 << 2) +#define RT5645_CAL_P_MASK (0x3) +#define RT5645_CAL_P_SFT 0 +#define RT5645_CAL_P_NONE (0x0) +#define RT5645_CAL_P_CAL (0x1) +#define RT5645_CAL_P_DAC_CAL (0x2) + +/* Soft volume and zero cross control 1 (0xd9) */ +#define RT5645_SV_MASK (0x1 << 15) +#define RT5645_SV_SFT 15 +#define RT5645_SV_DIS (0x0 << 15) +#define RT5645_SV_EN (0x1 << 15) +#define RT5645_SPO_SV_MASK (0x1 << 14) +#define RT5645_SPO_SV_SFT 14 +#define RT5645_SPO_SV_DIS (0x0 << 14) +#define RT5645_SPO_SV_EN (0x1 << 14) +#define RT5645_OUT_SV_MASK (0x1 << 13) +#define RT5645_OUT_SV_SFT 13 +#define RT5645_OUT_SV_DIS (0x0 << 13) +#define RT5645_OUT_SV_EN (0x1 << 13) +#define RT5645_HP_SV_MASK (0x1 << 12) +#define RT5645_HP_SV_SFT 12 +#define RT5645_HP_SV_DIS (0x0 << 12) +#define RT5645_HP_SV_EN (0x1 << 12) +#define RT5645_ZCD_DIG_MASK (0x1 << 11) +#define RT5645_ZCD_DIG_SFT 11 +#define RT5645_ZCD_DIG_DIS (0x0 << 11) +#define RT5645_ZCD_DIG_EN (0x1 << 11) +#define RT5645_ZCD_MASK (0x1 << 10) +#define RT5645_ZCD_SFT 10 +#define RT5645_ZCD_PD (0x0 << 10) +#define RT5645_ZCD_PU (0x1 << 10) +#define RT5645_M_ZCD_MASK (0x3f << 4) +#define RT5645_M_ZCD_SFT 4 +#define RT5645_M_ZCD_RM_L (0x1 << 9) +#define RT5645_M_ZCD_RM_R (0x1 << 8) +#define RT5645_M_ZCD_SM_L (0x1 << 7) +#define RT5645_M_ZCD_SM_R (0x1 << 6) +#define RT5645_M_ZCD_OM_L (0x1 << 5) +#define RT5645_M_ZCD_OM_R (0x1 << 4) +#define RT5645_SV_DLY_MASK (0xf) +#define RT5645_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0xda) */ +#define RT5645_ZCD_HP_MASK (0x1 << 15) +#define RT5645_ZCD_HP_SFT 15 +#define RT5645_ZCD_HP_DIS (0x0 << 15) +#define RT5645_ZCD_HP_EN (0x1 << 15) + + +/* Codec Private Register definition */ +/* 3D Speaker Control (0x63) */ +#define RT5645_3D_SPK_MASK (0x1 << 15) +#define RT5645_3D_SPK_SFT 15 +#define RT5645_3D_SPK_DIS (0x0 << 15) +#define RT5645_3D_SPK_EN (0x1 << 15) +#define RT5645_3D_SPK_M_MASK (0x3 << 13) +#define RT5645_3D_SPK_M_SFT 13 +#define RT5645_3D_SPK_CG_MASK (0x1f << 8) +#define RT5645_3D_SPK_CG_SFT 8 +#define RT5645_3D_SPK_SG_MASK (0x1f) +#define RT5645_3D_SPK_SG_SFT 0 + +/* Wind Noise Detection Control 1 (0x6c) */ +#define RT5645_WND_MASK (0x1 << 15) +#define RT5645_WND_SFT 15 +#define RT5645_WND_DIS (0x0 << 15) +#define RT5645_WND_EN (0x1 << 15) + +/* Wind Noise Detection Control 2 (0x6d) */ +#define RT5645_WND_FC_NW_MASK (0x3f << 10) +#define RT5645_WND_FC_NW_SFT 10 +#define RT5645_WND_FC_WK_MASK (0x3f << 4) +#define RT5645_WND_FC_WK_SFT 4 + +/* Wind Noise Detection Control 3 (0x6e) */ +#define RT5645_HPF_FC_MASK (0x3f << 6) +#define RT5645_HPF_FC_SFT 6 +#define RT5645_WND_FC_ST_MASK (0x3f) +#define RT5645_WND_FC_ST_SFT 0 + +/* Wind Noise Detection Control 4 (0x6f) */ +#define RT5645_WND_TH_LO_MASK (0x3ff) +#define RT5645_WND_TH_LO_SFT 0 + +/* Wind Noise Detection Control 5 (0x70) */ +#define RT5645_WND_TH_HI_MASK (0x3ff) +#define RT5645_WND_TH_HI_SFT 0 + +/* Wind Noise Detection Control 8 (0x73) */ +#define RT5645_WND_WIND_MASK (0x1 << 13) /* Read-Only */ +#define RT5645_WND_WIND_SFT 13 +#define RT5645_WND_STRONG_MASK (0x1 << 12) /* Read-Only */ +#define RT5645_WND_STRONG_SFT 12 +enum { + RT5645_NO_WIND, + RT5645_BREEZE, + RT5645_STORM, +}; + +/* Dipole Speaker Interface (0x75) */ +#define RT5645_DP_ATT_MASK (0x3 << 14) +#define RT5645_DP_ATT_SFT 14 +#define RT5645_DP_SPK_MASK (0x1 << 10) +#define RT5645_DP_SPK_SFT 10 +#define RT5645_DP_SPK_DIS (0x0 << 10) +#define RT5645_DP_SPK_EN (0x1 << 10) + +/* EQ Pre Volume Control (0xb3) */ +#define RT5645_EQ_PRE_VOL_MASK (0xffff) +#define RT5645_EQ_PRE_VOL_SFT 0 + +/* EQ Post Volume Control (0xb4) */ +#define RT5645_EQ_PST_VOL_MASK (0xffff) +#define RT5645_EQ_PST_VOL_SFT 0 + +/* Jack Detect Control 3 (0xf8) */ +#define RT5645_CMP_MIC_IN_DET_MASK (0x7 << 12) +#define RT5645_JD_CBJ_EN (0x1 << 7) +#define RT5645_JD_CBJ_POL (0x1 << 6) +#define RT5645_JD_TRI_CBJ_SEL_MASK (0x7 << 3) +#define RT5645_JD_TRI_CBJ_SEL_SFT (3) +#define RT5645_JD_TRI_HPO_SEL_MASK (0x7) +#define RT5645_JD_TRI_HPO_SEL_SFT (0) +#define RT5645_JD_F_GPIO_JD1 (0x0) +#define RT5645_JD_F_JD1_1 (0x1) +#define RT5645_JD_F_JD1_2 (0x2) +#define RT5645_JD_F_JD2 (0x3) +#define RT5645_JD_F_JD3 (0x4) +#define RT5645_JD_F_GPIO_JD2 (0x5) +#define RT5645_JD_F_MX0B_12 (0x6) + +/* Digital Misc Control (0xfa) */ +#define RT5645_RST_DSP (0x1 << 13) +#define RT5645_IF1_ADC1_IN1_SEL (0x1 << 12) +#define RT5645_IF1_ADC1_IN1_SFT 12 +#define RT5645_IF1_ADC1_IN2_SEL (0x1 << 11) +#define RT5645_IF1_ADC1_IN2_SFT 11 +#define RT5645_IF1_ADC2_IN1_SEL (0x1 << 10) +#define RT5645_IF1_ADC2_IN1_SFT 10 +#define RT5645_DIG_GATE_CTRL 0x1 + +/* General Control2 (0xfb) */ +#define RT5645_RXDC_SRC_MASK (0x1 << 7) +#define RT5645_RXDC_SRC_STO (0x0 << 7) +#define RT5645_RXDC_SRC_MONO (0x1 << 7) +#define RT5645_RXDC_SRC_SFT (7) +#define RT5645_RXDP2_SEL_MASK (0x1 << 3) +#define RT5645_RXDP2_SEL_IF2 (0x0 << 3) +#define RT5645_RXDP2_SEL_ADC (0x1 << 3) +#define RT5645_RXDP2_SEL_SFT (3) + + +/* Vendor ID (0xfd) */ +#define RT5645_VER_C 0x2 +#define RT5645_VER_D 0x3 + + +/* Volume Rescale */ +#define RT5645_VOL_RSCL_MAX 0x27 +#define RT5645_VOL_RSCL_RANGE 0x1F +/* Debug String Length */ +#define RT5645_REG_DISP_LEN 23 + + +/* System Clock Source */ +enum { + RT5645_SCLK_S_MCLK, + RT5645_SCLK_S_PLL1, + RT5645_SCLK_S_RCCLK, +}; + +/* PLL1 Source */ +enum { + RT5645_PLL1_S_MCLK, + RT5645_PLL1_S_BCLK1, + RT5645_PLL1_S_BCLK2, +}; + +enum { + RT5645_AIF1, + RT5645_AIF2, + RT5645_AIFS, +}; + +enum { + RT5645_DMIC_DIS, + RT5645_DMIC1, + RT5645_DMIC2, +}; + +struct rt5645_pll_code { + bool m_bp; /* Indicates bypass m code or not. */ + int m_code; + int n_code; + int k_code; +}; + +struct rt5645_priv { + struct snd_soc_codec *codec; + struct regmap *regmap; + + int sysclk; + int sysclk_src; + int lrck[RT5645_AIFS]; + int bclk[RT5645_AIFS]; + int master[RT5645_AIFS]; + + int pll_src; + int pll_in; + int pll_out; +}; + +#endif /* __RT5645_H__ */ -- 1.8.1.1.439.g50a6b54

3 4

[alsa-devel] [PATCH v2 00/20] ASoC: omap: Bind the omap-pcm platform driver to dai device
by Peter Ujfalusi 22 Apr '14

22 Apr '14

Hi, Changes since v1: - Add resource managed snd_soc_register_platform() - Use snd_soc_dai_init_dma_data() - omap-pcm is using the devm_snd_soc_register_platform() - Patch for davinci-mcasp since Version4 is using omap-pcm Cover letter from v1: This should have been done a long time ago... Add support for binding the omap-pcm platform to a dai device. IN this way we can remove the 'omap-pcm-audio' platform device/driver need. The change will also going to simplify cases when we boot with DT and we can use the same node for cpu_dai and platform. Regards, Peter --- Peter Ujfalusi (20): ASoC: Add resource managed snd_soc_register_platform() ASoC: omap-pcm: Support for binding the platform driver to dai devices ASoC: omap-mcpdm: Assign the dai DMA data at earlier time ASoC: omap-mcpdm: Bind the platform driver to the dai driver when loading ASoC: omap-mcbsp: Assign the dai DMA data at earlier time ASoC: omap-mcbsp: Bind the platform driver to the dai driver when loading ASoC: omap-dmic: Assign the dai DMA data at earlier time ASoC: omap-dmic: Bind the platform driver to the dai driver when loading ASoC: omap-hdmi: Bind the platform driver to the dai driver when loading ASoC: davinci-mcasp: Update MCASP_VERSION_4 platform driver registration ASoC: omap-abe-twl6040: Use the cpu_dai node to specify the platform driver ASoC: omap-twl4030: Use the same name/node for platform as the cpu_dai ASoC: am3517evm: Use the same name for platform as the cpu_dai ASoC: ams-delta: Use the same name for platform as the cpu_dai ASoC: n810: Use the same name for platform as the cpu_dai ASoC: omap3pandora: Use the same name for platform as the cpu_dai ASoC: osk5912: Use the same name for platform as the cpu_dai ASoC: rx51: Use the same name for platform as the cpu_dai ASoC: omap-hdmi-card: Use the same name for platform as the cpu_dai ASoC: omap-pcm: Drop the platform driver init code include/sound/soc.h | 2 ++ sound/soc/davinci/davinci-mcasp.c | 37 +++++++++++++++++++++++++++++++------ sound/soc/omap/am3517evm.c | 2 +- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/n810.c | 2 +- sound/soc/omap/omap-abe-twl6040.c | 8 ++------ sound/soc/omap/omap-dmic.c | 9 ++++++++- sound/soc/omap/omap-hdmi-card.c | 2 +- sound/soc/omap/omap-hdmi.c | 7 ++++++- sound/soc/omap/omap-mcbsp.c | 20 +++++++++++++------- sound/soc/omap/omap-mcpdm.c | 16 ++++++++++++---- sound/soc/omap/omap-pcm.c | 25 +++---------------------- sound/soc/omap/omap-pcm.h | 30 ++++++++++++++++++++++++++++++ sound/soc/omap/omap-twl4030.c | 10 ++++++++-- sound/soc/omap/omap3pandora.c | 4 ++-- sound/soc/omap/osk5912.c | 2 +- sound/soc/omap/rx51.c | 2 +- sound/soc/soc-devres.c | 34 ++++++++++++++++++++++++++++++++++ 18 files changed, 157 insertions(+), 57 deletions(-) create mode 100644 sound/soc/omap/omap-pcm.h -- 1.9.2

3 23

[alsa-devel] sound disappears or noise heard after long run
by narayanan best 22 Apr '14

22 Apr '14

Hi, I am facing issue with alsa similar to the issue reported here http://mailman.alsa-project.org/pipermail/alsa-devel/2007-June/001726.html This is seen with latest alsa library and driver. Alsa is configured at 44100 (alsa.conf) and application plays at the rate of 48000. When the pointers cross the boundary value (after 6-7hrs), a continuous noise (jack hammer noise) or silence is heard. Issuing SIGSTOP and SIGCONT recovers the application and it starts playing normally. Pulseaudio is not running in our setup. It could be problem with calculation in the rate plugin. Let me know if you need further details.

1 0

[alsa-devel] [PATCHv4 2/2] ALSA: hda - Add driver for Tegra SoC HDA
by Dylan Reid 22 Apr '14

22 Apr '14

This adds a driver for the HDA block in Tegra SoCs. The HDA bus is used to communicate with the HDMI codec on Tegra124. Most of the code is re-used from the Intel/PCI HDA driver. It brings over only two of the module params, power_save and probe_mask. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Changes since v3: Remove some runtime-pm only code from the regular pm path. --- Changes since v2: Remove runtime PM. The on-chip codec doesn't support D3 stop clock so the pm_runtime code was never run. --- Changes since v1: Kconfig depends on OF and ARCH_TEGRA. Fix warning on 64 bit build. Integrate a bunch of good suggestions from Thierry, including: Add resets property to device tree entry. More consistent/better naming. Remove probe mask that isn't needed. Cleaner/more readable handling of dword write restriction. --- .../bindings/sound/nvidia,tegra30-hda.txt | 27 + sound/pci/hda/Kconfig | 15 + sound/pci/hda/Makefile | 2 + sound/pci/hda/hda_tegra.c | 585 +++++++++++++++++++++ 4 files changed, 629 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt create mode 100644 sound/pci/hda/hda_tegra.c diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt new file mode 100644 index 0000000..97f6832 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt @@ -0,0 +1,27 @@ +NVIDIA Tegra30 HDA controller + +Required properties: +- compatible : "nvidia,tegra30-hda" +- reg : Should contain the HDA registers location and length. +- interrupts : The interrupt from the hda controller. +- clocks : Must contain an entry for each required entry in clock-names. +- clock-names : Must include the following entries: hda, hdacodec_2x, hda2hdmi +- resets : Must contain an entry for each entry in reset-names. + See ../reset/reset.txt for details. +- reset-names : Must include the following entries: hda, hdacodec_2x, hda2hdmi + +Example: + +hda@70030000 { + compatible = "nvidia,tegra124-hda", "nvidia,tegra30-hda"; + reg = <0x0 0x70030000 0x10000>; + interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&tegra_car TEGRA124_CLK_HDA>, + <&tegra_car TEGRA124_CLK_HDA2HDMI>, + <&tegra_car TEGRA124_CLK_HDA2CODEC_2X>; + clock-names = "hda", "hda2hdmi", "hdacodec_2x"; + resets = <&tegra_car 125>, /* hda */ + <&tegra_car 128>; /* hda2hdmi */ + <&tegra_car 111>, /* hda2codec_2x */ + reset-names = "hda", "hda2hdmi", "hdacodec_2x"; +}; diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index ac17c3f..f17c016 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -20,6 +20,21 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. +config SND_HDA_TEGRA + tristate "NVIDIA Tegra HD Audio" + depends on OF && ARCH_TEGRA + select SND_HDA + help + Say Y here to support the HDA controller present in NVIDIA + Tegra SoCs + + This options enables support for the HD Audio controller + present in some NVIDIA Tegra SoCs, used to communicate audio + to the HDMI output. + + To compile this driver as a module, choose M here: the module + will be called snd-hda-tegra. + if SND_HDA config SND_HDA_DSP_LOADER diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index d0d0c19..194f3093 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,5 +1,6 @@ snd-hda-intel-objs := hda_intel.o snd-hda-controller-objs := hda_controller.o +snd-hda-tegra-objs := hda_tegra.o # for haswell power well snd-hda-intel-$(CONFIG_SND_HDA_I915) += hda_i915.o @@ -47,3 +48,4 @@ obj-$(CONFIG_SND_HDA_CODEC_HDMI) += snd-hda-codec-hdmi.o # otherwise the codec patches won't be hooked before the PCI probe # when built in kernel obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-intel.o +obj-$(CONFIG_SND_HDA_TEGRA) += snd-hda-tegra.o diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c new file mode 100644 index 0000000..afdf5e1 --- /dev/null +++ b/sound/pci/hda/hda_tegra.c @@ -0,0 +1,585 @@ +/* + * + * Implementation of primary alsa driver code base for NVIDIA Tegra HDA. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + */ + +#include <linux/clk.h> +#include <linux/clocksource.h> +#include <linux/completion.h> +#include <linux/delay.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/io.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/mutex.h> +#include <linux/of_device.h> +#include <linux/reboot.h> +#include <linux/slab.h> +#include <linux/time.h> + +#include <sound/core.h> +#include <sound/initval.h> + +#include "hda_codec.h" +#include "hda_controller.h" +#include "hda_priv.h" + +/* Defines for Nvidia Tegra HDA support */ +#define HDA_BAR0 0x8000 + +#define HDA_CFG_CMD 0x1004 +#define HDA_CFG_BAR0 0x1010 + +#define HDA_ENABLE_IO_SPACE (1 << 0) +#define HDA_ENABLE_MEM_SPACE (1 << 1) +#define HDA_ENABLE_BUS_MASTER (1 << 2) +#define HDA_ENABLE_SERR (1 << 8) +#define HDA_DISABLE_INTR (1 << 10) +#define HDA_BAR0_INIT_PROGRAM 0xFFFFFFFF +#define HDA_BAR0_FINAL_PROGRAM (1 << 14) + +/* IPFS */ +#define HDA_IPFS_CONFIG 0x180 +#define HDA_IPFS_EN_FPCI 0x1 + +#define HDA_IPFS_FPCI_BAR0 0x80 +#define HDA_FPCI_BAR0_START 0x40 + +#define HDA_IPFS_INTR_MASK 0x188 +#define HDA_IPFS_EN_INTR (1 << 16) + +struct hda_tegra { + struct azx chip; + struct device *dev; + struct clk *hda_clk; + struct clk *hda2codec_2x_clk; + struct clk *hda2hdmi_clk; + void __iomem *regs; +}; + +static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; +module_param(power_save, bint, 0644); +MODULE_PARM_DESC(power_save, + "Automatic power-saving timeout (in seconds, 0 = disable)."); + +/* + * DMA page allocation ops. + */ +static int dma_alloc_pages(struct azx *chip, int type, size_t size, + struct snd_dma_buffer *buf) +{ + return snd_dma_alloc_pages(type, chip->card->dev, size, buf); +} + +static void dma_free_pages(struct azx *chip, struct snd_dma_buffer *buf) +{ + snd_dma_free_pages(buf); +} + +static int substream_alloc_pages(struct azx *chip, + struct snd_pcm_substream *substream, + size_t size) +{ + struct azx_dev *azx_dev = get_azx_dev(substream); + + azx_dev->bufsize = 0; + azx_dev->period_bytes = 0; + azx_dev->format_val = 0; + return snd_pcm_lib_malloc_pages(substream, size); +} + +static int substream_free_pages(struct azx *chip, + struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +/* + * Register access ops. Tegra HDA register access is DWORD only. + */ +static void hda_tegra_writel(u32 value, u32 *addr) +{ + writel(value, addr); +} + +static u32 hda_tegra_readl(u32 *addr) +{ + return readl(addr); +} + +static void hda_tegra_writew(u16 value, u16 *addr) +{ + unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; + void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + u32 v; + + v = readl(dword_addr); + v &= ~(0xffff << shift); + v |= value << shift; + writel(v, dword_addr); +} + +static u16 hda_tegra_readw(u16 *addr) +{ + unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; + void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + u32 v; + + v = readl(dword_addr); + return (v >> shift) & 0xffff; +} + +static void hda_tegra_writeb(u8 value, u8 *addr) +{ + unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; + void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + u32 v; + + v = readl(dword_addr); + v &= ~(0xff << shift); + v |= value << shift; + writel(v, dword_addr); +} + +static u8 hda_tegra_readb(u8 *addr) +{ + unsigned int shift = ((unsigned long)(addr) & 0x3) << 3; + void *dword_addr = (void *)((unsigned long)(addr) & ~0x3); + u32 v; + + v = readl(dword_addr); + return (v >> shift) & 0xff; +} + +static const struct hda_controller_ops hda_tegra_ops = { + .reg_writel = hda_tegra_writel, + .reg_readl = hda_tegra_readl, + .reg_writew = hda_tegra_writew, + .reg_readw = hda_tegra_readw, + .reg_writeb = hda_tegra_writeb, + .reg_readb = hda_tegra_readb, + .dma_alloc_pages = dma_alloc_pages, + .dma_free_pages = dma_free_pages, + .substream_alloc_pages = substream_alloc_pages, + .substream_free_pages = substream_free_pages, +}; + +static void hda_tegra_init(struct hda_tegra *hda) +{ + u32 v; + + /*Enable the PCI access */ + v = readl(hda->regs + HDA_IPFS_CONFIG); + v |= HDA_IPFS_EN_FPCI; + writel(v, hda->regs + HDA_IPFS_CONFIG); + + /* Enable MEM/IO space and bus master */ + v = readl(hda->regs + HDA_CFG_CMD); + v &= ~HDA_DISABLE_INTR; + v |= HDA_ENABLE_MEM_SPACE | HDA_ENABLE_IO_SPACE | + HDA_ENABLE_BUS_MASTER | HDA_ENABLE_SERR; + writel(v, hda->regs + HDA_CFG_CMD); + + writel(HDA_BAR0_INIT_PROGRAM, hda->regs + HDA_CFG_BAR0); + writel(HDA_BAR0_FINAL_PROGRAM, hda->regs + HDA_CFG_BAR0); + writel(HDA_FPCI_BAR0_START, hda->regs + HDA_IPFS_FPCI_BAR0); + + v = readl(hda->regs + HDA_IPFS_INTR_MASK); + v |= HDA_IPFS_EN_INTR; + writel(v, hda->regs + HDA_IPFS_INTR_MASK); +} + +static int hda_tegra_enable_clocks(struct hda_tegra *data) +{ + int rc; + + rc = clk_prepare_enable(data->hda_clk); + if (rc) + return rc; + rc = clk_prepare_enable(data->hda2codec_2x_clk); + if (rc) + goto disable_hda; + rc = clk_prepare_enable(data->hda2hdmi_clk); + if (rc) + goto disable_codec_2x; + + return 0; + +disable_codec_2x: + clk_disable_unprepare(data->hda2codec_2x_clk); +disable_hda: + clk_disable_unprepare(data->hda_clk); + return rc; +} + +static void hda_tegra_disable_clocks(struct hda_tegra *data) +{ + clk_disable_unprepare(data->hda2hdmi_clk); + clk_disable_unprepare(data->hda2codec_2x_clk); + clk_disable_unprepare(data->hda_clk); +} + +#ifdef CONFIG_PM_SLEEP +/* + * power management + */ +static int hda_tegra_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct azx_pcm *p; + struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + list_for_each_entry(p, &chip->pcm_list, list) + snd_pcm_suspend_all(p->pcm); + if (chip->initialized) + snd_hda_suspend(chip->bus); + + azx_stop_chip(chip); + azx_enter_link_reset(chip); + hda_tegra_disable_clocks(hda); + + return 0; +} + +static int hda_tegra_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); + int status; + + hda_tegra_enable_clocks(hda); + + /* Read STATESTS before controller reset */ + status = azx_readw(chip, STATESTS); + + hda_tegra_init(hda); + + azx_init_chip(chip, 1); + + snd_hda_resume(chip->bus); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops hda_tegra_pm = { + SET_SYSTEM_SLEEP_PM_OPS(hda_tegra_suspend, hda_tegra_resume) +}; + +/* + * reboot notifier for hang-up problem at power-down + */ +static int hda_tegra_halt(struct notifier_block *nb, unsigned long event, + void *buf) +{ + struct azx *chip = container_of(nb, struct azx, reboot_notifier); + snd_hda_bus_reboot_notify(chip->bus); + azx_stop_chip(chip); + return NOTIFY_OK; +} + +static void hda_tegra_notifier_register(struct azx *chip) +{ + chip->reboot_notifier.notifier_call = hda_tegra_halt; + register_reboot_notifier(&chip->reboot_notifier); +} + +static void hda_tegra_notifier_unregister(struct azx *chip) +{ + if (chip->reboot_notifier.notifier_call) + unregister_reboot_notifier(&chip->reboot_notifier); +} + +/* + * destructor + */ +static int hda_tegra_dev_free(struct snd_device *device) +{ + int i; + struct azx *chip = device->device_data; + + hda_tegra_notifier_unregister(chip); + + if (chip->initialized) { + for (i = 0; i < chip->num_streams; i++) + azx_stream_stop(chip, &chip->azx_dev[i]); + azx_stop_chip(chip); + } + + azx_free_stream_pages(chip); + + return 0; +} + +static int hda_tegra_init_chip(struct azx *chip, struct platform_device *pdev) +{ + struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); + struct device *dev = hda->dev; + struct resource *res; + int err; + + hda->hda_clk = devm_clk_get(dev, "hda"); + if (IS_ERR(hda->hda_clk)) + return PTR_ERR(hda->hda_clk); + hda->hda2codec_2x_clk = devm_clk_get(dev, "hda2codec_2x"); + if (IS_ERR(hda->hda2codec_2x_clk)) + return PTR_ERR(hda->hda2codec_2x_clk); + hda->hda2hdmi_clk = devm_clk_get(dev, "hda2hdmi"); + if (IS_ERR(hda->hda2hdmi_clk)) + return PTR_ERR(hda->hda2hdmi_clk); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + hda->regs = devm_ioremap_resource(dev, res); + if (IS_ERR(chip->remap_addr)) + return PTR_ERR(chip->remap_addr); + + chip->remap_addr = hda->regs + HDA_BAR0; + chip->addr = res->start + HDA_BAR0; + + err = hda_tegra_enable_clocks(hda); + if (err) + return err; + + hda_tegra_init(hda); + + return 0; +} + +/* + * The codecs were powered up in snd_hda_codec_new(). + * Now all initialization done, so turn them down if possible + */ +static void power_down_all_codecs(struct azx *chip) +{ + struct hda_codec *codec; + list_for_each_entry(codec, &chip->bus->codec_list, list) + snd_hda_power_down(codec); +} + +static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) +{ + struct snd_card *card = chip->card; + int err; + unsigned short gcap; + int irq_id = platform_get_irq(pdev, 0); + + err = hda_tegra_init_chip(chip, pdev); + if (err) + return err; + + err = devm_request_irq(chip->card->dev, irq_id, azx_interrupt, + IRQF_SHARED, KBUILD_MODNAME, chip); + if (err) { + dev_err(chip->card->dev, + "unable to request IRQ %d, disabling device\n", + irq_id); + return err; + } + chip->irq = irq_id; + + synchronize_irq(chip->irq); + + gcap = azx_readw(chip, GCAP); + dev_dbg(card->dev, "chipset global capabilities = 0x%x\n", gcap); + + /* read number of streams from GCAP register instead of using + * hardcoded value + */ + chip->capture_streams = (gcap >> 8) & 0x0f; + chip->playback_streams = (gcap >> 12) & 0x0f; + if (!chip->playback_streams && !chip->capture_streams) { + /* gcap didn't give any info, switching to old method */ + chip->playback_streams = ICH6_NUM_PLAYBACK; + chip->capture_streams = ICH6_NUM_CAPTURE; + } + chip->capture_index_offset = 0; + chip->playback_index_offset = chip->capture_streams; + chip->num_streams = chip->playback_streams + chip->capture_streams; + chip->azx_dev = devm_kcalloc(card->dev, chip->num_streams, + sizeof(*chip->azx_dev), GFP_KERNEL); + if (!chip->azx_dev) + return -ENOMEM; + + err = azx_alloc_stream_pages(chip); + if (err < 0) + return err; + + /* initialize streams */ + azx_init_stream(chip); + + /* initialize chip */ + azx_init_chip(chip, 1); + + /* codec detection */ + if (!chip->codec_mask) { + dev_err(card->dev, "no codecs found!\n"); + return -ENODEV; + } + + strcpy(card->driver, "tegra-hda"); + strcpy(card->shortname, "tegra-hda"); + snprintf(card->longname, sizeof(card->longname), + "%s at 0x%lx irq %i", + card->shortname, chip->addr, chip->irq); + + return 0; +} + +/* + * constructor + */ +static int hda_tegra_create(struct snd_card *card, + unsigned int driver_caps, + const struct hda_controller_ops *hda_ops, + struct hda_tegra *hda) +{ + static struct snd_device_ops ops = { + .dev_free = hda_tegra_dev_free, + }; + struct azx *chip; + int err; + + chip = &hda->chip; + + spin_lock_init(&chip->reg_lock); + mutex_init(&chip->open_mutex); + chip->card = card; + chip->ops = hda_ops; + chip->irq = -1; + chip->driver_caps = driver_caps; + chip->driver_type = driver_caps & 0xff; + chip->dev_index = 0; + INIT_LIST_HEAD(&chip->pcm_list); + INIT_LIST_HEAD(&chip->list); + + chip->position_fix[0] = POS_FIX_AUTO; + chip->position_fix[1] = POS_FIX_AUTO; + chip->codec_probe_mask = -1; + + chip->single_cmd = false; + chip->snoop = true; + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + dev_err(card->dev, "Error creating device\n"); + return err; + } + + return 0; +} + +static const struct of_device_id hda_tegra_match[] = { + { .compatible = "nvidia,tegra30-hda" }, + {}, +}; + +static int hda_tegra_probe(struct platform_device *pdev) +{ + struct snd_card *card; + struct azx *chip; + struct hda_tegra *hda; + const struct of_device_id *of_id; + int err; + const unsigned int driver_flags = AZX_DCAPS_RIRB_DELAY; + + of_id = of_match_device(hda_tegra_match, &pdev->dev); + if (!of_id) + return -ENODEV; + + hda = devm_kzalloc(&pdev->dev, sizeof(*hda), GFP_KERNEL); + if (!hda) + return -ENOMEM; + hda->dev = &pdev->dev; + chip = &hda->chip; + + err = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + THIS_MODULE, 0, &card); + if (err < 0) { + dev_err(&pdev->dev, "Error creating card!\n"); + return err; + } + + err = hda_tegra_create(card, driver_flags, &hda_tegra_ops, hda); + if (err < 0) + goto out_free; + card->private_data = chip; + + dev_set_drvdata(&pdev->dev, card); + + err = hda_tegra_first_init(chip, pdev); + if (err < 0) + goto out_free; + + /* create codec instances */ + err = azx_codec_create(chip, NULL, 0, &power_save); + if (err < 0) + goto out_free; + + err = azx_codec_configure(chip); + if (err < 0) + goto out_free; + + /* create PCM streams */ + err = snd_hda_build_pcms(chip->bus); + if (err < 0) + goto out_free; + + /* create mixer controls */ + err = azx_mixer_create(chip); + if (err < 0) + goto out_free; + + err = snd_card_register(chip->card); + if (err < 0) + goto out_free; + + chip->running = 1; + power_down_all_codecs(chip); + hda_tegra_notifier_register(chip); + + return 0; + +out_free: + snd_card_free(card); + return err; +} + +static int hda_tegra_remove(struct platform_device *pdev) +{ + return snd_card_free(dev_get_drvdata(&pdev->dev)); +} + +static struct platform_driver tegra_platform_hda = { + .driver = { + .name = "tegra-hda", + .pm = &hda_tegra_pm, + .of_match_table = hda_tegra_match, + }, + .probe = hda_tegra_probe, + .remove = hda_tegra_remove, +}; +module_platform_driver(tegra_platform_hda); + +MODULE_DESCRIPTION("Tegra HDA bus driver"); +MODULE_LICENSE("GPL v2"); +MODULE_DEVICE_TABLE(of, tegra_platform_hda_match); -- 1.8.1.3.605.g02339dd

2 2

[alsa-devel] How to create a proc entry for an ASoc codec?
by Lin, Mengdong 22 Apr '14

22 Apr '14

Hi, How to create a proc entry for an ASoc codec under the sound card? We hope to create an proc entry for an I2S codec under /proc/asound/cardx, to debug the codec status at runtime. So after the machine driver calls snd_soc_register_card(&soc_card), I calls snd_card_proc_new(soc_card .snd_card, ....) But no codec entry is created although dmesg shows this function succeeds. How ASoC creates the proc entries such as 'id' for the sound card? Thanks Mengdong

3 2

[alsa-devel] [PATCHv4 1/2] ALSA: hda/hdmi - Add Nvidia Tegra124 HDMI support
by Dylan Reid 21 Apr '14

21 Apr '14

From: Sumit Bhattacharya <sumitb(a)nvidia.com> Add the Tegra 12x HDMI controller id to patch_hdmi. Signed-off-by: Sumit Bhattacharya <sumitb(a)nvidia.com> Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- No changes since v1. --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0cb5b89..b5d1371 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3322,6 +3322,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .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 = 0x10de0028, .name = "Tegra12x HDMI", .patch = patch_nvhdmi }, { .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 }, @@ -3377,6 +3378,7 @@ 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"); -- 1.8.1.3.605.g02339dd

2 1

[alsa-devel] [PATCH 0/7] ASoC: Move IO and kcontrols to the component level
by Lars-Peter Clausen 21 Apr '14

21 Apr '14

This series contains the remaining patches from the previous series that have not been applied yet. The series has been rebased ontop of asoc/for-next and depends on topic/sta350, topic/core, topic/dapm, topic/cache and topic/component. Some minor updates were made to the patches and three new patches: * One new user of snd_kcontrol_chip() showed up, the patch converts it to snd_soc_kcontrol_codec() * Change return type of snd_soc_write() from unsigned int to int * Remove ASoC level tracing Original cover letter: This series is the first step towards full componentisation of the ASoC core. It moves both the IO abstraction layers within ASoC as well as the standard set of kcontrols to the component level. This for example means we can get rid of constructs like if (w->codec) snd_soc_read(....) else if(w->platform) snd_soc_platform_read(...) Moving the kcontrols to the component level means we can use the same implementation also for other non-CODEC components. E.g. there seems to be an increasing amount of CPU components that have basic signal processing and things like volume controls etc. whose register layout is similar to those used in CODECs. Currently each CPU component driver re-implements these controls by hand. The first two patches introduce two new helper functions which hide the actual implementation on how the CODEC or platform struct that register a control can be obtained from the control. This means that when the actual implementation is changed only the two helper functions need to be updated and not every single driver. The patches that follow that are just cleanups removing unused IO stuff and move all IO functions to soc-io.c. The next step is to make platforms also components. And then finally first the IO abstraction layers in ASoC are unified at the component level and then on top of that the kcontrol helpers are moved to the component level. The series depends on quite a few topic branches related to changes to the core and cleanups for individual drivers. It is probably best to place it on top of asoc-v3.15-2. The patch that moves the kcontrols to the component level also has a runtime dependency on the not yet applied patches that move the ams-delta and mfld_machine controls to the card level. - Lars Lars-Peter Clausen (7): ASoC: sta350: Use snd_soc_kcontrol_codec() ASoC: Change return type of snd_soc_write() to int ASoC: Move IO abstraction to the component level ASoC: Move standard kcontrol helpers to the component level ASoC: Remove snd_soc_update_bits_locked() ASoC: dapm: Rename soc_widget_update_bits_locked() to soc_widget_update_bits() ASoC: Remove ASoC level IO tracing include/sound/soc-dapm.h | 1 + include/sound/soc.h | 55 ++++++-- include/trace/events/asoc.h | 92 ------------- sound/soc/codecs/sta350.c | 4 +- sound/soc/soc-core.c | 284 +++++++++++++++++++++++++------------- sound/soc/soc-dapm.c | 91 ++---------- sound/soc/soc-io.c | 327 ++++++++++++++++++++++++++------------------ 7 files changed, 450 insertions(+), 404 deletions(-) -- 1.8.0

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[alsa-devel] [dapm question] About "playback_widget" power on or not
by 胡洪兵 21 Apr '14

21 Apr '14

hi dapm developers, I have a question about "playback_widget". Nobody gives me right answer, so email the question to devel maillist. I am sorry to disturb you and thanks for your explain Question: my machine driver export two pcm devices, eg. "pcm0", "pcm1". Open the "pcm0" for Playback, just active AIF3 playback_widget "AIF3 Playbck". at same time，one path of pcm1 is also power on。 the path is “Headset Mic -> IN3R -> AIF1TX Input 1 -> AIF1TX Mixer -> AIF1TX -> AIF1 Playbck -> AIF1RX1 -> Headphone Input 1 -> Headphone Mixer -> Headphone” my question is "the widgets of the path should be powered or not ?" I guess that the path should not be powered, because "pcm1" is not open, all paths of it should not be actvie. Maybe I am wrong， but let me known. Thank you very much! ***************************************************************case snd_soc_dapm_dai_in: w->power_check = dapm_dac_check_power; /* Check to see if a DAC has power */static int dapm_dac_check_power(struct snd_soc_dapm_widget *w){ int out; DAPM_UPDATE_STAT(w, power_checks); if (w->active) { out = is_connected_output_ep(w, NULL); dapm_clear_walk_output(w->dapm, &w->sinks); return out != 0; } else { return dapm_generic_check_power(w); }} ***************************************************************

1 0

[alsa-devel] alsa-utils aplay/arecord can't be interrupted when streaming is not active
by jiwang 21 Apr '14

21 Apr '14

Hi All With latest 1.0.27.2 alsa-utils, I am facing problem to interrupt aplay/arecord when the streaming is not active (no data can be written to or read from cards) further more, if I start aplay/arecord with "-N" nonblocking option, then aplay/arecord can never exit, unless I do a system reset I found if I revert the following patch /commit 1d0042d7e948815f8015c62bc53143eed3b83206// //Author: Jaroslav Kysela <perex(a)perex.cz>// //Date: Mon Apr 8 13:30:54 2013 +0200// // // aplay/arecord: change the interrupt handling using snd_pcm_abort()// //// // It is required (exclude the fatal SIGABRT) to call snd_pcm_close() and// // the end of work (outside of the interrupt handler). Use new snd_pcm_abort()// // function to inform alsa-lib to not ignore EINTR and move the in_aborting// // variable to the global scope to be checked in the i/o loops.// //// // Signed-off-by: Jaroslav Kysela <perex(a)perex.cz>// / http://git.alsa-project.org/?p=alsa-utils.git;a=commit;h=1d0042d7e948815f80… then aplay/arecord can be interrupted correctly Is it a defect in aplay.c or did I miss something? Thanks, Jiada

3 6

[alsa-devel] alsa-devel@alsa-project.org
by Andrew Reyes 19 Apr '14

19 Apr '14

Is there anything else I can add to help fix this issue? I would really like to get this resolved.

1 0

[alsa-devel] [PATCH 0/8] ARM: at91: sama5d3: enable sound support
by Bo Shen 19 Apr '14

19 Apr '14

This patch add sound support on sama5d3xek board. This patch series based on the following patch set: clk: at91: better support for the PCKs http://comments.gmane.org/gmane.linux.kernel/1664477 Bo Shen (5): ARM: at91: sama5d3: add DMA property for SSC devices ARM: at91: sama5d3: disable sound by default ARM: at91: sama5d3: correct the sound compatible string ARM: at91: sama5d3: add the missing property ARM: at91: sama5d3: clock for ssc from rk pin Boris BREZILLON (3): ASoC: atmel: support CCF based clks ASoC: atmel: document clock properties of the wm8904 driver ARM: at91/dt: add clock properties to the wm8904 codec node .../devicetree/bindings/sound/atmel-wm8904.txt | 2 ++ arch/arm/boot/dts/sama5d3.dtsi | 6 ++++++ arch/arm/boot/dts/sama5d3xmb.dtsi | 11 +++++++++- sound/soc/atmel/atmel_wm8904.c | 25 +++++++++++----------- 4 files changed, 31 insertions(+), 13 deletions(-) -- 1.8.5.2

4 7

Re: [alsa-devel] [PATCHv3 2/2] ALSA: hda - Add driver for Tegra SoC HDA
by Dylan Reid 19 Apr '14

19 Apr '14

On Fri, Apr 18, 2014 at 1:22 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > At Wed, 16 Apr 2014 14:39:32 -0700, > Dylan Reid wrote: >> >> This adds a driver for the HDA block in Tegra SoCs. The HDA bus is >> used to communicate with the HDMI codec on Tegra124. >> >> Most of the code is re-used from the Intel/PCI HDA driver. It brings >> over only the power_save module parameter. >> >> Signed-off-by: Dylan Reid <dgreid(a)chromium.org> > > Thanks, this looks better, but a few comments below. > Besides, I'd like to have a review in DT part. Thanks for looking. I'll address the comments below and send a v4 with some more DT maintainers CCed. > > >> +/* >> + * Register access ops. Tegra HDA register access is DWORD only. >> + */ >> +static void hda_tegra_writel(u32 value, u32 *addr) >> +{ >> + writel(value, addr); >> +} >> + >> +static u32 hda_tegra_readl(u32 *addr) >> +{ >> + return readl(addr); >> +} >> + >> +static void hda_tegra_writew(u16 value, u16 *addr) >> +{ >> + unsigned long shift = ((uintptr_t)(addr) & 0x3) << 3; >> + void *dword_addr = (void *)((uintptr_t)(addr) & ~0x3); >> + u32 v; >> + >> + v = readl(dword_addr); >> + v &= ~(0xffff << shift); >> + v |= value << shift; >> + writel(v, dword_addr); >> +} > > The shift doesn't have to be long at all, and the cast in kernel is > usually just unsigned long instead of uintptr_t. > > >> +#ifdef CONFIG_PM_SLEEP >> +/* >> + * power management >> + */ >> +static int hda_tegra_suspend(struct device *dev) >> +{ >> + struct snd_card *card = dev_get_drvdata(dev); >> + struct azx *chip = card->private_data; >> + struct azx_pcm *p; >> + struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); >> + >> + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); >> + list_for_each_entry(p, &chip->pcm_list, list) >> + snd_pcm_suspend_all(p->pcm); >> + if (chip->initialized) >> + snd_hda_suspend(chip->bus); >> + >> + /* enable controller wake up event */ >> + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | >> + STATESTS_INT_MASK); > > I guess WAKEEN doesn't work for Tegra? I'm unsure, but this doesn't belong in the regular resume path, so it will disappear in v4. > >> + >> + azx_stop_chip(chip); >> + azx_enter_link_reset(chip); >> + hda_tegra_disable_clocks(hda); >> + >> + return 0; >> +} >> + >> +static int hda_tegra_resume(struct device *dev) >> +{ >> + struct snd_card *card = dev_get_drvdata(dev); >> + struct azx *chip = card->private_data; >> + struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); >> + int status; >> + struct hda_codec *codec; >> + >> + hda_tegra_enable_clocks(hda); >> + >> + /* Read STATESTS before controller reset */ >> + status = azx_readw(chip, STATESTS); >> + >> + hda_tegra_init(hda); >> + >> + if (status && chip->bus) { >> + list_for_each_entry(codec, &chip->bus->codec_list, list) >> + if (status & (1 << codec->addr)) >> + queue_delayed_work(codec->bus->workq, >> + &codec->jackpoll_work, >> + codec->jackpoll_interval); >> + } > > This is superfluous for the regular resume. It was some missing call > only in the runtime resume. > >> + >> + /* disable controller Wake Up event*/ >> + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & ~STATESTS_INT_MASK); > > This can be removed like the suspend. > > > thanks, > > Takashi

1 0

[alsa-devel] [PATCH] ASoC: rt5651: Staticise non-exported symbols
by Mark Brown 18 Apr '14

18 Apr '14

From: Mark Brown <broonie(a)linaro.org> Make the dai_ops const too since we can. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/rt5651.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c index a1577369c49c..f785b81238ae 100644 --- a/sound/soc/codecs/rt5651.c +++ b/sound/soc/codecs/rt5651.c @@ -1736,14 +1736,14 @@ static int rt5651_resume(struct snd_soc_codec *codec) #define RT5651_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -struct snd_soc_dai_ops rt5651_aif_dai_ops = { +static const struct snd_soc_dai_ops rt5651_aif_dai_ops = { .hw_params = rt5651_hw_params, .set_fmt = rt5651_set_dai_fmt, .set_sysclk = rt5651_set_dai_sysclk, .set_pll = rt5651_set_dai_pll, }; -struct snd_soc_dai_driver rt5651_dai[] = { +static struct snd_soc_dai_driver rt5651_dai[] = { { .name = "rt5651-aif1", .id = RT5651_AIF1, @@ -1882,7 +1882,7 @@ static int rt5651_i2c_remove(struct i2c_client *i2c) return 0; } -struct i2c_driver rt5651_i2c_driver = { +static struct i2c_driver rt5651_i2c_driver = { .driver = { .name = "rt5651", .owner = THIS_MODULE, -- 1.9.2

1 0

[alsa-devel] [PATCH] ASoC: add RT5651 CODEC driver
by bardliao＠realtek.com 18 Apr '14

