- Alsa-devel - mailman.alsa-project.org

[PATCH for-next] ASoC: tlv320adcx140: Fix mic_bias and vref device tree verification
by Dan Murphy 05 Mar '20

05 Mar '20

Fix the range verification check for the mic_bias and vref device tree entries. Fixes 37bde5acf040 ("ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family") Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tlv320adcx140.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 93a0cb8e662c..38897568ee96 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -748,9 +748,8 @@ static int adcx140_codec_probe(struct snd_soc_component *component) if (ret) bias_source = ADCX140_MIC_BIAS_VAL_VREF; - if (bias_source != ADCX140_MIC_BIAS_VAL_VREF && - bias_source != ADCX140_MIC_BIAS_VAL_VREF_1096 && - bias_source != ADCX140_MIC_BIAS_VAL_AVDD) { + if (bias_source < ADCX140_MIC_BIAS_VAL_VREF || + bias_source > ADCX140_MIC_BIAS_VAL_AVDD) { dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); return -EINVAL; } @@ -760,9 +759,8 @@ static int adcx140_codec_probe(struct snd_soc_component *component) if (ret) vref_source = ADCX140_MIC_BIAS_VREF_275V; - if (vref_source != ADCX140_MIC_BIAS_VREF_275V && - vref_source != ADCX140_MIC_BIAS_VREF_25V && - vref_source != ADCX140_MIC_BIAS_VREF_1375V) { + if (vref_source < ADCX140_MIC_BIAS_VREF_275V || + vref_source > ADCX140_MIC_BIAS_VREF_1375V) { dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); return -EINVAL; } -- 2.25.1

2 1

[PATCH] ASoC: sprd: Allow the MCDT driver to build into modules
by Baolin Wang 05 Mar '20

05 Mar '20

Change the config to 'tristate' for MCDT driver to allow it to build into modules, as well as changing to use IS_ENABLED() to validate if need supply dummy functions when building the MCDT driver as a module. Signed-off-by: Baolin Wang <baolin.wang7(a)gmail.com> --- sound/soc/sprd/Kconfig | 2 +- sound/soc/sprd/sprd-mcdt.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sprd/Kconfig b/sound/soc/sprd/Kconfig index 5474fd3..5e0ac82 100644 --- a/sound/soc/sprd/Kconfig +++ b/sound/soc/sprd/Kconfig @@ -8,7 +8,7 @@ config SND_SOC_SPRD the Spreadtrum SoCs' Audio interfaces. config SND_SOC_SPRD_MCDT - bool "Spreadtrum multi-channel data transfer support" + tristate "Spreadtrum multi-channel data transfer support" depends on SND_SOC_SPRD help Say y here to enable multi-channel data transfer support. It diff --git a/sound/soc/sprd/sprd-mcdt.h b/sound/soc/sprd/sprd-mcdt.h index 9cc7e207..679e3af 100644 --- a/sound/soc/sprd/sprd-mcdt.h +++ b/sound/soc/sprd/sprd-mcdt.h @@ -48,7 +48,7 @@ struct sprd_mcdt_chan { struct list_head list; }; -#ifdef CONFIG_SND_SOC_SPRD_MCDT +#if IS_ENABLED(CONFIG_SND_SOC_SPRD_MCDT) struct sprd_mcdt_chan *sprd_mcdt_request_chan(u8 channel, enum sprd_mcdt_channel_type type); void sprd_mcdt_free_chan(struct sprd_mcdt_chan *chan); -- 1.9.1

2 1

Applied "ASoC: soc-pcm: use defined stream" to the asoc tree
by Mark Brown 05 Mar '20

05 Mar '20

The patch ASoC: soc-pcm: use defined stream has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 25c2f5156dd57f03aee2de079248c23a56222c92 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 27 Feb 2020 10:54:38 +0900 Subject: [PATCH] ASoC: soc-pcm: use defined stream Many functions defines "stream = substream->stream", but some of them is using "substream->stream" instead of "stream". It is pointless. This patch uses defined stream. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/87mu947q1t.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-pcm.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 90857138c823..8c27eb4d5e4c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -644,8 +644,7 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) * bailed out on a higher level, since there would be no * CODEC to support the transfer direction in that case. */ - if (!snd_soc_dai_stream_valid(codec_dai, - substream->stream)) + if (!snd_soc_dai_stream_valid(codec_dai, stream)) continue; codec_stream = snd_soc_dai_get_pcm_stream(codec_dai, stream); @@ -2149,7 +2148,7 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); - ret = dpcm_be_dai_startup(fe, fe_substream->stream); + ret = dpcm_be_dai_startup(fe, stream); if (ret < 0) { dev_err(fe->dev,"ASoC: failed to start some BEs %d\n", ret); goto be_err; @@ -2180,7 +2179,7 @@ static int dpcm_fe_dai_startup(struct snd_pcm_substream *fe_substream) return 0; unwind: - dpcm_be_dai_startup_unwind(fe, fe_substream->stream); + dpcm_be_dai_startup_unwind(fe, stream); be_err: dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); return ret; @@ -2234,7 +2233,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); /* shutdown the BEs */ - dpcm_be_dai_shutdown(fe, substream->stream); + dpcm_be_dai_shutdown(fe, stream); dev_dbg(fe->dev, "ASoC: close FE %s\n", fe->dai_link->name); @@ -2412,9 +2411,9 @@ static int dpcm_fe_dai_hw_params(struct snd_pcm_substream *substream, mutex_lock_nested(&fe->card->mutex, SND_SOC_CARD_CLASS_RUNTIME); dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_FE); - memcpy(&fe->dpcm[substream->stream].hw_params, params, + memcpy(&fe->dpcm[stream].hw_params, params, sizeof(struct snd_pcm_hw_params)); - ret = dpcm_be_dai_hw_params(fe, substream->stream); + ret = dpcm_be_dai_hw_params(fe, stream); if (ret < 0) { dev_err(fe->dev,"ASoC: hw_params BE failed %d\n", ret); goto out; @@ -2736,7 +2735,7 @@ static int dpcm_fe_dai_prepare(struct snd_pcm_substream *substream) goto out; } - ret = dpcm_be_dai_prepare(fe, substream->stream); + ret = dpcm_be_dai_prepare(fe, stream); if (ret < 0) goto out; -- 2.20.1

1 0

Applied "ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths()" to the asoc tree
by Mark Brown 05 Mar '20

05 Mar '20

The patch ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths() has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 6e02feb0d2663c1b7caa5e271c2a60e219f0ca07 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Thu, 27 Feb 2020 10:54:48 +0900 Subject: [PATCH] ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths() dpcm_add_paths() checks returned be from dpcm_get_be() static int dpcm_add_paths(...) { ... for_each_dapm_widgets(list, i, widget) { ... be = dpcm_get_be(...); ... /* make sure BE is a real BE */ => if (!be->dai_link->no_pcm) continue; ... } ... } But, dpcm_get_be() itself is checking it already. dpcm_get_be(...) { ... for_each_card_rtds(card, be) { => if (!be->dai_link->no_pcm) continue; ... if (...) => return be; } return NULL } This patch removes duplicate check Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Link: https://lore.kernel.org/r/87lfoo7q1j.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-pcm.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 8c27eb4d5e4c..e3a2c4f7757b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1690,10 +1690,6 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; } - /* make sure BE is a real BE */ - if (!be->dai_link->no_pcm) - continue; - /* don't connect if FE is not running */ if (!fe->dpcm[stream].runtime && !fe->fe_compr) continue; -- 2.20.1

1 0

sof-hda-dsp: Support systems without integrated graphics audio
by GitHub pull_request - opened 05 Mar '20

05 Mar '20

alsa-project/alsa-ucm-conf pull request #15 was opened from kv2019i: On systems where integrated graphics audio is not present or is disabled, the HDMI PCM nodes are disabled. Add rules to detect these systems by checking presence of HDMI jack controls with UCM2 rules. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/15 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/15.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 0/8] ASoC: soc-pcm cleanup step5
by Kuninori Morimoto 05 Mar '20

05 Mar '20

Hi Mark These are v2 of soc-pcm cleanup step5. These are based on latest for-5.7 branch. Mainly v2 fixed the points reviewed in v1. [7/8] patch might exchange the behavior Kuninori Morimoto (8): 1) ASoC: soc-pcm: use defined stream 2) ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths() 3) ASoC: soc-pcm: move dpcm_fe_dai_close() 4) ASoC: soc-pcm: add dpcm_fe_dai_cleanup() 5) ASoC: soc-pcm: use snd_soc_dai_get_pcm_stream() at dpcm_set_fe_runtime() 6) ASoC: soc-pcm: check DAI's activity more simply 7) ASoC: soc-pcm: Do Digital Mute for both CPU/Codec in same timing. 8) ASoC: soc-pcm: tidyup dulicate handing at dpcm_fe_dai_startup() sound/soc/soc-pcm.c | 114 +++++++++++++++++++------------------------- 1 file changed, 50 insertions(+), 64 deletions(-) -- 2.17.1

2 12

[PATCH] ASoC: wm8741: Fix typo in Kconfig prompt
by Axel Lin 05 Mar '20

05 Mar '20

Fix trivial copy-n-paste mistake. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6aee70ed43df..fd09656d2f6b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1486,7 +1486,7 @@ config SND_SOC_WM8737 depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8741 - tristate "Wolfson Microelectronics WM8737 DAC" + tristate "Wolfson Microelectronics WM8741 DAC" depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8750 -- 2.20.1

2 2

[PATCH] ASoc: amd: acp3x: Add missing include <linux/io.h>
by YueHaibing 05 Mar '20

05 Mar '20

gcc 7.4.0 build fails: In file included from sound/soc/amd/acp3x-rt5682-max9836.c:20:0: sound/soc/amd/raven/acp3x.h: In function rv_readl: sound/soc/amd/raven/acp3x.h:113:9: error: implicit declaration of function readl; did you mean rv_readl? [-Werror=implicit-function-declaration] return readl(base_addr - ACP3x_PHY_BASE_ADDRESS); ^~~~~ rv_readl sound/soc/amd/raven/acp3x.h: In function rv_writel: sound/soc/amd/raven/acp3x.h:118:2: error: implicit declaration of function writel; did you mean rv_writel? [-Werror=implicit-function-declaration] writel(val, base_addr - ACP3x_PHY_BASE_ADDRESS); ^~~~~~ rv_writel Add <linux/io.h> to fix this. Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 6b8e4e7db3cd ("ASoC: amd: Add machine driver for Raven based platform") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 511b8b1..b4f68c5 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -16,6 +16,7 @@ #include <linux/module.h> #include <linux/i2c.h> #include <linux/input.h> +#include <linux/io.h> #include <linux/acpi.h> #include "raven/acp3x.h" -- 2.7.4

3 2

[PATCH] ASoC: Fix SND_SOC_ALL_CODECS imply ac97 fallout
by Geert Uytterhoeven 05 Mar '20

05 Mar '20

On i386 randconfig: sound/soc/codecs/wm9705.o: In function `wm9705_soc_resume': wm9705.c:(.text+0x128): undefined reference to `snd_ac97_reset' sound/soc/codecs/wm9712.o: In function `wm9712_soc_resume': wm9712.c:(.text+0x2d1): undefined reference to `snd_ac97_reset' sound/soc/codecs/wm9713.o: In function `wm9713_soc_resume': wm9713.c:(.text+0x820): undefined reference to `snd_ac97_reset' Fix this by adding the missing dependencies on SND_SOC_AC97_BUS. Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Geert Uytterhoeven <geert(a)linux-m68k.org> --- Before commit ea00d95200d02ece ("ASoC: Use imply for SND_SOC_ALL_CODECS"), SND_SOC_ALL_CODECS used: select SND_SOC_WM9705 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) select SND_SOC_WM9712 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) select SND_SOC_WM9713 if (SND_SOC_AC97_BUS || SND_SOC_AC97_BUS_NEW) but SND_SOC_AC97_BUS_NEW never existed in upstream. Should there be another dependency> See also "non-existent SND_SOC_AC97_BUS_NEW (was: Re: [PATCH v9] ASoC: pxa: switch to new ac97 bus support)" http://lore.kernel.org/r/CAMuHMdU3uxfBwKd8SkOtZSDV5Ai3CKc3CWRhDy0Cz94T1Hn0i… --- sound/soc/codecs/Kconfig | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9e9d54e4576ce5ba..a7e89567edbe8b47 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1610,16 +1610,19 @@ config SND_SOC_WM9090 config SND_SOC_WM9705 tristate + depends on SND_SOC_AC97_BUS select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW config SND_SOC_WM9712 tristate + depends on SND_SOC_AC97_BUS select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW config SND_SOC_WM9713 tristate + depends on SND_SOC_AC97_BUS select REGMAP_AC97 select AC97_BUS_COMPAT if AC97_BUS_NEW -- 2.17.1

5 6

Re: [PATCH v4 00/10] PCI: Add and use constant PCI_STATUS_ERROR_BITS and helper pci_status_get_and_clear_errors
by David Miller 05 Mar '20

05 Mar '20

Bjorn, please review and let me know if it is OK to merge this via the networking tree. Thank you.

2 2

Problematic motherboard: Supermicro MBD-X11SAT-F-O
by D. R. Evans 04 Mar '20

04 Mar '20

Where is the correct place to post detailed information about a motherboard that provides only unacceptably low volume with ALSA? I realise that the correct place probably isn't here, but I didn't know where else to ask for a pointer to where I should send the detailed information. I initially raised the problem on alsa-users: https://sourceforge.net/p/alsa/mailman/alsa-user/thread/db705c61-87ba-8d66-… but the suggestions made there led to no progress being made, and I am now as sure as I can be that the issue is not user error. The motherboard is a Supermicro MBD-X11SAT-F-O, and the basic high-level device information is: --------- card 0: PCH [HDA Intel PCH], device 0: ALC888-VD Analog [ALC888-VD Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 --------- Doc Evans -- Web: http://enginehousebooks.com/drevans

1 0

[PATCH] ASoC: tas571x: use symbolic register offsets
by Hermann Lauer 04 Mar '20

04 Mar '20

convert the numeric register offsets in tas5711_reg_defaults to the defined constants TAS571X_* from tas571x.h Tested with a tas5713 chip for which the driver works, too. Signed-off-by: Hermann Lauer <Hermann.Lauer(a)iwr.uni-heidelberg.de> diff --git a/sound/soc/codecs/tas571x.c b/sound/soc/codecs/tas571x.c --- a/sound/soc/codecs/tas571x.c +++ b/sound/soc/codecs/tas571x.c @@ -424,13 +424,13 @@ }; static const struct reg_default tas5711_reg_defaults[] = { - { 0x04, 0x05 }, - { 0x05, 0x40 }, - { 0x06, 0x00 }, - { 0x07, 0xff }, - { 0x08, 0x30 }, - { 0x09, 0x30 }, - { 0x1b, 0x82 }, + { TAS571X_SDI_REG, 0x05 }, + { TAS571X_SYS_CTRL_2_REG, 0x40 }, + { TAS571X_SOFT_MUTE_REG, 0x00 }, + { TAS571X_MVOL_REG, 0xff }, + { TAS571X_CH1_VOL_REG, 0x30 }, + { TAS571X_CH2_VOL_REG, 0x30 }, + { TAS571X_OSC_TRIM_REG, 0x82 }, }; static const struct regmap_config tas5711_regmap_config = {

1 0

Re: [PATCH v4 09/10] PCI: pci-bridge-emul: Use new constant PCI_STATUS_ERROR_BITS
by Bjorn Helgaas 04 Mar '20

04 Mar '20

On Sat, Feb 29, 2020 at 11:28:18PM +0100, Heiner Kallweit wrote: > Use new constant PCI_STATUS_ERROR_BITS to simplify the code. > > Signed-off-by: Heiner Kallweit <hkallweit1(a)gmail.com> Acked-by: Bjorn Helgaas <bhelgaas(a)google.com> > --- > drivers/pci/pci-bridge-emul.c | 14 ++------------ > 1 file changed, 2 insertions(+), 12 deletions(-) > > diff --git a/drivers/pci/pci-bridge-emul.c b/drivers/pci/pci-bridge-emul.c > index fffa77093..4f4f54bc7 100644 > --- a/drivers/pci/pci-bridge-emul.c > +++ b/drivers/pci/pci-bridge-emul.c > @@ -50,12 +50,7 @@ static const struct pci_bridge_reg_behavior pci_regs_behavior[] = { > (PCI_STATUS_CAP_LIST | PCI_STATUS_66MHZ | > PCI_STATUS_FAST_BACK | PCI_STATUS_DEVSEL_MASK) << 16), > .rsvd = GENMASK(15, 10) | ((BIT(6) | GENMASK(3, 0)) << 16), > - .w1c = (PCI_STATUS_PARITY | > - PCI_STATUS_SIG_TARGET_ABORT | > - PCI_STATUS_REC_TARGET_ABORT | > - PCI_STATUS_REC_MASTER_ABORT | > - PCI_STATUS_SIG_SYSTEM_ERROR | > - PCI_STATUS_DETECTED_PARITY) << 16, > + .w1c = PCI_STATUS_ERROR_BITS << 16, > }, > [PCI_CLASS_REVISION / 4] = { .ro = ~0 }, > > @@ -100,12 +95,7 @@ static const struct pci_bridge_reg_behavior pci_regs_behavior[] = { > PCI_STATUS_DEVSEL_MASK) << 16) | > GENMASK(11, 8) | GENMASK(3, 0)), > > - .w1c = (PCI_STATUS_PARITY | > - PCI_STATUS_SIG_TARGET_ABORT | > - PCI_STATUS_REC_TARGET_ABORT | > - PCI_STATUS_REC_MASTER_ABORT | > - PCI_STATUS_SIG_SYSTEM_ERROR | > - PCI_STATUS_DETECTED_PARITY) << 16, > + .w1c = PCI_STATUS_ERROR_BITS << 16, > > .rsvd = ((BIT(6) | GENMASK(4, 0)) << 16), > }, > -- > 2.25.1 > >

1 0

Re: [PATCH v4 05/10] PCI: Add pci_status_get_and_clear_errors
by Bjorn Helgaas 04 Mar '20

04 Mar '20

On Sat, Feb 29, 2020 at 11:24:23PM +0100, Heiner Kallweit wrote: > Several drivers use the following code sequence: > 1. Read PCI_STATUS > 2. Mask out non-error bits > 3. Action based on error bits set > 4. Write back set error bits to clear them > > As this is a repeated pattern, add a helper to the PCI core. > > Signed-off-by: Heiner Kallweit <hkallweit1(a)gmail.com> Acked-by: Bjorn Helgaas <bhelgaas(a)google.com> Nit: if you have any reason to revise this, I would prefer "pci_status_get_and_clear_errors()" (with parens) in the subject. > --- > v4: > - improve commit description > --- > drivers/pci/pci.c | 23 +++++++++++++++++++++++ > include/linux/pci.h | 1 + > 2 files changed, 24 insertions(+) > > diff --git a/drivers/pci/pci.c b/drivers/pci/pci.c > index d828ca835..c16b0ba2a 100644 > --- a/drivers/pci/pci.c > +++ b/drivers/pci/pci.c > @@ -173,6 +173,29 @@ unsigned char pci_bus_max_busnr(struct pci_bus *bus) > } > EXPORT_SYMBOL_GPL(pci_bus_max_busnr); > > +/** > + * pci_status_get_and_clear_errors - return and clear error bits in PCI_STATUS > + * @pdev: the PCI device > + * > + * Returns error bits set in PCI_STATUS and clears them. > + */ > +int pci_status_get_and_clear_errors(struct pci_dev *pdev) > +{ > + u16 status; > + int ret; > + > + ret = pci_read_config_word(pdev, PCI_STATUS, &status); > + if (ret != PCIBIOS_SUCCESSFUL) > + return -EIO; > + > + status &= PCI_STATUS_ERROR_BITS; > + if (status) > + pci_write_config_word(pdev, PCI_STATUS, status); > + > + return status; > +} > +EXPORT_SYMBOL_GPL(pci_status_get_and_clear_errors); > + > #ifdef CONFIG_HAS_IOMEM > void __iomem *pci_ioremap_bar(struct pci_dev *pdev, int bar) > { > diff --git a/include/linux/pci.h b/include/linux/pci.h > index 101d71e0a..7beaf51e9 100644 > --- a/include/linux/pci.h > +++ b/include/linux/pci.h > @@ -1210,6 +1210,7 @@ int pci_select_bars(struct pci_dev *dev, unsigned long flags); > bool pci_device_is_present(struct pci_dev *pdev); > void pci_ignore_hotplug(struct pci_dev *dev); > struct pci_dev *pci_real_dma_dev(struct pci_dev *dev); > +int pci_status_get_and_clear_errors(struct pci_dev *pdev); > > int __printf(6, 7) pci_request_irq(struct pci_dev *dev, unsigned int nr, > irq_handler_t handler, irq_handler_t thread_fn, void *dev_id, > -- > 2.25.1 > >

1 0

Re: [PATCH v4 04/10] PCI: Add constant PCI_STATUS_ERROR_BITS
by Bjorn Helgaas 04 Mar '20

04 Mar '20

On Sat, Feb 29, 2020 at 11:23:44PM +0100, Heiner Kallweit wrote: > This collection of PCI error bits is used in more than one driver, > so move it to the PCI core. > > Signed-off-by: Heiner Kallweit <hkallweit1(a)gmail.com> Acked-by: Bjorn Helgaas <bhelgaas(a)google.com> > --- > v4: > - move new constant to include/linux/pci.h > - improve commit description > --- > drivers/net/ethernet/marvell/skge.h | 7 ------- > drivers/net/ethernet/marvell/sky2.h | 7 ------- > include/linux/pci.h | 7 +++++++ > 3 files changed, 7 insertions(+), 14 deletions(-) > > diff --git a/drivers/net/ethernet/marvell/skge.h b/drivers/net/ethernet/marvell/skge.h > index e76c03c87..e149bdfe1 100644 > --- a/drivers/net/ethernet/marvell/skge.h > +++ b/drivers/net/ethernet/marvell/skge.h > @@ -15,13 +15,6 @@ > #define PCI_VPD_ROM_SZ 7L<<14 /* VPD ROM size 0=256, 1=512, ... */ > #define PCI_REV_DESC 1<<2 /* Reverse Descriptor bytes */ > > -#define PCI_STATUS_ERROR_BITS (PCI_STATUS_DETECTED_PARITY | \ > - PCI_STATUS_SIG_SYSTEM_ERROR | \ > - PCI_STATUS_REC_MASTER_ABORT | \ > - PCI_STATUS_REC_TARGET_ABORT | \ > - PCI_STATUS_SIG_TARGET_ABORT | \ > - PCI_STATUS_PARITY) > - > enum csr_regs { > B0_RAP = 0x0000, > B0_CTST = 0x0004, > diff --git a/drivers/net/ethernet/marvell/sky2.h b/drivers/net/ethernet/marvell/sky2.h > index aee87f838..851d8ed34 100644 > --- a/drivers/net/ethernet/marvell/sky2.h > +++ b/drivers/net/ethernet/marvell/sky2.h > @@ -252,13 +252,6 @@ enum { > }; > > > -#define PCI_STATUS_ERROR_BITS (PCI_STATUS_DETECTED_PARITY | \ > - PCI_STATUS_SIG_SYSTEM_ERROR | \ > - PCI_STATUS_REC_MASTER_ABORT | \ > - PCI_STATUS_REC_TARGET_ABORT | \ > - PCI_STATUS_SIG_TARGET_ABORT | \ > - PCI_STATUS_PARITY) > - > enum csr_regs { > B0_RAP = 0x0000, > B0_CTST = 0x0004, > diff --git a/include/linux/pci.h b/include/linux/pci.h > index 3840a541a..101d71e0a 100644 > --- a/include/linux/pci.h > +++ b/include/linux/pci.h > @@ -42,6 +42,13 @@ > > #include <linux/pci_ids.h> > > +#define PCI_STATUS_ERROR_BITS (PCI_STATUS_DETECTED_PARITY | \ > + PCI_STATUS_SIG_SYSTEM_ERROR | \ > + PCI_STATUS_REC_MASTER_ABORT | \ > + PCI_STATUS_REC_TARGET_ABORT | \ > + PCI_STATUS_SIG_TARGET_ABORT | \ > + PCI_STATUS_PARITY) > + > /* > * The PCI interface treats multi-function devices as independent > * devices. The slot/function address of each device is encoded > -- > 2.25.1 > >

1 0

Applied "ASoC: stm32: sai: manage rebind issue" to the asoc tree
by Mark Brown 04 Mar '20

04 Mar '20

The patch ASoC: stm32: sai: manage rebind issue has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 0d6defc7e0e437a9fd53622f7fd85740f38d5693 Mon Sep 17 00:00:00 2001 From: Olivier Moysan <olivier.moysan(a)st.com> Date: Wed, 4 Mar 2020 11:24:06 +0100 Subject: [PATCH] ASoC: stm32: sai: manage rebind issue The commit e894efef9ac7 ("ASoC: core: add support to card rebind") allows to rebind the sound card after a rebind of one of its component. With this commit, the sound card is actually rebound, but may be no more functional. The following problems have been seen with STM32 SAI driver. 1) DMA channel is not requested: With the sound card rebind the simplified call sequence is: stm32_sai_sub_probe snd_soc_register_component snd_soc_try_rebind_card snd_soc_instantiate_card devm_snd_dmaengine_pcm_register The problem occurs because the pcm must be registered, before snd_soc_instantiate_card() is called. Modify SAI driver, to change the call sequence as follows: stm32_sai_sub_probe devm_snd_dmaengine_pcm_register snd_soc_register_component snd_soc_try_rebind_card 2) DMA channel is not released: dma_release_channel() is not called when devm_dmaengine_pcm_release() is executed. This occurs because SND_DMAENGINE_PCM_DRV_NAME component, has already been released through devm_component_release(). devm_dmaengine_pcm_release() should be called before devm_component_release() to avoid this problem. Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component() explicitly from SAI driver, to have the right sequence. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> Message-Id: <20200304102406.8093-1-olivier.moysan(a)st.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/stm/stm32_sai_sub.c | 18 ++++++++++-------- 1 file changed, 10 insertions(+), 8 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 30bcd5d3a32a..10eb4b8e8e7e 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1543,20 +1543,20 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return ret; } - ret = devm_snd_soc_register_component(&pdev->dev, &stm32_component, - &sai->cpu_dai_drv, 1); + ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0); + if (ret) { + dev_err(&pdev->dev, "Could not register pcm dma\n"); + return ret; + } + + ret = snd_soc_register_component(&pdev->dev, &stm32_component, + &sai->cpu_dai_drv, 1); if (ret) return ret; if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) conf = &stm32_sai_pcm_config_spdif; - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, conf, 0); - if (ret) { - dev_err(&pdev->dev, "Could not register pcm dma\n"); - return ret; - } - return 0; } @@ -1565,6 +1565,8 @@ static int stm32_sai_sub_remove(struct platform_device *pdev) struct stm32_sai_sub_data *sai = dev_get_drvdata(&pdev->dev); clk_unprepare(sai->pdata->pclk); + snd_dmaengine_pcm_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } -- 2.20.1

1 0

[PATCH] ASoC: stm32: sai: manage rebind issue
by Olivier Moysan 04 Mar '20

04 Mar '20

The commit e894efef9ac7 ("ASoC: core: add support to card rebind") allows to rebind the sound card after a rebind of one of its component. With this commit, the sound card is actually rebound, but may be no more functional. The following problems have been seen with STM32 SAI driver. 1) DMA channel is not requested: With the sound card rebind the simplified call sequence is: stm32_sai_sub_probe snd_soc_register_component snd_soc_try_rebind_card snd_soc_instantiate_card devm_snd_dmaengine_pcm_register The problem occurs because the pcm must be registered, before snd_soc_instantiate_card() is called. Modify SAI driver, to change the call sequence as follows: stm32_sai_sub_probe devm_snd_dmaengine_pcm_register snd_soc_register_component snd_soc_try_rebind_card 2) DMA channel is not released: dma_release_channel() is not called when devm_dmaengine_pcm_release() is executed. This occurs because SND_DMAENGINE_PCM_DRV_NAME component, has already been released through devm_component_release(). devm_dmaengine_pcm_release() should be called before devm_component_release() to avoid this problem. Call snd_dmaengine_pcm_unregister() and snd_soc_unregister_component() explicitly from SAI driver, to have the right sequence. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai_sub.c | 20 +++++++++++--------- 1 file changed, 11 insertions(+), 9 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 0bbf9ed5e48b..48dfcdd13574 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -1547,21 +1547,21 @@ static int stm32_sai_sub_probe(struct platform_device *pdev) return ret; } - ret = devm_snd_soc_register_component(&pdev->dev, &stm32_component, - &sai->cpu_dai_drv, 1); - if (ret) - return ret; - - if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) - conf = &stm32_sai_pcm_config_spdif; - - ret = devm_snd_dmaengine_pcm_register(&pdev->dev, conf, 0); + ret = snd_dmaengine_pcm_register(&pdev->dev, conf, 0); if (ret) { if (ret != -EPROBE_DEFER) dev_err(&pdev->dev, "PCM DMA register error %d\n", ret); return ret; } + ret = snd_soc_register_component(&pdev->dev, &stm32_component, + &sai->cpu_dai_drv, 1); + if (ret) + return ret; + + if (STM_SAI_PROTOCOL_IS_SPDIF(sai)) + conf = &stm32_sai_pcm_config_spdif; + return 0; } @@ -1570,6 +1570,8 @@ static int stm32_sai_sub_remove(struct platform_device *pdev) struct stm32_sai_sub_data *sai = dev_get_drvdata(&pdev->dev); clk_unprepare(sai->pdata->pclk); + snd_dmaengine_pcm_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } -- 2.17.1

1 0

Build is failing with python 3.8
by GitHub issues - opened 04 Mar '20

04 Mar '20

alsa-project/alsa-lib issue #33 was opened from seb128: One build log from Ubuntu (1.1.9 but the current version has the same issue) https://launchpadlibrarian.net/462093809/buildlog_ubuntu-focal-amd64.alsa-l… The Ubuntu package has been fixed with that change http://launchpadlibrarian.net/462101751/alsa-lib_1.2.1.2-2_1.2.1.2-2ubuntu1… The issue/needed changed is explained on https://docs.python.org/3/whatsnew/3.8.html > To embed Python into an application, a new --embed option must be passed to python3-config --libs --embed to get -lpython3.8 (link the application to libpython). To support both 3.8 and older, try python3-config --libs --embed first and fallback to python3-config --libs (without --embed) if the previous command fails. The Ubuntu patch doesn't have that fallback logic though which is why I'm not proposing it with a merge request Issue URL : https://github.com/alsa-project/alsa-lib/issues/33 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: SOF: Fix snd_sof_ipc_stream_posn()
by Dan Carpenter 04 Mar '20

04 Mar '20

We're passing "&posn" instead of "posn" so it ends up corrupting memory instead of doing something useful. Fixes: 53e0c72d98ba ("ASoC: SOF: Add support for IPC IO between DSP and Host") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/sof/ipc.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/ipc.c b/sound/soc/sof/ipc.c index 22d296f95761..d6bf53dcd80b 100644 --- a/sound/soc/sof/ipc.c +++ b/sound/soc/sof/ipc.c @@ -524,7 +524,7 @@ int snd_sof_ipc_stream_posn(struct snd_soc_component *scomp, /* send IPC to the DSP */ err = sof_ipc_tx_message(sdev->ipc, - stream.hdr.cmd, &stream, sizeof(stream), &posn, + stream.hdr.cmd, &stream, sizeof(stream), posn, sizeof(*posn)); if (err < 0) { dev_err(sdev->dev, "error: failed to get stream %d position\n", -- 2.11.0

6 5

[PATCH] ASoC: soc-pcm: Revert "call snd_soc_component_open/close() once"
by Kai Vehmanen 04 Mar '20

04 Mar '20

ASoC component open/close and snd_soc_component_module_get/put are called independently for each component-substream pair, so the logic in the reverted patch was not sufficient and led to PCM playback and module unload errors. Fixes: dd03907bf129 ("ASoC: soc-pcm: call snd_soc_component_open/close() once") Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- include/sound/soc-component.h | 7 ++----- sound/soc/soc-component.c | 35 +++++++---------------------------- sound/soc/soc-pcm.c | 19 +++++++++++++------ 3 files changed, 22 insertions(+), 39 deletions(-) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 1866ecc8e94b..154d02fbbfed 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -147,6 +147,8 @@ struct snd_soc_component { unsigned int active; + unsigned int suspended:1; /* is in suspend PM state */ + struct list_head list; struct list_head card_aux_list; /* for auxiliary bound components */ struct list_head card_list; @@ -180,11 +182,6 @@ struct snd_soc_component { struct dentry *debugfs_root; const char *debugfs_prefix; #endif - - /* bit field */ - unsigned int suspended:1; /* is in suspend PM state */ - unsigned int opened:1; - unsigned int module:1; }; #define for_each_component_dais(component, dai)\ diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index ee00c09df5e7..14e175cdeeb8 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -297,55 +297,34 @@ EXPORT_SYMBOL_GPL(snd_soc_component_set_jack); int snd_soc_component_module_get(struct snd_soc_component *component, int upon_open) { - if (component->module) - return 0; - if (component->driver->module_get_upon_open == !!upon_open && !try_module_get(component->dev->driver->owner)) return -ENODEV; - component->module = 1; - return 0; } void snd_soc_component_module_put(struct snd_soc_component *component, int upon_open) { - if (component->module && - component->driver->module_get_upon_open == !!upon_open) + if (component->driver->module_get_upon_open == !!upon_open) module_put(component->dev->driver->owner); - - component->module = 0; } int snd_soc_component_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - int ret = 0; - - if (!component->opened && - component->driver->open) - ret = component->driver->open(component, substream); - - if (ret == 0) - component->opened = 1; - - return ret; + if (component->driver->open) + return component->driver->open(component, substream); + return 0; } int snd_soc_component_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - int ret = 0; - - if (component->opened && - component->driver->close) - ret = component->driver->close(component, substream); - - component->opened = 0; - - return ret; + if (component->driver->close) + return component->driver->close(component, substream); + return 0; } int snd_soc_component_prepare(struct snd_soc_component *component, diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 63f67eb7c077..0c3a12d6290a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -466,13 +466,16 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) hw->rate_max = min_not_zero(hw->rate_max, rate_max); } -static int soc_pcm_components_open(struct snd_pcm_substream *substream) +static int soc_pcm_components_open(struct snd_pcm_substream *substream, + struct snd_soc_component **last) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; int i, ret = 0; for_each_rtd_components(rtd, i, component) { + *last = component; + ret = snd_soc_component_module_get_when_open(component); if (ret < 0) { dev_err(component->dev, @@ -489,17 +492,21 @@ static int soc_pcm_components_open(struct snd_pcm_substream *substream) return ret; } } - + *last = NULL; return 0; } -static int soc_pcm_components_close(struct snd_pcm_substream *substream) +static int soc_pcm_components_close(struct snd_pcm_substream *substream, + struct snd_soc_component *last) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_component *component; int i, r, ret = 0; for_each_rtd_components(rtd, i, component) { + if (component == last) + break; + r = snd_soc_component_close(component, substream); if (r < 0) ret = r; /* use last ret */ @@ -536,7 +543,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) soc_rtd_shutdown(rtd, substream); - soc_pcm_components_close(substream); + soc_pcm_components_close(substream, NULL); snd_soc_dapm_stream_stop(rtd, substream->stream); @@ -577,7 +584,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->card->pcm_mutex, rtd->card->pcm_subclass); - ret = soc_pcm_components_open(substream); + ret = soc_pcm_components_open(substream, &component); if (ret < 0) goto component_err; @@ -682,7 +689,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) soc_rtd_shutdown(rtd, substream); component_err: - soc_pcm_components_close(substream); + soc_pcm_components_close(substream, component); mutex_unlock(&rtd->card->pcm_mutex); -- 2.17.1

6 26

[PATCH] ASoC: rt1015: modify pre-divider for sysclk
by jack.yu＠realtek.com 03 Mar '20

03 Mar '20

From: Jack Yu <jack.yu(a)realtek.com> Modify pre-divider for system clock. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index 5fa21d8ab620..bb310bc7febd 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -664,7 +664,7 @@ static int rt1015_hw_params(struct snd_pcm_substream *substream, snd_soc_component_update_bits(component, RT1015_TDM_MASTER, RT1015_I2S_DL_MASK, val_len); snd_soc_component_update_bits(component, RT1015_CLK2, - RT1015_FS_PD_MASK, pre_div); + RT1015_FS_PD_MASK, pre_div << RT1015_FS_PD_SFT); return 0; } -- 2.25.1

2 2

[PATCH] ASoC: amd: AMD RV RT5682 should depends on CROS_EC
by Enric Balletbo i Serra 03 Mar '20

03 Mar '20

If SND_SOC_AMD_RV_RT5682_MACH=y, below kconfig and build errors can be seen: WARNING: unmet direct dependencies detected for SND_SOC_CROS_EC_CODEC WARNING: unmet direct dependencies detected for I2C_CROS_EC_TUNNEL ld: drivers/i2c/busses/i2c-cros-ec-tunnel.o: in function `ec_i2c_xfer': i2c-cros-ec-tunnel.c:(.text+0x2fc): undefined reference to `cros_ec_cmd_xfer_status' ld: sound/soc/codecs/cros_ec_codec.o: in function `wov_host_event': cros_ec_codec.c:(.text+0x4fb): undefined reference to `cros_ec_get_host_event' ld: sound/soc/codecs/cros_ec_codec.o: in function `send_ec_host_command': cros_ec_codec.c:(.text+0x831): undefined reference to `cros_ec_cmd_xfer_status' This is because it will select SND_SOC_CROS_EC_CODEC and I2c_CROS_EC_TUNNEL but both depends on CROS_EC. Fixes: 6b8e4e7db3cd ("ASoC: amd: Add machine driver for Raven based platform") Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> --- sound/soc/amd/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index b29ef1373946..bce4cee5cb54 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -33,6 +33,6 @@ config SND_SOC_AMD_RV_RT5682_MACH select SND_SOC_MAX98357A select SND_SOC_CROS_EC_CODEC select I2C_CROS_EC_TUNNEL - depends on SND_SOC_AMD_ACP3x && I2C + depends on SND_SOC_AMD_ACP3x && I2C && CROS_EC help This option enables machine driver for RT5682 and MAX9835. -- 2.25.1

3 2

Applied "ASoC: amd: Fix compile warning of argument type" to the asoc tree
by Mark Brown 03 Mar '20

03 Mar '20

The patch ASoC: amd: Fix compile warning of argument type has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From d7729c40b376ab7c1ad4b0498a8bbf44bed3cbf4 Mon Sep 17 00:00:00 2001 From: Akshu Agrawal <akshu.agrawal(a)amd.com> Date: Tue, 3 Mar 2020 14:34:37 +0530 Subject: [PATCH] ASoC: amd: Fix compile warning of argument type Fixes: >> sound/soc//amd/acp3x-rt5682-max9836.c:341:23: warning: format '%d' >> expects argument of type 'int', but argument 3 has type 'long int' >> [-Wformat=] dev_err(&pdev->dev, "DMIC gpio failed err=%d\n", Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Link: https://lore.kernel.org/r/20200303090444.95805-1-akshu.agrawal@amd.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/acp3x-rt5682-max9836.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 511b8b1722aa..521c9ab4c29c 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -338,7 +338,7 @@ static int acp3x_probe(struct platform_device *pdev) dmic_sel = devm_gpiod_get(&pdev->dev, "dmic", GPIOD_OUT_LOW); if (IS_ERR(dmic_sel)) { - dev_err(&pdev->dev, "DMIC gpio failed err=%d\n", + dev_err(&pdev->dev, "DMIC gpio failed err=%ld\n", PTR_ERR(dmic_sel)); return PTR_ERR(dmic_sel); } -- 2.20.1

1 0

[PATCH v3 0/3] simple-audio-card codec2codec support
by Samuel Holland 03 Mar '20

03 Mar '20

We are currently using simple-audio-card on the Allwinner A64 SoC. The digital audio codec there (sun8i-codec) has 3 AIFs, one each for the CPU, the modem, and Bluetooth. Adding support for the secondary AIFs requires adding codec2codec DAI links. Since the modem and bt-sco codec DAI drivers only have one set of possible PCM parameters (namely, 8kHz mono S16LE), there's no real need for a machine driver to specify the DAI link configuration. The parameters for these "simple" DAI links can be chosen automatically. This series adds codec2codec DAI link support to simple-audio-card. Codec to codec links are automatically detected when all DAIs in the link belong to codec components. I tried to reuse as much code as possible, so the first two patches refactor a couple of helper functions to be more generic. The last patch adds the new feature and its documentation. Changes in v3: - Update use of for_each_rtd_components for v5.6 Changes in v2: - Drop patch 1 as it was merged - Automatically detect codec2codec links instead of using a DT property Samuel Holland (3): ALSA: pcm: Add a standalone version of snd_pcm_limit_hw_rates ASoC: pcm: Export parameter intersection logic ASoC: simple-card: Add support for codec2codec DAI links Documentation/sound/soc/codec-to-codec.rst | 9 +++- include/sound/pcm.h | 9 +++- include/sound/soc.h | 3 ++ sound/core/pcm_misc.c | 18 +++---- sound/soc/generic/simple-card-utils.c | 49 +++++++++++++++++++ sound/soc/soc-pcm.c | 55 +++++++++++++++------- 6 files changed, 115 insertions(+), 28 deletions(-) -- 2.24.1

2 6

[RFC PATCH 0/2] Introduce multi PM domains helpers
by Daniel Baluta 03 Mar '20

03 Mar '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> i.MX8QXP/i.MX8QM has IPs that need multiple power domains to be up in order to work. In order to help drivers, we introduce multi PM domains helpers that are able to activate/deactivate multi PM domains. First patch introduces the helpers and second patch demonstrates how a driver can use them instead of hardcoding the PM domains handling. Daniel Baluta (2): PM / domains: Introduce multi PM domains helpers ASoC: SOF: Use multi PM domains helpers drivers/base/power/common.c | 93 +++++++++++++++++++++++++++++++++++++ include/linux/pm_domain.h | 19 ++++++++ sound/soc/sof/imx/imx8.c | 54 +++------------------ 3 files changed, 118 insertions(+), 48 deletions(-) -- 2.17.1

3 6

[PATCH][next] ASoc: amd: use correct format specifier for a long int
by Colin King 03 Mar '20

03 Mar '20

From: Colin Ian King <colin.king(a)canonical.com> Currently the format specifier %d is being used for a long int. Fix the by using %ld instead. Fixes warning: sound/soc/amd/acp3x-rt5682-max9836.c:341:23: warning: format '%d' expects argument of type 'int', but argument 3 has type 'long int' [-Wformat=] Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 511b8b1722aa..521c9ab4c29c 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -338,7 +338,7 @@ static int acp3x_probe(struct platform_device *pdev) dmic_sel = devm_gpiod_get(&pdev->dev, "dmic", GPIOD_OUT_LOW); if (IS_ERR(dmic_sel)) { - dev_err(&pdev->dev, "DMIC gpio failed err=%d\n", + dev_err(&pdev->dev, "DMIC gpio failed err=%ld\n", PTR_ERR(dmic_sel)); return PTR_ERR(dmic_sel); } -- 2.25.0

2 1

Change PCM device number of Asus Xonar U5
by GitHub pull_request - opened 03 Mar '20

03 Mar '20

alsa-project/alsa-lib pull request #32 was opened from be-marc: Asus Xonar U5 does not use the first PCM device for digital data like the U7. Request URL : https://github.com/alsa-project/alsa-lib/pull/32 Patch URL : https://github.com/alsa-project/alsa-lib/pull/32.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: amd: Fix compile warning of argument type
by Akshu Agrawal 03 Mar '20

03 Mar '20

Fixes: >> sound/soc//amd/acp3x-rt5682-max9836.c:341:23: warning: format '%d' >> expects argument of type 'int', but argument 3 has type 'long int' >> [-Wformat=] dev_err(&pdev->dev, "DMIC gpio failed err=%d\n", Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Reported-by: kbuild test robot <lkp(a)intel.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 511b8b1722aa..521c9ab4c29c 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -338,7 +338,7 @@ static int acp3x_probe(struct platform_device *pdev) dmic_sel = devm_gpiod_get(&pdev->dev, "dmic", GPIOD_OUT_LOW); if (IS_ERR(dmic_sel)) { - dev_err(&pdev->dev, "DMIC gpio failed err=%d\n", + dev_err(&pdev->dev, "DMIC gpio failed err=%ld\n", PTR_ERR(dmic_sel)); return PTR_ERR(dmic_sel); } -- 2.17.1

1 0

[PATCH v2] soundwire: bus: provide correct return value on error
by Pierre-Louis Bossart 03 Mar '20

03 Mar '20

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> It seems to be a typo. It makes more sense to return the return value of sdw_update() instead of the value we want to update. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v2: address Vinod's comment and use single return drivers/soundwire/bus.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index ccaa590df61e..488c3c9e4947 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1056,13 +1056,10 @@ static int sdw_initialize_slave(struct sdw_slave *slave) val |= SDW_DP0_INT_PORT_READY | SDW_DP0_INT_BRA_FAILURE; ret = sdw_update(slave, SDW_DP0_INTMASK, val, val); - if (ret < 0) { + if (ret < 0) dev_err(slave->bus->dev, "SDW_DP0_INTMASK read failed:%d\n", ret); - return val; - } - - return 0; + return ret; } static int sdw_handle_dp0_interrupt(struct sdw_slave *slave, u8 *slave_status) -- 2.20.1

2 1

[PATCH AUTOSEL 4.19 01/32] ALSA: hda: do not override bus codec_mask in link_get()
by Sasha Levin 03 Mar '20

03 Mar '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 43bcb1c0507858cdc95e425017dcc33f8105df39 ] snd_hdac_ext_bus_link_get() does not work correctly in case there are multiple codecs on the bus. It unconditionally resets the bus->codec_mask value. As per documentation in hdaudio.h and existing use in client code, this field should be used to store bit flag of detected codecs on the bus. By overwriting value of the codec_mask, information on all detected codecs is lost. No current user of hdac is impacted, but use of bus->codec_mask is planned in future patches for SOF. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200206200223.7715-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/ext/hdac_ext_controller.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index 60cb00fd0c693..84b44cdae28a1 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -262,6 +262,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down_all); int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, struct hdac_ext_link *link) { + unsigned long codec_mask; int ret = 0; mutex_lock(&bus->lock); @@ -283,9 +284,11 @@ int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, * HDA spec section 4.3 - Codec Discovery */ udelay(521); - bus->codec_mask = snd_hdac_chip_readw(bus, STATESTS); - dev_dbg(bus->dev, "codec_mask = 0x%lx\n", bus->codec_mask); - snd_hdac_chip_writew(bus, STATESTS, bus->codec_mask); + codec_mask = snd_hdac_chip_readw(bus, STATESTS); + dev_dbg(bus->dev, "codec_mask = 0x%lx\n", codec_mask); + snd_hdac_chip_writew(bus, STATESTS, codec_mask); + if (!bus->codec_mask) + bus->codec_mask = codec_mask; } mutex_unlock(&bus->lock); -- 2.20.1

1 0

[PATCH AUTOSEL 5.4 01/58] ALSA: hda: do not override bus codec_mask in link_get()
by Sasha Levin 03 Mar '20

03 Mar '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 43bcb1c0507858cdc95e425017dcc33f8105df39 ] snd_hdac_ext_bus_link_get() does not work correctly in case there are multiple codecs on the bus. It unconditionally resets the bus->codec_mask value. As per documentation in hdaudio.h and existing use in client code, this field should be used to store bit flag of detected codecs on the bus. By overwriting value of the codec_mask, information on all detected codecs is lost. No current user of hdac is impacted, but use of bus->codec_mask is planned in future patches for SOF. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200206200223.7715-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/ext/hdac_ext_controller.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index cfab60d88c921..09ff209df4a30 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -254,6 +254,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down_all); int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, struct hdac_ext_link *link) { + unsigned long codec_mask; int ret = 0; mutex_lock(&bus->lock); @@ -280,9 +281,11 @@ int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, * HDA spec section 4.3 - Codec Discovery */ udelay(521); - bus->codec_mask = snd_hdac_chip_readw(bus, STATESTS); - dev_dbg(bus->dev, "codec_mask = 0x%lx\n", bus->codec_mask); - snd_hdac_chip_writew(bus, STATESTS, bus->codec_mask); + codec_mask = snd_hdac_chip_readw(bus, STATESTS); + dev_dbg(bus->dev, "codec_mask = 0x%lx\n", codec_mask); + snd_hdac_chip_writew(bus, STATESTS, codec_mask); + if (!bus->codec_mask) + bus->codec_mask = codec_mask; } mutex_unlock(&bus->lock); -- 2.20.1

1 0

[PATCH AUTOSEL 5.5 01/66] ALSA: hda: do not override bus codec_mask in link_get()
by Sasha Levin 03 Mar '20

03 Mar '20

From: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> [ Upstream commit 43bcb1c0507858cdc95e425017dcc33f8105df39 ] snd_hdac_ext_bus_link_get() does not work correctly in case there are multiple codecs on the bus. It unconditionally resets the bus->codec_mask value. As per documentation in hdaudio.h and existing use in client code, this field should be used to store bit flag of detected codecs on the bus. By overwriting value of the codec_mask, information on all detected codecs is lost. No current user of hdac is impacted, but use of bus->codec_mask is planned in future patches for SOF. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Link: https://lore.kernel.org/r/20200206200223.7715-1-kai.vehmanen@linux.intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/hda/ext/hdac_ext_controller.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/hda/ext/hdac_ext_controller.c b/sound/hda/ext/hdac_ext_controller.c index cfab60d88c921..09ff209df4a30 100644 --- a/sound/hda/ext/hdac_ext_controller.c +++ b/sound/hda/ext/hdac_ext_controller.c @@ -254,6 +254,7 @@ EXPORT_SYMBOL_GPL(snd_hdac_ext_bus_link_power_down_all); int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, struct hdac_ext_link *link) { + unsigned long codec_mask; int ret = 0; mutex_lock(&bus->lock); @@ -280,9 +281,11 @@ int snd_hdac_ext_bus_link_get(struct hdac_bus *bus, * HDA spec section 4.3 - Codec Discovery */ udelay(521); - bus->codec_mask = snd_hdac_chip_readw(bus, STATESTS); - dev_dbg(bus->dev, "codec_mask = 0x%lx\n", bus->codec_mask); - snd_hdac_chip_writew(bus, STATESTS, bus->codec_mask); + codec_mask = snd_hdac_chip_readw(bus, STATESTS); + dev_dbg(bus->dev, "codec_mask = 0x%lx\n", codec_mask); + snd_hdac_chip_writew(bus, STATESTS, codec_mask); + if (!bus->codec_mask) + bus->codec_mask = codec_mask; } mutex_unlock(&bus->lock); -- 2.20.1

1 0

[PATCH 0/9] ASoC: Intel: machine driver updates for 5.7
by Pierre-Louis Bossart 02 Mar '20

02 Mar '20

Fix HDaudio/HDMI Kconfig dependencies, cleanup code, and add support for 2 new configs (DA7219+Maxim 98360A and RT5692+RT1015) Guennadi Liakhovetski (3): ASoC: codecs: hdac_hdmi: (cosmetic) remove redundant variable initialisations ASoC: Intel: skylake: (cosmetic) remove redundant variable initialisations ASoC: Intel: (cosmetic) simplify structure member access Kai Vehmanen (4): ASoC: Intel: boards: drop reverse deps for SND_HDA_CODEC_HDMI ASoC: Intel: sof_pcm512x: drop reverse deps for SND_HDA_CODEC_HDMI ASoC: Intel: sof_pcm512x: make HDMI optional for all platforms ASoC: SOF: Intel: hda: remove SND_SOC_SOF_HDA_COMMON_HDMI_CODEC Yong Zhi (2): ASoC: Intel: sof_da7219_max98373: Add support for max98360a speaker amp ASoC: Intel: sof_rt5682: Add rt1015 speaker amp support sound/soc/codecs/hdac_hdmi.c | 6 +- sound/soc/intel/boards/Kconfig | 20 ++-- sound/soc/intel/boards/bdw-rt5650.c | 2 +- sound/soc/intel/boards/bdw-rt5677.c | 2 +- sound/soc/intel/boards/broadwell.c | 2 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 2 +- sound/soc/intel/boards/bxt_rt298.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 2 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_rt5663_max98927.c | 2 +- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 +- .../soc/intel/boards/skl_nau88l25_max98357a.c | 2 +- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 2 +- sound/soc/intel/boards/sof_da7219_max98373.c | 76 ++++++++++-- sound/soc/intel/boards/sof_pcm512x.c | 38 ++++-- sound/soc/intel/boards/sof_rt5682.c | 110 +++++++++++++++++- .../intel/common/soc-acpi-intel-jsl-match.c | 34 +++++- sound/soc/intel/skylake/skl.c | 4 +- sound/soc/sof/intel/Kconfig | 11 -- sound/soc/sof/intel/hda.c | 3 +- 26 files changed, 267 insertions(+), 69 deletions(-) base-commit: 6941b0b5f919e9839e8c25efaeb53854efee14e5 -- 2.20.1

2 12

[PATCH] ASoC: rt1015: add operation callback function for rt1015_dai[]
by jack.yu＠realtek.com 02 Mar '20

02 Mar '20

From: Jack Yu <jack.yu(a)realtek.com> Add operation callback function for rt1015_dai[]. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index c118d030bd2d..5fa21d8ab620 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -857,6 +857,7 @@ static struct snd_soc_dai_driver rt1015_dai[] = { .rates = RT1015_STEREO_RATES, .formats = RT1015_FORMATS, }, + .ops = &rt1015_aif_dai_ops, } }; -- 2.25.1

2 1

[PATCH 0/4] ASoC: SOF: updates for 5.7
by Pierre-Louis Bossart 02 Mar '20

02 Mar '20

4 unrelated improvements grouped in one bundle. Jaska Uimonen (1): ASoC: SOF: ipc: check ipc return value before data copy Keyon Jie (2): ASoC: SOF: pcm: skip DMA buffer pre-allocation ASoC: SOF: Intel: hda-loader: clear the IPC ack bit after FW_PURGE done Tomasz Lauda (1): ASoC: SOF: add core id to sof_ipc_comp include/sound/sof/topology.h | 3 ++- include/uapi/sound/sof/abi.h | 2 +- sound/soc/sof/intel/hda-loader.c | 6 ++++++ sound/soc/sof/ipc.c | 12 +++++++----- sound/soc/sof/pcm.c | 6 ++---- 5 files changed, 18 insertions(+), 11 deletions(-) base-commit: 6941b0b5f919e9839e8c25efaeb53854efee14e5 -- 2.20.1

2 8

[PATCH] ASoC: SOF: Intel: hda: use snd_sof_dsp_set_power_state() op
by Ranjani Sridharan 02 Mar '20

02 Mar '20

Replace the calls to hda_dsp_set_power_state() with the top-level SOF op snd_sof_set_power_state(). Along with this, modify the hda_dsp_resume() function to return the value of snd_sof_set_power_state() directly. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Jaska Uimonen <jaska.uimonen(a)linux.intel.com> --- sound/soc/sof/intel/hda-dsp.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index 0e61c27785a3..79ce52c32ef1 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -662,7 +662,7 @@ int hda_dsp_resume(struct snd_sof_dev *sdev) hda_codec_i915_display_power(sdev, true); /* Set DSP power state */ - ret = hda_dsp_set_power_state(sdev, &target_state); + ret = snd_sof_dsp_set_power_state(sdev, &target_state); if (ret < 0) { dev_err(sdev->dev, "error: setting dsp state %d substate %d\n", target_state.state, target_state.substate); @@ -686,8 +686,7 @@ int hda_dsp_resume(struct snd_sof_dev *sdev) if (ret < 0) return ret; - hda_dsp_set_power_state(sdev, &target_state); - return ret; + return snd_sof_dsp_set_power_state(sdev, &target_state); } int hda_dsp_runtime_resume(struct snd_sof_dev *sdev) @@ -702,7 +701,7 @@ int hda_dsp_runtime_resume(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hda_dsp_set_power_state(sdev, &target_state); + return snd_sof_dsp_set_power_state(sdev, &target_state); } int hda_dsp_runtime_idle(struct snd_sof_dev *sdev) @@ -730,7 +729,7 @@ int hda_dsp_runtime_suspend(struct snd_sof_dev *sdev) if (ret < 0) return ret; - return hda_dsp_set_power_state(sdev, &target_state); + return snd_sof_dsp_set_power_state(sdev, &target_state); } int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) @@ -753,7 +752,7 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) hda_codec_i915_display_power(sdev, false); /* Set DSP power state */ - ret = hda_dsp_set_power_state(sdev, &target_dsp_state); + ret = snd_sof_dsp_set_power_state(sdev, &target_dsp_state); if (ret < 0) { dev_err(sdev->dev, "error: setting dsp state %d substate %d\n", target_dsp_state.state, @@ -781,7 +780,7 @@ int hda_dsp_suspend(struct snd_sof_dev *sdev, u32 target_state) return ret; } - return hda_dsp_set_power_state(sdev, &target_dsp_state); + return snd_sof_dsp_set_power_state(sdev, &target_dsp_state); } int hda_dsp_set_hw_params_upon_resume(struct snd_sof_dev *sdev) @@ -849,7 +848,7 @@ void hda_dsp_d0i3_work(struct work_struct *work) return; /* This can fail but error cannot be propagated */ - ret = hda_dsp_set_power_state(sdev, &target_state); + ret = snd_sof_dsp_set_power_state(sdev, &target_state); if (ret < 0) dev_err_ratelimited(sdev->dev, "error: failed to set DSP state %d substate %d\n", -- 2.17.1

2 1

[PATCH] ASoc: amd: Add DMIC switch capability to machine driver
by Akshu Agrawal 02 Mar '20

02 Mar '20

Switch between DMIC0 and DMIC1 based on recording device selected. This is done by toggling the dmic select gpio. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp3x-rt5682-max9836.c | 53 ++++++++++++++++++++++++---- 1 file changed, 47 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c index 96fbcd29e3ed..511b8b1722aa 100644 --- a/sound/soc/amd/acp3x-rt5682-max9836.c +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -12,6 +12,7 @@ #include <sound/jack.h> #include <linux/clk.h> #include <linux/gpio.h> +#include <linux/gpio/consumer.h> #include <linux/module.h> #include <linux/i2c.h> #include <linux/input.h> @@ -27,6 +28,7 @@ static struct snd_soc_jack pco_jack; static struct clk *rt5682_dai_wclk; static struct clk *rt5682_dai_bclk; +static struct gpio_desc *dmic_sel; static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) { @@ -176,7 +178,7 @@ static int acp3x_max_startup(struct snd_pcm_substream *substream) return rt5682_clk_enable(substream); } -static int acp3x_ec_startup(struct snd_pcm_substream *substream) +static int acp3x_ec_dmic0_startup(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_card *card = rtd->card; @@ -185,6 +187,23 @@ static int acp3x_ec_startup(struct snd_pcm_substream *substream) machine->cap_i2s_instance = I2S_BT_INSTANCE; snd_soc_dai_set_bclk_ratio(codec_dai, 64); + if (dmic_sel) + gpiod_set_value(dmic_sel, 0); + + return rt5682_clk_enable(substream); +} + +static int acp3x_ec_dmic1_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->cap_i2s_instance = I2S_BT_INSTANCE; + snd_soc_dai_set_bclk_ratio(codec_dai, 64); + if (dmic_sel) + gpiod_set_value(dmic_sel, 1); return rt5682_clk_enable(substream); } @@ -204,8 +223,13 @@ static const struct snd_soc_ops acp3x_max_play_ops = { .shutdown = rt5682_shutdown, }; -static const struct snd_soc_ops acp3x_ec_cap_ops = { - .startup = acp3x_ec_startup, +static const struct snd_soc_ops acp3x_ec_cap0_ops = { + .startup = acp3x_ec_dmic0_startup, + .shutdown = rt5682_shutdown, +}; + +static const struct snd_soc_ops acp3x_ec_cap1_ops = { + .startup = acp3x_ec_dmic1_startup, .shutdown = rt5682_shutdown, }; @@ -246,12 +270,21 @@ static struct snd_soc_dai_link acp3x_dai_5682_98357[] = { SND_SOC_DAILINK_REG(acp3x_bt, max, platform), }, { - .name = "acp3x-ec-capture", - .stream_name = "Capture", + .name = "acp3x-ec-dmic0-capture", + .stream_name = "Capture DMIC0", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .dpcm_capture = 1, + .ops = &acp3x_ec_cap0_ops, + SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), + }, + { + .name = "acp3x-ec-dmic1-capture", + .stream_name = "Capture DMIC1", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, .dpcm_capture = 1, - .ops = &acp3x_ec_cap_ops, + .ops = &acp3x_ec_cap1_ops, SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), }, }; @@ -302,6 +335,14 @@ static int acp3x_probe(struct platform_device *pdev) acp3x_card.dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); + + dmic_sel = devm_gpiod_get(&pdev->dev, "dmic", GPIOD_OUT_LOW); + if (IS_ERR(dmic_sel)) { + dev_err(&pdev->dev, "DMIC gpio failed err=%d\n", + PTR_ERR(dmic_sel)); + return PTR_ERR(dmic_sel); + } + ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_card); if (ret) { dev_err(&pdev->dev, -- 2.17.1

3 2

Applied "ASoC: rt1015: set snd_soc_dai_ops in rt1015_dai driver" to the asoc tree
by Mark Brown 02 Mar '20

02 Mar '20

The patch ASoC: rt1015: set snd_soc_dai_ops in rt1015_dai driver has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a79ee2e095c0a60c32d5b1fce39d58e0fc4d9ec5 Mon Sep 17 00:00:00 2001 From: YueHaibing <yuehaibing(a)huawei.com> Date: Mon, 2 Mar 2020 15:05:22 +0800 Subject: [PATCH] ASoC: rt1015: set snd_soc_dai_ops in rt1015_dai driver snd_soc_dai_driver should set ops in rt1015_dai driver. Also make the two variable static to fix sparse warnings. Fixes: df31007400c3 ("ASoC: rt1015: add rt1015 amplifier driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> Link: https://lore.kernel.org/r/20200302070522.48104-1-yuehaibing@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/rt1015.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index c118d030bd2d..100b8c89d537 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -857,6 +857,7 @@ static struct snd_soc_dai_driver rt1015_dai[] = { .rates = RT1015_STEREO_RATES, .formats = RT1015_FORMATS, }, + .ops = &rt1015_aif_dai_ops, } }; -- 2.20.1

1 0

alsa-gobject: add support for i386 build job to workflow of Github Actions
by GitHub pull_request - opened 02 Mar '20

02 Mar '20

alsa-project/alsa-gobject pull request #9 was opened from takaswie: In lxd repository for Ubuntu Cloud Images[1], Ubuntu i386 container is available for applications of Intel386 ABI. This commit uses the container to test build for Intel386 ABI in workflow of Github Actions. I note that no official Fedora i386 container is in dockerhub registry[2]. It's needed to find the way to build application for Intel386 ABI on AMD64 architecture (perhaps multiarch) if we are going to support it. [1] https://cloud-images.ubuntu.com/releases/ [2] https://hub.docker.com/_/fedora Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Request URL : https://github.com/alsa-project/alsa-gobject/pull/9 Patch URL : https://github.com/alsa-project/alsa-gobject/pull/9.patch Repository URL: https://github.com/alsa-project/alsa-gobject

1 0

[PATCH -next] ASoC: rt1015: set snd_soc_dai_ops in rt1015_dai driver
by YueHaibing 02 Mar '20

02 Mar '20

snd_soc_dai_driver should set ops in rt1015_dai driver. Also make the two variable static to fix sparse warnings. Fixes: df31007400c3 ("ASoC: rt1015: add rt1015 amplifier driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/rt1015.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index d300b41..100b8c8 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -841,12 +841,12 @@ static void rt1015_remove(struct snd_soc_component *component) #define RT1015_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 rt1015_aif_dai_ops = { +static struct snd_soc_dai_ops rt1015_aif_dai_ops = { .hw_params = rt1015_hw_params, .set_fmt = rt1015_set_dai_fmt, }; -struct snd_soc_dai_driver rt1015_dai[] = { +static struct snd_soc_dai_driver rt1015_dai[] = { { .name = "rt1015-aif", .id = 0, @@ -857,6 +857,7 @@ struct snd_soc_dai_driver rt1015_dai[] = { .rates = RT1015_STEREO_RATES, .formats = RT1015_FORMATS, }, + .ops = &rt1015_aif_dai_ops, } }; -- 2.7.4

1 0

RE: [alsa-devel] No sound with Realtek ALC298 on a Samsung Galaxy book
by Kailang 02 Mar '20

02 Mar '20

> -----Original Message----- > From: Aurélien Croc <aurelien(a)ap2c.com> > Sent: Saturday, February 29, 2020 1:40 AM > To: alsa-devel(a)alsa-project.org > Cc: Kailang <kailang(a)realtek.com>; Takashi Iwai <tiwai(a)suse.de> > Subject: Re: [alsa-devel] No sound with Realtek ALC298 on a Samsung Galaxy > book > > > Sorry. > > Samsung didn't request this platform need to support Linux. > > I can understand that you can't work on my problem if your client (Samsung) > didn't paid for that. However is it possible to get ALC298 specs in order to > understand how-to configure it (in particular a document which explains its > COEF registers and their meaning) and I will do the job. > Hi, OK. I could try to request spec from our AE. > Thank you in advance. > Regards, > > Aurélien > > > > ------Please consider the environment before printing this e-mail.

1 0

conf: namehint - add noargs to the hint section
by GitHub pull_request - opened 01 Mar '20

01 Mar '20

alsa-project/alsa-lib pull request #31 was opened from perexg: Do not list simple surround devices in the namehint section by default. Fixes: https://github.com/alsa-project/alsa-lib/issues/27 Request URL : https://github.com/alsa-project/alsa-lib/pull/31 Patch URL : https://github.com/alsa-project/alsa-lib/pull/31.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Remove duplicate surround entries. Fix for #27
by GitHub pull_request - opened 01 Mar '20

01 Mar '20

alsa-project/alsa-lib pull request #30 was opened from DarkWav: Try to remove duplicate surround entries to fix crashes with certain apps by changing the config a bit. This would have multiple benefits. 1. It allows all the bonus devices (such as samplerate) gained from Version 1.2.2 to stay. 2. It still keeps surround devices around, just removes duplicate entries if they exist. 3. It fixes Issue #27 and all Unity3D-based games. I'm not sure about side effects tho, but they seem to be reasonably low from what I've tested. Thanks for taking a look. Request URL : https://github.com/alsa-project/alsa-lib/pull/30 Patch URL : https://github.com/alsa-project/alsa-lib/pull/30.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Assertion 'jack' failed at modules/alsa/alsa-ucm.c:2158
by GitHub issues - edited 01 Mar '20

01 Mar '20

alsa-project/alsa-ucm-conf issue #14 was edited from JeremyFiggins: Hi, I have the following hardware: > ~> cat /proc/asound/cards > 0 [NVidia ]: HDA-Intel - HDA NVidia > HDA NVidia at 0xf7080000 irq 90 > 1 [Generic ]: HDA-Intel - HD-Audio Generic > HDAudio-Gigabyte-ALC1220DualCodecs and I'm running: > ~> cat /etc/os-release > NAME="openSUSE Tumbleweed" > VERSION="20200228" > ID="opensuse-tumbleweed" > ID_LIKE="opensuse suse" > VERSION_ID="20200228" > PRETTY_NAME="openSUSE Tumbleweed" > ANSI_COLOR="0;32" > CPE_NAME="cpe:/o:opensuse:tumbleweed:20200228" > BUG_REPORT_URL="https://bugs.opensuse.org" > HOME_URL="https://www.opensuse.org/" > LOGO="distributor-logo" Audio was working until a recent update. Now, I have no audio and when starting PulseAudio, I get: > ~> pulseaudio > W: [pulseaudio] pid.c: Stale PID file, overwriting. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Mic1, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Mic2, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Line2, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Headphones, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Line1, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Speaker, assuming stereo duplex. > E: [pulseaudio] alsa-ucm.c: Assertion 'jack' failed at modules/alsa/alsa-ucm.c:2158, function device_add_hw_mute_jack(). Aborting. > Aborted (core dumped) I tracked it down to /usr/share/alsa/ucm2/HDA-Intel/HiFi-dual.conf. If I revert the two commits here (https://github.com/alsa-project/alsa-ucm-conf/commits/master/ucm2/HDA-Intel…) and then use the version of HiFi-dual.conf before it was moved, then PulseAudio starts normally and I have sound. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/14 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Assertion 'jack' failed at modules/alsa/alsa-ucm.c:2158
by GitHub issues - opened 01 Mar '20

01 Mar '20

alsa-project/alsa-ucm-conf issue #14 was opened from JeremyFiggins: Hi, I have the following hardware: > ~> cat /proc/asound/cards > 0 [NVidia ]: HDA-Intel - HDA NVidia > HDA NVidia at 0xf7080000 irq 90 > 1 [Generic ]: HDA-Intel - HD-Audio Generic > HDAudio-Gigabyte-ALC1220DualCodecs and I'm running: > ~> cat /etc/os-release > NAME="openSUSE Tumbleweed" > # VERSION="20200228" > ID="opensuse-tumbleweed" > ID_LIKE="opensuse suse" > VERSION_ID="20200228" > PRETTY_NAME="openSUSE Tumbleweed" > ANSI_COLOR="0;32" > CPE_NAME="cpe:/o:opensuse:tumbleweed:20200228" > BUG_REPORT_URL="https://bugs.opensuse.org" > HOME_URL="https://www.opensuse.org/" > LOGO="distributor-logo" Audio was working until a recent update. Now, I have no audio and when starting PulseAudio, I get: > ~> pulseaudio > W: [pulseaudio] pid.c: Stale PID file, overwriting. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Mic1, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Mic2, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Line2, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Headphones, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Line1, assuming stereo duplex. > W: [pulseaudio] alsa-ucm.c: UCM file does not specify 'PlaybackChannels' or 'CaptureChannels'for device Speaker, assuming stereo duplex. > E: [pulseaudio] alsa-ucm.c: Assertion 'jack' failed at modules/alsa/alsa-ucm.c:2158, function device_add_hw_mute_jack(). Aborting. > Aborted (core dumped) I tracked it down to /usr/share/alsa/ucm2/HDA-Intel/HiFi-dual.conf. If I revert the two commits here (https://github.com/alsa-project/alsa-ucm-conf/commits/master/ucm2/HDA-Intel…) and then use the version of HiFi-dual.conf before it was moved, then PulseAudio starts normally and I have sound. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/14 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 0/3] Clean up RK3328 audio codec GPIO control
by Robin Murphy 01 Mar '20

01 Mar '20

Hi all, Since nobody said anything resembling "that looks like a terrible idea!" to what I propsed on top of v1, resending as a full v2 per Mark's request. Robin. Robin Murphy (3): ASoC: dt-bindings: Make RK3328 codec GPIO explicit ASoC: rockchip: Make RK3328 GPIO_MUTE control explicit arm64: dts: rockchip: Describe RK3328 GPIO_MUTE users .../bindings/sound/rockchip,rk3328-codec.txt | 7 ++++- arch/arm64/boot/dts/rockchip/rk3328-a1.dts | 1 + .../arm64/boot/dts/rockchip/rk3328-rock64.dts | 1 + sound/soc/codecs/rk3328_codec.c | 31 ++++++++++--------- 4 files changed, 24 insertions(+), 16 deletions(-) -- 2.17.1

4 8

alsa-lib 1.2.2 kills my unity3d games
by Thomas J. Moore 29 Feb '20

29 Feb '20

Sorry if this has already been reported, but I only really subscribed to report this bug, and nothing recent in the archive jumped out at me. Some time after Jan 12, some change in my (Gentoo Linux) system caused 67 of my 72 commercial, closed-source unity3d games to crash on startup. gdb was no help, but strace showed alsa activity before two crashes, and, sure enough, reverting alsa-lib to 1.2.1.2 cured it. I bisected to this commit (5ee5ef31b "namehint: correct the @args check"): https://git.alsa-project.org/?p=alsa-lib.git;a=blobdiff;f=src/control/nameh… I don't know what the exact problem is, but reverting just this commit was enough to make everything work again (and its later modification in 6055f8a58 which doesn't make sense looking at the code as the function always returns 0 and str may still be NULL on exit, but adding a check for NULL str is not enough to stop unity3d from crashing).

