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[PATCH AUTOSEL 5.14 200/252] ASoC: Intel: Skylake: Fix module configuration for KPB and MIXER
by Sasha Levin 09 Sep '21

09 Sep '21

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit e4e0633bcadc950b4b4af06c7f1bb7f7e3e86321 ] KeyPhrasebuffer, Mixin and Mixout modules configuration is described by firmware's basic module configuration structure. There are no extended parameters required. Update functions taking part in building INIT_INSTANCE IPC payload to reflect that. Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Tested-by: Lukasz Majczak <lma(a)semihalf.com> Link: https://lore.kernel.org/r/20210818075742.1515155-6-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/skylake/skl-messages.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 476ef1897961..79c6cf2c14bf 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -802,9 +802,12 @@ static u16 skl_get_module_param_size(struct skl_dev *skl, case SKL_MODULE_TYPE_BASE_OUTFMT: case SKL_MODULE_TYPE_MIC_SELECT: - case SKL_MODULE_TYPE_KPB: return sizeof(struct skl_base_outfmt_cfg); + case SKL_MODULE_TYPE_MIXER: + case SKL_MODULE_TYPE_KPB: + return sizeof(struct skl_base_cfg); + default: /* * return only base cfg when no specific module type is @@ -857,10 +860,14 @@ static int skl_set_module_format(struct skl_dev *skl, case SKL_MODULE_TYPE_BASE_OUTFMT: case SKL_MODULE_TYPE_MIC_SELECT: - case SKL_MODULE_TYPE_KPB: skl_set_base_outfmt_format(skl, module_config, *param_data); break; + case SKL_MODULE_TYPE_MIXER: + case SKL_MODULE_TYPE_KPB: + skl_set_base_module_format(skl, module_config, *param_data); + break; + default: skl_set_base_module_format(skl, module_config, *param_data); break; -- 2.30.2

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[PATCH AUTOSEL 5.14 199/252] soundwire: intel: fix potential race condition during power down
by Sasha Levin 09 Sep '21

09 Sep '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit ea6942dad4b2a7e1735aa0f10f3d0b04b847750f ] The power down sequence sets the link_up flag as false outside of the mutex_lock. This is potentially unsafe. In additional the flow in that sequence can be improved by first testing if the link was powered, setting the link_up flag as false and proceeding with the power down. In case the CPA bits cannot be cleared, we only flag an error since we cannot deal with interrupts any longer. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20210818024954.16873-2-yung-chuan.liao@linux.inte… Signed-off-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/intel.c | 23 +++++++++++++---------- 1 file changed, 13 insertions(+), 10 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index c11e3d8cd308..f156de765c68 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -537,12 +537,14 @@ static int intel_link_power_down(struct sdw_intel *sdw) mutex_lock(sdw->link_res->shim_lock); - intel_shim_master_ip_to_glue(sdw); - if (!(*shim_mask & BIT(link_id))) dev_err(sdw->cdns.dev, "%s: Unbalanced power-up/down calls\n", __func__); + sdw->cdns.link_up = false; + + intel_shim_master_ip_to_glue(sdw); + *shim_mask &= ~BIT(link_id); if (!*shim_mask) { @@ -559,18 +561,19 @@ static int intel_link_power_down(struct sdw_intel *sdw) link_control &= spa_mask; ret = intel_clear_bit(shim, SDW_SHIM_LCTL, link_control, cpa_mask); + if (ret < 0) { + dev_err(sdw->cdns.dev, "%s: could not power down link\n", __func__); + + /* + * we leave the sdw->cdns.link_up flag as false since we've disabled + * the link at this point and cannot handle interrupts any longer. + */ + } } mutex_unlock(sdw->link_res->shim_lock); - if (ret < 0) { - dev_err(sdw->cdns.dev, "%s: could not power down link\n", __func__); - - return ret; - } - - sdw->cdns.link_up = false; - return 0; + return ret; } static void intel_shim_sync_arm(struct sdw_intel *sdw) -- 2.30.2

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[PATCH AUTOSEL 5.14 179/252] ASoC: rsnd: adg: clearly handle clock error / NULL case
by Sasha Levin 09 Sep '21