18 Apr '14

From: Bard Liao <bardliao(a)realtek.com> This patch adds the Realtek ALC5651 codec driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- include/sound/rt5651.h | 21 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5651.c | 1898 +++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt5651.h | 2081 +++++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 4006 insertions(+) create mode 100644 include/sound/rt5651.h create mode 100644 sound/soc/codecs/rt5651.c create mode 100644 sound/soc/codecs/rt5651.h diff --git a/include/sound/rt5651.h b/include/sound/rt5651.h new file mode 100644 index 0000000..d35de75 --- /dev/null +++ b/include/sound/rt5651.h @@ -0,0 +1,21 @@ +/* + * linux/sound/rt286.h -- Platform data for RT286 + * + * Copyright 2013 Realtek Microelectronics + * + * 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. + */ + +#ifndef __LINUX_SND_RT5651_H +#define __LINUX_SND_RT5651_H + +struct rt5651_platform_data { + /* IN2 can optionally be differential */ + bool in2_diff; + + bool dmic_en; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c7b853f..6bdacaf 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -71,6 +71,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM512x_SPI if SPI_MASTER select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C + select SND_SOC_RT5651 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SIRF_AUDIO_CODEC @@ -397,6 +398,9 @@ config SND_SOC_RT5631 config SND_SOC_RT5640 tristate +config SND_SOC_RT5651 + tristate + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate "Freescale SGTL5000 CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index efdb4d0..923b6b5e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -60,6 +60,7 @@ snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o +snd-soc-rt5651-objs := rt5651.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -212,6 +213,7 @@ obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o +obj-$(CONFIG_SND_SOC_RT5651) += snd-soc-rt5651.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o diff --git a/sound/soc/codecs/rt5651.c b/sound/soc/codecs/rt5651.c new file mode 100644 index 0000000..a157736 --- /dev/null +++ b/sound/soc/codecs/rt5651.c @@ -0,0 +1,1898 @@ +/* + * rt5651.c -- RT5651 ALSA SoC audio codec driver + * + * Copyright 2014 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.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 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <sound/core.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 "rt5651.h" + +#define RT5651_DEVICE_ID_VALUE 0x6281 + +#define RT5651_PR_RANGE_BASE (0xff + 1) +#define RT5651_PR_SPACING 0x100 + +#define RT5651_PR_BASE (RT5651_PR_RANGE_BASE + (0 * RT5651_PR_SPACING)) + +static const struct regmap_range_cfg rt5651_ranges[] = { + { .name = "PR", .range_min = RT5651_PR_BASE, + .range_max = RT5651_PR_BASE + 0xb4, + .selector_reg = RT5651_PRIV_INDEX, + .selector_mask = 0xff, + .selector_shift = 0x0, + .window_start = RT5651_PRIV_DATA, + .window_len = 0x1, }, +}; + +static struct reg_default init_list[] = { + {RT5651_PR_BASE + 0x3d, 0x3e00}, +}; + +static const struct reg_default rt5651_reg[] = { + { 0x00, 0x0000 }, + { 0x02, 0xc8c8 }, + { 0x03, 0xc8c8 }, + { 0x05, 0x0000 }, + { 0x0d, 0x0000 }, + { 0x0e, 0x0000 }, + { 0x0f, 0x0808 }, + { 0x10, 0x0808 }, + { 0x19, 0xafaf }, + { 0x1a, 0xafaf }, + { 0x1b, 0x0c00 }, + { 0x1c, 0x2f2f }, + { 0x1d, 0x2f2f }, + { 0x1e, 0x0000 }, + { 0x27, 0x7860 }, + { 0x28, 0x7070 }, + { 0x29, 0x8080 }, + { 0x2a, 0x5252 }, + { 0x2b, 0x5454 }, + { 0x2f, 0x0000 }, + { 0x30, 0x5000 }, + { 0x3b, 0x0000 }, + { 0x3c, 0x006f }, + { 0x3d, 0x0000 }, + { 0x3e, 0x006f }, + { 0x45, 0x6000 }, + { 0x4d, 0x0000 }, + { 0x4e, 0x0000 }, + { 0x4f, 0x0279 }, + { 0x50, 0x0000 }, + { 0x51, 0x0000 }, + { 0x52, 0x0279 }, + { 0x53, 0xf000 }, + { 0x61, 0x0000 }, + { 0x62, 0x0000 }, + { 0x63, 0x00c0 }, + { 0x64, 0x0000 }, + { 0x65, 0x0000 }, + { 0x66, 0x0000 }, + { 0x70, 0x8000 }, + { 0x71, 0x8000 }, + { 0x73, 0x1104 }, + { 0x74, 0x0c00 }, + { 0x75, 0x1400 }, + { 0x77, 0x0c00 }, + { 0x78, 0x4000 }, + { 0x79, 0x0123 }, + { 0x80, 0x0000 }, + { 0x81, 0x0000 }, + { 0x82, 0x0000 }, + { 0x83, 0x0800 }, + { 0x84, 0x0000 }, + { 0x85, 0x0008 }, + { 0x89, 0x0000 }, + { 0x8e, 0x0004 }, + { 0x8f, 0x1100 }, + { 0x90, 0x0000 }, + { 0x93, 0x2000 }, + { 0x94, 0x0200 }, + { 0xb0, 0x2080 }, + { 0xb1, 0x0000 }, + { 0xb4, 0x2206 }, + { 0xb5, 0x1f00 }, + { 0xb6, 0x0000 }, + { 0xbb, 0x0000 }, + { 0xbc, 0x0000 }, + { 0xbd, 0x0000 }, + { 0xbe, 0x0000 }, + { 0xbf, 0x0000 }, + { 0xc0, 0x0400 }, + { 0xc1, 0x0000 }, + { 0xc2, 0x0000 }, + { 0xcf, 0x0013 }, + { 0xd0, 0x0680 }, + { 0xd1, 0x1c17 }, + { 0xd3, 0xb320 }, + { 0xd9, 0x0809 }, + { 0xfa, 0x0010 }, + { 0xfe, 0x10ec }, + { 0xff, 0x6281 }, +}; + +static bool rt5651_volatile_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5651_ranges); i++) { + if ((reg >= rt5651_ranges[i].window_start && + reg <= rt5651_ranges[i].window_start + + rt5651_ranges[i].window_len) || + (reg >= rt5651_ranges[i].range_min && + reg <= rt5651_ranges[i].range_max)) { + return true; + } + } + + switch (reg) { + case RT5651_RESET: + case RT5651_PRIV_DATA: + case RT5651_EQ_CTRL1: + case RT5651_ALC_1: + case RT5651_IRQ_CTRL2: + case RT5651_INT_IRQ_ST: + case RT5651_PGM_REG_ARR1: + case RT5651_PGM_REG_ARR3: + case RT5651_VENDOR_ID: + case RT5651_DEVICE_ID: + return true; + default: + return false; + } +} + +static bool rt5651_readable_register(struct device *dev, unsigned int reg) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(rt5651_ranges); i++) { + if ((reg >= rt5651_ranges[i].window_start && + reg <= rt5651_ranges[i].window_start + + rt5651_ranges[i].window_len) || + (reg >= rt5651_ranges[i].range_min && + reg <= rt5651_ranges[i].range_max)) { + return true; + } + } + + switch (reg) { + case RT5651_RESET: + case RT5651_VERSION_ID: + case RT5651_VENDOR_ID: + case RT5651_DEVICE_ID: + case RT5651_HP_VOL: + case RT5651_LOUT_CTRL1: + case RT5651_LOUT_CTRL2: + case RT5651_IN1_IN2: + case RT5651_IN3: + case RT5651_INL1_INR1_VOL: + case RT5651_INL2_INR2_VOL: + case RT5651_DAC1_DIG_VOL: + case RT5651_DAC2_DIG_VOL: + case RT5651_DAC2_CTRL: + case RT5651_ADC_DIG_VOL: + case RT5651_ADC_DATA: + case RT5651_ADC_BST_VOL: + case RT5651_STO1_ADC_MIXER: + case RT5651_STO2_ADC_MIXER: + case RT5651_AD_DA_MIXER: + case RT5651_STO_DAC_MIXER: + case RT5651_DD_MIXER: + case RT5651_DIG_INF_DATA: + case RT5651_PDM_CTL: + case RT5651_REC_L1_MIXER: + case RT5651_REC_L2_MIXER: + case RT5651_REC_R1_MIXER: + case RT5651_REC_R2_MIXER: + case RT5651_HPO_MIXER: + case RT5651_OUT_L1_MIXER: + case RT5651_OUT_L2_MIXER: + case RT5651_OUT_L3_MIXER: + case RT5651_OUT_R1_MIXER: + case RT5651_OUT_R2_MIXER: + case RT5651_OUT_R3_MIXER: + case RT5651_LOUT_MIXER: + case RT5651_PWR_DIG1: + case RT5651_PWR_DIG2: + case RT5651_PWR_ANLG1: + case RT5651_PWR_ANLG2: + case RT5651_PWR_MIXER: + case RT5651_PWR_VOL: + case RT5651_PRIV_INDEX: + case RT5651_PRIV_DATA: + case RT5651_I2S1_SDP: + case RT5651_I2S2_SDP: + case RT5651_ADDA_CLK1: + case RT5651_ADDA_CLK2: + case RT5651_DMIC: + case RT5651_TDM_CTL_1: + case RT5651_TDM_CTL_2: + case RT5651_TDM_CTL_3: + case RT5651_GLB_CLK: + case RT5651_PLL_CTRL1: + case RT5651_PLL_CTRL2: + case RT5651_PLL_MODE_1: + case RT5651_PLL_MODE_2: + case RT5651_PLL_MODE_3: + case RT5651_PLL_MODE_4: + case RT5651_PLL_MODE_5: + case RT5651_PLL_MODE_6: + case RT5651_PLL_MODE_7: + case RT5651_DEPOP_M1: + case RT5651_DEPOP_M2: + case RT5651_DEPOP_M3: + case RT5651_CHARGE_PUMP: + case RT5651_MICBIAS: + case RT5651_A_JD_CTL1: + case RT5651_EQ_CTRL1: + case RT5651_EQ_CTRL2: + case RT5651_ALC_1: + case RT5651_ALC_2: + case RT5651_ALC_3: + case RT5651_JD_CTRL1: + case RT5651_JD_CTRL2: + case RT5651_IRQ_CTRL1: + case RT5651_IRQ_CTRL2: + case RT5651_INT_IRQ_ST: + case RT5651_GPIO_CTRL1: + case RT5651_GPIO_CTRL2: + case RT5651_GPIO_CTRL3: + case RT5651_PGM_REG_ARR1: + case RT5651_PGM_REG_ARR2: + case RT5651_PGM_REG_ARR3: + case RT5651_PGM_REG_ARR4: + case RT5651_PGM_REG_ARR5: + case RT5651_SCB_FUNC: + case RT5651_SCB_CTRL: + case RT5651_BASE_BACK: + case RT5651_MP3_PLUS1: + case RT5651_MP3_PLUS2: + case RT5651_ADJ_HPF_CTRL1: + case RT5651_ADJ_HPF_CTRL2: + case RT5651_HP_CALIB_AMP_DET: + case RT5651_HP_CALIB2: + case RT5651_SV_ZCD1: + case RT5651_SV_ZCD2: + case RT5651_D_MISC: + case RT5651_DUMMY2: + case RT5651_DUMMY3: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -4650, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(in_vol_tlv, -3450, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static unsigned int bst_tlv[] = { + TLV_DB_RANGE_HEAD(7), + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), +}; + +/* Interface data select */ +static const char * const rt5651_data_select[] = { + "Normal", "Swap", "left copy to right", "right copy to left"}; + +static SOC_ENUM_SINGLE_DECL(rt5651_if2_dac_enum, RT5651_DIG_INF_DATA, + RT5651_IF2_DAC_SEL_SFT, rt5651_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5651_if2_adc_enum, RT5651_DIG_INF_DATA, + RT5651_IF2_ADC_SEL_SFT, rt5651_data_select); + +static const struct snd_kcontrol_new rt5651_snd_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE_TLV("HP Playback Volume", RT5651_HP_VOL, + RT5651_L_VOL_SFT, RT5651_R_VOL_SFT, 39, 1, out_vol_tlv), + /* OUTPUT Control */ + SOC_DOUBLE_TLV("OUT Playback Volume", RT5651_LOUT_CTRL1, + RT5651_L_VOL_SFT, RT5651_R_VOL_SFT, 39, 1, out_vol_tlv), + + /* DAC Digital Volume */ + SOC_DOUBLE("DAC2 Playback Switch", RT5651_DAC2_CTRL, + RT5651_M_DAC_L2_VOL_SFT, RT5651_M_DAC_R2_VOL_SFT, 1, 1), + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5651_DAC1_DIG_VOL, + RT5651_L_VOL_SFT, RT5651_R_VOL_SFT, + 175, 0, dac_vol_tlv), + SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5651_DAC2_DIG_VOL, + RT5651_L_VOL_SFT, RT5651_R_VOL_SFT, + 175, 0, dac_vol_tlv), + /* IN1/IN2 Control */ + SOC_SINGLE_TLV("IN1 Boost", RT5651_IN1_IN2, + RT5651_BST_SFT1, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN2 Boost", RT5651_IN1_IN2, + RT5651_BST_SFT2, 8, 0, bst_tlv), + /* INL/INR Volume Control */ + SOC_DOUBLE_TLV("IN Capture Volume", RT5651_INL1_INR1_VOL, + RT5651_INL_VOL_SFT, RT5651_INR_VOL_SFT, + 31, 1, in_vol_tlv), + /* ADC Digital Volume Control */ + SOC_DOUBLE("ADC Capture Switch", RT5651_ADC_DIG_VOL, + RT5651_L_MUTE_SFT, RT5651_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("ADC Capture Volume", RT5651_ADC_DIG_VOL, + RT5651_L_VOL_SFT, RT5651_R_VOL_SFT, + 127, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5651_ADC_DATA, + RT5651_L_VOL_SFT, RT5651_R_VOL_SFT, + 127, 0, adc_vol_tlv), + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("ADC Boost Gain", RT5651_ADC_BST_VOL, + RT5651_ADC_L_BST_SFT, RT5651_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), + + /* ASRC */ + SOC_SINGLE("IF1 ASRC Switch", RT5651_PLL_MODE_1, + RT5651_STO1_T_SFT, 1, 0), + SOC_SINGLE("IF2 ASRC Switch", RT5651_PLL_MODE_1, + RT5651_STO2_T_SFT, 1, 0), + SOC_SINGLE("DMIC ASRC Switch", RT5651_PLL_MODE_1, + RT5651_DMIC_1_M_SFT, 1, 0), + + SOC_ENUM("ADC IF2 Data Switch", rt5651_if2_adc_enum), + SOC_ENUM("DAC IF2 Data Switch", rt5651_if2_dac_enum), +}; + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); + int div[] = {2, 3, 4, 6, 8, 12}, idx = -EINVAL; + int i, rate, red, bound, temp; + + rate = rt5651->sysclk; + red = 3000000 * 12; + for (i = 0; i < ARRAY_SIZE(div); i++) { + bound = div[i] * 3000000; + if (rate > bound) + continue; + temp = bound - rate; + if (temp < red) { + red = temp; + idx = i; + } + } + if (idx < 0) + dev_err(codec->dev, "Failed to set DMIC clock\n"); + else + snd_soc_update_bits(codec, RT5651_DMIC, RT5651_DMIC_CLK_MASK, + idx << RT5651_DMIC_CLK_SFT); + + return idx; +} + +static int is_sysclk_from_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + + val = snd_soc_read(source->codec, RT5651_GLB_CLK); + val &= RT5651_SCLK_SRC_MASK; + if (val == RT5651_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5651_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5651_STO1_ADC_MIXER, + RT5651_M_STO1_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5651_STO1_ADC_MIXER, + RT5651_M_STO1_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5651_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5651_STO1_ADC_MIXER, + RT5651_M_STO1_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5651_STO1_ADC_MIXER, + RT5651_M_STO1_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5651_sto2_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5651_STO2_ADC_MIXER, + RT5651_M_STO2_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5651_STO2_ADC_MIXER, + RT5651_M_STO2_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5651_sto2_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5651_STO2_ADC_MIXER, + RT5651_M_STO2_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5651_STO2_ADC_MIXER, + RT5651_M_STO2_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5651_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5651_AD_DA_MIXER, + RT5651_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INF1 Switch", RT5651_AD_DA_MIXER, + RT5651_M_IF1_DAC_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5651_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5651_AD_DA_MIXER, + RT5651_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INF1 Switch", RT5651_AD_DA_MIXER, + RT5651_M_IF1_DAC_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5651_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5651_STO_DAC_MIXER, + RT5651_M_DAC_L1_MIXL_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5651_STO_DAC_MIXER, + RT5651_M_DAC_L2_MIXL_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5651_STO_DAC_MIXER, + RT5651_M_DAC_R1_MIXL_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5651_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5651_STO_DAC_MIXER, + RT5651_M_DAC_R1_MIXR_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5651_STO_DAC_MIXER, + RT5651_M_DAC_R2_MIXR_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5651_STO_DAC_MIXER, + RT5651_M_DAC_L1_MIXR_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5651_dd_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5651_DD_MIXER, + RT5651_M_STO_DD_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5651_DD_MIXER, + RT5651_M_STO_DD_L2_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5651_DD_MIXER, + RT5651_M_STO_DD_R2_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5651_dd_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5651_DD_MIXER, + RT5651_M_STO_DD_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5651_DD_MIXER, + RT5651_M_STO_DD_R2_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5651_DD_MIXER, + RT5651_M_STO_DD_L2_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5651_rec_l_mix[] = { + SOC_DAPM_SINGLE("INL1 Switch", RT5651_REC_L2_MIXER, + RT5651_M_IN1_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST3 Switch", RT5651_REC_L2_MIXER, + RT5651_M_BST3_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5651_REC_L2_MIXER, + RT5651_M_BST2_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5651_REC_L2_MIXER, + RT5651_M_BST1_RM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5651_rec_r_mix[] = { + SOC_DAPM_SINGLE("INR1 Switch", RT5651_REC_R2_MIXER, + RT5651_M_IN1_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST3 Switch", RT5651_REC_R2_MIXER, + RT5651_M_BST3_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5651_REC_R2_MIXER, + RT5651_M_BST2_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5651_REC_R2_MIXER, + RT5651_M_BST1_RM_R_SFT, 1, 1), +}; + +/* Analog Output Mixer */ + +static const struct snd_kcontrol_new rt5651_out_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", RT5651_OUT_L3_MIXER, + RT5651_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST2 Switch", RT5651_OUT_L3_MIXER, + RT5651_M_BST2_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL1 Switch", RT5651_OUT_L3_MIXER, + RT5651_M_IN1_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXL Switch", RT5651_OUT_L3_MIXER, + RT5651_M_RM_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5651_OUT_L3_MIXER, + RT5651_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5651_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5651_OUT_R3_MIXER, + RT5651_M_BST2_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5651_OUT_R3_MIXER, + RT5651_M_BST1_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR1 Switch", RT5651_OUT_R3_MIXER, + RT5651_M_IN1_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXR Switch", RT5651_OUT_R3_MIXER, + RT5651_M_RM_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5651_OUT_R3_MIXER, + RT5651_M_DAC_R1_OM_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5651_hpo_mix[] = { + SOC_DAPM_SINGLE("HPO MIX DAC1 Switch", RT5651_HPO_MIXER, + RT5651_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPO MIX HPVOL Switch", RT5651_HPO_MIXER, + RT5651_M_HPVOL_HM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5651_lout_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5651_LOUT_MIXER, + RT5651_M_DAC_L1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5651_LOUT_MIXER, + RT5651_M_DAC_R1_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTVOL L Switch", RT5651_LOUT_MIXER, + RT5651_M_OV_L_LM_SFT, 1, 1), + SOC_DAPM_SINGLE("OUTVOL R Switch", RT5651_LOUT_MIXER, + RT5651_M_OV_R_LM_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new outvol_l_control = + SOC_DAPM_SINGLE("Switch", RT5651_LOUT_CTRL1, + RT5651_VOL_L_SFT, 1, 1); + +static const struct snd_kcontrol_new outvol_r_control = + SOC_DAPM_SINGLE("Switch", RT5651_LOUT_CTRL1, + RT5651_VOL_R_SFT, 1, 1); + +static const struct snd_kcontrol_new lout_l_mute_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5651_LOUT_CTRL1, + RT5651_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new lout_r_mute_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5651_LOUT_CTRL1, + RT5651_R_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hpovol_l_control = + SOC_DAPM_SINGLE("Switch", RT5651_HP_VOL, + RT5651_VOL_L_SFT, 1, 1); + +static const struct snd_kcontrol_new hpovol_r_control = + SOC_DAPM_SINGLE("Switch", RT5651_HP_VOL, + RT5651_VOL_R_SFT, 1, 1); + +static const struct snd_kcontrol_new hpo_l_mute_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5651_HP_VOL, + RT5651_L_MUTE_SFT, 1, 1); + +static const struct snd_kcontrol_new hpo_r_mute_control = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5651_HP_VOL, + RT5651_R_MUTE_SFT, 1, 1); + +/* INL/R source */ +static const char * const rt5651_inl_src[] = {"IN2P", "HPOVOLLP"}; + +static SOC_ENUM_SINGLE_DECL( + rt5651_inl_enum, RT5651_INL1_INR1_VOL, + RT5651_INL_SEL_SFT, rt5651_inl_src); + +static const struct snd_kcontrol_new rt5651_inl1_mux = + SOC_DAPM_ENUM("INL1 source", rt5651_inl_enum); + +static const char * const rt5651_inr1_src[] = {"IN2N", "HPOVOLRP"}; + +static SOC_ENUM_SINGLE_DECL( + rt5651_inr1_enum, RT5651_INL1_INR1_VOL, + RT5651_INR_SEL_SFT, rt5651_inr1_src); + +static const struct snd_kcontrol_new rt5651_inr1_mux = + SOC_DAPM_ENUM("INR1 source", rt5651_inr1_enum); + +static const char * const rt5651_inl2_src[] = {"IN3P", "OUTVOLLP"}; + +static SOC_ENUM_SINGLE_DECL( + rt5651_inl2_enum, RT5651_INL2_INR2_VOL, + RT5651_INL_SEL_SFT, rt5651_inl2_src); + +static const struct snd_kcontrol_new rt5651_inl2_mux = + SOC_DAPM_ENUM("INL2 source", rt5651_inl2_enum); + +static const char * const rt5651_inr2_src[] = {"IN3N", "OUTVOLRP"}; + +static SOC_ENUM_SINGLE_DECL( + rt5651_inr2_enum, RT5651_INL2_INR2_VOL, + RT5651_INR_SEL_SFT, rt5651_inr2_src); + +static const struct snd_kcontrol_new rt5651_inr2_mux = + SOC_DAPM_ENUM("INR2 source", rt5651_inr2_enum); + + +/* Stereo ADC source */ +static const char * const rt5651_stereo1_adc1_src[] = {"DD MIX", "ADC"}; + +static SOC_ENUM_SINGLE_DECL( + rt5651_stereo1_adc1_enum, RT5651_STO1_ADC_MIXER, + RT5651_STO1_ADC_1_SRC_SFT, rt5651_stereo1_adc1_src); + +static const struct snd_kcontrol_new rt5651_sto1_adc_l1_mux = + SOC_DAPM_ENUM("Stereo1 ADC L1 source", rt5651_stereo1_adc1_enum); + +static const struct snd_kcontrol_new rt5651_sto1_adc_r1_mux = + SOC_DAPM_ENUM("Stereo1 ADC R1 source", rt5651_stereo1_adc1_enum); + +static const char * const rt5651_stereo1_adc2_src[] = {"DMIC", "DD MIX"}; + +static SOC_ENUM_SINGLE_DECL( + rt5651_stereo1_adc2_enum, RT5651_STO1_ADC_MIXER, + RT5651_STO1_ADC_2_SRC_SFT, rt5651_stereo1_adc2_src); + +static const struct snd_kcontrol_new rt5651_sto1_adc_l2_mux = + SOC_DAPM_ENUM("Stereo1 ADC L2 source", rt5651_stereo1_adc2_enum); + +static const struct snd_kcontrol_new rt5651_sto1_adc_r2_mux = + SOC_DAPM_ENUM("Stereo1 ADC R2 source", rt5651_stereo1_adc2_enum); + +/* Mono ADC source */ +static const char * const rt5651_sto2_adc_l1_src[] = {"DD MIXL", "ADCL"}; + +static SOC_ENUM_SINGLE_DECL( + rt5651_sto2_adc_l1_enum, RT5651_STO1_ADC_MIXER, + RT5651_STO2_ADC_L1_SRC_SFT, rt5651_sto2_adc_l1_src); + +static const struct snd_kcontrol_new rt5651_sto2_adc_l1_mux = + SOC_DAPM_ENUM("Stereo2 ADC1 left source", rt5651_sto2_adc_l1_enum); + +static const char * const rt5651_sto2_adc_l2_src[] = {"DMIC L", "DD MIXL"}; + +static SOC_ENUM_SINGLE_DECL( + rt5651_sto2_adc_l2_enum, RT5651_STO1_ADC_MIXER, + RT5651_STO2_ADC_L2_SRC_SFT, rt5651_sto2_adc_l2_src); + +static const struct snd_kcontrol_new rt5651_sto2_adc_l2_mux = + SOC_DAPM_ENUM("Stereo2 ADC2 left source", rt5651_sto2_adc_l2_enum); + +static const char * const rt5651_sto2_adc_r1_src[] = {"DD MIXR", "ADCR"}; + +static SOC_ENUM_SINGLE_DECL( + rt5651_sto2_adc_r1_enum, RT5651_STO1_ADC_MIXER, + RT5651_STO2_ADC_R1_SRC_SFT, rt5651_sto2_adc_r1_src); + +static const struct snd_kcontrol_new rt5651_sto2_adc_r1_mux = + SOC_DAPM_ENUM("Stereo2 ADC1 right source", rt5651_sto2_adc_r1_enum); + +static const char * const rt5651_sto2_adc_r2_src[] = {"DMIC R", "DD MIXR"}; + +static SOC_ENUM_SINGLE_DECL( + rt5651_sto2_adc_r2_enum, RT5651_STO1_ADC_MIXER, + RT5651_STO2_ADC_R2_SRC_SFT, rt5651_sto2_adc_r2_src); + +static const struct snd_kcontrol_new rt5651_sto2_adc_r2_mux = + SOC_DAPM_ENUM("Stereo2 ADC2 right source", rt5651_sto2_adc_r2_enum); + +/* DAC2 channel source */ + +static const char * const rt5651_dac_src[] = {"IF1", "IF2"}; + +static SOC_ENUM_SINGLE_DECL(rt5651_dac_l2_enum, RT5651_DAC2_CTRL, + RT5651_SEL_DAC_L2_SFT, rt5651_dac_src); + +static const struct snd_kcontrol_new rt5651_dac_l2_mux = + SOC_DAPM_ENUM("DAC2 left channel source", rt5651_dac_l2_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5651_dac_r2_enum, RT5651_DAC2_CTRL, + RT5651_SEL_DAC_R2_SFT, rt5651_dac_src); + +static const struct snd_kcontrol_new rt5651_dac_r2_mux = + SOC_DAPM_ENUM("DAC2 right channel source", rt5651_dac_r2_enum); + +/* IF2_ADC channel source */ + +static const char * const rt5651_adc_src[] = {"IF1 ADC1", "IF1 ADC2"}; + +static SOC_ENUM_SINGLE_DECL(rt5651_if2_adc_src_enum, RT5651_DIG_INF_DATA, + RT5651_IF2_ADC_SRC_SFT, rt5651_adc_src); + +static const struct snd_kcontrol_new rt5651_if2_adc_src_mux = + SOC_DAPM_ENUM("IF2 ADC channel source", rt5651_if2_adc_src_enum); + +/* PDM select */ +static const char * const rt5651_pdm_sel[] = {"DD MIX", "Stereo DAC MIX"}; + +static SOC_ENUM_SINGLE_DECL( + rt5651_pdm_l_sel_enum, RT5651_PDM_CTL, + RT5651_PDM_L_SEL_SFT, rt5651_pdm_sel); + +static SOC_ENUM_SINGLE_DECL( + rt5651_pdm_r_sel_enum, RT5651_PDM_CTL, + RT5651_PDM_R_SEL_SFT, rt5651_pdm_sel); + +static const struct snd_kcontrol_new rt5651_pdm_l_mux = + SOC_DAPM_ENUM("PDM L select", rt5651_pdm_l_sel_enum); + +static const struct snd_kcontrol_new rt5651_pdm_r_mux = + SOC_DAPM_ENUM("PDM R select", rt5651_pdm_r_sel_enum); + +static int rt5651_amp_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* depop parameters */ + regmap_update_bits(rt5651->regmap, RT5651_PR_BASE + + RT5651_CHPUMP_INT_REG1, 0x0700, 0x0200); + regmap_update_bits(rt5651->regmap, RT5651_DEPOP_M2, + RT5651_DEPOP_MASK, RT5651_DEPOP_MAN); + regmap_update_bits(rt5651->regmap, RT5651_DEPOP_M1, + RT5651_HP_CP_MASK | RT5651_HP_SG_MASK | + RT5651_HP_CB_MASK, RT5651_HP_CP_PU | + RT5651_HP_SG_DIS | RT5651_HP_CB_PU); + regmap_write(rt5651->regmap, RT5651_PR_BASE + + RT5651_HP_DCC_INT1, 0x9f00); + /* headphone amp power on */ + regmap_update_bits(rt5651->regmap, RT5651_PWR_ANLG1, + RT5651_PWR_FV1 | RT5651_PWR_FV2, 0); + regmap_update_bits(rt5651->regmap, RT5651_PWR_ANLG1, + RT5651_PWR_HA, + RT5651_PWR_HA); + usleep_range(10000, 15000); + regmap_update_bits(rt5651->regmap, RT5651_PWR_ANLG1, + RT5651_PWR_FV1 | RT5651_PWR_FV2 , + RT5651_PWR_FV1 | RT5651_PWR_FV2); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5651_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* headphone unmute sequence */ + regmap_update_bits(rt5651->regmap, RT5651_DEPOP_M2, + RT5651_DEPOP_MASK | RT5651_DIG_DP_MASK, + RT5651_DEPOP_AUTO | RT5651_DIG_DP_EN); + regmap_update_bits(rt5651->regmap, RT5651_CHARGE_PUMP, + RT5651_PM_HP_MASK, RT5651_PM_HP_HV); + + regmap_update_bits(rt5651->regmap, RT5651_DEPOP_M3, + RT5651_CP_FQ1_MASK | RT5651_CP_FQ2_MASK | + RT5651_CP_FQ3_MASK, + (RT5651_CP_FQ_192_KHZ << RT5651_CP_FQ1_SFT) | + (RT5651_CP_FQ_12_KHZ << RT5651_CP_FQ2_SFT) | + (RT5651_CP_FQ_192_KHZ << RT5651_CP_FQ3_SFT)); + + regmap_write(rt5651->regmap, RT5651_PR_BASE + + RT5651_MAMP_INT_REG2, 0x1c00); + regmap_update_bits(rt5651->regmap, RT5651_DEPOP_M1, + RT5651_HP_CP_MASK | RT5651_HP_SG_MASK, + RT5651_HP_CP_PD | RT5651_HP_SG_EN); + regmap_update_bits(rt5651->regmap, RT5651_PR_BASE + + RT5651_CHPUMP_INT_REG1, 0x0700, 0x0400); + rt5651->hp_mute = 0; + break; + + case SND_SOC_DAPM_PRE_PMD: + rt5651->hp_mute = 1; + usleep_range(70000, 75000); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5651_hp_post_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (!rt5651->hp_mute) + usleep_range(80000, 85000); + + break; + + default: + return 0; + } + + return 0; +} + +static int rt5651_bst1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5651_PWR_ANLG2, + RT5651_PWR_BST1_OP2, RT5651_PWR_BST1_OP2); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5651_PWR_ANLG2, + RT5651_PWR_BST1_OP2, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5651_bst2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5651_PWR_ANLG2, + RT5651_PWR_BST2_OP2, RT5651_PWR_BST2_OP2); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5651_PWR_ANLG2, + RT5651_PWR_BST2_OP2, 0); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5651_bst3_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, RT5651_PWR_ANLG2, + RT5651_PWR_BST3_OP2, RT5651_PWR_BST3_OP2); + break; + + case SND_SOC_DAPM_PRE_PMD: + snd_soc_update_bits(codec, RT5651_PWR_ANLG2, + RT5651_PWR_BST3_OP2, 0); + break; + + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt5651_dapm_widgets[] = { + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5651_PLL_MODE_2, + 15, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("I2S2 ASRC", 1, RT5651_PLL_MODE_2, + 14, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("STO1 DAC ASRC", 1, RT5651_PLL_MODE_2, + 13, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("STO2 DAC ASRC", 1, RT5651_PLL_MODE_2, + 12, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC ASRC", 1, RT5651_PLL_MODE_2, + 11, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("PLL1", RT5651_PWR_ANLG2, + RT5651_PWR_PLL_BIT, 0, NULL, 0), + /* Input Side */ + /* micbias */ + SND_SOC_DAPM_SUPPLY("LDO", RT5651_PWR_ANLG1, + RT5651_PWR_LDO_BIT, 0, NULL, 0), + SND_SOC_DAPM_MICBIAS("micbias1", RT5651_PWR_ANLG2, + RT5651_PWR_MB1_BIT, 0), + /* Input Lines */ + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + + SND_SOC_DAPM_INPUT("IN1P"), + SND_SOC_DAPM_INPUT("IN2P"), + SND_SOC_DAPM_INPUT("IN2N"), + SND_SOC_DAPM_INPUT("IN3P"), + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + SND_SOC_DAPM_SUPPLY("DMIC CLK", RT5651_DMIC, RT5651_DMIC_1_EN_SFT, + 0, set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + /* Boost */ + SND_SOC_DAPM_PGA_E("BST1", RT5651_PWR_ANLG2, + RT5651_PWR_BST1_BIT, 0, NULL, 0, rt5651_bst1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("BST2", RT5651_PWR_ANLG2, + RT5651_PWR_BST2_BIT, 0, NULL, 0, rt5651_bst2_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("BST3", RT5651_PWR_ANLG2, + RT5651_PWR_BST3_BIT, 0, NULL, 0, rt5651_bst3_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + /* Input Volume */ + SND_SOC_DAPM_PGA("INL1 VOL", RT5651_PWR_VOL, + RT5651_PWR_IN1_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR1 VOL", RT5651_PWR_VOL, + RT5651_PWR_IN1_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INL2 VOL", RT5651_PWR_VOL, + RT5651_PWR_IN2_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR2 VOL", RT5651_PWR_VOL, + RT5651_PWR_IN2_R_BIT, 0, NULL, 0), + /* IN Mux */ + SND_SOC_DAPM_MUX("INL1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl1_mux), + SND_SOC_DAPM_MUX("INR1 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr1_mux), + SND_SOC_DAPM_MUX("INL2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inl2_mux), + SND_SOC_DAPM_MUX("INR2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_inr2_mux), + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIXL", RT5651_PWR_MIXER, RT5651_PWR_RM_L_BIT, 0, + rt5651_rec_l_mix, ARRAY_SIZE(rt5651_rec_l_mix)), + SND_SOC_DAPM_MIXER("RECMIXR", RT5651_PWR_MIXER, RT5651_PWR_RM_R_BIT, 0, + rt5651_rec_r_mix, ARRAY_SIZE(rt5651_rec_r_mix)), + /* ADCs */ + SND_SOC_DAPM_ADC("ADC L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC R", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC L Power", RT5651_PWR_DIG1, + RT5651_PWR_ADC_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC R Power", RT5651_PWR_DIG1, + RT5651_PWR_ADC_R_BIT, 0, NULL, 0), + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5651_sto1_adc_l2_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5651_sto1_adc_r2_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5651_sto1_adc_l1_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5651_sto1_adc_r1_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5651_sto2_adc_l2_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5651_sto2_adc_l1_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5651_sto2_adc_r1_mux), + SND_SOC_DAPM_MUX("Stereo2 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5651_sto2_adc_r2_mux), + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("Stereo1 Filter", RT5651_PWR_DIG2, + RT5651_PWR_ADC_STO1_F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Stereo2 Filter", RT5651_PWR_DIG2, + RT5651_PWR_ADC_STO2_F_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5651_sto1_adc_l_mix, + ARRAY_SIZE(rt5651_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5651_sto1_adc_r_mix, + ARRAY_SIZE(rt5651_sto1_adc_r_mix)), + SND_SOC_DAPM_MIXER("Stereo2 ADC MIXL", SND_SOC_NOPM, 0, 0, + rt5651_sto2_adc_l_mix, + ARRAY_SIZE(rt5651_sto2_adc_l_mix)), + SND_SOC_DAPM_MIXER("Stereo2 ADC MIXR", SND_SOC_NOPM, 0, 0, + rt5651_sto2_adc_r_mix, + ARRAY_SIZE(rt5651_sto2_adc_r_mix)), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5651_PWR_DIG1, + RT5651_PWR_I2S1_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC2 R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 ADC2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S2", RT5651_PWR_DIG1, + RT5651_PWR_I2S2_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF2 DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MUX("IF2 ADC", SND_SOC_NOPM, 0, 0, + &rt5651_if2_adc_src_mux), + + /* Digital Interface Select */ + + SND_SOC_DAPM_MUX("PDM L Mux", RT5651_PDM_CTL, + RT5651_M_PDM_L_SFT, 1, &rt5651_pdm_l_mux), + SND_SOC_DAPM_MUX("PDM R Mux", RT5651_PDM_CTL, + RT5651_M_PDM_R_SFT, 1, &rt5651_pdm_r_mux), + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + /* Audio DSP */ + SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5651_dac_l_mix, ARRAY_SIZE(rt5651_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5651_dac_r_mix, ARRAY_SIZE(rt5651_dac_r_mix)), + + /* DAC2 channel Mux */ + SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_dac_l2_mux), + SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, &rt5651_dac_r2_mux), + SND_SOC_DAPM_PGA("DAC L2 Volume", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DAC R2 Volume", SND_SOC_NOPM, 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("Stero1 DAC Power", RT5651_PWR_DIG2, + RT5651_PWR_DAC_STO1_F_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Stero2 DAC Power", RT5651_PWR_DIG2, + RT5651_PWR_DAC_STO2_F_BIT, 0, NULL, 0), + /* DAC Mixer */ + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5651_sto_dac_l_mix, + ARRAY_SIZE(rt5651_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5651_sto_dac_r_mix, + ARRAY_SIZE(rt5651_sto_dac_r_mix)), + SND_SOC_DAPM_MIXER("DD MIXL", SND_SOC_NOPM, 0, 0, + rt5651_dd_dac_l_mix, + ARRAY_SIZE(rt5651_dd_dac_l_mix)), + SND_SOC_DAPM_MIXER("DD MIXR", SND_SOC_NOPM, 0, 0, + rt5651_dd_dac_r_mix, + ARRAY_SIZE(rt5651_dd_dac_r_mix)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("DAC L1 Power", RT5651_PWR_DIG1, + RT5651_PWR_DAC_L1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R1 Power", RT5651_PWR_DIG1, + RT5651_PWR_DAC_R1_BIT, 0, NULL, 0), + /* OUT Mixer */ + SND_SOC_DAPM_MIXER("OUT MIXL", RT5651_PWR_MIXER, RT5651_PWR_OM_L_BIT, + 0, rt5651_out_l_mix, ARRAY_SIZE(rt5651_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5651_PWR_MIXER, RT5651_PWR_OM_R_BIT, + 0, rt5651_out_r_mix, ARRAY_SIZE(rt5651_out_r_mix)), + /* Ouput Volume */ + SND_SOC_DAPM_SWITCH("OUTVOL L", RT5651_PWR_VOL, + RT5651_PWR_OV_L_BIT, 0, &outvol_l_control), + SND_SOC_DAPM_SWITCH("OUTVOL R", RT5651_PWR_VOL, + RT5651_PWR_OV_R_BIT, 0, &outvol_r_control), + SND_SOC_DAPM_SWITCH("HPOVOL L", RT5651_PWR_VOL, + RT5651_PWR_HV_L_BIT, 0, &hpovol_l_control), + SND_SOC_DAPM_SWITCH("HPOVOL R", RT5651_PWR_VOL, + RT5651_PWR_HV_R_BIT, 0, &hpovol_r_control), + SND_SOC_DAPM_PGA("INL1", RT5651_PWR_VOL, + RT5651_PWR_IN1_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR1", RT5651_PWR_VOL, + RT5651_PWR_IN1_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INL2", RT5651_PWR_VOL, + RT5651_PWR_IN2_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("INR2", RT5651_PWR_VOL, + RT5651_PWR_IN2_R_BIT, 0, NULL, 0), + /* HPO/LOUT/Mono Mixer */ + SND_SOC_DAPM_MIXER("HPOL MIX", SND_SOC_NOPM, 0, 0, + rt5651_hpo_mix, ARRAY_SIZE(rt5651_hpo_mix)), + SND_SOC_DAPM_MIXER("HPOR MIX", SND_SOC_NOPM, 0, 0, + rt5651_hpo_mix, ARRAY_SIZE(rt5651_hpo_mix)), + SND_SOC_DAPM_SUPPLY("HP L Amp", RT5651_PWR_ANLG1, + RT5651_PWR_HP_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP R Amp", RT5651_PWR_ANLG1, + RT5651_PWR_HP_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_MIXER("LOUT MIX", RT5651_PWR_ANLG1, RT5651_PWR_LM_BIT, 0, + rt5651_lout_mix, ARRAY_SIZE(rt5651_lout_mix)), + + SND_SOC_DAPM_SUPPLY("Amp Power", RT5651_PWR_ANLG1, + RT5651_PWR_HA_BIT, 0, rt5651_amp_power_event, + SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5651_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SWITCH("HPO L Playback", SND_SOC_NOPM, 0, 0, + &hpo_l_mute_control), + SND_SOC_DAPM_SWITCH("HPO R Playback", SND_SOC_NOPM, 0, 0, + &hpo_r_mute_control), + SND_SOC_DAPM_SWITCH("LOUT L Playback", SND_SOC_NOPM, 0, 0, + &lout_l_mute_control), + SND_SOC_DAPM_SWITCH("LOUT R Playback", SND_SOC_NOPM, 0, 0, + &lout_r_mute_control), + SND_SOC_DAPM_POST("HP Post", rt5651_hp_post_event), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + SND_SOC_DAPM_OUTPUT("LOUTL"), + SND_SOC_DAPM_OUTPUT("LOUTR"), + SND_SOC_DAPM_OUTPUT("PDML"), + SND_SOC_DAPM_OUTPUT("PDMR"), +}; + +static const struct snd_soc_dapm_route rt5651_dapm_routes[] = { + {"Stero1 DAC Power", NULL, "STO1 DAC ASRC"}, + {"Stero2 DAC Power", NULL, "STO2 DAC ASRC"}, + {"I2S1", NULL, "I2S1 ASRC"}, + {"I2S2", NULL, "I2S2 ASRC"}, + + {"IN1P", NULL, "LDO"}, + {"IN2P", NULL, "LDO"}, + {"IN3P", NULL, "LDO"}, + + {"IN1P", NULL, "MIC1"}, + {"IN2P", NULL, "MIC2"}, + {"IN2N", NULL, "MIC2"}, + {"IN3P", NULL, "MIC3"}, + + {"BST1", NULL, "IN1P"}, + {"BST2", NULL, "IN2P"}, + {"BST2", NULL, "IN2N"}, + {"BST3", NULL, "IN3P"}, + + {"INL1 VOL", NULL, "IN2P"}, + {"INR1 VOL", NULL, "IN2N"}, + + {"RECMIXL", "INL1 Switch", "INL1 VOL"}, + {"RECMIXL", "BST3 Switch", "BST3"}, + {"RECMIXL", "BST2 Switch", "BST2"}, + {"RECMIXL", "BST1 Switch", "BST1"}, + + {"RECMIXR", "INR1 Switch", "INR1 VOL"}, + {"RECMIXR", "BST3 Switch", "BST3"}, + {"RECMIXR", "BST2 Switch", "BST2"}, + {"RECMIXR", "BST1 Switch", "BST1"}, + + {"ADC L", NULL, "RECMIXL"}, + {"ADC L", NULL, "ADC L Power"}, + {"ADC R", NULL, "RECMIXR"}, + {"ADC R", NULL, "ADC R Power"}, + + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC CLK"}, + + {"Stereo1 ADC L2 Mux", "DMIC", "DMIC L1"}, + {"Stereo1 ADC L2 Mux", "DD MIX", "DD MIXL"}, + {"Stereo1 ADC L1 Mux", "ADC", "ADC L"}, + {"Stereo1 ADC L1 Mux", "DD MIX", "DD MIXL"}, + + {"Stereo1 ADC R1 Mux", "ADC", "ADC R"}, + {"Stereo1 ADC R1 Mux", "DD MIX", "DD MIXR"}, + {"Stereo1 ADC R2 Mux", "DMIC", "DMIC R1"}, + {"Stereo1 ADC R2 Mux", "DD MIX", "DD MIXR"}, + + {"Stereo2 ADC L2 Mux", "DMIC L", "DMIC L1"}, + {"Stereo2 ADC L2 Mux", "DD MIXL", "DD MIXL"}, + {"Stereo2 ADC L1 Mux", "DD MIXL", "DD MIXL"}, + {"Stereo2 ADC L1 Mux", "ADCL", "ADC L"}, + + {"Stereo2 ADC R1 Mux", "DD MIXR", "DD MIXR"}, + {"Stereo2 ADC R1 Mux", "ADCR", "ADC R"}, + {"Stereo2 ADC R2 Mux", "DMIC R", "DMIC R1"}, + {"Stereo2 ADC R2 Mux", "DD MIXR", "DD MIXR"}, + + {"Stereo1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux"}, + {"Stereo1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux"}, + {"Stereo1 ADC MIXL", NULL, "Stereo1 Filter"}, + {"Stereo1 Filter", NULL, "PLL1", is_sysclk_from_pll}, + {"Stereo1 Filter", NULL, "ADC ASRC"}, + + {"Stereo1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux"}, + {"Stereo1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux"}, + {"Stereo1 ADC MIXR", NULL, "Stereo1 Filter"}, + + {"Stereo2 ADC MIXL", "ADC1 Switch", "Stereo2 ADC L1 Mux"}, + {"Stereo2 ADC MIXL", "ADC2 Switch", "Stereo2 ADC L2 Mux"}, + {"Stereo2 ADC MIXL", NULL, "Stereo2 Filter"}, + {"Stereo2 Filter", NULL, "PLL1", is_sysclk_from_pll}, + {"Stereo2 Filter", NULL, "ADC ASRC"}, + + {"Stereo2 ADC MIXR", "ADC1 Switch", "Stereo2 ADC R1 Mux"}, + {"Stereo2 ADC MIXR", "ADC2 Switch", "Stereo2 ADC R2 Mux"}, + {"Stereo2 ADC MIXR", NULL, "Stereo2 Filter"}, + + {"IF1 ADC2", NULL, "Stereo2 ADC MIXL"}, + {"IF1 ADC2", NULL, "Stereo2 ADC MIXR"}, + {"IF1 ADC1", NULL, "Stereo1 ADC MIXL"}, + {"IF1 ADC1", NULL, "Stereo1 ADC MIXR"}, + + {"IF1 ADC1", NULL, "I2S1"}, + + {"IF2 ADC", "IF1 ADC1", "IF1 ADC1"}, + {"IF2 ADC", "IF1 ADC2", "IF1 ADC2"}, + {"IF2 ADC", NULL, "I2S2"}, + + {"AIF1TX", NULL, "IF1 ADC1"}, + {"AIF1TX", NULL, "IF1 ADC2"}, + {"AIF2TX", NULL, "IF2 ADC"}, + + {"IF1 DAC", NULL, "AIF1RX"}, + {"IF1 DAC", NULL, "I2S1"}, + {"IF2 DAC", NULL, "AIF2RX"}, + {"IF2 DAC", NULL, "I2S2"}, + + {"IF1 DAC1 L", NULL, "IF1 DAC"}, + {"IF1 DAC1 R", NULL, "IF1 DAC"}, + {"IF1 DAC2 L", NULL, "IF1 DAC"}, + {"IF1 DAC2 R", NULL, "IF1 DAC"}, + {"IF2 DAC L", NULL, "IF2 DAC"}, + {"IF2 DAC R", NULL, "IF2 DAC"}, + + {"DAC MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL"}, + {"DAC MIXL", "INF1 Switch", "IF1 DAC1 L"}, + {"DAC MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR"}, + {"DAC MIXR", "INF1 Switch", "IF1 DAC1 R"}, + + {"Audio DSP", NULL, "DAC MIXL"}, + {"Audio DSP", NULL, "DAC MIXR"}, + + {"DAC L2 Mux", "IF1", "IF1 DAC2 L"}, + {"DAC L2 Mux", "IF2", "IF2 DAC L"}, + {"DAC L2 Volume", NULL, "DAC L2 Mux"}, + + {"DAC R2 Mux", "IF1", "IF1 DAC2 R"}, + {"DAC R2 Mux", "IF2", "IF2 DAC R"}, + {"DAC R2 Volume", NULL, "DAC R2 Mux"}, + + {"Stereo DAC MIXL", "DAC L1 Switch", "Audio DSP"}, + {"Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Volume"}, + {"Stereo DAC MIXL", "DAC R1 Switch", "DAC MIXR"}, + {"Stereo DAC MIXL", NULL, "Stero1 DAC Power"}, + {"Stereo DAC MIXL", NULL, "Stero2 DAC Power"}, + {"Stereo DAC MIXR", "DAC R1 Switch", "Audio DSP"}, + {"Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Volume"}, + {"Stereo DAC MIXR", "DAC L1 Switch", "DAC MIXL"}, + {"Stereo DAC MIXR", NULL, "Stero1 DAC Power"}, + {"Stereo DAC MIXR", NULL, "Stero2 DAC Power"}, + + {"PDM L Mux", "Stereo DAC MIX", "Stereo DAC MIXL"}, + {"PDM L Mux", "DD MIX", "DAC MIXL"}, + {"PDM R Mux", "Stereo DAC MIX", "Stereo DAC MIXR"}, + {"PDM R Mux", "DD MIX", "DAC MIXR"}, + + {"DAC L1", NULL, "Stereo DAC MIXL"}, + {"DAC L1", NULL, "PLL1", is_sysclk_from_pll}, + {"DAC L1", NULL, "DAC L1 Power"}, + {"DAC R1", NULL, "Stereo DAC MIXR"}, + {"DAC R1", NULL, "PLL1", is_sysclk_from_pll}, + {"DAC R1", NULL, "DAC R1 Power"}, + + {"DD MIXL", "DAC L1 Switch", "DAC MIXL"}, + {"DD MIXL", "DAC L2 Switch", "DAC L2 Volume"}, + {"DD MIXL", "DAC R2 Switch", "DAC R2 Volume"}, + {"DD MIXL", NULL, "Stero2 DAC Power"}, + + {"DD MIXR", "DAC R1 Switch", "DAC MIXR"}, + {"DD MIXR", "DAC R2 Switch", "DAC R2 Volume"}, + {"DD MIXR", "DAC L2 Switch", "DAC L2 Volume"}, + {"DD MIXR", NULL, "Stero2 DAC Power"}, + + {"OUT MIXL", "BST1 Switch", "BST1"}, + {"OUT MIXL", "BST2 Switch", "BST2"}, + {"OUT MIXL", "INL1 Switch", "INL1 VOL"}, + {"OUT MIXL", "REC MIXL Switch", "RECMIXL"}, + {"OUT MIXL", "DAC L1 Switch", "DAC L1"}, + + {"OUT MIXR", "BST2 Switch", "BST2"}, + {"OUT MIXR", "BST1 Switch", "BST1"}, + {"OUT MIXR", "INR1 Switch", "INR1 VOL"}, + {"OUT MIXR", "REC MIXR Switch", "RECMIXR"}, + {"OUT MIXR", "DAC R1 Switch", "DAC R1"}, + + {"HPOVOL L", "Switch", "OUT MIXL"}, + {"HPOVOL R", "Switch", "OUT MIXR"}, + {"OUTVOL L", "Switch", "OUT MIXL"}, + {"OUTVOL R", "Switch", "OUT MIXR"}, + + {"HPOL MIX", "HPO MIX DAC1 Switch", "DAC L1"}, + {"HPOL MIX", "HPO MIX HPVOL Switch", "HPOVOL L"}, + {"HPOL MIX", NULL, "HP L Amp"}, + {"HPOR MIX", "HPO MIX DAC1 Switch", "DAC R1"}, + {"HPOR MIX", "HPO MIX HPVOL Switch", "HPOVOL R"}, + {"HPOR MIX", NULL, "HP R Amp"}, + + {"LOUT MIX", "DAC L1 Switch", "DAC L1"}, + {"LOUT MIX", "DAC R1 Switch", "DAC R1"}, + {"LOUT MIX", "OUTVOL L Switch", "OUTVOL L"}, + {"LOUT MIX", "OUTVOL R Switch", "OUTVOL R"}, + + {"HP Amp", NULL, "HPOL MIX"}, + {"HP Amp", NULL, "HPOR MIX"}, + {"HP Amp", NULL, "Amp Power"}, + {"HPO L Playback", "Switch", "HP Amp"}, + {"HPO R Playback", "Switch", "HP Amp"}, + {"HPOL", NULL, "HPO L Playback"}, + {"HPOR", NULL, "HPO R Playback"}, + + {"LOUT L Playback", "Switch", "LOUT MIX"}, + {"LOUT R Playback", "Switch", "LOUT MIX"}, + {"LOUTL", NULL, "LOUT L Playback"}, + {"LOUTL", NULL, "Amp Power"}, + {"LOUTR", NULL, "LOUT R Playback"}, + {"LOUTR", NULL, "Amp Power"}, + + {"PDML", NULL, "PDM L Mux"}, + {"PDMR", NULL, "PDM R Mux"}, +}; + +static int get_clk_info(int sclk, int rate) +{ + int i, pd[] = {1, 2, 3, 4, 6, 8, 12, 16}; + + if (sclk <= 0 || rate <= 0) + return -EINVAL; + + rate = rate << 8; + for (i = 0; i < ARRAY_SIZE(pd); i++) + if (sclk == rate * pd[i]) + return i; + + return -EINVAL; +} + +static int rt5651_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0, val_clk, mask_clk; + int pre_div, bclk_ms, frame_size; + + rt5651->lrck[dai->id] = params_rate(params); + pre_div = get_clk_info(rt5651->sysclk, rt5651->lrck[dai->id]); + + if (pre_div < 0) { + dev_err(codec->dev, "Unsupported clock setting\n"); + return -EINVAL; + } + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(codec->dev, "Unsupported frame size: %d\n", frame_size); + return -EINVAL; + } + bclk_ms = frame_size > 32 ? 1 : 0; + rt5651->bclk[dai->id] = rt5651->lrck[dai->id] * (32 << bclk_ms); + + dev_dbg(dai->dev, "bclk is %dHz and lrck is %dHz\n", + rt5651->bclk[dai->id], rt5651->lrck[dai->id]); + dev_dbg(dai->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val_len |= RT5651_I2S_DL_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val_len |= RT5651_I2S_DL_24; + break; + case SNDRV_PCM_FORMAT_S8: + val_len |= RT5651_I2S_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5651_AIF1: + mask_clk = RT5651_I2S_PD1_MASK; + val_clk = pre_div << RT5651_I2S_PD1_SFT; + snd_soc_update_bits(codec, RT5651_I2S1_SDP, + RT5651_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5651_ADDA_CLK1, mask_clk, val_clk); + break; + case RT5651_AIF2: + mask_clk = RT5651_I2S_BCLK_MS2_MASK | RT5651_I2S_PD2_MASK; + val_clk = pre_div << RT5651_I2S_PD2_SFT; + snd_soc_update_bits(codec, RT5651_I2S2_SDP, + RT5651_I2S_DL_MASK, val_len); + snd_soc_update_bits(codec, RT5651_ADDA_CLK1, mask_clk, val_clk); + break; + default: + dev_err(codec->dev, "Wrong dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5651_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5651->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT5651_I2S_MS_S; + rt5651->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5651_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5651_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5651_I2S_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5651_I2S_DF_PCM_B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5651_AIF1: + snd_soc_update_bits(codec, RT5651_I2S1_SDP, + RT5651_I2S_MS_MASK | RT5651_I2S_BP_MASK | + RT5651_I2S_DF_MASK, reg_val); + break; + case RT5651_AIF2: + snd_soc_update_bits(codec, RT5651_I2S2_SDP, + RT5651_I2S_MS_MASK | RT5651_I2S_BP_MASK | + RT5651_I2S_DF_MASK, reg_val); + break; + default: + dev_err(codec->dev, "Wrong dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt5651_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); + unsigned int reg_val = 0; + + if (freq == rt5651->sysclk && clk_id == rt5651->sysclk_src) + return 0; + + switch (clk_id) { + case RT5651_SCLK_S_MCLK: + reg_val |= RT5651_SCLK_SRC_MCLK; + break; + case RT5651_SCLK_S_PLL1: + reg_val |= RT5651_SCLK_SRC_PLL1; + break; + case RT5651_SCLK_S_RCCLK: + reg_val |= RT5651_SCLK_SRC_RCCLK; + break; + default: + dev_err(codec->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_update_bits(codec, RT5651_GLB_CLK, + RT5651_SCLK_SRC_MASK, reg_val); + rt5651->sysclk = freq; + rt5651->sysclk_src = clk_id; + + dev_dbg(dai->dev, "Sysclk is %dHz and clock id is %d\n", freq, clk_id); + + return 0; +} + +/** + * rt5651_pll_calc - Calcualte PLL M/N/K code. + * @freq_in: external clock provided to codec. + * @freq_out: target clock which codec works on. + * @pll_code: Pointer to structure with M, N, K and bypass flag. + * + * Calcualte M/N/K code to configure PLL for codec. And K is assigned to 2 + * which make calculation more efficiently. + * + * Returns 0 for success or negative error code. + */ +static int rt5651_pll_calc(const unsigned int freq_in, + const unsigned int freq_out, struct rt5651_pll_code *pll_code) +{ + int max_n = RT5651_PLL_N_MAX, max_m = RT5651_PLL_M_MAX; + int n = 0, m = 0, red, n_t, m_t, in_t, out_t; + int red_t = abs(freq_out - freq_in); + bool bypass = false; + + if (RT5651_PLL_INP_MAX < freq_in || RT5651_PLL_INP_MIN > freq_in) + return -EINVAL; + + for (n_t = 0; n_t <= max_n; n_t++) { + in_t = (freq_in >> 1) + (freq_in >> 2) * n_t; + if (in_t < 0) + continue; + if (in_t == freq_out) { + bypass = true; + n = n_t; + goto code_find; + } + for (m_t = 0; m_t <= max_m; m_t++) { + out_t = in_t / (m_t + 2); + red = abs(out_t - freq_out); + if (red < red_t) { + n = n_t; + m = m_t; + if (red == 0) + goto code_find; + red_t = red; + } + } + } + pr_debug("Only get approximation about PLL\n"); + +code_find: + pll_code->m_bp = bypass; + pll_code->m_code = m; + pll_code->n_code = n; + pll_code->k_code = 2; + return 0; +} + +static int rt5651_set_dai_pll(struct snd_soc_dai *dai, int pll_id, int source, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); + struct rt5651_pll_code *pll_code = &rt5651->pll_code; + int ret; + + if (source == rt5651->pll_src && freq_in == rt5651->pll_in && + freq_out == rt5651->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(codec->dev, "PLL disabled\n"); + + rt5651->pll_in = 0; + rt5651->pll_out = 0; + snd_soc_update_bits(codec, RT5651_GLB_CLK, + RT5651_SCLK_SRC_MASK, RT5651_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5651_PLL1_S_MCLK: + snd_soc_update_bits(codec, RT5651_GLB_CLK, + RT5651_PLL1_SRC_MASK, RT5651_PLL1_SRC_MCLK); + break; + case RT5651_PLL1_S_BCLK1: + snd_soc_update_bits(codec, RT5651_GLB_CLK, + RT5651_PLL1_SRC_MASK, RT5651_PLL1_SRC_BCLK1); + break; + case RT5651_PLL1_S_BCLK2: + snd_soc_update_bits(codec, RT5651_GLB_CLK, + RT5651_PLL1_SRC_MASK, RT5651_PLL1_SRC_BCLK2); + break; + default: + dev_err(codec->dev, "Unknown PLL source %d\n", source); + return -EINVAL; + } + + ret = rt5651_pll_calc(freq_in, freq_out, pll_code); + if (ret < 0) { + dev_err(codec->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(codec->dev, "bypass=%d m=%d n=%d k=2\n", pll_code->m_bp, + (pll_code->m_bp ? 0 : pll_code->m_code), pll_code->n_code); + + snd_soc_write(codec, RT5651_PLL_CTRL1, + pll_code->n_code << RT5651_PLL_N_SFT | pll_code->k_code); + snd_soc_write(codec, RT5651_PLL_CTRL2, + (pll_code->m_bp ? 0 : pll_code->m_code) << RT5651_PLL_M_SFT | + pll_code->m_bp << RT5651_PLL_M_BP_SFT); + + rt5651->pll_in = freq_in; + rt5651->pll_out = freq_out; + rt5651->pll_src = source; + + return 0; +} + +static int rt5651_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) { + snd_soc_update_bits(codec, RT5651_PWR_ANLG1, + RT5651_PWR_VREF1 | RT5651_PWR_MB | + RT5651_PWR_BG | RT5651_PWR_VREF2, + RT5651_PWR_VREF1 | RT5651_PWR_MB | + RT5651_PWR_BG | RT5651_PWR_VREF2); + usleep_range(10000, 15000); + snd_soc_update_bits(codec, RT5651_PWR_ANLG1, + RT5651_PWR_FV1 | RT5651_PWR_FV2, + RT5651_PWR_FV1 | RT5651_PWR_FV2); + snd_soc_update_bits(codec, RT5651_PWR_ANLG1, + RT5651_PWR_LDO_DVO_MASK, + RT5651_PWR_LDO_DVO_1_2V); + snd_soc_update_bits(codec, RT5651_D_MISC, 0x1, 0x1); + if (snd_soc_read(codec, RT5651_PLL_MODE_1) & 0x9200) + snd_soc_update_bits(codec, RT5651_D_MISC, + 0xc00, 0xc00); + } + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_write(codec, RT5651_D_MISC, 0x0010); + snd_soc_write(codec, RT5651_PWR_DIG1, 0x0000); + snd_soc_write(codec, RT5651_PWR_DIG2, 0x0000); + snd_soc_write(codec, RT5651_PWR_VOL, 0x0000); + snd_soc_write(codec, RT5651_PWR_MIXER, 0x0000); + snd_soc_write(codec, RT5651_PWR_ANLG1, 0x0000); + snd_soc_write(codec, RT5651_PWR_ANLG2, 0x0000); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt5651_probe(struct snd_soc_codec *codec) +{ + struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); + + rt5651->codec = codec; + + snd_soc_update_bits(codec, RT5651_PWR_ANLG1, + RT5651_PWR_VREF1 | RT5651_PWR_MB | + RT5651_PWR_BG | RT5651_PWR_VREF2, + RT5651_PWR_VREF1 | RT5651_PWR_MB | + RT5651_PWR_BG | RT5651_PWR_VREF2); + usleep_range(10000, 15000); + snd_soc_update_bits(codec, RT5651_PWR_ANLG1, + RT5651_PWR_FV1 | RT5651_PWR_FV2, + RT5651_PWR_FV1 | RT5651_PWR_FV2); + + rt5651_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +#ifdef CONFIG_PM +static int rt5651_suspend(struct snd_soc_codec *codec) +{ + struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5651->regmap, true); + regcache_mark_dirty(rt5651->regmap); + return 0; +} + +static int rt5651_resume(struct snd_soc_codec *codec) +{ + struct rt5651_priv *rt5651 = snd_soc_codec_get_drvdata(codec); + + regcache_cache_only(rt5651->regmap, false); + snd_soc_cache_sync(codec); + + return 0; +} +#else +#define rt5651_suspend NULL +#define rt5651_resume NULL +#endif + +#define RT5651_STEREO_RATES SNDRV_PCM_RATE_8000_96000 +#define RT5651_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_ops rt5651_aif_dai_ops = { + .hw_params = rt5651_hw_params, + .set_fmt = rt5651_set_dai_fmt, + .set_sysclk = rt5651_set_dai_sysclk, + .set_pll = rt5651_set_dai_pll, +}; + +struct snd_soc_dai_driver rt5651_dai[] = { + { + .name = "rt5651-aif1", + .id = RT5651_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5651_STEREO_RATES, + .formats = RT5651_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5651_STEREO_RATES, + .formats = RT5651_FORMATS, + }, + .ops = &rt5651_aif_dai_ops, + }, + { + .name = "rt5651-aif2", + .id = RT5651_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5651_STEREO_RATES, + .formats = RT5651_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5651_STEREO_RATES, + .formats = RT5651_FORMATS, + }, + .ops = &rt5651_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt5651 = { + .probe = rt5651_probe, + .suspend = rt5651_suspend, + .resume = rt5651_resume, + .set_bias_level = rt5651_set_bias_level, + .idle_bias_off = true, + .controls = rt5651_snd_controls, + .num_controls = ARRAY_SIZE(rt5651_snd_controls), + .dapm_widgets = rt5651_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5651_dapm_widgets), + .dapm_routes = rt5651_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5651_dapm_routes), +}; + +static const struct regmap_config rt5651_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = RT5651_DEVICE_ID + 1 + (ARRAY_SIZE(rt5651_ranges) * + RT5651_PR_SPACING), + .volatile_reg = rt5651_volatile_register, + .readable_reg = rt5651_readable_register, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5651_reg, + .num_reg_defaults = ARRAY_SIZE(rt5651_reg), + .ranges = rt5651_ranges, + .num_ranges = ARRAY_SIZE(rt5651_ranges), +}; + +static const struct i2c_device_id rt5651_i2c_id[] = { + { "rt5651", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5651_i2c_id); + +static int rt5651_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5651_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5651_priv *rt5651; + int ret; + + rt5651 = devm_kzalloc(&i2c->dev, sizeof(*rt5651), + GFP_KERNEL); + if (NULL == rt5651) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5651); + + if (pdata) + rt5651->pdata = *pdata; + + rt5651->regmap = devm_regmap_init_i2c(i2c, &rt5651_regmap); + if (IS_ERR(rt5651->regmap)) { + ret = PTR_ERR(rt5651->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt5651->regmap, RT5651_DEVICE_ID, &ret); + if (ret != RT5651_DEVICE_ID_VALUE) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt5651\n", ret); + return -ENODEV; + } + + regmap_write(rt5651->regmap, RT5651_RESET, 0); + + ret = regmap_register_patch(rt5651->regmap, init_list, + ARRAY_SIZE(init_list)); + if (ret != 0) + dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); + + if (rt5651->pdata.in2_diff) + regmap_update_bits(rt5651->regmap, RT5651_IN1_IN2, + RT5651_IN_DF2, RT5651_IN_DF2); + + if (rt5651->pdata.dmic_en) + regmap_update_bits(rt5651->regmap, RT5651_GPIO_CTRL1, + RT5651_GP2_PIN_MASK, RT5651_GP2_PIN_DMIC1_SCL); + + rt5651->hp_mute = 1; + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5651, + rt5651_dai, ARRAY_SIZE(rt5651_dai)); + + return ret; +} + +static int rt5651_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +struct i2c_driver rt5651_i2c_driver = { + .driver = { + .name = "rt5651", + .owner = THIS_MODULE, + }, + .probe = rt5651_i2c_probe, + .remove = rt5651_i2c_remove, + .id_table = rt5651_i2c_id, +}; +module_i2c_driver(rt5651_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5651 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5651.h b/sound/soc/codecs/rt5651.h new file mode 100644 index 0000000..a28bd0c --- /dev/null +++ b/sound/soc/codecs/rt5651.h @@ -0,0 +1,2081 @@ +/* + * rt5651.h -- RT5651 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu <johnnyhsu(a)realtek.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 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5651_H__ +#define __RT5651_H__ + +#include <sound/rt5651.h> + +/* Info */ +#define RT5651_RESET 0x00 +#define RT5651_VERSION_ID 0xfd +#define RT5651_VENDOR_ID 0xfe +#define RT5651_DEVICE_ID 0xff +/* I/O - Output */ +#define RT5651_HP_VOL 0x02 +#define RT5651_LOUT_CTRL1 0x03 +#define RT5651_LOUT_CTRL2 0x05 +/* I/O - Input */ +#define RT5651_IN1_IN2 0x0d +#define RT5651_IN3 0x0e +#define RT5651_INL1_INR1_VOL 0x0f +#define RT5651_INL2_INR2_VOL 0x10 +/* I/O - ADC/DAC/DMIC */ +#define RT5651_DAC1_DIG_VOL 0x19 +#define RT5651_DAC2_DIG_VOL 0x1a +#define RT5651_DAC2_CTRL 0x1b +#define RT5651_ADC_DIG_VOL 0x1c +#define RT5651_ADC_DATA 0x1d +#define RT5651_ADC_BST_VOL 0x1e +/* Mixer - D-D */ +#define RT5651_STO1_ADC_MIXER 0x27 +#define RT5651_STO2_ADC_MIXER 0x28 +#define RT5651_AD_DA_MIXER 0x29 +#define RT5651_STO_DAC_MIXER 0x2a +#define RT5651_DD_MIXER 0x2b +#define RT5651_DIG_INF_DATA 0x2f +/* PDM */ +#define RT5651_PDM_CTL 0x30 +#define RT5651_PDM_I2C_CTL1 0x31 +#define RT5651_PDM_I2C_CTL2 0x32 +#define RT5651_PDM_I2C_DATA_W 0x33 +#define RT5651_PDM_I2C_DATA_R 0x34 +/* Mixer - ADC */ +#define RT5651_REC_L1_MIXER 0x3b +#define RT5651_REC_L2_MIXER 0x3c +#define RT5651_REC_R1_MIXER 0x3d +#define RT5651_REC_R2_MIXER 0x3e +/* Mixer - DAC */ +#define RT5651_HPO_MIXER 0x45 +#define RT5651_OUT_L1_MIXER 0x4d +#define RT5651_OUT_L2_MIXER 0x4e +#define RT5651_OUT_L3_MIXER 0x4f +#define RT5651_OUT_R1_MIXER 0x50 +#define RT5651_OUT_R2_MIXER 0x51 +#define RT5651_OUT_R3_MIXER 0x52 +#define RT5651_LOUT_MIXER 0x53 +/* Power */ +#define RT5651_PWR_DIG1 0x61 +#define RT5651_PWR_DIG2 0x62 +#define RT5651_PWR_ANLG1 0x63 +#define RT5651_PWR_ANLG2 0x64 +#define RT5651_PWR_MIXER 0x65 +#define RT5651_PWR_VOL 0x66 +/* Private Register Control */ +#define RT5651_PRIV_INDEX 0x6a +#define RT5651_PRIV_DATA 0x6c +/* Format - ADC/DAC */ +#define RT5651_I2S1_SDP 0x70 +#define RT5651_I2S2_SDP 0x71 +#define RT5651_ADDA_CLK1 0x73 +#define RT5651_ADDA_CLK2 0x74 +#define RT5651_DMIC 0x75 +/* TDM Control */ +#define RT5651_TDM_CTL_1 0x77 +#define RT5651_TDM_CTL_2 0x78 +#define RT5651_TDM_CTL_3 0x79 +/* Function - Analog */ +#define RT5651_GLB_CLK 0x80 +#define RT5651_PLL_CTRL1 0x81 +#define RT5651_PLL_CTRL2 0x82 +#define RT5651_PLL_MODE_1 0x83 +#define RT5651_PLL_MODE_2 0x84 +#define RT5651_PLL_MODE_3 0x85 +#define RT5651_PLL_MODE_4 0x86 +#define RT5651_PLL_MODE_5 0x87 +#define RT5651_PLL_MODE_6 0x89 +#define RT5651_PLL_MODE_7 0x8a +#define RT5651_DEPOP_M1 0x8e +#define RT5651_DEPOP_M2 0x8f +#define RT5651_DEPOP_M3 0x90 +#define RT5651_CHARGE_PUMP 0x91 +#define RT5651_MICBIAS 0x93 +#define RT5651_A_JD_CTL1 0x94 +/* Function - Digital */ +#define RT5651_EQ_CTRL1 0xb0 +#define RT5651_EQ_CTRL2 0xb1 +#define RT5651_ALC_1 0xb4 +#define RT5651_ALC_2 0xb5 +#define RT5651_ALC_3 0xb6 +#define RT5651_JD_CTRL1 0xbb +#define RT5651_JD_CTRL2 0xbc +#define RT5651_IRQ_CTRL1 0xbd +#define RT5651_IRQ_CTRL2 0xbe +#define RT5651_INT_IRQ_ST 0xbf +#define RT5651_GPIO_CTRL1 0xc0 +#define RT5651_GPIO_CTRL2 0xc1 +#define RT5651_GPIO_CTRL3 0xc2 +#define RT5651_PGM_REG_ARR1 0xc8 +#define RT5651_PGM_REG_ARR2 0xc9 +#define RT5651_PGM_REG_ARR3 0xca +#define RT5651_PGM_REG_ARR4 0xcb +#define RT5651_PGM_REG_ARR5 0xcc +#define RT5651_SCB_FUNC 0xcd +#define RT5651_SCB_CTRL 0xce +#define RT5651_BASE_BACK 0xcf +#define RT5651_MP3_PLUS1 0xd0 +#define RT5651_MP3_PLUS2 0xd1 +#define RT5651_ADJ_HPF_CTRL1 0xd3 +#define RT5651_ADJ_HPF_CTRL2 0xd4 +#define RT5651_HP_CALIB_AMP_DET 0xd6 +#define RT5651_HP_CALIB2 0xd7 +#define RT5651_SV_ZCD1 0xd9 +#define RT5651_SV_ZCD2 0xda +#define RT5651_D_MISC 0xfa +/* Dummy Register */ +#define RT5651_DUMMY2 0xfb +#define RT5651_DUMMY3 0xfc + + +/* Index of Codec Private Register definition */ +#define RT5651_BIAS_CUR1 0x12 +#define RT5651_BIAS_CUR3 0x14 +#define RT5651_CLSD_INT_REG1 0x1c +#define RT5651_CHPUMP_INT_REG1 0x24 +#define RT5651_MAMP_INT_REG2 0x37 +#define RT5651_CHOP_DAC_ADC 0x3d +#define RT5651_3D_SPK 0x63 +#define RT5651_WND_1 0x6c +#define RT5651_WND_2 0x6d +#define RT5651_WND_3 0x6e +#define RT5651_WND_4 0x6f +#define RT5651_WND_5 0x70 +#define RT5651_WND_8 0x73 +#define RT5651_DIP_SPK_INF 0x75 +#define RT5651_HP_DCC_INT1 0x77 +#define RT5651_EQ_BW_LOP 0xa0 +#define RT5651_EQ_GN_LOP 0xa1 +#define RT5651_EQ_FC_BP1 0xa2 +#define RT5651_EQ_BW_BP1 0xa3 +#define RT5651_EQ_GN_BP1 0xa4 +#define RT5651_EQ_FC_BP2 0xa5 +#define RT5651_EQ_BW_BP2 0xa6 +#define RT5651_EQ_GN_BP2 0xa7 +#define RT5651_EQ_FC_BP3 0xa8 +#define RT5651_EQ_BW_BP3 0xa9 +#define RT5651_EQ_GN_BP3 0xaa +#define RT5651_EQ_FC_BP4 0xab +#define RT5651_EQ_BW_BP4 0xac +#define RT5651_EQ_GN_BP4 0xad +#define RT5651_EQ_FC_HIP1 0xae +#define RT5651_EQ_GN_HIP1 0xaf +#define RT5651_EQ_FC_HIP2 0xb0 +#define RT5651_EQ_BW_HIP2 0xb1 +#define RT5651_EQ_GN_HIP2 0xb2 +#define RT5651_EQ_PRE_VOL 0xb3 +#define RT5651_EQ_PST_VOL 0xb4 + + +/* global definition */ +#define RT5651_L_MUTE (0x1 << 15) +#define RT5651_L_MUTE_SFT 15 +#define RT5651_VOL_L_MUTE (0x1 << 14) +#define RT5651_VOL_L_SFT 14 +#define RT5651_R_MUTE (0x1 << 7) +#define RT5651_R_MUTE_SFT 7 +#define RT5651_VOL_R_MUTE (0x1 << 6) +#define RT5651_VOL_R_SFT 6 +#define RT5651_L_VOL_MASK (0x3f << 8) +#define RT5651_L_VOL_SFT 8 +#define RT5651_R_VOL_MASK (0x3f) +#define RT5651_R_VOL_SFT 0 + +/* LOUT Control 2(0x05) */ +#define RT5651_EN_DFO (0x1 << 15) + +/* IN1 and IN2 Control (0x0d) */ +/* IN3 and IN4 Control (0x0e) */ +#define RT5651_BST_MASK1 (0xf<<12) +#define RT5651_BST_SFT1 12 +#define RT5651_BST_MASK2 (0xf<<8) +#define RT5651_BST_SFT2 8 +#define RT5651_IN_DF1 (0x1 << 7) +#define RT5651_IN_SFT1 7 +#define RT5651_IN_DF2 (0x1 << 6) +#define RT5651_IN_SFT2 6 + +/* INL1 and INR1 Volume Control (0x0f) */ +/* INL2 and INR2 Volume Control (0x10) */ +#define RT5651_INL_SEL_MASK (0x1 << 15) +#define RT5651_INL_SEL_SFT 15 +#define RT5651_INL_SEL_IN4P (0x0 << 15) +#define RT5651_INL_SEL_MONOP (0x1 << 15) +#define RT5651_INL_VOL_MASK (0x1f << 8) +#define RT5651_INL_VOL_SFT 8 +#define RT5651_INR_SEL_MASK (0x1 << 7) +#define RT5651_INR_SEL_SFT 7 +#define RT5651_INR_SEL_IN4N (0x0 << 7) +#define RT5651_INR_SEL_MONON (0x1 << 7) +#define RT5651_INR_VOL_MASK (0x1f) +#define RT5651_INR_VOL_SFT 0 + +/* DAC1 Digital Volume (0x19) */ +#define RT5651_DAC_L1_VOL_MASK (0xff << 8) +#define RT5651_DAC_L1_VOL_SFT 8 +#define RT5651_DAC_R1_VOL_MASK (0xff) +#define RT5651_DAC_R1_VOL_SFT 0 + +/* DAC2 Digital Volume (0x1a) */ +#define RT5651_DAC_L2_VOL_MASK (0xff << 8) +#define RT5651_DAC_L2_VOL_SFT 8 +#define RT5651_DAC_R2_VOL_MASK (0xff) +#define RT5651_DAC_R2_VOL_SFT 0 + +/* DAC2 Control (0x1b) */ +#define RT5651_M_DAC_L2_VOL (0x1 << 13) +#define RT5651_M_DAC_L2_VOL_SFT 13 +#define RT5651_M_DAC_R2_VOL (0x1 << 12) +#define RT5651_M_DAC_R2_VOL_SFT 12 +#define RT5651_SEL_DAC_L2 (0x1 << 11) +#define RT5651_IF2_DAC_L2 (0x1 << 11) +#define RT5651_IF1_DAC_L2 (0x0 << 11) +#define RT5651_SEL_DAC_L2_SFT 11 +#define RT5651_SEL_DAC_R2 (0x1 << 10) +#define RT5651_IF2_DAC_R2 (0x1 << 11) +#define RT5651_IF1_DAC_R2 (0x0 << 11) +#define RT5651_SEL_DAC_R2_SFT 10 + +/* ADC Digital Volume Control (0x1c) */ +#define RT5651_ADC_L_VOL_MASK (0x7f << 8) +#define RT5651_ADC_L_VOL_SFT 8 +#define RT5651_ADC_R_VOL_MASK (0x7f) +#define RT5651_ADC_R_VOL_SFT 0 + +/* Mono ADC Digital Volume Control (0x1d) */ +#define RT5651_M_MONO_ADC_L (0x1 << 15) +#define RT5651_M_MONO_ADC_L_SFT 15 +#define RT5651_MONO_ADC_L_VOL_MASK (0x7f << 8) +#define RT5651_MONO_ADC_L_VOL_SFT 8 +#define RT5651_M_MONO_ADC_R (0x1 << 7) +#define RT5651_M_MONO_ADC_R_SFT 7 +#define RT5651_MONO_ADC_R_VOL_MASK (0x7f) +#define RT5651_MONO_ADC_R_VOL_SFT 0 + +/* ADC Boost Volume Control (0x1e) */ +#define RT5651_ADC_L_BST_MASK (0x3 << 14) +#define RT5651_ADC_L_BST_SFT 14 +#define RT5651_ADC_R_BST_MASK (0x3 << 12) +#define RT5651_ADC_R_BST_SFT 12 +#define RT5651_ADC_COMP_MASK (0x3 << 10) +#define RT5651_ADC_COMP_SFT 10 + +/* Stereo ADC1 Mixer Control (0x27) */ +#define RT5651_M_STO1_ADC_L1 (0x1 << 14) +#define RT5651_M_STO1_ADC_L1_SFT 14 +#define RT5651_M_STO1_ADC_L2 (0x1 << 13) +#define RT5651_M_STO1_ADC_L2_SFT 13 +#define RT5651_STO1_ADC_1_SRC_MASK (0x1 << 12) +#define RT5651_STO1_ADC_1_SRC_SFT 12 +#define RT5651_STO1_ADC_1_SRC_ADC (0x1 << 12) +#define RT5651_STO1_ADC_1_SRC_DACMIX (0x0 << 12) +#define RT5651_STO1_ADC_2_SRC_MASK (0x1 << 11) +#define RT5651_STO1_ADC_2_SRC_SFT 11 +#define RT5651_STO1_ADC_2_SRC_DMIC (0x0 << 11) +#define RT5651_STO1_ADC_2_SRC_DACMIXR (0x1 << 11) +#define RT5651_M_STO1_ADC_R1 (0x1 << 6) +#define RT5651_M_STO1_ADC_R1_SFT 6 +#define RT5651_M_STO1_ADC_R2 (0x1 << 5) +#define RT5651_M_STO1_ADC_R2_SFT 5 + +/* Stereo ADC2 Mixer Control (0x28) */ +#define RT5651_M_STO2_ADC_L1 (0x1 << 14) +#define RT5651_M_STO2_ADC_L1_SFT 14 +#define RT5651_M_STO2_ADC_L2 (0x1 << 13) +#define RT5651_M_STO2_ADC_L2_SFT 13 +#define RT5651_STO2_ADC_L1_SRC_MASK (0x1 << 12) +#define RT5651_STO2_ADC_L1_SRC_SFT 12 +#define RT5651_STO2_ADC_L1_SRC_DACMIXL (0x0 << 12) +#define RT5651_STO2_ADC_L1_SRC_ADCL (0x1 << 12) +#define RT5651_STO2_ADC_L2_SRC_MASK (0x1 << 11) +#define RT5651_STO2_ADC_L2_SRC_SFT 11 +#define RT5651_STO2_ADC_L2_SRC_DMIC (0x0 << 11) +#define RT5651_STO2_ADC_L2_SRC_DACMIXR (0x1 << 11) +#define RT5651_M_STO2_ADC_R1 (0x1 << 6) +#define RT5651_M_STO2_ADC_R1_SFT 6 +#define RT5651_M_STO2_ADC_R2 (0x1 << 5) +#define RT5651_M_STO2_ADC_R2_SFT 5 +#define RT5651_STO2_ADC_R1_SRC_MASK (0x1 << 4) +#define RT5651_STO2_ADC_R1_SRC_SFT 4 +#define RT5651_STO2_ADC_R1_SRC_ADCR (0x1 << 4) +#define RT5651_STO2_ADC_R1_SRC_DACMIXR (0x0 << 4) +#define RT5651_STO2_ADC_R2_SRC_MASK (0x1 << 3) +#define RT5651_STO2_ADC_R2_SRC_SFT 3 +#define RT5651_STO2_ADC_R2_SRC_DMIC (0x0 << 3) +#define RT5651_STO2_ADC_R2_SRC_DACMIXR (0x1 << 3) + +/* ADC Mixer to DAC Mixer Control (0x29) */ +#define RT5651_M_ADCMIX_L (0x1 << 15) +#define RT5651_M_ADCMIX_L_SFT 15 +#define RT5651_M_IF1_DAC_L (0x1 << 14) +#define RT5651_M_IF1_DAC_L_SFT 14 +#define RT5651_M_ADCMIX_R (0x1 << 7) +#define RT5651_M_ADCMIX_R_SFT 7 +#define RT5651_M_IF1_DAC_R (0x1 << 6) +#define RT5651_M_IF1_DAC_R_SFT 6 + +/* Stereo DAC Mixer Control (0x2a) */ +#define RT5651_M_DAC_L1_MIXL (0x1 << 14) +#define RT5651_M_DAC_L1_MIXL_SFT 14 +#define RT5651_DAC_L1_STO_L_VOL_MASK (0x1 << 13) +#define RT5651_DAC_L1_STO_L_VOL_SFT 13 +#define RT5651_M_DAC_L2_MIXL (0x1 << 12) +#define RT5651_M_DAC_L2_MIXL_SFT 12 +#define RT5651_DAC_L2_STO_L_VOL_MASK (0x1 << 11) +#define RT5651_DAC_L2_STO_L_VOL_SFT 11 +#define RT5651_M_DAC_R1_MIXL (0x1 << 9) +#define RT5651_M_DAC_R1_MIXL_SFT 9 +#define RT5651_DAC_R1_STO_L_VOL_MASK (0x1 << 8) +#define RT5651_DAC_R1_STO_L_VOL_SFT 8 +#define RT5651_M_DAC_R1_MIXR (0x1 << 6) +#define RT5651_M_DAC_R1_MIXR_SFT 6 +#define RT5651_DAC_R1_STO_R_VOL_MASK (0x1 << 5) +#define RT5651_DAC_R1_STO_R_VOL_SFT 5 +#define RT5651_M_DAC_R2_MIXR (0x1 << 4) +#define RT5651_M_DAC_R2_MIXR_SFT 4 +#define RT5651_DAC_R2_STO_R_VOL_MASK (0x1 << 3) +#define RT5651_DAC_R2_STO_R_VOL_SFT 3 +#define RT5651_M_DAC_L1_MIXR (0x1 << 1) +#define RT5651_M_DAC_L1_MIXR_SFT 1 +#define RT5651_DAC_L1_STO_R_VOL_MASK (0x1) +#define RT5651_DAC_L1_STO_R_VOL_SFT 0 + +/* DD Mixer Control (0x2b) */ +#define RT5651_M_STO_DD_L1 (0x1 << 14) +#define RT5651_M_STO_DD_L1_SFT 14 +#define RT5651_STO_DD_L1_VOL_MASK (0x1 << 13) +#define RT5651_DAC_DD_L1_VOL_SFT 13 +#define RT5651_M_STO_DD_L2 (0x1 << 12) +#define RT5651_M_STO_DD_L2_SFT 12 +#define RT5651_STO_DD_L2_VOL_MASK (0x1 << 11) +#define RT5651_STO_DD_L2_VOL_SFT 11 +#define RT5651_M_STO_DD_R2_L (0x1 << 10) +#define RT5651_M_STO_DD_R2_L_SFT 10 +#define RT5651_STO_DD_R2_L_VOL_MASK (0x1 << 9) +#define RT5651_STO_DD_R2_L_VOL_SFT 9 +#define RT5651_M_STO_DD_R1 (0x1 << 6) +#define RT5651_M_STO_DD_R1_SFT 6 +#define RT5651_STO_DD_R1_VOL_MASK (0x1 << 5) +#define RT5651_STO_DD_R1_VOL_SFT 5 +#define RT5651_M_STO_DD_R2 (0x1 << 4) +#define RT5651_M_STO_DD_R2_SFT 4 +#define RT5651_STO_DD_R2_VOL_MASK (0x1 << 3) +#define RT5651_STO_DD_R2_VOL_SFT 3 +#define RT5651_M_STO_DD_L2_R (0x1 << 2) +#define RT5651_M_STO_DD_L2_R_SFT 2 +#define RT5651_STO_DD_L2_R_VOL_MASK (0x1 << 1) +#define RT5651_STO_DD_L2_R_VOL_SFT 1 + +/* Digital Mixer Control (0x2c) */ +#define RT5651_M_STO_L_DAC_L (0x1 << 15) +#define RT5651_M_STO_L_DAC_L_SFT 15 +#define RT5651_STO_L_DAC_L_VOL_MASK (0x1 << 14) +#define RT5651_STO_L_DAC_L_VOL_SFT 14 +#define RT5651_M_DAC_L2_DAC_L (0x1 << 13) +#define RT5651_M_DAC_L2_DAC_L_SFT 13 +#define RT5651_DAC_L2_DAC_L_VOL_MASK (0x1 << 12) +#define RT5651_DAC_L2_DAC_L_VOL_SFT 12 +#define RT5651_M_STO_R_DAC_R (0x1 << 11) +#define RT5651_M_STO_R_DAC_R_SFT 11 +#define RT5651_STO_R_DAC_R_VOL_MASK (0x1 << 10) +#define RT5651_STO_R_DAC_R_VOL_SFT 10 +#define RT5651_M_DAC_R2_DAC_R (0x1 << 9) +#define RT5651_M_DAC_R2_DAC_R_SFT 9 +#define RT5651_DAC_R2_DAC_R_VOL_MASK (0x1 << 8) +#define RT5651_DAC_R2_DAC_R_VOL_SFT 8 + +/* DSP Path Control 1 (0x2d) */ +#define RT5651_RXDP_SRC_MASK (0x1 << 15) +#define RT5651_RXDP_SRC_SFT 15 +#define RT5651_RXDP_SRC_NOR (0x0 << 15) +#define RT5651_RXDP_SRC_DIV3 (0x1 << 15) +#define RT5651_TXDP_SRC_MASK (0x1 << 14) +#define RT5651_TXDP_SRC_SFT 14 +#define RT5651_TXDP_SRC_NOR (0x0 << 14) +#define RT5651_TXDP_SRC_DIV3 (0x1 << 14) + +/* DSP Path Control 2 (0x2e) */ +#define RT5651_DAC_L2_SEL_MASK (0x3 << 14) +#define RT5651_DAC_L2_SEL_SFT 14 +#define RT5651_DAC_L2_SEL_IF2 (0x0 << 14) +#define RT5651_DAC_L2_SEL_IF3 (0x1 << 14) +#define RT5651_DAC_L2_SEL_TXDC (0x2 << 14) +#define RT5651_DAC_L2_SEL_BASS (0x3 << 14) +#define RT5651_DAC_R2_SEL_MASK (0x3 << 12) +#define RT5651_DAC_R2_SEL_SFT 12 +#define RT5651_DAC_R2_SEL_IF2 (0x0 << 12) +#define RT5651_DAC_R2_SEL_IF3 (0x1 << 12) +#define RT5651_DAC_R2_SEL_TXDC (0x2 << 12) +#define RT5651_IF2_ADC_L_SEL_MASK (0x1 << 11) +#define RT5651_IF2_ADC_L_SEL_SFT 11 +#define RT5651_IF2_ADC_L_SEL_TXDP (0x0 << 11) +#define RT5651_IF2_ADC_L_SEL_PASS (0x1 << 11) +#define RT5651_IF2_ADC_R_SEL_MASK (0x1 << 10) +#define RT5651_IF2_ADC_R_SEL_SFT 10 +#define RT5651_IF2_ADC_R_SEL_TXDP (0x0 << 10) +#define RT5651_IF2_ADC_R_SEL_PASS (0x1 << 10) +#define RT5651_RXDC_SEL_MASK (0x3 << 8) +#define RT5651_RXDC_SEL_SFT 8 +#define RT5651_RXDC_SEL_NOR (0x0 << 8) +#define RT5651_RXDC_SEL_L2R (0x1 << 8) +#define RT5651_RXDC_SEL_R2L (0x2 << 8) +#define RT5651_RXDC_SEL_SWAP (0x3 << 8) +#define RT5651_RXDP_SEL_MASK (0x3 << 6) +#define RT5651_RXDP_SEL_SFT 6 +#define RT5651_RXDP_SEL_NOR (0x0 << 6) +#define RT5651_RXDP_SEL_L2R (0x1 << 6) +#define RT5651_RXDP_SEL_R2L (0x2 << 6) +#define RT5651_RXDP_SEL_SWAP (0x3 << 6) +#define RT5651_TXDC_SEL_MASK (0x3 << 4) +#define RT5651_TXDC_SEL_SFT 4 +#define RT5651_TXDC_SEL_NOR (0x0 << 4) +#define RT5651_TXDC_SEL_L2R (0x1 << 4) +#define RT5651_TXDC_SEL_R2L (0x2 << 4) +#define RT5651_TXDC_SEL_SWAP (0x3 << 4) +#define RT5651_TXDP_SEL_MASK (0x3 << 2) +#define RT5651_TXDP_SEL_SFT 2 +#define RT5651_TXDP_SEL_NOR (0x0 << 2) +#define RT5651_TXDP_SEL_L2R (0x1 << 2) +#define RT5651_TXDP_SEL_R2L (0x2 << 2) +#define RT5651_TRXDP_SEL_SWAP (0x3 << 2) + +/* Digital Interface Data Control (0x2f) */ +#define RT5651_IF2_DAC_SEL_MASK (0x3 << 10) +#define RT5651_IF2_DAC_SEL_SFT 10 +#define RT5651_IF2_DAC_SEL_NOR (0x0 << 10) +#define RT5651_IF2_DAC_SEL_SWAP (0x1 << 10) +#define RT5651_IF2_DAC_SEL_L2R (0x2 << 10) +#define RT5651_IF2_DAC_SEL_R2L (0x3 << 10) +#define RT5651_IF2_ADC_SEL_MASK (0x3 << 8) +#define RT5651_IF2_ADC_SEL_SFT 8 +#define RT5651_IF2_ADC_SEL_NOR (0x0 << 8) +#define RT5651_IF2_ADC_SEL_SWAP (0x1 << 8) +#define RT5651_IF2_ADC_SEL_L2R (0x2 << 8) +#define RT5651_IF2_ADC_SEL_R2L (0x3 << 8) +#define RT5651_IF2_ADC_SRC_MASK (0x1 << 7) +#define RT5651_IF2_ADC_SRC_SFT 7 +#define RT5651_IF1_ADC1 (0x0 << 7) +#define RT5651_IF1_ADC2 (0x1 << 7) + +/* PDM Output Control (0x30) */ +#define RT5651_PDM_L_SEL_MASK (0x1 << 15) +#define RT5651_PDM_L_SEL_SFT 15 +#define RT5651_PDM_L_SEL_DD_L (0x0 << 15) +#define RT5651_PDM_L_SEL_STO_L (0x1 << 15) +#define RT5651_M_PDM_L (0x1 << 14) +#define RT5651_M_PDM_L_SFT 14 +#define RT5651_PDM_R_SEL_MASK (0x1 << 13) +#define RT5651_PDM_R_SEL_SFT 13 +#define RT5651_PDM_R_SEL_DD_L (0x0 << 13) +#define RT5651_PDM_R_SEL_STO_L (0x1 << 13) +#define RT5651_M_PDM_R (0x1 << 12) +#define RT5651_M_PDM_R_SFT 12 +#define RT5651_PDM_BUSY (0x1 << 6) +#define RT5651_PDM_BUSY_SFT 6 +#define RT5651_PDM_PATTERN_SEL_MASK (0x1 << 5) +#define RT5651_PDM_PATTERN_SEL_64 (0x0 << 5) +#define RT5651_PDM_PATTERN_SEL_128 (0x1 << 5) +#define RT5651_PDM_VOL_MASK (0x1 << 4) +#define RT5651_PDM_VOL_SFT 4 +#define RT5651_PDM_DIV_MASK (0x3) +#define RT5651_PDM_DIV_SFT 0 +#define RT5651_PDM_DIV_1 0 +#define RT5651_PDM_DIV_2 1 +#define RT5651_PDM_DIV_3 2 +#define RT5651_PDM_DIV_4 3 + +/* PDM I2C/Data Control 1 (0x31) */ +#define RT5651_PDM_I2C_ID_MASK (0xf << 12) +#define PT5631_PDM_CMD_EXE (0x1 << 11) +#define RT5651_PDM_I2C_CMD_MASK (0x1 << 10) +#define RT5651_PDM_I2C_CMD_R (0x0 << 10) +#define RT5651_PDM_I2C_CMD_W (0x1 << 10) +#define RT5651_PDM_I2C_CMD_EXE (0x1 << 9) +#define RT5651_PDM_I2C_NORMAL (0x0 << 8) +#define RT5651_PDM_I2C_BUSY (0x1 << 8) + +/* PDM I2C/Data Control 2 (0x32) */ +#define RT5651_PDM_I2C_ADDR (0xff << 8) +#define RT5651_PDM_I2C_CMD_PATTERN (0xff) + + +/* REC Left Mixer Control 1 (0x3b) */ +#define RT5651_G_LN_L2_RM_L_MASK (0x7 << 13) +#define RT5651_G_IN_L2_RM_L_SFT 13 +#define RT5651_G_LN_L1_RM_L_MASK (0x7 << 10) +#define RT5651_G_IN_L1_RM_L_SFT 10 +#define RT5651_G_BST3_RM_L_MASK (0x7 << 4) +#define RT5651_G_BST3_RM_L_SFT 4 +#define RT5651_G_BST2_RM_L_MASK (0x7 << 1) +#define RT5651_G_BST2_RM_L_SFT 1 + +/* REC Left Mixer Control 2 (0x3c) */ +#define RT5651_G_BST1_RM_L_MASK (0x7 << 13) +#define RT5651_G_BST1_RM_L_SFT 13 +#define RT5651_G_OM_L_RM_L_MASK (0x7 << 10) +#define RT5651_G_OM_L_RM_L_SFT 10 +#define RT5651_M_IN2_L_RM_L (0x1 << 6) +#define RT5651_M_IN2_L_RM_L_SFT 6 +#define RT5651_M_IN1_L_RM_L (0x1 << 5) +#define RT5651_M_IN1_L_RM_L_SFT 5 +#define RT5651_M_BST3_RM_L (0x1 << 3) +#define RT5651_M_BST3_RM_L_SFT 3 +#define RT5651_M_BST2_RM_L (0x1 << 2) +#define RT5651_M_BST2_RM_L_SFT 2 +#define RT5651_M_BST1_RM_L (0x1 << 1) +#define RT5651_M_BST1_RM_L_SFT 1 +#define RT5651_M_OM_L_RM_L (0x1) +#define RT5651_M_OM_L_RM_L_SFT 0 + +/* REC Right Mixer Control 1 (0x3d) */ +#define RT5651_G_IN2_R_RM_R_MASK (0x7 << 13) +#define RT5651_G_IN2_R_RM_R_SFT 13 +#define RT5651_G_IN1_R_RM_R_MASK (0x7 << 10) +#define RT5651_G_IN1_R_RM_R_SFT 10 +#define RT5651_G_BST3_RM_R_MASK (0x7 << 4) +#define RT5651_G_BST3_RM_R_SFT 4 +#define RT5651_G_BST2_RM_R_MASK (0x7 << 1) +#define RT5651_G_BST2_RM_R_SFT 1 + +/* REC Right Mixer Control 2 (0x3e) */ +#define RT5651_G_BST1_RM_R_MASK (0x7 << 13) +#define RT5651_G_BST1_RM_R_SFT 13 +#define RT5651_G_OM_R_RM_R_MASK (0x7 << 10) +#define RT5651_G_OM_R_RM_R_SFT 10 +#define RT5651_M_IN2_R_RM_R (0x1 << 6) +#define RT5651_M_IN2_R_RM_R_SFT 6 +#define RT5651_M_IN1_R_RM_R (0x1 << 5) +#define RT5651_M_IN1_R_RM_R_SFT 5 +#define RT5651_M_BST3_RM_R (0x1 << 3) +#define RT5651_M_BST3_RM_R_SFT 3 +#define RT5651_M_BST2_RM_R (0x1 << 2) +#define RT5651_M_BST2_RM_R_SFT 2 +#define RT5651_M_BST1_RM_R (0x1 << 1) +#define RT5651_M_BST1_RM_R_SFT 1 +#define RT5651_M_OM_R_RM_R (0x1) +#define RT5651_M_OM_R_RM_R_SFT 0 + +/* HPMIX Control (0x45) */ +#define RT5651_M_DAC1_HM (0x1 << 14) +#define RT5651_M_DAC1_HM_SFT 14 +#define RT5651_M_HPVOL_HM (0x1 << 13) +#define RT5651_M_HPVOL_HM_SFT 13 +#define RT5651_G_HPOMIX_MASK (0x1 << 12) +#define RT5651_G_HPOMIX_SFT 12 + +/* SPK Left Mixer Control (0x46) */ +#define RT5651_G_RM_L_SM_L_MASK (0x3 << 14) +#define RT5651_G_RM_L_SM_L_SFT 14 +#define RT5651_G_IN_L_SM_L_MASK (0x3 << 12) +#define RT5651_G_IN_L_SM_L_SFT 12 +#define RT5651_G_DAC_L1_SM_L_MASK (0x3 << 10) +#define RT5651_G_DAC_L1_SM_L_SFT 10 +#define RT5651_G_DAC_L2_SM_L_MASK (0x3 << 8) +#define RT5651_G_DAC_L2_SM_L_SFT 8 +#define RT5651_G_OM_L_SM_L_MASK (0x3 << 6) +#define RT5651_G_OM_L_SM_L_SFT 6 +#define RT5651_M_RM_L_SM_L (0x1 << 5) +#define RT5651_M_RM_L_SM_L_SFT 5 +#define RT5651_M_IN_L_SM_L (0x1 << 4) +#define RT5651_M_IN_L_SM_L_SFT 4 +#define RT5651_M_DAC_L1_SM_L (0x1 << 3) +#define RT5651_M_DAC_L1_SM_L_SFT 3 +#define RT5651_M_DAC_L2_SM_L (0x1 << 2) +#define RT5651_M_DAC_L2_SM_L_SFT 2 +#define RT5651_M_OM_L_SM_L (0x1 << 1) +#define RT5651_M_OM_L_SM_L_SFT 1 + +/* SPK Right Mixer Control (0x47) */ +#define RT5651_G_RM_R_SM_R_MASK (0x3 << 14) +#define RT5651_G_RM_R_SM_R_SFT 14 +#define RT5651_G_IN_R_SM_R_MASK (0x3 << 12) +#define RT5651_G_IN_R_SM_R_SFT 12 +#define RT5651_G_DAC_R1_SM_R_MASK (0x3 << 10) +#define RT5651_G_DAC_R1_SM_R_SFT 10 +#define RT5651_G_DAC_R2_SM_R_MASK (0x3 << 8) +#define RT5651_G_DAC_R2_SM_R_SFT 8 +#define RT5651_G_OM_R_SM_R_MASK (0x3 << 6) +#define RT5651_G_OM_R_SM_R_SFT 6 +#define RT5651_M_RM_R_SM_R (0x1 << 5) +#define RT5651_M_RM_R_SM_R_SFT 5 +#define RT5651_M_IN_R_SM_R (0x1 << 4) +#define RT5651_M_IN_R_SM_R_SFT 4 +#define RT5651_M_DAC_R1_SM_R (0x1 << 3) +#define RT5651_M_DAC_R1_SM_R_SFT 3 +#define RT5651_M_DAC_R2_SM_R (0x1 << 2) +#define RT5651_M_DAC_R2_SM_R_SFT 2 +#define RT5651_M_OM_R_SM_R (0x1 << 1) +#define RT5651_M_OM_R_SM_R_SFT 1 + +/* SPOLMIX Control (0x48) */ +#define RT5651_M_DAC_R1_SPM_L (0x1 << 15) +#define RT5651_M_DAC_R1_SPM_L_SFT 15 +#define RT5651_M_DAC_L1_SPM_L (0x1 << 14) +#define RT5651_M_DAC_L1_SPM_L_SFT 14 +#define RT5651_M_SV_R_SPM_L (0x1 << 13) +#define RT5651_M_SV_R_SPM_L_SFT 13 +#define RT5651_M_SV_L_SPM_L (0x1 << 12) +#define RT5651_M_SV_L_SPM_L_SFT 12 +#define RT5651_M_BST1_SPM_L (0x1 << 11) +#define RT5651_M_BST1_SPM_L_SFT 11 + +/* SPORMIX Control (0x49) */ +#define RT5651_M_DAC_R1_SPM_R (0x1 << 13) +#define RT5651_M_DAC_R1_SPM_R_SFT 13 +#define RT5651_M_SV_R_SPM_R (0x1 << 12) +#define RT5651_M_SV_R_SPM_R_SFT 12 +#define RT5651_M_BST1_SPM_R (0x1 << 11) +#define RT5651_M_BST1_SPM_R_SFT 11 + +/* SPOLMIX / SPORMIX Ratio Control (0x4a) */ +#define RT5651_SPO_CLSD_RATIO_MASK (0x7) +#define RT5651_SPO_CLSD_RATIO_SFT 0 + +/* Mono Output Mixer Control (0x4c) */ +#define RT5651_M_DAC_R2_MM (0x1 << 15) +#define RT5651_M_DAC_R2_MM_SFT 15 +#define RT5651_M_DAC_L2_MM (0x1 << 14) +#define RT5651_M_DAC_L2_MM_SFT 14 +#define RT5651_M_OV_R_MM (0x1 << 13) +#define RT5651_M_OV_R_MM_SFT 13 +#define RT5651_M_OV_L_MM (0x1 << 12) +#define RT5651_M_OV_L_MM_SFT 12 +#define RT5651_M_BST1_MM (0x1 << 11) +#define RT5651_M_BST1_MM_SFT 11 +#define RT5651_G_MONOMIX_MASK (0x1 << 10) +#define RT5651_G_MONOMIX_SFT 10 + +/* Output Left Mixer Control 1 (0x4d) */ +#define RT5651_G_BST2_OM_L_MASK (0x7 << 10) +#define RT5651_G_BST2_OM_L_SFT 10 +#define RT5651_G_BST1_OM_L_MASK (0x7 << 7) +#define RT5651_G_BST1_OM_L_SFT 7 +#define RT5651_G_IN1_L_OM_L_MASK (0x7 << 4) +#define RT5651_G_IN1_L_OM_L_SFT 4 +#define RT5651_G_RM_L_OM_L_MASK (0x7 << 1) +#define RT5651_G_RM_L_OM_L_SFT 1 + +/* Output Left Mixer Control 2 (0x4e) */ +#define RT5651_G_DAC_L1_OM_L_MASK (0x7 << 7) +#define RT5651_G_DAC_L1_OM_L_SFT 7 +#define RT5651_G_IN2_L_OM_L_MASK (0x7 << 4) +#define RT5651_G_IN2_L_OM_L_SFT 4 + +/* Output Left Mixer Control 3 (0x4f) */ +#define RT5651_M_IN2_L_OM_L (0x1 << 9) +#define RT5651_M_IN2_L_OM_L_SFT 9 +#define RT5651_M_BST2_OM_L (0x1 << 6) +#define RT5651_M_BST2_OM_L_SFT 6 +#define RT5651_M_BST1_OM_L (0x1 << 5) +#define RT5651_M_BST1_OM_L_SFT 5 +#define RT5651_M_IN1_L_OM_L (0x1 << 4) +#define RT5651_M_IN1_L_OM_L_SFT 4 +#define RT5651_M_RM_L_OM_L (0x1 << 3) +#define RT5651_M_RM_L_OM_L_SFT 3 +#define RT5651_M_DAC_L1_OM_L (0x1) +#define RT5651_M_DAC_L1_OM_L_SFT 0 + +/* Output Right Mixer Control 1 (0x50) */ +#define RT5651_G_BST2_OM_R_MASK (0x7 << 10) +#define RT5651_G_BST2_OM_R_SFT 10 +#define RT5651_G_BST1_OM_R_MASK (0x7 << 7) +#define RT5651_G_BST1_OM_R_SFT 7 +#define RT5651_G_IN1_R_OM_R_MASK (0x7 << 4) +#define RT5651_G_IN1_R_OM_R_SFT 4 +#define RT5651_G_RM_R_OM_R_MASK (0x7 << 1) +#define RT5651_G_RM_R_OM_R_SFT 1 + +/* Output Right Mixer Control 2 (0x51) */ +#define RT5651_G_DAC_R1_OM_R_MASK (0x7 << 7) +#define RT5651_G_DAC_R1_OM_R_SFT 7 +#define RT5651_G_IN2_R_OM_R_MASK (0x7 << 4) +#define RT5651_G_IN2_R_OM_R_SFT 4 + +/* Output Right Mixer Control 3 (0x52) */ +#define RT5651_M_IN2_R_OM_R (0x1 << 9) +#define RT5651_M_IN2_R_OM_R_SFT 9 +#define RT5651_M_BST2_OM_R (0x1 << 6) +#define RT5651_M_BST2_OM_R_SFT 6 +#define RT5651_M_BST1_OM_R (0x1 << 5) +#define RT5651_M_BST1_OM_R_SFT 5 +#define RT5651_M_IN1_R_OM_R (0x1 << 4) +#define RT5651_M_IN1_R_OM_R_SFT 4 +#define RT5651_M_RM_R_OM_R (0x1 << 3) +#define RT5651_M_RM_R_OM_R_SFT 3 +#define RT5651_M_DAC_R1_OM_R (0x1) +#define RT5651_M_DAC_R1_OM_R_SFT 0 + +/* LOUT Mixer Control (0x53) */ +#define RT5651_M_DAC_L1_LM (0x1 << 15) +#define RT5651_M_DAC_L1_LM_SFT 15 +#define RT5651_M_DAC_R1_LM (0x1 << 14) +#define RT5651_M_DAC_R1_LM_SFT 14 +#define RT5651_M_OV_L_LM (0x1 << 13) +#define RT5651_M_OV_L_LM_SFT 13 +#define RT5651_M_OV_R_LM (0x1 << 12) +#define RT5651_M_OV_R_LM_SFT 12 +#define RT5651_G_LOUTMIX_MASK (0x1 << 11) +#define RT5651_G_LOUTMIX_SFT 11 + +/* Power Management for Digital 1 (0x61) */ +#define RT5651_PWR_I2S1 (0x1 << 15) +#define RT5651_PWR_I2S1_BIT 15 +#define RT5651_PWR_I2S2 (0x1 << 14) +#define RT5651_PWR_I2S2_BIT 14 +#define RT5651_PWR_DAC_L1 (0x1 << 12) +#define RT5651_PWR_DAC_L1_BIT 12 +#define RT5651_PWR_DAC_R1 (0x1 << 11) +#define RT5651_PWR_DAC_R1_BIT 11 +#define RT5651_PWR_ADC_L (0x1 << 2) +#define RT5651_PWR_ADC_L_BIT 2 +#define RT5651_PWR_ADC_R (0x1 << 1) +#define RT5651_PWR_ADC_R_BIT 1 + +/* Power Management for Digital 2 (0x62) */ +#define RT5651_PWR_ADC_STO1_F (0x1 << 15) +#define RT5651_PWR_ADC_STO1_F_BIT 15 +#define RT5651_PWR_ADC_STO2_F (0x1 << 14) +#define RT5651_PWR_ADC_STO2_F_BIT 14 +#define RT5651_PWR_DAC_STO1_F (0x1 << 11) +#define RT5651_PWR_DAC_STO1_F_BIT 11 +#define RT5651_PWR_DAC_STO2_F (0x1 << 10) +#define RT5651_PWR_DAC_STO2_F_BIT 10 +#define RT5651_PWR_PDM (0x1 << 9) +#define RT5651_PWR_PDM_BIT 9 + +/* Power Management for Analog 1 (0x63) */ +#define RT5651_PWR_VREF1 (0x1 << 15) +#define RT5651_PWR_VREF1_BIT 15 +#define RT5651_PWR_FV1 (0x1 << 14) +#define RT5651_PWR_FV1_BIT 14 +#define RT5651_PWR_MB (0x1 << 13) +#define RT5651_PWR_MB_BIT 13 +#define RT5651_PWR_LM (0x1 << 12) +#define RT5651_PWR_LM_BIT 12 +#define RT5651_PWR_BG (0x1 << 11) +#define RT5651_PWR_BG_BIT 11 +#define RT5651_PWR_HP_L (0x1 << 7) +#define RT5651_PWR_HP_L_BIT 7 +#define RT5651_PWR_HP_R (0x1 << 6) +#define RT5651_PWR_HP_R_BIT 6 +#define RT5651_PWR_HA (0x1 << 5) +#define RT5651_PWR_HA_BIT 5 +#define RT5651_PWR_VREF2 (0x1 << 4) +#define RT5651_PWR_VREF2_BIT 4 +#define RT5651_PWR_FV2 (0x1 << 3) +#define RT5651_PWR_FV2_BIT 3 +#define RT5651_PWR_LDO (0x1 << 2) +#define RT5651_PWR_LDO_BIT 2 +#define RT5651_PWR_LDO_DVO_MASK (0x3) +#define RT5651_PWR_LDO_DVO_1_0V 0 +#define RT5651_PWR_LDO_DVO_1_1V 1 +#define RT5651_PWR_LDO_DVO_1_2V 2 +#define RT5651_PWR_LDO_DVO_1_3V 3 + +/* Power Management for Analog 2 (0x64) */ +#define RT5651_PWR_BST1 (0x1 << 15) +#define RT5651_PWR_BST1_BIT 15 +#define RT5651_PWR_BST2 (0x1 << 14) +#define RT5651_PWR_BST2_BIT 14 +#define RT5651_PWR_BST3 (0x1 << 13) +#define RT5651_PWR_BST3_BIT 13 +#define RT5651_PWR_MB1 (0x1 << 11) +#define RT5651_PWR_MB1_BIT 11 +#define RT5651_PWR_PLL (0x1 << 9) +#define RT5651_PWR_PLL_BIT 9 +#define RT5651_PWR_BST1_OP2 (0x1 << 5) +#define RT5651_PWR_BST1_OP2_BIT 5 +#define RT5651_PWR_BST2_OP2 (0x1 << 4) +#define RT5651_PWR_BST2_OP2_BIT 4 +#define RT5651_PWR_BST3_OP2 (0x1 << 3) +#define RT5651_PWR_BST3_OP2_BIT 3 +#define RT5651_PWR_JD_M (0x1 << 2) +#define RT5651_PWM_JD_M_BIT 2 +#define RT5651_PWR_JD2 (0x1 << 1) +#define RT5651_PWM_JD2_BIT 1 +#define RT5651_PWR_JD3 (0x1) +#define RT5651_PWM_JD3_BIT 0 + +/* Power Management for Mixer (0x65) */ +#define RT5651_PWR_OM_L (0x1 << 15) +#define RT5651_PWR_OM_L_BIT 15 +#define RT5651_PWR_OM_R (0x1 << 14) +#define RT5651_PWR_OM_R_BIT 14 +#define RT5651_PWR_RM_L (0x1 << 11) +#define RT5651_PWR_RM_L_BIT 11 +#define RT5651_PWR_RM_R (0x1 << 10) +#define RT5651_PWR_RM_R_BIT 10 + +/* Power Management for Volume (0x66) */ +#define RT5651_PWR_OV_L (0x1 << 13) +#define RT5651_PWR_OV_L_BIT 13 +#define RT5651_PWR_OV_R (0x1 << 12) +#define RT5651_PWR_OV_R_BIT 12 +#define RT5651_PWR_HV_L (0x1 << 11) +#define RT5651_PWR_HV_L_BIT 11 +#define RT5651_PWR_HV_R (0x1 << 10) +#define RT5651_PWR_HV_R_BIT 10 +#define RT5651_PWR_IN1_L (0x1 << 9) +#define RT5651_PWR_IN1_L_BIT 9 +#define RT5651_PWR_IN1_R (0x1 << 8) +#define RT5651_PWR_IN1_R_BIT 8 +#define RT5651_PWR_IN2_L (0x1 << 7) +#define RT5651_PWR_IN2_L_BIT 7 +#define RT5651_PWR_IN2_R (0x1 << 6) +#define RT5651_PWR_IN2_R_BIT 6 + +/* I2S1/2/3 Audio Serial Data Port Control (0x70 0x71) */ +#define RT5651_I2S_MS_MASK (0x1 << 15) +#define RT5651_I2S_MS_SFT 15 +#define RT5651_I2S_MS_M (0x0 << 15) +#define RT5651_I2S_MS_S (0x1 << 15) +#define RT5651_I2S_O_CP_MASK (0x3 << 10) +#define RT5651_I2S_O_CP_SFT 10 +#define RT5651_I2S_O_CP_OFF (0x0 << 10) +#define RT5651_I2S_O_CP_U_LAW (0x1 << 10) +#define RT5651_I2S_O_CP_A_LAW (0x2 << 10) +#define RT5651_I2S_I_CP_MASK (0x3 << 8) +#define RT5651_I2S_I_CP_SFT 8 +#define RT5651_I2S_I_CP_OFF (0x0 << 8) +#define RT5651_I2S_I_CP_U_LAW (0x1 << 8) +#define RT5651_I2S_I_CP_A_LAW (0x2 << 8) +#define RT5651_I2S_BP_MASK (0x1 << 7) +#define RT5651_I2S_BP_SFT 7 +#define RT5651_I2S_BP_NOR (0x0 << 7) +#define RT5651_I2S_BP_INV (0x1 << 7) +#define RT5651_I2S_DL_MASK (0x3 << 2) +#define RT5651_I2S_DL_SFT 2 +#define RT5651_I2S_DL_16 (0x0 << 2) +#define RT5651_I2S_DL_20 (0x1 << 2) +#define RT5651_I2S_DL_24 (0x2 << 2) +#define RT5651_I2S_DL_8 (0x3 << 2) +#define RT5651_I2S_DF_MASK (0x3) +#define RT5651_I2S_DF_SFT 0 +#define RT5651_I2S_DF_I2S (0x0) +#define RT5651_I2S_DF_LEFT (0x1) +#define RT5651_I2S_DF_PCM_A (0x2) +#define RT5651_I2S_DF_PCM_B (0x3) + +/* ADC/DAC Clock Control 1 (0x73) */ +#define RT5651_I2S_PD1_MASK (0x7 << 12) +#define RT5651_I2S_PD1_SFT 12 +#define RT5651_I2S_PD1_1 (0x0 << 12) +#define RT5651_I2S_PD1_2 (0x1 << 12) +#define RT5651_I2S_PD1_3 (0x2 << 12) +#define RT5651_I2S_PD1_4 (0x3 << 12) +#define RT5651_I2S_PD1_6 (0x4 << 12) +#define RT5651_I2S_PD1_8 (0x5 << 12) +#define RT5651_I2S_PD1_12 (0x6 << 12) +#define RT5651_I2S_PD1_16 (0x7 << 12) +#define RT5651_I2S_BCLK_MS2_MASK (0x1 << 11) +#define RT5651_I2S_BCLK_MS2_SFT 11 +#define RT5651_I2S_BCLK_MS2_32 (0x0 << 11) +#define RT5651_I2S_BCLK_MS2_64 (0x1 << 11) +#define RT5651_I2S_PD2_MASK (0x7 << 8) +#define RT5651_I2S_PD2_SFT 8 +#define RT5651_I2S_PD2_1 (0x0 << 8) +#define RT5651_I2S_PD2_2 (0x1 << 8) +#define RT5651_I2S_PD2_3 (0x2 << 8) +#define RT5651_I2S_PD2_4 (0x3 << 8) +#define RT5651_I2S_PD2_6 (0x4 << 8) +#define RT5651_I2S_PD2_8 (0x5 << 8) +#define RT5651_I2S_PD2_12 (0x6 << 8) +#define RT5651_I2S_PD2_16 (0x7 << 8) +#define RT5651_DAC_OSR_MASK (0x3 << 2) +#define RT5651_DAC_OSR_SFT 2 +#define RT5651_DAC_OSR_128 (0x0 << 2) +#define RT5651_DAC_OSR_64 (0x1 << 2) +#define RT5651_DAC_OSR_32 (0x2 << 2) +#define RT5651_DAC_OSR_128_3 (0x3 << 2) +#define RT5651_ADC_OSR_MASK (0x3) +#define RT5651_ADC_OSR_SFT 0 +#define RT5651_ADC_OSR_128 (0x0) +#define RT5651_ADC_OSR_64 (0x1) +#define RT5651_ADC_OSR_32 (0x2) +#define RT5651_ADC_OSR_128_3 (0x3) + +/* ADC/DAC Clock Control 2 (0x74) */ +#define RT5651_DAHPF_EN (0x1 << 11) +#define RT5651_DAHPF_EN_SFT 11 +#define RT5651_ADHPF_EN (0x1 << 10) +#define RT5651_ADHPF_EN_SFT 10 + +/* Digital Microphone Control (0x75) */ +#define RT5651_DMIC_1_EN_MASK (0x1 << 15) +#define RT5651_DMIC_1_EN_SFT 15 +#define RT5651_DMIC_1_DIS (0x0 << 15) +#define RT5651_DMIC_1_EN (0x1 << 15) +#define RT5651_DMIC_1L_LH_MASK (0x1 << 13) +#define RT5651_DMIC_1L_LH_SFT 13 +#define RT5651_DMIC_1L_LH_FALLING (0x0 << 13) +#define RT5651_DMIC_1L_LH_RISING (0x1 << 13) +#define RT5651_DMIC_1R_LH_MASK (0x1 << 12) +#define RT5651_DMIC_1R_LH_SFT 12 +#define RT5651_DMIC_1R_LH_FALLING (0x0 << 12) +#define RT5651_DMIC_1R_LH_RISING (0x1 << 12) +#define RT5651_DMIC_1_DP_MASK (0x3 << 10) +#define RT5651_DMIC_1_DP_SFT 10 +#define RT5651_DMIC_1_DP_GPIO6 (0x0 << 10) +#define RT5651_DMIC_1_DP_IN1P (0x1 << 10) +#define RT5651_DMIC_2_DP_GPIO8 (0x2 << 10) +#define RT5651_DMIC_CLK_MASK (0x7 << 5) +#define RT5651_DMIC_CLK_SFT 5 + +/* TDM Control 1 (0x77) */ +#define RT5651_TDM_INTEL_SEL_MASK (0x1 << 15) +#define RT5651_TDM_INTEL_SEL_SFT 15 +#define RT5651_TDM_INTEL_SEL_64 (0x0 << 15) +#define RT5651_TDM_INTEL_SEL_50 (0x1 << 15) +#define RT5651_TDM_MODE_SEL_MASK (0x1 << 14) +#define RT5651_TDM_MODE_SEL_SFT 14 +#define RT5651_TDM_MODE_SEL_NOR (0x0 << 14) +#define RT5651_TDM_MODE_SEL_TDM (0x1 << 14) +#define RT5651_TDM_CH_NUM_SEL_MASK (0x3 << 12) +#define RT5651_TDM_CH_NUM_SEL_SFT 12 +#define RT5651_TDM_CH_NUM_SEL_2 (0x0 << 12) +#define RT5651_TDM_CH_NUM_SEL_4 (0x1 << 12) +#define RT5651_TDM_CH_NUM_SEL_6 (0x2 << 12) +#define RT5651_TDM_CH_NUM_SEL_8 (0x3 << 12) +#define RT5651_TDM_CH_LEN_SEL_MASK (0x3 << 10) +#define RT5651_TDM_CH_LEN_SEL_SFT 10 +#define RT5651_TDM_CH_LEN_SEL_16 (0x0 << 10) +#define RT5651_TDM_CH_LEN_SEL_20 (0x1 << 10) +#define RT5651_TDM_CH_LEN_SEL_24 (0x2 << 10) +#define RT5651_TDM_CH_LEN_SEL_32 (0x3 << 10) +#define RT5651_TDM_ADC_SEL_MASK (0x1 << 9) +#define RT5651_TDM_ADC_SEL_SFT 9 +#define RT5651_TDM_ADC_SEL_NOR (0x0 << 9) +#define RT5651_TDM_ADC_SEL_SWAP (0x1 << 9) +#define RT5651_TDM_ADC_START_SEL_MASK (0x1 << 8) +#define RT5651_TDM_ADC_START_SEL_SFT 8 +#define RT5651_TDM_ADC_START_SEL_SL0 (0x0 << 8) +#define RT5651_TDM_ADC_START_SEL_SL4 (0x1 << 8) +#define RT5651_TDM_I2S_CH2_SEL_MASK (0x3 << 6) +#define RT5651_TDM_I2S_CH2_SEL_SFT 6 +#define RT5651_TDM_I2S_CH2_SEL_LR (0x0 << 6) +#define RT5651_TDM_I2S_CH2_SEL_RL (0x1 << 6) +#define RT5651_TDM_I2S_CH2_SEL_LL (0x2 << 6) +#define RT5651_TDM_I2S_CH2_SEL_RR (0x3 << 6) +#define RT5651_TDM_I2S_CH4_SEL_MASK (0x3 << 4) +#define RT5651_TDM_I2S_CH4_SEL_SFT 4 +#define RT5651_TDM_I2S_CH4_SEL_LR (0x0 << 4) +#define RT5651_TDM_I2S_CH4_SEL_RL (0x1 << 4) +#define RT5651_TDM_I2S_CH4_SEL_LL (0x2 << 4) +#define RT5651_TDM_I2S_CH4_SEL_RR (0x3 << 4) +#define RT5651_TDM_I2S_CH6_SEL_MASK (0x3 << 2) +#define RT5651_TDM_I2S_CH6_SEL_SFT 2 +#define RT5651_TDM_I2S_CH6_SEL_LR (0x0 << 2) +#define RT5651_TDM_I2S_CH6_SEL_RL (0x1 << 2) +#define RT5651_TDM_I2S_CH6_SEL_LL (0x2 << 2) +#define RT5651_TDM_I2S_CH6_SEL_RR (0x3 << 2) +#define RT5651_TDM_I2S_CH8_SEL_MASK (0x3) +#define RT5651_TDM_I2S_CH8_SEL_SFT 0 +#define RT5651_TDM_I2S_CH8_SEL_LR (0x0) +#define RT5651_TDM_I2S_CH8_SEL_RL (0x1) +#define RT5651_TDM_I2S_CH8_SEL_LL (0x2) +#define RT5651_TDM_I2S_CH8_SEL_RR (0x3) + +/* TDM Control 2 (0x78) */ +#define RT5651_TDM_LRCK_POL_SEL_MASK (0x1 << 15) +#define RT5651_TDM_LRCK_POL_SEL_SFT 15 +#define RT5651_TDM_LRCK_POL_SEL_NOR (0x0 << 15) +#define RT5651_TDM_LRCK_POL_SEL_INV (0x1 << 15) +#define RT5651_TDM_CH_VAL_SEL_MASK (0x1 << 14) +#define RT5651_TDM_CH_VAL_SEL_SFT 14 +#define RT5651_TDM_CH_VAL_SEL_CH01 (0x0 << 14) +#define RT5651_TDM_CH_VAL_SEL_CH0123 (0x1 << 14) +#define RT5651_TDM_CH_VAL_EN (0x1 << 13) +#define RT5651_TDM_CH_VAL_SFT 13 +#define RT5651_TDM_LPBK_EN (0x1 << 12) +#define RT5651_TDM_LPBK_SFT 12 +#define RT5651_TDM_LRCK_PULSE_SEL_MASK (0x1 << 11) +#define RT5651_TDM_LRCK_PULSE_SEL_SFT 11 +#define RT5651_TDM_LRCK_PULSE_SEL_BCLK (0x0 << 11) +#define RT5651_TDM_LRCK_PULSE_SEL_CH (0x1 << 11) +#define RT5651_TDM_END_EDGE_SEL_MASK (0x1 << 10) +#define RT5651_TDM_END_EDGE_SEL_SFT 10 +#define RT5651_TDM_END_EDGE_SEL_POS (0x0 << 10) +#define RT5651_TDM_END_EDGE_SEL_NEG (0x1 << 10) +#define RT5651_TDM_END_EDGE_EN (0x1 << 9) +#define RT5651_TDM_END_EDGE_EN_SFT 9 +#define RT5651_TDM_TRAN_EDGE_SEL_MASK (0x1 << 8) +#define RT5651_TDM_TRAN_EDGE_SEL_SFT 8 +#define RT5651_TDM_TRAN_EDGE_SEL_POS (0x0 << 8) +#define RT5651_TDM_TRAN_EDGE_SEL_NEG (0x1 << 8) +#define RT5651_M_TDM2_L (0x1 << 7) +#define RT5651_M_TDM2_L_SFT 7 +#define RT5651_M_TDM2_R (0x1 << 6) +#define RT5651_M_TDM2_R_SFT 6 +#define RT5651_M_TDM4_L (0x1 << 5) +#define RT5651_M_TDM4_L_SFT 5 +#define RT5651_M_TDM4_R (0x1 << 4) +#define RT5651_M_TDM4_R_SFT 4 + +/* TDM Control 3 (0x79) */ +#define RT5651_CH2_L_SEL_MASK (0x7 << 12) +#define RT5651_CH2_L_SEL_SFT 12 +#define RT5651_CH2_L_SEL_SL0 (0x0 << 12) +#define RT5651_CH2_L_SEL_SL1 (0x1 << 12) +#define RT5651_CH2_L_SEL_SL2 (0x2 << 12) +#define RT5651_CH2_L_SEL_SL3 (0x3 << 12) +#define RT5651_CH2_L_SEL_SL4 (0x4 << 12) +#define RT5651_CH2_L_SEL_SL5 (0x5 << 12) +#define RT5651_CH2_L_SEL_SL6 (0x6 << 12) +#define RT5651_CH2_L_SEL_SL7 (0x7 << 12) +#define RT5651_CH2_R_SEL_MASK (0x7 << 8) +#define RT5651_CH2_R_SEL_SFT 8 +#define RT5651_CH2_R_SEL_SL0 (0x0 << 8) +#define RT5651_CH2_R_SEL_SL1 (0x1 << 8) +#define RT5651_CH2_R_SEL_SL2 (0x2 << 8) +#define RT5651_CH2_R_SEL_SL3 (0x3 << 8) +#define RT5651_CH2_R_SEL_SL4 (0x4 << 8) +#define RT5651_CH2_R_SEL_SL5 (0x5 << 8) +#define RT5651_CH2_R_SEL_SL6 (0x6 << 8) +#define RT5651_CH2_R_SEL_SL7 (0x7 << 8) +#define RT5651_CH4_L_SEL_MASK (0x7 << 4) +#define RT5651_CH4_L_SEL_SFT 4 +#define RT5651_CH4_L_SEL_SL0 (0x0 << 4) +#define RT5651_CH4_L_SEL_SL1 (0x1 << 4) +#define RT5651_CH4_L_SEL_SL2 (0x2 << 4) +#define RT5651_CH4_L_SEL_SL3 (0x3 << 4) +#define RT5651_CH4_L_SEL_SL4 (0x4 << 4) +#define RT5651_CH4_L_SEL_SL5 (0x5 << 4) +#define RT5651_CH4_L_SEL_SL6 (0x6 << 4) +#define RT5651_CH4_L_SEL_SL7 (0x7 << 4) +#define RT5651_CH4_R_SEL_MASK (0x7) +#define RT5651_CH4_R_SEL_SFT 0 +#define RT5651_CH4_R_SEL_SL0 (0x0) +#define RT5651_CH4_R_SEL_SL1 (0x1) +#define RT5651_CH4_R_SEL_SL2 (0x2) +#define RT5651_CH4_R_SEL_SL3 (0x3) +#define RT5651_CH4_R_SEL_SL4 (0x4) +#define RT5651_CH4_R_SEL_SL5 (0x5) +#define RT5651_CH4_R_SEL_SL6 (0x6) +#define RT5651_CH4_R_SEL_SL7 (0x7) + +/* Global Clock Control (0x80) */ +#define RT5651_SCLK_SRC_MASK (0x3 << 14) +#define RT5651_SCLK_SRC_SFT 14 +#define RT5651_SCLK_SRC_MCLK (0x0 << 14) +#define RT5651_SCLK_SRC_PLL1 (0x1 << 14) +#define RT5651_SCLK_SRC_RCCLK (0x2 << 14) +#define RT5651_PLL1_SRC_MASK (0x3 << 12) +#define RT5651_PLL1_SRC_SFT 12 +#define RT5651_PLL1_SRC_MCLK (0x0 << 12) +#define RT5651_PLL1_SRC_BCLK1 (0x1 << 12) +#define RT5651_PLL1_SRC_BCLK2 (0x2 << 12) +#define RT5651_PLL1_PD_MASK (0x1 << 3) +#define RT5651_PLL1_PD_SFT 3 +#define RT5651_PLL1_PD_1 (0x0 << 3) +#define RT5651_PLL1_PD_2 (0x1 << 3) + +#define RT5651_PLL_INP_MAX 40000000 +#define RT5651_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x81) */ +#define RT5651_PLL_N_MAX 0x1ff +#define RT5651_PLL_N_MASK (RT5651_PLL_N_MAX << 7) +#define RT5651_PLL_N_SFT 7 +#define RT5651_PLL_K_MAX 0x1f +#define RT5651_PLL_K_MASK (RT5651_PLL_K_MAX) +#define RT5651_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x82) */ +#define RT5651_PLL_M_MAX 0xf +#define RT5651_PLL_M_MASK (RT5651_PLL_M_MAX << 12) +#define RT5651_PLL_M_SFT 12 +#define RT5651_PLL_M_BP (0x1 << 11) +#define RT5651_PLL_M_BP_SFT 11 + +/* PLL tracking mode 1 (0x83) */ +#define RT5651_STO1_T_MASK (0x1 << 15) +#define RT5651_STO1_T_SFT 15 +#define RT5651_STO1_T_SCLK (0x0 << 15) +#define RT5651_STO1_T_LRCK1 (0x1 << 15) +#define RT5651_STO2_T_MASK (0x1 << 12) +#define RT5651_STO2_T_SFT 12 +#define RT5651_STO2_T_I2S2 (0x0 << 12) +#define RT5651_STO2_T_LRCK2 (0x1 << 12) +#define RT5651_ASRC2_REF_MASK (0x1 << 11) +#define RT5651_ASRC2_REF_SFT 11 +#define RT5651_ASRC2_REF_LRCK2 (0x0 << 11) +#define RT5651_ASRC2_REF_LRCK1 (0x1 << 11) +#define RT5651_DMIC_1_M_MASK (0x1 << 9) +#define RT5651_DMIC_1_M_SFT 9 +#define RT5651_DMIC_1_M_NOR (0x0 << 9) +#define RT5651_DMIC_1_M_ASYN (0x1 << 9) + +/* PLL tracking mode 2 (0x84) */ +#define RT5651_STO1_ASRC_EN (0x1 << 15) +#define RT5651_STO1_ASRC_EN_SFT 15 +#define RT5651_STO2_ASRC_EN (0x1 << 14) +#define RT5651_STO2_ASRC_EN_SFT 14 +#define RT5651_STO1_DAC_M_MASK (0x1 << 13) +#define RT5651_STO1_DAC_M_SFT 13 +#define RT5651_STO1_DAC_M_NOR (0x0 << 13) +#define RT5651_STO1_DAC_M_ASRC (0x1 << 13) +#define RT5651_STO2_DAC_M_MASK (0x1 << 12) +#define RT5651_STO2_DAC_M_SFT 12 +#define RT5651_STO2_DAC_M_NOR (0x0 << 12) +#define RT5651_STO2_DAC_M_ASRC (0x1 << 12) +#define RT5651_ADC_M_MASK (0x1 << 11) +#define RT5651_ADC_M_SFT 11 +#define RT5651_ADC_M_NOR (0x0 << 11) +#define RT5651_ADC_M_ASRC (0x1 << 11) +#define RT5651_I2S1_R_D_MASK (0x1 << 4) +#define RT5651_I2S1_R_D_SFT 4 +#define RT5651_I2S1_R_D_DIS (0x0 << 4) +#define RT5651_I2S1_R_D_EN (0x1 << 4) +#define RT5651_I2S2_R_D_MASK (0x1 << 3) +#define RT5651_I2S2_R_D_SFT 3 +#define RT5651_I2S2_R_D_DIS (0x0 << 3) +#define RT5651_I2S2_R_D_EN (0x1 << 3) +#define RT5651_PRE_SCLK_MASK (0x3) +#define RT5651_PRE_SCLK_SFT 0 +#define RT5651_PRE_SCLK_512 (0x0) +#define RT5651_PRE_SCLK_1024 (0x1) +#define RT5651_PRE_SCLK_2048 (0x2) + +/* PLL tracking mode 3 (0x85) */ +#define RT5651_I2S1_RATE_MASK (0xf << 12) +#define RT5651_I2S1_RATE_SFT 12 +#define RT5651_I2S2_RATE_MASK (0xf << 8) +#define RT5651_I2S2_RATE_SFT 8 +#define RT5651_G_ASRC_LP_MASK (0x1 << 3) +#define RT5651_G_ASRC_LP_SFT 3 +#define RT5651_ASRC_LP_F_M (0x1 << 2) +#define RT5651_ASRC_LP_F_SFT 2 +#define RT5651_ASRC_LP_F_NOR (0x0 << 2) +#define RT5651_ASRC_LP_F_SB (0x1 << 2) +#define RT5651_FTK_PH_DET_MASK (0x3) +#define RT5651_FTK_PH_DET_SFT 0 +#define RT5651_FTK_PH_DET_DIV1 (0x0) +#define RT5651_FTK_PH_DET_DIV2 (0x1) +#define RT5651_FTK_PH_DET_DIV4 (0x2) +#define RT5651_FTK_PH_DET_DIV8 (0x3) + +/*PLL tracking mode 6 (0x89) */ +#define RT5651_I2S1_PD_MASK (0x7 << 12) +#define RT5651_I2S1_PD_SFT 12 +#define RT5651_I2S2_PD_MASK (0x7 << 8) +#define RT5651_I2S2_PD_SFT 8 + +/*PLL tracking mode 7 (0x8a) */ +#define RT5651_FSI1_RATE_MASK (0xf << 12) +#define RT5651_FSI1_RATE_SFT 12 +#define RT5651_FSI2_RATE_MASK (0xf << 8) +#define RT5651_FSI2_RATE_SFT 8 + +/* HPOUT Over Current Detection (0x8b) */ +#define RT5651_HP_OVCD_MASK (0x1 << 10) +#define RT5651_HP_OVCD_SFT 10 +#define RT5651_HP_OVCD_DIS (0x0 << 10) +#define RT5651_HP_OVCD_EN (0x1 << 10) +#define RT5651_HP_OC_TH_MASK (0x3 << 8) +#define RT5651_HP_OC_TH_SFT 8 +#define RT5651_HP_OC_TH_90 (0x0 << 8) +#define RT5651_HP_OC_TH_105 (0x1 << 8) +#define RT5651_HP_OC_TH_120 (0x2 << 8) +#define RT5651_HP_OC_TH_135 (0x3 << 8) + +/* Depop Mode Control 1 (0x8e) */ +#define RT5651_SMT_TRIG_MASK (0x1 << 15) +#define RT5651_SMT_TRIG_SFT 15 +#define RT5651_SMT_TRIG_DIS (0x0 << 15) +#define RT5651_SMT_TRIG_EN (0x1 << 15) +#define RT5651_HP_L_SMT_MASK (0x1 << 9) +#define RT5651_HP_L_SMT_SFT 9 +#define RT5651_HP_L_SMT_DIS (0x0 << 9) +#define RT5651_HP_L_SMT_EN (0x1 << 9) +#define RT5651_HP_R_SMT_MASK (0x1 << 8) +#define RT5651_HP_R_SMT_SFT 8 +#define RT5651_HP_R_SMT_DIS (0x0 << 8) +#define RT5651_HP_R_SMT_EN (0x1 << 8) +#define RT5651_HP_CD_PD_MASK (0x1 << 7) +#define RT5651_HP_CD_PD_SFT 7 +#define RT5651_HP_CD_PD_DIS (0x0 << 7) +#define RT5651_HP_CD_PD_EN (0x1 << 7) +#define RT5651_RSTN_MASK (0x1 << 6) +#define RT5651_RSTN_SFT 6 +#define RT5651_RSTN_DIS (0x0 << 6) +#define RT5651_RSTN_EN (0x1 << 6) +#define RT5651_RSTP_MASK (0x1 << 5) +#define RT5651_RSTP_SFT 5 +#define RT5651_RSTP_DIS (0x0 << 5) +#define RT5651_RSTP_EN (0x1 << 5) +#define RT5651_HP_CO_MASK (0x1 << 4) +#define RT5651_HP_CO_SFT 4 +#define RT5651_HP_CO_DIS (0x0 << 4) +#define RT5651_HP_CO_EN (0x1 << 4) +#define RT5651_HP_CP_MASK (0x1 << 3) +#define RT5651_HP_CP_SFT 3 +#define RT5651_HP_CP_PD (0x0 << 3) +#define RT5651_HP_CP_PU (0x1 << 3) +#define RT5651_HP_SG_MASK (0x1 << 2) +#define RT5651_HP_SG_SFT 2 +#define RT5651_HP_SG_DIS (0x0 << 2) +#define RT5651_HP_SG_EN (0x1 << 2) +#define RT5651_HP_DP_MASK (0x1 << 1) +#define RT5651_HP_DP_SFT 1 +#define RT5651_HP_DP_PD (0x0 << 1) +#define RT5651_HP_DP_PU (0x1 << 1) +#define RT5651_HP_CB_MASK (0x1) +#define RT5651_HP_CB_SFT 0 +#define RT5651_HP_CB_PD (0x0) +#define RT5651_HP_CB_PU (0x1) + +/* Depop Mode Control 2 (0x8f) */ +#define RT5651_DEPOP_MASK (0x1 << 13) +#define RT5651_DEPOP_SFT 13 +#define RT5651_DEPOP_AUTO (0x0 << 13) +#define RT5651_DEPOP_MAN (0x1 << 13) +#define RT5651_RAMP_MASK (0x1 << 12) +#define RT5651_RAMP_SFT 12 +#define RT5651_RAMP_DIS (0x0 << 12) +#define RT5651_RAMP_EN (0x1 << 12) +#define RT5651_BPS_MASK (0x1 << 11) +#define RT5651_BPS_SFT 11 +#define RT5651_BPS_DIS (0x0 << 11) +#define RT5651_BPS_EN (0x1 << 11) +#define RT5651_FAST_UPDN_MASK (0x1 << 10) +#define RT5651_FAST_UPDN_SFT 10 +#define RT5651_FAST_UPDN_DIS (0x0 << 10) +#define RT5651_FAST_UPDN_EN (0x1 << 10) +#define RT5651_MRES_MASK (0x3 << 8) +#define RT5651_MRES_SFT 8 +#define RT5651_MRES_15MO (0x0 << 8) +#define RT5651_MRES_25MO (0x1 << 8) +#define RT5651_MRES_35MO (0x2 << 8) +#define RT5651_MRES_45MO (0x3 << 8) +#define RT5651_VLO_MASK (0x1 << 7) +#define RT5651_VLO_SFT 7 +#define RT5651_VLO_3V (0x0 << 7) +#define RT5651_VLO_32V (0x1 << 7) +#define RT5651_DIG_DP_MASK (0x1 << 6) +#define RT5651_DIG_DP_SFT 6 +#define RT5651_DIG_DP_DIS (0x0 << 6) +#define RT5651_DIG_DP_EN (0x1 << 6) +#define RT5651_DP_TH_MASK (0x3 << 4) +#define RT5651_DP_TH_SFT 4 + +/* Depop Mode Control 3 (0x90) */ +#define RT5651_CP_SYS_MASK (0x7 << 12) +#define RT5651_CP_SYS_SFT 12 +#define RT5651_CP_FQ1_MASK (0x7 << 8) +#define RT5651_CP_FQ1_SFT 8 +#define RT5651_CP_FQ2_MASK (0x7 << 4) +#define RT5651_CP_FQ2_SFT 4 +#define RT5651_CP_FQ3_MASK (0x7) +#define RT5651_CP_FQ3_SFT 0 +#define RT5651_CP_FQ_1_5_KHZ 0 +#define RT5651_CP_FQ_3_KHZ 1 +#define RT5651_CP_FQ_6_KHZ 2 +#define RT5651_CP_FQ_12_KHZ 3 +#define RT5651_CP_FQ_24_KHZ 4 +#define RT5651_CP_FQ_48_KHZ 5 +#define RT5651_CP_FQ_96_KHZ 6 +#define RT5651_CP_FQ_192_KHZ 7 + +/* HPOUT charge pump (0x91) */ +#define RT5651_OSW_L_MASK (0x1 << 11) +#define RT5651_OSW_L_SFT 11 +#define RT5651_OSW_L_DIS (0x0 << 11) +#define RT5651_OSW_L_EN (0x1 << 11) +#define RT5651_OSW_R_MASK (0x1 << 10) +#define RT5651_OSW_R_SFT 10 +#define RT5651_OSW_R_DIS (0x0 << 10) +#define RT5651_OSW_R_EN (0x1 << 10) +#define RT5651_PM_HP_MASK (0x3 << 8) +#define RT5651_PM_HP_SFT 8 +#define RT5651_PM_HP_LV (0x0 << 8) +#define RT5651_PM_HP_MV (0x1 << 8) +#define RT5651_PM_HP_HV (0x2 << 8) +#define RT5651_IB_HP_MASK (0x3 << 6) +#define RT5651_IB_HP_SFT 6 +#define RT5651_IB_HP_125IL (0x0 << 6) +#define RT5651_IB_HP_25IL (0x1 << 6) +#define RT5651_IB_HP_5IL (0x2 << 6) +#define RT5651_IB_HP_1IL (0x3 << 6) + +/* Micbias Control (0x93) */ +#define RT5651_MIC1_BS_MASK (0x1 << 15) +#define RT5651_MIC1_BS_SFT 15 +#define RT5651_MIC1_BS_9AV (0x0 << 15) +#define RT5651_MIC1_BS_75AV (0x1 << 15) +#define RT5651_MIC1_CLK_MASK (0x1 << 13) +#define RT5651_MIC1_CLK_SFT 13 +#define RT5651_MIC1_CLK_DIS (0x0 << 13) +#define RT5651_MIC1_CLK_EN (0x1 << 13) +#define RT5651_MIC1_OVCD_MASK (0x1 << 11) +#define RT5651_MIC1_OVCD_SFT 11 +#define RT5651_MIC1_OVCD_DIS (0x0 << 11) +#define RT5651_MIC1_OVCD_EN (0x1 << 11) +#define RT5651_MIC1_OVTH_MASK (0x3 << 9) +#define RT5651_MIC1_OVTH_SFT 9 +#define RT5651_MIC1_OVTH_600UA (0x0 << 9) +#define RT5651_MIC1_OVTH_1500UA (0x1 << 9) +#define RT5651_MIC1_OVTH_2000UA (0x2 << 9) +#define RT5651_PWR_MB_MASK (0x1 << 5) +#define RT5651_PWR_MB_SFT 5 +#define RT5651_PWR_MB_PD (0x0 << 5) +#define RT5651_PWR_MB_PU (0x1 << 5) +#define RT5651_PWR_CLK12M_MASK (0x1 << 4) +#define RT5651_PWR_CLK12M_SFT 4 +#define RT5651_PWR_CLK12M_PD (0x0 << 4) +#define RT5651_PWR_CLK12M_PU (0x1 << 4) + +/* Analog JD Control 1 (0x94) */ +#define RT5651_JD2_CMP_MASK (0x7 << 12) +#define RT5651_JD2_CMP_SFT 12 +#define RT5651_JD_PU (0x1 << 11) +#define RT5651_JD_PU_SFT 11 +#define RT5651_JD_PD (0x1 << 10) +#define RT5651_JD_PD_SFT 10 +#define RT5651_JD_MODE_SEL_MASK (0x3 << 8) +#define RT5651_JD_MODE_SEL_SFT 8 +#define RT5651_JD_MODE_SEL_M0 (0x0 << 8) +#define RT5651_JD_MODE_SEL_M1 (0x1 << 8) +#define RT5651_JD_MODE_SEL_M2 (0x2 << 8) +#define RT5651_JD_M_CMP (0x7 << 4) +#define RT5651_JD_M_CMP_SFT 4 +#define RT5651_JD_M_PU (0x1 << 3) +#define RT5651_JD_M_PU_SFT 3 +#define RT5651_JD_M_PD (0x1 << 2) +#define RT5651_JD_M_PD_SFT 2 +#define RT5651_JD_M_MODE_SEL_MASK (0x3) +#define RT5651_JD_M_MODE_SEL_SFT 0 +#define RT5651_JD_M_MODE_SEL_M0 (0x0) +#define RT5651_JD_M_MODE_SEL_M1 (0x1) +#define RT5651_JD_M_MODE_SEL_M2 (0x2) + +/* Analog JD Control 2 (0x95) */ +#define RT5651_JD3_CMP_MASK (0x7 << 12) +#define RT5651_JD3_CMP_SFT 12 + +/* EQ Control 1 (0xb0) */ +#define RT5651_EQ_SRC_MASK (0x1 << 15) +#define RT5651_EQ_SRC_SFT 15 +#define RT5651_EQ_SRC_DAC (0x0 << 15) +#define RT5651_EQ_SRC_ADC (0x1 << 15) +#define RT5651_EQ_UPD (0x1 << 14) +#define RT5651_EQ_UPD_BIT 14 +#define RT5651_EQ_CD_MASK (0x1 << 13) +#define RT5651_EQ_CD_SFT 13 +#define RT5651_EQ_CD_DIS (0x0 << 13) +#define RT5651_EQ_CD_EN (0x1 << 13) +#define RT5651_EQ_DITH_MASK (0x3 << 8) +#define RT5651_EQ_DITH_SFT 8 +#define RT5651_EQ_DITH_NOR (0x0 << 8) +#define RT5651_EQ_DITH_LSB (0x1 << 8) +#define RT5651_EQ_DITH_LSB_1 (0x2 << 8) +#define RT5651_EQ_DITH_LSB_2 (0x3 << 8) +#define RT5651_EQ_CD_F (0x1 << 7) +#define RT5651_EQ_CD_F_BIT 7 +#define RT5651_EQ_STA_HP2 (0x1 << 6) +#define RT5651_EQ_STA_HP2_BIT 6 +#define RT5651_EQ_STA_HP1 (0x1 << 5) +#define RT5651_EQ_STA_HP1_BIT 5 +#define RT5651_EQ_STA_BP4 (0x1 << 4) +#define RT5651_EQ_STA_BP4_BIT 4 +#define RT5651_EQ_STA_BP3 (0x1 << 3) +#define RT5651_EQ_STA_BP3_BIT 3 +#define RT5651_EQ_STA_BP2 (0x1 << 2) +#define RT5651_EQ_STA_BP2_BIT 2 +#define RT5651_EQ_STA_BP1 (0x1 << 1) +#define RT5651_EQ_STA_BP1_BIT 1 +#define RT5651_EQ_STA_LP (0x1) +#define RT5651_EQ_STA_LP_BIT 0 + +/* EQ Control 2 (0xb1) */ +#define RT5651_EQ_HPF1_M_MASK (0x1 << 8) +#define RT5651_EQ_HPF1_M_SFT 8 +#define RT5651_EQ_HPF1_M_HI (0x0 << 8) +#define RT5651_EQ_HPF1_M_1ST (0x1 << 8) +#define RT5651_EQ_LPF1_M_MASK (0x1 << 7) +#define RT5651_EQ_LPF1_M_SFT 7 +#define RT5651_EQ_LPF1_M_LO (0x0 << 7) +#define RT5651_EQ_LPF1_M_1ST (0x1 << 7) +#define RT5651_EQ_HPF2_MASK (0x1 << 6) +#define RT5651_EQ_HPF2_SFT 6 +#define RT5651_EQ_HPF2_DIS (0x0 << 6) +#define RT5651_EQ_HPF2_EN (0x1 << 6) +#define RT5651_EQ_HPF1_MASK (0x1 << 5) +#define RT5651_EQ_HPF1_SFT 5 +#define RT5651_EQ_HPF1_DIS (0x0 << 5) +#define RT5651_EQ_HPF1_EN (0x1 << 5) +#define RT5651_EQ_BPF4_MASK (0x1 << 4) +#define RT5651_EQ_BPF4_SFT 4 +#define RT5651_EQ_BPF4_DIS (0x0 << 4) +#define RT5651_EQ_BPF4_EN (0x1 << 4) +#define RT5651_EQ_BPF3_MASK (0x1 << 3) +#define RT5651_EQ_BPF3_SFT 3 +#define RT5651_EQ_BPF3_DIS (0x0 << 3) +#define RT5651_EQ_BPF3_EN (0x1 << 3) +#define RT5651_EQ_BPF2_MASK (0x1 << 2) +#define RT5651_EQ_BPF2_SFT 2 +#define RT5651_EQ_BPF2_DIS (0x0 << 2) +#define RT5651_EQ_BPF2_EN (0x1 << 2) +#define RT5651_EQ_BPF1_MASK (0x1 << 1) +#define RT5651_EQ_BPF1_SFT 1 +#define RT5651_EQ_BPF1_DIS (0x0 << 1) +#define RT5651_EQ_BPF1_EN (0x1 << 1) +#define RT5651_EQ_LPF_MASK (0x1) +#define RT5651_EQ_LPF_SFT 0 +#define RT5651_EQ_LPF_DIS (0x0) +#define RT5651_EQ_LPF_EN (0x1) +#define RT5651_EQ_CTRL_MASK (0x7f) + +/* Memory Test (0xb2) */ +#define RT5651_MT_MASK (0x1 << 15) +#define RT5651_MT_SFT 15 +#define RT5651_MT_DIS (0x0 << 15) +#define RT5651_MT_EN (0x1 << 15) + +/* ALC Control 1 (0xb4) */ +#define RT5651_ALC_P_MASK (0x1 << 15) +#define RT5651_ALC_P_SFT 15 +#define RT5651_ALC_P_DAC (0x0 << 15) +#define RT5651_ALC_P_ADC (0x1 << 15) +#define RT5651_ALC_MASK (0x1 << 14) +#define RT5651_ALC_SFT 14 +#define RT5651_ALC_DIS (0x0 << 14) +#define RT5651_ALC_EN (0x1 << 14) +#define RT5651_ALC_UPD (0x1 << 13) +#define RT5651_ALC_UPD_BIT 13 +#define RT5651_ALC_AR_MASK (0x1f << 8) +#define RT5651_ALC_AR_SFT 8 +#define RT5651_ALC_R_MASK (0x7 << 5) +#define RT5651_ALC_R_SFT 5 +#define RT5651_ALC_R_48K (0x1 << 5) +#define RT5651_ALC_R_96K (0x2 << 5) +#define RT5651_ALC_R_192K (0x3 << 5) +#define RT5651_ALC_R_441K (0x5 << 5) +#define RT5651_ALC_R_882K (0x6 << 5) +#define RT5651_ALC_R_1764K (0x7 << 5) +#define RT5651_ALC_RC_MASK (0x1f) +#define RT5651_ALC_RC_SFT 0 + +/* ALC Control 2 (0xb5) */ +#define RT5651_ALC_POB_MASK (0x3f << 8) +#define RT5651_ALC_POB_SFT 8 +#define RT5651_ALC_DRC_MASK (0x1 << 7) +#define RT5651_ALC_DRC_SFT 7 +#define RT5651_ALC_DRC_DIS (0x0 << 7) +#define RT5651_ALC_DRC_EN (0x1 << 7) +#define RT5651_ALC_CPR_MASK (0x3 << 5) +#define RT5651_ALC_CPR_SFT 5 +#define RT5651_ALC_CPR_1_1 (0x0 << 5) +#define RT5651_ALC_CPR_1_2 (0x1 << 5) +#define RT5651_ALC_CPR_1_4 (0x2 << 5) +#define RT5651_ALC_CPR_1_8 (0x3 << 5) +#define RT5651_ALC_PRB_MASK (0x1f) +#define RT5651_ALC_PRB_SFT 0 + +/* ALC Control 3 (0xb6) */ +#define RT5651_ALC_NGB_MASK (0xf << 12) +#define RT5651_ALC_NGB_SFT 12 +#define RT5651_ALC_TAR_MASK (0x1f << 7) +#define RT5651_ALC_TAR_SFT 7 +#define RT5651_ALC_NG_MASK (0x1 << 6) +#define RT5651_ALC_NG_SFT 6 +#define RT5651_ALC_NG_DIS (0x0 << 6) +#define RT5651_ALC_NG_EN (0x1 << 6) +#define RT5651_ALC_NGH_MASK (0x1 << 5) +#define RT5651_ALC_NGH_SFT 5 +#define RT5651_ALC_NGH_DIS (0x0 << 5) +#define RT5651_ALC_NGH_EN (0x1 << 5) +#define RT5651_ALC_NGT_MASK (0x1f) +#define RT5651_ALC_NGT_SFT 0 + +/* Jack Detect Control 1 (0xbb) */ +#define RT5651_JD_MASK (0x7 << 13) +#define RT5651_JD_SFT 13 +#define RT5651_JD_DIS (0x0 << 13) +#define RT5651_JD_GPIO1 (0x1 << 13) +#define RT5651_JD_GPIO2 (0x2 << 13) +#define RT5651_JD_GPIO3 (0x3 << 13) +#define RT5651_JD_GPIO4 (0x4 << 13) +#define RT5651_JD_GPIO5 (0x5 << 13) +#define RT5651_JD_GPIO6 (0x6 << 13) +#define RT5651_JD_HP_MASK (0x1 << 11) +#define RT5651_JD_HP_SFT 11 +#define RT5651_JD_HP_DIS (0x0 << 11) +#define RT5651_JD_HP_EN (0x1 << 11) +#define RT5651_JD_HP_TRG_MASK (0x1 << 10) +#define RT5651_JD_HP_TRG_SFT 10 +#define RT5651_JD_HP_TRG_LO (0x0 << 10) +#define RT5651_JD_HP_TRG_HI (0x1 << 10) +#define RT5651_JD_SPL_MASK (0x1 << 9) +#define RT5651_JD_SPL_SFT 9 +#define RT5651_JD_SPL_DIS (0x0 << 9) +#define RT5651_JD_SPL_EN (0x1 << 9) +#define RT5651_JD_SPL_TRG_MASK (0x1 << 8) +#define RT5651_JD_SPL_TRG_SFT 8 +#define RT5651_JD_SPL_TRG_LO (0x0 << 8) +#define RT5651_JD_SPL_TRG_HI (0x1 << 8) +#define RT5651_JD_SPR_MASK (0x1 << 7) +#define RT5651_JD_SPR_SFT 7 +#define RT5651_JD_SPR_DIS (0x0 << 7) +#define RT5651_JD_SPR_EN (0x1 << 7) +#define RT5651_JD_SPR_TRG_MASK (0x1 << 6) +#define RT5651_JD_SPR_TRG_SFT 6 +#define RT5651_JD_SPR_TRG_LO (0x0 << 6) +#define RT5651_JD_SPR_TRG_HI (0x1 << 6) +#define RT5651_JD_LO_MASK (0x1 << 3) +#define RT5651_JD_LO_SFT 3 +#define RT5651_JD_LO_DIS (0x0 << 3) +#define RT5651_JD_LO_EN (0x1 << 3) +#define RT5651_JD_LO_TRG_MASK (0x1 << 2) +#define RT5651_JD_LO_TRG_SFT 2 +#define RT5651_JD_LO_TRG_LO (0x0 << 2) +#define RT5651_JD_LO_TRG_HI (0x1 << 2) + +/* Jack Detect Control 2 (0xbc) */ +#define RT5651_JD_TRG_SEL_MASK (0x7 << 9) +#define RT5651_JD_TRG_SEL_SFT 9 +#define RT5651_JD_TRG_SEL_GPIO (0x0 << 9) +#define RT5651_JD_TRG_SEL_JD1_1 (0x1 << 9) +#define RT5651_JD_TRG_SEL_JD1_2 (0x2 << 9) +#define RT5651_JD_TRG_SEL_JD2 (0x3 << 9) +#define RT5651_JD_TRG_SEL_JD3 (0x4 << 9) +#define RT5651_JD3_IRQ_EN (0x1 << 8) +#define RT5651_JD3_IRQ_EN_SFT 8 +#define RT5651_JD3_EN_STKY (0x1 << 7) +#define RT5651_JD3_EN_STKY_SFT 7 +#define RT5651_JD3_INV (0x1 << 6) +#define RT5651_JD3_INV_SFT 6 + +/* IRQ Control 1 (0xbd) */ +#define RT5651_IRQ_JD_MASK (0x1 << 15) +#define RT5651_IRQ_JD_SFT 15 +#define RT5651_IRQ_JD_BP (0x0 << 15) +#define RT5651_IRQ_JD_NOR (0x1 << 15) +#define RT5651_JD_STKY_MASK (0x1 << 13) +#define RT5651_JD_STKY_SFT 13 +#define RT5651_JD_STKY_DIS (0x0 << 13) +#define RT5651_JD_STKY_EN (0x1 << 13) +#define RT5651_JD_P_MASK (0x1 << 11) +#define RT5651_JD_P_SFT 11 +#define RT5651_JD_P_NOR (0x0 << 11) +#define RT5651_JD_P_INV (0x1 << 11) +#define RT5651_JD1_1_IRQ_EN (0x1 << 9) +#define RT5651_JD1_1_IRQ_EN_SFT 9 +#define RT5651_JD1_1_EN_STKY (0x1 << 8) +#define RT5651_JD1_1_EN_STKY_SFT 8 +#define RT5651_JD1_1_INV (0x1 << 7) +#define RT5651_JD1_1_INV_SFT 7 +#define RT5651_JD1_2_IRQ_EN (0x1 << 6) +#define RT5651_JD1_2_IRQ_EN_SFT 6 +#define RT5651_JD1_2_EN_STKY (0x1 << 5) +#define RT5651_JD1_2_EN_STKY_SFT 5 +#define RT5651_JD1_2_INV (0x1 << 4) +#define RT5651_JD1_2_INV_SFT 4 +#define RT5651_JD2_IRQ_EN (0x1 << 3) +#define RT5651_JD2_IRQ_EN_SFT 3 +#define RT5651_JD2_EN_STKY (0x1 << 2) +#define RT5651_JD2_EN_STKY_SFT 2 +#define RT5651_JD2_INV (0x1 << 1) +#define RT5651_JD2_INV_SFT 1 + +/* IRQ Control 2 (0xbe) */ +#define RT5651_IRQ_MB1_OC_MASK (0x1 << 15) +#define RT5651_IRQ_MB1_OC_SFT 15 +#define RT5651_IRQ_MB1_OC_BP (0x0 << 15) +#define RT5651_IRQ_MB1_OC_NOR (0x1 << 15) +#define RT5651_MB1_OC_STKY_MASK (0x1 << 11) +#define RT5651_MB1_OC_STKY_SFT 11 +#define RT5651_MB1_OC_STKY_DIS (0x0 << 11) +#define RT5651_MB1_OC_STKY_EN (0x1 << 11) +#define RT5651_MB1_OC_P_MASK (0x1 << 7) +#define RT5651_MB1_OC_P_SFT 7 +#define RT5651_MB1_OC_P_NOR (0x0 << 7) +#define RT5651_MB1_OC_P_INV (0x1 << 7) +#define RT5651_MB2_OC_P_MASK (0x1 << 6) +#define RT5651_MB1_OC_CLR (0x1 << 3) +#define RT5651_MB1_OC_CLR_SFT 3 +#define RT5651_STA_GPIO8 (0x1) +#define RT5651_STA_GPIO8_BIT 0 + +/* Internal Status and GPIO status (0xbf) */ +#define RT5651_STA_JD3 (0x1 << 15) +#define RT5651_STA_JD3_BIT 15 +#define RT5651_STA_JD2 (0x1 << 14) +#define RT5651_STA_JD2_BIT 14 +#define RT5651_STA_JD1_2 (0x1 << 13) +#define RT5651_STA_JD1_2_BIT 13 +#define RT5651_STA_JD1_1 (0x1 << 12) +#define RT5651_STA_JD1_1_BIT 12 +#define RT5651_STA_GP7 (0x1 << 11) +#define RT5651_STA_GP7_BIT 11 +#define RT5651_STA_GP6 (0x1 << 10) +#define RT5651_STA_GP6_BIT 10 +#define RT5651_STA_GP5 (0x1 << 9) +#define RT5651_STA_GP5_BIT 9 +#define RT5651_STA_GP1 (0x1 << 8) +#define RT5651_STA_GP1_BIT 8 +#define RT5651_STA_GP2 (0x1 << 7) +#define RT5651_STA_GP2_BIT 7 +#define RT5651_STA_GP3 (0x1 << 6) +#define RT5651_STA_GP3_BIT 6 +#define RT5651_STA_GP4 (0x1 << 5) +#define RT5651_STA_GP4_BIT 5 +#define RT5651_STA_GP_JD (0x1 << 4) +#define RT5651_STA_GP_JD_BIT 4 + +/* GPIO Control 1 (0xc0) */ +#define RT5651_GP1_PIN_MASK (0x1 << 15) +#define RT5651_GP1_PIN_SFT 15 +#define RT5651_GP1_PIN_GPIO1 (0x0 << 15) +#define RT5651_GP1_PIN_IRQ (0x1 << 15) +#define RT5651_GP2_PIN_MASK (0x1 << 14) +#define RT5651_GP2_PIN_SFT 14 +#define RT5651_GP2_PIN_GPIO2 (0x0 << 14) +#define RT5651_GP2_PIN_DMIC1_SCL (0x1 << 14) +#define RT5651_GPIO_M_MASK (0x1 << 9) +#define RT5651_GPIO_M_SFT 9 +#define RT5651_GPIO_M_FLT (0x0 << 9) +#define RT5651_GPIO_M_PH (0x1 << 9) +#define RT5651_I2S2_SEL_MASK (0x1 << 8) +#define RT5651_I2S2_SEL_SFT 8 +#define RT5651_I2S2_SEL_I2S (0x0 << 8) +#define RT5651_I2S2_SEL_GPIO (0x1 << 8) +#define RT5651_GP5_PIN_MASK (0x1 << 7) +#define RT5651_GP5_PIN_SFT 7 +#define RT5651_GP5_PIN_GPIO5 (0x0 << 7) +#define RT5651_GP5_PIN_IRQ (0x1 << 7) +#define RT5651_GP6_PIN_MASK (0x1 << 6) +#define RT5651_GP6_PIN_SFT 6 +#define RT5651_GP6_PIN_GPIO6 (0x0 << 6) +#define RT5651_GP6_PIN_DMIC_SDA (0x1 << 6) +#define RT5651_GP7_PIN_MASK (0x1 << 5) +#define RT5651_GP7_PIN_SFT 5 +#define RT5651_GP7_PIN_GPIO7 (0x0 << 5) +#define RT5651_GP7_PIN_IRQ (0x1 << 5) +#define RT5651_GP8_PIN_MASK (0x1 << 4) +#define RT5651_GP8_PIN_SFT 4 +#define RT5651_GP8_PIN_GPIO8 (0x0 << 4) +#define RT5651_GP8_PIN_DMIC_SDA (0x1 << 4) +#define RT5651_GPIO_PDM_SEL_MASK (0x1 << 3) +#define RT5651_GPIO_PDM_SEL_SFT 3 +#define RT5651_GPIO_PDM_SEL_GPIO (0x0 << 3) +#define RT5651_GPIO_PDM_SEL_PDM (0x1 << 3) + +/* GPIO Control 2 (0xc1) */ +#define RT5651_GP5_DR_MASK (0x1 << 14) +#define RT5651_GP5_DR_SFT 14 +#define RT5651_GP5_DR_IN (0x0 << 14) +#define RT5651_GP5_DR_OUT (0x1 << 14) +#define RT5651_GP5_OUT_MASK (0x1 << 13) +#define RT5651_GP5_OUT_SFT 13 +#define RT5651_GP5_OUT_LO (0x0 << 13) +#define RT5651_GP5_OUT_HI (0x1 << 13) +#define RT5651_GP5_P_MASK (0x1 << 12) +#define RT5651_GP5_P_SFT 12 +#define RT5651_GP5_P_NOR (0x0 << 12) +#define RT5651_GP5_P_INV (0x1 << 12) +#define RT5651_GP4_DR_MASK (0x1 << 11) +#define RT5651_GP4_DR_SFT 11 +#define RT5651_GP4_DR_IN (0x0 << 11) +#define RT5651_GP4_DR_OUT (0x1 << 11) +#define RT5651_GP4_OUT_MASK (0x1 << 10) +#define RT5651_GP4_OUT_SFT 10 +#define RT5651_GP4_OUT_LO (0x0 << 10) +#define RT5651_GP4_OUT_HI (0x1 << 10) +#define RT5651_GP4_P_MASK (0x1 << 9) +#define RT5651_GP4_P_SFT 9 +#define RT5651_GP4_P_NOR (0x0 << 9) +#define RT5651_GP4_P_INV (0x1 << 9) +#define RT5651_GP3_DR_MASK (0x1 << 8) +#define RT5651_GP3_DR_SFT 8 +#define RT5651_GP3_DR_IN (0x0 << 8) +#define RT5651_GP3_DR_OUT (0x1 << 8) +#define RT5651_GP3_OUT_MASK (0x1 << 7) +#define RT5651_GP3_OUT_SFT 7 +#define RT5651_GP3_OUT_LO (0x0 << 7) +#define RT5651_GP3_OUT_HI (0x1 << 7) +#define RT5651_GP3_P_MASK (0x1 << 6) +#define RT5651_GP3_P_SFT 6 +#define RT5651_GP3_P_NOR (0x0 << 6) +#define RT5651_GP3_P_INV (0x1 << 6) +#define RT5651_GP2_DR_MASK (0x1 << 5) +#define RT5651_GP2_DR_SFT 5 +#define RT5651_GP2_DR_IN (0x0 << 5) +#define RT5651_GP2_DR_OUT (0x1 << 5) +#define RT5651_GP2_OUT_MASK (0x1 << 4) +#define RT5651_GP2_OUT_SFT 4 +#define RT5651_GP2_OUT_LO (0x0 << 4) +#define RT5651_GP2_OUT_HI (0x1 << 4) +#define RT5651_GP2_P_MASK (0x1 << 3) +#define RT5651_GP2_P_SFT 3 +#define RT5651_GP2_P_NOR (0x0 << 3) +#define RT5651_GP2_P_INV (0x1 << 3) +#define RT5651_GP1_DR_MASK (0x1 << 2) +#define RT5651_GP1_DR_SFT 2 +#define RT5651_GP1_DR_IN (0x0 << 2) +#define RT5651_GP1_DR_OUT (0x1 << 2) +#define RT5651_GP1_OUT_MASK (0x1 << 1) +#define RT5651_GP1_OUT_SFT 1 +#define RT5651_GP1_OUT_LO (0x0 << 1) +#define RT5651_GP1_OUT_HI (0x1 << 1) +#define RT5651_GP1_P_MASK (0x1) +#define RT5651_GP1_P_SFT 0 +#define RT5651_GP1_P_NOR (0x0) +#define RT5651_GP1_P_INV (0x1) + +/* GPIO Control 3 (0xc2) */ +#define RT5651_GP8_DR_MASK (0x1 << 8) +#define RT5651_GP8_DR_SFT 8 +#define RT5651_GP8_DR_IN (0x0 << 8) +#define RT5651_GP8_DR_OUT (0x1 << 8) +#define RT5651_GP8_OUT_MASK (0x1 << 7) +#define RT5651_GP8_OUT_SFT 7 +#define RT5651_GP8_OUT_LO (0x0 << 7) +#define RT5651_GP8_OUT_HI (0x1 << 7) +#define RT5651_GP8_P_MASK (0x1 << 6) +#define RT5651_GP8_P_SFT 6 +#define RT5651_GP8_P_NOR (0x0 << 6) +#define RT5651_GP8_P_INV (0x1 << 6) +#define RT5651_GP7_DR_MASK (0x1 << 5) +#define RT5651_GP7_DR_SFT 5 +#define RT5651_GP7_DR_IN (0x0 << 5) +#define RT5651_GP7_DR_OUT (0x1 << 5) +#define RT5651_GP7_OUT_MASK (0x1 << 4) +#define RT5651_GP7_OUT_SFT 4 +#define RT5651_GP7_OUT_LO (0x0 << 4) +#define RT5651_GP7_OUT_HI (0x1 << 4) +#define RT5651_GP7_P_MASK (0x1 << 3) +#define RT5651_GP7_P_SFT 3 +#define RT5651_GP7_P_NOR (0x0 << 3) +#define RT5651_GP7_P_INV (0x1 << 3) +#define RT5651_GP6_DR_MASK (0x1 << 2) +#define RT5651_GP6_DR_SFT 2 +#define RT5651_GP6_DR_IN (0x0 << 2) +#define RT5651_GP6_DR_OUT (0x1 << 2) +#define RT5651_GP6_OUT_MASK (0x1 << 1) +#define RT5651_GP6_OUT_SFT 1 +#define RT5651_GP6_OUT_LO (0x0 << 1) +#define RT5651_GP6_OUT_HI (0x1 << 1) +#define RT5651_GP6_P_MASK (0x1) +#define RT5651_GP6_P_SFT 0 +#define RT5651_GP6_P_NOR (0x0) +#define RT5651_GP6_P_INV (0x1) + +/* Scramble Control (0xce) */ +#define RT5651_SCB_SWAP_MASK (0x1 << 15) +#define RT5651_SCB_SWAP_SFT 15 +#define RT5651_SCB_SWAP_DIS (0x0 << 15) +#define RT5651_SCB_SWAP_EN (0x1 << 15) +#define RT5651_SCB_MASK (0x1 << 14) +#define RT5651_SCB_SFT 14 +#define RT5651_SCB_DIS (0x0 << 14) +#define RT5651_SCB_EN (0x1 << 14) + +/* Baseback Control (0xcf) */ +#define RT5651_BB_MASK (0x1 << 15) +#define RT5651_BB_SFT 15 +#define RT5651_BB_DIS (0x0 << 15) +#define RT5651_BB_EN (0x1 << 15) +#define RT5651_BB_CT_MASK (0x7 << 12) +#define RT5651_BB_CT_SFT 12 +#define RT5651_BB_CT_A (0x0 << 12) +#define RT5651_BB_CT_B (0x1 << 12) +#define RT5651_BB_CT_C (0x2 << 12) +#define RT5651_BB_CT_D (0x3 << 12) +#define RT5651_M_BB_L_MASK (0x1 << 9) +#define RT5651_M_BB_L_SFT 9 +#define RT5651_M_BB_R_MASK (0x1 << 8) +#define RT5651_M_BB_R_SFT 8 +#define RT5651_M_BB_HPF_L_MASK (0x1 << 7) +#define RT5651_M_BB_HPF_L_SFT 7 +#define RT5651_M_BB_HPF_R_MASK (0x1 << 6) +#define RT5651_M_BB_HPF_R_SFT 6 +#define RT5651_G_BB_BST_MASK (0x3f) +#define RT5651_G_BB_BST_SFT 0 + +/* MP3 Plus Control 1 (0xd0) */ +#define RT5651_M_MP3_L_MASK (0x1 << 15) +#define RT5651_M_MP3_L_SFT 15 +#define RT5651_M_MP3_R_MASK (0x1 << 14) +#define RT5651_M_MP3_R_SFT 14 +#define RT5651_M_MP3_MASK (0x1 << 13) +#define RT5651_M_MP3_SFT 13 +#define RT5651_M_MP3_DIS (0x0 << 13) +#define RT5651_M_MP3_EN (0x1 << 13) +#define RT5651_EG_MP3_MASK (0x1f << 8) +#define RT5651_EG_MP3_SFT 8 +#define RT5651_MP3_HLP_MASK (0x1 << 7) +#define RT5651_MP3_HLP_SFT 7 +#define RT5651_MP3_HLP_DIS (0x0 << 7) +#define RT5651_MP3_HLP_EN (0x1 << 7) +#define RT5651_M_MP3_ORG_L_MASK (0x1 << 6) +#define RT5651_M_MP3_ORG_L_SFT 6 +#define RT5651_M_MP3_ORG_R_MASK (0x1 << 5) +#define RT5651_M_MP3_ORG_R_SFT 5 + +/* MP3 Plus Control 2 (0xd1) */ +#define RT5651_MP3_WT_MASK (0x1 << 13) +#define RT5651_MP3_WT_SFT 13 +#define RT5651_MP3_WT_1_4 (0x0 << 13) +#define RT5651_MP3_WT_1_2 (0x1 << 13) +#define RT5651_OG_MP3_MASK (0x1f << 8) +#define RT5651_OG_MP3_SFT 8 +#define RT5651_HG_MP3_MASK (0x3f) +#define RT5651_HG_MP3_SFT 0 + +/* 3D HP Control 1 (0xd2) */ +#define RT5651_3D_CF_MASK (0x1 << 15) +#define RT5651_3D_CF_SFT 15 +#define RT5651_3D_CF_DIS (0x0 << 15) +#define RT5651_3D_CF_EN (0x1 << 15) +#define RT5651_3D_HP_MASK (0x1 << 14) +#define RT5651_3D_HP_SFT 14 +#define RT5651_3D_HP_DIS (0x0 << 14) +#define RT5651_3D_HP_EN (0x1 << 14) +#define RT5651_3D_BT_MASK (0x1 << 13) +#define RT5651_3D_BT_SFT 13 +#define RT5651_3D_BT_DIS (0x0 << 13) +#define RT5651_3D_BT_EN (0x1 << 13) +#define RT5651_3D_1F_MIX_MASK (0x3 << 11) +#define RT5651_3D_1F_MIX_SFT 11 +#define RT5651_3D_HP_M_MASK (0x1 << 10) +#define RT5651_3D_HP_M_SFT 10 +#define RT5651_3D_HP_M_SUR (0x0 << 10) +#define RT5651_3D_HP_M_FRO (0x1 << 10) +#define RT5651_M_3D_HRTF_MASK (0x1 << 9) +#define RT5651_M_3D_HRTF_SFT 9 +#define RT5651_M_3D_D2H_MASK (0x1 << 8) +#define RT5651_M_3D_D2H_SFT 8 +#define RT5651_M_3D_D2R_MASK (0x1 << 7) +#define RT5651_M_3D_D2R_SFT 7 +#define RT5651_M_3D_REVB_MASK (0x1 << 6) +#define RT5651_M_3D_REVB_SFT 6 + +/* Adjustable high pass filter control 1 (0xd3) */ +#define RT5651_2ND_HPF_MASK (0x1 << 15) +#define RT5651_2ND_HPF_SFT 15 +#define RT5651_2ND_HPF_DIS (0x0 << 15) +#define RT5651_2ND_HPF_EN (0x1 << 15) +#define RT5651_HPF_CF_L_MASK (0x7 << 12) +#define RT5651_HPF_CF_L_SFT 12 +#define RT5651_HPF_CF_R_MASK (0x7 << 8) +#define RT5651_HPF_CF_R_SFT 8 +#define RT5651_ZD_T_MASK (0x3 << 6) +#define RT5651_ZD_T_SFT 6 +#define RT5651_ZD_F_MASK (0x3 << 4) +#define RT5651_ZD_F_SFT 4 +#define RT5651_ZD_F_IM (0x0 << 4) +#define RT5651_ZD_F_ZC_IM (0x1 << 4) +#define RT5651_ZD_F_ZC_IOD (0x2 << 4) +#define RT5651_ZD_F_UN (0x3 << 4) + +/* Adjustable high pass filter control 2 (0xd4) */ +#define RT5651_HPF_CF_L_NUM_MASK (0x3f << 8) +#define RT5651_HPF_CF_L_NUM_SFT 8 +#define RT5651_HPF_CF_R_NUM_MASK (0x3f) +#define RT5651_HPF_CF_R_NUM_SFT 0 + +/* HP calibration control and Amp detection (0xd6) */ +#define RT5651_SI_DAC_MASK (0x1 << 11) +#define RT5651_SI_DAC_SFT 11 +#define RT5651_SI_DAC_AUTO (0x0 << 11) +#define RT5651_SI_DAC_TEST (0x1 << 11) +#define RT5651_DC_CAL_M_MASK (0x1 << 10) +#define RT5651_DC_CAL_M_SFT 10 +#define RT5651_DC_CAL_M_NOR (0x0 << 10) +#define RT5651_DC_CAL_M_CAL (0x1 << 10) +#define RT5651_DC_CAL_MASK (0x1 << 9) +#define RT5651_DC_CAL_SFT 9 +#define RT5651_DC_CAL_DIS (0x0 << 9) +#define RT5651_DC_CAL_EN (0x1 << 9) +#define RT5651_HPD_RCV_MASK (0x7 << 6) +#define RT5651_HPD_RCV_SFT 6 +#define RT5651_HPD_PS_MASK (0x1 << 5) +#define RT5651_HPD_PS_SFT 5 +#define RT5651_HPD_PS_DIS (0x0 << 5) +#define RT5651_HPD_PS_EN (0x1 << 5) +#define RT5651_CAL_M_MASK (0x1 << 4) +#define RT5651_CAL_M_SFT 4 +#define RT5651_CAL_M_DEP (0x0 << 4) +#define RT5651_CAL_M_CAL (0x1 << 4) +#define RT5651_CAL_MASK (0x1 << 3) +#define RT5651_CAL_SFT 3 +#define RT5651_CAL_DIS (0x0 << 3) +#define RT5651_CAL_EN (0x1 << 3) +#define RT5651_CAL_TEST_MASK (0x1 << 2) +#define RT5651_CAL_TEST_SFT 2 +#define RT5651_CAL_TEST_DIS (0x0 << 2) +#define RT5651_CAL_TEST_EN (0x1 << 2) +#define RT5651_CAL_P_MASK (0x3) +#define RT5651_CAL_P_SFT 0 +#define RT5651_CAL_P_NONE (0x0) +#define RT5651_CAL_P_CAL (0x1) +#define RT5651_CAL_P_DAC_CAL (0x2) + +/* Soft volume and zero cross control 1 (0xd9) */ +#define RT5651_SV_MASK (0x1 << 15) +#define RT5651_SV_SFT 15 +#define RT5651_SV_DIS (0x0 << 15) +#define RT5651_SV_EN (0x1 << 15) +#define RT5651_OUT_SV_MASK (0x1 << 13) +#define RT5651_OUT_SV_SFT 13 +#define RT5651_OUT_SV_DIS (0x0 << 13) +#define RT5651_OUT_SV_EN (0x1 << 13) +#define RT5651_HP_SV_MASK (0x1 << 12) +#define RT5651_HP_SV_SFT 12 +#define RT5651_HP_SV_DIS (0x0 << 12) +#define RT5651_HP_SV_EN (0x1 << 12) +#define RT5651_ZCD_DIG_MASK (0x1 << 11) +#define RT5651_ZCD_DIG_SFT 11 +#define RT5651_ZCD_DIG_DIS (0x0 << 11) +#define RT5651_ZCD_DIG_EN (0x1 << 11) +#define RT5651_ZCD_MASK (0x1 << 10) +#define RT5651_ZCD_SFT 10 +#define RT5651_ZCD_PD (0x0 << 10) +#define RT5651_ZCD_PU (0x1 << 10) +#define RT5651_M_ZCD_MASK (0x3f << 4) +#define RT5651_M_ZCD_SFT 4 +#define RT5651_M_ZCD_OM_L (0x1 << 7) +#define RT5651_M_ZCD_OM_R (0x1 << 6) +#define RT5651_M_ZCD_RM_L (0x1 << 5) +#define RT5651_M_ZCD_RM_R (0x1 << 4) +#define RT5651_SV_DLY_MASK (0xf) +#define RT5651_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0xda) */ +#define RT5651_ZCD_HP_MASK (0x1 << 15) +#define RT5651_ZCD_HP_SFT 15 +#define RT5651_ZCD_HP_DIS (0x0 << 15) +#define RT5651_ZCD_HP_EN (0x1 << 15) + +/* Digital Misc Control (0xfa) */ +#define RT5651_I2S2_MS_SP_MASK (0x1 << 8) +#define RT5651_I2S2_MS_SP_SEL 8 +#define RT5651_I2S2_MS_SP_64 (0x0 << 8) +#define RT5651_I2S2_MS_SP_50 (0x1 << 8) +#define RT5651_CLK_DET_EN (0x1 << 3) +#define RT5651_CLK_DET_EN_SFT 3 +#define RT5651_AMP_DET_EN (0x1 << 1) +#define RT5651_AMP_DET_EN_SFT 1 +#define RT5651_D_GATE_EN (0x1) +#define RT5651_D_GATE_EN_SFT 0 + +/* Codec Private Register definition */ +/* 3D Speaker Control (0x63) */ +#define RT5651_3D_SPK_MASK (0x1 << 15) +#define RT5651_3D_SPK_SFT 15 +#define RT5651_3D_SPK_DIS (0x0 << 15) +#define RT5651_3D_SPK_EN (0x1 << 15) +#define RT5651_3D_SPK_M_MASK (0x3 << 13) +#define RT5651_3D_SPK_M_SFT 13 +#define RT5651_3D_SPK_CG_MASK (0x1f << 8) +#define RT5651_3D_SPK_CG_SFT 8 +#define RT5651_3D_SPK_SG_MASK (0x1f) +#define RT5651_3D_SPK_SG_SFT 0 + +/* Wind Noise Detection Control 1 (0x6c) */ +#define RT5651_WND_MASK (0x1 << 15) +#define RT5651_WND_SFT 15 +#define RT5651_WND_DIS (0x0 << 15) +#define RT5651_WND_EN (0x1 << 15) + +/* Wind Noise Detection Control 2 (0x6d) */ +#define RT5651_WND_FC_NW_MASK (0x3f << 10) +#define RT5651_WND_FC_NW_SFT 10 +#define RT5651_WND_FC_WK_MASK (0x3f << 4) +#define RT5651_WND_FC_WK_SFT 4 + +/* Wind Noise Detection Control 3 (0x6e) */ +#define RT5651_HPF_FC_MASK (0x3f << 6) +#define RT5651_HPF_FC_SFT 6 +#define RT5651_WND_FC_ST_MASK (0x3f) +#define RT5651_WND_FC_ST_SFT 0 + +/* Wind Noise Detection Control 4 (0x6f) */ +#define RT5651_WND_TH_LO_MASK (0x3ff) +#define RT5651_WND_TH_LO_SFT 0 + +/* Wind Noise Detection Control 5 (0x70) */ +#define RT5651_WND_TH_HI_MASK (0x3ff) +#define RT5651_WND_TH_HI_SFT 0 + +/* Wind Noise Detection Control 8 (0x73) */ +#define RT5651_WND_WIND_MASK (0x1 << 13) /* Read-Only */ +#define RT5651_WND_WIND_SFT 13 +#define RT5651_WND_STRONG_MASK (0x1 << 12) /* Read-Only */ +#define RT5651_WND_STRONG_SFT 12 +enum { + RT5651_NO_WIND, + RT5651_BREEZE, + RT5651_STORM, +}; + +/* Dipole Speaker Interface (0x75) */ +#define RT5651_DP_ATT_MASK (0x3 << 14) +#define RT5651_DP_ATT_SFT 14 +#define RT5651_DP_SPK_MASK (0x1 << 10) +#define RT5651_DP_SPK_SFT 10 +#define RT5651_DP_SPK_DIS (0x0 << 10) +#define RT5651_DP_SPK_EN (0x1 << 10) + +/* EQ Pre Volume Control (0xb3) */ +#define RT5651_EQ_PRE_VOL_MASK (0xffff) +#define RT5651_EQ_PRE_VOL_SFT 0 + +/* EQ Post Volume Control (0xb4) */ +#define RT5651_EQ_PST_VOL_MASK (0xffff) +#define RT5651_EQ_PST_VOL_SFT 0 + +/* System Clock Source */ +enum { + RT5651_SCLK_S_MCLK, + RT5651_SCLK_S_PLL1, + RT5651_SCLK_S_RCCLK, +}; + +/* PLL1 Source */ +enum { + RT5651_PLL1_S_MCLK, + RT5651_PLL1_S_BCLK1, + RT5651_PLL1_S_BCLK2, +}; + +enum { + RT5651_AIF1, + RT5651_AIF2, + RT5651_AIFS, +}; + +struct rt5651_pll_code { + bool m_bp; /* Indicates bypass m code or not. */ + int m_code; + int n_code; + int k_code; +}; + +struct rt5651_priv { + struct snd_soc_codec *codec; + struct rt5651_platform_data pdata; + struct regmap *regmap; + + int sysclk; + int sysclk_src; + int lrck[RT5651_AIFS]; + int bclk[RT5651_AIFS]; + int master[RT5651_AIFS]; + + struct rt5651_pll_code pll_code; + int pll_src; + int pll_in; + int pll_out; + + int dmic_en; + bool hp_mute; +}; + +#endif /* __RT5651_H__ */ -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: dapm: Fix widget double free with auto-disable DAPM kcontrol
by Jarkko Nikula 18 Apr '14