2 1

[alsa-devel] No sound with Realtek ALC298 on a Samsung Galaxy book
by Aurélien Croc 29 Feb '20

29 Feb '20

Dear ALSA community, I just bought a Samsung Galaxy book 12" laptop and I installed the latest Fedora version (31) on it. Unfortunately when I try to listen something there is absolutely no sound from speaker nor headsets. I checked that volumes of the different channels were max. I also checked for muting system but I didn't found anything. Of course it works perfectly under Windows. I saw on internet that many ALC298 users had troubles with it. I tried different options to the intel hda driver (especially the model one) but it changes nothing. Here is the alsa-info.sh output: http://alsa-project.org/db/? f=871881e295972b9ecf252b25e90d659e38d939b8 I would appreciate some help in order to find a solution. Tell me if you need more information. Thanks in advance. Aurélien

3 8

alsa-lib 1.2.2 breaks Shovel Knight, 1.2.1.2 is still working.
by GitHub issues - reopened 29 Feb '20

29 Feb '20

alsa-project/alsa-lib issue #27 was reopened from DarkWav: So, today after upgrading to alsa-lib 1.2.2 I noticed the game "Shovel Knight" does no longer launch. Any Ideas why this could be happening? Sadly steam or the game executable won't provide any usable debug info when looking at terminal output :( . Can I maybe use GDB to gather more info or is that useless in this case? Issue URL : https://github.com/alsa-project/alsa-lib/issues/27 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

alsa-lib 1.2.2 breaks Shovel Knight, 1.2.1.2 is still working.
by GitHub issues - reopened 29 Feb '20

29 Feb '20

alsa-project/alsa-lib issue #27 was reopened from DarkWav: So, today after upgrading to alsa-lib 1.2.2 I noticed the game "Shovel Knight" does no longer launch. Any Ideas why this could be happening? Sadly steam or the game executable won't provide any usable debug info when looking at terminal output :( . Can I maybe use GDB to gather more info or is that useless in this case? Issue URL : https://github.com/alsa-project/alsa-lib/issues/27 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH] ASoC: dapm: Correct DAPM handling of active widgets during shutdown
by Charles Keepax 28 Feb '20

28 Feb '20

commit c2caa4da46a4 ("ASoC: Fix widget powerdown on shutdown") added a set of the power state during snd_soc_dapm_shutdown to ensure the widgets powered off. However, when commit 39eb5fd13dff ("ASoC: dapm: Delay w->power update until the changes are written") added the new_power member of the widget structure, to differentiate between the current power state and the target power state, it did not update the shutdown to use the new_power member. As new_power has not updated it will be left in the state set by the last DAPM sequence, ie. 1 for active widgets. So as the DAPM sequence for the shutdown proceeds it will turn the widgets on (despite them already being on) rather than turning them off. Fixes: 39eb5fd13dff ("ASoC: dapm: Delay w->power update until the changes are written") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.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 6ce024d52170e..83e7ce9e4aba8 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4794,7 +4794,7 @@ static void soc_dapm_shutdown_dapm(struct snd_soc_dapm_context *dapm) continue; if (w->power) { dapm_seq_insert(w, &down_list, false); - w->power = 0; + w->new_power = 0; powerdown = 1; } } -- 2.11.0

2 1

[PATCH] ASoC: soc-component: tidyup snd_soc_pcm_component_sync_stop()
by Kuninori Morimoto 28 Feb '20

28 Feb '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 1e5ddb6ba73894 ("ASoC: component: Add sync_stop PCM ops") added snd_soc_pcm_component_sync_stop(), but it is checking ioctrl instead of sync_stop. This is bug. This patch fixup it. Fixes: commit 1e5ddb6ba73894 ("ASoC: component: Add sync_stop PCM ops") Cc: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-component.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index eed07a4d85ed..76f4b953563c 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -339,7 +339,7 @@ int snd_soc_pcm_component_sync_stop(struct snd_pcm_substream *substream) int i, ret; for_each_rtd_components(rtd, i, component) { - if (component->driver->ioctl) { + if (component->driver->sync_stop) { ret = component->driver->sync_stop(component, substream); if (ret < 0) -- 2.17.1

2 1

[PATCH] ASoC: rt1015: modify some structure to be static.
by jack.yu＠realtek.com 28 Feb '20

28 Feb '20

From: Jack Yu <jack.yu(a)realtek.com> Modify rt1015_aif_dai_ops and rt1015_dai[] to be static. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index d300b417dd50..c118d030bd2d 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -841,12 +841,12 @@ static void rt1015_remove(struct snd_soc_component *component) #define RT1015_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 rt1015_aif_dai_ops = { +static struct snd_soc_dai_ops rt1015_aif_dai_ops = { .hw_params = rt1015_hw_params, .set_fmt = rt1015_set_dai_fmt, }; -struct snd_soc_dai_driver rt1015_dai[] = { +static struct snd_soc_dai_driver rt1015_dai[] = { { .name = "rt1015-aif", .id = 0, -- 2.25.1

2 1

[PATCH -next] ASoC: rt5682: Make rt5682_clock_config static
by YueHaibing 28 Feb '20

28 Feb '20

Fix sparse warning: sound/soc/codecs/rt5682-sdw.c:163:5: warning: symbol 'rt5682_clock_config' was not declared. Should it be static? Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/codecs/rt5682-sdw.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c index fc31d04..1d6963d 100644 --- a/sound/soc/codecs/rt5682-sdw.c +++ b/sound/soc/codecs/rt5682-sdw.c @@ -160,7 +160,7 @@ static int rt5682_read_prop(struct sdw_slave *slave) #define RT5682_CLK_FREQ_2400000HZ 2400000 #define RT5682_CLK_FREQ_12288000HZ 12288000 -int rt5682_clock_config(struct device *dev) +static int rt5682_clock_config(struct device *dev) { struct rt5682_priv *rt5682 = dev_get_drvdata(dev); unsigned int clk_freq, value; -- 2.7.4

2 1

[PATCH v2] ASoC: samsung: Silence warnings during deferred probe
by Marek Szyprowski 28 Feb '20

28 Feb '20

Don't confuse user with meaningless warning about the failure in getting resources and registering card in case of deferred probe. Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> Reviewed-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/arndale.c | 4 +++- sound/soc/samsung/littlemill.c | 2 +- sound/soc/samsung/lowland.c | 2 +- sound/soc/samsung/odroid.c | 4 +++- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/smdk_wm8994pcm.c | 2 +- sound/soc/samsung/snow.c | 4 +++- sound/soc/samsung/speyside.c | 2 +- sound/soc/samsung/tm2_wm5110.c | 3 ++- sound/soc/samsung/tobermory.c | 2 +- 10 files changed, 17 insertions(+), 10 deletions(-) diff --git a/sound/soc/samsung/arndale.c b/sound/soc/samsung/arndale.c index d64602950cbd..6e6d67d6e0ab 100644 --- a/sound/soc/samsung/arndale.c +++ b/sound/soc/samsung/arndale.c @@ -174,7 +174,9 @@ static int arndale_audio_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(card->dev, card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "snd_soc_register_card() failed: %d\n", ret); goto err_put_of_nodes; } return 0; diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 59904f44118b..2f2f83a8c23a 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -325,7 +325,7 @@ static int littlemill_probe(struct platform_device *pdev) card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index 098eefc764db..fcc7897ee7d0 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -183,7 +183,7 @@ static int lowland_probe(struct platform_device *pdev) card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index f0f5fa9c27d3..30c7e1bc2a30 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -311,7 +311,9 @@ static int odroid_audio_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(dev, card); if (ret < 0) { - dev_err(dev, "snd_soc_register_card() failed: %d\n", ret); + if (ret != -EPROBE_DEFER) + dev_err(dev, "snd_soc_register_card() failed: %d\n", + ret); goto err_put_clk_i2s; } diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 28f8be000aa1..8fa5f6b387ad 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -178,7 +178,7 @@ static int smdk_audio_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret); return ret; diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index 2e3dc7320c62..6e44f7927852 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -118,7 +118,7 @@ static int snd_smdk_probe(struct platform_device *pdev) smdk_pcm.dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, &smdk_pcm); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret); return ret; diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index f075aae9561a..bebcf0a4d608 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -216,7 +216,9 @@ static int snow_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(dev, card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "snd_soc_register_card failed (%d)\n", ret); return ret; } diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index ea0d1ec67f01..8f175f204eb7 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -330,7 +330,7 @@ static int speyside_probe(struct platform_device *pdev) card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); diff --git a/sound/soc/samsung/tm2_wm5110.c b/sound/soc/samsung/tm2_wm5110.c index 10ff14b856f2..043a287728b3 100644 --- a/sound/soc/samsung/tm2_wm5110.c +++ b/sound/soc/samsung/tm2_wm5110.c @@ -611,7 +611,8 @@ static int tm2_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(dev, card); if (ret < 0) { - dev_err(dev, "Failed to register card: %d\n", ret); + if (ret != -EPROBE_DEFER) + dev_err(dev, "Failed to register card: %d\n", ret); goto dai_node_put; } diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index fdce28cc26c4..1aa3fdb4b152 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -229,7 +229,7 @@ static int tobermory_probe(struct platform_device *pdev) card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); -- 2.17.1

3 5

[PATCH] ASoC: soc-dapm: don't use rtd->cpu_dai on for_each_rtd_cpu_dai()
by Kuninori Morimoto 28 Feb '20

28 Feb '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc_dapm_stream_event() is using for_each_rtd_cpu_dais(). It should use "cpu_dai", instead of "rtd->cpu_dai". This patch fixup it. Fixes: commit de6214a33633d ("ASoC: Add multiple CPU DAI support in DAPM") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.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 6ce024d52170..9a809f2caa10 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4438,7 +4438,7 @@ static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, int i; for_each_rtd_cpu_dai(rtd, i, cpu_dai) - soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event); + soc_dapm_dai_stream_event(cpu_dai, stream, event); for_each_rtd_codec_dai(rtd, i, codec_dai) soc_dapm_dai_stream_event(codec_dai, stream, event); -- 2.17.1

2 1

Applied "ASoC: zte: zx-spdif: remove redundant dev_err message" to the asoc tree
by Mark Brown 28 Feb '20

28 Feb '20

The patch ASoC: zte: zx-spdif: remove redundant dev_err message has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a57ec83a7104eab6f08215702067fbcbef90c0a0 Mon Sep 17 00:00:00 2001 From: tangbin <tangbin(a)cmss.chinamobile.com> Date: Thu, 27 Feb 2020 23:07:01 +0800 Subject: [PATCH] ASoC: zte: zx-spdif: remove redundant dev_err message devm_ioremap_resource has already contains error message, so remove the redundant dev_err message Signed-off-by: tangbin <tangbin(a)cmss.chinamobile.com> Link: https://lore.kernel.org/r/20200227150701.15652-1-tangbin@cmss.chinamobile.c… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/zte/zx-spdif.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/zte/zx-spdif.c b/sound/soc/zte/zx-spdif.c index 60382ec23832..a3a07c0730e6 100644 --- a/sound/soc/zte/zx-spdif.c +++ b/sound/soc/zte/zx-spdif.c @@ -322,7 +322,6 @@ static int zx_spdif_probe(struct platform_device *pdev) zx_spdif->mapbase = res->start; zx_spdif->reg_base = devm_ioremap_resource(&pdev->dev, res); if (IS_ERR(zx_spdif->reg_base)) { - dev_err(&pdev->dev, "ioremap failed!\n"); return PTR_ERR(zx_spdif->reg_base); } -- 2.20.1

1 0

[PATCH 0/9] ASoC: meson: convert dt-bindings to schema
by Jerome Brunet 28 Feb '20

28 Feb '20

This patchset converts the amlogic sound devices to schema. Jerome Brunet (9): ASoC: meson: gx-card: fix sound-dai dt schema ASoC: meson: convert axg tdm interface to schema ASoC: meson: convert axg tdm formatters to schema ASoC: meson: convert axg pdm to schema ASoC: meson: convert axg fifo to schema ASoC: meson: convert axg spdif input to schema ASoC: meson: convert axg spdif output to schema ASoC: meson: convert g12a tohdmitx control to schema ASoC: meson: convert axg sound card control to schema .../bindings/sound/amlogic,axg-fifo.txt | 34 ---- .../bindings/sound/amlogic,axg-fifo.yaml | 111 +++++++++++ .../bindings/sound/amlogic,axg-pdm.txt | 29 --- .../bindings/sound/amlogic,axg-pdm.yaml | 79 ++++++++ .../bindings/sound/amlogic,axg-sound-card.txt | 124 ------------ .../sound/amlogic,axg-sound-card.yaml | 181 ++++++++++++++++++ .../bindings/sound/amlogic,axg-spdifin.txt | 27 --- .../bindings/sound/amlogic,axg-spdifin.yaml | 84 ++++++++ .../bindings/sound/amlogic,axg-spdifout.txt | 25 --- .../bindings/sound/amlogic,axg-spdifout.yaml | 77 ++++++++ .../sound/amlogic,axg-tdm-formatters.txt | 36 ---- .../sound/amlogic,axg-tdm-formatters.yaml | 92 +++++++++ .../bindings/sound/amlogic,axg-tdm-iface.txt | 22 --- .../bindings/sound/amlogic,axg-tdm-iface.yaml | 57 ++++++ .../bindings/sound/amlogic,g12a-tohdmitx.txt | 58 ------ .../bindings/sound/amlogic,g12a-tohdmitx.yaml | 53 +++++ .../bindings/sound/amlogic,gx-sound-card.yaml | 4 +- 17 files changed, 736 insertions(+), 357 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-fifo.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-fifo.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-pdm.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-pdm.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-sound-card.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifin.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-spdifout.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-formatters.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,axg-tdm-iface.yaml delete mode 100644 Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt create mode 100644 Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.yaml -- 2.24.1

2 15

[PATCH 0/2 V2] TAS5756M codec driver
by Charles-Antoine Couret 28 Feb '20

28 Feb '20

V2 changes: * Fix bug in driver itself * Change device tree attribute names * Convert device tree documentation from txt to YAML format Charles-Antoine Couret (2): ASoC: add TAS5756m driver ASoC: tas5756m: Add DT binding document .../devicetree/bindings/sound/tas5756m.yaml | 64 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5756m.c | 2161 ++++++++ sound/soc/codecs/tas5756m.h | 351 ++ sound/soc/codecs/tas5756m_hf3_cfg.h | 4508 ++++++++++++++++ sound/soc/codecs/tas5756m_hf4_cfg.h | 4510 +++++++++++++++++ sound/soc/codecs/tas5756m_hf6_cfg.h | 4510 +++++++++++++++++ sound/soc/codecs/tas5756m_hf7_cfg.h | 4510 +++++++++++++++++ 9 files changed, 20624 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas5756m.yaml create mode 100644 sound/soc/codecs/tas5756m.c create mode 100644 sound/soc/codecs/tas5756m.h create mode 100644 sound/soc/codecs/tas5756m_hf3_cfg.h create mode 100644 sound/soc/codecs/tas5756m_hf4_cfg.h create mode 100644 sound/soc/codecs/tas5756m_hf6_cfg.h create mode 100644 sound/soc/codecs/tas5756m_hf7_cfg.h -- 2.24.1

2 2

[PATCH] ASoC: samsung: Silence warnings during deferred probe
by Marek Szyprowski 28 Feb '20

28 Feb '20

Don't confuse user with meaningless warning about the failure in getting resources and registering card in case of deferred probe. Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- sound/soc/samsung/arndale.c | 4 +++- sound/soc/samsung/littlemill.c | 2 +- sound/soc/samsung/lowland.c | 2 +- sound/soc/samsung/odroid.c | 4 +++- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/smdk_wm8994pcm.c | 2 +- sound/soc/samsung/snow.c | 4 +++- sound/soc/samsung/speyside.c | 2 +- sound/soc/samsung/tobermory.c | 2 +- 9 files changed, 15 insertions(+), 9 deletions(-) diff --git a/sound/soc/samsung/arndale.c b/sound/soc/samsung/arndale.c index d64602950cbd..6e6d67d6e0ab 100644 --- a/sound/soc/samsung/arndale.c +++ b/sound/soc/samsung/arndale.c @@ -174,7 +174,9 @@ static int arndale_audio_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(card->dev, card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "snd_soc_register_card() failed: %d\n", ret); goto err_put_of_nodes; } return 0; diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index 59904f44118b..2f2f83a8c23a 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -325,7 +325,7 @@ static int littlemill_probe(struct platform_device *pdev) card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); diff --git a/sound/soc/samsung/lowland.c b/sound/soc/samsung/lowland.c index 098eefc764db..fcc7897ee7d0 100644 --- a/sound/soc/samsung/lowland.c +++ b/sound/soc/samsung/lowland.c @@ -183,7 +183,7 @@ static int lowland_probe(struct platform_device *pdev) card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index f0f5fa9c27d3..30c7e1bc2a30 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -311,7 +311,9 @@ static int odroid_audio_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(dev, card); if (ret < 0) { - dev_err(dev, "snd_soc_register_card() failed: %d\n", ret); + if (ret != -EPROBE_DEFER) + dev_err(dev, "snd_soc_register_card() failed: %d\n", + ret); goto err_put_clk_i2s; } diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 28f8be000aa1..8fa5f6b387ad 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -178,7 +178,7 @@ static int smdk_audio_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret); return ret; diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c index 2e3dc7320c62..6e44f7927852 100644 --- a/sound/soc/samsung/smdk_wm8994pcm.c +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -118,7 +118,7 @@ static int snd_smdk_probe(struct platform_device *pdev) smdk_pcm.dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, &smdk_pcm); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret); return ret; diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index f075aae9561a..bebcf0a4d608 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -216,7 +216,9 @@ static int snow_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(dev, card); if (ret) { - dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "snd_soc_register_card failed (%d)\n", ret); return ret; } diff --git a/sound/soc/samsung/speyside.c b/sound/soc/samsung/speyside.c index ea0d1ec67f01..8f175f204eb7 100644 --- a/sound/soc/samsung/speyside.c +++ b/sound/soc/samsung/speyside.c @@ -330,7 +330,7 @@ static int speyside_probe(struct platform_device *pdev) card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); diff --git a/sound/soc/samsung/tobermory.c b/sound/soc/samsung/tobermory.c index fdce28cc26c4..1aa3fdb4b152 100644 --- a/sound/soc/samsung/tobermory.c +++ b/sound/soc/samsung/tobermory.c @@ -229,7 +229,7 @@ static int tobermory_probe(struct platform_device *pdev) card->dev = &pdev->dev; ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) + if (ret && ret != -EPROBE_DEFER) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); -- 2.17.1

2 1

[PATCH 1/2] mfd: wm8994: Silence warning about supplies during deferred probe
by Marek Szyprowski 28 Feb '20

28 Feb '20

Don't confuse user with meaningless warning about the failure in getting supplies in case of deferred probe. Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- drivers/mfd/wm8994-core.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index 1e9fe7d92597..bfee2f8ba17f 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -393,7 +393,9 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) ret = regulator_bulk_get(wm8994->dev, wm8994->num_supplies, wm8994->supplies); if (ret != 0) { - dev_err(wm8994->dev, "Failed to get supplies: %d\n", ret); + if (ret != -EPROBE_DEFER) + dev_err(wm8994->dev, "Failed to get supplies: %d\n", + ret); goto err; } -- 2.17.1

1 1

libavresample is deprecated since January 2017 swith to libswresample
by GitHub issues - opened 28 Feb '20

28 Feb '20

alsa-project/alsa-plugins issue #7 was opened from marillat: As the subject said, would be nice if you can switch from libavresample to libswresample Issue URL : https://github.com/alsa-project/alsa-plugins/issues/7 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH v3 00/10] add ASoC components for AHUB
by Sameer Pujar 28 Feb '20

28 Feb '20

Overview ======== The Audio Hub (AHUB) is part of the Audio Processing Engine (APE) which comprises a collection of hardware accelerators for audio pre-processing and post-processing. It also includes a programmable full crossbar for routing audio data across these accelerators. This series exposes some of these below mentioned HW devices as ASoC components for Tegra platforms from Tegra210 onwards. * ADMAIF : The interface between ADMA and AHUB * XBAR : Crossbar for routing audio samples across various modules * I2S : Inter-IC Sound Controller * DMIC : Digital Microphone * DSPK : Digital Speaker Following is the summary of current series. 1. Add YAML DT binding documentation for above mentioned modules. 2. Helper function for ACIF programming is exposed for Tegra210 and later. 3. Add ASoC driver components for each of the above modules. 4. Add DT entries for above components for Tegra210, Tegra186 and Tegra194. 5. Enable these components for Jetson-Tx1, Jetson-Tx2 and Jetson-Xavier. Machine driver series will be sent separately. Changelog ========= v2 -> v3 -------- * [1/10] "dt-bindings: sound: tegra: add DT binding for AHUB - Updated licence - Removed redundancy w.r.t items/const/enum - Added constraints wherever needed with "pattern" property * [2/10] "ASoC: tegra: add support for CIF programming" - Removed tegra_cif.c - Instead added inline helper function in tegra_cif.h * common changes (for patch [3/10] to [7/10]) - Replace LATE system calls with Normal sleep - Remove explicit RPM suspend in driver remove() call - Use devm_kzalloc() instead of devm_kcalloc() for single element - Replace 'ret' with 'err' for better reading - Consistent error printing style across drivers - Minor formating fixes * [8/10] "arm64: tegra: add AHUB components for few Tegra chips" - no change * [9/10] "arm64: tegra: enable AHUB modules for few Tegra chips" - no change * [10/10] "arm64: defconfig: enable AHUB components for Tegra210 and later" (New patch) - Enables ACONNECT and AHUB components. With this AHUB and components are registered with ASoC core. v1 -> v2 -------- * [1/9] "dt-bindings: sound: tegra: add DT binding for AHUB" - no changes * [2/9] "ASoC: tegra: add support for CIF programming" - removed CIF programming changes for legacy chips. - this patch now exposes helper function for CIF programming, which can be used on Tegra210 later. - later tegra_cif.c can be extended for legacy chips as well. - updated commit message accordingly * [3/9] "ASoC: tegra: add Tegra210 based DMIC driver" - removed unnecessary initialization of 'ret' in probe() * [4/9] "ASoC: tegra: add Tegra210 based I2S driver" - removed unnecessary initialization of 'ret' in probe() - fixed indentation - added consistent bracing for if-else clauses - updated 'rx_fifo_th' type to 'unsigned int' - used BIT() macro for defines like '1 << {x}' in tegra210_i2s.h * [5/9] "ASoC: tegra: add Tegra210 based AHUB driver" - used of_device_get_match_data() to get 'soc_data' and removed explicit of_match_device() - used devm_platform_ioremap_resource() and removed explicit platform_get_resource() - fixed indentation for devm_snd_soc_register_component() - updated commit message - updated commit message to reflect compatible binding for Tegra186 and Tegra194. * [6/9] "ASoC: tegra: add Tegra186 based DSPK driver" - removed unnecessary initialization of 'ret' in probe() - updated 'max_th' to 'unsigned int' - shortened lengthy macro names to avoid wrapping in tegra186_dspk_wr_reg() and to be consistent * [7/9] "ASoC: tegra: add Tegra210 based ADMAIF driver" - used of_device_get_match_data() and removed explicit of_match_device() - used BIT() macro for defines like '1 << {x}' in tegra210_admaif.h - updated commit message to reflect compatible binding for Tegra186 and Tegra194. * [8/9] "arm64: tegra: add AHUB components for few Tegra chips" - no change * [9/9] "arm64: tegra: enable AHUB modules for few Tegra chips" - no change * common changes for patch [3/9] to [7/9] - sorted headers in alphabetical order - moved MODULE_DEVICE_TABLE() right below *_of_match table - removed macro DRV_NAME - removed explicit 'owner' field from platform_driver structure - added 'const' to snd_soc_dai_ops structure ================= Sameer Pujar (10): dt-bindings: sound: tegra: add DT binding for AHUB ASoC: tegra: add support for CIF programming ASoC: tegra: add Tegra210 based DMIC driver ASoC: tegra: add Tegra210 based I2S driver ASoC: tegra: add Tegra210 based AHUB driver ASoC: tegra: add Tegra186 based DSPK driver ASoC: tegra: add Tegra210 based ADMAIF driver arm64: tegra: add AHUB components for few Tegra chips arm64: tegra: enable AHUB modules for few Tegra chips arm64: defconfig: enable AHUB components for Tegra210 and later .../bindings/sound/nvidia,tegra186-dspk.yaml | 88 ++ .../bindings/sound/nvidia,tegra210-admaif.yaml | 163 ++++ .../bindings/sound/nvidia,tegra210-ahub.yaml | 110 +++ .../bindings/sound/nvidia,tegra210-dmic.yaml | 88 ++ .../bindings/sound/nvidia,tegra210-i2s.yaml | 101 +++ arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 48 ++ arch/arm64/boot/dts/nvidia/tegra186.dtsi | 231 ++++- arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 36 + arch/arm64/boot/dts/nvidia/tegra194.dtsi | 239 +++++- arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 40 + arch/arm64/boot/dts/nvidia/tegra210.dtsi | 145 ++++ arch/arm64/configs/defconfig | 7 + sound/soc/tegra/Kconfig | 56 ++ sound/soc/tegra/Makefile | 10 + sound/soc/tegra/tegra186_dspk.c | 510 +++++++++++ sound/soc/tegra/tegra186_dspk.h | 73 ++ sound/soc/tegra/tegra210_admaif.c | 884 +++++++++++++++++++ sound/soc/tegra/tegra210_admaif.h | 164 ++++ sound/soc/tegra/tegra210_ahub.c | 651 ++++++++++++++ sound/soc/tegra/tegra210_ahub.h | 125 +++ sound/soc/tegra/tegra210_dmic.c | 515 +++++++++++ sound/soc/tegra/tegra210_dmic.h | 85 ++ sound/soc/tegra/tegra210_i2s.c | 938 +++++++++++++++++++++ sound/soc/tegra/tegra210_i2s.h | 132 +++ sound/soc/tegra/tegra_cif.h | 63 ++ sound/soc/tegra/tegra_pcm.c | 222 ++++- sound/soc/tegra/tegra_pcm.h | 23 +- 27 files changed, 5743 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml create mode 100644 sound/soc/tegra/tegra186_dspk.c create mode 100644 sound/soc/tegra/tegra186_dspk.h create mode 100644 sound/soc/tegra/tegra210_admaif.c create mode 100644 sound/soc/tegra/tegra210_admaif.h create mode 100644 sound/soc/tegra/tegra210_ahub.c create mode 100644 sound/soc/tegra/tegra210_ahub.h create mode 100644 sound/soc/tegra/tegra210_dmic.c create mode 100644 sound/soc/tegra/tegra210_dmic.h create mode 100644 sound/soc/tegra/tegra210_i2s.c create mode 100644 sound/soc/tegra/tegra210_i2s.h create mode 100644 sound/soc/tegra/tegra_cif.h -- 2.7.4

7 38

[PATCH v3 0/4] ASoC: Add new module driver for new ASRC
by Shengjiu Wang 28 Feb '20

28 Feb '20

Add new module driver for new ASRC in i.MX8MN Shengjiu Wang (4): ASoC: fsl_asrc: Change asrc_width to asrc_format ASoC: fsl_asrc: Move common definition to fsl_asrc_common ASoC: dt-bindings: fsl_easrc: Add document for EASRC ASoC: fsl_easrc: Add EASRC ASoC CPU DAI and platform drivers changes in v3 - add new commit "ASoC: fsl_asrc: Change asrc_width to asrc_format" - modify binding doc to yaml format - remove fsl_easrc_dma.c, make fsl_asrc_dma.c useable for easrc. changes in v2 - change i.MX815 to i.MX8MN - Add changes in Kconfig and Makefile .../devicetree/bindings/sound/fsl,easrc.yaml | 96 + sound/soc/fsl/Kconfig | 11 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_asrc.c | 104 +- sound/soc/fsl/fsl_asrc.h | 74 +- sound/soc/fsl/fsl_asrc_common.h | 105 + sound/soc/fsl/fsl_asrc_dma.c | 27 +- sound/soc/fsl/fsl_easrc.c | 2119 +++++++++++++++++ sound/soc/fsl/fsl_easrc.h | 651 +++++ 9 files changed, 3067 insertions(+), 122 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,easrc.yaml create mode 100644 sound/soc/fsl/fsl_asrc_common.h create mode 100644 sound/soc/fsl/fsl_easrc.c create mode 100644 sound/soc/fsl/fsl_easrc.h -- 2.21.0

3 12

[alsa-devel] [PATCH AUTOSEL 5.5 406/542] ASoC: SOF: Intel: hda: Fix SKL dai count
by Sasha Levin 28 Feb '20

28 Feb '20

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit a6947c9d86bcfd61b758b5693eba58defe7fd2ae ] With fourth pin added for iDisp for skl_dai, update SOF_SKL_DAI_NUM to account for the change. Without this, dais from the bottom of the list are skipped. In current state that's the case for 'Alt Analog CPU DAI'. Fixes: ac42b142cd76 ("ASoC: SOF: Intel: hda: Add iDisp4 DAI") Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200113114054.9716-1-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sof/intel/hda.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 63df888dddb6c..de0115294c74e 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -348,7 +348,7 @@ /* Number of DAIs */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) -#define SOF_SKL_NUM_DAIS 14 +#define SOF_SKL_NUM_DAIS 15 #else #define SOF_SKL_NUM_DAIS 8 #endif -- 2.20.1

2 1

sof-bdw-rt5677: initial port to UCM2
by GitHub pull_request - opened 27 Feb '20

27 Feb '20

alsa-project/alsa-ucm-conf pull request #13 was opened from plbossart: Tested on Pixel 2015/SAMUS Chromebook. Known limitations: Left/Right confusion (probably a firmware issue) PulseAudio does not switch capture devices on headset plug Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/13 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/13.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: linux-next: Tree for Feb 27 (sound/soc/codecs/: SND_SOC_MAX98357A)
by Randy Dunlap 27 Feb '20

27 Feb '20

On 2/26/20 8:22 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20200226: > on x86_64: # CONFIG_GPIOLIB is not set SND_SOC_MAX98357A depends on GPIOLIB, which is not enabled. Some driver(s) is selecting GPIOLIB in error. WARNING: unmet direct dependencies detected for SND_SOC_MAX98357A Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && GPIOLIB [=n] Selected by [m]: - SND_SOC_AMD_RV_RT5682_MACH [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_SOC_AMD_ACP3x [=m] && I2C [=y] - SND_SOC_STORM [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_SOC_QCOM [=m] Full randconfig file is attached. -- ~Randy Reported-by: Randy Dunlap <rdunlap(a)infradead.org>

1 0

[PATCH] ASoC: SOF - topology - do not change the link trigger order for pre-1.4 firmware
by Jaroslav Kysela 27 Feb '20

27 Feb '20

This patch is for SOF v1.3 firmware. The DSP firmware will crash (DSP oops) without this patch. The 1.4.1 firmare has this issue fixed. The ABI version is used for the comparison, because the firmware version for the firmware files before 1.4.2 was not set properly (git hash was used). BugLink: https://github.com/thesofproject/sof/issues/2102 Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/topology.c | 10 +++++++--- 1 file changed, 7 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 9f4f8868b386..58ff4766b47b 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -3111,6 +3111,7 @@ static int sof_link_load(struct snd_soc_component *scomp, int index, struct snd_soc_tplg_private *private = &cfg->priv; struct sof_ipc_dai_config config; struct snd_soc_tplg_hw_config *hw_config; + struct sof_ipc_fw_version *v = &sdev->fw_ready.version; int num_hw_configs; int ret; int i = 0; @@ -3128,9 +3129,12 @@ static int sof_link_load(struct snd_soc_component *scomp, int index, if (!link->no_pcm) { link->nonatomic = true; - /* set trigger order */ - link->trigger[0] = SND_SOC_DPCM_TRIGGER_POST; - link->trigger[1] = SND_SOC_DPCM_TRIGGER_POST; + /* this causes DSP panic on firmware v1.3 */ + if (v->abi_version > SOF_ABI_VER(3, 7, 0)) { + /* set trigger order */ + link->trigger[0] = SND_SOC_DPCM_TRIGGER_POST; + link->trigger[1] = SND_SOC_DPCM_TRIGGER_POST; + } /* nothing more to do for FE dai links */ return 0; -- 2.24.1

5 8

Re: [PATCH] zte:zx-spdif:remove redundant dev_err message
by Mark Brown 27 Feb '20

27 Feb '20

On Thu, Feb 27, 2020 at 11:07:01PM +0800, tangbin wrote: > devm_ioremap_resource has already contains error message, so remove > the redundant dev_err message Please submit patches using subject lines reflecting the style for the subsystem, this makes it easier for people to identify relevant patches. Look at what existing commits in the area you're changing are doing and make sure your subject lines visually resemble what they're doing. There's no need to resubmit to fix this alone.

1 0

[PATCH v2 1/2] mfd: wm8994: Fix driver operation if loaded as modules
by Marek Szyprowski 27 Feb '20

27 Feb '20

WM8994 chip has built-in regulators, which might be used for chip operation. They are controlled by a separate wm8994-regulator driver, which should be loaded before this driver calls regulator_get(), because that driver also provides consumer-supply mapping for the them. If that driver is not yet loaded, regulator core substitute them with dummy regulator, what breaks chip operation, because the built-in regulators are never enabled. Fix this by annotating this driver with MODULE_SOFTDEP() "pre" dependency to "wm8994_regulator" module. Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- v2: - replaced request_module() call with MODULE_SOFTDEP() annotation --- drivers/mfd/wm8994-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index 1e9fe7d92597..737dede4a95c 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -690,3 +690,4 @@ module_i2c_driver(wm8994_i2c_driver); MODULE_DESCRIPTION("Core support for the WM8994 audio CODEC"); MODULE_LICENSE("GPL"); MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_SOFTDEP("pre: wm8994_regulator"); -- 2.17.1

2 3

[PATCH for-next] ASoC: tlv320adcx140: Fix MIC_BIAS defines for ADC full scale
by Dan Murphy 27 Feb '20

27 Feb '20

Change the #defines for the ADC full scale bits from MIC_BIAS to ADC_FSCALE. This also changes the error message to incidate ADC full scale value error as opposed to the Mic bias. Reported-by: Ricard Wanderlof <ricardw(a)axis.com> Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tlv320adcx140.c | 12 ++++++------ sound/soc/codecs/tlv320adcx140.h | 8 ++++---- 2 files changed, 10 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 93a0cb8e662c..825ace9b5fa7 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -758,12 +758,12 @@ static int adcx140_codec_probe(struct snd_soc_component *component) ret = device_property_read_u8(adcx140->dev, "ti,vref-source", &vref_source); if (ret) - vref_source = ADCX140_MIC_BIAS_VREF_275V; + vref_source = ADCX140_ADC_FSCALE_VREF_275V; - if (vref_source != ADCX140_MIC_BIAS_VREF_275V && - vref_source != ADCX140_MIC_BIAS_VREF_25V && - vref_source != ADCX140_MIC_BIAS_VREF_1375V) { - dev_err(adcx140->dev, "Mic Bias source value is invalid\n"); + if (vref_source != ADCX140_ADC_FSCALE_VREF_275V && + vref_source != ADCX140_ADC_FSCALE_VREF_25V && + vref_source != ADCX140_ADC_FSCALE_VREF_1375V) { + dev_err(adcx140->dev, "ADC full scale setting is invalid\n"); return -EINVAL; } @@ -787,7 +787,7 @@ static int adcx140_codec_probe(struct snd_soc_component *component) ret = regmap_update_bits(adcx140->regmap, ADCX140_BIAS_CFG, ADCX140_MIC_BIAS_VAL_MSK | - ADCX140_MIC_BIAS_VREF_MSK, bias_source); + ADCX140_ADC_FSCALE_VREF_MSK, bias_source); if (ret) dev_err(adcx140->dev, "setting MIC bias failed %d\n", ret); out: diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h index 6d055e55909e..adb9513900b1 100644 --- a/sound/soc/codecs/tlv320adcx140.h +++ b/sound/soc/codecs/tlv320adcx140.h @@ -117,10 +117,10 @@ #define ADCX140_MIC_BIAS_VAL_AVDD 6 #define ADCX140_MIC_BIAS_VAL_MSK GENMASK(6, 4) -#define ADCX140_MIC_BIAS_VREF_275V 0 -#define ADCX140_MIC_BIAS_VREF_25V 1 -#define ADCX140_MIC_BIAS_VREF_1375V 2 -#define ADCX140_MIC_BIAS_VREF_MSK GENMASK(1, 0) +#define ADCX140_ADC_FSCALE_VREF_275V 0 +#define ADCX140_ADC_FSCALE_VREF_25V 1 +#define ADCX140_ADC_FSCALE_VREF_1375V 2 +#define ADCX140_ADC_FSCALE_VREF_MSK GENMASK(1, 0) #define ADCX140_PWR_CFG_BIAS_PDZ BIT(7) #define ADCX140_PWR_CFG_ADC_PDZ BIT(6) -- 2.25.0

2 1

snd_rawmidi_params_set_no_active_sensing()
by GitHub issues - opened 27 Feb '20

27 Feb '20

alsa-project/alsa-lib issue #29 was opened from MusicMaker: Does this filter out receiving 0xFE incoming and outgoing messages when enabled ? Still see incoming active sensing messages after using this. Issue URL : https://github.com/alsa-project/alsa-lib/issues/29 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/8] ASoC: soc-pcm cleanup step5
by Kuninori Morimoto 27 Feb '20

27 Feb '20

Hi Mark These are soc-pcm cleanup step5. These are based on Bard Liao's "ASoC: Add Multi CPU DAI support" patch-set. [7/8] patch might exchange the behavior Kuninori Morimoto (8): 1) ASoC: soc-pcm: use defined stream 2) ASoC: soc-pcm: remove duplicate be check from dpcm_add_paths() 3) ASoC: soc-pcm: move dpcm_fe_dai_close() 4) ASoC: soc-pcm: add dpcm_fe_dai_clean() 5) ASoC: soc-pcm: use snd_soc_dai_get_pcm_stream() at dpcm_set_fe_runtime() 6) ASoC: soc-pcm: check DAI's activity more simply 7) ASoC: soc-pcm: Do Digital Mute for both CPU/Codec in same timing. 8) ASoC: soc-pcm: tidyup dulicate handing at dpcm_fe_dai_startup() sound/soc/soc-pcm.c | 107 ++++++++++++++++++++------------------------ 1 file changed, 48 insertions(+), 59 deletions(-) -- 2.17.1

3 20

Applied "ASoC: tas2562: Fix sample rate error message" to the asoc tree
by Mark Brown 26 Feb '20

26 Feb '20

The patch ASoC: tas2562: Fix sample rate error message has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 38b6a714941a285319734b9ccc30d34e107d5d7e Mon Sep 17 00:00:00 2001 From: Dan Murphy <dmurphy(a)ti.com> Date: Wed, 26 Feb 2020 07:03:05 -0600 Subject: [PATCH] ASoC: tas2562: Fix sample rate error message Fix error message for setting the sample rate. It says bitwidth but should say sample rate. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> Link: https://lore.kernel.org/r/20200226130305.12043-3-dmurphy@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tas2562.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index 5b4803a76719..be52886a5edb 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -252,7 +252,7 @@ static int tas2562_hw_params(struct snd_pcm_substream *substream, ret = tas2562_set_samplerate(tas2562, params_rate(params)); if (ret) - dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret); + dev_err(tas2562->dev, "set sample rate failed, %d\n", ret); return ret; } -- 2.20.1

1 0

[PATCH v5 0/6] ASoC: Add Multi CPU DAI support
by Bard Liao 26 Feb '20

26 Feb '20

As discussed in [1], ASoC core supports multi codec DAIs on a DAI link. However it does not do so for CPU DAIs. So, add support for multi CPU DAIs on a DAI Link by adding multi CPU DAI in Card instantiation, suspend and resume functions, PCM ops, stream handling functions and DAPM. [1]: https://www.spinics.net/lists/alsa-devel/msg71369.html changes in v5: - rebase to latest kernel base Bard Liao (2): ASoC: Return error if the function does not support multi-cpu ASoC: pcm: check if cpu-dai supports a given stream Shreyas NC (4): ASoC: Add initial support for multiple CPU DAIs ASoC: Add multiple CPU DAI support for PCM ops ASoC: Add dapm_add_valid_dai_widget helper ASoC: Add multiple CPU DAI support in DAPM include/sound/soc.h | 15 + sound/soc/soc-compress.c | 5 +- sound/soc/soc-core.c | 168 +++++----- sound/soc/soc-dapm.c | 133 ++++---- sound/soc/soc-generic-dmaengine-pcm.c | 18 + sound/soc/soc-pcm.c | 463 ++++++++++++++++++-------- 6 files changed, 531 insertions(+), 271 deletions(-) -- 2.17.1

2 10

Applied "ASoC: Add dapm_add_valid_dai_widget helper" to the asoc tree
by Mark Brown 26 Feb '20

26 Feb '20

The patch ASoC: Add dapm_add_valid_dai_widget helper has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 6c4b13b51aa36aab023dd0bf24bf5582c9ba091e Mon Sep 17 00:00:00 2001 From: Shreyas NC <shreyas.nc(a)intel.com> Date: Tue, 25 Feb 2020 21:39:14 +0800 Subject: [PATCH] ASoC: Add dapm_add_valid_dai_widget helper Adding a helper to connect widget for a specific cpu and codec dai The helper will help dapm_connect_dai_link_widgets() to reduce indents. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Shreyas NC <shreyas.nc(a)intel.com> Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/20200225133917.21314-4-yung-chuan.liao@linux.inte… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 119 +++++++++++++++++++++++-------------------- 1 file changed, 63 insertions(+), 56 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 58c318c9debb..539a1eaebeac 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4277,16 +4277,15 @@ int snd_soc_dapm_link_dai_widgets(struct snd_soc_card *card) return 0; } -static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, - struct snd_soc_pcm_runtime *rtd) +static void dapm_add_valid_dai_widget(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *codec_dai, + struct snd_soc_dai *cpu_dai) { - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai; struct snd_soc_dapm_widget *playback = NULL, *capture = NULL; struct snd_soc_dapm_widget *codec, *playback_cpu, *capture_cpu; struct snd_pcm_substream *substream; struct snd_pcm_str *streams = rtd->pcm->streams; - int i; if (rtd->dai_link->params) { playback_cpu = cpu_dai->capture_widget; @@ -4298,67 +4297,75 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, capture_cpu = capture; } - for_each_rtd_codec_dai(rtd, i, codec_dai) { - /* connect BE DAI playback if widgets are valid */ - codec = codec_dai->playback_widget; - - if (playback_cpu && codec) { - if (!playback) { - substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - playback = snd_soc_dapm_new_dai(card, substream, - "playback"); - if (IS_ERR(playback)) { - dev_err(rtd->dev, - "ASoC: Failed to create DAI %s: %ld\n", - codec_dai->name, - PTR_ERR(playback)); - continue; - } - - snd_soc_dapm_add_path(&card->dapm, playback_cpu, - playback, NULL, NULL); + /* connect BE DAI playback if widgets are valid */ + codec = codec_dai->playback_widget; + + if (playback_cpu && codec) { + if (!playback) { + substream = streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + playback = snd_soc_dapm_new_dai(card, substream, + "playback"); + if (IS_ERR(playback)) { + dev_err(rtd->dev, + "ASoC: Failed to create DAI %s: %ld\n", + codec_dai->name, + PTR_ERR(playback)); + goto capture; } - dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - cpu_dai->component->name, playback_cpu->name, - codec_dai->component->name, codec->name); - - snd_soc_dapm_add_path(&card->dapm, playback, codec, - NULL, NULL); + snd_soc_dapm_add_path(&card->dapm, playback_cpu, + playback, NULL, NULL); } - } - for_each_rtd_codec_dai(rtd, i, codec_dai) { - /* connect BE DAI capture if widgets are valid */ - codec = codec_dai->capture_widget; - - if (codec && capture_cpu) { - if (!capture) { - substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream; - capture = snd_soc_dapm_new_dai(card, substream, - "capture"); - if (IS_ERR(capture)) { - dev_err(rtd->dev, - "ASoC: Failed to create DAI %s: %ld\n", - codec_dai->name, - PTR_ERR(capture)); - continue; - } - - snd_soc_dapm_add_path(&card->dapm, capture, - capture_cpu, NULL, NULL); + dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", + cpu_dai->component->name, playback_cpu->name, + codec_dai->component->name, codec->name); + + snd_soc_dapm_add_path(&card->dapm, playback, codec, + NULL, NULL); + } + +capture: + /* connect BE DAI capture if widgets are valid */ + codec = codec_dai->capture_widget; + + if (codec && capture_cpu) { + if (!capture) { + substream = streams[SNDRV_PCM_STREAM_CAPTURE].substream; + capture = snd_soc_dapm_new_dai(card, substream, + "capture"); + if (IS_ERR(capture)) { + dev_err(rtd->dev, + "ASoC: Failed to create DAI %s: %ld\n", + codec_dai->name, + PTR_ERR(capture)); + return; } - dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", - codec_dai->component->name, codec->name, - cpu_dai->component->name, capture_cpu->name); - - snd_soc_dapm_add_path(&card->dapm, codec, capture, - NULL, NULL); + snd_soc_dapm_add_path(&card->dapm, capture, + capture_cpu, NULL, NULL); } + + dev_dbg(rtd->dev, "connected DAI link %s:%s -> %s:%s\n", + codec_dai->component->name, codec->name, + cpu_dai->component->name, capture_cpu->name); + + snd_soc_dapm_add_path(&card->dapm, codec, capture, + NULL, NULL); } } +static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, + struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_dai *codec_dai; + int i; + + for_each_rtd_codec_dai(rtd, i, codec_dai) + dapm_add_valid_dai_widget(card, rtd, + codec_dai, rtd->cpu_dais[0]); +} + static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, int event) { -- 2.20.1

1 0

Applied "ASoC: Add multiple CPU DAI support in DAPM" to the asoc tree
by Mark Brown 26 Feb '20

26 Feb '20

The patch ASoC: Add multiple CPU DAI support in DAPM has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From de6214a33633d8ce1c1490336f8e798e75ccd004 Mon Sep 17 00:00:00 2001 From: Shreyas NC <shreyas.nc(a)intel.com> Date: Tue, 25 Feb 2020 21:39:15 +0800 Subject: [PATCH] ASoC: Add multiple CPU DAI support in DAPM DAPM handles DAIs during soc_dapm_stream_event() and during addition and creation of DAI widgets i.e., dapm_add_valid_dai_widget() and dapm_connect_dai_link_widgets(). Extend these functions to handle multiple cpu dai. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Shreyas NC <shreyas.nc(a)intel.com> Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Link: https://lore.kernel.org/r/20200225133917.21314-5-yung-chuan.liao@linux.inte… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-dapm.c | 20 ++++++++++++++++---- 1 file changed, 16 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 539a1eaebeac..6ce024d52170 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -4361,9 +4361,19 @@ static void dapm_connect_dai_link_widgets(struct snd_soc_card *card, struct snd_soc_dai *codec_dai; int i; - for_each_rtd_codec_dai(rtd, i, codec_dai) - dapm_add_valid_dai_widget(card, rtd, - codec_dai, rtd->cpu_dais[0]); + if (rtd->num_cpus == 1) { + for_each_rtd_codec_dai(rtd, i, codec_dai) + dapm_add_valid_dai_widget(card, rtd, codec_dai, + rtd->cpu_dais[0]); + } else if (rtd->num_codecs == rtd->num_cpus) { + for_each_rtd_codec_dai(rtd, i, codec_dai) + dapm_add_valid_dai_widget(card, rtd, codec_dai, + rtd->cpu_dais[i]); + } else { + dev_err(card->dev, + "N cpus to M codecs link is not supported yet\n"); + } + } static void soc_dapm_dai_stream_event(struct snd_soc_dai *dai, int stream, @@ -4424,9 +4434,11 @@ static void soc_dapm_stream_event(struct snd_soc_pcm_runtime *rtd, int stream, int event) { struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai; int i; - soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event); + for_each_rtd_cpu_dai(rtd, i, cpu_dai) + soc_dapm_dai_stream_event(rtd->cpu_dai, stream, event); for_each_rtd_codec_dai(rtd, i, codec_dai) soc_dapm_dai_stream_event(codec_dai, stream, event); -- 2.20.1

1 0

[PATCH 0/3] SoundWire: ASoC interfaces for multi-cpu dais and DisCo helpers
by Pierre-Louis Bossart 26 Feb '20

26 Feb '20

The first two patches prepare the support of multi-cpu dais for synchronized playback and capture. We remove an unused set of prototypes and add a get_sdw_stream() callback prototype currently missing (the implementation will come later as part of the synchronized playback) The last exposes macros used internally, so that they can be reused to extract information from the _ADR 64-bit values in SOF platform drivers and related machine drivers. I think it's simpler if all these simple patches are merged through the SoundWire tree. With the additional changes to remove the platform drivers and the merge of interrupt handling, that will result in a single immutable tag provided to Mark Brown. Pierre-Louis Bossart (3): soundwire: cadence: remove useless prototypes ASoC: soc-dai: add get_sdw_stream() callback soundwire: add helper macros for devID fields drivers/soundwire/bus.c | 21 +++++---------------- drivers/soundwire/cadence_master.h | 8 -------- include/linux/soundwire/sdw.h | 23 +++++++++++++++++++++++ include/sound/soc-dai.h | 21 +++++++++++++++++++++ 4 files changed, 49 insertions(+), 24 deletions(-) -- 2.20.1

3 7

[alsa-devel] [PATCH] ASoC: amd: Add machine driver for Raven based platform
by Akshu Agrawal 26 Feb '20

26 Feb '20

Add machine driver for Raven based platform using RT5682 + MAX9836 + CROS_EC codecs Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- This patch is dependent on https://patchwork.kernel.org/patch/11381839/ sound/soc/amd/Kconfig | 10 + sound/soc/amd/Makefile | 2 + sound/soc/amd/acp3x-rt5682-max9836.c | 334 +++++++++++++++++++++++++++ 3 files changed, 346 insertions(+) create mode 100644 sound/soc/amd/acp3x-rt5682-max9836.c diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 5f40517717c4..b29ef1373946 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -26,3 +26,13 @@ config SND_SOC_AMD_ACP3x depends on X86 && PCI help This option enables ACP v3.x I2S support on AMD platform + +config SND_SOC_AMD_RV_RT5682_MACH + tristate "AMD RV support for RT5682" + select SND_SOC_RT5682 + select SND_SOC_MAX98357A + select SND_SOC_CROS_EC_CODEC + select I2C_CROS_EC_TUNNEL + depends on SND_SOC_AMD_ACP3x && I2C + help + This option enables machine driver for RT5682 and MAX9835. diff --git a/sound/soc/amd/Makefile b/sound/soc/amd/Makefile index c4ddc6adb6f0..e6f3d9b469f3 100644 --- a/sound/soc/amd/Makefile +++ b/sound/soc/amd/Makefile @@ -2,8 +2,10 @@ acp_audio_dma-objs := acp-pcm-dma.o snd-soc-acp-da7219mx98357-mach-objs := acp-da7219-max98357a.o snd-soc-acp-rt5645-mach-objs := acp-rt5645.o +snd-soc-acp-rt5682-mach-objs := acp3x-rt5682-max9836.o obj-$(CONFIG_SND_SOC_AMD_ACP) += acp_audio_dma.o obj-$(CONFIG_SND_SOC_AMD_CZ_DA7219MX98357_MACH) += snd-soc-acp-da7219mx98357-mach.o obj-$(CONFIG_SND_SOC_AMD_CZ_RT5645_MACH) += snd-soc-acp-rt5645-mach.o obj-$(CONFIG_SND_SOC_AMD_ACP3x) += raven/ +obj-$(CONFIG_SND_SOC_AMD_RV_RT5682_MACH) += snd-soc-acp-rt5682-mach.o diff --git a/sound/soc/amd/acp3x-rt5682-max9836.c b/sound/soc/amd/acp3x-rt5682-max9836.c new file mode 100644 index 000000000000..96fbcd29e3ed --- /dev/null +++ b/sound/soc/amd/acp3x-rt5682-max9836.c @@ -0,0 +1,334 @@ +// SPDX-License-Identifier: GPL-2.0+ +// +// Machine driver for AMD ACP Audio engine using DA7219 & MAX98357 codec. +// +//Copyright 2016 Advanced Micro Devices, Inc. + +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <linux/clk.h> +#include <linux/gpio.h> +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/input.h> +#include <linux/acpi.h> + +#include "raven/acp3x.h" +#include "../codecs/rt5682.h" + +#define PCO_PLAT_CLK 48000000 +#define RT5682_PLL_FREQ (48000 * 512) +#define DUAL_CHANNEL 2 + +static struct snd_soc_jack pco_jack; +static struct clk *rt5682_dai_wclk; +static struct clk *rt5682_dai_bclk; + +static int acp3x_5682_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_component *component = codec_dai->component; + + dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); + + /* set rt5682 dai fmt */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S + | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) { + dev_err(rtd->card->dev, + "Failed to set rt5682 dai fmt: %d\n", ret); + return ret; + } + + /* set codec PLL */ + ret = snd_soc_dai_set_pll(codec_dai, RT5682_PLL2, RT5682_PLL2_S_MCLK, + PCO_PLAT_CLK, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(rtd->dev, "can't set rt5682 PLL: %d\n", ret); + return ret; + } + + /* Set codec sysclk */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL2, + RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, + "Failed to set rt5682 SYSCLK: %d\n", ret); + return ret; + } + + /* Set tdm/i2s1 master bclk ratio */ + ret = snd_soc_dai_set_bclk_ratio(codec_dai, 64); + if (ret < 0) { + dev_err(rtd->dev, + "Failed to set rt5682 tdm bclk ratio: %d\n", ret); + return ret; + } + + rt5682_dai_wclk = clk_get(component->dev, "rt5682-dai-wclk"); + rt5682_dai_bclk = clk_get(component->dev, "rt5682-dai-bclk"); + + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_LINEOUT | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &pco_jack, NULL, 0); + if (ret) { + dev_err(card->dev, "HP jack creation failed %d\n", ret); + return ret; + } + + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(pco_jack.jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + + ret = snd_soc_component_set_jack(component, &pco_jack, NULL); + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return ret; +} + +static int rt5682_clk_enable(struct snd_pcm_substream *substream) +{ + int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* RT5682 will support only 48K output with 48M mclk */ + clk_set_rate(rt5682_dai_wclk, 48000); + clk_set_rate(rt5682_dai_bclk, 48000 * 64); + ret = clk_prepare_enable(rt5682_dai_wclk); + if (ret < 0) { + dev_err(rtd->dev, "can't enable wclk %d\n", ret); + return ret; + } + + return ret; +} + +static void rt5682_clk_disable(void) +{ + clk_disable_unprepare(rt5682_dai_wclk); +} + +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const unsigned int rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static int acp3x_5682_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->play_i2s_instance = I2S_SP_INSTANCE; + machine->cap_i2s_instance = I2S_SP_INSTANCE; + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + return rt5682_clk_enable(substream); +} + +static int acp3x_max_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->play_i2s_instance = I2S_BT_INSTANCE; + + runtime->hw.channels_max = DUAL_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels); + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + return rt5682_clk_enable(substream); +} + +static int acp3x_ec_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct acp3x_platform_info *machine = snd_soc_card_get_drvdata(card); + + machine->cap_i2s_instance = I2S_BT_INSTANCE; + snd_soc_dai_set_bclk_ratio(codec_dai, 64); + + return rt5682_clk_enable(substream); +} + +static void rt5682_shutdown(struct snd_pcm_substream *substream) +{ + rt5682_clk_disable(); +} + +static const struct snd_soc_ops acp3x_5682_ops = { + .startup = acp3x_5682_startup, + .shutdown = rt5682_shutdown, +}; + +static const struct snd_soc_ops acp3x_max_play_ops = { + .startup = acp3x_max_startup, + .shutdown = rt5682_shutdown, +}; + +static const struct snd_soc_ops acp3x_ec_cap_ops = { + .startup = acp3x_ec_startup, + .shutdown = rt5682_shutdown, +}; + +SND_SOC_DAILINK_DEF(acp3x_i2s, + DAILINK_COMP_ARRAY(COMP_CPU("acp3x_i2s_playcap.0"))); +SND_SOC_DAILINK_DEF(acp3x_bt, + DAILINK_COMP_ARRAY(COMP_CPU("acp3x_i2s_playcap.2"))); + +SND_SOC_DAILINK_DEF(rt5682, + DAILINK_COMP_ARRAY(COMP_CODEC("i2c-10EC5682:00", "rt5682-aif1"))); +SND_SOC_DAILINK_DEF(max, + DAILINK_COMP_ARRAY(COMP_CODEC("MX98357A:00", "HiFi"))); +SND_SOC_DAILINK_DEF(cros_ec, + DAILINK_COMP_ARRAY(COMP_CODEC("GOOG0013:00", "EC Codec I2S RX"))); + +SND_SOC_DAILINK_DEF(platform, + DAILINK_COMP_ARRAY(COMP_PLATFORM("acp3x_rv_i2s_dma.0"))); + +static struct snd_soc_dai_link acp3x_dai_5682_98357[] = { + { + .name = "acp3x-5682-play", + .stream_name = "Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .init = acp3x_5682_init, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &acp3x_5682_ops, + SND_SOC_DAILINK_REG(acp3x_i2s, rt5682, platform), + }, + { + .name = "acp3x-max98357-play", + .stream_name = "HiFi Playback", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .dpcm_playback = 1, + .ops = &acp3x_max_play_ops, + SND_SOC_DAILINK_REG(acp3x_bt, max, platform), + }, + { + .name = "acp3x-ec-capture", + .stream_name = "Capture", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .dpcm_capture = 1, + .ops = &acp3x_ec_cap_ops, + SND_SOC_DAILINK_REG(acp3x_bt, cros_ec, platform), + }, +}; + +static const struct snd_soc_dapm_widget acp3x_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Spk", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), +}; + +static const struct snd_soc_dapm_route acp3x_audio_route[] = { + {"Headphone Jack", NULL, "HPOL"}, + {"Headphone Jack", NULL, "HPOR"}, + {"IN1P", NULL, "Headset Mic"}, + {"Spk", NULL, "Speaker"}, +}; + +static const struct snd_kcontrol_new acp3x_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Spk"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), +}; + +static struct snd_soc_card acp3x_card = { + .name = "acp3xalc5682m98357", + .owner = THIS_MODULE, + .dai_link = acp3x_dai_5682_98357, + .num_links = ARRAY_SIZE(acp3x_dai_5682_98357), + .dapm_widgets = acp3x_widgets, + .num_dapm_widgets = ARRAY_SIZE(acp3x_widgets), + .dapm_routes = acp3x_audio_route, + .num_dapm_routes = ARRAY_SIZE(acp3x_audio_route), + .controls = acp3x_mc_controls, + .num_controls = ARRAY_SIZE(acp3x_mc_controls), +}; + +static int acp3x_probe(struct platform_device *pdev) +{ + int ret; + struct snd_soc_card *card; + struct acp3x_platform_info *machine; + + machine = devm_kzalloc(&pdev->dev, sizeof(*machine), GFP_KERNEL); + if (!machine) + return -ENOMEM; + + card = &acp3x_card; + acp3x_card.dev = &pdev->dev; + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, machine); + ret = devm_snd_soc_register_card(&pdev->dev, &acp3x_card); + if (ret) { + dev_err(&pdev->dev, + "devm_snd_soc_register_card(%s) failed: %d\n", + acp3x_card.name, ret); + return ret; + } + return 0; +} + +static const struct acpi_device_id acp3x_audio_acpi_match[] = { + { "AMDI5682", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, acp3x_audio_acpi_match); + +static struct platform_driver acp3x_audio = { + .driver = { + .name = "acp3x-alc5682-max98357", + .acpi_match_table = ACPI_PTR(acp3x_audio_acpi_match), + .pm = &snd_soc_pm_ops, + }, + .probe = acp3x_probe, +}; + +module_platform_driver(acp3x_audio); + +MODULE_AUTHOR("akshu.agrawal(a)amd.com"); +MODULE_DESCRIPTION("ALC5682 & MAX98357 audio support"); +MODULE_LICENSE("GPL v2"); -- 2.17.1

3 3

[PATCH 1/2] ASoC: rt5682: Add the soundwire support
by Oder Chiou 26 Feb '20

26 Feb '20

This patch adds the soundwire support for ALC5682. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5682-sdw.c | 333 +++++++++++++++++++++ sound/soc/codecs/rt5682-sdw.h | 20 ++ sound/soc/codecs/rt5682.c | 526 +++++++++++++++++++++++++++++++--- sound/soc/codecs/rt5682.h | 49 ++++ 6 files changed, 897 insertions(+), 40 deletions(-) create mode 100644 sound/soc/codecs/rt5682-sdw.c create mode 100644 sound/soc/codecs/rt5682-sdw.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d957fd6980b1..2e9afe12fc8f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -168,6 +168,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_RT5670 imply SND_SOC_RT5677 imply SND_SOC_RT5682 + imply SND_SOC_RT5682_SDW imply SND_SOC_RT700_SDW imply SND_SOC_RT711_SDW imply SND_SOC_RT715_SDW @@ -1135,6 +1136,12 @@ config SND_SOC_RT5682 tristate depends on I2C +config SND_SOC_RT5682_SDW + tristate "Realtek RT5682 Codec - SDW" + depends on SOUNDWIRE + select SND_SOC_RT5682 + select REGMAP_SOUNDWIRE + config SND_SOC_RT700 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ba1b4b3fa2da..c0561f584f5b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -177,6 +177,7 @@ snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o snd-soc-rt5682-objs := rt5682.o +snd-soc-rt5682-sdw-objs := rt5682-sdw.o snd-soc-rt700-objs := rt700.o rt700-sdw.o snd-soc-rt711-objs := rt711.o rt711-sdw.o snd-soc-rt715-objs := rt715.o rt715-sdw.o @@ -476,6 +477,7 @@ obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o obj-$(CONFIG_SND_SOC_RT5682) += snd-soc-rt5682.o +obj-$(CONFIG_SND_SOC_RT5682_SDW) += snd-soc-rt5682-sdw.o obj-$(CONFIG_SND_SOC_RT700) += snd-soc-rt700.o obj-$(CONFIG_SND_SOC_RT711) += snd-soc-rt711.o obj-$(CONFIG_SND_SOC_RT715) += snd-soc-rt715.o diff --git a/sound/soc/codecs/rt5682-sdw.c b/sound/soc/codecs/rt5682-sdw.c new file mode 100644 index 000000000000..fc31d04b5203 --- /dev/null +++ b/sound/soc/codecs/rt5682-sdw.c @@ -0,0 +1,333 @@ +// SPDX-License-Identifier: GPL-2.0-only +// +// rt5682-sdw.c -- RT5682 ALSA SoC audio component driver +// +// Copyright 2019 Realtek Semiconductor Corp. +// Author: Oder Chiou <oder_chiou(a)realtek.com> +// + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/acpi.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/regulator/consumer.h> +#include <linux/mutex.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.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 "rt5682.h" +#include "rt5682-sdw.h" + +static bool rt5682_sdw_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case 0x00e0: + case 0x00f0: + case 0x3000: + case 0x3001: + case 0x3004: + case 0x3005: + case 0x3008: + return true; + default: + return false; + } +} + +const struct regmap_config rt5682_sdw_regmap = { + .name = "sdw", + .reg_bits = 32, + .val_bits = 8, + .max_register = RT5682_I2C_MODE, + .readable_reg = rt5682_sdw_readable_register, + .cache_type = REGCACHE_NONE, + .use_single_read = true, + .use_single_write = true, +}; + +static int rt5682_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev); + + /* Update the status */ + rt5682->status = status; + + if (status == SDW_SLAVE_UNATTACHED) + rt5682->hw_init = false; + + /* + * Perform initialization only if slave status is present and + * hw_init flag is false + */ + if (rt5682->hw_init || rt5682->status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return rt5682_io_init(&slave->dev, slave); +} + +static int rt5682_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + int nval, i, num_of_ports = 1; + u32 bit; + unsigned long addr; + struct sdw_dpn_prop *dpn; + + prop->paging_support = false; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = 0x4; /* BITMAP: 00000100 */ + prop->sink_ports = 0x2; /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + num_of_ports += nval; + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), + GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + num_of_ports += nval; + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), + GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* Allocate port_ready based on num_of_ports */ + slave->port_ready = devm_kcalloc(&slave->dev, num_of_ports, + sizeof(*slave->port_ready), + GFP_KERNEL); + if (!slave->port_ready) + return -ENOMEM; + + /* Initialize completion */ + for (i = 0; i < num_of_ports; i++) + init_completion(&slave->port_ready[i]); + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + /* wake-up event */ + prop->wake_capable = 1; + + return 0; +} + +/* Bus clock frequency */ +#define RT5682_CLK_FREQ_9600000HZ 9600000 +#define RT5682_CLK_FREQ_12000000HZ 12000000 +#define RT5682_CLK_FREQ_6000000HZ 6000000 +#define RT5682_CLK_FREQ_4800000HZ 4800000 +#define RT5682_CLK_FREQ_2400000HZ 2400000 +#define RT5682_CLK_FREQ_12288000HZ 12288000 + +int rt5682_clock_config(struct device *dev) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(dev); + unsigned int clk_freq, value; + + clk_freq = (rt5682->params.curr_dr_freq >> 1); + + switch (clk_freq) { + case RT5682_CLK_FREQ_12000000HZ: + value = 0x0; + break; + case RT5682_CLK_FREQ_6000000HZ: + value = 0x1; + break; + case RT5682_CLK_FREQ_9600000HZ: + value = 0x2; + break; + case RT5682_CLK_FREQ_4800000HZ: + value = 0x3; + break; + case RT5682_CLK_FREQ_2400000HZ: + value = 0x4; + break; + case RT5682_CLK_FREQ_12288000HZ: + value = 0x5; + break; + default: + return -EINVAL; + } + + regmap_write(rt5682->sdw_regmap, 0xe0, value); + regmap_write(rt5682->sdw_regmap, 0xf0, value); + + dev_dbg(dev, "%s complete, clk_freq=%d\n", __func__, clk_freq); + + return 0; +} + +static int rt5682_bus_config(struct sdw_slave *slave, + struct sdw_bus_params *params) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev); + int ret; + + memcpy(&rt5682->params, params, sizeof(*params)); + + ret = rt5682_clock_config(&slave->dev); + if (ret < 0) + dev_err(&slave->dev, "Invalid clk config"); + + return ret; +} + +static int rt5682_interrupt_callback(struct sdw_slave *slave, + struct sdw_slave_intr_status *status) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev); + + dev_dbg(&slave->dev, + "%s control_port_stat=%x", __func__, status->control_port); + + if (status->control_port & 0x4) { + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(250)); + } + + return 0; +} + +static struct sdw_slave_ops rt5682_slave_ops = { + .read_prop = rt5682_read_prop, + .interrupt_callback = rt5682_interrupt_callback, + .update_status = rt5682_update_status, + .bus_config = rt5682_bus_config, +}; + +static int rt5682_sdw_probe(struct sdw_slave *slave, + const struct sdw_device_id *id) +{ + struct regmap *regmap; + + /* Assign ops */ + slave->ops = &rt5682_slave_ops; + + /* Regmap Initialization */ + regmap = devm_regmap_init_sdw(slave, &rt5682_sdw_regmap); + if (IS_ERR(regmap)) + return -EINVAL; + + rt5682_sdw_init(&slave->dev, regmap, slave); + + return 0; +} + +static int rt5682_sdw_remove(struct sdw_slave *slave) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(&slave->dev); + + if (rt5682 && rt5682->hw_init) + cancel_delayed_work(&rt5682->jack_detect_work); + + return 0; +} + +static const struct sdw_device_id rt5682_id[] = { + SDW_SLAVE_ENTRY(0x025d, 0x5682, 0), + {}, +}; +MODULE_DEVICE_TABLE(sdw, rt5682_id); + +static int rt5682_dev_suspend(struct device *dev) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(dev); + + if (!rt5682->hw_init) + return 0; + + regcache_cache_only(rt5682->regmap, true); + regcache_mark_dirty(rt5682->regmap); + + return 0; +} + +static int rt5682_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct rt5682_priv *rt5682 = dev_get_drvdata(dev); + unsigned long time; + + if (!rt5682->hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(RT5682_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete, timed out\n"); + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(rt5682->regmap, false); + regcache_sync(rt5682->regmap); + + return 0; +} + +static const struct dev_pm_ops rt5682_pm = { + SET_SYSTEM_SLEEP_PM_OPS(rt5682_dev_suspend, rt5682_dev_resume) + SET_RUNTIME_PM_OPS(rt5682_dev_suspend, rt5682_dev_resume, NULL) +}; + +static struct sdw_driver rt5682_sdw_driver = { + .driver = { + .name = "rt5682", + .owner = THIS_MODULE, + .pm = &rt5682_pm, + }, + .probe = rt5682_sdw_probe, + .remove = rt5682_sdw_remove, + .ops = &rt5682_slave_ops, + .id_table = rt5682_id, +}; +module_sdw_driver(rt5682_sdw_driver); + +MODULE_DESCRIPTION("ASoC RT5682 driver SDW"); +MODULE_AUTHOR("Oder Chiou <oder_chiou(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5682-sdw.h b/sound/soc/codecs/rt5682-sdw.h new file mode 100644 index 000000000000..76e6f607066e --- /dev/null +++ b/sound/soc/codecs/rt5682-sdw.h @@ -0,0 +1,20 @@ +/* SPDX-License-Identifier: GPL-2.0-only + * + * rt5682-sdw.h -- RT5682 SDW ALSA SoC audio driver + * + * Copyright 2019 Realtek Semiconductor Corp. + * Author: Oder Chiou <oder_chiou(a)realtek.com> + */ + +#ifndef __RT5682_SDW_H__ +#define __RT5682_SDW_H__ + +#define RT5682_SDW_ADDR_L 0x3000 +#define RT5682_SDW_ADDR_H 0x3001 +#define RT5682_SDW_DATA_L 0x3004 +#define RT5682_SDW_DATA_H 0x3005 +#define RT5682_SDW_CMD 0x3008 + +#define RT5682_PROBE_TIMEOUT 2000 + +#endif /* __RT5682_SDW_H__ */ diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index 6774813e0eea..1795a8bbea1a 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -11,13 +11,13 @@ #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> +#include <linux/pm_runtime.h> #include <linux/i2c.h> #include <linux/platform_device.h> #include <linux/spi/spi.h> #include <linux/acpi.h> #include <linux/gpio.h> #include <linux/of_gpio.h> -#include <linux/regulator/consumer.h> #include <linux/mutex.h> #include <sound/core.h> #include <sound/pcm.h> @@ -27,15 +27,11 @@ #include <sound/soc-dapm.h> #include <sound/initval.h> #include <sound/tlv.h> -#include <linux/clk.h> -#include <linux/clkdev.h> -#include <linux/clk-provider.h> #include <sound/rt5682.h> #include "rl6231.h" #include "rt5682.h" - -#define RT5682_NUM_SUPPLIES 3 +#include "rt5682-sdw.h" static const char *rt5682_supply_names[RT5682_NUM_SUPPLIES] = { "AVDD", @@ -52,37 +48,6 @@ static const struct rt5682_platform_data i2s_default_platform_data = { .dai_clk_names[RT5682_DAI_BCLK_IDX] = "rt5682-dai-bclk", }; -struct rt5682_priv { - struct snd_soc_component *component; - struct rt5682_platform_data pdata; - struct regmap *regmap; - struct snd_soc_jack *hs_jack; - struct regulator_bulk_data supplies[RT5682_NUM_SUPPLIES]; - struct delayed_work jack_detect_work; - struct delayed_work jd_check_work; - struct mutex calibrate_mutex; - bool is_sdw; - -#ifdef CONFIG_COMMON_CLK - struct clk_hw dai_clks_hw[RT5682_DAI_NUM_CLKS]; - struct clk_lookup *dai_clks_lookup[RT5682_DAI_NUM_CLKS]; - struct clk *dai_clks[RT5682_DAI_NUM_CLKS]; - struct clk *mclk; -#endif - - int sysclk; - int sysclk_src; - int lrck[RT5682_AIFS]; - int bclk[RT5682_AIFS]; - int master[RT5682_AIFS]; - - int pll_src[RT5682_PLLS]; - int pll_in[RT5682_PLLS]; - int pll_out[RT5682_PLLS]; - - int jack_type; -}; - static const struct reg_sequence patch_list[] = { {RT5682_HP_IMP_SENS_CTRL_19, 0x1000}, {RT5682_DAC_ADC_DIG_VOL1, 0xa020}, @@ -819,6 +784,22 @@ static const struct snd_kcontrol_new rt5682_if1_45_adc_swap_mux = static const struct snd_kcontrol_new rt5682_if1_67_adc_swap_mux = SOC_DAPM_ENUM("IF1 67 ADC Swap Mux", rt5682_if1_67_adc_enum); +static const char * const rt5682_dac_select[] = { + "IF1", "SOUND" +}; + +static SOC_ENUM_SINGLE_DECL(rt5682_dacl_enum, + RT5682_AD_DA_MIXER, RT5682_DAC1_L_SEL_SFT, rt5682_dac_select); + +static const struct snd_kcontrol_new rt5682_dac_l_mux = + SOC_DAPM_ENUM("DAC L Mux", rt5682_dacl_enum); + +static SOC_ENUM_SINGLE_DECL(rt5682_dacr_enum, + RT5682_AD_DA_MIXER, RT5682_DAC1_R_SEL_SFT, rt5682_dac_select); + +static const struct snd_kcontrol_new rt5682_dac_r_mux = + SOC_DAPM_ENUM("DAC R Mux", rt5682_dacr_enum); + static void rt5682_reset(struct rt5682_priv *rt5682) { regmap_write(rt5682->regmap, RT5682_RESET, 0); @@ -1271,6 +1252,9 @@ static int set_filter_clk(struct snd_soc_dapm_widget *w, static const int div_f[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48}; static const int div_o[] = {1, 2, 4, 6, 8, 12, 16, 24, 32, 48}; + if (rt5682->is_sdw) + return 0; + val = snd_soc_component_read32(component, RT5682_GPIO_CTRL_1) & RT5682_GP4_PIN_MASK; if (w->shift == RT5682_PWR_ADC_S1F_BIT && @@ -1743,6 +1727,8 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_PGA("IF1 DAC1", 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("SOUND DAC L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("SOUND DAC R", SND_SOC_NOPM, 0, 0, NULL, 0), /* Digital Interface Select */ SND_SOC_DAPM_MUX("IF1 01 ADC Swap Mux", SND_SOC_NOPM, 0, 0, @@ -1759,12 +1745,19 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_MUX("ADCDAT Mux", SND_SOC_NOPM, 0, 0, &rt5682_adcdat_pin_ctrl), + SND_SOC_DAPM_MUX("DAC L Mux", SND_SOC_NOPM, 0, 0, + &rt5682_dac_l_mux), + SND_SOC_DAPM_MUX("DAC R Mux", SND_SOC_NOPM, 0, 0, + &rt5682_dac_r_mux), + /* Audio Interface */ SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, RT5682_I2S1_SDP, RT5682_SEL_ADCDAT_SFT, 1), SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, RT5682_I2S2_SDP, RT5682_I2S2_PIN_CFG_SFT, 1), SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("SDWRX", "SDW Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("SDWTX", "SDW Capture", 0, SND_SOC_NOPM, 0, 0), /* Output Side */ /* DAC mixer before sound effect */ @@ -1921,8 +1914,8 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"IF1_ADC Mux", "Slot 2", "IF1 23 ADC Swap Mux"}, {"IF1_ADC Mux", "Slot 4", "IF1 45 ADC Swap Mux"}, {"IF1_ADC Mux", "Slot 6", "IF1 67 ADC Swap Mux"}, - {"IF1_ADC Mux", NULL, "I2S1"}, {"ADCDAT Mux", "ADCDAT1", "IF1_ADC Mux"}, + {"AIF1TX", NULL, "I2S1"}, {"AIF1TX", NULL, "ADCDAT Mux"}, {"IF2 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, {"IF2 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, @@ -1931,6 +1924,10 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"ADCDAT Mux", "ADCDAT2", "IF2 ADC Swap Mux"}, {"AIF2TX", NULL, "ADCDAT Mux"}, + {"SDWTX", NULL, "PLL2B"}, + {"SDWTX", NULL, "PLL2F"}, + {"SDWTX", NULL, "ADCDAT Mux"}, + {"IF1 DAC1 L", NULL, "AIF1RX"}, {"IF1 DAC1 L", NULL, "I2S1"}, {"IF1 DAC1 L", NULL, "DAC Stereo1 Filter"}, @@ -1938,10 +1935,24 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"IF1 DAC1 R", NULL, "I2S1"}, {"IF1 DAC1 R", NULL, "DAC Stereo1 Filter"}, + {"SOUND DAC L", NULL, "SDWRX"}, + {"SOUND DAC L", NULL, "DAC Stereo1 Filter"}, + {"SOUND DAC L", NULL, "PLL2B"}, + {"SOUND DAC L", NULL, "PLL2F"}, + {"SOUND DAC R", NULL, "SDWRX"}, + {"SOUND DAC R", NULL, "DAC Stereo1 Filter"}, + {"SOUND DAC R", NULL, "PLL2B"}, + {"SOUND DAC R", NULL, "PLL2F"}, + + {"DAC L Mux", "IF1", "IF1 DAC1 L"}, + {"DAC L Mux", "SOUND", "SOUND DAC L"}, + {"DAC R Mux", "IF1", "IF1 DAC1 R"}, + {"DAC R Mux", "SOUND", "SOUND DAC R"}, + {"DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL"}, - {"DAC1 MIXL", "DAC1 Switch", "IF1 DAC1 L"}, + {"DAC1 MIXL", "DAC1 Switch", "DAC L Mux"}, {"DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR"}, - {"DAC1 MIXR", "DAC1 Switch", "IF1 DAC1 R"}, + {"DAC1 MIXR", "DAC1 Switch", "DAC R Mux"}, {"Stereo1 DAC MIXL", "DAC L1 Switch", "DAC1 MIXL"}, {"Stereo1 DAC MIXL", "DAC R1 Switch", "DAC1 MIXR"}, @@ -2826,6 +2837,8 @@ static int rt5682_register_dai_clks(struct snd_soc_component *component) static int rt5682_probe(struct snd_soc_component *component) { struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct sdw_slave *slave; + unsigned long time; #ifdef CONFIG_COMMON_CLK int ret; @@ -2852,6 +2865,17 @@ static int rt5682_probe(struct snd_soc_component *component) rt5682->lrck[RT5682_AIF1] = CLK_48; #endif + if (rt5682->is_sdw) { + slave = rt5682->slave; + time = wait_for_completion_timeout( + &slave->initialization_complete, + msecs_to_jiffies(RT5682_PROBE_TIMEOUT)); + if (!time) { + dev_err(&slave->dev, "Initialization not complete, timed out\n"); + return -ETIMEDOUT; + } + } + return 0; } @@ -2914,6 +2938,194 @@ static const struct snd_soc_dai_ops rt5682_aif2_dai_ops = { .set_bclk_ratio = rt5682_set_bclk2_ratio, }; +#if IS_ENABLED(CONFIG_SND_SOC_RT5682_SDW) +struct sdw_stream_data { + struct sdw_stream_runtime *sdw_stream; +}; + +static int rt5682_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_stream, + int direction) +{ + struct sdw_stream_data *stream; + + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (!stream) + return -ENOMEM; + + stream->sdw_stream = (struct sdw_stream_runtime *)sdw_stream; + + /* Use tx_mask or rx_mask to configure stream tag and set dma_data */ + if (direction == SNDRV_PCM_STREAM_PLAYBACK) + dai->playback_dma_data = stream; + else + dai->capture_dma_data = stream; + + return 0; +} + +static void rt5682_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sdw_stream_data *stream; + + stream = snd_soc_dai_get_dma_data(dai, substream); + snd_soc_dai_set_dma_data(dai, substream, NULL); + kfree(stream); +} + +static int rt5682_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config; + struct sdw_port_config port_config; + enum sdw_data_direction direction; + struct sdw_stream_data *stream; + int retval, port, num_channels; + unsigned int val_p = 0, val_c = 0, osr_p = 0, osr_c = 0; + + dev_dbg(dai->dev, "%s %s", __func__, dai->name); + stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!stream) + return -ENOMEM; + + if (!rt5682->slave) + return -EINVAL; + + /* SoundWire specific configuration */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + direction = SDW_DATA_DIR_RX; + port = 1; + } else { + direction = SDW_DATA_DIR_TX; + port = 2; + } + + stream_config.frame_rate = params_rate(params); + stream_config.ch_count = params_channels(params); + stream_config.bps = snd_pcm_format_width(params_format(params)); + stream_config.direction = direction; + + num_channels = params_channels(params); + port_config.ch_mask = (1 << (num_channels)) - 1; + port_config.num = port; + + retval = sdw_stream_add_slave(rt5682->slave, &stream_config, + &port_config, 1, stream->sdw_stream); + if (retval) { + dev_err(dai->dev, "Unable to configure port\n"); + return retval; + } + + switch (params_rate(params)) { + case 48000: + val_p = RT5682_SDW_REF_1_48K; + val_c = RT5682_SDW_REF_2_48K; + break; + case 96000: + val_p = RT5682_SDW_REF_1_96K; + val_c = RT5682_SDW_REF_2_96K; + break; + case 192000: + val_p = RT5682_SDW_REF_1_192K; + val_c = RT5682_SDW_REF_2_192K; + break; + case 32000: + val_p = RT5682_SDW_REF_1_32K; + val_c = RT5682_SDW_REF_2_32K; + break; + case 24000: + val_p = RT5682_SDW_REF_1_24K; + val_c = RT5682_SDW_REF_2_24K; + break; + case 16000: + val_p = RT5682_SDW_REF_1_16K; + val_c = RT5682_SDW_REF_2_16K; + break; + case 12000: + val_p = RT5682_SDW_REF_1_12K; + val_c = RT5682_SDW_REF_2_12K; + break; + case 8000: + val_p = RT5682_SDW_REF_1_8K; + val_c = RT5682_SDW_REF_2_8K; + break; + case 44100: + val_p = RT5682_SDW_REF_1_44K; + val_c = RT5682_SDW_REF_2_44K; + break; + case 88200: + val_p = RT5682_SDW_REF_1_88K; + val_c = RT5682_SDW_REF_2_88K; + break; + case 176400: + val_p = RT5682_SDW_REF_1_176K; + val_c = RT5682_SDW_REF_2_176K; + break; + case 22050: + val_p = RT5682_SDW_REF_1_22K; + val_c = RT5682_SDW_REF_2_22K; + break; + case 11025: + val_p = RT5682_SDW_REF_1_11K; + val_c = RT5682_SDW_REF_2_11K; + break; + default: + return -EINVAL; + } + + if (params_rate(params) <= 48000) { + osr_p = RT5682_DAC_OSR_D_8; + osr_c = RT5682_ADC_OSR_D_8; + } else if (params_rate(params) <= 96000) { + osr_p = RT5682_DAC_OSR_D_4; + osr_c = RT5682_ADC_OSR_D_4; + } else { + osr_p = RT5682_DAC_OSR_D_2; + osr_c = RT5682_ADC_OSR_D_2; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + regmap_update_bits(rt5682->regmap, RT5682_SDW_REF_CLK, + RT5682_SDW_REF_1_MASK, val_p); + regmap_update_bits(rt5682->regmap, RT5682_ADDA_CLK_1, + RT5682_DAC_OSR_MASK, osr_p); + } else { + regmap_update_bits(rt5682->regmap, RT5682_SDW_REF_CLK, + RT5682_SDW_REF_2_MASK, val_c); + regmap_update_bits(rt5682->regmap, RT5682_ADDA_CLK_1, + RT5682_ADC_OSR_MASK, osr_c); + } + + return retval; +} + +static int rt5682_sdw_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt5682_priv *rt5682 = snd_soc_component_get_drvdata(component); + struct sdw_stream_data *stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!rt5682->slave) + return -EINVAL; + + sdw_stream_remove_slave(rt5682->slave, stream->sdw_stream); + return 0; +} + +static struct snd_soc_dai_ops rt5682_sdw_ops = { + .hw_params = rt5682_sdw_hw_params, + .hw_free = rt5682_sdw_hw_free, + .set_sdw_stream = rt5682_set_sdw_stream, + .shutdown = rt5682_sdw_shutdown, +}; +#endif + static struct snd_soc_dai_driver rt5682_dai[] = { { .name = "rt5682-aif1", @@ -2946,6 +3158,27 @@ static struct snd_soc_dai_driver rt5682_dai[] = { }, .ops = &rt5682_aif2_dai_ops, }, +#if IS_ENABLED(CONFIG_SND_SOC_RT5682_SDW) + { + .name = "rt5682-sdw", + .id = RT5682_SDW, + .playback = { + .stream_name = "SDW Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682_STEREO_RATES, + .formats = RT5682_FORMATS, + }, + .capture = { + .stream_name = "SDW Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5682_STEREO_RATES, + .formats = RT5682_FORMATS, + }, + .ops = &rt5682_sdw_ops, + }, +#endif }; static const struct snd_soc_component_driver soc_component_dev_rt5682 = { @@ -3064,6 +3297,219 @@ static void rt5682_calibrate(struct rt5682_priv *rt5682) } +#if IS_ENABLED(CONFIG_SND_SOC_RT5682_SDW) +static int rt5682_sdw_read(void *context, unsigned int reg, unsigned int *val) +{ + struct device *dev = context; + struct rt5682_priv *rt5682 = dev_get_drvdata(dev); + unsigned int data_l, data_h; + + regmap_write(rt5682->sdw_regmap, RT5682_SDW_CMD, 0); + regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_H, (reg >> 8) & 0xff); + regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_L, (reg & 0xff)); + regmap_read(rt5682->sdw_regmap, RT5682_SDW_DATA_H, &data_h); + regmap_read(rt5682->sdw_regmap, RT5682_SDW_DATA_L, &data_l); + + *val = (data_h << 8) | data_l; + + dev_vdbg(dev, "[%s] %04x => %04x\n", __func__, reg, *val); + + return 0; +} + +static int rt5682_sdw_write(void *context, unsigned int reg, unsigned int val) +{ + struct device *dev = context; + struct rt5682_priv *rt5682 = dev_get_drvdata(dev); + + regmap_write(rt5682->sdw_regmap, RT5682_SDW_CMD, 1); + regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_H, (reg >> 8) & 0xff); + regmap_write(rt5682->sdw_regmap, RT5682_SDW_ADDR_L, (reg & 0xff)); + regmap_write(rt5682->sdw_regmap, RT5682_SDW_DATA_H, (val >> 8) & 0xff); + regmap_write(rt5682->sdw_regmap, RT5682_SDW_DATA_L, (val & 0xff)); + + dev_vdbg(dev, "[%s] %04x <= %04x\n", __func__, reg, val); + + return 0; +} + +static const struct regmap_config rt5682_sdw_regmap = { + .reg_bits = 16, + .val_bits = 16, + .max_register = RT5682_I2C_MODE, + .volatile_reg = rt5682_volatile_register, + .readable_reg = rt5682_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5682_reg, + .num_reg_defaults = ARRAY_SIZE(rt5682_reg), + .use_single_read = true, + .use_single_write = true, + .reg_read = rt5682_sdw_read, + .reg_write = rt5682_sdw_write, +}; + +int rt5682_sdw_init(struct device *dev, struct regmap *regmap, + struct sdw_slave *slave) +{ + struct rt5682_priv *rt5682; + int ret; + + rt5682 = devm_kzalloc(dev, sizeof(*rt5682), GFP_KERNEL); + if (!rt5682) + return -ENOMEM; + + dev_set_drvdata(dev, rt5682); + rt5682->slave = slave; + rt5682->sdw_regmap = regmap; + rt5682->is_sdw = true; + + rt5682->regmap = devm_regmap_init(dev, NULL, dev, &rt5682_sdw_regmap); + if (IS_ERR(rt5682->regmap)) { + ret = PTR_ERR(rt5682->regmap); + dev_err(dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + rt5682->hw_init = false; + rt5682->first_hw_init = false; + + mutex_init(&rt5682->calibrate_mutex); + INIT_DELAYED_WORK(&rt5682->jack_detect_work, + rt5682_jack_detect_handler); + + ret = devm_snd_soc_register_component(dev, &soc_component_dev_rt5682, + rt5682_dai, ARRAY_SIZE(rt5682_dai)); + + dev_dbg(&slave->dev, "%s\n", __func__); + + return ret; +} +EXPORT_SYMBOL_GPL(rt5682_sdw_init); + +int rt5682_io_init(struct device *dev, struct sdw_slave *slave) +{ + struct rt5682_priv *rt5682 = dev_get_drvdata(dev); + int ret = 0; + unsigned int val; + + if (rt5682->hw_init) + return 0; + + regmap_read(rt5682->regmap, RT5682_DEVICE_ID, &val); + if (val != DEVICE_ID) { + pr_err("Device with ID register %x is not rt5682\n", val); + return -ENODEV; + } + + /* + * PM runtime is only enabled when a Slave reports as Attached + */ + if (!rt5682->first_hw_init) { + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(&slave->dev, 3000); + pm_runtime_use_autosuspend(&slave->dev); + + /* update count of parent 'active' children */ + pm_runtime_set_active(&slave->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(&slave->dev); + + pm_runtime_enable(&slave->dev); + } + + pm_runtime_get_noresume(&slave->dev); + + rt5682_reset(rt5682); + + if (rt5682->first_hw_init) { + regcache_cache_only(rt5682->regmap, false); + regcache_cache_bypass(rt5682->regmap, true); + } + + rt5682_calibrate(rt5682); + + if (rt5682->first_hw_init) { + regcache_cache_bypass(rt5682->regmap, false); + regcache_mark_dirty(rt5682->regmap); + regcache_sync(rt5682->regmap); + + /* volatile registers */ + regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2, + RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); + + goto reinit; + } + + ret = regmap_multi_reg_write(rt5682->regmap, patch_list, + ARRAY_SIZE(patch_list)); + if (ret != 0) + dev_warn(dev, "Failed to apply regmap patch: %d\n", ret); + + regmap_write(rt5682->regmap, RT5682_DEPOP_1, 0x0000); + + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_1, + RT5682_LDO1_DVO_MASK | RT5682_HP_DRIVER_MASK, + RT5682_LDO1_DVO_12 | RT5682_HP_DRIVER_5X); + regmap_write(rt5682->regmap, RT5682_MICBIAS_2, 0x0380); + regmap_write(rt5682->regmap, RT5682_TEST_MODE_CTRL_1, 0x0000); + regmap_update_bits(rt5682->regmap, RT5682_BIAS_CUR_CTRL_8, + RT5682_HPA_CP_BIAS_CTRL_MASK, RT5682_HPA_CP_BIAS_3UA); + regmap_update_bits(rt5682->regmap, RT5682_CHARGE_PUMP_1, + RT5682_CP_CLK_HP_MASK, RT5682_CP_CLK_HP_300KHZ); + + /* Soundwire */ + regmap_write(rt5682->regmap, RT5682_PLL2_INTERNAL, 0xa266); + regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_1, 0x1700); + regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_2, 0x0006); + regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_3, 0x2600); + regmap_write(rt5682->regmap, RT5682_PLL2_CTRL_4, 0x0c8f); + regmap_write(rt5682->regmap, RT5682_PLL_TRACK_2, 0x3000); + regmap_write(rt5682->regmap, RT5682_PLL_TRACK_3, 0x4000); + regmap_update_bits(rt5682->regmap, RT5682_GLB_CLK, + RT5682_SCLK_SRC_MASK | RT5682_PLL2_SRC_MASK, + RT5682_SCLK_SRC_PLL2 | RT5682_PLL2_SRC_SDW); + + regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_2, + RT5682_EXT_JD_SRC, RT5682_EXT_JD_SRC_MANUAL); + regmap_write(rt5682->regmap, RT5682_CBJ_CTRL_1, 0xd042); + regmap_update_bits(rt5682->regmap, RT5682_CBJ_CTRL_3, + RT5682_CBJ_IN_BUF_EN, RT5682_CBJ_IN_BUF_EN); + regmap_update_bits(rt5682->regmap, RT5682_SAR_IL_CMD_1, + RT5682_SAR_POW_MASK, RT5682_SAR_POW_EN); + regmap_update_bits(rt5682->regmap, RT5682_RC_CLK_CTRL, + RT5682_POW_IRQ | RT5682_POW_JDH | + RT5682_POW_ANA, RT5682_POW_IRQ | + RT5682_POW_JDH | RT5682_POW_ANA); + regmap_update_bits(rt5682->regmap, RT5682_PWR_ANLG_2, + RT5682_PWR_JDH, RT5682_PWR_JDH); + regmap_update_bits(rt5682->regmap, RT5682_IRQ_CTRL_2, + RT5682_JD1_EN_MASK | RT5682_JD1_IRQ_MASK, + RT5682_JD1_EN | RT5682_JD1_IRQ_PUL); + +reinit: + mod_delayed_work(system_power_efficient_wq, + &rt5682->jack_detect_work, msecs_to_jiffies(250)); + + /* Mark Slave initialization complete */ + rt5682->hw_init = true; + rt5682->first_hw_init = true; + + pm_runtime_mark_last_busy(&slave->dev); + pm_runtime_put_autosuspend(&slave->dev); + + dev_dbg(&slave->dev, "%s hw_init complete\n", __func__); + + return ret; +} +EXPORT_SYMBOL_GPL(rt5682_io_init); +#endif + static int rt5682_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index f82126a6f211..43de6e802309 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -10,6 +10,12 @@ #define __RT5682_H__ #include <sound/rt5682.h> +#include <linux/regulator/consumer.h> +#include <linux/clk.h> +#include <linux/clkdev.h> +#include <linux/clk-provider.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_type.h> #define DEVICE_ID 0x6530 @@ -1355,6 +1361,7 @@ enum { enum { RT5682_AIF1, RT5682_AIF2, + RT5682_SDW, RT5682_AIFS }; @@ -1370,7 +1377,49 @@ enum { RT5682_CLK_SEL_I2S2_ASRC, }; +#define RT5682_NUM_SUPPLIES 3 + +struct rt5682_priv { + struct snd_soc_component *component; + struct rt5682_platform_data pdata; + struct regmap *regmap; + struct regmap *sdw_regmap; + struct snd_soc_jack *hs_jack; + struct regulator_bulk_data supplies[RT5682_NUM_SUPPLIES]; + struct delayed_work jack_detect_work; + struct delayed_work jd_check_work; + struct mutex calibrate_mutex; + struct sdw_slave *slave; + enum sdw_slave_status status; + struct sdw_bus_params params; + bool hw_init; + bool first_hw_init; + bool is_sdw; + +#ifdef CONFIG_COMMON_CLK + struct clk_hw dai_clks_hw[RT5682_DAI_NUM_CLKS]; + struct clk_lookup *dai_clks_lookup[RT5682_DAI_NUM_CLKS]; + struct clk *dai_clks[RT5682_DAI_NUM_CLKS]; + struct clk *mclk; +#endif + + int sysclk; + int sysclk_src; + int lrck[RT5682_AIFS]; + int bclk[RT5682_AIFS]; + int master[RT5682_AIFS]; + + int pll_src[RT5682_PLLS]; + int pll_in[RT5682_PLLS]; + int pll_out[RT5682_PLLS]; + + int jack_type; +}; + int rt5682_sel_asrc_clk_src(struct snd_soc_component *component, unsigned int filter_mask, unsigned int clk_src); +int rt5682_sdw_init(struct device *dev, struct regmap *regmap, + struct sdw_slave *slave); +int rt5682_io_init(struct device *dev, struct sdw_slave *slave); #endif /* __RT5682_H__ */ -- 2.25.0