09 Sep '21

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> [ Upstream commit cc64c390b215b404524725a94857d6fb58d9a62a ] This driver is assuming that all adg->clk[i] is not NULL. Because of this prerequisites, for_each_rsnd_clk() is possible to work for all clk without checking NULL. In other words, all adg->clk[i] should not NULL. Some SoC might doesn't have clk_a/b/c/i. devm_clk_get() returns error in such case. This driver calls rsnd_adg_null_clk_get() and use null_clk instead of NULL in such cases. But devm_clk_get() might returns NULL even though such clocks exist, but it doesn't mean error (user deliberately chose to disable the feature). NULL clk itself is not error from clk point of view, but is error from this driver point of view because it is not assuming such case. But current code is using IS_ERR() which doesn't care NULL. This driver uses IS_ERR_OR_NULL() instead of IS_ERR() for clk check. And it uses ERR_CAST() to clarify null_clk error. One concern here is that it unconditionally uses null_clk if clk_a/b/c/i was error. It is correct if it doesn't exist, but is not correct if it returns error even though it exist. It needs to check "clock-names" from DT before calling devm_clk_get() to handling such case. But let's assume it is overkill so far. Link: https://lore.kernel.org/r/YMCmhfQUimHCSH/n@mwanda Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Link: https://lore.kernel.org/r/87v940wyf9.wl-kuninori.morimoto.gx@renesas.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/sh/rcar/adg.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 0ebee1ed06a9..5f1e72edfee0 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -391,9 +391,9 @@ static struct clk *rsnd_adg_create_null_clk(struct rsnd_priv *priv, struct clk *clk; clk = clk_register_fixed_rate(dev, name, parent, 0, 0); - if (IS_ERR(clk)) { + if (IS_ERR_OR_NULL(clk)) { dev_err(dev, "create null clk error\n"); - return NULL; + return ERR_CAST(clk); } return clk; @@ -430,9 +430,9 @@ static int rsnd_adg_get_clkin(struct rsnd_priv *priv) for (i = 0; i < CLKMAX; i++) { clk = devm_clk_get(dev, clk_name[i]); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) clk = rsnd_adg_null_clk_get(priv); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) goto err; adg->clk[i] = clk; @@ -582,7 +582,7 @@ static int rsnd_adg_get_clkout(struct rsnd_priv *priv) if (!count) { clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT], parent_clk_name, 0, req_rate[0]); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) goto err; adg->clkout[CLKOUT] = clk; @@ -596,7 +596,7 @@ static int rsnd_adg_get_clkout(struct rsnd_priv *priv) clk = clk_register_fixed_rate(dev, clkout_name[i], parent_clk_name, 0, req_rate[0]); - if (IS_ERR(clk)) + if (IS_ERR_OR_NULL(clk)) goto err; adg->clkout[i] = clk; -- 2.30.2

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[PATCH AUTOSEL 5.14 102/252] ASoC: Intel: update sof_pcm512x quirks
by Sasha Levin 09 Sep '21

09 Sep '21

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> [ Upstream commit 22414cade8dfec25ab94df52b3a4f7aa8edb6120 ] The default SOF topology enables SSP capture and DMICs, even though both of these hardware capabilities are not always available in hardware (specific versions of HiFiberry and DMIC kit needed). For the SSP capture, this leads to annoying "SP5-Codec: ASoC: no backend capture" and "streamSSP5-Codec: ASoC: no users capture at close - state 0" errors. Update the quirks to match what the topology needs, which also allows for the ability to remove SSP capture and DMIC support. BugLink: https://github.com/thesofproject/linux/issues/3061 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Guennadi Liakhovetski <guennadi.liakhovetski(a)linux.intel.com> Link: https://lore.kernel.org/r/20210802152151.15832-4-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_pcm512x.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/sof_pcm512x.c b/sound/soc/intel/boards/sof_pcm512x.c index 2ec9c62366e2..6815204e58d5 100644 --- a/sound/soc/intel/boards/sof_pcm512x.c +++ b/sound/soc/intel/boards/sof_pcm512x.c @@ -26,11 +26,16 @@ #define SOF_PCM512X_SSP_CODEC(quirk) ((quirk) & GENMASK(3, 0)) #define SOF_PCM512X_SSP_CODEC_MASK (GENMASK(3, 0)) +#define SOF_PCM512X_ENABLE_SSP_CAPTURE BIT(4) +#define SOF_PCM512X_ENABLE_DMIC BIT(5) #define IDISP_CODEC_MASK 0x4 /* Default: SSP5 */ -static unsigned long sof_pcm512x_quirk = SOF_PCM512X_SSP_CODEC(5); +static unsigned long sof_pcm512x_quirk = + SOF_PCM512X_SSP_CODEC(5) | + SOF_PCM512X_ENABLE_SSP_CAPTURE | + SOF_PCM512X_ENABLE_DMIC; static bool is_legacy_cpu; @@ -244,8 +249,9 @@ static struct snd_soc_dai_link *sof_card_dai_links_create(struct device *dev, links[id].dpcm_playback = 1; /* * capture only supported with specific versions of the Hifiberry DAC+ - * links[id].dpcm_capture = 1; */ + if (sof_pcm512x_quirk & SOF_PCM512X_ENABLE_SSP_CAPTURE) + links[id].dpcm_capture = 1; links[id].no_pcm = 1; links[id].cpus = &cpus[id]; links[id].num_cpus = 1; @@ -380,6 +386,9 @@ static int sof_audio_probe(struct platform_device *pdev) ssp_codec = sof_pcm512x_quirk & SOF_PCM512X_SSP_CODEC_MASK; + if (!(sof_pcm512x_quirk & SOF_PCM512X_ENABLE_DMIC)) + dmic_be_num = 0; + /* compute number of dai links */ sof_audio_card_pcm512x.num_links = 1 + dmic_be_num + hdmi_num; -- 2.30.2