18 Apr '14

Commit 9e1fda4ae158 ("ASoC: dapm: Implement mixer input auto-disable") is trying to free the widget it allocated by snd_soc_dapm_new_control() call in dapm_kcontrol_data_alloc() by adding kfree(data->widget) to dapm_kcontrol_free(). This is causing a widget double free with auto-disabled DAPM kcontrols in sound card unregistration because widgets are already freed before dapm_kcontrol_free() is called. Reason for that is all widgets are added into dapm->card->widgets list in snd_soc_dapm_new_control() and freed in dapm_free_widgets() during execution of snd_soc_dapm_free(). Now snd_soc_dapm_free() calls for different DAPM contexts happens before snd_card_free() call from where the call chain to dapm_kcontrol_free() begins: soc_cleanup_card_resources() soc_remove_dai_links() soc_remove_link_dais() snd_soc_dapm_free(&cpu_dai->dapm) soc_remove_link_components() soc_remove_platform() snd_soc_dapm_free(&platform->dapm) soc_remove_codec() snd_soc_dapm_free(&codec->dapm) snd_soc_dapm_free(&card->dapm) snd_card_free() snd_card_do_free() snd_device_free_all() snd_device_free() snd_ctl_dev_free() snd_ctl_remove() snd_ctl_free_one() dapm_kcontrol_free() This wasn't making harm with ordinary DAPM kcontrols since data->widget is NULL for them. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Cc: stable <stable(a)vger.kernel.org> # 3.12+ --- sound/soc/soc-dapm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c8a780d0d057..7769b0a2bc5a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -254,7 +254,6 @@ static int dapm_kcontrol_data_alloc(struct snd_soc_dapm_widget *widget, static void dapm_kcontrol_free(struct snd_kcontrol *kctl) { struct dapm_kcontrol_data *data = snd_kcontrol_chip(kctl); - kfree(data->widget); kfree(data->wlist); kfree(data); } -- 1.9.1

3 2

[alsa-devel] [PATCH] ASoC: core: use PTR_ERR instead of PTR_RET
by Christoph Jaeger 18 Apr '14

18 Apr '14

PTR_RET is deprecated. PTR_ERR_OR_ZERO should be used instead. However, we already know that IS_ERR is true, and thus PTR_ERR_OR_ZERO would never yield zero, so we can use PTR_ERR here. Signed-off-by: Christoph Jaeger <christophjaeger(a)linux.com> --- sound/soc/soc-core.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 051c006..2dc69845 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2153,28 +2153,28 @@ static int snd_soc_ac97_parse_pinctl(struct device *dev, p = devm_pinctrl_get(dev); if (IS_ERR(p)) { dev_err(dev, "Failed to get pinctrl\n"); - return PTR_RET(p); + return PTR_ERR(p); } cfg->pctl = p; state = pinctrl_lookup_state(p, "ac97-reset"); if (IS_ERR(state)) { dev_err(dev, "Can't find pinctrl state ac97-reset\n"); - return PTR_RET(state); + return PTR_ERR(state); } cfg->pstate_reset = state; state = pinctrl_lookup_state(p, "ac97-warm-reset"); if (IS_ERR(state)) { dev_err(dev, "Can't find pinctrl state ac97-warm-reset\n"); - return PTR_RET(state); + return PTR_ERR(state); } cfg->pstate_warm_reset = state; state = pinctrl_lookup_state(p, "ac97-running"); if (IS_ERR(state)) { dev_err(dev, "Can't find pinctrl state ac97-running\n"); - return PTR_RET(state); + return PTR_ERR(state); } cfg->pstate_run = state; -- 1.9.0

2 1

[alsa-devel] [PATCH] ASoC: Intel: Fix incorrect sizeof() in sst_hsw_stream_get_volume()
by Christian Engelmayer 18 Apr '14

18 Apr '14

Fix an incorrect sizeof() usage in sst_hsw_stream_get_volume(). sst_dsp_read() is called to read into a variable of type u32, but is passed sizeof(u32 *) for argument 'size_t bytes'. Detected by Coverity: CID 1195260. Signed-off-by: Christian Engelmayer <cengelma(a)gmx.at> --- sound/soc/intel/sst-haswell-ipc.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index f46bb4d..455a185 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -991,7 +991,8 @@ int sst_hsw_stream_get_volume(struct sst_hsw *hsw, struct sst_hsw_stream *stream return -EINVAL; sst_dsp_read(hsw->dsp, volume, - stream->reply.volume_register_address[channel], sizeof(volume)); + stream->reply.volume_register_address[channel], + sizeof(*volume)); return 0; } -- 1.8.3.2

3 4

[alsa-devel] [PATCH] ASoC: cs42l51: Fix mask for REVID
by Axel Lin 18 Apr '14

18 Apr '14

The REVID mask was changed by commit a1253ef6d3fa "ASoC: cs42l51: split i2c from codec driver". Fix it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l51.c | 3 ++- sound/soc/codecs/cs42l51.h | 1 + 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index dadc20d..09488d9 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -548,7 +548,8 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap) ret = -ENODEV; goto error; } - dev_info(dev, "Cirrus Logic CS42L51, Revision: %02X\n", val & 0xFF); + dev_info(dev, "Cirrus Logic CS42L51, Revision: %02X\n", + val & CS42L51_CHIP_REV_MASK); ret = snd_soc_register_codec(dev, &soc_codec_device_cs42l51, &cs42l51_dai, 1); diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h index 641ef18..8c55bf3 100644 --- a/sound/soc/codecs/cs42l51.h +++ b/sound/soc/codecs/cs42l51.h @@ -26,6 +26,7 @@ int cs42l51_probe(struct device *dev, struct regmap *regmap); #define CS42L51_CHIP_ID 0x1B #define CS42L51_CHIP_REV_A 0x00 #define CS42L51_CHIP_REV_B 0x01 +#define CS42L51_CHIP_REV_MASK 0x07 #define CS42L51_CHIP_REV_ID 0x01 #define CS42L51_MK_CHIP_REV(a, b) ((a)<<3|(b)) -- 1.8.3.2

3 2

[alsa-devel] [PATCH] ASoC: DAPM: Allow update_bits use 32 bits reg
by bardliao＠realtek.com 18 Apr '14

18 Apr '14

From: Bard Liao <bardliao(a)realtek.com> This patch change reg's type from unsigned short to int. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c8a780d..f4ba7b4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -426,7 +426,7 @@ static void soc_dapm_async_complete(struct snd_soc_dapm_context *dapm) } static int soc_widget_update_bits_locked(struct snd_soc_dapm_widget *w, - unsigned short reg, unsigned int mask, unsigned int value) + int reg, unsigned int mask, unsigned int value) { bool change; unsigned int old, new; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 1/2] ASoC: cs42l52: Convert to use devm_gpio_request_one
by Axel Lin 18 Apr '14

18 Apr '14

Current code missed a gpio_free() call in cs42l52_i2c_remove(). Convert to use devm_gpio_request_one() to fix it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l52.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 460d355..2213a03 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1229,8 +1229,10 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, } if (cs42l52->pdata.reset_gpio) { - ret = gpio_request_one(cs42l52->pdata.reset_gpio, - GPIOF_OUT_INIT_HIGH, "CS42L52 /RST"); + ret = devm_gpio_request_one(&i2c_client->dev, + cs42l52->pdata.reset_gpio, + GPIOF_OUT_INIT_HIGH, + "CS42L52 /RST"); if (ret < 0) { dev_err(&i2c_client->dev, "Failed to request /RST %d: %d\n", cs42l52->pdata.reset_gpio, ret); -- 1.8.3.2

2 3

Re: [alsa-devel] [PATCH] ASoC: cs42l51: Fix SOC_DOUBLE_R_SX_TLV shift values for ADC, PCM, and Analog kcontrols
by Brian Austin 18 Apr '14

18 Apr '14

On Fri, 18 Apr 2014, Thomas Petazzoni wrote: > Simple mixer control 'Analog',0 > Capabilities: pvolume > Playback channels: Front Left - Front Right > Limits: Playback 0 - 177 This is wrong > Mono: > Front Left: Playback 204 [115%] [0.00dB] > Front Right: Playback 204 [115%] [0.00dB] root@beagleboard:~# amixer sget 'Analog' Simple mixer control 'Analog',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 229 Mono: Front Left: Playback 204 [89%] [0.00dB] Front Right: Playback 204 [89%] [0.00dB] root@beagleboard:~# i2cdump -y -f 1 0x4b No size specified (using byte-data access) 0 1 2 3 4 5 6 7 8 9 a b c d e f 0123456789abcdef 00: 00 d9 7e ae 00 00 a0 00 60 44 00 00 00 00 84 84 .?~?..?.`D....?? 10: 19 19 00 00 00 88 00 00 00 00 7f 00 01 3f 00 00 ??...?....?.??.. [...] As you can see the output of amixer is wrong on your system. There are 229 steps for gain on this control. At 89% we are at 0dB which is correct according to I2C Tools (0h) root@beagleboard:~# amixer sset 'Analog' 0 Simple mixer control 'Analog',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 229 Mono: Front Left: Playback 0 [0%] [-102.00dB] Front Right: Playback 0 [0%] [-102.00dB] root@beagleboard:~# i2cdump -y -f 1 0x4b No size specified (using byte-data access) 0 1 2 3 4 5 6 7 8 9 a b c d e f 0123456789abcdef 00: 00 d9 7e ae 00 00 a0 00 60 44 00 00 00 00 84 84 .?~?..?.`D....?? 10: 19 19 00 00 00 88 34 34 00 00 7f 00 01 3f 00 00 ??...?44..?.??.. -102dB gives the correct output and register writes as well. (34h) My regmap output maps correctly too. Weird. So lets get a few versions of some things to see what and where things went wrong. What kernel version? Alsa-lib version? What distro are you using? Thanks, Brian

2 1

[alsa-devel] [PATCH] ASoC: wm_adsp: Remove uneeded semicolon
by Charles Keepax 18 Apr '14

18 Apr '14

Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index d9686dc..ff15eec 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1625,7 +1625,7 @@ int wm_adsp2_early_event(struct snd_soc_dapm_widget *w, break; default: break; - }; + } return 0; } -- 1.7.2.5

2 1

[alsa-devel] [PATCH] ASoC: tlv320aic31xx: document that the regulators are mandatory
by Alexandre Belloni 18 Apr '14

18 Apr '14

The driver will not probe if regulators are not provided. Move the regulator properties to the required section. Signed-off-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- Documentation/devicetree/bindings/sound/tlv320aic31xx.txt | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt index 74c66dee3e14..eff12be5e789 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt @@ -13,6 +13,9 @@ Required properties: "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP) - reg - <int> - I2C slave address +- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply, + DVDD-supply : power supplies for the device as covered in + Documentation/devicetree/bindings/regulator/regulator.txt Optional properties: @@ -24,9 +27,6 @@ Optional properties: 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD If this node is not mentioned or if the value is unknown, then micbias is set to 2.0V. -- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply, - DVDD-supply : power supplies for the device as covered in - Documentation/devicetree/bindings/regulator/regulator.txt CODEC output pins: * HPL -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: fsl_spdif: Fix wrong OFFSET of STC_SYSCLK_DIV
by Nicolin Chen 18 Apr '14

18 Apr '14

It should use STC_SYSCLK_DIV_OFFSET. Thus fix it. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_spdif.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.h b/sound/soc/fsl/fsl_spdif.h index b126679..605a10b 100644 --- a/sound/soc/fsl/fsl_spdif.h +++ b/sound/soc/fsl/fsl_spdif.h @@ -144,8 +144,8 @@ enum spdif_gainsel { /* SPDIF Clock register */ #define STC_SYSCLK_DIV_OFFSET 11 -#define STC_SYSCLK_DIV_MASK (0x1ff << STC_TXCLK_SRC_OFFSET) -#define STC_SYSCLK_DIV(x) ((((x) - 1) << STC_TXCLK_DIV_OFFSET) & STC_SYSCLK_DIV_MASK) +#define STC_SYSCLK_DIV_MASK (0x1ff << STC_SYSCLK_DIV_OFFSET) +#define STC_SYSCLK_DIV(x) ((((x) - 1) << STC_SYSCLK_DIV_OFFSET) & STC_SYSCLK_DIV_MASK) #define STC_TXCLK_SRC_OFFSET 8 #define STC_TXCLK_SRC_MASK (0x7 << STC_TXCLK_SRC_OFFSET) #define STC_TXCLK_SRC_SET(x) ((x << STC_TXCLK_SRC_OFFSET) & STC_TXCLK_SRC_MASK) -- 1.8.4

2 1

Re: [alsa-devel] [PATCH] ASoC: cs42l51: Fix SOC_DOUBLE_R_SX_TLV shift values for ADC, PCM, and Analog kcontrols
by Austin, Brian 18 Apr '14

18 Apr '14

On Apr 18, 2014, at 4:23 AM, Thomas Petazzoni <thomas.petazzoni(a)free-electrons.com> wrote: > Dear Brian Austin, > > On Thu, 17 Apr 2014 10:08:23 -0500, Brian Austin wrote: > >> can you try this quick check please? >> >> static const struct snd_kcontrol_new cs42l51_snd_controls[] = { >> SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", >> CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, >> - 0, 0x19, 0x7F, adc_pcm_tlv), >> + 0, 0x19, 0x80, adc_pcm_tlv), >> SOC_DOUBLE_R("PCM Playback Switch", >> CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1), >> SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", >> CS42L51_AOUTA_VOL, CS42L51_AOUTB_VOL, >> - 0, 0x34, 0xE4, aout_tlv), >> + 0, 0x34, 0xE5, aout_tlv), >> SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", >> CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, >> - 0, 0x19, 0x7F, adc_pcm_tlv), >> + 0, 0x19, 0x80, adc_pcm_tlv), >> SOC_DOUBLE_R("ADC Mixer Switch", >> CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1), >> SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, >> 0), > > Unfortunately, it still doesn't work: at boot time the PCM and ADC > mixer values are 100 which looks OK, but the Analog value is 170, which > doesn't look ok. And inspecting the values set at the register level > for the PCM control seems to reveal that things are still not working > correctly (read on below). > > But this time around, I've collected a lot of debugging informations, > which hopefully will help understand the problem. My understanding is > that: > > SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", > CS42L51_AOUTA_VOL, CS42L51_AOUTB_VOL, > 0, 0x34, 0xE5, aout_tlv), > > cannot work, because it assumes that the range -102.0 dB to 12.0 dB > described in the datasheet is represented by using register values from > 0x34 to 0xE5. Which misses the fact that when going from -102.0 dB to 0 > dB, the register value goes from 0x34 to 0, and then when going from 0 > to 12.0 dB, the register value should go from 0 to 0xE5. > > Ok, Thanks for the output. Something is for sure broken, but I could have sworn I checked this. I’ll hook up my L51 board today and see what the deal is. Thanks, Brian

1 0

[alsa-devel] [PATCH 1/6] sound: lx_core: Remove unused defines
by Maxime Ripard 18 Apr '14

18 Apr '14

Commit f9367f3fbe3c ("ALSA: lx6464es: Remove unused function in pci/lx6464es/lx_core.c") removed the lx_dsp_es_check_pipeline function that was the only user of these defines. Since they're useless now, simply remove them. Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- sound/pci/lx6464es/lx_core.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index 2d8e95e9fbe5..6a1d90a308ea 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -429,11 +429,6 @@ int lx_dsp_read_async_events(struct lx6464es *chip, u32 *data) return ret; } -#define CSES_TIMEOUT 100 /* microseconds */ -#define CSES_CE 0x0001 -#define CSES_BROADCAST 0x0002 -#define CSES_UPDATE_LDSV 0x0004 - #define PIPE_INFO_TO_CMD(capture, pipe) \ ((u32)((u32)(pipe) | ((capture) ? ID_IS_CAPTURE : 0L)) << ID_OFFSET) -- 1.9.1

2 7

[alsa-devel] [PATCHv3 1/2] ALSA: hda/hdmi - Add Nvidia Tegra124 HDMI support
by Dylan Reid 18 Apr '14

18 Apr '14

From: Sumit Bhattacharya <sumitb(a)nvidia.com> Add the Tegra 12x HDMI controller id to patch_hdmi. Signed-off-by: Sumit Bhattacharya <sumitb(a)nvidia.com> Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- No changes since v1. --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0cb5b89..b5d1371 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3322,6 +3322,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .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 = 0x10de0028, .name = "Tegra12x HDMI", .patch = patch_nvhdmi }, { .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 }, @@ -3377,6 +3378,7 @@ 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"); -- 1.8.1.3.605.g02339dd

2 2

[alsa-devel] Behringer Uphoria UMC204 usb audio capture only pops and clicks
by Andrew Reyes 18 Apr '14

18 Apr '14

Does the output help?

1 0

[alsa-devel] [PATCH] ASoC: cs42l51: Fix SOC_DOUBLE_R_SX_TLV shift values for ADC, PCM, and Analog kcontrols
by Brian Austin 17 Apr '14

17 Apr '14

The shift values for the ADC,PCM, and Analog kcontrols were wrong causing wrong values for the SOC_DOUBLE_R_SX_TLV macros Fixed the TLV for aout_tlv to show -102dB correctly Reported-by: Thomas Petazzoni <thomas.petazzoni(a)free-electrons.com> Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42l51.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 3eab460..65fc37c 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -106,9 +106,8 @@ static int cs42l51_set_chan_mix(struct snd_kcontrol *kcontrol, static const DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -5150, 50, 0); static const DECLARE_TLV_DB_SCALE(tone_tlv, -1050, 150, 0); -/* This is a lie. after -102 db, it stays at -102 */ -/* maybe a range would be better */ -static const DECLARE_TLV_DB_SCALE(aout_tlv, -11550, 50, 0); + +static const DECLARE_TLV_DB_SCALE(aout_tlv, -10200, 50, 0); static const DECLARE_TLV_DB_SCALE(boost_tlv, 1600, 1600, 0); static const char *chan_mix[] = { @@ -123,7 +122,7 @@ static const struct soc_enum cs42l51_chan_mix = static const struct snd_kcontrol_new cs42l51_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, - 6, 0x19, 0x7F, adc_pcm_tlv), + 0, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("PCM Playback Switch", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1), SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", @@ -131,7 +130,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0x34, 0xE4, aout_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, - 6, 0x19, 0x7F, adc_pcm_tlv), + 0, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("ADC Mixer Switch", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), -- 1.7.9.5

3 4

[alsa-devel] [PATCH] sound: lx_core: Few misc cleanups
by Maxime Ripard 17 Apr '14

17 Apr '14

This file had a few comments in french that are now in english, and took the opportunity to cleanup a bunch of #if 0 .. #endif and commented out code Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> --- sound/pci/lx6464es/lx_core.c | 70 +++++++++++++++++--------------------------- 1 file changed, 27 insertions(+), 43 deletions(-) diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index 626ecad4dae7..002049770ff7 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -22,8 +22,7 @@ * */ -/* #define RMH_DEBUG 1 */ - +#include <linux/bitops.h> #include <linux/module.h> #include <linux/pci.h> #include <linux/delay.h> @@ -496,15 +495,17 @@ int lx_dsp_es_check_pipeline(struct lx6464es *chip) int i; for (i = 0; i != CSES_TIMEOUT; ++i) { - /* - * le bit CSES_UPDATE_LDSV est Ã 1 dÃ©s que le macprog - * est pret. il re-passe Ã 0 lorsque le premier read a - * Ã©tÃ© fait. pour l'instant on retire le test car ce bit - * passe a 1 environ 200 Ã 400 ms aprÃ©s que le registre - * confES Ã Ã©tÃ© Ã©crit (kick du xilinx ES). + /* + * The bit CSES_UPDATE_LDSV is set to 1 as soon as the + * macprog is ready. He goes back to 0 whenever the + * first read has been issued. + * + * For now, we don't bother testing this bit because + * it's set to 1 between 200 to 400ms after the confES + * register has been written to (xilinx ES kick). * - * On ne teste que le bit CE. - * */ + * We only test the CE bit. + */ u32 cses = lx_dsp_reg_read(chip, eReg_CSES); @@ -602,8 +603,7 @@ int lx_buffer_ask(struct lx6464es *chip, u32 pipe, int is_capture, /* free */ *r_needed += 1; } - -#if 0 +#if DEBUG snd_printdd(LXP "CMD_08_ASK_BUFFERS: needed %d, freed %d\n", *r_needed, *r_freed); for (i = 0; i < MAX_STREAM_BUFFER; ++i) { @@ -1049,9 +1049,9 @@ int lx_level_peaks(struct lx6464es *chip, int is_capture, int channels, /* interrupt handling */ #define PCX_IRQ_NONE 0 -#define IRQCS_ACTIVE_PCIDB 0x00002000L /* Bit nÃÂ¸ 13 */ -#define IRQCS_ENABLE_PCIIRQ 0x00000100L /* Bit nÃÂ¸ 08 */ -#define IRQCS_ENABLE_PCIDB 0x00000200L /* Bit nÃÂ¸ 09 */ +#define IRQCS_ACTIVE_PCIDB BIT(13) +#define IRQCS_ENABLE_PCIIRQ BIT(8) +#define IRQCS_ENABLE_PCIDB BIT(9) static u32 lx_interrupt_test_ack(struct lx6464es *chip) { @@ -1108,25 +1108,21 @@ static int lx_interrupt_handle_async_events(struct lx6464es *chip, u32 irqsrc, int err; u32 stat[9]; /* answer from CMD_04_GET_EVENT */ - /* On peut optimiser pour ne pas lire les evenements vides - * les mots de rÃÂ©ponse sont dans l'ordre suivant : - * Stat[0] mot de status gÃÂ©nÃÂ©ral - * Stat[1] fin de buffer OUT pF - * Stat[2] fin de buffer OUT pf - * Stat[3] fin de buffer IN pF - * Stat[4] fin de buffer IN pf - * Stat[5] underrun poid fort - * Stat[6] underrun poid faible - * Stat[7] overrun poid fort - * Stat[8] overrun poid faible + /* We can optimize this to not read dumb events. + * Answer words are in the following order: + * Stat[0] general status + * Stat[1] end of buffer OUT pF + * Stat[2] end of buffer OUT pf + * Stat[3] end of buffer IN pF + * Stat[4] end of buffer IN pf + * Stat[5] MSB underrun + * Stat[6] LSB underrun + * Stat[7] MSB overrun + * Stat[8] LSB overrun * */ u64 orun_mask; u64 urun_mask; -#if 0 - int has_underrun = (irqsrc & MASK_SYS_STATUS_URUN) ? 1 : 0; - int has_overrun = (irqsrc & MASK_SYS_STATUS_ORUN) ? 1 : 0; -#endif int eb_pending_out = (irqsrc & MASK_SYS_STATUS_EOBO) ? 1 : 0; int eb_pending_in = (irqsrc & MASK_SYS_STATUS_EOBI) ? 1 : 0; @@ -1272,7 +1268,7 @@ irqreturn_t lx_interrupt(int irq, void *dev_id) if (irqsrc & MASK_SYS_STATUS_CMD_DONE) goto exit; -#if 0 +#if DEBUG if (irqsrc & MASK_SYS_STATUS_EOBI) snd_printdd(LXP "interrupt: EOBI\n"); @@ -1310,18 +1306,6 @@ irqreturn_t lx_interrupt(int irq, void *dev_id) "error during audio transfer\n"); } - if (async_escmd) { -#if 0 - /* backdoor for ethersound commands - * - * for now, we do not need this - * - * */ - - snd_printdd("lx6464es: interrupt requests escmd handling\n"); -#endif - } - exit: spin_unlock(&chip->lock); return IRQ_HANDLED; /* this device caused the interrupt */ -- 1.9.1

2 2

Re: [alsa-devel] Tenor TE8802, USB : clics and crackles during music play
by Clemens Ladisch 17 Apr '14

17 Apr '14

mick wrote: > I tried the patch but it does not help. Still crackles. Do you get changing values, and any out-of-range values, when you execute this while it's playing? (while sleep 0.123; do grep 'Momentary freq' /proc/asound/card*/stream0; done) | uniq Regards, Clemens

3 2

[alsa-devel] [PATCH] ASoC: rt5640: Add minimal support for RT5642
by Jarkko Nikula 17 Apr '14

17 Apr '14

We have been using rt5640.c codec driver with RT5642 codec chip before commit 022d21f004c1 ("ASoC: rt5640: add rt5639 support"). That commits starts using device ID reading in reset register for adding device specific controls and routes runtime. Now since device ID appears to be different between RT5640 and RT5642 the driver doesn't add those controls and routes that are valid also on RT5642. Fix this by adding a device ID found by debugging and minimal code for supporting RT5642. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/codecs/rt5640.c | 2 ++ sound/soc/codecs/rt5640.h | 1 + 2 files changed, 3 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index b947c820697d..ffde5bf11789 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2000,6 +2000,7 @@ static int rt5640_probe(struct snd_soc_codec *codec) switch (snd_soc_read(codec, RT5640_RESET)) { case RT5640_RESET_ID: + case RT5642_RESET_ID: snd_soc_add_codec_controls(codec, rt5640_specific_snd_controls, ARRAY_SIZE(rt5640_specific_snd_controls)); @@ -2150,6 +2151,7 @@ static const struct regmap_config rt5640_regmap = { static const struct i2c_device_id rt5640_i2c_id[] = { { "rt5640", 0 }, { "rt5639", 0 }, + { "rt5642", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index 3b50459a83b4..d31faeaa637e 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -16,6 +16,7 @@ #define RT5639_RESET_ID 0x0008 #define RT5640_RESET_ID 0x000c +#define RT5642_RESET_ID 0x0006 /* Info */ #define RT5640_RESET 0x00 -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fsl: Add explicit include of of.h
by Mark Brown 17 Apr '14

17 Apr '14

From: Mark Brown <broonie(a)linaro.org> Hopefully fixing a build failure reported by Stephen Rothwell - though quite why the other OF headers don't include this as well I'm not sure. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/fsl_ssi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 5428a1fda260..2fbbd8416c00 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -40,6 +40,7 @@ #include <linux/delay.h> #include <linux/slab.h> #include <linux/spinlock.h> +#include <linux/of.h> #include <linux/of_address.h> #include <linux/of_irq.h> #include <linux/of_platform.h> -- 1.9.1

2 2

[alsa-devel] [PATCH] of: add empty of_find_node_by_path() for !OF
by Alexander Shiyan 16 Apr '14

16 Apr '14

Add an empty version of of_find_node_by_path(). This fixes following build error for asoc tree: sound/soc/fsl/fsl_ssi.c: In function 'fsl_ssi_probe': sound/soc/fsl/fsl_ssi.c:1471:2: error: implicit declaration of function 'of_find_node_by_path' [-Werror=implicit-function-declaration] sprop = of_get_property(of_find_node_by_path("/"), "compatible", NULL); Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- include/linux/of.h | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/include/linux/of.h b/include/linux/of.h index 919bf21..3bad8d1 100644 --- a/include/linux/of.h +++ b/include/linux/of.h @@ -374,6 +374,11 @@ static inline struct device_node *of_find_matching_node_and_match( return NULL; } +static inline struct device_node *of_find_node_by_path(const char *path) +{ + return NULL; +} + static inline struct device_node *of_get_parent(const struct device_node *node) { return NULL; -- 1.8.3.2

3 7

[alsa-devel] [PATCH v2 linux-next] ALSA: pcm: 'BUG:' message unnecessarily triggers kerneloops
by Tim Gardner 16 Apr '14

16 Apr '14

BugLink: http://bugs.launchpad.net/bugs/1305480 The kerneloops-daemon scans dmesg for common crash signatures, among which is 'BUG:'. The message emitted by the PCM library is really a warning, so the most expedient thing to do seems to be to change the string. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Mark Brown <broonie(a)linaro.org> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: JongHo Kim <furmuwon(a)gmail.com> Signed-off-by: Tim Gardner <tim.gardner(a)canonical.com> --- v2: Changed string preface from "OVER RUN" to "XRUN". As Takashi pointed out, 'Use a term "XRUN", as it's not always an overrun.' sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index ce83def..9acc77e 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -345,7 +345,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, snd_pcm_debug_name(substream, name, sizeof(name)); xrun_log_show(substream); pcm_err(substream->pcm, - "BUG: %s, pos = %ld, buffer size = %ld, period size = %ld\n", + "XRUN: %s, pos = %ld, buffer size = %ld, period size = %ld\n", name, pos, runtime->buffer_size, runtime->period_size); } -- 1.9.1

2 1

[alsa-devel] Behringer Uphoria UMC204 usb audio capture only pops and clicks
by Andrew Reyes 16 Apr '14

16 Apr '14

>Could you send the output of 'lsusb -v', please? Here is a pastebin link to the output: -http://pastebin.com/nRULYjZh

1 0

[alsa-devel] [PATCH 00/18] ASoC: omap: Bind the omap-pcm platform driver to dai device
by Peter Ujfalusi 16 Apr '14

16 Apr '14

Hi, This should have been done a long time ago... Add support for binding the omap-pcm platform to a dai device. IN this way we can remove the 'omap-pcm-audio' platform device/driver need. The change will also going to simplify cases when we boot with DT and we can use the same node for cpu_dai and platform. Regards, Peter --- Peter Ujfalusi (18): ASoC: omap-pcm: Support for binding the platform driver to dai devices ASoC: omap-mcpdm: Assign the dai DMA data at earlier time ASoC: omap-mcpdm: Bind the platform driver to the dai driver when loading ASoC: omap-mcbsp: Assign the dai DMA data at earlier time ASoC: omap-mcbsp: Bind the platform driver to the dai driver when loading ASoC: omap-dmic: Assign the dai DMA data at earlier time ASoC: omap-dmic: Bind the platform driver to the dai driver when loading ASoC: omap-hdmi: Bind the platform driver to the dai driver when loading ASoC: omap-abe-twl6040: Use the cpu_dai node to specify the platform driver ASoC: omap-twl4030: Use the same name/node for platform as the cpu_dai ASoC: am3517evm: Use the same name for platform as the cpu_dai ASoC: ams-delta: Use the same name for platform as the cpu_dai ASoC: n810: Use the same name for platform as the cpu_dai ASoC: omap3pandora: Use the same name for platform as the cpu_dai ASoC: osk5912: Use the same name for platform as the cpu_dai ASoC: rx51: Use the same name for platform as the cpu_dai ASoC: omap-hdmi-card: Use the same name for platform as the cpu_dai ASoC: omap-pcm: Drop the platform driver init code sound/soc/omap/am3517evm.c | 2 +- sound/soc/omap/ams-delta.c | 2 +- sound/soc/omap/n810.c | 2 +- sound/soc/omap/omap-abe-twl6040.c | 8 ++------ sound/soc/omap/omap-dmic.c | 10 +++++++++- sound/soc/omap/omap-hdmi-card.c | 2 +- sound/soc/omap/omap-hdmi.c | 7 ++++++- sound/soc/omap/omap-mcbsp.c | 19 ++++++++++++------- sound/soc/omap/omap-mcpdm.c | 15 +++++++++++---- sound/soc/omap/omap-pcm.c | 25 ++++++------------------- sound/soc/omap/omap-pcm.h | 35 +++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-twl4030.c | 10 ++++++++-- sound/soc/omap/omap3pandora.c | 4 ++-- sound/soc/omap/osk5912.c | 2 +- sound/soc/omap/rx51.c | 2 +- 15 files changed, 97 insertions(+), 48 deletions(-) create mode 100644 sound/soc/omap/omap-pcm.h -- 1.9.2

3 23

[alsa-devel] Using simple-card to replace kirkwood-t5325.c
by Andrew Lunn 16 Apr '14

16 Apr '14

Hi Folks I'm an ALSA newbie, so if i say anything stupid, please let me know. I'm trying to replace sound/soc/kirkwood/kirkwood-t5325.c with DT, using simple-card. I can get near, but i'm missing two things: There does not appear to be a way to represent this in DT: static int t5325_dai_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; snd_soc_dapm_enable_pin(dapm, "Mic Jack"); snd_soc_dapm_enable_pin(dapm, "Headphone Jack"); snd_soc_dapm_enable_pin(dapm, "Speaker"); return 0; } It appears that quite a few drivers need to enable pins. It also seems common to disable pins and to set them to NC. Is this a generic enough feature it could be added to the DT binding? The second thing i need is: static int t5325_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; unsigned int freq; freq = params_rate(params) * 256; return snd_soc_dai_set_sysclk(codec_dai, 0, freq, SND_SOC_CLOCK_IN); } static struct snd_soc_ops t5325_ops = { .hw_params = t5325_hw_params, }; This seems a lot less common requirements. All the Marvell SoCs need it, but not many others. So i don't think it makes sense to add it directly to simple-card, otherwise simple-card quickly becomes complex-card as everybody else wants there quirks adding. Has there been any thoughts about turning simple-card.c into a library, or adding hooks so that a driver can modify the created dai_link structures to add in ops like this? Thanks Andrew

3 4

[alsa-devel] Behringer Uphoria UMC204 usb audio capture only pops and clicks
by Andrew Reyes 16 Apr '14

16 Apr '14

I have a usb Behringer Uphoria UMC204 and it seems to work fine at 44.1khz full duplex but anything higher than that I get consistent pops and clicks. I notice it's on the capure side of things as when I have it in playback only it's quiet. I have to say I'm testing this using JACK so this is how I noticed it on the capture only side of things. I'm not sure exactly whats needed to debug this issue. Any help would be appreciated. Thanks in advance.

2 1

[alsa-devel] Add support for headset Mic of Dell
by Kailang 16 Apr '14

16 Apr '14

Hi Takashi, Add support for Dell machine for Headset Mic support. SSID: 0x0696 and 0x0698 Thanks. BR, Kailang

2 1

[alsa-devel] [PATCH v2 1/2] ALSA: sound/atmel/ac97c.c: Convert to module_platform_driver
by Alexander Stein 16 Apr '14

16 Apr '14

This reduces some boilerplate code. Signed-off-by: Alexander Stein <alexanders83(a)web.de> Acked-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> --- sound/atmel/ac97c.c | 15 ++------------- 1 file changed, 2 insertions(+), 13 deletions(-) diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 05ec049..a04d2317 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -1198,6 +1198,7 @@ static int atmel_ac97c_remove(struct platform_device *pdev) } static struct platform_driver atmel_ac97c_driver = { + .probe = atmel_ac97c_probe, .remove = atmel_ac97c_remove, .driver = { .name = "atmel_ac97c", @@ -1205,19 +1206,7 @@ static struct platform_driver atmel_ac97c_driver = { .pm = ATMEL_AC97C_PM_OPS, }, }; - -static int __init atmel_ac97c_init(void) -{ - return platform_driver_probe(&atmel_ac97c_driver, - atmel_ac97c_probe); -} -module_init(atmel_ac97c_init); - -static void __exit atmel_ac97c_exit(void) -{ - platform_driver_unregister(&atmel_ac97c_driver); -} -module_exit(atmel_ac97c_exit); +module_platform_driver(atmel_ac97c_driver); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Driver for Atmel AC97 controller"); -- 1.9.2

3 4

[alsa-devel] [PATCH] ALSA: hda - add headset mic detect quirk for a Dell laptop
by Hui Wang 16 Apr '14

16 Apr '14

When we plug a 3-ring headset on the Dell machine (VID: 0x10ec0255, SID: 0x1028067f), the headset mic can't be detected, after apply this patch, the headset mic can work well. BugLink: https://bugs.launchpad.net/bugs/1297581 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f7766fe..26895d1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4621,6 +4621,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0667, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0668, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0669, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x067f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 1.8.1.2

2 1

[alsa-devel] [PATCH] ASoC: cs42l51: split i2c from codec driver
by Brian Austin 16 Apr '14

16 Apr '14

This patch removes the i2c bus code from the codec driver and creates seperate i2c driver. Signed-off-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/Kconfig | 6 +++- sound/soc/codecs/Makefile | 2 ++ sound/soc/codecs/cs42l51-i2c.c | 59 ++++++++++++++++++++++++++++++ sound/soc/codecs/cs42l51.c | 77 +++++++++++----------------------------- sound/soc/codecs/cs42l51.h | 5 +++ 5 files changed, 91 insertions(+), 58 deletions(-) create mode 100644 sound/soc/codecs/cs42l51-i2c.c diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087a..5ee2eb5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -32,7 +32,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ALC5623 if I2C select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC - select SND_SOC_CS42L51 if I2C + select SND_SOC_CS42L51_I2C if I2C select SND_SOC_CS42L52 if I2C select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C @@ -233,6 +233,10 @@ config SND_SOC_CQ0093VC config SND_SOC_CS42L51 tristate +config SND_SOC_CS42L51_I2C + tristate + select SND_SOC_CS42L51 + config SND_SOC_CS42L52 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc12676..3046dc0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -19,6 +19,7 @@ snd-soc-ak5386-objs := ak5386.o snd-soc-arizona-objs := arizona.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o +snd-soc-cs42l51-i2c-objs := cs42l51-i2c.o snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o @@ -152,6 +153,7 @@ obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o +obj-$(CONFIG_SND_SOC_CS42L51_I2C) += snd-soc-cs42l51-i2c.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o diff --git a/sound/soc/codecs/cs42l51-i2c.c b/sound/soc/codecs/cs42l51-i2c.c new file mode 100644 index 0000000..50d42ef --- /dev/null +++ b/sound/soc/codecs/cs42l51-i2c.c @@ -0,0 +1,59 @@ +/* + * cs42l56.c -- CS42L51 ALSA SoC I2C audio driver + * + * Copyright 2014 CirrusLogic, Inc. + * + * Author: Brian Austin <brian.austin(a)cirrus.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 as + * published by the Free Software Foundation. + * + */ + +#include <linux/i2c.h> +#include <linux/module.h> +#include <sound/soc.h> + +#include "cs42l51.h" + +static struct i2c_device_id cs42l51_i2c_id[] = { + {"cs42l51", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs42l51_i2c_id); + +static int cs42l51_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap_config config; + + config = cs42l51_regmap; + config.val_bits = 8; + config.reg_bits = 8; + + return cs42l51_probe(&i2c->dev, devm_regmap_init_i2c(i2c, &config)); +} + +static int cs42l51_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +static struct i2c_driver cs42l51_i2c_driver = { + .driver = { + .name = "cs42l51", + .owner = THIS_MODULE, + }, + .probe = cs42l51_i2c_probe, + .remove = cs42l51_i2c_remove, + .id_table = cs42l51_i2c_id, +}; + +module_i2c_driver(cs42l51_i2c_driver); + +MODULE_DESCRIPTION("ASoC CS42L51 I2C Driver"); +MODULE_AUTHOR("Brian Austin, Cirrus Logic Inc, <brian.austin(a)cirrus.com>"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 65fc37c..68ab2eb 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -29,7 +29,6 @@ #include <sound/initval.h> #include <sound/pcm_params.h> #include <sound/pcm.h> -#include <linux/i2c.h> #include <linux/regmap.h> #include "cs42l51.h" @@ -484,7 +483,7 @@ static struct snd_soc_dai_driver cs42l51_dai = { .ops = &cs42l51_dai_ops, }; -static int cs42l51_probe(struct snd_soc_codec *codec) +static int cs42l51_codec_probe(struct snd_soc_codec *codec) { int ret, reg; @@ -511,7 +510,7 @@ static int cs42l51_probe(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_device_cs42l51 = { - .probe = cs42l51_probe, + .probe = cs42l51_codec_probe, .controls = cs42l51_snd_controls, .num_controls = ARRAY_SIZE(cs42l51_snd_controls), @@ -521,91 +520,55 @@ static struct snd_soc_codec_driver soc_codec_device_cs42l51 = { .num_dapm_routes = ARRAY_SIZE(cs42l51_routes), }; -static const struct regmap_config cs42l51_regmap = { - .reg_bits = 8, - .val_bits = 8, - +const struct regmap_config cs42l51_regmap = { .max_register = CS42L51_CHARGE_FREQ, .cache_type = REGCACHE_RBTREE, }; +EXPORT_SYMBOL_GPL(cs42l51_regmap); -static int cs42l51_i2c_probe(struct i2c_client *i2c_client, - const struct i2c_device_id *id) +int cs42l51_probe(struct device *dev, struct regmap *regmap) { struct cs42l51_private *cs42l51; - struct regmap *regmap; unsigned int val; int ret; - - regmap = devm_regmap_init_i2c(i2c_client, &cs42l51_regmap); - if (IS_ERR(regmap)) { - ret = PTR_ERR(regmap); - dev_err(&i2c_client->dev, "Failed to create regmap: %d\n", - ret); - return ret; - } + + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + cs42l51 = devm_kzalloc(dev, sizeof(struct cs42l51_private), + GFP_KERNEL); + if (!cs42l51) + return -ENOMEM; + + dev_set_drvdata(dev, cs42l51); /* Verify that we have a CS42L51 */ ret = regmap_read(regmap, CS42L51_CHIP_REV_ID, &val); if (ret < 0) { - dev_err(&i2c_client->dev, "failed to read I2C\n"); + dev_err(dev, "failed to read I2C\n"); goto error; } if ((val != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_A)) && (val != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_B))) { - dev_err(&i2c_client->dev, "Invalid chip id: %x\n", val); + dev_err(dev, "Invalid chip id: %x\n", val); ret = -ENODEV; goto error; } + dev_info(dev, "Cirrus Logic CS42L51, Revision: %02X\n", val & 0xFF); - dev_info(&i2c_client->dev, "found device cs42l51 rev %d\n", - val & 7); - - cs42l51 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l51_private), - GFP_KERNEL); - if (!cs42l51) - return -ENOMEM; - - i2c_set_clientdata(i2c_client, cs42l51); - - ret = snd_soc_register_codec(&i2c_client->dev, + ret = snd_soc_register_codec(dev, &soc_codec_device_cs42l51, &cs42l51_dai, 1); error: return ret; } - -static int cs42l51_i2c_remove(struct i2c_client *client) -{ - snd_soc_unregister_codec(&client->dev); - return 0; -} - -static const struct i2c_device_id cs42l51_id[] = { - {"cs42l51", 0}, - {} -}; -MODULE_DEVICE_TABLE(i2c, cs42l51_id); +EXPORT_SYMBOL_GPL(cs42l51_probe); static const struct of_device_id cs42l51_of_match[] = { { .compatible = "cirrus,cs42l51", }, { } }; MODULE_DEVICE_TABLE(of, cs42l51_of_match); - -static struct i2c_driver cs42l51_i2c_driver = { - .driver = { - .name = "cs42l51-codec", - .owner = THIS_MODULE, - .of_match_table = cs42l51_of_match, - }, - .id_table = cs42l51_id, - .probe = cs42l51_i2c_probe, - .remove = cs42l51_i2c_remove, -}; - -module_i2c_driver(cs42l51_i2c_driver); - MODULE_AUTHOR("Arnaud Patard <arnaud.patard(a)rtp-net.org>"); MODULE_DESCRIPTION("Cirrus Logic CS42L51 ALSA SoC Codec Driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42l51.h b/sound/soc/codecs/cs42l51.h index 2beeb17..641ef18 100644 --- a/sound/soc/codecs/cs42l51.h +++ b/sound/soc/codecs/cs42l51.h @@ -18,6 +18,11 @@ #ifndef _CS42L51_H #define _CS42L51_H +struct device; + +extern const struct regmap_config cs42l51_regmap; +int cs42l51_probe(struct device *dev, struct regmap *regmap); + #define CS42L51_CHIP_ID 0x1B #define CS42L51_CHIP_REV_A 0x00 #define CS42L51_CHIP_REV_B 0x01 -- 1.7.9.5

3 4

[alsa-devel] [PATCH] ASoC: Fix snd_soc_kcontrol_platform() return type
by Lars-Peter Clausen 16 Apr '14

16 Apr '14

This should obviously be snd_soc_platform * and not snd_soc_codec * Fixes: f6272ff8a5f4 ("ASoC: Add snd_soc_kcontrol_platform() helper function") Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 3396594..488e749 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1252,7 +1252,7 @@ static inline struct snd_soc_codec *snd_soc_kcontrol_codec( * registered with snd_soc_add_platform_controls() or via table based setup of * a snd_soc_platform_driver. Otherwise the behavior is undefined. */ -static inline struct snd_soc_codec *snd_soc_kcontrol_platform( +static inline struct snd_soc_platform *snd_soc_kcontrol_platform( struct snd_kcontrol *kcontrol) { return snd_kcontrol_chip(kcontrol); -- 1.8.0

2 1

[alsa-devel] [PATCH 0/5] DT support for N900 soundcard (rx51-audio)
by Sebastian Reichel 15 Apr '14

15 Apr '14

Hi, This patchset adds DT support in rx51-audio. I tested it on my Nokia N900 and was able to play a wav file using aplay. I have not tested the whole functionality, but output via speakers, headphones and the related enable controls seem to work. The patches can be applied cleanly to torvalds/linux.git:master and broonie/sound.git:for-next. -- Sebastian Pali Rohár (1): ASoC: omap: rx51: Use devm_snd_soc_register_card Sebastian Reichel (4): ASoC: Allow Aux Codecs to be specified using DT ASoC: RX-51: Add DT support to sound driver ASoC: RX-51: Convert to table based DAPM setup ASoC: tlv320aic3x: fix shared reset pin for DT .../devicetree/bindings/sound/nokia,rx51.txt | 27 +++ include/sound/rx51.h | 18 ++ include/sound/soc.h | 13 +- sound/soc/codecs/tlv320aic3x.c | 9 +- sound/soc/omap/omap-mcbsp.c | 5 +- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/rx51.c | 248 +++++++++++++-------- sound/soc/soc-core.c | 68 +++--- 8 files changed, 266 insertions(+), 124 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nokia,rx51.txt create mode 100644 include/sound/rx51.h -- 1.9.1

3 11

[alsa-devel] [PATCH] ASoC: alc5623: Fix regmap endianness
by Andrew Lunn 15 Apr '14

15 Apr '14

Commit 0cd257bf9b9b0cbb4fa1a5c988a232506997867c, "ASoC: alc5623: Convert to direct regmap API usage" broke probing of the codec, because of wrong endinness of the ID and codec version read from the device. Fix this by removing the existing flipping of the endiannes, and extracting the codec type byte from the word from the regmap. Signed-off-by: Andrew Lunn <andrew(a)lunn.ch> --- sound/soc/codecs/alc5623.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index f500905e9373..2acf82f4a08a 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -1018,13 +1018,13 @@ static int alc5623_i2c_probe(struct i2c_client *client, dev_err(&client->dev, "failed to read vendor ID1: %d\n", ret); return ret; } - vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID2, &vid2); if (ret < 0) { dev_err(&client->dev, "failed to read vendor ID2: %d\n", ret); return ret; } + vid2 >>= 8; if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) { dev_err(&client->dev, "unknown or wrong codec\n"); -- 1.9.2

2 1

[alsa-devel] [PATCH] ASoC: Intel: Fix a self assignment in sst_mem_block_alloc_scratch()
by Christian Engelmayer 15 Apr '14

15 Apr '14

Remove a self assignment in sst_mem_block_alloc_scratch(). When calculating buffer sizes there is no need for statements without effect. Detected by Coverity: CID 1195249. Signed-off-by: Christian Engelmayer <cengelma(a)gmx.at> --- sound/soc/intel/sst-firmware.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/intel/sst-firmware.c b/sound/soc/intel/sst-firmware.c index f768710..f24619a 100644 --- a/sound/soc/intel/sst-firmware.c +++ b/sound/soc/intel/sst-firmware.c @@ -505,9 +505,7 @@ struct sst_module *sst_mem_block_alloc_scratch(struct sst_dsp *dsp) /* calculate required scratch size */ list_for_each_entry(sst_module, &dsp->module_list, list) { - if (scratch->s.size > sst_module->s.size) - scratch->s.size = scratch->s.size; - else + if (scratch->s.size < sst_module->s.size) scratch->s.size = sst_module->s.size; } -- 1.8.3.2

2 1

[alsa-devel] [PATCH] ASoC: wm8997: Fix compile error
by Lars-Peter Clausen 15 Apr '14

15 Apr '14

The macro's name is SOC_ENUM, not SOC_VALUE. Fixes: e13dd8ce ("ASoC: wm8997: Replace usage deprecated MUX/ENUM macros") Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- I'm getting to old for this... --- sound/soc/codecs/wm8997.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 61926cb..bb9b47b 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -245,8 +245,8 @@ SND_SOC_BYTES("LHPF2 Coefficients", ARIZONA_HPLPF2_2, 1), SND_SOC_BYTES("LHPF3 Coefficients", ARIZONA_HPLPF3_2, 1), SND_SOC_BYTES("LHPF4 Coefficients", ARIZONA_HPLPF4_2, 1), -SOC_VALUE("ISRC1 FSL", arizona_isrc_fsl[0]), -SOC_VALUE("ISRC2 FSL", arizona_isrc_fsl[1]), +SOC_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), +SOC_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), @@ -286,8 +286,8 @@ SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, 0xbf, 0, digital_tlv), -SOC_VALUE("HPOUT1 OSR", wm8997_hpout_osr[0]), -SOC_VALUE("EPOUT OSR", wm8997_hpout_osr[1]), +SOC_ENUM("HPOUT1 OSR", wm8997_hpout_osr[0]), +SOC_ENUM("EPOUT OSR", wm8997_hpout_osr[1]), SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), -- 1.8.0

2 1

[alsa-devel] [PATCH] ASoC: fsl_sai: Fix incorrect condition check in trigger()
by Nicolin Chen 15 Apr '14

15 Apr '14

Patch ASoC: fsl_sai: Fix buggy configurations in trigger() doesn't entirely fix the condition: FRDE of the current substream direction is being cleared while the code is still using the non-updated one. Thus this patch fixes this issue by checking the opposite one's FRDE alone since the current one's is absolutely disabled. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 9858d42..1716a41 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -343,7 +343,8 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), FSL_SAI_CSR_xIE_MASK, 0); - if (!(tcsr & FSL_SAI_CSR_FRDE || rcsr & FSL_SAI_CSR_FRDE)) { + /* Check if the opposite FRDE is also disabled */ + if (!(tx ? rcsr & FSL_SAI_CSR_FRDE : tcsr & FSL_SAI_CSR_FRDE)) { regmap_update_bits(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_TERE, 0); regmap_update_bits(sai->regmap, FSL_SAI_RCSR, -- 1.8.4

2 2

[alsa-devel] [PATCH v2 0/2] ASoC: fsl_sai: Add bus clock and mclks with clock controls
by Nicolin Chen 15 Apr '14

15 Apr '14

This series of patches add clock controls to fsl_sai driver and updates the vf610.dtsi accordingly. Changelog v2: * Appended two extra mclks to the driver since SAI actually has three. * Renamed clock name to 'bus' and 'mclk' according to the reference manual. Nicolin Chen (2): ASoC: fsl_sai: Add clock controls for SAI ARM: dts: Append clock bindings for sai2 on VF610 platform .../devicetree/bindings/sound/fsl-sai.txt | 8 ++-- arch/arm/boot/dts/vf610.dtsi | 6 ++- sound/soc/fsl/fsl_sai.c | 50 ++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 4 ++ 4 files changed, 59 insertions(+), 9 deletions(-) -- 1.8.4

4 9

[alsa-devel] [PATCH RFC v2 0/2] Fix simple-card *-master DT parameter handling
by Jyri Sarha 15 Apr '14

15 Apr '14

Since RFC: - fixed commit msg typo - added include/sound/soc.h changes too The sematics of bitclock-master and frame-master DT parameters should depend on whether they are found from a cpu-dai or codec sub-node. - bitclock-master in cpu-dai node means Codec-Bitclock-Slave - frame-master in cpu-dai node means Codec-Frame-Slave - bitclock-master in codec node means Codec-Bitclock-Master - frame-master in codec node means Codec-Frame-Master For example in a cpu-dai mode bitclock-master parameter should produce SND_SOC_DAIFMT_CBS_* daifmt flags and a codec node SND_SOC_DAIFMT_CBM_* flags. Best regards, Jyri Jyri Sarha (2): ASoC: core: Add is_cpu_dai_node-parameter to snd_soc_of_parse_daifmt() ASoC: simple-card: Take snd_soc_of_parse_daifmt() change in to account include/sound/soc.h | 3 ++- sound/soc/generic/simple-card.c | 13 ++++++++----- sound/soc/soc-core.c | 8 +++++++- 3 files changed, 17 insertions(+), 7 deletions(-) -- 1.7.9.5

4 12

[alsa-devel] [PATCH] ASoC: fsl_sai: Use FSL_SAI_xXR() and regmap_update_bits() to simplify code
by Nicolin Chen 14 Apr '14

14 Apr '14

By doing this, the driver can drop around 50 lines and become neater. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 101 ++++++++++++------------------------------------ 1 file changed, 25 insertions(+), 76 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5fff2e1..9858d42 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -93,16 +93,8 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int fsl_dir) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 val_cr2, reg_cr2; - - if (fsl_dir == FSL_FMT_TRANSMITTER) - reg_cr2 = FSL_SAI_TCR2; - else - reg_cr2 = FSL_SAI_RCR2; - - regmap_read(sai->regmap, reg_cr2, &val_cr2); - - val_cr2 &= ~FSL_SAI_CR2_MSEL_MASK; + bool tx = fsl_dir == FSL_FMT_TRANSMITTER; + u32 val_cr2 = 0; switch (clk_id) { case FSL_SAI_CLK_BUS: @@ -121,7 +113,8 @@ static int fsl_sai_set_dai_sysclk_tr(struct snd_soc_dai *cpu_dai, return -EINVAL; } - regmap_write(sai->regmap, reg_cr2, val_cr2); + regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx), + FSL_SAI_CR2_MSEL_MASK, val_cr2); return 0; } @@ -153,22 +146,10 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, unsigned int fmt, int fsl_dir) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 val_cr2, val_cr4, reg_cr2, reg_cr4; - - if (fsl_dir == FSL_FMT_TRANSMITTER) { - reg_cr2 = FSL_SAI_TCR2; - reg_cr4 = FSL_SAI_TCR4; - } else { - reg_cr2 = FSL_SAI_RCR2; - reg_cr4 = FSL_SAI_RCR4; - } + bool tx = fsl_dir == FSL_FMT_TRANSMITTER; + u32 val_cr2 = 0, val_cr4 = 0; - regmap_read(sai->regmap, reg_cr2, &val_cr2); - regmap_read(sai->regmap, reg_cr4, &val_cr4); - - if (sai->big_endian_data) - val_cr4 &= ~FSL_SAI_CR4_MF; - else + if (!sai->big_endian_data) val_cr4 |= FSL_SAI_CR4_MF; /* DAI mode */ @@ -189,7 +170,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * frame sync asserts with the first bit of the frame. */ val_cr2 |= FSL_SAI_CR2_BCP; - val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP); break; case SND_SOC_DAIFMT_DSP_A: /* @@ -199,7 +179,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * data word. */ val_cr2 |= FSL_SAI_CR2_BCP; - val_cr4 &= ~FSL_SAI_CR4_FSP; val_cr4 |= FSL_SAI_CR4_FSE; sai->is_dsp_mode = true; break; @@ -209,7 +188,6 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * frame sync asserts with the first bit of the frame. */ val_cr2 |= FSL_SAI_CR2_BCP; - val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP); sai->is_dsp_mode = true; break; case SND_SOC_DAIFMT_RIGHT_J: @@ -247,23 +225,22 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, val_cr4 |= FSL_SAI_CR4_FSD_MSTR; break; case SND_SOC_DAIFMT_CBM_CFM: - val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR; - val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR; break; case SND_SOC_DAIFMT_CBS_CFM: val_cr2 |= FSL_SAI_CR2_BCD_MSTR; - val_cr4 &= ~FSL_SAI_CR4_FSD_MSTR; break; case SND_SOC_DAIFMT_CBM_CFS: - val_cr2 &= ~FSL_SAI_CR2_BCD_MSTR; val_cr4 |= FSL_SAI_CR4_FSD_MSTR; break; default: return -EINVAL; } - regmap_write(sai->regmap, reg_cr2, val_cr2); - regmap_write(sai->regmap, reg_cr4, val_cr4); + regmap_update_bits(sai->regmap, FSL_SAI_xCR2(tx), + FSL_SAI_CR2_BCP | FSL_SAI_CR2_BCD_MSTR, val_cr2); + regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx), + FSL_SAI_CR4_MF | FSL_SAI_CR4_FSE | + FSL_SAI_CR4_FSP | FSL_SAI_CR4_FSD_MSTR, val_cr4); return 0; } @@ -290,29 +267,10 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 val_cr4, val_cr5, val_mr, reg_cr4, reg_cr5, reg_mr; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; unsigned int channels = params_channels(params); u32 word_width = snd_pcm_format_width(params_format(params)); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - reg_cr4 = FSL_SAI_TCR4; - reg_cr5 = FSL_SAI_TCR5; - reg_mr = FSL_SAI_TMR; - } else { - reg_cr4 = FSL_SAI_RCR4; - reg_cr5 = FSL_SAI_RCR5; - reg_mr = FSL_SAI_RMR; - } - - regmap_read(sai->regmap, reg_cr4, &val_cr4); - regmap_read(sai->regmap, reg_cr4, &val_cr5); - - val_cr4 &= ~FSL_SAI_CR4_SYWD_MASK; - val_cr4 &= ~FSL_SAI_CR4_FRSZ_MASK; - - val_cr5 &= ~FSL_SAI_CR5_WNW_MASK; - val_cr5 &= ~FSL_SAI_CR5_W0W_MASK; - val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; + u32 val_cr4 = 0, val_cr5 = 0; if (!sai->is_dsp_mode) val_cr4 |= FSL_SAI_CR4_SYWD(word_width); @@ -320,18 +278,20 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 |= FSL_SAI_CR5_WNW(word_width); val_cr5 |= FSL_SAI_CR5_W0W(word_width); - val_cr5 &= ~FSL_SAI_CR5_FBT_MASK; if (sai->big_endian_data) val_cr5 |= FSL_SAI_CR5_FBT(0); else val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); val_cr4 |= FSL_SAI_CR4_FRSZ(channels); - val_mr = ~0UL - ((1 << channels) - 1); - regmap_write(sai->regmap, reg_cr4, val_cr4); - regmap_write(sai->regmap, reg_cr5, val_cr5); - regmap_write(sai->regmap, reg_mr, val_mr); + regmap_update_bits(sai->regmap, FSL_SAI_xCR4(tx), + FSL_SAI_CR4_SYWD_MASK | FSL_SAI_CR4_FRSZ_MASK, + val_cr4); + regmap_update_bits(sai->regmap, FSL_SAI_xCR5(tx), + FSL_SAI_CR5_WNW_MASK | FSL_SAI_CR5_W0W_MASK | + FSL_SAI_CR5_FBT_MASK, val_cr5); + regmap_write(sai->regmap, FSL_SAI_xMR(tx), ~0UL - ((1 << channels) - 1)); return 0; } @@ -401,8 +361,8 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; struct device *dev = &sai->pdev->dev; - u32 reg; int ret; ret = clk_prepare_enable(sai->bus_clk); @@ -411,12 +371,7 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, return ret; } - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - reg = FSL_SAI_TCR3; - else - reg = FSL_SAI_RCR3; - - regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, + regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx), FSL_SAI_CR3_TRCE, FSL_SAI_CR3_TRCE); return 0; @@ -426,15 +381,9 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 reg; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - reg = FSL_SAI_TCR3; - else - reg = FSL_SAI_RCR3; + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, - ~FSL_SAI_CR3_TRCE); + regmap_update_bits(sai->regmap, FSL_SAI_xCR3(tx), FSL_SAI_CR3_TRCE, 0); clk_disable_unprepare(sai->bus_clk); } -- 1.8.4

2 1

[alsa-devel] [PATCHv2 0/4] Add ESAI & SPDIF Vybrid and LS1+ support.
by Xiubo Li 14 Apr '14

14 Apr '14

Changes in V2: - Revise some issue from Nicolin Chen and Fabio Estevam. Xiubo Li (4): ASoC: esai: use the precise definition of 'ret'. ASoC: spdif: Sort the header files alphabetically. ASoC: esai: Add VF610+ compatibles support. ASoC: spdif: Add VF610+ compatibles support. sound/soc/fsl/fsl_esai.c | 9 ++++++--- sound/soc/fsl/fsl_spdif.c | 9 +++++---- 2 files changed, 11 insertions(+), 7 deletions(-) -- 1.8.4

3 6

[alsa-devel] [PATCH] ASoC: dapm: don't skip evaluate the power of widget with force set
by Xiang Xiao 14 Apr '14

14 Apr '14

Signed-off-by: Xiang Xiao <xiaoxiang(a)xiaomi.com> --- sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c8a780d..14af77e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1673,7 +1673,7 @@ static void dapm_widget_set_peer_power(struct snd_soc_dapm_widget *peer, /* If the peer is already in the state we're moving to then we * won't have an impact on it. */ - if (power != peer->power) + if (peer->force || power != peer->power) dapm_mark_dirty(peer, "peer state change"); } @@ -1683,7 +1683,7 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, { struct snd_soc_dapm_path *path; - if (w->power == power) + if (!w->force && w->power == power) return; trace_snd_soc_dapm_widget_power(w, power); -- 1.8.1.2

2 1

[alsa-devel] [RFT v2 0/6] ASoC: core: Add support for DAI multicodec
by Benoit Cousson 14 Apr '14

14 Apr '14

Hi Mark and Liam, Here is the second version of the multicodec series. I mostly rebased the code to asoc-next, added smaller patches to introduce helpers before the main patch and thus reduce the overal size of that patch and modified the TDM slot fixup. Please note that I have a very simple setup based on BBB to check if the series is working, so it will be good to test the series on various setup to ensure this is not generating any regression. Hence the RFT for this version. The series is based on asoc-v3.15-2-245-g876dea8. Test and comments are welcome. Thanks, Benoit v1: http://comments.gmane.org/gmane.linux.alsa.devel/120532 v2: - Split the first version in several patches. - Remove the TDM fixup --- Misael Lopez Cruz (6): ASoC: core: Add helpers for codec and codec_dai search ASoC: core: Add helpers for codec DAI probe & remove ASoC: core: Add helper for DAI widgets linking ASoC: core: Add function for ac97 codec registration ASoC: core: Add helpers for dai link and aux dev init ASoC: core: Add support for DAI multicodec include/sound/soc-dai.h | 5 + include/sound/soc.h | 16 ++ sound/soc/soc-core.c | 578 ++++++++++++++++++++++++++++++++++-------------- sound/soc/soc-dapm.c | 39 +++- sound/soc/soc-pcm.c | 438 +++++++++++++++++++++++++----------- 5 files changed, 761 insertions(+), 315 deletions(-) -- 1.8.3.2

3 11

[alsa-devel] [PATCH] ASoC: jack: export gpio detect
by cfreeman＠nvidia.com 14 Apr '14

14 Apr '14

From: Christopher Freeman <cfreeman(a)nvidia.com> Export the gpio detect function so machine drivers may call it. Interrupts for the jack may be disabled during sleep, so this allows a machine driver to have the jack status updated during resume. Signed-off-by: Christopher Freeman <cfreeman(a)nvidia.com> --- include/sound/soc.h | 1 + sound/soc/soc-jack.c | 3 ++- 2 files changed, 3 insertions(+), 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index a6a059c..36383cb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -442,6 +442,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, struct snd_soc_jack_gpio *gpios); +void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio); #endif /* codec register bit access */ diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index 0bb5ccc..fd97cc5 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -231,7 +231,7 @@ EXPORT_SYMBOL_GPL(snd_soc_jack_notifier_unregister); #ifdef CONFIG_GPIOLIB /* gpio detect */ -static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) +void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) { struct snd_soc_jack *jack = gpio->jack; int enable; @@ -251,6 +251,7 @@ static void snd_soc_jack_gpio_detect(struct snd_soc_jack_gpio *gpio) snd_soc_jack_report(jack, report, gpio->report); } +EXPORT_SYMBOL_GPL(snd_soc_jack_gpio_detect); /* irq handler for gpio pin */ static irqreturn_t gpio_handler(int irq, void *data) -- 1.8.3.2 ----------------------------------------------------------------------------------- This email message is for the sole use of the intended recipient(s) and may contain confidential information. Any unauthorized review, use, disclosure or distribution is prohibited. If you are not the intended recipient, please contact the sender by reply email and destroy all copies of the original message. -----------------------------------------------------------------------------------

3 5

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Do not touch 0x04 register above McASP_VERSION_2
by Peter Ujfalusi 14 Apr '14

14 Apr '14

This register is not defined in TI81xx and on AM335x/AM437x it is the SYSCONFIG register which should not be touched by drivers since it is related to PM and handled by the generic PM code. This register write was there since the first time the davinci-mcasp driver was appeared in the kernel. The reason why it did not caused any issues on AM335x/AM437x is that it sets bit 1 in SYSCONFIG register which in turn will enable the smart-idle mode. This is the default mode and this is the mode McASP should be in also when in use. On TI81xx the register is not defined. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index ce0e55a4304c..50a824d9dbb7 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -495,7 +495,7 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, int active_serializers, numevt, n; u32 reg; /* Default configuration */ - if (mcasp->version != MCASP_VERSION_4) + if (mcasp->version < MCASP_VERSION_3) mcasp_set_bits(mcasp, DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); /* All PINS as McASP */ -- 1.9.1

2 1

[alsa-devel] [PATCH 00/17] Removed deprecated ENUM/MUX macros
by Lars-Peter Clausen 14 Apr '14

14 Apr '14

Hi, This is a follow-up to the enum consolidation series. It updates all users of the now redundant and deprecated macros and once that is done removes the macros. - Lars Lars-Peter Clausen (17): ASoC: adau1373: Replace usage deprecated MUX/ENUM macros ASoC: adav80x: Replace usage deprecated MUX/ENUM macros ASoC: arizona: Replace usage deprecated MUX/ENUM macros ASoC: max98090: Replace usage deprecated MUX/ENUM macros ASoC: mc13783: Replace usage deprecated MUX/ENUM macros ASoC: pcm512x: Replace usage deprecated SOC_VALUE_ENUM macro ASoC: rt5640: Replace usage deprecated MUX/ENUM macros ASoC: wm2200: Replace usage deprecated MUX/ENUM macros ASoC: wm5100: Replace usage deprecated MUX/ENUM macros ASoC: wm5102: Replace usage deprecated MUX/ENUM macros ASoC: wm5110: Replace usage deprecated MUX/ENUM macros ASoC: wm8988: Replace usage deprecated MUX/ENUM macros ASoC: wm5102: Replace usage deprecated SOC_VALUE_ENUM macro ASoC: wm8994: Replace usage deprecated MUX/ENUM macros ASoC: wm8995: Replace usage deprecated MUX/ENUM macros ASoC: wm8997: Replace usage deprecated MUX/ENUM macros ASoC: Remove deprecated ENUM/MUX macros include/sound/soc-dapm.h | 12 ------------ include/sound/soc.h | 2 -- sound/soc/codecs/adau1373.c | 7 +++---- sound/soc/codecs/adav80x.c | 8 ++++---- sound/soc/codecs/arizona.h | 4 ++-- sound/soc/codecs/max98090.c | 5 ++--- sound/soc/codecs/mc13783.c | 8 ++++---- sound/soc/codecs/pcm512x.c | 2 +- sound/soc/codecs/rt5640.c | 4 ++-- sound/soc/codecs/wm2200.c | 4 ++-- sound/soc/codecs/wm5100.c | 4 ++-- sound/soc/codecs/wm5102.c | 14 +++++++------- sound/soc/codecs/wm5110.c | 18 +++++++++--------- sound/soc/codecs/wm8988.c | 8 ++++---- sound/soc/codecs/wm8994.c | 12 ++++++------ sound/soc/codecs/wm8995.c | 10 ++++------ sound/soc/codecs/wm8997.c | 12 ++++++------ 17 files changed, 58 insertions(+), 76 deletions(-) -- 1.8.0

2 18

[alsa-devel] [PATCH 1/6] ASoC: pcm512x: Use CONFIG_PM_RUNTIME macro
by Sachin Kamat 14 Apr '14

14 Apr '14

Fixes the following compilation warnings: sound/soc/codecs/pcm512x.c:520:12: warning: ‘pcm512x_suspend’ defined but not used [-Wunused-function] sound/soc/codecs/pcm512x.c:545:12: warning: ‘pcm512x_resume’ defined but not used [-Wunused-function] Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/codecs/pcm512x.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 4b4c0c7..5112420 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -517,6 +517,7 @@ void pcm512x_remove(struct device *dev) } EXPORT_SYMBOL_GPL(pcm512x_remove); +#ifdef CONFIG_PM_RUNTIME static int pcm512x_suspend(struct device *dev) { struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); @@ -578,6 +579,7 @@ static int pcm512x_resume(struct device *dev) return 0; } +#endif const struct dev_pm_ops pcm512x_pm_ops = { SET_RUNTIME_PM_OPS(pcm512x_suspend, pcm512x_resume, NULL) -- 1.7.9.5

2 11

[alsa-devel] [PATCH 0/7] ASoC:davinci-mcasp: bc polarity fix and proper format support
by Peter Ujfalusi 14 Apr '14

14 Apr '14

Hi, The bit clock polarity has been configured incorrectly in the McASP driver for a long time. IB_NF, NB_IF and IB_IF was not correct on the receive side since they were selecting the same edge as we used for the tx. The driver only had support for DSP_B mode (and probably AC97, but I can not test that). All other formats were broken (DSP_A, I2S, LEFT_J, etc). With this series the following formats will be supported by the McASP driver: - DSP_B - AC97 - DSP_A (new) - I2S (new) - LEFT_J (new) Non supported modes will return error from now on. Regards, Peter --- Peter Ujfalusi (7): ASoC: davinci-mcasp: Fix bit clock polarity settings ASoC: davinci-mcasp: Format data delay configuration enhancement ASoC: davinci-mcasp: Support for DSP_A format ASoC: davinci-mcasp: Move the FS polarity change out from the switch case ASoC: davinci-mcasp: Add support for I2S format ASoC: davinci-mcasp: Support for LEFT_J format ASoC: davinci-mcasp: Remove excess empty lines from davinci_mcasp_set_dai_fmt() sound/soc/davinci/davinci-mcasp.c | 76 +++++++++++++++++++++++++-------------- 1 file changed, 50 insertions(+), 26 deletions(-) -- 1.9.1

3 13

[alsa-devel] iMX233 audio capture: glitches on task switch?
by Gilles Grégoire 14 Apr '14

14 Apr '14

Dear alsa developers, I need help debugging a driver for the builtin codec of Freescale's iMX233 SoC. I am facing an issue when capturing audio: garbage data (glitches) appears periodically in the recorded audio data. Audio playback looks fine, as far as I can tell. I am using kernel 3.13.6. A driver by Freescale exists for a 2.6.35 kernel but it was never submitted to mainline. Therefore, the driver I use is a port of Freescale's driver to kernel 3.13.6 by Michal Ulianko. You can find it in the attached patch+dts. Please note that this is still a work in progress. The driver is actually quite simple: it simply sets up the codec/cpu DAI and uses the generic dma engine through snd_dmaengine_pcm_register() to make the audio run. I am using arecord to capture the audio from the default input (the mic), which is _not_ connected in my setup. The command line I am using is the following: $ arecord --disable-format --disable-resample --fatal-errors --disable-channels --buffer-size=32768 --period-size=4096 -r 22050 -f S16_LE -c 2 -d 1 /tmp/test.wav What happens: the recorded audio data contains glitches at periodic intervals which contain garbage data. No underrun/overrun occurs during capture. I suspect that the problem is not in the driver's code itself, but could be the symptom of a bug somewhere else (possibly in mxs dma code), because of the following observations I did: * there are 2 kind of glitches: short (~3 samples) and long (~25 samples) depending on the kernel configuration. Both kinds may exists on the same capture. * short glitches exist only with CONFIG_HZ_PERIODIC=y. The period of those glitches is _exactly_ 4ms, which corresponds to the HZ value in my kernel config (250Hz). * long glitches are always there. The period of those glitches is directly related to the --period-bytes parameter of arecord: halving the --period-bytes parameter halves the period of the glitches. You can find attached a screenshot showing the glitches for the "dynticks" case (where only long glitches exist) and for the "periodic timer" case (where both kinds are visible). These observations lead me to believe that the glitches are related to task switching. Here is my best guess: * the short glitches come from the HZ tick invoking the scheduler. * the long glitches come from the kernel code waking up the arecord process when enough audio data was received. I spent a long time trying to narrow down the problem with no success. My main problem is that I have no idea what kind of bug could cause these symptoms. This is where I need help: I would welcome any advice on what to look for. Best regards, -- Gilles

2 2

[alsa-devel] snd-usb-audio issues??
by Richard Storer 14 Apr '14

14 Apr '14

Hi, this is my first time post on the ML so apologies if my message is not to the required standard. I am currently using a Raspberry Pi ARM computer it's running a customised version of Debian called Raspbian: Linux raspberrypi 3.10.36+ #665 PREEMPT Wed Apr 9 15:01:53 BST 2014 armv6l GNU/Linux For audio playback and capture I am using a Creative Sound Blaster Live 24bit external: ID 041e:3040 Creative Technology, Ltd SoundBlaster Live! 24-bit External SB0490 I am currently experiencing 2 problems with this device which I believe are related to the snd-usb-audio driver. My first issue is for some reason booting Linux with my device already attached to the USB port produces no audio output, however capture is functioning. I've tried mute, unmute via alsamixer but it seems like the amplifier circuit inside hasn't turned itself on properly. The cure for this (and 100% reproducible) would be to unplug / replug the USB cable or simply connect to USB after Linux has fully loaded. I don't understand why this should be the case surely it doesn't matter at what stage the device is connected?? My next issue I feel is a little bit more serious and from what I understand affects all versions of Linux even the x86 distros. I use my Sound Blaster to capture stereo audio via Line-In which I then stream in realtime via Darkice. While Line-In capture does function on this device the signal levels are extremely low. Unfortunately there is no Line-In mixer control for my device the only capture control I have is Mic and that doesn't control Line-In. I have search for many months for a solution to the lack of Line-In control but I have come to the conclusion that something is broken in the driver or it got missed out. Is anyone aware of these issues or could at least offer a solution to increase the Line-In mixer levels maybe communicate with the USB device directly without ALSA's assistance?? Richard S.