2 3

[PATCH] ASoC: amd: Allow I2S wake event after ACP is powerd On
by Akshu Agrawal 26 Feb '20

26 Feb '20

ACP_PME_EN allows wake interrupt to be generated when I2S wake feature is enabled. On turning ACP On, ACP_PME_EN gets cleared. Setting the bit back ensures that wake event can be received when ACP is On. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/raven/pci-acp3x.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/pci-acp3x.c b/sound/soc/amd/raven/pci-acp3x.c index da60e2ec5535..f25ce50f1a90 100644 --- a/sound/soc/amd/raven/pci-acp3x.c +++ b/sound/soc/amd/raven/pci-acp3x.c @@ -38,8 +38,13 @@ static int acp3x_power_on(void __iomem *acp3x_base) timeout = 0; while (++timeout < 500) { val = rv_readl(acp3x_base + mmACP_PGFSM_STATUS); - if (!val) + if (!val) { + /* Set PME_EN as after ACP power On, + * PME_EN gets cleared + */ + rv_writel(0x1, acp3x_base + mmACP_PME_EN); return 0; + } udelay(1); } return -ETIMEDOUT; -- 2.17.1

2 1

Applied "ASoC: dt-bindings: Add TAS2563 compatible to the TAS2562 binding" to the asoc tree
by Mark Brown 26 Feb '20

26 Feb '20

The patch ASoC: dt-bindings: Add TAS2563 compatible to the TAS2562 binding has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f87cdb1f9937e6f5234e3300804ac156e639bc00 Mon Sep 17 00:00:00 2001 From: Dan Murphy <dmurphy(a)ti.com> Date: Wed, 26 Feb 2020 07:03:03 -0600 Subject: [PATCH] ASoC: dt-bindings: Add TAS2563 compatible to the TAS2562 binding Add the Texas Instruments TAS2563 audio amplifier to the TAS262 binding. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> CC: Rob Herring <robh(a)kernel.org> Link: https://lore.kernel.org/r/20200226130305.12043-1-dmurphy@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/tas2562.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt index 658e1fb18a99..94796b547184 100644 --- a/Documentation/devicetree/bindings/sound/tas2562.txt +++ b/Documentation/devicetree/bindings/sound/tas2562.txt @@ -8,7 +8,7 @@ real time monitoring of loudspeaker behavior. Required properties: - #address-cells - Should be <1>. - #size-cells - Should be <0>. - - compatible: - Should contain "ti,tas2562". + - compatible: - Should contain "ti,tas2562", "ti,tas2563". - reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f. - ti,imon-slot-no:- TDM TX current sense time slot. -- 2.20.1

1 0

Applied "ASoC: tas2562: Add entries for the TAS2563 audio amplifier" to the asoc tree
by Mark Brown 26 Feb '20

26 Feb '20

The patch ASoC: tas2562: Add entries for the TAS2563 audio amplifier has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 14f8c8d8fd62207f081549d45099a90dd3717696 Mon Sep 17 00:00:00 2001 From: Dan Murphy <dmurphy(a)ti.com> Date: Wed, 26 Feb 2020 07:03:04 -0600 Subject: [PATCH] ASoC: tas2562: Add entries for the TAS2563 audio amplifier The TAS2563 is register compatible with the TAS2562. The main difference is the TAS2563 has a programmable DSP to manage different audio profiles. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> Link: https://lore.kernel.org/r/20200226130305.12043-2-dmurphy@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tas2562.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index d5e04030a0c1..79c3c3d79766 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -55,6 +55,11 @@ struct tas2562_data { int volume_lvl; }; +enum tas256x_model { + TAS2562, + TAS2563, +}; + static int tas2562_set_bias_level(struct snd_soc_component *component, enum snd_soc_bias_level level) { @@ -664,13 +669,15 @@ static int tas2562_probe(struct i2c_client *client, } static const struct i2c_device_id tas2562_id[] = { - { "tas2562", 0 }, + { "tas2562", TAS2562 }, + { "tas2563", TAS2563 }, { } }; MODULE_DEVICE_TABLE(i2c, tas2562_id); static const struct of_device_id tas2562_of_match[] = { { .compatible = "ti,tas2562", }, + { .compatible = "ti,tas2563", }, { }, }; MODULE_DEVICE_TABLE(of, tas2562_of_match); -- 2.20.1

1 0

[RESEND PATCH for-next 1/3] ASoC: tas2562: Fix sample rate error message
by Dan Murphy 26 Feb '20

26 Feb '20

Fix error message for setting the sample rate. It says bitwidth but should say sample rate. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tas2562.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index d5e04030a0c1..c7a4b3b74654 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -271,7 +271,7 @@ static int tas2562_hw_params(struct snd_pcm_substream *substream, ret = tas2562_set_samplerate(tas2562, params_rate(params)); if (ret) - dev_err(tas2562->dev, "set bitwidth failed, %d\n", ret); + dev_err(tas2562->dev, "set sample rate failed, %d\n", ret); return ret; } -- 2.25.0

1 2

[PATCH for-next 1/3] ASoC: dt-bindings: Add TAS2563 compatible to the TAS2562 binding
by Dan Murphy 26 Feb '20

26 Feb '20

Add the Texas Instruments TAS2563 audio amplifier to the TAS262 binding. CC: Rob Herring <robh(a)kernel.org> Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- Documentation/devicetree/bindings/sound/tas2562.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tas2562.txt b/Documentation/devicetree/bindings/sound/tas2562.txt index 658e1fb18a99..94796b547184 100644 --- a/Documentation/devicetree/bindings/sound/tas2562.txt +++ b/Documentation/devicetree/bindings/sound/tas2562.txt @@ -8,7 +8,7 @@ real time monitoring of loudspeaker behavior. Required properties: - #address-cells - Should be <1>. - #size-cells - Should be <0>. - - compatible: - Should contain "ti,tas2562". + - compatible: - Should contain "ti,tas2562", "ti,tas2563". - reg: - The i2c address. Should be 0x4c, 0x4d, 0x4e or 0x4f. - ti,imon-slot-no:- TDM TX current sense time slot. -- 2.25.0

2 4

[alsa-devel] [RFC PATCH 00/34] sun8i-codec fixes and new features
by Samuel Holland 26 Feb '20

26 Feb '20

Hello all, The sun8i-codec driver, as used in the Allwinner A33 and A64, currently only exposes a small subset of the available hardware features. In order to use the A64 in a smartphone (the PinePhone), I've added the necessary functionality to the driver: * The full set of supported DAI format options * Support for AIF2 and AIF3 * Additional routing knobs * Additional volume controls Unfortunately, due to preexisting issues with the driver, there are some breaking changes, as explained further in the commit messages: * The LRCK inversion issue means we need a new compatible for the A64. * Some controls are named inaccurately, so they are renamed. * Likewise, the DAPM widgets used in device trees were either named wrong, or the device trees were using the wrong widgets in the first place. (Specifically, the links between the analog codec and digital codec happen at the ADC and DAC, not AIF1.) I tended to take the philosophy of "while I'm breaking things, I might as well do them right", so I've probably made a few more changes than absolutely necessary. I'm not sure about where all of the policy boundaries are, about how far I should go to maintain compatibility. For example, for the DT widget usage, I could: * Rename everything and update the DTS files (which is what I did) * Keep the old (misleading/wrong) name for the widgets, but repurpose them to work correctly (i.e. "ADC Left" would be named "AIF1 Slot 0 Left ADC", but it would work just like "ADC Left" does in this patchset) * Keep the old widgets around as a compatibility layer, but add new widgets and update the in-tree DTS files to use them (i.e. "ADC Left" would have a path from "AIF1 Slot 0 Left ADC", but "AIF1 Slot 0 Left ADC" would be a no-op widget) * Something else entirely There are several trivial fixes in here, and there are several commits that just add new features without changing any existing behavior, but there is enough changing that I thought it would be best to send the whole thing as an RFC. I'm more than happy to reorganize this into one or several patchsets in future revisions. It doesn't have to all go in at once. This has all been tested on the PinePhone. For example, I've successfully routed a voice call between the modem at AIF2 and a Bluetooth headset at AIF3, merged with an audio file played from AIF1. Unfortunately, the PinePhone does not have a DTS upstream yet (partially due to the audio driver situation), so I don't have an example DTS in this patchset. Thanks for any feedback, Samuel Samuel Holland (34): ASoC: dt-bindings: Add a separate compatible for the A64 codec ASoC: sun8i-codec: LRCK is not inverted on A64 arm64: dts: allwinner: a64: Fix the audio codec compatible ASoC: sun8i-codec: Remove unused dev from codec struct ASoC: sun8i-codec: Remove incorrect SND_SOC_DAIFMT_DSP_B ASoC: sun8i-codec: Fix setting DAI data format ASoC: sun8i-codec: Remove extraneous widgets ASoC: sun8i-codec: Fix direction of AIF1 outputs ASoC: sun8i-codec: Fix broken DAPM routing ASoC: sun8i-codec: Advertise only hardware-supported rates ASoC: sun8i-codec: Enforce parameter symmetry ASoC: sun8i-codec: Fix AIF1 MODCLK widget name ASoC: sun8i-codec: Fix AIF1_ADCDAT_CTRL field names ASoC: sun8i-codec: Fix AIF1_MXR_SRC field names ASoC: sun8i-codec: Fix ADC_DIG_CTRL field name ASoC: sun8i-codec: Fix field bit number indentation ASoC: sun8i-codec: Sort masks in a consistent order ASoC: sun8i-codec: Allow all clock inversion permutations ASoC: sun8i-codec: Support mono DAI configurations ASoC: sun8i-codec: Support 8/20/24-bit word sizes ASoC: sun8i-codec: Clean up module/clock hierarchy ASoC: sun8i-codec: Clean up AIF1 Slot 0 widgets ASoC: sun8i-codec: Clean up DAC widgets ASoC: sun8i-codec: Prepare to support multiple AIFs ASoC: sun8i-codec: Add support for AIF2 ASoC: sun8i-codec: Add support for AIF3 ASoC: sun8i-codec: Add AIF mono/stereo controls ASoC: sun8i-codec: Add AIF loopback controls ASoC: sun8i-codec: Add AIF, ADC, and DAC volume controls ASoC: dt-bindings: Bump sound-dai-cells on sun8i-codec ARM: dts: sun8i-a33: Allow using multiple codec DAIs arm64: dts: allwinner: a64: Allow using multiple codec DAIs arm64: dts: allwinner: a64: Allow multiple DAI links arm64: dts: allwinner: a64: Add pinmux for AIF2/AIF3 .../sound/allwinner,sun8i-a33-codec.yaml | 6 +- arch/arm/boot/dts/sun8i-a33-olinuxino.dts | 6 +- arch/arm/boot/dts/sun8i-a33.dtsi | 10 +- .../dts/allwinner/sun50i-a64-bananapi-m64.dts | 8 +- .../dts/allwinner/sun50i-a64-orangepi-win.dts | 8 +- .../boot/dts/allwinner/sun50i-a64-pine64.dts | 8 +- .../dts/allwinner/sun50i-a64-pinebook.dts | 8 +- .../boot/dts/allwinner/sun50i-a64-pinetab.dts | 14 +- .../allwinner/sun50i-a64-sopine-baseboard.dts | 8 +- .../boot/dts/allwinner/sun50i-a64-teres-i.dts | 8 +- arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 47 +- sound/soc/sunxi/sun8i-codec.c | 1056 +++++++++++++---- 12 files changed, 890 insertions(+), 297 deletions(-) -- 2.24.1

5 68

[PATCH] soundwire: bus: provide correct return value on error
by Pierre-Louis Bossart 26 Feb '20

26 Feb '20

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> It seems to be a typo. It makes more sense to return the return value of sdw_update() instead of the value we want to update. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- drivers/soundwire/bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index 13887713f311..b8a7a84aca1c 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -1070,7 +1070,7 @@ static int sdw_initialize_slave(struct sdw_slave *slave) if (ret < 0) { dev_err(slave->bus->dev, "SDW_DP0_INTMASK read failed:%d\n", ret); - return val; + return ret; } return 0; -- 2.20.1

2 2

[PATCH v3 0/2] Introduce the TLV320ADCx140 codec family
by Dan Murphy 26 Feb '20

26 Feb '20

Hello Introducing the Texas Instruments TLV320ADCx140 quad channel audio ADC. This device supports 4 analog inputs, 8 digital inputs or a combination of analog and digital microphones. TLV320ADC3140 - http://www.ti.com/lit/gpn/tlv320adc3140 TLV320ADC5140 - http://www.ti.com/lit/gpn/tlv320adc5140 TLV320ADC6140 - http://www.ti.com/lit/gpn/tlv320adc6140 Dan Murphy (2): dt-bindings: sound: Add TLV320ADCx140 dt bindings ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family .../bindings/sound/tlv320adcx140.yaml | 83 ++ sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adcx140.c | 849 ++++++++++++++++++ sound/soc/codecs/tlv320adcx140.h | 130 +++ 5 files changed, 1073 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320adcx140.yaml create mode 100644 sound/soc/codecs/tlv320adcx140.c create mode 100644 sound/soc/codecs/tlv320adcx140.h -- 2.25.0

3 9

[PATCH v2 0/3] Add new module driver for new ASRC
by Shengjiu Wang 26 Feb '20

26 Feb '20

Add new module driver for new ASRC in i.MX8MN. Shengjiu Wang (3): ASoC: fsl_asrc: Move common definition to fsl_asrc_common ASoC: dt-bindings: fsl_easrc: Add document for EASRC ASoC: fsl_easrc: Add EASRC ASoC CPU DAI and platform drivers changes in v2 - change i.MX815 to i.MX8MN - Add changes in Kconfig and Makefile .../devicetree/bindings/sound/fsl,easrc.txt | 57 + sound/soc/fsl/Kconfig | 10 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_asrc.h | 11 +- sound/soc/fsl/fsl_asrc_common.h | 22 + sound/soc/fsl/fsl_easrc.c | 2265 +++++++++++++++++ sound/soc/fsl/fsl_easrc.h | 668 +++++ sound/soc/fsl/fsl_easrc_dma.c | 440 ++++ 8 files changed, 3465 insertions(+), 10 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/fsl,easrc.txt create mode 100644 sound/soc/fsl/fsl_asrc_common.h create mode 100644 sound/soc/fsl/fsl_easrc.c create mode 100644 sound/soc/fsl/fsl_easrc.h create mode 100644 sound/soc/fsl/fsl_easrc_dma.c -- 2.21.0

3 5

[PATCH] ASoC: rt1015: modify some structure to be static and add operation callback function for rt1015_dai
by jack.yu＠realtek.com 26 Feb '20

26 Feb '20

From: Jack Yu <jack.yu(a)realtek.com> 1. Add "static" for rt1015_aif_dai_ops. 2. Add "static" for rt1015_dai[]. 3. Add operation callback for rt1015_dai[]. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- sound/soc/codecs/rt1015.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c index d300b417dd50..bb310bc7febd 100644 --- a/sound/soc/codecs/rt1015.c +++ b/sound/soc/codecs/rt1015.c @@ -664,7 +664,7 @@ static int rt1015_hw_params(struct snd_pcm_substream *substream, snd_soc_component_update_bits(component, RT1015_TDM_MASTER, RT1015_I2S_DL_MASK, val_len); snd_soc_component_update_bits(component, RT1015_CLK2, - RT1015_FS_PD_MASK, pre_div); + RT1015_FS_PD_MASK, pre_div << RT1015_FS_PD_SFT); return 0; } @@ -841,12 +841,12 @@ static void rt1015_remove(struct snd_soc_component *component) #define RT1015_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 rt1015_aif_dai_ops = { +static struct snd_soc_dai_ops rt1015_aif_dai_ops = { .hw_params = rt1015_hw_params, .set_fmt = rt1015_set_dai_fmt, }; -struct snd_soc_dai_driver rt1015_dai[] = { +static struct snd_soc_dai_driver rt1015_dai[] = { { .name = "rt1015-aif", .id = 0, @@ -857,6 +857,7 @@ struct snd_soc_dai_driver rt1015_dai[] = { .rates = RT1015_STEREO_RATES, .formats = RT1015_FORMATS, }, + .ops = &rt1015_aif_dai_ops, } }; -- 2.25.1

2 1

[alsa-devel] Question about the various mixer options in UCM
by Tanu Kaskinen 26 Feb '20

26 Feb '20

What's the difference between PlaybackVolume, PlaybackMixerElem and PlaybackMasterElem? Other than the obvious difference that PlaybackVolume is used only to configure the volume control, whereas PlaybackMixerElem and PlaybackMasterElem are used also to configure the mute control. -- Tanu https://www.patreon.com/tanuk https://liberapay.com/tanuk

2 8

Re: [PATCH 0/1] Fix silent output on Gigabyte X570 Aorus Master
by Chris 26 Feb '20

26 Feb '20

Hello, on Sunday, Feb 23, I sent a patch (Fix silent output on Gigabyte X570 Aorus Master) for a sound-problem my motherboard has and made the mistake of not registering to alsa-devel beforehand. As a result, my mails are pending for approval in mailman and wont show up in the mailinglist. I did the registration now. Can someone please approve my previous mails or should I just resend my patches? I am really sorry for the mishap - it wont happen again. Greetings, Chris On Sun, Feb 23, 2020 at 10:27 AM Christian Lachner <gladiac(a)gmail.com> wrote: > > The Gigabyte X570 Aorus Master motherboard with ALC1220 codec > requires a similar workaround for Clevo laptops to enforce the > DAC/mixer connection path. This patch sets up a quirk entry for > that. It was tested by myself on my own hardware for some time > and it works just fine. We should probably rename > ALC1220_FIXUP_CLEVO_P950 at some point as the amount of > non-Clevo hardware needing this workaround is growing. > > Christian Lachner (1): > ALSA: hda/realtek - Fix silent output on Gigabyte X570 Aorus Master > > sound/pci/hda/patch_realtek.c | 1 + > 1 file changed, 1 insertion(+) > > -- > 2.25.1 >

1 0

Most speakers don't work
by GitHub issues - opened 26 Feb '20

26 Feb '20

alsa-project/alsa-lib issue #28 was opened from JonathanReeve: I have a Lenovo Yoga C-930, which has a Dolby sound bar, and it seems that only one or two out of six speakers work. Is there a way I can make this work? Here's my alsa-info: http://alsa-project.org/db/?f=87931920f5a8cbfcaa9a2490b88d8c1ed6337e9f Issue URL : https://github.com/alsa-project/alsa-lib/issues/28 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] Add headset Mic no shutup for ALC283
by Kailang 26 Feb '20

26 Feb '20

Hi Takashi, ChromeBox machine had humming noise from external skeaker plugin at codec D3 state. BR, Kailang

3 2

Re: [PATCH v2 3/3] ASoC: fsl_easrc: Add EASRC ASoC CPU DAI and platform drivers
by S.j. Wang 26 Feb '20

26 Feb '20

Hi > > > > Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > > --- > > sound/soc/fsl/Kconfig | 10 + > > sound/soc/fsl/Makefile | 2 + > > sound/soc/fsl/fsl_asrc_common.h | 1 + > > sound/soc/fsl/fsl_easrc.c | 2265 +++++++++++++++++++++++++++++++ > > sound/soc/fsl/fsl_easrc.h | 668 +++++++++ > > sound/soc/fsl/fsl_easrc_dma.c | 440 ++++++ > > I see a 90% similarity between fsl_asrc_dma and fsl_easrc_dma files. > Would it be possible reuse the existing code? Could share structures from > my point of view, just like it reuses "enum asrc_pair_index", I know > differentiating "pair" and "context" is a big point here though. > > A possible quick solution for that, off the top of my head, could be: > > 1) in fsl_asrc_common.h > > struct fsl_asrc { > .... > }; > > struct fsl_asrc_pair { > .... > }; > > 2) in fsl_easrc.h > > /* Renaming shared structures */ > #define fsl_easrc fsl_asrc > #define fsl_easrc_context fsl_asrc_pair > > May be a good idea to see if others have some opinion too. > We need to modify the fsl_asrc and fsl_asrc_pair, let them To be used by both driver, also we need to put the specific Definition for each module to same struct, right? > > > +static const struct regmap_config fsl_easrc_regmap_config = { > > + .readable_reg = fsl_easrc_readable_reg, > > + .volatile_reg = fsl_easrc_volatile_reg, > > + .writeable_reg = fsl_easrc_writeable_reg, > > Can we use regmap_range and regmap_access_table? > Can the regmap_range support discontinuous registers? The reg_stride = 4. Best regards Wang shengjiu

3 3

[alsa-devel] motu microbook iic
by thomas rost 26 Feb '20

26 Feb '20

Hello, can anybody help me get the 'class compliant' motu microbook iic to run? this is my alsa-info.sh output: http://alsa-project.org/db/?f=0fd9a8df7bae3400d83b6af8b75c45aa0760d84d thanks in advance. thomas

2 1

[alsa-devel] [PATCH] ASoC: rt1015: add rt1015 amplifier driver
by jack.yu＠realtek.com 26 Feb '20

26 Feb '20

From: Jack Yu <jack.yu(a)realtek.com> This is initial amplifier driver for rt1015. Signed-off-by: Jack Yu <jack.yu(a)realtek.com> --- .../devicetree/bindings/sound/rt1015.txt | 17 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt1015.c | 1016 +++++++++++++++++ sound/soc/codecs/rt1015.h | 368 ++++++ 5 files changed, 1409 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt1015.txt create mode 100644 sound/soc/codecs/rt1015.c create mode 100644 sound/soc/codecs/rt1015.h diff --git a/Documentation/devicetree/bindings/sound/rt1015.txt b/Documentation/devicetree/bindings/sound/rt1015.txt new file mode 100644 index 000000000000..fcfd02d8d32f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt1015.txt @@ -0,0 +1,17 @@ +RT1015 Mono Class D Audio Amplifier + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt1015". + +- reg : The I2C address of the device. + + +Example: + +rt1015: codec@28 { + compatible = "realtek,rt1015"; + reg = <0x28>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index cdfd912d5f8f..d7475e06c3e6 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -149,6 +149,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT286 if I2C select SND_SOC_RT298 if I2C select SND_SOC_RT1011 if I2C + select SND_SOC_RT1015 if I2C select SND_SOC_RT1305 if I2C select SND_SOC_RT1308 if I2C select SND_SOC_RT5514 if I2C @@ -956,6 +957,7 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5677=y default y if SND_SOC_RT5682=y default y if SND_SOC_RT1011=y + default y if SND_SOC_RT1015=y default y if SND_SOC_RT1305=y default y if SND_SOC_RT1308=y default m if SND_SOC_RT5514=m @@ -972,6 +974,7 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5677=m default m if SND_SOC_RT5682=m default m if SND_SOC_RT1011=m + default m if SND_SOC_RT1015=m default m if SND_SOC_RT1305=m default m if SND_SOC_RT1308=m @@ -999,6 +1002,9 @@ config SND_SOC_RT298 config SND_SOC_RT1011 tristate +config SND_SOC_RT1015 + tristate + config SND_SOC_RT1305 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index ce285b33a806..3efe6a5797e2 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -152,6 +152,7 @@ snd-soc-rk3328-objs := rk3328_codec.o snd-soc-rl6231-objs := rl6231.o snd-soc-rl6347a-objs := rl6347a.o snd-soc-rt1011-objs := rt1011.o +snd-soc-rt1015-objs := rt1015.o snd-soc-rt1305-objs := rt1305.o snd-soc-rt1308-objs := rt1308.o snd-soc-rt274-objs := rt274.o @@ -446,6 +447,7 @@ obj-$(CONFIG_SND_SOC_RK3328) += snd-soc-rk3328.o obj-$(CONFIG_SND_SOC_RL6231) += snd-soc-rl6231.o obj-$(CONFIG_SND_SOC_RL6347A) += snd-soc-rl6347a.o obj-$(CONFIG_SND_SOC_RT1011) += snd-soc-rt1011.o +obj-$(CONFIG_SND_SOC_RT1015) += snd-soc-rt1015.o obj-$(CONFIG_SND_SOC_RT1305) += snd-soc-rt1305.o obj-$(CONFIG_SND_SOC_RT1308) += snd-soc-rt1308.o obj-$(CONFIG_SND_SOC_RT274) += snd-soc-rt274.o diff --git a/sound/soc/codecs/rt1015.c b/sound/soc/codecs/rt1015.c new file mode 100644 index 000000000000..317c914cbfc9 --- /dev/null +++ b/sound/soc/codecs/rt1015.c @@ -0,0 +1,1016 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * rt1015.c -- RT1015 ALSA SoC audio amplifier driver + * + * Copyright 2019 Realtek Semiconductor Corp. + * Author: Jack Yu <jack.yu(a)realtek.com> + * + */ + +#include <linux/fs.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/regmap.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/firmware.h> +#include <linux/gpio.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 "rl6231.h" +#include "rt1015.h" + +static const struct reg_sequence init_list[] = { + { 0x0000, 0x0000 }, + { 0x0104, 0xA213 }, + { 0x010E, 0x0058 }, + { 0x0111, 0x2000 }, + { 0x0322, 0xF7DF }, + { 0x1302, 0x12F7 }, + { 0x0336, 0x0010 }, + { 0x04FC, 0x35CF }, + { 0x0102, 0xed02 }, +}; +#define RT1015_INIT_REG_LEN ARRAY_SIZE(init_list) + +static const struct reg_default rt1015_reg[] = { + { 0x0000, 0x0000 }, + { 0x0004, 0xa000 }, + { 0x0006, 0x0003 }, + { 0x000a, 0x0802 }, + { 0x000c, 0x0020 }, + { 0x000e, 0x0000 }, + { 0x0010, 0x0000 }, + { 0x0012, 0x0000 }, + { 0x0020, 0x8000 }, + { 0x0022, 0x471b }, + { 0x006a, 0x0000 }, + { 0x006c, 0x4020 }, + { 0x0076, 0x0000 }, + { 0x0078, 0x0000 }, + { 0x007a, 0x0000 }, + { 0x007c, 0x10ec }, + { 0x007d, 0x1015 }, + { 0x00f0, 0x5000 }, + { 0x00f2, 0x0774 }, + { 0x00f3, 0x8400 }, + { 0x00f4, 0x0000 }, + { 0x0100, 0x0028 }, + { 0x0102, 0xff02 }, + { 0x0104, 0x8232 }, + { 0x0106, 0x200c }, + { 0x010c, 0x002f }, + { 0x010e, 0xc000 }, + { 0x0111, 0x0200 }, + { 0x0112, 0x0400 }, + { 0x0114, 0x0022 }, + { 0x0116, 0x0000 }, + { 0x0118, 0x0000 }, + { 0x011a, 0x0123 }, + { 0x011c, 0x4567 }, + { 0x0300, 0xdddd }, + { 0x0302, 0x0000 }, + { 0x0311, 0x9330 }, + { 0x0313, 0x0000 }, + { 0x0314, 0x0000 }, + { 0x031a, 0x00a0 }, + { 0x031c, 0x001f }, + { 0x031d, 0xffff }, + { 0x031e, 0x0000 }, + { 0x031f, 0x0000 }, + { 0x0321, 0x0000 }, + { 0x0322, 0x0000 }, + { 0x0328, 0x0000 }, + { 0x0329, 0x0000 }, + { 0x032a, 0x0000 }, + { 0x032b, 0x0000 }, + { 0x032c, 0x0000 }, + { 0x032d, 0x0000 }, + { 0x032e, 0x030e }, + { 0x0330, 0x0080 }, + { 0x0332, 0x0034 }, + { 0x0334, 0x0000 }, + { 0x0336, 0x0000 }, + { 0x0506, 0x04ff }, + { 0x0508, 0x0030 }, + { 0x050a, 0x0018 }, + { 0x0519, 0x307f }, + { 0x051a, 0xffff }, + { 0x051b, 0x4000 }, + { 0x051d, 0x0000 }, + { 0x051f, 0x0000 }, + { 0x0536, 0x1000 }, + { 0x0538, 0x0000 }, + { 0x053a, 0x0000 }, + { 0x053c, 0x0000 }, + { 0x053d, 0x0000 }, + { 0x053e, 0x0000 }, + { 0x053f, 0x0000 }, + { 0x0540, 0x0000 }, + { 0x0541, 0x0000 }, + { 0x0542, 0x0000 }, + { 0x0543, 0x0000 }, + { 0x0544, 0x0000 }, + { 0x0568, 0x0000 }, + { 0x056a, 0x0000 }, + { 0x1000, 0x0000 }, + { 0x1002, 0x6505 }, + { 0x1006, 0x5515 }, + { 0x1007, 0x003f }, + { 0x1009, 0x770f }, + { 0x100a, 0x01ff }, + { 0x100c, 0x0000 }, + { 0x100d, 0x0003 }, + { 0x1010, 0xa433 }, + { 0x1020, 0x0000 }, + { 0x1200, 0x3d02 }, + { 0x1202, 0x0813 }, + { 0x1204, 0x0211 }, + { 0x1206, 0x0000 }, + { 0x1208, 0x0000 }, + { 0x120a, 0x0000 }, + { 0x120c, 0x0000 }, + { 0x120e, 0x0000 }, + { 0x1210, 0x0000 }, + { 0x1212, 0x0000 }, + { 0x1300, 0x0701 }, + { 0x1302, 0x12f9 }, + { 0x1304, 0x3405 }, + { 0x1305, 0x0844 }, + { 0x1306, 0x1611 }, + { 0x1308, 0x555e }, + { 0x130a, 0x0000 }, + { 0x130c, 0x2400}, + { 0x130e, 0x7700 }, + { 0x130f, 0x0000 }, + { 0x1310, 0x0000 }, + { 0x1312, 0x0000 }, + { 0x1314, 0x0000 }, + { 0x1316, 0x0000 }, + { 0x1318, 0x0000 }, + { 0x131a, 0x0000 }, + { 0x1322, 0x0029 }, + { 0x1323, 0x4a52 }, + { 0x1324, 0x002c }, + { 0x1325, 0x0b02 }, + { 0x1326, 0x002d }, + { 0x1327, 0x6b5a }, + { 0x1328, 0x002e }, + { 0x1329, 0xcbb2 }, + { 0x132a, 0x0030 }, + { 0x132b, 0x2c0b }, + { 0x1330, 0x0031 }, + { 0x1331, 0x8c63 }, + { 0x1332, 0x0032 }, + { 0x1333, 0xecbb }, + { 0x1334, 0x0034 }, + { 0x1335, 0x4d13 }, + { 0x1336, 0x0037 }, + { 0x1337, 0x0dc3 }, + { 0x1338, 0x003d }, + { 0x1339, 0xef7b }, + { 0x133a, 0x0044 }, + { 0x133b, 0xd134 }, + { 0x133c, 0x0047 }, + { 0x133d, 0x91e4 }, + { 0x133e, 0x004d }, + { 0x133f, 0xc370 }, + { 0x1340, 0x0053 }, + { 0x1341, 0xf4fd }, + { 0x1342, 0x0060 }, + { 0x1343, 0x5816 }, + { 0x1344, 0x006c }, + { 0x1345, 0xbb2e }, + { 0x1346, 0x0072 }, + { 0x1347, 0xecbb }, + { 0x1348, 0x0076 }, + { 0x1349, 0x5d97 }, +}; + +static int rt1015_reg_init(struct snd_soc_component *component) +{ + struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component); + + regmap_multi_reg_write(rt1015->regmap, init_list, RT1015_INIT_REG_LEN); + return 0; +} + +static bool rt1015_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT1015_RESET: + case RT1015_CLK_DET: + case RT1015_SIL_DET: + case RT1015_VER_ID: + case RT1015_VENDOR_ID: + case RT1015_DEVICE_ID: + case RT1015_PRO_ALT: + case RT1015_DAC3: + case RT1015_VBAT_TEST_OUT1: + case RT1015_VBAT_TEST_OUT2: + case RT1015_VBAT_PROT_ATT: + case RT1015_VBAT_DET_CODE: + case RT1015_SMART_BST_CTRL1: + case RT1015_SPK_DC_DETECT1: + case RT1015_SPK_DC_DETECT4: + case RT1015_SPK_DC_DETECT5: + case RT1015_DC_CALIB_CLSD1: + case RT1015_DC_CALIB_CLSD5: + case RT1015_DC_CALIB_CLSD6: + case RT1015_DC_CALIB_CLSD7: + case RT1015_DC_CALIB_CLSD8: + case RT1015_S_BST_TIMING_INTER1: + return true; + + default: + return false; + } +} + +static bool rt1015_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT1015_RESET: + case RT1015_CLK2: + case RT1015_CLK3: + case RT1015_PLL1: + case RT1015_PLL2: + case RT1015_CLK_DET: + case RT1015_SIL_DET: + case RT1015_CUSTOMER_ID: + case RT1015_PCODE_FWVER: + case RT1015_VER_ID: + case RT1015_VENDOR_ID: + case RT1015_DEVICE_ID: + case RT1015_PAD_DRV1: + case RT1015_PAD_DRV2: + case RT1015_GAT_BOOST: + case RT1015_PRO_ALT: + case RT1015_MAN_I2C: + case RT1015_DAC1: + case RT1015_DAC2: + case RT1015_DAC3: + case RT1015_ADC1: + case RT1015_ADC2: + case RT1015_TDM_MASTER: + case RT1015_TDM_TCON: + case RT1015_TDM1_1: + case RT1015_TDM1_2: + case RT1015_TDM1_3: + case RT1015_TDM1_4: + case RT1015_TDM1_5: + case RT1015_MIXER1: + case RT1015_MIXER2: + case RT1015_ANA_PROTECT1: + case RT1015_ANA_CTRL_SEQ1: + case RT1015_ANA_CTRL_SEQ2: + case RT1015_VBAT_DET_DEB: + case RT1015_VBAT_VOLT_DET1: + case RT1015_VBAT_VOLT_DET2: + case RT1015_VBAT_TEST_OUT1: + case RT1015_VBAT_TEST_OUT2: + case RT1015_VBAT_PROT_ATT: + case RT1015_VBAT_DET_CODE: + case RT1015_PWR1: + case RT1015_PWR4: + case RT1015_PWR5: + case RT1015_PWR6: + case RT1015_PWR7: + case RT1015_PWR8: + case RT1015_PWR9: + case RT1015_CLASSD_SEQ: + case RT1015_SMART_BST_CTRL1: + case RT1015_SMART_BST_CTRL2: + case RT1015_ANA_CTRL1: + case RT1015_ANA_CTRL2: + case RT1015_SPK_VOL: + case RT1015_SHORT_DETTOP1: + case RT1015_SHORT_DETTOP2: + case RT1015_SPK_DC_DETECT1: + case RT1015_SPK_DC_DETECT2: + case RT1015_SPK_DC_DETECT3: + case RT1015_SPK_DC_DETECT4: + case RT1015_SPK_DC_DETECT5: + case RT1015_BAT_RPO_STEP1: + case RT1015_BAT_RPO_STEP2: + case RT1015_BAT_RPO_STEP3: + case RT1015_BAT_RPO_STEP4: + case RT1015_BAT_RPO_STEP5: + case RT1015_BAT_RPO_STEP6: + case RT1015_BAT_RPO_STEP7: + case RT1015_BAT_RPO_STEP8: + case RT1015_BAT_RPO_STEP9: + case RT1015_BAT_RPO_STEP10: + case RT1015_BAT_RPO_STEP11: + case RT1015_BAT_RPO_STEP12: + case RT1015_SPREAD_SPEC1: + case RT1015_SPREAD_SPEC2: + case RT1015_PAD_STATUS: + case RT1015_PADS_PULLING_CTRL1: + case RT1015_PADS_DRIVING: + case RT1015_SYS_RST1: + case RT1015_SYS_RST2: + case RT1015_SYS_GATING1: + case RT1015_TEST_MODE1: + case RT1015_TEST_MODE2: + case RT1015_TIMING_CTRL1: + case RT1015_PLL_INT: + case RT1015_TEST_OUT1: + case RT1015_DC_CALIB_CLSD1: + case RT1015_DC_CALIB_CLSD2: + case RT1015_DC_CALIB_CLSD3: + case RT1015_DC_CALIB_CLSD4: + case RT1015_DC_CALIB_CLSD5: + case RT1015_DC_CALIB_CLSD6: + case RT1015_DC_CALIB_CLSD7: + case RT1015_DC_CALIB_CLSD8: + case RT1015_DC_CALIB_CLSD9: + case RT1015_DC_CALIB_CLSD10: + case RT1015_CLSD_INTERNAL1: + case RT1015_CLSD_INTERNAL2: + case RT1015_CLSD_INTERNAL3: + case RT1015_CLSD_INTERNAL4: + case RT1015_CLSD_INTERNAL5: + case RT1015_CLSD_INTERNAL6: + case RT1015_CLSD_INTERNAL7: + case RT1015_CLSD_INTERNAL8: + case RT1015_CLSD_INTERNAL9: + case RT1015_CLSD_OCP_CTRL: + case RT1015_VREF_LV: + case RT1015_MBIAS1: + case RT1015_MBIAS2: + case RT1015_MBIAS3: + case RT1015_MBIAS4: + case RT1015_VREF_LV1: + case RT1015_S_BST_TIMING_INTER1: + case RT1015_S_BST_TIMING_INTER2: + case RT1015_S_BST_TIMING_INTER3: + case RT1015_S_BST_TIMING_INTER4: + case RT1015_S_BST_TIMING_INTER5: + case RT1015_S_BST_TIMING_INTER6: + case RT1015_S_BST_TIMING_INTER7: + case RT1015_S_BST_TIMING_INTER8: + case RT1015_S_BST_TIMING_INTER9: + case RT1015_S_BST_TIMING_INTER10: + case RT1015_S_BST_TIMING_INTER11: + case RT1015_S_BST_TIMING_INTER12: + case RT1015_S_BST_TIMING_INTER13: + case RT1015_S_BST_TIMING_INTER14: + case RT1015_S_BST_TIMING_INTER15: + case RT1015_S_BST_TIMING_INTER16: + case RT1015_S_BST_TIMING_INTER17: + case RT1015_S_BST_TIMING_INTER18: + case RT1015_S_BST_TIMING_INTER19: + case RT1015_S_BST_TIMING_INTER20: + case RT1015_S_BST_TIMING_INTER21: + case RT1015_S_BST_TIMING_INTER22: + case RT1015_S_BST_TIMING_INTER23: + case RT1015_S_BST_TIMING_INTER24: + case RT1015_S_BST_TIMING_INTER25: + case RT1015_S_BST_TIMING_INTER26: + case RT1015_S_BST_TIMING_INTER27: + case RT1015_S_BST_TIMING_INTER28: + case RT1015_S_BST_TIMING_INTER29: + case RT1015_S_BST_TIMING_INTER30: + case RT1015_S_BST_TIMING_INTER31: + case RT1015_S_BST_TIMING_INTER32: + case RT1015_S_BST_TIMING_INTER33: + case RT1015_S_BST_TIMING_INTER34: + case RT1015_S_BST_TIMING_INTER35: + case RT1015_S_BST_TIMING_INTER36: + return true; + + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -9525, 75, 0); + +static const char * const rt1015_din_source_select[] = { + "Left", + "Right", + "Left + Right average", +}; + +static SOC_ENUM_SINGLE_DECL(rt1015_mono_lr_sel, RT1015_PAD_DRV2, 4, + rt1015_din_source_select); + +static const char * const rt1015_boost_mode[] = { + "Bypass", "Adaptive", "Fixed Adaptive" +}; + +static const SOC_ENUM_SINGLE_DECL(rt1015_boost_mode_enum, 0, 0, + rt1015_boost_mode); + +static int rt1015_boost_mode_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct rt1015_priv *rt1015 = + snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = rt1015->boost_mode; + + return 0; +} + +static int rt1015_boost_mode_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct rt1015_priv *rt1015 = + snd_soc_component_get_drvdata(component); + + rt1015->boost_mode = ucontrol->value.integer.value[0]; + if (rt1015->boost_mode == 0) { + /* Bypass */ + snd_soc_component_update_bits(component, + RT1015_SMART_BST_CTRL1, RT1015_ABST_AUTO_EN_MASK | + RT1015_ABST_FIX_TGT_MASK | RT1015_BYPASS_SWR_REG_MASK, + RT1015_ABST_REG_MODE | RT1015_ABST_FIX_TGT_DIS | + RT1015_BYPASS_SWRREG_BYPASS); + } else if (rt1015->boost_mode == 1) { + /* Adaptive */ + snd_soc_component_update_bits(component, + RT1015_SMART_BST_CTRL1, RT1015_ABST_AUTO_EN_MASK | + RT1015_ABST_FIX_TGT_MASK | RT1015_BYPASS_SWR_REG_MASK, + RT1015_ABST_AUTO_MODE | RT1015_ABST_FIX_TGT_DIS | + RT1015_BYPASS_SWRREG_PASS); + } else { + /* Fixed Adaptive */ + snd_soc_component_update_bits(component, + RT1015_SMART_BST_CTRL1, RT1015_ABST_AUTO_EN_MASK | + RT1015_ABST_FIX_TGT_MASK | RT1015_BYPASS_SWR_REG_MASK, + RT1015_ABST_AUTO_MODE | RT1015_ABST_FIX_TGT_EN | + RT1015_BYPASS_SWRREG_PASS); + } + + return 0; +} + +static int rt5518_bypass_boost_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct rt1015_priv *rt1015 = + snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = rt1015->bypass_boost; + + return 0; +} + +static int rt5518_bypass_boost_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = + snd_soc_kcontrol_component(kcontrol); + struct rt1015_priv *rt1015 = + snd_soc_component_get_drvdata(component); + + rt1015->bypass_boost = ucontrol->value.integer.value[0]; + if (rt1015->bypass_boost == 1) { + snd_soc_component_write(component, + RT1015_PWR4, 0x00b2); + snd_soc_component_write(component, + RT1015_CLSD_INTERNAL8, 0x2008); + snd_soc_component_write(component, + RT1015_CLSD_INTERNAL9, 0x0140); + snd_soc_component_write(component, + RT1015_GAT_BOOST, 0x00fe); + snd_soc_component_write(component, + RT1015_PWR_STATE_CTRL, 0x000d); + msleep(500); + snd_soc_component_write(component, + RT1015_PWR_STATE_CTRL, 0x000e); + } + + return 0; +} + +static const struct snd_kcontrol_new rt1015_snd_controls[] = { + SOC_SINGLE_TLV("DAC Playback Volume", RT1015_DAC1, RT1015_DAC_VOL_SFT, + 127, 0, dac_vol_tlv), + SOC_DOUBLE("DAC Playback Switch", RT1015_DAC3, + RT1015_DA_MUTE_SFT, RT1015_DVOL_MUTE_FLAG_SFT, 1, 1), + SOC_ENUM_EXT("Boost Mode", rt1015_boost_mode_enum, + rt1015_boost_mode_get, rt1015_boost_mode_put), + SOC_ENUM("Mono LR Select", rt1015_mono_lr_sel), + SOC_SINGLE_EXT("Bypass Boost", SND_SOC_NOPM, 0, 1, 0, + rt5518_bypass_boost_get, rt5518_bypass_boost_put), +}; + +static int rt1015_is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(source->dapm); + struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component); + + if (rt1015->sysclk_src == RT1015_SCLK_S_PLL) + return 1; + else + return 0; +} + +static int r1015_dac_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (rt1015->bypass_boost == 0) { + snd_soc_component_write(component, + RT1015_SYS_RST1, 0x05f7); + snd_soc_component_write(component, + RT1015_GAT_BOOST, 0xacfe); + snd_soc_component_write(component, + RT1015_PWR9, 0xaa00); + snd_soc_component_write(component, + RT1015_GAT_BOOST, 0xecfe); + } else { + snd_soc_component_write(component, + RT1015_SYS_RST1, 0x05f7); + snd_soc_component_write(component, + RT1015_PWR_STATE_CTRL, 0x026e); + } + break; + + case SND_SOC_DAPM_POST_PMD: + if (rt1015->bypass_boost == 0) { + snd_soc_component_write(component, + RT1015_PWR9, 0xa800); + snd_soc_component_write(component, + RT1015_SYS_RST1, 0x05f5); + } else { + snd_soc_component_write(component, + RT1015_PWR_STATE_CTRL, 0x0268); + snd_soc_component_write(component, + RT1015_SYS_RST1, 0x05f5); + } + break; + + default: + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget rt1015_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO2", RT1015_PWR1, RT1015_PWR_LDO2_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("INT RC CLK", RT1015_PWR1, RT1015_PWR_INTCLK_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ISENSE", RT1015_PWR1, RT1015_PWR_ISENSE_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("VSENSE", RT1015_PWR1, RT1015_PWR_VSENSE_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL", RT1015_PWR1, RT1015_PWR_PLL_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("BG1 BG2", RT1015_PWR1, RT1015_PWR_BG_1_2_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("MBIAS BG", RT1015_PWR1, RT1015_PWR_MBIAS_BG_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("VBAT", RT1015_PWR1, RT1015_PWR_VBAT_BIT, 0, NULL, + 0), + SND_SOC_DAPM_SUPPLY("MBIAS", RT1015_PWR1, RT1015_PWR_MBIAS_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("ADCV", RT1015_PWR1, RT1015_PWR_ADCV_BIT, 0, NULL, + 0), + SND_SOC_DAPM_SUPPLY("MIXERV", RT1015_PWR1, RT1015_PWR_MIXERV_BIT, 0, + NULL, 0), + SND_SOC_DAPM_SUPPLY("SUMV", RT1015_PWR1, RT1015_PWR_SUMV_BIT, 0, NULL, + 0), + SND_SOC_DAPM_SUPPLY("VREFLV", RT1015_PWR1, RT1015_PWR_VREFLV_BIT, 0, + NULL, 0), + + SND_SOC_DAPM_AIF_IN("AIFRX", "AIF Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC_E("DAC", NULL, RT1015_PWR1, RT1015_PWR_DAC_BIT, 0, + r1015_dac_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_OUTPUT("SPO"), +}; + +static const struct snd_soc_dapm_route rt1015_dapm_routes[] = { + { "DAC", NULL, "AIFRX" }, + { "DAC", NULL, "LDO2" }, + { "DAC", NULL, "PLL", rt1015_is_sys_clk_from_pll}, + { "DAC", NULL, "INT RC CLK" }, + { "DAC", NULL, "ISENSE" }, + { "DAC", NULL, "VSENSE" }, + { "DAC", NULL, "BG1 BG2" }, + { "DAC", NULL, "MBIAS BG" }, + { "DAC", NULL, "VBAT" }, + { "DAC", NULL, "MBIAS" }, + { "DAC", NULL, "ADCV" }, + { "DAC", NULL, "MIXERV" }, + { "DAC", NULL, "SUMV" }, + { "DAC", NULL, "VREFLV" }, + { "SPO", NULL, "DAC" }, +}; + +static int rt1015_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component); + int pre_div, bclk_ms, frame_size; + unsigned int val_len = 0; + + rt1015->lrck = params_rate(params); + pre_div = rl6231_get_clk_info(rt1015->sysclk, rt1015->lrck); + if (pre_div < 0) { + dev_err(component->dev, "Unsupported clock rate\n"); + return -EINVAL; + } + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(component->dev, "Unsupported frame size: %d\n", + frame_size); + return -EINVAL; + } + + bclk_ms = frame_size > 32; + rt1015->bclk = rt1015->lrck * (32 << bclk_ms); + + dev_dbg(component->dev, "bclk_ms is %d and pre_div is %d for iis %d\n", + bclk_ms, pre_div, dai->id); + + dev_dbg(component->dev, "lrck is %dHz and pre_div is %d for iis %d\n", + rt1015->lrck, pre_div, dai->id); + + switch (params_width(params)) { + case 16: + break; + case 20: + val_len = RT1015_I2S_DL_20; + break; + case 24: + val_len = RT1015_I2S_DL_24; + break; + case 8: + val_len = RT1015_I2S_DL_8; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT1015_TDM_MASTER, + RT1015_I2S_DL_MASK, val_len); + snd_soc_component_update_bits(component, RT1015_CLK2, + RT1015_FS_PD_MASK, pre_div); + + return 0; +} + +static int rt1015_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + unsigned int reg_val = 0, reg_val2 = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + reg_val |= RT1015_TCON_TDM_MS_M; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg_val |= RT1015_TCON_TDM_MS_S; + 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_val2 |= RT1015_TDM_INV_BCLK; + 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 |= RT1015_I2S_M_DF_LEFT; + break; + + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT1015_I2S_M_DF_PCM_A; + break; + + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT1015_I2S_M_DF_PCM_B; + break; + + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT1015_TDM_MASTER, + RT1015_TCON_TDM_MS_MASK | RT1015_I2S_M_DF_MASK, + reg_val); + snd_soc_component_update_bits(component, RT1015_TDM1_1, + RT1015_TDM_INV_BCLK_MASK, reg_val2); + + return 0; +} + +static int rt1015_set_component_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir) +{ + struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0; + + if (freq == rt1015->sysclk && clk_id == rt1015->sysclk_src) + return 0; + + switch (clk_id) { + case RT1015_SCLK_S_MCLK: + reg_val |= RT1015_CLK_SYS_PRE_SEL_MCLK; + break; + + case RT1015_SCLK_S_PLL: + reg_val |= RT1015_CLK_SYS_PRE_SEL_PLL; + break; + + default: + dev_err(component->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + rt1015->sysclk = freq; + rt1015->sysclk_src = clk_id; + + dev_dbg(component->dev, "Sysclk is %dHz and clock id is %d\n", + freq, clk_id); + + snd_soc_component_update_bits(component, RT1015_CLK2, + RT1015_CLK_SYS_PRE_SEL_MASK, reg_val); + + return 0; +} + +static int rt1015_set_component_pll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component); + struct rl6231_pll_code pll_code; + int ret; + + if (!freq_in || !freq_out) { + dev_dbg(component->dev, "PLL disabled\n"); + + rt1015->pll_in = 0; + rt1015->pll_out = 0; + + return 0; + } + + if (source == rt1015->pll_src && freq_in == rt1015->pll_in && + freq_out == rt1015->pll_out) + return 0; + + switch (source) { + case RT1015_PLL_S_MCLK: + snd_soc_component_update_bits(component, RT1015_CLK2, + RT1015_PLL_SEL_MASK, RT1015_PLL_SEL_PLL_SRC2); + break; + + case RT1015_PLL_S_BCLK: + snd_soc_component_update_bits(component, RT1015_CLK2, + RT1015_PLL_SEL_MASK, RT1015_PLL_SEL_BCLK); + break; + + default: + dev_err(component->dev, "Unknown PLL Source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(component->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_component_write(component, RT1015_PLL1, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT1015_PLL_M_SFT | + pll_code.m_bp << RT1015_PLL_M_BP_SFT | pll_code.n_code); + snd_soc_component_write(component, RT1015_PLL2, + pll_code.k_code); + + rt1015->pll_in = freq_in; + rt1015->pll_out = freq_out; + rt1015->pll_src = source; + + return 0; +} + +static int rt1015_probe(struct snd_soc_component *component) +{ + struct rt1015_priv *rt1015 = + snd_soc_component_get_drvdata(component); + + rt1015->component = component; + + switch (snd_soc_component_read32(component, RT1015_VER_ID) + & RT1015_ID_MASK) { + case RT1015_ID_VERA: + rt1015_reg_init(component); + break; + case RT1015_ID_VERB: + snd_soc_component_write(component, RT1015_BAT_RPO_STEP1, + 0x061c); + break; + default: + dev_err(component->dev, "Unknown version id!\n"); + } + + return 0; +} + +static void rt1015_remove(struct snd_soc_component *component) +{ + struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component); + + regmap_write(rt1015->regmap, RT1015_RESET, 0); +} + +#define RT1015_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT1015_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 rt1015_aif_dai_ops = { + .hw_params = rt1015_hw_params, + .set_fmt = rt1015_set_dai_fmt, +}; + +struct snd_soc_dai_driver rt1015_dai[] = { + { + .name = "rt1015-aif", + .id = 0, + .playback = { + .stream_name = "AIF Playback", + .channels_min = 1, + .channels_max = 4, + .rates = RT1015_STEREO_RATES, + .formats = RT1015_FORMATS, + }, + } +}; + +#ifdef CONFIG_PM +static int rt1015_suspend(struct snd_soc_component *component) +{ + struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt1015->regmap, true); + regcache_mark_dirty(rt1015->regmap); + + return 0; +} + +static int rt1015_resume(struct snd_soc_component *component) +{ + struct rt1015_priv *rt1015 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt1015->regmap, false); + regcache_sync(rt1015->regmap); + return 0; +} +#else +#define rt1015_suspend NULL +#define rt1015_resume NULL +#endif + +static const struct snd_soc_component_driver soc_component_dev_rt1015 = { + .probe = rt1015_probe, + .remove = rt1015_remove, + .suspend = rt1015_suspend, + .resume = rt1015_resume, + .controls = rt1015_snd_controls, + .num_controls = ARRAY_SIZE(rt1015_snd_controls), + .dapm_widgets = rt1015_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt1015_dapm_widgets), + .dapm_routes = rt1015_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt1015_dapm_routes), + .set_sysclk = rt1015_set_component_sysclk, + .set_pll = rt1015_set_component_pll, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config rt1015_regmap = { + .reg_bits = 16, + .val_bits = 16, + .max_register = RT1015_S_BST_TIMING_INTER36, + .volatile_reg = rt1015_volatile_register, + .readable_reg = rt1015_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt1015_reg, + .num_reg_defaults = ARRAY_SIZE(rt1015_reg), +}; + +static const struct i2c_device_id rt1015_i2c_id[] = { + { "rt1015", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, rt1015_i2c_id); + +#if defined(CONFIG_OF) +static const struct of_device_id rt1015_of_match[] = { + { .compatible = "realtek,rt1015", }, + {}, +}; +MODULE_DEVICE_TABLE(of, rt1015_of_match); +#endif + +#ifdef CONFIG_ACPI +static struct acpi_device_id rt1015_acpi_match[] = { + {"10EC1015", 0,}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt1015_acpi_match); +#endif + +static int rt1015_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt1015_priv *rt1015; + int ret; + unsigned int val; + + rt1015 = devm_kzalloc(&i2c->dev, sizeof(struct rt1015_priv), + GFP_KERNEL); + if (rt1015 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt1015); + + rt1015->regmap = devm_regmap_init_i2c(i2c, &rt1015_regmap); + if (IS_ERR(rt1015->regmap)) { + ret = PTR_ERR(rt1015->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + regmap_read(rt1015->regmap, RT1015_DEVICE_ID, &val); + if ((val != RT1015_DEVICE_ID_VAL) && (val != RT1015_DEVICE_ID_VAL2)) { + dev_err(&i2c->dev, + "Device with ID register %x is not rt1015\n", val); + return -ENODEV; + } + + return devm_snd_soc_register_component(&i2c->dev, + &soc_component_dev_rt1015, + rt1015_dai, ARRAY_SIZE(rt1015_dai)); +} + +static void rt1015_i2c_shutdown(struct i2c_client *client) +{ + struct rt1015_priv *rt1015 = i2c_get_clientdata(client); + + regmap_write(rt1015->regmap, RT1015_RESET, 0); +} + +static struct i2c_driver rt1015_i2c_driver = { + .driver = { + .name = "rt1015", + .of_match_table = of_match_ptr(rt1015_of_match), + .acpi_match_table = ACPI_PTR(rt1015_acpi_match), + }, + .probe = rt1015_i2c_probe, + .shutdown = rt1015_i2c_shutdown, + .id_table = rt1015_i2c_id, +}; +module_i2c_driver(rt1015_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT1015 driver"); +MODULE_AUTHOR("Jack Yu <jack.yu(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt1015.h b/sound/soc/codecs/rt1015.h new file mode 100644 index 000000000000..8d75cadaa0d5 --- /dev/null +++ b/sound/soc/codecs/rt1015.h @@ -0,0 +1,368 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * rt1015.h -- RT1015 ALSA SoC audio amplifier driver + * + * Copyright 2019 Realtek Semiconductor Corp. + * Author: Jack Yu <jack.yu(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 __RT1015_H__ +#define __RT1015_H__ + +#define RT1015_DEVICE_ID_VAL 0x1011 +#define RT1015_DEVICE_ID_VAL2 0x1015 + +#define RT1015_RESET 0x0000 +#define RT1015_CLK2 0x0004 +#define RT1015_CLK3 0x0006 +#define RT1015_PLL1 0x000a +#define RT1015_PLL2 0x000c +#define RT1015_CLK_DET 0x0020 +#define RT1015_SIL_DET 0x0022 +#define RT1015_CUSTOMER_ID 0x0076 +#define RT1015_PCODE_FWVER 0x0078 +#define RT1015_VER_ID 0x007a +#define RT1015_VENDOR_ID 0x007c +#define RT1015_DEVICE_ID 0x007d +#define RT1015_PAD_DRV1 0x00f0 +#define RT1015_PAD_DRV2 0x00f2 +#define RT1015_GAT_BOOST 0x00f3 +#define RT1015_PRO_ALT 0x00f4 +#define RT1015_MAN_I2C 0x0100 +#define RT1015_DAC1 0x0102 +#define RT1015_DAC2 0x0104 +#define RT1015_DAC3 0x0106 +#define RT1015_ADC1 0x010c +#define RT1015_ADC2 0x010e +#define RT1015_TDM_MASTER 0x0111 +#define RT1015_TDM_TCON 0x0112 +#define RT1015_TDM1_1 0x0114 +#define RT1015_TDM1_2 0x0116 +#define RT1015_TDM1_3 0x0118 +#define RT1015_TDM1_4 0x011a +#define RT1015_TDM1_5 0x011c +#define RT1015_MIXER1 0x0300 +#define RT1015_MIXER2 0x0302 +#define RT1015_ANA_PROTECT1 0x0311 +#define RT1015_ANA_CTRL_SEQ1 0x0313 +#define RT1015_ANA_CTRL_SEQ2 0x0314 +#define RT1015_VBAT_DET_DEB 0x031a +#define RT1015_VBAT_VOLT_DET1 0x031c +#define RT1015_VBAT_VOLT_DET2 0x031d +#define RT1015_VBAT_TEST_OUT1 0x031e +#define RT1015_VBAT_TEST_OUT2 0x031f +#define RT1015_VBAT_PROT_ATT 0x0320 +#define RT1015_VBAT_DET_CODE 0x0321 +#define RT1015_PWR1 0x0322 +#define RT1015_PWR4 0x0328 +#define RT1015_PWR5 0x0329 +#define RT1015_PWR6 0x032a +#define RT1015_PWR7 0x032b +#define RT1015_PWR8 0x032c +#define RT1015_PWR9 0x032d +#define RT1015_CLASSD_SEQ 0x032e +#define RT1015_SMART_BST_CTRL1 0x0330 +#define RT1015_SMART_BST_CTRL2 0x0332 +#define RT1015_ANA_CTRL1 0x0334 +#define RT1015_ANA_CTRL2 0x0336 +#define RT1015_PWR_STATE_CTRL 0x0338 +#define RT1015_SPK_VOL 0x0506 +#define RT1015_SHORT_DETTOP1 0x0508 +#define RT1015_SHORT_DETTOP2 0x050a +#define RT1015_SPK_DC_DETECT1 0x0519 +#define RT1015_SPK_DC_DETECT2 0x051a +#define RT1015_SPK_DC_DETECT3 0x051b +#define RT1015_SPK_DC_DETECT4 0x051d +#define RT1015_SPK_DC_DETECT5 0x051f +#define RT1015_BAT_RPO_STEP1 0x0536 +#define RT1015_BAT_RPO_STEP2 0x0538 +#define RT1015_BAT_RPO_STEP3 0x053a +#define RT1015_BAT_RPO_STEP4 0x053c +#define RT1015_BAT_RPO_STEP5 0x053d +#define RT1015_BAT_RPO_STEP6 0x053e +#define RT1015_BAT_RPO_STEP7 0x053f +#define RT1015_BAT_RPO_STEP8 0x0540 +#define RT1015_BAT_RPO_STEP9 0x0541 +#define RT1015_BAT_RPO_STEP10 0x0542 +#define RT1015_BAT_RPO_STEP11 0x0543 +#define RT1015_BAT_RPO_STEP12 0x0544 +#define RT1015_SPREAD_SPEC1 0x0568 +#define RT1015_SPREAD_SPEC2 0x056a +#define RT1015_PAD_STATUS 0x1000 +#define RT1015_PADS_PULLING_CTRL1 0x1002 +#define RT1015_PADS_DRIVING 0x1006 +#define RT1015_SYS_RST1 0x1007 +#define RT1015_SYS_RST2 0x1009 +#define RT1015_SYS_GATING1 0x100a +#define RT1015_TEST_MODE1 0x100c +#define RT1015_TEST_MODE2 0x100d +#define RT1015_TIMING_CTRL1 0x100e +#define RT1015_PLL_INT 0x1010 +#define RT1015_TEST_OUT1 0x1020 +#define RT1015_DC_CALIB_CLSD1 0x1200 +#define RT1015_DC_CALIB_CLSD2 0x1202 +#define RT1015_DC_CALIB_CLSD3 0x1204 +#define RT1015_DC_CALIB_CLSD4 0x1206 +#define RT1015_DC_CALIB_CLSD5 0x1208 +#define RT1015_DC_CALIB_CLSD6 0x120a +#define RT1015_DC_CALIB_CLSD7 0x120c +#define RT1015_DC_CALIB_CLSD8 0x120e +#define RT1015_DC_CALIB_CLSD9 0x1210 +#define RT1015_DC_CALIB_CLSD10 0x1212 +#define RT1015_CLSD_INTERNAL1 0x1300 +#define RT1015_CLSD_INTERNAL2 0x1302 +#define RT1015_CLSD_INTERNAL3 0x1304 +#define RT1015_CLSD_INTERNAL4 0x1305 +#define RT1015_CLSD_INTERNAL5 0x1306 +#define RT1015_CLSD_INTERNAL6 0x1308 +#define RT1015_CLSD_INTERNAL7 0x130a +#define RT1015_CLSD_INTERNAL8 0x130c +#define RT1015_CLSD_INTERNAL9 0x130e +#define RT1015_CLSD_OCP_CTRL 0x130f +#define RT1015_VREF_LV 0x1310 +#define RT1015_MBIAS1 0x1312 +#define RT1015_MBIAS2 0x1314 +#define RT1015_MBIAS3 0x1316 +#define RT1015_MBIAS4 0x1318 +#define RT1015_VREF_LV1 0x131a +#define RT1015_S_BST_TIMING_INTER1 0x1322 +#define RT1015_S_BST_TIMING_INTER2 0x1323 +#define RT1015_S_BST_TIMING_INTER3 0x1324 +#define RT1015_S_BST_TIMING_INTER4 0x1325 +#define RT1015_S_BST_TIMING_INTER5 0x1326 +#define RT1015_S_BST_TIMING_INTER6 0x1327 +#define RT1015_S_BST_TIMING_INTER7 0x1328 +#define RT1015_S_BST_TIMING_INTER8 0x1329 +#define RT1015_S_BST_TIMING_INTER9 0x132a +#define RT1015_S_BST_TIMING_INTER10 0x132b +#define RT1015_S_BST_TIMING_INTER11 0x1330 +#define RT1015_S_BST_TIMING_INTER12 0x1331 +#define RT1015_S_BST_TIMING_INTER13 0x1332 +#define RT1015_S_BST_TIMING_INTER14 0x1333 +#define RT1015_S_BST_TIMING_INTER15 0x1334 +#define RT1015_S_BST_TIMING_INTER16 0x1335 +#define RT1015_S_BST_TIMING_INTER17 0x1336 +#define RT1015_S_BST_TIMING_INTER18 0x1337 +#define RT1015_S_BST_TIMING_INTER19 0x1338 +#define RT1015_S_BST_TIMING_INTER20 0x1339 +#define RT1015_S_BST_TIMING_INTER21 0x133a +#define RT1015_S_BST_TIMING_INTER22 0x133b +#define RT1015_S_BST_TIMING_INTER23 0x133c +#define RT1015_S_BST_TIMING_INTER24 0x133d +#define RT1015_S_BST_TIMING_INTER25 0x133e +#define RT1015_S_BST_TIMING_INTER26 0x133f +#define RT1015_S_BST_TIMING_INTER27 0x1340 +#define RT1015_S_BST_TIMING_INTER28 0x1341 +#define RT1015_S_BST_TIMING_INTER29 0x1342 +#define RT1015_S_BST_TIMING_INTER30 0x1343 +#define RT1015_S_BST_TIMING_INTER31 0x1344 +#define RT1015_S_BST_TIMING_INTER32 0x1345 +#define RT1015_S_BST_TIMING_INTER33 0x1346 +#define RT1015_S_BST_TIMING_INTER34 0x1347 +#define RT1015_S_BST_TIMING_INTER35 0x1348 +#define RT1015_S_BST_TIMING_INTER36 0x1349 + +/* 0x0004 */ +#define RT1015_CLK_SYS_PRE_SEL_MASK (0x3 << 14) +#define RT1015_CLK_SYS_PRE_SEL_SFT 14 +#define RT1015_CLK_SYS_PRE_SEL_MCLK (0x0 << 14) +#define RT1015_CLK_SYS_PRE_SEL_PLL (0x2 << 14) +#define RT1015_PLL_SEL_MASK (0x1 << 13) +#define RT1015_PLL_SEL_SFT 13 +#define RT1015_PLL_SEL_PLL_SRC2 (0x0 << 13) +#define RT1015_PLL_SEL_BCLK (0x1 << 13) +#define RT1015_FS_PD_MASK (0x7 << 4) +#define RT1015_FS_PD_SFT 4 + +/* 0x000a */ +#define RT1015_PLL_M_MAX 0xf +#define RT1015_PLL_M_MASK (RT1015_PLL_M_MAX << 12) +#define RT1015_PLL_M_SFT 12 +#define RT1015_PLL_M_BP (0x1 << 11) +#define RT1015_PLL_M_BP_SFT 11 +#define RT1015_PLL_N_MAX 0x1ff +#define RT1015_PLL_N_MASK (RT1015_PLL_N_MAX << 0) +#define RT1015_PLL_N_SFT 0 + +/* 0x000c */ +#define RT1015_PLL_BPK_MASK (0x1 << 5) +#define RT1015_PLL_BPK (0x0 << 5) +#define RT1015_PLL_K_MAX 0x1f +#define RT1015_PLL_K_MASK (RT1015_PLL_K_MAX) +#define RT1015_PLL_K_SFT 0 + +/* 0x007a */ +#define RT1015_ID_MASK 0xff +#define RT1015_ID_VERA 0x0 +#define RT1015_ID_VERB 0x1 + +/* 0x0102 */ +#define RT1015_DAC_VOL_MASK (0x7f << 9) +#define RT1015_DAC_VOL_SFT 9 + +/* 0x0104 */ +#define RT1015_DAC_CLK (0x1 << 13) +#define RT1015_DAC_CLK_BIT 13 + +/* 0x0106 */ +#define RT1015_DAC_MUTE_MASK (0x1 << 15) +#define RT1015_DA_MUTE_SFT 15 +#define RT1015_DVOL_MUTE_FLAG_SFT 12 + +/* 0x0111 */ +#define RT1015_TCON_TDM_MS_MASK (0x1 << 14) +#define RT1015_TCON_TDM_MS_SFT 14 +#define RT1015_TCON_TDM_MS_S (0x0 << 14) +#define RT1015_TCON_TDM_MS_M (0x1 << 14) +#define RT1015_I2S_DL_MASK (0x7 << 8) +#define RT1015_I2S_DL_SFT 8 +#define RT1015_I2S_DL_16 (0x0 << 8) +#define RT1015_I2S_DL_20 (0x1 << 8) +#define RT1015_I2S_DL_24 (0x2 << 8) +#define RT1015_I2S_DL_8 (0x3 << 8) +#define RT1015_I2S_M_DF_MASK (0x7 << 0) +#define RT1015_I2S_M_DF_SFT 0 +#define RT1015_I2S_M_DF_I2S (0x0) +#define RT1015_I2S_M_DF_LEFT (0x1) +#define RT1015_I2S_M_DF_PCM_A (0x2) +#define RT1015_I2S_M_DF_PCM_B (0x3) +#define RT1015_I2S_M_DF_PCM_A_N (0x6) +#define RT1015_I2S_M_DF_PCM_B_N (0x7) + +/* TDM_tcon Setting (0x0112) */ +#define RT1015_I2S_TCON_DF_MASK (0x7 << 13) +#define RT1015_I2S_TCON_DF_SFT 13 +#define RT1015_I2S_TCON_DF_I2S (0x0 << 13) +#define RT1015_I2S_TCON_DF_LEFT (0x1 << 13) +#define RT1015_I2S_TCON_DF_PCM_A (0x2 << 13) +#define RT1015_I2S_TCON_DF_PCM_B (0x3 << 13) +#define RT1015_I2S_TCON_DF_PCM_A_N (0x6 << 13) +#define RT1015_I2S_TCON_DF_PCM_B_N (0x7 << 13) +#define RT1015_TCON_BCLK_SEL_MASK (0x3 << 10) +#define RT1015_TCON_BCLK_SEL_SFT 10 +#define RT1015_TCON_BCLK_SEL_32FS (0x0 << 10) +#define RT1015_TCON_BCLK_SEL_64FS (0x1 << 10) +#define RT1015_TCON_BCLK_SEL_128FS (0x2 << 10) +#define RT1015_TCON_BCLK_SEL_256FS (0x3 << 10) +#define RT1015_TCON_CH_LEN_MASK (0x3 << 5) +#define RT1015_TCON_CH_LEN_SFT 5 +#define RT1015_TCON_CH_LEN_16B (0x0 << 5) +#define RT1015_TCON_CH_LEN_20B (0x1 << 5) +#define RT1015_TCON_CH_LEN_24B (0x2 << 5) +#define RT1015_TCON_CH_LEN_32B (0x3 << 5) +#define RT1015_TCON_BCLK_MST_MASK (0x1 << 4) +#define RT1015_TCON_BCLK_MST_SFT 4 +#define RT1015_TCON_BCLK_MST_INV (0x1 << 4) + +/* TDM1 Setting-1 (0x0114) */ +#define RT1015_TDM_INV_BCLK_MASK (0x1 << 15) +#define RT1015_TDM_INV_BCLK_SFT 15 +#define RT1015_TDM_INV_BCLK (0x1 << 15) + +/* 0x0330 */ +#define RT1015_ABST_AUTO_EN_MASK (0x1 << 13) +#define RT1015_ABST_AUTO_MODE (0x1 << 13) +#define RT1015_ABST_REG_MODE (0x0 << 13) +#define RT1015_ABST_FIX_TGT_MASK (0x1 << 12) +#define RT1015_ABST_FIX_TGT_EN (0x1 << 12) +#define RT1015_ABST_FIX_TGT_DIS (0x0 << 12) +#define RT1015_BYPASS_SWR_REG_MASK (0x1 << 7) +#define RT1015_BYPASS_SWRREG_BYPASS (0x1 << 7) +#define RT1015_BYPASS_SWRREG_PASS (0x0 << 7) + +/* 0x0322 */ +#define RT1015_PWR_LDO2 (0x1 << 15) +#define RT1015_PWR_LDO2_BIT 15 +#define RT1015_PWR_DAC (0x1 << 14) +#define RT1015_PWR_DAC_BIT 14 +#define RT1015_PWR_INTCLK (0x1 << 13) +#define RT1015_PWR_INTCLK_BIT 13 +#define RT1015_PWR_ISENSE (0x1 << 12) +#define RT1015_PWR_ISENSE_BIT 12 +#define RT1015_PWR_VSENSE (0x1 << 10) +#define RT1015_PWR_VSENSE_BIT 10 +#define RT1015_PWR_PLL (0x1 << 9) +#define RT1015_PWR_PLL_BIT 9 +#define RT1015_PWR_BG_1_2 (0x1 << 8) +#define RT1015_PWR_BG_1_2_BIT 8 +#define RT1015_PWR_MBIAS_BG (0x1 << 7) +#define RT1015_PWR_MBIAS_BG_BIT 7 +#define RT1015_PWR_VBAT (0x1 << 6) +#define RT1015_PWR_VBAT_BIT 6 +#define RT1015_PWR_MBIAS (0x1 << 4) +#define RT1015_PWR_MBIAS_BIT 4 +#define RT1015_PWR_ADCV (0x1 << 3) +#define RT1015_PWR_ADCV_BIT 3 +#define RT1015_PWR_MIXERV (0x1 << 2) +#define RT1015_PWR_MIXERV_BIT 2 +#define RT1015_PWR_SUMV (0x1 << 1) +#define RT1015_PWR_SUMV_BIT 1 +#define RT1015_PWR_VREFLV (0x1 << 0) +#define RT1015_PWR_VREFLV_BIT 0 + +/* 0x0324 */ +#define RT1015_PWR_BASIC (0x1 << 15) +#define RT1015_PWR_BASIC_BIT 15 +#define RT1015_PWR_SD (0x1 << 14) +#define RT1015_PWR_SD_BIT 14 +#define RT1015_PWR_IBIAS (0x1 << 13) +#define RT1015_PWR_IBIAS_BIT 13 +#define RT1015_PWR_VCM (0x1 << 11) +#define RT1015_PWR_VCM_BIT 11 + +/* 0x0328 */ +#define RT1015_PWR_SWR (0x1 << 12) +#define RT1015_PWR_SWR_BIT 12 + +/* 0x1300 */ +#define RT1015_PWR_CLSD (0x1 << 12) +#define RT1015_PWR_CLSD_BIT 12 + +/* 0x007a */ +#define RT1015_ID_MASK 0xff +#define RT1015_ID_VERA 0x0 +#define RT1015_ID_VERB 0x1 + +/* System Clock Source */ +enum { + RT1015_SCLK_S_MCLK, + RT1015_SCLK_S_PLL, +}; + +/* PLL1 Source */ +enum { + RT1015_PLL_S_MCLK, + RT1015_PLL_S_BCLK, +}; + +enum { + RT1015_AIF1, + RT1015_AIFS, +}; + +enum { + RT1015_VERA, + RT1015_VERB, +}; + +struct rt1015_priv { + struct snd_soc_component *component; + struct regmap *regmap; + int sysclk; + int sysclk_src; + int lrck; + int bclk; + int id; + int pll_src; + int pll_in; + int pll_out; + int boost_mode; + int bypass_boost; + int amp_ver; +}; + +#endif /* __RT1015_H__ */ -- 2.24.1