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[PATCH AUTOSEL 5.14 100/252] ASoC: Intel: bytcr_rt5640: Move "Platform Clock" routes to the maps for the matching in-/output
by Sasha Levin 09 Sep '21

09 Sep '21

From: Hans de Goede <hdegoede(a)redhat.com> [ Upstream commit dccd1dfd0770bfd494b68d1135b4547b2c602c42 ] Move the "Platform Clock" routes for the "Internal Mic" and "Speaker" routes to the intmic_*_map[] / *_spk_map[] arrays. This ensures that these "Platform Clock" routes do not get added when the BYT_RT5640_NO_INTERNAL_MIC_MAP / BYT_RT5640_NO_SPEAKERS quirks are used. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20210802142501.991985-2-hdegoede@redhat.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/bytcr_rt5640.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 91a6d712eb58..c403fb672594 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -290,9 +290,6 @@ static const struct snd_soc_dapm_widget byt_rt5640_widgets[] = { static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { {"Headphone", NULL, "Platform Clock"}, {"Headset Mic", NULL, "Platform Clock"}, - {"Internal Mic", NULL, "Platform Clock"}, - {"Speaker", NULL, "Platform Clock"}, - {"Headset Mic", NULL, "MICBIAS1"}, {"IN2P", NULL, "Headset Mic"}, {"Headphone", NULL, "HPOL"}, @@ -300,19 +297,23 @@ static const struct snd_soc_dapm_route byt_rt5640_audio_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC1", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_dmic2_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"DMIC2", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in1_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN1P", NULL, "Internal Mic"}, }; static const struct snd_soc_dapm_route byt_rt5640_intmic_in3_map[] = { + {"Internal Mic", NULL, "Platform Clock"}, {"Internal Mic", NULL, "MICBIAS1"}, {"IN3P", NULL, "Internal Mic"}, }; @@ -354,6 +355,7 @@ static const struct snd_soc_dapm_route byt_rt5640_ssp0_aif2_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, {"Speaker", NULL, "SPORP"}, @@ -361,6 +363,7 @@ static const struct snd_soc_dapm_route byt_rt5640_stereo_spk_map[] = { }; static const struct snd_soc_dapm_route byt_rt5640_mono_spk_map[] = { + {"Speaker", NULL, "Platform Clock"}, {"Speaker", NULL, "SPOLP"}, {"Speaker", NULL, "SPOLN"}, }; -- 2.30.2

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[PATCH AUTOSEL 5.14 013/252] ASoC: atmel: ATMEL drivers don't need HAS_DMA
by Sasha Levin 09 Sep '21