3 8

[alsa-devel] [PATCH 1/1] driver/snd: es18xx driver should use udelay error
by Li, Zhen-Hua 14 Apr '14

14 Apr '14

From: "Li, Zhen-Hua" <zhen-hual(a)hp.com> udelay with more than 20000 may cause __bad_udelay. Use mdelay for instead. Signed-off-by: Li, Zhen-Hua <zhen-hual(a)hp.com> --- sound/isa/es18xx.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 1c16830..8d3a3b6 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -520,7 +520,7 @@ static int snd_es18xx_playback1_trigger(struct snd_es18xx *chip, snd_es18xx_mixer_write(chip, 0x78, 0x93); #ifdef AVOID_POPS /* Avoid pops */ - udelay(100000); + mudelay(100); if (chip->caps & ES18XX_PCM2) /* Restore Audio 2 volume */ snd_es18xx_mixer_write(chip, 0x7C, chip->audio2_vol); @@ -537,7 +537,7 @@ static int snd_es18xx_playback1_trigger(struct snd_es18xx *chip, /* Stop DMA */ snd_es18xx_mixer_write(chip, 0x78, 0x00); #ifdef AVOID_POPS - udelay(25000); + mdelay(25); if (chip->caps & ES18XX_PCM2) /* Set Audio 2 volume to 0 */ snd_es18xx_mixer_write(chip, 0x7C, 0); @@ -596,7 +596,7 @@ static int snd_es18xx_capture_prepare(struct snd_pcm_substream *substream) snd_es18xx_write(chip, 0xA5, count >> 8); #ifdef AVOID_POPS - udelay(100000); + mdelay(100); #endif /* Set format */ @@ -691,7 +691,7 @@ static int snd_es18xx_playback2_trigger(struct snd_es18xx *chip, snd_es18xx_write(chip, 0xB8, 0x05); #ifdef AVOID_POPS /* Avoid pops */ - udelay(100000); + mdelay(100); /* Enable Audio 1 */ snd_es18xx_dsp_command(chip, 0xD1); #endif @@ -705,7 +705,7 @@ static int snd_es18xx_playback2_trigger(struct snd_es18xx *chip, snd_es18xx_write(chip, 0xB8, 0x00); #ifdef AVOID_POPS /* Avoid pops */ - udelay(25000); + mdelay(25); /* Disable Audio 1 */ snd_es18xx_dsp_command(chip, 0xD3); #endif -- 1.7.10.4

3 2

[alsa-devel] [PATCH 1/1] driver/snd: es18xx driver should use udelay error
by Li, Zhen-Hua 14 Apr '14

14 Apr '14

From: "Li, Zhen-Hua" <zhen-hual(a)hp.com> udelay with more than 20000 may cause __bad_udelay. Use mdelay for instead. Signed-off-by: Li, Zhen-Hua <zhen-hual(a)hp.com> --- sound/isa/es18xx.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index 1c16830..8d3a3b6 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -520,7 +520,7 @@ static int snd_es18xx_playback1_trigger(struct snd_es18xx *chip, snd_es18xx_mixer_write(chip, 0x78, 0x93); #ifdef AVOID_POPS /* Avoid pops */ - udelay(100000); + mudelay(100); if (chip->caps & ES18XX_PCM2) /* Restore Audio 2 volume */ snd_es18xx_mixer_write(chip, 0x7C, chip->audio2_vol); @@ -537,7 +537,7 @@ static int snd_es18xx_playback1_trigger(struct snd_es18xx *chip, /* Stop DMA */ snd_es18xx_mixer_write(chip, 0x78, 0x00); #ifdef AVOID_POPS - udelay(25000); + mdelay(25); if (chip->caps & ES18XX_PCM2) /* Set Audio 2 volume to 0 */ snd_es18xx_mixer_write(chip, 0x7C, 0); @@ -596,7 +596,7 @@ static int snd_es18xx_capture_prepare(struct snd_pcm_substream *substream) snd_es18xx_write(chip, 0xA5, count >> 8); #ifdef AVOID_POPS - udelay(100000); + mdelay(100); #endif /* Set format */ @@ -691,7 +691,7 @@ static int snd_es18xx_playback2_trigger(struct snd_es18xx *chip, snd_es18xx_write(chip, 0xB8, 0x05); #ifdef AVOID_POPS /* Avoid pops */ - udelay(100000); + mdelay(100); /* Enable Audio 1 */ snd_es18xx_dsp_command(chip, 0xD1); #endif @@ -705,7 +705,7 @@ static int snd_es18xx_playback2_trigger(struct snd_es18xx *chip, snd_es18xx_write(chip, 0xB8, 0x00); #ifdef AVOID_POPS /* Avoid pops */ - udelay(25000); + mdelay(25); /* Disable Audio 1 */ snd_es18xx_dsp_command(chip, 0xD3); #endif -- 1.7.10.4

3 2

[alsa-devel] Add support of alc288
by Kailang 14 Apr '14

14 Apr '14

Hi Takashi, We have new codec release for ALC288. Thanks. BR, Kailang

1 0

[alsa-devel] [PATCH] ASoC: rsnd: fix clock prepare/unprepare
by Kuninori Morimoto 14 Apr '14

14 Apr '14

From: Ben Dooks <ben.dooks(a)codethink.co.uk> As with the previous commit, before a clock can be used it must be prepared for use. Change from clk_enable() and clk_disable() to the versions of the calls which also prepare and un-prepare the clocks. Will fix warnings from the clock code when this is used. Signed-off-by: Ben Dooks <ben.dooks(a)codethink.co.uk> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This patch is for current linus/master tree sound/soc/sh/rcar/src.c | 4 ++-- sound/soc/sh/rcar/ssi.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index 6232b7d..4d0720e 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -258,7 +258,7 @@ static int rsnd_src_init(struct rsnd_mod *mod, { struct rsnd_src *src = rsnd_mod_to_src(mod); - clk_enable(src->clk); + clk_prepare_enable(src->clk); return 0; } @@ -269,7 +269,7 @@ static int rsnd_src_quit(struct rsnd_mod *mod, { struct rsnd_src *src = rsnd_mod_to_src(mod); - clk_disable(src->clk); + clk_disable_unprepare(src->clk); return 0; } diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 4b7e206..1d8387c 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -171,7 +171,7 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, u32 cr; if (0 == ssi->usrcnt) { - clk_enable(ssi->clk); + clk_prepare_enable(ssi->clk); if (rsnd_dai_is_clk_master(rdai)) { if (rsnd_ssi_clk_from_parent(ssi)) @@ -230,7 +230,7 @@ static void rsnd_ssi_hw_stop(struct rsnd_ssi *ssi, rsnd_ssi_master_clk_stop(ssi); } - clk_disable(ssi->clk); + clk_disable_unprepare(ssi->clk); } dev_dbg(dev, "ssi%d hw stopped\n", rsnd_mod_id(&ssi->mod)); -- 1.7.9.5

2 3

[alsa-devel] Behringer Uphoria UMC204 usb audio capture only pops and clicks
by Andrew Reyes 14 Apr '14

14 Apr '14

I have a usb Behringer Uphoria UMC204 and it seems to work fine at 44.1khz full duplex but anything higher than that I get consistent pops and clicks. I notice it's on the capure side of things as when I have it in playback only it's quiet. I have to say I'm testing this using JACK so this is how I noticed it on the capture only side of things. I'm not sure exactly whats needed to debug this issue. Any help would be appreciated. Thanks in advance.

1 0

[alsa-devel] Invalid PCM pointer with CMI8738 and pcm.c buffer underuns
by andoru 13 Apr '14

13 Apr '14

Continued from here: http://article.gmane.org/gmane.linux.alsa.devel/121575 > http://www.alsa-project.org/main/index.php/XRUN_Debug > > # Enable basic debugging and dump stack, check hardware pointer on the > period update > # Usefull to just see, if PCM stream is stopped for a reason (usually wrong > audio process timing from scheduler) > # And to check the values from driver > > echo 11 > /proc/asound/card0/pcm0p/xrun_debug Okay, I not compiled my own kernel with CONFIG_SND_PCM_XRUN_DEBUG, CONFIG_SND_VERBOSE_PROCFS, CONFIG_SND_DEBUG all enabled in the kernel configuration, I have an xrun_debug file, but it doesn't seem to list anything, then I also tried with cat instead of echo, and just a 0 showed up: ___________________________________________________________________ $ echo 11 > /proc/asound/card0/pcm0p/xrun_debug bash: /proc/asound/card0/pcm0p/xrun_debug: Permission denied # echo 11 > /proc/asound/card0/pcm0p/xrun_debug bash: /proc/asound/card0/pcm0p/xrun_debug: Permission denied $ cat /proc/asound/card0/pcm0p/xrun_debug 0 ___________________________________________________________________

2 3

[alsa-devel] Behringer Uphoria UMC204 usb audio capture only pops and clicks
by Andrew Reyes 13 Apr '14

13 Apr '14

I have a usb Behringer Uphoria UMC204 and it seems to work fine at 44.1khz full duplex but anything higher than that I get consistent pops and clicks. I notice it's on the capure side of things as when I have it in playback only it's quiet. I have to say I'm testing this using JACK so this is how I noticed it on the capture only side of things. I'm not sure exactly whats needed to debug this issue. Any help would be appreciated. Thanks in advance.

1 0

[alsa-devel] [PATCH] ALSA: pcm: 'BUG:' message unnecessarily triggers kerneloops
by Tim Gardner 11 Apr '14

11 Apr '14

BugLink: http://bugs.launchpad.net/bugs/1305480 The kerneloops-daemon scans dmesg for common crash signatures, among which is 'BUG:'. The message emitted by the PCM library is really a warning, so the most expedient thing to do seems to be to change the string. Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Mark Brown <broonie(a)linaro.org> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: JongHo Kim <furmuwon(a)gmail.com> Signed-off-by: Tim Gardner <tim.gardner(a)canonical.com> --- sound/core/pcm_lib.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index ce83def..2a0563d 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -345,7 +345,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, snd_pcm_debug_name(substream, name, sizeof(name)); xrun_log_show(substream); pcm_err(substream->pcm, - "BUG: %s, pos = %ld, buffer size = %ld, period size = %ld\n", + "OVER RUN: %s, pos = %ld, buffer size = %ld, period size = %ld\n", name, pos, runtime->buffer_size, runtime->period_size); } -- 1.7.9.5

2 1

[alsa-devel] [PATCH v6] ASoC: fsl_sai: Add clock controls for SAI
by Nicolin Chen 11 Apr '14

11 Apr '14

The SAI mainly has the following clocks: bus clock control and configure registers and to generate synchronous interrupts and DMA requests. mclk1, mclk2, mclk3 to generate the bit clock when the receiver or transmitter is configured for an internally generated bit clock. So this patch adds these clocks and their clock controls to the driver. [ To concern the old DTB cases, I've added a bit of extra code to make the driver compatible with them. And by marking clock NULL if failed to get, the clk_prepare() or clk_get_rate() would easily return 0 so no further path should be broken. -- by Nicolin ] Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> Acked-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Changelog v6: * Dropped the dtsi change since it's no more demanded with this patch. * Removed Shawn's ACK since we no long change dtsi in this case. v5: * Dropped mclk preparing and enabling since we are not using them currecntly. * Made the change compatible to the old DTB. * Added returned error value print. v4: * Merged into single patch. * Fixed bus clock ID on vf610. v3: * Use int type for ret instead of u32. * Added Acked-by and Tested-by from Xiubo Li. v2: * Appended two extra mclks to the driver since SAI actually has three. * Renamed clock name to 'bus' and 'mclk' according to the reference manual. .../devicetree/bindings/sound/fsl-sai.txt | 9 +++-- sound/soc/fsl/fsl_sai.c | 38 ++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 4 +++ 3 files changed, 46 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 35c09fe..0f4e238 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -10,7 +10,8 @@ Required properties: - compatible: Compatible list, contains "fsl,vf610-sai" or "fsl,imx6sx-sai". - reg: Offset and length of the register set for the device. - clocks: Must contain an entry for each entry in clock-names. -- clock-names : Must include the "sai" entry. +- clock-names : Must include the "bus" for register access and "mclk1" "mclk2" + "mclk3" for bit clock and frame clock providing. - dmas : Generic dma devicetree binding as described in Documentation/devicetree/bindings/dma/dma.txt. - dma-names : Two dmas have to be defined, "tx" and "rx". @@ -30,8 +31,10 @@ sai2: sai@40031000 { reg = <0x40031000 0x1000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_sai2_1>; - clocks = <&clks VF610_CLK_SAI2>; - clock-names = "sai"; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; dma-names = "tx", "rx"; dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index db9f75e..5fff2e1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -401,7 +401,15 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + struct device *dev = &sai->pdev->dev; u32 reg; + int ret; + + ret = clk_prepare_enable(sai->bus_clk); + if (ret) { + dev_err(dev, "failed to enable bus clock: %d\n", ret); + return ret; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = FSL_SAI_TCR3; @@ -427,6 +435,8 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, ~FSL_SAI_CR3_TRCE); + + clk_disable_unprepare(sai->bus_clk); } static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { @@ -559,7 +569,8 @@ static int fsl_sai_probe(struct platform_device *pdev) struct fsl_sai *sai; struct resource *res; void __iomem *base; - int irq, ret; + char tmp[8]; + int irq, ret, i; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -582,12 +593,35 @@ static int fsl_sai_probe(struct platform_device *pdev) return PTR_ERR(base); sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, - "sai", base, &fsl_sai_regmap_config); + "bus", base, &fsl_sai_regmap_config); + + /* Compatible with old DTB cases */ + if (IS_ERR(sai->regmap)) + sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, + "sai", base, &fsl_sai_regmap_config); if (IS_ERR(sai->regmap)) { dev_err(&pdev->dev, "regmap init failed\n"); return PTR_ERR(sai->regmap); } + /* No error out for old DTB cases but only mark the clock NULL */ + sai->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(sai->bus_clk)) { + dev_err(&pdev->dev, "failed to get bus clock: %ld\n", + PTR_ERR(sai->bus_clk)); + sai->bus_clk = NULL; + } + + for (i = 0; i < FSL_SAI_MCLK_MAX; i++) { + sprintf(tmp, "mclk%d", i + 1); + sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); + if (IS_ERR(sai->mclk_clk[i])) { + dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", + i + 1, PTR_ERR(sai->mclk_clk[i])); + sai->mclk_clk[i] = NULL; + } + } + irq = platform_get_irq(pdev, 0); if (irq < 0) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 677670d..0e6c9f5 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -119,6 +119,8 @@ #define FSL_SAI_CLK_MAST2 2 #define FSL_SAI_CLK_MAST3 3 +#define FSL_SAI_MCLK_MAX 3 + /* SAI data transfer numbers per DMA request */ #define FSL_SAI_MAXBURST_TX 6 #define FSL_SAI_MAXBURST_RX 6 @@ -126,6 +128,8 @@ struct fsl_sai { struct platform_device *pdev; struct regmap *regmap; + struct clk *bus_clk; + struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; bool big_endian_regs; bool big_endian_data; -- 1.8.4

2 1

[alsa-devel] Broken alc562x codec in linus/master
by Andrew Lunn 10 Apr '14

10 Apr '14

Hi Mark I found a few issues with the changes you have made to sound/soc/codecs/alc5623.c which is currently in linus master branch. The patch causing the trouble seems to be: commit 0cd257bf9b9b0cbb4fa1a5c988a232506997867c Author: Mark Brown <broonie(a)linaro.org> Date: Thu Feb 20 09:04:06 2014 +0900 ASoC: alc5623: Convert to direct regmap API usage Convert to directly use the regmap API, allowing us to eliminate the last user of the ASoC level I/O implementations (there are still open coded I/O implementations in drivers), avoiding duplicating code in regmap. We no longer cache the entire CODEC register map on probe since the more advanced cache infrastructure in regmap is able to fill the cache on demand. Signed-off-by: Mark Brown <broonie(a)linaro.org> At boot i see: alc562x-codec 0-001a: unknown or wrong codec alc562x-codec 0-001a: Expected 10ec:21, got ec10:2103 There are two issues here: - vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1); + ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID1, &vid1); It looks like these are returning different endianness. This explains the 10ec vs ec10. Can the regmap flip the endianness? The second is: - vid2 = i2c_smbus_read_byte_data(client, ALC5623_VENDOR_ID2); - if (vid2 < 0) { - dev_err(&client->dev, "failed to read I2C\n"); - return -EIO; + ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID2, &vid2); + if (ret < 0) { + dev_err(&client->dev, "failed to read vendor ID2: %d\n", ret); + return ret; It was reading a byte, but now it seems to get a word. Hence 21 vs 2103. Once the endianness is sorted out, we can mask for the lower byte. Thanks Andrew

1 0

[alsa-devel] [PATCH v6] ASoC: fsl_sai: Add clock controls for SAI
by Nicolin Chen 10 Apr '14

10 Apr '14

The SAI mainly has the following clocks: bus clock control and configure registers and to generate synchronous interrupts and DMA requests. mclk1, mclk2, mclk3 to generate the bit clock when the receiver or transmitter is configured for an internally generated bit clock. So this patch adds these clocks and their clock controls to the driver. [ To concern the old DTB cases, I've added a bit of extra code to make the driver compatible with them. And by marking clock NULL if failed to get, the clk_prepare() or clk_get_rate() would easily return 0 so no further path should be broken. -- by Nicolin ] Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> Acked-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- Changelog v6: * Dropped the dtsi change since it's no more demanded with this patch. * Removed Shawn's ACK since we no long change dtsi in this case. v5: * Dropped mclk preparing and enabling since we are not using them currecntly. * Made the change compatible to the old DTB. * Added returned error value print. v4: * Merged into single patch. * Fixed bus clock ID on vf610. v3: * Use int type for ret instead of u32. * Added Acked-by and Tested-by from Xiubo Li. v2: * Appended two extra mclks to the driver since SAI actually has three. * Renamed clock name to 'bus' and 'mclk' according to the reference manual. .../devicetree/bindings/sound/fsl-sai.txt | 9 +++-- sound/soc/fsl/fsl_sai.c | 38 ++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 4 +++ 3 files changed, 46 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 35c09fe..0f4e238 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -10,7 +10,8 @@ Required properties: - compatible: Compatible list, contains "fsl,vf610-sai" or "fsl,imx6sx-sai". - reg: Offset and length of the register set for the device. - clocks: Must contain an entry for each entry in clock-names. -- clock-names : Must include the "sai" entry. +- clock-names : Must include the "bus" for register access and "mclk1" "mclk2" + "mclk3" for bit clock and frame clock providing. - dmas : Generic dma devicetree binding as described in Documentation/devicetree/bindings/dma/dma.txt. - dma-names : Two dmas have to be defined, "tx" and "rx". @@ -30,8 +31,10 @@ sai2: sai@40031000 { reg = <0x40031000 0x1000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_sai2_1>; - clocks = <&clks VF610_CLK_SAI2>; - clock-names = "sai"; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; dma-names = "tx", "rx"; dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index db9f75e..5fff2e1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -401,7 +401,15 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + struct device *dev = &sai->pdev->dev; u32 reg; + int ret; + + ret = clk_prepare_enable(sai->bus_clk); + if (ret) { + dev_err(dev, "failed to enable bus clock: %d\n", ret); + return ret; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = FSL_SAI_TCR3; @@ -427,6 +435,8 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, ~FSL_SAI_CR3_TRCE); + + clk_disable_unprepare(sai->bus_clk); } static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { @@ -559,7 +569,8 @@ static int fsl_sai_probe(struct platform_device *pdev) struct fsl_sai *sai; struct resource *res; void __iomem *base; - int irq, ret; + char tmp[8]; + int irq, ret, i; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -582,12 +593,35 @@ static int fsl_sai_probe(struct platform_device *pdev) return PTR_ERR(base); sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, - "sai", base, &fsl_sai_regmap_config); + "bus", base, &fsl_sai_regmap_config); + + /* Compatible with old DTB cases */ + if (IS_ERR(sai->regmap)) + sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, + "sai", base, &fsl_sai_regmap_config); if (IS_ERR(sai->regmap)) { dev_err(&pdev->dev, "regmap init failed\n"); return PTR_ERR(sai->regmap); } + /* No error out for old DTB cases but only mark the clock NULL */ + sai->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(sai->bus_clk)) { + dev_err(&pdev->dev, "failed to get bus clock: %ld\n", + PTR_ERR(sai->bus_clk)); + sai->bus_clk = NULL; + } + + for (i = 0; i < FSL_SAI_MCLK_MAX; i++) { + sprintf(tmp, "mclk%d", i + 1); + sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); + if (IS_ERR(sai->mclk_clk[i])) { + dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", + i + 1, PTR_ERR(sai->mclk_clk[i])); + sai->mclk_clk[i] = NULL; + } + } + irq = platform_get_irq(pdev, 0); if (irq < 0) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 677670d..0e6c9f5 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -119,6 +119,8 @@ #define FSL_SAI_CLK_MAST2 2 #define FSL_SAI_CLK_MAST3 3 +#define FSL_SAI_MCLK_MAX 3 + /* SAI data transfer numbers per DMA request */ #define FSL_SAI_MAXBURST_TX 6 #define FSL_SAI_MAXBURST_RX 6 @@ -126,6 +128,8 @@ struct fsl_sai { struct platform_device *pdev; struct regmap *regmap; + struct clk *bus_clk; + struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; bool big_endian_regs; bool big_endian_data; -- 1.8.4

1 0

[alsa-devel] [PATCH v5] ASoC: fsl_sai: Add clock controls for SAI
by Nicolin Chen 10 Apr '14

10 Apr '14

The SAI mainly has the following clocks: bus clock control and configure registers and to generate synchronous interrupts and DMA requests. mclk1, mclk2, mclk3 to generate the bit clock when the receiver or transmitter is configured for an internally generated bit clock. So this patch adds these clocks and their clock controls to the driver, meanwhile, corrects the existing DTS accordingly so those platforms can benifit from the further feature with different clock sources. [ To concern the old DTB cases, I've added a bit of extra code to make the driver compatible with them. And by marking clock NULL if failed to get, the clk_prepare() or clk_get_rate() would easily return 0 so no further path should be broken. -- by Nicolin ] Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> Acked-by: Xiubo Li <Li.Xiubo(a)freescale.com> Acked-by: Shawn Guo <shawn.guo(a)linaro.org> --- Changelog v5: * Dropped mclk preparing and enabling since we are not using them currecntly. * Made the change compatible to the old DTB. * Added returned error value print. v4: * Merged into single patch. * Fixed bus clock ID on vf610. v3: * Use int type for ret instead of u32. * Added Acked-by and Tested-by from Xiubo Li. v2: * Appended two extra mclks to the driver since SAI actually has three. * Renamed clock name to 'bus' and 'mclk' according to the reference manual. .../devicetree/bindings/sound/fsl-sai.txt | 9 +++-- arch/arm/boot/dts/vf610.dtsi | 6 ++-- sound/soc/fsl/fsl_sai.c | 38 ++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 4 +++ 4 files changed, 50 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 35c09fe..0f4e238 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -10,7 +10,8 @@ Required properties: - compatible: Compatible list, contains "fsl,vf610-sai" or "fsl,imx6sx-sai". - reg: Offset and length of the register set for the device. - clocks: Must contain an entry for each entry in clock-names. -- clock-names : Must include the "sai" entry. +- clock-names : Must include the "bus" for register access and "mclk1" "mclk2" + "mclk3" for bit clock and frame clock providing. - dmas : Generic dma devicetree binding as described in Documentation/devicetree/bindings/dma/dma.txt. - dma-names : Two dmas have to be defined, "tx" and "rx". @@ -30,8 +31,10 @@ sai2: sai@40031000 { reg = <0x40031000 0x1000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_sai2_1>; - clocks = <&clks VF610_CLK_SAI2>; - clock-names = "sai"; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; dma-names = "tx", "rx"; dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; diff --git a/arch/arm/boot/dts/vf610.dtsi b/arch/arm/boot/dts/vf610.dtsi index d31ce1b..4c3cd59 100644 --- a/arch/arm/boot/dts/vf610.dtsi +++ b/arch/arm/boot/dts/vf610.dtsi @@ -139,8 +139,10 @@ compatible = "fsl,vf610-sai"; reg = <0x40031000 0x1000>; interrupts = <0 86 0x04>; - clocks = <&clks VF610_CLK_SAI2>; - clock-names = "sai"; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; status = "disabled"; }; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index db9f75e..5fff2e1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -401,7 +401,15 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + struct device *dev = &sai->pdev->dev; u32 reg; + int ret; + + ret = clk_prepare_enable(sai->bus_clk); + if (ret) { + dev_err(dev, "failed to enable bus clock: %d\n", ret); + return ret; + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = FSL_SAI_TCR3; @@ -427,6 +435,8 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, ~FSL_SAI_CR3_TRCE); + + clk_disable_unprepare(sai->bus_clk); } static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { @@ -559,7 +569,8 @@ static int fsl_sai_probe(struct platform_device *pdev) struct fsl_sai *sai; struct resource *res; void __iomem *base; - int irq, ret; + char tmp[8]; + int irq, ret, i; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -582,12 +593,35 @@ static int fsl_sai_probe(struct platform_device *pdev) return PTR_ERR(base); sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, - "sai", base, &fsl_sai_regmap_config); + "bus", base, &fsl_sai_regmap_config); + + /* Compatible with old DTB cases */ + if (IS_ERR(sai->regmap)) + sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, + "sai", base, &fsl_sai_regmap_config); if (IS_ERR(sai->regmap)) { dev_err(&pdev->dev, "regmap init failed\n"); return PTR_ERR(sai->regmap); } + /* No error out for old DTB cases but only mark the clock NULL */ + sai->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(sai->bus_clk)) { + dev_err(&pdev->dev, "failed to get bus clock: %ld\n", + PTR_ERR(sai->bus_clk)); + sai->bus_clk = NULL; + } + + for (i = 0; i < FSL_SAI_MCLK_MAX; i++) { + sprintf(tmp, "mclk%d", i + 1); + sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); + if (IS_ERR(sai->mclk_clk[i])) { + dev_err(&pdev->dev, "failed to get mclk%d clock: %ld\n", + i + 1, PTR_ERR(sai->mclk_clk[i])); + sai->mclk_clk[i] = NULL; + } + } + irq = platform_get_irq(pdev, 0); if (irq < 0) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 677670d..0e6c9f5 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -119,6 +119,8 @@ #define FSL_SAI_CLK_MAST2 2 #define FSL_SAI_CLK_MAST3 3 +#define FSL_SAI_MCLK_MAX 3 + /* SAI data transfer numbers per DMA request */ #define FSL_SAI_MAXBURST_TX 6 #define FSL_SAI_MAXBURST_RX 6 @@ -126,6 +128,8 @@ struct fsl_sai { struct platform_device *pdev; struct regmap *regmap; + struct clk *bus_clk; + struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; bool big_endian_regs; bool big_endian_data; -- 1.8.4

2 2

[alsa-devel] [PATCH 1/2] ASoC: rt5640: Add the string "rt5639" to the list of I2C device IDs
by Oder Chiou 10 Apr '14

10 Apr '14

The patch adds the string "rt5639" to the list of I2C device IDs. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index d672f44..2e7132b 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1,5 +1,5 @@ /* - * rt5640.c -- RT5640 ALSA SoC audio codec driver + * rt5640.c -- RT5640/RT5639 ALSA SoC audio codec driver * * Copyright 2011 Realtek Semiconductor Corp. * Author: Johnny Hsu <johnnyhsu(a)realtek.com> @@ -2164,6 +2164,7 @@ static const struct regmap_config rt5640_regmap = { static const struct i2c_device_id rt5640_i2c_id[] = { { "rt5640", 0 }, + { "rt5639", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); @@ -2322,6 +2323,6 @@ static struct i2c_driver rt5640_i2c_driver = { }; module_i2c_driver(rt5640_i2c_driver); -MODULE_DESCRIPTION("ASoC RT5640 driver"); +MODULE_DESCRIPTION("ASoC RT5640/RT5639 driver"); MODULE_AUTHOR("Johnny Hsu <johnnyhsu(a)realtek.com>"); MODULE_LICENSE("GPL v2"); -- 1.8.1.1.439.g50a6b54

2 3

[alsa-devel] [PATCH v4] ASoC: fsl_sai: Add clock controls for SAI
by Nicolin Chen 10 Apr '14

10 Apr '14

The SAI mainly has the following clocks: bus clock control and configure registers and to generate synchronous interrupts and DMA requests. mclk1, mclk2, mclk3 to generate the bit clock when the receiver or transmitter is configured for an internally generated bit clock. So this patch adds these clocks and their clock controls to the driver, meanwhile, corrects the existing DTS accordingly so those platforms can benifit from the further feature with different clock sources. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> CC: Xiubo Li <Li.Xiubo(a)freescale.com> Acked-by: Shawn Guo <shawn.guo(a)linaro.org> --- @Xiubo Even though you've tested it before, I'd still like to wait for your test result to this newer version. Changelog v4: * Merged into single patch. * Fixed bus clock ID on vf610. v3: * Use int type for ret instead of u32. * Added Acked-by and Tested-by from Xiubo Li. v2: * Appended two extra mclks to the driver since SAI actually has three. * Renamed clock name to 'bus' and 'mclk' according to the reference manual. .../devicetree/bindings/sound/fsl-sai.txt | 9 ++-- arch/arm/boot/dts/vf610.dtsi | 6 ++- sound/soc/fsl/fsl_sai.c | 51 ++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 4 ++ 4 files changed, 61 insertions(+), 9 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 35c09fe..0f4e238 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -10,7 +10,8 @@ Required properties: - compatible: Compatible list, contains "fsl,vf610-sai" or "fsl,imx6sx-sai". - reg: Offset and length of the register set for the device. - clocks: Must contain an entry for each entry in clock-names. -- clock-names : Must include the "sai" entry. +- clock-names : Must include the "bus" for register access and "mclk1" "mclk2" + "mclk3" for bit clock and frame clock providing. - dmas : Generic dma devicetree binding as described in Documentation/devicetree/bindings/dma/dma.txt. - dma-names : Two dmas have to be defined, "tx" and "rx". @@ -30,8 +31,10 @@ sai2: sai@40031000 { reg = <0x40031000 0x1000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_sai2_1>; - clocks = <&clks VF610_CLK_SAI2>; - clock-names = "sai"; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; dma-names = "tx", "rx"; dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; diff --git a/arch/arm/boot/dts/vf610.dtsi b/arch/arm/boot/dts/vf610.dtsi index d31ce1b..4c3cd59 100644 --- a/arch/arm/boot/dts/vf610.dtsi +++ b/arch/arm/boot/dts/vf610.dtsi @@ -139,8 +139,10 @@ compatible = "fsl,vf610-sai"; reg = <0x40031000 0x1000>; interrupts = <0 86 0x04>; - clocks = <&clks VF610_CLK_SAI2>; - clock-names = "sai"; + clocks = <&clks VF610_CLK_PLATFORM_BUS>, + <&clks VF610_CLK_SAI2>, + <&clks 0>, <&clks 0>; + clock-names = "bus", "mclk1", "mclk2", "mclk3"; status = "disabled"; }; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index db9f75e..7cd4af9 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -401,7 +401,23 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 reg; + struct device *dev = &sai->pdev->dev; + u32 reg, i; + int ret; + + ret = clk_prepare_enable(sai->bus_clk); + if (ret) { + dev_err(dev, "failed to enable bus clock\n"); + return ret; + } + + for (i = 0; i < FSL_SAI_MCLK_MAX; i++) { + ret = clk_prepare_enable(sai->mclk_clk[i]); + if (ret) { + dev_err(dev, "failed to enable mclk%d clock\n", i + 1); + goto err; + } + } if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = FSL_SAI_TCR3; @@ -412,13 +428,20 @@ static int fsl_sai_startup(struct snd_pcm_substream *substream, FSL_SAI_CR3_TRCE); return 0; + +err: + for (; i > 0; i--) + clk_disable_unprepare(sai->mclk_clk[i - 1]); + clk_disable_unprepare(sai->bus_clk); + + return ret; } static void fsl_sai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); - u32 reg; + u32 reg, i; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = FSL_SAI_TCR3; @@ -427,6 +450,10 @@ static void fsl_sai_shutdown(struct snd_pcm_substream *substream, regmap_update_bits(sai->regmap, reg, FSL_SAI_CR3_TRCE, ~FSL_SAI_CR3_TRCE); + + for (i = 0; i < FSL_SAI_MCLK_MAX; i++) + clk_disable_unprepare(sai->mclk_clk[i]); + clk_disable_unprepare(sai->bus_clk); } static const struct snd_soc_dai_ops fsl_sai_pcm_dai_ops = { @@ -559,7 +586,8 @@ static int fsl_sai_probe(struct platform_device *pdev) struct fsl_sai *sai; struct resource *res; void __iomem *base; - int irq, ret; + char tmp[8]; + int irq, ret, i; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -582,12 +610,27 @@ static int fsl_sai_probe(struct platform_device *pdev) return PTR_ERR(base); sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, - "sai", base, &fsl_sai_regmap_config); + "bus", base, &fsl_sai_regmap_config); if (IS_ERR(sai->regmap)) { dev_err(&pdev->dev, "regmap init failed\n"); return PTR_ERR(sai->regmap); } + sai->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(sai->bus_clk)) { + dev_err(&pdev->dev, "failed to get bus clock\n"); + return PTR_ERR(sai->bus_clk); + } + + for (i = 0; i < FSL_SAI_MCLK_MAX; i++) { + sprintf(tmp, "mclk%d", i + 1); + sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); + if (IS_ERR(sai->mclk_clk[i])) { + dev_err(&pdev->dev, "failed to get mclk%d clock\n", i + 1); + return PTR_ERR(sai->mclk_clk[i]); + } + } + irq = platform_get_irq(pdev, 0); if (irq < 0) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 677670d..0e6c9f5 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -119,6 +119,8 @@ #define FSL_SAI_CLK_MAST2 2 #define FSL_SAI_CLK_MAST3 3 +#define FSL_SAI_MCLK_MAX 3 + /* SAI data transfer numbers per DMA request */ #define FSL_SAI_MAXBURST_TX 6 #define FSL_SAI_MAXBURST_RX 6 @@ -126,6 +128,8 @@ struct fsl_sai { struct platform_device *pdev; struct regmap *regmap; + struct clk *bus_clk; + struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; bool big_endian_regs; bool big_endian_data; -- 1.8.4

2 1

[alsa-devel] [PATCH] ASoC: core: Fix Sparse incompatible types warning
by Krzysztof Kozlowski 10 Apr '14

10 Apr '14

Fix following Sparse warning: sound/soc/soc-core.c:252:20: error: incompatible types in comparison expression (different type sizes) Signed-off-by: Krzysztof Kozlowski <k.kozlowski(a)samsung.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 051c006281f5..41f6178249df 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -249,7 +249,7 @@ static ssize_t codec_reg_write_file(struct file *file, struct snd_soc_codec *codec = file->private_data; int ret; - buf_size = min(count, (sizeof(buf)-1)); + buf_size = min(count, (size_t)(sizeof(buf)-1)); if (copy_from_user(buf, user_buf, buf_size)) return -EFAULT; buf[buf_size] = 0; -- 1.8.3.2

4 7

[alsa-devel] [PATCH 2/2] ALSA: hda - Add driver for Tegra SoC HDA
by Dylan Reid 10 Apr '14

10 Apr '14

This adds a driver for the HDA block in Tegra SoCs. The HDA bus is used to communicate with the HDMI codec on Tegra124. Most of the code is re-used from the Intel/PCI HDA driver. It brings over only two of thw module params, power_save and probe_mask. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- .../bindings/sound/nvidia,tegra124-hda.txt | 20 + sound/pci/hda/Kconfig | 14 + sound/pci/hda/Makefile | 2 + sound/pci/hda/hda_tegra.c | 695 +++++++++++++++++++++ 4 files changed, 731 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra124-hda.txt create mode 100644 sound/pci/hda/hda_tegra.c diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra124-hda.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra124-hda.txt new file mode 100644 index 0000000..c9256d3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra124-hda.txt @@ -0,0 +1,20 @@ +NVIDIA Tegra124 HDA controller + +Required properties: +- compatible : "nvidia,tegra124-hda" +- reg : Should contain the HDA registers location and length. +- interrupts : The interrupt from the hda controller. +- clocks : Must contain an entry for each required entry in clock-names. +- clock-names : Must include the following entries: hda, hdacodec_2x, hda2hdmi + +Example: + +hda@70030000 { + compatible = "nvidia,tegra124-hda"; + reg = <0x0 0x70030000 0x10000>; + interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&tegra_car TEGRA124_CLK_HDA>, + <&tegra_car TEGRA124_CLK_HDA2HDMI>, + <&tegra_car TEGRA124_CLK_HDA2CODEC_2X>; + clock-names = "hda", "hda2hdmi", "hdacodec_2x"; +}; diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index ac17c3f..4f6ee6b 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -20,6 +20,20 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. +config SND_HDA_TEGRA + tristate "Tegra HD Audio" + select SND_HDA + help + Say Y here to support the HDA controller present in Nvidia + Tegra SoCs + + This options enables support for the HD Audio controller + present in some Nvidia Tegra SoCs, used to communicate audio + to the HDMI output. + + To compile this driver as a module, choose M here: the module + will be called snd-hda-tegra. + if SND_HDA config SND_HDA_DSP_LOADER diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index d0d0c19..194f3093 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,5 +1,6 @@ snd-hda-intel-objs := hda_intel.o snd-hda-controller-objs := hda_controller.o +snd-hda-tegra-objs := hda_tegra.o # for haswell power well snd-hda-intel-$(CONFIG_SND_HDA_I915) += hda_i915.o @@ -47,3 +48,4 @@ obj-$(CONFIG_SND_HDA_CODEC_HDMI) += snd-hda-codec-hdmi.o # otherwise the codec patches won't be hooked before the PCI probe # when built in kernel obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-intel.o +obj-$(CONFIG_SND_HDA_TEGRA) += snd-hda-tegra.o diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c new file mode 100644 index 0000000..c36bbca --- /dev/null +++ b/sound/pci/hda/hda_tegra.c @@ -0,0 +1,695 @@ +/* + * + * Implementation of primary alsa driver code base for Tegra HDA. + * + * 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. + * + * This program is distributed in the hope that it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + */ + +#include <linux/clk.h> +#include <linux/clocksource.h> +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/dma-mapping.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/mutex.h> +#include <linux/reboot.h> +#include <linux/io.h> +#include <linux/pm_runtime.h> +#include <linux/of_device.h> +#include <linux/time.h> +#include <linux/completion.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <linux/firmware.h> +#include "hda_codec.h" +#include "hda_controller.h" +#include "hda_priv.h" + +#define DRV_NAME "tegra-hda" + +/* Defines for Nvidia Tegra HDA support */ +#define NVIDIA_TEGRA_HDA_BAR0_OFFSET 0x8000 + +#define NVIDIA_TEGRA_HDA_CFG_CMD_OFFSET 0x1004 +#define NVIDIA_TEGRA_HDA_CFG_BAR0_OFFSET 0x1010 + +#define NVIDIA_TEGRA_HDA_ENABLE_IO_SPACE (1 << 0) +#define NVIDIA_TEGRA_HDA_ENABLE_MEM_SPACE (1 << 1) +#define NVIDIA_TEGRA_HDA_ENABLE_BUS_MASTER (1 << 2) +#define NVIDIA_TEGRA_HDA_ENABLE_SERR (1 << 8) +#define NVIDIA_TEGRA_HDA_DISABLE_INTR (1 << 10) +#define NVIDIA_TEGRA_HDA_BAR0_INIT_PROGRAM 0xFFFFFFFF +#define NVIDIA_TEGRA_HDA_BAR0_FINAL_PROGRAM (1 << 14) + +/* IPFS */ +#define NVIDIA_TEGRA_HDA_IPFS_CONFIG 0x180 +#define NVIDIA_TEGRA_HDA_IPFS_EN_FPCI 0x1 + +#define NVIDIA_TEGRA_HDA_IPFS_FPCI_BAR0 0x80 +#define NVIDIA_TEGRA_HDA_FPCI_BAR0_START 0x40 + +#define NVIDIA_TEGRA_HDA_IPFS_INTR_MASK 0x188 +#define NVIDIA_TEGRA_HDA_IPFS_EN_INTR (1 << 16) + +struct hda_tegra_data { + struct azx chip; + struct platform_device *pdev; + struct clk **platform_clks; + int platform_clk_count; + void __iomem *remap_config_addr; +}; + +static int probe_mask[SNDRV_CARDS] = {[0 ... (SNDRV_CARDS-1)] = -1}; +module_param_array(probe_mask, int, NULL, 0444); +MODULE_PARM_DESC(probe_mask, "Bitmask to probe codecs (default = -1)."); + +static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; +static int *power_save_addr = &power_save; +module_param(power_save, bint, 0644); +MODULE_PARM_DESC(power_save, + "Automatic power-saving timeout (in seconds, 0 = disable)."); + +/* + * DMA page allocation ops. + */ +static int dma_alloc_pages(struct azx *chip, + int type, + size_t size, + struct snd_dma_buffer *buf) +{ + return snd_dma_alloc_pages(type, + chip->card->dev, + size, buf); +} + +static void dma_free_pages(struct azx *chip, struct snd_dma_buffer *buf) +{ + snd_dma_free_pages(buf); +} + +static int substream_alloc_pages(struct azx *chip, + struct snd_pcm_substream *substream, + size_t size) +{ + struct azx_dev *azx_dev = get_azx_dev(substream); + + azx_dev->bufsize = 0; + azx_dev->period_bytes = 0; + azx_dev->format_val = 0; + return snd_pcm_lib_malloc_pages(substream, size); +} + +static int substream_free_pages(struct azx *chip, + struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +/* + * Register access ops. Tegra HDA register access is DWORD only. + */ +#define MASK_LONG_ALIGN 0x3UL +#define SHIFT_BYTE 3 +#define SHIFT_BITS(addr) \ + (((unsigned int)(addr) & MASK_LONG_ALIGN) << SHIFT_BYTE) +#define ADDR_ALIGN_L(addr) \ + (void *)((unsigned int)(addr) & ~MASK_LONG_ALIGN) +#define MASK(bits) (BIT(bits) - 1) +#define MASK_REG(addr, bits) (MASK(bits) << SHIFT_BITS(addr)) + +static void tegra_hda_writel(u32 value, u32 *addr) +{ + writel(value, addr); +} + +static u32 tegra_hda_readl(u32 *addr) +{ + return readl(addr); +} + +static void tegra_hda_writew(u16 value, u16 *addr) +{ + unsigned int shift_bits = SHIFT_BITS(addr); + writel((readl(ADDR_ALIGN_L(addr)) & ~MASK_REG(addr, 16)) | + ((value) << shift_bits), ADDR_ALIGN_L(addr)); +} + +static u16 tegra_hda_readw(u16 *addr) +{ + return (readl(ADDR_ALIGN_L(addr)) >> SHIFT_BITS(addr)) & MASK(16); +} + +static void tegra_hda_writeb(u8 value, u8 *addr) +{ + writel((readl(ADDR_ALIGN_L(addr)) & ~MASK_REG(addr, 8)) | + ((value) << SHIFT_BITS(addr)), ADDR_ALIGN_L(addr)); +} + +static u8 tegra_hda_readb(u8 *addr) +{ + return (readl(ADDR_ALIGN_L(addr)) >> SHIFT_BITS(addr)) & MASK(8); +} + +static const struct hda_controller_ops tegra_hda_reg_ops = { + .reg_writel = tegra_hda_writel, + .reg_readl = tegra_hda_readl, + .reg_writew = tegra_hda_writew, + .reg_readw = tegra_hda_readw, + .reg_writeb = tegra_hda_writeb, + .reg_readb = tegra_hda_readb, + .dma_alloc_pages = dma_alloc_pages, + .dma_free_pages = dma_free_pages, + .substream_alloc_pages = substream_alloc_pages, + .substream_free_pages = substream_free_pages, +}; + +static int tegra_hda_acquire_irq(struct azx *chip, int do_disconnect) +{ + struct hda_tegra_data *tdata = + container_of(chip, struct hda_tegra_data, chip); + int irq_id = platform_get_irq(tdata->pdev, 0); + + if (devm_request_irq(chip->card->dev, irq_id, azx_interrupt, + IRQF_SHARED, KBUILD_MODNAME, chip)) { + dev_err(chip->card->dev, + "unable to grab IRQ %d, disabling device\n", + irq_id); + if (do_disconnect) + snd_card_disconnect(chip->card); + return -1; + } + chip->irq = irq_id; + return 0; +} + +static void tegra_hda_reg_update_bits(void __iomem *base, unsigned int reg, + unsigned int mask, unsigned int val) +{ + unsigned int data; + + data = readl(base + reg); + data &= ~mask; + data |= (val & mask); + writel(data, base + reg); +} + +static void hda_tegra_init(struct hda_tegra_data *tdata) +{ + /*Enable the PCI access */ + tegra_hda_reg_update_bits(tdata->remap_config_addr, + NVIDIA_TEGRA_HDA_IPFS_CONFIG, + NVIDIA_TEGRA_HDA_IPFS_EN_FPCI, + NVIDIA_TEGRA_HDA_IPFS_EN_FPCI); + /* Enable MEM/IO space and bus master */ + tegra_hda_reg_update_bits(tdata->remap_config_addr, + NVIDIA_TEGRA_HDA_CFG_CMD_OFFSET, 0x507, + NVIDIA_TEGRA_HDA_ENABLE_MEM_SPACE | + NVIDIA_TEGRA_HDA_ENABLE_IO_SPACE | + NVIDIA_TEGRA_HDA_ENABLE_BUS_MASTER | + NVIDIA_TEGRA_HDA_ENABLE_SERR); + tegra_hda_reg_update_bits(tdata->remap_config_addr, + NVIDIA_TEGRA_HDA_CFG_BAR0_OFFSET, 0xFFFFFFFF, + NVIDIA_TEGRA_HDA_BAR0_INIT_PROGRAM); + tegra_hda_reg_update_bits(tdata->remap_config_addr, + NVIDIA_TEGRA_HDA_CFG_BAR0_OFFSET, 0xFFFFFFFF, + NVIDIA_TEGRA_HDA_BAR0_FINAL_PROGRAM); + tegra_hda_reg_update_bits(tdata->remap_config_addr, + NVIDIA_TEGRA_HDA_IPFS_FPCI_BAR0, 0xFFFFFFFF, + NVIDIA_TEGRA_HDA_FPCI_BAR0_START); + tegra_hda_reg_update_bits(tdata->remap_config_addr, + NVIDIA_TEGRA_HDA_IPFS_INTR_MASK, + NVIDIA_TEGRA_HDA_IPFS_EN_INTR, + NVIDIA_TEGRA_HDA_IPFS_EN_INTR); + + return; +} + +static void hda_tegra_enable_clocks(struct hda_tegra_data *data) +{ + int i; + + for (i = 0; i < data->platform_clk_count; i++) + clk_prepare_enable(data->platform_clks[i]); +} + +static void hda_tegra_disable_clocks(struct hda_tegra_data *data) +{ + int i; + + for (i = 0; i < data->platform_clk_count; i++) + clk_disable_unprepare(data->platform_clks[i]); +} + +#if CONFIG_PM_SLEEP +/* + * power management + */ +static int tegra_hda_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct azx_pcm *p; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + list_for_each_entry(p, &chip->pcm_list, list) + snd_pcm_suspend_all(p->pcm); + if (chip->initialized) + snd_hda_suspend(chip->bus); + + if (pm_runtime_active(card->dev)) { + azx_stop_chip(chip); + azx_enter_link_reset(chip); + } + + return 0; +} + +static int tegra_hda_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct hda_tegra_data *tdata = + container_of(chip, struct hda_tegra_data, chip); + + if (pm_runtime_active(card->dev)) { + hda_tegra_init(tdata); + + azx_init_chip(chip, 1); + + snd_hda_resume(chip->bus); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + } + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + +#ifdef CONFIG_PM_RUNTIME +static int tegra_hda_runtime_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct hda_tegra_data *tdata = + container_of(chip, struct hda_tegra_data, chip); + + /* enable controller wake up event */ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | + STATESTS_INT_MASK); + + azx_stop_chip(chip); + azx_enter_link_reset(chip); + hda_tegra_disable_clocks(tdata); + return 0; +} + +static int tegra_hda_runtime_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct hda_bus *bus; + struct hda_codec *codec; + struct hda_tegra_data *tdata = + container_of(chip, struct hda_tegra_data, chip); + int status; + + hda_tegra_enable_clocks(tdata); + + /* Read STATESTS before controller reset */ + status = azx_readw(chip, STATESTS); + + hda_tegra_init(tdata); + + azx_init_chip(chip, 1); + + snd_hda_resume(chip->bus); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + + bus = chip->bus; + if (status && bus) { + list_for_each_entry(codec, &bus->codec_list, list) + if (status & (1 << codec->addr)) + queue_delayed_work(codec->bus->workq, + &codec->jackpoll_work, + codec->jackpoll_interval); + } + + /* disable controller Wake Up event*/ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & ~STATESTS_INT_MASK); + + return 0; +} + +#endif /* CONFIG_PM_RUNTIME */ + +static const struct dev_pm_ops azx_pm = { + SET_SYSTEM_SLEEP_PM_OPS(tegra_hda_suspend, tegra_hda_resume) + SET_RUNTIME_PM_OPS(tegra_hda_runtime_suspend, + tegra_hda_runtime_resume, + NULL) +}; + +/* + * reboot notifier for hang-up problem at power-down + */ +static int tegra_hda_halt(struct notifier_block *nb, unsigned long event, + void *buf) +{ + struct azx *chip = container_of(nb, struct azx, reboot_notifier); + snd_hda_bus_reboot_notify(chip->bus); + azx_stop_chip(chip); + return NOTIFY_OK; +} + +static void tegra_hda_notifier_register(struct azx *chip) +{ + chip->reboot_notifier.notifier_call = tegra_hda_halt; + register_reboot_notifier(&chip->reboot_notifier); +} + +static void tegra_hda_notifier_unregister(struct azx *chip) +{ + if (chip->reboot_notifier.notifier_call) + unregister_reboot_notifier(&chip->reboot_notifier); +} + +/* + * destructor + */ +static int tegra_hda_free(struct azx *chip) +{ + int i; + + if (chip->running) + pm_runtime_get_noresume(chip->card->dev); + + tegra_hda_notifier_unregister(chip); + + if (chip->initialized) { + for (i = 0; i < chip->num_streams; i++) + azx_stream_stop(chip, &chip->azx_dev[i]); + azx_stop_chip(chip); + } + + azx_free_stream_pages(chip); + + return 0; +} + +static int tegra_hda_dev_free(struct snd_device *device) +{ + return tegra_hda_free(device->device_data); +} + +static const char * const tegra_clk_names[] = { + "hda", + "hda2codec_2x", + "hda2hdmi", +}; + +static int hda_tegra_init_chip(struct azx *chip) +{ + struct hda_tegra_data *tdata = + container_of(chip, struct hda_tegra_data, chip); + struct device *dev = &tdata->pdev->dev; + struct resource *res, *region; + int i; + + tdata->platform_clk_count = ARRAY_SIZE(tegra_clk_names); + tdata->platform_clks = devm_kzalloc(dev, + tdata->platform_clk_count * + sizeof(*tdata->platform_clks), + GFP_KERNEL); + if (!tdata->platform_clks) + return -ENOMEM; + + for (i = 0; i < tdata->platform_clk_count; i++) { + tdata->platform_clks[i] = devm_clk_get(dev, tegra_clk_names[i]); + if (IS_ERR(tdata->platform_clks[i])) + return PTR_ERR(tdata->platform_clks[i]); + } + + res = platform_get_resource(tdata->pdev, IORESOURCE_MEM, 0); + if (!res) + return -EINVAL; + + region = devm_request_mem_region(dev, res->start, + resource_size(res), + tdata->pdev->name); + if (!region) + return -ENOMEM; + + chip->addr = res->start; + chip->remap_addr = devm_ioremap(dev, res->start, resource_size(res)); + if (!chip->remap_addr) + return -ENXIO; + + tdata->remap_config_addr = chip->remap_addr; + chip->remap_addr += NVIDIA_TEGRA_HDA_BAR0_OFFSET; + chip->addr += NVIDIA_TEGRA_HDA_BAR0_OFFSET; + + hda_tegra_enable_clocks(tdata); + + hda_tegra_init(tdata); + + return 0; +} + +static void power_down_all_codecs(struct azx *chip) +{ + /* The codecs were powered up in snd_hda_codec_new(). + * Now all initialization done, so turn them down if possible + */ + struct hda_codec *codec; + list_for_each_entry(codec, &chip->bus->codec_list, list) + snd_hda_power_down(codec); +} + +static int tegra_hda_first_init(struct azx *chip) +{ + struct snd_card *card = chip->card; + int err; + unsigned short gcap; + + err = hda_tegra_init_chip(chip); + if (err) + return err; + + if (tegra_hda_acquire_irq(chip, 0) < 0) + return -EBUSY; + + synchronize_irq(chip->irq); + + gcap = azx_readw(chip, GCAP); + dev_dbg(card->dev, "chipset global capabilities = 0x%x\n", gcap); + + /* read number of streams from GCAP register instead of using + * hardcoded value + */ + chip->capture_streams = (gcap >> 8) & 0x0f; + chip->playback_streams = (gcap >> 12) & 0x0f; + if (!chip->playback_streams && !chip->capture_streams) { + /* gcap didn't give any info, switching to old method */ + chip->playback_streams = ICH6_NUM_PLAYBACK; + chip->capture_streams = ICH6_NUM_CAPTURE; + } + chip->capture_index_offset = 0; + chip->playback_index_offset = chip->capture_streams; + chip->num_streams = chip->playback_streams + chip->capture_streams; + chip->azx_dev = devm_kzalloc(card->dev, + chip->num_streams * sizeof(*chip->azx_dev), + GFP_KERNEL); + if (!chip->azx_dev) + return -ENOMEM; + + err = azx_alloc_stream_pages(chip); + if (err < 0) + return err; + + /* initialize streams */ + azx_init_stream(chip); + + /* initialize chip */ + azx_init_chip(chip, 1); + + /* codec detection */ + if (!chip->codec_mask) { + dev_err(card->dev, "no codecs found!\n"); + return -ENODEV; + } + + strcpy(card->driver, "HDA-Tegra"); + strcpy(card->shortname, "HDA-Tegra"); + snprintf(card->longname, sizeof(card->longname), + "%s at 0x%lx irq %i", + card->shortname, chip->addr, chip->irq); + + return 0; +} + +/* + * constructor + */ +static int tegra_hda_create(struct snd_card *card, + int dev, + struct platform_device *pdev, + unsigned int driver_caps, + const struct hda_controller_ops *hda_ops, + struct hda_tegra_data *tdata) +{ + static struct snd_device_ops ops = { + .dev_free = tegra_hda_dev_free, + }; + struct azx *chip; + int err; + + chip = &tdata->chip; + + spin_lock_init(&chip->reg_lock); + mutex_init(&chip->open_mutex); + chip->card = card; + chip->ops = hda_ops; + chip->irq = -1; + chip->driver_caps = driver_caps; + chip->driver_type = driver_caps & 0xff; + chip->dev_index = dev; + INIT_LIST_HEAD(&chip->pcm_list); + INIT_LIST_HEAD(&chip->list); + + chip->position_fix[0] = POS_FIX_AUTO; + chip->position_fix[1] = POS_FIX_AUTO; + chip->codec_probe_mask = probe_mask[dev]; + + chip->single_cmd = false; + chip->snoop = true; + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + dev_err(card->dev, "Error creating device [card]!\n"); + return err; + } + + return 0; +} + +static unsigned int tegra_driver_flags = AZX_DCAPS_RIRB_DELAY | + AZX_DCAPS_PM_RUNTIME; + +static const struct of_device_id tegra_platform_hda_match[] = { + { .compatible = "nvidia,tegra124-hda", .data = &tegra_driver_flags }, + {}, +}; + +static int hda_tegra_probe(struct platform_device *pdev) +{ + static int dev; + struct snd_card *card; + struct azx *chip; + struct hda_tegra_data *tdata; + const struct of_device_id *of_id; + const unsigned int *driver_data; + int err; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + + of_id = of_match_device(tegra_platform_hda_match, &pdev->dev); + if (!of_id) + return -EINVAL; + + tdata = devm_kzalloc(&pdev->dev, sizeof(*tdata), GFP_KERNEL); + if (!tdata) + return -ENOMEM; + tdata->pdev = pdev; + chip = &tdata->chip; + + err = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + THIS_MODULE, 0, &card); + if (err < 0) { + dev_err(&pdev->dev, "Error creating card!\n"); + return err; + } + + driver_data = of_id->data; + err = tegra_hda_create(card, dev, pdev, *driver_data, + &tegra_hda_reg_ops, tdata); + if (err < 0) + goto out_free; + card->private_data = chip; + + dev_set_drvdata(&pdev->dev, card); + + err = tegra_hda_first_init(chip); + if (err < 0) + goto out_free; + + /* create codec instances */ + err = azx_codec_create(chip, NULL, 0, power_save_addr); + if (err < 0) + goto out_free; + + err = azx_codec_configure(chip); + if (err < 0) + goto out_free; + + /* create PCM streams */ + err = snd_hda_build_pcms(chip->bus); + if (err < 0) + goto out_free; + + /* create mixer controls */ + err = azx_mixer_create(chip); + if (err < 0) + goto out_free; + + err = snd_card_register(chip->card); + if (err < 0) + goto out_free; + + chip->running = 1; + pm_runtime_enable(chip->card->dev); + hda_tegra_disable_clocks(tdata); + power_down_all_codecs(chip); + tegra_hda_notifier_register(chip); + + dev++; + return 0; + +out_free: + snd_card_free(card); + return err; +} + +static int hda_tegra_remove(struct platform_device *pdev) +{ + return snd_card_free(dev_get_drvdata(&pdev->dev)); +} + +static struct platform_driver tegra_platform_hda = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + .pm = &azx_pm, + .of_match_table = tegra_platform_hda_match, + }, + .probe = hda_tegra_probe, + .remove = hda_tegra_remove, +}; +module_platform_driver(tegra_platform_hda); + +MODULE_DESCRIPTION("Tegra HDA bus driver"); +MODULE_LICENSE("GPL v2"); +MODULE_DEVICE_TABLE(of, tegra_platform_hda_match); -- 1.8.1.3.605.g02339dd

3 6

[alsa-devel] [PATCH v2 2/2] ALSA: hda - Add driver for Tegra SoC HDA
by Dylan Reid 10 Apr '14

10 Apr '14

This adds a driver for the HDA block in Tegra SoCs. The HDA bus is used to communicate with the HDMI codec on Tegra124. Most of the code is re-used from the Intel/PCI HDA driver. It brings over only two of the module params, power_save and probe_mask. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- Changes since v1: Kconfig depends on OF and ARCH_TEGRA. Fix warning on 64 bit build. Integrate a bunch of good suggestions from Thierry, including: Add resets property to device tree entry. More consistent/better naming. Remove probe mask that isn't needed. Cleaner/more readable handling of dword write restriction. --- .../bindings/sound/nvidia,tegra30-hda.txt | 27 + sound/pci/hda/Kconfig | 15 + sound/pci/hda/Makefile | 2 + sound/pci/hda/hda_tegra.c | 644 +++++++++++++++++++++ 4 files changed, 688 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt create mode 100644 sound/pci/hda/hda_tegra.c diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt new file mode 100644 index 0000000..97f6832 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.txt @@ -0,0 +1,27 @@ +NVIDIA Tegra30 HDA controller + +Required properties: +- compatible : "nvidia,tegra30-hda" +- reg : Should contain the HDA registers location and length. +- interrupts : The interrupt from the hda controller. +- clocks : Must contain an entry for each required entry in clock-names. +- clock-names : Must include the following entries: hda, hdacodec_2x, hda2hdmi +- resets : Must contain an entry for each entry in reset-names. + See ../reset/reset.txt for details. +- reset-names : Must include the following entries: hda, hdacodec_2x, hda2hdmi + +Example: + +hda@70030000 { + compatible = "nvidia,tegra124-hda", "nvidia,tegra30-hda"; + reg = <0x0 0x70030000 0x10000>; + interrupts = <GIC_SPI 81 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&tegra_car TEGRA124_CLK_HDA>, + <&tegra_car TEGRA124_CLK_HDA2HDMI>, + <&tegra_car TEGRA124_CLK_HDA2CODEC_2X>; + clock-names = "hda", "hda2hdmi", "hdacodec_2x"; + resets = <&tegra_car 125>, /* hda */ + <&tegra_car 128>; /* hda2hdmi */ + <&tegra_car 111>, /* hda2codec_2x */ + reset-names = "hda", "hda2hdmi", "hdacodec_2x"; +}; diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index ac17c3f..f17c016 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -20,6 +20,21 @@ config SND_HDA_INTEL To compile this driver as a module, choose M here: the module will be called snd-hda-intel. +config SND_HDA_TEGRA + tristate "NVIDIA Tegra HD Audio" + depends on OF && ARCH_TEGRA + select SND_HDA + help + Say Y here to support the HDA controller present in NVIDIA + Tegra SoCs + + This options enables support for the HD Audio controller + present in some NVIDIA Tegra SoCs, used to communicate audio + to the HDMI output. + + To compile this driver as a module, choose M here: the module + will be called snd-hda-tegra. + if SND_HDA config SND_HDA_DSP_LOADER diff --git a/sound/pci/hda/Makefile b/sound/pci/hda/Makefile index d0d0c19..194f3093 100644 --- a/sound/pci/hda/Makefile +++ b/sound/pci/hda/Makefile @@ -1,5 +1,6 @@ snd-hda-intel-objs := hda_intel.o snd-hda-controller-objs := hda_controller.o +snd-hda-tegra-objs := hda_tegra.o # for haswell power well snd-hda-intel-$(CONFIG_SND_HDA_I915) += hda_i915.o @@ -47,3 +48,4 @@ obj-$(CONFIG_SND_HDA_CODEC_HDMI) += snd-hda-codec-hdmi.o # otherwise the codec patches won't be hooked before the PCI probe # when built in kernel obj-$(CONFIG_SND_HDA_INTEL) += snd-hda-intel.o +obj-$(CONFIG_SND_HDA_TEGRA) += snd-hda-tegra.o diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c new file mode 100644 index 0000000..75f246b --- /dev/null +++ b/sound/pci/hda/hda_tegra.c @@ -0,0 +1,644 @@ +/* + * + * Implementation of primary alsa driver code base for NVIDIA Tegra HDA. + * + * This program is free software; you can redistribute it and/or modify it + * under the terms and conditions of the GNU General Public License, + * version 2, as published by the Free Software Foundation. + * + * This program is distributed in the hope it will be useful, but WITHOUT + * ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or + * FITNESS FOR A PARTICULAR PURPOSE. See the GNU General Public License for + * more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + */ + +#include <linux/clk.h> +#include <linux/clocksource.h> +#include <linux/completion.h> +#include <linux/delay.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/io.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/mutex.h> +#include <linux/of_device.h> +#include <linux/pm_runtime.h> +#include <linux/reboot.h> +#include <linux/slab.h> +#include <linux/time.h> + +#include <sound/core.h> +#include <sound/initval.h> + +#include "hda_codec.h" +#include "hda_controller.h" +#include "hda_priv.h" + +/* Defines for NVIDIA Tegra HDA support */ +#define HDA_BAR0 0x8000 + +#define HDA_CFG_CMD 0x1004 +#define HDA_CFG_BAR0 0x1010 + +#define HDA_ENABLE_IO_SPACE (1 << 0) +#define HDA_ENABLE_MEM_SPACE (1 << 1) +#define HDA_ENABLE_BUS_MASTER (1 << 2) +#define HDA_ENABLE_SERR (1 << 8) +#define HDA_DISABLE_INTR (1 << 10) +#define HDA_BAR0_INIT_PROGRAM 0xFFFFFFFF +#define HDA_BAR0_FINAL_PROGRAM (1 << 14) + +/* IPFS */ +#define HDA_IPFS_CONFIG 0x180 +#define HDA_IPFS_EN_FPCI 0x1 + +#define HDA_IPFS_FPCI_BAR0 0x80 +#define HDA_FPCI_BAR0_START 0x40 + +#define HDA_IPFS_INTR_MASK 0x188 +#define HDA_IPFS_EN_INTR (1 << 16) + +struct hda_tegra { + struct azx chip; + struct device *dev; + struct clk *hda_clk; + struct clk *hda2codec_2x_clk; + struct clk *hda2hdmi_clk; + void __iomem *regs; +}; + +static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; +module_param(power_save, bint, 0644); +MODULE_PARM_DESC(power_save, + "Automatic power-saving timeout (in seconds, 0 = disable)."); + +/* + * DMA page allocation ops. + */ +static int dma_alloc_pages(struct azx *chip, int type, size_t size, + struct snd_dma_buffer *buf) +{ + return snd_dma_alloc_pages(type, chip->card->dev, size, buf); +} + +static void dma_free_pages(struct azx *chip, struct snd_dma_buffer *buf) +{ + snd_dma_free_pages(buf); +} + +static int substream_alloc_pages(struct azx *chip, + struct snd_pcm_substream *substream, + size_t size) +{ + struct azx_dev *azx_dev = get_azx_dev(substream); + + azx_dev->bufsize = 0; + azx_dev->period_bytes = 0; + azx_dev->format_val = 0; + return snd_pcm_lib_malloc_pages(substream, size); +} + +static int substream_free_pages(struct azx *chip, + struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +/* + * Register access ops. Tegra HDA register access is DWORD only. + */ +static void hda_tegra_writel(u32 value, u32 *addr) +{ + writel(value, addr); +} + +static u32 hda_tegra_readl(u32 *addr) +{ + return readl(addr); +} + +static void hda_tegra_writew(u16 value, u16 *addr) +{ + unsigned long shift = ((uintptr_t)(addr) & 0x3) << 3; + void *dword_addr = (void *)((uintptr_t)(addr) & ~0x3); + u32 v; + + v = readl(dword_addr); + v &= ~(0xffff << shift); + v |= value << shift; + writel(v, dword_addr); +} + +static u16 hda_tegra_readw(u16 *addr) +{ + unsigned long shift = ((uintptr_t)(addr) & 0x3) << 3; + void *dword_addr = (void *)((uintptr_t)(addr) & ~0x3); + u32 v; + + v = readl(dword_addr); + return (v >> shift) & 0xffff; +} + +static void hda_tegra_writeb(u8 value, u8 *addr) +{ + unsigned long shift = ((uintptr_t)(addr) & 0x3) << 3; + void *dword_addr = (void *)((uintptr_t)(addr) & ~0x3); + u32 v; + + v = readl(dword_addr); + v &= ~(0xff << shift); + v |= value << shift; + writel(v, dword_addr); +} + +static u8 hda_tegra_readb(u8 *addr) +{ + unsigned long shift = ((uintptr_t)(addr) & 0x3) << 3; + void *dword_addr = (void *)((uintptr_t)(addr) & ~0x3); + u32 v; + + v = readl(dword_addr); + return (v >> shift) & 0xff; +} + +static const struct hda_controller_ops hda_tegra_ops = { + .reg_writel = hda_tegra_writel, + .reg_readl = hda_tegra_readl, + .reg_writew = hda_tegra_writew, + .reg_readw = hda_tegra_readw, + .reg_writeb = hda_tegra_writeb, + .reg_readb = hda_tegra_readb, + .dma_alloc_pages = dma_alloc_pages, + .dma_free_pages = dma_free_pages, + .substream_alloc_pages = substream_alloc_pages, + .substream_free_pages = substream_free_pages, +}; + +static void hda_tegra_init(struct hda_tegra *hda) +{ + u32 v; + + /*Enable the PCI access */ + v = readl(hda->regs + HDA_IPFS_CONFIG); + v |= HDA_IPFS_EN_FPCI; + writel(v, hda->regs + HDA_IPFS_CONFIG); + + /* Enable MEM/IO space and bus master */ + v = readl(hda->regs + HDA_CFG_CMD); + v &= ~HDA_DISABLE_INTR; + v |= HDA_ENABLE_MEM_SPACE | HDA_ENABLE_IO_SPACE | + HDA_ENABLE_BUS_MASTER | HDA_ENABLE_SERR; + writel(v, hda->regs + HDA_CFG_CMD); + + writel(HDA_BAR0_INIT_PROGRAM, hda->regs + HDA_CFG_BAR0); + writel(HDA_BAR0_FINAL_PROGRAM, hda->regs + HDA_CFG_BAR0); + writel(HDA_FPCI_BAR0_START, hda->regs + HDA_IPFS_FPCI_BAR0); + + v = readl(hda->regs + HDA_IPFS_INTR_MASK); + v |= HDA_IPFS_EN_INTR; + writel(v, hda->regs + HDA_IPFS_INTR_MASK); +} + +static int hda_tegra_enable_clocks(struct hda_tegra *data) +{ + int rc; + + rc = clk_prepare_enable(data->hda_clk); + if (rc) + return rc; + rc = clk_prepare_enable(data->hda2codec_2x_clk); + if (rc) + goto disable_hda; + rc = clk_prepare_enable(data->hda2hdmi_clk); + if (rc) + goto disable_codec_2x; + + return 0; + +disable_codec_2x: + clk_disable_unprepare(data->hda2codec_2x_clk); +disable_hda: + clk_disable_unprepare(data->hda_clk); + return rc; +} + +static void hda_tegra_disable_clocks(struct hda_tegra *data) +{ + clk_disable_unprepare(data->hda2hdmi_clk); + clk_disable_unprepare(data->hda2codec_2x_clk); + clk_disable_unprepare(data->hda_clk); +} + +#ifdef CONFIG_PM_SLEEP +/* + * power management + */ +static int hda_tegra_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct azx_pcm *p; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + list_for_each_entry(p, &chip->pcm_list, list) + snd_pcm_suspend_all(p->pcm); + if (chip->initialized) + snd_hda_suspend(chip->bus); + + if (pm_runtime_active(card->dev)) { + azx_stop_chip(chip); + azx_enter_link_reset(chip); + } + + return 0; +} + +static int hda_tegra_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); + + if (pm_runtime_active(card->dev)) { + hda_tegra_init(hda); + + azx_init_chip(chip, 1); + + snd_hda_resume(chip->bus); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + } + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + +#ifdef CONFIG_PM_RUNTIME +static int hda_tegra_runtime_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); + + /* enable controller wake up event */ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) | + STATESTS_INT_MASK); + + azx_stop_chip(chip); + azx_enter_link_reset(chip); + hda_tegra_disable_clocks(hda); + return 0; +} + +static int hda_tegra_runtime_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct hda_codec *codec; + struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); + int status; + + hda_tegra_enable_clocks(hda); + + /* Read STATESTS before controller reset */ + status = azx_readw(chip, STATESTS); + + hda_tegra_init(hda); + + azx_init_chip(chip, 1); + + snd_hda_resume(chip->bus); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + + if (status && chip->bus) { + list_for_each_entry(codec, &chip->bus->codec_list, list) + if (status & (1 << codec->addr)) + queue_delayed_work(codec->bus->workq, + &codec->jackpoll_work, + codec->jackpoll_interval); + } + + /* disable controller Wake Up event*/ + azx_writew(chip, WAKEEN, azx_readw(chip, WAKEEN) & ~STATESTS_INT_MASK); + + return 0; +} + +#endif /* CONFIG_PM_RUNTIME */ + +static const struct dev_pm_ops hda_tegra_pm = { + SET_SYSTEM_SLEEP_PM_OPS(hda_tegra_suspend, hda_tegra_resume) + SET_RUNTIME_PM_OPS(hda_tegra_runtime_suspend, + hda_tegra_runtime_resume, + NULL) +}; + +/* + * reboot notifier for hang-up problem at power-down + */ +static int hda_tegra_halt(struct notifier_block *nb, unsigned long event, + void *buf) +{ + struct azx *chip = container_of(nb, struct azx, reboot_notifier); + snd_hda_bus_reboot_notify(chip->bus); + azx_stop_chip(chip); + return NOTIFY_OK; +} + +static void hda_tegra_notifier_register(struct azx *chip) +{ + chip->reboot_notifier.notifier_call = hda_tegra_halt; + register_reboot_notifier(&chip->reboot_notifier); +} + +static void hda_tegra_notifier_unregister(struct azx *chip) +{ + if (chip->reboot_notifier.notifier_call) + unregister_reboot_notifier(&chip->reboot_notifier); +} + +/* + * destructor + */ +static int hda_tegra_dev_free(struct snd_device *device) +{ + int i; + struct azx *chip = device->device_data; + + if (chip->running) + pm_runtime_get_noresume(chip->card->dev); + + hda_tegra_notifier_unregister(chip); + + if (chip->initialized) { + for (i = 0; i < chip->num_streams; i++) + azx_stream_stop(chip, &chip->azx_dev[i]); + azx_stop_chip(chip); + } + + azx_free_stream_pages(chip); + + return 0; +} + +static int hda_tegra_init_chip(struct azx *chip, struct platform_device *pdev) +{ + struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); + struct device *dev = hda->dev; + struct resource *res; + int err; + + hda->hda_clk = devm_clk_get(dev, "hda"); + if (IS_ERR(hda->hda_clk)) + return PTR_ERR(hda->hda_clk); + hda->hda2codec_2x_clk = devm_clk_get(dev, "hda2codec_2x"); + if (IS_ERR(hda->hda2codec_2x_clk)) + return PTR_ERR(hda->hda2codec_2x_clk); + hda->hda2hdmi_clk = devm_clk_get(dev, "hda2hdmi"); + if (IS_ERR(hda->hda2hdmi_clk)) + return PTR_ERR(hda->hda2hdmi_clk); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + hda->regs = devm_ioremap_resource(dev, res); + if (IS_ERR(chip->remap_addr)) + return PTR_ERR(chip->remap_addr); + + chip->remap_addr = hda->regs + HDA_BAR0; + chip->addr = res->start + HDA_BAR0; + + err = hda_tegra_enable_clocks(hda); + if (err) + return err; + + hda_tegra_init(hda); + + return 0; +} + +/* + * The codecs were powered up in snd_hda_codec_new(). + * Now all initialization done, so turn them down if possible + */ +static void power_down_all_codecs(struct azx *chip) +{ + struct hda_codec *codec; + list_for_each_entry(codec, &chip->bus->codec_list, list) + snd_hda_power_down(codec); +} + +static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) +{ + struct snd_card *card = chip->card; + int err; + unsigned short gcap; + int irq_id = platform_get_irq(pdev, 0); + + err = hda_tegra_init_chip(chip, pdev); + if (err) + return err; + + err = devm_request_irq(chip->card->dev, irq_id, azx_interrupt, + IRQF_SHARED, KBUILD_MODNAME, chip); + if (err) { + dev_err(chip->card->dev, + "unable to request IRQ %d, disabling device\n", + irq_id); + return err; + } + chip->irq = irq_id; + + synchronize_irq(chip->irq); + + gcap = azx_readw(chip, GCAP); + dev_dbg(card->dev, "chipset global capabilities = 0x%x\n", gcap); + + /* read number of streams from GCAP register instead of using + * hardcoded value + */ + chip->capture_streams = (gcap >> 8) & 0x0f; + chip->playback_streams = (gcap >> 12) & 0x0f; + if (!chip->playback_streams && !chip->capture_streams) { + /* gcap didn't give any info, switching to old method */ + chip->playback_streams = ICH6_NUM_PLAYBACK; + chip->capture_streams = ICH6_NUM_CAPTURE; + } + chip->capture_index_offset = 0; + chip->playback_index_offset = chip->capture_streams; + chip->num_streams = chip->playback_streams + chip->capture_streams; + chip->azx_dev = devm_kcalloc(card->dev, chip->num_streams, + sizeof(*chip->azx_dev), GFP_KERNEL); + if (!chip->azx_dev) + return -ENOMEM; + + err = azx_alloc_stream_pages(chip); + if (err < 0) + return err; + + /* initialize streams */ + azx_init_stream(chip); + + /* initialize chip */ + azx_init_chip(chip, 1); + + /* codec detection */ + if (!chip->codec_mask) { + dev_err(card->dev, "no codecs found!\n"); + return -ENODEV; + } + + strcpy(card->driver, "tegra-hda"); + strcpy(card->shortname, "tegra-hda"); + snprintf(card->longname, sizeof(card->longname), + "%s at 0x%lx irq %i", + card->shortname, chip->addr, chip->irq); + + return 0; +} + +/* + * constructor + */ +static int hda_tegra_create(struct snd_card *card, + unsigned int driver_caps, + const struct hda_controller_ops *hda_ops, + struct hda_tegra *hda) +{ + static struct snd_device_ops ops = { + .dev_free = hda_tegra_dev_free, + }; + struct azx *chip; + int err; + + chip = &hda->chip; + + spin_lock_init(&chip->reg_lock); + mutex_init(&chip->open_mutex); + chip->card = card; + chip->ops = hda_ops; + chip->irq = -1; + chip->driver_caps = driver_caps; + chip->driver_type = driver_caps & 0xff; + chip->dev_index = 0; + INIT_LIST_HEAD(&chip->pcm_list); + INIT_LIST_HEAD(&chip->list); + + chip->position_fix[0] = POS_FIX_AUTO; + chip->position_fix[1] = POS_FIX_AUTO; + chip->codec_probe_mask = -1; + + chip->single_cmd = false; + chip->snoop = true; + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (err < 0) { + dev_err(card->dev, "Error creating device\n"); + return err; + } + + return 0; +} + +static const struct of_device_id hda_tegra_match[] = { + { .compatible = "nvidia,tegra30-hda" }, + {}, +}; + +static int hda_tegra_probe(struct platform_device *pdev) +{ + struct snd_card *card; + struct azx *chip; + struct hda_tegra *hda; + const struct of_device_id *of_id; + int err; + const unsigned int driver_flags = AZX_DCAPS_RIRB_DELAY | + AZX_DCAPS_PM_RUNTIME; + + of_id = of_match_device(hda_tegra_match, &pdev->dev); + if (!of_id) + return -ENODEV; + + hda = devm_kzalloc(&pdev->dev, sizeof(*hda), GFP_KERNEL); + if (!hda) + return -ENOMEM; + hda->dev = &pdev->dev; + chip = &hda->chip; + + err = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, + THIS_MODULE, 0, &card); + if (err < 0) { + dev_err(&pdev->dev, "Error creating card!\n"); + return err; + } + + err = hda_tegra_create(card, driver_flags, &hda_tegra_ops, hda); + if (err < 0) + goto out_free; + card->private_data = chip; + + dev_set_drvdata(&pdev->dev, card); + + err = hda_tegra_first_init(chip, pdev); + if (err < 0) + goto out_free; + + /* create codec instances */ + err = azx_codec_create(chip, NULL, 0, &power_save); + if (err < 0) + goto out_free; + + err = azx_codec_configure(chip); + if (err < 0) + goto out_free; + + /* create PCM streams */ + err = snd_hda_build_pcms(chip->bus); + if (err < 0) + goto out_free; + + /* create mixer controls */ + err = azx_mixer_create(chip); + if (err < 0) + goto out_free; + + err = snd_card_register(chip->card); + if (err < 0) + goto out_free; + + chip->running = 1; + pm_runtime_enable(chip->card->dev); + hda_tegra_disable_clocks(hda); + power_down_all_codecs(chip); + hda_tegra_notifier_register(chip); + + return 0; + +out_free: + snd_card_free(card); + return err; +} + +static int hda_tegra_remove(struct platform_device *pdev) +{ + return snd_card_free(dev_get_drvdata(&pdev->dev)); +} + +static struct platform_driver tegra_platform_hda = { + .driver = { + .name = "tegra-hda", + .pm = &hda_tegra_pm, + .of_match_table = hda_tegra_match, + }, + .probe = hda_tegra_probe, + .remove = hda_tegra_remove, +}; +module_platform_driver(tegra_platform_hda); + +MODULE_DESCRIPTION("Tegra HDA bus driver"); +MODULE_LICENSE("GPL v2"); +MODULE_DEVICE_TABLE(of, tegra_platform_hda_match); -- 1.8.1.3.605.g02339dd

1 0

[alsa-devel] [PATCH v2 1/2] ALSA: hda/hdmi - Add Nvidia Tegra124 HDMI support
by Dylan Reid 10 Apr '14

10 Apr '14

From: Sumit Bhattacharya <sumitb(a)nvidia.com> Add the Tegra 12x HDMI controller id to patch_hdmi. Signed-off-by: Sumit Bhattacharya <sumitb(a)nvidia.com> Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- No changes since v1 --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0cb5b89..b5d1371 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3322,6 +3322,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .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 = 0x10de0028, .name = "Tegra12x HDMI", .patch = patch_nvhdmi }, { .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 }, @@ -3377,6 +3378,7 @@ 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"); -- 1.8.1.3.605.g02339dd

1 0

[alsa-devel] [PATCH] ASoC: rt5640: add more settings for rt5639
by Oder Chiou 10 Apr '14

10 Apr '14

The patch adds the the list of OF compatible strings, the binding document and the list of I2C device IDs for rt5639. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5639.txt | 48 ++++++++++ sound/soc/codecs/rt5640.c | 103 ++++++++++++++++++++- 2 files changed, 147 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rt5639.txt diff --git a/Documentation/devicetree/bindings/sound/rt5639.txt b/Documentation/devicetree/bindings/sound/rt5639.txt new file mode 100644 index 0000000..d63c662 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5639.txt @@ -0,0 +1,48 @@ +RT5639 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5639". + +- reg : The I2C address of the device. + +- interrupts : The CODEC's interrupt output. + +Optional properties: + +- realtek,in1-differential +- realtek,in2-differential + Boolean. Indicate MIC1/2 input are differential, rather than single-ended. + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. + +Pins on the device (for linking into audio routes): + + * DMIC1 + * DMIC2 + * MICBIAS1 + * IN1P + * IN1R + * IN2P + * IN2R + * HPOL + * HPOR + * LOUTL + * LOUTR + * SPOLP + * SPOLN + * SPORP + * SPORN + +Example: + +rt5639 { + compatible = "realtek,rt5639"; + reg = <0x1c>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(W, 3) GPIO_ACTIVE_HIGH>; + realtek,ldo1-en-gpios = + <&gpio TEGRA_GPIO(V, 3) GPIO_ACTIVE_HIGH>; +}; diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index d672f44..d7d6cad 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1,5 +1,5 @@ /* - * rt5640.c -- RT5640 ALSA SoC audio codec driver + * rt5640.c -- RT5640/RT5639 ALSA SoC audio codec driver * * Copyright 2011 Realtek Semiconductor Corp. * Author: Johnny Hsu <johnnyhsu(a)realtek.com> @@ -2131,6 +2131,47 @@ static struct snd_soc_dai_driver rt5640_dai[] = { }, }; +static struct snd_soc_dai_driver rt5639_dai[] = { + { + .name = "rt5639-aif1", + .id = RT5640_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5640_STEREO_RATES, + .formats = RT5640_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5640_STEREO_RATES, + .formats = RT5640_FORMATS, + }, + .ops = &rt5640_aif_dai_ops, + }, + { + .name = "rt5639-aif2", + .id = RT5640_AIF2, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5640_STEREO_RATES, + .formats = RT5640_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5640_STEREO_RATES, + .formats = RT5640_FORMATS, + }, + .ops = &rt5640_aif_dai_ops, + }, +}; + static struct snd_soc_codec_driver soc_codec_dev_rt5640 = { .probe = rt5640_probe, .remove = rt5640_remove, @@ -2168,12 +2209,38 @@ static const struct i2c_device_id rt5640_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); +static const struct i2c_device_id rt5639_i2c_id[] = { + { "rt5639", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id rt5640_of_match[] = { + { .compatible = "realtek,rt5640", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5640_of_match); + +static const struct of_device_id rt5639_of_match[] = { + { .compatible = "realtek,rt5639", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5639_of_match); +#endif + #ifdef CONFIG_ACPI static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5640_acpi_match); + +static struct acpi_device_id rt5639_acpi_match[] = { + { "INT33CA", 0 }, + { }, +}; +MODULE_DEVICE_TABLE(acpi, rt5639_acpi_match); #endif static int rt5640_parse_dt(struct rt5640_priv *rt5640, struct device_node *np) @@ -2293,8 +2360,18 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, rt5640->hp_mute = 1; - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, - rt5640_dai, ARRAY_SIZE(rt5640_dai)); + regmap_read(rt5640->regmap, RT5640_RESET, &val); + switch (val) { + case RT5640_RESET_ID: + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, + rt5640_dai, ARRAY_SIZE(rt5640_dai)); + break; + case RT5639_RESET_ID: + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, + rt5639_dai, ARRAY_SIZE(rt5639_dai)); + break; + } + if (ret < 0) goto err; @@ -2315,6 +2392,9 @@ static struct i2c_driver rt5640_i2c_driver = { .name = "rt5640", .owner = THIS_MODULE, .acpi_match_table = ACPI_PTR(rt5640_acpi_match), +#if defined(CONFIG_OF) + .of_match_table = of_match_ptr(rt5640_of_match), +#endif }, .probe = rt5640_i2c_probe, .remove = rt5640_i2c_remove, @@ -2322,6 +2402,21 @@ static struct i2c_driver rt5640_i2c_driver = { }; module_i2c_driver(rt5640_i2c_driver); -MODULE_DESCRIPTION("ASoC RT5640 driver"); +static struct i2c_driver rt5639_i2c_driver = { + .driver = { + .name = "rt5639", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt5639_acpi_match), +#if defined(CONFIG_OF) + .of_match_table = of_match_ptr(rt5639_of_match), +#endif + }, + .probe = rt5640_i2c_probe, + .remove = rt5640_i2c_remove, + .id_table = rt5639_i2c_id, +}; +module_i2c_driver(rt5639_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5640/RT5639 driver"); MODULE_AUTHOR("Johnny Hsu <johnnyhsu(a)realtek.com>"); MODULE_LICENSE("GPL v2"); -- 1.8.1.1.439.g50a6b54

3 3

[alsa-devel] move alc892 to patch_alc882
by Kailang 10 Apr '14

10 Apr '14

Hi Takashi, The original was patch to patch_alc662. Move it into patch_alc882. Thanks. BR, Kailang

3 21

[alsa-devel] [PATCH linux-next] ALSA: usb-audio: Suppress repetitive debug messages from retire_playback_urb()
by Tim Gardner 09 Apr '14

09 Apr '14

BugLink: http://bugs.launchpad.net/bugs/1305133 Malfunctioning or slow devices can cause a flood of dmesg SPAM. I've ignored checkpatch.pl complaints about the use of printk_ratelimit() in favour of prior art in sound/usb/pcm.c. WARNING: Prefer printk_ratelimited or pr_<level>_ratelimited to printk_ratelimit + if (printk_ratelimit() && Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Eldad Zack <eldad(a)fogrefinery.com> Cc: Daniel Mack <zonque(a)gmail.com> Cc: Clemens Ladisch <clemens(a)ladisch.de> Signed-off-by: Tim Gardner <tim.gardner(a)canonical.com> --- sound/usb/pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 49de5c1..131336d 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1501,7 +1501,8 @@ static void retire_playback_urb(struct snd_usb_substream *subs, * The error should be lower than 2ms since the estimate relies * on two reads of a counter updated every ms. */ - if (abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2) + if (printk_ratelimit() && + abs(est_delay - subs->last_delay) * 1000 > runtime->rate * 2) dev_dbg(&subs->dev->dev, "delay: estimated %d, actual %d\n", est_delay, subs->last_delay); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: dapm: Add support for multi register mux
by Arun Shamanna Lakshmi 09 Apr '14