3 2

Re: [PATCH v3 0/8] PCI: add and use constant PCI_STATUS_ERROR_BITS and helper pci_status_get_and_clear_errors
by Bjorn Helgaas 25 Feb '20

25 Feb '20

On Tue, Feb 25, 2020 at 03:03:05PM +0100, Heiner Kallweit wrote: > Few drivers have own definitions for this constant, so move it to the > PCI core. In addition there are several places where the following > code sequence is used: > 1. Read PCI_STATUS > 2. Mask out non-error bits > 3. Action based on set error bits > 4. Write back set error bits to clear them > > As this is a repeated pattern, add a helper to the PCI core. > > Most affected drivers are network drivers. But as it's about core > PCI functionality, I suppose the series should go through the PCI > tree. Makes good sense to me, thanks for doing this. > v2: > - fix formal issue with cover letter > v3: > - fix dumb typo in patch 7 > > Heiner Kallweit (8): > PCI: add constant PCI_STATUS_ERROR_BITS > PCI: add pci_status_get_and_clear_errors > r8169: use pci_status_get_and_clear_errors > net: cassini: use pci_status_get_and_clear_errors > net: sungem: use pci_status_get_and_clear_errors > net: skfp: use PCI_STATUS_ERROR_BITS > PCI: pci-bridge-emul: use PCI_STATUS_ERROR_BITS > sound: bt87x: use pci_status_get_and_clear_errors > > drivers/net/ethernet/marvell/skge.h | 6 ----- > drivers/net/ethernet/marvell/sky2.h | 6 ----- > drivers/net/ethernet/realtek/r8169_main.c | 15 +++++------- > drivers/net/ethernet/sun/cassini.c | 28 ++++++++------------- > drivers/net/ethernet/sun/sungem.c | 30 +++++++---------------- > drivers/net/fddi/skfp/drvfbi.c | 2 +- > drivers/net/fddi/skfp/h/skfbi.h | 5 ---- > drivers/pci/pci-bridge-emul.c | 14 ++--------- > drivers/pci/pci.c | 23 +++++++++++++++++ > include/linux/pci.h | 1 + > include/uapi/linux/pci_regs.h | 7 ++++++ > sound/pci/bt87x.c | 7 +----- > 12 files changed, 60 insertions(+), 84 deletions(-) > > -- > 2.25.1 > > >

1 0

Re: [PATCH v3 2/8] PCI: add pci_status_get_and_clear_errors
by Bjorn Helgaas 25 Feb '20

25 Feb '20

On Tue, Feb 25, 2020 at 03:04:23PM +0100, Heiner Kallweit wrote: > Few drivers use the following code sequence: This reads like "Not very many drivers". I think "Several drivers" would capture the sense of this better. > 1. Read PCI_STATUS > 2. Mask out non-error bits > 3. Action based on error bits set > 4. Write back set error bits to clear them > > As this is a repeated pattern, add a helper to the PCI core. > > Signed-off-by: Heiner Kallweit <hkallweit1(a)gmail.com> > --- > drivers/pci/pci.c | 23 +++++++++++++++++++++++ > include/linux/pci.h | 1 + > 2 files changed, 24 insertions(+) > > diff --git a/drivers/pci/pci.c b/drivers/pci/pci.c > index d828ca835..c16b0ba2a 100644 > --- a/drivers/pci/pci.c > +++ b/drivers/pci/pci.c > @@ -173,6 +173,29 @@ unsigned char pci_bus_max_busnr(struct pci_bus *bus) > } > EXPORT_SYMBOL_GPL(pci_bus_max_busnr); > > +/** > + * pci_status_get_and_clear_errors - return and clear error bits in PCI_STATUS > + * @pdev: the PCI device > + * > + * Returns error bits set in PCI_STATUS and clears them. > + */ > +int pci_status_get_and_clear_errors(struct pci_dev *pdev) > +{ > + u16 status; > + int ret; > + > + ret = pci_read_config_word(pdev, PCI_STATUS, &status); > + if (ret != PCIBIOS_SUCCESSFUL) > + return -EIO; > + > + status &= PCI_STATUS_ERROR_BITS; > + if (status) > + pci_write_config_word(pdev, PCI_STATUS, status); > + > + return status; > +} > +EXPORT_SYMBOL_GPL(pci_status_get_and_clear_errors); > + > #ifdef CONFIG_HAS_IOMEM > void __iomem *pci_ioremap_bar(struct pci_dev *pdev, int bar) > { > diff --git a/include/linux/pci.h b/include/linux/pci.h > index 3840a541a..7a75aae04 100644 > --- a/include/linux/pci.h > +++ b/include/linux/pci.h > @@ -1203,6 +1203,7 @@ int pci_select_bars(struct pci_dev *dev, unsigned long flags); > bool pci_device_is_present(struct pci_dev *pdev); > void pci_ignore_hotplug(struct pci_dev *dev); > struct pci_dev *pci_real_dma_dev(struct pci_dev *dev); > +int pci_status_get_and_clear_errors(struct pci_dev *pdev); > > int __printf(6, 7) pci_request_irq(struct pci_dev *dev, unsigned int nr, > irq_handler_t handler, irq_handler_t thread_fn, void *dev_id, > -- > 2.25.1 > > > >

1 0

Re: [PATCH v3 1/8] PCI: add constant PCI_STATUS_ERROR_BITS
by Bjorn Helgaas 25 Feb '20

25 Feb '20

On Tue, Feb 25, 2020 at 03:03:44PM +0100, Heiner Kallweit wrote: Run "git log --oneline drivers/pci" and make yours match. In particular, capitalize the first word ("Add"). Same for the other PCI patches. I don't know the drivers/net convention, but please find and follow that as well. > This constant is used (with different names) in more than one driver, > so move it to the PCI core. The driver constants in *this* patch at least use the same name. > Signed-off-by: Heiner Kallweit <hkallweit1(a)gmail.com> > --- > drivers/net/ethernet/marvell/skge.h | 6 ------ > drivers/net/ethernet/marvell/sky2.h | 6 ------ > include/uapi/linux/pci_regs.h | 7 +++++++ > 3 files changed, 7 insertions(+), 12 deletions(-) > > diff --git a/drivers/net/ethernet/marvell/skge.h b/drivers/net/ethernet/marvell/skge.h > index 6fa7b6a34..e149bdfe1 100644 > --- a/drivers/net/ethernet/marvell/skge.h > +++ b/drivers/net/ethernet/marvell/skge.h > @@ -15,12 +15,6 @@ > #define PCI_VPD_ROM_SZ 7L<<14 /* VPD ROM size 0=256, 1=512, ... */ > #define PCI_REV_DESC 1<<2 /* Reverse Descriptor bytes */ > > -#define PCI_STATUS_ERROR_BITS (PCI_STATUS_DETECTED_PARITY | \ > - PCI_STATUS_SIG_SYSTEM_ERROR | \ > - PCI_STATUS_REC_MASTER_ABORT | \ > - PCI_STATUS_REC_TARGET_ABORT | \ > - PCI_STATUS_PARITY) > - > enum csr_regs { > B0_RAP = 0x0000, > B0_CTST = 0x0004, > diff --git a/drivers/net/ethernet/marvell/sky2.h b/drivers/net/ethernet/marvell/sky2.h > index b02b65230..851d8ed34 100644 > --- a/drivers/net/ethernet/marvell/sky2.h > +++ b/drivers/net/ethernet/marvell/sky2.h > @@ -252,12 +252,6 @@ enum { > }; > > > -#define PCI_STATUS_ERROR_BITS (PCI_STATUS_DETECTED_PARITY | \ > - PCI_STATUS_SIG_SYSTEM_ERROR | \ > - PCI_STATUS_REC_MASTER_ABORT | \ > - PCI_STATUS_REC_TARGET_ABORT | \ > - PCI_STATUS_PARITY) > - > enum csr_regs { > B0_RAP = 0x0000, > B0_CTST = 0x0004, > diff --git a/include/uapi/linux/pci_regs.h b/include/uapi/linux/pci_regs.h > index 543769048..9b84a1278 100644 > --- a/include/uapi/linux/pci_regs.h > +++ b/include/uapi/linux/pci_regs.h > @@ -68,6 +68,13 @@ > #define PCI_STATUS_SIG_SYSTEM_ERROR 0x4000 /* Set when we drive SERR */ > #define PCI_STATUS_DETECTED_PARITY 0x8000 /* Set on parity error */ > > +#define PCI_STATUS_ERROR_BITS (PCI_STATUS_DETECTED_PARITY | \ > + PCI_STATUS_SIG_SYSTEM_ERROR | \ > + PCI_STATUS_REC_MASTER_ABORT | \ > + PCI_STATUS_REC_TARGET_ABORT | \ > + PCI_STATUS_SIG_TARGET_ABORT | \ > + PCI_STATUS_PARITY) This actually *adds* PCI_STATUS_SIG_TARGET_ABORT, which is not in the driver definitions. At the very least that should be mentioned in the commit log. Ideally the addition would be in its own patch so it's obvious and bisectable, but I see the problem -- the subsequent patches consolidate things that aren't really quite the same. One alternative would be to have preliminary patches that change the drivers individually so they all use this new mask, then do the consolidation afterwards. There is pitifully little documentation about the use of include/uapi, but AFAICT that is for the user API, and this isn't part of that. I think this #define could go in include/linux/pci.h instead. > + > #define PCI_CLASS_REVISION 0x08 /* High 24 bits are class, low 8 revision */ > #define PCI_REVISION_ID 0x08 /* Revision ID */ > #define PCI_CLASS_PROG 0x09 /* Reg. Level Programming Interface */ > -- > 2.25.1 > > > >

1 0

[alsa-devel] [PATCH 00/10] soundwire: bus: fix race conditions, add suspend-resume
by Pierre-Louis Bossart 25 Feb '20

25 Feb '20

The existing mainline code for SoundWire does not handle critical race conditions, and does not have any support for pm_runtime suspend or clock-stop modes needed for e.g. jack detection or external VAD. As suggested by Vinod, these patches for the bus are shared first - with the risk that they are separated from their actual use in Intel drivers, so reviewers might wonder why they are needed in the first place. For reference, the complete set of 90+ patches required for SoundWire on Intel platforms is available here: https://github.com/thesofproject/linux/pull/1692 These patches are not Intel-specific and are likely required for e.g. Qualcomm-based implementations. All the patches in this series were generated during the joint Intel-Realtek validation effort on Intel reference designs and form-factor devices. The support for the initialization_complete signaling is already available in the Realtek codecs drivers merged in the ASoC tree (rt700, rt711, rt1308, rt715) Pierre-Louis Bossart (8): soundwire: bus: fix race condition with probe_complete signaling soundwire: bus: fix race condition with enumeration_complete signaling soundwire: bus: fix race condition with initialization_complete signaling soundwire: bus: add PM/no-PM versions of read/write functions soundwire: bus: write Slave Device Number without runtime_pm soundwire: bus: add helper to clear Slave status to UNATTACHED soundwire: bus: disable pm_runtime in sdw_slave_delete soundwire: bus: don't treat CMD_IGNORED as error on ClockStop Rander Wang (2): soundwire: bus: fix io error when processing alert event soundwire: bus: add clock stop helpers drivers/soundwire/bus.c | 509 ++++++++++++++++++++++++++++++++-- drivers/soundwire/bus.h | 9 + drivers/soundwire/bus_type.c | 5 + drivers/soundwire/slave.c | 4 + include/linux/soundwire/sdw.h | 24 ++ 5 files changed, 526 insertions(+), 25 deletions(-) -- 2.20.1

2 13

[alsa-devel] [PATCH] ASoC: fsl: Add generic CPU DAI driver
by Daniel Baluta (OSS) 25 Feb '20

25 Feb '20

From: Daniel Baluta <daniel.baluta(a)nxp.com> On i.MX8 platforms that have a DSP the DAI handling is taken care of by two entities: * Application Processor (AP), which runs Linux * DSP, which runs a firmware (typically Sound Open Firmware) The DSP has access to DAI IP registers, but it cannot easily handle resources like: * clocks * power domain management * pinctrl For this reason we introduce a generic FSL DAI driver which will take care of the resources above. --- sound/soc/fsl/Kconfig | 8 ++ sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_dai.c | 276 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 286 insertions(+) create mode 100644 sound/soc/fsl/fsl_dai.c diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 65e8cd4be930..bea8ab2c24f9 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -75,6 +75,14 @@ config SND_SOC_FSL_ESAI This option is only useful for out-of-tree drivers since in-tree drivers select it automatically. +config SND_SOC_FSL_DAI + tristate "Generic FSL DAI support for Sound Open Firmware" + help + Say Y if you want to enable generic FSL DAI support to be used + with Sound Open Firmware. This module takes care of enabling + clocks, power domain, pinctrl for FSL DAIs. The rest of DAI + control is taken care of by SOF firmware. + config SND_SOC_FSL_MICFIL tristate "Pulse Density Modulation Microphone Interface (MICFIL) module support" select REGMAP_MMIO diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 8cde88c72d93..e4ed253b6657 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -20,6 +20,7 @@ snd-soc-fsl-ssi-y := fsl_ssi.o snd-soc-fsl-ssi-$(CONFIG_DEBUG_FS) += fsl_ssi_dbg.o snd-soc-fsl-spdif-objs := fsl_spdif.o snd-soc-fsl-esai-objs := fsl_esai.o +snd-soc-fsl-dai-objs := fsl_dai.o snd-soc-fsl-micfil-objs := fsl_micfil.o snd-soc-fsl-utils-objs := fsl_utils.o snd-soc-fsl-dma-objs := fsl_dma.o @@ -32,6 +33,7 @@ obj-$(CONFIG_SND_SOC_FSL_SAI) += snd-soc-fsl-sai.o obj-$(CONFIG_SND_SOC_FSL_SSI) += snd-soc-fsl-ssi.o obj-$(CONFIG_SND_SOC_FSL_SPDIF) += snd-soc-fsl-spdif.o obj-$(CONFIG_SND_SOC_FSL_ESAI) += snd-soc-fsl-esai.o +obj-$(CONFIG_SND_SOC_FSL_DAI) += snd-soc-fsl-dai.o obj-$(CONFIG_SND_SOC_FSL_MICFIL) += snd-soc-fsl-micfil.o obj-$(CONFIG_SND_SOC_FSL_UTILS) += snd-soc-fsl-utils.o obj-$(CONFIG_SND_SOC_FSL_MQS) += snd-soc-fsl-mqs.o diff --git a/sound/soc/fsl/fsl_dai.c b/sound/soc/fsl/fsl_dai.c new file mode 100644 index 000000000000..f70889939801 --- /dev/null +++ b/sound/soc/fsl/fsl_dai.c @@ -0,0 +1,276 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Freescale Generic DAI driver for DSP +// +// Copyright 2019 NXP +// Author: Daniel Baluta <daniel.baluta(a)nxp.com> + +#include <linux/clk.h> +#include <linux/module.h> +#include <linux/pm_runtime.h> +#include <linux/pm_domain.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +enum fsl_dai_type { + FSL_DAI_TYPE_NONE, + FSL_DAI_TYPE_SAI, + FSL_DAI_TYPE_ESAI, +}; + +#define FSL_DAI_ESAI_CLK_NUM 4 +static const char *esai_clks[FSL_DAI_ESAI_CLK_NUM] = { + "core", + "extal", + "fsys", + "spba", +}; + +#define FSL_DAI_SAI_CLK_NUM 5 +static const char *sai_clks[FSL_DAI_SAI_CLK_NUM] = { + "bus", + "mclk0", + "mclk1", + "mclk2", + "mclk3", +}; + +struct fsl_dai { + struct platform_device *pdev; + + /* DAI clocks */ + struct clk **clks; + const char **clk_names; + int num_clks; + + /* Power Domain handling */ + int num_domains; + struct device **pd_dev; + struct device_link **link; + + /* DAIS */ + struct snd_soc_dai_driver *dai_drv; + int num_drv; +}; + +static struct snd_soc_dai_driver fsl_esai_dai = { + .name = "esai0", +}; + +static struct snd_soc_dai_driver fsl_sai_dai = { + .name = "sai1", +}; + +static const struct snd_soc_component_driver fsl_dai_component = { + .name = "fsl-dai", +}; + +static int fsl_dai_init_clocks(struct fsl_dai *dai_priv) +{ + struct device *dev = &dai_priv->pdev->dev; + int i; + + dai_priv->clks = devm_kcalloc(dev, dai_priv->num_clks, + sizeof(*dai_priv->clks), GFP_KERNEL); + if (!dai_priv->clks) + return -ENOMEM; + + for (i = 0; i < dai_priv->num_clks; i++) { + dai_priv->clks[i] = devm_clk_get(dev, dai_priv->clk_names[i]); + if (IS_ERR(dai_priv->clks[i])) { + dev_dbg(dev, "Failed to get clk %s\n", + dai_priv->clk_names[i]); + dai_priv->clks[i] = NULL; + } + } + + return 0; +} + +int fsl_get_dai_type(struct fsl_dai *dai_priv) +{ + struct device_node *np = dai_priv->pdev->dev.of_node; + + if (of_device_is_compatible(np, "fsl,esai-dai")) + return FSL_DAI_TYPE_ESAI; + + if (of_device_is_compatible(np, "fsl,sai-dai")) + return FSL_DAI_TYPE_SAI; + + return FSL_DAI_TYPE_NONE; +} + +static int fsl_dai_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct fsl_dai *priv; + int dai_type; + int ret; + int i; + + priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->pdev = pdev; + + dev_set_drvdata(&pdev->dev, priv); + + dai_type = fsl_get_dai_type(priv); + switch (dai_type) { + case FSL_DAI_TYPE_ESAI: + priv->clk_names = esai_clks; + priv->num_clks = FSL_DAI_ESAI_CLK_NUM; + priv->dai_drv = &fsl_esai_dai; + priv->num_drv = 1; + break; + case FSL_DAI_TYPE_SAI: + priv->clk_names = sai_clks; + priv->num_clks = FSL_DAI_SAI_CLK_NUM; + priv->dai_drv = &fsl_sai_dai; + priv->num_drv = 1; + break; + default: + dev_err(&pdev->dev, "Invalid DAI type %d\n", dai_type); + return -EINVAL; + } + + ret = fsl_dai_init_clocks(priv); + if (ret < 0) { + dev_err(&pdev->dev, "Error at init clocks\n"); + return ret; + } + + priv->num_domains = of_count_phandle_with_args(np, "power-domains", + "#power-domain-cells"); + if (priv->num_domains < 0) { + dev_err(&pdev->dev, "no power-domains property in %pOF\n", np); + return priv->num_domains; + } + + priv->pd_dev = devm_kmalloc_array(&pdev->dev, priv->num_domains, + sizeof(*priv->pd_dev), GFP_KERNEL); + if (!priv->pd_dev) + return -ENOMEM; + + priv->link = devm_kmalloc_array(&pdev->dev, priv->num_domains, + sizeof(*priv->link), GFP_KERNEL); + if (!priv->link) + return -ENOMEM; + + for (i = 0; i < priv->num_domains; i++) { + priv->pd_dev[i] = dev_pm_domain_attach_by_id(&pdev->dev, i); + if (IS_ERR(priv->pd_dev[i])) { + ret = PTR_ERR(priv->pd_dev[i]); + goto unroll_pm; + } + + priv->link[i] = device_link_add(&pdev->dev, priv->pd_dev[i], + DL_FLAG_STATELESS | + DL_FLAG_PM_RUNTIME | + DL_FLAG_RPM_ACTIVE); + if (!priv->link[i]) { + ret = -EINVAL; + dev_pm_domain_detach(priv->pd_dev[i], false); + goto unroll_pm; + } + } + + pm_runtime_enable(&pdev->dev); + + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_dai_component, + priv->dai_drv, priv->num_drv); + if (ret < 0) { + dev_err(&pdev->dev, "Failed to register DAI ret = %d\n", ret); + return ret; + } + return 0; + +unroll_pm: + while (--i >= 0) { + device_link_del(priv->link[i]); + dev_pm_domain_detach(priv->pd_dev[i], false); + } + return ret; +} + +static int fsl_dai_remove(struct platform_device *pdev) +{ + struct fsl_dai *priv = platform_get_drvdata(pdev); + int i; + + pm_runtime_disable(&priv->pdev->dev); + + for (i = 0; i < priv->num_domains; i++) { + device_link_del(priv->link[i]); + dev_pm_domain_detach(priv->pd_dev[i], false); + } + + return 0; +} + +static const struct of_device_id fsl_dai_dt_ids[] = { + { .compatible = "fsl,esai-dai", }, + { .compatible = "fsl,sai-dai", }, + {} +}; +MODULE_DEVICE_TABLE(of, fsl_dai_dt_ids); + +#ifdef CONFIG_PM +static int fsl_dai_runtime_resume(struct device *dev) +{ + struct fsl_dai *priv = dev_get_drvdata(dev); + int i, ret; + + for (i = 0; i < priv->num_clks; i++) { + ret = clk_prepare_enable(priv->clks[i]); + if (ret < 0) { + dev_err(dev, "Failed to enable clk %s\n", + priv->clk_names[i]); + goto out; + } + } + return 0; +out: + while (--i >= 0) + clk_disable_unprepare(priv->clks[i]); + + return ret; +} + +static int fsl_dai_runtime_suspend(struct device *dev) +{ + struct fsl_dai *priv = dev_get_drvdata(dev); + int i; + + for (i = 0; i < priv->num_clks; i++) + clk_disable_unprepare(priv->clks[i]); + + return 0; +} +#endif /* CONFIG_PM */ + +static const struct dev_pm_ops fsl_dai_pm_ops = { + SET_RUNTIME_PM_OPS(fsl_dai_runtime_suspend, + fsl_dai_runtime_resume, NULL) + SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, + pm_runtime_force_resume) +}; + +static struct platform_driver fsl_dai_driver = { + .probe = fsl_dai_probe, + .remove = fsl_dai_remove, + .driver = { + .name = "fsl-dai", + .pm = &fsl_dai_pm_ops, + .of_match_table = fsl_dai_dt_ids, + }, +}; + +module_platform_driver(fsl_dai_driver); + +MODULE_ALIAS("platform:fsl-dai"); + +MODULE_AUTHOR("Daniel Baluta <daniel.baluta(a)nxp.com>"); +MODULE_DESCRIPTION("FSL Generic DAI driver for DSP"); +MODULE_LICENSE("GPL v2"); -- 2.17.1

4 3

[alsa-devel] [PATCH] ASoC: Intel: mrfld: fix incorrect check on p->sink
by Colin King 25 Feb '20

25 Feb '20

From: Colin Ian King <colin.king(a)canonical.com> The check on p->sink looks bogus, I believe it should be p->source since the following code blocks are related to p->source. Fix this by replacing p->sink with p->source. Addresses-Coverity: ("Copy-paste error") Fixes: 24c8d14192cc ("ASoC: Intel: mrfld: add DSP core controls") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- [ Note: this has not been tested ] --- sound/soc/intel/atom/sst-atom-controls.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/atom/sst-atom-controls.c b/sound/soc/intel/atom/sst-atom-controls.c index baef461a99f1..f883c9340eee 100644 --- a/sound/soc/intel/atom/sst-atom-controls.c +++ b/sound/soc/intel/atom/sst-atom-controls.c @@ -1333,7 +1333,7 @@ int sst_send_pipe_gains(struct snd_soc_dai *dai, int stream, int mute) dai->capture_widget->name); w = dai->capture_widget; snd_soc_dapm_widget_for_each_source_path(w, p) { - if (p->connected && !p->connected(w, p->sink)) + if (p->connected && !p->connected(w, p->source)) continue; if (p->connect && p->source->power && -- 2.24.0

3 2

Samsung Galaxy book and and Realtek ALC298 failed in output mode
by Aurélien Croc 25 Feb '20

25 Feb '20

Dear ALSA developers, Several days ago I wrote messages here because the sound card of my new laptop Samsung Galaxy book 12 (W720) doesn't let me listening to music. It is an Intel Sunrine Point-LP HD audio with Realtek ACL298 chip. After many tests, thanks to Takashi's help, internal and headset microphone work very well but there is no sound when I play something (internal speaker and headphones). So it seems that problem doesn't come from pin configuration but something like COEF setup.. Thus I used qemu (with modifications of the code I did) in order to grab communication between Windows driver and the PCI audio sound card. In particular I extracted the VERB exchanges. Output of this job is attached to this email: - corbRirbBuffer2.txt: the whole VERB commands from startup to shut down - corbStart2.txt: VERB commands during startup (before any sound is played) - corbSound2.txt: VERB commands when output mixer is updated and a sound is played - corbEnd2.txt: VERB commands when the system is powered down - similar "output" files which contains all PCI writes for these different stages Since I'm not familiar with the hda ALSA driver and what it is currently set up with the current driver I don't know how to find the problem. The only thing I did is to execute each of the startup VERBs through hda-verb command. I saw that some of these VERBs don't return the same value as those got under Windows.. (almost 0x020c0000). Does someone has an idea about what's going wrong or what could I do in order to understand what could be problem or how to debug it deeper? Thank you for your help. Regards, Aurélien

1 0

Re: [PATCH v2 7/8] PCI: pci-bridge-emul: use PCI_STATUS_ERROR_BITS
by kbuild test robot 25 Feb '20

25 Feb '20

Hi Heiner, I love your patch! Yet something to improve: [auto build test ERROR on pci/next] [also build test ERROR on sound/for-next net-next/master net/master linus/master v5.6-rc3 next-20200224] [if your patch is applied to the wrong git tree, please drop us a note to help improve the system. BTW, we also suggest to use '--base' option to specify the base tree in git format-patch, please see https://stackoverflow.com/a/37406982] url: https://github.com/0day-ci/linux/commits/Heiner-Kallweit/PCI-add-and-use-co… base: https://git.kernel.org/pub/scm/linux/kernel/git/helgaas/pci.git next config: arm-mvebu_v7_defconfig (attached as .config) compiler: arm-linux-gnueabi-gcc (GCC) 7.5.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross # save the attached .config to linux build tree GCC_VERSION=7.5.0 make.cross ARCH=arm If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): >> drivers//pci/pci-bridge-emul.c:53:37: error: expected '}' before ';' token .w1c = PCI_STATUS_ERROR_BITS << 16; ^ drivers//pci/pci-bridge-emul.c:98:37: error: expected '}' before ';' token .w1c = PCI_STATUS_ERROR_BITS << 16; ^ vim +53 drivers//pci/pci-bridge-emul.c 40 41 static const struct pci_bridge_reg_behavior pci_regs_behavior[] = { 42 [PCI_VENDOR_ID / 4] = { .ro = ~0 }, 43 [PCI_COMMAND / 4] = { 44 .rw = (PCI_COMMAND_IO | PCI_COMMAND_MEMORY | 45 PCI_COMMAND_MASTER | PCI_COMMAND_PARITY | 46 PCI_COMMAND_SERR), 47 .ro = ((PCI_COMMAND_SPECIAL | PCI_COMMAND_INVALIDATE | 48 PCI_COMMAND_VGA_PALETTE | PCI_COMMAND_WAIT | 49 PCI_COMMAND_FAST_BACK) | 50 (PCI_STATUS_CAP_LIST | PCI_STATUS_66MHZ | 51 PCI_STATUS_FAST_BACK | PCI_STATUS_DEVSEL_MASK) << 16), 52 .rsvd = GENMASK(15, 10) | ((BIT(6) | GENMASK(3, 0)) << 16), > 53 .w1c = PCI_STATUS_ERROR_BITS << 16; 54 }, 55 [PCI_CLASS_REVISION / 4] = { .ro = ~0 }, 56 57 /* 58 * Cache Line Size register: implement as read-only, we do not 59 * pretend implementing "Memory Write and Invalidate" 60 * transactions" 61 * 62 * Latency Timer Register: implemented as read-only, as "A 63 * bridge that is not capable of a burst transfer of more than 64 * two data phases on its primary interface is permitted to 65 * hardwire the Latency Timer to a value of 16 or less" 66 * 67 * Header Type: always read-only 68 * 69 * BIST register: implemented as read-only, as "A bridge that 70 * does not support BIST must implement this register as a 71 * read-only register that returns 0 when read" 72 */ 73 [PCI_CACHE_LINE_SIZE / 4] = { .ro = ~0 }, 74 75 /* 76 * Base Address registers not used must be implemented as 77 * read-only registers that return 0 when read. 78 */ 79 [PCI_BASE_ADDRESS_0 / 4] = { .ro = ~0 }, 80 [PCI_BASE_ADDRESS_1 / 4] = { .ro = ~0 }, 81 82 [PCI_PRIMARY_BUS / 4] = { 83 /* Primary, secondary and subordinate bus are RW */ 84 .rw = GENMASK(24, 0), 85 /* Secondary latency is read-only */ 86 .ro = GENMASK(31, 24), 87 }, 88 89 [PCI_IO_BASE / 4] = { 90 /* The high four bits of I/O base/limit are RW */ 91 .rw = (GENMASK(15, 12) | GENMASK(7, 4)), 92 93 /* The low four bits of I/O base/limit are RO */ 94 .ro = (((PCI_STATUS_66MHZ | PCI_STATUS_FAST_BACK | 95 PCI_STATUS_DEVSEL_MASK) << 16) | 96 GENMASK(11, 8) | GENMASK(3, 0)), 97 98 .w1c = PCI_STATUS_ERROR_BITS << 16; 99 100 .rsvd = ((BIT(6) | GENMASK(4, 0)) << 16), 101 }, 102 103 [PCI_MEMORY_BASE / 4] = { 104 /* The high 12-bits of mem base/limit are RW */ 105 .rw = GENMASK(31, 20) | GENMASK(15, 4), 106 107 /* The low four bits of mem base/limit are RO */ 108 .ro = GENMASK(19, 16) | GENMASK(3, 0), 109 }, 110 111 [PCI_PREF_MEMORY_BASE / 4] = { 112 /* The high 12-bits of pref mem base/limit are RW */ 113 .rw = GENMASK(31, 20) | GENMASK(15, 4), 114 115 /* The low four bits of pref mem base/limit are RO */ 116 .ro = GENMASK(19, 16) | GENMASK(3, 0), 117 }, 118 119 [PCI_PREF_BASE_UPPER32 / 4] = { 120 .rw = ~0, 121 }, 122 123 [PCI_PREF_LIMIT_UPPER32 / 4] = { 124 .rw = ~0, 125 }, 126 127 [PCI_IO_BASE_UPPER16 / 4] = { 128 .rw = ~0, 129 }, 130 131 [PCI_CAPABILITY_LIST / 4] = { 132 .ro = GENMASK(7, 0), 133 .rsvd = GENMASK(31, 8), 134 }, 135 136 [PCI_ROM_ADDRESS1 / 4] = { 137 .rw = GENMASK(31, 11) | BIT(0), 138 .rsvd = GENMASK(10, 1), 139 }, 140 141 /* 142 * Interrupt line (bits 7:0) are RW, interrupt pin (bits 15:8) 143 * are RO, and bridge control (31:16) are a mix of RW, RO, 144 * reserved and W1C bits 145 */ 146 [PCI_INTERRUPT_LINE / 4] = { 147 /* Interrupt line is RW */ 148 .rw = (GENMASK(7, 0) | 149 ((PCI_BRIDGE_CTL_PARITY | 150 PCI_BRIDGE_CTL_SERR | 151 PCI_BRIDGE_CTL_ISA | 152 PCI_BRIDGE_CTL_VGA | 153 PCI_BRIDGE_CTL_MASTER_ABORT | 154 PCI_BRIDGE_CTL_BUS_RESET | 155 BIT(8) | BIT(9) | BIT(11)) << 16)), 156 157 /* Interrupt pin is RO */ 158 .ro = (GENMASK(15, 8) | ((PCI_BRIDGE_CTL_FAST_BACK) << 16)), 159 160 .w1c = BIT(10) << 16, 161 162 .rsvd = (GENMASK(15, 12) | BIT(4)) << 16, 163 }, 164 }; 165 --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

1 0

Re: [linux-sunxi] [PATCH v3 1/1] drm: sun4i: hdmi: Add support for sun4i HDMI encoder audio
by Maxime Ripard 25 Feb '20

25 Feb '20

On Tue, Feb 25, 2020 at 11:45:28AM +0200, Stefan Mavrodiev wrote: > > On 1/29/20 6:34 PM, Chen-Yu Tsai wrote: > > On Tue, Jan 28, 2020 at 10:07 PM Stefan Mavrodiev <stefan(a)olimex.com> wrote: > > > Add HDMI audio support for the sun4i-hdmi encoder, used on > > > the older Allwinner chips - A10, A20, A31. > > > > > > Most of the code is based on the BSP implementation. In it > > > dditional formats are supported (S20_3LE and S24_LE), however > > > there where some problems with them and only S16_LE is left. > > > > > > Signed-off-by: Stefan Mavrodiev <stefan(a)olimex.com> > > > --- > > > Changes for v3: > > > - Instead of platfrom_driver dynammicly register/unregister card > > > - Add Kconfig dependencies > > > - Restrore drvdata after card unregistering > > > > > > Changes for v2: > > > - Create a new platform driver instead of using the HDMI encoder > > > - Expose a new kcontrol to the userspace holding the ELD data > > > - Wrap all macro arguments in parentheses > > > > > > drivers/gpu/drm/sun4i/Kconfig | 11 + > > > drivers/gpu/drm/sun4i/Makefile | 3 + > > > drivers/gpu/drm/sun4i/sun4i_hdmi.h | 37 ++ > > > drivers/gpu/drm/sun4i/sun4i_hdmi_audio.c | 450 +++++++++++++++++++++++ > > > drivers/gpu/drm/sun4i/sun4i_hdmi_enc.c | 14 + > > > 5 files changed, 515 insertions(+) > > > create mode 100644 drivers/gpu/drm/sun4i/sun4i_hdmi_audio.c > > Since this is actually an audio driver, please include all the ALSA / ASoC > > maintainers and mailing lists (CC-ed). > > > > > diff --git a/drivers/gpu/drm/sun4i/Kconfig b/drivers/gpu/drm/sun4i/Kconfig > > > index 37e90e42943f..ca2ab5d53dd4 100644 > > > --- a/drivers/gpu/drm/sun4i/Kconfig > > > +++ b/drivers/gpu/drm/sun4i/Kconfig > > > @@ -23,6 +23,17 @@ config DRM_SUN4I_HDMI > > > Choose this option if you have an Allwinner SoC with an HDMI > > > controller. > > > > > > +config DRM_SUN4I_HDMI_AUDIO > > > + bool "Allwinner A10 HDMI Audio Support" > > > + default y > > > + depends on DRM_SUN4I_HDMI > > > + depends on SND_SOC=y || SND_SOC=DRM_SUN4I_HDMI > > > + select SND_PCM_ELD > > > + select SND_SOC_GENERIC_DMAENGINE_PCM > > > + help > > > + Choose this option if you have an Allwinner SoC with an HDMI > > > + controller and want to use audio. > > > + > > > config DRM_SUN4I_HDMI_CEC > > > bool "Allwinner A10 HDMI CEC Support" > > > depends on DRM_SUN4I_HDMI > > > diff --git a/drivers/gpu/drm/sun4i/Makefile b/drivers/gpu/drm/sun4i/Makefile > > > index 0d04f2447b01..492bfd28ad2e 100644 > > > --- a/drivers/gpu/drm/sun4i/Makefile > > > +++ b/drivers/gpu/drm/sun4i/Makefile > > > @@ -5,6 +5,9 @@ sun4i-frontend-y += sun4i_frontend.o > > > sun4i-drm-y += sun4i_drv.o > > > sun4i-drm-y += sun4i_framebuffer.o > > > > > > +ifdef CONFIG_DRM_SUN4I_HDMI_AUDIO > > > +sun4i-drm-hdmi-y += sun4i_hdmi_audio.o > > > +endif > > > sun4i-drm-hdmi-y += sun4i_hdmi_ddc_clk.o > > > sun4i-drm-hdmi-y += sun4i_hdmi_enc.o > > > sun4i-drm-hdmi-y += sun4i_hdmi_i2c.o > > > diff --git a/drivers/gpu/drm/sun4i/sun4i_hdmi.h b/drivers/gpu/drm/sun4i/sun4i_hdmi.h > > > index 7ad3f06c127e..28621d289655 100644 > > > --- a/drivers/gpu/drm/sun4i/sun4i_hdmi.h > > > +++ b/drivers/gpu/drm/sun4i/sun4i_hdmi.h > > > @@ -42,7 +42,32 @@ > > > #define SUN4I_HDMI_VID_TIMING_POL_VSYNC BIT(1) > > > #define SUN4I_HDMI_VID_TIMING_POL_HSYNC BIT(0) > > > > > > +#define SUN4I_HDMI_AUDIO_CTRL_REG 0x040 > > > +#define SUN4I_HDMI_AUDIO_CTRL_ENABLE BIT(31) > > > +#define SUN4I_HDMI_AUDIO_CTRL_RESET BIT(30) > > > + > > > +#define SUN4I_HDMI_AUDIO_FMT_REG 0x048 > > > +#define SUN4I_HDMI_AUDIO_FMT_SRC BIT(31) > > > +#define SUN4I_HDMI_AUDIO_FMT_LAYOUT BIT(3) > > > +#define SUN4I_HDMI_AUDIO_FMT_CH_CFG(n) ((n) - 1) > > > +#define SUN4I_HDMI_AUDIO_FMT_CH_CFG_MASK GENMASK(2, 0) > > > + > > > +#define SUN4I_HDMI_AUDIO_PCM_REG 0x4c > > > +#define SUN4I_HDMI_AUDIO_PCM_CH_MAP(n, m) (((m) - 1) << ((n) * 4)) > > > +#define SUN4I_HDMI_AUDIO_PCM_CH_MAP_MASK(n) (GENMASK(2, 0) << ((n) * 4)) > > > + > > > +#define SUN4I_HDMI_AUDIO_CTS_REG 0x050 > > > +#define SUN4I_HDMI_AUDIO_CTS(n) ((n) & GENMASK(19, 0)) > > > + > > > +#define SUN4I_HDMI_AUDIO_N_REG 0x054 > > > +#define SUN4I_HDMI_AUDIO_N(n) ((n) & GENMASK(19, 0)) > > > + > > > +#define SUN4I_HDMI_AUDIO_STAT0_REG 0x58 > > > +#define SUN4I_HDMI_AUDIO_STAT0_FREQ(n) ((n) << 24) > > > +#define SUN4I_HDMI_AUDIO_STAT0_FREQ_MASK GENMASK(27, 24) > > > + > > > #define SUN4I_HDMI_AVI_INFOFRAME_REG(n) (0x080 + (n)) > > > +#define SUN4I_HDMI_AUDIO_INFOFRAME_REG(n) (0x0a0 + (n)) > > > > > > #define SUN4I_HDMI_PAD_CTRL0_REG 0x200 > > > #define SUN4I_HDMI_PAD_CTRL0_BIASEN BIT(31) > > > @@ -242,6 +267,11 @@ struct sun4i_hdmi_variant { > > > bool ddc_fifo_has_dir; > > > }; > > > > > > +struct sun4i_hdmi_audio { > > > + struct snd_soc_card *card; > > > + u8 channels; > > > +}; > > > + > > > struct sun4i_hdmi { > > > struct drm_connector connector; > > > struct drm_encoder encoder; > > > @@ -283,9 +313,14 @@ struct sun4i_hdmi { > > > struct regmap_field *field_ddc_sda_en; > > > struct regmap_field *field_ddc_sck_en; > > > > > > + > > > struct sun4i_drv *drv; > > > > > > bool hdmi_monitor; > > > + bool hdmi_audio; > > > + > > > + struct sun4i_hdmi_audio audio; > > > + > > > struct cec_adapter *cec_adap; > > > > > > const struct sun4i_hdmi_variant *variant; > > > @@ -294,5 +329,7 @@ struct sun4i_hdmi { > > > int sun4i_ddc_create(struct sun4i_hdmi *hdmi, struct clk *clk); > > > int sun4i_tmds_create(struct sun4i_hdmi *hdmi); > > > int sun4i_hdmi_i2c_create(struct device *dev, struct sun4i_hdmi *hdmi); > > > +int sun4i_hdmi_audio_create(struct sun4i_hdmi *hdmi); > > > +void sun4i_hdmi_audio_destroy(struct sun4i_hdmi *hdmi); > > > > > > #endif /* _SUN4I_HDMI_H_ */ > > > diff --git a/drivers/gpu/drm/sun4i/sun4i_hdmi_audio.c b/drivers/gpu/drm/sun4i/sun4i_hdmi_audio.c > > > new file mode 100644 > > > index 000000000000..f42f2cea4e9e > > > --- /dev/null > > > +++ b/drivers/gpu/drm/sun4i/sun4i_hdmi_audio.c > > > @@ -0,0 +1,450 @@ > > > +// SPDX-License-Identifier: GPL-2.0-or-later > > > +/* > > > + * Copyright (C) 2020 Olimex Ltd. > > > + * Author: Stefan Mavrodiev <stefan(a)olimex.com> > > > + */ > > > +#include <linux/dma-mapping.h> > > > +#include <linux/dmaengine.h> > > > +#include <linux/module.h> > > > +#include <linux/of_dma.h> > > > +#include <linux/regmap.h> > > > + > > > +#include <drm/drm_print.h> > > > + > > > +#include <sound/dmaengine_pcm.h> > > > +#include <sound/pcm_drm_eld.h> > > > +#include <sound/pcm_params.h> > > > +#include <sound/soc.h> > > > + > > > +#include "sun4i_hdmi.h" > > > + > > > +static int sun4i_hdmi_audio_ctl_eld_info(struct snd_kcontrol *kcontrol, > > > + struct snd_ctl_elem_info *uinfo) > > > +{ > > > + uinfo->type = SNDRV_CTL_ELEM_TYPE_BYTES; > > > + uinfo->count = MAX_ELD_BYTES; > > > + return 0; > > > +} > > > + > > > +static int sun4i_hdmi_audio_ctl_eld_get(struct snd_kcontrol *kcontrol, > > > + struct snd_ctl_elem_value *ucontrol) > > > +{ > > > + struct snd_soc_component *component = snd_kcontrol_chip(kcontrol); > > > + struct snd_soc_card *card = snd_soc_component_get_drvdata(component); > > > + struct sun4i_hdmi *hdmi = snd_soc_card_get_drvdata(card); > > > + > > > + memcpy(ucontrol->value.bytes.data, > > > + hdmi->connector.eld, > > > + MAX_ELD_BYTES); > > > + > > > + return 0; > > > +} > > > + > > > +static const struct snd_kcontrol_new sun4i_hdmi_audio_controls[] = { > > > + { > > > + .access = SNDRV_CTL_ELEM_ACCESS_READ | > > > + SNDRV_CTL_ELEM_ACCESS_VOLATILE, > > > + .iface = SNDRV_CTL_ELEM_IFACE_PCM, > > > + .name = "ELD", > > > + .info = sun4i_hdmi_audio_ctl_eld_info, > > > + .get = sun4i_hdmi_audio_ctl_eld_get, > > > + }, > > > +}; > > > + > > > +static const struct snd_soc_dapm_widget sun4i_hdmi_audio_widgets[] = { > > > + SND_SOC_DAPM_OUTPUT("TX"), > > > +}; > > > + > > > +static const struct snd_soc_dapm_route sun4i_hdmi_audio_routes[] = { > > > + { "TX", NULL, "Playback" }, > > > +}; > > > + > > > +static const struct snd_soc_component_driver sun4i_hdmi_audio_component = { > > > + .controls = sun4i_hdmi_audio_controls, > > > + .num_controls = ARRAY_SIZE(sun4i_hdmi_audio_controls), > > > + .dapm_widgets = sun4i_hdmi_audio_widgets, > > > + .num_dapm_widgets = ARRAY_SIZE(sun4i_hdmi_audio_widgets), > > > + .dapm_routes = sun4i_hdmi_audio_routes, > > > + .num_dapm_routes = ARRAY_SIZE(sun4i_hdmi_audio_routes), > > > +}; > > > + > > > +static int sun4i_hdmi_audio_startup(struct snd_pcm_substream *substream, > > > + struct snd_soc_dai *dai) > > > +{ > > > + struct snd_soc_card *card = snd_soc_dai_get_drvdata(dai); > > > + struct sun4i_hdmi *hdmi = snd_soc_card_get_drvdata(card); > > > + u32 reg; > > > + int ret; > > > + > > > + regmap_write(hdmi->regmap, SUN4I_HDMI_AUDIO_CTRL_REG, 0); > > > + regmap_write(hdmi->regmap, > > > + SUN4I_HDMI_AUDIO_CTRL_REG, > > > + SUN4I_HDMI_AUDIO_CTRL_RESET); > > > + ret = regmap_read_poll_timeout(hdmi->regmap, > > > + SUN4I_HDMI_AUDIO_CTRL_REG, > > > + reg, !reg, 100, 50000); > > > + if (ret < 0) { > > > + DRM_ERROR("Failed to reset HDMI Audio\n"); > > > + return ret; > > > + } > > > + > > > + regmap_write(hdmi->regmap, > > > + SUN4I_HDMI_AUDIO_CTRL_REG, > > > + SUN4I_HDMI_AUDIO_CTRL_ENABLE); > > > + > > > + return snd_pcm_hw_constraint_eld(substream->runtime, > > > + hdmi->connector.eld); > > > +} > > > + > > > +static void sun4i_hdmi_audio_shutdown(struct snd_pcm_substream *substream, > > > + struct snd_soc_dai *dai) > > > +{ > > > + struct snd_soc_card *card = snd_soc_dai_get_drvdata(dai); > > > + struct sun4i_hdmi *hdmi = snd_soc_card_get_drvdata(card); > > > + > > > + regmap_write(hdmi->regmap, SUN4I_HDMI_AUDIO_CTRL_REG, 0); > > > +} > > > + > > > +static int sun4i_hdmi_setup_audio_infoframes(struct sun4i_hdmi *hdmi) > > > +{ > > > + union hdmi_infoframe frame; > > > + u8 buffer[14]; > > > + int i, ret; > > > + > > > + ret = hdmi_audio_infoframe_init(&frame.audio); > > > + if (ret < 0) { > > > + DRM_ERROR("Failed to init HDMI audio infoframe\n"); > > > + return ret; > > > + } > > > + > > > + frame.audio.coding_type = HDMI_AUDIO_CODING_TYPE_STREAM; > > > + frame.audio.sample_frequency = HDMI_AUDIO_SAMPLE_FREQUENCY_STREAM; > > > + frame.audio.sample_size = HDMI_AUDIO_SAMPLE_SIZE_STREAM; > > > + frame.audio.channels = hdmi->audio.channels; > > > + > > > + ret = hdmi_infoframe_pack(&frame, buffer, sizeof(buffer)); > > > + if (ret < 0) { > > > + DRM_ERROR("Failed to pack HDMI audio infoframe\n"); > > > + return ret; > > > + } > > > + > > > + for (i = 0; i < sizeof(buffer); i++) > > > + writeb(buffer[i], > > > + hdmi->base + SUN4I_HDMI_AUDIO_INFOFRAME_REG(i)); > > > + > > > + return 0; > > > +} > > > + > > > +static void sun4i_hdmi_audio_set_cts_n(struct sun4i_hdmi *hdmi, > > > + struct snd_pcm_hw_params *params) > > > +{ > > > + struct drm_encoder *encoder = &hdmi->encoder; > > > + struct drm_crtc *crtc = encoder->crtc; > > > + const struct drm_display_mode *mode = &crtc->state->adjusted_mode; > > > + u32 rate = params_rate(params); > > > + u32 n, cts; > > > + u64 tmp; > > > + > > > + /** > > > + * Calculate Cycle Time Stamp (CTS) and Numerator (N): > > > + * > > > + * N = 128 * Samplerate / 1000 > > > + * CTS = (Ftdms * N) / (128 * Samplerate) > > > + */ > > > + > > > + n = 128 * rate / 1000; > > > + tmp = (u64)(mode->clock * 1000) * n; > > > + do_div(tmp, 128 * rate); > > > + cts = tmp; > > > + > > > + regmap_write(hdmi->regmap, > > > + SUN4I_HDMI_AUDIO_CTS_REG, > > > + SUN4I_HDMI_AUDIO_CTS(cts)); > > > + > > > + regmap_write(hdmi->regmap, > > > + SUN4I_HDMI_AUDIO_N_REG, > > > + SUN4I_HDMI_AUDIO_N(n)); > > > +} > > > + > > > +static int sun4i_hdmi_audio_set_hw_rate(struct sun4i_hdmi *hdmi, > > > + struct snd_pcm_hw_params *params) > > > +{ > > > + u32 rate = params_rate(params); > > > + u32 val; > > > + > > > + switch (rate) { > > > + case 44100: > > > + val = 0x0; > > > + break; > > > + case 48000: > > > + val = 0x2; > > > + break; > > > + case 32000: > > > + val = 0x3; > > > + break; > > > + case 88200: > > > + val = 0x8; > > > + break; > > > + case 96000: > > > + val = 0x9; > > > + break; > > > + case 176400: > > > + val = 0xc; > > > + break; > > > + case 192000: > > > + val = 0xe; > > > + break; > > > + default: > > > + return -EINVAL; > > > + } > > > + > > > + regmap_update_bits(hdmi->regmap, > > > + SUN4I_HDMI_AUDIO_STAT0_REG, > > > + SUN4I_HDMI_AUDIO_STAT0_FREQ_MASK, > > > + SUN4I_HDMI_AUDIO_STAT0_FREQ(val)); > > > + > > > + return 0; > > > +} > > > + > > > +static int sun4i_hdmi_audio_set_hw_channels(struct sun4i_hdmi *hdmi, > > > + struct snd_pcm_hw_params *params) > > > +{ > > > + u32 channels = params_channels(params); > > > + > > > + if (channels > 8) > > > + return -EINVAL; > > > + > > > + hdmi->audio.channels = channels; > > > + > > > + regmap_update_bits(hdmi->regmap, > > > + SUN4I_HDMI_AUDIO_FMT_REG, > > > + SUN4I_HDMI_AUDIO_FMT_LAYOUT, > > > + (channels > 2) ? SUN4I_HDMI_AUDIO_FMT_LAYOUT : 0); > > > + > > > + regmap_update_bits(hdmi->regmap, > > > + SUN4I_HDMI_AUDIO_FMT_REG, > > > + SUN4I_HDMI_AUDIO_FMT_CH_CFG_MASK, > > > + SUN4I_HDMI_AUDIO_FMT_CH_CFG(channels)); > > > + > > > + regmap_write(hdmi->regmap, SUN4I_HDMI_AUDIO_PCM_REG, 0x76543210); > > > + > > > + /** > > > + * If only one channel is required, send the same sample > > > + * to the sink device as a left and right channel. > > > + */ > > > + if (channels == 1) > > > + regmap_update_bits(hdmi->regmap, > > > + SUN4I_HDMI_AUDIO_PCM_REG, > > > + SUN4I_HDMI_AUDIO_PCM_CH_MAP_MASK(1), > > > + SUN4I_HDMI_AUDIO_PCM_CH_MAP(1, 1)); > > > + > > > + return 0; > > > +} > > > + > > > +static int sun4i_hdmi_audio_hw_params(struct snd_pcm_substream *substream, > > > + struct snd_pcm_hw_params *params, > > > + struct snd_soc_dai *dai) > > > +{ > > > + struct snd_soc_card *card = snd_soc_dai_get_drvdata(dai); > > > + struct sun4i_hdmi *hdmi = snd_soc_card_get_drvdata(card); > > > + int ret; > > > + > > > + ret = sun4i_hdmi_audio_set_hw_rate(hdmi, params); > > > + if (ret < 0) > > > + return ret; > > > + > > > + ret = sun4i_hdmi_audio_set_hw_channels(hdmi, params); > > > + if (ret < 0) > > > + return ret; > > > + > > > + sun4i_hdmi_audio_set_cts_n(hdmi, params); > > > + > > > + return 0; > > > +} > > > + > > > +static int sun4i_hdmi_audio_trigger(struct snd_pcm_substream *substream, > > > + int cmd, > > > + struct snd_soc_dai *dai) > > > +{ > > > + struct snd_soc_card *card = snd_soc_dai_get_drvdata(dai); > > > + struct sun4i_hdmi *hdmi = snd_soc_card_get_drvdata(card); > > > + int ret = 0; > > > + > > > + switch (cmd) { > > > + case SNDRV_PCM_TRIGGER_START: > > > + ret = sun4i_hdmi_setup_audio_infoframes(hdmi); > > > + break; > > > + default: > > > + break; > > > + } > > > + > > > + return ret; > > > +} > > > + > > > +static const struct snd_soc_dai_ops sun4i_hdmi_audio_dai_ops = { > > > + .startup = sun4i_hdmi_audio_startup, > > > + .shutdown = sun4i_hdmi_audio_shutdown, > > > + .hw_params = sun4i_hdmi_audio_hw_params, > > > + .trigger = sun4i_hdmi_audio_trigger, > > > +}; > > > + > > > +static int sun4i_hdmi_audio_dai_probe(struct snd_soc_dai *dai) > > > +{ > > > + struct snd_dmaengine_dai_dma_data *dma_data; > > > + > > > + dma_data = devm_kzalloc(dai->dev, sizeof(*dma_data), GFP_KERNEL); > > > + if (!dma_data) > > > + return -ENOMEM; > > > + > > > + dma_data->addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; > > > + dma_data->maxburst = 8; > > > + > > > + snd_soc_dai_init_dma_data(dai, dma_data, NULL); > > > + > > > + return 0; > > > +} > > > + > > > +static struct snd_soc_dai_driver sun4i_hdmi_audio_dai = { > > > + .name = "HDMI", > > > + .ops = &sun4i_hdmi_audio_dai_ops, > > > + .probe = sun4i_hdmi_audio_dai_probe, > > > + .playback = { > > > + .stream_name = "Playback", > > > + .channels_min = 1, > > > + .channels_max = 8, > > > + .formats = SNDRV_PCM_FMTBIT_S16_LE, > > > + .rates = SNDRV_PCM_RATE_8000_192000, > > > + }, > > > +}; > > > + > > > +static const struct snd_pcm_hardware sun4i_hdmi_audio_pcm_hardware = { > > > + .info = SNDRV_PCM_INFO_INTERLEAVED | > > > + SNDRV_PCM_INFO_BLOCK_TRANSFER | > > > + SNDRV_PCM_INFO_MMAP | > > > + SNDRV_PCM_INFO_MMAP_VALID | > > > + SNDRV_PCM_INFO_PAUSE | > > > + SNDRV_PCM_INFO_RESUME, > > > + .formats = SNDRV_PCM_FMTBIT_S16_LE, > > > + .rates = SNDRV_PCM_RATE_8000_192000, > > > + .rate_min = 8000, > > > + .rate_max = 192000, > > > + .channels_min = 1, > > > + .channels_max = 8, > > > + .buffer_bytes_max = 128 * 1024, > > > + .period_bytes_min = 4 * 1024, > > > + .period_bytes_max = 32 * 1024, > > > + .periods_min = 2, > > > + .periods_max = 8, > > > + .fifo_size = 128, > > > +}; > > > + > > > +static const struct snd_dmaengine_pcm_config sun4i_hdmi_audio_pcm_config = { > > > + .chan_names[SNDRV_PCM_STREAM_PLAYBACK] = "audio-tx", > > > + .pcm_hardware = &sun4i_hdmi_audio_pcm_hardware, > > > + .prealloc_buffer_size = 128 * 1024, > > > + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, > > > +}; > > > + > > > +struct snd_soc_card sun4i_hdmi_audio_card = { > > > + .name = "sun4i-hdmi", > > > +}; > > > + > > > +int sun4i_hdmi_audio_create(struct sun4i_hdmi *hdmi) > > > +{ > > > + struct snd_soc_card *card = &sun4i_hdmi_audio_card; > > > + struct snd_soc_dai_link_component *comp; > > > + struct snd_soc_dai_link *link; > > > + int ret; > > > + > > > + ret = snd_dmaengine_pcm_register(hdmi->dev, > > > + &sun4i_hdmi_audio_pcm_config, 0); > > > + if (ret < 0) { > > > + DRM_ERROR("Could not register PCM\n"); > > > + return ret; > > > + } > > > + > > > + ret = snd_soc_register_component(hdmi->dev, > > > + &sun4i_hdmi_audio_component, > > > + &sun4i_hdmi_audio_dai, 1); > > > + if (ret < 0) { > > > + DRM_ERROR("Could not register DAI\n"); > > > + goto unregister_pcm; > > > + } > > > + > > > + link = devm_kzalloc(hdmi->dev, sizeof(*link), GFP_KERNEL); > > > + if (!link) { > > > + ret = -ENOMEM; > > > + goto unregister_component; > > > + } > > > + > > > + comp = devm_kzalloc(hdmi->dev, sizeof(*comp) * 3, GFP_KERNEL); > > > + if (!comp) { > > > + ret = -ENOMEM; > > > + goto unregister_component; > > > + } > > > + > > > + link->cpus = &comp[0]; > > > + link->codecs = &comp[1]; > > > + link->platforms = &comp[2]; > > > + > > > + link->num_cpus = 1; > > > + link->num_codecs = 1; > > > + link->num_platforms = 1; > > > + > > > + link->playback_only = 1; > > > + > > > + link->name = "SUN4I-HDMI"; > > > + link->stream_name = "SUN4I-HDMI PCM"; > > > + > > > + link->codecs->name = dev_name(hdmi->dev); > > > + link->codecs->dai_name = sun4i_hdmi_audio_dai.name; > > > + > > > + link->cpus->dai_name = dev_name(hdmi->dev); > > > + > > > + link->platforms->name = dev_name(hdmi->dev); > > > + > > > + link->dai_fmt = SND_SOC_DAIFMT_I2S; > > > + > > > + card->dai_link = link; > > > + card->num_links = 1; > > > + card->dev = hdmi->dev; > > > + > > > + hdmi->audio.card = card; > > > + > > > + /** > > > + * snd_soc_register_card() will overwrite the driver_data pointer. > > > + * So before registering the card, store the original pointer in > > > + * card->drvdata. > > > + */ > > > + snd_soc_card_set_drvdata(card, hdmi); > > > + ret = snd_soc_register_card(card); > > > + if (ret) > > > + goto unregister_component; > > So using ASoC with all the components IMHO is just adding dead weight. The > > audio interface for this particular hardware is just a FIFO that needs to > > be written to by an external DMA engine, and a bunch of controls to setup > > the parameters of the HDMI audio stream. There's no power sequencing to do, > > and no actual individual components to control. There's no reason you couldn't > > use just the ALSA DMAENGINE helpers to create a simple ALSA sound card. > > > > (Maybe we could clean it up after it's merged? Would there be any issues > > with backward compatibility?) > > > > I think the only example of this besides ASoC is the PXA2xx sound library > > and AC97 driver: > > > > - sound/arm/pxa2xx-pcm-lib.c > > - sound/arm/pxa2xx-ac97.c > > I'd like to hear Maxime opinion on this. I get it why it's overkill to use > ASoC. > However it will take only couple of lines to make the changes for v4. On the > other > side with dropping the ASoC, the driver must be almost rewritten. It's not > such a big deal, > but I want to know that this is the right thing to do. I don't really have an opinion on this, either way is fine for me, so if it's simpler to go for ALSA, then let's do it :) Maxime

1 0

[alsa-devel] [PATCH 0/7] ASoC: soc-pcm cleanup step2
by Kuninori Morimoto 25 Feb '20

25 Feb '20

Hi Mark These are soc-pcm cleanup step2. Very random cleanup... Kuninori Morimoto (7): ASoC: soc-pcm: add snd_soc_runtime_action() ASoC: soc-pcm: adjustment for DAI member 0 reset ASoC: soc-pcm: add for_each_dapm_widgets() macro ASoC: soc-pcm: goto error after trying for_each_rtd_codec_dai ASoC: soc-pcm: goto error after trying all component open ASoC: soc-pcm: move soc_pcm_close() next to soc_pcm_open() ASoC: soc-pcm: tidyup soc_pcm_open() order include/sound/soc-dapm.h | 5 + sound/soc/soc-dapm.c | 8 +- sound/soc/soc-pcm.c | 266 ++++++++++++++++++++--------------------------- 3 files changed, 118 insertions(+), 161 deletions(-) -- 2.7.4

5 17

[alsa-devel] [PATCH RFC v4 0/6] ASoC: Add Multi CPU DAI support
by Bard Liao 25 Feb '20

25 Feb '20

As discussed in [1], ASoC core supports multi codec DAIs on a DAI link. However it does not do so for CPU DAIs. So, add support for multi CPU DAIs on a DAI Link by adding multi CPU DAI in Card instantiation, suspend and resume functions, PCM ops, stream handling functions and DAPM. [1]: https://www.spinics.net/lists/alsa-devel/msg71369.html changes in v4: - fix "rtd->num_codecs > 1" error in ASoC: Add Multi CPU DAI support - split "ASoC: pcm: check if cpu DAI support the given stream" from "ASoC: Add multiple CPU DAI support for PCM ops" - add helper "ASoC: Add dapm_add_valid_dai_widget helper" for "ASoC: Add multiple CPU DAI support in DAPM" Bard Liao (2): ASoC: return error if the function is not support multi cpu yet. ASoC: pcm: check if cpu DAI support the given stream Shreyas NC (4): ASoC: Add initial support for multiple CPU DAIs ASoC: Add multiple CPU DAI support for PCM ops ASoC: Add dapm_add_valid_dai_widget helper ASoC: Add multiple CPU DAI support in DAPM include/sound/soc.h | 15 + sound/soc/soc-compress.c | 5 +- sound/soc/soc-core.c | 205 +++++----- sound/soc/soc-dapm.c | 133 ++++--- sound/soc/soc-generic-dmaengine-pcm.c | 18 + sound/soc/soc-pcm.c | 554 ++++++++++++++++++-------- 6 files changed, 615 insertions(+), 315 deletions(-) -- 2.17.1

3 9

Re: [PATCH 0/9] PCI: add and use constant PCI_STATUS_ERROR_BITS and helper pci_status_get_and_clear_errors
by David Miller 25 Feb '20

25 Feb '20

From: Heiner Kallweit <hkallweit1(a)gmail.com> Date: Mon, 24 Feb 2020 22:20:08 +0100 > Few drivers have own definitions for this constant, so move it to the > PCI core. In addition there are several places where the following > code sequence is used: > 1. Read PCI_STATUS > 2. Mask out non-error bits > 3. Action based on set error bits > 4. Write back set error bits to clear them > > As this is a repeated pattern, add a helper to the PCI core. > > Most affected drivers are network drivers. But as it's about core > PCI functionality, I suppose the series should go through the PCI > tree. Heiner, something is up with this submission. The subject line here says 0/9, but the patches say N/8 and patch #8 never showed up on the list. Sort out what this should be and resubmit, thank you.