09 Sep '21

From: Randy Dunlap <rdunlap(a)infradead.org> [ Upstream commit 6c5c659dfe3f02e08054a6c20019e3886618b512 ] On a config (such as arch/sh/) which does not set HAS_DMA when MMU is not set, several ATMEL ASoC drivers select symbols that cause kconfig warnings. There is one "depends on HAS_DMA" which is no longer needed. Dropping it eliminates the kconfig warnings and still builds with no problems reported. Fix the following kconfig warnings: WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] - SND_ATMEL_SOC_SSC_PDC [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_PDC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_AT91_SOC_SAM9G20_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_SSC_DMA [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] WARNING: unmet direct dependencies detected for SND_ATMEL_SOC_SSC_DMA Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && ATMEL_SSC [=m] && HAS_DMA [=n] Selected by [m]: - SND_ATMEL_SOC_WM8904 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && I2C [=m] - SND_AT91_SOC_SAM9X5_WM8731 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_ATMEL_SOC [=m] && (ARCH_AT91 || COMPILE_TEST [=y]) && ATMEL_SSC [=m] && SND_SOC_I2C_AND_SPI [=m] Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Reviewed-by: Codrin Ciubotariu <codrin.ciubotariu(a)microchip.com> Acked-by: Alexandre Belloni <alexandre.belloni(a)bootlin.com> Link: https://lore.kernel.org/r/20210707214752.3831-1-rdunlap@infradead.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/atmel/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index ec04e3386bc0..8617793ed955 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -11,7 +11,6 @@ if SND_ATMEL_SOC config SND_ATMEL_SOC_PDC bool - depends on HAS_DMA config SND_ATMEL_SOC_DMA bool -- 2.30.2

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[PATCH AUTOSEL 5.14 012/252] ASoC: ti: davinci-mcasp: Fix DIT mode support
by Sasha Levin 09 Sep '21