09 Apr '14

1. Modify soc_enum struct to handle pointers for reg and mask 2. Add dapm get and put APIs for multi register one hot encoded mux 3. Update snd_soc_dapm_update struct to support multiple reg update Signed-off-by: Arun S L <aruns(a)nvidia.com> Signed-off-by: Songhee Baek <sbaek(a)nvidia.com> --- include/sound/soc-dapm.h | 20 ++++- include/sound/soc.h | 33 +++++++-- sound/soc/soc-core.c | 12 +-- sound/soc/soc-dapm.c | 181 +++++++++++++++++++++++++++++++++++++++------- 4 files changed, 203 insertions(+), 43 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ef78f56..c255349 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -305,6 +305,12 @@ struct device; .get = snd_soc_dapm_get_enum_double, \ .put = snd_soc_dapm_put_enum_double, \ .private_value = (unsigned long)&xenum } +#define SOC_DAPM_ENUM_ONEHOT(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_enum_double, \ + .get = snd_soc_dapm_get_enum_onehot, \ + .put = snd_soc_dapm_put_enum_onehot, \ + .private_value = (unsigned long)&xenum } #define SOC_DAPM_ENUM_VIRT(xname, xenum) \ SOC_DAPM_ENUM(xname, xenum) #define SOC_DAPM_ENUM_EXT(xname, xenum, xget, xput) \ @@ -352,6 +358,9 @@ struct device; /* regulator widget flags */ #define SND_SOC_DAPM_REGULATOR_BYPASS 0x1 /* bypass when disabled */ +/* maximum registers to update */ +#define SND_SOC_DAPM_UPDATE_MAX_REG 3 + struct snd_soc_dapm_widget; enum snd_soc_dapm_type; struct snd_soc_dapm_path; @@ -378,6 +387,10 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_dapm_get_enum_onehot(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_dapm_put_enum_onehot(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_info_pin_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, @@ -590,9 +603,10 @@ struct snd_soc_dapm_widget { struct snd_soc_dapm_update { struct snd_kcontrol *kcontrol; - int reg; - int mask; - int val; + int reg[SND_SOC_DAPM_UPDATE_MAX_REG]; + int mask[SND_SOC_DAPM_UPDATE_MAX_REG]; + int val[SND_SOC_DAPM_UPDATE_MAX_REG]; + unsigned int num_regs; }; /* DAPM context */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 0b83168..dcdf1cb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -177,18 +177,24 @@ {.reg = xreg, .min = xmin, .max = xmax, \ .platform_max = xmax} } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xitems, xtexts) \ -{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ +{ .reg = &(int){(xreg)}, .shift_l = xshift_l, .shift_r = xshift_r, \ .items = xitems, .texts = xtexts, \ - .mask = xitems ? roundup_pow_of_two(xitems) - 1 : 0} + .mask = &(unsigned int){(xitems ? roundup_pow_of_two(xitems) - 1 : 0)}, \ + .num_regs = 1, .type = SND_SOC_ENUM_BINARY } #define SOC_ENUM_SINGLE(xreg, xshift, xitems, xtexts) \ SOC_ENUM_DOUBLE(xreg, xshift, xshift, xitems, xtexts) #define SOC_ENUM_SINGLE_EXT(xitems, xtexts) \ { .items = xitems, .texts = xtexts } #define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xitems, xtexts, xvalues) \ -{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ - .mask = xmask, .items = xitems, .texts = xtexts, .values = xvalues} +{ .reg = &(int){(xreg)}, .shift_l = xshift_l, .shift_r = xshift_r, \ + .mask = &(unsigned int){(xmask)}, .items = xitems, .texts = xtexts, \ + .values = xvalues, .num_regs = 1, .type = SND_SOC_ENUM_BINARY } #define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xnitmes, xtexts, xvalues) \ SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xnitmes, xtexts, xvalues) +#define SOC_VALUE_ENUM_ONEHOT(xregs, xmasks, xnum_regs, xitems, xtexts, xvalues) \ +{ .reg = xregs, .mask = xmasks, .num_regs = xnum_regs, \ + .items = xitems, .texts = xtexts, .values = xvalues, \ + .type = SND_SOC_ENUM_ONEHOT } #define SOC_ENUM_SINGLE_VIRT(xitems, xtexts) \ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, xitems, xtexts) #define SOC_ENUM(xname, xenum) \ @@ -293,6 +299,9 @@ #define SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xmask, xtexts, xvalues) \ const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \ ARRAY_SIZE(xtexts), xtexts, xvalues) +#define SOC_VALUE_ENUM_ONEHOT_DECL(name, xregs, xmasks, xnum_regs, xtexts, xvalues) \ + const struct soc_enum name = SOC_VALUE_ENUM_ONEHOT(xregs, xmasks, xnum_regs, \ + ARRAY_SIZE(xtexts), xtexts, xvalues) #define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \ SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) #define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \ @@ -326,6 +335,16 @@ enum snd_soc_bias_level { SND_SOC_BIAS_ON = 3, }; +/* + * enum snd_soc_enum_type - Type of ASoC enum control + * @SND_SOC_ENUM_BINARY: soc_enum type for SINGLE, DOUBLE or VIRTUAL mux + * @SND_SOC_ENUM_ONEHOT: soc_enum type for ONEHOT encoding mux + */ +enum snd_soc_enum_type { + SND_SOC_ENUM_BINARY = 0, + SND_SOC_ENUM_ONEHOT = 1, +}; + struct device_node; struct snd_jack; struct snd_soc_card; @@ -1098,13 +1117,15 @@ struct soc_mreg_control { /* enumerated kcontrol */ struct soc_enum { - int reg; + int *reg; unsigned char shift_l; unsigned char shift_r; unsigned int items; - unsigned int mask; + unsigned int *mask; const char * const *texts; const unsigned int *values; + enum snd_soc_enum_type type; + unsigned int num_regs; }; /** diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index caebd63..cf29722 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2601,12 +2601,12 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, unsigned int val, item; unsigned int reg_val; - reg_val = snd_soc_read(codec, e->reg); - val = (reg_val >> e->shift_l) & e->mask; + reg_val = snd_soc_read(codec, e->reg[0]); + val = (reg_val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[0] = item; if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_l) & e->mask; + val = (reg_val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = item; } @@ -2636,15 +2636,15 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, if (item[0] >= e->items) return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; - mask = e->mask << e->shift_l; + mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] >= e->items) return -EINVAL; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_r; - mask |= e->mask << e->shift_r; + mask |= e->mask[0] << e->shift_r; } - return snd_soc_update_bits_locked(codec, e->reg, mask, val); + return snd_soc_update_bits_locked(codec, e->reg[0], mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c8a780d..be9cd63 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -511,13 +511,26 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, const struct snd_kcontrol_new *kcontrol) { struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, item; + unsigned int val, item, bit_pos = 0; int i; - if (e->reg != SND_SOC_NOPM) { - soc_widget_read(dest, e->reg, &val); - val = (val >> e->shift_l) & e->mask; - item = snd_soc_enum_val_to_item(e, val); + if (e->reg[0] != SND_SOC_NOPM) { + if (e->type == SND_SOC_ENUM_ONEHOT) { + for (i = 0; i < e->num_regs; i++) { + soc_widget_read(dest, e->reg[i], &val); + val = val & e->mask[i]; + if (val != 0) { + bit_pos = ffs(val) + + (8 * dest->codec->val_bytes * i); + break; + } + } + item = snd_soc_enum_val_to_item(e, bit_pos); + } else { + soc_widget_read(dest, e->reg[0], &val); + val = (val >> e->shift_l) & e->mask[0]; + item = snd_soc_enum_val_to_item(e, val); + } } else { /* since a virtual mux has no backing registers to * decide which path to connect, it will try to match @@ -1553,8 +1566,8 @@ static void dapm_widget_update(struct snd_soc_card *card) struct snd_soc_dapm_update *update = card->update; struct snd_soc_dapm_widget_list *wlist; struct snd_soc_dapm_widget *w = NULL; - unsigned int wi; - int ret; + unsigned int wi, i; + int ret = 0; if (!update || !dapm_kcontrol_is_powered(update->kcontrol)) return; @@ -1575,11 +1588,16 @@ static void dapm_widget_update(struct snd_soc_card *card) if (!w) return; - ret = soc_widget_update_bits_locked(w, update->reg, update->mask, - update->val); - if (ret < 0) - dev_err(w->dapm->dev, "ASoC: %s DAPM update failed: %d\n", - w->name, ret); + /* dapm update for multiple registers */ + for (i = 0; i < update->num_regs; i++) { + ret = soc_widget_update_bits_locked(w, update->reg[i], + update->mask[i], update->val[i]); + if (ret < 0) { + dev_err(w->dapm->dev, "ASoC: %s DAPM update failed: %d\n", + w->name, ret); + break; + } + } for (wi = 0; wi < wlist->num_widgets; wi++) { w = wlist->widgets[wi]; @@ -2866,10 +2884,10 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, if (change) { if (reg != SND_SOC_NOPM) { update.kcontrol = kcontrol; - update.reg = reg; - update.mask = mask; - update.val = val; - + update.reg[0] = reg; + update.mask[0] = mask; + update.val[0] = val; + update.num_regs = 1; card->update = &update; } @@ -2903,15 +2921,15 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val; - if (e->reg != SND_SOC_NOPM) - reg_val = snd_soc_read(codec, e->reg); + if (e->reg[0] != SND_SOC_NOPM) + reg_val = snd_soc_read(codec, e->reg[0]); else reg_val = dapm_kcontrol_get_value(kcontrol); - val = (reg_val >> e->shift_l) & e->mask; + val = (reg_val >> e->shift_l) & e->mask[0]; ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val); if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_r) & e->mask; + val = (reg_val >> e->shift_r) & e->mask[0]; val = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = val; } @@ -2945,27 +2963,28 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; - mask = e->mask << e->shift_l; + mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] > e->items) return -EINVAL; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_l; - mask |= e->mask << e->shift_r; + mask |= e->mask[0] << e->shift_r; } mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (e->reg != SND_SOC_NOPM) - change = snd_soc_test_bits(codec, e->reg, mask, val); + if (e->reg[0] != SND_SOC_NOPM) + change = snd_soc_test_bits(codec, e->reg[0], mask, val); else change = dapm_kcontrol_set_value(kcontrol, val); if (change) { - if (e->reg != SND_SOC_NOPM) { + if (e->reg[0] != SND_SOC_NOPM) { update.kcontrol = kcontrol; - update.reg = e->reg; - update.mask = mask; - update.val = val; + update.reg[0] = e->reg[0]; + update.mask[0] = mask; + update.val[0] = val; + update.num_regs = 1; card->update = &update; } @@ -2984,6 +3003,112 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); /** + * snd_soc_dapm_get_enum_onehot - dapm enumerated onehot mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a dapm enumerated onehot encoded mixer control + * + * Returns 0 for success. + */ +int snd_soc_dapm_get_enum_onehot(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int reg_val, val, bit_pos = 0, reg_idx; + + for (reg_idx = 0; reg_idx < e->num_regs; reg_idx++) { + reg_val = snd_soc_read(codec, e->reg[reg_idx]); + val = reg_val & e->mask[reg_idx]; + if (val != 0) { + bit_pos = ffs(val) + (8 * codec->val_bytes * reg_idx); + break; + } + } + + ucontrol->value.enumerated.item[0] = + snd_soc_enum_val_to_item(e, bit_pos); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_onehot); + +/** + * snd_soc_dapm_put_enum_onehot - dapm enumerated onehot mixer put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to put the value of a dapm enumerated onehot encoded mixer control + * + * Returns 0 for success. + */ +int snd_soc_dapm_put_enum_onehot(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct snd_soc_card *card = codec->card; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int change = 0, reg_idx = 0, value, bit_pos; + unsigned int i, reg_val = 0, update_idx = 0; + struct snd_soc_dapm_update update; + int ret = 0; + + if (item[0] >= e->items || e->num_regs > SND_SOC_DAPM_UPDATE_MAX_REG) + return -EINVAL; + + value = snd_soc_enum_item_to_val(e, item[0]); + + if (value) { + /* get the register index and value to set */ + reg_idx = (value - 1) / (8 * codec->val_bytes); + bit_pos = (value - 1) % (8 * codec->val_bytes); + reg_val = BIT(bit_pos); + } + + for (i = 0; i < e->num_regs; i++) { + if (i == reg_idx) { + change |= snd_soc_test_bits(codec, e->reg[i], + e->mask[i], reg_val); + /* set the selected register */ + update.reg[e->num_regs - 1] = e->reg[reg_idx]; + update.mask[e->num_regs - 1] = e->mask[reg_idx]; + update.val[e->num_regs - 1] = reg_val; + } else { + /* accumulate the change to update the DAPM path + when none is selected */ + change |= snd_soc_test_bits(codec, e->reg[i], + e->mask[i], 0); + + /* clear the register when not selected */ + update.reg[update_idx] = e->reg[i]; + update.mask[update_idx] = e->mask[i]; + update.val[update_idx++] = 0; + } + } + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + if (change) { + update.kcontrol = kcontrol; + update.num_regs = e->num_regs; + card->update = &update; + + ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); + card->update = NULL; + } + + mutex_unlock(&card->dapm_mutex); + + if (ret > 0) + soc_dpcm_runtime_update(card); + + return change; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_onehot); + +/** * snd_soc_dapm_info_pin_switch - Info for a pin switch * * @kcontrol: mixer control -- 1.7.9.5

4 3

[alsa-devel] ASoC updates for v3.15
by Mark Brown 09 Apr '14

09 Apr '14

The following changes since commit 455c6fdbd219161bd09b1165f11699d6d73de11c: Linux 3.14 (2014-03-30 20:40:15 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.15-5 for you to fetch changes up to ab5d6fbdb71c183add7431243c7f615c42b643bb: Merge remote-tracking branches 'asoc/fix/rt5640', 'asoc/fix/samsung', 'asoc/fix/tlv320aic23' and 'asoc/fix/warn' into asoc-linus (2014-04-08 21:22:14 +0100) Some extra merges coming in due to the way my script ensures that previously sent fixes are included in my branch still not interacting well with signed tags, I'll need to figure that out. ---------------------------------------------------------------- ASoC: Fixes for v3.15 A smattering of device specific fixes, nothing stands out here except for the multiplatform fixes for Samsung and the device IDs being added by Stephen Warren - there's no real code changes from those and they give better robustness to the enumeration with DT. ---------------------------------------------------------------- Axel Lin (3): ASoC: cs42xx8: Check return value of regmap_read and report correct chipid value ASoC: cs42l52: Fix mask for REVID ASoC: da732x: Print correct major id Christoph Jaeger (1): ASoC: alc56(23|32): fix undefined return value of probing code Mark Brown (6): Merge tag 'asoc-v3.15' into asoc-linus Merge tag 'asoc-v3.15-2' into asoc-linus Merge tag 'asoc-v3.15-3' into asoc-linus Merge tag 'asoc-v3.15-4' into asoc-linus Merge remote-tracking branches 'asoc/fix/alc5632', 'asoc/fix/cs42l52', 'asoc/fix/cs42xxx8', 'asoc/fix/da732x', 'asoc/fix/davinci', 'asoc/fix/fsl-sai', 'asoc/fix/fsl-ssi' and 'asoc/fix/max98090' into asoc-linus Merge remote-tracking branches 'asoc/fix/rt5640', 'asoc/fix/samsung', 'asoc/fix/tlv320aic23' and 'asoc/fix/warn' into asoc-linus Markus Pargmann (1): ASoC: fsl-ssi: Update binding documentation Nicolin Chen (2): ASoC: fsl_sai: Add isr to deal with error flag ASoC: fsl_sai: Fix Bit Clock Polarity configurations Peter Ujfalusi (1): ASoC: davinci-mcasp: Fix bit clock polarity settings Sachin Kamat (1): ASoC: samsung: Fix build on multiplatform Stephen Warren (5): ASoC: rt5640: add an of_match table ASoC: alc5632: fix uninit var in alc5632_probe() ASoC: alc5632: add an of_match table ASoC: max98090: add an of_match table ASoC: tlv320aic23: add an of_match table .../devicetree/bindings/i2c/trivial-devices.txt | 1 + .../devicetree/bindings/misc/atmel-ssc.txt | 8 + .../bindings/sound/armada-370db-audio.txt | 27 + .../devicetree/bindings/sound/cs42xx8.txt | 28 + Documentation/devicetree/bindings/sound/da9055.txt | 22 + .../bindings/sound/davinci-evm-audio.txt | 9 +- .../devicetree/bindings/sound/eukrea-tlv320.txt | 21 + .../devicetree/bindings/sound/fsl,esai.txt | 5 + .../devicetree/bindings/sound/fsl,spdif.txt | 5 + .../devicetree/bindings/sound/fsl,ssi.txt | 21 +- .../devicetree/bindings/sound/mvebu-audio.txt | 1 + .../devicetree/bindings/sound/pcm512x.txt | 30 + .../devicetree/bindings/sound/renesas,rsnd.txt | 105 ++ .../devicetree/bindings/sound/simple-card.txt | 65 +- .../devicetree/bindings/sound/sirf-audio-codec.txt | 17 + .../devicetree/bindings/sound/sirf-audio-port.txt | 20 + .../devicetree/bindings/sound/sirf-audio.txt | 41 + .../devicetree/bindings/sound/tdm-slot.txt | 20 + .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + .../devicetree/bindings/sound/tlv320aic32x4.txt | 30 + .../devicetree/bindings/sound/tlv320aic3x.txt | 1 - .../devicetree/bindings/sound/widgets.txt | 20 + MAINTAINERS | 7 + arch/arm/mach-shmobile/board-armadillo800eva.c | 4 +- arch/arm/mach-shmobile/board-bockw.c | 16 +- arch/arm/mach-shmobile/board-kzm9g.c | 4 +- arch/arm/mach-shmobile/board-mackerel.c | 6 +- arch/sh/boards/mach-ecovec24/setup.c | 4 +- arch/sh/boards/mach-se/7724/setup.c | 4 +- drivers/base/regmap/regmap.c | 12 + drivers/misc/atmel-ssc.c | 6 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 8 + include/linux/atmel-ssc.h | 1 + include/linux/mfd/arizona/registers.h | 6 +- include/linux/platform_data/adau1977.h | 45 + include/linux/platform_data/asoc-s3c.h | 3 +- include/linux/platform_data/asoc-s3c24xx_simtec.h | 3 +- include/linux/platform_data/davinci_asp.h | 4 +- include/linux/regmap.h | 9 + include/sound/rcar_snd.h | 36 +- include/sound/simple_card.h | 6 +- include/sound/soc-dai.h | 3 + include/sound/soc-dapm.h | 40 +- include/sound/soc.h | 152 +- include/trace/events/hswadsp.h | 384 +++++ include/trace/events/intel-sst.h | 148 ++ sound/core/pcm_lib.c | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/Kconfig | 2 +- sound/soc/atmel/atmel_ssc_dai.c | 13 +- sound/soc/atmel/sam9g20_wm8731.c | 20 +- sound/soc/blackfin/Kconfig | 9 +- sound/soc/cirrus/Kconfig | 4 +- sound/soc/cirrus/snappercl15.c | 18 +- sound/soc/codecs/88pm860x-codec.c | 119 +- sound/soc/codecs/Kconfig | 195 ++- sound/soc/codecs/Makefile | 39 +- sound/soc/codecs/ad1836.c | 4 +- sound/soc/codecs/ad193x-i2c.c | 54 + sound/soc/codecs/ad193x-spi.c | 48 + sound/soc/codecs/ad193x.c | 154 +- sound/soc/codecs/ad193x.h | 7 + sound/soc/codecs/adau1373.c | 39 +- sound/soc/codecs/adau1977-i2c.c | 59 + sound/soc/codecs/adau1977-spi.c | 76 + sound/soc/codecs/adau1977.c | 1018 +++++++++++ sound/soc/codecs/adau1977.h | 37 + sound/soc/codecs/adav801.c | 53 + sound/soc/codecs/adav803.c | 50 + sound/soc/codecs/adav80x.c | 159 +- sound/soc/codecs/adav80x.h | 7 + sound/soc/codecs/ak4104.c | 2 - sound/soc/codecs/ak4535.c | 9 - sound/soc/codecs/ak4641.c | 24 +- sound/soc/codecs/ak4642.c | 8 - sound/soc/codecs/ak4671.c | 250 ++- sound/soc/codecs/ak4671.h | 2 - sound/soc/codecs/alc5623.c | 123 +- sound/soc/codecs/alc5632.c | 58 +- sound/soc/codecs/arizona.c | 325 ++-- sound/soc/codecs/cq93vc.c | 3 +- sound/soc/codecs/cs4270.c | 9 - sound/soc/codecs/cs4271.c | 63 +- sound/soc/codecs/cs42l51.c | 99 +- sound/soc/codecs/cs42l52.c | 111 +- sound/soc/codecs/cs42l52.h | 2 +- sound/soc/codecs/cs42l73.c | 72 +- sound/soc/codecs/cs42xx8-i2c.c | 64 + sound/soc/codecs/cs42xx8.c | 601 +++++++ sound/soc/codecs/cs42xx8.h | 238 +++ sound/soc/codecs/da7210.c | 28 +- sound/soc/codecs/da7213.c | 159 +- sound/soc/codecs/da732x.c | 199 +-- sound/soc/codecs/da732x.h | 3 - sound/soc/codecs/da9055.c | 100 +- sound/soc/codecs/isabelle.c | 19 +- sound/soc/codecs/lm4857.c | 3 +- sound/soc/codecs/lm49453.c | 47 +- sound/soc/codecs/max9768.c | 5 - sound/soc/codecs/max98088.c | 47 +- sound/soc/codecs/max98090.c | 188 ++- sound/soc/codecs/max98090.h | 1 - sound/soc/codecs/max98095.c | 60 +- sound/soc/codecs/max9850.c | 8 - sound/soc/codecs/mc13783.c | 30 +- sound/soc/codecs/ml26124.c | 22 +- sound/soc/codecs/pcm1681.c | 15 +- sound/soc/codecs/pcm1792a.c | 33 +- sound/soc/codecs/pcm512x-i2c.c | 71 + sound/soc/codecs/pcm512x-spi.c | 69 + sound/soc/codecs/pcm512x.c | 589 +++++++ sound/soc/codecs/pcm512x.h | 171 ++ sound/soc/codecs/rt5631.c | 84 +- sound/soc/codecs/rt5640.c | 95 +- sound/soc/codecs/sgtl5000.c | 18 +- sound/soc/codecs/si476x.c | 6 +- sound/soc/codecs/sirf-audio-codec.c | 524 ++++++ sound/soc/codecs/sirf-audio-codec.h | 75 + sound/soc/codecs/sn95031.c | 46 +- sound/soc/codecs/ssm2518.c | 24 +- sound/soc/codecs/ssm2602-i2c.c | 57 + sound/soc/codecs/ssm2602-spi.c | 41 + sound/soc/codecs/ssm2602.c | 180 +- sound/soc/codecs/ssm2602.h | 14 + sound/soc/codecs/sta32x.c | 14 - sound/soc/codecs/sta529.c | 15 +- sound/soc/codecs/stac9766.c | 38 +- sound/soc/codecs/tlv320aic23-i2c.c | 66 + sound/soc/codecs/tlv320aic23-spi.c | 56 + sound/soc/codecs/tlv320aic23.c | 79 +- sound/soc/codecs/tlv320aic23.h | 6 + sound/soc/codecs/tlv320aic26.c | 7 +- sound/soc/codecs/tlv320aic31xx.c | 1280 ++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 +++ sound/soc/codecs/tlv320aic32x4.c | 234 ++- sound/soc/codecs/tlv320aic3x.c | 6 - sound/soc/codecs/tlv320dac33.c | 35 +- sound/soc/codecs/twl4030.c | 101 +- sound/soc/codecs/twl6040.c | 17 +- sound/soc/codecs/uda134x.c | 3 +- sound/soc/codecs/uda1380.c | 43 +- sound/soc/codecs/wl1273.c | 9 +- sound/soc/codecs/wm2000.c | 2 - sound/soc/codecs/wm2200.c | 25 +- sound/soc/codecs/wm5100.c | 43 +- sound/soc/codecs/wm5102.c | 32 +- sound/soc/codecs/wm5110.c | 22 +- sound/soc/codecs/wm8350.c | 4 +- sound/soc/codecs/wm8400.c | 3 +- sound/soc/codecs/wm8510.c | 10 +- sound/soc/codecs/wm8523.c | 11 +- sound/soc/codecs/wm8580.c | 9 +- sound/soc/codecs/wm8711.c | 8 +- sound/soc/codecs/wm8728.c | 11 +- sound/soc/codecs/wm8731.c | 11 +- sound/soc/codecs/wm8737.c | 56 +- sound/soc/codecs/wm8741.c | 40 +- sound/soc/codecs/wm8750.c | 6 - sound/soc/codecs/wm8753.c | 12 +- sound/soc/codecs/wm8770.c | 6 - sound/soc/codecs/wm8776.c | 6 - sound/soc/codecs/wm8804.c | 10 +- sound/soc/codecs/wm8900.c | 8 +- sound/soc/codecs/wm8903.c | 118 +- sound/soc/codecs/wm8904.c | 86 +- sound/soc/codecs/wm8940.c | 26 +- sound/soc/codecs/wm8955.c | 19 +- sound/soc/codecs/wm8958-dsp2.c | 8 +- sound/soc/codecs/wm8960.c | 6 - sound/soc/codecs/wm8961.c | 23 +- sound/soc/codecs/wm8962.c | 87 +- sound/soc/codecs/wm8971.c | 6 - sound/soc/codecs/wm8974.c | 10 +- sound/soc/codecs/wm8978.c | 38 +- sound/soc/codecs/wm8983.c | 51 +- sound/soc/codecs/wm8985.c | 46 +- sound/soc/codecs/wm8988.c | 70 +- sound/soc/codecs/wm8990.c | 49 +- sound/soc/codecs/wm8991.c | 52 +- sound/soc/codecs/wm8993.c | 73 +- sound/soc/codecs/wm8994.c | 55 +- sound/soc/codecs/wm8995.c | 50 +- sound/soc/codecs/wm8996.c | 87 +- sound/soc/codecs/wm8997.c | 29 +- sound/soc/codecs/wm9081.c | 34 +- sound/soc/codecs/wm9090.c | 10 - sound/soc/codecs/wm9705.c | 12 +- sound/soc/codecs/wm_adsp.c | 50 +- sound/soc/codecs/wm_hubs.c | 16 +- sound/soc/davinci/davinci-evm.c | 80 +- sound/soc/davinci/davinci-mcasp.c | 219 ++- sound/soc/davinci/edma-pcm.c | 57 + sound/soc/davinci/edma-pcm.h | 25 + sound/soc/fsl/Kconfig | 14 +- sound/soc/fsl/eukrea-tlv320.c | 108 +- sound/soc/fsl/fsl_esai.c | 34 +- sound/soc/fsl/fsl_sai.c | 397 +++-- sound/soc/fsl/fsl_sai.h | 63 +- sound/soc/fsl/fsl_spdif.c | 9 +- sound/soc/fsl/fsl_utils.c | 27 + sound/soc/fsl/fsl_utils.h | 4 +- sound/soc/fsl/imx-pcm-fiq.c | 7 +- sound/soc/fsl/imx-ssi.c | 2 + sound/soc/fsl/wm1133-ev1.c | 11 +- sound/soc/generic/simple-card.c | 385 +++-- sound/soc/intel/Kconfig | 42 +- sound/soc/intel/Makefile | 27 +- sound/soc/intel/byt-rt5640.c | 187 ++ sound/soc/intel/haswell.c | 235 +++ sound/soc/intel/mfld_machine.c | 108 +- sound/soc/intel/sst-acpi.c | 284 ++++ sound/soc/intel/sst-baytrail-dsp.c | 372 ++++ sound/soc/intel/sst-baytrail-ipc.c | 867 ++++++++++ sound/soc/intel/sst-baytrail-ipc.h | 69 + sound/soc/intel/sst-baytrail-pcm.c | 422 +++++ sound/soc/intel/sst-dsp-priv.h | 309 ++++ sound/soc/intel/sst-dsp.c | 385 +++++ sound/soc/intel/sst-dsp.h | 233 +++ sound/soc/intel/sst-firmware.c | 587 +++++++ sound/soc/intel/sst-haswell-dsp.c | 517 ++++++ sound/soc/intel/sst-haswell-ipc.c | 1785 ++++++++++++++++++++ sound/soc/intel/sst-haswell-ipc.h | 488 ++++++ sound/soc/intel/sst-haswell-pcm.c | 872 ++++++++++ sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} | 8 +- .../intel/{sst_platform.c => sst-mfld-platform.c} | 8 +- .../intel/{sst_platform.h => sst-mfld-platform.h} | 4 +- sound/soc/kirkwood/Kconfig | 11 +- sound/soc/kirkwood/Makefile | 2 + sound/soc/kirkwood/armada-370-db.c | 148 ++ sound/soc/kirkwood/kirkwood-i2s.c | 1 + sound/soc/omap/Kconfig | 4 +- sound/soc/omap/ams-delta.c | 55 +- sound/soc/omap/n810.c | 22 +- sound/soc/omap/omap-abe-twl6040.c | 3 +- sound/soc/omap/rx51.c | 22 +- sound/soc/pxa/corgi.c | 49 +- sound/soc/pxa/e740_wm9705.c | 10 +- sound/soc/pxa/e750_wm9705.c | 10 +- sound/soc/pxa/e800_wm9712.c | 19 +- sound/soc/pxa/magician.c | 60 +- sound/soc/pxa/mioa701_wm9713.c | 19 +- sound/soc/pxa/poodle.c | 7 +- sound/soc/pxa/spitz.c | 58 +- sound/soc/pxa/tosa.c | 67 +- sound/soc/pxa/zylonite.c | 17 +- sound/soc/s6000/s6105-ipcam.c | 28 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/ac97.c | 6 +- sound/soc/samsung/dma.h | 6 +- sound/soc/samsung/h1940_uda1380.c | 7 +- sound/soc/samsung/i2s.c | 6 +- sound/soc/samsung/neo1973_wm8753.c | 168 +- sound/soc/samsung/pcm.c | 5 +- sound/soc/samsung/rx1950_uda1380.c | 5 - sound/soc/samsung/s3c2412-i2s.c | 4 +- sound/soc/samsung/s3c24xx-i2s.c | 4 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/spdif.c | 3 +- sound/soc/samsung/tobermory.c | 2 + sound/soc/sh/fsi.c | 2 +- sound/soc/sh/migor.c | 19 +- sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/adg.c | 229 ++- sound/soc/sh/rcar/core.c | 426 +++-- sound/soc/sh/rcar/gen.c | 113 +- sound/soc/sh/rcar/rsnd.h | 206 ++- sound/soc/sh/rcar/scu.c | 384 ----- sound/soc/sh/rcar/src.c | 727 ++++++++ sound/soc/sh/rcar/ssi.c | 388 ++--- sound/soc/sirf/Kconfig | 14 + sound/soc/sirf/Makefile | 5 + sound/soc/sirf/sirf-audio-port.c | 194 +++ sound/soc/sirf/sirf-audio-port.h | 62 + sound/soc/sirf/sirf-audio.c | 156 ++ sound/soc/soc-cache.c | 13 +- sound/soc/soc-compress.c | 65 +- sound/soc/soc-core.c | 603 ++++--- sound/soc/soc-dapm.c | 465 ++--- sound/soc/soc-io.c | 99 +- sound/soc/soc-jack.c | 5 +- sound/soc/soc-pcm.c | 112 +- sound/soc/spear/spdif_out.c | 10 +- sound/soc/tegra/Kconfig | 2 +- sound/soc/tegra/tegra20_ac97.c | 17 +- sound/soc/tegra/tegra20_ac97.h | 1 - sound/soc/tegra/tegra20_das.c | 2 +- sound/soc/tegra/tegra20_i2s.c | 2 +- sound/soc/tegra/tegra20_spdif.c | 2 +- sound/soc/tegra/tegra30_ahub.c | 4 +- sound/soc/tegra/tegra30_i2s.c | 2 +- sound/soc/tegra/tegra_wm9712.c | 17 +- 292 files changed, 21319 insertions(+), 5575 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/armada-370db-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/cs42xx8.txt create mode 100644 Documentation/devicetree/bindings/sound/da9055.txt create mode 100644 Documentation/devicetree/bindings/sound/eukrea-tlv320.txt create mode 100644 Documentation/devicetree/bindings/sound/pcm512x.txt create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-port.txt create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/tdm-slot.txt create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt create mode 100644 Documentation/devicetree/bindings/sound/widgets.txt create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 include/linux/platform_data/adau1977.h create mode 100644 include/trace/events/hswadsp.h create mode 100644 include/trace/events/intel-sst.h create mode 100644 sound/soc/codecs/ad193x-i2c.c create mode 100644 sound/soc/codecs/ad193x-spi.c create mode 100644 sound/soc/codecs/adau1977-i2c.c create mode 100644 sound/soc/codecs/adau1977-spi.c create mode 100644 sound/soc/codecs/adau1977.c create mode 100644 sound/soc/codecs/adau1977.h create mode 100644 sound/soc/codecs/adav801.c create mode 100644 sound/soc/codecs/adav803.c create mode 100644 sound/soc/codecs/cs42xx8-i2c.c create mode 100644 sound/soc/codecs/cs42xx8.c create mode 100644 sound/soc/codecs/cs42xx8.h create mode 100644 sound/soc/codecs/pcm512x-i2c.c create mode 100644 sound/soc/codecs/pcm512x-spi.c create mode 100644 sound/soc/codecs/pcm512x.c create mode 100644 sound/soc/codecs/pcm512x.h create mode 100644 sound/soc/codecs/sirf-audio-codec.c create mode 100644 sound/soc/codecs/sirf-audio-codec.h create mode 100644 sound/soc/codecs/ssm2602-i2c.c create mode 100644 sound/soc/codecs/ssm2602-spi.c create mode 100644 sound/soc/codecs/tlv320aic23-i2c.c create mode 100644 sound/soc/codecs/tlv320aic23-spi.c create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h create mode 100644 sound/soc/davinci/edma-pcm.c create mode 100644 sound/soc/davinci/edma-pcm.h create mode 100644 sound/soc/intel/byt-rt5640.c create mode 100644 sound/soc/intel/haswell.c create mode 100644 sound/soc/intel/sst-acpi.c create mode 100644 sound/soc/intel/sst-baytrail-dsp.c create mode 100644 sound/soc/intel/sst-baytrail-ipc.c create mode 100644 sound/soc/intel/sst-baytrail-ipc.h create mode 100644 sound/soc/intel/sst-baytrail-pcm.c create mode 100644 sound/soc/intel/sst-dsp-priv.h create mode 100644 sound/soc/intel/sst-dsp.c create mode 100644 sound/soc/intel/sst-dsp.h create mode 100644 sound/soc/intel/sst-firmware.c create mode 100644 sound/soc/intel/sst-haswell-dsp.c create mode 100644 sound/soc/intel/sst-haswell-ipc.c create mode 100644 sound/soc/intel/sst-haswell-ipc.h create mode 100644 sound/soc/intel/sst-haswell-pcm.c rename sound/soc/intel/{sst_dsp.h => sst-mfld-dsp.h} (96%) rename sound/soc/intel/{sst_platform.c => sst-mfld-platform.c} (99%) rename sound/soc/intel/{sst_platform.h => sst-mfld-platform.h} (97%) create mode 100644 sound/soc/kirkwood/armada-370-db.c delete mode 100644 sound/soc/sh/rcar/scu.c create mode 100644 sound/soc/sh/rcar/src.c create mode 100644 sound/soc/sirf/Kconfig create mode 100644 sound/soc/sirf/Makefile create mode 100644 sound/soc/sirf/sirf-audio-port.c create mode 100644 sound/soc/sirf/sirf-audio-port.h create mode 100644 sound/soc/sirf/sirf-audio.c

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[alsa-devel] [PATCH] ALSA: hda - Make full_reset boolean
by Thierry Reding 09 Apr '14

09 Apr '14

From: Thierry Reding <treding(a)nvidia.com> The full_reset argument to azx_init_chip() carries boolean rather than numerical information, so update the type to reflect that. Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- Changes in v2: - update prototype in hda_controller.h sound/pci/hda/hda_controller.c | 8 ++++---- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 4 ++-- 3 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 97993e17f46a..98d31d153703 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1604,7 +1604,7 @@ static void azx_exit_link_reset(struct azx *chip) } /* reset codec link */ -static int azx_reset(struct azx *chip, int full_reset) +static int azx_reset(struct azx *chip, bool full_reset) { if (!full_reset) goto __skip; @@ -1701,7 +1701,7 @@ static void azx_int_clear(struct azx *chip) /* * reset and start the controller registers */ -void azx_init_chip(struct azx *chip, int full_reset) +void azx_init_chip(struct azx *chip, bool full_reset) { if (chip->initialized) return; @@ -1841,7 +1841,7 @@ static void azx_bus_reset(struct hda_bus *bus) bus->in_reset = 1; azx_stop_chip(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); #ifdef CONFIG_PM if (chip->initialized) { struct azx_pcm *p; @@ -1948,7 +1948,7 @@ int azx_codec_create(struct azx *chip, const char *model, * get back to the sanity state. */ azx_stop_chip(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); } } } diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index 1d2e3be2bae6..baf0e77330af 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -37,7 +37,7 @@ int azx_alloc_stream_pages(struct azx *chip); void azx_free_stream_pages(struct azx *chip); /* Low level azx interface */ -void azx_init_chip(struct azx *chip, int full_reset); +void azx_init_chip(struct azx *chip, bool full_reset); void azx_stop_chip(struct azx *chip); void azx_enter_link_reset(struct azx *chip); irqreturn_t azx_interrupt(int irq, void *dev_id); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 77ca894f8284..d6bca62ef387 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -636,7 +636,7 @@ static int azx_resume(struct device *dev) return -EIO; azx_init_pci(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); snd_hda_resume(chip->bus); snd_power_change_state(card, SNDRV_CTL_POWER_D0); @@ -689,7 +689,7 @@ static int azx_runtime_resume(struct device *dev) status = azx_readw(chip, STATESTS); azx_init_pci(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); bus = chip->bus; if (status && bus) { -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Make full_reset boolean
by Thierry Reding 09 Apr '14

09 Apr '14

From: Thierry Reding <treding(a)nvidia.com> The full_reset argument to azx_init_chip() carries boolean rather than numerical information, so update the type to reflect that. Signed-off-by: Thierry Reding <treding(a)nvidia.com> --- sound/pci/hda/hda_controller.c | 8 ++++---- sound/pci/hda/hda_intel.c | 4 ++-- 2 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c index 97993e17f46a..98d31d153703 100644 --- a/sound/pci/hda/hda_controller.c +++ b/sound/pci/hda/hda_controller.c @@ -1604,7 +1604,7 @@ static void azx_exit_link_reset(struct azx *chip) } /* reset codec link */ -static int azx_reset(struct azx *chip, int full_reset) +static int azx_reset(struct azx *chip, bool full_reset) { if (!full_reset) goto __skip; @@ -1701,7 +1701,7 @@ static void azx_int_clear(struct azx *chip) /* * reset and start the controller registers */ -void azx_init_chip(struct azx *chip, int full_reset) +void azx_init_chip(struct azx *chip, bool full_reset) { if (chip->initialized) return; @@ -1841,7 +1841,7 @@ static void azx_bus_reset(struct hda_bus *bus) bus->in_reset = 1; azx_stop_chip(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); #ifdef CONFIG_PM if (chip->initialized) { struct azx_pcm *p; @@ -1948,7 +1948,7 @@ int azx_codec_create(struct azx *chip, const char *model, * get back to the sanity state. */ azx_stop_chip(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); } } } diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 77ca894f8284..d6bca62ef387 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -636,7 +636,7 @@ static int azx_resume(struct device *dev) return -EIO; azx_init_pci(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); snd_hda_resume(chip->bus); snd_power_change_state(card, SNDRV_CTL_POWER_D0); @@ -689,7 +689,7 @@ static int azx_runtime_resume(struct device *dev) status = azx_readw(chip, STATESTS); azx_init_pci(chip); - azx_init_chip(chip, 1); + azx_init_chip(chip, true); bus = chip->bus; if (status && bus) { -- 1.9.1

2 2

[alsa-devel] [PATCH v3 0/2] ASoC: fsl_sai: Add bus clock and mclks with clock controls
by Nicolin Chen 09 Apr '14

09 Apr '14

This series of patches add clock controls to fsl_sai driver and updates the vf610.dtsi accordingly. Changelog v3: * Use int type for ret instead of u32. * Added Acked-by and Tested-by from Xiubo Li. v2: * Appended two extra mclks to the driver since SAI actually has three. * Renamed clock name to 'bus' and 'mclk' according to the reference manual. Nicolin Chen (2): ASoC: fsl_sai: Add clock controls for SAI ARM: dts: Append clock bindings for sai2 on VF610 platform .../devicetree/bindings/sound/fsl-sai.txt | 8 ++-- arch/arm/boot/dts/vf610.dtsi | 6 ++- sound/soc/fsl/fsl_sai.c | 51 ++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 4 ++ 4 files changed, 60 insertions(+), 9 deletions(-) -- 1.8.4

2 6

[alsa-devel] [PATCH] ALSA: hda - add headset mic detect quirk for a Dell laptop
by Hui Wang 09 Apr '14

09 Apr '14

When we plug a 3-ring headset on the Dell machine (VID: 0x10ec0283, SID: 0x10280667), the headset mic can't be detected, after apply this patch, the headset mic can work well. BugLink: https://bugs.launchpad.net/bugs/1297581 Cc: David Henningsson <david.henningsson(a)canonical.com> Cc: stable(a)vger.kernel.org Signed-off-by: Hui Wang <hui.wang(a)canonical.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0d72dbf..14ae979 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4618,6 +4618,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0658, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x065f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0662, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0667, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0668, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0669, "Dell", ALC255_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), -- 1.8.1.2

2 1

[alsa-devel] [PATCH] sound: dmasound: use module_platform_driver_probe()
by Christoph Jaeger 09 Apr '14

09 Apr '14

Eliminate boilerplate code by using module_platform_driver_probe(). Signed-off-by: Christoph Jaeger <christophjaeger(a)linux.com> --- sound/oss/dmasound/dmasound_paula.c | 14 +------------- 1 file changed, 1 insertion(+), 13 deletions(-) diff --git a/sound/oss/dmasound/dmasound_paula.c b/sound/oss/dmasound/dmasound_paula.c index 87910e9..c2d45a5 100644 --- a/sound/oss/dmasound/dmasound_paula.c +++ b/sound/oss/dmasound/dmasound_paula.c @@ -733,19 +733,7 @@ static struct platform_driver amiga_audio_driver = { }, }; -static int __init amiga_audio_init(void) -{ - return platform_driver_probe(&amiga_audio_driver, amiga_audio_probe); -} - -module_init(amiga_audio_init); - -static void __exit amiga_audio_exit(void) -{ - platform_driver_unregister(&amiga_audio_driver); -} - -module_exit(amiga_audio_exit); +module_platform_driver_probe(amiga_audio_driver, amiga_audio_probe); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:amiga-audio"); -- 1.9.0

2 1

[alsa-devel] [PATCH] ALSA: au1x00: use module_platform_driver()
by Christoph Jaeger 09 Apr '14

09 Apr '14

Eliminate boilerplate code by using module_platform_driver(). Signed-off-by: Christoph Jaeger <christophjaeger(a)linux.com> --- sound/mips/au1x00.c | 13 +------------ 1 file changed, 1 insertion(+), 12 deletions(-) diff --git a/sound/mips/au1x00.c b/sound/mips/au1x00.c index a7cc49e..d10ef76 100644 --- a/sound/mips/au1x00.c +++ b/sound/mips/au1x00.c @@ -725,15 +725,4 @@ struct platform_driver au1000_ac97c_driver = { .remove = au1000_ac97_remove, }; -static int __init au1000_ac97_load(void) -{ - return platform_driver_register(&au1000_ac97c_driver); -} - -static void __exit au1000_ac97_unload(void) -{ - platform_driver_unregister(&au1000_ac97c_driver); -} - -module_init(au1000_ac97_load); -module_exit(au1000_ac97_unload); +module_platform_driver(au1000_ac97c_driver); -- 1.9.0

2 1

[alsa-devel] 2.1 audio not working in Kubuntu 13.10
by Tomas Black 09 Apr '14

09 Apr '14

Dear friends, I have a Dell L702X laptop which comes with a 2.1 sound configuration - 2 speakers + subwoofer from JBL. Sound is suberb in Windows, but I can't get the subwoofer to work in Linux! Here's the information I think you need: &&&&&@dell-xps:~$ aplay -l **** Lista de Dispositivos PLAYBACK Hardware **** placa 0: PCH [HDA Intel PCH], dispositivo 0: ALC665 Analog [ALC665 Analog] Dispositivo secundário: 1/1 Dispositivo secundário #0: subdevice #0 placa 0: PCH [HDA Intel PCH], dispositivo 1: ALC665 Digital [ALC665 Digital] Dispositivo secundário: 1/1 Dispositivo secundário #0: subdevice #0 placa 0: PCH [HDA Intel PCH], dispositivo 3: HDMI 0 [HDMI 0] Dispositivo secundário: 1/1 Dispositivo secundário #0: subdevice #0 #######@dell-xps:~$ aplay -L default Playback/recording through the PulseAudio sound server sysdefault:CARD=PCH HDA Intel PCH, ALC665 Analog Default Audio Device front:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog Front speakers surround40:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Digital IEC958 (S/PDIF) Digital Audio Output hdmi:CARD=PCH,DEV=0 HDA Intel PCH, HDMI 0 HDMI Audio Output dmix:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog Direct sample mixing device dmix:CARD=PCH,DEV=1 HDA Intel PCH, ALC665 Digital Direct sample mixing device dmix:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Direct sample mixing device dsnoop:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog Direct sample snooping device dsnoop:CARD=PCH,DEV=1 HDA Intel PCH, ALC665 Digital Direct sample snooping device dsnoop:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Direct sample snooping device hw:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog Direct hardware device without any conversions hw:CARD=PCH,DEV=1 HDA Intel PCH, ALC665 Digital Direct hardware device without any conversions hw:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Direct hardware device without any conversions plughw:CARD=PCH,DEV=0 HDA Intel PCH, ALC665 Analog Hardware device with all software conversions plughw:CARD=PCH,DEV=1 HDA Intel PCH, ALC665 Digital Hardware device with all software conversions plughw:CARD=PCH,DEV=3 HDA Intel PCH, HDMI 0 Hardware device with all software conversions ))))))))@dell-xps:~$ lspci -v | grep -A7 -i audio 00:1b.0 Audio device: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller (rev 05) Subsystem: Dell Device 04b8 Flags: bus master, fast devsel, latency 0, IRQ 54 Memory at f3c00000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel 00:1c.0 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 1 (rev b5) (prog-if 00 [Normal decode]) ^^^^^^@dell-xps:~$ cat /proc/asound/card0/codec* | grep Codec Codec: Realtek ALC665 Codec: Intel CougarPoint HDMI I've installed Realtek audio drivers and pretty much tried all tweaks on alsa .conf files a pulse .conf files. As I said, currently the stereo sound is good on both speakers, but how can I get the subwoofer to work? THanks a million in advance, Tomas

2 3

[alsa-devel] [PATCH] ASoC: rt5640: add rt5639 support
by Oder Chiou 08 Apr '14

08 Apr '14

This patch adds the rt5639 support Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 286 ++++++++++++++++++++++++++++++++++------------ sound/soc/codecs/rt5640.h | 3 + 2 files changed, 213 insertions(+), 76 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6b8778f..d672f44 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -399,18 +399,13 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("OUT Playback Volume", RT5640_OUTPUT, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), - /* MONO Output Control */ - SOC_SINGLE("Mono Playback Switch", RT5640_MONO_OUT, - RT5640_L_MUTE_SFT, 1, 1), + /* DAC Digital Volume */ SOC_DOUBLE("DAC2 Playback Switch", RT5640_DAC2_CTRL, RT5640_M_DAC_L2_VOL_SFT, RT5640_M_DAC_R2_VOL_SFT, 1, 1), SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5640_DAC1_DIG_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 175, 0, dac_vol_tlv), - SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5640_DAC2_DIG_VOL, - RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, - 175, 0, dac_vol_tlv), /* IN1/IN2 Control */ SOC_SINGLE_TLV("IN1 Boost", RT5640_IN1_IN2, RT5640_BST_SFT1, 8, 0, bst_tlv), @@ -442,6 +437,15 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { SOC_ENUM("DAC IF2 Data Switch", rt5640_if2_dac_enum), }; +static const struct snd_kcontrol_new rt5640_specific_snd_controls[] = { + /* MONO Output Control */ + SOC_SINGLE("Mono Playback Switch", RT5640_MONO_OUT, RT5640_L_MUTE_SFT, + 1, 1), + + SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5640_DAC2_DIG_VOL, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 175, 0, dac_vol_tlv), +}; + /** * set_dmic_clk - Set parameter of dmic. * @@ -555,6 +559,20 @@ static const struct snd_kcontrol_new rt5640_sto_dac_r_mix[] = { RT5640_M_ANC_DAC_R_SFT, 1, 1), }; +static const struct snd_kcontrol_new rt5639_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_R2_SFT, 1, 1), +}; + static const struct snd_kcontrol_new rt5640_mono_dac_l_mix[] = { SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_MONO_DAC_MIXER, RT5640_M_DAC_L1_MONO_L_SFT, 1, 1), @@ -677,6 +695,30 @@ static const struct snd_kcontrol_new rt5640_out_r_mix[] = { RT5640_M_DAC_R1_OM_R_SFT, 1, 1), }; +static const struct snd_kcontrol_new rt5639_out_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5640_OUT_L3_MIXER, + RT5640_M_IN_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXL Switch", RT5640_OUT_L3_MIXER, + RT5640_M_RM_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_BST4_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_BST1_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5640_OUT_R3_MIXER, + RT5640_M_IN_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXR Switch", RT5640_OUT_R3_MIXER, + RT5640_M_RM_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_DAC_R1_OM_R_SFT, 1, 1), +}; + static const struct snd_kcontrol_new rt5640_spo_l_mix[] = { SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_SPO_L_MIXER, RT5640_M_DAC_R1_SPM_L_SFT, 1, 1), @@ -708,6 +750,13 @@ static const struct snd_kcontrol_new rt5640_hpo_mix[] = { RT5640_M_HPVOL_HM_SFT, 1, 1), }; +static const struct snd_kcontrol_new rt5639_hpo_mix[] = { + SOC_DAPM_SINGLE("HPO MIX DAC1 Switch", RT5640_HPO_MIXER, + RT5640_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPO MIX HPVOL Switch", RT5640_HPO_MIXER, + RT5640_M_HPVOL_HM_SFT, 1, 1), +}; + static const struct snd_kcontrol_new rt5640_lout_mix[] = { SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_LOUT_MIXER, RT5640_M_DAC_L1_LM_SFT, 1, 1), @@ -1103,26 +1152,15 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), - /* Audio DSP */ - SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), - /* ANC */ - SND_SOC_DAPM_PGA("ANC", SND_SOC_NOPM, 0, 0, NULL, 0), + /* Output Side */ /* DAC mixer before sound effect */ SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, rt5640_dac_l_mix, ARRAY_SIZE(rt5640_dac_l_mix)), SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, rt5640_dac_r_mix, ARRAY_SIZE(rt5640_dac_r_mix)), - /* DAC2 channel Mux */ - SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, - &rt5640_dac_l2_mux), - SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, - &rt5640_dac_r2_mux), + /* DAC Mixer */ - SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, - rt5640_sto_dac_l_mix, ARRAY_SIZE(rt5640_sto_dac_l_mix)), - SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, - rt5640_sto_dac_r_mix, ARRAY_SIZE(rt5640_sto_dac_r_mix)), SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0, rt5640_mono_dac_l_mix, ARRAY_SIZE(rt5640_mono_dac_l_mix)), SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0, @@ -1134,21 +1172,14 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { /* DACs */ SND_SOC_DAPM_DAC("DAC L1", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_L1_BIT, 0), - SND_SOC_DAPM_DAC("DAC L2", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_L2_BIT, 0), SND_SOC_DAPM_DAC("DAC R1", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_R1_BIT, 0), - SND_SOC_DAPM_DAC("DAC R2", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_R2_BIT, 0), + /* SPK/OUT Mixer */ SND_SOC_DAPM_MIXER("SPK MIXL", RT5640_PWR_MIXER, RT5640_PWR_SM_L_BIT, 0, rt5640_spk_l_mix, ARRAY_SIZE(rt5640_spk_l_mix)), SND_SOC_DAPM_MIXER("SPK MIXR", RT5640_PWR_MIXER, RT5640_PWR_SM_R_BIT, 0, rt5640_spk_r_mix, ARRAY_SIZE(rt5640_spk_r_mix)), - SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, - 0, rt5640_out_l_mix, ARRAY_SIZE(rt5640_out_l_mix)), - SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, - 0, rt5640_out_r_mix, ARRAY_SIZE(rt5640_out_r_mix)), /* Ouput Volume */ SND_SOC_DAPM_PGA("SPKVOL L", RT5640_PWR_VOL, RT5640_PWR_SV_L_BIT, 0, NULL, 0), @@ -1167,16 +1198,8 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { 0, rt5640_spo_l_mix, ARRAY_SIZE(rt5640_spo_l_mix)), SND_SOC_DAPM_MIXER("SPOR MIX", SND_SOC_NOPM, 0, 0, rt5640_spo_r_mix, ARRAY_SIZE(rt5640_spo_r_mix)), - SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, - rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), - SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, - rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), SND_SOC_DAPM_MIXER("LOUT MIX", RT5640_PWR_ANLG1, RT5640_PWR_LM_BIT, 0, rt5640_lout_mix, ARRAY_SIZE(rt5640_lout_mix)), - SND_SOC_DAPM_MIXER("Mono MIX", RT5640_PWR_ANLG1, RT5640_PWR_MM_BIT, 0, - rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), - SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, - RT5640_PWR_MA_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, @@ -1208,10 +1231,69 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HPOR"), SND_SOC_DAPM_OUTPUT("LOUTL"), SND_SOC_DAPM_OUTPUT("LOUTR"), +}; + +static const struct snd_soc_dapm_widget rt5640_specific_dapm_widgets[] = { + /* Audio DSP */ + SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), + /* ANC */ + SND_SOC_DAPM_PGA("ANC", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DAC2 channel Mux */ + SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dac_l2_mux), + SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dac_r2_mux), + + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5640_sto_dac_l_mix, ARRAY_SIZE(rt5640_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5640_sto_dac_r_mix, ARRAY_SIZE(rt5640_sto_dac_r_mix)), + + SND_SOC_DAPM_DAC("DAC R2", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_R2_BIT, + 0), + SND_SOC_DAPM_DAC("DAC L2", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_L2_BIT, + 0), + + SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, + 0, rt5640_out_l_mix, ARRAY_SIZE(rt5640_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, + 0, rt5640_out_r_mix, ARRAY_SIZE(rt5640_out_r_mix)), + + SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), + + SND_SOC_DAPM_MIXER("Mono MIX", RT5640_PWR_ANLG1, RT5640_PWR_MM_BIT, 0, + rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), + SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, + RT5640_PWR_MA_BIT, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("MONOP"), SND_SOC_DAPM_OUTPUT("MONON"), }; +static const struct snd_soc_dapm_widget rt5639_specific_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5639_sto_dac_l_mix, ARRAY_SIZE(rt5639_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5639_sto_dac_r_mix, ARRAY_SIZE(rt5639_sto_dac_r_mix)), + + SND_SOC_DAPM_SUPPLY("DAC L2 Filter", RT5640_PWR_DIG1, + RT5640_PWR_DAC_L2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R2 Filter", RT5640_PWR_DIG1, + RT5640_PWR_DAC_R2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, + 0, rt5639_out_l_mix, ARRAY_SIZE(rt5639_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, + 0, rt5639_out_r_mix, ARRAY_SIZE(rt5639_out_r_mix)), + + SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, + rt5639_hpo_mix, ARRAY_SIZE(rt5639_hpo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, + rt5639_hpo_mix, ARRAY_SIZE(rt5639_hpo_mix)), +}; + static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"IN1P", NULL, "LDO2"}, {"IN2P", NULL, "LDO2"}, @@ -1353,71 +1435,38 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"DAC MIXR", "Stereo ADC Switch", "Stereo ADC MIXR"}, {"DAC MIXR", "INF1 Switch", "IF1 DAC R"}, - {"ANC", NULL, "Stereo ADC MIXL"}, - {"ANC", NULL, "Stereo ADC MIXR"}, - - {"Audio DSP", NULL, "DAC MIXL"}, - {"Audio DSP", NULL, "DAC MIXR"}, - - {"DAC L2 Mux", "IF2", "IF2 DAC L"}, - {"DAC L2 Mux", "Base L/R", "Audio DSP"}, - - {"DAC R2 Mux", "IF2", "IF2 DAC R"}, - {"Stereo DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, - {"Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, - {"Stereo DAC MIXL", "ANC Switch", "ANC"}, {"Stereo DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, - {"Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, - {"Stereo DAC MIXR", "ANC Switch", "ANC"}, {"Mono DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, - {"Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, - {"Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Mux"}, {"Mono DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, - {"Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, - {"Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Mux"}, {"DIG MIXL", "DAC L1 Switch", "DAC MIXL"}, - {"DIG MIXL", "DAC L2 Switch", "DAC L2 Mux"}, {"DIG MIXR", "DAC R1 Switch", "DAC MIXR"}, - {"DIG MIXR", "DAC R2 Switch", "DAC R2 Mux"}, {"DAC L1", NULL, "Stereo DAC MIXL"}, {"DAC L1", NULL, "PLL1", is_sys_clk_from_pll}, {"DAC R1", NULL, "Stereo DAC MIXR"}, {"DAC R1", NULL, "PLL1", is_sys_clk_from_pll}, - {"DAC L2", NULL, "Mono DAC MIXL"}, - {"DAC L2", NULL, "PLL1", is_sys_clk_from_pll}, - {"DAC R2", NULL, "Mono DAC MIXR"}, - {"DAC R2", NULL, "PLL1", is_sys_clk_from_pll}, {"SPK MIXL", "REC MIXL Switch", "RECMIXL"}, {"SPK MIXL", "INL Switch", "INL VOL"}, {"SPK MIXL", "DAC L1 Switch", "DAC L1"}, - {"SPK MIXL", "DAC L2 Switch", "DAC L2"}, {"SPK MIXL", "OUT MIXL Switch", "OUT MIXL"}, {"SPK MIXR", "REC MIXR Switch", "RECMIXR"}, {"SPK MIXR", "INR Switch", "INR VOL"}, {"SPK MIXR", "DAC R1 Switch", "DAC R1"}, - {"SPK MIXR", "DAC R2 Switch", "DAC R2"}, {"SPK MIXR", "OUT MIXR Switch", "OUT MIXR"}, - {"OUT MIXL", "SPK MIXL Switch", "SPK MIXL"}, {"OUT MIXL", "BST1 Switch", "BST1"}, {"OUT MIXL", "INL Switch", "INL VOL"}, {"OUT MIXL", "REC MIXL Switch", "RECMIXL"}, - {"OUT MIXL", "DAC R2 Switch", "DAC R2"}, - {"OUT MIXL", "DAC L2 Switch", "DAC L2"}, {"OUT MIXL", "DAC L1 Switch", "DAC L1"}, - {"OUT MIXR", "SPK MIXR Switch", "SPK MIXR"}, {"OUT MIXR", "BST2 Switch", "BST2"}, {"OUT MIXR", "BST1 Switch", "BST1"}, {"OUT MIXR", "INR Switch", "INR VOL"}, {"OUT MIXR", "REC MIXR Switch", "RECMIXR"}, - {"OUT MIXR", "DAC L2 Switch", "DAC L2"}, - {"OUT MIXR", "DAC R2 Switch", "DAC R2"}, {"OUT MIXR", "DAC R1 Switch", "DAC R1"}, {"SPKVOL L", NULL, "SPK MIXL"}, @@ -1436,11 +1485,9 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"SPOR MIX", "SPKVOL R Switch", "SPKVOL R"}, {"SPOR MIX", "BST1 Switch", "BST1"}, - {"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"}, {"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"}, {"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"}, {"HPO MIX L", NULL, "HP L Amp"}, - {"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"}, {"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"}, {"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"}, {"HPO MIX R", NULL, "HP R Amp"}, @@ -1450,12 +1497,6 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"LOUT MIX", "OUTVOL L Switch", "OUTVOL L"}, {"LOUT MIX", "OUTVOL R Switch", "OUTVOL R"}, - {"Mono MIX", "DAC R2 Switch", "DAC R2"}, - {"Mono MIX", "DAC L2 Switch", "DAC L2"}, - {"Mono MIX", "OUTVOL R Switch", "OUTVOL R"}, - {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, - {"Mono MIX", "BST1 Switch", "BST1"}, - {"HP Amp", NULL, "HPO MIX L"}, {"HP Amp", NULL, "HPO MIX R"}, @@ -1480,11 +1521,82 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPOR", NULL, "HP R Playback"}, {"LOUTL", NULL, "LOUT MIX"}, {"LOUTR", NULL, "LOUT MIX"}, +}; + +static const struct snd_soc_dapm_route rt5640_specific_dapm_routes[] = { + {"ANC", NULL, "Stereo ADC MIXL"}, + {"ANC", NULL, "Stereo ADC MIXR"}, + + {"Audio DSP", NULL, "DAC MIXL"}, + {"Audio DSP", NULL, "DAC MIXR"}, + + {"DAC L2 Mux", "IF2", "IF2 DAC L"}, + {"DAC L2 Mux", "Base L/R", "Audio DSP"}, + + {"DAC R2 Mux", "IF2", "IF2 DAC R"}, + + {"Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + {"Stereo DAC MIXL", "ANC Switch", "ANC"}, + {"Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"Stereo DAC MIXR", "ANC Switch", "ANC"}, + + {"Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + {"Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Mux"}, + + {"Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Mux"}, + + {"DIG MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"DIG MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + + {"DAC L2", NULL, "Mono DAC MIXL"}, + {"DAC L2", NULL, "PLL1", is_sys_clk_from_pll}, + {"DAC R2", NULL, "Mono DAC MIXR"}, + {"DAC R2", NULL, "PLL1", is_sys_clk_from_pll}, + + {"SPK MIXL", "DAC L2 Switch", "DAC L2"}, + {"SPK MIXR", "DAC R2 Switch", "DAC R2"}, + + {"OUT MIXL", "SPK MIXL Switch", "SPK MIXL"}, + {"OUT MIXR", "SPK MIXR Switch", "SPK MIXR"}, + + {"OUT MIXL", "DAC R2 Switch", "DAC R2"}, + {"OUT MIXL", "DAC L2 Switch", "DAC L2"}, + + {"OUT MIXR", "DAC L2 Switch", "DAC L2"}, + {"OUT MIXR", "DAC R2 Switch", "DAC R2"}, + + {"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"}, + {"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"}, + + {"Mono MIX", "DAC R2 Switch", "DAC R2"}, + {"Mono MIX", "DAC L2 Switch", "DAC L2"}, + {"Mono MIX", "OUTVOL R Switch", "OUTVOL R"}, + {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, + {"Mono MIX", "BST1 Switch", "BST1"}, + {"MONOP", NULL, "Mono MIX"}, {"MONON", NULL, "Mono MIX"}, {"MONOP", NULL, "Improve MONO Amp Drv"}, }; +static const struct snd_soc_dapm_route rt5639_specific_dapm_routes[] = { + {"Stereo DAC MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"Stereo DAC MIXR", "DAC R2 Switch", "IF2 DAC R"}, + + {"Mono DAC MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"Mono DAC MIXL", "DAC R2 Switch", "IF2 DAC R"}, + + {"Mono DAC MIXR", "DAC R2 Switch", "IF2 DAC R"}, + {"Mono DAC MIXR", "DAC L2 Switch", "IF2 DAC L"}, + + {"DIG MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"DIG MIXR", "DAC R2 Switch", "IF2 DAC R"}, + + {"IF2 DAC L", NULL, "DAC L2 Filter"}, + {"IF2 DAC R", NULL, "DAC R2 Filter"}, +}; + static int get_sdp_info(struct snd_soc_codec *codec, int dai_id) { int ret = 0, val; @@ -1901,6 +2013,28 @@ static int rt5640_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); + switch (snd_soc_read(codec, RT5640_RESET)) { + case RT5640_RESET_ID: + snd_soc_add_codec_controls(codec, + rt5640_specific_snd_controls, + ARRAY_SIZE(rt5640_specific_snd_controls)); + snd_soc_dapm_new_controls(&codec->dapm, + rt5640_specific_dapm_widgets, + ARRAY_SIZE(rt5640_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5640_specific_dapm_routes, + ARRAY_SIZE(rt5640_specific_dapm_routes)); + break; + case RT5639_RESET_ID: + snd_soc_dapm_new_controls(&codec->dapm, + rt5639_specific_dapm_widgets, + ARRAY_SIZE(rt5639_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5639_specific_dapm_routes, + ARRAY_SIZE(rt5639_specific_dapm_routes)); + break; + } + return 0; } diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index d7bd525..3b50459 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -14,6 +14,9 @@ #include <sound/rt5640.h> +#define RT5639_RESET_ID 0x0008 +#define RT5640_RESET_ID 0x000c + /* Info */ #define RT5640_RESET 0x00 #define RT5640_VENDOR_ID 0xfd -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda/hdmi - Add Nvidia Tegra124 HDMI support
by Dylan Reid 08 Apr '14

08 Apr '14

From: Sumit Bhattacharya <sumitb(a)nvidia.com> Add the Tegra 12x HDMI controller id to patch_hdmi. Signed-off-by: Sumit Bhattacharya <sumitb(a)nvidia.com> Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0cb5b89..b5d1371 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3322,6 +3322,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .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 = 0x10de0028, .name = "Tegra12x HDMI", .patch = patch_nvhdmi }, { .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 }, @@ -3377,6 +3378,7 @@ 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"); -- 1.8.1.3.605.g02339dd

1 0

[alsa-devel] [PATCH Resend 1/1] ASoC: Samsung: Fix build on multiplatform
by Sachin Kamat 08 Apr '14

08 Apr '14

PCM and S/PDIF drivers referenced mach headers for a trivial data structure. This caused build errors on multiplatform builds as machine headers are not accessible from driver files. Move the data structure definition to the driver header and remove the dependency. While at it rename the structure to avoid multiple definition errors as the same structure is also used by the platform code. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/samsung/ac97.c | 6 +++--- sound/soc/samsung/dma.h | 6 +++++- sound/soc/samsung/i2s.c | 6 +++--- sound/soc/samsung/pcm.c | 5 ++--- sound/soc/samsung/s3c2412-i2s.c | 4 ++-- sound/soc/samsung/s3c24xx-i2s.c | 4 ++-- sound/soc/samsung/spdif.c | 3 +-- 7 files changed, 18 insertions(+), 16 deletions(-) diff --git a/sound/soc/samsung/ac97.c b/sound/soc/samsung/ac97.c index 4a88e36c82ec..76b072bd4ba2 100644 --- a/sound/soc/samsung/ac97.c +++ b/sound/soc/samsung/ac97.c @@ -39,15 +39,15 @@ struct s3c_ac97_info { }; static struct s3c_ac97_info s3c_ac97; -static struct s3c2410_dma_client s3c_dma_client_out = { +static struct s3c_dma_client s3c_dma_client_out = { .name = "AC97 PCMOut" }; -static struct s3c2410_dma_client s3c_dma_client_in = { +static struct s3c_dma_client s3c_dma_client_in = { .name = "AC97 PCMIn" }; -static struct s3c2410_dma_client s3c_dma_client_micin = { +static struct s3c_dma_client s3c_dma_client_micin = { .name = "AC97 MicIn" }; diff --git a/sound/soc/samsung/dma.h b/sound/soc/samsung/dma.h index 225e5378014e..ad7c0f04f00d 100644 --- a/sound/soc/samsung/dma.h +++ b/sound/soc/samsung/dma.h @@ -14,8 +14,12 @@ #include <sound/dmaengine_pcm.h> +struct s3c_dma_client { + char *name; +}; + struct s3c_dma_params { - struct s3c2410_dma_client *client; /* stream identifier */ + struct s3c_dma_client *client; /* stream identifier */ int channel; /* Channel ID */ dma_addr_t dma_addr; int dma_size; /* Size of the DMA transfer */ diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 0a9b44c940ce..048ead967199 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -1211,10 +1211,10 @@ static int samsung_i2s_probe(struct platform_device *pdev) pri_dai->dma_playback.dma_addr = regs_base + I2STXD; pri_dai->dma_capture.dma_addr = regs_base + I2SRXD; pri_dai->dma_playback.client = - (struct s3c2410_dma_client *)&pri_dai->dma_playback; + (struct s3c_dma_client *)&pri_dai->dma_playback; pri_dai->dma_playback.ch_name = "tx"; pri_dai->dma_capture.client = - (struct s3c2410_dma_client *)&pri_dai->dma_capture; + (struct s3c_dma_client *)&pri_dai->dma_capture; pri_dai->dma_capture.ch_name = "rx"; pri_dai->dma_playback.dma_size = 4; pri_dai->dma_capture.dma_size = 4; @@ -1233,7 +1233,7 @@ static int samsung_i2s_probe(struct platform_device *pdev) } sec_dai->dma_playback.dma_addr = regs_base + I2STXDS; sec_dai->dma_playback.client = - (struct s3c2410_dma_client *)&sec_dai->dma_playback; + (struct s3c_dma_client *)&sec_dai->dma_playback; sec_dai->dma_playback.ch_name = "tx-sec"; if (!np) { diff --git a/sound/soc/samsung/pcm.c b/sound/soc/samsung/pcm.c index 6a5e4bf6ac96..ab54e297957c 100644 --- a/sound/soc/samsung/pcm.c +++ b/sound/soc/samsung/pcm.c @@ -20,7 +20,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "pcm.h" @@ -132,11 +131,11 @@ struct s3c_pcm_info { struct s3c_dma_params *dma_capture; }; -static struct s3c2410_dma_client s3c_pcm_dma_client_out = { +static struct s3c_dma_client s3c_pcm_dma_client_out = { .name = "PCM Stereo out" }; -static struct s3c2410_dma_client s3c_pcm_dma_client_in = { +static struct s3c_dma_client s3c_pcm_dma_client_in = { .name = "PCM Stereo in" }; diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index d0794458963a..e9bb5d7a71ee 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -33,11 +33,11 @@ #include "regs-i2s-v2.h" #include "s3c2412-i2s.h" -static struct s3c2410_dma_client s3c2412_dma_client_out = { +static struct s3c_dma_client s3c2412_dma_client_out = { .name = "I2S PCM Stereo out" }; -static struct s3c2410_dma_client s3c2412_dma_client_in = { +static struct s3c_dma_client s3c2412_dma_client_in = { .name = "I2S PCM Stereo in" }; diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index f31e916dd8c4..d7b8457b5650 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -31,11 +31,11 @@ #include "dma.h" #include "s3c24xx-i2s.h" -static struct s3c2410_dma_client s3c24xx_dma_client_out = { +static struct s3c_dma_client s3c24xx_dma_client_out = { .name = "I2S PCM Stereo out" }; -static struct s3c2410_dma_client s3c24xx_dma_client_in = { +static struct s3c_dma_client s3c24xx_dma_client_in = { .name = "I2S PCM Stereo in" }; diff --git a/sound/soc/samsung/spdif.c b/sound/soc/samsung/spdif.c index 28487dcc4538..cfe63b7bcc9f 100644 --- a/sound/soc/samsung/spdif.c +++ b/sound/soc/samsung/spdif.c @@ -18,7 +18,6 @@ #include <sound/pcm_params.h> #include <linux/platform_data/asoc-s3c.h> -#include <mach/dma.h> #include "dma.h" #include "spdif.h" @@ -94,7 +93,7 @@ struct samsung_spdif_info { struct s3c_dma_params *dma_playback; }; -static struct s3c2410_dma_client spdif_dma_client_out = { +static struct s3c_dma_client spdif_dma_client_out = { .name = "S/PDIF Stereo out", }; -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: ice1712: Fix boundary checks in PCM pointer ops
by Takashi Iwai 08 Apr '14

08 Apr '14

PCM pointer callbacks in ice1712 driver check the buffer size boundary wrongly between bytes and frames. This leads to PCM core warnings like: snd_pcm_update_hw_ptr0: 105 callbacks suppressed ALSA pcm_lib.c:352 BUG: pcmC3D0c:0, pos = 5461, buffer size = 5461, period size = 2730 This patch fixes these checks to be placed after the proper unit conversions. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/ice1712/ice1712.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index da005493f060..d9b9e4595f17 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -685,9 +685,10 @@ static snd_pcm_uframes_t snd_ice1712_playback_pointer(struct snd_pcm_substream * if (!(snd_ice1712_read(ice, ICE1712_IREG_PBK_CTRL) & 1)) return 0; ptr = runtime->buffer_size - inw(ice->ddma_port + 4); + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static snd_pcm_uframes_t snd_ice1712_playback_ds_pointer(struct snd_pcm_substream *substream) @@ -704,9 +705,10 @@ static snd_pcm_uframes_t snd_ice1712_playback_ds_pointer(struct snd_pcm_substrea addr = ICE1712_DSC_ADDR0; ptr = snd_ice1712_ds_read(ice, substream->number * 2, addr) - ice->playback_con_virt_addr[substream->number]; + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == substream->runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static snd_pcm_uframes_t snd_ice1712_capture_pointer(struct snd_pcm_substream *substream) @@ -717,9 +719,10 @@ static snd_pcm_uframes_t snd_ice1712_capture_pointer(struct snd_pcm_substream *s if (!(snd_ice1712_read(ice, ICE1712_IREG_CAP_CTRL) & 1)) return 0; ptr = inl(ICEREG(ice, CONCAP_ADDR)) - ice->capture_con_virt_addr; + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == substream->runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static const struct snd_pcm_hardware snd_ice1712_playback = { @@ -1116,9 +1119,10 @@ static snd_pcm_uframes_t snd_ice1712_playback_pro_pointer(struct snd_pcm_substre if (!(inl(ICEMT(ice, PLAYBACK_CONTROL)) & ICE1712_PLAYBACK_START)) return 0; ptr = ice->playback_pro_size - (inw(ICEMT(ice, PLAYBACK_SIZE)) << 2); + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == substream->runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static snd_pcm_uframes_t snd_ice1712_capture_pro_pointer(struct snd_pcm_substream *substream) @@ -1129,9 +1133,10 @@ static snd_pcm_uframes_t snd_ice1712_capture_pro_pointer(struct snd_pcm_substrea if (!(inl(ICEMT(ice, PLAYBACK_CONTROL)) & ICE1712_CAPTURE_START_SHADOW)) return 0; ptr = ice->capture_pro_size - (inw(ICEMT(ice, CAPTURE_SIZE)) << 2); + ptr = bytes_to_frames(substream->runtime, ptr); if (ptr == substream->runtime->buffer_size) ptr = 0; - return bytes_to_frames(substream->runtime, ptr); + return ptr; } static const struct snd_pcm_hardware snd_ice1712_playback_pro = { -- 1.9.1

1 0

[alsa-devel] (JOB) ALSA, ASoC driver development
by Adam Ćwiek 08 Apr '14

08 Apr '14

Hi, I am developing multi channel audio dsp platform for high level audio processing. Now I am searching for someone, who can deal with audio ALSA, ASoC driver development (of course paid offer). I am electronic engeneer, so I will take care of hardware. I am going to use audio codecs, like wolfsons WM8510, CS42448 from Cirrus Logic or PCM3168 from Texas. My goal is to be able to send / receive as many analog inputs/outputs as possible. As mainboard i want to use ready made modules as OLinuXino , Raspberry Pi, or BeagleBone. Looking forward to hearing from you. Adam Cwiek

1 0

[alsa-devel] [PATCH] ASoC: fsl_sai: Fix Bit Clock Polarity configurations
by Nicolin Chen 08 Apr '14

08 Apr '14

The BCP bit in TCR4/RCR4 register rules as followings: 0 Bit clock is active high with drive outputs on rising edge and sample inputs on falling edge. 1 Bit clock is active low with drive outputs on falling edge and sample inputs on rising edge. For all formats currently supported in the fsl_sai driver, they're exactly sending data on the falling edge and sampling on the rising edge. However, the driver clears this BCP bit for all of them which results click noise when working with SGTL5000 and big noise with WM8962. Thus this patch corrects the BCP settings for all the formats here to fix the nosie issue. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 99051c7..9bbebea 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -180,7 +180,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * that is, together with the last bit of the previous * data word. */ - val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr2 |= FSL_SAI_CR2_BCP; val_cr4 |= FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP; break; case SND_SOC_DAIFMT_LEFT_J: @@ -188,7 +188,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * Frame high, one word length for frame sync, * frame sync asserts with the first bit of the frame. */ - val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr2 |= FSL_SAI_CR2_BCP; val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP); break; case SND_SOC_DAIFMT_DSP_A: @@ -198,7 +198,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * that is, together with the last bit of the previous * data word. */ - val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr2 |= FSL_SAI_CR2_BCP; val_cr4 &= ~FSL_SAI_CR4_FSP; val_cr4 |= FSL_SAI_CR4_FSE; sai->is_dsp_mode = true; @@ -208,7 +208,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, * Frame high, one bit for frame sync, * frame sync asserts with the first bit of the frame. */ - val_cr2 &= ~FSL_SAI_CR2_BCP; + val_cr2 |= FSL_SAI_CR2_BCP; val_cr4 &= ~(FSL_SAI_CR4_FSE | FSL_SAI_CR4_FSP); sai->is_dsp_mode = true; break; -- 1.8.4

3 6

[alsa-devel] [PATCH] ASoC: rt5640: add rt5639 support
by Oder Chiou 08 Apr '14

08 Apr '14

This patch adds the rt5639 support Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 365 +++++++++++++++++++++++++++++++--------------- sound/soc/codecs/rt5640.h | 3 + 2 files changed, 250 insertions(+), 118 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 6b8778f..f5b9ecc 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -361,24 +361,23 @@ static unsigned int bst_tlv[] = { static const char * const rt5640_data_select[] = { "Normal", "left copy to right", "right copy to left", "Swap"}; -static const SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA, - RT5640_IF1_DAC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA, + RT5640_IF1_DAC_SEL_SFT, rt5640_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5640_if1_adc_enum, RT5640_DIG_INF_DATA, - RT5640_IF1_ADC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if1_adc_enum, RT5640_DIG_INF_DATA, + RT5640_IF1_ADC_SEL_SFT, rt5640_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5640_if2_dac_enum, RT5640_DIG_INF_DATA, - RT5640_IF2_DAC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if2_dac_enum, RT5640_DIG_INF_DATA, + RT5640_IF2_DAC_SEL_SFT, rt5640_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA, - RT5640_IF2_ADC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA, + RT5640_IF2_ADC_SEL_SFT, rt5640_data_select); /* Class D speaker gain ratio */ static const char * const rt5640_clsd_spk_ratio[] = {"1.66x", "1.83x", "1.94x", "2x", "2.11x", "2.22x", "2.33x", "2.44x", "2.55x", "2.66x", "2.77x"}; -static const SOC_ENUM_SINGLE_DECL( - rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT, +static SOC_ENUM_SINGLE_DECL(rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT, RT5640_CLSD_RATIO_SFT, rt5640_clsd_spk_ratio); static const struct snd_kcontrol_new rt5640_snd_controls[] = { @@ -387,11 +386,13 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("Speaker Playback Volume", RT5640_SPK_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), + /* Headphone Output Volume */ SOC_DOUBLE("HP Channel Switch", RT5640_HP_VOL, RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("HP Playback Volume", RT5640_HP_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), + /* OUTPUT Control */ SOC_DOUBLE("OUT Playback Switch", RT5640_OUTPUT, RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), @@ -399,49 +400,54 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { RT5640_VOL_L_SFT, RT5640_VOL_R_SFT, 1, 1), SOC_DOUBLE_TLV("OUT Playback Volume", RT5640_OUTPUT, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 39, 1, out_vol_tlv), - /* MONO Output Control */ - SOC_SINGLE("Mono Playback Switch", RT5640_MONO_OUT, - RT5640_L_MUTE_SFT, 1, 1), + /* DAC Digital Volume */ SOC_DOUBLE("DAC2 Playback Switch", RT5640_DAC2_CTRL, RT5640_M_DAC_L2_VOL_SFT, RT5640_M_DAC_R2_VOL_SFT, 1, 1), SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5640_DAC1_DIG_VOL, - RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, - 175, 0, dac_vol_tlv), - SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5640_DAC2_DIG_VOL, - RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, - 175, 0, dac_vol_tlv), + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 175, 0, dac_vol_tlv), + /* IN1/IN2 Control */ - SOC_SINGLE_TLV("IN1 Boost", RT5640_IN1_IN2, - RT5640_BST_SFT1, 8, 0, bst_tlv), - SOC_SINGLE_TLV("IN2 Boost", RT5640_IN3_IN4, - RT5640_BST_SFT2, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN1 Boost", RT5640_IN1_IN2, RT5640_BST_SFT1, 8, 0, + bst_tlv), + SOC_SINGLE_TLV("IN2 Boost", RT5640_IN3_IN4, RT5640_BST_SFT2, 8, 0, + bst_tlv), + /* INL/INR Volume Control */ SOC_DOUBLE_TLV("IN Capture Volume", RT5640_INL_INR_VOL, - RT5640_INL_VOL_SFT, RT5640_INR_VOL_SFT, - 31, 1, in_vol_tlv), + RT5640_INL_VOL_SFT, RT5640_INR_VOL_SFT, 31, 1, in_vol_tlv), + /* ADC Digital Volume Control */ SOC_DOUBLE("ADC Capture Switch", RT5640_ADC_DIG_VOL, RT5640_L_MUTE_SFT, RT5640_R_MUTE_SFT, 1, 1), SOC_DOUBLE_TLV("ADC Capture Volume", RT5640_ADC_DIG_VOL, - RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, - 127, 0, adc_vol_tlv), + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 127, 0, adc_vol_tlv), SOC_DOUBLE_TLV("Mono ADC Capture Volume", RT5640_ADC_DATA, - RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, - 127, 0, adc_vol_tlv), + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 127, 0, adc_vol_tlv), + /* ADC Boost Volume Control */ SOC_DOUBLE_TLV("ADC Boost Gain", RT5640_ADC_BST_VOL, - RT5640_ADC_L_BST_SFT, RT5640_ADC_R_BST_SFT, - 3, 0, adc_bst_tlv), + RT5640_ADC_L_BST_SFT, RT5640_ADC_R_BST_SFT, 3, 0, adc_bst_tlv), + /* Class D speaker gain ratio */ SOC_ENUM("Class D SPK Ratio Control", rt5640_clsd_spk_ratio_enum), + /* Data Swap */ SOC_ENUM("ADC IF1 Data Switch", rt5640_if1_adc_enum), SOC_ENUM("DAC IF1 Data Switch", rt5640_if1_dac_enum), SOC_ENUM("ADC IF2 Data Switch", rt5640_if2_adc_enum), SOC_ENUM("DAC IF2 Data Switch", rt5640_if2_dac_enum), }; +static const struct snd_kcontrol_new rt5640_specific_snd_controls[] = { + /* MONO Output Control */ + SOC_SINGLE("Mono Playback Switch", RT5640_MONO_OUT, RT5640_L_MUTE_SFT, + 1, 1), + + SOC_DOUBLE_TLV("Mono DAC Playback Volume", RT5640_DAC2_DIG_VOL, + RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 175, 0, dac_vol_tlv), +}; + /** * set_dmic_clk - Set parameter of dmic. * @@ -555,6 +561,20 @@ static const struct snd_kcontrol_new rt5640_sto_dac_r_mix[] = { RT5640_M_ANC_DAC_R_SFT, 1, 1), }; +static const struct snd_kcontrol_new rt5639_sto_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L2 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_sto_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R2 Switch", RT5640_STO_DAC_MIXER, + RT5640_M_DAC_R2_SFT, 1, 1), +}; + static const struct snd_kcontrol_new rt5640_mono_dac_l_mix[] = { SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_MONO_DAC_MIXER, RT5640_M_DAC_L1_MONO_L_SFT, 1, 1), @@ -677,6 +697,30 @@ static const struct snd_kcontrol_new rt5640_out_r_mix[] = { RT5640_M_DAC_R1_OM_R_SFT, 1, 1), }; +static const struct snd_kcontrol_new rt5639_out_l_mix[] = { + SOC_DAPM_SINGLE("BST1 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_BST1_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("INL Switch", RT5640_OUT_L3_MIXER, + RT5640_M_IN_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXL Switch", RT5640_OUT_L3_MIXER, + RT5640_M_RM_L_OM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_OUT_L3_MIXER, + RT5640_M_DAC_L1_OM_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5639_out_r_mix[] = { + SOC_DAPM_SINGLE("BST2 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_BST4_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST1 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_BST1_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("INR Switch", RT5640_OUT_R3_MIXER, + RT5640_M_IN_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("REC MIXR Switch", RT5640_OUT_R3_MIXER, + RT5640_M_RM_R_OM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_OUT_R3_MIXER, + RT5640_M_DAC_R1_OM_R_SFT, 1, 1), +}; + static const struct snd_kcontrol_new rt5640_spo_l_mix[] = { SOC_DAPM_SINGLE("DAC R1 Switch", RT5640_SPO_L_MIXER, RT5640_M_DAC_R1_SPM_L_SFT, 1, 1), @@ -708,6 +752,13 @@ static const struct snd_kcontrol_new rt5640_hpo_mix[] = { RT5640_M_HPVOL_HM_SFT, 1, 1), }; +static const struct snd_kcontrol_new rt5639_hpo_mix[] = { + SOC_DAPM_SINGLE("HPO MIX DAC1 Switch", RT5640_HPO_MIXER, + RT5640_M_DAC1_HM_SFT, 1, 1), + SOC_DAPM_SINGLE("HPO MIX HPVOL Switch", RT5640_HPO_MIXER, + RT5640_M_HPVOL_HM_SFT, 1, 1), +}; + static const struct snd_kcontrol_new rt5640_lout_mix[] = { SOC_DAPM_SINGLE("DAC L1 Switch", RT5640_LOUT_MIXER, RT5640_M_DAC_L1_LM_SFT, 1, 1), @@ -753,8 +804,7 @@ static const char * const rt5640_stereo_adc1_src[] = { "DIG MIX", "ADC" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_stereo_adc1_enum, RT5640_STO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5640_stereo_adc1_enum, RT5640_STO_ADC_MIXER, RT5640_ADC_1_SRC_SFT, rt5640_stereo_adc1_src); static const struct snd_kcontrol_new rt5640_sto_adc_1_mux = @@ -764,8 +814,7 @@ static const char * const rt5640_stereo_adc2_src[] = { "DMIC1", "DMIC2", "DIG MIX" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_stereo_adc2_enum, RT5640_STO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5640_stereo_adc2_enum, RT5640_STO_ADC_MIXER, RT5640_ADC_2_SRC_SFT, rt5640_stereo_adc2_src); static const struct snd_kcontrol_new rt5640_sto_adc_2_mux = @@ -776,8 +825,7 @@ static const char * const rt5640_mono_adc_l1_src[] = { "Mono DAC MIXL", "ADCL" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_l1_enum, RT5640_MONO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_l1_enum, RT5640_MONO_ADC_MIXER, RT5640_MONO_ADC_L1_SRC_SFT, rt5640_mono_adc_l1_src); static const struct snd_kcontrol_new rt5640_mono_adc_l1_mux = @@ -787,8 +835,7 @@ static const char * const rt5640_mono_adc_l2_src[] = { "DMIC L1", "DMIC L2", "Mono DAC MIXL" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_l2_enum, RT5640_MONO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_l2_enum, RT5640_MONO_ADC_MIXER, RT5640_MONO_ADC_L2_SRC_SFT, rt5640_mono_adc_l2_src); static const struct snd_kcontrol_new rt5640_mono_adc_l2_mux = @@ -798,8 +845,7 @@ static const char * const rt5640_mono_adc_r1_src[] = { "Mono DAC MIXR", "ADCR" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_r1_enum, RT5640_MONO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_r1_enum, RT5640_MONO_ADC_MIXER, RT5640_MONO_ADC_R1_SRC_SFT, rt5640_mono_adc_r1_src); static const struct snd_kcontrol_new rt5640_mono_adc_r1_mux = @@ -809,8 +855,7 @@ static const char * const rt5640_mono_adc_r2_src[] = { "DMIC R1", "DMIC R2", "Mono DAC MIXR" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_r2_enum, RT5640_MONO_ADC_MIXER, +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_r2_enum, RT5640_MONO_ADC_MIXER, RT5640_MONO_ADC_R2_SRC_SFT, rt5640_mono_adc_r2_src); static const struct snd_kcontrol_new rt5640_mono_adc_r2_mux = @@ -1103,26 +1148,15 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), - /* Audio DSP */ - SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), - /* ANC */ - SND_SOC_DAPM_PGA("ANC", SND_SOC_NOPM, 0, 0, NULL, 0), + /* Output Side */ /* DAC mixer before sound effect */ SND_SOC_DAPM_MIXER("DAC MIXL", SND_SOC_NOPM, 0, 0, rt5640_dac_l_mix, ARRAY_SIZE(rt5640_dac_l_mix)), SND_SOC_DAPM_MIXER("DAC MIXR", SND_SOC_NOPM, 0, 0, rt5640_dac_r_mix, ARRAY_SIZE(rt5640_dac_r_mix)), - /* DAC2 channel Mux */ - SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, - &rt5640_dac_l2_mux), - SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, - &rt5640_dac_r2_mux), + /* DAC Mixer */ - SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, - rt5640_sto_dac_l_mix, ARRAY_SIZE(rt5640_sto_dac_l_mix)), - SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, - rt5640_sto_dac_r_mix, ARRAY_SIZE(rt5640_sto_dac_r_mix)), SND_SOC_DAPM_MIXER("Mono DAC MIXL", SND_SOC_NOPM, 0, 0, rt5640_mono_dac_l_mix, ARRAY_SIZE(rt5640_mono_dac_l_mix)), SND_SOC_DAPM_MIXER("Mono DAC MIXR", SND_SOC_NOPM, 0, 0, @@ -1132,23 +1166,15 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_MIXER("DIG MIXR", SND_SOC_NOPM, 0, 0, rt5640_dig_r_mix, ARRAY_SIZE(rt5640_dig_r_mix)), /* DACs */ - SND_SOC_DAPM_DAC("DAC L1", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_L1_BIT, 0), - SND_SOC_DAPM_DAC("DAC L2", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_L2_BIT, 0), - SND_SOC_DAPM_DAC("DAC R1", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_R1_BIT, 0), - SND_SOC_DAPM_DAC("DAC R2", NULL, RT5640_PWR_DIG1, - RT5640_PWR_DAC_R2_BIT, 0), + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_L1_BIT, + 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_R1_BIT, + 0), /* SPK/OUT Mixer */ SND_SOC_DAPM_MIXER("SPK MIXL", RT5640_PWR_MIXER, RT5640_PWR_SM_L_BIT, 0, rt5640_spk_l_mix, ARRAY_SIZE(rt5640_spk_l_mix)), SND_SOC_DAPM_MIXER("SPK MIXR", RT5640_PWR_MIXER, RT5640_PWR_SM_R_BIT, 0, rt5640_spk_r_mix, ARRAY_SIZE(rt5640_spk_r_mix)), - SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, - 0, rt5640_out_l_mix, ARRAY_SIZE(rt5640_out_l_mix)), - SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, - 0, rt5640_out_r_mix, ARRAY_SIZE(rt5640_out_r_mix)), /* Ouput Volume */ SND_SOC_DAPM_PGA("SPKVOL L", RT5640_PWR_VOL, RT5640_PWR_SV_L_BIT, 0, NULL, 0), @@ -1167,16 +1193,8 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { 0, rt5640_spo_l_mix, ARRAY_SIZE(rt5640_spo_l_mix)), SND_SOC_DAPM_MIXER("SPOR MIX", SND_SOC_NOPM, 0, 0, rt5640_spo_r_mix, ARRAY_SIZE(rt5640_spo_r_mix)), - SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, - rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), - SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, - rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), SND_SOC_DAPM_MIXER("LOUT MIX", RT5640_PWR_ANLG1, RT5640_PWR_LM_BIT, 0, rt5640_lout_mix, ARRAY_SIZE(rt5640_lout_mix)), - SND_SOC_DAPM_MIXER("Mono MIX", RT5640_PWR_ANLG1, RT5640_PWR_MM_BIT, 0, - rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), - SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, - RT5640_PWR_MA_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("Improve HP Amp Drv", 1, SND_SOC_NOPM, 0, 0, rt5640_hp_power_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, @@ -1199,6 +1217,7 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("HP R Playback", SND_SOC_NOPM, 0, 0, &hp_r_enable_control), SND_SOC_DAPM_POST("HP Post", rt5640_hp_post_event), + /* Output Lines */ SND_SOC_DAPM_OUTPUT("SPOLP"), SND_SOC_DAPM_OUTPUT("SPOLN"), @@ -1208,10 +1227,69 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HPOR"), SND_SOC_DAPM_OUTPUT("LOUTL"), SND_SOC_DAPM_OUTPUT("LOUTR"), +}; + +static const struct snd_soc_dapm_widget rt5640_specific_dapm_widgets[] = { + /* Audio DSP */ + SND_SOC_DAPM_PGA("Audio DSP", SND_SOC_NOPM, 0, 0, NULL, 0), + /* ANC */ + SND_SOC_DAPM_PGA("ANC", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* DAC2 channel Mux */ + SND_SOC_DAPM_MUX("DAC L2 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dac_l2_mux), + SND_SOC_DAPM_MUX("DAC R2 Mux", SND_SOC_NOPM, 0, 0, &rt5640_dac_r2_mux), + + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5640_sto_dac_l_mix, ARRAY_SIZE(rt5640_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5640_sto_dac_r_mix, ARRAY_SIZE(rt5640_sto_dac_r_mix)), + + SND_SOC_DAPM_DAC("DAC R2", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_R2_BIT, + 0), + SND_SOC_DAPM_DAC("DAC L2", NULL, RT5640_PWR_DIG1, RT5640_PWR_DAC_L2_BIT, + 0), + + SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, + 0, rt5640_out_l_mix, ARRAY_SIZE(rt5640_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, + 0, rt5640_out_r_mix, ARRAY_SIZE(rt5640_out_r_mix)), + + SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), + + SND_SOC_DAPM_MIXER("Mono MIX", RT5640_PWR_ANLG1, RT5640_PWR_MM_BIT, 0, + rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), + SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, + RT5640_PWR_MA_BIT, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("MONOP"), SND_SOC_DAPM_OUTPUT("MONON"), }; +static const struct snd_soc_dapm_widget rt5639_specific_dapm_widgets[] = { + SND_SOC_DAPM_MIXER("Stereo DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5639_sto_dac_l_mix, ARRAY_SIZE(rt5639_sto_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5639_sto_dac_r_mix, ARRAY_SIZE(rt5639_sto_dac_r_mix)), + + SND_SOC_DAPM_SUPPLY("DAC L2 Filter", RT5640_PWR_DIG1, + RT5640_PWR_DAC_L2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R2 Filter", RT5640_PWR_DIG1, + RT5640_PWR_DAC_R2_BIT, 0, NULL, 0), + + SND_SOC_DAPM_MIXER("OUT MIXL", RT5640_PWR_MIXER, RT5640_PWR_OM_L_BIT, + 0, rt5639_out_l_mix, ARRAY_SIZE(rt5639_out_l_mix)), + SND_SOC_DAPM_MIXER("OUT MIXR", RT5640_PWR_MIXER, RT5640_PWR_OM_R_BIT, + 0, rt5639_out_r_mix, ARRAY_SIZE(rt5639_out_r_mix)), + + SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, + rt5639_hpo_mix, ARRAY_SIZE(rt5639_hpo_mix)), + SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, + rt5639_hpo_mix, ARRAY_SIZE(rt5639_hpo_mix)), +}; + static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"IN1P", NULL, "LDO2"}, {"IN2P", NULL, "LDO2"}, @@ -1353,71 +1431,38 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"DAC MIXR", "Stereo ADC Switch", "Stereo ADC MIXR"}, {"DAC MIXR", "INF1 Switch", "IF1 DAC R"}, - {"ANC", NULL, "Stereo ADC MIXL"}, - {"ANC", NULL, "Stereo ADC MIXR"}, - - {"Audio DSP", NULL, "DAC MIXL"}, - {"Audio DSP", NULL, "DAC MIXR"}, - - {"DAC L2 Mux", "IF2", "IF2 DAC L"}, - {"DAC L2 Mux", "Base L/R", "Audio DSP"}, - - {"DAC R2 Mux", "IF2", "IF2 DAC R"}, - {"Stereo DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, - {"Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, - {"Stereo DAC MIXL", "ANC Switch", "ANC"}, {"Stereo DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, - {"Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, - {"Stereo DAC MIXR", "ANC Switch", "ANC"}, {"Mono DAC MIXL", "DAC L1 Switch", "DAC MIXL"}, - {"Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, - {"Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Mux"}, {"Mono DAC MIXR", "DAC R1 Switch", "DAC MIXR"}, - {"Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, - {"Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Mux"}, {"DIG MIXL", "DAC L1 Switch", "DAC MIXL"}, - {"DIG MIXL", "DAC L2 Switch", "DAC L2 Mux"}, {"DIG MIXR", "DAC R1 Switch", "DAC MIXR"}, - {"DIG MIXR", "DAC R2 Switch", "DAC R2 Mux"}, {"DAC L1", NULL, "Stereo DAC MIXL"}, {"DAC L1", NULL, "PLL1", is_sys_clk_from_pll}, {"DAC R1", NULL, "Stereo DAC MIXR"}, {"DAC R1", NULL, "PLL1", is_sys_clk_from_pll}, - {"DAC L2", NULL, "Mono DAC MIXL"}, - {"DAC L2", NULL, "PLL1", is_sys_clk_from_pll}, - {"DAC R2", NULL, "Mono DAC MIXR"}, - {"DAC R2", NULL, "PLL1", is_sys_clk_from_pll}, {"SPK MIXL", "REC MIXL Switch", "RECMIXL"}, {"SPK MIXL", "INL Switch", "INL VOL"}, {"SPK MIXL", "DAC L1 Switch", "DAC L1"}, - {"SPK MIXL", "DAC L2 Switch", "DAC L2"}, {"SPK MIXL", "OUT MIXL Switch", "OUT MIXL"}, {"SPK MIXR", "REC MIXR Switch", "RECMIXR"}, {"SPK MIXR", "INR Switch", "INR VOL"}, {"SPK MIXR", "DAC R1 Switch", "DAC R1"}, - {"SPK MIXR", "DAC R2 Switch", "DAC R2"}, {"SPK MIXR", "OUT MIXR Switch", "OUT MIXR"}, - {"OUT MIXL", "SPK MIXL Switch", "SPK MIXL"}, {"OUT MIXL", "BST1 Switch", "BST1"}, {"OUT MIXL", "INL Switch", "INL VOL"}, {"OUT MIXL", "REC MIXL Switch", "RECMIXL"}, - {"OUT MIXL", "DAC R2 Switch", "DAC R2"}, - {"OUT MIXL", "DAC L2 Switch", "DAC L2"}, {"OUT MIXL", "DAC L1 Switch", "DAC L1"}, - {"OUT MIXR", "SPK MIXR Switch", "SPK MIXR"}, {"OUT MIXR", "BST2 Switch", "BST2"}, {"OUT MIXR", "BST1 Switch", "BST1"}, {"OUT MIXR", "INR Switch", "INR VOL"}, {"OUT MIXR", "REC MIXR Switch", "RECMIXR"}, - {"OUT MIXR", "DAC L2 Switch", "DAC L2"}, - {"OUT MIXR", "DAC R2 Switch", "DAC R2"}, {"OUT MIXR", "DAC R1 Switch", "DAC R1"}, {"SPKVOL L", NULL, "SPK MIXL"}, @@ -1436,11 +1481,9 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"SPOR MIX", "SPKVOL R Switch", "SPKVOL R"}, {"SPOR MIX", "BST1 Switch", "BST1"}, - {"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"}, {"HPO MIX L", "HPO MIX DAC1 Switch", "DAC L1"}, {"HPO MIX L", "HPO MIX HPVOL Switch", "HPOVOL L"}, {"HPO MIX L", NULL, "HP L Amp"}, - {"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"}, {"HPO MIX R", "HPO MIX DAC1 Switch", "DAC R1"}, {"HPO MIX R", "HPO MIX HPVOL Switch", "HPOVOL R"}, {"HPO MIX R", NULL, "HP R Amp"}, @@ -1450,12 +1493,6 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"LOUT MIX", "OUTVOL L Switch", "OUTVOL L"}, {"LOUT MIX", "OUTVOL R Switch", "OUTVOL R"}, - {"Mono MIX", "DAC R2 Switch", "DAC R2"}, - {"Mono MIX", "DAC L2 Switch", "DAC L2"}, - {"Mono MIX", "OUTVOL R Switch", "OUTVOL R"}, - {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, - {"Mono MIX", "BST1 Switch", "BST1"}, - {"HP Amp", NULL, "HPO MIX L"}, {"HP Amp", NULL, "HPO MIX R"}, @@ -1480,11 +1517,82 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"HPOR", NULL, "HP R Playback"}, {"LOUTL", NULL, "LOUT MIX"}, {"LOUTR", NULL, "LOUT MIX"}, +}; + +static const struct snd_soc_dapm_route rt5640_specific_dapm_routes[] = { + {"ANC", NULL, "Stereo ADC MIXL"}, + {"ANC", NULL, "Stereo ADC MIXR"}, + + {"Audio DSP", NULL, "DAC MIXL"}, + {"Audio DSP", NULL, "DAC MIXR"}, + + {"DAC L2 Mux", "IF2", "IF2 DAC L"}, + {"DAC L2 Mux", "Base L/R", "Audio DSP"}, + + {"DAC R2 Mux", "IF2", "IF2 DAC R"}, + + {"Stereo DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + {"Stereo DAC MIXL", "ANC Switch", "ANC"}, + {"Stereo DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"Stereo DAC MIXR", "ANC Switch", "ANC"}, + + {"Mono DAC MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + {"Mono DAC MIXL", "DAC R2 Switch", "DAC R2 Mux"}, + + {"Mono DAC MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"Mono DAC MIXR", "DAC L2 Switch", "DAC L2 Mux"}, + + {"DIG MIXR", "DAC R2 Switch", "DAC R2 Mux"}, + {"DIG MIXL", "DAC L2 Switch", "DAC L2 Mux"}, + + {"DAC L2", NULL, "Mono DAC MIXL"}, + {"DAC L2", NULL, "PLL1", is_sys_clk_from_pll}, + {"DAC R2", NULL, "Mono DAC MIXR"}, + {"DAC R2", NULL, "PLL1", is_sys_clk_from_pll}, + + {"SPK MIXL", "DAC L2 Switch", "DAC L2"}, + {"SPK MIXR", "DAC R2 Switch", "DAC R2"}, + + {"OUT MIXL", "SPK MIXL Switch", "SPK MIXL"}, + {"OUT MIXR", "SPK MIXR Switch", "SPK MIXR"}, + + {"OUT MIXL", "DAC R2 Switch", "DAC R2"}, + {"OUT MIXL", "DAC L2 Switch", "DAC L2"}, + + {"OUT MIXR", "DAC L2 Switch", "DAC L2"}, + {"OUT MIXR", "DAC R2 Switch", "DAC R2"}, + + {"HPO MIX L", "HPO MIX DAC2 Switch", "DAC L2"}, + {"HPO MIX R", "HPO MIX DAC2 Switch", "DAC R2"}, + + {"Mono MIX", "DAC R2 Switch", "DAC R2"}, + {"Mono MIX", "DAC L2 Switch", "DAC L2"}, + {"Mono MIX", "OUTVOL R Switch", "OUTVOL R"}, + {"Mono MIX", "OUTVOL L Switch", "OUTVOL L"}, + {"Mono MIX", "BST1 Switch", "BST1"}, + {"MONOP", NULL, "Mono MIX"}, {"MONON", NULL, "Mono MIX"}, {"MONOP", NULL, "Improve MONO Amp Drv"}, }; +static const struct snd_soc_dapm_route rt5639_specific_dapm_routes[] = { + {"Stereo DAC MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"Stereo DAC MIXR", "DAC R2 Switch", "IF2 DAC R"}, + + {"Mono DAC MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"Mono DAC MIXL", "DAC R2 Switch", "IF2 DAC R"}, + + {"Mono DAC MIXR", "DAC R2 Switch", "IF2 DAC R"}, + {"Mono DAC MIXR", "DAC L2 Switch", "IF2 DAC L"}, + + {"DIG MIXL", "DAC L2 Switch", "IF2 DAC L"}, + {"DIG MIXR", "DAC R2 Switch", "IF2 DAC R"}, + + {"IF2 DAC L", NULL, "DAC L2 Filter"}, + {"IF2 DAC R", NULL, "DAC R2 Filter"}, +}; + static int get_sdp_info(struct snd_soc_codec *codec, int dai_id) { int ret = 0, val; @@ -1551,8 +1659,7 @@ static int get_clk_info(int sclk, int rate) static int rt5640_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); unsigned int val_len = 0, val_clk, mask_clk; int dai_sel, pre_div, bclk_ms, frame_size; @@ -1901,6 +2008,28 @@ static int rt5640_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, RT5640_MICBIAS, 0x0030, 0x0030); snd_soc_update_bits(codec, RT5640_DSP_PATH2, 0xfc00, 0x0c00); + switch (snd_soc_read(codec, RT5640_RESET)) { + case RT5640_RESET_ID: + snd_soc_add_codec_controls(codec, + rt5640_specific_snd_controls, + ARRAY_SIZE(rt5640_specific_snd_controls)); + snd_soc_dapm_new_controls(&codec->dapm, + rt5640_specific_dapm_widgets, + ARRAY_SIZE(rt5640_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5640_specific_dapm_routes, + ARRAY_SIZE(rt5640_specific_dapm_routes)); + break; + case RT5639_RESET_ID: + snd_soc_dapm_new_controls(&codec->dapm, + rt5639_specific_dapm_widgets, + ARRAY_SIZE(rt5639_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5639_specific_dapm_routes, + ARRAY_SIZE(rt5639_specific_dapm_routes)); + break; + } + return 0; } diff --git a/sound/soc/codecs/rt5640.h b/sound/soc/codecs/rt5640.h index d7bd525..3b50459 100644 --- a/sound/soc/codecs/rt5640.h +++ b/sound/soc/codecs/rt5640.h @@ -14,6 +14,9 @@ #include <sound/rt5640.h> +#define RT5639_RESET_ID 0x0008 +#define RT5640_RESET_ID 0x000c + /* Info */ #define RT5640_RESET 0x00 #define RT5640_VENDOR_ID 0xfd -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] [PATCH] ASoC: core: Allow snd_soc_update_bits use 32 bits register
by Bard Liao 08 Apr '14