1 0

[PATCH] ASoC: soc-pcm/soc-compress: don't use snd_soc_dapm_stream_stop()
by Kuninori Morimoto 25 Feb '20

25 Feb '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit b0edff42360ab4 ("ASoC: soc-pcm/soc-compress: use snd_soc_dapm_stream_stop() for SND_SOC_DAPM_STREAM_STOP") uses snd_soc_dapm_stream_stop() for soc_compr_free_fe() and dpcm_fe_dai_shutdown() because it didn't care about pmdown_time. But, it didn't need to care. This patch rollback to original code. Some system will wait unneeded timed-out without this patch. Special Thanks for reporting to Chris Gorman. ... intel_sst_acpi 808622A8:00: Wait timed-out condition:0x0, msg_id:0x1 fw_state 0x3 intel_sst_acpi 808622A8:00: fw returned err -16 sst-mfld-platform sst-mfld-platform: ASoC: PRE_PMD: pcm0_in event failed: -16 ... Fixes: commit b0edff42360ab4 ("ASoC: soc-pcm/soc-compress: use snd_soc_dapm_stream_stop() for SND_SOC_DAPM_STREAM_STOP") Reported-by: Chris Gorman <chrisjohgorman(a)gmail.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- > Mark This patch is for asoc/for-5.6 branch sound/soc/soc-compress.c | 2 +- sound/soc/soc-pcm.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 223cd045719e..392a1c5b15d3 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -299,7 +299,7 @@ static int soc_compr_free_fe(struct snd_compr_stream *cstream) for_each_dpcm_be(fe, stream, dpcm) dpcm->state = SND_SOC_DPCM_LINK_STATE_FREE; - snd_soc_dapm_stream_stop(fe, stream); + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; fe->dpcm[stream].runtime_update = SND_SOC_DPCM_UPDATE_NO; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 42fb03a99cb1..6d400bf02ca1 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2006,7 +2006,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) soc_pcm_close(substream); /* run the stream event for each BE */ - snd_soc_dapm_stream_stop(fe, stream); + dpcm_dapm_stream_event(fe, stream, SND_SOC_DAPM_STREAM_STOP); fe->dpcm[stream].state = SND_SOC_DPCM_STATE_CLOSE; dpcm_set_fe_update_state(fe, stream, SND_SOC_DPCM_UPDATE_NO); -- 2.17.1

2 1

[PATCH] ASoC: Intel: Skylake: Fix available clock counter incrementation
by Amadeusz Sławiński 25 Feb '20

25 Feb '20

Incrementation of avail_clk_cnt was incorrectly moved to error path. Put it back to success path. Fixes: 6ee927f2f01466 ('ASoC: Intel: Skylake: Fix NULL ptr dereference when unloading clk dev') Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/intel/skylake/skl-ssp-clk.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-ssp-clk.c b/sound/soc/intel/skylake/skl-ssp-clk.c index 1c0e5226cb5b..bd43885f3805 100644 --- a/sound/soc/intel/skylake/skl-ssp-clk.c +++ b/sound/soc/intel/skylake/skl-ssp-clk.c @@ -384,9 +384,11 @@ static int skl_clk_dev_probe(struct platform_device *pdev) &clks[i], clk_pdata, i); if (IS_ERR(data->clk[data->avail_clk_cnt])) { - ret = PTR_ERR(data->clk[data->avail_clk_cnt++]); + ret = PTR_ERR(data->clk[data->avail_clk_cnt]); goto err_unreg_skl_clk; } + + data->avail_clk_cnt++; } platform_set_drvdata(pdev, data); -- 2.17.1

4 6

[PATCH 00/13] ASoC: soc-pcm cleanup step4
by Kuninori Morimoto 25 Feb '20

25 Feb '20

Hi Mark These are soc-pcm cleanup step4. [01/13] - [06/13] are posted but not yet accepted patches. To avoid conflict, I merged these into step4. These patches are already received Reviewed-by from Pierre-Louis. Subject: [PATCH 0/6] ASoC: use for_each_rtd_codec_dai() macro Date: Thu, 13 Feb 2020 13:08:07 +0900 Kuninori Morimoto (13): 1) ASoC: soundwaire: qcom: use for_each_rtd_codec_dai() macro 2) ASoC: qcom: sdm845: use for_each_rtd_codec_dai() macro 3) ASoC: qcom: apq8016_sbc: use for_each_rtd_codec_dai() macro 4) ASoC: intel: cml_rt1011_rt5682: use for_each_rtd_codec_dai() macro 5) ASoC: intel: kbl_da7219_max98927: use for_each_rtd_codec_dai() macro 6) ASoC: mediatek: mt8183-da7219-max98357: use for_each_rtd_codec_dai() macro 7) ASoC: soc-pcm: add snd_soc_dai_get_pcm_stream() 8) ASoC: soc-pcm: cleanup soc_pcm_apply_msb() 9) ASoC: soc-pcm: add snd_soc_dai_get_widget() 10) ASoC: soc-pcm: merge dpcm_run_new/old_update() into dpcm_fe_runtime_update() 11) ASoC: soc-pcm: move dpcm_path_put() to soc-pcm.c 12) ASoC: soc-pcm: move CONFIG_DEBUG_FS functions to top side 13) ASoC: soc-pcm: add dpcm_create/remove_debugfs_state() drivers/soundwire/qcom.c | 7 +- include/sound/soc-dai.h | 15 + include/sound/soc-dapm.h | 1 + include/sound/soc-dpcm.h | 7 +- sound/soc/intel/boards/cml_rt1011_rt5682.c | 3 +- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 +- sound/soc/intel/skylake/skl-pcm.c | 10 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 8 +- sound/soc/qcom/apq8016_sbc.c | 7 +- sound/soc/qcom/sdm845.c | 20 +- sound/soc/soc-dai.c | 7 +- sound/soc/soc-dapm.c | 20 +- sound/soc/soc-pcm.c | 462 ++++++++---------- 13 files changed, 266 insertions(+), 309 deletions(-) -- 2.17.1 Thank you for your help !! Best regards --- Kuninori Morimoto

2 20

[PATCH] ASoC: soc-pcm: fix state tracking error in snd_soc_component_open/close()
by Kai Vehmanen 25 Feb '20

25 Feb '20

ASoC component open/close and snd_soc_component_module_get/put are called independently for each component-substream pair, so the logic added in commit dd03907bf129 ("ASoC: soc-pcm: call snd_soc_component_open/close() once") was not sufficient and led to PCM playback and module unload errors. Implement handling of failures directly in soc_pcm_components_open(), so that any successfully opened components are closed upon error with other components. This allows to clean up error handling in soc_pcm_open() without adding more state tracking. Fixes: dd03907bf129 ("ASoC: soc-pcm: call snd_soc_component_open/close() once") Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- include/sound/soc-component.h | 7 ++----- sound/soc/soc-component.c | 35 +++++++---------------------------- sound/soc/soc-pcm.c | 27 +++++++++++++++++++++------ 3 files changed, 30 insertions(+), 39 deletions(-) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index 1866ecc8e94b..154d02fbbfed 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -147,6 +147,8 @@ struct snd_soc_component { unsigned int active; + unsigned int suspended:1; /* is in suspend PM state */ + struct list_head list; struct list_head card_aux_list; /* for auxiliary bound components */ struct list_head card_list; @@ -180,11 +182,6 @@ struct snd_soc_component { struct dentry *debugfs_root; const char *debugfs_prefix; #endif - - /* bit field */ - unsigned int suspended:1; /* is in suspend PM state */ - unsigned int opened:1; - unsigned int module:1; }; #define for_each_component_dais(component, dai)\ diff --git a/sound/soc/soc-component.c b/sound/soc/soc-component.c index ee00c09df5e7..14e175cdeeb8 100644 --- a/sound/soc/soc-component.c +++ b/sound/soc/soc-component.c @@ -297,55 +297,34 @@ EXPORT_SYMBOL_GPL(snd_soc_component_set_jack); int snd_soc_component_module_get(struct snd_soc_component *component, int upon_open) { - if (component->module) - return 0; - if (component->driver->module_get_upon_open == !!upon_open && !try_module_get(component->dev->driver->owner)) return -ENODEV; - component->module = 1; - return 0; } void snd_soc_component_module_put(struct snd_soc_component *component, int upon_open) { - if (component->module && - component->driver->module_get_upon_open == !!upon_open) + if (component->driver->module_get_upon_open == !!upon_open) module_put(component->dev->driver->owner); - - component->module = 0; } int snd_soc_component_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - int ret = 0; - - if (!component->opened && - component->driver->open) - ret = component->driver->open(component, substream); - - if (ret == 0) - component->opened = 1; - - return ret; + if (component->driver->open) + return component->driver->open(component, substream); + return 0; } int snd_soc_component_close(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - int ret = 0; - - if (component->opened && - component->driver->close) - ret = component->driver->close(component, substream); - - component->opened = 0; - - return ret; + if (component->driver->close) + return component->driver->close(component, substream); + return 0; } int snd_soc_component_prepare(struct snd_soc_component *component, diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 63f67eb7c077..9e761f932887 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -469,28 +469,43 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) static int soc_pcm_components_open(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *last = NULL; struct snd_soc_component *component; int i, ret = 0; for_each_rtd_components(rtd, i, component) { + last = component; + ret = snd_soc_component_module_get_when_open(component); if (ret < 0) { dev_err(component->dev, "ASoC: can't get module %s\n", component->name); - return ret; + break; } ret = snd_soc_component_open(component, substream); if (ret < 0) { + snd_soc_component_module_put_when_close(component); dev_err(component->dev, "ASoC: can't open component %s: %d\n", component->name, ret); - return ret; + break; } } - return 0; + if (ret < 0) { + /* rollback on error */ + for_each_rtd_components(rtd, i, component) { + if (component == last) + break; + + snd_soc_component_close(component, substream); + snd_soc_component_module_put_when_close(component); + } + } + + return ret; } static int soc_pcm_components_close(struct snd_pcm_substream *substream) @@ -585,7 +600,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) if (ret < 0) { pr_err("ASoC: %s startup failed: %d\n", rtd->dai_link->name, ret); - goto component_err; + goto rtd_startup_err; } /* startup the audio subsystem */ @@ -681,9 +696,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) snd_soc_dai_shutdown(cpu_dai, substream); soc_rtd_shutdown(rtd, substream); -component_err: +rtd_startup_err: soc_pcm_components_close(substream); - +component_err: mutex_unlock(&rtd->card->pcm_mutex); for_each_rtd_components(rtd, i, component) { -- 2.17.1

4 4

Re: [PATCH] docs: dt: fix several broken doc references
by Rob Herring 24 Feb '20

24 Feb '20

On Sun, Feb 23, 2020 at 09:59:53AM +0100, Mauro Carvalho Chehab wrote: > There are several DT doc references that require manual fixes. > I found 3 cases fixed on this patch: > > - directory named "binding/" instead of "bindings/"; > - .txt to .yaml renames; > - file renames (still on txt format); > > Signed-off-by: Mauro Carvalho Chehab <mchehab+huawei(a)kernel.org> > --- > .../devicetree/bindings/mtd/cadence-nand-controller.txt | 2 +- > .../devicetree/bindings/net/brcm,bcm7445-switch-v4.0.txt | 2 +- > Documentation/devicetree/bindings/sound/st,stm32-sai.txt | 2 +- > Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt | 2 +- > Documentation/devicetree/bindings/spi/st,stm32-spi.yaml | 2 +- > MAINTAINERS | 4 ++-- > .../devicetree/bindings/net/wireless/siliabs,wfx.txt | 2 +- > 7 files changed, 8 insertions(+), 8 deletions(-) Applied. Rob

1 0

[alsa-devel] [PATCH] ASoC: codecs: make PCM5102A codec selectable
by David Douard 24 Feb '20

24 Feb '20

The PCM5102A codec entry in the Kconfig file lacks a title/description to make it selectable in menuconfig (since it has no dependent i2s/spi sub-modules like e.g. the pcm512x one). --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7e90f5d83097..676fa62e9b7f 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -934,7 +934,7 @@ config SND_SOC_PCM3168A_SPI select REGMAP_SPI config SND_SOC_PCM5102A - tristate + tristate "Texas Instruments PCM5102A CODEC" config SND_SOC_PCM512x tristate -- 2.20.1

2 3

[PATCH 1/2] mfd: wm8994: Fix driver operation if loaded as modules
by Marek Szyprowski 24 Feb '20

24 Feb '20

WM8994 chip has built-in regulators, which might be used for chip operation. They are controlled by a separate wm8994-regulator driver, which should be loaded before this driver calls regulator_get(), because that driver also provides consumer-supply mapping for the them. If that driver is not yet loaded, regulator core substitute them with dummy regulator, what breaks chip operation, because the built-in regulators are never enabled. Fix this by adding a comment and a call to module_request() helper. Signed-off-by: Marek Szyprowski <m.szyprowski(a)samsung.com> --- drivers/mfd/wm8994-core.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index 1e9fe7d92597..f15f12d8bccc 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -321,6 +321,13 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) int ret, i, patch_regs = 0; int pulls = 0; + /* + * Request a driver for built-in regulators first, because they are + * default supplies for "AVDD1" and "DCVDD". Otherwise, getting them + * fails due to lack of consumer mapping defined by that driver. + */ + request_module("wm8994_regulator"); + if (dev_get_platdata(wm8994->dev)) { pdata = dev_get_platdata(wm8994->dev); wm8994->pdata = *pdata; -- 2.17.1

1 1

[PATCH for-next 1/2] ASoC: tlv320adcx140: Add DRE and AGC support
by Dan Murphy 24 Feb '20

24 Feb '20

The TLV320ADCx140 parts support Dynamic Range Enhancer (DRE) as defined in Section 8.3.2 of the data sheets. The DRE achieves a complete-channel dynamic range as high as 120 dB. At a system level, the DRE scheme enables far-field, high-fidelity recording of audio signals in very quiet environments and low-distortion recording in loud environments. There are 2 enables for DRE. The first is a global setting that enables the DRE engine in the device and the other enable is per channel. If the DRE is enabled globally then either DRE or AGC can be used per each configured channel. If global DRE is disabled then even setting the DRE enable bit in the channel config register will have no effect. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tlv320adcx140.c | 55 ++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320adcx140.h | 1 + 2 files changed, 56 insertions(+) diff --git a/sound/soc/codecs/tlv320adcx140.c b/sound/soc/codecs/tlv320adcx140.c index 8182c584de9c..105e51be6fe6 100644 --- a/sound/soc/codecs/tlv320adcx140.c +++ b/sound/soc/codecs/tlv320adcx140.c @@ -108,6 +108,7 @@ static const struct reg_default adcx140_reg_defaults[] = { { ADCX140_DSP_CFG0, 0x01 }, { ADCX140_DSP_CFG1, 0x40 }, { ADCX140_DRE_CFG0, 0x7b }, + { ADCX140_AGC_CFG0, 0xe7 }, { ADCX140_IN_CH_EN, 0xf0 }, { ADCX140_ASI_OUT_CH_EN, 0x00 }, { ADCX140_PWR_CFG, 0x00 }, @@ -158,6 +159,16 @@ static DECLARE_TLV_DB_SCALE(dig_vol_tlv, -10000, 50, 0); /* ADC gain. From 0 to 42 dB in 1 dB steps */ static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 100, 0); +/* DRE Level. From -12 dB to -66 dB in 1 dB steps */ +static DECLARE_TLV_DB_SCALE(dre_thresh_tlv, -6600, 100, 0); +/* DRE Max Gain. From 2 dB to 26 dB in 2 dB steps */ +static DECLARE_TLV_DB_SCALE(dre_gain_tlv, 200, 200, 0); + +/* AGC Level. From -6 dB to -36 dB in 2 dB steps */ +static DECLARE_TLV_DB_SCALE(agc_thresh_tlv, -3600, 200, 0); +/* AGC Max Gain. From 3 dB to 42 dB in 3 dB steps */ +static DECLARE_TLV_DB_SCALE(agc_gain_tlv, 300, 300, 0); + static const char * const resistor_text[] = { "2.5 kOhm", "10 kOhm", "20 kOhm" }; @@ -281,6 +292,18 @@ static const struct snd_kcontrol_new adcx140_dapm_ch3_en_switch = static const struct snd_kcontrol_new adcx140_dapm_ch4_en_switch = SOC_DAPM_SINGLE("Switch", ADCX140_ASI_OUT_CH_EN, 4, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch1_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH1_CFG0, 0, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch2_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH2_CFG0, 0, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch3_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH3_CFG0, 0, 1, 0); +static const struct snd_kcontrol_new adcx140_dapm_ch4_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_CH4_CFG0, 0, 1, 0); + +static const struct snd_kcontrol_new adcx140_dapm_dre_en_switch = + SOC_DAPM_SINGLE("Switch", ADCX140_DSP_CFG1, 3, 1, 0); + /* Output Mixer */ static const struct snd_kcontrol_new adcx140_output_mixer_controls[] = { SOC_DAPM_SINGLE("Digital CH1 Switch", 0, 0, 0, 0), @@ -361,6 +384,18 @@ static const struct snd_soc_dapm_widget adcx140_dapm_widgets[] = { SND_SOC_DAPM_SWITCH("CH4_ASI_EN", SND_SOC_NOPM, 0, 0, &adcx140_dapm_ch4_en_switch), + SND_SOC_DAPM_SWITCH("DRE_ENABLE", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_dre_en_switch), + + SND_SOC_DAPM_SWITCH("CH1_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch1_dre_en_switch), + SND_SOC_DAPM_SWITCH("CH2_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch2_dre_en_switch), + SND_SOC_DAPM_SWITCH("CH3_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch3_dre_en_switch), + SND_SOC_DAPM_SWITCH("CH4_DRE_EN", SND_SOC_NOPM, 0, 0, + &adcx140_dapm_ch4_dre_en_switch), + SND_SOC_DAPM_MUX("IN1 Analog Mic Resistor", SND_SOC_NOPM, 0, 0, in1_resistor_controls), SND_SOC_DAPM_MUX("IN2 Analog Mic Resistor", SND_SOC_NOPM, 0, 0, @@ -383,6 +418,16 @@ static const struct snd_soc_dapm_route adcx140_audio_map[] = { {"CH3_ASI_EN", "Switch", "CH3_ADC"}, {"CH4_ASI_EN", "Switch", "CH4_ADC"}, + {"DRE_ENABLE", "Switch", "CH1_DRE_EN"}, + {"DRE_ENABLE", "Switch", "CH2_DRE_EN"}, + {"DRE_ENABLE", "Switch", "CH3_DRE_EN"}, + {"DRE_ENABLE", "Switch", "CH4_DRE_EN"}, + + {"CH1_DRE_EN", "Switch", "CH1_ADC"}, + {"CH2_DRE_EN", "Switch", "CH2_ADC"}, + {"CH3_DRE_EN", "Switch", "CH3_ADC"}, + {"CH4_DRE_EN", "Switch", "CH4_ADC"}, + /* Mic input */ {"CH1_ADC", NULL, "MIC_GAIN_CTL_CH1"}, {"CH2_ADC", NULL, "MIC_GAIN_CTL_CH2"}, @@ -455,6 +500,16 @@ static const struct snd_kcontrol_new adcx140_snd_controls[] = { SOC_SINGLE_TLV("Analog CH4 Mic Gain Volume", ADCX140_CH1_CFG4, 2, 42, 0, adc_tlv), + SOC_SINGLE_TLV("DRE Threshold", ADCX140_DRE_CFG0, 4, 9, 0, + dre_thresh_tlv), + SOC_SINGLE_TLV("DRE Max Gain", ADCX140_DRE_CFG0, 0, 12, 0, + dre_gain_tlv), + + SOC_SINGLE_TLV("AGC Threshold", ADCX140_AGC_CFG0, 4, 15, 0, + agc_thresh_tlv), + SOC_SINGLE_TLV("AGC Max Gain", ADCX140_AGC_CFG0, 0, 13, 0, + agc_gain_tlv), + SOC_SINGLE_TLV("Digital CH1 Out Volume", ADCX140_CH1_CFG2, 0, 0xff, 0, dig_vol_tlv), SOC_SINGLE_TLV("Digital CH2 Out Volume", ADCX140_CH2_CFG2, diff --git a/sound/soc/codecs/tlv320adcx140.h b/sound/soc/codecs/tlv320adcx140.h index 66b1c3b33f1e..6d055e55909e 100644 --- a/sound/soc/codecs/tlv320adcx140.h +++ b/sound/soc/codecs/tlv320adcx140.h @@ -84,6 +84,7 @@ #define ADCX140_DSP_CFG0 0x6b #define ADCX140_DSP_CFG1 0x6c #define ADCX140_DRE_CFG0 0x6d +#define ADCX140_AGC_CFG0 0x70 #define ADCX140_IN_CH_EN 0x73 #define ADCX140_ASI_OUT_CH_EN 0x74 #define ADCX140_PWR_CFG 0x75 -- 2.25.0

2 3

Applied "ASoC: meson: aiu: fix semicolon.cocci warnings" to the asoc tree
by Mark Brown 24 Feb '20

24 Feb '20

The patch ASoC: meson: aiu: fix semicolon.cocci warnings has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 1640c8df0bbac9e5156132e24c0f0a932c2b2865 Mon Sep 17 00:00:00 2001 From: kbuild test robot <lkp(a)intel.com> Date: Sun, 23 Feb 2020 01:01:54 +0800 Subject: [PATCH] ASoC: meson: aiu: fix semicolon.cocci warnings sound/soc/meson/aiu-encoder-i2s.c:129:2-3: Unneeded semicolon Remove unneeded semicolon. Generated by: scripts/coccinelle/misc/semicolon.cocci Fixes: 3e25c44598aa ("ASoC: meson: aiu: add support for the Meson8 and Meson8b SoC families") Signed-off-by: kbuild test robot <lkp(a)intel.com> CC: Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> Link: https://lore.kernel.org/r/20200222170154.GA119396@e50d7db646c3 Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/meson/aiu-encoder-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/meson/aiu-encoder-i2s.c b/sound/soc/meson/aiu-encoder-i2s.c index cc73b5d5c2b7..832e22d275fe 100644 --- a/sound/soc/meson/aiu-encoder-i2s.c +++ b/sound/soc/meson/aiu-encoder-i2s.c @@ -126,7 +126,7 @@ static int aiu_encoder_i2s_set_legacy_div(struct snd_soc_component *component, default: dev_err(component->dev, "Unsupported i2s divider: %u\n", bs); return -EINVAL; - }; + } snd_soc_component_update_bits(component, AIU_CLK_CTRL, AIU_CLK_CTRL_I2S_DIV, -- 2.20.1

1 0

Re: [PATCH] docs: dt: fix several broken doc references
by Mark Brown 24 Feb '20

24 Feb '20

On Sun, Feb 23, 2020 at 09:59:53AM +0100, Mauro Carvalho Chehab wrote: > There are several DT doc references that require manual fixes. > I found 3 cases fixed on this patch: > > - directory named "binding/" instead of "bindings/"; > - .txt to .yaml renames; > - file renames (still on txt format); This seems like it should've been split up a bit. :/ Acked-by: Mark Brown <broonie(a)kernel.org>

1 0

[alsa-devel] [PATCH v9] ASoC: pxa: switch to new ac97 bus support
by Robert Jarzmik 24 Feb '20

24 Feb '20

Switch to the new ac97 bus support in sound/ac97 instead of the legacy snd_ac97 one. Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v8: fixed the trivial compilation error v7 was in https://patchwork.kernel.org/patch/9951919/ --- sound/arm/Kconfig | 1 - sound/soc/pxa/Kconfig | 5 ++--- sound/soc/pxa/pxa2xx-ac97.c | 48 +++++++++++++++++++++++---------------------- 3 files changed, 27 insertions(+), 27 deletions(-) diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 65171f6657a2..f1f25704fe52 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -36,7 +36,6 @@ endif # SND_ARM config SND_PXA2XX_LIB tristate - select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 select SND_DMAENGINE_PCM config SND_PXA2XX_LIB_AC97 diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 484ab3c2ad67..1a0b55beb282 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -20,13 +20,12 @@ config SND_MMP_SOC config SND_PXA2XX_AC97 tristate - select SND_AC97_CODEC config SND_PXA2XX_SOC_AC97 tristate - select AC97_BUS + select AC97_BUS_NEW select SND_PXA2XX_LIB_AC97 - select SND_SOC_AC97_BUS + select SND_SOC_AC97_BUS_NEW config SND_PXA2XX_SOC_I2S tristate diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 1b41c0f2a8fb..80e4757176d8 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -17,6 +17,7 @@ #include <linux/dmaengine.h> #include <linux/dma/pxa-dma.h> +#include <sound/ac97/controller.h> #include <sound/core.h> #include <sound/ac97_codec.h> #include <sound/soc.h> @@ -27,43 +28,35 @@ #include <mach/regs-ac97.h> #include <mach/audio.h> -static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_warm_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_warm_reset(); pxa2xx_ac97_finish_reset(); } -static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_cold_reset(struct ac97_controller *adrv) { pxa2xx_ac97_try_cold_reset(); pxa2xx_ac97_finish_reset(); } -static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97, - unsigned short reg) +static int pxa2xx_ac97_read_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg) { - int ret; - - ret = pxa2xx_ac97_read(ac97->num, reg); - if (ret < 0) - return 0; - else - return (unsigned short)(ret & 0xffff); + return pxa2xx_ac97_read(slot, reg); } -static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97, - unsigned short reg, unsigned short val) +static int pxa2xx_ac97_write_actrl(struct ac97_controller *adrv, int slot, + unsigned short reg, unsigned short val) { - int ret; - - ret = pxa2xx_ac97_write(ac97->num, reg, val); + return pxa2xx_ac97_write(slot, reg, val); } -static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { - .read = pxa2xx_ac97_legacy_read, - .write = pxa2xx_ac97_legacy_write, +static struct ac97_controller_ops pxa2xx_ac97_ops = { + .read = pxa2xx_ac97_read_actrl, + .write = pxa2xx_ac97_write_actrl, .warm_reset = pxa2xx_ac97_warm_reset, .reset = pxa2xx_ac97_cold_reset, }; @@ -219,6 +212,9 @@ static const struct snd_soc_component_driver pxa_ac97_component = { static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) { int ret; + struct ac97_controller *ctrl; + pxa2xx_audio_ops_t *pdata = pdev->dev.platform_data; + void **codecs_pdata; if (pdev->id != -1) { dev_err(&pdev->dev, "PXA2xx has only one AC97 port.\n"); @@ -231,10 +227,14 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) return ret; } - ret = snd_soc_set_ac97_ops(&pxa2xx_ac97_ops); - if (ret != 0) - return ret; + codecs_pdata = pdata ? pdata->codec_pdata : NULL; + ctrl = snd_ac97_controller_register(&pxa2xx_ac97_ops, &pdev->dev, + AC97_SLOTS_AVAILABLE_ALL, + codecs_pdata); + if (IS_ERR(ctrl)) + return PTR_ERR(ctrl); + platform_set_drvdata(pdev, ctrl); /* Punt most of the init to the SoC probe; we may need the machine * driver to do interesting things with the clocking to get us up * and running. @@ -245,8 +245,10 @@ static int pxa2xx_ac97_dev_probe(struct platform_device *pdev) static int pxa2xx_ac97_dev_remove(struct platform_device *pdev) { + struct ac97_controller *ctrl = platform_get_drvdata(pdev); + snd_soc_unregister_component(&pdev->dev); - snd_soc_set_ac97_ops(NULL); + snd_ac97_controller_unregister(ctrl); pxa2xx_ac97_hw_remove(pdev); return 0; } -- 2.11.0

2 1

Re: [PATCH] Intel: Skylake: Fix inconsistent IS_ERR and PTR_ERR
by Joe Perches 24 Feb '20

24 Feb '20

On Fri, 2020-02-21 at 18:11 +0800, Xu Wang wrote: > PTR_ERR should access the value just tested by IS_ERR. > In skl_clk_dev_probe(),it is inconsistent. [] > diff --git a/sound/soc/intel/skylake/skl-ssp-clk.c b/sound/soc/intel/skylake/skl-ssp-clk.c [] > @@ -384,7 +384,7 @@ static int skl_clk_dev_probe(struct platform_device *pdev) > &clks[i], clk_pdata, i); > > if (IS_ERR(data->clk[data->avail_clk_cnt])) { > - ret = PTR_ERR(data->clk[data->avail_clk_cnt++]); > + ret = PTR_ERR(data->clk[data->avail_clk_cnt]); NAK. This is not inconsistent and you are removing the ++ which is a post increment. Likely that is necessary. You could write the access and the increment as two separate statements if it confuses you.

4 3

Re: [PATCH] docs: dt: fix several broken doc references
by Miquel Raynal 24 Feb '20

24 Feb '20

Hi Mauro, Mauro Carvalho Chehab <mchehab+huawei(a)kernel.org> wrote on Sun, 23 Feb 2020 09:59:53 +0100: > There are several DT doc references that require manual fixes. > I found 3 cases fixed on this patch: > > - directory named "binding/" instead of "bindings/"; > - .txt to .yaml renames; > - file renames (still on txt format); > > Signed-off-by: Mauro Carvalho Chehab <mchehab+huawei(a)kernel.org> > --- > .../devicetree/bindings/mtd/cadence-nand-controller.txt | 2 +- > .../devicetree/bindings/net/brcm,bcm7445-switch-v4.0.txt | 2 +- > Documentation/devicetree/bindings/sound/st,stm32-sai.txt | 2 +- > Documentation/devicetree/bindings/sound/st,stm32-spdifrx.txt | 2 +- > Documentation/devicetree/bindings/spi/st,stm32-spi.yaml | 2 +- > MAINTAINERS | 4 ++-- > .../devicetree/bindings/net/wireless/siliabs,wfx.txt | 2 +- > 7 files changed, 8 insertions(+), 8 deletions(-) For the Cadence file, Reviewed-by: Miquel Raynal <miquel.raynal(a)bootlin.com> Thanks, Miquèl

1 0

[alsa-devel] [PATCH 0/8] ASoC: sun50i-codec-analog: Cleanup and power management
by Samuel Holland 23 Feb '20

23 Feb '20

This series performs some minor cleanup on the driver for the analog codec in the Allwinner A64, and hooks up the existing mute switches to DAPM widgets, in order to provide improved power management. Samuel Holland (8): ASoC: sun50i-codec-analog: Fix duplicate use of ADC enable bits ASoC: sun50i-codec-analog: Gate the amplifier clock during suspend ASoC: sun50i-codec-analog: Group and sort mixer routes ASoC: sun50i-codec-analog: Make headphone routes stereo ASoC: sun50i-codec-analog: Enable DAPM for headphone switch ASoC: sun50i-codec-analog: Make line out routes stereo ASoC: sun50i-codec-analog: Enable DAPM for line out switch ASoC: sun50i-codec-analog: Enable DAPM for earpiece switch sound/soc/sunxi/sun50i-codec-analog.c | 174 ++++++++++++++++---------- 1 file changed, 109 insertions(+), 65 deletions(-) -- 2.24.1

2 19

alsa-lib 1.2.2 breaks Shovel Knight, 1.2.1.2 is still working.
by GitHub issues - edited 22 Feb '20

22 Feb '20

alsa-project/alsa-lib issue #27 was edited from DarkWav: So, today after upgrading to alsa-lib 1.2.2 I noticed the game "Shovel Knight" does no longer launch. Any Ideas why this could be happening? Sadly steam or the game executable won't provide any usable debug info when looking at terminal output :( . Can I maybe use GDB to gather more info or is that useless in this case? Issue URL : https://github.com/alsa-project/alsa-lib/issues/27 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

alsa-lib 1.2.2 breaks Shovel Knight, 1.2.1.2 is still working.
by GitHub issues - opened 22 Feb '20

22 Feb '20

alsa-project/alsa-lib issue #27 was opened from DarkWav: So, today after upgrading to alsa-lib 1.2.2 I noticed the game "Shovel Knight" does no longer launch. Any Ideas why this could be happening? Sadly steam or the game executable won't provide any usable debug info :( . Issue URL : https://github.com/alsa-project/alsa-lib/issues/27 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[asoc:for-5.7 134/139] sound/soc/meson/aiu-encoder-i2s.c:129:2-3: Unneeded semicolon
by kbuild test robot 22 Feb '20

22 Feb '20

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.7 head: b38c4a8a0291c31a660cd77761202ebb18332fb7 commit: 3e25c44598aa44134207ad7b3c5ad6b586135777 [134/139] ASoC: meson: aiu: add support for the Meson8 and Meson8b SoC families If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> coccinelle warnings: (new ones prefixed by >>) >> sound/soc/meson/aiu-encoder-i2s.c:129:2-3: Unneeded semicolon Please review and possibly fold the followup patch. --- 0-DAY CI Kernel Test Service, Intel Corporation https://lists.01.org/hyperkitty/list/kbuild-all@lists.01.org

1 1

amixer not setting Master volume at a given percentage value (check with 85% to 90%)
by GitHub issues - opened 22 Feb '20

22 Feb '20

alsa-project/alsa-utils issue #35 was opened from immortalaxay: I am using Ubuntu 18.04 (alsa-utils/amixer version 1.1.3) and trying to set volume using amixer command but amixer not setting Master volume at the given percentage value. $ amixer -M -c PCH sset Master **85**% Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 81 [**83**%] [-4.50dB] [on] $ amixer -M -c PCH sset Master **86**% Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 82 [**85**%] [-3.75dB] [on] $ amixer -M -c PCH sset Master **87**% Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 82 [**85**%] [-3.75dB] [on] $ amixer -M -c PCH sset Master **88**% Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 82 [**85**%] [-3.75dB] [on] $ amixer -M -c PCH sset Master **89**% Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 83 [**88**%] [-3.00dB] [on] $ amixer -M -c PCH sset Master **90**% Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 87 Mono: Playback 83 [**88**%] [-3.00dB] [on] Issue URL : https://github.com/alsa-project/alsa-utils/issues/35 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[alsa-devel] [PATCH] ASoC: ti: Allocate dais dynamically for TDM and audio graph card
by Tony Lindgren 21 Feb '20

21 Feb '20

We can have multiple connections on a single McBSP instance configured with audio graph card when using TDM (Time Division Multiplexing). Let's allow that by configuring dais dynamically. See Documentation/devicetree/bindings/sound/audio-graph-card.txt and Documentation/devicetree/bindings/graph.txt for more details for multiple endpoints. I've tested this with droid4 where cpcap pmic and modem voice are both both wired to mcbsp3. I've also tested this on droid4 both with and without the pending modem audio codec driver that is waiting for n_gsm serdev dependencies to clear. Cc: Aaro Koskinen <aaro.koskinen(a)iki.fi> Cc: Arthur D. <spinal.by(a)gmail.com> Cc: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Cc: Merlijn Wajer <merlijn(a)wizzup.org> Cc: Pavel Machek <pavel(a)ucw.cz> Cc: Peter Ujfalusi <peter.ujfalusi(a)ti.com> Cc: Sebastian Reichel <sre(a)kernel.org> Signed-off-by: Tony Lindgren <tony(a)atomide.com> --- sound/soc/ti/omap-mcbsp-priv.h | 2 + sound/soc/ti/omap-mcbsp.c | 76 ++++++++++++++++++++++++---------- 2 files changed, 55 insertions(+), 23 deletions(-) diff --git a/sound/soc/ti/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h --- a/sound/soc/ti/omap-mcbsp-priv.h +++ b/sound/soc/ti/omap-mcbsp-priv.h @@ -262,6 +262,8 @@ struct omap_mcbsp { struct omap_mcbsp_platform_data *pdata; struct omap_mcbsp_st_data *st_data; struct omap_mcbsp_reg_cfg cfg_regs; + struct snd_soc_dai_driver *dais; + int dai_count; struct snd_dmaengine_dai_dma_data dma_data[2]; unsigned int dma_req[2]; int dma_op_mode; diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c --- a/sound/soc/ti/omap-mcbsp.c +++ b/sound/soc/ti/omap-mcbsp.c @@ -14,6 +14,7 @@ #include <linux/pm_runtime.h> #include <linux/of.h> #include <linux/of_device.h> +#include <linux/of_graph.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -1304,23 +1305,53 @@ static int omap_mcbsp_remove(struct snd_soc_dai *dai) return 0; } -static struct snd_soc_dai_driver omap_mcbsp_dai = { - .probe = omap_mcbsp_probe, - .remove = omap_mcbsp_remove, - .playback = { - .channels_min = 1, - .channels_max = 16, - .rates = OMAP_MCBSP_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, - }, - .capture = { - .channels_min = 1, - .channels_max = 16, - .rates = OMAP_MCBSP_RATES, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, - }, - .ops = &mcbsp_dai_ops, -}; +static int omap_mcbsp_init_dais(struct omap_mcbsp *mcbsp) +{ + struct device_node *np = mcbsp->dev->of_node; + int i; + + if (np) + mcbsp->dai_count = of_graph_get_endpoint_count(np); + + if (!mcbsp->dai_count) + mcbsp->dai_count = 1; + + mcbsp->dais = devm_kcalloc(mcbsp->dev, mcbsp->dai_count, + sizeof(*mcbsp->dais), GFP_KERNEL); + if (!mcbsp->dais) + return -ENOMEM; + + for (i = 0; i < mcbsp->dai_count; i++) { + struct snd_soc_dai_driver *dai = &mcbsp->dais[i]; + + dai->name = devm_kasprintf(mcbsp->dev, GFP_KERNEL, "%s-dai%i", + dev_name(mcbsp->dev), i); + + if (i == 0) { + dai->probe = omap_mcbsp_probe; + dai->remove = omap_mcbsp_remove; + dai->ops = &mcbsp_dai_ops; + } + dai->playback.channels_min = 1; + dai->playback.channels_max = 16; + dai->playback.rates = OMAP_MCBSP_RATES; + if (mcbsp->pdata->reg_size == 2) + dai->playback.formats = SNDRV_PCM_FMTBIT_S16_LE; + else + dai->playback.formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE; + dai->capture.channels_min = 1; + dai->capture.channels_max = 16; + dai->capture.rates = OMAP_MCBSP_RATES; + if (mcbsp->pdata->reg_size == 2) + dai->capture.formats = SNDRV_PCM_FMTBIT_S16_LE; + else + dai->capture.formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE; + } + + return 0; +} static const struct snd_soc_component_driver omap_mcbsp_component = { .name = "omap-mcbsp", @@ -1409,18 +1440,17 @@ static int asoc_mcbsp_probe(struct platform_device *pdev) mcbsp->dev = &pdev->dev; platform_set_drvdata(pdev, mcbsp); - ret = omap_mcbsp_init(pdev); + ret = omap_mcbsp_init_dais(mcbsp); if (ret) return ret; - if (mcbsp->pdata->reg_size == 2) { - omap_mcbsp_dai.playback.formats = SNDRV_PCM_FMTBIT_S16_LE; - omap_mcbsp_dai.capture.formats = SNDRV_PCM_FMTBIT_S16_LE; - } + ret = omap_mcbsp_init(pdev); + if (ret) + return ret; ret = devm_snd_soc_register_component(&pdev->dev, &omap_mcbsp_component, - &omap_mcbsp_dai, 1); + mcbsp->dais, mcbsp->dai_count); if (ret) return ret; -- 2.25.0

6 32

Applied "ASoC: meson: g12a: add toacodec dt-binding documentation" to the asoc tree
by Mark Brown 21 Feb '20

21 Feb '20

The patch ASoC: meson: g12a: add toacodec dt-binding documentation has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From bd56e593da19de22284951c33ce5c419258171bf Mon Sep 17 00:00:00 2001 From: Jerome Brunet <jbrunet(a)baylibre.com> Date: Fri, 21 Feb 2020 16:36:05 +0100 Subject: [PATCH] ASoC: meson: g12a: add toacodec dt-binding documentation Add the DT bindings and documentation of the internal audio DAC glue found on Amlogic g12a and sm1 SoC families Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Link: https://lore.kernel.org/r/20200221153607.1585499-2-jbrunet@baylibre.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/amlogic,g12a-toacodec.yaml | 51 +++++++++++++++++++ .../dt-bindings/sound/meson-g12a-toacodec.h | 10 ++++ 2 files changed, 61 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml create mode 100644 include/dt-bindings/sound/meson-g12a-toacodec.h diff --git a/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml b/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml new file mode 100644 index 000000000000..f778d3371fde --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml @@ -0,0 +1,51 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,g12a-toacodec.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic G12a Internal DAC Control Glue + +maintainers: + - Jerome Brunet <jbrunet(a)baylibre.com> + +properties: + $nodename: + pattern: "^audio-controller@.*" + + "#sound-dai-cells": + const: 1 + + compatible: + oneOf: + - items: + - const: + amlogic,g12a-toacodec + - items: + - enum: + - amlogic,sm1-toacodec + - const: + amlogic,g12a-toacodec + + reg: + maxItems: 1 + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - resets + +examples: + - | + #include <dt-bindings/reset/amlogic,meson-g12a-audio-reset.h> + + toacodec: audio-controller@740 { + compatible = "amlogic,g12a-toacodec"; + reg = <0x0 0x740 0x0 0x4>; + #sound-dai-cells = <1>; + resets = <&clkc_audio AUD_RESET_TOACODEC>; + }; diff --git a/include/dt-bindings/sound/meson-g12a-toacodec.h b/include/dt-bindings/sound/meson-g12a-toacodec.h new file mode 100644 index 000000000000..69d7a75592a2 --- /dev/null +++ b/include/dt-bindings/sound/meson-g12a-toacodec.h @@ -0,0 +1,10 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __DT_MESON_G12A_TOACODEC_H +#define __DT_MESON_G12A_TOACODEC_H + +#define TOACODEC_IN_A 0 +#define TOACODEC_IN_B 1 +#define TOACODEC_IN_C 2 +#define TOACODEC_OUT 3 + +#endif /* __DT_MESON_G12A_TOACODEC_H */ -- 2.20.1

1 0

[PATCH v2 0/3] ASoC: meson: g12a: add internal audio DAC support
by Jerome Brunet 21 Feb '20

21 Feb '20

Like the gxl, the Amlogic g12a and sm1 SoC families have a t9015 internal audio DAC. On these more recent SoCs, any of the 3 TDM outputs can be routed to the internal DAC. This routing is done by a small glue device called 'toacodec'. This patchset adds support for it. This was tested on the amlogic reference design g12a-u200. Changes since v1 [0]: * Fixup patch 2 which was left in an intermediate state after rebasing, missing part of the changes. Thanks to Sergey Bolshakov for reporting it. [0]: https://lore.kernel.org/r/20200221122242.1500093-1-jbrunet@baylibre.com Jerome Brunet (3): ASoC: meson: g12a: add toacodec dt-binding documentation ASoC: meson: g12a: add internal DAC glue driver ASoC: meson: axg-card: add toacodec support .../bindings/sound/amlogic,g12a-toacodec.yaml | 51 ++++ .../dt-bindings/sound/meson-g12a-toacodec.h | 10 + sound/soc/meson/Kconfig | 9 + sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-card.c | 3 +- sound/soc/meson/g12a-toacodec.c | 252 ++++++++++++++++++ 6 files changed, 326 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml create mode 100644 include/dt-bindings/sound/meson-g12a-toacodec.h create mode 100644 sound/soc/meson/g12a-toacodec.c -- 2.24.1

2 4

Applied "ASoC: meson: axg-card: add toacodec support" to the asoc tree
by Mark Brown 21 Feb '20

21 Feb '20

The patch ASoC: meson: axg-card: add toacodec support has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b38c4a8a0291c31a660cd77761202ebb18332fb7 Mon Sep 17 00:00:00 2001 From: Jerome Brunet <jbrunet(a)baylibre.com> Date: Fri, 21 Feb 2020 16:36:07 +0100 Subject: [PATCH] ASoC: meson: axg-card: add toacodec support Make sure the axg audio card driver recognise the dai_link as a codec-to-codec link if the cpu dai is the internal dac glue. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Link: https://lore.kernel.org/r/20200221153607.1585499-4-jbrunet@baylibre.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/meson/axg-card.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index 372dc696cc8e..48651631bdcf 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -303,7 +303,8 @@ static int axg_card_cpu_is_tdm_iface(struct device_node *np) static int axg_card_cpu_is_codec(struct device_node *np) { - return of_device_is_compatible(np, DT_PREFIX "g12a-tohdmitx"); + return of_device_is_compatible(np, DT_PREFIX "g12a-tohdmitx") || + of_device_is_compatible(np, DT_PREFIX "g12a-toacodec"); } static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, -- 2.20.1

1 0

Re: [PATCH 3/3] ASoC: meson: aiu: add support for the Meson8 and Meson8b SoC families
by Jerome Brunet 21 Feb '20

21 Feb '20

On Thu 20 Feb 2020 at 21:57, Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> wrote: > The AIU audio controller on the Meson8 and Meson8b SoC families is > compatible with the one found in the later GXBB family. Add compatible > strings for these two older SoC families so the driver can be loaded for > them. > > Instead of using the I2S divider from the AIU_CLK_CTRL_MORE register we > need to use the I2S divider from the AIU_CLK_CTRL register. This older > register is less flexible because it only supports four divider settings > (1, 2, 4, 8) compared to the AIU_CLK_CTRL_MORE register (which supports > dividers in the range 0..64). > > Signed-off-by: Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> > --- > sound/soc/meson/Kconfig | 2 +- > sound/soc/meson/aiu-encoder-i2s.c | 92 +++++++++++++++++++++++-------- > sound/soc/meson/aiu.c | 9 +++ > sound/soc/meson/aiu.h | 1 + > 4 files changed, 81 insertions(+), 23 deletions(-) > > diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig > index 897a706dcda0..d27e9180b453 100644 > --- a/sound/soc/meson/Kconfig > +++ b/sound/soc/meson/Kconfig > @@ -10,7 +10,7 @@ config SND_MESON_AIU > imply SND_SOC_HDMI_CODEC if DRM_MESON_DW_HDMI > help > Select Y or M to add support for the Audio output subsystem found > - in the Amlogic GX SoC family > + in the Amlogic Meson8, Meson8b and GX SoC families > > config SND_MESON_AXG_FIFO > tristate > diff --git a/sound/soc/meson/aiu-encoder-i2s.c b/sound/soc/meson/aiu-encoder-i2s.c > index 4900e38e7e49..cc73b5d5c2b7 100644 > --- a/sound/soc/meson/aiu-encoder-i2s.c > +++ b/sound/soc/meson/aiu-encoder-i2s.c > @@ -111,34 +111,40 @@ static int aiu_encoder_i2s_setup_desc(struct snd_soc_component *component, > return 0; > } > > -static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, > - struct snd_pcm_hw_params *params) > +static int aiu_encoder_i2s_set_legacy_div(struct snd_soc_component *component, > + struct snd_pcm_hw_params *params, > + unsigned int bs) > { > - struct aiu *aiu = snd_soc_component_get_drvdata(component); > - unsigned int srate = params_rate(params); > - unsigned int fs, bs; > - > - /* Get the oversampling factor */ > - fs = DIV_ROUND_CLOSEST(clk_get_rate(aiu->i2s.clks[MCLK].clk), srate); > + switch (bs) { > + case 1: > + case 2: > + case 4: > + case 8: > + /* These are the only valid legacy dividers */ > + break; > > - if (fs % 64) > + default: > + dev_err(component->dev, "Unsupported i2s divider: %u\n", bs); > return -EINVAL; > + }; Not major but this semicolon should be removed (Thx coccinelle!) > > - /* Send data MSB first */ > - snd_soc_component_update_bits(component, AIU_I2S_DAC_CFG, > - AIU_I2S_DAC_CFG_MSB_FIRST, > - AIU_I2S_DAC_CFG_MSB_FIRST); > + snd_soc_component_update_bits(component, AIU_CLK_CTRL, > + AIU_CLK_CTRL_I2S_DIV, > + FIELD_PREP(AIU_CLK_CTRL_I2S_DIV, > + __ffs(bs))); > > - /* Set bclk to lrlck ratio */ > - snd_soc_component_update_bits(component, AIU_CODEC_DAC_LRCLK_CTRL, > - AIU_CODEC_DAC_LRCLK_CTRL_DIV, > - FIELD_PREP(AIU_CODEC_DAC_LRCLK_CTRL_DIV, > - 64 - 1)); > + snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, > + AIU_CLK_CTRL_MORE_I2S_DIV, > + FIELD_PREP(AIU_CLK_CTRL_MORE_I2S_DIV, > + 0)); > > - /* Use CLK_MORE for mclk to bclk divider */ > - snd_soc_component_update_bits(component, AIU_CLK_CTRL, > - AIU_CLK_CTRL_I2S_DIV, 0); > + return 0; > +} > > +static int aiu_encoder_i2s_set_more_div(struct snd_soc_component *component, > + struct snd_pcm_hw_params *params, > + unsigned int bs) > +{ > /* > * NOTE: this HW is odd. > * In most configuration, the i2s divider is 'mclk / blck'. > @@ -146,7 +152,6 @@ static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, > * increased by 50% to get the correct output rate. > * No idea why ! > */ > - bs = fs / 64; > if (params_width(params) == 16 && params_channels(params) == 8) { > if (bs % 2) { > dev_err(component->dev, > @@ -156,11 +161,54 @@ static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, > bs += bs / 2; > } > > + /* Use CLK_MORE for mclk to bclk divider */ > + snd_soc_component_update_bits(component, AIU_CLK_CTRL, > + AIU_CLK_CTRL_I2S_DIV, > + FIELD_PREP(AIU_CLK_CTRL_I2S_DIV, 0)); > + > snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, > AIU_CLK_CTRL_MORE_I2S_DIV, > FIELD_PREP(AIU_CLK_CTRL_MORE_I2S_DIV, > bs - 1)); > > + return 0; > +} > + > +static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, > + struct snd_pcm_hw_params *params) > +{ > + struct aiu *aiu = snd_soc_component_get_drvdata(component); > + unsigned int srate = params_rate(params); > + unsigned int fs, bs; > + int ret; > + > + /* Get the oversampling factor */ > + fs = DIV_ROUND_CLOSEST(clk_get_rate(aiu->i2s.clks[MCLK].clk), srate); > + > + if (fs % 64) > + return -EINVAL; > + > + /* Send data MSB first */ > + snd_soc_component_update_bits(component, AIU_I2S_DAC_CFG, > + AIU_I2S_DAC_CFG_MSB_FIRST, > + AIU_I2S_DAC_CFG_MSB_FIRST); > + > + /* Set bclk to lrlck ratio */ > + snd_soc_component_update_bits(component, AIU_CODEC_DAC_LRCLK_CTRL, > + AIU_CODEC_DAC_LRCLK_CTRL_DIV, > + FIELD_PREP(AIU_CODEC_DAC_LRCLK_CTRL_DIV, > + 64 - 1)); > + > + bs = fs / 64; > + > + if (aiu->platform->has_clk_ctrl_more_i2s_div) > + ret = aiu_encoder_i2s_set_more_div(component, params, bs); > + else > + ret = aiu_encoder_i2s_set_legacy_div(component, params, bs); > + > + if (ret) > + return ret; > + > /* Make sure amclk is used for HDMI i2s as well */ > snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, > AIU_CLK_CTRL_MORE_HDMI_AMCLK, > diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c > index 38209312a8c3..dc35ca79021c 100644 > --- a/sound/soc/meson/aiu.c > +++ b/sound/soc/meson/aiu.c > @@ -351,15 +351,24 @@ static int aiu_remove(struct platform_device *pdev) > > static const struct aiu_platform_data aiu_gxbb_pdata = { > .has_acodec = false, > + .has_clk_ctrl_more_i2s_div = true, > }; > > static const struct aiu_platform_data aiu_gxl_pdata = { > .has_acodec = true, > + .has_clk_ctrl_more_i2s_div = true, > +}; > + > +static const struct aiu_platform_data aiu_meson8_pdata = { > + .has_acodec = false, > + .has_clk_ctrl_more_i2s_div = false, > }; > > static const struct of_device_id aiu_of_match[] = { > { .compatible = "amlogic,aiu-gxbb", .data = &aiu_gxbb_pdata }, > { .compatible = "amlogic,aiu-gxl", .data = &aiu_gxl_pdata }, > + { .compatible = "amlogic,aiu-meson8", .data = &aiu_meson8_pdata }, > + { .compatible = "amlogic,aiu-meson8b", .data = &aiu_meson8_pdata }, > {} > }; > MODULE_DEVICE_TABLE(of, aiu_of_match); > diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h > index ab003638d5e5..87aa19ac4af3 100644 > --- a/sound/soc/meson/aiu.h > +++ b/sound/soc/meson/aiu.h > @@ -29,6 +29,7 @@ struct aiu_interface { > > struct aiu_platform_data { > bool has_acodec; > + bool has_clk_ctrl_more_i2s_div; > }; > > struct aiu {

1 0

[PATCH 0/3] ASoC: meson: g12a: add internal audio DAC support
by Jerome Brunet 21 Feb '20

21 Feb '20

Like the gxl, the Amlogic g12a and sm1 SoC families have a t9015 internal audio DAC. On these more recent SoCs, any of the 3 TDM outputs can be routed to the internal DAC. This routing is done by a small glue device called 'toacodec'. This patchset adds support for it. Jerome Brunet (3): ASoC: meson: g12a: add toacodec dt-binding documentation ASoC: meson: g12a: add internal DAC glue driver ASoC: meson: axg-card: add toacodec support .../bindings/sound/amlogic,g12a-toacodec.yaml | 51 ++++ .../dt-bindings/sound/meson-g12a-toacodec.h | 10 + sound/soc/meson/Kconfig | 9 + sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-card.c | 3 +- sound/soc/meson/g12a-toacodec.c | 240 ++++++++++++++++++ 6 files changed, 314 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,g12a-toacodec.yaml create mode 100644 include/dt-bindings/sound/meson-g12a-toacodec.h create mode 100644 sound/soc/meson/g12a-toacodec.c -- 2.24.1

1 4

[PATCH for-next v3] ASoC: tas2562: Add support for digital volume control
by Dan Murphy 21 Feb '20

21 Feb '20

Add support for digital volume control. There is no dedicated register for volume control but instead there are 4. The values of the registers are determined with exponential floating point math. So a table was created with register values for 2dB step increments from -110dB to 0dB. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- v3 - Removed volume init in codec_probe. sound/soc/codecs/tas2562.c | 78 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2562.h | 6 ++- 2 files changed, 82 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index b2682c2360b6..d5e04030a0c1 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -26,6 +26,24 @@ #define TAS2562_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FORMAT_S32_LE) +/* DVC equation involves floating point math + * round(10^(volume in dB/20)*2^30) + * so create a lookup table for 2dB step + */ +static const unsigned int float_vol_db_lookup[] = { +0x00000d43, 0x000010b2, 0x00001505, 0x00001a67, 0x00002151, +0x000029f1, 0x000034cd, 0x00004279, 0x000053af, 0x0000695b, +0x0000695b, 0x0000a6fa, 0x0000d236, 0x000108a4, 0x00014d2a, +0x0001a36e, 0x00021008, 0x000298c0, 0x000344df, 0x00041d8f, +0x00052e5a, 0x000685c8, 0x00083621, 0x000a566d, 0x000d03a7, +0x0010624d, 0x0014a050, 0x0019f786, 0x0020b0bc, 0x0029279d, +0x0033cf8d, 0x004139d3, 0x00521d50, 0x00676044, 0x0082248a, +0x00a3d70a, 0x00ce4328, 0x0103ab3d, 0x0146e75d, 0x019b8c27, +0x02061b89, 0x028c423f, 0x03352529, 0x0409c2b0, 0x05156d68, +0x080e9f96, 0x0a24b062, 0x0cc509ab, 0x10137987, 0x143d1362, +0x197a967f, 0x2013739e, 0x28619ae9, 0x32d64617, 0x40000000 +}; + struct tas2562_data { struct snd_soc_component *component; struct gpio_desc *sdz_gpio; @@ -34,6 +52,7 @@ struct tas2562_data { struct i2c_client *client; int v_sense_slot; int i_sense_slot; + int volume_lvl; }; static int tas2562_set_bias_level(struct snd_soc_component *component, @@ -414,6 +433,50 @@ static int tas2562_dac_event(struct snd_soc_dapm_widget *w, return 0; } +static int tas2562_volume_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = tas2562->volume_lvl; + return 0; +} + +static int tas2562_volume_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; + u32 reg_val; + + reg_val = float_vol_db_lookup[ucontrol->value.integer.value[0]/2]; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG4, + (reg_val & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG3, + ((reg_val >> 8) & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG2, + ((reg_val >> 16) & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG1, + ((reg_val >> 24) & 0xff)); + if (ret) + return ret; + + tas2562->volume_lvl = ucontrol->value.integer.value[0]; + + return ret; +} + +/* Digital Volume Control. From 0 dB to -110 dB in 1 dB steps */ +static const DECLARE_TLV_DB_SCALE(dvc_tlv, -11000, 100, 0); + static DECLARE_TLV_DB_SCALE(tas2562_dac_tlv, 850, 50, 0); static const struct snd_kcontrol_new isense_switch = @@ -427,6 +490,17 @@ static const struct snd_kcontrol_new vsense_switch = static const struct snd_kcontrol_new tas2562_snd_controls[] = { SOC_SINGLE_TLV("Amp Gain Volume", TAS2562_PB_CFG1, 0, 0x1c, 0, tas2562_dac_tlv), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Volume Control", + .index = 0, + .tlv.p = dvc_tlv, + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_soc_info_volsw, + .get = tas2562_volume_control_get, + .put = tas2562_volume_control_put, + .private_value = SOC_SINGLE_VALUE(TAS2562_DVC_CFG1, 0, 110, 0, 0) , + }, }; static const struct snd_soc_dapm_widget tas2562_dapm_widgets[] = { @@ -517,6 +591,10 @@ static const struct reg_default tas2562_reg_defaults[] = { { TAS2562_PB_CFG1, 0x20 }, { TAS2562_TDM_CFG0, 0x09 }, { TAS2562_TDM_CFG1, 0x02 }, + { TAS2562_DVC_CFG1, 0x40 }, + { TAS2562_DVC_CFG2, 0x40 }, + { TAS2562_DVC_CFG3, 0x00 }, + { TAS2562_DVC_CFG4, 0x00 }, }; static const struct regmap_config tas2562_regmap_config = { diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 6f55ebcf19ea..28e75fc431d0 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -35,8 +35,10 @@ #define TAS2562_REV_ID TAS2562_REG(0, 0x7d) /* Page 2 */ -#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x01) -#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x02) +#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x0c) +#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x0d) +#define TAS2562_DVC_CFG3 TAS2562_REG(2, 0x0e) +#define TAS2562_DVC_CFG4 TAS2562_REG(2, 0x0f) #define TAS2562_RESET BIT(0) -- 2.25.0

2 1

[PATCH] ASoC: meson: g12a: add tohdmitx reset
by Jerome Brunet 21 Feb '20

21 Feb '20

Reset the g12a hdmi codec glue on probe. This ensure a sane startup state. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/meson/g12a-tohdmitx.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c index f8853f2fba08..9b2b59536ced 100644 --- a/sound/soc/meson/g12a-tohdmitx.c +++ b/sound/soc/meson/g12a-tohdmitx.c @@ -8,6 +8,7 @@ #include <linux/module.h> #include <sound/pcm_params.h> #include <linux/regmap.h> +#include <linux/reset.h> #include <sound/soc.h> #include <sound/soc-dai.h> @@ -245,6 +246,11 @@ static int g12a_tohdmitx_probe(struct platform_device *pdev) struct device *dev = &pdev->dev; void __iomem *regs; struct regmap *map; + int ret; + + ret = device_reset(dev); + if (ret) + return ret; regs = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(regs)) -- 2.24.1

2 1

Applied "ASoC: meson: aiu: Document Meson8 and Meson8b support in the dt-bindings" to the asoc tree
by Mark Brown 21 Feb '20

21 Feb '20

The patch ASoC: meson: aiu: Document Meson8 and Meson8b support in the dt-bindings has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 62209c9ad2ac29431e91392afb0bd6332ae4c33e Mon Sep 17 00:00:00 2001 From: Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> Date: Thu, 20 Feb 2020 21:57:09 +0100 Subject: [PATCH] ASoC: meson: aiu: Document Meson8 and Meson8b support in the dt-bindings The AIU audio output controller on the Meson8 and Meson8b SoC families is compatible with the one found in the GXBB family. Document the compatible string for these two older SoCs. Signed-off-by: Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> Reviewed-by: Jerome Brunet <jbrunet(a)baylibre.com> Link: https://lore.kernel.org/r/20200220205711.77953-2-martin.blumenstingl@google… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/amlogic,aiu.yaml | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml b/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml index 3ef7632dcb59..a61bccf915d8 100644 --- a/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml +++ b/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml @@ -21,6 +21,8 @@ properties: - enum: - amlogic,aiu-gxbb - amlogic,aiu-gxl + - amlogic,aiu-meson8 + - amlogic,aiu-meson8b - const: amlogic,aiu -- 2.20.1

1 0

Applied "ASoC: meson: aiu: introduce a struct for platform specific information" to the asoc tree
by Mark Brown 21 Feb '20

21 Feb '20

The patch ASoC: meson: aiu: introduce a struct for platform specific information has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From edc761805302db6d63916694d0cdb7468864a47a Mon Sep 17 00:00:00 2001 From: Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> Date: Thu, 20 Feb 2020 21:57:10 +0100 Subject: [PATCH] ASoC: meson: aiu: introduce a struct for platform specific information Introduce a struct aiu_platform_data to make the driver aware of platform specific information. Convert the existing check for the internal stereo audio codec (only available on GXL) to this new struct. Support for the 32-bit SoCs will need this as well because the AIU_CLK_CTRL_MORE register doesn't have the I2S divider bits (and we need to use the I2S divider from AIU_CLK_CTRL instead). Signed-off-by: Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> Reviewed-by: Jerome Brunet <jbrunet(a)baylibre.com> Link: https://lore.kernel.org/r/20200220205711.77953-3-martin.blumenstingl@google… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/meson/aiu.c | 19 ++++++++++++++++--- sound/soc/meson/aiu.h | 5 +++++ 2 files changed, 21 insertions(+), 3 deletions(-) diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index d3e2d40e9562..38209312a8c3 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -273,6 +273,11 @@ static int aiu_probe(struct platform_device *pdev) aiu = devm_kzalloc(dev, sizeof(*aiu), GFP_KERNEL); if (!aiu) return -ENOMEM; + + aiu->platform = device_get_match_data(dev); + if (!aiu->platform) + return -ENODEV; + platform_set_drvdata(pdev, aiu); ret = device_reset(dev); @@ -322,7 +327,7 @@ static int aiu_probe(struct platform_device *pdev) } /* Register the internal dac control component on gxl */ - if (of_device_is_compatible(dev->of_node, "amlogic,aiu-gxl")) { + if (aiu->platform->has_acodec) { ret = aiu_acodec_ctrl_register_component(dev); if (ret) { dev_err(dev, @@ -344,9 +349,17 @@ static int aiu_remove(struct platform_device *pdev) return 0; } +static const struct aiu_platform_data aiu_gxbb_pdata = { + .has_acodec = false, +}; + +static const struct aiu_platform_data aiu_gxl_pdata = { + .has_acodec = true, +}; + static const struct of_device_id aiu_of_match[] = { - { .compatible = "amlogic,aiu-gxbb", }, - { .compatible = "amlogic,aiu-gxl", }, + { .compatible = "amlogic,aiu-gxbb", .data = &aiu_gxbb_pdata }, + { .compatible = "amlogic,aiu-gxl", .data = &aiu_gxl_pdata }, {} }; MODULE_DEVICE_TABLE(of, aiu_of_match); diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h index 06a968c55728..ab003638d5e5 100644 --- a/sound/soc/meson/aiu.h +++ b/sound/soc/meson/aiu.h @@ -27,11 +27,16 @@ struct aiu_interface { int irq; }; +struct aiu_platform_data { + bool has_acodec; +}; + struct aiu { struct clk *pclk; struct clk *spdif_mclk; struct aiu_interface i2s; struct aiu_interface spdif; + const struct aiu_platform_data *platform; }; #define AIU_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ -- 2.20.1

1 0

Applied "ASoC: meson: aiu: add support for the Meson8 and Meson8b SoC families" to the asoc tree
by Mark Brown 21 Feb '20

21 Feb '20

The patch ASoC: meson: aiu: add support for the Meson8 and Meson8b SoC families has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3e25c44598aa44134207ad7b3c5ad6b586135777 Mon Sep 17 00:00:00 2001 From: Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> Date: Thu, 20 Feb 2020 21:57:11 +0100 Subject: [PATCH] ASoC: meson: aiu: add support for the Meson8 and Meson8b SoC families The AIU audio controller on the Meson8 and Meson8b SoC families is compatible with the one found in the later GXBB family. Add compatible strings for these two older SoC families so the driver can be loaded for them. Instead of using the I2S divider from the AIU_CLK_CTRL_MORE register we need to use the I2S divider from the AIU_CLK_CTRL register. This older register is less flexible because it only supports four divider settings (1, 2, 4, 8) compared to the AIU_CLK_CTRL_MORE register (which supports dividers in the range 0..64). Signed-off-by: Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> Reviewed-by: Jerome Brunet <jbrunet(a)baylibre.com> Link: https://lore.kernel.org/r/20200220205711.77953-4-martin.blumenstingl@google… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/meson/Kconfig | 2 +- sound/soc/meson/aiu-encoder-i2s.c | 92 +++++++++++++++++++++++-------- sound/soc/meson/aiu.c | 9 +++ sound/soc/meson/aiu.h | 1 + 4 files changed, 81 insertions(+), 23 deletions(-) diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig index 897a706dcda0..d27e9180b453 100644 --- a/sound/soc/meson/Kconfig +++ b/sound/soc/meson/Kconfig @@ -10,7 +10,7 @@ config SND_MESON_AIU imply SND_SOC_HDMI_CODEC if DRM_MESON_DW_HDMI help Select Y or M to add support for the Audio output subsystem found - in the Amlogic GX SoC family + in the Amlogic Meson8, Meson8b and GX SoC families config SND_MESON_AXG_FIFO tristate diff --git a/sound/soc/meson/aiu-encoder-i2s.c b/sound/soc/meson/aiu-encoder-i2s.c index 4900e38e7e49..cc73b5d5c2b7 100644 --- a/sound/soc/meson/aiu-encoder-i2s.c +++ b/sound/soc/meson/aiu-encoder-i2s.c @@ -111,34 +111,40 @@ static int aiu_encoder_i2s_setup_desc(struct snd_soc_component *component, return 0; } -static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, - struct snd_pcm_hw_params *params) +static int aiu_encoder_i2s_set_legacy_div(struct snd_soc_component *component, + struct snd_pcm_hw_params *params, + unsigned int bs) { - struct aiu *aiu = snd_soc_component_get_drvdata(component); - unsigned int srate = params_rate(params); - unsigned int fs, bs; - - /* Get the oversampling factor */ - fs = DIV_ROUND_CLOSEST(clk_get_rate(aiu->i2s.clks[MCLK].clk), srate); + switch (bs) { + case 1: + case 2: + case 4: + case 8: + /* These are the only valid legacy dividers */ + break; - if (fs % 64) + default: + dev_err(component->dev, "Unsupported i2s divider: %u\n", bs); return -EINVAL; + }; - /* Send data MSB first */ - snd_soc_component_update_bits(component, AIU_I2S_DAC_CFG, - AIU_I2S_DAC_CFG_MSB_FIRST, - AIU_I2S_DAC_CFG_MSB_FIRST); + snd_soc_component_update_bits(component, AIU_CLK_CTRL, + AIU_CLK_CTRL_I2S_DIV, + FIELD_PREP(AIU_CLK_CTRL_I2S_DIV, + __ffs(bs))); - /* Set bclk to lrlck ratio */ - snd_soc_component_update_bits(component, AIU_CODEC_DAC_LRCLK_CTRL, - AIU_CODEC_DAC_LRCLK_CTRL_DIV, - FIELD_PREP(AIU_CODEC_DAC_LRCLK_CTRL_DIV, - 64 - 1)); + snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, + AIU_CLK_CTRL_MORE_I2S_DIV, + FIELD_PREP(AIU_CLK_CTRL_MORE_I2S_DIV, + 0)); - /* Use CLK_MORE for mclk to bclk divider */ - snd_soc_component_update_bits(component, AIU_CLK_CTRL, - AIU_CLK_CTRL_I2S_DIV, 0); + return 0; +} +static int aiu_encoder_i2s_set_more_div(struct snd_soc_component *component, + struct snd_pcm_hw_params *params, + unsigned int bs) +{ /* * NOTE: this HW is odd. * In most configuration, the i2s divider is 'mclk / blck'. @@ -146,7 +152,6 @@ static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, * increased by 50% to get the correct output rate. * No idea why ! */ - bs = fs / 64; if (params_width(params) == 16 && params_channels(params) == 8) { if (bs % 2) { dev_err(component->dev, @@ -156,11 +161,54 @@ static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, bs += bs / 2; } + /* Use CLK_MORE for mclk to bclk divider */ + snd_soc_component_update_bits(component, AIU_CLK_CTRL, + AIU_CLK_CTRL_I2S_DIV, + FIELD_PREP(AIU_CLK_CTRL_I2S_DIV, 0)); + snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, AIU_CLK_CTRL_MORE_I2S_DIV, FIELD_PREP(AIU_CLK_CTRL_MORE_I2S_DIV, bs - 1)); + return 0; +} + +static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, + struct snd_pcm_hw_params *params) +{ + struct aiu *aiu = snd_soc_component_get_drvdata(component); + unsigned int srate = params_rate(params); + unsigned int fs, bs; + int ret; + + /* Get the oversampling factor */ + fs = DIV_ROUND_CLOSEST(clk_get_rate(aiu->i2s.clks[MCLK].clk), srate); + + if (fs % 64) + return -EINVAL; + + /* Send data MSB first */ + snd_soc_component_update_bits(component, AIU_I2S_DAC_CFG, + AIU_I2S_DAC_CFG_MSB_FIRST, + AIU_I2S_DAC_CFG_MSB_FIRST); + + /* Set bclk to lrlck ratio */ + snd_soc_component_update_bits(component, AIU_CODEC_DAC_LRCLK_CTRL, + AIU_CODEC_DAC_LRCLK_CTRL_DIV, + FIELD_PREP(AIU_CODEC_DAC_LRCLK_CTRL_DIV, + 64 - 1)); + + bs = fs / 64; + + if (aiu->platform->has_clk_ctrl_more_i2s_div) + ret = aiu_encoder_i2s_set_more_div(component, params, bs); + else + ret = aiu_encoder_i2s_set_legacy_div(component, params, bs); + + if (ret) + return ret; + /* Make sure amclk is used for HDMI i2s as well */ snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, AIU_CLK_CTRL_MORE_HDMI_AMCLK, diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index 38209312a8c3..dc35ca79021c 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -351,15 +351,24 @@ static int aiu_remove(struct platform_device *pdev) static const struct aiu_platform_data aiu_gxbb_pdata = { .has_acodec = false, + .has_clk_ctrl_more_i2s_div = true, }; static const struct aiu_platform_data aiu_gxl_pdata = { .has_acodec = true, + .has_clk_ctrl_more_i2s_div = true, +}; + +static const struct aiu_platform_data aiu_meson8_pdata = { + .has_acodec = false, + .has_clk_ctrl_more_i2s_div = false, }; static const struct of_device_id aiu_of_match[] = { { .compatible = "amlogic,aiu-gxbb", .data = &aiu_gxbb_pdata }, { .compatible = "amlogic,aiu-gxl", .data = &aiu_gxl_pdata }, + { .compatible = "amlogic,aiu-meson8", .data = &aiu_meson8_pdata }, + { .compatible = "amlogic,aiu-meson8b", .data = &aiu_meson8_pdata }, {} }; MODULE_DEVICE_TABLE(of, aiu_of_match); diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h index ab003638d5e5..87aa19ac4af3 100644 --- a/sound/soc/meson/aiu.h +++ b/sound/soc/meson/aiu.h @@ -29,6 +29,7 @@ struct aiu_interface { struct aiu_platform_data { bool has_acodec; + bool has_clk_ctrl_more_i2s_div; }; struct aiu { -- 2.20.1

1 0

Re: [PATCH 3/3] ASoC: meson: aiu: add support for the Meson8 and Meson8b SoC families
by Jerome Brunet 21 Feb '20