09 Sep '21

From: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> [ Upstream commit bbdd3f4dbe81e19b9123bc54e23ed54517615524 ] The DIT mode support has not been tested due to lack of platform where it can be tested. To be able to use the McASP on OMAP4/5 (only supporting DIT mode) we need to have DIT mode working in the McASP driver on a know platform. After hacking around (on BBW, mcasp1.axr1 can be routed out for this) it appeared that DIT mode is broken. This patch fixes it up and 16/24 bit audio works along with passthrough, but I have only tested with DTS example and test files. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> Link: https://lore.kernel.org/r/20210705194249.2385-2-peter.ujfalusi@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/ti/davinci-mcasp.c | 150 ++++++++++++++++++++++++++++++----- 1 file changed, 129 insertions(+), 21 deletions(-) diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c index 017a5a5e56cd..64ec6d485834 100644 --- a/sound/soc/ti/davinci-mcasp.c +++ b/sound/soc/ti/davinci-mcasp.c @@ -83,6 +83,8 @@ struct davinci_mcasp { struct snd_pcm_substream *substreams[2]; unsigned int dai_fmt; + u32 iec958_status; + /* Audio can not be enabled due to missing parameter(s) */ bool missing_audio_param; @@ -757,6 +759,9 @@ static int davinci_mcasp_set_tdm_slot(struct snd_soc_dai *dai, { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) + return 0; + dev_dbg(mcasp->dev, "%s() tx_mask 0x%08x rx_mask 0x%08x slots %d width %d\n", __func__, tx_mask, rx_mask, slots, slot_width); @@ -827,6 +832,20 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXROT(rx_rotate), RXROT(7)); mcasp_set_reg(mcasp, DAVINCI_MCASP_RXMASK_REG, mask); + } else { + /* + * according to the TRM it should be TXROT=0, this one works: + * 16 bit to 23-8 (TXROT=6, rotate 24 bits) + * 24 bit to 23-0 (TXROT=0, rotate 0 bits) + * + * TXROT = 0 only works with 24bit samples + */ + tx_rotate = (sample_width / 4 + 2) & 0x7; + + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(tx_rotate), + TXROT(7)); + mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXSSZ(15), + TXSSZ(0x0F)); } mcasp_set_reg(mcasp, DAVINCI_MCASP_TXMASK_REG, mask); @@ -842,10 +861,16 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, u8 tx_ser = 0; u8 rx_ser = 0; u8 slots = mcasp->tdm_slots; - u8 max_active_serializers = (channels + slots - 1) / slots; - u8 max_rx_serializers, max_tx_serializers; + u8 max_active_serializers, max_rx_serializers, max_tx_serializers; int active_serializers, numevt; u32 reg; + + /* In DIT mode we only allow maximum of one serializers for now */ + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) + max_active_serializers = 1; + else + max_active_serializers = (channels + slots - 1) / slots; + /* Default configuration */ if (mcasp->version < MCASP_VERSION_3) mcasp_set_bits(mcasp, DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -1031,16 +1056,18 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream, static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, unsigned int rate) { - u32 cs_value = 0; - u8 *cs_bytes = (u8*) &cs_value; + u8 *cs_bytes = (u8 *)&mcasp->iec958_status; - /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 - and LSB first */ - mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15)); + if (!mcasp->dat_port) + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXSEL); + else + mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXSEL); /* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */ mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE | FSXMOD(0x180)); + mcasp_set_reg(mcasp, DAVINCI_MCASP_TXMASK_REG, 0xFFFF); + /* Set the TX tdm : for all the slots */ mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF); @@ -1049,16 +1076,8 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, mcasp_clr_bits(mcasp, DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS); - /* Only 44100 and 48000 are valid, both have the same setting */ - mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXDIV(3)); - - /* Enable the DIT */ - mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); - /* Set S/PDIF channel status bits */ - cs_bytes[0] = IEC958_AES0_CON_NOT_COPYRIGHT; - cs_bytes[1] = IEC958_AES1_CON_PCM_CODER; - + cs_bytes[3] &= ~IEC958_AES3_CON_FS; switch (rate) { case 22050: cs_bytes[3] |= IEC958_AES3_CON_FS_22050; @@ -1088,12 +1107,15 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, cs_bytes[3] |= IEC958_AES3_CON_FS_192000; break; default: - printk(KERN_WARNING "unsupported sampling rate: %d\n", rate); + dev_err(mcasp->dev, "unsupported sampling rate: %d\n", rate); return -EINVAL; } - mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG, cs_value); - mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG, cs_value); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG, mcasp->iec958_status); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG, mcasp->iec958_status); + + /* Enable the DIT */ + mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); return 0; } @@ -1237,12 +1259,18 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, int slots = mcasp->tdm_slots; int rate = params_rate(params); int sbits = params_width(params); + unsigned int bclk_target; if (mcasp->slot_width) sbits = mcasp->slot_width; + if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) + bclk_target = rate * sbits * slots; + else + bclk_target = rate * 128; + davinci_mcasp_calc_clk_div(mcasp, mcasp->sysclk_freq, - rate * sbits * slots, true); + bclk_target, true); } ret = mcasp_common_hw_param(mcasp, substream->stream, @@ -1598,6 +1626,77 @@ static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = { .set_tdm_slot = davinci_mcasp_set_tdm_slot, }; +static int davinci_mcasp_iec958_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + + return 0; +} + +static int davinci_mcasp_iec958_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uctl) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + + memcpy(uctl->value.iec958.status, &mcasp->iec958_status, + sizeof(mcasp->iec958_status)); + + return 0; +} + +static int davinci_mcasp_iec958_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uctl) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + + memcpy(&mcasp->iec958_status, uctl->value.iec958.status, + sizeof(mcasp->iec958_status)); + + return 0; +} + +static int davinci_mcasp_iec958_con_mask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol); + struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai); + + memset(ucontrol->value.iec958.status, 0xff, sizeof(mcasp->iec958_status)); + return 0; +} + +static const struct snd_kcontrol_new davinci_mcasp_iec958_ctls[] = { + { + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_VOLATILE), + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT), + .info = davinci_mcasp_iec958_info, + .get = davinci_mcasp_iec958_get, + .put = davinci_mcasp_iec958_put, + }, { + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK), + .info = davinci_mcasp_iec958_info, + .get = davinci_mcasp_iec958_con_mask_get, + }, +}; + +static void davinci_mcasp_init_iec958_status(struct davinci_mcasp *mcasp) +{ + unsigned char *cs = (u8 *)&mcasp->iec958_status; + + cs[0] = IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_NONE; + cs[1] = IEC958_AES1_CON_PCM_CODER; + cs[2] = IEC958_AES2_CON_SOURCE_UNSPEC | IEC958_AES2_CON_CHANNEL_UNSPEC; + cs[3] = IEC958_AES3_CON_CLOCK_1000PPM; +} + static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) { struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai); @@ -1605,6 +1704,12 @@ static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai) dai->playback_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK]; dai->capture_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE]; + if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) { + davinci_mcasp_init_iec958_status(mcasp); + snd_soc_add_dai_controls(dai, davinci_mcasp_iec958_ctls, + ARRAY_SIZE(davinci_mcasp_iec958_ctls)); + } + return 0; } @@ -1651,7 +1756,8 @@ static struct snd_soc_dai_driver davinci_mcasp_dai[] = { .channels_min = 1, .channels_max = 384, .rates = DAVINCI_MCASP_RATES, - .formats = DAVINCI_MCASP_PCM_FMTS, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE, }, .ops = &davinci_mcasp_dai_ops, }, @@ -1871,6 +1977,8 @@ static int davinci_mcasp_get_config(struct davinci_mcasp *mcasp, } else { mcasp->tdm_slots = pdata->tdm_slots; } + } else { + mcasp->tdm_slots = 32; } mcasp->num_serializer = pdata->num_serializer; -- 2.30.2