08 Apr '14

Change reg's type from unsigned short to unsigned int. So that we can use 32 bits reg value in snd_soc_update_bits. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- include/sound/soc.h | 6 +++--- sound/soc/soc-core.c | 8 ++++---- 2 files changed, 7 insertions(+), 7 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 5878410..b18cf27 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -469,12 +469,12 @@ static inline void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, #endif /* codec register bit access */ -int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, +int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value); int snd_soc_update_bits_locked(struct snd_soc_codec *codec, - unsigned short reg, unsigned int mask, + unsigned int reg, unsigned int mask, unsigned int value); -int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, +int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value); int snd_soc_new_ac97_codec(struct snd_soc_codec *codec, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cd52d52..7e2734d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2316,7 +2316,7 @@ EXPORT_SYMBOL_GPL(snd_soc_write); * * Returns 1 for change, 0 for no change, or negative error code. */ -int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, +int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value) { bool change; @@ -2357,7 +2357,7 @@ EXPORT_SYMBOL_GPL(snd_soc_update_bits); * Returns 1 for change else 0. */ int snd_soc_update_bits_locked(struct snd_soc_codec *codec, - unsigned short reg, unsigned int mask, + unsigned int reg, unsigned int mask, unsigned int value) { int change; @@ -2382,7 +2382,7 @@ EXPORT_SYMBOL_GPL(snd_soc_update_bits_locked); * * Returns 1 for change else 0. */ -int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg, +int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value) { int change; @@ -2897,7 +2897,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, int min = mc->min; int mask = (1 << (fls(min + max) - 1)) - 1; int err = 0; - unsigned short val, val_mask, val2 = 0; + unsigned int val, val_mask, val2 = 0; val_mask = mask << shift; val = (ucontrol->value.integer.value[0] + min) & mask; -- 1.8.1.1.439.g50a6b54

2 1

Re: [alsa-devel] Tenor TE8802, USB : clics and crackles during music play
by mick 08 Apr '14

08 Apr '14

Clemens, please help me out: are you asking me personally to test or is this a general call for the group? If you need me to test, what version of kernel sources do I patch? I am probably not running the latest. Forgive the questions, I do not routinely do these things. Also: Your patch looks quite different from the solution I pointed out. Why? Did you test it yourself? In particular the assertion: > + * The TEAC UD-H01 firmware sometimes changes the feedback value > + * by +/- 0x1.0000. is quite different from the assertion "the 3rd byte of the feadback value is getting corrupted and probably changing when it should not " from http://forums.slimdevices.com/showthread.php?93310-Async-USB-Dac-Experiment…. Also, you made it quite explicit for the TEAC UD-H01 but I think it should be more general. I have the Teac A-H01 and it also has the problem. As far as I gather from the various discussions on the Web it probably is a general Tenor 8802 issue. For instance with Nuforce DAC-9 here: http://www.spinics.net/linux/fedora/alsa-user/msg11387.html Mick On Mon, 07 Apr 2014 22:09:21 +0200, Clemens Ladisch <clemens(a)ladisch.de> wrote: > Please test this patch: > > --8<----------------------------------------------------------------->8-- > ALSA: usb-audio: work around corrupted TEAC UD-H01 feedback data > > The TEAC UD-H01 firmware sends wrong feedback frequency values, thus > causing the PC to send the samples at a wrong rate, which results in > clicks and crackles in the output. > > Add a workaround to detect and fix the corruption. > > Signed-off-by: Clemens Ladisch <clemens(a)ladisch.de> > Tested-by: Mick <mick37(a)gmx.de> > Cc: <stable(a)vger.kernel.org> > --- > sound/usb/card.h | 1 + > sound/usb/endpoint.c | 15 ++++++++++++++- > 2 files changed, 15 insertions(+), 1 deletion(-) > > diff --git a/sound/usb/card.h b/sound/usb/card.h > index 9867ab8..97acb90 100644 > --- a/sound/usb/card.h > +++ b/sound/usb/card.h > @@ -92,6 +92,7 @@ struct snd_usb_endpoint { > unsigned int curframesize; /* current packet size in frames (for > capture) */ > unsigned int syncmaxsize; /* sync endpoint packet size */ > unsigned int fill_max:1; /* fill max packet size always */ > + unsigned int udh01_fb_quirk:1; /* corrupted feedback data */ > unsigned int datainterval; /* log_2 of data packet interval */ > unsigned int syncinterval; /* P for adaptive mode, 0 otherwise */ > unsigned char silence_value; > diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c > index e70a87e..7b400eb 100644 > --- a/sound/usb/endpoint.c > +++ b/sound/usb/endpoint.c > @@ -471,6 +471,10 @@ struct snd_usb_endpoint > *snd_usb_add_endpoint(struct snd_usb_audio *chip, > ep->syncinterval = 3; > > ep->syncmaxsize = le16_to_cpu(get_endpoint(alts, 1)->wMaxPacketSize); > + > + if (chip->usb_id == USB_ID(0x0644, 0x8038) /* TEAC UD-H01 */ && > + ep->syncmaxsize == 4) > + ep->udh01_fb_quirk = 1; > } > > list_add_tail(&ep->list, &chip->ep_list); > @@ -1105,7 +1109,16 @@ void snd_usb_handle_sync_urb(struct > snd_usb_endpoint *ep, > if (f == 0) > return; > > - if (unlikely(ep->freqshift == INT_MIN)) { > + if (unlikely(ep->udh01_fb_quirk)) { > + /* > + * The TEAC UD-H01 firmware sometimes changes the feedback value > + * by +/- 0x1.0000. > + */ > + if (f < ep->freqn - 0x8000) > + f += 0x10000; > + else if (f > ep->freqn + 0x8000) > + f -= 0x10000; > + } else if (unlikely(ep->freqshift == INT_MIN)) { > /* > * The first time we see a feedback value, determine its format > * by shifting it left or right until it matches the nominal -- Erstellt mit Operas E-Mail-Modul: http://www.opera.com/mail/

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Re: [alsa-devel] [PATCH] ALSA: hda - Do not assign streams in reverse order
by Takashi Iwai 08 Apr '14

08 Apr '14

At Mon, 7 Apr 2014 22:36:38 +0300, Anssi Hannula wrote: > > Currently stream numbers are assigned in reverse order. > > Unfortunately commit 7546abfb8e1f9933b5 ("ALSA: hda - Increment > default stream numbers for AMD HDMI controllers") assumed this was not > the case (specifically, it had the "old cards had single device only" > => "extra unused stream numbers do not matter" assumption), causing > non-working audio regressions for AMD Radeon HDMI users on v3.14. > > Change the stream numbers to be assigned in increasing order instead. > > Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=77002 > Reported-by: Christian Güdel <cg(a)dmesg.ch> > Signed-off-by: Anssi Hannula <anssi.hannula(a)iki.fi> > Tested-by: Christian Güdel <cg(a)dmesg.ch> > Cc: Alex Deucher <alexander.deucher(a)amd.com> > Cc: <stable(a)vger.kernel.org> # 3.14 > --- > > Takashi, please check I didn't miss anything (I'm not that familiar > with this low-level HDA code). :) In that case, let's revert the commit instead. There was a reason to behave in that way, although I don't remember well. That said, this change might lead to another regression. Thus, if a fix is needed for stable, the revert would be the best choice as a first aid. Of course, this change is good to have, so let's implement this for 3.15 and see whether it works well for all nasty controllers. Besides, this patch can't be applied any longer to the current code base since it's been rewritten a lot; most of codes have been moved to another file hda_controller.c. Could you rebase the patch for the latest tree? Meanwhile I'll prepare a revert that is applied only to 3.14.x. thanks, Takashi > > > sound/pci/hda/hda_intel.c | 9 +++++---- > 1 file changed, 5 insertions(+), 4 deletions(-) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index e354ab1ec20f..7a7b7b10dddf 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -1856,13 +1856,14 @@ azx_assign_device(struct azx *chip, struct snd_pcm_substream *substream) > struct azx_dev *azx_dev = &chip->azx_dev[dev]; > dsp_lock(azx_dev); > if (!azx_dev->opened && !dsp_is_locked(azx_dev)) { > - res = azx_dev; > - if (res->assigned_key == key) { > - res->opened = 1; > - res->assigned_key = key; > + if (azx_dev->assigned_key == key) { > + azx_dev->opened = 1; > + azx_dev->assigned_key = key; > dsp_unlock(azx_dev); > return azx_dev; > } > + if (!res) > + res = azx_dev; > } > dsp_unlock(azx_dev); > } > -- > 1.8.4.5 >

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Re: [alsa-devel] Audio codec with multiple I2C devices
by Lars-Peter Clausen 08 Apr '14

08 Apr '14

On 04/07/2014 09:29 PM, Gilles wrote: > On Apr 7, 2014, at 12:03 , Lars-Peter Clausen <lars(a)metafoo.de> wrote: > >> On 04/07/2014 08:32 PM, Gilles wrote: >> [...] >>> Doesn't it make perfect sense to have the Volume be a control of the Codec driver even if it's provided by a separate chip on the same sound card? >> >> Not if they are standalone components that can also be used in different configurations on different sound cards. E.g. you code suggests that you are using a AK4113 and a DAC121C085. The CODEC driver is not the sound card driver, the sound card driver is the piece of code that glues all the different components that are found on the sound card together. >> >> i2c_new_dummy() is typically used for devices with multiple I2C addresses. Or where you have e.g. multiple devices with different addresses on the same die so they become a inseparable unit. >> >> - Lars > > Yes, but the functionality is very specific to the way the components are wired on the hardware. They are wired to work in conjunction with the ASIC Codec. > > The DAC121C085 is a very specific case as the DAC is actually wired to control the voltage of 4 quadrant multiplier which is hardwired to the Codec ASIC. This is a prototype. In the final product, the DAC chip may be built into the ASIC. For this one, I feel like the "Volume" should be a control of the same driver as the ASIC. > > This said I see your point about the AK4113 which is a generic multi channel receiver. In this specific application it is wired so only inputs 1,2 and 3 are hard wired (this actually isn't a sound card, but rather a finished product using Linux as the SoC OS as can be seen on the home page of http://birdland.com right now). > > There is already a working prototype, but as of now, the inputs are selected by the application talking directly to the I2C AKM. I'm just in the process of streamlining the code, and naturally, I envisioned the input selection be part of the sound card codec as I'd much rather have the sound card have a enum control with the 4 inputs shown by name show under amixer (this would enable my application to be more generic). > > Can you please point me to an example of a Sound board driver that uses multiple codecs? I suppose I can modify my sound board driver to use a codec for the audio and another for the AKM input selector). Anything that uses aux devs. E.g. samsung/speyside.c There is also this series https://lkml.org/lkml/2014/4/5/92 which adds DT support for aux devs. - Lars

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[alsa-devel] [PATCH 00/44 v3] Enhancement of support for Firewire devices
by Takashi Sakamoto 08 Apr '14

08 Apr '14

This series is an update for my previous series: [alsa-devel] [GIT PULL][PATCH 00/39 v2] Enhancement of support for Firewire devices http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/073868.html Updates since previous pull-request: firewire-lib: - remove packet sorting - remove adding transfer delay to out-packets - fix a bug of MIDI capturing for several ports - add detecting discontinuity of packets - add a support for deferred transaction - add some flags for device's quirk related to AMDTP packets - improve return value of some AV/C general commands fireworks: - add a workaround for out-of-specification dbc meaning - add a workaround for AudioFire9 firmware - add a workaround for firmware version 5.5 bebob: - add some flags for device's quirk related to AMDTP packets - add a workaround for discontinuity of packets at bus reset - add workarounds for M-Audio spefial Firewire series This time I intentionally add no restriction of version for Fireworks because I confirmed that my model doesn't always transmit packets with disorder. I expect users to retry starging streams several times till detecting no discontinuity of packets. [alsa-devel] [PATCH 09/39] firewire-lib: Add sort function for transmitted packet http://mailman.alsa-project.org/pipermail/alsa-devel/2014-March/074157.html I note that a workaround for M-Audio special Firewire series at [44/44] relates to Linux Firewire subsystem. Please follow: Generating bus-reset by kernel module in the other subsystem http://sourceforge.net/p/linux1394/mailman/linux1394-devel/thread/5326B651.… ---------------------------------------------------------------- The following changes since commit 2df6742f613840a0b0a1590fb28f7af5b058a673: ALSA: hda - verify pin:cvt connection on preparing a stream for Intel HDMI codec (2014-03-20 07:36:17 +0100) are available in the git repository at: https://github.com/takaswie/sound.git request for you to fetch changes up to aa0564fdd013e5754b73ba89b50c722bbedbc4ac: firewire/bebob: Add a workaround for M-Audio special Firewire series (2014-03-21 19:35:04 +0900) ---------------------------------------------------------------- Takashi Sakamoto (44): firewire-lib: Add macros instead of fixed value for AMDTP firewire-lib: Add 'direction' member to 'amdtp_stream' structure firewire-lib: Split some codes into functions to reuse for both streams firewire-lib: Add support for AMDTP in-stream and PCM capture firewire-lib: Add support for MIDI capture/playback firewire-lib: Give syt value as parameter to handle_out_packet() firewire-lib: Add support for duplex streams synchronization in blocking mode firewire-lib: Add support for channel mapping firewire-lib: Restrict calling flush_context_completion() when context exists firewire-lib: Rename macros, variables and functions for CMP firewire-lib: Add 'direction' member to 'cmp_connection' structure firewire-lib: Add handling output connection by CMP firewire-lib: Add a new function to check others' connection firewire-lib: Add support for deferred transaction firewire-lib: Add some AV/C general commands fireworks: Add skelton for Fireworks based devices fireworks: Add transaction and some commands fireworks: Add connection and stream management fireworks/firewire-lib: Add a quirk for empty packet with TAG0 fireworks/firewire-lib: Add a quirk for the meaning of dbc fireworks/firewire-lib: Add a quirk for wrong dbs in tx packets fireworks/firewire-libs: Add a quirk for fixed interval of reported dbc fireworks: Add proc interface for debugging purpose fireworks: Add MIDI interface fireworks/firewire-lib: Add a quirk for data blocks for MIDI in out-stream fireworks: Add PCM interface fireworks: Add hwdep interface fireworks: Add command/response functionality into hwdep interface bebob: Add skelton for BeBoB based devices bebob: Add commands and connections/streams management bebob/firewire-lib: Add a quirk for discontinuity at bus reset bebob: Add proc interface for debugging purpose bebob: Add MIDI interface bebob: Add PCM interface bebob: Add hwdep interface bebob: Prepare for device specific operations bebob: Add support for Terratec PHASE, EWS series and Aureon bebob: Add support for Yamaha GO series bebob: Add support for Focusrite Saffire/SaffirePro series bebob: Add support for M-Audio usual Firewire series bebob: Add support for M-Audio special Firewire series bebob/firewire-lib: Add a quirk of wrong dbc in empty packet for M-Audio special Firewire series bebob: Send a cue to load firmware for M-Audio Firewire series firewire/bebob: Add a workaround for M-Audio special Firewire series drivers/firewire/core.h | 1 - include/linux/firewire.h | 3 + include/uapi/sound/asound.h | 4 +- include/uapi/sound/firewire.h | 22 +- sound/firewire/Kconfig | 59 ++ sound/firewire/Makefile | 2 + sound/firewire/amdtp.c | 668 ++++++++++++---- sound/firewire/amdtp.h | 124 ++- sound/firewire/bebob/Makefile | 4 + sound/firewire/bebob/bebob.c | 470 ++++++++++++ sound/firewire/bebob/bebob.h | 257 +++++++ sound/firewire/bebob/bebob_command.c | 311 ++++++++ sound/firewire/bebob/bebob_focusrite.c | 287 +++++++ sound/firewire/bebob/bebob_hwdep.c | 197 +++++ sound/firewire/bebob/bebob_maudio.c | 832 ++++++++++++++++++++ sound/firewire/bebob/bebob_midi.c | 168 ++++ sound/firewire/bebob/bebob_pcm.c | 451 +++++++++++ sound/firewire/bebob/bebob_proc.c | 201 +++++ sound/firewire/bebob/bebob_stream.c | 929 +++++++++++++++++++++++ sound/firewire/bebob/bebob_terratec.c | 75 ++ sound/firewire/bebob/bebob_yamaha.c | 54 ++ sound/firewire/cmp.c | 203 +++-- sound/firewire/cmp.h | 14 +- sound/firewire/dice.c | 2 +- sound/firewire/fcp.c | 192 ++++- sound/firewire/fcp.h | 21 + sound/firewire/fireworks/Makefile | 4 + sound/firewire/fireworks/fireworks.c | 331 ++++++++ sound/firewire/fireworks/fireworks.h | 227 ++++++ sound/firewire/fireworks/fireworks_command.c | 397 ++++++++++ sound/firewire/fireworks/fireworks_hwdep.c | 294 +++++++ sound/firewire/fireworks/fireworks_midi.c | 179 +++++ sound/firewire/fireworks/fireworks_pcm.c | 477 ++++++++++++ sound/firewire/fireworks/fireworks_proc.c | 233 ++++++ sound/firewire/fireworks/fireworks_stream.c | 347 +++++++++ sound/firewire/fireworks/fireworks_transaction.c | 327 ++++++++ sound/firewire/speakers.c | 48 +- 37 files changed, 8157 insertions(+), 258 deletions(-) create mode 100644 sound/firewire/bebob/Makefile create mode 100644 sound/firewire/bebob/bebob.c create mode 100644 sound/firewire/bebob/bebob.h create mode 100644 sound/firewire/bebob/bebob_command.c create mode 100644 sound/firewire/bebob/bebob_focusrite.c create mode 100644 sound/firewire/bebob/bebob_hwdep.c create mode 100644 sound/firewire/bebob/bebob_maudio.c create mode 100644 sound/firewire/bebob/bebob_midi.c create mode 100644 sound/firewire/bebob/bebob_pcm.c create mode 100644 sound/firewire/bebob/bebob_proc.c create mode 100644 sound/firewire/bebob/bebob_stream.c create mode 100644 sound/firewire/bebob/bebob_terratec.c create mode 100644 sound/firewire/bebob/bebob_yamaha.c create mode 100644 sound/firewire/fireworks/Makefile create mode 100644 sound/firewire/fireworks/fireworks.c create mode 100644 sound/firewire/fireworks/fireworks.h create mode 100644 sound/firewire/fireworks/fireworks_command.c create mode 100644 sound/firewire/fireworks/fireworks_hwdep.c create mode 100644 sound/firewire/fireworks/fireworks_midi.c create mode 100644 sound/firewire/fireworks/fireworks_pcm.c create mode 100644 sound/firewire/fireworks/fireworks_proc.c create mode 100644 sound/firewire/fireworks/fireworks_stream.c create mode 100644 sound/firewire/fireworks/fireworks_transaction.c -- 1.8.3.2

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[alsa-devel] Audio codec with multiple I2C devices
by Gilles 07 Apr '14

07 Apr '14

Folks, I'm writing an ALSA SoC codec which uses a 3 distinct I2C devices (a DAC, a CAG for volume and a input selector) all 3 at different I2C addresses. My existing code relies on DT to declare the primary I2C address (in this case the DAC) and it works. But I now need to add support for the extended functionality. Can someone point me to the best way to declare the second I2C address in the DT and make use of it in the driver? I know I could always hard code it and add a new i2c_new_device() in the probe() into the private data but I'm trying to make it a bit more DT friendly. Not having much luck on finding examples of this. I'm attaching the existing code code below if anyone wants to reference to snipets of it. As you can see, the set/get volume are declared as _EXT since the volume is controlled by a secondary GC I2C device. Similarily, the input selector talks to a third I2C device. Any help would be appreciated. Thanks! /* // ALSA SoC codec driver for Birdland Audio SDM-50 Digital Amplifier // Copyright (C) 2012 Birdland Audio - http://birdland.com // Author: Gilles Gameiro <gilles(a)whospot.com> // */ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> #include <linux/gpio.h> #include <linux/of_gpio.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/initval.h> #include <sound/tlv.h> #define DRV_NAME "sdm2x-audio-codec" #define BIRDLAND_MODEL_250 0 #define SDM2X_REG0_VOLUME 0 #define SDM2X_REG1_MUXCLK 1 #define SDM2X_INPUT_EXTERNAL 0x80 #define SDM2X_PLAYRATE_48000 0x08 #define SDM2X_RATES ( 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 SDM2X_FORMATS ( SNDRV_PCM_FMTBIT_S16_LE \ | SNDRV_PCM_FMTBIT_S20_3LE \ | SNDRV_PCM_FMTBIT_S24_3LE \ | SNDRV_PCM_FMTBIT_S32_LE ) /* codec private data */ struct sdm2x_priv { struct snd_soc_codec *codec; enum snd_soc_control_type control_type; u16 model; unsigned int sysclk; u8 input_cache; // Caches the back panel input selected (Controlled by AK4113) u8 volume_cache; // Caches the current value of the volume (Controlled by DAC121C085) }; // FPGA I2C Slave IP is write only. Use register cache #define SDM2X_RCACHE_SIZE 4 static const u8 sdm2x_reg_cache[SDM2X_RCACHE_SIZE] = { 0x00, (SDM2X_INPUT_EXTERNAL | SDM2X_PLAYRATE_48000), 0x00, 0x00, }; static int sdm250_get_volume(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); ucontrol->value.integer.value[0] = sdm2x->volume_cache; return 0; } static int sdm250_set_volume(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); int volume = ucontrol->value.integer.value[0] & 0x7f; sdm2x->volume_cache = volume; // TODO: Write it to GA return 0; } static int sdm250_get_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); ucontrol->value.integer.value[0] = sdm2x->input_cache; return 0; } static int sdm250_set_input(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); int input = ucontrol->value.integer.value[0] & 0x7f; sdm2x->input_cache = volume; // TODO: Write it to AK4113 return 0; } /* General Volume -127dB to 0dB on 1dB steps */ static DECLARE_TLV_DB_SCALE(general_volume_tlv, -12700, 100, 0); /* Enum declarations */ static const char *sdm2x_srcin[] = {"Internal", "External"}; static const struct soc_enum sdm2x_srcin_enum = SOC_ENUM_SINGLE(SDM2X_REG1_MUXCLK, 7, 2, sdm2x_srcin); static const char *sdm2x_i2sin[] = {"Optical 1", "Optical 2", "Optical 3", "XLR" }; static const struct soc_enum sdm2x_i2sin_enum = SOC_ENUM_SINGLE(SDM2X_REG1_MUXCLK, 0, 3, sdm2x_i2sin); static const struct snd_kcontrol_new sdm2x_snd_controls[] = { /* General Volume: This is set outside of the FPGA so use getter/setter functions */ SOC_SINGLE_EXT_TLV("Amplifier Volume", SDM2X_REG0_VOLUME, 0, 127, 0, sdm250_get_volume, sdm250_set_volume, general_volume_tlv), /* FPGA Manged Controls (primary I2C of this driver) */ SOC_SINGLE("Digital Mute", SDM2X_REG0_VOLUME, 7, 1, 0), // TODO: Make this a DAI op? SOC_ENUM("SRC Input Source", sdm2x_srcin_enum), /* Back Pannel Controls (AK4113) */ SOC_ENUM_EXT("Back Panel Input", sdm2x_i2sin_enum, sdm250_get_input, sdm250_set_input) }; static int sdm2x_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 sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); int fsref; unsigned char divreg; // Set Data word length not needed in I2S) fsref = params_format(params); // Set Sampling Rate: I haven't figured out how to make the mcasp I2S master for TX and slave for RX so let the FPGA generate the clocks Sampling Rate switch (fsref) { case SNDRV_PCM_RATE_32000: divreg = 0x0C; break; case SNDRV_PCM_RATE_44100: case SNDRV_PCM_RATE_48000: divreg = 0x08; break; case SNDRV_PCM_RATE_64000: divreg = 0x06; break; case SNDRV_PCM_RATE_88200: case SNDRV_PCM_RATE_96000: divreg = 0x04; break; case SNDRV_PCM_RATE_176400: case SNDRV_PCM_RATE_192000: divreg = 0x02; break; default: printk(KERN_ERR "sdm2x-codec.%s(): Requested rate 0x%08x not supported\n", __func__, fsref); return -EINVAL; } if (fsref % 11025 == 0) divreg |= 0x10; snd_soc_update_bits(codec, SDM2X_REG1_MUXCLK, 0x1F, divreg); return 0; } // DAI Ops static int sdm2x_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { //struct snd_soc_codec *codec = codec_dai->codec; //struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); // SDM-50 only supports vanilla I2S format. switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF): break; default: printk(KERN_ERR "sdm2x-codec.%s(): Requested Format 0x%08x not supported\n", __func__, (fmt & SND_SOC_DAIFMT_FORMAT_MASK)); return -EINVAL; } // SDM-50 Uses normal I2S Clock polarities switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: break; default: printk(KERN_ERR "sdm2x-codec.%s(): Requested Clock Polarity 0x%08x not supported\n", __func__, (fmt & SND_SOC_DAIFMT_INV_MASK)); return -EINVAL; } // And SDM-50 is the I2S Clock Master for both Bit and Frame switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case (SND_SOC_DAIFMT_CBM_CFM): break; default: printk(KERN_ERR "sdm2x-codec.%s(): Requested Master/Slave Clock 0x%08x not supported\n", __func__, (fmt & SND_SOC_DAIFMT_MASTER_MASK)); return -EINVAL; } return 0; } static int sdm2x_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 sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); sdm2x->sysclk = freq; return 0; } static const struct snd_soc_dai_ops sdm2x_dai_ops = { .hw_params = sdm2x_hw_params, .set_fmt = sdm2x_set_dai_fmt, .set_sysclk = sdm2x_set_dai_sysclk, //.digital_mute = sdm2x_mute, }; static struct snd_soc_dai_driver sdm2x_dai = { .name = "sdm2x-hifi", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = SDM2X_RATES, .formats = SDM2X_FORMATS, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = SDM2X_RATES, .formats = SDM2X_FORMATS, }, .ops = &sdm2x_dai_ops, .symmetric_rates = 1, }; static int sdm2x_suspend(struct snd_soc_codec *codec) { return 0; } static int sdm2x_resume(struct snd_soc_codec *codec) { return 0; } /* PROBE: Setup private data, initialize codec and attach it to ASoC */ static int sdm2x_probe(struct snd_soc_codec *codec) { int ret; struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); sdm2x->codec = codec; // This belongs in the core but not sure if 3.8.13 supports it already.. ret = snd_soc_codec_set_cache_io(codec, 8, 8, sdm2x->control_type); if (ret != 0) { printk(KERN_ERR "sdm2x-codec.%s(): Failed to set cache I/O: %d\n", __func__, ret); return ret; } //codec->cache_only = 1; // Init... snd_soc_write(codec, SDM2X_REG0_VOLUME, 0); snd_soc_write(codec, SDM2X_REG1_MUXCLK, (SDM2X_INPUT_EXTERNAL | SDM2X_PLAYRATE_48000)); //snd_soc_update_bits(codec, SDM250_REG2_STATUS, SOFT_RESET, SOFT_RESET); // Auto Clears after reset complete (might not work with cached registers) // Add AMIXER controls (Volume, etc..) snd_soc_add_codec_controls(codec, sdm2x_snd_controls, ARRAY_SIZE(sdm2x_snd_controls)); return 0; } static int sdm2x_remove(struct snd_soc_codec *codec) { // struct sdm2x_priv *sdm2x = snd_soc_codec_get_drvdata(codec); // TODO: Where is private data freed up? return 0; } static struct snd_soc_codec_driver soc_codec_dev_sdm2x = { .reg_cache_size = ARRAY_SIZE(sdm2x_reg_cache), .reg_word_size = sizeof(u8), .reg_cache_default = sdm2x_reg_cache, .probe = sdm2x_probe, .remove = sdm2x_remove, .suspend = sdm2x_suspend, .resume = sdm2x_resume, }; static const struct i2c_device_id sdm2x_i2c_id[] = { { "sdm2x", BIRDLAND_MODEL_250 }, { } }; MODULE_DEVICE_TABLE(i2c, sdm2x_i2c_id); /* If the i2c layer weren't so broken, we could pass this kind of data around */ static int sdm2x_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct sdm2x_priv *sdm2x; struct device_node *np = i2c->dev.of_node; int ret; sdm2x = devm_kzalloc(&i2c->dev, sizeof(struct sdm2x_priv), GFP_KERNEL); if (sdm2x == NULL) { printk(KERN_ERR "sdm2x-codec.%s(): Failed to set cache I/O\n", __func__); return -ENOMEM; } // This driver doesn't support the old style platform data if (!np) { printk(KERN_ERR "sdm2x-codec.%s(): missing DT data info\n", __func__); return -EINVAL; } sdm2x->control_type = SND_SOC_I2C; i2c_set_clientdata(i2c, sdm2x); sdm2x->model = id->driver_data; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_sdm2x, &sdm2x_dai, 1); return ret; } static int sdm2x_i2c_remove(struct i2c_client *client) { // TODO: Where is private data freed up? snd_soc_unregister_codec(&client->dev); return 0; } #if defined(CONFIG_OF) static const struct of_device_id sdm2x_of_match[] = { { .compatible = "birdland,sdm2x", }, {}, }; MODULE_DEVICE_TABLE(of, sdm2x_of_match); #endif /* machine i2c codec control layer */ static struct i2c_driver sdm2x_i2c_driver = { .driver = { .name = "sdm2x-codec", .owner = THIS_MODULE, .of_match_table = of_match_ptr(sdm2x_of_match), }, .probe = sdm2x_i2c_probe, .remove = sdm2x_i2c_remove, .id_table = sdm2x_i2c_id, }; module_i2c_driver(sdm2x_i2c_driver); MODULE_DESCRIPTION("ASoC SDM2x codec driver"); MODULE_AUTHOR("Gilles Gameiro"); MODULE_LICENSE("GPL");

3 3

[alsa-devel] [PATCH v9] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 07 Apr '14

07 Apr '14

This patch adds initial support for the Behringer BCD2000 USB DJ controller. At the moment, only the MIDI part of the device is working, i.e. knobs, buttons and LEDs. I also plan to add support for the audio part, but I assume that this will require more effort than the rather simple MIDI interface. Progress can be tracked at https://github.com/anyc/snd-usb-bcd2000. Signed-off-by: Mario Kicherer <dev(a)kicherer.org> --- Changes since v8: - module_usb_driver() macro for module initialization - improved some comments Changes since v7: - replaced snd_card_create with snd_card_new Changes since v6: - applied more style improvements Changes since v5: - use kernel bitmap functions for devices_used Changes since v4: - devices_used as array to support arbitrary number of SNDRV_CARDS - removed unused array "enable" Changes since v3: - applied style and snd_printk changes as suggested by Daniel Mack Changes since v2: - applied more changes from Daniel Mack and Clemens Ladisch Changes since v1: - fixed the various code style issues, thanks to Daniel Mack and checkpatch.pl. sound/usb/Kconfig | 13 ++ sound/usb/Makefile | 2 +- sound/usb/bcd2000/Makefile | 3 + sound/usb/bcd2000/bcd2000.c | 461 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 478 insertions(+), 1 deletion(-) create mode 100644 sound/usb/bcd2000/Makefile create mode 100644 sound/usb/bcd2000/bcd2000.c diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index e05a86b..d393153 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -147,5 +147,18 @@ config SND_USB_HIFACE To compile this driver as a module, choose M here: the module will be called snd-usb-hiface. +config SND_BCD2000 + tristate "Behringer BCD2000 MIDI driver" + select SND_RAWMIDI + help + Say Y here to include MIDI support for the Behringer BCD2000 DJ + controller. + + Audio support is still work-in-progress at + https://github.com/anyc/snd-usb-bcd2000 + + To compile this driver as a module, choose M here: the module + will be called snd-bcd2000. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index abe668f..2b92f0d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ bcd2000/ diff --git a/sound/usb/bcd2000/Makefile b/sound/usb/bcd2000/Makefile new file mode 100644 index 0000000..f09ccc0 --- /dev/null +++ b/sound/usb/bcd2000/Makefile @@ -0,0 +1,3 @@ +snd-bcd2000-y := bcd2000.o + +obj-$(CONFIG_SND_BCD2000) += snd-bcd2000.o \ No newline at end of file diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c new file mode 100644 index 0000000..820d6ca --- /dev/null +++ b/sound/usb/bcd2000/bcd2000.c @@ -0,0 +1,461 @@ +/* + * Behringer BCD2000 driver + * + * Copyright (C) 2014 Mario Kicherer (dev(a)kicherer.org) + * + * 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. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/bitmap.h> +#include <linux/usb.h> +#include <linux/usb/audio.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/rawmidi.h> + +#define PREFIX "snd-bcd2000: " +#define BUFSIZE 64 + +static struct usb_device_id id_table[] = { + { USB_DEVICE(0x1397, 0x00bd) }, + { }, +}; + +static unsigned char device_cmd_prefix[] = {0x03, 0x00}; + +static unsigned char bcd2000_init_sequence[] = { + 0x07, 0x00, 0x00, 0x00, 0x78, 0x48, 0x1c, 0x81, + 0xc4, 0x00, 0x00, 0x00, 0x5e, 0x53, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff, 0x6c, 0xf3, 0x90, 0xff, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, + 0x18, 0xfa, 0x11, 0xff, 0x14, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0xf2, 0x34, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff +}; + +struct bcd2000 { + struct usb_device *dev; + struct snd_card *card; + struct usb_interface *intf; + int card_index; + + int midi_out_active; + struct snd_rawmidi *rmidi; + struct snd_rawmidi_substream *midi_receive_substream; + struct snd_rawmidi_substream *midi_out_substream; + + unsigned char midi_in_buf[BUFSIZE]; + unsigned char midi_out_buf[BUFSIZE]; + + struct urb *midi_out_urb; + struct urb *midi_in_urb; + + struct usb_anchor anchor; +}; + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; + +static DEFINE_MUTEX(devices_mutex); +DECLARE_BITMAP(devices_used, SNDRV_CARDS); +static struct usb_driver bcd2000_driver; + +#ifdef CONFIG_SND_DEBUG +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) +{ + print_hex_dump(KERN_DEBUG, prefix, + DUMP_PREFIX_NONE, 16, 1, + buf, len, false); +} +#else +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) {} +#endif + +static int bcd2000_midi_input_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_input_close(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +/* (de)register midi substream from client */ +static void bcd2000_midi_input_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + bcd2k->midi_receive_substream = up ? substream : NULL; +} + +static void bcd2000_midi_handle_input(struct bcd2000 *bcd2k, + const unsigned char *buf, unsigned int buf_len) +{ + unsigned int payload_length, tocopy; + struct snd_rawmidi_substream *midi_receive_substream; + + midi_receive_substream = ACCESS_ONCE(bcd2k->midi_receive_substream); + if (!midi_receive_substream) + return; + + bcd2000_dump_buffer(PREFIX "received from device: ", buf, buf_len); + + if (buf_len < 2) + return; + + payload_length = buf[0]; + + /* ignore packets without payload */ + if (payload_length == 0) + return; + + tocopy = min(payload_length, buf_len-1); + + bcd2000_dump_buffer(PREFIX "sending to userspace: ", + &buf[1], tocopy); + + snd_rawmidi_receive(midi_receive_substream, + &buf[1], tocopy); +} + +static void bcd2000_midi_send(struct bcd2000 *bcd2k) +{ + int len, ret; + struct snd_rawmidi_substream *midi_out_substream; + + BUILD_BUG_ON(sizeof(device_cmd_prefix) >= BUFSIZE); + + midi_out_substream = ACCESS_ONCE(bcd2k->midi_out_substream); + if (!midi_out_substream) + return; + + /* copy command prefix bytes */ + memcpy(bcd2k->midi_out_buf, device_cmd_prefix, + sizeof(device_cmd_prefix)); + + /* + * get MIDI packet and leave space for command prefix + * and payload length + */ + len = snd_rawmidi_transmit(midi_out_substream, + bcd2k->midi_out_buf + 3, BUFSIZE - 3); + + if (len < 0) + dev_err(&bcd2k->dev->dev, "%s: snd_rawmidi_transmit error %d\n", + __func__, len); + + if (len <= 0) + return; + + /* set payload length */ + bcd2k->midi_out_buf[2] = len; + bcd2k->midi_out_urb->transfer_buffer_length = BUFSIZE; + + bcd2000_dump_buffer(PREFIX "sending to device: ", + bcd2k->midi_out_buf, len+3); + + /* send packet to the BCD2000 */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s (%p): usb_submit_urb() failed, ret=%d, len=%d\n", + __func__, midi_out_substream, ret, len); + else + bcd2k->midi_out_active = 1; +} + +static int bcd2000_midi_output_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_output_close(struct snd_rawmidi_substream *substream) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (bcd2k->midi_out_active) { + usb_kill_urb(bcd2k->midi_out_urb); + bcd2k->midi_out_active = 0; + } + + return 0; +} + +/* (de)register midi substream from client */ +static void bcd2000_midi_output_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (up) { + bcd2k->midi_out_substream = substream; + /* check if there is data userspace wants to send */ + if (!bcd2k->midi_out_active) + bcd2000_midi_send(bcd2k); + } else { + bcd2k->midi_out_substream = NULL; + } +} + +static void bcd2000_output_complete(struct urb *urb) +{ + struct bcd2000 *bcd2k = urb->context; + + bcd2k->midi_out_active = 0; + + if (urb->status) + dev_warn(&urb->dev->dev, + PREFIX "output urb->status: %d\n", urb->status); + + if (urb->status == -ESHUTDOWN) + return; + + /* check if there is more data userspace wants to send */ + bcd2000_midi_send(bcd2k); +} + +static void bcd2000_input_complete(struct urb *urb) +{ + int ret; + struct bcd2000 *bcd2k = urb->context; + + if (urb->status) + dev_warn(&urb->dev->dev, + PREFIX "input urb->status: %i\n", urb->status); + + if (!bcd2k || urb->status == -ESHUTDOWN) + return; + + if (urb->actual_length > 0) + bcd2000_midi_handle_input(bcd2k, urb->transfer_buffer, + urb->actual_length); + + /* return URB to device */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() failed, ret=%d\n", + __func__, ret); +} + +static struct snd_rawmidi_ops bcd2000_midi_output = { + .open = bcd2000_midi_output_open, + .close = bcd2000_midi_output_close, + .trigger = bcd2000_midi_output_trigger, +}; + +static struct snd_rawmidi_ops bcd2000_midi_input = { + .open = bcd2000_midi_input_open, + .close = bcd2000_midi_input_close, + .trigger = bcd2000_midi_input_trigger, +}; + +static void bcd2000_init_device(struct bcd2000 *bcd2k) +{ + int ret; + + init_usb_anchor(&bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_out_urb, &bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_in_urb, &bcd2k->anchor); + + /* copy init sequence into buffer */ + memcpy(bcd2k->midi_out_buf, bcd2000_init_sequence, 52); + bcd2k->midi_out_urb->transfer_buffer_length = 52; + + /* submit sequence */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() out failed, ret=%d: ", + __func__, ret); + else + bcd2k->midi_out_active = 1; + + /* pass URB to device to enable button and controller events */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() in failed, ret=%d: ", + __func__, ret); + + /* ensure initialization is finished */ + usb_wait_anchor_empty_timeout(&bcd2k->anchor, 1000); +} + +static int bcd2000_init_midi(struct bcd2000 *bcd2k) +{ + int ret; + struct snd_rawmidi *rmidi; + + ret = snd_rawmidi_new(bcd2k->card, bcd2k->card->shortname, 0, + 1, /* output */ + 1, /* input */ + &rmidi); + + if (ret < 0) + return ret; + + strlcpy(rmidi->name, bcd2k->card->shortname, sizeof(rmidi->name)); + + rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; + rmidi->private_data = bcd2k; + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &bcd2000_midi_output); + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &bcd2000_midi_input); + + bcd2k->rmidi = rmidi; + + bcd2k->midi_in_urb = usb_alloc_urb(0, GFP_KERNEL); + bcd2k->midi_out_urb = usb_alloc_urb(0, GFP_KERNEL); + + if (!bcd2k->midi_in_urb || !bcd2k->midi_out_urb) { + dev_err(&bcd2k->dev->dev, PREFIX "usb_alloc_urb failed\n"); + return -ENOMEM; + } + + usb_fill_int_urb(bcd2k->midi_in_urb, bcd2k->dev, + usb_rcvintpipe(bcd2k->dev, 0x81), + bcd2k->midi_in_buf, BUFSIZE, + bcd2000_input_complete, bcd2k, 1); + + usb_fill_int_urb(bcd2k->midi_out_urb, bcd2k->dev, + usb_sndintpipe(bcd2k->dev, 0x1), + bcd2k->midi_out_buf, BUFSIZE, + bcd2000_output_complete, bcd2k, 1); + + bcd2000_init_device(bcd2k); + + return 0; +} + +static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k, + struct usb_interface *interface) +{ + /* usb_kill_urb not necessary, urb is aborted automatically */ + + usb_free_urb(bcd2k->midi_out_urb); + usb_free_urb(bcd2k->midi_in_urb); + + if (bcd2k->intf) { + usb_set_intfdata(bcd2k->intf, NULL); + bcd2k->intf = NULL; + } +} + +static int bcd2000_probe(struct usb_interface *interface, + const struct usb_device_id *usb_id) +{ + struct snd_card *card; + struct bcd2000 *bcd2k; + unsigned int card_index; + char usb_path[32]; + int err; + + mutex_lock(&devices_mutex); + + for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) + if (!test_bit(card_index, devices_used)) + break; + + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + + err = snd_card_new(&interface->dev, index[card_index], id[card_index], + THIS_MODULE, sizeof(*bcd2k), &card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return err; + } + + bcd2k = card->private_data; + bcd2k->dev = interface_to_usbdev(interface); + bcd2k->card = card; + bcd2k->card_index = card_index; + bcd2k->intf = interface; + + snd_card_set_dev(card, &interface->dev); + + strncpy(card->driver, "snd-bcd2000", sizeof(card->driver)); + strncpy(card->shortname, "BCD2000", sizeof(card->shortname)); + usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); + snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), + "Behringer BCD2000 at %s", + usb_path); + + err = bcd2000_init_midi(bcd2k); + if (err < 0) + goto probe_error; + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + usb_set_intfdata(interface, bcd2k); + set_bit(card_index, devices_used); + + mutex_unlock(&devices_mutex); + return 0; + +probe_error: + dev_info(&bcd2k->dev->dev, PREFIX "error during probing"); + bcd2000_free_usb_related_resources(bcd2k, interface); + snd_card_free(card); + mutex_unlock(&devices_mutex); + return err; +} + +static void bcd2000_disconnect(struct usb_interface *interface) +{ + struct bcd2000 *bcd2k = usb_get_intfdata(interface); + + if (!bcd2k) + return; + + mutex_lock(&devices_mutex); + + /* make sure that userspace cannot create new requests */ + snd_card_disconnect(bcd2k->card); + + bcd2000_free_usb_related_resources(bcd2k, interface); + + clear_bit(bcd2k->card_index, devices_used); + + snd_card_free_when_closed(bcd2k->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_driver bcd2000_driver = { + .name = "snd-bcd2000", + .probe = bcd2000_probe, + .disconnect = bcd2000_disconnect, + .id_table = id_table, +}; + +module_usb_driver(bcd2000_driver); + +MODULE_DEVICE_TABLE(usb, id_table); +MODULE_AUTHOR("Mario Kicherer, dev(a)kicherer.org"); +MODULE_DESCRIPTION("Behringer BCD2000 driver"); +MODULE_LICENSE("GPL"); -- 1.8.3.2

4 5

[alsa-devel] Tenor TE8802, USB : clics and crackles during music play
by Eric F 07 Apr '14

07 Apr '14

Hello, Been searching for days on Internet for this issue, without luck :( I own a Teac UD-H01 : when connected via USB => crackles and clicks

4 7

[alsa-devel] [PATCH] ASoC: da732x: Print correct major id
by Axel Lin 07 Apr '14

07 Apr '14

DA732X_ID_MAJOR_MASK is 0xF0, so the major id is (reg & DA732X_ID_MAJOR_MASK) >> 4. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/da732x.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 7d168ec..48f3fef 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1571,7 +1571,8 @@ static int da732x_i2c_probe(struct i2c_client *i2c, } dev_info(&i2c->dev, "Revision: %d.%d\n", - (reg & DA732X_ID_MAJOR_MASK), (reg & DA732X_ID_MINOR_MASK)); + (reg & DA732X_ID_MAJOR_MASK) >> 4, + (reg & DA732X_ID_MINOR_MASK)); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da732x, da732x_dai, ARRAY_SIZE(da732x_dai)); -- 1.8.3.2

3 2

[alsa-devel] [PATCH alsa-lib] pcm: rate: Don't return negative frame count on success in rewind
by Andrew Eikum 07 Apr '14

07 Apr '14

snd_pcm_rewind is documented to return <0 on failure and >=0 on success. Signed-off-by: Andrew Eikum <aeikum(a)codeweavers.com> --- I checked all other rewind implementations, and this was the only one that obviously had this bug. diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 54a3e67..2eb4b1b 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -702,7 +702,7 @@ static snd_pcm_sframes_t snd_pcm_rate_rewind(snd_pcm_t *pcm, snd_pcm_uframes_t f snd_atomic_write_begin(&rate->watom); n = snd_pcm_rate_move_applptr(pcm, -frames); snd_atomic_write_end(&rate->watom); - return n; + return n < 0 ? -n : n; } static snd_pcm_sframes_t snd_pcm_rate_forward(snd_pcm_t *pcm, snd_pcm_uframes_t frames) -- 1.9.1

3 2

[alsa-devel] [PATCH] Allow snd_soc_update_bits use 32 bits register
by bardliao＠realtek.com 07 Apr '14

07 Apr '14

From: Bard Liao <bardliao(a)realtek.com> Change reg's type from unsigned short to unsigned int. So that we can use 32 bits reg value in snd_soc_update_bits. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- include/sound/soc.h | 2 +- sound/soc/soc-core.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 0b83168..687dec0 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -469,7 +469,7 @@ static inline void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count, #endif /* codec register bit access */ -int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, +int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value); int snd_soc_update_bits_locked(struct snd_soc_codec *codec, unsigned short reg, unsigned int mask, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index caebd63..fde636c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2316,7 +2316,7 @@ EXPORT_SYMBOL_GPL(snd_soc_write); * * Returns 1 for change, 0 for no change, or negative error code. */ -int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg, +int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned int reg, unsigned int mask, unsigned int value) { bool change; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: cs42l52: Fix mask for REVID
by Axel Lin 07 Apr '14

07 Apr '14

BIT[0:2] of register 01h is REVID, so the mask for REVID should be 0x7. Also updates the code to use CS42L52_CHIP_REV_MASK. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l52.c | 2 +- sound/soc/codecs/cs42l52.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index f0ca6be..460d355 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1259,7 +1259,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, } dev_info(&i2c_client->dev, "Cirrus Logic CS42L52, Revision: %02X\n", - reg & 0xFF); + reg & CS42L52_CHIP_REV_MASK); /* Set Platform Data */ if (cs42l52->pdata.mica_diff_cfg) diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h index 6fb8f00..ac445993 100644 --- a/sound/soc/codecs/cs42l52.h +++ b/sound/soc/codecs/cs42l52.h @@ -37,7 +37,7 @@ #define CS42L52_CHIP_REV_A0 0x00 #define CS42L52_CHIP_REV_A1 0x01 #define CS42L52_CHIP_REV_B0 0x02 -#define CS42L52_CHIP_REV_MASK 0x03 +#define CS42L52_CHIP_REV_MASK 0x07 #define CS42L52_PWRCTL1 0x02 #define CS42L52_PWRCTL1_PDN_ALL 0x9F -- 1.8.3.2

3 2

[alsa-devel] [PATCH] ASoC: wm5110: Add in OSR controls for OUT5/6
by Charles Keepax 07 Apr '14

07 Apr '14

There are no OSR controls on outputs 1-4 on wm5110, however when these were removed the ones on output 5 and 6 were also accidentally removed, but those actually exist. This patch adds these controls back in. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 5 +++++ 1 files changed, 5 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index df5a38d..83a7e2f 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -367,6 +367,11 @@ SOC_SINGLE("HPOUT2 SC Protect Switch", ARIZONA_HP2_SHORT_CIRCUIT_CTRL, SOC_SINGLE("HPOUT3 SC Protect Switch", ARIZONA_HP3_SHORT_CIRCUIT_CTRL, ARIZONA_HP3_SC_ENA_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, + ARIZONA_OUT5_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L, + ARIZONA_OUT6_OSR_SHIFT, 1, 0), + SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, -- 1.7.2.5

2 1

[alsa-devel] Invalid PCM pointer with CMI8738 and pcm.c buffer underuns
by andoru 06 Apr '14

06 Apr '14

Continued from here: http://article.gmane.org/gmane.linux.alsa.devel/121379 After a few days now I finally got VLC to log everything that happens, and finally the bug reappeared, and unlike previously thought, it happens when I watch a video as well. But unfortunately VLC didn't give anything specific that happened at the time, just some vague ALSA errors about broken pipes (at line 394) : http://pastebin.com/NgZiG9yX At the time I added the video to VLC's queue I was doing something in the background (which I don't remember now unfortunately), but this has happened even when listening to something without doing anything in the background, and not having any background-running apps except Transmission. Also interestingly lately I've heard a few crackles/pops when listening to stuff, followed by a second or so of silence then back to normal when I would run CPU intensive stuff. At those points I thought the bug would be invoked again, but it didn't happen so. I still have the crackling problem, not able to change the PCM channel volume and screeching on audio that has clipped portions or very high frequency waves. Can anyone give me any instructions on what I could do to further diagnose this problem? And more importantly, as I stated in the linked post above, how could I shut down the sound daemon/server/whatever so I could bring back the OS to stability without restarting.

2 4

[alsa-devel] [PATCH] ASoC: cs42l52: Report correct chip id and revision
by Axel Lin 05 Apr '14

05 Apr '14

According to the datasheet: Chip I.D and Revision Register (Address 01h) BIT[0:2] REVID BIT[3:7] CHIPID (CS42L52: 11100) REVID takes 3 bits, so CS42L52_CHIP_REV_MASK should be 0x07. While at it, also adds define for CS42L52_CHIP_REV_B1. The CHIPID takes BIT[3:7], so this patch updates the defines and the code to show correct chip id for this chip. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42l52.c | 4 ++-- sound/soc/codecs/cs42l52.h | 7 ++++--- 2 files changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index f0ca6be..277df17 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1249,7 +1249,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, ret); ret = regmap_read(cs42l52->regmap, CS42L52_CHIP, &reg); - devid = reg & CS42L52_CHIP_ID_MASK; + devid = (reg & CS42L52_CHIP_ID_MASK) >> 3; if (devid != CS42L52_CHIP_ID) { ret = -ENODEV; dev_err(&i2c_client->dev, @@ -1259,7 +1259,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, } dev_info(&i2c_client->dev, "Cirrus Logic CS42L52, Revision: %02X\n", - reg & 0xFF); + reg & CS42L52_CHIP_REV_MASK); /* Set Platform Data */ if (cs42l52->pdata.mica_diff_cfg) diff --git a/sound/soc/codecs/cs42l52.h b/sound/soc/codecs/cs42l52.h index 6fb8f00..ce7ce5a 100644 --- a/sound/soc/codecs/cs42l52.h +++ b/sound/soc/codecs/cs42l52.h @@ -32,12 +32,13 @@ #define CS42L52_FIX_BITS_CTL 0x00 #define CS42L52_CHIP 0x01 -#define CS42L52_CHIP_ID 0xE0 -#define CS42L52_CHIP_ID_MASK 0xF8 +#define CS42L52_CHIP_ID 0x1C +#define CS42L52_CHIP_ID_MASK (0x1F << 3) #define CS42L52_CHIP_REV_A0 0x00 #define CS42L52_CHIP_REV_A1 0x01 #define CS42L52_CHIP_REV_B0 0x02 -#define CS42L52_CHIP_REV_MASK 0x03 +#define CS42L52_CHIP_REV_B1 0x03 +#define CS42L52_CHIP_REV_MASK 0x07 #define CS42L52_PWRCTL1 0x02 #define CS42L52_PWRCTL1_PDN_ALL 0x9F -- 1.8.3.2

3 9

[alsa-devel] [PATCH] sound/oss: Remove uncompilable DBG macro use
by Joe Perches 04 Apr '14

04 Apr '14

Most of it duplicates function tracing and one of them has an uncompilable printf %P use. Others have format/argument mismatches. Remove unused DBG1 macro definition Neaten uart401.c use of ok test around this DBG macro removal. Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/oss/ad1848.c | 4 ---- sound/oss/opl3.c | 3 --- sound/oss/pas2_mixer.c | 9 --------- sound/oss/pas2_pcm.c | 18 ------------------ sound/oss/sb_common.c | 4 ---- sound/oss/sb_ess.c | 4 ---- sound/oss/sequencer.c | 6 ------ sound/oss/sound_config.h | 4 ---- sound/oss/soundcard.c | 6 ------ sound/oss/uart401.c | 11 ++--------- 10 files changed, 2 insertions(+), 67 deletions(-) diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index 4918b71..ec1ee07 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -50,8 +50,6 @@ #include <linux/pnp.h> #include <linux/spinlock.h> -#define DEB(x) -#define DEB1(x) #include "sound_config.h" #include "ad1848.h" @@ -1016,8 +1014,6 @@ static void ad1848_close(int dev) ad1848_info *devc = (ad1848_info *) audio_devs[dev]->devc; ad1848_port_info *portc = (ad1848_port_info *) audio_devs[dev]->portc; - DEB(printk("ad1848_close(void)\n")); - devc->intr_active = 0; ad1848_halt(dev); diff --git a/sound/oss/opl3.c b/sound/oss/opl3.c index c5c2440..4709e59 100644 --- a/sound/oss/opl3.c +++ b/sound/oss/opl3.c @@ -275,7 +275,6 @@ static int opl3_kill_note (int devno, int voice, int note, int velocity) devc->v_alloc->map[voice] = 0; map = &pv_map[devc->lv_map[voice]]; - DEB(printk("Kill note %d\n", voice)); if (map->voice_mode == 0) return 0; @@ -873,8 +872,6 @@ static void opl3_aftertouch(int dev, int voice, int pressure) map = &pv_map[devc->lv_map[voice]]; - DEB(printk("Aftertouch %d\n", voice)); - if (map->voice_mode == 0) return; diff --git a/sound/oss/pas2_mixer.c b/sound/oss/pas2_mixer.c index a0bcb85..50b5bd5 100644 --- a/sound/oss/pas2_mixer.c +++ b/sound/oss/pas2_mixer.c @@ -21,10 +21,6 @@ #include "pas2.h" -#ifndef DEB -#define DEB(what) /* (what) */ -#endif - extern int pas_translate_code; extern char pas_model; extern int *pas_osp; @@ -120,8 +116,6 @@ pas_mixer_set(int whichDev, unsigned int level) { int left, right, devmask, changed, i, mixer = 0; - DEB(printk("static int pas_mixer_set(int whichDev = %d, unsigned int level = %X)\n", whichDev, level)); - left = level & 0x7f; right = (level & 0x7f00) >> 8; @@ -207,8 +201,6 @@ pas_mixer_reset(void) { int foo; - DEB(printk("pas2_mixer.c: void pas_mixer_reset(void)\n")); - for (foo = 0; foo < SOUND_MIXER_NRDEVICES; foo++) pas_mixer_set(foo, levels[foo]); @@ -220,7 +212,6 @@ static int pas_mixer_ioctl(int dev, unsigned int cmd, void __user *arg) int level,v ; int __user *p = (int __user *)arg; - DEB(printk("pas2_mixer.c: int pas_mixer_ioctl(unsigned int cmd = %X, unsigned int arg = %X)\n", cmd, arg)); if (cmd == SOUND_MIXER_PRIVATE1) { /* Set loudness bit */ if (get_user(level, p)) return -EFAULT; diff --git a/sound/oss/pas2_pcm.c b/sound/oss/pas2_pcm.c index 6f13ab4..474803b 100644 --- a/sound/oss/pas2_pcm.c +++ b/sound/oss/pas2_pcm.c @@ -22,10 +22,6 @@ #include "pas2.h" -#ifndef DEB -#define DEB(WHAT) -#endif - #define PAS_PCM_INTRBITS (0x08) /* * Sample buffer timer interrupt enable @@ -156,8 +152,6 @@ static int pas_audio_ioctl(int dev, unsigned int cmd, void __user *arg) int val, ret; int __user *p = arg; - DEB(printk("pas2_pcm.c: static int pas_audio_ioctl(unsigned int cmd = %X, unsigned int arg = %X)\n", cmd, arg)); - switch (cmd) { case SOUND_PCM_WRITE_RATE: @@ -204,8 +198,6 @@ static int pas_audio_ioctl(int dev, unsigned int cmd, void __user *arg) static void pas_audio_reset(int dev) { - DEB(printk("pas2_pcm.c: static void pas_audio_reset(void)\n")); - pas_write(pas_read(0xF8A) & ~0x40, 0xF8A); /* Disable PCM */ } @@ -214,8 +206,6 @@ static int pas_audio_open(int dev, int mode) int err; unsigned long flags; - DEB(printk("pas2_pcm.c: static int pas_audio_open(int mode = %X)\n", mode)); - spin_lock_irqsave(&pas_lock, flags); if (pcm_busy) { @@ -239,8 +229,6 @@ static void pas_audio_close(int dev) { unsigned long flags; - DEB(printk("pas2_pcm.c: static void pas_audio_close(void)\n")); - spin_lock_irqsave(&pas_lock, flags); pas_audio_reset(dev); @@ -256,8 +244,6 @@ static void pas_audio_output_block(int dev, unsigned long buf, int count, { unsigned long flags, cnt; - DEB(printk("pas2_pcm.c: static void pas_audio_output_block(char *buf = %P, int count = %X)\n", buf, count)); - cnt = count; if (audio_devs[dev]->dmap_out->dma > 3) cnt >>= 1; @@ -303,8 +289,6 @@ static void pas_audio_start_input(int dev, unsigned long buf, int count, unsigned long flags; int cnt; - DEB(printk("pas2_pcm.c: static void pas_audio_start_input(char *buf = %P, int count = %X)\n", buf, count)); - cnt = count; if (audio_devs[dev]->dmap_out->dma > 3) cnt >>= 1; @@ -388,8 +372,6 @@ static struct audio_driver pas_audio_driver = void __init pas_pcm_init(struct address_info *hw_config) { - DEB(printk("pas2_pcm.c: long pas_pcm_init()\n")); - pcm_bitsok = 8; if (pas_read(0xEF8B) & 0x08) pcm_bitsok |= 16; diff --git a/sound/oss/sb_common.c b/sound/oss/sb_common.c index 851a1da..3d50fb4 100644 --- a/sound/oss/sb_common.c +++ b/sound/oss/sb_common.c @@ -226,8 +226,6 @@ int sb_dsp_reset(sb_devc * devc) { int loopc; - DEB(printk("Entered sb_dsp_reset()\n")); - if (devc->model == MDL_ESS) return ess_dsp_reset (devc); /* This is only for non-ESS chips */ @@ -246,8 +244,6 @@ int sb_dsp_reset(sb_devc * devc) return 0; /* Sorry */ } - DEB(printk("sb_dsp_reset() OK\n")); - return 1; } diff --git a/sound/oss/sb_ess.c b/sound/oss/sb_ess.c index 0e7254b..b47a690 100644 --- a/sound/oss/sb_ess.c +++ b/sound/oss/sb_ess.c @@ -865,8 +865,6 @@ printk(KERN_INFO "FKS: ess_dsp_reset 1\n"); ess_show_mixerregs (devc); #endif - DEB(printk("Entered ess_dsp_reset()\n")); - outb(3, DSP_RESET); /* Reset FIFO too */ udelay(10); @@ -881,8 +879,6 @@ ess_show_mixerregs (devc); } ess_extended (devc); - DEB(printk("sb_dsp_reset() OK\n")); - #ifdef FKS_LOGGING printk(KERN_INFO "FKS: dsp_reset 2\n"); ess_show_mixerregs (devc); diff --git a/sound/oss/sequencer.c b/sound/oss/sequencer.c index 9b9f7d3..c0eea1d 100644 --- a/sound/oss/sequencer.c +++ b/sound/oss/sequencer.c @@ -216,8 +216,6 @@ int sequencer_write(int dev, struct file *file, const char __user *buf, int coun dev = dev >> 4; - DEB(printk("sequencer_write(dev=%d, count=%d)\n", dev, count)); - if (mode == OPEN_READ) return -EIO; @@ -959,8 +957,6 @@ int sequencer_open(int dev, struct file *file) dev = dev >> 4; mode = translate_mode(file); - DEB(printk("sequencer_open(dev=%d)\n", dev)); - if (!sequencer_ok) { /* printk("Sound card: sequencer not initialized\n");*/ @@ -1133,8 +1129,6 @@ void sequencer_release(int dev, struct file *file) dev = dev >> 4; - DEB(printk("sequencer_release(dev=%d)\n", dev)); - /* * Wait until the queue is empty (if we don't have nonblock) */ diff --git a/sound/oss/sound_config.h b/sound/oss/sound_config.h index 9d35c4c..f2554ab 100644 --- a/sound/oss/sound_config.h +++ b/sound/oss/sound_config.h @@ -123,10 +123,6 @@ static inline int translate_mode(struct file *file) #include "sound_calls.h" #include "dev_table.h" -#ifndef DEB -#define DEB(x) -#endif - #ifndef DDB #define DDB(x) do {} while (0) #endif diff --git a/sound/oss/soundcard.c b/sound/oss/soundcard.c index e778034..b70c7c8 100644 --- a/sound/oss/soundcard.c +++ b/sound/oss/soundcard.c @@ -154,7 +154,6 @@ static ssize_t sound_read(struct file *file, char __user *buf, size_t count, lof mutex_lock(&soundcard_mutex); - DEB(printk("sound_read(dev=%d, count=%d)\n", dev, count)); switch (dev & 0x0f) { case SND_DEV_DSP: case SND_DEV_DSP16: @@ -180,7 +179,6 @@ static ssize_t sound_write(struct file *file, const char __user *buf, size_t cou int ret = -EINVAL; mutex_lock(&soundcard_mutex); - DEB(printk("sound_write(dev=%d, count=%d)\n", dev, count)); switch (dev & 0x0f) { case SND_DEV_SEQ: case SND_DEV_SEQ2: @@ -206,7 +204,6 @@ static int sound_open(struct inode *inode, struct file *file) int dev = iminor(inode); int retval; - DEB(printk("sound_open(dev=%d)\n", dev)); if ((dev >= SND_NDEVS) || (dev < 0)) { printk(KERN_ERR "Invalid minor device %d\n", dev); return -ENXIO; @@ -257,7 +254,6 @@ static int sound_release(struct inode *inode, struct file *file) int dev = iminor(inode); mutex_lock(&soundcard_mutex); - DEB(printk("sound_release(dev=%d)\n", dev)); switch (dev & 0x0f) { case SND_DEV_CTL: module_put(mixer_devs[dev >> 4]->owner); @@ -351,7 +347,6 @@ static long sound_ioctl(struct file *file, unsigned int cmd, unsigned long arg) if (!access_ok(VERIFY_WRITE, p, len)) return -EFAULT; } - DEB(printk("sound_ioctl(dev=%d, cmd=0x%x, arg=0x%x)\n", dev, cmd, arg)); if (cmd == OSS_GETVERSION) return __put_user(SOUND_VERSION, (int __user *)p); @@ -409,7 +404,6 @@ static unsigned int sound_poll(struct file *file, poll_table * wait) struct inode *inode = file_inode(file); int dev = iminor(inode); - DEB(printk("sound_poll(dev=%d)\n", dev)); switch (dev & 0x0f) { case SND_DEV_SEQ: case SND_DEV_SEQ2: diff --git a/sound/oss/uart401.c b/sound/oss/uart401.c index 5433c6f..62b8869 100644 --- a/sound/oss/uart401.c +++ b/sound/oss/uart401.c @@ -274,19 +274,12 @@ static int reset_uart401(uart401_devc * devc) } } - + /* Flush input before enabling interrupts */ if (ok) - { - DEB(printk("Reset UART401 OK\n")); - } + uart401_input_loop(devc); else DDB(printk("Reset UART401 failed - No hardware detected.\n")); - if (ok) - uart401_input_loop(devc); /* - * Flush input before enabling interrupts - */ - return ok; }

2 1

[alsa-devel] [PATCH 1/2] snd-ice1712: restore AK4xxx volumes on resume
by Ondrej Zary 04 Apr '14

04 Apr '14

Also restore AK4xxx mixer volumes on resume for M-Audio ICE1712-based cards. This fixes incorrect (sound working) zero mixer volumes after resume. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/ice1712/delta.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/pci/ice1712/delta.c b/sound/pci/ice1712/delta.c index 3764a27..bb34608 100644 --- a/sound/pci/ice1712/delta.c +++ b/sound/pci/ice1712/delta.c @@ -578,7 +578,8 @@ static struct snd_ak4xxx_private akm_vx442_priv = { #ifdef CONFIG_PM_SLEEP static int snd_ice1712_delta_resume(struct snd_ice1712 *ice) { - unsigned char akm_backup[AK4XXX_IMAGE_SIZE]; + unsigned char akm_img_bak[AK4XXX_IMAGE_SIZE]; + unsigned char akm_vol_bak[AK4XXX_IMAGE_SIZE]; /* init spdif */ switch (ice->eeprom.subvendor) { @@ -603,9 +604,11 @@ static int snd_ice1712_delta_resume(struct snd_ice1712 *ice) /* init codec and restore registers */ if (ice->akm_codecs) { - memcpy(akm_backup, ice->akm->images, sizeof(akm_backup)); + memcpy(akm_img_bak, ice->akm->images, sizeof(akm_img_bak)); + memcpy(akm_vol_bak, ice->akm->volumes, sizeof(akm_vol_bak)); snd_akm4xxx_init(ice->akm); - memcpy(ice->akm->images, akm_backup, sizeof(akm_backup)); + memcpy(ice->akm->images, akm_img_bak, sizeof(akm_img_bak)); + memcpy(ice->akm->volumes, akm_vol_bak, sizeof(akm_vol_bak)); snd_akm4xxx_reset(ice->akm, 0); } -- Ondrej Zary

2 2

[alsa-devel] [PATCH v2 00/16] ASoC: fsl-ssi: Driver cleanup
by Markus Pargmann 04 Apr '14

04 Apr '14

Hi, This new version of the cleanup is mainly a reordering of the patches. It moves the regmap patch to the end of the series so none of the other patches depend on it. This also fixes a bug I introduced in "Remove fsl_ssi_setup" which also removed the switching between network and normal mode depending on the number of channels. I reversed this code removal. Another new patch was added "Fix i2s_mode variable setup" which fixes the i2s_mode assignment in set_dai_fmt(). Best regards, Markus Changes in v2: - Move "Use regmap" patch to the end of the series - Add normal/network mode switching code to again (previously removed in "Remove fsl_ssi_setup" - Add patch "Fix i2s_mode variable setup" Markus Pargmann (16): ASoC: fsl-ssi: Remove fsl_ssi_setup ASoC: fsl-ssi: Fix i2s_mode variable setup ASoC: fsl-ssi: Move debugging to seperate file ASoC: fsl-ssi: Use dev_name for DAI driver struct ASoC: fsl-ssi: Move imx-specific probe to seperate function ASoC: fsl-ssi: Remove useless DMA code ASoC: fsl-ssi: baud clock error handling ASoC: fsl-ssi: Cleanup probe function ASoC: fsl-ssi: Remove unnecessary variables from ssi_private ASoC: fsl-ssi: reorder and document fsl_ssi_private ASoC: fsl-ssi: Fix register values when disabling ASoC: fsl-ssi: Only enable baudclk when used ASoC: fsl-ssi: Set default dai-fmts ASoC: fsl-ssi: Transmit enable synchronization ASoC: fsl-ssi: Update binding documentation ASoC: fsl-ssi: Use regmap .../devicetree/bindings/sound/fsl,ssi.txt | 21 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_ssi.c | 1173 +++++++++----------- sound/soc/fsl/fsl_ssi.h | 112 +- sound/soc/fsl/fsl_ssi_dbg.c | 163 +++ 6 files changed, 759 insertions(+), 713 deletions(-) create mode 100644 sound/soc/fsl/fsl_ssi_dbg.c -- 1.9.0

6 33

[alsa-devel] [PATCH] ASoC: alc56(23|32): fix undefined return value of probing code
by Christoph Jaeger 04 Apr '14

04 Apr '14

Commit 5d6be5aa ("ASoC: codec: Simplify ASoC probe code.") left variable 'ret', whose value is returned, uninitialized. Since it is not used otherwise, remove it. Signed-off-by: Christoph Jaeger <christophjaeger(a)linux.com> --- sound/soc/codecs/alc5623.c | 3 +-- sound/soc/codecs/alc5632.c | 3 +-- 2 files changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 09f7e77..f500905 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -902,7 +902,6 @@ static int alc5623_probe(struct snd_soc_codec *codec) { struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret; alc5623_reset(codec); @@ -961,7 +960,7 @@ static int alc5623_probe(struct snd_soc_codec *codec) return -EINVAL; } - return ret; + return 0; } /* power down chip */ diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index ec071a6..3ee8d4e 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1061,7 +1061,6 @@ static int alc5632_resume(struct snd_soc_codec *codec) static int alc5632_probe(struct snd_soc_codec *codec) { struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); - int ret; /* power on device */ alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1075,7 +1074,7 @@ static int alc5632_probe(struct snd_soc_codec *codec) return -EINVAL; } - return ret; + return 0; } /* power down chip */ -- 1.9.0

2 1

[alsa-devel] [PATCH] ASoC: rsnd: SSI needs to prepare clocks before enabling
by Ben Dooks 04 Apr '14

04 Apr '14

From: William Towle <william.towle(a)codethink.co.uk> The rsnd_ssi_hw_{stop|start} functions need to call clk_prepare_enable() and clk_disable_unprepare(), otherwise the following warning occurs on attempt to play sound: WARNING: CPU: 2 PID: 1176 at drivers/clk/clk.c:926 __clk_enable+0x30/0x9c() due to drivers/clk/clk.c: 926 if (WARN_ON(clk->prepare_count == 0)) 927 return -ESHUTDOWN; Signed-off-by: William Towle <william.towle(a)codethink.co.uk> [ben.dooks(a)codethink.co.uk: add more bug info to report] Tested-by: Ben Dooks <ben.dooks(a)codethink.co.uk> --- sound/soc/sh/rcar/ssi.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 4b8cf7c..1c07ef7 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -250,7 +250,7 @@ static void rsnd_ssi_hw_start(struct rsnd_ssi *ssi, u32 cr; if (0 == ssi->usrcnt) { - clk_enable(ssi->clk); + clk_prepare_enable(ssi->clk); if (rsnd_rdai_is_clk_master(rdai)) { if (rsnd_ssi_clk_from_parent(ssi)) @@ -309,7 +309,7 @@ static void rsnd_ssi_hw_stop(struct rsnd_ssi *ssi, rsnd_ssi_master_clk_stop(ssi); } - clk_disable(ssi->clk); + clk_disable_unprepare(ssi->clk); } dev_dbg(dev, "ssi%d hw stopped\n", rsnd_mod_id(&ssi->mod)); -- 1.9.0

3 2

[alsa-devel] [PATCH] ASoC: sirf: Move the tx rx enable from port to codec, that will not need register sharing
by RongJun Ying 04 Apr '14

04 Apr '14

From: Rongjun Ying <rongjun.ying(a)csr.com> The port driver only used to register component and dmaengine pcm. Signed-off-by: Rongjun Ying <rongjun.ying(a)csr.com> --- sound/soc/codecs/sirf-audio-codec.c | 74 +++++++++++++++++++++--- sound/soc/codecs/sirf-audio-codec.h | 50 ++++++++++++++++ sound/soc/sirf/sirf-audio-port.c | 107 ----------------------------------- sound/soc/sirf/sirf-audio-port.h | 62 -------------------- 4 files changed, 116 insertions(+), 177 deletions(-) delete mode 100644 sound/soc/sirf/sirf-audio-port.h diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index 58e7c1f..c5177bc 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -279,13 +279,63 @@ static const struct snd_soc_dapm_route sirf_audio_codec_map[] = { {"Mic input mode mux", "Differential", "MICIN1"}, }; +static void sirf_audio_codec_tx_enable(struct sirf_audio_codec *sirf_audio_codec) +{ + regmap_update_bits(sirf_audio_codec->regmap, AUDIO_PORT_IC_TXFIFO_OP, + AUDIO_FIFO_RESET, AUDIO_FIFO_RESET); + regmap_update_bits(sirf_audio_codec->regmap, AUDIO_PORT_IC_TXFIFO_OP, + AUDIO_FIFO_RESET, ~AUDIO_FIFO_RESET); + regmap_write(sirf_audio_codec->regmap, AUDIO_PORT_IC_TXFIFO_INT_MSK, 0); + regmap_write(sirf_audio_codec->regmap, AUDIO_PORT_IC_TXFIFO_OP, 0); + regmap_update_bits(sirf_audio_codec->regmap, AUDIO_PORT_IC_TXFIFO_OP, + AUDIO_FIFO_START, AUDIO_FIFO_START); + regmap_update_bits(sirf_audio_codec->regmap, + AUDIO_PORT_IC_CODEC_TX_CTRL, IC_TX_ENABLE, IC_TX_ENABLE); +} + +static void sirf_audio_codec_tx_disable(struct sirf_audio_codec *sirf_audio_codec) +{ + regmap_write(sirf_audio_codec->regmap, AUDIO_PORT_IC_TXFIFO_OP, 0); + regmap_update_bits(sirf_audio_codec->regmap, + AUDIO_PORT_IC_CODEC_TX_CTRL, IC_TX_ENABLE, ~IC_TX_ENABLE); +} + +static void sirf_audio_codec_rx_enable(struct sirf_audio_codec *sirf_audio_codec, + int channels) +{ + regmap_update_bits(sirf_audio_codec->regmap, AUDIO_PORT_IC_RXFIFO_OP, + AUDIO_FIFO_RESET, AUDIO_FIFO_RESET); + regmap_update_bits(sirf_audio_codec->regmap, AUDIO_PORT_IC_RXFIFO_OP, + AUDIO_FIFO_RESET, ~AUDIO_FIFO_RESET); + regmap_write(sirf_audio_codec->regmap, + AUDIO_PORT_IC_RXFIFO_INT_MSK, 0); + regmap_write(sirf_audio_codec->regmap, AUDIO_PORT_IC_RXFIFO_OP, 0); + regmap_update_bits(sirf_audio_codec->regmap, AUDIO_PORT_IC_RXFIFO_OP, + AUDIO_FIFO_START, AUDIO_FIFO_START); + if (channels == 1) + regmap_update_bits(sirf_audio_codec->regmap, + AUDIO_PORT_IC_CODEC_RX_CTRL, + IC_RX_ENABLE_MONO, IC_RX_ENABLE_MONO); + else + regmap_update_bits(sirf_audio_codec->regmap, + AUDIO_PORT_IC_CODEC_RX_CTRL, + IC_RX_ENABLE_STEREO, IC_RX_ENABLE_STEREO); +} + +static void sirf_audio_codec_rx_disable(struct sirf_audio_codec *sirf_audio_codec) +{ + regmap_update_bits(sirf_audio_codec->regmap, + AUDIO_PORT_IC_CODEC_RX_CTRL, + IC_RX_ENABLE_STEREO, ~IC_RX_ENABLE_STEREO); +} + static int sirf_audio_codec_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; struct snd_soc_codec *codec = dai->codec; - u32 val = 0; + struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec); + int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; /* * This is a workaround, When stop playback, @@ -295,20 +345,28 @@ static int sirf_audio_codec_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (playback) { + snd_soc_update_bits(codec, AUDIO_IC_CODEC_CTRL0, + IC_HSLEN | IC_HSREN, 0); + sirf_audio_codec_tx_disable(sirf_audio_codec); + } else + sirf_audio_codec_rx_disable(sirf_audio_codec); break; case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (playback) - val = IC_HSLEN | IC_HSREN; + if (playback) { + sirf_audio_codec_tx_enable(sirf_audio_codec); + snd_soc_update_bits(codec, AUDIO_IC_CODEC_CTRL0, + IC_HSLEN | IC_HSREN, IC_HSLEN | IC_HSREN); + } else + sirf_audio_codec_rx_enable(sirf_audio_codec, + substream->runtime->channels); break; default: return -EINVAL; } - if (playback) - snd_soc_update_bits(codec, AUDIO_IC_CODEC_CTRL0, - IC_HSLEN | IC_HSREN, val); return 0; } @@ -392,7 +450,7 @@ static const struct regmap_config sirf_audio_codec_regmap_config = { .reg_bits = 32, .reg_stride = 4, .val_bits = 32, - .max_register = AUDIO_IC_CODEC_CTRL3, + .max_register = AUDIO_PORT_IC_RXFIFO_INT_MSK, .cache_type = REGCACHE_NONE, }; diff --git a/sound/soc/codecs/sirf-audio-codec.h b/sound/soc/codecs/sirf-audio-codec.h index d4c187b..ba1adc0 100644 --- a/sound/soc/codecs/sirf-audio-codec.h +++ b/sound/soc/codecs/sirf-audio-codec.h @@ -72,4 +72,54 @@ #define IC_RXPGAR 0x7B #define IC_RXPGAL 0x7B +#define AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK 0x3F +#define AUDIO_PORT_TX_FIFO_SC_OFFSET 0 +#define AUDIO_PORT_TX_FIFO_LC_OFFSET 10 +#define AUDIO_PORT_TX_FIFO_HC_OFFSET 20 + +#define TX_FIFO_SC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_PORT_TX_FIFO_SC_OFFSET) +#define TX_FIFO_LC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_PORT_TX_FIFO_LC_OFFSET) +#define TX_FIFO_HC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_PORT_TX_FIFO_HC_OFFSET) + +#define AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK 0x0F +#define AUDIO_PORT_RX_FIFO_SC_OFFSET 0 +#define AUDIO_PORT_RX_FIFO_LC_OFFSET 10 +#define AUDIO_PORT_RX_FIFO_HC_OFFSET 20 + +#define RX_FIFO_SC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_PORT_RX_FIFO_SC_OFFSET) +#define RX_FIFO_LC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_PORT_RX_FIFO_LC_OFFSET) +#define RX_FIFO_HC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ + << AUDIO_PORT_RX_FIFO_HC_OFFSET) +#define AUDIO_PORT_IC_CODEC_TX_CTRL (0x00F4) +#define AUDIO_PORT_IC_CODEC_RX_CTRL (0x00F8) + +#define AUDIO_PORT_IC_TXFIFO_OP (0x00FC) +#define AUDIO_PORT_IC_TXFIFO_LEV_CHK (0x0100) +#define AUDIO_PORT_IC_TXFIFO_STS (0x0104) +#define AUDIO_PORT_IC_TXFIFO_INT (0x0108) +#define AUDIO_PORT_IC_TXFIFO_INT_MSK (0x010C) + +#define AUDIO_PORT_IC_RXFIFO_OP (0x0110) +#define AUDIO_PORT_IC_RXFIFO_LEV_CHK (0x0114) +#define AUDIO_PORT_IC_RXFIFO_STS (0x0118) +#define AUDIO_PORT_IC_RXFIFO_INT (0x011C) +#define AUDIO_PORT_IC_RXFIFO_INT_MSK (0x0120) + +#define AUDIO_FIFO_START (1 << 0) +#define AUDIO_FIFO_RESET (1 << 1) + +#define AUDIO_FIFO_FULL (1 << 0) +#define AUDIO_FIFO_EMPTY (1 << 1) +#define AUDIO_FIFO_OFLOW (1 << 2) +#define AUDIO_FIFO_UFLOW (1 << 3) + +#define IC_TX_ENABLE (0x03) +#define IC_RX_ENABLE_MONO (0x01) +#define IC_RX_ENABLE_STEREO (0x03) + #endif /*__SIRF_AUDIO_CODEC_H*/ diff --git a/sound/soc/sirf/sirf-audio-port.c b/sound/soc/sirf/sirf-audio-port.c index b04a53f..b4afa31 100644 --- a/sound/soc/sirf/sirf-audio-port.c +++ b/sound/soc/sirf/sirf-audio-port.c @@ -6,60 +6,15 @@ * Licensed under GPLv2 or later. */ #include <linux/module.h> -#include <linux/io.h> -#include <linux/regmap.h> #include <sound/soc.h> #include <sound/dmaengine_pcm.h> -#include "sirf-audio-port.h" - struct sirf_audio_port { struct regmap *regmap; struct snd_dmaengine_dai_dma_data playback_dma_data; struct snd_dmaengine_dai_dma_data capture_dma_data; }; -static void sirf_audio_port_tx_enable(struct sirf_audio_port *port) -{ - regmap_update_bits(port->regmap, AUDIO_PORT_IC_TXFIFO_OP, - AUDIO_FIFO_RESET, AUDIO_FIFO_RESET); - regmap_write(port->regmap, AUDIO_PORT_IC_TXFIFO_INT_MSK, 0); - regmap_write(port->regmap, AUDIO_PORT_IC_TXFIFO_OP, 0); - regmap_update_bits(port->regmap, AUDIO_PORT_IC_TXFIFO_OP, - AUDIO_FIFO_START, AUDIO_FIFO_START); - regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_TX_CTRL, - IC_TX_ENABLE, IC_TX_ENABLE); -} - -static void sirf_audio_port_tx_disable(struct sirf_audio_port *port) -{ - regmap_write(port->regmap, AUDIO_PORT_IC_TXFIFO_OP, 0); - regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_TX_CTRL, - IC_TX_ENABLE, ~IC_TX_ENABLE); -} - -static void sirf_audio_port_rx_enable(struct sirf_audio_port *port, - int channels) -{ - regmap_update_bits(port->regmap, AUDIO_PORT_IC_RXFIFO_OP, - AUDIO_FIFO_RESET, AUDIO_FIFO_RESET); - regmap_write(port->regmap, AUDIO_PORT_IC_RXFIFO_INT_MSK, 0); - regmap_write(port->regmap, AUDIO_PORT_IC_RXFIFO_OP, 0); - regmap_update_bits(port->regmap, AUDIO_PORT_IC_RXFIFO_OP, - AUDIO_FIFO_START, AUDIO_FIFO_START); - if (channels == 1) - regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_RX_CTRL, - IC_RX_ENABLE_MONO, IC_RX_ENABLE_MONO); - else - regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_RX_CTRL, - IC_RX_ENABLE_STEREO, IC_RX_ENABLE_STEREO); -} - -static void sirf_audio_port_rx_disable(struct sirf_audio_port *port) -{ - regmap_update_bits(port->regmap, AUDIO_PORT_IC_CODEC_RX_CTRL, - IC_RX_ENABLE_STEREO, ~IC_RX_ENABLE_STEREO); -} static int sirf_audio_port_dai_probe(struct snd_soc_dai *dai) { @@ -69,41 +24,6 @@ static int sirf_audio_port_dai_probe(struct snd_soc_dai *dai) return 0; } -static int sirf_audio_port_trigger(struct snd_pcm_substream *substream, int cmd, - struct snd_soc_dai *dai) -{ - struct sirf_audio_port *port = snd_soc_dai_get_drvdata(dai); - int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; - - switch (cmd) { - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (playback) - sirf_audio_port_tx_disable(port); - else - sirf_audio_port_rx_disable(port); - break; - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (playback) - sirf_audio_port_tx_enable(port); - else - sirf_audio_port_rx_enable(port, - substream->runtime->channels); - break; - default: - return -EINVAL; - } - - return 0; -} - -static const struct snd_soc_dai_ops sirf_audio_port_dai_ops = { - .trigger = sirf_audio_port_trigger, -}; - static struct snd_soc_dai_driver sirf_audio_port_dai = { .probe = sirf_audio_port_dai_probe, .name = "sirf-audio-port", @@ -120,49 +40,22 @@ static struct snd_soc_dai_driver sirf_audio_port_dai = { .rates = SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .ops = &sirf_audio_port_dai_ops, }; static const struct snd_soc_component_driver sirf_audio_port_component = { .name = "sirf-audio-port", }; -static const struct regmap_config sirf_audio_port_regmap_config = { - .reg_bits = 32, - .reg_stride = 4, - .val_bits = 32, - .max_register = AUDIO_PORT_IC_RXFIFO_INT_MSK, - .cache_type = REGCACHE_NONE, -}; - static int sirf_audio_port_probe(struct platform_device *pdev) { int ret; struct sirf_audio_port *port; - void __iomem *base; - struct resource *mem_res; port = devm_kzalloc(&pdev->dev, sizeof(struct sirf_audio_port), GFP_KERNEL); if (!port) return -ENOMEM; - mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem_res) { - dev_err(&pdev->dev, "no mem resource?\n"); - return -ENODEV; - } - - base = devm_ioremap(&pdev->dev, mem_res->start, - resource_size(mem_res)); - if (base == NULL) - return -ENOMEM; - - port->regmap = devm_regmap_init_mmio(&pdev->dev, base, - &sirf_audio_port_regmap_config); - if (IS_ERR(port->regmap)) - return PTR_ERR(port->regmap); - ret = devm_snd_soc_register_component(&pdev->dev, &sirf_audio_port_component, &sirf_audio_port_dai, 1); if (ret) diff --git a/sound/soc/sirf/sirf-audio-port.h b/sound/soc/sirf/sirf-audio-port.h deleted file mode 100644 index f32dc54..0000000 --- a/sound/soc/sirf/sirf-audio-port.h +++ /dev/null @@ -1,62 +0,0 @@ -/* - * SiRF Audio port controllers define - * - * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. - * - * Licensed under GPLv2 or later. - */ - -#ifndef _SIRF_AUDIO_PORT_H -#define _SIRF_AUDIO_PORT_H - -#define AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK 0x3F -#define AUDIO_PORT_TX_FIFO_SC_OFFSET 0 -#define AUDIO_PORT_TX_FIFO_LC_OFFSET 10 -#define AUDIO_PORT_TX_FIFO_HC_OFFSET 20 - -#define TX_FIFO_SC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_TX_FIFO_SC_OFFSET) -#define TX_FIFO_LC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_TX_FIFO_LC_OFFSET) -#define TX_FIFO_HC(x) (((x) & AUDIO_PORT_TX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_TX_FIFO_HC_OFFSET) - -#define AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK 0x0F -#define AUDIO_PORT_RX_FIFO_SC_OFFSET 0 -#define AUDIO_PORT_RX_FIFO_LC_OFFSET 10 -#define AUDIO_PORT_RX_FIFO_HC_OFFSET 20 - -#define RX_FIFO_SC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_RX_FIFO_SC_OFFSET) -#define RX_FIFO_LC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_RX_FIFO_LC_OFFSET) -#define RX_FIFO_HC(x) (((x) & AUDIO_PORT_RX_FIFO_LEVEL_CHECK_MASK) \ - << AUDIO_PORT_RX_FIFO_HC_OFFSET) -#define AUDIO_PORT_IC_CODEC_TX_CTRL (0x00F4) -#define AUDIO_PORT_IC_CODEC_RX_CTRL (0x00F8) - -#define AUDIO_PORT_IC_TXFIFO_OP (0x00FC) -#define AUDIO_PORT_IC_TXFIFO_LEV_CHK (0x0100) -#define AUDIO_PORT_IC_TXFIFO_STS (0x0104) -#define AUDIO_PORT_IC_TXFIFO_INT (0x0108) -#define AUDIO_PORT_IC_TXFIFO_INT_MSK (0x010C) - -#define AUDIO_PORT_IC_RXFIFO_OP (0x0110) -#define AUDIO_PORT_IC_RXFIFO_LEV_CHK (0x0114) -#define AUDIO_PORT_IC_RXFIFO_STS (0x0118) -#define AUDIO_PORT_IC_RXFIFO_INT (0x011C) -#define AUDIO_PORT_IC_RXFIFO_INT_MSK (0x0120) - -#define AUDIO_FIFO_START (1 << 0) -#define AUDIO_FIFO_RESET (1 << 1) - -#define AUDIO_FIFO_FULL (1 << 0) -#define AUDIO_FIFO_EMPTY (1 << 1) -#define AUDIO_FIFO_OFLOW (1 << 2) -#define AUDIO_FIFO_UFLOW (1 << 3) - -#define IC_TX_ENABLE (0x03) -#define IC_RX_ENABLE_MONO (0x01) -#define IC_RX_ENABLE_STEREO (0x03) - -#endif /*__SIRF_AUDIO_PORT_H*/ -- 1.7.5.4

2 3

[alsa-devel] [PATCH v2] ASoC: cs42xx8: Check return value of regmap_read and report correct chipid value
by Axel Lin 04 Apr '14

04 Apr '14

Fix checking return value of regmap_read(). Also fix reporting the chip_id value. CS42XX8_CHIPID_CHIP_ID_MASK is 0xF0, so the chip_id value is (val & CS42XX8_CHIPID_CHIP_ID_MASK) >> 4). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> Acked-by: Paul Handrigan <paul.handrigan(a)cirrus.com> Acked-by: Brian Austin <brian.austin(a)cirrus.com> --- sound/soc/codecs/cs42xx8.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index 082299a..8502032 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -495,17 +495,16 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap) regcache_cache_bypass(cs42xx8->regmap, true); /* Validate the chip ID */ - regmap_read(cs42xx8->regmap, CS42XX8_CHIPID, &val); - if (val < 0) { - dev_err(dev, "failed to get device ID: %x", val); - ret = -EINVAL; + ret = regmap_read(cs42xx8->regmap, CS42XX8_CHIPID, &val); + if (ret < 0) { + dev_err(dev, "failed to get device ID, ret = %d", ret); goto err_enable; } /* The top four bits of the chip ID should be 0000 */ - if ((val & CS42XX8_CHIPID_CHIP_ID_MASK) != 0x00) { + if (((val & CS42XX8_CHIPID_CHIP_ID_MASK) >> 4) != 0x00) { dev_err(dev, "unmatched chip ID: %d\n", - val & CS42XX8_CHIPID_CHIP_ID_MASK); + (val & CS42XX8_CHIPID_CHIP_ID_MASK) >> 4); ret = -EINVAL; goto err_enable; } -- 1.8.3.2

3 2

[alsa-devel] [PATCH 0/2] Add ipg clock control to sai driver
by Nicolin Chen 04 Apr '14

04 Apr '14

This series of patches add ipg clock control to fsl_sai driver and updates the vf610.dtsi accordingly. @Shawn I'm not sure if VF610 currently does full works with SAI audio function. The PATCH-2 is based on broonie/for-next. Nicolin Chen (2): ASoC: fsl_sai: Add clock control for SAI ARM: dts: Add ipg clock for sai2 on VF610 platform .../devicetree/bindings/sound/fsl-sai.txt | 7 ++-- arch/arm/boot/dts/vf610.dtsi | 4 +-- sound/soc/fsl/fsl_sai.c | 37 ++++++++++++++++++++-- sound/soc/fsl/fsl_sai.h | 2 ++ 4 files changed, 43 insertions(+), 7 deletions(-) -- 1.8.4

3 7

[alsa-devel] [PATCH 1/3] ASoC: alc5632: add an of_match table
by Stephen Warren 04 Apr '14

04 Apr '14

From: Stephen Warren <swarren(a)nvidia.com> Add a device tree match table. This serves to make the driver's support of device tree more explicit. Perhaps the fallback for DT matching to using the i2c_device_id table will go away one day, since it fails in face of devices from different vendors with the same name. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/alc5632.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 3ee8d4e41a99..85942ca36cbf 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1190,11 +1190,18 @@ static const struct i2c_device_id alc5632_i2c_table[] = { }; MODULE_DEVICE_TABLE(i2c, alc5632_i2c_table); +static const struct of_device_id alc5632_of_match[] = { + { .compatible = "realtek,alc5632", }, + { } +}; +MODULE_DEVICE_TABLE(of, alc5632_of_match); + /* i2c codec control layer */ static struct i2c_driver alc5632_i2c_driver = { .driver = { .name = "alc5632", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(alc5632_of_match), }, .probe = alc5632_i2c_probe, .remove = alc5632_i2c_remove, -- 1.8.1.5

4 13

[alsa-devel] [PATCH] ASoC: dapm: Add support for multi register mux
by Arun Shamanna Lakshmi 04 Apr '14