21 Feb '20

On Thu 20 Feb 2020 at 21:57, Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> wrote: > The AIU audio controller on the Meson8 and Meson8b SoC families is > compatible with the one found in the later GXBB family. Add compatible > strings for these two older SoC families so the driver can be loaded for > them. > > Instead of using the I2S divider from the AIU_CLK_CTRL_MORE register we > need to use the I2S divider from the AIU_CLK_CTRL register. This older > register is less flexible because it only supports four divider settings > (1, 2, 4, 8) compared to the AIU_CLK_CTRL_MORE register (which supports > dividers in the range 0..64). > > Signed-off-by: Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> Reviewed-by: Jerome Brunet <jbrunet(a)baylibre.com> > --- > sound/soc/meson/Kconfig | 2 +- > sound/soc/meson/aiu-encoder-i2s.c | 92 +++++++++++++++++++++++-------- > sound/soc/meson/aiu.c | 9 +++ > sound/soc/meson/aiu.h | 1 + > 4 files changed, 81 insertions(+), 23 deletions(-) > > diff --git a/sound/soc/meson/Kconfig b/sound/soc/meson/Kconfig > index 897a706dcda0..d27e9180b453 100644 > --- a/sound/soc/meson/Kconfig > +++ b/sound/soc/meson/Kconfig > @@ -10,7 +10,7 @@ config SND_MESON_AIU > imply SND_SOC_HDMI_CODEC if DRM_MESON_DW_HDMI > help > Select Y or M to add support for the Audio output subsystem found > - in the Amlogic GX SoC family > + in the Amlogic Meson8, Meson8b and GX SoC families > > config SND_MESON_AXG_FIFO > tristate > diff --git a/sound/soc/meson/aiu-encoder-i2s.c b/sound/soc/meson/aiu-encoder-i2s.c > index 4900e38e7e49..cc73b5d5c2b7 100644 > --- a/sound/soc/meson/aiu-encoder-i2s.c > +++ b/sound/soc/meson/aiu-encoder-i2s.c > @@ -111,34 +111,40 @@ static int aiu_encoder_i2s_setup_desc(struct snd_soc_component *component, > return 0; > } > > -static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, > - struct snd_pcm_hw_params *params) > +static int aiu_encoder_i2s_set_legacy_div(struct snd_soc_component *component, > + struct snd_pcm_hw_params *params, > + unsigned int bs) > { > - struct aiu *aiu = snd_soc_component_get_drvdata(component); > - unsigned int srate = params_rate(params); > - unsigned int fs, bs; > - > - /* Get the oversampling factor */ > - fs = DIV_ROUND_CLOSEST(clk_get_rate(aiu->i2s.clks[MCLK].clk), srate); > + switch (bs) { > + case 1: > + case 2: > + case 4: > + case 8: > + /* These are the only valid legacy dividers */ > + break; I wonder how it will work with the 8ch mode and 16bits but we can deal with this later on. > > - if (fs % 64) > + default: > + dev_err(component->dev, "Unsupported i2s divider: %u\n", bs); > return -EINVAL; > + }; > > - /* Send data MSB first */ > - snd_soc_component_update_bits(component, AIU_I2S_DAC_CFG, > - AIU_I2S_DAC_CFG_MSB_FIRST, > - AIU_I2S_DAC_CFG_MSB_FIRST); > + snd_soc_component_update_bits(component, AIU_CLK_CTRL, > + AIU_CLK_CTRL_I2S_DIV, > + FIELD_PREP(AIU_CLK_CTRL_I2S_DIV, > + __ffs(bs))); > > - /* Set bclk to lrlck ratio */ > - snd_soc_component_update_bits(component, AIU_CODEC_DAC_LRCLK_CTRL, > - AIU_CODEC_DAC_LRCLK_CTRL_DIV, > - FIELD_PREP(AIU_CODEC_DAC_LRCLK_CTRL_DIV, > - 64 - 1)); > + snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, > + AIU_CLK_CTRL_MORE_I2S_DIV, > + FIELD_PREP(AIU_CLK_CTRL_MORE_I2S_DIV, > + 0)); > > - /* Use CLK_MORE for mclk to bclk divider */ > - snd_soc_component_update_bits(component, AIU_CLK_CTRL, > - AIU_CLK_CTRL_I2S_DIV, 0); > + return 0; > +} > > +static int aiu_encoder_i2s_set_more_div(struct snd_soc_component *component, > + struct snd_pcm_hw_params *params, > + unsigned int bs) > +{ > /* > * NOTE: this HW is odd. > * In most configuration, the i2s divider is 'mclk / blck'. > @@ -146,7 +152,6 @@ static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, > * increased by 50% to get the correct output rate. > * No idea why ! > */ > - bs = fs / 64; > if (params_width(params) == 16 && params_channels(params) == 8) { > if (bs % 2) { > dev_err(component->dev, > @@ -156,11 +161,54 @@ static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, > bs += bs / 2; > } > > + /* Use CLK_MORE for mclk to bclk divider */ > + snd_soc_component_update_bits(component, AIU_CLK_CTRL, > + AIU_CLK_CTRL_I2S_DIV, > + FIELD_PREP(AIU_CLK_CTRL_I2S_DIV, 0)); > + > snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, > AIU_CLK_CTRL_MORE_I2S_DIV, > FIELD_PREP(AIU_CLK_CTRL_MORE_I2S_DIV, > bs - 1)); > > + return 0; > +} > + > +static int aiu_encoder_i2s_set_clocks(struct snd_soc_component *component, > + struct snd_pcm_hw_params *params) > +{ > + struct aiu *aiu = snd_soc_component_get_drvdata(component); > + unsigned int srate = params_rate(params); > + unsigned int fs, bs; > + int ret; > + > + /* Get the oversampling factor */ > + fs = DIV_ROUND_CLOSEST(clk_get_rate(aiu->i2s.clks[MCLK].clk), srate); > + > + if (fs % 64) > + return -EINVAL; > + > + /* Send data MSB first */ > + snd_soc_component_update_bits(component, AIU_I2S_DAC_CFG, > + AIU_I2S_DAC_CFG_MSB_FIRST, > + AIU_I2S_DAC_CFG_MSB_FIRST); > + > + /* Set bclk to lrlck ratio */ > + snd_soc_component_update_bits(component, AIU_CODEC_DAC_LRCLK_CTRL, > + AIU_CODEC_DAC_LRCLK_CTRL_DIV, > + FIELD_PREP(AIU_CODEC_DAC_LRCLK_CTRL_DIV, > + 64 - 1)); > + > + bs = fs / 64; > + > + if (aiu->platform->has_clk_ctrl_more_i2s_div) > + ret = aiu_encoder_i2s_set_more_div(component, params, bs); > + else > + ret = aiu_encoder_i2s_set_legacy_div(component, params, bs); > + > + if (ret) > + return ret; > + > /* Make sure amclk is used for HDMI i2s as well */ > snd_soc_component_update_bits(component, AIU_CLK_CTRL_MORE, > AIU_CLK_CTRL_MORE_HDMI_AMCLK, > diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c > index 38209312a8c3..dc35ca79021c 100644 > --- a/sound/soc/meson/aiu.c > +++ b/sound/soc/meson/aiu.c > @@ -351,15 +351,24 @@ static int aiu_remove(struct platform_device *pdev) > > static const struct aiu_platform_data aiu_gxbb_pdata = { > .has_acodec = false, > + .has_clk_ctrl_more_i2s_div = true, > }; > > static const struct aiu_platform_data aiu_gxl_pdata = { > .has_acodec = true, > + .has_clk_ctrl_more_i2s_div = true, > +}; > + > +static const struct aiu_platform_data aiu_meson8_pdata = { > + .has_acodec = false, > + .has_clk_ctrl_more_i2s_div = false, > }; > > static const struct of_device_id aiu_of_match[] = { > { .compatible = "amlogic,aiu-gxbb", .data = &aiu_gxbb_pdata }, > { .compatible = "amlogic,aiu-gxl", .data = &aiu_gxl_pdata }, > + { .compatible = "amlogic,aiu-meson8", .data = &aiu_meson8_pdata }, > + { .compatible = "amlogic,aiu-meson8b", .data = &aiu_meson8_pdata }, > {} > }; > MODULE_DEVICE_TABLE(of, aiu_of_match); > diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h > index ab003638d5e5..87aa19ac4af3 100644 > --- a/sound/soc/meson/aiu.h > +++ b/sound/soc/meson/aiu.h > @@ -29,6 +29,7 @@ struct aiu_interface { > > struct aiu_platform_data { > bool has_acodec; > + bool has_clk_ctrl_more_i2s_div; > }; > > struct aiu {

1 0

Re: [PATCH 2/3] ASoC: meson: aiu: introduce a struct for platform specific information
by Jerome Brunet 21 Feb '20

21 Feb '20

On Thu 20 Feb 2020 at 21:57, Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> wrote: > Introduce a struct aiu_platform_data to make the driver aware of > platform specific information. Convert the existing check for the > internal stereo audio codec (only available on GXL) to this new struct. > Support for the 32-bit SoCs will need this as well because the > AIU_CLK_CTRL_MORE register doesn't have the I2S divider bits (and we > need to use the I2S divider from AIU_CLK_CTRL instead). > > Signed-off-by: Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> Reviewed-by: Jerome Brunet <jbrunet(a)baylibre.com> > --- > sound/soc/meson/aiu.c | 19 ++++++++++++++++--- > sound/soc/meson/aiu.h | 5 +++++ > 2 files changed, 21 insertions(+), 3 deletions(-) > > diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c > index d3e2d40e9562..38209312a8c3 100644 > --- a/sound/soc/meson/aiu.c > +++ b/sound/soc/meson/aiu.c > @@ -273,6 +273,11 @@ static int aiu_probe(struct platform_device *pdev) > aiu = devm_kzalloc(dev, sizeof(*aiu), GFP_KERNEL); > if (!aiu) > return -ENOMEM; > + > + aiu->platform = device_get_match_data(dev); > + if (!aiu->platform) > + return -ENODEV; > + > platform_set_drvdata(pdev, aiu); > > ret = device_reset(dev); > @@ -322,7 +327,7 @@ static int aiu_probe(struct platform_device *pdev) > } > > /* Register the internal dac control component on gxl */ > - if (of_device_is_compatible(dev->of_node, "amlogic,aiu-gxl")) { > + if (aiu->platform->has_acodec) { > ret = aiu_acodec_ctrl_register_component(dev); > if (ret) { > dev_err(dev, > @@ -344,9 +349,17 @@ static int aiu_remove(struct platform_device *pdev) > return 0; > } > > +static const struct aiu_platform_data aiu_gxbb_pdata = { > + .has_acodec = false, > +}; > + > +static const struct aiu_platform_data aiu_gxl_pdata = { > + .has_acodec = true, > +}; > + > static const struct of_device_id aiu_of_match[] = { > - { .compatible = "amlogic,aiu-gxbb", }, > - { .compatible = "amlogic,aiu-gxl", }, > + { .compatible = "amlogic,aiu-gxbb", .data = &aiu_gxbb_pdata }, > + { .compatible = "amlogic,aiu-gxl", .data = &aiu_gxl_pdata }, > {} > }; > MODULE_DEVICE_TABLE(of, aiu_of_match); > diff --git a/sound/soc/meson/aiu.h b/sound/soc/meson/aiu.h > index 06a968c55728..ab003638d5e5 100644 > --- a/sound/soc/meson/aiu.h > +++ b/sound/soc/meson/aiu.h > @@ -27,11 +27,16 @@ struct aiu_interface { > int irq; > }; > > +struct aiu_platform_data { > + bool has_acodec; > +}; > + > struct aiu { > struct clk *pclk; > struct clk *spdif_mclk; > struct aiu_interface i2s; > struct aiu_interface spdif; > + const struct aiu_platform_data *platform; > }; > > #define AIU_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \

1 0

Re: [PATCH 1/3] ASoC: meson: aiu: Document Meson8 and Meson8b support in the dt-bindings
by Jerome Brunet 21 Feb '20

21 Feb '20

On Thu 20 Feb 2020 at 21:57, Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> wrote: > The AIU audio output controller on the Meson8 and Meson8b SoC families > is compatible with the one found in the GXBB family. Document the > compatible string for these two older SoCs. > > Signed-off-by: Martin Blumenstingl <martin.blumenstingl(a)googlemail.com> Reviewed-by: Jerome Brunet <jbrunet(a)baylibre.com> > --- > Documentation/devicetree/bindings/sound/amlogic,aiu.yaml | 2 ++ > 1 file changed, 2 insertions(+) > > diff --git a/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml b/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml > index 3ef7632dcb59..a61bccf915d8 100644 > --- a/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml > +++ b/Documentation/devicetree/bindings/sound/amlogic,aiu.yaml > @@ -21,6 +21,8 @@ properties: > - enum: > - amlogic,aiu-gxbb > - amlogic,aiu-gxl > + - amlogic,aiu-meson8 > + - amlogic,aiu-meson8b > - const: > amlogic,aiu

1 0

[PATCH 5.5 241/399] ASoC: txx9: Remove unused rtd variable
by Greg Kroah-Hartman 21 Feb '20

21 Feb '20

From: Paul Burton <paulburton(a)kernel.org> [ Upstream commit ec0f6a4c4a987aa20b2e77e0db2ae555276e45e6 ] Commit a857e073ffc6 ("ASoC: txx9: txx9aclc: remove snd_pcm_ops") removed the last use of the rtd variable but didn't remove its definition, leading to the following warning/error for MIPS rbtx49xx_defconfig builds: sound/soc/txx9/txx9aclc.c: In function 'txx9aclc_pcm_hw_params': sound/soc/txx9/txx9aclc.c:54:30: error: unused variable 'rtd' [-Werror=unused-variable] struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); ^~~ Resolve this by removing the unused variable. Signed-off-by: Paul Burton <paulburton(a)kernel.org> Fixes: a857e073ffc6 ("ASoC: txx9: txx9aclc: remove snd_pcm_ops") Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Link: https://lore.kernel.org/r/20200109191422.334516-1-paulburton@kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/txx9/txx9aclc.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 33c78d33e5a1d..9a55926ebf07b 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -51,7 +51,6 @@ static int txx9aclc_pcm_hw_params(struct snd_soc_component *component, struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { - struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; struct txx9aclc_dmadata *dmadata = runtime->private_data; int ret; -- 2.20.1

1 0

[alsa-devel] [PATCH v2] ASoC: dt-bindings: simple-card: switch to yaml base Documentation
by Kuninori Morimoto 21 Feb '20

21 Feb '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - add missing setup .../devicetree/bindings/sound/simple-card.txt | 351 ------------- .../bindings/sound/simple-card.yaml | 484 ++++++++++++++++++ 2 files changed, 484 insertions(+), 351 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/simple-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-card.yaml diff --git a/Documentation/devicetree/bindings/sound/simple-card.txt b/Documentation/devicetree/bindings/sound/simple-card.txt deleted file mode 100644 index 79954cd6e37b..000000000000 --- a/Documentation/devicetree/bindings/sound/simple-card.txt +++ /dev/null @@ -1,351 +0,0 @@ -Simple-Card: - -Simple-Card specifies audio DAI connections of SoC <-> codec. - -Required properties: - -- compatible : "simple-audio-card" - -Optional properties: - -- simple-audio-card,name : User specified audio sound card name, one string - property. -- simple-audio-card,widgets : Please refer to widgets.txt. -- simple-audio-card,routing : A list of the connections between audio components. - Each entry is a pair of strings, the first being the - connection's sink, the second being the connection's - source. -- simple-audio-card,mclk-fs : Multiplication factor between stream rate and codec - mclk. When defined, mclk-fs property defined in - dai-link sub nodes are ignored. -- simple-audio-card,hp-det-gpio : Reference to GPIO that signals when - headphones are attached. -- simple-audio-card,mic-det-gpio : Reference to GPIO that signals when - a microphone is attached. -- simple-audio-card,aux-devs : List of phandles pointing to auxiliary devices, such - as amplifiers, to be added to the sound card. -- simple-audio-card,pin-switches : List of strings containing the widget names for - which pin switches must be created. - -Optional subnodes: - -- simple-audio-card,dai-link : Container for dai-link level - properties and the CPU and CODEC - sub-nodes. This container may be - omitted when the card has only one - DAI link. See the examples and the - section below. - -Dai-link subnode properties and subnodes: - -If dai-link subnode is omitted and the subnode properties are directly -under "sound"-node the subnode property and subnode names have to be -prefixed with "simple-audio-card,"-prefix. - -Required dai-link subnodes: - -- cpu : CPU sub-node -- codec : CODEC sub-node - -Optional dai-link subnode properties: - -- format : CPU/CODEC common audio format. - "i2s", "right_j", "left_j" , "dsp_a" - "dsp_b", "ac97", "pdm", "msb", "lsb" -- frame-master : Indicates dai-link frame master. - phandle to a cpu or codec subnode. -- bitclock-master : Indicates dai-link bit clock master. - phandle to a cpu or codec subnode. -- bitclock-inversion : bool property. Add this if the - dai-link uses bit clock inversion. -- frame-inversion : bool property. Add this if the - dai-link uses frame clock inversion. -- mclk-fs : Multiplication factor between stream - rate and codec mclk, applied only for - the dai-link. - -For backward compatibility the frame-master and bitclock-master -properties can be used as booleans in codec subnode to indicate if the -codec is the dai-link frame or bit clock master. In this case there -should be no dai-link node, the same properties should not be present -at sound-node level, and the bitclock-inversion and frame-inversion -properties should also be placed in the codec node if needed. - -Required CPU/CODEC subnodes properties: - -- sound-dai : phandle and port of CPU/CODEC - -Optional CPU/CODEC subnodes properties: - -- dai-tdm-slot-num : Please refer to tdm-slot.txt. -- dai-tdm-slot-width : Please refer to tdm-slot.txt. -- clocks / system-clock-frequency : specify subnode's clock if needed. - it can be specified via "clocks" if system has - clock node (= common clock), or "system-clock-frequency" - (if system doens't support common clock) - If a clock is specified, it is - enabled with clk_prepare_enable() - in dai startup() and disabled with - clk_disable_unprepare() in dai - shutdown(). - If a clock is specified and a - multiplication factor is given with - mclk-fs, the clock will be set to the - calculated mclk frequency when the - stream starts. -- system-clock-direction-out : specifies clock direction as 'out' on - initialization. It is useful for some aCPUs with - fixed clocks. - -------------------------------------------- -Example 1 - single DAI link: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "VF610-Tower-Sound-Card"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dailink0_master>; - simple-audio-card,frame-master = <&dailink0_master>; - simple-audio-card,widgets = - "Microphone", "Microphone Jack", - "Headphone", "Headphone Jack", - "Speaker", "External Speaker"; - simple-audio-card,routing = - "MIC_IN", "Microphone Jack", - "Headphone Jack", "HP_OUT", - "External Speaker", "LINE_OUT"; - - simple-audio-card,cpu { - sound-dai = <&sh_fsi2 0>; - }; - - dailink0_master: simple-audio-card,codec { - sound-dai = <&ak4648>; - clocks = <&osc>; - }; -}; - -&i2c0 { - ak4648: ak4648@12 { - #sound-dai-cells = <0>; - compatible = "asahi-kasei,ak4648"; - reg = <0x12>; - }; -}; - -sh_fsi2: sh_fsi2@ec230000 { - #sound-dai-cells = <1>; - compatible = "renesas,sh_fsi2"; - reg = <0xec230000 0x400>; - interrupt-parent = <&gic>; - interrupts = <0 146 0x4>; -}; - -------------------------------------------- -Example 2 - many DAI links: -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - simple-audio-card,name = "Cubox Audio"; - - simple-audio-card,dai-link@0 { /* I2S - HDMI */ - reg = <0>; - format = "i2s"; - cpu { - sound-dai = <&audio1 0>; - }; - codec { - sound-dai = <&tda998x 0>; - }; - }; - - simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ - reg = <1>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&tda998x 1>; - }; - }; - - simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ - reg = <2>; - cpu { - sound-dai = <&audio1 1>; - }; - codec { - sound-dai = <&spdif_codec>; - }; - }; -}; - -------------------------------------------- -Example 3 - route audio from IMX6 SSI2 through TLV320DAC3100 codec -through TPA6130A2 amplifier to headphones: -------------------------------------------- - -&i2c0 { - codec: tlv320dac3100@18 { - compatible = "ti,tlv320dac3100"; - ... - } - - amp: tpa6130a2@60 { - compatible = "ti,tpa6130a2"; - ... - } -} - -sound { - compatible = "simple-audio-card"; - ... - simple-audio-card,widgets = - "Headphone", "Headphone Jack"; - simple-audio-card,routing = - "Headphone Jack", "HPLEFT", - "Headphone Jack", "HPRIGHT", - "LEFTIN", "HPL", - "RIGHTIN", "HPR"; - simple-audio-card,aux-devs = <&amp>; - simple-audio-card,cpu { - sound-dai = <&ssi2>; - }; - simple-audio-card,codec { - sound-dai = <&codec>; - clocks = ... - }; -}; - -------------------------------------------- -Example 4. Sampling Rate Conversion -------------------------------------------- - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - simple-audio-card,convert-rate = <48000>; - - simple-audio-card,prefix = "ak4642"; - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "DAI0 Capture", "ak4642 Capture"; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&ak4643>; - system-clock-frequency = <11289600>; - }; -}; - -------------------------------------------- -Example 5. 2 CPU 1 Codec (Mixing) -------------------------------------------- -sound { - compatible = "simple-audio-card"; - - simple-audio-card,name = "rsnd-ak4643"; - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&dpcmcpu>; - simple-audio-card,frame-master = <&dpcmcpu>; - - simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", - "ak4642 Playback", "DAI1 Playback"; - - dpcmcpu: cpu@0 { - sound-dai = <&rcar_sound 0>; - }; - - cpu@1 { - sound-dai = <&rcar_sound 1>; - }; - - codec { - prefix = "ak4642"; - sound-dai = <&ak4643>; - clocks = <&audio_clock>; - }; -}; - -------------------------------------------- -Example 6 - many DAI links with DPCM: -------------------------------------------- - -CPU0 ------ ak4613 -CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ -CPU2 --/ /* DPCM 3ch/4ch */ -CPU3 --/ /* DPCM 5ch/6ch */ -CPU4 --/ /* DPCM 7ch/8ch */ -CPU5 ------ PCM3168A-c - -sound { - compatible = "simple-audio-card"; - - simple-audio-card,routing = - "pcm3168a Playback", "DAI1 Playback", - "pcm3168a Playback", "DAI2 Playback", - "pcm3168a Playback", "DAI3 Playback", - "pcm3168a Playback", "DAI4 Playback"; - - simple-audio-card,dai-link@0 { - format = "left_j"; - bitclock-master = <&sndcpu0>; - frame-master = <&sndcpu0>; - - sndcpu0: cpu { - sound-dai = <&rcar_sound 0>; - }; - codec { - sound-dai = <&ak4613>; - }; - }; - simple-audio-card,dai-link@1 { - format = "i2s"; - bitclock-master = <&sndcpu1>; - frame-master = <&sndcpu1>; - - convert-channels = <8>; /* TDM Split */ - - sndcpu1: cpu@0 { - sound-dai = <&rcar_sound 1>; - }; - cpu@1 { - sound-dai = <&rcar_sound 2>; - }; - cpu@2 { - sound-dai = <&rcar_sound 3>; - }; - cpu@3 { - sound-dai = <&rcar_sound 4>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - dai-tdm-slot-num = <8>; - sound-dai = <&pcm3168a 0>; - }; - }; - simple-audio-card,dai-link@2 { - format = "i2s"; - bitclock-master = <&sndcpu2>; - frame-master = <&sndcpu2>; - - sndcpu2: cpu { - sound-dai = <&rcar_sound 5>; - }; - codec { - mclk-fs = <512>; - prefix = "pcm3168a"; - sound-dai = <&pcm3168a 1>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/simple-card.yaml b/Documentation/devicetree/bindings/sound/simple-card.yaml new file mode 100644 index 000000000000..0a2a2c878c4a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/simple-card.yaml @@ -0,0 +1,484 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/simple-card.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Simple Audio Card Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +definitions: + + frame-master: + description: Indicates dai-link frame master. + $ref: /schemas/types.yaml#/definitions/phandle-array + + bitclock-master: + description: Indicates dai-link bit clock master + $ref: /schemas/types.yaml#/definitions/phandle-array + + frame-inversion: + description: dai-link uses frame clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + bitclock-inversion: + description: dai-link uses bit clock inversion + $ref: /schemas/types.yaml#/definitions/flag + + dai-tdm-slot-num: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + dai-tdm-slot-width: + description: see tdm-slot.txt. + $ref: /schemas/types.yaml#/definitions/uint32 + + clocks: + description: | + If a clock is specified and a multiplication factor is given with + mclk-fs, the clock will be set to the calculated mclk frequency + when the stream starts. + $ref: /schemas/types.yaml#/definitions/phandle-array + + system-clock-frequency: + description: | + If a clock is specified and a multiplication factor is given with + mclk-fs, the clock will be set to the calculated mclk frequency + when the stream starts. + $ref: /schemas/types.yaml#/definitions/uint32 + + system-clock-direction-out: + description: | + specifies clock direction as 'out' on initialization. + It is useful for some aCPUs with fixed clocks. + $ref: /schemas/types.yaml#/definitions/flag + + mclk-fs: + description: | + Multiplication factor between stream rate and codec mclk. + When defined, mclk-fs property defined in dai-link sub nodes are ignored. + $ref: /schemas/types.yaml#/definitions/uint32 + + aux-devs: + description: | + List of phandles pointing to auxiliary devices, such + as amplifiers, to be added to the sound card. + $ref: /schemas/types.yaml#/definitions/phandle-array + + convert-rate: + description: CPU to Codec rate convert. + $ref: /schemas/types.yaml#/definitions/uint32 + + convert-channels: + description: CPU to Codec rate channels. + $ref: /schemas/types.yaml#/definitions/uint32 + + prefix: + description: "device name prefix" + $ref: /schemas/types.yaml#/definitions/string + + label: + description: "device name" + $ref: /schemas/types.yaml#/definitions/string + + routing: + description: | + A list of the connections between audio components. + Each entry is a pair of strings, the first being the + connection's sink, the second being the connection's source. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + widgets: + description: User specified audio sound widgets. + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + + pin-switches: + description: the widget names for which pin switches must be created. + $ref: /schemas/types.yaml#/definitions/string-array + + format: + description: audio format. + items: + enum: + - i2s + - right_j + - left_j + - dsp_a + - dsp_b + - ac97 + - pdm + - msb + - lsb + + dai: + type: object + properties: + sound-dai: + $ref: /schemas/types.yaml#/definitions/phandle-array + + # common properties + mclk-fs: + $ref: "#/definitions/mclk-fs" + prefix: + $ref: "#/definitions/prefix" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + frame-master: + $ref: /schemas/types.yaml#/definitions/flag + bitclock-master: + $ref: /schemas/types.yaml#/definitions/flag + + dai-tdm-slot-num: + $ref: "#/definitions/dai-tdm-slot-num" + dai-tdm-slot-width: + $ref: "#/definitions/dai-tdm-slot-width" + clocks: + $ref: "#/definitions/clocks" + system-clock-frequency: + $ref: "#/definitions/system-clock-frequency" + system-clock-direction-out: + $ref: "#/definitions/system-clock-direction-out" + required: + - sound-dai + +properties: + compatible: + contains: + enum: + - simple-audio-card + - simple-scu-audio-card + + "#address-cells": true + "#size-cells": true + + label: + $ref: "#/definitions/label" + + simple-audio-card,name: + description: User specified audio sound card name. + $ref: /schemas/types.yaml#/definitions/string + +# use patternProperties to use "#definitions/xxx" +patternProperties: + "^simple-audio-card,widgets$": + $ref: "#/definitions/widgets" + "^simple-audio-card,routing$": + $ref: "#/definitions/routing" + "^simple-audio-card,cpu(@.*)?": + $ref: "#/definitions/dai" + "^simple-audio-card,codec(@.*)?": + $ref: "#/definitions/dai" + + # common properties + "^simple-audio-card,frame-master$": + $ref: "#/definitions/frame-master" + "^simple-audio-card,bitclock-master$": + $ref: "#/definitions/bitclock-master" + "^simple-audio-card,frame-inversion$": + $ref: "#/definitions/frame-inversion" + "^simple-audio-card,bitclock-inversion$": + $ref: "#/definitions/bitclock-inversion" + "^simple-audio-card,format$": + $ref: "#/definitions/format" + "^simple-audio-card,mclk-fs$": + $ref: "#/definitions/mclk-fs" + "^simple-audio-card,aux-devs$": + $ref: "#/definitions/aux-devs" + "^simple-audio-card,convert-rate$": + $ref: "#/definitions/convert-rate" + "^simple-audio-card,convert-channels$": + $ref: "#/definitions/convert-channels" + "^simple-audio-card,prefix$": + $ref: "#/definitions/prefix" + "^simple-audio-card,pin-switches$": + $ref: "#/definitions/pin-switches" +# use common *-gpio for +# simple-audio-card,hp-det-gpio: +# simple-audio-card,mic-det-gpio: + + "^simple-audio-card,dai-link(@.*)?": + description: | + Container for dai-link level properties and the CPU and CODEC sub-nodes. + This container may be omitted when the card has only one DAI link. + type: object + properties: + reg: + $ref: /schemas/types.yaml#/definitions/uint32 + + # common properties + frame-master: + $ref: "#/definitions/frame-master" + bitclock-master: + $ref: "#/definitions/bitclock-master" + frame-inversion: + $ref: "#/definitions/frame-inversion" + bitclock-inversion: + $ref: "#/definitions/bitclock-inversion" + format: + $ref: "#/definitions/format" + mclk-fs: + $ref: "#/definitions/mclk-fs" + aux-devs: + $ref: "#/definitions/aux-devs" + convert-rate: + $ref: "#/definitions/convert-rate" + convert-channels: + $ref: "#/definitions/convert-channels" + prefix: + $ref: "#/definitions/prefix" + pin-switches: + $ref: "#/definitions/pin-switches" + # use common *-gpio for + # hp-det-gpio: + # mic-det-gpio: + + patternProperties: + "^cpu(@.*)?": + $ref: "#/definitions/dai" + "^codec(@.*)?": + $ref: "#/definitions/dai" + additionalProperties: false + +required: + - compatible + +additionalProperties: false + +examples: +#-------------------- +# single DAI link +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "VF610-Tower-Sound-Card"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dailink0_master>; + simple-audio-card,frame-master = <&dailink0_master>; + simple-audio-card,widgets = + "Microphone", "Microphone Jack", + "Headphone", "Headphone Jack", + "Speaker", "External Speaker"; + simple-audio-card,routing = + "MIC_IN", "Microphone Jack", + "Headphone Jack", "HP_OUT", + "External Speaker", "LINE_OUT"; + + simple-audio-card,cpu { + sound-dai = <&sh_fsi2 0>; + }; + + dailink0_master: simple-audio-card,codec { + sound-dai = <&ak4648>; + clocks = <&osc>; + }; + }; + +#-------------------- +# Multi DAI links +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "Cubox Audio"; + + #address-cells = <1>; + #size-cells = <0>; + + simple-audio-card,dai-link@0 { /* I2S - HDMI */ + reg = <0>; + format = "i2s"; + cpu { + sound-dai = <&audio0>; + }; + codec { + sound-dai = <&tda998x0>; + }; + }; + + simple-audio-card,dai-link@1 { /* S/PDIF - HDMI */ + reg = <1>; + cpu { + sound-dai = <&audio1>; + }; + codec { + sound-dai = <&tda998x1>; + }; + }; + + simple-audio-card,dai-link@2 { /* S/PDIF - S/PDIF */ + reg = <2>; + cpu { + sound-dai = <&audio2>; + }; + codec { + sound-dai = <&spdif_codec>; + }; + }; + }; + +#-------------------- +# route audio from IMX6 SSI2 through TLV320DAC3100 codec +# through TPA6130A2 amplifier to headphones: +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,widgets = + "Headphone", "Headphone Jack"; + simple-audio-card,routing = + "Headphone Jack", "HPLEFT", + "Headphone Jack", "HPRIGHT", + "LEFTIN", "HPL", + "RIGHTIN", "HPR"; + simple-audio-card,aux-devs = <&amp>; + simple-audio-card,cpu { + sound-dai = <&ssi2>; + }; + simple-audio-card,codec { + sound-dai = <&codec>; + clocks = <&clocks>; + }; + }; + +#-------------------- +# Sampling Rate Conversion +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&sndcodec>; + simple-audio-card,frame-master = <&sndcodec>; + + simple-audio-card,convert-rate = <48000>; + + simple-audio-card,prefix = "ak4642"; + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "DAI0 Capture", "ak4642 Capture"; + + sndcpu: simple-audio-card,cpu { + sound-dai = <&rcar_sound>; + }; + + sndcodec: simple-audio-card,codec { + sound-dai = <&ak4643>; + system-clock-frequency = <11289600>; + }; + }; + +#-------------------- +# 2 CPU 1 Codec (Mixing) +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,name = "rsnd-ak4643"; + simple-audio-card,format = "left_j"; + simple-audio-card,bitclock-master = <&dpcmcpu>; + simple-audio-card,frame-master = <&dpcmcpu>; + + simple-audio-card,convert-rate = <48000>; + simple-audio-card,convert-channels = <2>; + + simple-audio-card,routing = "ak4642 Playback", "DAI0 Playback", + "ak4642 Playback", "DAI1 Playback"; + + dpcmcpu: simple-audio-card,cpu@0 { + sound-dai = <&rcar_sound 0>; + }; + + simple-audio-card,cpu@1 { + sound-dai = <&rcar_sound 1>; + }; + + simple-audio-card,codec { + prefix = "ak4642"; + sound-dai = <&ak4643>; + clocks = <&audio_clock>; + }; + }; + +#-------------------- +# Multi DAI links with DPCM: +# +# CPU0 ------ ak4613 +# CPU1 ------ PCM3168A-p /* DPCM 1ch/2ch */ +# CPU2 --/ /* DPCM 3ch/4ch */ +# CPU3 --/ /* DPCM 5ch/6ch */ +# CPU4 --/ /* DPCM 7ch/8ch */ +# CPU5 ------ PCM3168A-c +#-------------------- + - | + sound { + compatible = "simple-audio-card"; + + simple-audio-card,routing = + "pcm3168a Playback", "DAI1 Playback", + "pcm3168a Playback", "DAI2 Playback", + "pcm3168a Playback", "DAI3 Playback", + "pcm3168a Playback", "DAI4 Playback"; + + simple-audio-card,dai-link@0 { + format = "left_j"; + bitclock-master = <&sndcpu0>; + frame-master = <&sndcpu0>; + + sndcpu0: cpu { + sound-dai = <&rcar_sound 0>; + }; + codec { + sound-dai = <&ak4613>; + }; + }; + + simple-audio-card,dai-link@1 { + format = "i2s"; + bitclock-master = <&sndcpu1>; + frame-master = <&sndcpu1>; + + convert-channels = <8>; /* TDM Split */ + + sndcpu1: cpu@0 { + sound-dai = <&rcar_sound 1>; + }; + cpu@1 { + sound-dai = <&rcar_sound 2>; + }; + cpu@2 { + sound-dai = <&rcar_sound 3>; + }; + cpu@3 { + sound-dai = <&rcar_sound 4>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + dai-tdm-slot-num = <8>; + sound-dai = <&pcm3168a 0>; + }; + }; + + simple-audio-card,dai-link@2 { + format = "i2s"; + bitclock-master = <&sndcpu2>; + frame-master = <&sndcpu2>; + + sndcpu2: cpu { + sound-dai = <&rcar_sound 5>; + }; + codec { + mclk-fs = <512>; + prefix = "pcm3168a"; + sound-dai = <&pcm3168a 1>; + }; + }; + }; -- 2.17.1

3 5

[alsa-devel] [PATCH v2] ASoC: dt-bindings: renesas, rsnd: switch to yaml base Documentation
by Kuninori Morimoto 21 Feb '20

21 Feb '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. It is still keeping detail explanations at .txt Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - fixup dtc error .../bindings/sound/renesas,rsnd.txt | 518 ----------------- .../bindings/sound/renesas,rsnd.yaml | 529 ++++++++++++++++++ 2 files changed, 529 insertions(+), 518 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/renesas,rsnd.yaml diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 797fd035434c..b731f16aea84 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -253,521 +253,3 @@ This is example of TDM 6ch. Driver can automatically switches TDM <-> stereo mode in this case. see "Example: simple sound card for TDM" - -============================================= -Required properties: -============================================= - -- compatible : "renesas,rcar_sound-<soctype>", fallbacks - "renesas,rcar_sound-gen1" if generation1, and - "renesas,rcar_sound-gen2" if generation2 (or RZ/G1) - "renesas,rcar_sound-gen3" if generation3 (or RZ/G2) - Examples with soctypes are: - - "renesas,rcar_sound-r8a7743" (RZ/G1M) - - "renesas,rcar_sound-r8a7744" (RZ/G1N) - - "renesas,rcar_sound-r8a7745" (RZ/G1E) - - "renesas,rcar_sound-r8a77470" (RZ/G1C) - - "renesas,rcar_sound-r8a774a1" (RZ/G2M) - - "renesas,rcar_sound-r8a774b1" (RZ/G2N) - - "renesas,rcar_sound-r8a774c0" (RZ/G2E) - - "renesas,rcar_sound-r8a7778" (R-Car M1A) - - "renesas,rcar_sound-r8a7779" (R-Car H1) - - "renesas,rcar_sound-r8a7790" (R-Car H2) - - "renesas,rcar_sound-r8a7791" (R-Car M2-W) - - "renesas,rcar_sound-r8a7793" (R-Car M2-N) - - "renesas,rcar_sound-r8a7794" (R-Car E2) - - "renesas,rcar_sound-r8a7795" (R-Car H3) - - "renesas,rcar_sound-r8a7796" (R-Car M3-W) - - "renesas,rcar_sound-r8a77965" (R-Car M3-N) - - "renesas,rcar_sound-r8a77990" (R-Car E3) - - "renesas,rcar_sound-r8a77995" (R-Car D3) -- reg : Should contain the register physical address. - required register is - SRU/ADG/SSI if generation1 - SRU/ADG/SSIU/SSI/AUDIO-DMAC-periperi if generation2/generation3 - Select extended AUDIO-DMAC-periperi address if SoC has it, - otherwise select normal AUDIO-DMAC-periperi address. -- reg-names : Should contain the register names. - scu/adg/ssi if generation1 - scu/adg/ssiu/ssi/audmapp if generation2/generation3 -- rcar_sound,ssi : Should contain SSI feature. - The number of SSI subnode should be same as HW. - see below for detail. -- rcar_sound,ssiu : Should contain SSIU feature. - The number of SSIU subnode should be same as HW. - see below for detail. -- rcar_sound,src : Should contain SRC feature. - The number of SRC subnode should be same as HW. - see below for detail. -- rcar_sound,ctu : Should contain CTU feature. - The number of CTU subnode should be same as HW. - see below for detail. -- rcar_sound,mix : Should contain MIX feature. - The number of MIX subnode should be same as HW. - see below for detail. -- rcar_sound,dvc : Should contain DVC feature. - The number of DVC subnode should be same as HW. - see below for detail. -- rcar_sound,dai : DAI contents. - The number of DAI subnode should be same as HW. - see below for detail. -- #sound-dai-cells : it must be 0 if your system is using single DAI - it must be 1 if your system is using multi DAI -- clocks : References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. -- clock-names : List of necessary clock names. - "ssi-all", "ssi.X", "src.X", "mix.X", "ctu.X", - "dvc.X", "clk_a", "clk_b", "clk_c", "clk_i" - -Optional properties: -- #clock-cells : it must be 0 if your system has audio_clkout - it must be 1 if your system has audio_clkout0/1/2/3 -- clock-frequency : for all audio_clkout0/1/2/3 -- clkout-lr-asynchronous : boolean property. it indicates that audio_clkoutn - is asynchronizes with lr-clock. -- resets : References to SSI resets. -- reset-names : List of valid reset names. - "ssi-all", "ssi.X" - -SSI subnode properties: -- interrupts : Should contain SSI interrupt for PIO transfer -- shared-pin : if shared clock pin -- pio-transfer : use PIO transfer mode -- no-busif : BUSIF is not ussed when [mem -> SSI] via DMA case -- dma : Should contain Audio DMAC entry -- dma-names : SSI case "rx" (=playback), "tx" (=capture) - Deprecated: see SSIU subnode properties - SSIU case "rxu" (=playback), "txu" (=capture) - -SSIU subnode properties: -- dma : Should contain Audio DMAC entry -- dma-names : "rx" (=playback), "tx" (=capture) - -SRC subnode properties: -- dma : Should contain Audio DMAC entry -- dma-names : "rx" (=playback), "tx" (=capture) - -DVC subnode properties: -- dma : Should contain Audio DMAC entry -- dma-names : "tx" (=playback/capture) - -DAI subnode properties: -- playback : list of playback modules -- capture : list of capture modules - - -============================================= -Example: -============================================= - -rcar_sound: sound@ec500000 { - #sound-dai-cells = <1>; - compatible = "renesas,rcar_sound-r8a7791", "renesas,rcar_sound-gen2"; - reg = <0 0xec500000 0 0x1000>, /* SCU */ - <0 0xec5a0000 0 0x100>, /* ADG */ - <0 0xec540000 0 0x1000>, /* SSIU */ - <0 0xec541000 0 0x1280>, /* SSI */ - <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ - reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; - - clocks = <&mstp10_clks R8A7790_CLK_SSI_ALL>, - <&mstp10_clks R8A7790_CLK_SSI9>, <&mstp10_clks R8A7790_CLK_SSI8>, - <&mstp10_clks R8A7790_CLK_SSI7>, <&mstp10_clks R8A7790_CLK_SSI6>, - <&mstp10_clks R8A7790_CLK_SSI5>, <&mstp10_clks R8A7790_CLK_SSI4>, - <&mstp10_clks R8A7790_CLK_SSI3>, <&mstp10_clks R8A7790_CLK_SSI2>, - <&mstp10_clks R8A7790_CLK_SSI1>, <&mstp10_clks R8A7790_CLK_SSI0>, - <&mstp10_clks R8A7790_CLK_SCU_SRC9>, <&mstp10_clks R8A7790_CLK_SCU_SRC8>, - <&mstp10_clks R8A7790_CLK_SCU_SRC7>, <&mstp10_clks R8A7790_CLK_SCU_SRC6>, - <&mstp10_clks R8A7790_CLK_SCU_SRC5>, <&mstp10_clks R8A7790_CLK_SCU_SRC4>, - <&mstp10_clks R8A7790_CLK_SCU_SRC3>, <&mstp10_clks R8A7790_CLK_SCU_SRC2>, - <&mstp10_clks R8A7790_CLK_SCU_SRC1>, <&mstp10_clks R8A7790_CLK_SCU_SRC0>, - <&mstp10_clks R8A7790_CLK_SCU_DVC0>, <&mstp10_clks R8A7790_CLK_SCU_DVC1>, - <&audio_clk_a>, <&audio_clk_b>, <&audio_clk_c>, <&m2_clk>; - clock-names = "ssi-all", - "ssi.9", "ssi.8", "ssi.7", "ssi.6", "ssi.5", - "ssi.4", "ssi.3", "ssi.2", "ssi.1", "ssi.0", - "src.9", "src.8", "src.7", "src.6", "src.5", - "src.4", "src.3", "src.2", "src.1", "src.0", - "dvc.0", "dvc.1", - "clk_a", "clk_b", "clk_c", "clk_i"; - - rcar_sound,dvc { - dvc0: dvc-0 { - dmas = <&audma0 0xbc>; - dma-names = "tx"; - }; - dvc1: dvc-1 { - dmas = <&audma0 0xbe>; - dma-names = "tx"; - }; - }; - - rcar_sound,mix { - mix0: mix-0 { }; - mix1: mix-1 { }; - }; - - rcar_sound,ctu { - ctu00: ctu-0 { }; - ctu01: ctu-1 { }; - ctu02: ctu-2 { }; - ctu03: ctu-3 { }; - ctu10: ctu-4 { }; - ctu11: ctu-5 { }; - ctu12: ctu-6 { }; - ctu13: ctu-7 { }; - }; - - rcar_sound,src { - src0: src-0 { - interrupts = <0 352 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x85>, <&audma1 0x9a>; - dma-names = "rx", "tx"; - }; - src1: src-1 { - interrupts = <0 353 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x87>, <&audma1 0x9c>; - dma-names = "rx", "tx"; - }; - src2: src-2 { - interrupts = <0 354 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x89>, <&audma1 0x9e>; - dma-names = "rx", "tx"; - }; - src3: src-3 { - interrupts = <0 355 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8b>, <&audma1 0xa0>; - dma-names = "rx", "tx"; - }; - src4: src-4 { - interrupts = <0 356 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8d>, <&audma1 0xb0>; - dma-names = "rx", "tx"; - }; - src5: src-5 { - interrupts = <0 357 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x8f>, <&audma1 0xb2>; - dma-names = "rx", "tx"; - }; - src6: src-6 { - interrupts = <0 358 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x91>, <&audma1 0xb4>; - dma-names = "rx", "tx"; - }; - src7: src-7 { - interrupts = <0 359 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x93>, <&audma1 0xb6>; - dma-names = "rx", "tx"; - }; - src8: src-8 { - interrupts = <0 360 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x95>, <&audma1 0xb8>; - dma-names = "rx", "tx"; - }; - src9: src-9 { - interrupts = <0 361 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x97>, <&audma1 0xba>; - dma-names = "rx", "tx"; - }; - }; - - rcar_sound,ssiu { - ssiu00: ssiu-0 { - dmas = <&audma0 0x15>, <&audma1 0x16>; - dma-names = "rx", "tx"; - }; - ssiu01: ssiu-1 { - dmas = <&audma0 0x35>, <&audma1 0x36>; - dma-names = "rx", "tx"; - }; - - ... - - ssiu95: ssiu-49 { - dmas = <&audma0 0xA5>, <&audma1 0xA6>; - dma-names = "rx", "tx"; - }; - ssiu96: ssiu-50 { - dmas = <&audma0 0xA7>, <&audma1 0xA8>; - dma-names = "rx", "tx"; - }; - ssiu97: ssiu-51 { - dmas = <&audma0 0xA9>, <&audma1 0xAA>; - dma-names = "rx", "tx"; - }; - }; - - rcar_sound,ssi { - ssi0: ssi-0 { - interrupts = <0 370 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x01>, <&audma1 0x02>; - dma-names = "rx", "tx"; - }; - ssi1: ssi-1 { - interrupts = <0 371 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x03>, <&audma1 0x04>; - dma-names = "rx", "tx"; - }; - - ... - - ssi8: ssi-8 { - interrupts = <0 378 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x11>, <&audma1 0x12>; - dma-names = "rx", "tx"; - }; - ssi9: ssi-9 { - interrupts = <0 379 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x13>, <&audma1 0x14>; - dma-names = "rx", "tx"; - }; - }; - - rcar_sound,dai { - dai0 { - playback = <&ssi5 &src5>; - capture = <&ssi6>; - }; - dai1 { - playback = <&ssi3>; - }; - dai2 { - capture = <&ssi4>; - }; - dai3 { - playback = <&ssi7>; - }; - dai4 { - capture = <&ssi8>; - }; - }; -}; - -============================================= -Example: simple sound card -============================================= - - rsnd_ak4643: sound { - compatible = "simple-audio-card"; - - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&ak4643>; - clocks = <&audio_clock>; - }; - }; - -&rcar_sound { - pinctrl-0 = <&sound_pins &sound_clk_pins>; - pinctrl-names = "default"; - - /* Single DAI */ - #sound-dai-cells = <0>; - - - rcar_sound,dai { - dai0 { - playback = <&ssi0 &src2 &dvc0>; - capture = <&ssi1 &src3 &dvc1>; - }; - }; -}; - -&ssi1 { - shared-pin; -}; - -============================================= -Example: simple sound card for Asynchronous mode -============================================= - -sound { - compatible = "simple-scu-audio-card"; - ... - /* - * SRC Asynchronous mode setting - * Playback: - * All input data will be converted to 48kHz - * Capture: - * Inputed 48kHz data will be converted to - * system specified Hz - */ - simple-audio-card,convert-rate = <48000>; - ... - simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - simple-audio-card,codec { - ... - }; -}; - -============================================= -Example: simple sound card for channel convert -============================================= - -sound { - compatible = "simple-scu-audio-card"; - ... - /* - * CTU setting - * All input data will be converted to 2ch - * as output data - */ - simple-audio-card,convert-channels = <2>; - ... - simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - }; - simple-audio-card,codec { - ... - }; -}; - -============================================= -Example: simple sound card for MIXer -============================================= - -sound { - compatible = "simple-scu-audio-card"; - ... - simple-audio-card,cpu@0 { - sound-dai = <&rcar_sound 0>; - }; - simple-audio-card,cpu@1 { - sound-dai = <&rcar_sound 1>; - }; - simple-audio-card,codec { - ... - }; -}; - -&rcar_sound { - ... - rcar_sound,dai { - dai0 { - playback = <&src1 &ctu02 &mix0 &dvc0 &ssi0>; - }; - dai1 { - playback = <&src2 &ctu03 &mix0 &dvc0 &ssi0>; - }; - }; -}; - -============================================= -Example: simple sound card for TDM -============================================= - -rsnd_tdm: sound { - compatible = "simple-audio-card"; - - simple-audio-card,format = "left_j"; - simple-audio-card,bitclock-master = <&sndcodec>; - simple-audio-card,frame-master = <&sndcodec>; - - sndcpu: simple-audio-card,cpu { - sound-dai = <&rcar_sound>; - dai-tdm-slot-num = <6>; - }; - - sndcodec: simple-audio-card,codec { - sound-dai = <&xxx>; - }; -}; - -============================================= -Example: simple sound card for TDM Split -============================================= - -sound_card: sound { - compatible = "audio-graph-scu-card"; - prefix = "xxxx"; - routing = "xxxx Playback", "DAI0 Playback", - "xxxx Playback", "DAI1 Playback", - "xxxx Playback", "DAI2 Playback", - "xxxx Playback", "DAI3 Playback"; - convert-channels = <8>; /* TDM Split */ - - dais = <&rsnd_port0 /* playback ch1/ch2 */ - &rsnd_port1 /* playback ch3/ch4 */ - &rsnd_port2 /* playback ch5/ch6 */ - &rsnd_port3 /* playback ch7/ch8 */ - >; -}; - -audio-codec { - ... - port { - codec_0: endpoint@1 { - remote-endpoint = <&rsnd_ep0>; - }; - codec_1: endpoint@2 { - remote-endpoint = <&rsnd_ep1>; - }; - codec_2: endpoint@3 { - remote-endpoint = <&rsnd_ep2>; - }; - codec_3: endpoint@4 { - remote-endpoint = <&rsnd_ep3>; - }; - }; -}; - -&rcar_sound { - ... - ports { - rsnd_port0: port@0 { - rsnd_ep0: endpoint { - remote-endpoint = <&codec_0>; - ... - playback = <&ssiu30 &ssi3>; - }; - }; - rsnd_port1: port@1 { - rsnd_ep1: endpoint { - remote-endpoint = <&codec_1>; - ... - playback = <&ssiu31 &ssi3>; - }; - }; - rsnd_port2: port@2 { - rsnd_ep2: endpoint { - remote-endpoint = <&codec_2>; - ... - playback = <&ssiu32 &ssi3>; - }; - }; - rsnd_port3: port@3 { - rsnd_ep3: endpoint { - remote-endpoint = <&codec_3>; - ... - playback = <&ssiu33 &ssi3>; - }; - }; - }; -}; - -============================================= -Example: simple sound card for Multi channel -============================================= - -&rcar_sound { - pinctrl-0 = <&sound_pins &sound_clk_pins>; - pinctrl-names = "default"; - - /* Single DAI */ - #sound-dai-cells = <0>; - - - rcar_sound,dai { - dai0 { - playback = <&ssi0 &ssi1 &ssi2 &src0 &dvc0>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml new file mode 100644 index 000000000000..0958255c8542 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.yaml @@ -0,0 +1,529 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/renesas,rsnd.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Renesas R-Car Sound Driver Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +properties: + $nodename: + pattern: "^sound@.*" + + compatible: + oneOf: + # for Gen1 SoC + - items: + - enum: + - renesas,rcar_sound-r8a7778 # R-Car M1A + - renesas,rcar_sound-r8a7779 # R-Car H1 + - enum: + - renesas,rcar_sound-gen1 + # for Gen2 SoC + - items: + - enum: + - renesas,rcar_sound-r8a7743 # RZ/G1M + - renesas,rcar_sound-r8a7744 # RZ/G1N + - renesas,rcar_sound-r8a7745 # RZ/G1E + - renesas,rcar_sound-r8a77470 # RZ/G1C + - renesas,rcar_sound-r8a7790 # R-Car H2 + - renesas,rcar_sound-r8a7791 # R-Car M2-W + - renesas,rcar_sound-r8a7793 # R-Car M2-N + - renesas,rcar_sound-r8a7794 # R-Car E2 + - enum: + - renesas,rcar_sound-gen2 + # for Gen3 SoC + - items: + - enum: + - renesas,rcar_sound-r8a774a1 # RZ/G2M + - renesas,rcar_sound-r8a774b1 # RZ/G2N + - renesas,rcar_sound-r8a774c0 # RZ/G2E + - renesas,rcar_sound-r8a7795 # R-Car H3 + - renesas,rcar_sound-r8a7796 # R-Car M3-W + - renesas,rcar_sound-r8a77965 # R-Car M3-N + - renesas,rcar_sound-r8a77990 # R-Car E3 + - renesas,rcar_sound-r8a77995 # R-Car D3 + - enum: + - renesas,rcar_sound-gen3 + # for Generic + - items: + - enum: + - renesas,rcar_sound-gen1 + - renesas,rcar_sound-gen2 + - renesas,rcar_sound-gen3 + + reg: + minItems: 3 # Gen1 + maxItems: 5 # Gen2/Gen3 + + reg-names: + description: | + Should contain the register names. + scu/adg/ssi if generation1 + scu/adg/ssiu/ssi/audmapp if generation2/generation3 + minItems: 3 # Gen1 + maxItems: 5 # Gen2/Gen3 + allOf: + - items: + enum: + - scu + - adg + - ssiu + - ssi + - audmapp + + "#sound-dai-cells": + description: | + it must be 0 if your system is using single DAI + it must be 1 if your system is using multi DAIs + enum: [0, 1] + + "#clock-cells": + description: | + it must be 0 if your system has audio_clkout + it must be 1 if your system has audio_clkout0/1/2/3 + enum: [0, 1] + + clock-frequency: + description: for audio_clkout0/1/2/3 + $ref: /schemas/types.yaml#/definitions/uint32-array + + clkout-lr-asynchronous: + description: audio_clkoutn is asynchronizes with lr-clock. + $ref: /schemas/types.yaml#/definitions/flag + + power-domains: true + + resets: + description: References to SSI resets + $ref: /schemas/types.yaml#/definitions/phandle-array + + reset-names: + description: List of valid reset names. + allOf: + - items: + enum: + - ssi-all + - ssi.9 + - ssi.8 + - ssi.7 + - ssi.6 + - ssi.5 + - ssi.4 + - ssi.3 + - ssi.2 + - ssi.1 + - ssi.0 + + clocks: + $ref: /schemas/types.yaml#/definitions/phandle-array + description: References to SSI/SRC/MIX/CTU/DVC/AUDIO_CLK clocks. + + clock-names: + description: List of necessary clock names. + allOf: + - items: + enum: + - ssi-all + - ssi.9 + - ssi.8 + - ssi.7 + - ssi.6 + - ssi.5 + - ssi.4 + - ssi.3 + - ssi.2 + - ssi.1 + - ssi.0 + - src.9 + - src.8 + - src.7 + - src.6 + - src.5 + - src.4 + - src.3 + - src.2 + - src.1 + - src.0 + - mix.1 + - mix.0 + - ctu.1 + - ctu.0 + - dvc.1 + - dvc.0 + - clk_a + - clk_b + - clk_c + - clk_i + + # For OF-graph + port: + description: OF-Graph subnode + type: object + properties: + reg: + $ref: /schemas/types.yaml#/definitions/uint32 + endpoint: + type: object + properties: + remote-endpoint: + $ref: /schemas/types.yaml#/definitions/phandle-array + dai-format: + $ref: "simple-card.yaml#/definitions/format" + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + required: + - remote-endpoint + + # For multi OF-graph + ports: + description: multi OF-Graph subnode + type: object + patternProperties: + "port(@.*)?": + $ref: "#properties/port" + +patternProperties: + "^rcar_sound,dvc$": + description: DVC subnode. + type: object + patternProperties: + "dvc-.": + type: object + properties: + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + const: "tx" + required: + - dmas + - dma-names + additionalProperties: false + + "^rcar_sound,mix$": + description: MIX subnode. + type: object + patternProperties: + "mix-.": + type: object + additionalProperties: false + + "^rcar_sound,ctu$": + description: CTU subnode. + type: object + patternProperties: + "ctu-.": + type: object + additionalProperties: false + + "^rcar_sound,src$": + description: SRC subnode. + type: object + patternProperties: + "src-.": + type: object + properties: + interrupts: + maxItems: 1 + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + allOf: + - items: + enum: + - tx + - rx + allOf: + - if: + properties: + status: + const: "ok" + then: + required: + - interrupts + - dmas + - dma-names + additionalProperties: false + + "^rcar_sound,ssiu$": + description: SSIU subnode. + type: object + patternProperties: + "ssiu-.*": + type: object + properties: + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + allOf: + - items: + enum: + - tx + - rx + required: + - dmas + - dma-names + additionalProperties: false + + "^rcar_sound,ssi$": + description: SSI subnode. + type: object + patternProperties: + "ssi-.": + type: object + properties: + interrupts: + maxItems: 1 + dmas: + $ref: /schemas/types.yaml#/definitions/phandle-array + dma-names: + allOf: + - items: + enum: + - tx + - rx + - txu # if no ssiu node + - rxu # if no ssiu node + + shared-pin: + description: shared clock pin + $ref: /schemas/types.yaml#/definitions/flag + pio-transfer: + description: PIO transfer mode + $ref: /schemas/types.yaml#/definitions/flag + no-busif: + description: BUSIF is not used when [mem -> SSI] via DMA case + $ref: /schemas/types.yaml#/definitions/flag + required: + - interrupts + - dmas + - dma-names + additionalProperties: false + + # For DAI base + "^rcar_sound,dai$": + description: DAI subnode. + type: object + patternProperties: + "dai.*": + type: object + properties: + playback: + $ref: /schemas/types.yaml#/definitions/phandle-array + capture: + $ref: /schemas/types.yaml#/definitions/phandle-array + anyOf: + - required: + - playback + - required: + - capture + additionalProperties: false + +required: + - compatible + - reg + - reg-names + - clocks + - clock-names + - "#sound-dai-cells" + +additionalProperties: false + +examples: + - | + rcar_sound: sound@ec500000 { + #sound-dai-cells = <1>; + compatible = "renesas,rcar_sound-r8a7790", "renesas,rcar_sound-gen2"; + reg = <0 0xec500000 0 0x1000>, /* SCU */ + <0 0xec5a0000 0 0x100>, /* ADG */ + <0 0xec540000 0 0x1000>, /* SSIU */ + <0 0xec541000 0 0x1280>, /* SSI */ + <0 0xec740000 0 0x200>; /* Audio DMAC peri peri*/ + reg-names = "scu", "adg", "ssiu", "ssi", "audmapp"; + + clocks = <&mstp10_clks 1005>, /* SSI-ALL */ + <&mstp10_clks 1006>, <&mstp10_clks 1007>, /* SSI9, SSI8 */ + <&mstp10_clks 1008>, <&mstp10_clks 1009>, /* SSI7, SSI6 */ + <&mstp10_clks 1010>, <&mstp10_clks 1011>, /* SSI5, SSI4 */ + <&mstp10_clks 1012>, <&mstp10_clks 1013>, /* SSI3, SSI2 */ + <&mstp10_clks 1014>, <&mstp10_clks 1015>, /* SSI1, SSI0 */ + <&mstp10_clks 1022>, <&mstp10_clks 1023>, /* SRC9, SRC8 */ + <&mstp10_clks 1024>, <&mstp10_clks 1025>, /* SRC7, SRC6 */ + <&mstp10_clks 1026>, <&mstp10_clks 1027>, /* SRC5, SRC4 */ + <&mstp10_clks 1028>, <&mstp10_clks 1029>, /* SRC3, SRC2 */ + <&mstp10_clks 1030>, <&mstp10_clks 1031>, /* SRC1, SRC0 */ + <&mstp10_clks 1020>, <&mstp10_clks 1021>, /* DVC0, DVC1 */ + <&mstp10_clks 1020>, <&mstp10_clks 1021>, /* CLKA, CLKB */ + <&mstp10_clks 1019>, <&mstp10_clks 1018>; /* CLKC, CLKI */ + + clock-names = "ssi-all", + "ssi.9", "ssi.8", + "ssi.7", "ssi.6", + "ssi.5", "ssi.4", + "ssi.3", "ssi.2", + "ssi.1", "ssi.0", + "src.9", "src.8", + "src.7", "src.6", + "src.5", "src.4", + "src.3", "src.2", + "src.1", "src.0", + "dvc.0", "dvc.1", + "clk_a", "clk_b", + "clk_c", "clk_i"; + + rcar_sound,dvc { + dvc0: dvc-0 { + dmas = <&audma0 0xbc>; + dma-names = "tx"; + }; + dvc1: dvc-1 { + dmas = <&audma0 0xbe>; + dma-names = "tx"; + }; + }; + + + rcar_sound,mix { + mix0: mix-0 { }; + mix1: mix-1 { }; + }; + + rcar_sound,ctu { + ctu00: ctu-0 { }; + ctu01: ctu-1 { }; + ctu02: ctu-2 { }; + ctu03: ctu-3 { }; + ctu10: ctu-4 { }; + ctu11: ctu-5 { }; + ctu12: ctu-6 { }; + ctu13: ctu-7 { }; + }; + + rcar_sound,src { + src0: src-0 { + status = "disabled"; + }; + src1: src-1 { + interrupts = <0 353 0>; + dmas = <&audma0 0x87>, <&audma1 0x9c>; + dma-names = "rx", "tx"; + }; + /* skip after src-2 */ + }; + + rcar_sound,ssiu { + ssiu00: ssiu-0 { + dmas = <&audma0 0x15>, <&audma1 0x16>; + dma-names = "rx", "tx"; + }; + ssiu01: ssiu-1 { + dmas = <&audma0 0x35>, <&audma1 0x36>; + dma-names = "rx", "tx"; + }; + /* skip after ssiu-2 */ + }; + + rcar_sound,ssi { + ssi0: ssi-0 { + interrupts = <0 370 1>; + dmas = <&audma0 0x01>, <&audma1 0x02>; + dma-names = "rx", "tx"; + }; + ssi1: ssi-1 { + interrupts = <0 371 1>; + dmas = <&audma0 0x03>, <&audma1 0x04>; + dma-names = "rx", "tx"; + }; + /* skip other ssi-2 */ + }; + + /* DAI base */ + rcar_sound,dai { + dai0 { + playback = <&ssi5 &src5>; + capture = <&ssi6>; + }; + dai1 { + playback = <&ssi3>; + }; + dai2 { + capture = <&ssi4>; + }; + dai3 { + playback = <&ssi7>; + }; + dai4 { + capture = <&ssi8>; + }; + }; + + /* OF-Graph */ + port { + rsnd_endpoint: endpoint { + remote-endpoint = <&codec_endpoint>; + + dai-format = "left_j"; + bitclock-master = <&rsnd_endpoint0>; + frame-master = <&rsnd_endpoint0>; + + playback = <&ssi0 &src0 &dvc0>; + capture = <&ssi1 &src1 &dvc1>; + }; + }; + + /* Multi OF-Graph */ + ports { + #address-cells = <1>; + #size-cells = <0>; + rsnd_port0: port@0 { + reg = <0>; + rsnd_endpoint0: endpoint { + remote-endpoint = <&codec_endpoint1>; + + dai-format = "left_j"; + bitclock-master = <&rsnd_endpoint0>; + frame-master = <&rsnd_endpoint0>; + + playback = <&ssi0 &src0 &dvc0>; + capture = <&ssi1 &src1 &dvc1>; + }; + }; + rsnd_port1: port@1 { + reg = <1>; + rsnd_endpoint1: endpoint { + remote-endpoint = <&codec_endpoint2>; + + dai-format = "i2s"; + playback = <&ssi0 &src0 &dvc0>; + capture = <&ssi1 &src1 &dvc1>; + }; + }; + }; + }; + + + /* For OF-Graph */ + codec { + ports { + port@0 { + codec_endpoint: endpoint { + remote-endpoint = <&rsnd_endpoint>; + }; + }; + port@1 { + codec_endpoint1: endpoint { + remote-endpoint = <&rsnd_endpoint0>; + }; + }; + port@2 { + codec_endpoint2: endpoint { + remote-endpoint = <&rsnd_endpoint1>; + }; + }; + }; + }; -- 2.17.1

3 5

[PATCH 0/2] SOF: Intel: hda: allow operation without i915 gfx
by Kai Vehmanen 21 Feb '20

21 Feb '20

Hi, here's a pair of patches that fixes SOF HDA driver load on devices that have: - digital mics connected to Intel HDA (so that SOF needs to be used instead of the legacy HDA driver), and - external GPU (internal i915 graphics not enabled) Kai Vehmanen (2): ASoC: intel/skl/hda - add no-HDMI cases to generic HDA driver ASoC: SOF: Intel: hda: allow operation without i915 gfx sound/soc/intel/boards/skl_hda_dsp_common.h | 4 ++++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 25 ++++++++++++++++---- sound/soc/sof/intel/hda-codec.c | 11 ++++++++- sound/soc/sof/intel/hda.c | 22 +++++++---------- 4 files changed, 42 insertions(+), 20 deletions(-) -- 2.17.1

2 4

[PATCH] ASoC: pcm512x: Fix unbalanced regulator enable call in probe error path
by Matthias Reichl 21 Feb '20

21 Feb '20

When we get a clock error during probe we have to call regulator_bulk_disable before bailing out, otherwise we trigger a warning in regulator_put. Fix this by using "goto err" like in the error cases above. Fixes: 5a3af1293194d ("ASoC: pcm512x: Add PCM512x driver") Signed-off-by: Matthias Reichl <hias(a)horus.com> --- sound/soc/codecs/pcm512x.c | 8 +++++--- 1 file changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 861210f6bf4fd..4cbef9affffda 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -1564,13 +1564,15 @@ int pcm512x_probe(struct device *dev, struct regmap *regmap) } pcm512x->sclk = devm_clk_get(dev, NULL); - if (PTR_ERR(pcm512x->sclk) == -EPROBE_DEFER) - return -EPROBE_DEFER; + if (PTR_ERR(pcm512x->sclk) == -EPROBE_DEFER) { + ret = -EPROBE_DEFER; + goto err; + } if (!IS_ERR(pcm512x->sclk)) { ret = clk_prepare_enable(pcm512x->sclk); if (ret != 0) { dev_err(dev, "Failed to enable SCLK: %d\n", ret); - return ret; + goto err; } } -- 2.20.1

3 2

Applied "dt-bindings: sound: Add TLV320ADCx140 dt bindings" to the asoc tree
by Mark Brown 21 Feb '20

21 Feb '20

The patch dt-bindings: sound: Add TLV320ADCx140 dt bindings has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 4ee67cbd97668ab1b17d86d85348302c0b7490cd Mon Sep 17 00:00:00 2001 From: Dan Murphy <dmurphy(a)ti.com> Date: Thu, 20 Feb 2020 15:07:58 -0600 Subject: [PATCH] dt-bindings: sound: Add TLV320ADCx140 dt bindings Add dt bindings for the TLV320ADCx140 Burr-Brown ADC. The initial support is for the following: TLV320ADC3140 - http://www.ti.com/lit/gpn/tlv320adc3140 TLV320ADC5140 - http://www.ti.com/lit/gpn/tlv320adc5140 TLV320ADC6140 - http://www.ti.com/lit/gpn/tlv320adc6140 Signed-off-by: Dan Murphy <dmurphy(a)ti.com> Link: https://lore.kernel.org/r/20200220210759.31466-2-dmurphy@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/tlv320adcx140.yaml | 83 +++++++++++++++++++ 1 file changed, 83 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320adcx140.yaml diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml new file mode 100644 index 000000000000..1433ff62b14f --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -0,0 +1,83 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +# Copyright (C) 2019 Texas Instruments Incorporated +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/tlv320adcx140.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Texas Instruments TLV320ADCX140 Quad Channel Analog-to-Digital Converter + +maintainers: + - Dan Murphy <dmurphy(a)ti.com> + +description: | + The TLV320ADCX140 are multichannel (4-ch analog recording or 8-ch digital + PDM microphones recording), high-performance audio, analog-to-digital + converter (ADC) with analog inputs supporting up to 2V RMS. The TLV320ADCX140 + family supports line and microphone Inputs, and offers a programmable + microphone bias or supply voltage generation. + + Specifications can be found at: + http://www.ti.com/lit/ds/symlink/tlv320adc3140.pdf + http://www.ti.com/lit/ds/symlink/tlv320adc5140.pdf + http://www.ti.com/lit/ds/symlink/tlv320adc6140.pdf + +properties: + compatible: + oneOf: + - const: ti,tlv320adc3140 + - const: ti,tlv320adc5140 + - const: ti,tlv320adc6140 + + reg: + maxItems: 1 + description: | + I2C addresss of the device can be one of these 0x4c, 0x4d, 0x4e or 0x4f + + reset-gpios: + description: | + GPIO used for hardware reset. + + areg-supply: + description: | + Regulator with AVDD at 3.3V. If not defined then the internal regulator + is enabled. + + ti,mic-bias-source: + description: | + Indicates the source for MIC Bias. + 0 - Mic bias is set to VREF + 1 - Mic bias is set to VREF × 1.096 + 6 - Mic bias is set to AVDD + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + - enum: [0, 1, 6] + + ti,vref-source: + description: | + Indicates the source for MIC Bias. + 0 - Set VREF to 2.75V + 1 - Set VREF to 2.5V + 2 - Set VREF to 1.375V + allOf: + - $ref: /schemas/types.yaml#/definitions/uint32 + - enum: [0, 1, 2] + +required: + - compatible + - reg + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + i2c0 { + #address-cells = <1>; + #size-cells = <0>; + codec: codec@4c { + compatible = "ti,tlv320adc5140"; + reg = <0x4c>; + ti,use-internal-areg; + ti,mic-bias-source = <6>; + reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>; + }; + }; -- 2.20.1

1 0

[PATCH for-next v3 0/2] Introduce the TLV320ADCx140 codec family
by Dan Murphy 21 Feb '20

21 Feb '20

Hello Introducing the Texas Instruments TLV320ADCx140 quad channel audio ADC. This device supports 4 analog inputs, 8 digital inputs or a combination of analog and digital microphones. TLV320ADC3140 - http://www.ti.com/lit/gpn/tlv320adc3140 TLV320ADC5140 - http://www.ti.com/lit/gpn/tlv320adc5140 TLV320ADC6140 - http://www.ti.com/lit/gpn/tlv320adc6140 Dan Murphy (2): dt-bindings: sound: Add TLV320ADCx140 dt bindings ASoC: tlv320adcx140: Add the tlv320adcx140 codec driver family .../bindings/sound/tlv320adcx140.yaml | 83 ++ sound/soc/codecs/Kconfig | 9 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320adcx140.c | 849 ++++++++++++++++++ sound/soc/codecs/tlv320adcx140.h | 130 +++ 5 files changed, 1073 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320adcx140.yaml create mode 100644 sound/soc/codecs/tlv320adcx140.c create mode 100644 sound/soc/codecs/tlv320adcx140.h -- 2.25.0

2 3

[PATCH v3] ASoC: dt-bindings: stm32: convert sai to json-schema
by Olivier Moysan 20 Feb '20

20 Feb '20

Convert the STM32 SAI bindings to DT schema format using json-schema. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- Changes in v2: - use pattern for compatible of child nodes - rework dmas and clocks properties - add "additionalProperties" Changes in v3: - move clocks properties for st,stm32h7-sai compatible, to 'else' clause --- .../bindings/sound/st,stm32-sai.txt | 107 ---------- .../bindings/sound/st,stm32-sai.yaml | 193 ++++++++++++++++++ 2 files changed, 193 insertions(+), 107 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.yaml diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt deleted file mode 100644 index 944743dd9212..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt +++ /dev/null @@ -1,107 +0,0 @@ -STMicroelectronics STM32 Serial Audio Interface (SAI). - -The SAI interface (Serial Audio Interface) offers a wide set of audio protocols -as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. -The SAI contains two independent audio sub-blocks. Each sub-block has -its own clock generator and I/O lines controller. - -Required properties: - - compatible: Should be "st,stm32f4-sai" or "st,stm32h7-sai" - - reg: Base address and size of SAI common register set. - - clocks: Must contain phandle and clock specifier pairs for each entry - in clock-names. - - clock-names: Must contain "pclk" "x8k" and "x11k" - "pclk": Clock which feeds the peripheral bus interface. - Mandatory for "st,stm32h7-sai" compatible. - Not used for "st,stm32f4-sai" compatible. - "x8k": SAI parent clock for sampling rates multiple of 8kHz. - "x11k": SAI parent clock for sampling rates multiple of 11.025kHz. - - interrupts: cpu DAI interrupt line shared by SAI sub-blocks - -Optional properties: - - resets: Reference to a reset controller asserting the SAI - -SAI subnodes: -Two subnodes corresponding to SAI sub-block instances A et B can be defined. -Subnode can be omitted for unsused sub-block. - -SAI subnodes required properties: - - compatible: Should be "st,stm32-sai-sub-a" or "st,stm32-sai-sub-b" - for SAI sub-block A or B respectively. - - reg: Base address and size of SAI sub-block register set. - - clocks: Must contain one phandle and clock specifier pair - for sai_ck which feeds the internal clock generator. - If the SAI shares a master clock, with another SAI set as MCLK - clock provider, SAI provider phandle must be specified here. - - clock-names: Must contain "sai_ck". - Must also contain "MCLK", if SAI shares a master clock, - with a SAI set as MCLK clock provider. - - dmas: see Documentation/devicetree/bindings/dma/stm32-dma.txt - - dma-names: identifier string for each DMA request line - "tx": if sai sub-block is configured as playback DAI - "rx": if sai sub-block is configured as capture DAI - - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml - -SAI subnodes Optional properties: - - st,sync: specify synchronization mode. - By default SAI sub-block is in asynchronous mode. - This property sets SAI sub-block as slave of another SAI sub-block. - Must contain the phandle and index of the sai sub-block providing - the synchronization. - - st,iec60958: support S/PDIF IEC6958 protocol for playback - IEC60958 protocol is not available for capture. - By default, custom protocol is assumed, meaning that protocol is - configured according to protocol defined in related DAI link node, - such as i2s, left justified, right justified, dsp and pdm protocols. - Note: ac97 protocol is not supported by SAI driver - - #clock-cells: should be 0. This property must be present if the SAI device - is a master clock provider, according to clocks bindings, described in - Documentation/devicetree/bindings/clock/clock-bindings.txt. - -The device node should contain one 'port' child node with one child 'endpoint' -node, according to the bindings defined in Documentation/devicetree/bindings/ -graph.txt. - -Example: -sound_card { - compatible = "audio-graph-card"; - dais = <&sai1b_port>; -}; - -sai1: sai1@40015800 { - compatible = "st,stm32h7-sai"; - #address-cells = <1>; - #size-cells = <1>; - ranges = <0 0x40015800 0x400>; - reg = <0x40015800 0x4>; - clocks = <&rcc SAI1_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>; - clock-names = "pclk", "x8k", "x11k"; - interrupts = <87>; - - sai1a: audio-controller@40015804 { - compatible = "st,stm32-sai-sub-a"; - reg = <0x4 0x1C>; - clocks = <&rcc SAI1_CK>; - clock-names = "sai_ck"; - dmas = <&dmamux1 1 87 0x400 0x0>; - dma-names = "tx"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_sai1a>; - - sai1b_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - format = "i2s"; - }; - }; - }; -}; - -audio-codec { - codec_port: port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml new file mode 100644 index 000000000000..cdea20150c1e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml @@ -0,0 +1,193 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 Serial Audio Interface (SAI) + +maintainers: + - Olivier Moysan <olivier.moysan(a)st.com> + +description: + The SAI interface (Serial Audio Interface) offers a wide set of audio + protocols as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. + The SAI contains two independent audio sub-blocks. Each sub-block has + its own clock generator and I/O lines controller. + +properties: + compatible: + enum: + - st,stm32f4-sai + - st,stm32h7-sai + + reg: + items: + - description: Base address and size of SAI common register set. + - description: Base address and size of SAI identification register set. + minItems: 1 + maxItems: 2 + + ranges: + maxItems: 1 + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + "#address-cells": + const: 1 + + "#size-cells": + const: 1 + +required: + - compatible + - reg + - ranges + - "#address-cells" + - "#size-cells" + - clocks + - clock-names + +patternProperties: + "^audio-controller@[0-9a-f]+$": + type: object + description: + Two subnodes corresponding to SAI sub-block instances A et B + can be defined. Subnode can be omitted for unsused sub-block. + + properties: + compatible: + description: Compatible for SAI sub-block A or B. + pattern: "st,stm32-sai-sub-[ab]" + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + items: + - description: sai_ck clock feeding the internal clock generator. + - description: MCLK clock from a SAI set as master clock provider. + minItems: 1 + maxItems: 2 + + clock-names: + items: + - const: sai_ck + - const: MCLK + minItems: 1 + maxItems: 2 + + dmas: + maxItems: 1 + + dma-names: + description: | + rx: SAI sub-block is configured as a capture DAI. + tx: SAI sub-block is configured as a playback DAI. + enum: [ rx, tx ] + + st,sync: + description: + Configure the SAI sub-block as slave of another SAI sub-block. + By default SAI sub-block is in asynchronous mode. + Must contain the phandle and index of the SAI sub-block providing + the synchronization. + allOf: + - $ref: /schemas/types.yaml#definitions/phandle-array + - maxItems: 1 + + st,iec60958: + description: + If set, support S/PDIF IEC6958 protocol for playback. + IEC60958 protocol is not available for capture. + By default, custom protocol is assumed, meaning that protocol is + configured according to protocol defined in related DAI link node, + such as i2s, left justified, right justified, dsp and pdm protocols. + allOf: + - $ref: /schemas/types.yaml#definitions/flag + + "#clock-cells": + description: Configure the SAI device as master clock provider. + const: 0 + + required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - dmas + - dma-names + + additionalProperties: false + +allOf: + - if: + properties: + compatible: + contains: + const: st,stm32f4-sai + + - then: + properties: + clocks: + items: + - description: x8k, SAI parent clock for sampling rates multiple of 8kHz. + - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - const: x8k + - const: x11k + + - else: + properties: + clocks: + items: + - description: pclk feeds the peripheral bus interface. + - description: x8k, SAI parent clock for sampling rates multiple of 8kHz. + - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - const: pclk + - const: x8k + - const: x11k + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + #include <dt-bindings/reset/stm32mp1-resets.h> + sai1: sai@4400a000 { + compatible = "st,stm32h7-sai"; + #address-cells = <1>; + #size-cells = <1>; + ranges = <0 0x4400a000 0x400>; + reg = <0x4400a000 0x4>, <0x4400a3f0 0x10>; + interrupts = <GIC_SPI 87 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&rcc SAI1>, <&rcc PLL1_Q>, <&rcc PLL2_P>; + clock-names = "pclk", "x8k", "x11k"; + resets = <&rcc SAI1_R>; + + sai1a: audio-controller@4400a004 { + compatible = "st,stm32-sai-sub-a"; + #sound-dai-cells = <0>; + reg = <0x4 0x1c>; + clocks = <&rcc SAI1_K>; + clock-names = "sai_ck"; + dmas = <&dmamux1 87 0x400 0x01>; + dma-names = "tx"; + }; + }; + +... -- 2.17.1

2 1

[PATCH] ASoC: tas2562: Add support for digital volume control
by Dan Murphy 20 Feb '20