1 0

[PATCH] ASoC: SOF: fix a typo in put operations for kcontrol
by Pierre-Louis Bossart 08 Sep '21

08 Sep '21

From: Rander Wang <rander.wang(a)intel.com> SOF_CTRL_TYPE_VALUE_CHAN_SET should be used for put operations for consistency. The current use of _GET is obviously incorrect but _GET and _SET result in the same action so there is no functional change introduced by this patch. Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Rander Wang <rander.wang(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/control.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sof/control.c b/sound/soc/sof/control.c index a5dd728c580a..504500dd4d43 100644 --- a/sound/soc/sof/control.c +++ b/sound/soc/sof/control.c @@ -108,7 +108,7 @@ int snd_sof_volume_put(struct snd_kcontrol *kcontrol, if (pm_runtime_active(scomp->dev)) snd_sof_ipc_set_get_comp_data(scontrol, SOF_IPC_COMP_SET_VALUE, - SOF_CTRL_TYPE_VALUE_CHAN_GET, + SOF_CTRL_TYPE_VALUE_CHAN_SET, SOF_CTRL_CMD_VOLUME, true); return change; @@ -179,7 +179,7 @@ int snd_sof_switch_put(struct snd_kcontrol *kcontrol, if (pm_runtime_active(scomp->dev)) snd_sof_ipc_set_get_comp_data(scontrol, SOF_IPC_COMP_SET_VALUE, - SOF_CTRL_TYPE_VALUE_CHAN_GET, + SOF_CTRL_TYPE_VALUE_CHAN_SET, SOF_CTRL_CMD_SWITCH, true); @@ -226,7 +226,7 @@ int snd_sof_enum_put(struct snd_kcontrol *kcontrol, if (pm_runtime_active(scomp->dev)) snd_sof_ipc_set_get_comp_data(scontrol, SOF_IPC_COMP_SET_VALUE, - SOF_CTRL_TYPE_VALUE_CHAN_GET, + SOF_CTRL_TYPE_VALUE_CHAN_SET, SOF_CTRL_CMD_ENUM, true); -- 2.25.1

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[GIT PULL] ASoC fixes for v5.15-rc1
by Mark Brown 08 Sep '21

08 Sep '21

The following changes since commit 38b7673000949ca784fcb8a9feb70d2a802befa6: Merge remote-tracking branch 'asoc/for-5.15' into asoc-linus (2021-08-30 12:30:33 +0100) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.15-rc1 for you to fetch changes up to fcb958ee8e832e9cdf43408535207e15f14af755: ASoC: rockchip: i2s: Fix concurrency between tx/rx (2021-09-03 16:19:01 +0100) ---------------------------------------------------------------- ASoC: Fixes for v5.15 A collection of fixes that came in during the merge window, nothing too remarkable but a reasonably large number of fixes. ---------------------------------------------------------------- Colin Ian King (1): ASoC: samsung: s3c24xx_simtec: fix spelling mistake "devicec" -> "device" Geert Uytterhoeven (1): ASoC: mediatek: SND_SOC_MT8195 should depend on ARCH_MEDIATEK Kuninori Morimoto (1): ASoC: audio-graph: respawn Platform Support Mark Brown (1): Merge tag 'asoc-v5.15' into asoc-5.15 Nathan Chancellor (1): ASoC: Intel: boards: Fix CONFIG_SND_SOC_SDW_MOCKUP select Shengjiu Wang (1): ASoC: dt-bindings: fsl_rpmsg: Add compatible string for i.MX8ULP Shuming Fan (1): ASoC: rt5682: fix headset background noise when S3 state Sugar Zhang (1): ASoC: rockchip: i2s: Fix concurrency between tx/rx Trevor Wu (3): ASoC: mediatek: mt8195: add MTK_PMIC_WRAP dependency ASoC: dt-bindings: mt8195: remove dependent headers in the example ASoC: mt8195: correct the dts parsing logic about DPTX and HDMITX .../devicetree/bindings/sound/fsl,rpmsg.yaml | 1 + .../devicetree/bindings/sound/mt8195-afe-pcm.yaml | 40 ++++++++++------------ sound/soc/codecs/rt5682.c | 3 -- sound/soc/generic/audio-graph-card.c | 6 ++++ sound/soc/intel/boards/Kconfig | 2 +- sound/soc/mediatek/Kconfig | 3 +- .../mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c | 22 ++++++------ sound/soc/rockchip/rockchip_i2s.c | 7 ++++ sound/soc/samsung/s3c24xx_simtec.c | 2 +- 9 files changed, 47 insertions(+), 39 deletions(-)