04 Apr '14

- Modify soc_enum struct to handle pointers for reg and mask - Add dapm get and put APIs for multi register mux with one hot encoding - Update snd_soc_dapm_update struct to support multiple reg update Signed-off-by: Arun Shamanna Lakshmi <aruns(a)nvidia.com> Signed-off-by: Songhee Baek <sbaek(a)nvidia.com> --- include/sound/soc-dapm.h | 17 ++++- include/sound/soc.h | 34 +++++++-- sound/soc/soc-core.c | 12 ++-- sound/soc/soc-dapm.c | 174 +++++++++++++++++++++++++++++++++++++++------- 4 files changed, 197 insertions(+), 40 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ef78f56..ded46732 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -305,6 +305,12 @@ struct device; .get = snd_soc_dapm_get_enum_double, \ .put = snd_soc_dapm_put_enum_double, \ .private_value = (unsigned long)&xenum } +#define SOC_DAPM_ENUM_ONEHOT(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_enum_double, \ + .get = snd_soc_dapm_get_enum_onehot, \ + .put = snd_soc_dapm_put_enum_onehot, \ + .private_value = (unsigned long)&xenum } #define SOC_DAPM_ENUM_VIRT(xname, xenum) \ SOC_DAPM_ENUM(xname, xenum) #define SOC_DAPM_ENUM_EXT(xname, xenum, xget, xput) \ @@ -378,6 +384,10 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_dapm_get_enum_onehot(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_dapm_put_enum_onehot(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_info_pin_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, @@ -590,9 +600,10 @@ struct snd_soc_dapm_widget { struct snd_soc_dapm_update { struct snd_kcontrol *kcontrol; - int reg; - int mask; - int val; + int reg[3]; + int mask[3]; + int val[3]; + int num_regs; }; /* DAPM context */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 0b83168..add326a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -177,18 +177,24 @@ {.reg = xreg, .min = xmin, .max = xmax, \ .platform_max = xmax} } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xitems, xtexts) \ -{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ +{ .reg = &(int){(xreg)}, .shift_l = xshift_l, .shift_r = xshift_r, \ .items = xitems, .texts = xtexts, \ - .mask = xitems ? roundup_pow_of_two(xitems) - 1 : 0} + .mask = &(unsigned int){(xitems ? roundup_pow_of_two(xitems) - 1 : 0)}, \ + .num_regs = 1, .type = SND_SOC_ENUM_NONE } #define SOC_ENUM_SINGLE(xreg, xshift, xitems, xtexts) \ SOC_ENUM_DOUBLE(xreg, xshift, xshift, xitems, xtexts) #define SOC_ENUM_SINGLE_EXT(xitems, xtexts) \ { .items = xitems, .texts = xtexts } #define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xitems, xtexts, xvalues) \ -{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ - .mask = xmask, .items = xitems, .texts = xtexts, .values = xvalues} +{ .reg = &(int){(xreg)}, .shift_l = xshift_l, .shift_r = xshift_r, \ + .mask = &(unsigned int){(xmask)}, .items = xitems, .texts = xtexts, \ + .values = xvalues, .num_regs = 1, .type = SND_SOC_ENUM_NONE } #define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xnitmes, xtexts, xvalues) \ SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xnitmes, xtexts, xvalues) +#define SOC_VALUE_ENUM_ONEHOT(xregs, xmasks, xnum_regs, xitems, xtexts, xvalues) \ +{ .reg = xregs, .mask = xmasks, .num_regs = xnum_regs, \ + .items = xitems, .texts = xtexts, .values = xvalues, \ + .type = SND_SOC_ENUM_ONEHOT } #define SOC_ENUM_SINGLE_VIRT(xitems, xtexts) \ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, xitems, xtexts) #define SOC_ENUM(xname, xenum) \ @@ -293,6 +299,9 @@ #define SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xmask, xtexts, xvalues) \ const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \ ARRAY_SIZE(xtexts), xtexts, xvalues) +#define SOC_VALUE_ENUM_ONEHOT_DECL(name, xregs, xmasks, xnum_regs, xtexts, xvalues) \ + const struct soc_enum name = SOC_VALUE_ENUM_ONEHOT(xregs, xmasks, xnum_regs, \ + ARRAY_SIZE(xtexts), xtexts, xvalues) #define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \ SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) #define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \ @@ -326,6 +335,17 @@ enum snd_soc_bias_level { SND_SOC_BIAS_ON = 3, }; +/* + * Soc Enum Type + * + * @NONE: soc_enum type for SINGLE, DOUBLE or VIRTUAL mux + * @ONEHOT: soc_enum type for one hot encoding mux + */ +enum snd_soc_enum_type { + SND_SOC_ENUM_NONE = 0, + SND_SOC_ENUM_ONEHOT = 1, +}; + struct device_node; struct snd_jack; struct snd_soc_card; @@ -1098,13 +1118,15 @@ struct soc_mreg_control { /* enumerated kcontrol */ struct soc_enum { - int reg; + int *reg; unsigned char shift_l; unsigned char shift_r; unsigned int items; - unsigned int mask; + unsigned int *mask; const char * const *texts; const unsigned int *values; + enum snd_soc_enum_type type; + unsigned int num_regs; }; /** diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index caebd63..cf29722 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2601,12 +2601,12 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, unsigned int val, item; unsigned int reg_val; - reg_val = snd_soc_read(codec, e->reg); - val = (reg_val >> e->shift_l) & e->mask; + reg_val = snd_soc_read(codec, e->reg[0]); + val = (reg_val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[0] = item; if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_l) & e->mask; + val = (reg_val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = item; } @@ -2636,15 +2636,15 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, if (item[0] >= e->items) return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; - mask = e->mask << e->shift_l; + mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] >= e->items) return -EINVAL; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_r; - mask |= e->mask << e->shift_r; + mask |= e->mask[0] << e->shift_r; } - return snd_soc_update_bits_locked(codec, e->reg, mask, val); + return snd_soc_update_bits_locked(codec, e->reg[0], mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c8a780d..19b004a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -511,13 +511,26 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, const struct snd_kcontrol_new *kcontrol) { struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned int val, item; + unsigned int val, item, bit_pos = -1; int i; - if (e->reg != SND_SOC_NOPM) { - soc_widget_read(dest, e->reg, &val); - val = (val >> e->shift_l) & e->mask; - item = snd_soc_enum_val_to_item(e, val); + if (e->reg[0] != SND_SOC_NOPM) { + if (e->type == SND_SOC_ENUM_ONEHOT) { + for (i = 0; i < e->num_regs; i++) { + soc_widget_read(dest, e->reg[i], &val); + val = val & e->mask[i]; + if (val != 0) { + bit_pos = __ffs(val) + + (8 * dest->codec->val_bytes * i); + break; + } + } + item = snd_soc_enum_val_to_item(e, bit_pos); + } else { + soc_widget_read(dest, e->reg[0], &val); + val = (val >> e->shift_l) & e->mask[0]; + item = snd_soc_enum_val_to_item(e, val); + } } else { /* since a virtual mux has no backing registers to * decide which path to connect, it will try to match @@ -1553,8 +1566,8 @@ static void dapm_widget_update(struct snd_soc_card *card) struct snd_soc_dapm_update *update = card->update; struct snd_soc_dapm_widget_list *wlist; struct snd_soc_dapm_widget *w = NULL; - unsigned int wi; - int ret; + unsigned int wi, i; + int ret = 0; if (!update || !dapm_kcontrol_is_powered(update->kcontrol)) return; @@ -1575,8 +1588,12 @@ static void dapm_widget_update(struct snd_soc_card *card) if (!w) return; - ret = soc_widget_update_bits_locked(w, update->reg, update->mask, - update->val); + /* dapm update for multiple registers */ + for (i = 0; i < update->num_regs; i++) { + ret |= soc_widget_update_bits_locked(w, update->reg[i], + update->mask[i], update->val[i]); + } + if (ret < 0) dev_err(w->dapm->dev, "ASoC: %s DAPM update failed: %d\n", w->name, ret); @@ -2866,10 +2883,10 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, if (change) { if (reg != SND_SOC_NOPM) { update.kcontrol = kcontrol; - update.reg = reg; - update.mask = mask; - update.val = val; - + update.reg[0] = reg; + update.mask[0] = mask; + update.val[0] = val; + update.num_regs = 1; card->update = &update; } @@ -2903,15 +2920,15 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val; - if (e->reg != SND_SOC_NOPM) - reg_val = snd_soc_read(codec, e->reg); + if (e->reg[0] != SND_SOC_NOPM) + reg_val = snd_soc_read(codec, e->reg[0]); else reg_val = dapm_kcontrol_get_value(kcontrol); - val = (reg_val >> e->shift_l) & e->mask; + val = (reg_val >> e->shift_l) & e->mask[0]; ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val); if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_r) & e->mask; + val = (reg_val >> e->shift_r) & e->mask[0]; val = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = val; } @@ -2945,27 +2962,28 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; - mask = e->mask << e->shift_l; + mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] > e->items) return -EINVAL; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_l; - mask |= e->mask << e->shift_r; + mask |= e->mask[0] << e->shift_r; } mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (e->reg != SND_SOC_NOPM) - change = snd_soc_test_bits(codec, e->reg, mask, val); + if (e->reg[0] != SND_SOC_NOPM) + change = snd_soc_test_bits(codec, e->reg[0], mask, val); else change = dapm_kcontrol_set_value(kcontrol, val); if (change) { - if (e->reg != SND_SOC_NOPM) { + if (e->reg[0] != SND_SOC_NOPM) { update.kcontrol = kcontrol; - update.reg = e->reg; - update.mask = mask; - update.val = val; + update.reg[0] = e->reg[0]; + update.mask[0] = mask; + update.val[0] = val; + update.num_regs = 1; card->update = &update; } @@ -2984,6 +3002,112 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); /** + * snd_soc_dapm_get_enum_onehot - dapm enumerated onehot mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a dapm enumerated onehot encoded mixer control + * + * Returns 0 for success. + */ +int snd_soc_dapm_get_enum_onehot(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int reg_val, val, bit_pos = -1, reg_idx; + + for (reg_idx = 0; reg_idx < e->num_regs; reg_idx++) { + reg_val = snd_soc_read(codec, e->reg[reg_idx]); + val = reg_val & e->mask[reg_idx]; + if (val != 0) { + bit_pos = __ffs(val) + (8 * codec->val_bytes * reg_idx); + break; + } + } + + ucontrol->value.enumerated.item[0] = + snd_soc_enum_val_to_item(e, bit_pos); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_onehot); + +/** + * snd_soc_dapm_put_enum_onehot - dapm enumerated onehot mixer put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to put the value of a dapm enumerated onehot encoded mixer control + * + * Returns 0 for success. + */ +int snd_soc_dapm_put_enum_onehot(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct snd_soc_card *card = codec->card; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int change = 0, reg_idx = 0, value, bit_pos; + struct snd_soc_dapm_update update; + int ret = 0, reg_val = 0, i, update_idx = 0; + + if (item[0] >= e->items) + return -EINVAL; + + value = snd_soc_enum_item_to_val(e, item[0]); + + if (value >= 0) { + /* get the register index and value to set */ + reg_idx = value / (8 * codec->val_bytes); + bit_pos = value % (8 * codec->val_bytes); + reg_val = BIT(bit_pos); + } + + for (i = 0; i < e->num_regs; i++) { + if (i == reg_idx) { + change = snd_soc_test_bits(codec, e->reg[i], + e->mask[i], reg_val); + /* set the selected register */ + update.reg[e->num_regs - 1] = e->reg[reg_idx]; + update.mask[e->num_regs - 1] = e->mask[reg_idx]; + update.val[e->num_regs - 1] = reg_val; + } else { + /* accumulate the change to update the DAPM path + when none is selected */ + change |= snd_soc_test_bits(codec, e->reg[i], + e->mask[i], 0); + + /* clear the register when not selected */ + update.reg[update_idx] = e->reg[i]; + update.mask[update_idx] = e->mask[i]; + update.val[update_idx++] = 0; + } + } + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + if (change) { + update.kcontrol = kcontrol; + update.num_regs = 3; + card->update = &update; + + ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); + + card->update = NULL; + } + + mutex_unlock(&card->dapm_mutex); + + if (ret > 0) + soc_dpcm_runtime_update(card); + + return change; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_onehot); + +/** * snd_soc_dapm_info_pin_switch - Info for a pin switch * * @kcontrol: mixer control -- 1.7.9.5

4 11

[alsa-devel] [PATCH 0/4] Add ESAI & SPDIF Vybird and LS1+ support.
by Xiubo Li 04 Apr '14

04 Apr '14

Xiubo Li (4): ASoC: esai: fix buggy of 'ret' type definition. ASoC: esai: Add Vybird and LS1+ compatibles support. ASoC: spdif: Sort the header files alphabetically. ASoC: spdif: Add Vybird and LS1+ compatibles support. sound/soc/fsl/fsl_esai.c | 10 +++++++--- sound/soc/fsl/fsl_spdif.c | 10 ++++++---- 2 files changed, 13 insertions(+), 7 deletions(-) -- 1.8.4

4 12

[alsa-devel] [PATCH RESEND 1/2] ASoC: fsl: Separation of the main audio options and boards
by Alexander Shiyan 04 Apr '14

04 Apr '14

This patch provide separation of the main audio options and boards. Signed-off-by: Alexander Shiyan <shc_work(a)mail.ru> --- sound/soc/fsl/Kconfig | 36 ++++++++++++++++++++++-------------- 1 file changed, 22 insertions(+), 14 deletions(-) diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 597962e..94c9350 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -1,3 +1,5 @@ +menu "SoC Audio for Freescale CPUs" + config SND_SOC_FSL_SAI tristate select REGMAP_MMIO @@ -17,13 +19,27 @@ config SND_SOC_FSL_ESAI config SND_SOC_FSL_UTILS tristate -menuconfig SND_POWERPC_SOC +config SND_SOC_IMX_PCM_DMA + tristate + select SND_SOC_GENERIC_DMAENGINE_PCM + +config SND_SOC_IMX_AUDMUX + tristate + +config SND_POWERPC_SOC tristate "SoC Audio for Freescale PowerPC CPUs" depends on FSL_SOC || PPC_MPC52xx help Say Y or M if you want to add support for codecs attached to the PowerPC CPUs. +config SND_IMX_SOC + tristate "SoC Audio for Freescale i.MX CPUs" + depends on ARCH_MXC || COMPILE_TEST + help + Say Y or M if you want to add support for codecs attached to + the i.MX CPUs. + if SND_POWERPC_SOC config SND_MPC52xx_DMA @@ -32,6 +48,8 @@ config SND_MPC52xx_DMA config SND_SOC_POWERPC_DMA tristate +comment "SoC Audio support for Freescale PPC boards:" + config SND_SOC_MPC8610_HPCD tristate "ALSA SoC support for the Freescale MPC8610 HPCD board" # I2C is necessary for the CS4270 driver @@ -109,13 +127,6 @@ config SND_MPC52xx_SOC_EFIKA endif # SND_POWERPC_SOC -menuconfig SND_IMX_SOC - tristate "SoC Audio for Freescale i.MX CPUs" - depends on ARCH_MXC || COMPILE_TEST - help - Say Y or M if you want to add support for codecs attached to - the i.MX CPUs. - if SND_IMX_SOC config SND_SOC_IMX_SSI @@ -125,12 +136,7 @@ config SND_SOC_IMX_PCM_FIQ tristate select FIQ -config SND_SOC_IMX_PCM_DMA - tristate - select SND_SOC_GENERIC_DMAENGINE_PCM - -config SND_SOC_IMX_AUDMUX - tristate +comment "SoC Audio support for Freescale i.MX boards:" config SND_MXC_SOC_WM1133_EV1 tristate "Audio on the i.MX31ADS with WM1133-EV1 fitted" @@ -223,3 +229,5 @@ config SND_SOC_IMX_MC13783 select SND_SOC_IMX_PCM_DMA endif # SND_IMX_SOC + +endmenu -- 1.8.3.2

2 2

[alsa-devel] [PATCH] ASoC: cs42xx8: Check return value of regmap_read and report correct chipid value
by Axel Lin 04 Apr '14

04 Apr '14

Fix checking return value of regmap_read(). Also fix reporting the chip_id value. CS42XX8_CHIPID_CHIP_ID_MASK is 0xF0, so the chip_id value is (val & CS42XX8_CHIPID_CHIP_ID_MASK) >> 4). Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/cs42xx8.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index 082299a..b1606cf 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -495,17 +495,16 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap) regcache_cache_bypass(cs42xx8->regmap, true); /* Validate the chip ID */ - regmap_read(cs42xx8->regmap, CS42XX8_CHIPID, &val); - if (val < 0) { + ret = regmap_read(cs42xx8->regmap, CS42XX8_CHIPID, &val); + if (ret < 0) { dev_err(dev, "failed to get device ID: %x", val); - ret = -EINVAL; goto err_enable; } /* The top four bits of the chip ID should be 0000 */ - if ((val & CS42XX8_CHIPID_CHIP_ID_MASK) != 0x00) { + if (((val & CS42XX8_CHIPID_CHIP_ID_MASK) >> 4) != 0x00) { dev_err(dev, "unmatched chip ID: %d\n", - val & CS42XX8_CHIPID_CHIP_ID_MASK); + (val & CS42XX8_CHIPID_CHIP_ID_MASK) >> 4); ret = -EINVAL; goto err_enable; } -- 1.8.3.2

5 4

Re: [alsa-devel] [Intel-gfx] [PATCH v2] drm/i915: move power domain init earlier during system resume
by Daniel Vetter 04 Apr '14

04 Apr '14

On Tue, Apr 01, 2014 at 09:50:43PM +0300, Imre Deak wrote: > On Tue, 2014-04-01 at 19:48 +0200, Daniel Vetter wrote: > > On Tue, Apr 01, 2014 at 07:55:22PM +0300, Imre Deak wrote: > > > During resume the intel hda audio driver depends on the i915 driver > > > reinitializing the audio power domain. Since the order of calling the > > > i915 resume handler wrt. that of the audio driver is not guaranteed, > > > move the power domain reinitialization step to the resume_early > > > handler. This is guaranteed to run before the resume handler of any > > > other driver. > > > > > > The power domain initialization in turn requires us to enable the i915 > > > pci device first, so move that part earlier too. > > > > > > Accordingly disabling of the i915 pci device should happen after the > > > audio suspend handler ran. So move the disabling later from the i915 > > > resume handler to the resume_late handler. > > > > > > v2: > > > - move intel_uncore_sanitize/early_sanitize earlier too, so they don't > > > get reordered wrt. intel_power_domains_init_hw() > > > > > > Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=76152 > > > Signed-off-by: Imre Deak <imre.deak(a)intel.com> > > > > So this is kinda why we should have gone with something proper, like a new > > hdmi sink platform device created by i915 and registered as a driver by > > snd-hda. Then the power domains stuff in the device core should take care > > of these kinds of ordering issues. Or at least snd-hda can tell it that it > > needs to wait for the hdmi-sink power domain to go on first before it can > > resume, I'm not really fluent on the details here. > > > > And having a hdmi sink bus would allow us to throw all kinds of crap into > > a clearly-defined interface, e.g. eld handling, hdcp synchronization, hpd > > forwarding and all the other fun stuff. > > > > So not sure what I should do with this here now. > > Right, I'm not too happy about this solution either, so if anything it > could be considered only a stop-gap fix. What you suggest seems to be a > cleaner way but it'd require more time to investigate/implement at least > on my part (but I'm ok to put it on my TODO list). We'd definitely need to discuss this with Takashi Iwai, since it would only really be useful if it's good enough to solve the general pile of sound/gfx coordination issues we have with hdmi. Adding him and alsa-dev. -Daniel -- Daniel Vetter Software Engineer, Intel Corporation +41 (0) 79 365 57 48 - http://blog.ffwll.ch

2 4

[alsa-devel] [PATCH 1/2] snd-cs8427: separate HW initialization
by Ondrej Zary 03 Apr '14

03 Apr '14

Separate HW initialization from device creation. This is needed for suspend/resume support. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- include/sound/cs8427.h | 1 + sound/i2c/cs8427.c | 57 ++++++++++++++++++++++++++++++++---------------- 2 files changed, 39 insertions(+), 19 deletions(-) diff --git a/include/sound/cs8427.h b/include/sound/cs8427.h index f862cff..0b6a187 100644 --- a/include/sound/cs8427.h +++ b/include/sound/cs8427.h @@ -188,6 +188,7 @@ struct snd_pcm_substream; +int snd_cs8427_init(struct snd_i2c_bus *bus, struct snd_i2c_device *device); int snd_cs8427_create(struct snd_i2c_bus *bus, unsigned char addr, unsigned int reset_timeout, struct snd_i2c_device **r_cs8427); int snd_cs8427_reg_write(struct snd_i2c_device *device, unsigned char reg, diff --git a/sound/i2c/cs8427.c b/sound/i2c/cs8427.c index 6c2dc38..d9c15d9 100644 --- a/sound/i2c/cs8427.c +++ b/sound/i2c/cs8427.c @@ -150,10 +150,8 @@ static void snd_cs8427_free(struct snd_i2c_device *device) kfree(device->private_data); } -int snd_cs8427_create(struct snd_i2c_bus *bus, - unsigned char addr, - unsigned int reset_timeout, - struct snd_i2c_device **r_cs8427) +int snd_cs8427_init(struct snd_i2c_bus *bus, + struct snd_i2c_device *device) { static unsigned char initvals1[] = { CS8427_REG_CONTROL1 | CS8427_REG_AUTOINC, @@ -200,22 +198,10 @@ int snd_cs8427_create(struct snd_i2c_bus *bus, Inhibit E->F transfers. */ CS8427_UD | CS8427_EFTUI | CS8427_DETUI, }; + struct cs8427 *chip = device->private_data; int err; - struct cs8427 *chip; - struct snd_i2c_device *device; unsigned char buf[24]; - if ((err = snd_i2c_device_create(bus, "CS8427", - CS8427_ADDR | (addr & 7), - &device)) < 0) - return err; - chip = device->private_data = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) { - snd_i2c_device_free(device); - return -ENOMEM; - } - device->private_free = snd_cs8427_free; - snd_i2c_lock(bus); err = snd_cs8427_reg_read(device, CS8427_REG_ID_AND_VER); if (err != CS8427_VER8427A) { @@ -264,10 +250,44 @@ int snd_cs8427_create(struct snd_i2c_bus *bus, snd_i2c_unlock(bus); /* turn on run bit and rock'n'roll */ + snd_cs8427_reset(device); + + return 0; + + __fail: + snd_i2c_unlock(bus); + + return err; +} + +EXPORT_SYMBOL(snd_cs8427_init); + +int snd_cs8427_create(struct snd_i2c_bus *bus, + unsigned char addr, + unsigned int reset_timeout, + struct snd_i2c_device **r_cs8427) +{ + int err; + struct cs8427 *chip; + struct snd_i2c_device *device; + + if ((err = snd_i2c_device_create(bus, "CS8427", + CS8427_ADDR | (addr & 7), + &device)) < 0) + return err; + chip = device->private_data = kzalloc(sizeof(*chip), GFP_KERNEL); + if (chip == NULL) { + snd_i2c_device_free(device); + return -ENOMEM; + } + device->private_free = snd_cs8427_free; + if (reset_timeout < 1) reset_timeout = 1; chip->reset_timeout = reset_timeout; - snd_cs8427_reset(device); + + if ((err = snd_cs8427_init(bus, device))) + goto __fail; #if 0 // it's nice for read tests { @@ -286,7 +306,6 @@ int snd_cs8427_create(struct snd_i2c_bus *bus, return 0; __fail: - snd_i2c_unlock(bus); snd_i2c_device_free(device); return err < 0 ? err : -EIO; } -- Ondrej Zary

2 6

[alsa-devel] [PATCH] ASoC: codec: fix the sparse check warnings.
by Xiubo Li 03 Apr '14

03 Apr '14

Some thing Likes: reproduce: make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/codecs/wm8997.c:1084:15: sparse: symbol \ 'wm8997_get_regmap' was not declared. Should it be static? Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/codecs/88pm860x-codec.c | 2 +- sound/soc/codecs/cq93vc.c | 2 +- sound/soc/codecs/mc13783.c | 2 +- sound/soc/codecs/si476x.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8350.c | 2 +- sound/soc/codecs/wm8400.c | 2 +- sound/soc/codecs/wm8994.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- 10 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index dc9e6b9..f073b6f 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1358,7 +1358,7 @@ static int pm860x_remove(struct snd_soc_codec *codec) return 0; } -struct regmap *pm860x_get_regmap(struct device *dev) +static struct regmap *pm860x_get_regmap(struct device *dev) { struct pm860x_priv *pm860x = dev_get_drvdata(dev); diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 5e5518d..5ee48c8 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -152,7 +152,7 @@ static int cq93vc_remove(struct snd_soc_codec *codec) return 0; } -struct regmap *cq93vc_get_regmap(struct device *dev) +static struct regmap *cq93vc_get_regmap(struct device *dev) { struct davinci_vc *davinci_vc = codec->dev->platform_data; diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 8c0cec3..9029e20 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -727,7 +727,7 @@ static struct snd_soc_dai_driver mc13783_dai_sync[] = { } }; -struct regmap *mc13783_get_regmap(struct device *dev) +static struct regmap *mc13783_get_regmap(struct device *dev) { return dev_get_regmap(dev->parent, NULL); } diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 961b7e8..f26befb 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -231,7 +231,7 @@ static struct snd_soc_dai_driver si476x_dai = { .ops = &si476x_dai_ops, }; -struct regmap *si476x_get_regmap(struct device *dev) +static struct regmap *si476x_get_regmap(struct device *dev) { return dev_get_regmap(dev->parent, NULL); } diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index aa60ba2..7a04653 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1798,7 +1798,7 @@ static unsigned int wm5102_digital_vu[] = { ARIZONA_DAC_DIGITAL_VOLUME_5R, }; -struct regmap *wm5102_get_regmap(struct device *dev) +static struct regmap *wm5102_get_regmap(struct device *dev) { struct wm5102_priv *priv = dev_get_drvdata(dev); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 4be5f99..97eb1bc 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1629,7 +1629,7 @@ static unsigned int wm5110_digital_vu[] = { ARIZONA_DAC_DIGITAL_VOLUME_6R, }; -struct regmap *wm5110_get_regmap(struct device *dev) +static struct regmap *wm5110_get_regmap(struct device *dev) { struct wm5110_priv *priv = dev_get_drvdata(dev); diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 6b31a9f..1bd14b6 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1606,7 +1606,7 @@ static int wm8350_codec_remove(struct snd_soc_codec *codec) return 0; } -struct regmap *wm8350_get_regmap(struct device *dev) +static struct regmap *wm8350_get_regmap(struct device *dev) { struct wm8350 *wm8350 = dev_get_platdata(dev); diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index e6410f2..5880d22 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1359,7 +1359,7 @@ static int wm8400_codec_remove(struct snd_soc_codec *codec) return 0; } -struct regmap *wm8400_get_regmap(struct device *dev) +static struct regmap *wm8400_get_regmap(struct device *dev) { struct wm8400 *wm8400 = dev_get_platdata(dev); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index daa4edc..6f336da 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4432,7 +4432,7 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) return 0; } -struct regmap *wm8994_get_regmap(struct device *dev) +static struct regmap *wm8994_get_regmap(struct device *dev) { struct wm8994 *control = dev_get_drvdata(dev->parent); diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 3d50621..09c4150 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1081,7 +1081,7 @@ static unsigned int wm8997_digital_vu[] = { ARIZONA_DAC_DIGITAL_VOLUME_5R, }; -struct regmap *wm8997_get_regmap(struct device *dev) +static struct regmap *wm8997_get_regmap(struct device *dev) { struct wm8997_priv *priv = dev_get_drvdata(dev); -- 1.8.4

2 1

[alsa-devel] [PATCH] src/pcm/pcm_ladspa.c: Delay LADSPA plugin activate call
by Matthias Larisch 03 Apr '14

03 Apr '14

Some LADSPA Plugins rely on connected control ports on activate call. While this is not okay by spec, the spec also encourages the activate call happening as late as possible. Signed-off-by: Matthias Larisch <mail(a)matthias-larisch.de> --- src/pcm/pcm_ladspa.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_ladspa.c b/src/pcm/pcm_ladspa.c index 9ce5242..7d1e3df 100644 --- a/src/pcm/pcm_ladspa.c +++ b/src/pcm/pcm_ladspa.c @@ -645,8 +645,6 @@ static int snd_pcm_ladspa_allocate_instances(snd_pcm_t *pcm, snd_pcm_ladspa_t *l return -EINVAL; } list_add_tail(&instance->list, &plugin->instances); - if (plugin->desc->activate) - plugin->desc->activate(instance->handle); if (plugin->policy == SND_PCM_LADSPA_POLICY_DUPLICATE) { err = snd_pcm_ladspa_connect_plugin_duplicate(plugin, &plugin->input, &plugin->output, instance, idx); if (err < 0) { @@ -664,6 +662,8 @@ static int snd_pcm_ladspa_allocate_instances(snd_pcm_t *pcm, snd_pcm_ladspa_t *l assert(err >= 0); err = snd_pcm_ladspa_connect_controls(plugin, &plugin->output, instance); assert(err >= 0); + if (plugin->desc->activate) + plugin->desc->activate(instance->handle); } err = snd_pcm_ladspa_check_connect(plugin, &plugin->input, &instance->input, depth); if (err < 0) -- 1.9.1

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix silent speaker output due to mute LED fixup
by Takashi Iwai 03 Apr '14

03 Apr '14

The recent fixups for HP laptops to support the mute LED made the speaker output silent on some machines. It turned out that they use the NID 0x18 for the speaker while it's also used for controlling the LED via VREF bits although the current driver code blindly assumes that such a node is a mic pin (where 0x18 is usually so). This patch fixes the problem by only changing the VREF bits and keeping the other pin ctl bits. Reported-and-tested-by: Hui Wang <hui.wang(a)canonical.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index dba297288398..053107786f33 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3371,8 +3371,9 @@ static void alc269_fixup_mic_mute_hook(void *private_data, int enabled) if (spec->mute_led_polarity) enabled = !enabled; - pinval = AC_PINCTL_IN_EN | - (enabled ? AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80); + pinval = snd_hda_codec_get_pin_target(codec, spec->mute_led_nid); + pinval &= ~AC_PINCTL_VREFEN; + pinval |= enabled ? AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80; if (spec->mute_led_nid) snd_hda_set_pin_ctl_cache(codec, spec->mute_led_nid, pinval); } -- 1.9.1

1 0

[alsa-devel] Fixed single output empty pin_sense issue for alc283
by Kailang 03 Apr '14

03 Apr '14

Hi Takashi, Some machine with single output. It has only one HP out on it. alc283_init and alc283_shutup will get empty hp_pin value. I add the check for the attachment patch. Thanks. BR, Kailang

2 1

[alsa-devel] [PATCH] ASoC: fsl_sai: Add imx6sx platform support
by Nicolin Chen 03 Apr '14

03 Apr '14

The next coming i.MX6 Solo X SoC also contains SAI module while we use imp_pcm_init() for i.MX platform. So this patch adds one compatible route for imx6sx and updates the DT doc accordingly. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- Documentation/devicetree/bindings/sound/fsl-sai.txt | 2 +- sound/soc/fsl/fsl_sai.c | 12 ++++++++++-- sound/soc/fsl/fsl_sai.h | 1 + 3 files changed, 12 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 98611a6..35c09fe 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -7,7 +7,7 @@ codec/DSP interfaces. Required properties: -- compatible: Compatible list, contains "fsl,vf610-sai". +- compatible: Compatible list, contains "fsl,vf610-sai" or "fsl,imx6sx-sai". - reg: Offset and length of the register set for the device. - clocks: Must contain an entry for each entry in clock-names. - clock-names : Must include the "sai" entry. diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index d64c33f..9ed6795 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -22,6 +22,7 @@ #include <sound/pcm_params.h> #include "fsl_sai.h" +#include "imx-pcm.h" #define FSL_SAI_FLAGS (FSL_SAI_CSR_SEIE |\ FSL_SAI_CSR_FEIE) @@ -592,6 +593,9 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->pdev = pdev; + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) + sai->sai_on_imx = true; + sai->big_endian_regs = of_property_read_bool(np, "big-endian-regs"); if (sai->big_endian_regs) fsl_sai_regmap_config.val_format_endian = REGMAP_ENDIAN_BIG; @@ -634,12 +638,16 @@ static int fsl_sai_probe(struct platform_device *pdev) if (ret) return ret; - return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); + if (sai->sai_on_imx) + return imx_pcm_dma_init(pdev); + else + return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, + SND_DMAENGINE_PCM_FLAG_NO_RESIDUE); } static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, + { .compatible = "fsl,imx6sx-sai", }, { /* sentinel */ } }; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index be26d46..677670d 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -130,6 +130,7 @@ struct fsl_sai { bool big_endian_regs; bool big_endian_data; bool is_dsp_mode; + bool sai_on_imx; struct snd_dmaengine_dai_dma_data dma_params_rx; struct snd_dmaengine_dai_dma_data dma_params_tx; -- 1.8.4

2 2

[alsa-devel] [PATCH v3] ASoC: sta350: Add codec driver
by Daniel Mack 02 Apr '14

02 Apr '14

From: Sven Brandau <brandau(a)gmx.de> The TI STA350 is an integrated 2.1-channel power amplifier that is controllable over I2C. This patch adds an ASoC driver for it. At a glance, this chip is very similar to the STA320 for which a driver already exists. In details, however, the register maps contain subtle differences which made a whole new driver easier to write and maintain. [daniel(a)zonque.org: cleanups, DT property rework, rebased on asoc-next] Signed-off-by: Sven Brandau <brandau(a)gmx.de> Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- v2 -> v3: * Renamed the GPIO DT properties to *-gpios * Use DECLARE_TLV_DB_RANGE for automatic TLV_DB_RANGE_HEAD calculation * Use the new gpiod API * Some more cleanups v1 -> v2: * Added my S-o-b * Made the Kconfig symbol visible * Documented regulator notes in DT bindings doc * Fixed spelling and other style nits * Converted to SOC_ENUM_SINGLE_DECL * Added a lock around the logic in sta350_coefficient_get() * Added a DAPM route from "DAC" to "Playback" * Use dev_dbg() rather than pr_debug() * Changed the code to make use of regmap_update_bits rather than open-coding the same functionality * Cleaned up the reset sequence logic * If a gpio was requested, the claiming has to succeed (handles EPROBE_DEFER cases) * Kick codec->control_data initialization * Skip initialization of default values in regcache * Apply power before running init sequences * Parse the DT without consulting of_match_device() .../devicetree/bindings/sound/st,sta350.txt | 107 ++ include/sound/sta350.h | 52 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sta350.c | 1266 ++++++++++++++++++++ sound/soc/codecs/sta350.h | 228 ++++ 6 files changed, 1660 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,sta350.txt create mode 100644 include/sound/sta350.h create mode 100644 sound/soc/codecs/sta350.c create mode 100644 sound/soc/codecs/sta350.h diff --git a/Documentation/devicetree/bindings/sound/st,sta350.txt b/Documentation/devicetree/bindings/sound/st,sta350.txt new file mode 100644 index 0000000..9501888 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,sta350.txt @@ -0,0 +1,107 @@ +STA350 audio CODEC + +The driver for this device only supports I2C. + +Required properties: + + - compatible: "st,sta350" + - reg: the I2C address of the device for I2C + - reset-gpios: a GPIO spec for the reset pin. If specified, it will be + deasserted before communication to the codec starts. + + - power-down-gpios: a GPIO spec for the power down pin. If specified, + it will be deasserted before communication to the codec + starts. + + - vdd-dig-supply: regulator spec, providing 3.3V + - vdd-pll-supply: regulator spec, providing 3.3V + - vcc-supply: regulator spec, providing 5V - 26V + +Optional properties: + + - st,output-conf: number, Selects the output configuration: + 0: 2-channel (full-bridge) power, 2-channel data-out + 1: 2 (half-bridge). 1 (full-bridge) on-board power + 2: 2 Channel (Full-Bridge) Power, 1 Channel FFX + 3: 1 Channel Mono-Parallel + If parameter is missing, mode 0 will be enabled. + + - st,ch1-output-mapping: Channel 1 output mapping + - st,ch2-output-mapping: Channel 2 output mapping + - st,ch3-output-mapping: Channel 3 output mapping + 0: Channel 1 + 1: Channel 2 + 2: Channel 3 + If parameter is missing, channel 1 is choosen. + + - st,thermal-warning-recover: + If present, thermal warning recovery is enabled. + + - st,thermal-warning-adjustment: + If present, thermal warning adjustment is enabled. + + - st,fault-detect-recovery: + If present, then fault recovery will be enabled. + + - st,ffx-power-output-mode: string + The FFX power output mode selects how the FFX output timing is + configured. Must be one of these values: + - "drop-compensation" + - "tapered-compensation" + - "full-power-mode" + - "variable-drop-compensation" (default) + + - st,drop-compensation-ns: number + Only required for "st,ffx-power-output-mode" == + "variable-drop-compensation". + Specifies the drop compensation in nanoseconds. + The value must be in the range of 0..300, and only + multiples of 20 are allowed. Default is 140ns. + + - st,overcurrent-warning-adjustment: + If present, overcurrent warning adjustment is enabled. + + - st,max-power-use-mpcc: + If present, then MPCC bits are used for MPC coefficients, + otherwise standard MPC coefficients are used. + + - st,max-power-corr: + If present, power bridge correction for THD reduction near maximum + power output is enabled. + + - st,am-reduction-mode: + If present, FFX mode runs in AM reduction mode, otherwise normal + FFX mode is used. + + - st,odd-pwm-speed-mode: + If present, PWM speed mode run on odd speed mode (341.3 kHz) on all + channels. If not present, normal PWM spped mode (384 kHz) will be used. + + - st,distortion-compensation: + If present, distortion compensation variable uses DCC coefficient. + If not present, preset DC coefficient is used. + + - st,invalid-input-detect-mute: + If not present, automatic invalid input detect mute is enabled. + + + +Example: + +codec: sta350@38 { + compatible = "st,sta350"; + reg = <0x1c>; + reset-gpios = <&gpio1 19 0>; + power-down-gpios = <&gpio1 16 0>; + st,output-conf = <0x3>; // set output to 2-channel + // (full-bridge) power, + // 2-channel data-out + st,ch1-output-mapping = <0>; // set channel 1 output ch 1 + st,ch2-output-mapping = <0>; // set channel 2 output ch 1 + st,ch3-output-mapping = <0>; // set channel 3 output ch 1 + st,max-power-correction; // enables power bridge + // correction for THD reduction + // near maximum power output + st,invalid-input-detect-mute; // mute if no valid digital + // audio signal is provided. +}; diff --git a/include/sound/sta350.h b/include/sound/sta350.h new file mode 100644 index 0000000..3a329810 --- /dev/null +++ b/include/sound/sta350.h @@ -0,0 +1,52 @@ +/* + * Platform data for ST STA350 ASoC codec driver. + * + * Copyright: 2014 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * 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 __LINUX_SND__STA350_H +#define __LINUX_SND__STA350_H + +#define STA350_OCFG_2CH 0 +#define STA350_OCFG_2_1CH 1 +#define STA350_OCFG_1CH 3 + +#define STA350_OM_CH1 0 +#define STA350_OM_CH2 1 +#define STA350_OM_CH3 2 + +#define STA350_THERMAL_ADJUSTMENT_ENABLE 1 +#define STA350_THERMAL_RECOVERY_ENABLE 2 +#define STA350_FAULT_DETECT_RECOVERY_BYPASS 1 + +#define STA350_FFX_PM_DROP_COMP 0 +#define STA350_FFX_PM_TAPERED_COMP 1 +#define STA350_FFX_PM_FULL_POWER 2 +#define STA350_FFX_PM_VARIABLE_DROP_COMP 3 + + +struct sta350_platform_data { + u8 output_conf; + u8 ch1_output_mapping; + u8 ch2_output_mapping; + u8 ch3_output_mapping; + u8 ffx_power_output_mode; + u8 drop_compensation_ns; + unsigned int thermal_warning_recovery:1; + unsigned int thermal_warning_adjustment:1; + unsigned int fault_detect_recovery:1; + unsigned int oc_warning_adjustment:1; + unsigned int max_power_use_mpcc:1; + unsigned int max_power_correction:1; + unsigned int am_reduction_mode:1; + unsigned int odd_pwm_speed_mode:1; + unsigned int distortion_compensation:1; + unsigned int invalid_input_detect_mute:1; +}; + +#endif /* __LINUX_SND__STA350_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f0e8401..c7b853f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -80,6 +80,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SSM2602_SPI if SPI_MASTER select SND_SOC_SSM2602_I2C if I2C select SND_SOC_STA32X if I2C + select SND_SOC_STA350 if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TAS5086 if I2C @@ -435,6 +436,10 @@ config SND_SOC_SSM2602_I2C config SND_SOC_STA32X tristate +config SND_SOC_STA350 + tristate "STA350 speaker amplifier" + depends on I2C + config SND_SOC_STA529 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3c4d275..efdb4d0 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -74,6 +74,7 @@ snd-soc-ssm2602-objs := ssm2602.o snd-soc-ssm2602-spi-objs := ssm2602-spi.o snd-soc-ssm2602-i2c-objs := ssm2602-i2c.o snd-soc-sta32x-objs := sta32x.o +snd-soc-sta350-objs := sta350.o snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o snd-soc-tas5086-objs := tas5086.o @@ -221,6 +222,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o obj-$(CONFIG_SND_SOC_SSM2602_I2C) += snd-soc-ssm2602-i2c.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o +obj-$(CONFIG_SND_SOC_STA350) += snd-soc-sta350.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o diff --git a/sound/soc/codecs/sta350.c b/sound/soc/codecs/sta350.c new file mode 100644 index 0000000..552e92a --- /dev/null +++ b/sound/soc/codecs/sta350.c @@ -0,0 +1,1266 @@ +/* + * Codec driver for ST STA350 2.1-channel high-efficiency digital audio system + * + * Copyright: 2014 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * based on code from: + * Raumfeld GmbH + * Johannes Stezenbach <js(a)sig21.net> + * Wolfson Microelectronics PLC. + * Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * Freescale Semiconductor, Inc. + * Timur Tabi <timur(a)freescale.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. + */ + +#define pr_fmt(fmt) KBUILD_MODNAME ":%s:%d: " fmt, __func__, __LINE__ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <linux/regmap.h> +#include <linux/regulator/consumer.h> +#include <linux/gpio/consumer.h> +#include <linux/slab.h> +#include <sound/core.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 <sound/sta350.h> +#include "sta350.h" + +#define STA350_RATES (SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +#define STA350_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE) + +/* Power-up register defaults */ +static const struct reg_default sta350_regs[] = { + { 0x0, 0x63 }, + { 0x1, 0x80 }, + { 0x2, 0xdf }, + { 0x3, 0x40 }, + { 0x4, 0xc2 }, + { 0x5, 0x5c }, + { 0x6, 0x00 }, + { 0x7, 0xff }, + { 0x8, 0x60 }, + { 0x9, 0x60 }, + { 0xa, 0x60 }, + { 0xb, 0x00 }, + { 0xc, 0x00 }, + { 0xd, 0x00 }, + { 0xe, 0x00 }, + { 0xf, 0x40 }, + { 0x10, 0x80 }, + { 0x11, 0x77 }, + { 0x12, 0x6a }, + { 0x13, 0x69 }, + { 0x14, 0x6a }, + { 0x15, 0x69 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, + { 0x18, 0x00 }, + { 0x19, 0x00 }, + { 0x1a, 0x00 }, + { 0x1b, 0x00 }, + { 0x1c, 0x00 }, + { 0x1d, 0x00 }, + { 0x1e, 0x00 }, + { 0x1f, 0x00 }, + { 0x20, 0x00 }, + { 0x21, 0x00 }, + { 0x22, 0x00 }, + { 0x23, 0x00 }, + { 0x24, 0x00 }, + { 0x25, 0x00 }, + { 0x26, 0x00 }, + { 0x27, 0x2a }, + { 0x28, 0xc0 }, + { 0x29, 0xf3 }, + { 0x2a, 0x33 }, + { 0x2b, 0x00 }, + { 0x2c, 0x0c }, + { 0x31, 0x00 }, + { 0x36, 0x00 }, + { 0x37, 0x00 }, + { 0x38, 0x00 }, + { 0x39, 0x01 }, + { 0x3a, 0xee }, + { 0x3b, 0xff }, + { 0x3c, 0x7e }, + { 0x3d, 0xc0 }, + { 0x3e, 0x26 }, + { 0x3f, 0x00 }, + { 0x48, 0x00 }, + { 0x49, 0x00 }, + { 0x4a, 0x00 }, + { 0x4b, 0x04 }, + { 0x4c, 0x00 }, +}; + +static const struct regmap_range sta350_write_regs_range[] = { + regmap_reg_range(STA350_CONFA, STA350_AUTO2), + regmap_reg_range(STA350_C1CFG, STA350_FDRC2), + regmap_reg_range(STA350_EQCFG, STA350_EVOLRES), + regmap_reg_range(STA350_NSHAPE, STA350_MISC2), +}; + +static const struct regmap_range sta350_read_regs_range[] = { + regmap_reg_range(STA350_CONFA, STA350_AUTO2), + regmap_reg_range(STA350_C1CFG, STA350_STATUS), + regmap_reg_range(STA350_EQCFG, STA350_EVOLRES), + regmap_reg_range(STA350_NSHAPE, STA350_MISC2), +}; + +static const struct regmap_range sta350_volatile_regs_range[] = { + regmap_reg_range(STA350_CFADDR2, STA350_CFUD), + regmap_reg_range(STA350_STATUS, STA350_STATUS), +}; + +static const struct regmap_access_table sta350_write_regs = { + .yes_ranges = sta350_write_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_write_regs_range), +}; + +static const struct regmap_access_table sta350_read_regs = { + .yes_ranges = sta350_read_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_read_regs_range), +}; + +static const struct regmap_access_table sta350_volatile_regs = { + .yes_ranges = sta350_volatile_regs_range, + .n_yes_ranges = ARRAY_SIZE(sta350_volatile_regs_range), +}; + +/* regulator power supply names */ +static const char * const sta350_supply_names[] = { + "vdd-dig", /* digital supply, 3.3V */ + "vdd-pll", /* pll supply, 3.3V */ + "vcc" /* power amp supply, 5V - 26V */ +}; + +/* codec private data */ +struct sta350_priv { + struct regmap *regmap; + struct regulator_bulk_data supplies[ARRAY_SIZE(sta350_supply_names)]; + struct sta350_platform_data *pdata; + + unsigned int mclk; + unsigned int format; + + u32 coef_shadow[STA350_COEF_COUNT]; + int shutdown; + + struct gpio_desc *gpiod_nreset; + struct gpio_desc *gpiod_power_down; + + struct mutex coeff_lock; +}; + +static const DECLARE_TLV_DB_SCALE(mvol_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(chvol_tlv, -7950, 50, 1); +static const DECLARE_TLV_DB_SCALE(tone_tlv, -1200, 200, 0); + +static const char * const sta350_drc_ac[] = { + "Anti-Clipping", "Dynamic Range Compression" +}; +static const char * const sta350_auto_gc_mode[] = { + "User", "AC no clipping", "AC limited clipping (10%)", + "DRC nighttime listening mode" +}; +static const char * const sta350_auto_xo_mode[] = { + "User", "80Hz", "100Hz", "120Hz", "140Hz", "160Hz", "180Hz", + "200Hz", "220Hz", "240Hz", "260Hz", "280Hz", "300Hz", "320Hz", + "340Hz", "360Hz" +}; +static const char * const sta350_binary_output[] = { + "FFX 3-state output - normal operation", "Binary output" +}; +static const char * const sta350_limiter_select[] = { + "Limiter Disabled", "Limiter #1", "Limiter #2" +}; +static const char * const sta350_limiter_attack_rate[] = { + "3.1584", "2.7072", "2.2560", "1.8048", "1.3536", "0.9024", + "0.4512", "0.2256", "0.1504", "0.1123", "0.0902", "0.0752", + "0.0645", "0.0564", "0.0501", "0.0451" +}; +static const char * const sta350_limiter_release_rate[] = { + "0.5116", "0.1370", "0.0744", "0.0499", "0.0360", "0.0299", + "0.0264", "0.0208", "0.0198", "0.0172", "0.0147", "0.0137", + "0.0134", "0.0117", "0.0110", "0.0104" +}; +static const char * const sta350_noise_shaper_type[] = { + "Third order", "Fourth order" +}; + +static DECLARE_TLV_DB_RANGE(sta350_limiter_ac_attack_tlv, + 0, 7, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(300, 100, 0), +); + +static DECLARE_TLV_DB_RANGE(sta350_limiter_ac_release_tlv, + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(-2900, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(-2000, 0, 0), + 3, 8, TLV_DB_SCALE_ITEM(-1400, 200, 0), + 8, 16, TLV_DB_SCALE_ITEM(-700, 100, 0), +); + +static DECLARE_TLV_DB_RANGE(sta350_limiter_drc_attack_tlv, + 0, 7, TLV_DB_SCALE_ITEM(-3100, 200, 0), + 8, 13, TLV_DB_SCALE_ITEM(-1600, 100, 0), + 14, 16, TLV_DB_SCALE_ITEM(-1000, 300, 0), +); + +static DECLARE_TLV_DB_RANGE(sta350_limiter_drc_release_tlv, + 0, 0, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 0), + 1, 2, TLV_DB_SCALE_ITEM(-3800, 200, 0), + 3, 4, TLV_DB_SCALE_ITEM(-3300, 200, 0), + 5, 12, TLV_DB_SCALE_ITEM(-3000, 200, 0), + 13, 16, TLV_DB_SCALE_ITEM(-1500, 300, 0), +); + +static SOC_ENUM_SINGLE_DECL(sta350_drc_ac_enum, + STA350_CONFD, STA350_CONFD_DRC_SHIFT, + sta350_drc_ac); +static SOC_ENUM_SINGLE_DECL(sta350_noise_shaper_enum, + STA350_CONFE, STA350_CONFE_NSBW_SHIFT, + sta350_noise_shaper_type); +static SOC_ENUM_SINGLE_DECL(sta350_auto_gc_enum, + STA350_AUTO1, STA350_AUTO1_AMGC_SHIFT, + sta350_auto_gc_mode); +static SOC_ENUM_SINGLE_DECL(sta350_auto_xo_enum, + STA350_AUTO2, STA350_AUTO2_XO_SHIFT, + sta350_auto_xo_mode); +static SOC_ENUM_SINGLE_DECL(sta350_binary_output_ch1_enum, + STA350_C1CFG, STA350_CxCFG_BO_SHIFT, + sta350_binary_output); +static SOC_ENUM_SINGLE_DECL(sta350_binary_output_ch2_enum, + STA350_C2CFG, STA350_CxCFG_BO_SHIFT, + sta350_binary_output); +static SOC_ENUM_SINGLE_DECL(sta350_binary_output_ch3_enum, + STA350_C3CFG, STA350_CxCFG_BO_SHIFT, + sta350_binary_output); +static SOC_ENUM_SINGLE_DECL(sta350_limiter_ch1_enum, + STA350_C1CFG, STA350_CxCFG_LS_SHIFT, + sta350_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta350_limiter_ch2_enum, + STA350_C2CFG, STA350_CxCFG_LS_SHIFT, + sta350_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta350_limiter_ch3_enum, + STA350_C3CFG, STA350_CxCFG_LS_SHIFT, + sta350_limiter_select); +static SOC_ENUM_SINGLE_DECL(sta350_limiter1_attack_rate_enum, + STA350_L1AR, STA350_LxA_SHIFT, + sta350_limiter_attack_rate); +static SOC_ENUM_SINGLE_DECL(sta350_limiter2_attack_rate_enum, + STA350_L2AR, STA350_LxA_SHIFT, + sta350_limiter_attack_rate); +static SOC_ENUM_SINGLE_DECL(sta350_limiter1_release_rate_enum, + STA350_L1AR, STA350_LxR_SHIFT, + sta350_limiter_release_rate); +static SOC_ENUM_SINGLE_DECL(sta350_limiter2_release_rate_enum, + STA350_L2AR, STA350_LxR_SHIFT, + sta350_limiter_release_rate); + +/* + * byte array controls for setting biquad, mixer, scaling coefficients; + * for biquads all five coefficients need to be set in one go, + * mixer and pre/postscale coefs can be set individually; + * each coef is 24bit, the bytes are ordered in the same way + * as given in the STA350 data sheet (big endian; b1, b2, a1, a2, b0) + */ + +static int sta350_coefficient_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int numcoef = kcontrol->private_value >> 16; + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; + uinfo->count = 3 * numcoef; + return 0; +} + +static int sta350_coefficient_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int numcoef = kcontrol->private_value >> 16; + int index = kcontrol->private_value & 0xffff; + unsigned int cfud, val; + int i, ret = 0; + + mutex_lock(&sta350->coeff_lock); + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + /* + * chip documentation does not say if the bits are self clearing, + * so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + + regmap_write(sta350->regmap, STA350_CFADDR2, index); + if (numcoef == 1) { + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x04); + } else if (numcoef == 5) { + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x08); + } else { + ret = -EINVAL; + goto exit_unlock; + } + + for (i = 0; i < 3 * numcoef; i++) { + regmap_read(sta350->regmap, STA350_B1CF1 + i, &val); + ucontrol->value.bytes.data[i] = val; + } + +exit_unlock: + mutex_unlock(&sta350->coeff_lock); + + return ret; +} + +static int sta350_coefficient_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int numcoef = kcontrol->private_value >> 16; + int index = kcontrol->private_value & 0xffff; + unsigned int cfud; + int i; + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + /* + * chip documentation does not say if the bits are self clearing, + * so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + + regmap_write(sta350->regmap, STA350_CFADDR2, index); + for (i = 0; i < numcoef && (index + i < STA350_COEF_COUNT); i++) + sta350->coef_shadow[index + i] = + (ucontrol->value.bytes.data[3 * i] << 16) + | (ucontrol->value.bytes.data[3 * i + 1] << 8) + | (ucontrol->value.bytes.data[3 * i + 2]); + for (i = 0; i < 3 * numcoef; i++) + regmap_write(sta350->regmap, STA350_B1CF1 + i, + ucontrol->value.bytes.data[i]); + if (numcoef == 1) + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x01); + else if (numcoef == 5) + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x02); + else + return -EINVAL; + + return 0; +} + +static int sta350_sync_coef_shadow(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int cfud; + int i; + + /* preserve reserved bits in STA350_CFUD */ + regmap_read(sta350->regmap, STA350_CFUD, &cfud); + cfud &= 0xf0; + + for (i = 0; i < STA350_COEF_COUNT; i++) { + regmap_write(sta350->regmap, STA350_CFADDR2, i); + regmap_write(sta350->regmap, STA350_B1CF1, + (sta350->coef_shadow[i] >> 16) & 0xff); + regmap_write(sta350->regmap, STA350_B1CF2, + (sta350->coef_shadow[i] >> 8) & 0xff); + regmap_write(sta350->regmap, STA350_B1CF3, + (sta350->coef_shadow[i]) & 0xff); + /* + * chip documentation does not say if the bits are + * self-clearing, so do it explicitly + */ + regmap_write(sta350->regmap, STA350_CFUD, cfud); + regmap_write(sta350->regmap, STA350_CFUD, cfud | 0x01); + } + return 0; +} + +static int sta350_cache_sync(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int mute; + int rc; + + /* mute during register sync */ + regmap_read(sta350->regmap, STA350_CFUD, &mute); + regmap_write(sta350->regmap, STA350_MMUTE, mute | STA350_MMUTE_MMUTE); + sta350_sync_coef_shadow(codec); + rc = regcache_sync(sta350->regmap); + regmap_write(sta350->regmap, STA350_MMUTE, mute); + return rc; +} + +#define SINGLE_COEF(xname, index) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = sta350_coefficient_info, \ + .get = sta350_coefficient_get,\ + .put = sta350_coefficient_put, \ + .private_value = index | (1 << 16) } + +#define BIQUAD_COEFS(xname, index) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = sta350_coefficient_info, \ + .get = sta350_coefficient_get,\ + .put = sta350_coefficient_put, \ + .private_value = index | (5 << 16) } + +static const struct snd_kcontrol_new sta350_snd_controls[] = { +SOC_SINGLE_TLV("Master Volume", STA350_MVOL, 0, 0xff, 1, mvol_tlv), +/* VOL */ +SOC_SINGLE_TLV("Ch1 Volume", STA350_C1VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch2 Volume", STA350_C2VOL, 0, 0xff, 1, chvol_tlv), +SOC_SINGLE_TLV("Ch3 Volume", STA350_C3VOL, 0, 0xff, 1, chvol_tlv), +/* CONFD */ +SOC_SINGLE("High Pass Filter Bypass Switch", + STA350_CONFD, STA350_CONFD_HPB_SHIFT, 1, 1), +SOC_SINGLE("De-emphasis Filter Switch", + STA350_CONFD, STA350_CONFD_DEMP_SHIFT, 1, 0), +SOC_SINGLE("DSP Bypass Switch", + STA350_CONFD, STA350_CONFD_DSPB_SHIFT, 1, 0), +SOC_SINGLE("Post-scale Link Switch", + STA350_CONFD, STA350_CONFD_PSL_SHIFT, 1, 0), +SOC_SINGLE("Biquad Coefficient Link Switch", + STA350_CONFD, STA350_CONFD_BQL_SHIFT, 1, 0), +SOC_ENUM("Compressor/Limiter Switch", sta350_drc_ac_enum), +SOC_ENUM("Noise Shaper Bandwidth", sta350_noise_shaper_enum), +SOC_SINGLE("Zero-detect Mute Enable Switch", + STA350_CONFD, STA350_CONFD_ZDE_SHIFT, 1, 0), +SOC_SINGLE("Submix Mode Switch", + STA350_CONFD, STA350_CONFD_SME_SHIFT, 1, 0), +/* CONFE */ +SOC_SINGLE("Zero Cross Switch", STA350_CONFE, STA350_CONFE_ZCE_SHIFT, 1, 0), +SOC_SINGLE("Soft Ramp Switch", STA350_CONFE, STA350_CONFE_SVE_SHIFT, 1, 0), +/* MUTE */ +SOC_SINGLE("Master Switch", STA350_MMUTE, STA350_MMUTE_MMUTE_SHIFT, 1, 1), +SOC_SINGLE("Ch1 Switch", STA350_MMUTE, STA350_MMUTE_C1M_SHIFT, 1, 1), +SOC_SINGLE("Ch2 Switch", STA350_MMUTE, STA350_MMUTE_C2M_SHIFT, 1, 1), +SOC_SINGLE("Ch3 Switch", STA350_MMUTE, STA350_MMUTE_C3M_SHIFT, 1, 1), +/* AUTOx */ +SOC_ENUM("Automode GC", sta350_auto_gc_enum), +SOC_ENUM("Automode XO", sta350_auto_xo_enum), +/* CxCFG */ +SOC_SINGLE("Ch1 Tone Control Bypass Switch", + STA350_C1CFG, STA350_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Tone Control Bypass Switch", + STA350_C2CFG, STA350_CxCFG_TCB_SHIFT, 1, 0), +SOC_SINGLE("Ch1 EQ Bypass Switch", + STA350_C1CFG, STA350_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 EQ Bypass Switch", + STA350_C2CFG, STA350_CxCFG_EQBP_SHIFT, 1, 0), +SOC_SINGLE("Ch1 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch2 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_SINGLE("Ch3 Master Volume Bypass Switch", + STA350_C1CFG, STA350_CxCFG_VBP_SHIFT, 1, 0), +SOC_ENUM("Ch1 Binary Output Select", sta350_binary_output_ch1_enum), +SOC_ENUM("Ch2 Binary Output Select", sta350_binary_output_ch2_enum), +SOC_ENUM("Ch3 Binary Output Select", sta350_binary_output_ch3_enum), +SOC_ENUM("Ch1 Limiter Select", sta350_limiter_ch1_enum), +SOC_ENUM("Ch2 Limiter Select", sta350_limiter_ch2_enum), +SOC_ENUM("Ch3 Limiter Select", sta350_limiter_ch3_enum), +/* TONE */ +SOC_SINGLE_RANGE_TLV("Bass Tone Control Volume", + STA350_TONE, STA350_TONE_BTC_SHIFT, 1, 13, 0, tone_tlv), +SOC_SINGLE_RANGE_TLV("Treble Tone Control Volume", + STA350_TONE, STA350_TONE_TTC_SHIFT, 1, 13, 0, tone_tlv), +SOC_ENUM("Limiter1 Attack Rate (dB/ms)", sta350_limiter1_attack_rate_enum), +SOC_ENUM("Limiter2 Attack Rate (dB/ms)", sta350_limiter2_attack_rate_enum), +SOC_ENUM("Limiter1 Release Rate (dB/ms)", sta350_limiter1_release_rate_enum), +SOC_ENUM("Limiter2 Release Rate (dB/ms)", sta350_limiter2_release_rate_enum), + +/* + * depending on mode, the attack/release thresholds have + * two different enum definitions; provide both + */ +SOC_SINGLE_TLV("Limiter1 Attack Threshold (AC Mode)", + STA350_L1ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (AC Mode)", + STA350_L2ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_ac_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (AC Mode)", + STA350_L1ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (AC Mode)", + STA350_L2ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_ac_release_tlv), +SOC_SINGLE_TLV("Limiter1 Attack Threshold (DRC Mode)", + STA350_L1ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter2 Attack Threshold (DRC Mode)", + STA350_L2ATRT, STA350_LxA_SHIFT, + 16, 0, sta350_limiter_drc_attack_tlv), +SOC_SINGLE_TLV("Limiter1 Release Threshold (DRC Mode)", + STA350_L1ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_drc_release_tlv), +SOC_SINGLE_TLV("Limiter2 Release Threshold (DRC Mode)", + STA350_L2ATRT, STA350_LxR_SHIFT, + 16, 0, sta350_limiter_drc_release_tlv), + +BIQUAD_COEFS("Ch1 - Biquad 1", 0), +BIQUAD_COEFS("Ch1 - Biquad 2", 5), +BIQUAD_COEFS("Ch1 - Biquad 3", 10), +BIQUAD_COEFS("Ch1 - Biquad 4", 15), +BIQUAD_COEFS("Ch2 - Biquad 1", 20), +BIQUAD_COEFS("Ch2 - Biquad 2", 25), +BIQUAD_COEFS("Ch2 - Biquad 3", 30), +BIQUAD_COEFS("Ch2 - Biquad 4", 35), +BIQUAD_COEFS("High-pass", 40), +BIQUAD_COEFS("Low-pass", 45), +SINGLE_COEF("Ch1 - Prescale", 50), +SINGLE_COEF("Ch2 - Prescale", 51), +SINGLE_COEF("Ch1 - Postscale", 52), +SINGLE_COEF("Ch2 - Postscale", 53), +SINGLE_COEF("Ch3 - Postscale", 54), +SINGLE_COEF("Thermal warning - Postscale", 55), +SINGLE_COEF("Ch1 - Mix 1", 56), +SINGLE_COEF("Ch1 - Mix 2", 57), +SINGLE_COEF("Ch2 - Mix 1", 58), +SINGLE_COEF("Ch2 - Mix 2", 59), +SINGLE_COEF("Ch3 - Mix 1", 60), +SINGLE_COEF("Ch3 - Mix 2", 61), +}; + +static const struct snd_soc_dapm_widget sta350_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_OUTPUT("LEFT"), +SND_SOC_DAPM_OUTPUT("RIGHT"), +SND_SOC_DAPM_OUTPUT("SUB"), +}; + +static const struct snd_soc_dapm_route sta350_dapm_routes[] = { + { "LEFT", NULL, "DAC" }, + { "RIGHT", NULL, "DAC" }, + { "SUB", NULL, "DAC" }, + { "DAC", NULL, "Playback" }, +}; + +/* MCLK interpolation ratio per fs */ +static struct { + int fs; + int ir; +} interpolation_ratios[] = { + { 32000, 0 }, + { 44100, 0 }, + { 48000, 0 }, + { 88200, 1 }, + { 96000, 1 }, + { 176400, 2 }, + { 192000, 2 }, +}; + +/* MCLK to fs clock ratios */ +static int mcs_ratio_table[3][6] = { + { 768, 512, 384, 256, 128, 576 }, + { 384, 256, 192, 128, 64, 0 }, + { 192, 128, 96, 64, 32, 0 }, +}; + +/** + * sta350_set_dai_sysclk - configure MCLK + * @codec_dai: the codec DAI + * @clk_id: the clock ID (ignored) + * @freq: the MCLK input frequency + * @dir: the clock direction (ignored) + * + * The value of MCLK is used to determine which sample rates are supported + * by the STA350, based on the mcs_ratio_table. + * + * This function must be called by the machine driver's 'startup' function, + * otherwise the list of supported sample rates will not be available in + * time for ALSA. + */ +static int sta350_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 sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + + dev_dbg(codec->dev, "mclk=%u\n", freq); + sta350->mclk = freq; + + return 0; +} + +/** + * sta350_set_dai_fmt - configure the codec for the selected audio format + * @codec_dai: the codec DAI + * @fmt: a SND_SOC_DAIFMT_x value indicating the data format + * + * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the + * codec accordingly. + */ +static int sta350_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + unsigned int confb = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + sta350->format = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + confb |= STA350_CONFB_C2IM; + break; + case SND_SOC_DAIFMT_NB_IF: + confb |= STA350_CONFB_C1IM; + break; + default: + return -EINVAL; + } + + return regmap_update_bits(sta350->regmap, STA350_CONFB, + STA350_CONFB_C1IM | STA350_CONFB_C2IM, confb); +} + +/** + * sta350_hw_params - program the STA350 with the given hardware parameters. + * @substream: the audio stream + * @params: the hardware parameters to set + * @dai: the SOC DAI (ignored) + * + * This function programs the hardware with the values provided. + * Specifically, the sample rate and the data format. + */ +static int sta350_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 sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int i, mcs = -EINVAL, ir = -EINVAL; + unsigned int confa, confb; + unsigned int rate, ratio; + int ret; + + if (!sta350->mclk) { + dev_err(codec->dev, + "sta350->mclk is unset. Unable to determine ratio\n"); + return -EIO; + } + + rate = params_rate(params); + ratio = sta350->mclk / rate; + dev_dbg(codec->dev, "rate: %u, ratio: %u\n", rate, ratio); + + for (i = 0; i < ARRAY_SIZE(interpolation_ratios); i++) { + if (interpolation_ratios[i].fs == rate) { + ir = interpolation_ratios[i].ir; + break; + } + } + + if (ir < 0) { + dev_err(codec->dev, "Unsupported samplerate: %u\n", rate); + return -EINVAL; + } + + for (i = 0; i < 6; i++) { + if (mcs_ratio_table[ir][i] == ratio) { + mcs = i; + break; + } + } + + if (mcs < 0) { + dev_err(codec->dev, "Unresolvable ratio: %u\n", ratio); + return -EINVAL; + } + + confa = (ir << STA350_CONFA_IR_SHIFT) | + (mcs << STA350_CONFA_MCS_SHIFT); + confb = 0; + + switch (params_width(params)) { + case 24: + dev_dbg(codec->dev, "24bit\n"); + /* fall through */ + case 32: + dev_dbg(codec->dev, "24bit or 32bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x1; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x2; + break; + } + + break; + case 20: + dev_dbg(codec->dev, "20bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x4; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x5; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0x6; + break; + } + + break; + case 18: + dev_dbg(codec->dev, "18bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x8; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0x9; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xa; + break; + } + + break; + case 16: + dev_dbg(codec->dev, "16bit\n"); + switch (sta350->format) { + case SND_SOC_DAIFMT_I2S: + confb |= 0x0; + break; + case SND_SOC_DAIFMT_LEFT_J: + confb |= 0xd; + break; + case SND_SOC_DAIFMT_RIGHT_J: + confb |= 0xe; + break; + } + + break; + default: + return -EINVAL; + } + + ret = regmap_update_bits(sta350->regmap, STA350_CONFA, + STA350_CONFA_MCS_MASK | STA350_CONFA_IR_MASK, + confa); + if (ret < 0) + return ret; + + ret = regmap_update_bits(sta350->regmap, STA350_CONFB, + STA350_CONFB_SAI_MASK | STA350_CONFB_SAIFB, + confb); + if (ret < 0) + return ret; + + return 0; +} + +static int sta350_startup_sequence(struct sta350_priv *sta350) +{ + if (sta350->gpiod_power_down) + gpiod_set_value(sta350->gpiod_power_down, 1); + + if (sta350->gpiod_nreset) { + gpiod_set_value(sta350->gpiod_nreset, 0); + mdelay(1); + gpiod_set_value(sta350->gpiod_nreset, 1); + mdelay(1); + } + + return 0; +} + +/** + * sta350_set_bias_level - DAPM callback + * @codec: the codec device + * @level: DAPM power level + * + * This is called by ALSA to put the codec into low power mode + * or to wake it up. If the codec is powered off completely + * all registers must be restored after power on. + */ +static int sta350_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + int ret; + + dev_dbg(codec->dev, "level = %d\n", level); + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + /* Full power on */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, + STA350_CONFF_PWDN | STA350_CONFF_EAPD); + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + ret = regulator_bulk_enable( + ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(codec->dev, + "Failed to enable supplies: %d\n", + ret); + return ret; + } + sta350_startup_sequence(sta350); + sta350_cache_sync(codec); + } + + /* Power down */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, + 0); + + break; + + case SND_SOC_BIAS_OFF: + /* The chip runs through the power down sequence for us */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_PWDN | STA350_CONFF_EAPD, 0); + + /* power down: low */ + if (sta350->gpiod_power_down) + gpiod_set_value(sta350->gpiod_power_down, 0); + + if (sta350->gpiod_nreset) + gpiod_set_value(sta350->gpiod_nreset, 0); + + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), + sta350->supplies); + break; + } + codec->dapm.bias_level = level; + return 0; +} + +static const struct snd_soc_dai_ops sta350_dai_ops = { + .hw_params = sta350_hw_params, + .set_sysclk = sta350_set_dai_sysclk, + .set_fmt = sta350_set_dai_fmt, +}; + +static struct snd_soc_dai_driver sta350_dai = { + .name = "sta350-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = STA350_RATES, + .formats = STA350_FORMATS, + }, + .ops = &sta350_dai_ops, +}; + +#ifdef CONFIG_PM +static int sta350_suspend(struct snd_soc_codec *codec) +{ + sta350_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int sta350_resume(struct snd_soc_codec *codec) +{ + sta350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} +#else +#define sta350_suspend NULL +#define sta350_resume NULL +#endif + +static int sta350_probe(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + struct sta350_platform_data *pdata = sta350->pdata; + int i, ret = 0, thermal = 0; + + ret = regulator_bulk_enable(ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + ret = sta350_startup_sequence(sta350); + if (ret < 0) { + dev_err(codec->dev, "Failed to startup device\n"); + return ret; + } + + /* CONFA */ + if (!pdata->thermal_warning_recovery) + thermal |= STA350_CONFA_TWAB; + if (!pdata->thermal_warning_adjustment) + thermal |= STA350_CONFA_TWRB; + if (!pdata->fault_detect_recovery) + thermal |= STA350_CONFA_FDRB; + regmap_update_bits(sta350->regmap, STA350_CONFA, + STA350_CONFA_TWAB | STA350_CONFA_TWRB | + STA350_CONFA_FDRB, + thermal); + + /* CONFC */ + regmap_update_bits(sta350->regmap, STA350_CONFC, + STA350_CONFC_OM_MASK, + pdata->ffx_power_output_mode + << STA350_CONFC_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_CONFC, + STA350_CONFC_CSZ_MASK, + pdata->drop_compensation_ns + << STA350_CONFC_CSZ_SHIFT); + regmap_update_bits(sta350->regmap, + STA350_CONFC, + STA350_CONFC_OCRB, + pdata->oc_warning_adjustment ? + STA350_CONFC_OCRB : 0); + + /* CONFE */ + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_MPCV, + pdata->max_power_use_mpcc ? + STA350_CONFE_MPCV : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_MPC, + pdata->max_power_correction ? + STA350_CONFE_MPC : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_AME, + pdata->am_reduction_mode ? + STA350_CONFE_AME : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_PWMS, + pdata->odd_pwm_speed_mode ? + STA350_CONFE_PWMS : 0); + regmap_update_bits(sta350->regmap, STA350_CONFE, + STA350_CONFE_DCCV, + pdata->distortion_compensation ? + STA350_CONFE_DCCV : 0); + /* CONFF */ + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_IDE, + pdata->invalid_input_detect_mute ? + STA350_CONFF_IDE : 0); + regmap_update_bits(sta350->regmap, STA350_CONFF, + STA350_CONFF_OCFG_MASK, + pdata->output_conf + << STA350_CONFF_OCFG_SHIFT); + + /* channel to output mapping */ + regmap_update_bits(sta350->regmap, STA350_C1CFG, + STA350_CxCFG_OM_MASK, + pdata->ch1_output_mapping + << STA350_CxCFG_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_C2CFG, + STA350_CxCFG_OM_MASK, + pdata->ch2_output_mapping + << STA350_CxCFG_OM_SHIFT); + regmap_update_bits(sta350->regmap, STA350_C3CFG, + STA350_CxCFG_OM_MASK, + pdata->ch3_output_mapping + << STA350_CxCFG_OM_SHIFT); + + /* initialize coefficient shadow RAM with reset values */ + for (i = 4; i <= 49; i += 5) + sta350->coef_shadow[i] = 0x400000; + for (i = 50; i <= 54; i++) + sta350->coef_shadow[i] = 0x7fffff; + sta350->coef_shadow[55] = 0x5a9df7; + sta350->coef_shadow[56] = 0x7fffff; + sta350->coef_shadow[59] = 0x7fffff; + sta350->coef_shadow[60] = 0x400000; + sta350->coef_shadow[61] = 0x400000; + + sta350_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + /* Bias level configuration will have done an extra enable */ + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), sta350->supplies); + + return 0; +} + +static int sta350_remove(struct snd_soc_codec *codec) +{ + struct sta350_priv *sta350 = snd_soc_codec_get_drvdata(codec); + + sta350_set_bias_level(codec, SND_SOC_BIAS_OFF); + regulator_bulk_disable(ARRAY_SIZE(sta350->supplies), sta350->supplies); + + return 0; +} + +static const struct snd_soc_codec_driver sta350_codec = { + .probe = sta350_probe, + .remove = sta350_remove, + .suspend = sta350_suspend, + .resume = sta350_resume, + .set_bias_level = sta350_set_bias_level, + .controls = sta350_snd_controls, + .num_controls = ARRAY_SIZE(sta350_snd_controls), + .dapm_widgets = sta350_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sta350_dapm_widgets), + .dapm_routes = sta350_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sta350_dapm_routes), +}; + +static const struct regmap_config sta350_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = STA350_MISC2, + .reg_defaults = sta350_regs, + .num_reg_defaults = ARRAY_SIZE(sta350_regs), + .cache_type = REGCACHE_RBTREE, + .wr_table = &sta350_write_regs, + .rd_table = &sta350_read_regs, + .volatile_table = &sta350_volatile_regs, +}; + +#ifdef CONFIG_OF +static const struct of_device_id st350_dt_ids[] = { + { .compatible = "st,sta350", }, + { } +}; +MODULE_DEVICE_TABLE(of, st350_dt_ids); + +static const char * const sta350_ffx_modes[] = { + [STA350_FFX_PM_DROP_COMP] = "drop-compensation", + [STA350_FFX_PM_TAPERED_COMP] = "tapered-compensation", + [STA350_FFX_PM_FULL_POWER] = "full-power-mode", + [STA350_FFX_PM_VARIABLE_DROP_COMP] = "variable-drop-compensation", +}; + +static int sta350_probe_dt(struct device *dev, struct sta350_priv *sta350) +{ + struct device_node *np = dev->of_node; + struct sta350_platform_data *pdata; + const char *ffx_power_mode; + u16 tmp; + + pdata = devm_kzalloc(dev, sizeof(*pdata), GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + of_property_read_u8(np, "st,output-conf", + &pdata->output_conf); + of_property_read_u8(np, "st,ch1-output-mapping", + &pdata->ch1_output_mapping); + of_property_read_u8(np, "st,ch2-output-mapping", + &pdata->ch2_output_mapping); + of_property_read_u8(np, "st,ch3-output-mapping", + &pdata->ch3_output_mapping); + + if (of_get_property(np, "st,thermal-warning-recovery", NULL)) + pdata->thermal_warning_recovery = 1; + if (of_get_property(np, "st,thermal-warning-adjustment", NULL)) + pdata->thermal_warning_adjustment = 1; + if (of_get_property(np, "st,fault-detect-recovery", NULL)) + pdata->fault_detect_recovery = 1; + + pdata->ffx_power_output_mode = STA350_FFX_PM_VARIABLE_DROP_COMP; + if (!of_property_read_string(np, "st,ffx-power-output-mode", + &ffx_power_mode)) { + int i, mode = -EINVAL; + + for (i = 0; i < ARRAY_SIZE(sta350_ffx_modes); i++) + if (!strcasecmp(ffx_power_mode, sta350_ffx_modes[i])) + mode = i; + + if (mode < 0) + dev_warn(dev, "Unsupported ffx output mode: %s\n", + ffx_power_mode); + else + pdata->ffx_power_output_mode = mode; + } + + tmp = 140; + of_property_read_u16(np, "st,drop-compensation-ns", &tmp); + pdata->drop_compensation_ns = clamp_t(u16, tmp, 0, 300) / 20; + + if (of_get_property(np, "st,overcurrent-warning-adjustment", NULL)) + pdata->oc_warning_adjustment = 1; + + /* CONFE */ + if (of_get_property(np, "st,max-power-use-mpcc", NULL)) + pdata->max_power_use_mpcc = 1; + + if (of_get_property(np, "st,max-power-correction", NULL)) + pdata->max_power_correction = 1; + + if (of_get_property(np, "st,am-reduction-mode", NULL)) + pdata->am_reduction_mode = 1; + + if (of_get_property(np, "st,odd-pwm-speed-mode", NULL)) + pdata->odd_pwm_speed_mode = 1; + + if (of_get_property(np, "st,distortion-compensation", NULL)) + pdata->distortion_compensation = 1; + + /* CONFF */ + if (of_get_property(np, "st,invalid-input-detect-mute", NULL)) + pdata->invalid_input_detect_mute = 1; + + sta350->pdata = pdata; + + return 0; +} +#endif + +static int sta350_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct sta350_priv *sta350; + int ret, i; + + sta350 = devm_kzalloc(dev, sizeof(struct sta350_priv), GFP_KERNEL); + if (!sta350) + return -ENOMEM; + + mutex_init(&sta350->coeff_lock); + sta350->pdata = dev_get_platdata(dev); + +#ifdef CONFIG_OF + if (dev->of_node) { + ret = sta350_probe_dt(dev, sta350); + if (ret < 0) + return ret; + } +#endif + + /* GPIOs */ + sta350->gpiod_nreset = devm_gpiod_get(dev, "reset"); + if (IS_ERR(sta350->gpiod_nreset)) { + ret = PTR_ERR(sta350->gpiod_nreset); + if (ret != -ENOENT && ret != -ENOSYS) + return ret; + + sta350->gpiod_nreset = NULL; + } else { + gpiod_direction_output(sta350->gpiod_nreset, 0); + } + + sta350->gpiod_power_down = devm_gpiod_get(dev, "power-down"); + if (IS_ERR(sta350->gpiod_power_down)) { + ret = PTR_ERR(sta350->gpiod_power_down); + if (ret != -ENOENT && ret != -ENOSYS) + return ret; + + sta350->gpiod_power_down = NULL; + } else { + gpiod_direction_output(sta350->gpiod_power_down, 0); + } + + /* regulators */ + for (i = 0; i < ARRAY_SIZE(sta350->supplies); i++) + sta350->supplies[i].supply = sta350_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(sta350->supplies), + sta350->supplies); + if (ret < 0) { + dev_err(dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + sta350->regmap = devm_regmap_init_i2c(i2c, &sta350_regmap); + if (IS_ERR(sta350->regmap)) { + ret = PTR_ERR(sta350->regmap); + dev_err(dev, "Failed to init regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(i2c, sta350); + + ret = snd_soc_register_codec(dev, &sta350_codec, &sta350_dai, 1); + if (ret < 0) + dev_err(dev, "Failed to register codec (%d)\n", ret); + + return ret; +} + +static int sta350_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id sta350_i2c_id[] = { + { "sta350", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, sta350_i2c_id); + +static struct i2c_driver sta350_i2c_driver = { + .driver = { + .name = "sta350", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(st350_dt_ids), + }, + .probe = sta350_i2c_probe, + .remove = sta350_i2c_remove, + .id_table = sta350_i2c_id, +}; + +module_i2c_driver(sta350_i2c_driver); + +MODULE_DESCRIPTION("ASoC STA350 driver"); +MODULE_AUTHOR("Sven Brandau <info(a)brandau.biz>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/sta350.h b/sound/soc/codecs/sta350.h new file mode 100644 index 0000000..c3248f0 --- /dev/null +++ b/sound/soc/codecs/sta350.h @@ -0,0 +1,228 @@ +/* + * Codec driver for ST STA350 2.1-channel high-efficiency digital audio system + * + * Copyright: 2011 Raumfeld GmbH + * Author: Sven Brandau <info(a)brandau.biz> + * + * based on code from: + * Raumfeld GmbH + * Johannes Stezenbach <js(a)sig21.net> + * Wolfson Microelectronics PLC. + * Mark Brown <broonie(a)opensource.wolfsonmicro.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 _ASOC_STA_350_H +#define _ASOC_STA_350_H + +/* STA50 register addresses */ + +#define STA350_REGISTER_COUNT 0x4D +#define STA350_COEF_COUNT 62 + +#define STA350_CONFA 0x00 +#define STA350_CONFB 0x01 +#define STA350_CONFC 0x02 +#define STA350_CONFD 0x03 +#define STA350_CONFE 0x04 +#define STA350_CONFF 0x05 +#define STA350_MMUTE 0x06 +#define STA350_MVOL 0x07 +#define STA350_C1VOL 0x08 +#define STA350_C2VOL 0x09 +#define STA350_C3VOL 0x0a +#define STA350_AUTO1 0x0b +#define STA350_AUTO2 0x0c +#define STA350_AUTO3 0x0d +#define STA350_C1CFG 0x0e +#define STA350_C2CFG 0x0f +#define STA350_C3CFG 0x10 +#define STA350_TONE 0x11 +#define STA350_L1AR 0x12 +#define STA350_L1ATRT 0x13 +#define STA350_L2AR 0x14 +#define STA350_L2ATRT 0x15 +#define STA350_CFADDR2 0x16 +#define STA350_B1CF1 0x17 +#define STA350_B1CF2 0x18 +#define STA350_B1CF3 0x19 +#define STA350_B2CF1 0x1a +#define STA350_B2CF2 0x1b +#define STA350_B2CF3 0x1c +#define STA350_A1CF1 0x1d +#define STA350_A1CF2 0x1e +#define STA350_A1CF3 0x1f +#define STA350_A2CF1 0x20 +#define STA350_A2CF2 0x21 +#define STA350_A2CF3 0x22 +#define STA350_B0CF1 0x23 +#define STA350_B0CF2 0x24 +#define STA350_B0CF3 0x25 +#define STA350_CFUD 0x26 +#define STA350_MPCC1 0x27 +#define STA350_MPCC2 0x28 +#define STA350_DCC1 0x29 +#define STA350_DCC2 0x2a +#define STA350_FDRC1 0x2b +#define STA350_FDRC2 0x2c +#define STA350_STATUS 0x2d +/* reserved: 0x2d - 0x30 */ +#define STA350_EQCFG 0x31 +#define STA350_EATH1 0x32 +#define STA350_ERTH1 0x33 +#define STA350_EATH2 0x34 +#define STA350_ERTH2 0x35 +#define STA350_CONFX 0x36 +#define STA350_SVCA 0x37 +#define STA350_SVCB 0x38 +#define STA350_RMS0A 0x39 +#define STA350_RMS0B 0x3a +#define STA350_RMS0C 0x3b +#define STA350_RMS1A 0x3c +#define STA350_RMS1B 0x3d +#define STA350_RMS1C 0x3e +#define STA350_EVOLRES 0x3f +/* reserved: 0x40 - 0x47 */ +#define STA350_NSHAPE 0x48 +#define STA350_CTXB4B1 0x49 +#define STA350_CTXB7B5 0x4a +#define STA350_MISC1 0x4b +#define STA350_MISC2 0x4c + +/* 0x00 CONFA */ +#define STA350_CONFA_MCS_MASK 0x03 +#define STA350_CONFA_MCS_SHIFT 0 +#define STA350_CONFA_IR_MASK 0x18 +#define STA350_CONFA_IR_SHIFT 3 +#define STA350_CONFA_TWRB BIT(5) +#define STA350_CONFA_TWAB BIT(6) +#define STA350_CONFA_FDRB BIT(7) + +/* 0x01 CONFB */ +#define STA350_CONFB_SAI_MASK 0x0f +#define STA350_CONFB_SAI_SHIFT 0 +#define STA350_CONFB_SAIFB BIT(4) +#define STA350_CONFB_DSCKE BIT(5) +#define STA350_CONFB_C1IM BIT(6) +#define STA350_CONFB_C2IM BIT(7) + +/* 0x02 CONFC */ +#define STA350_CONFC_OM_MASK 0x03 +#define STA350_CONFC_OM_SHIFT 0 +#define STA350_CONFC_CSZ_MASK 0x3c +#define STA350_CONFC_CSZ_SHIFT 2 +#define STA350_CONFC_OCRB BIT(7) + +/* 0x03 CONFD */ +#define STA350_CONFD_HPB_SHIFT 0 +#define STA350_CONFD_DEMP_SHIFT 1 +#define STA350_CONFD_DSPB_SHIFT 2 +#define STA350_CONFD_PSL_SHIFT 3 +#define STA350_CONFD_BQL_SHIFT 4 +#define STA350_CONFD_DRC_SHIFT 5 +#define STA350_CONFD_ZDE_SHIFT 6 +#define STA350_CONFD_SME_SHIFT 7 + +/* 0x04 CONFE */ +#define STA350_CONFE_MPCV BIT(0) +#define STA350_CONFE_MPCV_SHIFT 0 +#define STA350_CONFE_MPC BIT(1) +#define STA350_CONFE_MPC_SHIFT 1 +#define STA350_CONFE_NSBW BIT(2) +#define STA350_CONFE_NSBW_SHIFT 2 +#define STA350_CONFE_AME BIT(3) +#define STA350_CONFE_AME_SHIFT 3 +#define STA350_CONFE_PWMS BIT(4) +#define STA350_CONFE_PWMS_SHIFT 4 +#define STA350_CONFE_DCCV BIT(5) +#define STA350_CONFE_DCCV_SHIFT 5 +#define STA350_CONFE_ZCE BIT(6) +#define STA350_CONFE_ZCE_SHIFT 6 +#define STA350_CONFE_SVE BIT(7) +#define STA350_CONFE_SVE_SHIFT 7 + +/* 0x05 CONFF */ +#define STA350_CONFF_OCFG_MASK 0x03 +#define STA350_CONFF_OCFG_SHIFT 0 +#define STA350_CONFF_IDE BIT(2) +#define STA350_CONFF_BCLE BIT(3) +#define STA350_CONFF_LDTE BIT(4) +#define STA350_CONFF_ECLE BIT(5) +#define STA350_CONFF_PWDN BIT(6) +#define STA350_CONFF_EAPD BIT(7) + +/* 0x06 MMUTE */ +#define STA350_MMUTE_MMUTE 0x01 +#define STA350_MMUTE_MMUTE_SHIFT 0 +#define STA350_MMUTE_C1M 0x02 +#define STA350_MMUTE_C1M_SHIFT 1 +#define STA350_MMUTE_C2M 0x04 +#define STA350_MMUTE_C2M_SHIFT 2 +#define STA350_MMUTE_C3M 0x08 +#define STA350_MMUTE_C3M_SHIFT 3 +#define STA350_MMUTE_LOC_MASK 0xC0 +#define STA350_MMUTE_LOC_SHIFT 6 + +/* 0x0b AUTO1 */ +#define STA350_AUTO1_AMGC_MASK 0x30 +#define STA350_AUTO1_AMGC_SHIFT 4 + +/* 0x0c AUTO2 */ +#define STA350_AUTO2_AMAME 0x01 +#define STA350_AUTO2_AMAM_MASK 0x0e +#define STA350_AUTO2_AMAM_SHIFT 1 +#define STA350_AUTO2_XO_MASK 0xf0 +#define STA350_AUTO2_XO_SHIFT 4 + +/* 0x0d AUTO3 */ +#define STA350_AUTO3_PEQ_MASK 0x1f +#define STA350_AUTO3_PEQ_SHIFT 0 + +/* 0x0e 0x0f 0x10 CxCFG */ +#define STA350_CxCFG_TCB_SHIFT 0 +#define STA350_CxCFG_EQBP_SHIFT 1 +#define STA350_CxCFG_VBP_SHIFT 2 +#define STA350_CxCFG_BO_SHIFT 3 +#define STA350_CxCFG_LS_SHIFT 4 +#define STA350_CxCFG_OM_MASK 0xc0 +#define STA350_CxCFG_OM_SHIFT 6 + +/* 0x11 TONE */ +#define STA350_TONE_BTC_SHIFT 0 +#define STA350_TONE_TTC_SHIFT 4 + +/* 0x12 0x13 0x14 0x15 limiter attack/release */ +#define STA350_LxA_SHIFT 0 +#define STA350_LxR_SHIFT 4 + +/* 0x26 CFUD */ +#define STA350_CFUD_W1 0x01 +#define STA350_CFUD_WA 0x02 +#define STA350_CFUD_R1 0x04 +#define STA350_CFUD_RA 0x08 + + +/* biquad filter coefficient table offsets */ +#define STA350_C1_BQ_BASE 0 +#define STA350_C2_BQ_BASE 20 +#define STA350_CH_BQ_NUM 4 +#define STA350_BQ_NUM_COEF 5 +#define STA350_XO_HP_BQ_BASE 40 +#define STA350_XO_LP_BQ_BASE 45 +#define STA350_C1_PRESCALE 50 +#define STA350_C2_PRESCALE 51 +#define STA350_C1_POSTSCALE 52 +#define STA350_C2_POSTSCALE 53 +#define STA350_C3_POSTSCALE 54 +#define STA350_TW_POSTSCALE 55 +#define STA350_C1_MIX1 56 +#define STA350_C1_MIX2 57 +#define STA350_C2_MIX1 58 +#define STA350_C2_MIX2 59 +#define STA350_C3_MIX1 60 +#define STA350_C3_MIX2 61 + +#endif /* _ASOC_STA_350_H */ -- 1.8.5.3

2 1

[alsa-devel] [PATCH 0/6] ASoC: davinci-mcasp: Dynamic AFIFO and DMA configuration
by Peter Ujfalusi 02 Apr '14

02 Apr '14

Hi, Apart from the first patch which is a rebased resend of a previous patch this series will change the way McASP driver will configure the AFIFO and thus the DMA burst. The AFIFO and DMA configuration will be in one function so it is going to be easier to see the code and debug if needed. The main change is that instead of static tx/rx numevt (which is provided via DT or pdata) we are going to switch to dynamic configuration. With this setup we do not need to place any constraint on the period or buffer size since the AFIFO numevt and the corresponding DMA burst size will be picked according to period size, active serializers and the preferred numevt level. The only case when the code can fail if the period size in words can not be divided evenly by the active serializers, which is unlikely to happen. Regards, Peter --- Peter Ujfalusi (6): ASoC: davinci-mcasp: Assign the dma_data earlier in dai_probe callback ASoC: davinci-mcasp: Fix debug typo in davinci_mcasp_hw_params() ASoC: davinci-mcasp: Simplify and clean up the AFIFO configuration code ASoC: davinci-mcasp: Configure the AFIFO and DMA burst size at the same place ASoC: davinic-mcasp: Adopt the AFIFO/DMA configuration to the stream (dynamic depth) ASoC: davinci-mcasp: Fine tune and correct the DMA burst configuration sound/soc/davinci/davinci-mcasp.c | 145 +++++++++++++++++++++++--------------- sound/soc/davinci/davinci-mcasp.h | 1 + 2 files changed, 89 insertions(+), 57 deletions(-) -- 1.9.1

2 7

[alsa-devel] [PATCHv2 00/11] Remove set_cache_io entirely from ASoC probe
by Xiubo Li 02 Apr '14

02 Apr '14

Change in v2: - Add .get_regmap() support for struct snd_soc_codec_driver. Xiubo Li (11): ASoC: core: Move the default regmap I/O setting to snd_soc_register_codec() ASoc: 88pm860x: Remove the set_cache_io() entirely from ASoC probe. ASoc: cq93vc: Remove the set_cache_io() entirely from ASoC probe. ASoc: mc13783: Remove the set_cache_io() entirely from ASoC probe. ASoc: si476x: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm5102: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm5110: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm8350: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm8400: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm8994: Remove the set_cache_io() entirely from ASoC probe. ASoc: wm8997: Remove the set_cache_io() entirely from ASoC probe. include/sound/soc.h | 1 + sound/soc/codecs/88pm860x-codec.c | 12 ++++++++---- sound/soc/codecs/cq93vc.c | 10 ++++++++-- sound/soc/codecs/mc13783.c | 14 ++++++-------- sound/soc/codecs/si476x.c | 14 ++++++-------- sound/soc/codecs/wm5102.c | 12 ++++++++---- sound/soc/codecs/wm5110.c | 12 ++++++++---- sound/soc/codecs/wm8350.c | 10 ++++++++-- sound/soc/codecs/wm8400.c | 10 ++++++++-- sound/soc/codecs/wm8994.c | 10 ++++++++-- sound/soc/codecs/wm8997.c | 13 ++++++++----- sound/soc/soc-core.c | 28 ++++++++++++++++++---------- sound/soc/soc-io.c | 9 +++------ 13 files changed, 98 insertions(+), 57 deletions(-) -- 1.8.4

4 14

[alsa-devel] [PATCH 00/13] ASoC: Move IO and kcontrols to the component level
by Lars-Peter Clausen 02 Apr '14

02 Apr '14

Hi, This series is the first step towards full componentisation of the ASoC core. It moves both the IO abstraction layers within ASoC as well as the standard set of kcontrols to the component level. This for example means we can get rid of constructs like if (w->codec) snd_soc_read(....) else if(w->platform) snd_soc_platform_read(...) Moving the kcontrols to the component level means we can use the same implementation also for other non-CODEC components. E.g. there seems to be an increasing amount of CPU components that have basic signal processing and things like volume controls etc. whose register layout is similar to those used in CODECs. Currently each CPU component driver re-implements these controls by hand. The first two patches introduce two new helper functions which hide the actual implementation on how the CODEC or platform struct that register a control can be obtained from the control. This means that when the actual implementation is changed only the two helper functions need to be updated and not every single driver. The patches that follow that are just cleanups removing unused IO stuff and move all IO functions to soc-io.c. The next step is to make platforms also components. And then finally first the IO abstraction layers in ASoC are unified at the component level and then on top of that the kcontrol helpers are moved to the component level. The series depends on quite a few topic branches related to changes to the core and cleanups for individual drivers. It is probably best to place it on top of asoc-v3.15-2. The patch that moves the kcontrols to the component level also has a runtime dependency on the not yet applied patches that move the ams-delta and mfld_machine controls to the card level. - Lars Lars-Peter Clausen (13): ASoC: Add snd_soc_kcontrol_codec() helper function ASoC: Add snd_soc_kcontrol_platform() helper function ASoC: Prepare SOC_SINGLE_XR_SX controls for regmap ASoC: Move IO functions to soc-io.c ASoC: Drop ASoC level caching from hw_write/hw_read ASoC: Remove IO register modifier callbacks ASoC: Add helper function to cast component back to CODEC ASoC: Track which components have been registered with snd_soc_register_component() ASoC: Let snd_soc_platform subclass snd_soc_component ASoC: Move IO abstraction to the component level ASoC: Move standard kcontrol helpers to the component level ASoC: Remove snd_soc_update_bits_locked() ASoC: dapm: Rename soc_widget_update_bits_locked() to soc_widget_update_bits() include/sound/soc-dapm.h | 1 + include/sound/soc.h | 114 +++++++-- sound/soc/codecs/88pm860x-codec.c | 8 +- sound/soc/codecs/ab8500-codec.c | 12 +- sound/soc/codecs/adav80x.c | 4 +- sound/soc/codecs/ak4641.c | 4 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/cs4271.c | 4 +- sound/soc/codecs/cs42l51.c | 4 +- sound/soc/codecs/da7210.c | 4 +- sound/soc/codecs/da7213.c | 4 +- sound/soc/codecs/da732x.c | 4 +- sound/soc/codecs/da9055.c | 2 +- sound/soc/codecs/lm4857.c | 4 +- sound/soc/codecs/max9768.c | 4 +- sound/soc/codecs/max98088.c | 12 +- sound/soc/codecs/max98090.c | 4 +- sound/soc/codecs/max98095.c | 16 +- sound/soc/codecs/pcm1681.c | 4 +- sound/soc/codecs/rt5631.c | 4 +- sound/soc/codecs/sgtl5000.c | 4 +- sound/soc/codecs/sta32x.c | 4 +- sound/soc/codecs/tas5086.c | 4 +- sound/soc/codecs/tlv320aic23.c | 4 +- sound/soc/codecs/tlv320dac33.c | 4 +- sound/soc/codecs/twl4030.c | 10 +- sound/soc/codecs/twl6040.c | 8 +- sound/soc/codecs/wl1273.c | 12 +- sound/soc/codecs/wm2000.c | 8 +- sound/soc/codecs/wm8350.c | 4 +- sound/soc/codecs/wm8400.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8731.c | 4 +- sound/soc/codecs/wm8753.c | 4 +- sound/soc/codecs/wm8804.c | 4 +- sound/soc/codecs/wm8903.c | 4 +- sound/soc/codecs/wm8904.c | 14 +- sound/soc/codecs/wm8955.c | 4 +- sound/soc/codecs/wm8958-dsp2.c | 32 +-- sound/soc/codecs/wm8960.c | 4 +- sound/soc/codecs/wm8962.c | 8 +- sound/soc/codecs/wm8983.c | 4 +- sound/soc/codecs/wm8985.c | 4 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm8991.c | 2 +- sound/soc/codecs/wm8994.c | 10 +- sound/soc/codecs/wm8996.c | 4 +- sound/soc/codecs/wm9081.c | 4 +- sound/soc/codecs/wm_adsp.c | 4 +- sound/soc/codecs/wm_hubs.c | 2 +- sound/soc/intel/sst-haswell-pcm.c | 8 +- sound/soc/soc-cache.c | 2 - sound/soc/soc-core.c | 520 ++++++++++++++++---------------------- sound/soc/soc-dapm.c | 91 +------ sound/soc/soc-io.c | 220 +++++++++++++--- 55 files changed, 644 insertions(+), 592 deletions(-) -- 1.8.0

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[alsa-devel] [PATCH v8] MIDI driver for Behringer BCD2000 USB device
by Mario Kicherer 02 Apr '14