20 Feb '20

Add support for digital volume control. There is no dedicated register for volume control but instead there are 4. The values of the registers are determined with exponential floating point math. So a table was created with register values for 2dB step increments from -110dB to 0dB. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tas2562.c | 94 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2562.h | 6 ++- 2 files changed, 98 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index b2682c2360b6..e577aa6f304a 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -26,6 +26,24 @@ #define TAS2562_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FORMAT_S32_LE) +/* DVC equation involves floating point math + * round(10^(volume in dB/20)*2^30) + * so create a lookup table for 2dB step + */ +static const unsigned int float_vol_db_lookup[] = { +0x00000d43, 0x000010b2, 0x00001505, 0x00001a67, 0x00002151, +0x000029f1, 0x000034cd, 0x00004279, 0x000053af, 0x0000695b, +0x0000695b, 0x0000a6fa, 0x0000d236, 0x000108a4, 0x00014d2a, +0x0001a36e, 0x00021008, 0x000298c0, 0x000344df, 0x00041d8f, +0x00052e5a, 0x000685c8, 0x00083621, 0x000a566d, 0x000d03a7, +0x0010624d, 0x0014a050, 0x0019f786, 0x0020b0bc, 0x0029279d, +0x0033cf8d, 0x004139d3, 0x00521d50, 0x00676044, 0x0082248a, +0x00a3d70a, 0x00ce4328, 0x0103ab3d, 0x0146e75d, 0x019b8c27, +0x02061b89, 0x028c423f, 0x03352529, 0x0409c2b0, 0x05156d68, +0x080e9f96, 0x0a24b062, 0x0cc509ab, 0x10137987, 0x143d1362, +0x197a967f, 0x2013739e, 0x28619ae9, 0x32d64617, 0x40000000 +}; + struct tas2562_data { struct snd_soc_component *component; struct gpio_desc *sdz_gpio; @@ -34,6 +52,7 @@ struct tas2562_data { struct i2c_client *client; int v_sense_slot; int i_sense_slot; + int volume_lvl; }; static int tas2562_set_bias_level(struct snd_soc_component *component, @@ -334,6 +353,22 @@ static int tas2562_codec_probe(struct snd_soc_component *component) if (ret < 0) return ret; + /* Set the Digital volume to -110dB */ + ret = snd_soc_component_write(component, TAS2562_DVC_CFG4, 0x00); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG3, 0x00); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG2, 0x0d); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG1, 0x43); + if (ret) + return ret; + + tas2562->volume_lvl = 0; + return 0; } @@ -414,6 +449,50 @@ static int tas2562_dac_event(struct snd_soc_dapm_widget *w, return 0; } +static int tas2562_volume_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = tas2562->volume_lvl; + return 0; +} + +static int tas2562_volume_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; + u32 reg_val; + + reg_val = float_vol_db_lookup[ucontrol->value.integer.value[0]/2]; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG4, + (reg_val & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG3, + ((reg_val >> 8) & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG2, + ((reg_val >> 16) & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG1, + ((reg_val >> 24) & 0xff)); + if (ret) + return ret; + + tas2562->volume_lvl = ucontrol->value.integer.value[0]; + + return ret; +} + +/* Digital Volume Control. From 0 dB to -110 dB in 1 dB steps */ +static const DECLARE_TLV_DB_SCALE(dvc_tlv, -11000, 100, 0); + static DECLARE_TLV_DB_SCALE(tas2562_dac_tlv, 850, 50, 0); static const struct snd_kcontrol_new isense_switch = @@ -427,6 +506,17 @@ static const struct snd_kcontrol_new vsense_switch = static const struct snd_kcontrol_new tas2562_snd_controls[] = { SOC_SINGLE_TLV("Amp Gain Volume", TAS2562_PB_CFG1, 0, 0x1c, 0, tas2562_dac_tlv), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Volume Control", + .index = 0, + .tlv.p = dvc_tlv, + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_soc_info_volsw, + .get = tas2562_volume_control_get, + .put = tas2562_volume_control_put, + .private_value = SOC_SINGLE_VALUE(TAS2562_DVC_CFG1, 0, 110, 0, 0) , + }, }; static const struct snd_soc_dapm_widget tas2562_dapm_widgets[] = { @@ -517,6 +607,10 @@ static const struct reg_default tas2562_reg_defaults[] = { { TAS2562_PB_CFG1, 0x20 }, { TAS2562_TDM_CFG0, 0x09 }, { TAS2562_TDM_CFG1, 0x02 }, + { TAS2562_DVC_CFG1, 0x40 }, + { TAS2562_DVC_CFG2, 0x40 }, + { TAS2562_DVC_CFG3, 0x00 }, + { TAS2562_DVC_CFG4, 0x00 }, }; static const struct regmap_config tas2562_regmap_config = { diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 6f55ebcf19ea..28e75fc431d0 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -35,8 +35,10 @@ #define TAS2562_REV_ID TAS2562_REG(0, 0x7d) /* Page 2 */ -#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x01) -#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x02) +#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x0c) +#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x0d) +#define TAS2562_DVC_CFG3 TAS2562_REG(2, 0x0e) +#define TAS2562_DVC_CFG4 TAS2562_REG(2, 0x0f) #define TAS2562_RESET BIT(0) -- 2.25.0

2 6

[PATCH] sound: Replace zero-length array with flexible-array member
by Gustavo A. R. Silva 20 Feb '20

20 Feb '20

The current codebase makes use of the zero-length array language extension to the C90 standard, but the preferred mechanism to declare variable-length types such as these ones is a flexible array member[1][2], introduced in C99: struct foo { int stuff; struct boo array[]; }; By making use of the mechanism above, we will get a compiler warning in case the flexible array does not occur last in the structure, which will help us prevent some kind of undefined behavior bugs from being inadvertently introduced[3] to the codebase from now on. Also, notice that, dynamic memory allocations won't be affected by this change: "Flexible array members have incomplete type, and so the sizeof operator may not be applied. As a quirk of the original implementation of zero-length arrays, sizeof evaluates to zero."[1] This issue was found with the help of Coccinelle. [1] https://gcc.gnu.org/onlinedocs/gcc/Zero-Length.html [2] https://github.com/KSPP/linux/issues/21 [3] commit 76497732932f ("cxgb3/l2t: Fix undefined behaviour") Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- include/sound/control.h | 2 +- include/sound/intel-nhlt.h | 6 +++--- include/sound/soc-dapm.h | 2 +- include/sound/soc.h | 2 +- include/uapi/sound/asound.h | 2 +- include/uapi/sound/skl-tplg-interface.h | 2 +- sound/core/oss/pcm_plugin.h | 2 +- sound/firewire/fireworks/fireworks.h | 2 +- sound/soc/intel/atom/sst-atom-controls.h | 2 +- sound/soc/intel/skylake/skl-i2s.h | 2 +- sound/soc/intel/skylake/skl-topology.h | 4 ++-- sound/soc/intel/skylake/skl.h | 2 +- sound/usb/usx2y/usbusx2y.h | 2 +- 13 files changed, 16 insertions(+), 16 deletions(-) diff --git a/include/sound/control.h b/include/sound/control.h index 11feeee31e35..aeaed2a05bae 100644 --- a/include/sound/control.h +++ b/include/sound/control.h @@ -75,7 +75,7 @@ struct snd_kcontrol { unsigned long private_value; void *private_data; void (*private_free)(struct snd_kcontrol *kcontrol); - struct snd_kcontrol_volatile vd[0]; /* volatile data */ + struct snd_kcontrol_volatile vd[]; /* volatile data */ }; #define snd_kcontrol(n) list_entry(n, struct snd_kcontrol, list) diff --git a/include/sound/intel-nhlt.h b/include/sound/intel-nhlt.h index f657fd8fc0ad..743c2f442280 100644 --- a/include/sound/intel-nhlt.h +++ b/include/sound/intel-nhlt.h @@ -50,7 +50,7 @@ enum nhlt_device_type { struct nhlt_specific_cfg { u32 size; - u8 caps[0]; + u8 caps[]; } __packed; struct nhlt_fmt_cfg { @@ -60,7 +60,7 @@ struct nhlt_fmt_cfg { struct nhlt_fmt { u8 fmt_count; - struct nhlt_fmt_cfg fmt_config[0]; + struct nhlt_fmt_cfg fmt_config[]; } __packed; struct nhlt_endpoint { @@ -80,7 +80,7 @@ struct nhlt_endpoint { struct nhlt_acpi_table { struct acpi_table_header header; u8 endpoint_count; - struct nhlt_endpoint desc[0]; + struct nhlt_endpoint desc[]; } __packed; struct nhlt_resource_desc { diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 1a9c5dd40228..f870f927b70c 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -688,7 +688,7 @@ struct snd_soc_dapm_context { /* A list of widgets associated with an object, typically a snd_kcontrol */ struct snd_soc_dapm_widget_list { int num_widgets; - struct snd_soc_dapm_widget *widgets[0]; + struct snd_soc_dapm_widget *widgets[]; }; #define for_each_dapm_widgets(list, i, widget) \ diff --git a/include/sound/soc.h b/include/sound/soc.h index f0e4f36f83bf..9944b978ae70 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1146,7 +1146,7 @@ struct snd_soc_pcm_runtime { unsigned int fe_compr:1; /* for Dynamic PCM */ int num_components; - struct snd_soc_component *components[0]; /* CPU/Codec/Platform */ + struct snd_soc_component *components[]; /* CPU/Codec/Platform */ }; #define for_each_rtd_components(rtd, i, component) \ for ((i) = 0; \ diff --git a/include/uapi/sound/asound.h b/include/uapi/sound/asound.h index 535a7229e1d9..3809993f6cb4 100644 --- a/include/uapi/sound/asound.h +++ b/include/uapi/sound/asound.h @@ -1071,7 +1071,7 @@ struct snd_ctl_elem_value { struct snd_ctl_tlv { unsigned int numid; /* control element numeric identification */ unsigned int length; /* in bytes aligned to 4 */ - unsigned int tlv[0]; /* first TLV */ + unsigned int tlv[]; /* first TLV */ }; #define SNDRV_CTL_IOCTL_PVERSION _IOR('U', 0x00, int) diff --git a/include/uapi/sound/skl-tplg-interface.h b/include/uapi/sound/skl-tplg-interface.h index 9eee32f5e407..62f76f154d19 100644 --- a/include/uapi/sound/skl-tplg-interface.h +++ b/include/uapi/sound/skl-tplg-interface.h @@ -149,7 +149,7 @@ struct skl_dfw_algo_data { __u32 rsvd:30; __u32 param_id; __u32 max; - char params[0]; + char params[]; } __packed; enum skl_tkn_dir { diff --git a/sound/core/oss/pcm_plugin.h b/sound/core/oss/pcm_plugin.h index 8d2f7a4e3ab6..46e273bd4a78 100644 --- a/sound/core/oss/pcm_plugin.h +++ b/sound/core/oss/pcm_plugin.h @@ -64,7 +64,7 @@ struct snd_pcm_plugin { char *buf; snd_pcm_uframes_t buf_frames; struct snd_pcm_plugin_channel *buf_channels; - char extra_data[0]; + char extra_data[]; }; int snd_pcm_plugin_build(struct snd_pcm_substream *handle, diff --git a/sound/firewire/fireworks/fireworks.h b/sound/firewire/fireworks/fireworks.h index dda797209a27..654e28a6669f 100644 --- a/sound/firewire/fireworks/fireworks.h +++ b/sound/firewire/fireworks/fireworks.h @@ -177,7 +177,7 @@ struct snd_efw_phys_meters { u32 in_meters; u32 reserved4; u32 reserved5; - u32 values[0]; + u32 values[]; } __packed; enum snd_efw_clock_source { SND_EFW_CLOCK_SOURCE_INTERNAL = 0, diff --git a/sound/soc/intel/atom/sst-atom-controls.h b/sound/soc/intel/atom/sst-atom-controls.h index 5356e954a732..620b48d2a064 100644 --- a/sound/soc/intel/atom/sst-atom-controls.h +++ b/sound/soc/intel/atom/sst-atom-controls.h @@ -410,7 +410,7 @@ struct sst_cmd_set_gain_dual { struct sst_cmd_set_params { struct sst_destination_id dst; u16 command_id; - char params[0]; + char params[]; } __packed; diff --git a/sound/soc/intel/skylake/skl-i2s.h b/sound/soc/intel/skylake/skl-i2s.h index d7c15873c0d4..dfce91e11be1 100644 --- a/sound/soc/intel/skylake/skl-i2s.h +++ b/sound/soc/intel/skylake/skl-i2s.h @@ -46,7 +46,7 @@ struct skl_i2s_config_mclk { struct skl_i2s_config_mclk_ext { u32 mdivctrl; u32 mdivr_count; - u32 mdivr[0]; + u32 mdivr[]; } __packed; struct skl_i2s_config_blob_signature { diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index e967800dbb62..d2cd8ef8e97f 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -119,7 +119,7 @@ struct skl_cpr_gtw_cfg { struct skl_dma_control { u32 node_id; u32 config_length; - u32 config_data[0]; + u32 config_data[]; } __packed; struct skl_cpr_cfg { @@ -152,7 +152,7 @@ struct skl_up_down_mixer_cfg { struct skl_algo_cfg { struct skl_base_cfg base_cfg; - char params[0]; + char params[]; } __packed; struct skl_base_outfmt_cfg { diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index 2bfbf59277c4..26057f38a014 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -49,7 +49,7 @@ struct skl_astate_param { struct skl_astate_config { u32 count; - struct skl_astate_param astate_table[0]; + struct skl_astate_param astate_table[]; }; struct skl_fw_config { diff --git a/sound/usb/usx2y/usbusx2y.h b/sound/usb/usx2y/usbusx2y.h index e0f77172ce8f..144b85f57bd2 100644 --- a/sound/usb/usx2y/usbusx2y.h +++ b/sound/usb/usx2y/usbusx2y.h @@ -18,7 +18,7 @@ struct snd_usX2Y_AsyncSeq { struct snd_usX2Y_urbSeq { int submitted; int len; - struct urb *urb[0]; + struct urb *urb[]; }; #include "usx2yhwdeppcm.h" -- 2.25.0

2 2

[PATCH v2] ASoC: tas2562: Add support for digital volume control
by Dan Murphy 20 Feb '20

20 Feb '20

Add support for digital volume control. There is no dedicated register for volume control but instead there are 4. The values of the registers are determined with exponential floating point math. So a table was created with register values for 2dB step increments from -110dB to 0dB. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- v2 - Fix init values to lower volume CFG4 is the LSB and CFG1 is MSB sound/soc/codecs/tas2562.c | 94 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2562.h | 6 ++- 2 files changed, 98 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index b2682c2360b6..92e4bf34b35c 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -26,6 +26,24 @@ #define TAS2562_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FORMAT_S32_LE) +/* DVC equation involves floating point math + * round(10^(volume in dB/20)*2^30) + * so create a lookup table for 2dB step + */ +static const unsigned int float_vol_db_lookup[] = { +0x00000d43, 0x000010b2, 0x00001505, 0x00001a67, 0x00002151, +0x000029f1, 0x000034cd, 0x00004279, 0x000053af, 0x0000695b, +0x0000695b, 0x0000a6fa, 0x0000d236, 0x000108a4, 0x00014d2a, +0x0001a36e, 0x00021008, 0x000298c0, 0x000344df, 0x00041d8f, +0x00052e5a, 0x000685c8, 0x00083621, 0x000a566d, 0x000d03a7, +0x0010624d, 0x0014a050, 0x0019f786, 0x0020b0bc, 0x0029279d, +0x0033cf8d, 0x004139d3, 0x00521d50, 0x00676044, 0x0082248a, +0x00a3d70a, 0x00ce4328, 0x0103ab3d, 0x0146e75d, 0x019b8c27, +0x02061b89, 0x028c423f, 0x03352529, 0x0409c2b0, 0x05156d68, +0x080e9f96, 0x0a24b062, 0x0cc509ab, 0x10137987, 0x143d1362, +0x197a967f, 0x2013739e, 0x28619ae9, 0x32d64617, 0x40000000 +}; + struct tas2562_data { struct snd_soc_component *component; struct gpio_desc *sdz_gpio; @@ -34,6 +52,7 @@ struct tas2562_data { struct i2c_client *client; int v_sense_slot; int i_sense_slot; + int volume_lvl; }; static int tas2562_set_bias_level(struct snd_soc_component *component, @@ -334,6 +353,22 @@ static int tas2562_codec_probe(struct snd_soc_component *component) if (ret < 0) return ret; + /* Set the Digital volume to -110dB */ + ret = snd_soc_component_write(component, TAS2562_DVC_CFG4, 0x43); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG3, 0x0d); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG2, 0x00); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG1, 0x00); + if (ret) + return ret; + + tas2562->volume_lvl = 0; + return 0; } @@ -414,6 +449,50 @@ static int tas2562_dac_event(struct snd_soc_dapm_widget *w, return 0; } +static int tas2562_volume_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + + ucontrol->value.integer.value[0] = tas2562->volume_lvl; + return 0; +} + +static int tas2562_volume_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_component *component = snd_soc_kcontrol_component(kcontrol); + struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; + u32 reg_val; + + reg_val = float_vol_db_lookup[ucontrol->value.integer.value[0]/2]; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG4, + (reg_val & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG3, + ((reg_val >> 8) & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG2, + ((reg_val >> 16) & 0xff)); + if (ret) + return ret; + ret = snd_soc_component_write(component, TAS2562_DVC_CFG1, + ((reg_val >> 24) & 0xff)); + if (ret) + return ret; + + tas2562->volume_lvl = ucontrol->value.integer.value[0]; + + return ret; +} + +/* Digital Volume Control. From 0 dB to -110 dB in 1 dB steps */ +static const DECLARE_TLV_DB_SCALE(dvc_tlv, -11000, 100, 0); + static DECLARE_TLV_DB_SCALE(tas2562_dac_tlv, 850, 50, 0); static const struct snd_kcontrol_new isense_switch = @@ -427,6 +506,17 @@ static const struct snd_kcontrol_new vsense_switch = static const struct snd_kcontrol_new tas2562_snd_controls[] = { SOC_SINGLE_TLV("Amp Gain Volume", TAS2562_PB_CFG1, 0, 0x1c, 0, tas2562_dac_tlv), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Digital Volume Control", + .index = 0, + .tlv.p = dvc_tlv, + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_soc_info_volsw, + .get = tas2562_volume_control_get, + .put = tas2562_volume_control_put, + .private_value = SOC_SINGLE_VALUE(TAS2562_DVC_CFG1, 0, 110, 0, 0) , + }, }; static const struct snd_soc_dapm_widget tas2562_dapm_widgets[] = { @@ -517,6 +607,10 @@ static const struct reg_default tas2562_reg_defaults[] = { { TAS2562_PB_CFG1, 0x20 }, { TAS2562_TDM_CFG0, 0x09 }, { TAS2562_TDM_CFG1, 0x02 }, + { TAS2562_DVC_CFG1, 0x40 }, + { TAS2562_DVC_CFG2, 0x40 }, + { TAS2562_DVC_CFG3, 0x00 }, + { TAS2562_DVC_CFG4, 0x00 }, }; static const struct regmap_config tas2562_regmap_config = { diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 6f55ebcf19ea..28e75fc431d0 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -35,8 +35,10 @@ #define TAS2562_REV_ID TAS2562_REG(0, 0x7d) /* Page 2 */ -#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x01) -#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x02) +#define TAS2562_DVC_CFG1 TAS2562_REG(2, 0x0c) +#define TAS2562_DVC_CFG2 TAS2562_REG(2, 0x0d) +#define TAS2562_DVC_CFG3 TAS2562_REG(2, 0x0e) +#define TAS2562_DVC_CFG4 TAS2562_REG(2, 0x0f) #define TAS2562_RESET BIT(0) -- 2.25.0

1 0

Re: [alsa-devel] No sound since 5.4 on skl_n88l25_s4567
by Kuninori Morimoto 20 Feb '20

20 Feb '20

Hi ojab > > Please take a look, I want my sound back ._. > > Forgot to mention, it's reproducible on linux master git. Hmm... This patch 1) removes unneeded if check 2) adds missing error check. 1) shouldn't have effect to your PC. Thus, maybe your PC failed by 2). It had been worked by accident before ? I guess your kernel is telling some message like this ? ASoC: no dapm match for %s --> %s --> %s Thank you for your help !! Best regards --- Kuninori Morimoto

4 13

[PATCH] ASoC: samsung: Update dependencies for Arizona machine drivers
by Charles Keepax 20 Feb '20

20 Feb '20

Currently it is possible to get the following bad config: WARNING: unmet direct dependencies detected for SND_SOC_WM5110 Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=y] && MFD_WM5110 [=n] commit ea00d95200d0 ("ASoC: Use imply for SND_SOC_ALL_CODECS") commit d8dd3f92a6ba ("ASoC: Fix SND_SOC_ALL_CODECS imply misc fallout") After these two patches the machine drivers still selects the SND_SOC_WM5110 symbol which doesn't take account of the dependency added on the MFD_WM5110 symbol, fix this by also adding a dependency on MFD_WM5110 itself. Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/samsung/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 1a0b163ca47b5..112911dc271ba 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -151,7 +151,7 @@ config SND_SOC_TOBERMORY config SND_SOC_BELLS tristate "Audio support for Wolfson Bells" - depends on MFD_ARIZONA && I2C && SPI_MASTER + depends on MFD_ARIZONA && MFD_WM5102 && MFD_WM5110 && I2C && SPI_MASTER depends on MACH_WLF_CRAGG_6410 || COMPILE_TEST select SND_SAMSUNG_I2S select SND_SOC_WM5102 @@ -204,7 +204,7 @@ config SND_SOC_ARNDALE config SND_SOC_SAMSUNG_TM2_WM5110 tristate "SoC I2S Audio support for WM5110 on TM2 board" - depends on SND_SOC_SAMSUNG && MFD_ARIZONA && I2C && SPI_MASTER + depends on SND_SOC_SAMSUNG && MFD_ARIZONA && MFD_WM5110 && I2C && SPI_MASTER depends on GPIOLIB || COMPILE_TEST select SND_SOC_MAX98504 select SND_SOC_WM5110 -- 2.11.0

2 1

[PATCH] ASoC: soc-core: fix for_rtd_codec_dai_rollback() macro
by Pierre-Louis Bossart 20 Feb '20

20 Feb '20

The use of parentheses to protect the argument is fine for (i)++ but not for (--i). Fixes: 83f94a2e293d61 ("ASoC: soc-core: add snd_soc_close_delayed_work()") Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- 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 f0e4f36f83bf..8a2266676b2d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1157,7 +1157,7 @@ struct snd_soc_pcm_runtime { ((i) < rtd->num_codecs) && ((dai) = rtd->codec_dais[i]); \ (i)++) #define for_each_rtd_codec_dai_rollback(rtd, i, dai) \ - for (; ((--i) >= 0) && ((dai) = rtd->codec_dais[i]);) + for (; (--(i) >= 0) && ((dai) = rtd->codec_dais[i]);) void snd_soc_close_delayed_work(struct snd_soc_pcm_runtime *rtd); -- 2.20.1

3 2

[PATCH] ASoC: dpcm: remove confusing trace in dpcm_get_be()
by Jerome Brunet 20 Feb '20

20 Feb '20

Now that dpcm_get_be() is used in dpcm_end_walk_at_be(), it is not a error if this function does not find a BE for the provided widget. Remove the related dev_err() trace which is confusing since things might be working as expected. When called from dpcm_add_paths(), it is an error if dpcm_get_be() fails to find a BE for the provided widget. The necessary error trace is already done in this case. Fixes: 027a48387183 ("ASoC: soc-pcm: use dpcm_get_be() at dpcm_end_walk_at_be()") Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/soc-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 63f67eb7c077..aff27c8599ef 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1270,9 +1270,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, } } - /* dai link name and stream name set correctly ? */ - dev_err(card->dev, "ASoC: can't get %s BE for %s\n", - stream ? "capture" : "playback", widget->name); + /* Widget provided is not a BE */ return NULL; } -- 2.24.1

4 3

[PATCH] ASoC: tas2562: Add support for ISENSE and VSENSE
by Dan Murphy 20 Feb '20

20 Feb '20

Add additional support for ISENSE and VSENSE feature for the TAS2562. This feature monitors the output to the loud speaker attempts to eliminate IR drop errors due to packaging. This feature is defined in Section 8.4.5 IV Sense of the data sheet. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- sound/soc/codecs/tas2562.c | 32 +++++++++++++++++++++++++++----- sound/soc/codecs/tas2562.h | 6 +++--- 2 files changed, 30 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index 5b4803a76719..b2682c2360b6 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -383,18 +383,34 @@ static int tas2562_dac_event(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct tas2562_data *tas2562 = snd_soc_component_get_drvdata(component); + int ret; switch (event) { case SND_SOC_DAPM_POST_PMU: - dev_info(tas2562->dev, "SND_SOC_DAPM_POST_PMU\n"); + ret = snd_soc_component_update_bits(component, + TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, + TAS2562_MUTE); + if (ret) + goto end; break; case SND_SOC_DAPM_PRE_PMD: - dev_info(tas2562->dev, "SND_SOC_DAPM_PRE_PMD\n"); + ret = snd_soc_component_update_bits(component, + TAS2562_PWR_CTRL, + TAS2562_MODE_MASK, + TAS2562_SHUTDOWN); + if (ret) + goto end; break; default: - break; + dev_err(tas2562->dev, "Not supported evevt\n"); + return -EINVAL; } +end: + if (ret < 0) + return ret; + return 0; } @@ -416,7 +432,6 @@ static const struct snd_kcontrol_new tas2562_snd_controls[] = { static const struct snd_soc_dapm_widget tas2562_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("ASI1", "ASI1 Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_MUX("ASI1 Sel", SND_SOC_NOPM, 0, 0, &tas2562_asi1_mux), - SND_SOC_DAPM_AIF_IN("DAC IN", "Playback", 0, SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC_E("DAC", NULL, SND_SOC_NOPM, 0, 0, tas2562_dac_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), SND_SOC_DAPM_SWITCH("ISENSE", TAS2562_PWR_CTRL, 3, 1, &isense_switch), @@ -431,7 +446,7 @@ static const struct snd_soc_dapm_route tas2562_audio_map[] = { {"ASI1 Sel", "Left", "ASI1"}, {"ASI1 Sel", "Right", "ASI1"}, {"ASI1 Sel", "LeftRightDiv2", "ASI1"}, - { "DAC", NULL, "DAC IN" }, + { "DAC", NULL, "ASI1 Sel" }, { "OUT", NULL, "DAC" }, {"ISENSE", "Switch", "IMON"}, {"VSENSE", "Switch", "VMON"}, @@ -472,6 +487,13 @@ static struct snd_soc_dai_driver tas2562_dai[] = { .rates = SNDRV_PCM_RATE_8000_192000, .formats = TAS2562_FORMATS, }, + .capture = { + .stream_name = "ASI1 Capture", + .channels_min = 0, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = TAS2562_FORMATS, + }, .ops = &tas2562_speaker_dai_ops, }, }; diff --git a/sound/soc/codecs/tas2562.h b/sound/soc/codecs/tas2562.h index 62e659ab786d..6f55ebcf19ea 100644 --- a/sound/soc/codecs/tas2562.h +++ b/sound/soc/codecs/tas2562.h @@ -40,7 +40,7 @@ #define TAS2562_RESET BIT(0) -#define TAS2562_MODE_MASK 0x3 +#define TAS2562_MODE_MASK GENMASK(1,0) #define TAS2562_ACTIVE 0x0 #define TAS2562_MUTE 0x1 #define TAS2562_SHUTDOWN 0x2 @@ -73,8 +73,8 @@ #define TAS2562_TDM_CFG2_RXWLEN_24B BIT(3) #define TAS2562_TDM_CFG2_RXWLEN_32B (BIT(2) | BIT(3)) -#define TAS2562_VSENSE_POWER_EN BIT(2) -#define TAS2562_ISENSE_POWER_EN BIT(3) +#define TAS2562_VSENSE_POWER_EN 2 +#define TAS2562_ISENSE_POWER_EN 3 #define TAS2562_TDM_CFG5_VSNS_EN BIT(6) #define TAS2562_TDM_CFG5_VSNS_SLOT_MASK GENMASK(5, 0) -- 2.25.0

2 3

[alsa-devel] Implicit feedback solution for Boss GT-1 and maybe other devices
by Mike Oliphant 20 Feb '20

20 Feb '20

I received a very helpful email response to my previous thread about trying to get the Boss GT-1 pedalboard working reliably, and as a result I now have a working solution. The issue seems to be that the GT-1 does not send any capture data until it gets some playback data first. This block in the "snd_usb_endpoint_start()" function in endpoint.c: "if (snd_usb_endpoint_implicit_feedback_sink(ep)) { ... }" causes the system to wait to send output until some capture urbs are received. As a result, both sides end up waiting on the other, and nothing happens. Simply commenting out that "if" block makes the GT-1 work perfectly with implicit feedback enabled. I plan to submit a patch for this after I do a bit of legwork to see if the fix works for other recent Boss devices (Katana, other GT devices, etc.). Perhaps making the relevant "if" block conditional on a flag, with a USB_ID switch statement to set the flag? Any feedback would be welcome. Thanks, Mike

2 1

Enable Headset Mic for Dell desktop
by Kailang 20 Feb '20

20 Feb '20

Hi Takashi, Attach patch was supported headset Mic. BR, Kailang

1 0

usage of snd_rawmidi_status_get_avail()
by GitHub issues - edited 20 Feb '20

20 Feb '20

alsa-project/alsa-lib issue #26 was edited from MusicMaker: In non blocked raw MIDI, is snd_rawmidi_status_get_avail supposed to return the number of available bytes that can be read using a snd_rawmidi_read. (port is opened for input only) It seems get_avail always returns 0 even there is MIDI data when using a read. Else is there a(nother) way to find the number of unread received bytes in the buffer? snd_rawmidi_status_alloca (&status); snd_rawmidi_status(client, status); snd_rawmidi_status_get_avail(status); Issue URL : https://github.com/alsa-project/alsa-lib/issues/26 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

usage of snd_rawmidi_status_get_avail()
by GitHub issues - edited 20 Feb '20

20 Feb '20

alsa-project/alsa-lib issue #26 was edited from MusicMaker: In non blocked raw MIDI, is snd_rawmidi_status_get_avail supposed to return the number of available bytes that can be read using a snd_rawmidi_read. (port is opened for read only) It seems get_avail always returns 0 even there is MIDI data when using a read. Else is there a(nother) way to find the number of unread received bytes in the buffer? snd_rawmidi_status_alloca (&status); snd_rawmidi_status(client, status); snd_rawmidi_status_get_avail(status); Issue URL : https://github.com/alsa-project/alsa-lib/issues/26 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

usage of snd_rawmidi_status_get_avail()
by GitHub issues - edited 20 Feb '20

20 Feb '20

alsa-project/alsa-lib issue #26 was edited from MusicMaker: In non blocked raw MIDI, is snd_rawmidi_status_get_avail supposed to return the number of available bytes that can be read via snd_rawmidi_read. (port is opened for read only) It seems get_avail always returns 0 even there is MIDI data when using a read. Else is there a(nother) way to find the number of unread received bytes in the buffer? snd_rawmidi_status_alloca (&status); snd_rawmidi_status(client, status); snd_rawmidi_status_get_avail(status); Issue URL : https://github.com/alsa-project/alsa-lib/issues/26 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

usage of snd_rawmidi_status_get_avail()
by GitHub issues - edited 20 Feb '20

20 Feb '20

alsa-project/alsa-lib issue #26 was edited from MusicMaker: In non blocked raw MIDI, is snd_rawmidi_status_get_avail supposed to return the number of available bytes than can read via snd_rawmidi_read. (port is opened for read only) It seems get_avail always returns 0 even there is MIDI data when using a read. Else is there a(nother) way to find the number of unread received bytes in the buffer? snd_rawmidi_status_alloca (&status); snd_rawmidi_status(client, status); snd_rawmidi_status_get_avail(status); Issue URL : https://github.com/alsa-project/alsa-lib/issues/26 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

usage of snd_rawmidi_status_get_avail()
by GitHub issues - edited 20 Feb '20

20 Feb '20

alsa-project/alsa-lib issue #26 was edited from MusicMaker: In non blocked raw MIDI, is snd_rawmidi_status_get_avail supposed to return the number of available bytes than can read via snd_rawmidi_read. (port is opened for read only) It seems always return 0 even there is MIDI data when using a read. Else is there a(nother) way to find the number of unread received bytes in the buffer? snd_rawmidi_status_alloca (&status); snd_rawmidi_status(client, status); snd_rawmidi_status_get_avail(status); Issue URL : https://github.com/alsa-project/alsa-lib/issues/26 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

usage of snd_rawmidi_status_get_avail()
by GitHub issues - opened 20 Feb '20

20 Feb '20

alsa-project/alsa-lib issue #26 was opened from MusicMaker: In non blocked raw MIDI mode, is snd_rawmidi_status_get_avail supposed to return the number of available bytes than can read via snd_rawmidi_read. (port is opened for read only) It seems always return 0 even there is MIDI data when using a read. Else is there a(nother) way to find the number of unread received bytes in the buffer? snd_rawmidi_status_alloca (&status); snd_rawmidi_status(client, status); snd_rawmidi_status_get_avail(status); Issue URL : https://github.com/alsa-project/alsa-lib/issues/26 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

sytem reboots when initializing Edirol FA-101 firewire audio interface
by Steve Morris 20 Feb '20

20 Feb '20

Hi All, I'm running Arch Linux x86_64 My system consistently reboots when I power on my FA-101 when running kernels 5.5.1-4. Downgrading to 5.4.15 allows everything to work properly. Here's the outpu of: journalctl | grep -E 'Reboot|firewire|fw|bebob|alsa|jack' This is a good init under 5.4.15: -- Reboot -- powered on interface while running 5.4.15, initialized properly Feb 18 18:35:37 hostname kernel: firewire_ohci 0000:05:00.0: enabling device (0080 -> 0083) Feb 18 18:35:37 hostname kernel: firewire_ohci 0000:05:00.0: added OHCI v1.10 device as card 0, 4 IR + 8 IT contexts, quirks 0x11 Feb 18 18:35:37 hostname kernel: firewire_core 0000:05:00.0: created device fw0: GUID 0011223333666677, S400 Feb 18 18:35:37 hostname systemd-udevd[541]: controlC0: Process '/usr/bin/alsactl restore 0' failed with exit code 99. Feb 18 18:35:37 hostname systemd-udevd[541]: controlC1: Process '/usr/bin/alsactl restore 1' failed with exit code 99. Feb 18 18:35:38 hostname kernel: amdgpu: [powerplay] smu driver if version = 0x00000033, smu fw if version = 0x00000035, smu fw version = 0x002a3200 (42.50.0) Feb 18 18:36:08 hostname kernel: firewire_ohci 0000:05:00.0: isochronous cycle inconsistent Feb 18 18:36:08 hostname kernel: firewire_core 0000:05:00.0: created device fw1: GUID 0040ab0000c20bc1, S400 Feb 18 18:36:08 hostname kernel: firewire_core 0000:05:00.0: phy config: new root=ffc1, gap_count=5 Feb 18 18:36:10 hostname kernel: firewire_core 0000:05:00.0: BM lock failed (timeout), making local node (ffc0) root Feb 18 18:36:10 hostname kernel: firewire_core 0000:05:00.0: phy config: new root=ffc0, gap_count=5 Feb 18 18:36:12 hostname systemd-udevd[1532]: controlC2: Process '/usr/bin/alsactl restore 2' failed with exit code 99. Feb 18 18:36:15 hostname kernel: snd-bebob fw1.0: transaction failed: no ack Feb 18 18:36:15 hostname kernel: snd-bebob fw1.0: fail to get an input for MSU in plug 7: -5 Feb 18 18:36:15 hostname kernel: snd-bebob fw1.0: transaction failed: no ack Feb 18 18:36:15 hostname kernel: snd-bebob fw1.0: fail to get an input for MSU in plug 7: -5 This is a bad init under 5.5.4: -- Reboot -- Powered on interface when machine was running. Feb 18 18:13:37 hostname kernel: firewire_ohci 0000:05:00.0: enabling device (0080 -> 0083) Feb 18 18:13:37 hostname kernel: firewire_ohci 0000:05:00.0: added OHCI v1.10 device as card 0, 4 IR + 8 IT contexts, quirks 0x11 Feb 18 18:13:37 hostname kernel: firewire_core 0000:05:00.0: created device fw0: GUID 0011223333666677, S400 Feb 18 18:13:37 hostname kernel: audit: type=1130 audit(1582078417.360:3): pid=1 uid=0 auid=4294967295 ses=4294967295 msg='unit=ufw comm="systemd" exe="/usr/lib/systemd/systemd" hostname=? addr=? terminal=? res=success' Feb 18 18:13:37 hostname audit[1]: SERVICE_START pid=1 uid=0 auid=4294967295 ses=4294967295 msg='unit=ufw comm="systemd" exe="/usr/lib/systemd/systemd" hostname=? addr=? terminal=? res=success' Feb 18 18:13:37 hostname systemd-udevd[539]: controlC0: Process '/usr/bin/alsactl restore 0' failed with exit code 99. Feb 18 18:13:37 hostname systemd-udevd[530]: controlC1: Process '/usr/bin/alsactl restore 1' failed with exit code 99. Feb 18 18:13:38 hostname kernel: amdgpu: [powerplay] smu driver if version = 0x00000033, smu fw if version = 0x00000035, smu fw version = 0x002a3200 (42.50.0) Feb 18 18:19:45 hostname kernel: firewire_core 0000:05:00.0: phy config: new root=ffc1, gap_count=5 Feb 18 18:19:48 hostname kernel: firewire_core 0000:05:00.0: phy config: new root=ffc1, gap_count=5 Feb 18 18:19:48 hostname kernel: firewire_core 0000:05:00.0: created device fw1: GUID 0040ab0000c20bc1, S400 Feb 18 18:19:52 hostname systemd-udevd[1682]: controlC2: Process '/usr/bin/alsactl restore 2' failed with exit code 99. Feb 18 18:20:08 hostname kernel: snd-bebob fw1.0: transaction failed: no ack Feb 18 18:20:08 hostname kernel: snd-bebob fw1.0: fail to get an input for MSU in plug 7: -5 Feb 18 18:20:08 hostname kernel: snd-bebob fw1.0: transaction failed: no ack Feb 18 18:20:08 hostname kernel: snd-bebob fw1.0: fail to get an input for MSU in plug 7: -5 Feb 18 18:20:12 hostname kernel: firewire_core 0000:05:00.0: phy config: new root=ffc1, gap_count=5 Feb 18 18:21:24 hostname kernel: firewire_ohci 0000:05:00.0: AMD-Vi: Event logged [IO_PAGE_FAULT domain=0x0001 address=0xd5080000 flags=0x0000] Followed by 125 more O_PAGE_FAULT and then the reboot, I'll be happy to provide additional information if needed. Steve

2 1

[PATCH 0/3] ASoC: snprintf() usage fixes
by Takashi Iwai 19 Feb '20

19 Feb '20

Hi, this is a series of patches I spotted during debugging other code. Intel skl and ASoC PCM core invokes succeeding calls of snprintf() with the incorrect expectation of the return value, which may lead to a buffer overflow. The first patch is yet another bug I found during working on this, and it's rather a real bug that can be seen clearly. Takashi === Takashi Iwai (3): ASoC: intel: skl: Fix pin debug prints ASoC: intel: skl: Fix possible buffer overflow in debug outputs ASoC: pcm: Fix possible buffer overflow in dpcm state sysfs output sound/soc/intel/skylake/skl-debug.c | 32 +++++++++++++++++--------------- sound/soc/soc-pcm.c | 16 ++++++++-------- 2 files changed, 25 insertions(+), 23 deletions(-) -- 2.16.4

3 7

[PATCH v2] ASoC: MT6660 update to 1.0.8_G
by Jeff Chang 19 Feb '20

19 Feb '20

From: Jeff Chang <jeff_chang(a)richtek.com> 1. add mt6660_component_settign for Component INIT Setting Signed-off-by: Jeff Chang <jeff_chang(a)richtek.com> --- sound/soc/codecs/mt6660.c | 78 +++++++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 75 insertions(+), 3 deletions(-) changelogs between v2 & v1 - remove parse DT and apply function about Chip INIT SETTING. - use hard coding for MT6660 Component INIT SETTING. diff --git a/sound/soc/codecs/mt6660.c b/sound/soc/codecs/mt6660.c index a36c416..88cf228 100644 --- a/sound/soc/codecs/mt6660.c +++ b/sound/soc/codecs/mt6660.c @@ -9,7 +9,6 @@ #include <linux/i2c.h> #include <linux/pm_runtime.h> #include <linux/delay.h> -#include <linux/debugfs.h> #include <sound/soc.h> #include <sound/tlv.h> #include <sound/pcm_params.h> @@ -225,14 +224,87 @@ static int _mt6660_chip_power_on(struct mt6660_chip *chip, int on_off) 0x01, on_off ? 0x00 : 0x01); } +struct reg_table { + uint32_t addr; + uint32_t mask; + uint32_t val; +}; + +static const struct reg_table mt6660_setting_table[] = { + { 0x20, 0x80, 0x00 }, + { 0x30, 0x01, 0x00 }, + { 0x50, 0x1c, 0x04 }, + { 0xB1, 0x0c, 0x00 }, + { 0xD3, 0x03, 0x03 }, + { 0xE0, 0x01, 0x00 }, + { 0x98, 0x44, 0x04 }, + { 0xB9, 0xff, 0x82 }, + { 0xB7, 0x7777, 0x7273 }, + { 0xB6, 0x07, 0x03 }, + { 0x6B, 0xe0, 0x20 }, + { 0x07, 0xff, 0x70 }, + { 0xBB, 0xff, 0x20 }, + { 0x69, 0xff, 0x40 }, + { 0xBD, 0xffff, 0x17f8 }, + { 0x70, 0xff, 0x15 }, + { 0x7C, 0xff, 0x00 }, + { 0x46, 0xff, 0x1d }, + { 0x1A, 0xffffffff, 0x7fdb7ffe }, + { 0x1B, 0xffffffff, 0x7fdb7ffe }, + { 0x51, 0xff, 0x58 }, + { 0xA2, 0xff, 0xce }, + { 0x33, 0xffff, 0x7fff }, + { 0x4C, 0xffff, 0x0116 }, + { 0x16, 0x1800, 0x0800 }, + { 0x68, 0x1f, 0x07 }, +}; + +static int mt6660_component_setting(struct snd_soc_component *component) +{ + struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int ret = 0; + size_t i = 0; + + ret = _mt6660_chip_power_on(chip, 1); + if (ret < 0) { + dev_err(component->dev, "%s chip power on failed\n", __func__); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(mt6660_setting_table); i++) { + ret = snd_soc_component_update_bits(component, + mt6660_setting_table[i].addr, + mt6660_setting_table[i].mask, + mt6660_setting_table[i].val); + if (ret < 0) { + dev_err(component->dev, "%s update 0x%02x failed\n", + __func__, mt6660_setting_table[i].addr); + return ret; + } + } + + ret = _mt6660_chip_power_on(chip, 0); + if (ret < 0) { + dev_err(component->dev, "%s chip power off failed\n", __func__); + return ret; + } + + return 0; +} + static int mt6660_component_probe(struct snd_soc_component *component) { struct mt6660_chip *chip = snd_soc_component_get_drvdata(component); + int ret; dev_dbg(component->dev, "%s\n", __func__); snd_soc_component_init_regmap(component, chip->regmap); - return 0; + ret = mt6660_component_setting(component); + if (ret < 0) + dev_err(chip->dev, "mt6660 component setting failed\n"); + + return ret; } static void mt6660_component_remove(struct snd_soc_component *component) @@ -506,4 +578,4 @@ module_i2c_driver(mt6660_i2c_driver); MODULE_AUTHOR("Jeff Chang <jeff_chang(a)richtek.com>"); MODULE_DESCRIPTION("MT6660 SPKAMP Driver"); MODULE_LICENSE("GPL"); -MODULE_VERSION("1.0.7_G"); +MODULE_VERSION("1.0.8_G"); -- 2.7.4

2 1

[PATCH] ASoC: fix card registration regression.
by Jerome Brunet 19 Feb '20

19 Feb '20

This reverts commit b2354e4009a773c00054b964d937e1b81cb92078. This change might have been desirable to ensure the uniqueness of the component name. It would have helped to better support linux devices which register multiple components, something is which more common than initially thought. However, some card driver are directly using dev_name() to fill the component names of the dai_link which is a problem if want to change the way ASoC generates the component names. Until we figure out the appropriate way to deal with this, revert the change and keep the names as they were. There might be a couple of warning related to debugfs (which were already present before the change) but it is still better than breaking working audio cards. Cc: Marek Szyprowski <m.szyprowski(a)samsung.com> Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/soc-core.c | 29 +---------------------------- 1 file changed, 1 insertion(+), 28 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 30c17fde14ca..518b652cf872 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2442,33 +2442,6 @@ static int snd_soc_register_dais(struct snd_soc_component *component, return ret; } -static char *snd_soc_component_unique_name(struct device *dev, - struct snd_soc_component *component) -{ - struct snd_soc_component *pos; - int count = 0; - char *name, *unique; - - name = fmt_single_name(dev, &component->id); - if (!name) - return name; - - /* Count the number of components registred by the device */ - for_each_component(pos) { - if (dev == pos->dev) - count++; - } - - /* Keep naming as it is for the 1st component */ - if (!count) - return name; - - unique = devm_kasprintf(dev, GFP_KERNEL, "%s-%d", name, count); - devm_kfree(dev, name); - - return unique; -} - static int snd_soc_component_initialize(struct snd_soc_component *component, const struct snd_soc_component_driver *driver, struct device *dev) { @@ -2477,7 +2450,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, INIT_LIST_HEAD(&component->card_list); mutex_init(&component->io_mutex); - component->name = snd_soc_component_unique_name(dev, component); + component->name = fmt_single_name(dev, &component->id); if (!component->name) { dev_err(dev, "ASoC: Failed to allocate name\n"); return -ENOMEM; -- 2.24.1

3 2

[RFC PATCH 0/2] ASoC: mediatek: mt8183-da7219: use SND_SOC_DAPM_PINCTRL
by Tzung-Bi Shih 19 Feb '20

19 Feb '20

In previous discussion[1], Mark thinks we should have something general pinctrl integration with DAPM. SND_SOC_DAPM_PINCTRL is exactly what we want but I failed to find existing usage (maybe in topology files?). To be equivalent to mt8183-da7219's original implementation, the 1st patch selects sleep-state at the beginning. The 2nd patch attaches the DAPM widget to playback stream. When the stream is about to start, the power on sequence selects the active-state. When the stream is about to stop, the power off sequence selects the sleep-state. Note: the DAI links will connect to hdmi-codec, that's why the playback stream name is "I2S Playback". Note: this is a RFC series. If it makes sense, will also apply to mt8183-mt6358. [1]: https://patchwork.kernel.org/patch/11379735/ Tzung-Bi Shih (2): ASoC: dapm: select sleep_state when initializing PINCTRL widget ASoC: mediatek: mt8183-da7219: use SND_SOC_DAPM_PINCTRL in TDM out .../mediatek/mt8183/mt8183-da7219-max98357.c | 98 ++----------------- sound/soc/soc-dapm.c | 3 + 2 files changed, 13 insertions(+), 88 deletions(-) -- 2.25.0.265.gbab2e86ba0-goog

2 4

Applied "ASoC: soundwaire: qcom: use for_each_rtd_codec_dai() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: soundwaire: qcom: use for_each_rtd_codec_dai() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From ce83baca8526689c332135c0abca2667ba056009 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 19 Feb 2020 15:55:53 +0900 Subject: [PATCH] ASoC: soundwaire: qcom: use for_each_rtd_codec_dai() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87o8tvjcbc.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/soundwire/qcom.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 1c6c6a2e0def..fb30bbec999a 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -594,6 +594,7 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, struct qcom_swrm_ctrl *ctrl = dev_get_drvdata(dai->dev); struct snd_soc_pcm_runtime *rtd = substream->private_data; struct sdw_stream_runtime *sruntime; + struct snd_soc_dai *codec_dai; int ret, i; sruntime = sdw_alloc_stream(dai->name); @@ -602,12 +603,12 @@ static int qcom_swrm_startup(struct snd_pcm_substream *substream, ctrl->sruntime[dai->id] = sruntime; - for (i = 0; i < rtd->num_codecs; i++) { - ret = snd_soc_dai_set_sdw_stream(rtd->codec_dais[i], sruntime, + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_set_sdw_stream(codec_dai, sruntime, substream->stream); if (ret < 0 && ret != -ENOTSUPP) { dev_err(dai->dev, "Failed to set sdw stream on %s", - rtd->codec_dais[i]->name); + codec_dai->name); sdw_release_stream(sruntime); return ret; } -- 2.20.1

1 0

Applied "ASoC: qcom: sdm845: use for_each_rtd_codec_dai() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: qcom: sdm845: use for_each_rtd_codec_dai() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From a4eb41eef331d31b8593defa10b249e155e0314f Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 19 Feb 2020 15:56:01 +0900 Subject: [PATCH] ASoC: qcom: sdm845: use for_each_rtd_codec_dai() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87mu9fjcb4.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/sdm845.c | 20 +++++++++----------- 1 file changed, 9 insertions(+), 11 deletions(-) diff --git a/sound/soc/qcom/sdm845.c b/sound/soc/qcom/sdm845.c index 3b5547a27aad..5a23597261ac 100644 --- a/sound/soc/qcom/sdm845.c +++ b/sound/soc/qcom/sdm845.c @@ -43,14 +43,14 @@ static int sdm845_slim_snd_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_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; u32 rx_ch[SLIM_MAX_RX_PORTS], tx_ch[SLIM_MAX_TX_PORTS]; u32 rx_ch_cnt = 0, tx_ch_cnt = 0; int ret = 0, i; - for (i = 0 ; i < dai_link->num_codecs; i++) { - ret = snd_soc_dai_get_channel_map(rtd->codec_dais[i], + for_each_rtd_codec_dai(rtd, i, codec_dai) { + ret = snd_soc_dai_get_channel_map(codec_dai, &tx_ch_cnt, tx_ch, &rx_ch_cnt, rx_ch); if (ret != 0 && ret != -ENOTSUPP) { @@ -77,6 +77,7 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; int ret = 0, j; int channels, slot_width; @@ -125,8 +126,7 @@ static int sdm845_tdm_snd_hw_params(struct snd_pcm_substream *substream, } } - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dai(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name_prefix, "Left")) { ret = snd_soc_dai_set_tdm_slot( @@ -214,7 +214,6 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct sdm845_snd_data *pdata = snd_soc_card_get_drvdata(card); struct snd_jack *jack; - struct snd_soc_dai_link *dai_link = rtd->dai_link; /* * Codec SLIMBUS configuration * RX1, RX2, RX3, RX4, RX5, RX6, RX7, RX8, RX9, RX10, RX11, RX12, RX13 @@ -266,8 +265,8 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) } break; case SLIMBUS_0_RX...SLIMBUS_6_TX: - for (i = 0 ; i < dai_link->num_codecs; i++) { - rval = snd_soc_dai_set_channel_map(rtd->codec_dais[i], + for_each_rtd_codec_dai(rtd, i, codec_dai) { + rval = snd_soc_dai_set_channel_map(codec_dai, ARRAY_SIZE(tx_ch), tx_ch, ARRAY_SIZE(rx_ch), @@ -275,7 +274,7 @@ static int sdm845_dai_init(struct snd_soc_pcm_runtime *rtd) if (rval != 0 && rval != -ENOTSUPP) return rval; - snd_soc_dai_set_sysclk(rtd->codec_dais[i], 0, + snd_soc_dai_set_sysclk(codec_dai, 0, WCD934X_DEFAULT_MCLK_RATE, SNDRV_PCM_STREAM_PLAYBACK); } @@ -345,8 +344,7 @@ static int sdm845_snd_startup(struct snd_pcm_substream *substream) codec_dai_fmt |= SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_DSP_B; - for (j = 0; j < rtd->num_codecs; j++) { - codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dai(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name_prefix, "Left")) { -- 2.20.1

1 0

Applied "ASoC: qcom: apq8016_sbc: use for_each_rtd_codec_dai() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: qcom: apq8016_sbc: use for_each_rtd_codec_dai() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From cf4dae032096f1299cf390fd55da489cb445dcbb Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 19 Feb 2020 15:56:09 +0900 Subject: [PATCH] ASoC: qcom: apq8016_sbc: use for_each_rtd_codec_dai() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87lfozjcaw.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/apq8016_sbc.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/qcom/apq8016_sbc.c b/sound/soc/qcom/apq8016_sbc.c index ac75838bbfab..2d064f3bc9b6 100644 --- a/sound/soc/qcom/apq8016_sbc.c +++ b/sound/soc/qcom/apq8016_sbc.c @@ -34,8 +34,8 @@ struct apq8016_sbc_data { static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai; struct snd_soc_component *component; - struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_card *card = rtd->card; struct apq8016_sbc_data *pdata = snd_soc_card_get_drvdata(card); int i, rval; @@ -90,10 +90,9 @@ static int apq8016_sbc_dai_init(struct snd_soc_pcm_runtime *rtd) pdata->jack_setup = true; } - for (i = 0 ; i < dai_link->num_codecs; i++) { - struct snd_soc_dai *dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { - component = dai->component; + component = codec_dai->component; /* Set default mclk for internal codec */ rval = snd_soc_component_set_sysclk(component, 0, 0, DEFAULT_MCLK_RATE, SND_SOC_CLOCK_IN); -- 2.20.1

1 0

Applied "ASoC: intel: cml_rt1011_rt5682: use for_each_rtd_codec_dai() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: intel: cml_rt1011_rt5682: use for_each_rtd_codec_dai() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 225c53a8cfb6fdd8defbbf72e8dcfb3801f7f51e Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 19 Feb 2020 15:56:15 +0900 Subject: [PATCH] ASoC: intel: cml_rt1011_rt5682: use for_each_rtd_codec_dai() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87k14jjcaq.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/cml_rt1011_rt5682.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/cml_rt1011_rt5682.c b/sound/soc/intel/boards/cml_rt1011_rt5682.c index dd80d0186a6c..02aa18d24319 100644 --- a/sound/soc/intel/boards/cml_rt1011_rt5682.c +++ b/sound/soc/intel/boards/cml_rt1011_rt5682.c @@ -164,8 +164,7 @@ static int cml_rt1011_hw_params(struct snd_pcm_substream *substream, srate = params_rate(params); - for (i = 0; i < rtd->num_codecs; i++) { - codec_dai = rtd->codec_dais[i]; + for_each_rtd_codec_dai(rtd, i, codec_dai) { /* 100 Fs to drive 24 bit data */ ret = snd_soc_dai_set_pll(codec_dai, 0, RT1011_PLL1_S_BCLK, -- 2.20.1

1 0

Applied "ASoC: intel: kbl_da7219_max98927: use for_each_rtd_codec_dai() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: intel: kbl_da7219_max98927: use for_each_rtd_codec_dai() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 89a2870f6be6aa75de2df22f8baa982c2d7d86e8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 19 Feb 2020 15:56:20 +0900 Subject: [PATCH] ASoC: intel: kbl_da7219_max98927: use for_each_rtd_codec_dai() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87imk3jcal.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/kbl_da7219_max98927.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 7a13e9b35187..88f69e3697d2 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -176,10 +176,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *runtime = substream->private_data; + struct snd_soc_dai *codec_dai; int ret, j; - for (j = 0; j < runtime->num_codecs; j++) { - struct snd_soc_dai *codec_dai = runtime->codec_dais[j]; + for_each_rtd_codec_dai(runtime, j, codec_dai) { if (!strcmp(codec_dai->component->name, MAX98927_DEV0_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); @@ -221,10 +221,10 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, static int kabylake_ssp0_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; int j, ret; - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dai(rtd, j, codec_dai) { const char *name = codec_dai->component->name; struct snd_soc_component *component = codec_dai->component; struct snd_soc_dapm_context *dapm = -- 2.20.1

1 0

Applied "ASoC: mediatek: mt8183-da7219-max98357: use for_each_rtd_codec_dai() macro" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: mediatek: mt8183-da7219-max98357: use for_each_rtd_codec_dai() macro has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 56f1003f65830697bd68ae1e26b5561e4cbe6523 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 19 Feb 2020 15:56:25 +0900 Subject: [PATCH] ASoC: mediatek: mt8183-da7219-max98357: use for_each_rtd_codec_dai() macro Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/87h7znjcag.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 03d104fbe185..4a5ef07e956b 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -40,6 +40,7 @@ static int mt8183_da7219_i2s_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; unsigned int rate = params_rate(params); unsigned int mclk_fs_ratio = 256; unsigned int mclk_fs = rate * mclk_fs_ratio; @@ -51,8 +52,7 @@ static int mt8183_da7219_i2s_hw_params(struct snd_pcm_substream *substream, if (ret < 0) dev_err(rtd->dev, "failed to set cpu dai sysclk\n"); - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dai(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, "da7219.5-001a")) { ret = snd_soc_dai_set_sysclk(codec_dai, @@ -82,10 +82,10 @@ static int mt8183_da7219_i2s_hw_params(struct snd_pcm_substream *substream, static int mt8183_da7219_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; int ret = 0, j; - for (j = 0; j < rtd->num_codecs; j++) { - struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + for_each_rtd_codec_dai(rtd, j, codec_dai) { if (!strcmp(codec_dai->component->name, "da7219.5-001a")) { ret = snd_soc_dai_set_pll(codec_dai, -- 2.20.1

1 0

Applied "ASoC: dt-bindings: Make RK3328 codec GPIO explicit" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: dt-bindings: Make RK3328 codec GPIO explicit has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e14980976534d9d94f5cddd70033707965482ede Mon Sep 17 00:00:00 2001 From: Robin Murphy <robin.murphy(a)arm.com> Date: Tue, 18 Feb 2020 21:31:58 +0000 Subject: [PATCH] ASoC: dt-bindings: Make RK3328 codec GPIO explicit Existing RK3328 codec drivers have overloaded the GRF phandle to assume implicit control of the limited-function GPIO_MUTE pin, which is usually used to enable an external audio line driver IC. Since this pin has a proper binding of its own (see gpio/rockchip,rk3328-grf-gpio.txt), make a GPIO explicit in the codec binding too. This will help avoid ambiguity on boards that use that pin for some other purpose. (and while touching the example, enforce the "don't include status" rule) Signed-off-by: Robin Murphy <robin.murphy(a)arm.com> Link: https://lore.kernel.org/r/5f7a399dea8a9dedef57f6f99f0f6ab1c1fdc56a.15813767… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/rockchip,rk3328-codec.txt | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt index 2469588c7ccb..1ecd75d2032a 100644 --- a/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt +++ b/Documentation/devicetree/bindings/sound/rockchip,rk3328-codec.txt @@ -10,6 +10,11 @@ Required properties: - clock-names: should be "pclk". - spk-depop-time-ms: speak depop time msec. +Optional properties: + +- mute-gpios: GPIO specifier for external line driver control (typically the + dedicated GPIO_MUTE pin) + Example for rk3328 internal codec: codec: codec@ff410000 { @@ -18,6 +23,6 @@ codec: codec@ff410000 { rockchip,grf = <&grf>; clocks = <&cru PCLK_ACODEC>; clock-names = "pclk"; + mute-gpios = <&grf_gpio 0 GPIO_ACTIVE_LOW>; spk-depop-time-ms = 100; - status = "disabled"; }; -- 2.20.1

1 0

[alsa-devel] [PATCH 0/2] ASoC: topology: Fix memory leaks with ABI version mismatch
by Dragos Tarcatu 19 Feb '20

19 Feb '20

The topology loader in soc-topology.c is currently accepting ABI version mismatches for some topology components in order to keep backwards compatibility with ABIv4. When dealing with these ABI mismatch situations, temporary copies of the v4 objects are allocated and passed on to the right driver topology handlers. After that they're supposed to be freed. However, memory leaks are currently possible in soc_tplg_link_elems_load() and also soc_tplg_manifest_load(). These patches fix these. I've originally created a PR on github for SOF at: https://github.com/thesofproject/linux/pull/1771 as my impression was this can affect older SOF toplogies. But as it turns out this may impact Skylake instead. Dragos Tarcatu (2): ASoC: topology: Fix memleak in soc_tplg_link_elems_load() ASoC: topology: Fix memleak in soc_tplg_manifest_load() sound/soc/soc-topology.c | 17 ++++++++++------- 1 file changed, 10 insertions(+), 7 deletions(-) -- 2.17.1

2 4

Applied "ASoC: meson: add t9015 internal codec binding documentation" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: meson: add t9015 internal codec binding documentation has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 5c36abcd2621adc3d50d05628f0ef0be6e7840a9 Mon Sep 17 00:00:00 2001 From: Jerome Brunet <jbrunet(a)baylibre.com> Date: Wed, 19 Feb 2020 18:35:02 +0100 Subject: [PATCH] ASoC: meson: add t9015 internal codec binding documentation Add the DT binding documention of the internal DAC found in the Amlogic gxl, g12a and sm1 SoC family. Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> Link: https://lore.kernel.org/r/20200219173503.1112561-2-jbrunet@baylibre.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../bindings/sound/amlogic,t9015.yaml | 58 +++++++++++++++++++ 1 file changed, 58 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,t9015.yaml diff --git a/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml new file mode 100644 index 000000000000..b7c38c2b5b54 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/amlogic,t9015.yaml @@ -0,0 +1,58 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/amlogic,t9015.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Amlogic T9015 Internal Audio DAC + +maintainers: + - Jerome Brunet <jbrunet(a)baylibre.com> + +properties: + $nodename: + pattern: "^audio-controller@.*" + + "#sound-dai-cells": + const: 0 + + compatible: + items: + - const: amlogic,t9015 + + clocks: + items: + - description: Peripheral clock + + clock-names: + items: + - const: pclk + + reg: + maxItems: 1 + + resets: + maxItems: 1 + +required: + - "#sound-dai-cells" + - compatible + - reg + - clocks + - clock-names + - resets + +examples: + - | + #include <dt-bindings/clock/g12a-clkc.h> + #include <dt-bindings/reset/amlogic,meson-g12a-reset.h> + + acodec: audio-controller@32000 { + compatible = "amlogic,t9015"; + reg = <0x0 0x32000 0x0 0x14>; + #sound-dai-cells = <0>; + clocks = <&clkc CLKID_AUDIO_CODEC>; + clock-names = "pclk"; + resets = <&reset RESET_AUDIO_CODEC>; + }; + -- 2.20.1

1 0

[PATCH v3 0/2] ASoC: meson: add internal DAC support
by Jerome Brunet 19 Feb '20

19 Feb '20

This patchset adds support for the internal audio DAC provided on the Amlogic gxl, g12a and sm1 SoC families. On each of these SoC families, there is glue between this codec on the audio provider. The architecture is similar to the one used for the synopsys hdmi codec on these SoCs Changes since v2 [1]: * Fix Mute kcontrol name Changes since v1 [0]: * Change some kcontrol names * Move DAC sources to DAPM [0]: https://lore.kernel.org/r/20200219133646.1035506-1-jbrunet@baylibre.com [1]: https://lore.kernel.org/r/20200219161625.1078051-1-jbrunet@baylibre.com Jerome Brunet (2): ASoC: meson: add t9015 internal codec binding documentation ASoC: meson: add t9015 internal DAC driver .../bindings/sound/amlogic,t9015.yaml | 58 +++ sound/soc/meson/Kconfig | 8 + sound/soc/meson/Makefile | 2 + sound/soc/meson/t9015.c | 333 ++++++++++++++++++ 4 files changed, 401 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,t9015.yaml create mode 100644 sound/soc/meson/t9015.c -- 2.24.1

2 3

[alsa-devel] [PATCH v2] ASoC: dt-bindings: stm32: convert i2s to json-schema
by Olivier Moysan 19 Feb '20

19 Feb '20

Convert the STM32 I2S bindings to DT schema format using json-schema. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- Changes in v2: - Define items order for clock and dma properties --- .../bindings/sound/st,stm32-i2s.txt | 62 ------------- .../bindings/sound/st,stm32-i2s.yaml | 87 +++++++++++++++++++ 2 files changed, 87 insertions(+), 62 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt deleted file mode 100644 index cbf24bcd1b8d..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt +++ /dev/null @@ -1,62 +0,0 @@ -STMicroelectronics STM32 SPI/I2S Controller - -The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode. -Only some SPI instances support I2S. - -Required properties: - - compatible: Must be "st,stm32h7-i2s" - - reg: Offset and length of the device's register set. - - interrupts: Must contain the interrupt line id. - - clocks: Must contain phandle and clock specifier pairs for each entry - in clock-names. - - clock-names: Must contain "i2sclk", "pclk", "x8k" and "x11k". - "i2sclk": clock which feeds the internal clock generator - "pclk": clock which feeds the peripheral bus interface - "x8k": I2S parent clock for sampling rates multiple of 8kHz. - "x11k": I2S parent clock for sampling rates multiple of 11.025kHz. - - dmas: DMA specifiers for tx and rx dma. - See Documentation/devicetree/bindings/dma/stm32-dma.txt. - - dma-names: Identifier for each DMA request line. Must be "tx" and "rx". - - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml - -Optional properties: - - resets: Reference to a reset controller asserting the reset controller - -The device node should contain one 'port' child node with one child 'endpoint' -node, according to the bindings defined in Documentation/devicetree/bindings/ -graph.txt. - -Example: -sound_card { - compatible = "audio-graph-card"; - dais = <&i2s2_port>; -}; - -i2s2: audio-controller@40003800 { - compatible = "st,stm32h7-i2s"; - reg = <0x40003800 0x400>; - interrupts = <36>; - clocks = <&rcc PCLK1>, <&rcc SPI2_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>; - clock-names = "pclk", "i2sclk", "x8k", "x11k"; - dmas = <&dmamux2 2 39 0x400 0x1>, - <&dmamux2 3 40 0x400 0x1>; - dma-names = "rx", "tx"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_i2s2>; - - i2s2_port: port@0 { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - format = "i2s"; - }; - }; -}; - -audio-codec { - codec_port: port@0 { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml new file mode 100644 index 000000000000..f32410890589 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-i2s.yaml @@ -0,0 +1,87 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 SPI/I2S Controller + +maintainers: + - Olivier Moysan <olivier.moysan(a)st.com> + +description: + The SPI/I2S block supports I2S/PCM protocols when configured on I2S mode. + Only some SPI instances support I2S. + +properties: + compatible: + enum: + - st,stm32h7-i2s + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + items: + - description: clock feeding the peripheral bus interface. + - description: clock feeding the internal clock generator. + - description: I2S parent clock for sampling rates multiple of 8kHz. + - description: I2S parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - const: pclk + - const: i2sclk + - const: x8k + - const: x11k + + interrupts: + maxItems: 1 + + dmas: + items: + - description: audio capture DMA. + - description: audio playback DMA. + + dma-names: + items: + - const: rx + - const: tx + + resets: + maxItems: 1 + +required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - interrupts + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + i2s2: audio-controller@4000b000 { + compatible = "st,stm32h7-i2s"; + #sound-dai-cells = <0>; + reg = <0x4000b000 0x400>; + clocks = <&rcc SPI2>, <&rcc SPI2_K>, <&rcc PLL3_Q>, <&rcc PLL3_R>; + clock-names = "pclk", "i2sclk", "x8k", "x11k"; + interrupts = <GIC_SPI 35 IRQ_TYPE_LEVEL_HIGH>; + dmas = <&dmamux1 39 0x400 0x01>, + <&dmamux1 40 0x400 0x01>; + dma-names = "rx", "tx"; + pinctrl-names = "default"; + pinctrl-0 = <&i2s2_pins_a>; + }; + +... -- 2.17.1

3 2

[alsa-devel] [PATCH] ASoC: cpcap: Implement set_tdm_slot for voice call support
by Tony Lindgren 19 Feb '20

19 Feb '20

For using cpcap for voice calls, we need to route audio directly from the modem to cpcap for TDM (Time Division Multiplexing). The voice call is direct data between the modem and cpcap with no CPU involvment. In this mode, the cpcap related audio mixer controls work for the speaker selection and volume though. To do this, we need to implement standard snd_soc_dai_set_tdm_slot() for cpcap. Then the modem codec driver can use snd_soc_dai_set_sysclk(), snd_soc_dai_set_fmt(), and snd_soc_dai_set_tdm_slot() to configure a voice call. Let's add cpcap_voice_set_tdm_slot() for this, and cpcap_voice_call() helper to configure the additional registers needed for voice call. Let's also clear CPCAP_REG_VAUDIOC on init in case we have the bit for CPCAP_BIT_VAUDIO_MODE0 set on init. Cc: Arthur D. <spinal.by(a)gmail.com> Cc: Merlijn Wajer <merlijn(a)wizzup.org> Cc: Pavel Machek <pavel(a)ucw.cz> Cc: Sebastian Reichel <sre(a)kernel.org> Signed-off-by: Tony Lindgren <tony(a)atomide.com> --- sound/soc/codecs/cpcap.c | 123 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 123 insertions(+) diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c --- a/sound/soc/codecs/cpcap.c +++ b/sound/soc/codecs/cpcap.c @@ -16,6 +16,14 @@ #include <sound/soc.h> #include <sound/tlv.h> +/* Register 512 CPCAP_REG_VAUDIOC --- Audio Regulator and Bias Voltage */ +#define CPCAP_BIT_AUDIO_LOW_PWR 6 +#define CPCAP_BIT_AUD_LOWPWR_SPEED 5 +#define CPCAP_BIT_VAUDIOPRISTBY 4 +#define CPCAP_BIT_VAUDIO_MODE1 2 +#define CPCAP_BIT_VAUDIO_MODE0 1 +#define CPCAP_BIT_V_AUDIO_EN 0 + /* Register 513 CPCAP_REG_CC --- CODEC */ #define CPCAP_BIT_CDC_CLK2 15 #define CPCAP_BIT_CDC_CLK1 14 @@ -221,6 +229,7 @@ struct cpcap_reg_info { }; static const struct cpcap_reg_info cpcap_default_regs[] = { + { CPCAP_REG_VAUDIOC, 0x003F, 0x0000 }, { CPCAP_REG_CC, 0xFFFF, 0x0000 }, { CPCAP_REG_CC, 0xFFFF, 0x0000 }, { CPCAP_REG_CDI, 0xBFFF, 0x0000 }, @@ -1370,6 +1379,119 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; } +/* + * Configure codec for voice call if requested. + * + * We can configure most with snd_soc_dai_set_sysclk(), snd_soc_dai_set_fmt() + * and snd_soc_dai_set_tdm_slot(). This function configures the rest of the + * cpcap related hardware as CPU is not involved in the voice call. + */ +static int cpcap_voice_call(struct cpcap_audio *cpcap, struct snd_soc_dai *dai, + bool voice_call) +{ + int mask, err; + + /* Modem to codec VAUDIO_MODE1 */ + mask = BIT(CPCAP_BIT_VAUDIO_MODE1); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_VAUDIOC, + mask, voice_call ? mask : 0); + if (err) + return err; + + /* Clear MIC1_MUX for call */ + mask = BIT(CPCAP_BIT_MIC1_MUX); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_TXI, + mask, voice_call ? 0 : mask); + if (err) + return err; + + /* Set MIC2_MUX for call */ + mask = BIT(CPCAP_BIT_MB_ON1L) | BIT(CPCAP_BIT_MB_ON1R) | + BIT(CPCAP_BIT_MIC2_MUX) | BIT(CPCAP_BIT_MIC2_PGA_EN); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_TXI, + mask, voice_call ? mask : 0); + if (err) + return err; + + /* Enable LDSP for call */ + mask = BIT(CPCAP_BIT_A2_LDSP_L_EN) | BIT(CPCAP_BIT_A2_LDSP_R_EN); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_RXOA, + mask, voice_call ? mask : 0); + if (err) + return err; + + /* Enable CPCAP_BIT_PGA_CDC_EN for call */ + mask = BIT(CPCAP_BIT_PGA_CDC_EN); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_RXCOA, + mask, voice_call ? mask : 0); + if (err) + return err; + + /* Unmute voice for call */ + if (dai) { + err = snd_soc_dai_digital_mute(dai, !voice_call, + SNDRV_PCM_STREAM_PLAYBACK); + if (err) + return err; + } + + /* Set modem to codec mic CDC and HPF for call */ + mask = BIT(CPCAP_BIT_MIC2_CDC_EN) | BIT(CPCAP_BIT_CDC_EN_RX) | + BIT(CPCAP_BIT_AUDOHPF_1) | BIT(CPCAP_BIT_AUDOHPF_0) | + BIT(CPCAP_BIT_AUDIHPF_1) | BIT(CPCAP_BIT_AUDIHPF_0); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CC, + mask, voice_call ? mask : 0); + if (err) + return err; + + /* Enable modem to codec CDC for call*/ + mask = BIT(CPCAP_BIT_CDC_CLK_EN); + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI, + mask, voice_call ? mask : 0); + + return err; +} + +static int cpcap_voice_set_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); + int err, ts_mask, mask; + bool voice_call; + + /* + * Primitive test for voice call, probably needs more checks + * later on for 16-bit calls detected, Bluetooth headset etc. + */ + if (tx_mask == 0 && rx_mask == 1 && slot_width == 8) + voice_call = true; + else + voice_call = false; + + ts_mask = 0x7 << CPCAP_BIT_MIC2_TIMESLOT0; + ts_mask |= 0x7 << CPCAP_BIT_MIC1_RX_TIMESLOT0; + + mask = (tx_mask & 0x7) << CPCAP_BIT_MIC2_TIMESLOT0; + mask |= (rx_mask & 0x7) << CPCAP_BIT_MIC1_RX_TIMESLOT0; + + err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI, + ts_mask, mask); + if (err) + return err; + + err = cpcap_set_samprate(cpcap, CPCAP_DAI_VOICE, slot_width * 1000); + if (err) + return err; + + err = cpcap_voice_call(cpcap, dai, voice_call); + if (err) + return err; + + return 0; +} + static int cpcap_voice_set_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_component *component = dai->component; @@ -1391,6 +1513,7 @@ static const struct snd_soc_dai_ops cpcap_dai_voice_ops = { .hw_params = cpcap_voice_hw_params, .set_sysclk = cpcap_voice_set_dai_sysclk, .set_fmt = cpcap_voice_set_dai_fmt, + .set_tdm_slot = cpcap_voice_set_tdm_slot, .digital_mute = cpcap_voice_set_mute, }; -- 2.25.0

4 13

[PATCH v2 0/2] ASoC: meson: add internal DAC support
by Jerome Brunet 19 Feb '20

19 Feb '20

This patchset adds support for the internal audio DAC provided on the Amlogic gxl, g12a and sm1 SoC families. On each of these SoC families, there is glue between this codec on the audio provider. The architecture is similar to the one used for the synopsys hdmi codec on these SoCs Changes since v1 [0]: * Change some kcontrol names * Move DAC sources to DAPM [0]: https://lore.kernel.org/r/20200219133646.1035506-1-jbrunet@baylibre.com/ Jerome Brunet (2): ASoC: meson: add t9015 internal codec binding documentation ASoC: meson: add t9015 internal DAC driver .../bindings/sound/amlogic,t9015.yaml | 58 +++ sound/soc/meson/Kconfig | 8 + sound/soc/meson/Makefile | 2 + sound/soc/meson/t9015.c | 333 ++++++++++++++++++ 4 files changed, 401 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,t9015.yaml create mode 100644 sound/soc/meson/t9015.c -- 2.24.1

2 4

[alsa-devel] ALSA 1.0.29 release
by Jaroslav Kysela 19 Feb '20

19 Feb '20

Hello all, all userspace packages (except alsa-oss) were released today. You may download them from the ALSA website http://www.alsa-project.org or directly from the FTP server ftp://ftp.alsa-project.org/pub . Full list of changes: http://www.alsa-project.org/main/index.php/Changes_v1.0.28_v1.0.29 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