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[PATCH v5 00/21] ASoC: qcom: Add AudioReach support
by Srinivas Kandagatla 08 Sep '21

08 Sep '21

Many thanks for reviewing v4. This version addresses all the comments raised as part of v4 review. This patchset adds ASoC driver support to configure signal processing framework ("AudioReach") which is integral part of Qualcomm next generation audio SDK and will be deployed on upcoming Qualcomm chipsets. It makes use of ASoC Topology to load graphs on to the DSP which is then managed by APM (Audio Processing Manager) service to prepare/start/stop. Here is simplified high-level block diagram of AudioReach: ___________________________________________________________ | CPU (Application Processor) | | +---------+ +---------+ +----------+ | | | q6apm | | q6apm | | q6apm | | | | dais | <------> | | <-----> |lpass-dais| | | +---------+ +---------+ +----------+ | | ^ ^ | | | | +---------+ | | +---------+ v +---------->|topology | | | | q6prm | +---------+ | | | | | |<-------->| GPR | +---------+ | | +---------+ +---------+ | | ^ ^ | | | | | | +----------+ | | | | q6prm | | | | |lpass-clks| | | | +----------+ | | |____________________________|______________________________| | | RPMSG (IPC over GLINK) ____________________________|______________________________ | | | | +-----------------------+ | | | | | | v v q6 (Audio DSP) | |+-----+ +----------------------------------+ | || PRM | | APM (Audio Processing Manager) | | |+-----+ | . Graph Management | | | | . Command Handing | | | | . Event Management | | | | ... | | | +----------------------------------+ | | ^ | |____________________________|______________________________| | | LPASS AIF ____________________________|______________________________ | | Audio I/O | | v | | +--------------------------------------------------+ | | | Audio devices | | | | CODEC | HDMI-TX | PCM | SLIMBUS | I2S |MI2S |...| | | | | | | +--------------------------------------------------+ | |___________________________________________________________| AudioReach has constructs of sub-graph, container and modules. Each sub-graph can have N containers and each Container can have N Modules and connections between them can be linear or non-linear. An audio function can be realized with one or many connected sub-graphs. There are also control/event paths between modules that can be wired up while building graph to achieve various control mechanism between modules. These concepts of Sub-Graph, Containers and Modules are represented in ASoC topology. Here is simple I2S graph with a Write Shared Memory and a Volume control module within a single Subgraph (1) with one Container (1) and 5 modules. ____________________________________________________________ | Sub-Graph [1] | | _______________________________________________________ | | | Container [1] | | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]-> [I2S-EP]| | | |_______________________________________________________| | |____________________________________________________________| For now this graph is split into two subgraphs to achieve dpcm like below: ________________________________________________ _________________ | Sub-Graph [1] | | Sub-Graph [2] | | ____________________________________________ | | _____________ | | | Container [1] | | | |Container [2]| | | | [WR_SH] -> [PCM DEC] -> [PCM CONV] -> [VOL]| | | | [I2S-EP] | | | |____________________________________________| | | |_____________| | |________________________________________________| |_________________| _________________ | Sub-Graph [3] | | _____________ | | |Container [3]| | | | [DMA-EP] | | | |_____________| | |_________________| This patchset adds very minimal support for AudioReach which includes supporting sub-graphs containing CODEC DMA ports and simple PCM Decoder/Encoder and Logger Modules. Additional capabilities will be built over time to expose features offered by AudioReach. This patchset is Tested on SM8250 SoC based Qualcomm Robotics Platform RB5 and SM9250 MTP with WSA881X Smart Speaker Amplifiers, DMICs connected via VA Macro and WCD938x Codec connected via TX and RX Macro and HDMI audio via I2S. First 10 Patches are mostly reorganization existing Old QDSP Audio Framework code and bindings so that we could reuse them on AudioReach. ASoC topology