02 Apr '14

This patch adds initial support for the Behringer BCD2000 USB DJ controller. At the moment, only the MIDI part of the device is working, i.e. knobs, buttons and LEDs. I also plan to add support for the audio part, but I assume that this will require more effort than the rather simple MIDI interface. Progress can be tracked at https://github.com/anyc/snd-usb-bcd2000. Daniel or Clemens: could I get another ACK on this? Thank you both! Best regards, Mario Changes since v7: - replaced snd_card_create with snd_card_new Changes since v6: - applied more style improvements Changes since v5: - use kernel bitmap functions for devices_used Changes since v4: - devices_used as array to support arbitrary number of SNDRV_CARDS - removed unused array "enable" Changes since v3: - applied style and snd_printk changes as suggested by Daniel Mack Changes since v2: - applied more changes from Daniel Mack and Clemens Ladisch Changes since v1: - fixed the various code style issues, thanks to Daniel Mack and checkpatch.pl. Signed-off-by: Mario Kicherer <dev(a)kicherer.org> --- sound/usb/Kconfig | 13 ++ sound/usb/Makefile | 2 +- sound/usb/bcd2000/Makefile | 3 + sound/usb/bcd2000/bcd2000.c | 477 ++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 494 insertions(+), 1 deletion(-) create mode 100644 sound/usb/bcd2000/Makefile create mode 100644 sound/usb/bcd2000/bcd2000.c diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index e05a86b..d393153 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -147,5 +147,18 @@ config SND_USB_HIFACE To compile this driver as a module, choose M here: the module will be called snd-usb-hiface. +config SND_BCD2000 + tristate "Behringer BCD2000 MIDI driver" + select SND_RAWMIDI + help + Say Y here to include MIDI support for the Behringer BCD2000 DJ + controller. + + Audio support is still work-in-progress at + https://github.com/anyc/snd-usb-bcd2000 + + To compile this driver as a module, choose M here: the module + will be called snd-bcd2000. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index abe668f..2b92f0d 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ bcd2000/ diff --git a/sound/usb/bcd2000/Makefile b/sound/usb/bcd2000/Makefile new file mode 100644 index 0000000..f09ccc0 --- /dev/null +++ b/sound/usb/bcd2000/Makefile @@ -0,0 +1,3 @@ +snd-bcd2000-y := bcd2000.o + +obj-$(CONFIG_SND_BCD2000) += snd-bcd2000.o \ No newline at end of file diff --git a/sound/usb/bcd2000/bcd2000.c b/sound/usb/bcd2000/bcd2000.c new file mode 100644 index 0000000..9fda92d --- /dev/null +++ b/sound/usb/bcd2000/bcd2000.c @@ -0,0 +1,477 @@ +/* + * Behringer BCD2000 driver + * + * Copyright (C) 2014 Mario Kicherer (dev(a)kicherer.org) + * + * 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. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/module.h> +#include <linux/bitmap.h> +#include <linux/usb.h> +#include <linux/usb/audio.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/rawmidi.h> + +#define PREFIX "snd-bcd2000: " +#define BUFSIZE 64 + +static struct usb_device_id id_table[] = { + { USB_DEVICE(0x1397, 0x00bd) }, + { }, +}; + +static unsigned char device_cmd_prefix[] = {0x03, 0x00}; + +static unsigned char bcd2000_init_sequence[] = { + 0x07, 0x00, 0x00, 0x00, 0x78, 0x48, 0x1c, 0x81, + 0xc4, 0x00, 0x00, 0x00, 0x5e, 0x53, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff, 0x6c, 0xf3, 0x90, 0xff, + 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, + 0x18, 0xfa, 0x11, 0xff, 0x14, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0xf2, 0x34, 0x4a, 0xf7, + 0x18, 0xfa, 0x11, 0xff +}; + +struct bcd2000 { + struct usb_device *dev; + struct snd_card *card; + struct usb_interface *intf; + int card_index; + + int midi_out_active; + struct snd_rawmidi *rmidi; + struct snd_rawmidi_substream *midi_receive_substream; + struct snd_rawmidi_substream *midi_out_substream; + + unsigned char midi_in_buf[BUFSIZE]; + unsigned char midi_out_buf[BUFSIZE]; + + struct urb *midi_out_urb; + struct urb *midi_in_urb; + + struct usb_anchor anchor; +}; + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; + +static DEFINE_MUTEX(devices_mutex); +DECLARE_BITMAP(devices_used, SNDRV_CARDS); +static struct usb_driver bcd2000_driver; + +#ifdef CONFIG_SND_DEBUG +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) +{ + print_hex_dump(KERN_DEBUG, prefix, + DUMP_PREFIX_NONE, 16, 1, + buf, len, false); +} +#else +static void bcd2000_dump_buffer(const char *prefix, const char *buf, int len) {} +#endif + +static int bcd2000_midi_input_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_input_close(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +/* register midi substream */ +static void bcd2000_midi_input_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + bcd2k->midi_receive_substream = up ? substream : NULL; +} + +static void bcd2000_midi_handle_input(struct bcd2000 *bcd2k, + const unsigned char *buf, unsigned int buf_len) +{ + unsigned int payload_length, tocopy; + struct snd_rawmidi_substream *midi_receive_substream; + + midi_receive_substream = ACCESS_ONCE(bcd2k->midi_receive_substream); + if (!midi_receive_substream) + return; + + bcd2000_dump_buffer(PREFIX "received from device: ", buf, buf_len); + + if (buf_len < 2) + return; + + payload_length = buf[0]; + + /* ignore packets without payload */ + if (payload_length == 0) + return; + + tocopy = min(payload_length, buf_len-1); + + bcd2000_dump_buffer(PREFIX "sending to userspace: ", + &buf[1], tocopy); + + snd_rawmidi_receive(midi_receive_substream, + &buf[1], tocopy); +} + +static void bcd2000_midi_send(struct bcd2000 *bcd2k) +{ + int len, ret; + struct snd_rawmidi_substream *midi_out_substream; + + BUILD_BUG_ON(sizeof(device_cmd_prefix) >= BUFSIZE); + + midi_out_substream = ACCESS_ONCE(bcd2k->midi_out_substream); + if (!midi_out_substream) + return; + + /* copy command prefix bytes */ + memcpy(bcd2k->midi_out_buf, device_cmd_prefix, + sizeof(device_cmd_prefix)); + + /* + * get MIDI packet and leave space for command prefix + * and payload length + */ + len = snd_rawmidi_transmit(midi_out_substream, + bcd2k->midi_out_buf + 3, BUFSIZE - 3); + + if (len < 0) + dev_err(&bcd2k->dev->dev, "%s: snd_rawmidi_transmit error %d\n", + __func__, len); + + if (len <= 0) + return; + + /* set payload length */ + bcd2k->midi_out_buf[2] = len; + bcd2k->midi_out_urb->transfer_buffer_length = BUFSIZE; + + bcd2000_dump_buffer(PREFIX "sending to device: ", + bcd2k->midi_out_buf, len+3); + + /* send packet to the BCD2000 */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s (%p): usb_submit_urb() failed, ret=%d, len=%d\n", + __func__, midi_out_substream, ret, len); + else + bcd2k->midi_out_active = 1; +} + +static int bcd2000_midi_output_open(struct snd_rawmidi_substream *substream) +{ + return 0; +} + +static int bcd2000_midi_output_close(struct snd_rawmidi_substream *substream) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (bcd2k->midi_out_active) { + usb_kill_urb(bcd2k->midi_out_urb); + bcd2k->midi_out_active = 0; + } + + return 0; +} + +/* register midi substream */ +static void bcd2000_midi_output_trigger(struct snd_rawmidi_substream *substream, + int up) +{ + struct bcd2000 *bcd2k = substream->rmidi->private_data; + + if (up) { + bcd2k->midi_out_substream = substream; + /* check if there is data userspace wants to send */ + if (!bcd2k->midi_out_active) + bcd2000_midi_send(bcd2k); + } else { + bcd2k->midi_out_substream = NULL; + } +} + +static void bcd2000_output_complete(struct urb *urb) +{ + struct bcd2000 *bcd2k = urb->context; + + bcd2k->midi_out_active = 0; + + if (urb->status) + dev_warn(&urb->dev->dev, + PREFIX "output urb->status: %d\n", urb->status); + + if (urb->status == -ESHUTDOWN) + return; + + /* check if there is more data userspace wants to send */ + bcd2000_midi_send(bcd2k); +} + +static void bcd2000_input_complete(struct urb *urb) +{ + int ret; + struct bcd2000 *bcd2k = urb->context; + + if (urb->status) + dev_warn(&urb->dev->dev, + PREFIX "input urb->status: %i\n", urb->status); + + if (!bcd2k || urb->status == -ESHUTDOWN) + return; + + if (urb->actual_length > 0) + bcd2000_midi_handle_input(bcd2k, urb->transfer_buffer, + urb->actual_length); + + /* acknowledge received packet */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_ATOMIC); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() failed, ret=%d\n", + __func__, ret); +} + +static struct snd_rawmidi_ops bcd2000_midi_output = { + .open = bcd2000_midi_output_open, + .close = bcd2000_midi_output_close, + .trigger = bcd2000_midi_output_trigger, +}; + +static struct snd_rawmidi_ops bcd2000_midi_input = { + .open = bcd2000_midi_input_open, + .close = bcd2000_midi_input_close, + .trigger = bcd2000_midi_input_trigger, +}; + +static void bcd2000_init_device(struct bcd2000 *bcd2k) +{ + int ret; + + init_usb_anchor(&bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_out_urb, &bcd2k->anchor); + usb_anchor_urb(bcd2k->midi_in_urb, &bcd2k->anchor); + + /* copy init sequence into buffer */ + memcpy(bcd2k->midi_out_buf, bcd2000_init_sequence, 52); + bcd2k->midi_out_urb->transfer_buffer_length = 52; + + /* submit sequence */ + ret = usb_submit_urb(bcd2k->midi_out_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() out failed, ret=%d: ", + __func__, ret); + else + bcd2k->midi_out_active = 1; + + /* send empty packet to enable button and controller events */ + ret = usb_submit_urb(bcd2k->midi_in_urb, GFP_KERNEL); + if (ret < 0) + dev_err(&bcd2k->dev->dev, PREFIX + "%s: usb_submit_urb() in failed, ret=%d: ", + __func__, ret); + + /* ensure initialization is finished */ + usb_wait_anchor_empty_timeout(&bcd2k->anchor, 1000); +} + +static int bcd2000_init_midi(struct bcd2000 *bcd2k) +{ + int ret; + struct snd_rawmidi *rmidi; + + ret = snd_rawmidi_new(bcd2k->card, bcd2k->card->shortname, 0, + 1, /* output */ + 1, /* input */ + &rmidi); + + if (ret < 0) + return ret; + + strlcpy(rmidi->name, bcd2k->card->shortname, sizeof(rmidi->name)); + + rmidi->info_flags = SNDRV_RAWMIDI_INFO_DUPLEX; + rmidi->private_data = bcd2k; + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_OUTPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, + &bcd2000_midi_output); + + rmidi->info_flags |= SNDRV_RAWMIDI_INFO_INPUT; + snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, + &bcd2000_midi_input); + + bcd2k->rmidi = rmidi; + + bcd2k->midi_in_urb = usb_alloc_urb(0, GFP_KERNEL); + bcd2k->midi_out_urb = usb_alloc_urb(0, GFP_KERNEL); + + if (!bcd2k->midi_in_urb || !bcd2k->midi_out_urb) { + dev_err(&bcd2k->dev->dev, PREFIX "usb_alloc_urb failed\n"); + return -ENOMEM; + } + + usb_fill_int_urb(bcd2k->midi_in_urb, bcd2k->dev, + usb_rcvintpipe(bcd2k->dev, 0x81), + bcd2k->midi_in_buf, BUFSIZE, + bcd2000_input_complete, bcd2k, 1); + + usb_fill_int_urb(bcd2k->midi_out_urb, bcd2k->dev, + usb_sndintpipe(bcd2k->dev, 0x1), + bcd2k->midi_out_buf, BUFSIZE, + bcd2000_output_complete, bcd2k, 1); + + bcd2000_init_device(bcd2k); + + return 0; +} + +static void bcd2000_free_usb_related_resources(struct bcd2000 *bcd2k, + struct usb_interface *interface) +{ + /* usb_kill_urb not necessary, urb is aborted automatically */ + + usb_free_urb(bcd2k->midi_out_urb); + usb_free_urb(bcd2k->midi_in_urb); + + if (bcd2k->intf) { + usb_set_intfdata(bcd2k->intf, NULL); + bcd2k->intf = NULL; + } +} + +static int bcd2000_probe(struct usb_interface *interface, + const struct usb_device_id *usb_id) +{ + struct snd_card *card; + struct bcd2000 *bcd2k; + unsigned int card_index; + char usb_path[32]; + int err; + + mutex_lock(&devices_mutex); + + for (card_index = 0; card_index < SNDRV_CARDS; ++card_index) + if (!test_bit(card_index, devices_used)) + break; + + if (card_index >= SNDRV_CARDS) { + mutex_unlock(&devices_mutex); + return -ENOENT; + } + + err = snd_card_new(&interface->dev, index[card_index], id[card_index], + THIS_MODULE, sizeof(*bcd2k), &card); + if (err < 0) { + mutex_unlock(&devices_mutex); + return err; + } + + bcd2k = card->private_data; + bcd2k->dev = interface_to_usbdev(interface); + bcd2k->card = card; + bcd2k->card_index = card_index; + bcd2k->intf = interface; + + snd_card_set_dev(card, &interface->dev); + + strncpy(card->driver, "snd-bcd2000", sizeof(card->driver)); + strncpy(card->shortname, "BCD2000", sizeof(card->shortname)); + usb_make_path(bcd2k->dev, usb_path, sizeof(usb_path)); + snprintf(bcd2k->card->longname, sizeof(bcd2k->card->longname), + "Behringer BCD2000 at %s", + usb_path); + + err = bcd2000_init_midi(bcd2k); + if (err < 0) + goto probe_error; + + err = snd_card_register(card); + if (err < 0) + goto probe_error; + + usb_set_intfdata(interface, bcd2k); + set_bit(card_index, devices_used); + + mutex_unlock(&devices_mutex); + return 0; + +probe_error: + dev_info(&bcd2k->dev->dev, PREFIX "error during probing"); + bcd2000_free_usb_related_resources(bcd2k, interface); + snd_card_free(card); + mutex_unlock(&devices_mutex); + return err; +} + +static void bcd2000_disconnect(struct usb_interface *interface) +{ + struct bcd2000 *bcd2k = usb_get_intfdata(interface); + + if (!bcd2k) + return; + + mutex_lock(&devices_mutex); + + /* make sure that userspace cannot create new requests */ + snd_card_disconnect(bcd2k->card); + + bcd2000_free_usb_related_resources(bcd2k, interface); + + clear_bit(bcd2k->card_index, devices_used); + + snd_card_free_when_closed(bcd2k->card); + + mutex_unlock(&devices_mutex); +} + +static struct usb_driver bcd2000_driver = { + .name = "snd-bcd2000", + .probe = bcd2000_probe, + .disconnect = bcd2000_disconnect, + .id_table = id_table, +}; + +static int __init bcd2000_init(void) +{ + int retval = 0; + + retval = usb_register(&bcd2000_driver); + if (retval) + pr_info(PREFIX "usb_register failed. Error: %d", retval); + return retval; +} + +static void __exit bcd2000_exit(void) +{ + usb_deregister(&bcd2000_driver); +} + +module_init(bcd2000_init); +module_exit(bcd2000_exit); + +MODULE_DEVICE_TABLE(usb, id_table); +MODULE_AUTHOR("Mario Kicherer, dev(a)kicherer.org"); +MODULE_DESCRIPTION("Behringer BCD2000 driver"); +MODULE_LICENSE("GPL"); -- 1.8.3.2

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Re: [alsa-devel] [PATCH] ASoC: DAPM: Add support for multi register mux
by Arun Shamanna Lakshmi 02 Apr '14

02 Apr '14

> -----Original Message----- > From: Lars-Peter Clausen [mailto:lars@metafoo.de] > Sent: Tuesday, April 01, 2014 12:48 AM > To: Arun Shamanna Lakshmi > Cc: lgirdwood(a)gmail.com; broonie(a)kernel.org; swarren(a)wwwdotorg.org; > perex(a)perex.cz; tiwai(a)suse.de; alsa- devel(a)alsa-project.org; > linux-kernel(a)vger.kernel.org; Songhee Baek > Subject: Re: [PATCH] ASoC: DAPM: Add support for multi register mux > > On 04/01/2014 08:21 AM, Arun Shamanna Lakshmi wrote: > > Modify soc_enum struct to handle pointers for reg and mask. Add dapm > > get and put APIs for multi register mux with one hot encoding. > > > > Signed-off-by: Arun Shamanna Lakshmi <aruns(a)nvidia.com> > > Signed-off-by: Songhee Baek <sbaek(a)nvidia.com> > > Looks in my opinion much better than the previous version :) Just a > few minor issues, comments inline > > > --- > > include/sound/soc-dapm.h | 10 ++++ > > include/sound/soc.h | 22 +++++-- > > sound/soc/soc-core.c | 12 ++-- > > sound/soc/soc-dapm.c | 143 > +++++++++++++++++++++++++++++++++++++++++----- > > 4 files changed, 162 insertions(+), 25 deletions(-) > > > > diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h > index > > ef78f56..983b0ab 100644 > > --- a/include/sound/soc-dapm.h > > +++ b/include/sound/soc-dapm.h > > @@ -305,6 +305,12 @@ struct device; > > .get = snd_soc_dapm_get_enum_double, \ > > .put = snd_soc_dapm_put_enum_double, \ > > .private_value = (unsigned long)&xenum } > > +#define SOC_DAPM_ENUM_WIDE(xname, xenum) \ > > maybe just call it ENUM_ONEHOT, since it doesn't actually have to be > more than one register. > > [...] > > diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index > > cd52d52..aba0094 100644 > > --- a/sound/soc/soc-core.c > > +++ b/sound/soc/soc-core.c > > @@ -2601,12 +2601,12 @@ int snd_soc_get_enum_double(struct > snd_kcontrol *kcontrol, > > unsigned int val, item; > > unsigned int reg_val; > > > > - reg_val = snd_soc_read(codec, e->reg); > > - val = (reg_val >> e->shift_l) & e->mask; > > + reg_val = snd_soc_read(codec, e->reg[0]); > > + val = (reg_val >> e->shift_l) & e->mask[0]; > > item = snd_soc_enum_val_to_item(e, val); > > ucontrol->value.enumerated.item[0] = item; > > if (e->shift_l != e->shift_r) { > > - val = (reg_val >> e->shift_l) & e->mask; > > + val = (reg_val >> e->shift_l) & e->mask[0]; > > item = snd_soc_enum_val_to_item(e, val); > > ucontrol->value.enumerated.item[1] = item; > > } > > @@ -2636,15 +2636,15 @@ int snd_soc_put_enum_double(struct > snd_kcontrol *kcontrol, > > if (item[0] >= e->items) > > return -EINVAL; > > val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; > > - mask = e->mask << e->shift_l; > > + mask = e->mask[0] << e->shift_l; > > if (e->shift_l != e->shift_r) { > > if (item[1] >= e->items) > > return -EINVAL; > > val |= snd_soc_enum_item_to_val(e, item[1]) << e- shift_r; > > - mask |= e->mask << e->shift_r; > > + mask |= e->mask[0] << e->shift_r; > > } > > > > - return snd_soc_update_bits_locked(codec, e->reg, mask, val); > > + return snd_soc_update_bits_locked(codec, e->reg[0], mask, val); > > } > > EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); > > > > diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index > > c8a780d..4d2b35c 100644 > > --- a/sound/soc/soc-dapm.c > > +++ b/sound/soc/soc-dapm.c > > @@ -514,9 +514,9 @@ static int dapm_connect_mux(struct > snd_soc_dapm_context *dapm, > > unsigned int val, item; > > int i; > > > > - if (e->reg != SND_SOC_NOPM) { > > - soc_widget_read(dest, e->reg, &val); > > - val = (val >> e->shift_l) & e->mask; > > + if (e->reg[0] != SND_SOC_NOPM) { > > + soc_widget_read(dest, e->reg[0], &val); > > + val = (val >> e->shift_l) & e->mask[0]; > > item = snd_soc_enum_val_to_item(e, val); > > This probably should handle the new enum type as well. You'll probably > need some kind of flag in the struct to distinguish between the two > enum types. Any suggestion on the flag name ? > > > } else { > > /* since a virtual mux has no backing registers to > [...] > > /** > > + * snd_soc_dapm_get_enum_wide - dapm semi enumerated multiple > > + registers > > What's a semi-enumerated register? > > > + * mixer get callback > > + * @kcontrol: mixer control > > + * @ucontrol: control element information > > + * > > + * Callback to get the value of a dapm semi enumerated multiple > > +register mixer > > + * control. > > + * > > + * semi enumerated multiple registers mixer: > > + * the mixer has multiple registers to set the enumerated items. > > +The enumerated > > + * items are referred as values. > > + * Can be used for handling bit field coded enumeration for example. > > + * > > + * Returns 0 for success. > > + */ > > +int snd_soc_dapm_get_enum_wide(struct snd_kcontrol *kcontrol, > > + struct snd_ctl_elem_value *ucontrol) { > > + struct snd_soc_codec *codec = > snd_soc_dapm_kcontrol_codec(kcontrol); > > + struct soc_enum *e = (struct soc_enum *)kcontrol- > >private_value; > > + unsigned int reg_val, val, bit_pos = 0, reg_idx; > > + > > + for (reg_idx = 0; reg_idx < e->num_regs; reg_idx++) { > > + reg_val = snd_soc_read(codec, e->reg[reg_idx]); > > + val = reg_val & e->mask[reg_idx]; > > + if (val != 0) { > > + bit_pos = ffs(val) + (e->reg_width * reg_idx); > > Should be __ffs. __ffs returns the bits zero-indexed and ffs one-indexed. > That will work better for cases where there is not additional value > table necessary, since it means bit 1 maps to value 0. > > > + break; > > + } > > + } > > + > > + ucontrol->value.enumerated.item[0] = > > + snd_soc_enum_val_to_item(e, bit_pos); > > + > > + return 0; > > +} > [...] > > +int snd_soc_dapm_put_enum_wide(struct snd_kcontrol *kcontrol, > > + struct snd_ctl_elem_value *ucontrol) { > > + struct snd_soc_codec *codec = > snd_soc_dapm_kcontrol_codec(kcontrol); > > + struct snd_soc_card *card = codec->card; > > + struct soc_enum *e = (struct soc_enum *)kcontrol- > >private_value; > > + unsigned int *item = ucontrol->value.enumerated.item; > > + unsigned int change = 0, reg_idx = 0, value, bit_pos; > > + struct snd_soc_dapm_update update; > > + int ret = 0, reg_val = 0, i; > > + > > + if (item[0] >= e->items) > > + return -EINVAL; > > + > > + value = snd_soc_enum_item_to_val(e, item[0]); > > + > > + if (value) { > > + /* get the register index and value to set */ > > + reg_idx = (value - 1) / e->reg_width; > > + bit_pos = (value - 1) % e->reg_width; > > Changing the ffs to __ffs also means you can drop the ' - 1' here. > > Also e->reg_width should be (codec->val_bytes * 8) and reg_width field > should be dropped from the enum struct. > > > + reg_val = BIT(bit_pos); > > + } > > + > > + for (i = 0; i < e->num_regs; i++) { > > + if (i == reg_idx) { > > + change = snd_soc_test_bits(codec, e->reg[i], > > + e->mask[i], > reg_val); > > + > > + } else { > > + /* accumulate the change to update the DAPM > path > > + when none is selected */ > > + change += snd_soc_test_bits(codec, e->reg[i], > > + e->mask[i], 0); > > change |= > > > + > > + /* clear the register when not selected */ > > + snd_soc_write(codec, e->reg[i], 0); > > I think this should happen as part of the DAPM update sequence like > you had earlier. Some special care should probably be take to make > sure that you de-select the previous mux input before selecting the > new one if the new one is in a different register than the previous one. I am not sure I follow this part. We are clearing the 'not selected' registers before we set the one we want. Do you want us to loop the logic of soc_dapm_mux_update_power for each register ? or do you want to change the dapm_update structure so that it takes all the regs, masks, and values together ? > > > + } > > + } > > + > > + mutex_lock_nested(&card->dapm_mutex, > SND_SOC_DAPM_CLASS_RUNTIME); > > + > [...]

3 8

[alsa-devel] [PATCH] ASoC: core: Only kmemdup binary control buffer if masking
by Charles Keepax 02 Apr '14

02 Apr '14

When writing a binary control we may apply a mask to the first register, as this requires modifying the data the buffer is duplicated, currently this is done for all binary control writes. As most binary controls don't use the mask facility and thus can freely use the original buffer, avoid the kmemdup for these cases. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 86 ++++++++++++++++++++++++------------------------- 1 files changed, 42 insertions(+), 44 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index caebd63..275bd71 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3229,67 +3229,65 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, len = params->num_regs * codec->val_bytes; - data = kmemdup(ucontrol->value.bytes.data, len, GFP_KERNEL | GFP_DMA); - if (!data) - return -ENOMEM; + if (!params->mask) + return regmap_raw_write(codec->control_data, params->base, + ucontrol->value.bytes.data, len); /* * If we've got a mask then we need to preserve the register * bits. We shouldn't modify the incoming data so take a * copy. */ - if (params->mask) { - ret = regmap_read(codec->control_data, params->base, &val); - if (ret != 0) - goto out; + data = kmemdup(ucontrol->value.bytes.data, len, + GFP_KERNEL | GFP_DMA); + if (!data) + return -ENOMEM; - val &= params->mask; + ret = regmap_read(codec->control_data, params->base, &val); + if (ret != 0) + goto out; - switch (codec->val_bytes) { - case 1: - ((u8 *)data)[0] &= ~params->mask; - ((u8 *)data)[0] |= val; - break; - case 2: - mask = ~params->mask; - ret = regmap_parse_val(codec->control_data, - &mask, &mask); - if (ret != 0) - goto out; + val &= params->mask; - ((u16 *)data)[0] &= mask; + switch (codec->val_bytes) { + case 1: + ((u8 *)data)[0] &= ~params->mask; + ((u8 *)data)[0] |= val; + break; + case 2: + mask = ~params->mask; + ret = regmap_parse_val(codec->control_data, &mask, &mask); + if (ret != 0) + goto out; - ret = regmap_parse_val(codec->control_data, - &val, &val); - if (ret != 0) - goto out; + ((u16 *)data)[0] &= mask; - ((u16 *)data)[0] |= val; - break; - case 4: - mask = ~params->mask; - ret = regmap_parse_val(codec->control_data, - &mask, &mask); - if (ret != 0) - goto out; + ret = regmap_parse_val(codec->control_data, &val, &val); + if (ret != 0) + goto out; - ((u32 *)data)[0] &= mask; + ((u16 *)data)[0] |= val; + break; + case 4: + mask = ~params->mask; + ret = regmap_parse_val(codec->control_data, &mask, &mask); + if (ret != 0) + goto out; - ret = regmap_parse_val(codec->control_data, - &val, &val); - if (ret != 0) - goto out; + ((u32 *)data)[0] &= mask; - ((u32 *)data)[0] |= val; - break; - default: - ret = -EINVAL; + ret = regmap_parse_val(codec->control_data, &val, &val); + if (ret != 0) goto out; - } + + ((u32 *)data)[0] |= val; + break; + default: + ret = -EINVAL; + goto out; } - ret = regmap_raw_write(codec->control_data, params->base, - data, len); + ret = regmap_raw_write(codec->control_data, params->base, data, len); out: kfree(data); -- 1.7.2.5

2 2

[alsa-devel] [PATCH] ASoC: core: Fail codec probe if we fail to add controls
by Nariman Poushin 02 Apr '14

02 Apr '14

Signed-off-by: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 35c0e71..7f9ae6c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1127,9 +1127,17 @@ static int soc_probe_codec(struct snd_soc_card *card, soc_init_codec_debugfs(codec); - if (driver->dapm_widgets) - snd_soc_dapm_new_controls(&codec->dapm, driver->dapm_widgets, - driver->num_dapm_widgets); + if (driver->dapm_widgets) { + ret = snd_soc_dapm_new_controls(&codec->dapm, + driver->dapm_widgets, + driver->num_dapm_widgets); + + if (ret != 0) { + dev_err(codec->dev, + "Failed to create new controls %d\n", ret); + goto err_probe; + } + } /* Create DAPM widgets for each DAI stream */ list_for_each_entry(dai, &codec->component.dai_list, list) { -- 1.8.5.3

2 2

Re: [alsa-devel] Invalid PCM pointer with CMI8738 and pcm.c buffer underuns
by Clemens Ladisch 02 Apr '14

02 Apr '14

Andoru wrote: > The last time you replied you said that it's a hardware error. If it > is that way then how come the card otherwise works flawlessly in > Windows, and this doesn't happen on Debian when watching videos or > with certain programs? It's still a hardware error, but, apparently, it happens only when the Linux driver accesses it in a certain way. Which programs work? Which do not? Regards, Clemens

3 3

[alsa-devel] [PATCH bisect 0/2] ASoC: fsl_sai: Overwrite trigger()
by Nicolin Chen 02 Apr '14

02 Apr '14

This series of patches are bisected from the preivous version so as to apply the PATCH-1 onto asoc-v3.15-4. * The patches are generated by using '-U2' because the default '-U3' would conflict the baseline without fsl_sai_isr patches. Nicolin Chen (2): ASoC: fsl_sai: Fix buggy configurations in trigger() ASoC: fsl_sai: Separately enable interrupts for Tx and Rx streams sound/soc/fsl/fsl_sai.c | 43 +++++++++++++++++++++++-------------------- sound/soc/fsl/fsl_sai.h | 11 +++++++++++ 2 files changed, 34 insertions(+), 20 deletions(-) -- 1.8.4

2 9

[alsa-devel] pulseaudio and rcar-h2 sound
by Ben Dooks 01 Apr '14

01 Apr '14

I am currently investigating an issue with the rcar-h2 audio driver and a hang when running pulse-audio. From what I can see the issue seems to be an interaction with the driver and the regmap code. The current issue is that whatever is happening causes CPU0 to stall, thus nothing seems to be able to continue. The following errors are detected by the rcu code: INFO: rcu_sched self-detected stall on CPU 0: (1 GPs behind) idle=415/2/0 softirq=3076/3077 (t=10500 jiffies g=520 c=519 q=51) I have been through the regmap code and the locking looks ok. I also enabled as much of the locking checks as I could and it does not produce any useful output as to why it would be deadlocking. I can run the system fine if I replace the rsnd_read, rsnd_write and rsnd_bset code with versions which do not directly use regmap. Does anyone have any experience with this, or any further ideas on how to continue debugging? -- Ben Dooks http://www.codethink.co.uk/ Senior Engineer Codethink - Providing Genius

1 0

Re: [alsa-devel] twl4030 latency update
by Peter Ujfalusi 01 Apr '14

01 Apr '14

Hi Leonardo, On 04/01/2014 12:41 PM, Leonardo Gabrielli wrote: > Dear Peter, > I actually managed to nearly halve the latency with McBSP2 and jackd with a > little trick: requesting 4 channels audio. Of course two will be zero, but the > FIFO will grow up more quick. Yes, this is expected. The FIFO is lost (or word) based. If you have mono audio, it is 1280 samples long, in stereo it is 640 samples with 4 channel it can hold up to 320 samples. > Outcome: > 4 channels, 44100, 64 input --> jack --> output takes 10.1ms latency, i.e.: > - 2 periods for input: 128 L/R frames > - 1280 FIFO words / 4 channels = 320 frames > Without the trick the latency was 17.4ms :) What you can also try is to reduce the max_rx_thres to let's say 10 samples (20 or 40 depending on the number of channels you are using). Keep the max_tx_thres as you have been using. > > CPU load as increased slightly in jackd, but of course any other jack client > will stay the same assumed that the 2 fake extra channels are unused. > > In this experiment sample size = 16-bit. If I try to default to 32bit nothing > changes (so I assume the FIFO words are 32-bit wide and can contain either > 16-bit zeropadded samples or 32, which I remember from the AM37xx tech guide > but I'm too lazy to check again ;) ) Yes it is like that. > Glitches may happen when some regular user tasks (scp, curl, ...) request some > cpu. I may try the new 3.14 kernel with SCHED_DEADLINE to see if jackd and > clients can really be not preempted by user tasks... > > Best > > Leonardo > > > > On 26/03/2014 13:51, Peter Ujfalusi wrote: >> On 03/26/2014 11:45 AM, Leonardo Gabrielli wrote: >>> On 26/03/2014 09:26, Peter Ujfalusi wrote: >>>> The McBSP2 FIFO will be always there. There's nothing can be done on that. >>>> The >>>> size on McBSP2 is 1280 words -> 640 stereo samples, ie ~29ms with 22050, >>>> 14.5ms with 44100. >>>> >>>> If you are staying in element mode this means that it is granted that the >>>> sample at the DMA pointer will out on the i2s line about the mentioned times. >>>> This is the delay caused by the FIFO itself. From where the rest is coming >>>> I'm >>>> not really sure. >>> BTW: I forgot to mention: the latency listed in my previous email is >>> input+output (i.e. I record pulses from the beagleboard input jack and the >>> delayed version to the beagleboard output jack). The twl4030 analog and >>> digital loopback features have been of course disabled, in order to get the >>> total latency due from A/D to D/A. >> This means that the McBSP latency in worst case is 1280 + selected rx >> threshold in words (so /2 in case of stereo.) If you lower the rx threshold >> you decrease the latency on the capture side. On the playback side there's >> nothing can be done. >> >>> So just to get confirm I understood the McBSP mechanism well: even though I >>> can transfer to/from DMA samples in bursts of <threshold> length, each sample >>> will always "travel along" the whole FIFO buffer length, (as if in a delay >>> line) and thus they will always have 640samples delay? >> On the playback side this is pretty much true. On capture side the threshold >> means that DMA will read from FIFO when threshold amount is available in it. >> >>> Would it be possible to workaround this, e.g. by putting 4-channel audio >>> frames instead of stereo frames in the FIFO (with 2 channels unused), in order >>> to fill up the FIFO more quickly and have less latency? Or is it pure craze? >> >From the FIFO McBSP takes data word by word. If you play stereo, you need to >> have stereo data in the FIFO. You can not skip two words with McBSP. >> >> The thing I tried for playback and did not worked AFAIR: >> In general the idea was to configure DMA to send threshold/channel to every >> request while configuring the McBSP threshold register to be 1280 - threshold. >> In case of threshold 80 (40 stereo samples) it would play out: >> transfer 40 samples to FIFO per DMA request >> assert the DMA request when we have space for 1260 (630 samples). The number >> is just a guess, keeping 10 samples in FIFO sounds safe enough >> This would keep the FIFO fill between 10 and 50 samples. >> But this does not work, I think McBSP is counting the received words also and >> deasserts the DMA request based on this count and not the FIFO level. >> >> Another thing which would be even more complicated is to play with the McBSP >> threshold runtime. With the same 40 sample: >> DMA is to transfer 40 samples per DMA requests. >> start >> 1. McBSP threshold to 80 >> 2. in dma interrupt callback McBSP threshold to 1260 >> 3. in McBSP warning interrupt (that we will be reaching the threshold soon) >> back to 80 >> 4. goto 2 >> >> If we could do the step between 3 and 4 within one sample time this might work >> but as soon as you are late the thing will fail. >> >> I know this is working in realtime systems like in DSPs and non linux >> systems... >> > -- Péter

1 0

[alsa-devel] [PATCH] ASoC: core: Fail probe if we fail to add dai widgets
by Nariman Poushin 01 Apr '14

01 Apr '14

Signed-off-by: Nariman Poushin <nariman(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b322cf2..5cf7107 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1132,8 +1132,15 @@ static int soc_probe_codec(struct snd_soc_card *card, driver->num_dapm_widgets); /* Create DAPM widgets for each DAI stream */ - list_for_each_entry(dai, &codec->component.dai_list, list) - snd_soc_dapm_new_dai_widgets(&codec->dapm, dai); + list_for_each_entry(dai, &codec->component.dai_list, list) { + ret = snd_soc_dapm_new_dai_widgets(&codec->dapm, dai); + + if (ret != 0) { + dev_err(codec->dev, + "Failed to create DAI widgets %d\n", ret); + goto err_probe; + } + } codec->dapm.idle_bias_off = driver->idle_bias_off; -- 1.8.5.3

2 1

[alsa-devel] [PATCH] ASoC: alc5632: fix uninit var in alc5632_probe()
by Stephen Warren 01 Apr '14

01 Apr '14

From: Stephen Warren <swarren(a)nvidia.com> alc5632_probe() returns ret, yet it is not initialized or set anywhere. This ends up causing the function to appear to fail, and audio not to work on the Toshiba AC100, with my compiler at least. This function used to set ret in all cases, but recent cleanup removed that. Cc: Xiubo Li <Li.Xiubo(a)freescale.com> Fixes: 5d6be5aa6bec ("ASoC: codec: Simplify ASoC probe code.") Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/alc5632.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index ec071a6306ef..3ee8d4e41a99 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1061,7 +1061,6 @@ static int alc5632_resume(struct snd_soc_codec *codec) static int alc5632_probe(struct snd_soc_codec *codec) { struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); - int ret; /* power on device */ alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1075,7 +1074,7 @@ static int alc5632_probe(struct snd_soc_codec *codec) return -EINVAL; } - return ret; + return 0; } /* power down chip */ -- 1.8.1.5

3 2

[alsa-devel] [PATCH] ASoC: DAPM: Add support for multi register mux
by Arun Shamanna Lakshmi 01 Apr '14

01 Apr '14

Modify soc_enum struct to handle pointers for reg and mask. Add dapm get and put APIs for multi register mux with one hot encoding. Signed-off-by: Arun Shamanna Lakshmi <aruns(a)nvidia.com> Signed-off-by: Songhee Baek <sbaek(a)nvidia.com> --- include/sound/soc-dapm.h | 10 ++++ include/sound/soc.h | 22 +++++-- sound/soc/soc-core.c | 12 ++-- sound/soc/soc-dapm.c | 143 +++++++++++++++++++++++++++++++++++++++++----- 4 files changed, 162 insertions(+), 25 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index ef78f56..983b0ab 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -305,6 +305,12 @@ struct device; .get = snd_soc_dapm_get_enum_double, \ .put = snd_soc_dapm_put_enum_double, \ .private_value = (unsigned long)&xenum } +#define SOC_DAPM_ENUM_WIDE(xname, xenum) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_enum_double, \ + .get = snd_soc_dapm_get_enum_wide, \ + .put = snd_soc_dapm_put_enum_wide, \ + .private_value = (unsigned long)&xenum } #define SOC_DAPM_ENUM_VIRT(xname, xenum) \ SOC_DAPM_ENUM(xname, xenum) #define SOC_DAPM_ENUM_EXT(xname, xenum, xget, xput) \ @@ -378,6 +384,10 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_dapm_get_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_dapm_put_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_soc_dapm_info_pin_switch(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_dapm_get_pin_switch(struct snd_kcontrol *kcontrol, diff --git a/include/sound/soc.h b/include/sound/soc.h index 5878410..5c274c4 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -177,18 +177,23 @@ {.reg = xreg, .min = xmin, .max = xmax, \ .platform_max = xmax} } #define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xitems, xtexts) \ -{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ +{ .reg = &(int){(xreg)}, .shift_l = xshift_l, .shift_r = xshift_r, \ .items = xitems, .texts = xtexts, \ - .mask = xitems ? roundup_pow_of_two(xitems) - 1 : 0} + .mask = &(unsigned int){(xitems ? roundup_pow_of_two(xitems) - 1 : 0)}, \ + .num_regs = 1 } #define SOC_ENUM_SINGLE(xreg, xshift, xitems, xtexts) \ SOC_ENUM_DOUBLE(xreg, xshift, xshift, xitems, xtexts) #define SOC_ENUM_SINGLE_EXT(xitems, xtexts) \ { .items = xitems, .texts = xtexts } #define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xitems, xtexts, xvalues) \ -{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ - .mask = xmask, .items = xitems, .texts = xtexts, .values = xvalues} +{ .reg = &(int){(xreg)}, .shift_l = xshift_l, .shift_r = xshift_r, \ + .mask = &(unsigned int){(xmask)}, .items = xitems, .texts = xtexts, \ + .values = xvalues, .num_regs = 1 } #define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xnitmes, xtexts, xvalues) \ SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xnitmes, xtexts, xvalues) +#define SOC_VALUE_ENUM_WIDE(xregs, xmasks, xnum_regs, xitems, xtexts, xvalues) \ +{ .reg = xregs, .mask = xmasks, .num_regs = xnum_regs, \ + .items = xitems, .texts = xtexts, .values = xvalues, .reg_width = 32 } #define SOC_ENUM_SINGLE_VIRT(xitems, xtexts) \ SOC_ENUM_SINGLE(SND_SOC_NOPM, 0, xitems, xtexts) #define SOC_ENUM(xname, xenum) \ @@ -293,6 +298,9 @@ #define SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift_l, xshift_r, xmask, xtexts, xvalues) \ const struct soc_enum name = SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, \ ARRAY_SIZE(xtexts), xtexts, xvalues) +#define SOC_VALUE_ENUM_WIDE_DECL(name, xregs, xmasks, xnum_regs, xtexts, xvalues) \ + const struct soc_enum name = SOC_VALUE_ENUM_WIDE(xregs, xmasks, xnum_regs, \ + ARRAY_SIZE(xtexts), xtexts, xvalues) #define SOC_VALUE_ENUM_SINGLE_DECL(name, xreg, xshift, xmask, xtexts, xvalues) \ SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) #define SOC_ENUM_SINGLE_VIRT_DECL(name, xtexts) \ @@ -1098,13 +1106,15 @@ struct soc_mreg_control { /* enumerated kcontrol */ struct soc_enum { - int reg; + int *reg; unsigned char shift_l; unsigned char shift_r; unsigned int items; - unsigned int mask; + unsigned int *mask; const char * const *texts; const unsigned int *values; + unsigned int reg_width; + unsigned int num_regs; }; /** diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cd52d52..aba0094 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2601,12 +2601,12 @@ int snd_soc_get_enum_double(struct snd_kcontrol *kcontrol, unsigned int val, item; unsigned int reg_val; - reg_val = snd_soc_read(codec, e->reg); - val = (reg_val >> e->shift_l) & e->mask; + reg_val = snd_soc_read(codec, e->reg[0]); + val = (reg_val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[0] = item; if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_l) & e->mask; + val = (reg_val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = item; } @@ -2636,15 +2636,15 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, if (item[0] >= e->items) return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; - mask = e->mask << e->shift_l; + mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] >= e->items) return -EINVAL; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_r; - mask |= e->mask << e->shift_r; + mask |= e->mask[0] << e->shift_r; } - return snd_soc_update_bits_locked(codec, e->reg, mask, val); + return snd_soc_update_bits_locked(codec, e->reg[0], mask, val); } EXPORT_SYMBOL_GPL(snd_soc_put_enum_double); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c8a780d..4d2b35c 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -514,9 +514,9 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, unsigned int val, item; int i; - if (e->reg != SND_SOC_NOPM) { - soc_widget_read(dest, e->reg, &val); - val = (val >> e->shift_l) & e->mask; + if (e->reg[0] != SND_SOC_NOPM) { + soc_widget_read(dest, e->reg[0], &val); + val = (val >> e->shift_l) & e->mask[0]; item = snd_soc_enum_val_to_item(e, val); } else { /* since a virtual mux has no backing registers to @@ -2903,15 +2903,15 @@ int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val; - if (e->reg != SND_SOC_NOPM) - reg_val = snd_soc_read(codec, e->reg); + if (e->reg[0] != SND_SOC_NOPM) + reg_val = snd_soc_read(codec, e->reg[0]); else reg_val = dapm_kcontrol_get_value(kcontrol); - val = (reg_val >> e->shift_l) & e->mask; + val = (reg_val >> e->shift_l) & e->mask[0]; ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val); if (e->shift_l != e->shift_r) { - val = (reg_val >> e->shift_r) & e->mask; + val = (reg_val >> e->shift_r) & e->mask[0]; val = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = val; } @@ -2945,25 +2945,25 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; - mask = e->mask << e->shift_l; + mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] > e->items) return -EINVAL; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_l; - mask |= e->mask << e->shift_r; + mask |= e->mask[0] << e->shift_r; } mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); - if (e->reg != SND_SOC_NOPM) - change = snd_soc_test_bits(codec, e->reg, mask, val); + if (e->reg[0] != SND_SOC_NOPM) + change = snd_soc_test_bits(codec, e->reg[0], mask, val); else change = dapm_kcontrol_set_value(kcontrol, val); if (change) { - if (e->reg != SND_SOC_NOPM) { + if (e->reg[0] != SND_SOC_NOPM) { update.kcontrol = kcontrol; - update.reg = e->reg; + update.reg = e->reg[0]; update.mask = mask; update.val = val; card->update = &update; @@ -2984,6 +2984,123 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_double); /** + * snd_soc_dapm_get_enum_wide - dapm semi enumerated multiple registers + * mixer get callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value of a dapm semi enumerated multiple register mixer + * control. + * + * semi enumerated multiple registers mixer: + * the mixer has multiple registers to set the enumerated items. The enumerated + * items are referred as values. + * Can be used for handling bit field coded enumeration for example. + * + * Returns 0 for success. + */ +int snd_soc_dapm_get_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int reg_val, val, bit_pos = 0, reg_idx; + + for (reg_idx = 0; reg_idx < e->num_regs; reg_idx++) { + reg_val = snd_soc_read(codec, e->reg[reg_idx]); + val = reg_val & e->mask[reg_idx]; + if (val != 0) { + bit_pos = ffs(val) + (e->reg_width * reg_idx); + break; + } + } + + ucontrol->value.enumerated.item[0] = + snd_soc_enum_val_to_item(e, bit_pos); + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_get_enum_wide); + +/** + * snd_soc_dapm_put_enum_wide - dapm semi enumerated multiple registers + * mixer put callback + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to put the value of a dapm semi enumerated multiple register mixer + * control. + * + * semi enumerated multiple registers mixer: + * the mixer has multiple registers to set the enumerated items. The enumerated + * items are referred as values. + * Can be used for handling bit field coded enumeration for example. + * + * Returns 0 for success. + */ +int snd_soc_dapm_put_enum_wide(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); + struct snd_soc_card *card = codec->card; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int change = 0, reg_idx = 0, value, bit_pos; + struct snd_soc_dapm_update update; + int ret = 0, reg_val = 0, i; + + if (item[0] >= e->items) + return -EINVAL; + + value = snd_soc_enum_item_to_val(e, item[0]); + + if (value) { + /* get the register index and value to set */ + reg_idx = (value - 1) / e->reg_width; + bit_pos = (value - 1) % e->reg_width; + reg_val = BIT(bit_pos); + } + + for (i = 0; i < e->num_regs; i++) { + if (i == reg_idx) { + change = snd_soc_test_bits(codec, e->reg[i], + e->mask[i], reg_val); + + } else { + /* accumulate the change to update the DAPM path + when none is selected */ + change += snd_soc_test_bits(codec, e->reg[i], + e->mask[i], 0); + + /* clear the register when not selected */ + snd_soc_write(codec, e->reg[i], 0); + } + } + + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + if (change) { + update.kcontrol = kcontrol; + update.reg = e->reg[reg_idx]; + update.mask = e->mask[reg_idx]; + update.val = reg_val; + card->update = &update; + + ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); + + card->update = NULL; + } + + mutex_unlock(&card->dapm_mutex); + + if (ret > 0) + soc_dpcm_runtime_update(card); + + return change; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_put_enum_wide); + +/** * snd_soc_dapm_info_pin_switch - Info for a pin switch * * @kcontrol: mixer control -- 1.7.9.5

2 1

Re: [alsa-devel] [PATCH] ASoC: Add support for multi register mux
by Songhee Baek 01 Apr '14

01 Apr '14

> -----Original Message----- > From: Songhee Baek > Sent: Friday, March 28, 2014 11:10 AM > To: 'Lars-Peter Clausen' > Cc: Arun Shamanna Lakshmi; 'lgirdwood(a)gmail.com'; 'broonie(a)kernel.org'; > 'swarren(a)wwwdotorg.org'; 'alsa-devel(a)alsa-project.org'; 'tiwai(a)suse.de'; > 'linux-kernel(a)vger.kernel.org' > Subject: RE: [alsa-devel] [PATCH] ASoC: Add support for multi register mux > > > > > On 03/26/2014 11:41 PM, Songhee Baek wrote: > > > >> On 03/26/2014 01:02 AM, Arun Shamanna Lakshmi wrote: > > > >> > > > >> The way you describe this it seems to me that a value array for > > > >> this kind of mux would look like. > > > >> > > > >> 0x00000000, 0x00000000, 0x00000001 0x00000000, 0x00000000, > > > >> 0x00000002 0x00000000, 0x00000000, 0x00000003 0x00000000, > > > >> 0x00000000, 0x00000004 0x00000000, 0x00000000, 0x00000008 ... > > > >> > > > >> That seems to be extremely tedious. If the MUX uses a one hot > > > >> encoding how about storing the index of the bit in the values > > > >> array and use (1 << value) when writing the value to the register? > > > > > > > > If we store the index of the bit, the value will be duplicated for > > > > each > > > registers inputs since register has 0 to 31bits to shift, then we > > > need to decode the index to interpret value for which registers to > > > set. If we need to interpret the decoded value of index, it is > > > better to have custom put/get function in our driver, isn't it? > > > > > > > > > > I'm not sure I understand. If you use (val / 32) to pick the > > > register and (val % > > > 32) to pick the bit in the register this should work just fine. > > > Maybe I'm missing something. Do you have a real world code example > > > of of the this type of enum is used? > > > > > > > I can use val/32 and val%32 for this multi register mux. > With this logic, put function would be easy however get function becomes tedious due to looping on each bit per register for 3 of them in our case. Rather, it would be easy to identify a unique MUX_OFFSET to distinguish between the 3 registers as shown in the code below. These get/put functions are updated from previous mail, now it works for multi register mux, please review these function whether I can add in current put/get function. #define MULTI_MUX_INPUT_OFFSET(n) (5 * n) int snd_soc_dapm_get_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int reg_val, val, reg_idx; if (e->reg[0] != SND_SOC_NOPM) { for (reg_idx = 0; reg_idx < e->num_regs; reg_idx++) { reg_val = snd_soc_read(codec, e->reg[reg_idx]); val = (reg_val >> e->shift_l) & e->mask[reg_idx]; if (val) { val += MULTI_MUX_INPUT_OFFSET(reg_idx); break; } } } else { reg_val = dapm_kcontrol_get_value(kcontrol); val = (reg_val >> e->shift_l) & e->mask[0]; } ucontrol->value.enumerated.item[0] = snd_soc_enum_val_to_item(e, val); if (e->shift_l != e->shift_r) { val = (reg_val >> e->shift_r) & e->mask[0]; val = snd_soc_enum_val_to_item(e, val); ucontrol->value.enumerated.item[1] = val; } return 0; } int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); struct snd_soc_card *card = codec->card; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int *item = ucontrol->value.enumerated.item; unsigned int change, i, value, val, update_idx = 0; unsigned int mask; struct snd_soc_dapm_update update; int ret = 0, reg_val; if (item[0] >= e->items) return -EINVAL; val = snd_soc_enum_item_to_val(e, item[0]) << e->shift_l; mask = e->mask[0] << e->shift_l; if (e->shift_l != e->shift_r) { if (item[1] > e->items) return -EINVAL; val |= snd_soc_enum_item_to_val(e, item[1]) << e->shift_l; mask |= e->mask[0] << e->shift_r; } if (e->num_regs < 2) { value = val; goto update_reg; } for (i = 0; i < e->num_regs; i++) { reg_val = val - MULTI_MUX_INPUT_OFFSET(i); /* checking reg_val is power of 2 : one-hot code */ /* if reg_val after subtract MULTI_MUX_INPUT_OFFSET is not power of 2, reg[i] should be zero */ if (reg_val & (reg_val - 1)) { /* clear the current input register */ snd_soc_write(codec, e->reg[i], 0); } else { /* reg_val is power of 2, store updated info */ value = reg_val; mask = e->mask[i]; update_idx = i; } } update_reg: mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); if (e->reg[update_idx] != SND_SOC_NOPM) change = snd_soc_test_bits(codec, e->reg[update_idx], mask, value); else change = dapm_kcontrol_set_value(kcontrol, value); if (change) { if (e->reg[update_idx] != SND_SOC_NOPM) { update.kcontrol = kcontrol; update.reg = e->reg[update_idx]; update.mask= mask; update.val = value; card->update = &update; } ret = soc_dapm_mux_update_power(card, kcontrol, item[0], e); card->update = NULL; } mutex_unlock(&card->dapm_mutex); if (ret > 0) soc_dpcm_runtime_update(card); return change; } > > > >>> - int reg; > > > >>> + int reg[SOC_ENUM_MAX_REGS]; > > > >>> unsigned char shift_l; > > > >>> unsigned char shift_r; > > > >>> unsigned int items; > > > >>> - unsigned int mask; > > > >>> + unsigned int mask[SOC_ENUM_MAX_REGS]; > > > >> > > > >> If you make mask and reg pointers instead of arrays this should > > > >> be much more flexible and not be limited to 3 registers. > We will make reg* and mask* instead of arrays and since we use the same structure, the plan is to share the get and put function code. Thanks. Songhee.

4 6

Re: [alsa-devel] [PATCH bisect 2/2] ASoC: fsl_sai: Separately enable interrupts for Tx and Rx streams
by Nicolin Chen 01 Apr '14

01 Apr '14

On Tue, Apr 01, 2014 at 11:48:16AM +0800, Wang Dongsheng-B40534 wrote: > > > > -----Original Message----- > > From: Nicolin Chen [mailto:Guangyu.Chen@freescale.com] > > Sent: Tuesday, April 01, 2014 11:14 AM > > To: Wang Dongsheng-B40534 > > Cc: broonie(a)kernel.org; alsa-devel(a)alsa-project.org; Xiubo Li-B47053; linuxppc- > > dev(a)lists.ozlabs.org; linux-kernel(a)vger.kernel.org; timur(a)tabi.org > > Subject: Re: [PATCH bisect 2/2] ASoC: fsl_sai: Separately enable interrupts for > > Tx and Rx streams > > > > On Tue, Apr 01, 2014 at 11:25:02AM +0800, Wang Dongsheng-B40534 wrote: > > > > Subject: [PATCH bisect 2/2] ASoC: fsl_sai: Separately enable interrupts for > > Tx > > > > and Rx streams > > > > > > > > We only enable one side interrupt for each stream since over/underrun > > > > on the opposite stream would be resulted from what we previously did, > > > > enabling TERE but remaining FRDE disabled, even though the xrun on the > > > > opposite direction will not break the current stream. > > > > > > > > Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> > > > > Acked-by: Xiubo Li <Li.Xiubo(a)freescale.com> > > > > --- > > > > sound/soc/fsl/fsl_sai.c | 8 ++++++-- > > > > sound/soc/fsl/fsl_sai.h | 1 + > > > > 2 files changed, 7 insertions(+), 2 deletions(-) > > > > > > > > diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c > > > > index bdfd497..d64c33f 100644 > > > > --- a/sound/soc/fsl/fsl_sai.c > > > > +++ b/sound/soc/fsl/fsl_sai.c > > > > @@ -397,4 +397,6 @@ static int fsl_sai_trigger(struct snd_pcm_substream > > > > *substream, int cmd, > > > > > > > > regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > > > + FSL_SAI_CSR_xIE_MASK, FSL_SAI_FLAGS); > > > > + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > > > FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE); > > > > break; > > > > @@ -404,4 +406,6 @@ static int fsl_sai_trigger(struct snd_pcm_substream > > > > *substream, int cmd, > > > > regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > > > FSL_SAI_CSR_FRDE, 0); > > > > + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > > > + FSL_SAI_CSR_xIE_MASK, 0); > > > > > > > > if (!(tcsr & FSL_SAI_CSR_FRDE || rcsr & FSL_SAI_CSR_FRDE)) { > > > > @@ -464,6 +468,6 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) > > > > struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); > > > > > > > > - regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, FSL_SAI_FLAGS); > > > > - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, FSL_SAI_FLAGS); > > > > + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); > > > > + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); > > > > > > Why are you remove this macro? Don't use magic number. > > > > It's pretty clear that the so-called magic number is to clear the settings > > in the registers for driver init as what this driver did at the first place > > -- no offense but I don't think you would ask this if you check the git-log > > of the driver. > > > ~FSL_SAI_MASK is better than 0x0. And you also replace 0xffffffff. I would later send a patch to reset SAI for a true init instead of these lines but not within this patch as it's focusing on the interrupts enabling. So please don't grasp the mask here. Just let me continue. Thank you, Nicolin Chen

1 0

Re: [alsa-devel] [PATCH bisect 2/2] ASoC: fsl_sai: Separately enable interrupts for Tx and Rx streams
by Nicolin Chen 01 Apr '14

01 Apr '14

On Tue, Apr 01, 2014 at 11:25:02AM +0800, Wang Dongsheng-B40534 wrote: > > Subject: [PATCH bisect 2/2] ASoC: fsl_sai: Separately enable interrupts for Tx > > and Rx streams > > > > We only enable one side interrupt for each stream since over/underrun > > on the opposite stream would be resulted from what we previously did, > > enabling TERE but remaining FRDE disabled, even though the xrun on the > > opposite direction will not break the current stream. > > > > Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> > > Acked-by: Xiubo Li <Li.Xiubo(a)freescale.com> > > --- > > sound/soc/fsl/fsl_sai.c | 8 ++++++-- > > sound/soc/fsl/fsl_sai.h | 1 + > > 2 files changed, 7 insertions(+), 2 deletions(-) > > > > diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c > > index bdfd497..d64c33f 100644 > > --- a/sound/soc/fsl/fsl_sai.c > > +++ b/sound/soc/fsl/fsl_sai.c > > @@ -397,4 +397,6 @@ static int fsl_sai_trigger(struct snd_pcm_substream > > *substream, int cmd, > > > > regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > + FSL_SAI_CSR_xIE_MASK, FSL_SAI_FLAGS); > > + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE); > > break; > > @@ -404,4 +406,6 @@ static int fsl_sai_trigger(struct snd_pcm_substream > > *substream, int cmd, > > regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > FSL_SAI_CSR_FRDE, 0); > > + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), > > + FSL_SAI_CSR_xIE_MASK, 0); > > > > if (!(tcsr & FSL_SAI_CSR_FRDE || rcsr & FSL_SAI_CSR_FRDE)) { > > @@ -464,6 +468,6 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) > > struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); > > > > - regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, FSL_SAI_FLAGS); > > - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, FSL_SAI_FLAGS); > > + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); > > + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); > > Why are you remove this macro? Don't use magic number. It's pretty clear that the so-called magic number is to clear the settings in the registers for driver init as what this driver did at the first place -- no offense but I don't think you would ask this if you check the git-log of the driver. Thank you, Nicolin Chen

1 0

[alsa-devel] [PATCH] ALSA: usb-audio - Capture and duplex support for Digidesign Mbox 1 sound card.
by Damien Zammit 01 Apr '14

01 Apr '14

This patch completes the support for Digidesign Mbox 1 sound card by adding capture. The issue with this card was that it has two endpoints on the same interface. This patch creates a dual endpoint quirk. The quirk interface needs a second audioformat struct for this to work which I called ".data2". Signed-off-by: Damien Zammit <damien(a)zamaudio.com>

4 13

[alsa-devel] [PATCH] ASoC: fsl_sai: Fix buggy configurations in trigger()
by Nicolin Chen 01 Apr '14

01 Apr '14

The current trigger() has two crucial problems: 1) The DMA request enabling operations (FSL_SAI_CSR_FRDE) for Tx and Rx are now totally exclusive: It would fail to run simultaneous Tx-Rx cases. 2) The TERE disabling operation depends on an incorrect condition -- active reference count that only gets increased in snd_pcm_open() and decreased in snd_pcm_close(): The TERE would never get cleared. So this patch overwrites the trigger function by following these rules: A) We continue to support tx-async-while-rx-sync-to-tx case alone, which's originally limited by this fsl_sai driver, but we make the code easy to modify for the further support of the opposite case. B) We enable both TE and RE for PLAYBACK stream or CAPTURE stream but only enabling the DMA request bit (FSL_SAI_CSR_FRDE) of the current direction due to the requirement of SAI -- For tx-async-while-rx-sync-to-tx case, the receiver is enabled only when both the transmitter and receiver are enabled. C) We only enable one side interrupt for each stream since over/underrun on the opposite stream would be resulted from what we've done in rule B: enabling TERE but remaining FRDE disabled, even though the xrun on the opposite direction will not break the current stream. Tested cases: a) aplay test.wav -d5 b) arecord -r44100 -c2 -fS16_LE test.wav -d5 c) arecord -r44100 -c2 -fS16_LE -d5 | aplay d) (aplay test2.wav &); sleep 1; arecord -r44100 -c2 -fS16_LE test.wav -d1 e) (arecord -r44100 -c2 -fS16_LE test.wav -d5 &); sleep 1; aplay test.wav -d1 Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_sai.c | 43 +++++++++++++++++++++++-------------------- sound/soc/fsl/fsl_sai.h | 11 +++++++++++ 2 files changed, 34 insertions(+), 20 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index f088545..d64c33f 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -365,6 +365,7 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = snd_soc_dai_get_drvdata(cpu_dai); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u32 tcsr, rcsr; /* @@ -379,14 +380,6 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_read(sai->regmap, FSL_SAI_TCSR, &tcsr); regmap_read(sai->regmap, FSL_SAI_RCSR, &rcsr); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - tcsr |= FSL_SAI_CSR_FRDE; - rcsr &= ~FSL_SAI_CSR_FRDE; - } else { - rcsr |= FSL_SAI_CSR_FRDE; - tcsr &= ~FSL_SAI_CSR_FRDE; - } - /* * It is recommended that the transmitter is the last enabled * and the first disabled. @@ -395,22 +388,32 @@ static int fsl_sai_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - tcsr |= FSL_SAI_CSR_TERE; - rcsr |= FSL_SAI_CSR_TERE; + if (!(tcsr & FSL_SAI_CSR_FRDE || rcsr & FSL_SAI_CSR_FRDE)) { + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, + FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, + FSL_SAI_CSR_TERE, FSL_SAI_CSR_TERE); + } - regmap_write(sai->regmap, FSL_SAI_RCSR, rcsr); - regmap_write(sai->regmap, FSL_SAI_TCSR, tcsr); + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), + FSL_SAI_CSR_xIE_MASK, FSL_SAI_FLAGS); + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), + FSL_SAI_CSR_FRDE, FSL_SAI_CSR_FRDE); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (!(cpu_dai->playback_active || cpu_dai->capture_active)) { - tcsr &= ~FSL_SAI_CSR_TERE; - rcsr &= ~FSL_SAI_CSR_TERE; + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), + FSL_SAI_CSR_FRDE, 0); + regmap_update_bits(sai->regmap, FSL_SAI_xCSR(tx), + FSL_SAI_CSR_xIE_MASK, 0); + + if (!(tcsr & FSL_SAI_CSR_FRDE || rcsr & FSL_SAI_CSR_FRDE)) { + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, + FSL_SAI_CSR_TERE, 0); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, + FSL_SAI_CSR_TERE, 0); } - - regmap_write(sai->regmap, FSL_SAI_TCSR, tcsr); - regmap_write(sai->regmap, FSL_SAI_RCSR, rcsr); break; default: return -EINVAL; @@ -464,8 +467,8 @@ static int fsl_sai_dai_probe(struct snd_soc_dai *cpu_dai) { struct fsl_sai *sai = dev_get_drvdata(cpu_dai->dev); - regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, FSL_SAI_FLAGS); - regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, FSL_SAI_FLAGS); + regmap_update_bits(sai->regmap, FSL_SAI_TCSR, 0xffffffff, 0x0); + regmap_update_bits(sai->regmap, FSL_SAI_RCSR, 0xffffffff, 0x0); regmap_update_bits(sai->regmap, FSL_SAI_TCR1, FSL_SAI_CR1_RFW_MASK, FSL_SAI_MAXBURST_TX * 2); regmap_update_bits(sai->regmap, FSL_SAI_RCR1, FSL_SAI_CR1_RFW_MASK, diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index a264185..be26d46 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -35,6 +35,16 @@ #define FSL_SAI_RFR 0xc0 /* SAI Receive FIFO */ #define FSL_SAI_RMR 0xe0 /* SAI Receive Mask */ +#define FSL_SAI_xCSR(tx) (tx ? FSL_SAI_TCSR : FSL_SAI_RCSR) +#define FSL_SAI_xCR1(tx) (tx ? FSL_SAI_TCR1 : FSL_SAI_RCR1) +#define FSL_SAI_xCR2(tx) (tx ? FSL_SAI_TCR2 : FSL_SAI_RCR2) +#define FSL_SAI_xCR3(tx) (tx ? FSL_SAI_TCR3 : FSL_SAI_RCR3) +#define FSL_SAI_xCR4(tx) (tx ? FSL_SAI_TCR4 : FSL_SAI_RCR4) +#define FSL_SAI_xCR5(tx) (tx ? FSL_SAI_TCR5 : FSL_SAI_RCR5) +#define FSL_SAI_xDR(tx) (tx ? FSL_SAI_TDR : FSL_SAI_RDR) +#define FSL_SAI_xFR(tx) (tx ? FSL_SAI_TFR : FSL_SAI_RFR) +#define FSL_SAI_xMR(tx) (tx ? FSL_SAI_TMR : FSL_SAI_RMR) + /* SAI Transmit/Recieve Control Register */ #define FSL_SAI_CSR_TERE BIT(31) #define FSL_SAI_CSR_FR BIT(25) @@ -48,6 +58,7 @@ #define FSL_SAI_CSR_FWF BIT(17) #define FSL_SAI_CSR_FRF BIT(16) #define FSL_SAI_CSR_xIE_SHIFT 8 +#define FSL_SAI_CSR_xIE_MASK (0x1f << FSL_SAI_CSR_xIE_SHIFT) #define FSL_SAI_CSR_WSIE BIT(12) #define FSL_SAI_CSR_SEIE BIT(11) #define FSL_SAI_CSR_FEIE BIT(10) -- 1.8.4

3 2

[alsa-devel] cloning snd_seq_t (or creating one from client id)
by Tobias Schlemmer 01 Apr '14

01 Apr '14

Hi, Recently, I came upon the following problem with the ALSA sequencer interface: I'm surprised that the upcoming version of RtMidi uses one ALSA client per MIDI port. As an end user I'd expect to see only one ALSA client per program instance (e.g. using aconnect -i - o -l). Asking for the reason of that change I got the answer that it has something to do with broken multithread support. Digging deeper into this problem I found out that sequencer handles (snd_seq_t) are not thread safe. So my question is: Could you provide some way to implement multithreaded ALSA clients? My idea is to provide some function int snd_seq_clone(snd_seq_t * old, snd_seq_t ** new) /* clone a sequencer */ and/or int snd_seq_create_from_client(int clientid, snd_seq_t ** new) /* create a sequencer from a given client id */ and/or int snd_seq_split_port(snd_seq_t * old, snd_seq_t ** new, int port) /* create a sequencer that steels a port from old */ For the semantics I think cloned sequencers should act as if they don't know each other. Each one has its own set of ports. While the generation of port numbers still uses the same counter. I don't expect creating a port to be that time critical that it can't use locks. If it helps to ease the implementation one could use a master sequencer with some slaves. If the master sequencer is deleted the slaves are automatically freed and must not be used anymore. And if a subsequencer tries to access a port of another subsequncer, ALSA could just return with an error. Regards, Tobias P.S.: please, CC to me in answers.

3 3

[alsa-devel] [PATCH] ASoC: rt5640: Change the setting method of idle_bias_off
by Oder Chiou 31 Mar '14

31 Mar '14

The patch moves the idle_bias_off setting to struct "soc_codec_dev_rt5640". Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5640.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index b088f14..6b8778f 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1895,7 +1895,6 @@ static int rt5640_probe(struct snd_soc_codec *codec) return ret; } - codec->dapm.idle_bias_off = 1; rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_update_bits(codec, RT5640_DUMMY1, 0x0301, 0x0301); @@ -2004,6 +2003,7 @@ static struct snd_soc_codec_driver soc_codec_dev_rt5640 = { .suspend = rt5640_suspend, .resume = rt5640_resume, .set_bias_level = rt5640_set_bias_level, + .idle_bias_off = true, .controls = rt5640_snd_controls, .num_controls = ARRAY_SIZE(rt5640_snd_controls), .dapm_widgets = rt5640_dapm_widgets, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: rt5640: add an of_match table
by Stephen Warren 31 Mar '14

31 Mar '14

From: Stephen Warren <swarren(a)nvidia.com> Add a device tree match table. This serves to make the driver's support of device tree more explicit. Perhaps the fallback for DT matching to using the i2c_device_id table will go away one day, since it fails in face of devices from different vendors with the same name. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/rt5640.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index b5014a2024e2..b6a02c16f100 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2018,6 +2018,14 @@ static const struct i2c_device_id rt5640_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, rt5640_i2c_id); +#if defined(CONFIG_OF) +static const struct of_device_id rt5640_of_match[] = { + { .compatible = "realtek,rt5640", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5640_of_match); +#endif + #ifdef CONFIG_ACPI static struct acpi_device_id rt5640_acpi_match[] = { { "INT33CA", 0 }, @@ -2166,6 +2174,7 @@ static struct i2c_driver rt5640_i2c_driver = { .name = "rt5640", .owner = THIS_MODULE, .acpi_match_table = ACPI_PTR(rt5640_acpi_match), + .of_match_table = of_match_ptr(rt5640_of_match), }, .probe = rt5640_i2c_probe, .remove = rt5640_i2c_remove, -- 1.8.1.5

3 3

[alsa-devel] [PATCH] ASoC: rt5640: Use the platform data for DMIC settings
by Oder Chiou 31 Mar '14

31 Mar '14

The patch uses the platform data for DMIC settings. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- include/sound/rt5640.h | 4 +++ sound/soc/codecs/rt5640.c | 77 ++++++++++++++--------------------------------- 2 files changed, 27 insertions(+), 54 deletions(-) diff --git a/include/sound/rt5640.h b/include/sound/rt5640.h index 27cc75e..59d26dd 100644 --- a/include/sound/rt5640.h +++ b/include/sound/rt5640.h @@ -16,6 +16,10 @@ struct rt5640_platform_data { bool in1_diff; bool in2_diff; + bool dmic_en; + bool dmic1_data_pin; /* 0 = IN1P; 1 = GPIO3 */ + bool dmic2_data_pin; /* 0 = IN1N; 1 = GPIO4 */ + int ldo1_en; /* GPIO for LDO1_EN */ }; diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index bcd6513..cf041e3 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -872,54 +872,6 @@ static SOC_ENUM_SINGLE_DECL(rt5640_sdi_sel_enum, RT5640_I2S2_SDP, static const struct snd_kcontrol_new rt5640_sdi_mux = SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum); -static int rt5640_set_dmic1_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - snd_soc_update_bits(codec, RT5640_GPIO_CTRL1, - RT5640_GP2_PIN_MASK | RT5640_GP3_PIN_MASK, - RT5640_GP2_PIN_DMIC1_SCL | RT5640_GP3_PIN_DMIC1_SDA); - snd_soc_update_bits(codec, RT5640_DMIC, - RT5640_DMIC_1L_LH_MASK | RT5640_DMIC_1R_LH_MASK | - RT5640_DMIC_1_DP_MASK, - RT5640_DMIC_1L_LH_FALLING | RT5640_DMIC_1R_LH_RISING | - RT5640_DMIC_1_DP_IN1P); - break; - - default: - return 0; - } - - return 0; -} - -static int rt5640_set_dmic2_event(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - struct snd_soc_codec *codec = w->codec; - - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - snd_soc_update_bits(codec, RT5640_GPIO_CTRL1, - RT5640_GP2_PIN_MASK | RT5640_GP4_PIN_MASK, - RT5640_GP2_PIN_DMIC1_SCL | RT5640_GP4_PIN_DMIC2_SDA); - snd_soc_update_bits(codec, RT5640_DMIC, - RT5640_DMIC_2L_LH_MASK | RT5640_DMIC_2R_LH_MASK | - RT5640_DMIC_2_DP_MASK, - RT5640_DMIC_2L_LH_FALLING | RT5640_DMIC_2R_LH_RISING | - RT5640_DMIC_2_DP_IN1N); - break; - - default: - return 0; - } - - return 0; -} - static void hp_amp_power_on(struct snd_soc_codec *codec) { struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); @@ -1054,12 +1006,10 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, set_dmic_clk, SND_SOC_DAPM_PRE_PMU), - SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5640_DMIC, - RT5640_DMIC_1_EN_SFT, 0, rt5640_set_dmic1_event, - SND_SOC_DAPM_PRE_PMU), - SND_SOC_DAPM_SUPPLY("DMIC2 Power", RT5640_DMIC, - RT5640_DMIC_2_EN_SFT, 0, rt5640_set_dmic2_event, - SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5640_DMIC, RT5640_DMIC_1_EN_SFT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DMIC2 Power", RT5640_DMIC, RT5640_DMIC_2_EN_SFT, 0, + NULL, 0), /* Boost */ SND_SOC_DAPM_PGA("BST1", RT5640_PWR_ANLG2, RT5640_PWR_BST1_BIT, 0, NULL, 0), @@ -2179,6 +2129,25 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4, RT5640_IN_DF2, RT5640_IN_DF2); + if (rt5640->pdata.dmic_en) { + regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, + RT5640_GP2_PIN_MASK, RT5640_GP2_PIN_DMIC1_SCL); + + if (rt5640->pdata.dmic1_data_pin) { + regmap_update_bits(rt5640->regmap, RT5640_DMIC, + RT5640_DMIC_1_DP_MASK, RT5640_DMIC_1_DP_GPIO3); + regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, + RT5640_GP3_PIN_MASK, RT5640_GP3_PIN_DMIC1_SDA); + } + + if (rt5640->pdata.dmic2_data_pin) { + regmap_update_bits(rt5640->regmap, RT5640_DMIC, + RT5640_DMIC_2_DP_MASK, RT5640_DMIC_2_DP_GPIO4); + regmap_update_bits(rt5640->regmap, RT5640_GPIO_CTRL1, + RT5640_GP4_PIN_MASK, RT5640_GP4_PIN_DMIC2_SDA); + } + } + rt5640->hp_mute = 1; ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, -- 1.8.1.1.439.g50a6b54

3 7

[alsa-devel] [GIT PULL] ASoC updates for v3.15
by Mark Brown 31 Mar '14

31 Mar '14

The following changes since commit e090d5b6ad20056ec0ef58727e3ae95fd82be090: Merge remote-tracking branches 'asoc/topic/simple' and 'asoc/topic/sirf' into asoc-next (2014-03-23 14:00:59 +0000) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.15-4 for you to fetch changes up to c159a85013afbb8283f0c7272812952e04d5c3a1: Merge remote-tracking branches 'asoc/topic/tdm' and 'asoc/topic/tegra' into asoc-next (2014-03-25 21:22:15 +0000) ---------------------------------------------------------------- ASoC: Final updates for v3.15 merge window A few more updates from last week - use of the tdm_slot mapping from Xiubo plus a few smaller fixes and cleanups. ---------------------------------------------------------------- Alex Smith (1): MIPS: Fix possible build error with transparent hugepages enabled Andreas Herrmann (1): MIPS: Octeon: Fix warning in of_device_alloc on cn3xxx Arnaldo Carvalho de Melo (1): perf bench numa: Make no args mean 'run all tests' Ben Pfaff (1): openvswitch: Correctly report flow used times for first 5 minutes after boot. Ben Widawsky (1): drm/i915: Fix PSR programming Benedikt Spranger (1): net: cpsw: do not register cpts twice Bjorn Helgaas (1): Revert "[PATCH] Insert GART region into resource map" Bjørn Mork (1): net: cdc_ncm: fix control message ordering Boris BREZILLON (1): ARM: at91: fix network interface ordering for sama5d36 Charles Keepax (1): ALSA: compress: Pass through return value of open ops callback Chris Wilson (1): drm/i915: Disable stolen memory when DMAR is active Christian Riesch (2): net: davinci_emac: Replace devm_request_irq with request_irq net: davinci_emac: Fix rollback of emac_dev_open() Clemens Ladisch (1): ALSA: oxygen: Xonar DG(X): fix Stereo Upmixing regression Colin Ian King (1): MIPS: Octeon: Fix fall through on bar type OCTEON_DMA_BAR_TYPE_SMALL Daniel Kurtz (1): drm/exynos: Fix (more) freeing issues in exynos_drm_drv.c Daniel Vetter (2): drm/i915: Don't enable display error interrupts from the start drm: Fix use-after-free in the shadow-attache exit code Dave Airlie (3): Merge tag 'drm-intel-fixes-2014-03-17' of git://anongit.freedesktop.org/drm-intel into drm-fixes Merge tag 'drm-intel-fixes-2014-03-19' of git://anongit.freedesktop.org/drm-intel into drm-fixes Merge branch 'exynos-drm-fixes' of git://git.kernel.org/.../daeinki/drm-exynos into drm-fixes Dave Jones (2): brcmfmac: fix skb leak in brcmf_sdio_txpkt_prep_sg error path. block: free q->flush_rq in blk_init_allocated_queue error paths Dave Kleikamp (1): sparc64: don't treat 64-bit syscall return codes as 32-bit David Miller (1): sparc: serial: Clean up the locking for -rt David S. Miller (6): Merge branch 'for-davem' of git://git.kernel.org/.../linville/wireless Merge branch 'master' of git://git.kernel.org/.../klassert/ipsec Merge branch 'cnic-net' Merge branch 'fixes' of git://git.kernel.org/.../jesse/openvswitch sparc64: Make sure %pil interrupts are enabled during hypervisor yield. Merge branch 'davinci_emac' David Stevens (2): vxlan: fix potential NULL dereference in arp_reduce() vxlan: fix nonfunctional neigh_reduce() Deng-Cheng Zhu (4): MIPS: APRP: Fix the linking of rtlx interrupt hook MIPS: APRP: Unregister rtlx interrupt hook at module exit MIPS: APRP: Choose the correct VPE loader by fixing the linking MIPS: math-emu: Fix prefx detection and COP1X function field definition Doug Wilson (1): sparc64:tsb.c:use array size macro rather than number Eric Dumazet (1): tcp: syncookies: do not use getnstimeofday() Erik Hugne (1): tipc: fix spinlock recursion bug for failed subscriptions Felix Fietkau (1): ath9k_hw: fix unreachable code in baseband hang detection code Fernando Luis Vázquez Cao (1): HID: hidraw: fix warning destroying hidraw device files after parent Frank Praznik (1): HID: sony: Fix work queue issues. Helge Deller (2): partly revert commit 8a10bc9: parisc/sti_console: prefer Linux fonts over built-in ROM fonts parisc: wire up sys_utimes Helmut Schaa (1): ath9k: Fix sequence number assignment for non-data frames Huacai Chen (1): MIPS: FPU: Fix conflict of register usage Hugh Dickins (2): mm: fix bad rss-counter if remap_file_pages raced migration mm: fix swapops.h:131 bug if remap_file_pages raced migration Ingo Molnar (1): Merge tag 'perf-urgent-for-mingo' of git://git.kernel.org/.../acme/linux into perf/urgent Jani Nikula (1): Revert "drm/i915: don't touch the VDD when disabling the panel" Jarno Rajahalme (2): openvswitch: Read tcp flags only then the tranport header is present. openvswitch: Fix race. Jean-Francois Moine (3): ASoC: simple-card: dynamically allocate the DAI link and properties ASoC: simple-card: Add DT documentation for multi-DAI links ASoC: simple-card: Handle many DAI links Jens Axboe (1): mtip32xx: fix bad use of smp_processor_id() Jim Quinlan (1): MIPS: Make local_irq_disable macro safe for non-Mipsr2 Jiri Pirko (1): ovs: fix dp check in ovs_dp_reset_user_features Joe Perches (1): MAINTAINERS: Add linux.nics(a)intel.com to INTEL ETHERNET DRIVERS John David Anglin (1): parisc: Remove unused CONFIG_PARISC_TMPALIAS code John W. Linville (1): Merge branch 'master' of git://git.kernel.org/.../linville/wireless into for-davem Kuninori Morimoto (3): ARM: bockw: fixup SND_SOC_DAIFMT_CBx_CFx flags ASoC: rcar: bugfix: it cares about the non-src case ASoC: rcar: subnode tidyup for renesas,rsnd.txt Lars Persson (1): MIPS: Fix syscall tracing interface Lars-Peter Clausen (1): ASoC: Remove name_prefix unset during DAI link init hack Li RongQing (1): netpoll: fix the skb check in pkt_is_ns Li Zefan (1): cgroup: fix a failure path in create_css() Linus Torvalds (18): Merge branch 'for-linus' of git://git.kernel.org/.../jikos/hid Merge branch 'upstream' of git://git.linux-mips.org/pub/scm/ralf/upstream-linus Merge branch 'for-linus' of git://git.kernel.dk/linux-block Merge tag 'for-linus' of git://git.kernel.org/pub/scm/virt/kvm/kvm Merge branch 'timers-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge branch 'for-3.14-fixes' of git://git.kernel.org/.../tj/cgroup Merge tag 'pci-v3.14-fixes-3' of git://git.kernel.org/.../helgaas/pci Merge tag 'sound-3.14' of git://git.kernel.org/.../tiwai/sound Merge branch 'upstream' of git://git.linux-mips.org/pub/scm/ralf/upstream-linus Merge tag 'trace-fixes-v3.14-rc7' of git://git.kernel.org/.../rostedt/linux-trace futex: revert back to the explicit waiter counting code Merge branch 'drm-fixes' of git://people.freedesktop.org/~airlied/linux Merge tag 'fixes-for-linus' of git://git.kernel.org/.../arm/arm-soc Merge branch 'perf-urgent-for-linus' of git://git.kernel.org/.../tip/tip Merge git://git.kernel.org/.../davem/net Merge git://git.kernel.org/.../davem/sparc Merge branch 'parisc-3.14' of git://git.kernel.org/.../deller/parisc-linux Linux 3.14-rc8 Lucas Stach (1): ASoC: tegra: move AC97 clock handling to the machine driver Manuel Lauss (1): MIPS: Alchemy: Fix unchecked kstrtoul return value Mark Brown (14): Merge remote-tracking branches 'asoc/fix/cs42l51', 'asoc/fix/cs42l52', 'asoc/fix/cs42l73', 'asoc/fix/rcar', 'asoc/fix/spear' and 'asoc/fix/tegra' into asoc-linus Merge tag 'asoc-v3.15' into asoc-next Merge tag 'asoc-v3.15-2' into asoc-next Merge tag 'asoc-v3.15-3' into asoc-next Merge remote-tracking branch 'asoc/topic/rcar' into asoc-next Merge remote-tracking branch 'asoc/fix/pcm' into asoc-linus Merge remote-tracking branches 'asoc/fix/cs42l51', 'asoc/fix/cs42l52', 'asoc/fix/cs42l73', 'asoc/fix/rcar', 'asoc/fix/spear' and 'asoc/fix/tegra' into asoc-linus Merge tag 'asoc-v3.15' into asoc-next Merge tag 'asoc-v3.15-2' into asoc-next Merge tag 'asoc-v3.15-3' into asoc-next Merge tag 'asoc-v3.15-rcar' into asoc-next Merge remote-tracking branches 'asoc/topic/prefix' and 'asoc/topic/rcar' into asoc-next Merge remote-tracking branch 'asoc/topic/simple' into asoc-next Merge remote-tracking branches 'asoc/topic/tdm' and 'asoc/topic/tegra' into asoc-next Markos Chandras (2): MIPS: bcm47xx: Include missing errno.h for ENXIO MIPS: asm: syscall: Fix copying system call arguments Michael Chan (3): cnic: Use proper ulp_ops for per device operations. cnic,bnx2i,bnx2fc: Fix inconsistent use of page size cnic: Update version to 2.5.20 and copyright year. Mike Snitzer (2): block: fix q->flush_rq NULL pointer crash on dm-mpath flush block: change flush sequence list addition back to front add Nicolas Dichtel (3): rtnetlink: fix fdb notification flags ipmr: fix mfc notification flags ip6mr: fix mfc notification flags Nikolay Aleksandrov (2): net: af_key: fix sleeping under rcu selinux: add gfp argument to security_xfrm_policy_alloc and fix callers Nishanth Menon (1): net: micrel : ks8851-ml: add vdd-supply support Patrick Palka (1): perf bench: Fix NULL pointer dereference in "perf bench all" Paul Bolle (2): MIPS: Replace CONFIG_MIPS64 and CONFIG_MIPS32_R2 isdn/capi: Make Middleware depend on CAPI2.0 Paul Burton (1): MIPS: mark O32+FP64 experimental for now Paul Mackerras (2): KVM: PPC: Book3S HV: Remove bogus duplicate code KVM: PPC: Book3S HV: Fix register usage when loading/saving VRSAVE Peter Senna Tschudin (2): ATHEROS-ALX: Use dma_set_mask_and_coherent and fix a bug ATHEROS-ATL1E: Convert iounmap to pci_iounmap Rafał Miłecki (1): MIPS: BCM47XX: Check all (32) GPIOs when looking for a pin Ralf Baechle (1): MIPS: Fix randconfig build error. Roger Luethi (1): via-rhine: Disable device in error path Sebastian Hesselbarth (1): net: phy: fix uninitalized ethtool_wolinfo in phy_suspend Shawn Guo (1): MAINTAINERS: update IMX kernel git tree Simon Wood (1): HID: hid-lg4ff: Support new version of G27 Stanislaw Gruszka (1): Revert "rt2x00: rt2800lib: Update BBP register initialization for RT53xx" Stefan Agner (1): clocksource: vf_pit_timer: use complement for sched_clock reading Vaibhav Nagarnaik (1): tracing: Fix array size mismatch in format string Ville Syrjälä (2): drm/i915: Add a workaround for HSW scanline counter weirdness drm/i915: Fix scanline counter fixup on BDW Viller Hsiao (2): MIPS: ftrace: Fix icache flush range error MIPS: ftrace: Tweak safe_load()/safe_store() macros Will Deacon (1): parisc: locks: remove redundant arch_*_relax operations Xiubo Li (4): ASoC: core: remove the 'of_' prefix of of_xlate_tdm_slot_mask. ASoC: fsl-utils: Add fsl_asoc_xlate_tdm_slot_mask() support. ASoC: fsl-esai: Add .xlate_tdm_slot_mask() support. ASoC: imx-ssi: Add .xlate_tdm_slot_mask() support. lucien (1): ipv6: ip6_append_data_mtu do not handle the mtu of the second fragment properly stephen hemminger (1): netlink: fix setsockopt in mmap examples in documentation .../devicetree/bindings/net/micrel-ks8851.txt | 1 + .../devicetree/bindings/sound/renesas,rsnd.txt | 15 +- .../devicetree/bindings/sound/simple-card.txt | 42 +++- Documentation/networking/netlink_mmap.txt | 4 +- MAINTAINERS | 5 +- Makefile | 2 +- arch/arm/boot/dts/sama5d36.dtsi | 2 +- arch/arm/mach-shmobile/board-bockw.c | 16 +- arch/mips/Kconfig | 24 ++- arch/mips/alchemy/board-gpr.c | 4 +- arch/mips/alchemy/board-mtx1.c | 4 +- arch/mips/bcm47xx/board.c | 1 + arch/mips/bcm47xx/nvram.c | 2 +- arch/mips/cavium-octeon/octeon-irq.c | 22 ++- arch/mips/include/asm/asmmacro.h | 15 +- arch/mips/include/asm/fpu.h | 2 +- arch/mips/include/asm/ftrace.h | 20 +- arch/mips/include/asm/syscall.h | 10 +- arch/mips/include/uapi/asm/inst.h | 4 +- arch/mips/kernel/ftrace.c | 5 +- arch/mips/kernel/r4k_fpu.S | 16 +- arch/mips/kernel/rtlx-cmp.c | 3 + arch/mips/kernel/rtlx-mt.c | 3 + arch/mips/math-emu/cp1emu.c | 6 +- arch/mips/mti-malta/malta-amon.c | 2 +- arch/mips/mti-malta/malta-int.c | 4 +- arch/mips/pci/msi-octeon.c | 1 + arch/parisc/include/asm/page.h | 11 -- arch/parisc/include/asm/spinlock.h | 4 - arch/parisc/include/uapi/asm/unistd.h | 4 +- arch/parisc/kernel/cache.c | 64 ------ arch/parisc/kernel/syscall_table.S | 1 + arch/powerpc/kvm/book3s_hv_rmhandlers.S | 71 +------ arch/sparc/kernel/process_64.c | 4 +- arch/sparc/kernel/syscalls.S | 4 +- arch/sparc/mm/tsb.c | 2 +- arch/x86/kernel/aperture_64.c | 20 +- block/blk-core.c | 25 ++- block/blk-flush.c | 11 +- drivers/block/mtip32xx/mtip32xx.c | 2 +- drivers/clocksource/vf_pit_timer.c | 2 +- drivers/gpu/drm/drm_pci.c | 2 +- drivers/gpu/drm/exynos/exynos_drm_drv.c | 10 +- drivers/gpu/drm/i915/i915_gem_stolen.c | 7 + drivers/gpu/drm/i915/i915_irq.c | 71 ++++--- drivers/gpu/drm/i915/intel_ddi.c | 1 + drivers/gpu/drm/i915/intel_dp.c | 12 +- drivers/hid/hid-lg4ff.c | 2 + drivers/hid/hid-sony.c | 27 ++- drivers/hid/hidraw.c | 4 +- drivers/isdn/capi/Kconfig | 18 +- drivers/net/ethernet/atheros/alx/main.c | 14 +- drivers/net/ethernet/atheros/atl1e/atl1e_main.c | 4 +- drivers/net/ethernet/broadcom/cnic.c | 111 +++++------ drivers/net/ethernet/broadcom/cnic.h | 2 +- drivers/net/ethernet/broadcom/cnic_defs.h | 2 +- drivers/net/ethernet/broadcom/cnic_if.h | 16 +- drivers/net/ethernet/micrel/ks8851.c | 30 ++- drivers/net/ethernet/ti/cpsw.c | 4 - drivers/net/ethernet/ti/davinci_cpdma.c | 4 +- drivers/net/ethernet/ti/davinci_emac.c | 53 +++-- drivers/net/ethernet/via/via-rhine.c | 8 +- drivers/net/phy/phy_device.c | 3 +- drivers/net/usb/cdc_ncm.c | 48 +++-- drivers/net/vxlan.c | 130 ++++++++++-- drivers/net/wireless/ath/ath9k/hw.c | 3 +- drivers/net/wireless/ath/ath9k/xmit.c | 4 +- drivers/net/wireless/brcm80211/brcmfmac/dhd_sdio.c | 4 +- drivers/net/wireless/rt2x00/rt2800lib.c | 6 +- drivers/scsi/bnx2fc/bnx2fc_io.c | 16 +- drivers/scsi/bnx2fc/bnx2fc_tgt.c | 38 ++-- drivers/scsi/bnx2i/bnx2i_hwi.c | 52 ++--- drivers/scsi/bnx2i/bnx2i_iscsi.c | 23 +-- drivers/tty/serial/sunhv.c | 22 +-- drivers/tty/serial/sunsab.c | 14 +- drivers/tty/serial/sunsu.c | 14 +- drivers/tty/serial/sunzilog.c | 14 +- include/linux/ftrace_event.h | 4 - include/linux/rmap.h | 3 +- include/linux/security.h | 10 +- include/linux/usb/cdc_ncm.h | 1 + include/net/tcp.h | 11 +- include/sound/soc-dai.h | 2 +- include/trace/ftrace.h | 7 +- kernel/cgroup.c | 11 +- kernel/futex.c | 53 ++++- kernel/trace/trace_events.c | 6 - kernel/trace/trace_export.c | 7 +- lib/fonts/Kconfig | 6 +- mm/fremap.c | 28 ++- mm/migrate.c | 32 +++ mm/rmap.c | 5 +- net/core/netpoll.c | 2 +- net/core/rtnetlink.c | 10 +- net/ipv4/ipmr.c | 13 +- net/ipv6/ip6_output.c | 14 +- net/ipv6/ip6mr.c | 13 +- net/key/af_key.c | 19 +- net/openvswitch/datapath.c | 9 +- net/openvswitch/flow.c | 3 +- net/tipc/subscr.c | 29 +-- net/xfrm/xfrm_user.c | 6 +- security/capability.c | 3 +- security/security.c | 6 +- security/selinux/hooks.c | 13 +- security/selinux/include/security.h | 2 +- security/selinux/include/xfrm.h | 3 +- security/selinux/selinuxfs.c | 28 ++- security/selinux/ss/services.c | 6 +- security/selinux/xfrm.c | 14 +- sound/core/compress_offload.c | 2 +- sound/pci/oxygen/xonar_dg.c | 30 ++- sound/soc/fsl/Kconfig | 2 + sound/soc/fsl/fsl_esai.c | 2 + sound/soc/fsl/fsl_utils.c | 27 +++ sound/soc/fsl/fsl_utils.h | 4 +- sound/soc/fsl/imx-ssi.c | 2 + sound/soc/generic/simple-card.c | 220 +++++++++++++-------- sound/soc/sh/rcar/src.c | 18 +- sound/soc/soc-core.c | 17 +- sound/soc/tegra/tegra20_ac97.c | 15 +- sound/soc/tegra/tegra20_ac97.h | 1 - sound/soc/tegra/tegra_wm9712.c | 17 +- tools/perf/bench/numa.c | 1 + tools/perf/builtin-bench.c | 2 +- 125 files changed, 1152 insertions(+), 840 deletions(-)

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[alsa-devel] [PATCH 1/2] snd-ice1712: add suspend support for ICE1712 chip
by Ondrej Zary 31 Mar '14

31 Mar '14

Add suspend/resume support for ICE1712 chip. Card-specific subdrivers need to enable it and provide callbacks that suspend/resume the codecs. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/ice1712/ice1712.c | 85 +++++++++++++++++++++++++++++++++++++++---- 1 file changed, 77 insertions(+), 8 deletions(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 28ec872..dbb0f96 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2428,6 +2428,13 @@ static int snd_ice1712_chip_init(struct snd_ice1712 *ice) snd_ice1712_write(ice, ICE1712_IREG_CONSUMER_POWERDOWN, 0); } snd_ice1712_set_pro_rate(ice, 48000, 1); + /* unmask used interrupts */ + outb(((ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_2xMPU401) == 0 ? + ICE1712_IRQ_MPU2 : 0) | + ((ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_NO_CON_AC97) ? + ICE1712_IRQ_PBKDS | ICE1712_IRQ_CONCAP | ICE1712_IRQ_CONPBK : 0), + ICEREG(ice, IRQMASK)); + outb(0x00, ICEMT(ice, IRQ)); return 0; } @@ -2589,6 +2596,7 @@ static int snd_ice1712_create(struct snd_card *card, ice->pci = pci; ice->irq = -1; pci_set_master(pci); + /* disable legacy emulation */ pci_write_config_word(ice->pci, 0x40, 0x807f); pci_write_config_word(ice->pci, 0x42, 0x0006); snd_ice1712_proc_init(ice); @@ -2625,14 +2633,6 @@ static int snd_ice1712_create(struct snd_card *card, return -EIO; } - /* unmask used interrupts */ - outb(((ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_2xMPU401) == 0 ? - ICE1712_IRQ_MPU2 : 0) | - ((ice->eeprom.data[ICE_EEP1_CODEC] & ICE1712_CFG_NO_CON_AC97) ? - ICE1712_IRQ_PBKDS | ICE1712_IRQ_CONCAP | ICE1712_IRQ_CONPBK : 0), - ICEREG(ice, IRQMASK)); - outb(0x00, ICEMT(ice, IRQ)); - err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, ice, &ops); if (err < 0) { snd_ice1712_free(ice); @@ -2809,11 +2809,80 @@ static void snd_ice1712_remove(struct pci_dev *pci) snd_card_free(card); } +#ifdef CONFIG_PM_SLEEP +static int snd_ice1712_suspend(struct device *dev) +{ + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); + struct snd_ice1712 *ice = card->private_data; + + if (!ice->pm_suspend_enabled) + return 0; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + + snd_pcm_suspend_all(ice->pcm); + snd_pcm_suspend_all(ice->pcm_pro); + snd_pcm_suspend_all(ice->pcm_ds); + snd_ac97_suspend(ice->ac97); + + if (ice->pm_suspend) + ice->pm_suspend(ice); + + pci_disable_device(pci); + pci_save_state(pci); + pci_set_power_state(pci, PCI_D3hot); + return 0; +} + +static int snd_ice1712_resume(struct device *dev) +{ + struct pci_dev *pci = to_pci_dev(dev); + struct snd_card *card = dev_get_drvdata(dev); + struct snd_ice1712 *ice = card->private_data; + + if (!ice->pm_suspend_enabled) + return 0; + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + + if (pci_enable_device(pci) < 0) { + snd_card_disconnect(card); + return -EIO; + } + + pci_set_master(pci); + + if (snd_ice1712_chip_init(ice) < 0) { + snd_card_disconnect(card); + return -EIO; + } + + if (ice->pm_resume) + ice->pm_resume(ice); + + if (ice->ac97) + snd_ac97_resume(ice->ac97); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} + +static SIMPLE_DEV_PM_OPS(snd_ice1712_pm, snd_ice1712_suspend, snd_ice1712_resume); +#define SND_VT1712_PM_OPS &snd_ice1712_pm +#else +#define SND_VT1712_PM_OPS NULL +#endif /* CONFIG_PM_SLEEP */ + static struct pci_driver ice1712_driver = { .name = KBUILD_MODNAME, .id_table = snd_ice1712_ids, .probe = snd_ice1712_probe, .remove = snd_ice1712_remove, + .driver = { + .pm = SND_VT1712_PM_OPS, + }, }; module_pci_driver(ice1712_driver); -- Ondrej Zary

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[alsa-devel] [PATCH] ASoC: rsnd: fix src clock prepare/unprepare
by Ben Dooks 31 Mar '14

31 Mar '14

As with the previous commit, before a clock can be used it must be prepared for use. Change from clk_enable() and clk_disable() to the versions of the calls which also prepare and un-prepare the clocks. Will fix warnings from the clock code when this is used. Signed-off-by: Ben Dooks <ben.dooks(a)codethink.co.uk> --- sound/soc/sh/rcar/scu.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/scu.c b/sound/soc/sh/rcar/scu.c index 9153a11..3b1d1d2 100644 --- a/sound/soc/sh/rcar/scu.c +++ b/sound/soc/sh/rcar/scu.c @@ -284,7 +284,7 @@ static int rsnd_scu_start(struct rsnd_mod *mod, return 0; } - clk_enable(scu->clk); + clk_prepare_enable(scu->clk); /* it use DMA transter */ @@ -317,7 +317,7 @@ static int rsnd_scu_stop(struct rsnd_mod *mod, rsnd_scu_transfer_stop(priv, mod, rdai, io); - clk_disable(scu->clk); + clk_disable_unprepare(scu->clk); return 0; } -- 1.9.0

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Re: [alsa-devel] is 440 Hz the true harmonic wave
by Joël Krähemann 31 Mar '14

31 Mar '14

On Mon, 2014-03-31 at 10:27 +0200, Felix Homann wrote: > 2014-03-31 10:20 GMT+02:00 Joël Krähemann <joel(a)weedlight.ch>: > > > I believe that 440 Hz is an approximation rather than true > harmonic on > it's pressure. Or am I wrong? Sine is harmonic in default arc. > > > > What are you talking about??? About scale. Am I wrong when I believe 440 Hz has physical property to be harmonic?

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