2 16

[alsa-devel] [PATCH 0/9] add ASoC components for AHUB
by Sameer Pujar 19 Feb '20

19 Feb '20

Overview ======== The Audio Hub (AHUB) is part of the Audio Processing Engine (APE) which comprises a collection of hardware accelerators for audio pre-processing and post-processing. It also includes a programmable full crossbar for routing audio data across these accelerators. This series exposes some of these below mentioned HW devices as ASoC components for Tegra platforms from Tegra210 onwards. * ADMAIF : The interface between ADMA and AHUB * XBAR : Crossbar for routing audio samples across various modules * I2S : Inter-IC Sound Controller * DMIC : Digital Microphone * DSPK : Digital Speaker Following is the summary of current series. 1. Add YAML DT binding documentation for above mentioned modules. 2. ACIF programming is same for Tegra generations and hence it is moved to a common file. 3. Add ASoC driver components for each of the above modules. 4. Add DT entries for above components for Tegra210, Tegra186 and Tegra194. 5. Enable these components for Jetson-Tx1, Jetson-Tx2 and Jetson-Xavier. Machine driver series will be sent separately. Sameer Pujar (9): dt-bindings: sound: tegra: add DT binding for AHUB ASoC: tegra: add support for CIF programming ASoC: tegra: add Tegra210 based DMIC driver ASoC: tegra: add Tegra210 based I2S driver ASoC: tegra: add Tegra210 based AHUB driver ASoC: tegra: add Tegra186 based DSPK driver ASoC: tegra: add Tegra210 based ADMAIF driver arm64: tegra: add AHUB components for few Tegra chips arm64: tegra: enable AHUB modules for few Tegra chips .../bindings/sound/nvidia,tegra186-dspk.yaml | 105 +++ .../bindings/sound/nvidia,tegra210-admaif.yaml | 165 ++++ .../bindings/sound/nvidia,tegra210-ahub.yaml | 130 +++ .../bindings/sound/nvidia,tegra210-dmic.yaml | 105 +++ .../bindings/sound/nvidia,tegra210-i2s.yaml | 112 +++ arch/arm64/boot/dts/nvidia/tegra186-p2771-0000.dts | 48 ++ arch/arm64/boot/dts/nvidia/tegra186.dtsi | 231 ++++- arch/arm64/boot/dts/nvidia/tegra194-p2972-0000.dts | 36 + arch/arm64/boot/dts/nvidia/tegra194.dtsi | 239 +++++- arch/arm64/boot/dts/nvidia/tegra210-p2371-2180.dts | 40 + arch/arm64/boot/dts/nvidia/tegra210.dtsi | 145 ++++ sound/soc/tegra/Kconfig | 56 ++ sound/soc/tegra/Makefile | 12 + sound/soc/tegra/tegra186_dspk.c | 516 +++++++++++ sound/soc/tegra/tegra186_dspk.h | 73 ++ sound/soc/tegra/tegra210_admaif.c | 896 ++++++++++++++++++++ sound/soc/tegra/tegra210_admaif.h | 164 ++++ sound/soc/tegra/tegra210_ahub.c | 667 +++++++++++++++ sound/soc/tegra/tegra210_ahub.h | 125 +++ sound/soc/tegra/tegra210_dmic.c | 522 ++++++++++++ sound/soc/tegra/tegra210_dmic.h | 85 ++ sound/soc/tegra/tegra210_i2s.c | 941 +++++++++++++++++++++ sound/soc/tegra/tegra210_i2s.h | 132 +++ sound/soc/tegra/tegra30_ahub.c | 94 +- sound/soc/tegra/tegra30_ahub.h | 129 --- sound/soc/tegra/tegra30_i2s.c | 35 +- sound/soc/tegra/tegra30_i2s.h | 7 - sound/soc/tegra/tegra_cif.c | 34 + sound/soc/tegra/tegra_cif.h | 50 ++ sound/soc/tegra/tegra_pcm.c | 224 ++++- sound/soc/tegra/tegra_pcm.h | 23 +- 31 files changed, 5897 insertions(+), 244 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra186-dspk.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-admaif.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-ahub.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-dmic.yaml create mode 100644 Documentation/devicetree/bindings/sound/nvidia,tegra210-i2s.yaml create mode 100644 sound/soc/tegra/tegra186_dspk.c create mode 100644 sound/soc/tegra/tegra186_dspk.h create mode 100644 sound/soc/tegra/tegra210_admaif.c create mode 100644 sound/soc/tegra/tegra210_admaif.h create mode 100644 sound/soc/tegra/tegra210_ahub.c create mode 100644 sound/soc/tegra/tegra210_ahub.h create mode 100644 sound/soc/tegra/tegra210_dmic.c create mode 100644 sound/soc/tegra/tegra210_dmic.h create mode 100644 sound/soc/tegra/tegra210_i2s.c create mode 100644 sound/soc/tegra/tegra210_i2s.h create mode 100644 sound/soc/tegra/tegra_cif.c create mode 100644 sound/soc/tegra/tegra_cif.h -- 2.7.4

3 42

[PATCH 0/2] ASoC: meson: add internal DAC support
by Jerome Brunet 19 Feb '20

19 Feb '20

This patchset adds support for the internal audio DAC provided on the Amlogic gxl, g12a and sm1 SoC families. On each of these SoC families, there is glue between this codec on the audio provider. The architecture is similar to the one used for the synopsys hdmi codec on these SoCs Jerome Brunet (2): ASoC: meson: add t9015 internal codec binding documentation ASoC: meson: add t9015 internal DAC driver .../bindings/sound/amlogic,t9015.yaml | 58 ++++ sound/soc/meson/Kconfig | 8 + sound/soc/meson/Makefile | 2 + sound/soc/meson/t9015.c | 320 ++++++++++++++++++ 4 files changed, 388 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,t9015.yaml create mode 100644 sound/soc/meson/t9015.c -- 2.24.1

2 5

[alsa-devel] [PATCH v2] ASoC: dt-bindings: stm32: convert sai to json-schema
by Olivier Moysan 19 Feb '20

19 Feb '20

Convert the STM32 SAI bindings to DT schema format using json-schema. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- Changes in v2: - use pattern for compatible of child nodes - rework dmas and clocks properties - add "additionalProperties" --- .../bindings/sound/st,stm32-sai.txt | 107 ---------- .../bindings/sound/st,stm32-sai.yaml | 191 ++++++++++++++++++ 2 files changed, 191 insertions(+), 107 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.txt create mode 100644 Documentation/devicetree/bindings/sound/st,stm32-sai.yaml diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt deleted file mode 100644 index 944743dd9212..000000000000 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt +++ /dev/null @@ -1,107 +0,0 @@ -STMicroelectronics STM32 Serial Audio Interface (SAI). - -The SAI interface (Serial Audio Interface) offers a wide set of audio protocols -as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. -The SAI contains two independent audio sub-blocks. Each sub-block has -its own clock generator and I/O lines controller. - -Required properties: - - compatible: Should be "st,stm32f4-sai" or "st,stm32h7-sai" - - reg: Base address and size of SAI common register set. - - clocks: Must contain phandle and clock specifier pairs for each entry - in clock-names. - - clock-names: Must contain "pclk" "x8k" and "x11k" - "pclk": Clock which feeds the peripheral bus interface. - Mandatory for "st,stm32h7-sai" compatible. - Not used for "st,stm32f4-sai" compatible. - "x8k": SAI parent clock for sampling rates multiple of 8kHz. - "x11k": SAI parent clock for sampling rates multiple of 11.025kHz. - - interrupts: cpu DAI interrupt line shared by SAI sub-blocks - -Optional properties: - - resets: Reference to a reset controller asserting the SAI - -SAI subnodes: -Two subnodes corresponding to SAI sub-block instances A et B can be defined. -Subnode can be omitted for unsused sub-block. - -SAI subnodes required properties: - - compatible: Should be "st,stm32-sai-sub-a" or "st,stm32-sai-sub-b" - for SAI sub-block A or B respectively. - - reg: Base address and size of SAI sub-block register set. - - clocks: Must contain one phandle and clock specifier pair - for sai_ck which feeds the internal clock generator. - If the SAI shares a master clock, with another SAI set as MCLK - clock provider, SAI provider phandle must be specified here. - - clock-names: Must contain "sai_ck". - Must also contain "MCLK", if SAI shares a master clock, - with a SAI set as MCLK clock provider. - - dmas: see Documentation/devicetree/bindings/dma/stm32-dma.txt - - dma-names: identifier string for each DMA request line - "tx": if sai sub-block is configured as playback DAI - "rx": if sai sub-block is configured as capture DAI - - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml - -SAI subnodes Optional properties: - - st,sync: specify synchronization mode. - By default SAI sub-block is in asynchronous mode. - This property sets SAI sub-block as slave of another SAI sub-block. - Must contain the phandle and index of the sai sub-block providing - the synchronization. - - st,iec60958: support S/PDIF IEC6958 protocol for playback - IEC60958 protocol is not available for capture. - By default, custom protocol is assumed, meaning that protocol is - configured according to protocol defined in related DAI link node, - such as i2s, left justified, right justified, dsp and pdm protocols. - Note: ac97 protocol is not supported by SAI driver - - #clock-cells: should be 0. This property must be present if the SAI device - is a master clock provider, according to clocks bindings, described in - Documentation/devicetree/bindings/clock/clock-bindings.txt. - -The device node should contain one 'port' child node with one child 'endpoint' -node, according to the bindings defined in Documentation/devicetree/bindings/ -graph.txt. - -Example: -sound_card { - compatible = "audio-graph-card"; - dais = <&sai1b_port>; -}; - -sai1: sai1@40015800 { - compatible = "st,stm32h7-sai"; - #address-cells = <1>; - #size-cells = <1>; - ranges = <0 0x40015800 0x400>; - reg = <0x40015800 0x4>; - clocks = <&rcc SAI1_CK>, <&rcc PLL1_Q>, <&rcc PLL2_P>; - clock-names = "pclk", "x8k", "x11k"; - interrupts = <87>; - - sai1a: audio-controller@40015804 { - compatible = "st,stm32-sai-sub-a"; - reg = <0x4 0x1C>; - clocks = <&rcc SAI1_CK>; - clock-names = "sai_ck"; - dmas = <&dmamux1 1 87 0x400 0x0>; - dma-names = "tx"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_sai1a>; - - sai1b_port: port { - cpu_endpoint: endpoint { - remote-endpoint = <&codec_endpoint>; - format = "i2s"; - }; - }; - }; -}; - -audio-codec { - codec_port: port { - codec_endpoint: endpoint { - remote-endpoint = <&cpu_endpoint>; - }; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml new file mode 100644 index 000000000000..51dd2b3bdeb1 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml @@ -0,0 +1,191 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/st,stm32-sai.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: STMicroelectronics STM32 Serial Audio Interface (SAI) + +maintainers: + - Olivier Moysan <olivier.moysan(a)st.com> + +description: + The SAI interface (Serial Audio Interface) offers a wide set of audio + protocols as I2S standards, LSB or MSB-justified, PCM/DSP, TDM, and AC'97. + The SAI contains two independent audio sub-blocks. Each sub-block has + its own clock generator and I/O lines controller. + +properties: + compatible: + enum: + - st,stm32f4-sai + - st,stm32h7-sai + + reg: + items: + - description: Base address and size of SAI common register set. + - description: Base address and size of SAI identification register set. + minItems: 1 + maxItems: 2 + + ranges: + maxItems: 1 + + interrupts: + maxItems: 1 + + resets: + maxItems: 1 + + "#address-cells": + const: 1 + + "#size-cells": + const: 1 + + clocks: + items: + - description: pclk feeds the peripheral bus interface. + - description: x8k, SAI parent clock for sampling rates multiple of 8kHz. + - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - const: pclk + - const: x8k + - const: x11k + +required: + - compatible + - reg + - ranges + - "#address-cells" + - "#size-cells" + - clocks + - clock-names + +patternProperties: + "^audio-controller@[0-9a-f]+$": + type: object + description: + Two subnodes corresponding to SAI sub-block instances A et B + can be defined. Subnode can be omitted for unsused sub-block. + + properties: + compatible: + description: Compatible for SAI sub-block A or B. + pattern: "st,stm32-sai-sub-[ab]" + + "#sound-dai-cells": + const: 0 + + reg: + maxItems: 1 + + clocks: + items: + - description: sai_ck clock feeding the internal clock generator. + - description: MCLK clock from a SAI set as master clock provider. + minItems: 1 + maxItems: 2 + + clock-names: + items: + - const: sai_ck + - const: MCLK + minItems: 1 + maxItems: 2 + + dmas: + maxItems: 1 + + dma-names: + description: | + rx: SAI sub-block is configured as a capture DAI. + tx: SAI sub-block is configured as a playback DAI. + enum: [ rx, tx ] + + st,sync: + description: + Configure the SAI sub-block as slave of another SAI sub-block. + By default SAI sub-block is in asynchronous mode. + Must contain the phandle and index of the SAI sub-block providing + the synchronization. + allOf: + - $ref: /schemas/types.yaml#definitions/phandle-array + - maxItems: 1 + + st,iec60958: + description: + If set, support S/PDIF IEC6958 protocol for playback. + IEC60958 protocol is not available for capture. + By default, custom protocol is assumed, meaning that protocol is + configured according to protocol defined in related DAI link node, + such as i2s, left justified, right justified, dsp and pdm protocols. + allOf: + - $ref: /schemas/types.yaml#definitions/flag + + "#clock-cells": + description: Configure the SAI device as master clock provider. + const: 0 + + required: + - compatible + - "#sound-dai-cells" + - reg + - clocks + - clock-names + - dmas + - dma-names + + additionalProperties: false + +allOf: + - if: + properties: + compatible: + contains: + const: st,stm32f4-sai + + - then: + properties: + clocks: + items: + - description: x8k, SAI parent clock for sampling rates multiple of 8kHz. + - description: x11k, SAI parent clock for sampling rates multiple of 11.025kHz. + + clock-names: + items: + - const: x8k + - const: x11k + +additionalProperties: false + +examples: + - | + #include <dt-bindings/interrupt-controller/arm-gic.h> + #include <dt-bindings/clock/stm32mp1-clks.h> + #include <dt-bindings/reset/stm32mp1-resets.h> + sai1: sai@4400a000 { + compatible = "st,stm32h7-sai"; + #address-cells = <1>; + #size-cells = <1>; + ranges = <0 0x4400a000 0x400>; + reg = <0x4400a000 0x4>, <0x4400a3f0 0x10>; + interrupts = <GIC_SPI 87 IRQ_TYPE_LEVEL_HIGH>; + clocks = <&rcc SAI1>, <&rcc PLL1_Q>, <&rcc PLL2_P>; + clock-names = "pclk", "x8k", "x11k"; + resets = <&rcc SAI1_R>; + + sai1a: audio-controller@4400a004 { + compatible = "st,stm32-sai-sub-a"; + #sound-dai-cells = <0>; + reg = <0x4 0x1c>; + clocks = <&rcc SAI1_K>; + clock-names = "sai_ck"; + dmas = <&dmamux1 87 0x400 0x01>; + dma-names = "tx"; + }; + }; + +... -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: da7219: check SRM lock in trigger callback
by Brent Lu 19 Feb '20

19 Feb '20

Intel sst firmware turns on BCLK/WCLK in START Ioctl call which timing is later than the DAPM SUPPLY event handler da7219_dai_event is called (in PREPARED state). Therefore, the SRM lock check always fail. Moving the check to trigger callback could ensure the SRM is locked before DSP starts to process data and avoid possisble noise. Signed-off-by: Brent Lu <brent.lu(a)intel.com> --- sound/soc/codecs/da7219.c | 68 +++++++++++++++++++++++++++++++---------------- 1 file changed, 45 insertions(+), 23 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index f83a6ea..0fb5ea5 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -794,9 +794,7 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); struct clk *bclk = da7219->dai_clks[DA7219_DAI_BCLK_IDX]; - u8 pll_ctrl, pll_status; - int i = 0, ret; - bool srm_lock = false; + int ret; switch (event) { case SND_SOC_DAPM_PRE_PMU: @@ -820,26 +818,6 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w, /* PC synchronised to DAI */ snd_soc_component_update_bits(component, DA7219_PC_COUNT, DA7219_PC_FREERUN_MASK, 0); - - /* Slave mode, if SRM not enabled no need for status checks */ - pll_ctrl = snd_soc_component_read32(component, DA7219_PLL_CTRL); - if ((pll_ctrl & DA7219_PLL_MODE_MASK) != DA7219_PLL_MODE_SRM) - return 0; - - /* Check SRM has locked */ - do { - pll_status = snd_soc_component_read32(component, DA7219_PLL_SRM_STS); - if (pll_status & DA7219_PLL_SRM_STS_SRM_LOCK) { - srm_lock = true; - } else { - ++i; - msleep(50); - } - } while ((i < DA7219_SRM_CHECK_RETRIES) && (!srm_lock)); - - if (!srm_lock) - dev_warn(component->dev, "SRM failed to lock\n"); - return 0; case SND_SOC_DAPM_POST_PMD: /* PC free-running */ @@ -1658,12 +1636,56 @@ static int da7219_hw_params(struct snd_pcm_substream *substream, return 0; } +static int da7219_set_dai_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + u8 pll_ctrl, pll_status; + int i = 0; + bool srm_lock = false; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* Slave mode, if SRM not enabled no need for status checks */ + pll_ctrl = snd_soc_component_read32(component, DA7219_PLL_CTRL); + if ((pll_ctrl & DA7219_PLL_MODE_MASK) != DA7219_PLL_MODE_SRM) + return 0; + + /* Check SRM has locked */ + do { + pll_status = snd_soc_component_read32(component, + DA7219_PLL_SRM_STS); + if (pll_status & DA7219_PLL_SRM_STS_SRM_LOCK) { + srm_lock = true; + } else { + ++i; + msleep(50); + } + } while ((i < DA7219_SRM_CHECK_RETRIES) && (!srm_lock)); + + if (!srm_lock) + dev_warn(component->dev, "SRM failed to lock\n"); + + break; + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + default: + break; + } + + return 0; +} + static const struct snd_soc_dai_ops da7219_dai_ops = { .hw_params = da7219_hw_params, .set_sysclk = da7219_set_dai_sysclk, .set_pll = da7219_set_dai_pll, .set_fmt = da7219_set_dai_fmt, .set_tdm_slot = da7219_set_dai_tdm_slot, + .trigger = da7219_set_dai_trigger, }; #define DA7219_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ -- 2.7.4

8 20

[PATCH] ALSA: hda/realtek - Fix a regression for mute led on Lenovo Carbon X1
by Hui Wang 19 Feb '20

19 Feb '20

Need to chain the THINKPAD_ACPI, otherwise the mute led will not work. Fixes: d2cd795c4ece ("ALSA: hda - fixup for the bass speaker on Lenovo Carbon X1 7th gen") Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 477589e7ec1d..31bee0512334 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6684,6 +6684,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC285_FIXUP_SPEAKER2_TO_DAC1] = { .type = HDA_FIXUP_FUNC, .v.func = alc285_fixup_speaker2_to_dac1, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI }, [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { .type = HDA_FIXUP_PINS, -- 2.17.1

2 1

ALSA raw MIDI API - client hangup and amidi exits midiC3D0 failed: Device or resource busy
by GitHub issues - opened 19 Feb '20

19 Feb '20

alsa-project/alsa-lib issue #25 was opened from MusicMaker: While developing a client using the raw MIDI interface, bumped into an issue that went away after rebooting Ubuntu 19.10 - 64 bit. Using a MIDISPORT 4x4 one of the 4 ports could not be opened. The client (written in C) call hang and the client thus went unresponsive. No timeout, no error, no nothing. CTRL-BREAK the client and retrying many times gave the exact same result, no other apps were using the port. The other 3 ports on the interface and any other MIDI USB device connected could be opened and used without any problem. When opening the port with amidi, an error was reported, it seems amidi did not gracefully exit as well. amidi -receive -p hw:3,0,1 ALSA lib rawmidi_hw.c:233:(snd_rawmidi_hw_open) open /dev/snd/midiC3D0 failed: Device or resource busy cannot open port "hw:3,0,1": Device or resource busy Looking at the rawmidi_hw.c code there is a GOTO and SYSERR statement. Not sure if those caused this . if (fd < 0) { snd_ctl_close(ctl); SYSERR("open %s failed", filename); return -errno; . After reboot the problem went away. The problem could have been caused while fixing some by coding issue. Maybe someone with more understanding of the code can spot an issue in the code. While many applications use the raw MIDI interface. believe Bitwig and most others use the sequencer API to open ports. Just wanted to log this just in case anyone else experiences this Issue URL : https://github.com/alsa-project/alsa-lib/issues/25 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ALSA raw MIDI API - client hangup and amidi exits midiC3D0 failed: Device or resource busy
by happy musicmaker 19 Feb '20

19 Feb '20

While developing a client using the raw MIDI interface, bumped into an issue that went away after rebooting Ubuntu 19.10 - 64 bit. Using a MIDISPORT 4x4 one of the 4 ports could not be opened. The client (written in C) call hang and the client thus went unresponsive. No timeout, no error, no nothing. CTRL-BREAK the client and retrying many times gave the exact same result, no other apps were using the port. The other 3 ports on the interface and any other MIDI USB device connected could be opened and used without any problem. When opening the port with amidi, an error was reported, it seems amidi did not gracefully exit as well. amidi -receive -p hw:3,0,1 ALSA lib rawmidi_hw.c:233:(snd_rawmidi_hw_open) open /dev/snd/midiC3D0 failed: Device or resource busy cannot open port "hw:3,0,1": Device or resource busy Looking at the rawmidi_hw.c code there is a GOTO and SYSERR statement. Not sure if those caused this . if (fd < 0) { snd_ctl_close(ctl); SYSERR("open %s failed", filename); return -errno; . After reboot the problem went away. The problem could have been caused while fixing some by coding issue. Maybe someone with more understanding of the code can spot an issue in the code. While many applications use the raw MIDI interface. believe Bitwig and most others use the sequencer API to open ports. Just wanted to log this just in case anyone else experiences this Thanks.

1 0

Applied "ASoC: tas2562: Return invalid for when bitwidth is invalid" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: tas2562: Return invalid for when bitwidth is invalid has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 310006cab991fa895cc3a42c53609778b0d50d46 Mon Sep 17 00:00:00 2001 From: Dan Murphy <dmurphy(a)ti.com> Date: Tue, 18 Feb 2020 12:52:52 -0600 Subject: [PATCH] ASoC: tas2562: Return invalid for when bitwidth is invalid If the bitwidth passed in to the set_bitwidth function is not supported then return an error. Fixes: 29b74236bd57 ("ASoC: tas2562: Introduce the TAS2562 amplifier") Signed-off-by: Dan Murphy <dmurphy(a)ti.com> Link: https://lore.kernel.org/r/20200218185252.26290-1-dmurphy@ti.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tas2562.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2562.c b/sound/soc/codecs/tas2562.c index 729acd874c48..5b4803a76719 100644 --- a/sound/soc/codecs/tas2562.c +++ b/sound/soc/codecs/tas2562.c @@ -215,7 +215,8 @@ static int tas2562_set_bitwidth(struct tas2562_data *tas2562, int bitwidth) break; default: - dev_info(tas2562->dev, "Not supported params format\n"); + dev_info(tas2562->dev, "Unsupported bitwidth format\n"); + return -EINVAL; } ret = snd_soc_component_update_bits(tas2562->component, -- 2.20.1

1 0

[PATCH] ASoC: soc-pcm: fix regression in soc_new_pcm()
by Stephan Gerhold 19 Feb '20

19 Feb '20

Commit af4bac11531f ("ASoC: soc-pcm: crash in snd_soc_dapm_new_dai") swapped the SNDRV_PCM_STREAM_* parameter in the snd_soc_dai_stream_valid(cpu_dai, ...) checks. But that works only for codec2codec links. For normal links it breaks registration of playback/capture-only PCM devices. E.g. on qcom/apq8016_sbc there is usually one playback-only and one capture-only PCM device, but they disappeared after the commit. The codec2codec case was added in commit a342031cdd08 ("ASoC: create pcm for codec2codec links as well") as an extra check (e.g. `playback = playback && cpu_playback->channels_min`). We should be able to simplify the code by checking directly for the correct stream type in the loop. This also fixes the regression because we check for PLAYBACK for both codec and cpu dai again when codec2codec is not used. Cc: Sameer Pujar <spujar(a)nvidia.com> Cc: Jerome Brunet <jbrunet(a)baylibre.com> Fixes: af4bac11531f ("ASoC: soc-pcm: crash in snd_soc_dapm_new_dai") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- Original report of the regression: https://lore.kernel.org/alsa-devel/20200217144120.GA243254@gerhold.net/ I'm still quite confused about the crash that was fixed by the commit that introduced the regression... If anything, the original code added in commit a342031cdd08 ("ASoC: create pcm for codec2codec links as well") should have been too "restrictive" since it checked for both PLAYBACK and CAPTURE for the cpu dai in the codec2codec case... Audio works fine gain on apq8016_sbc with this patch, but I don't have any device with codec2codec to test this. --- sound/soc/soc-pcm.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ff1b7c7078e5..cfa24d214a9c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2888,22 +2888,19 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture = rtd->dai_link->dpcm_capture; } else { /* Adapt stream for codec2codec links */ - struct snd_soc_pcm_stream *cpu_capture = rtd->dai_link->params ? - &cpu_dai->driver->playback : &cpu_dai->driver->capture; - struct snd_soc_pcm_stream *cpu_playback = rtd->dai_link->params ? - &cpu_dai->driver->capture : &cpu_dai->driver->playback; + int cpu_capture = rtd->dai_link->params ? + SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE; + int cpu_playback = rtd->dai_link->params ? + SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK; for_each_rtd_codec_dai(rtd, i, codec_dai) { if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_CAPTURE)) + snd_soc_dai_stream_valid(cpu_dai, cpu_playback)) playback = 1; if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_CAPTURE) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_PLAYBACK)) + snd_soc_dai_stream_valid(cpu_dai, cpu_capture)) capture = 1; } - - capture = capture && cpu_capture->channels_min; - playback = playback && cpu_playback->channels_min; } if (rtd->dai_link->playback_only) { -- 2.25.0

3 2

Applied "ALSA: core: Expand DMA buffer information" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ALSA: core: Expand DMA buffer information has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 386dd54b3a2eedb91aa6e465e7c3a57db04f3960 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski <cezary.rojewski(a)intel.com> Date: Tue, 18 Feb 2020 15:39:16 +0100 Subject: [PATCH] ALSA: core: Expand DMA buffer information Update DMA buffer definition for snd_compr_runtime so it is represented similarly as in snd_pcm_runtime. While at it, modify snd_compr_set_runtime_buffer to account for newly added members. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Acked-by: Vinod Koul <vkoul(a)kernel.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200218143924.10565-2-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/compress_driver.h | 35 ++++++++++++++++++++++++--------- 1 file changed, 26 insertions(+), 9 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index bc88d6f964da..00f633c0c3ba 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -23,7 +23,6 @@ struct snd_compr_ops; * struct snd_compr_runtime: runtime stream description * @state: stream state * @ops: pointer to DSP callbacks - * @dma_buffer_p: runtime dma buffer pointer * @buffer: pointer to kernel buffer, valid only when not in mmap mode or * DSP doesn't implement copy * @buffer_size: size of the above buffer @@ -34,11 +33,14 @@ struct snd_compr_ops; * @total_bytes_transferred: cumulative bytes transferred by offload DSP * @sleep: poll sleep * @private_data: driver private data pointer + * @dma_area: virtual buffer address + * @dma_addr: physical buffer address (not accessible from main CPU) + * @dma_bytes: size of DMA area + * @dma_buffer_p: runtime dma buffer pointer */ struct snd_compr_runtime { snd_pcm_state_t state; struct snd_compr_ops *ops; - struct snd_dma_buffer *dma_buffer_p; void *buffer; u64 buffer_size; u32 fragment_size; @@ -47,6 +49,11 @@ struct snd_compr_runtime { u64 total_bytes_transferred; wait_queue_head_t sleep; void *private_data; + + unsigned char *dma_area; + dma_addr_t dma_addr; + size_t dma_bytes; + struct snd_dma_buffer *dma_buffer_p; }; /** @@ -180,19 +187,29 @@ static inline void snd_compr_drain_notify(struct snd_compr_stream *stream) /** * snd_compr_set_runtime_buffer - Set the Compress runtime buffer - * @substream: compress substream to set + * @stream: compress stream to set * @bufp: the buffer information, NULL to clear * * Copy the buffer information to runtime buffer when @bufp is non-NULL. * Otherwise it clears the current buffer information. */ -static inline void snd_compr_set_runtime_buffer( - struct snd_compr_stream *substream, - struct snd_dma_buffer *bufp) +static inline void +snd_compr_set_runtime_buffer(struct snd_compr_stream *stream, + struct snd_dma_buffer *bufp) { - struct snd_compr_runtime *runtime = substream->runtime; - - runtime->dma_buffer_p = bufp; + struct snd_compr_runtime *runtime = stream->runtime; + + if (bufp) { + runtime->dma_buffer_p = bufp; + runtime->dma_area = bufp->area; + runtime->dma_addr = bufp->addr; + runtime->dma_bytes = bufp->bytes; + } else { + runtime->dma_buffer_p = NULL; + runtime->dma_area = NULL; + runtime->dma_addr = 0; + runtime->dma_bytes = 0; + } } int snd_compr_stop_error(struct snd_compr_stream *stream, -- 2.20.1

1 0

Applied "ALSA: core: Implement compress page allocation and free routines" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ALSA: core: Implement compress page allocation and free routines has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b9759ef2fd1acb0d3f3dce7991c44a4c5e9e68a3 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski <cezary.rojewski(a)intel.com> Date: Tue, 18 Feb 2020 15:39:17 +0100 Subject: [PATCH] ALSA: core: Implement compress page allocation and free routines Add simple malloc and free methods for memory management for compress streams. Based on snd_pcm_lib_malloc_pages and snd_pcm_lib_free_pages implementation. Signed-off-by: Divya Prakash <divya1.prakash(a)intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Reviewed-by: Takashi Iwai <tiwai(a)suse.de> Acked-by: Vinod Koul <vkoul(a)kernel.org> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200218143924.10565-3-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/compress_driver.h | 5 ++++ sound/core/compress_offload.c | 42 +++++++++++++++++++++++++++++++++ 2 files changed, 47 insertions(+) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 00f633c0c3ba..6ce8effa0b12 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -67,6 +67,7 @@ struct snd_compr_runtime { * @metadata_set: metadata set flag, true when set * @next_track: has userspace signal next track transition, true when set * @private_data: pointer to DSP private data + * @dma_buffer: allocated buffer if any */ struct snd_compr_stream { const char *name; @@ -78,6 +79,7 @@ struct snd_compr_stream { bool metadata_set; bool next_track; void *private_data; + struct snd_dma_buffer dma_buffer; }; /** @@ -212,6 +214,9 @@ snd_compr_set_runtime_buffer(struct snd_compr_stream *stream, } } +int snd_compr_malloc_pages(struct snd_compr_stream *stream, size_t size); +int snd_compr_free_pages(struct snd_compr_stream *stream); + int snd_compr_stop_error(struct snd_compr_stream *stream, snd_pcm_state_t state); diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 9de1c9a0173e..509290f2efa8 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -488,6 +488,48 @@ snd_compr_get_codec_caps(struct snd_compr_stream *stream, unsigned long arg) } #endif /* !COMPR_CODEC_CAPS_OVERFLOW */ +int snd_compr_malloc_pages(struct snd_compr_stream *stream, size_t size) +{ + struct snd_dma_buffer *dmab; + int ret; + + if (snd_BUG_ON(!(stream) || !(stream)->runtime)) + return -EINVAL; + dmab = kzalloc(sizeof(*dmab), GFP_KERNEL); + if (!dmab) + return -ENOMEM; + dmab->dev = stream->dma_buffer.dev; + ret = snd_dma_alloc_pages(dmab->dev.type, dmab->dev.dev, size, dmab); + if (ret < 0) { + kfree(dmab); + return ret; + } + + snd_compr_set_runtime_buffer(stream, dmab); + stream->runtime->dma_bytes = size; + return 1; +} +EXPORT_SYMBOL(snd_compr_malloc_pages); + +int snd_compr_free_pages(struct snd_compr_stream *stream) +{ + struct snd_compr_runtime *runtime = stream->runtime; + + if (snd_BUG_ON(!(stream) || !(stream)->runtime)) + return -EINVAL; + if (runtime->dma_area == NULL) + return 0; + if (runtime->dma_buffer_p != &stream->dma_buffer) { + /* It's a newly allocated buffer. Release it now. */ + snd_dma_free_pages(runtime->dma_buffer_p); + kfree(runtime->dma_buffer_p); + } + + snd_compr_set_runtime_buffer(stream, NULL); + return 0; +} +EXPORT_SYMBOL(snd_compr_free_pages); + /* revisit this with snd_pcm_preallocate_xxx */ static int snd_compr_allocate_buffer(struct snd_compr_stream *stream, struct snd_compr_params *params) -- 2.20.1

1 0

[PATCH v5 0/9] ASoC: SOF: Data probing
by Cezary Rojewski 19 Feb '20

19 Feb '20

This set of patches achieves few goals in order to enable data probing feature for audio DSP. First, provide new and alter existing interfaces (page allocation, runtime flow adjustments) to make them compress friendly. Finally, the end goal which are the probe APIs and usage itself. Probes can be treated as endpoints which allow for data extraction from or injection to target module - a great ally when debugging problematic audio issues such as distortions, glitches or gaps. Compress streams are a weapon of choice here to provide a lightweight implementation. While all available IPCs have been defined, current implementation covers extraction only, with injection scheduled for a later date. Initial review and development of probes can be found under: https://github.com/thesofproject/linux/pull/1276 with the hda-compress-enable set of patches being separated and reviewed on: https://github.com/thesofproject/linux/pull/1571 Tested on CML-U with rt5682 i2s board. Cahnges in v5: - Removed SND_SOC_SOF_HDA_LINK requirement for hda probes. Instead of sound/hda/ext, sof hda operations are used - In consequence of above, patches: ALSA: hda: Allow for compress stream to hdac_ext_stream assignment ALSA: hda: Prepare for compress stream support ALSA: hda: Interrupt servicing and BDL setup for compress streams have been dropped. - In order to reuse pcm utility functions for hda compress, new patch has been added: ASoC: SOF: Intel: Expose SDnFMT helpers - Addressed u64/ u32 concerns from Pierre when calculating bytes transferred during runtime Changes in v4: - No changes to ALSA core and hda patches - 'probe' source added in patch "ASoC: SOF: Implement Probe IPC API" now only compiled if SND_SOC_SOF_DEBUG_PROBES is enabled. Said kconfig initial declaration has been moved to that patch too - 'extractor_stream_tag' field now decorated with _DEBUG_PROBES to account for the change above - Debugfs probe handlers now check if extractor stream is running before continuing. To achieve this, 'extractor_stream_tag' is both initialized and set to SOF_PROBE_INVALID_NODE_ID on sof_probe_compr_free(). Suggestion provided by Pierre - Reverted SND_SOC_SOF_HDA_LINK requirement-removal change for HDA probes. Code requires SND_HDA_EXT_CORE in order to function on Intel platforms - Relocated probe-compress related code from /sof/pcm.c directly to /sof/compress.c as requested by Daniel Changes in v3: - No changes to ALSA core and hda patches - Addressed sparse and all doc related warnings as reported by Pierre - Moved _DEBUG_PROBES config outside of _DEVELOPER_SUPPORT block - Renamed 'extractor' field to 'extractor_stream_tag' as requested - Relocated 'extractor_stream_tag' declaration from patch 07 to 08 Changes in v2: - No changes to ALSA core and hda patches - Removed "ASoC: Intel: sof_rt5682: Add compress probe DAI links" from the patchset list as requested by Pierre - updated copyright header for newly added files (dates) - probes over HDA no longer require SND_SOC_SOF_HDA_LINK enabled - renamed debugfs probe functions as requested by Pierre - probe IPC API has been updated to align with newest SOF FW & probe debug app (struct_size macro has been enlisted to make the size calculations transparent). This targets only "ASoC: SOF: Implement Probe IPC API" patch Cezary Rojewski (9): ALSA: core: Expand DMA buffer information ALSA: core: Implement compress page allocation and free routines ASoC: SOF: Intel: Account for compress streams when servicing IRQs ASoC: SOF: Implement Probe IPC API ASoC: SOF: Generic probe compress operations ASoC: SOF: Intel: Expose SDnFMT helpers ASoC: SOF: Intel: Probe compress operations ASoC: SOF: Provide probe debugfs support ASoC: SOF: Intel: Add Probe compress CPU DAIs include/sound/compress_driver.h | 40 +++- include/sound/hdaudio.h | 2 + include/sound/sof/header.h | 11 ++ sound/core/compress_offload.c | 42 +++++ sound/soc/sof/Kconfig | 9 + sound/soc/sof/Makefile | 1 + sound/soc/sof/compress.c | 146 +++++++++++++++ sound/soc/sof/compress.h | 31 ++++ sound/soc/sof/core.c | 6 + sound/soc/sof/debug.c | 226 +++++++++++++++++++++++ sound/soc/sof/intel/Kconfig | 9 + sound/soc/sof/intel/Makefile | 1 + sound/soc/sof/intel/apl.c | 9 + sound/soc/sof/intel/cnl.c | 9 + sound/soc/sof/intel/hda-compress.c | 114 ++++++++++++ sound/soc/sof/intel/hda-dai.c | 28 +++ sound/soc/sof/intel/hda-ipc.c | 4 +- sound/soc/sof/intel/hda-pcm.c | 8 +- sound/soc/sof/intel/hda-stream.c | 25 ++- sound/soc/sof/intel/hda.h | 32 ++++ sound/soc/sof/ops.h | 43 +++++ sound/soc/sof/pcm.c | 7 + sound/soc/sof/probe.c | 286 +++++++++++++++++++++++++++++ sound/soc/sof/probe.h | 85 +++++++++ sound/soc/sof/sof-priv.h | 25 +++ 25 files changed, 1183 insertions(+), 16 deletions(-) create mode 100644 sound/soc/sof/compress.c create mode 100644 sound/soc/sof/compress.h create mode 100644 sound/soc/sof/intel/hda-compress.c create mode 100644 sound/soc/sof/probe.c create mode 100644 sound/soc/sof/probe.h -- 2.17.1

3 13

Applied "ASoC: SOF: Implement Probe IPC API" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: SOF: Implement Probe IPC API has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f3b433e4699fa358ce5b7bd7688bebe36068c199 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski <cezary.rojewski(a)intel.com> Date: Tue, 18 Feb 2020 15:39:19 +0100 Subject: [PATCH] ASoC: SOF: Implement Probe IPC API Add all required types and methods to support each and every request that driver could sent to firmware. Probe is one of SOF firmware features which allows for data extraction and injection directly from or to DMA stream. Exposes eight IPCs: - addition and removal of injection DMAs - addition and removal of probe points - info retrieval of injection DMAs and probe points - probe initialization and cleanup Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200218143924.10565-5-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/sof/header.h | 11 ++ sound/soc/sof/Kconfig | 8 + sound/soc/sof/Makefile | 1 + sound/soc/sof/intel/hda-ipc.c | 4 +- sound/soc/sof/probe.c | 286 ++++++++++++++++++++++++++++++++++ sound/soc/sof/probe.h | 85 ++++++++++ 6 files changed, 394 insertions(+), 1 deletion(-) create mode 100644 sound/soc/sof/probe.c create mode 100644 sound/soc/sof/probe.h diff --git a/include/sound/sof/header.h b/include/sound/sof/header.h index bf3edd9c08b4..b79479575cc8 100644 --- a/include/sound/sof/header.h +++ b/include/sound/sof/header.h @@ -51,6 +51,7 @@ #define SOF_IPC_GLB_TRACE_MSG SOF_GLB_TYPE(0x9U) #define SOF_IPC_GLB_GDB_DEBUG SOF_GLB_TYPE(0xAU) #define SOF_IPC_GLB_TEST_MSG SOF_GLB_TYPE(0xBU) +#define SOF_IPC_GLB_PROBE SOF_GLB_TYPE(0xCU) /* * DSP Command Message Types @@ -102,6 +103,16 @@ #define SOF_IPC_STREAM_VORBIS_PARAMS SOF_CMD_TYPE(0x010) #define SOF_IPC_STREAM_VORBIS_FREE SOF_CMD_TYPE(0x011) +/* probe */ +#define SOF_IPC_PROBE_INIT SOF_CMD_TYPE(0x001) +#define SOF_IPC_PROBE_DEINIT SOF_CMD_TYPE(0x002) +#define SOF_IPC_PROBE_DMA_ADD SOF_CMD_TYPE(0x003) +#define SOF_IPC_PROBE_DMA_INFO SOF_CMD_TYPE(0x004) +#define SOF_IPC_PROBE_DMA_REMOVE SOF_CMD_TYPE(0x005) +#define SOF_IPC_PROBE_POINT_ADD SOF_CMD_TYPE(0x006) +#define SOF_IPC_PROBE_POINT_INFO SOF_CMD_TYPE(0x007) +#define SOF_IPC_PROBE_POINT_REMOVE SOF_CMD_TYPE(0x008) + /* trace */ #define SOF_IPC_TRACE_DMA_PARAMS SOF_CMD_TYPE(0x001) #define SOF_IPC_TRACE_DMA_POSITION SOF_CMD_TYPE(0x002) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 827b0ec92522..65c3cfbcb812 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -41,6 +41,14 @@ config SND_SOC_SOF_OF required to enable i.MX8 devices. Say Y if you need this option. If unsure select "N". +config SND_SOC_SOF_DEBUG_PROBES + bool "SOF enable data probing" + help + This option enables the data probing feature that can be used to + gather data directly from specific points of the audio pipeline. + Say Y if you want to enable probes. + If unsure, select "N". + config SND_SOC_SOF_DEVELOPER_SUPPORT bool "SOF developer options support" depends on EXPERT diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile index 0a8bc72c28a5..18d7cab9046e 100644 --- a/sound/soc/sof/Makefile +++ b/sound/soc/sof/Makefile @@ -2,6 +2,7 @@ snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\ control.o trace.o utils.o sof-audio.o +snd-sof-$(CONFIG_SND_SOC_SOF_DEBUG_PROBES) += probe.o snd-sof-pci-objs := sof-pci-dev.o snd-sof-acpi-objs := sof-acpi-dev.o diff --git a/sound/soc/sof/intel/hda-ipc.c b/sound/soc/sof/intel/hda-ipc.c index 1837f66e361f..922052883b0a 100644 --- a/sound/soc/sof/intel/hda-ipc.c +++ b/sound/soc/sof/intel/hda-ipc.c @@ -106,7 +106,9 @@ void hda_dsp_ipc_get_reply(struct snd_sof_dev *sdev) ret = reply.error; } else { /* reply correct size ? */ - if (reply.hdr.size != msg->reply_size) { + if (reply.hdr.size != msg->reply_size && + /* getter payload is never known upfront */ + !(reply.hdr.cmd & SOF_IPC_GLB_PROBE)) { dev_err(sdev->dev, "error: reply expected %zu got %u bytes\n", msg->reply_size, reply.hdr.size); ret = -EINVAL; diff --git a/sound/soc/sof/probe.c b/sound/soc/sof/probe.c new file mode 100644 index 000000000000..2b2f3dcfc7e9 --- /dev/null +++ b/sound/soc/sof/probe.c @@ -0,0 +1,286 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2019-2020 Intel Corporation. All rights reserved. +// +// Author: Cezary Rojewski <cezary.rojewski(a)intel.com> +// + +#include "sof-priv.h" +#include "probe.h" + +/** + * sof_ipc_probe_init - initialize data probing + * @sdev: SOF sound device + * @stream_tag: Extractor stream tag + * @buffer_size: DMA buffer size to set for extractor + * + * Host chooses whether extraction is supported or not by providing + * valid stream tag to DSP. Once specified, stream described by that + * tag will be tied to DSP for extraction for the entire lifetime of + * probe. + * + * Probing is initialized only once and each INIT request must be + * matched by DEINIT call. + */ +int sof_ipc_probe_init(struct snd_sof_dev *sdev, + u32 stream_tag, size_t buffer_size) +{ + struct sof_ipc_probe_dma_add_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, dma, 1); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_INIT; + msg->num_elems = 1; + msg->dma[0].stream_tag = stream_tag; + msg->dma[0].dma_buffer_size = buffer_size; + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_init); + +/** + * sof_ipc_probe_deinit - cleanup after data probing + * @sdev: SOF sound device + * + * Host sends DEINIT request to free previously initialized probe + * on DSP side once it is no longer needed. DEINIT only when there + * are no probes connected and with all injectors detached. + */ +int sof_ipc_probe_deinit(struct snd_sof_dev *sdev) +{ + struct sof_ipc_cmd_hdr msg; + struct sof_ipc_reply reply; + + msg.size = sizeof(msg); + msg.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DEINIT; + + return sof_ipc_tx_message(sdev->ipc, msg.cmd, &msg, msg.size, + &reply, sizeof(reply)); +} +EXPORT_SYMBOL(sof_ipc_probe_deinit); + +static int sof_ipc_probe_info(struct snd_sof_dev *sdev, unsigned int cmd, + void **params, size_t *num_params) +{ + struct sof_ipc_probe_info_params msg = {{{0}}}; + struct sof_ipc_probe_info_params *reply; + size_t bytes; + int ret; + + *params = NULL; + *num_params = 0; + + reply = kzalloc(SOF_IPC_MSG_MAX_SIZE, GFP_KERNEL); + if (!reply) + return -ENOMEM; + msg.rhdr.hdr.size = sizeof(msg); + msg.rhdr.hdr.cmd = SOF_IPC_GLB_PROBE | cmd; + + ret = sof_ipc_tx_message(sdev->ipc, msg.rhdr.hdr.cmd, &msg, + msg.rhdr.hdr.size, reply, SOF_IPC_MSG_MAX_SIZE); + if (ret < 0 || reply->rhdr.error < 0) + goto exit; + + if (!reply->num_elems) + goto exit; + + bytes = reply->num_elems * sizeof(reply->dma[0]); + *params = kmemdup(&reply->dma[0], bytes, GFP_KERNEL); + if (!*params) { + ret = -ENOMEM; + goto exit; + } + *num_params = msg.num_elems; + +exit: + kfree(reply); + return ret; +} + +/** + * sof_ipc_probe_dma_info - retrieve list of active injection dmas + * @sdev: SOF sound device + * @dma: Returned list of active dmas + * @num_dma: Returned count of active dmas + * + * Host sends DMA_INFO request to obtain list of injection dmas it + * can use to transfer data over with. + * + * Note that list contains only injection dmas as there is only one + * extractor (dma) and it is always assigned on probing init. + * DSP knows exactly where data from extraction probes is going to, + * which is not the case for injection where multiple streams + * could be engaged. + */ +int sof_ipc_probe_dma_info(struct snd_sof_dev *sdev, + struct sof_probe_dma **dma, size_t *num_dma) +{ + return sof_ipc_probe_info(sdev, SOF_IPC_PROBE_DMA_INFO, + (void **)dma, num_dma); +} +EXPORT_SYMBOL(sof_ipc_probe_dma_info); + +/** + * sof_ipc_probe_dma_add - attach to specified dmas + * @sdev: SOF sound device + * @dma: List of streams (dmas) to attach to + * @num_dma: Number of elements in @dma + * + * Contrary to extraction, injection streams are never assigned + * on init. Before attempting any data injection, host is responsible + * for specifying streams which will be later used to transfer data + * to connected probe points. + */ +int sof_ipc_probe_dma_add(struct snd_sof_dev *sdev, + struct sof_probe_dma *dma, size_t num_dma) +{ + struct sof_ipc_probe_dma_add_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, dma, num_dma); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->num_elems = num_dma; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DMA_ADD; + memcpy(&msg->dma[0], dma, size - sizeof(*msg)); + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_dma_add); + +/** + * sof_ipc_probe_dma_remove - detach from specified dmas + * @sdev: SOF sound device + * @stream_tag: List of stream tags to detach from + * @num_stream_tag: Number of elements in @stream_tag + * + * Host sends DMA_REMOVE request to free previously attached stream + * from being occupied for injection. Each detach operation should + * match equivalent DMA_ADD. Detach only when all probes tied to + * given stream have been disconnected. + */ +int sof_ipc_probe_dma_remove(struct snd_sof_dev *sdev, + unsigned int *stream_tag, size_t num_stream_tag) +{ + struct sof_ipc_probe_dma_remove_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, stream_tag, num_stream_tag); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->num_elems = num_stream_tag; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_DMA_REMOVE; + memcpy(&msg->stream_tag[0], stream_tag, size - sizeof(*msg)); + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_dma_remove); + +/** + * sof_ipc_probe_points_info - retrieve list of active probe points + * @sdev: SOF sound device + * @desc: Returned list of active probes + * @num_desc: Returned count of active probes + * + * Host sends PROBE_POINT_INFO request to obtain list of active probe + * points, valid for disconnection when given probe is no longer + * required. + */ +int sof_ipc_probe_points_info(struct snd_sof_dev *sdev, + struct sof_probe_point_desc **desc, size_t *num_desc) +{ + return sof_ipc_probe_info(sdev, SOF_IPC_PROBE_POINT_INFO, + (void **)desc, num_desc); +} +EXPORT_SYMBOL(sof_ipc_probe_points_info); + +/** + * sof_ipc_probe_points_add - connect specified probes + * @sdev: SOF sound device + * @desc: List of probe points to connect + * @num_desc: Number of elements in @desc + * + * Dynamically connects to provided set of endpoints. Immediately + * after connection is established, host must be prepared to + * transfer data from or to target stream given the probing purpose. + * + * Each probe point should be removed using PROBE_POINT_REMOVE + * request when no longer needed. + */ +int sof_ipc_probe_points_add(struct snd_sof_dev *sdev, + struct sof_probe_point_desc *desc, size_t num_desc) +{ + struct sof_ipc_probe_point_add_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, desc, num_desc); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->num_elems = num_desc; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_POINT_ADD; + memcpy(&msg->desc[0], desc, size - sizeof(*msg)); + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_points_add); + +/** + * sof_ipc_probe_points_remove - disconnect specified probes + * @sdev: SOF sound device + * @buffer_id: List of probe points to disconnect + * @num_buffer_id: Number of elements in @desc + * + * Removes previously connected probes from list of active probe + * points and frees all resources on DSP side. + */ +int sof_ipc_probe_points_remove(struct snd_sof_dev *sdev, + unsigned int *buffer_id, size_t num_buffer_id) +{ + struct sof_ipc_probe_point_remove_params *msg; + struct sof_ipc_reply reply; + size_t size = struct_size(msg, buffer_id, num_buffer_id); + int ret; + + msg = kmalloc(size, GFP_KERNEL); + if (!msg) + return -ENOMEM; + msg->hdr.size = size; + msg->num_elems = num_buffer_id; + msg->hdr.cmd = SOF_IPC_GLB_PROBE | SOF_IPC_PROBE_POINT_REMOVE; + memcpy(&msg->buffer_id[0], buffer_id, size - sizeof(*msg)); + + ret = sof_ipc_tx_message(sdev->ipc, msg->hdr.cmd, msg, msg->hdr.size, + &reply, sizeof(reply)); + kfree(msg); + return ret; +} +EXPORT_SYMBOL(sof_ipc_probe_points_remove); diff --git a/sound/soc/sof/probe.h b/sound/soc/sof/probe.h new file mode 100644 index 000000000000..45daa5552834 --- /dev/null +++ b/sound/soc/sof/probe.h @@ -0,0 +1,85 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2019-2020 Intel Corporation. All rights reserved. + * + * Author: Cezary Rojewski <cezary.rojewski(a)intel.com> + */ + +#ifndef __SOF_PROBE_H +#define __SOF_PROBE_H + +#include <sound/sof/header.h> + +struct snd_sof_dev; + +#define SOF_PROBE_INVALID_NODE_ID UINT_MAX + +struct sof_probe_dma { + unsigned int stream_tag; + unsigned int dma_buffer_size; +} __packed; + +enum sof_connection_purpose { + SOF_CONNECTION_PURPOSE_EXTRACT = 1, + SOF_CONNECTION_PURPOSE_INJECT, +}; + +struct sof_probe_point_desc { + unsigned int buffer_id; + unsigned int purpose; + unsigned int stream_tag; +} __packed; + +struct sof_ipc_probe_dma_add_params { + struct sof_ipc_cmd_hdr hdr; + unsigned int num_elems; + struct sof_probe_dma dma[0]; +} __packed; + +struct sof_ipc_probe_info_params { + struct sof_ipc_reply rhdr; + unsigned int num_elems; + union { + struct sof_probe_dma dma[0]; + struct sof_probe_point_desc desc[0]; + }; +} __packed; + +struct sof_ipc_probe_dma_remove_params { + struct sof_ipc_cmd_hdr hdr; + unsigned int num_elems; + unsigned int stream_tag[0]; +} __packed; + +struct sof_ipc_probe_point_add_params { + struct sof_ipc_cmd_hdr hdr; + unsigned int num_elems; + struct sof_probe_point_desc desc[0]; +} __packed; + +struct sof_ipc_probe_point_remove_params { + struct sof_ipc_cmd_hdr hdr; + unsigned int num_elems; + unsigned int buffer_id[0]; +} __packed; + +int sof_ipc_probe_init(struct snd_sof_dev *sdev, + u32 stream_tag, size_t buffer_size); +int sof_ipc_probe_deinit(struct snd_sof_dev *sdev); +int sof_ipc_probe_dma_info(struct snd_sof_dev *sdev, + struct sof_probe_dma **dma, size_t *num_dma); +int sof_ipc_probe_dma_add(struct snd_sof_dev *sdev, + struct sof_probe_dma *dma, size_t num_dma); +int sof_ipc_probe_dma_remove(struct snd_sof_dev *sdev, + unsigned int *stream_tag, size_t num_stream_tag); +int sof_ipc_probe_points_info(struct snd_sof_dev *sdev, + struct sof_probe_point_desc **desc, size_t *num_desc); +int sof_ipc_probe_points_add(struct snd_sof_dev *sdev, + struct sof_probe_point_desc *desc, size_t num_desc); +int sof_ipc_probe_points_remove(struct snd_sof_dev *sdev, + unsigned int *buffer_id, size_t num_buffer_id); + +#endif -- 2.20.1

1 0

Applied "ASoC: SOF: Generic probe compress operations" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: SOF: Generic probe compress operations has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e145e9af231adff081e0e16e1dacfb6e3c4e968f Mon Sep 17 00:00:00 2001 From: Cezary Rojewski <cezary.rojewski(a)intel.com> Date: Tue, 18 Feb 2020 15:39:20 +0100 Subject: [PATCH] ASoC: SOF: Generic probe compress operations Define system-agnostic probe compress flow which serves as a base for actual, hardware-dependent implementations. As per firmware spec, maximum of one extraction stream is allowed, while for injection, there can be plenty. Apart from probe_pointer, all probe compress operations are mandatory. Copy operation is defined as unified as its flow should be shared across all SOF systems. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200218143924.10565-6-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/Kconfig | 1 + sound/soc/sof/Makefile | 2 +- sound/soc/sof/compress.c | 141 +++++++++++++++++++++++++++++++++++++++ sound/soc/sof/compress.h | 29 ++++++++ sound/soc/sof/core.c | 6 ++ sound/soc/sof/ops.h | 43 ++++++++++++ sound/soc/sof/sof-priv.h | 25 +++++++ 7 files changed, 246 insertions(+), 1 deletion(-) create mode 100644 sound/soc/sof/compress.c create mode 100644 sound/soc/sof/compress.h diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index 65c3cfbcb812..4dda4b62509f 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -43,6 +43,7 @@ config SND_SOC_SOF_OF config SND_SOC_SOF_DEBUG_PROBES bool "SOF enable data probing" + select SND_SOC_COMPRESS help This option enables the data probing feature that can be used to gather data directly from specific points of the audio pipeline. diff --git a/sound/soc/sof/Makefile b/sound/soc/sof/Makefile index 18d7cab9046e..8eca2f85c90e 100644 --- a/sound/soc/sof/Makefile +++ b/sound/soc/sof/Makefile @@ -2,7 +2,7 @@ snd-sof-objs := core.o ops.o loader.o ipc.o pcm.o pm.o debug.o topology.o\ control.o trace.o utils.o sof-audio.o -snd-sof-$(CONFIG_SND_SOC_SOF_DEBUG_PROBES) += probe.o +snd-sof-$(CONFIG_SND_SOC_SOF_DEBUG_PROBES) += probe.o compress.o snd-sof-pci-objs := sof-pci-dev.o snd-sof-acpi-objs := sof-acpi-dev.o diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c new file mode 100644 index 000000000000..e87cc81a0599 --- /dev/null +++ b/sound/soc/sof/compress.c @@ -0,0 +1,141 @@ +// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) +// +// This file is provided under a dual BSD/GPLv2 license. When using or +// redistributing this file, you may do so under either license. +// +// Copyright(c) 2019-2020 Intel Corporation. All rights reserved. +// +// Author: Cezary Rojewski <cezary.rojewski(a)intel.com> +// + +#include <sound/soc.h> +#include "compress.h" +#include "ops.h" +#include "probe.h" + +int sof_probe_compr_open(struct snd_compr_stream *cstream, + struct snd_soc_dai *dai) +{ + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + int ret; + + ret = snd_sof_probe_compr_assign(sdev, cstream, dai); + if (ret < 0) { + dev_err(dai->dev, "Failed to assign probe stream: %d\n", ret); + return ret; + } + + sdev->extractor_stream_tag = ret; + return 0; +} +EXPORT_SYMBOL(sof_probe_compr_open); + +int sof_probe_compr_free(struct snd_compr_stream *cstream, + struct snd_soc_dai *dai) +{ + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + struct sof_probe_point_desc *desc; + size_t num_desc; + int i, ret; + + /* disconnect all probe points */ + ret = sof_ipc_probe_points_info(sdev, &desc, &num_desc); + if (ret < 0) { + dev_err(dai->dev, "Failed to get probe points: %d\n", ret); + goto exit; + } + + for (i = 0; i < num_desc; i++) + sof_ipc_probe_points_remove(sdev, &desc[i].buffer_id, 1); + kfree(desc); + +exit: + ret = sof_ipc_probe_deinit(sdev); + if (ret < 0) + dev_err(dai->dev, "Failed to deinit probe: %d\n", ret); + + sdev->extractor_stream_tag = SOF_PROBE_INVALID_NODE_ID; + snd_compr_free_pages(cstream); + + return snd_sof_probe_compr_free(sdev, cstream, dai); +} +EXPORT_SYMBOL(sof_probe_compr_free); + +int sof_probe_compr_set_params(struct snd_compr_stream *cstream, + struct snd_compr_params *params, struct snd_soc_dai *dai) +{ + struct snd_compr_runtime *rtd = cstream->runtime; + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + int ret; + + cstream->dma_buffer.dev.type = SNDRV_DMA_TYPE_DEV_SG; + cstream->dma_buffer.dev.dev = sdev->dev; + ret = snd_compr_malloc_pages(cstream, rtd->buffer_size); + if (ret < 0) + return ret; + + ret = snd_sof_probe_compr_set_params(sdev, cstream, params, dai); + if (ret < 0) + return ret; + + ret = sof_ipc_probe_init(sdev, sdev->extractor_stream_tag, + rtd->dma_bytes); + if (ret < 0) { + dev_err(dai->dev, "Failed to init probe: %d\n", ret); + return ret; + } + + return 0; +} +EXPORT_SYMBOL(sof_probe_compr_set_params); + +int sof_probe_compr_trigger(struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + + return snd_sof_probe_compr_trigger(sdev, cstream, cmd, dai); +} +EXPORT_SYMBOL(sof_probe_compr_trigger); + +int sof_probe_compr_pointer(struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai) +{ + struct snd_sof_dev *sdev = + snd_soc_component_get_drvdata(dai->component); + + return snd_sof_probe_compr_pointer(sdev, cstream, tstamp, dai); +} +EXPORT_SYMBOL(sof_probe_compr_pointer); + +int sof_probe_compr_copy(struct snd_compr_stream *cstream, + char __user *buf, size_t count) +{ + struct snd_compr_runtime *rtd = cstream->runtime; + unsigned int offset, n; + void *ptr; + int ret; + + if (count > rtd->buffer_size) + count = rtd->buffer_size; + + div_u64_rem(rtd->total_bytes_transferred, rtd->buffer_size, &offset); + ptr = rtd->dma_area + offset; + n = rtd->buffer_size - offset; + + if (count < n) { + ret = copy_to_user(buf, ptr, count); + } else { + ret = copy_to_user(buf, ptr, n); + ret += copy_to_user(buf + n, rtd->dma_area, count - n); + } + + if (ret) + return count - ret; + return count; +} +EXPORT_SYMBOL(sof_probe_compr_copy); diff --git a/sound/soc/sof/compress.h b/sound/soc/sof/compress.h new file mode 100644 index 000000000000..dccc9e008f81 --- /dev/null +++ b/sound/soc/sof/compress.h @@ -0,0 +1,29 @@ +/* SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) */ +/* + * This file is provided under a dual BSD/GPLv2 license. When using or + * redistributing this file, you may do so under either license. + * + * Copyright(c) 2019-2020 Intel Corporation. All rights reserved. + * + * Author: Cezary Rojewski <cezary.rojewski(a)intel.com> + */ + +#ifndef __SOF_COMPRESS_H +#define __SOF_COMPRESS_H + +#include <sound/compress_driver.h> + +int sof_probe_compr_open(struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); +int sof_probe_compr_free(struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); +int sof_probe_compr_set_params(struct snd_compr_stream *cstream, + struct snd_compr_params *params, struct snd_soc_dai *dai); +int sof_probe_compr_trigger(struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai); +int sof_probe_compr_pointer(struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai); +int sof_probe_compr_copy(struct snd_compr_stream *cstream, + char __user *buf, size_t count); + +#endif diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 1d07450aff77..91acfae7935c 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -14,6 +14,9 @@ #include <sound/sof.h> #include "sof-priv.h" #include "ops.h" +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) +#include "probe.h" +#endif /* see SOF_DBG_ flags */ int sof_core_debug; @@ -292,6 +295,9 @@ int snd_sof_device_probe(struct device *dev, struct snd_sof_pdata *plat_data) sdev->pdata = plat_data; sdev->first_boot = true; sdev->fw_state = SOF_FW_BOOT_NOT_STARTED; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + sdev->extractor_stream_tag = SOF_PROBE_INVALID_NODE_ID; +#endif dev_set_drvdata(dev, sdev); /* check all mandatory ops */ diff --git a/sound/soc/sof/ops.h b/sound/soc/sof/ops.h index 7f532bcc8e9d..a771500ac442 100644 --- a/sound/soc/sof/ops.h +++ b/sound/soc/sof/ops.h @@ -393,6 +393,49 @@ snd_sof_pcm_platform_pointer(struct snd_sof_dev *sdev, return 0; } +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) +static inline int +snd_sof_probe_compr_assign(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, struct snd_soc_dai *dai) +{ + return sof_ops(sdev)->probe_assign(sdev, cstream, dai); +} + +static inline int +snd_sof_probe_compr_free(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, struct snd_soc_dai *dai) +{ + return sof_ops(sdev)->probe_free(sdev, cstream, dai); +} + +static inline int +snd_sof_probe_compr_set_params(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_params *params, struct snd_soc_dai *dai) +{ + return sof_ops(sdev)->probe_set_params(sdev, cstream, params, dai); +} + +static inline int +snd_sof_probe_compr_trigger(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai) +{ + return sof_ops(sdev)->probe_trigger(sdev, cstream, cmd, dai); +} + +static inline int +snd_sof_probe_compr_pointer(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, struct snd_soc_dai *dai) +{ + if (sof_ops(sdev) && sof_ops(sdev)->probe_pointer) + return sof_ops(sdev)->probe_pointer(sdev, cstream, tstamp, dai); + + return 0; +} +#endif + /* machine driver */ static inline int snd_sof_machine_register(struct snd_sof_dev *sdev, void *pdata) diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 00084471d0de..5d16f668d16a 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -170,6 +170,27 @@ struct snd_sof_dsp_ops { snd_pcm_uframes_t (*pcm_pointer)(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream); /* optional */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + /* Except for probe_pointer, all probe ops are mandatory */ + int (*probe_assign)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); /* mandatory */ + int (*probe_free)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_soc_dai *dai); /* mandatory */ + int (*probe_set_params)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_params *params, + struct snd_soc_dai *dai); /* mandatory */ + int (*probe_trigger)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, int cmd, + struct snd_soc_dai *dai); /* mandatory */ + int (*probe_pointer)(struct snd_sof_dev *sdev, + struct snd_compr_stream *cstream, + struct snd_compr_tstamp *tstamp, + struct snd_soc_dai *dai); /* optional */ +#endif + /* host read DSP stream data */ void (*ipc_msg_data)(struct snd_sof_dev *sdev, struct snd_pcm_substream *substream, @@ -405,6 +426,10 @@ struct snd_sof_dev { wait_queue_head_t waitq; int code_loading; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + unsigned int extractor_stream_tag; +#endif + /* DMA for Trace */ struct snd_dma_buffer dmatb; struct snd_dma_buffer dmatp; -- 2.20.1

1 0

Applied "ASoC: SOF: Provide probe debugfs support" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: SOF: Provide probe debugfs support has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 394695f410c1cd3208906451ba5420f45c420a51 Mon Sep 17 00:00:00 2001 From: Cezary Rojewski <cezary.rojewski(a)intel.com> Date: Tue, 18 Feb 2020 15:39:23 +0100 Subject: [PATCH] ASoC: SOF: Provide probe debugfs support Define debugfs subdirectory delegated for IPC communication with DSP. Input format: uint,uint,(...) which are later translated into DWORDS sequence and further into instances of struct of interest given the IPC type. For Extractor probes, following have been enabled: - PROBE_POINT_ADD (echo <..> probe_points) - PROBE_POINT_REMOVE (echo <..> probe_points_remove) - PROBE_POINT_INFO (cat probe_points) Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200218143924.10565-9-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/debug.c | 226 ++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 226 insertions(+) diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index d2b3b99d3a20..b5c0d6cf72cc 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -17,6 +17,221 @@ #include "sof-priv.h" #include "ops.h" +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) +#include "probe.h" + +/** + * strsplit_u32 - Split string into sequence of u32 tokens + * @buf: String to split into tokens. + * @delim: String containing delimiter characters. + * @tkns: Returned u32 sequence pointer. + * @num_tkns: Returned number of tokens obtained. + */ +static int +strsplit_u32(char **buf, const char *delim, u32 **tkns, size_t *num_tkns) +{ + char *s; + u32 *data, *tmp; + size_t count = 0; + size_t cap = 32; + int ret = 0; + + *tkns = NULL; + *num_tkns = 0; + data = kcalloc(cap, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + while ((s = strsep(buf, delim)) != NULL) { + ret = kstrtouint(s, 0, data + count); + if (ret) + goto exit; + if (++count >= cap) { + cap *= 2; + tmp = krealloc(data, cap * sizeof(*data), GFP_KERNEL); + if (!tmp) { + ret = -ENOMEM; + goto exit; + } + data = tmp; + } + } + + if (!count) + goto exit; + *tkns = kmemdup(data, count * sizeof(*data), GFP_KERNEL); + if (*tkns == NULL) { + ret = -ENOMEM; + goto exit; + } + *num_tkns = count; + +exit: + kfree(data); + return ret; +} + +static int tokenize_input(const char __user *from, size_t count, + loff_t *ppos, u32 **tkns, size_t *num_tkns) +{ + char *buf; + int ret; + + buf = kmalloc(count + 1, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + ret = simple_write_to_buffer(buf, count, ppos, from, count); + if (ret != count) { + ret = ret >= 0 ? -EIO : ret; + goto exit; + } + + buf[count] = '\0'; + ret = strsplit_u32((char **)&buf, ",", tkns, num_tkns); +exit: + kfree(buf); + return ret; +} + +static ssize_t probe_points_read(struct file *file, + char __user *to, size_t count, loff_t *ppos) +{ + struct snd_sof_dfsentry *dfse = file->private_data; + struct snd_sof_dev *sdev = dfse->sdev; + struct sof_probe_point_desc *desc; + size_t num_desc, len = 0; + char *buf; + int i, ret; + + if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) { + dev_warn(sdev->dev, "no extractor stream running\n"); + return -ENOENT; + } + + buf = kzalloc(PAGE_SIZE, GFP_KERNEL); + if (!buf) + return -ENOMEM; + + ret = sof_ipc_probe_points_info(sdev, &desc, &num_desc); + if (ret < 0) + goto exit; + + for (i = 0; i < num_desc; i++) { + ret = snprintf(buf + len, PAGE_SIZE - len, + "Id: %#010x Purpose: %d Node id: %#x\n", + desc[i].buffer_id, desc[i].purpose, desc[i].stream_tag); + if (ret < 0) + goto free_desc; + len += ret; + } + + ret = simple_read_from_buffer(to, count, ppos, buf, len); +free_desc: + kfree(desc); +exit: + kfree(buf); + return ret; +} + +static ssize_t probe_points_write(struct file *file, + const char __user *from, size_t count, loff_t *ppos) +{ + struct snd_sof_dfsentry *dfse = file->private_data; + struct snd_sof_dev *sdev = dfse->sdev; + struct sof_probe_point_desc *desc; + size_t num_tkns, bytes; + u32 *tkns; + int ret; + + if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) { + dev_warn(sdev->dev, "no extractor stream running\n"); + return -ENOENT; + } + + ret = tokenize_input(from, count, ppos, &tkns, &num_tkns); + if (ret < 0) + return ret; + bytes = sizeof(*tkns) * num_tkns; + if (!num_tkns || (bytes % sizeof(*desc))) { + ret = -EINVAL; + goto exit; + } + + desc = (struct sof_probe_point_desc *)tkns; + ret = sof_ipc_probe_points_add(sdev, + desc, bytes / sizeof(*desc)); + if (!ret) + ret = count; +exit: + kfree(tkns); + return ret; +} + +static const struct file_operations probe_points_fops = { + .open = simple_open, + .read = probe_points_read, + .write = probe_points_write, + .llseek = default_llseek, +}; + +static ssize_t probe_points_remove_write(struct file *file, + const char __user *from, size_t count, loff_t *ppos) +{ + struct snd_sof_dfsentry *dfse = file->private_data; + struct snd_sof_dev *sdev = dfse->sdev; + size_t num_tkns; + u32 *tkns; + int ret; + + if (sdev->extractor_stream_tag == SOF_PROBE_INVALID_NODE_ID) { + dev_warn(sdev->dev, "no extractor stream running\n"); + return -ENOENT; + } + + ret = tokenize_input(from, count, ppos, &tkns, &num_tkns); + if (ret < 0) + return ret; + if (!num_tkns) { + ret = -EINVAL; + goto exit; + } + + ret = sof_ipc_probe_points_remove(sdev, tkns, num_tkns); + if (!ret) + ret = count; +exit: + kfree(tkns); + return ret; +} + +static const struct file_operations probe_points_remove_fops = { + .open = simple_open, + .write = probe_points_remove_write, + .llseek = default_llseek, +}; + +static int snd_sof_debugfs_probe_item(struct snd_sof_dev *sdev, + const char *name, mode_t mode, + const struct file_operations *fops) +{ + struct snd_sof_dfsentry *dfse; + + dfse = devm_kzalloc(sdev->dev, sizeof(*dfse), GFP_KERNEL); + if (!dfse) + return -ENOMEM; + + dfse->type = SOF_DFSENTRY_TYPE_BUF; + dfse->sdev = sdev; + + debugfs_create_file(name, mode, sdev->debugfs_root, dfse, fops); + /* add to dfsentry list */ + list_add(&dfse->list, &sdev->dfsentry_list); + + return 0; +} +#endif + #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST) #define MAX_IPC_FLOOD_DURATION_MS 1000 #define MAX_IPC_FLOOD_COUNT 10000 @@ -436,6 +651,17 @@ int snd_sof_dbg_init(struct snd_sof_dev *sdev) return err; } +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + err = snd_sof_debugfs_probe_item(sdev, "probe_points", + 0644, &probe_points_fops); + if (err < 0) + return err; + err = snd_sof_debugfs_probe_item(sdev, "probe_points_remove", + 0200, &probe_points_remove_fops); + if (err < 0) + return err; +#endif + #if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST) /* create read-write ipc_flood_count debugfs entry */ err = snd_sof_debugfs_buf_item(sdev, NULL, 0, -- 2.20.1

1 0

Applied "ASoC: SOF: Intel: Add Probe compress CPU DAIs" to the asoc tree
by Mark Brown 19 Feb '20

19 Feb '20

The patch ASoC: SOF: Intel: Add Probe compress CPU DAIs has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 70368106467cd8c420176bf3ab0acc797f6584bf Mon Sep 17 00:00:00 2001 From: Cezary Rojewski <cezary.rojewski(a)intel.com> Date: Tue, 18 Feb 2020 15:39:24 +0100 Subject: [PATCH] ASoC: SOF: Intel: Add Probe compress CPU DAIs Declare extraction CPU DAI as well as sof_probe_compr_ops. FE DAIs can link against these new CPU DAI to create new compress devices. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200218143924.10565-10-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/compress.c | 5 +++++ sound/soc/sof/compress.h | 2 ++ sound/soc/sof/intel/hda-dai.c | 28 ++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 6 ++++++ sound/soc/sof/pcm.c | 7 +++++++ 5 files changed, 48 insertions(+) diff --git a/sound/soc/sof/compress.c b/sound/soc/sof/compress.c index e87cc81a0599..7354dc6a49cf 100644 --- a/sound/soc/sof/compress.c +++ b/sound/soc/sof/compress.c @@ -13,6 +13,11 @@ #include "ops.h" #include "probe.h" +struct snd_compr_ops sof_probe_compressed_ops = { + .copy = sof_probe_compr_copy, +}; +EXPORT_SYMBOL(sof_probe_compressed_ops); + int sof_probe_compr_open(struct snd_compr_stream *cstream, struct snd_soc_dai *dai) { diff --git a/sound/soc/sof/compress.h b/sound/soc/sof/compress.h index dccc9e008f81..800f163603e1 100644 --- a/sound/soc/sof/compress.h +++ b/sound/soc/sof/compress.h @@ -13,6 +13,8 @@ #include <sound/compress_driver.h> +extern struct snd_compr_ops sof_probe_compressed_ops; + int sof_probe_compr_open(struct snd_compr_stream *cstream, struct snd_soc_dai *dai); int sof_probe_compr_free(struct snd_compr_stream *cstream, diff --git a/sound/soc/sof/intel/hda-dai.c b/sound/soc/sof/intel/hda-dai.c index 9c6e3f990ee3..ed5e7d2c0d43 100644 --- a/sound/soc/sof/intel/hda-dai.c +++ b/sound/soc/sof/intel/hda-dai.c @@ -399,6 +399,19 @@ static const struct snd_soc_dai_ops hda_link_dai_ops = { .trigger = hda_link_pcm_trigger, .prepare = hda_link_pcm_prepare, }; + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) +#include "../compress.h" + +static struct snd_soc_cdai_ops sof_probe_compr_ops = { + .startup = sof_probe_compr_open, + .shutdown = sof_probe_compr_free, + .set_params = sof_probe_compr_set_params, + .trigger = sof_probe_compr_trigger, + .pointer = sof_probe_compr_pointer, +}; + +#endif #endif /* @@ -460,5 +473,20 @@ struct snd_soc_dai_driver skl_dai[] = { .name = "Alt Analog CPU DAI", .ops = &hda_link_dai_ops, }, +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) +{ + .name = "Probe Extraction CPU DAI", + .compress_new = snd_soc_new_compress, + .cops = &sof_probe_compr_ops, + .capture = { + .stream_name = "Probe Extraction", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + }, +}, +#endif #endif }; diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index ca44ecb76534..537c0a930a15 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -349,7 +349,13 @@ /* Number of DAIs */ #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA_PROBES) +#define SOF_SKL_NUM_DAIS 16 +#else #define SOF_SKL_NUM_DAIS 15 +#endif + #else #define SOF_SKL_NUM_DAIS 8 #endif diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index db3df02c7398..b239bbff4b5c 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -16,6 +16,9 @@ #include "sof-priv.h" #include "sof-audio.h" #include "ops.h" +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) +#include "compress.h" +#endif /* Create DMA buffer page table for DSP */ static int create_page_table(struct snd_soc_component *component, @@ -787,6 +790,10 @@ void snd_sof_new_platform_drv(struct snd_sof_dev *sdev) #if IS_ENABLED(CONFIG_SND_SOC_SOF_COMPRESS) pd->compr_ops = &sof_compressed_ops; +#endif +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_PROBES) + /* override cops when probe support is enabled */ + pd->compr_ops = &sof_probe_compressed_ops; #endif pd->pcm_construct = sof_pcm_new; pd->ignore_machine = drv_name; -- 2.20.1

1 0