graphs for DragonBoard RB5 and SM8250 MTP are available at https://git.linaro.org/people/srinivas.kandagatla/audioreach-topology.git/ Thanks, srini Changes since v4: only apr dt-binding changes. - deprecated qcom,apr-domain dt-bindings to make it more reusable - added more protection-domain list that have been missed during yaml conversion. - remove "qcom,apr" compatible that was added by mistake during yaml conversion. Srinivas Kandagatla (21): soc: dt-bindings: qcom: apr: convert to yaml soc: dt-bindings: qcom: apr: deprecate qcom,apr-domain property soc: qcom: apr: make code more reuseable soc: dt-bindings: qcom: add gpr bindings soc: qcom: apr: Add GPR support ASoC: dt-bindings: move LPASS dai related bindings out of q6afe ASoC: dt-bindings: move LPASS clocks related bindings out of q6afe ASoC: dt-bindings: rename q6afe.h to q6dsp-lpass-ports.h ASoC: qdsp6: q6afe-dai: move lpass audio ports to common file ASoC: qdsp6: q6afe-clocks: move audio-clocks to common file ASoC: dt-bindings: q6dsp: add q6apm-lpass-dai compatible ASoC: dt-bindings: lpass-clocks: add q6prm clocks compatible ASoC: dt-bindings: add q6apm digital audio stream bindings ASoC: qdsp6: audioreach: add basic pkt alloc support ASoC: qdsp6: audioreach: add q6apm support ASoC: qdsp6: audioreach: add module configuration command helpers ASoC: qdsp6: audioreach: add topology support ASoC: qdsp6: audioreach: add q6apm-dai support ASoC: qdsp6: audioreach: add q6apm lpass dai support ASoC: qdsp6: audioreach: add q6prm support ASoC: qdsp6: audioreach: add support for q6prm-clocks .../devicetree/bindings/soc/qcom/qcom,apr.txt | 134 -- .../bindings/soc/qcom/qcom,apr.yaml | 169 +++ .../devicetree/bindings/sound/qcom,q6afe.txt | 181 --- .../bindings/sound/qcom,q6apm-dai.yaml | 50 + .../sound/qcom,q6dsp-lpass-clocks.yaml | 74 ++ .../sound/qcom,q6dsp-lpass-ports.yaml | 202 +++ drivers/soc/qcom/Kconfig | 8 + drivers/soc/qcom/apr.c | 289 +++- include/dt-bindings/soc/qcom,gpr.h | 18 + include/dt-bindings/sound/qcom,q6afe.h | 203 +-- .../sound/qcom,q6dsp-lpass-ports.h | 208 +++ include/linux/soc/qcom/apr.h | 70 +- include/uapi/sound/snd_ar_tokens.h | 203 +++ sound/soc/qcom/Kconfig | 22 + sound/soc/qcom/qdsp6/Makefile | 11 +- sound/soc/qcom/qdsp6/audioreach.c | 1170 +++++++++++++++++ sound/soc/qcom/qdsp6/audioreach.h | 692 ++++++++++ sound/soc/qcom/qdsp6/q6afe-clocks.c | 187 +-- sound/soc/qcom/qdsp6/q6afe-dai.c | 687 +--------- sound/soc/qcom/qdsp6/q6apm-dai.c | 504 +++++++ sound/soc/qcom/qdsp6/q6apm-lpass-dais.c | 266 ++++ sound/soc/qcom/qdsp6/q6apm.c | 916 +++++++++++++ sound/soc/qcom/qdsp6/q6apm.h | 154 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c | 186 +++ sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h | 30 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 627 +++++++++ sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 22 + sound/soc/qcom/qdsp6/q6prm-clocks.c | 85 ++ sound/soc/qcom/qdsp6/q6prm.c | 256 ++++ sound/soc/qcom/qdsp6/q6prm.h | 78 ++ sound/soc/qcom/qdsp6/topology.c | 1118 ++++++++++++++++ 31 files changed, 7406 insertions(+), 1414 deletions(-) delete mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.txt create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm-dai.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-clocks.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6dsp-lpass-ports.yaml create mode 100644 include/dt-bindings/soc/qcom,gpr.h create mode 100644 include/dt-bindings/sound/qcom,q6dsp-lpass-ports.h create mode 100644 include/uapi/sound/snd_ar_tokens.h create mode 100644 sound/soc/qcom/qdsp6/audioreach.c create mode 100644 sound/soc/qcom/qdsp6/audioreach.h create mode 100644 sound/soc/qcom/qdsp6/q6apm-dai.c create mode 100644 sound/soc/qcom/qdsp6/q6apm-lpass-dais.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.c create mode 100644 sound/soc/qcom/qdsp6/q6apm.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-clocks.h create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c create mode 100644 sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h create mode 100644 sound/soc/qcom/qdsp6/q6prm-clocks.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.c create mode 100644 sound/soc/qcom/qdsp6/q6prm.h create mode 100644 sound/soc/qcom/qdsp6/topology.c -- 2.21.0

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