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[PATCH] ASoC: fsl_spdif: keep all TxClk sources by txclk array
by Shengjiu Wang 10 Mar '22

10 Mar '22

From: Viorel Suman <viorel.suman(a)nxp.com> Use txclk array to keep all TxClk sources instead of keeping clocks per rate - need to do this in order to avoid multiple prepare_enable/disable_unprepare of the same clock during suspend/resume. Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_spdif.c | 41 +++++++++++++++++++-------------------- 1 file changed, 20 insertions(+), 21 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index 57c41b2f7d17..e0acce6b2213 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -125,7 +125,7 @@ struct fsl_spdif_priv { u16 sysclk_df[SPDIF_TXRATE_MAX]; u8 txclk_src[SPDIF_TXRATE_MAX]; u8 rxclk_src; - struct clk *txclk[SPDIF_TXRATE_MAX]; + struct clk *txclk[STC_TXCLK_SRC_MAX]; struct clk *rxclk; struct clk *coreclk; struct clk *sysclk; @@ -526,7 +526,7 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, goto clk_set_bypass; /* The S/PDIF block needs a clock of 64 * fs * txclk_df */ - ret = clk_set_rate(spdif_priv->txclk[rate], + ret = clk_set_rate(spdif_priv->txclk[clk], 64 * sample_rate * txclk_df); if (ret) { dev_err(&pdev->dev, "failed to set tx clock rate\n"); @@ -537,7 +537,7 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, dev_dbg(&pdev->dev, "expected clock rate = %d\n", (64 * sample_rate * txclk_df * sysclk_df)); dev_dbg(&pdev->dev, "actual clock rate = %ld\n", - clk_get_rate(spdif_priv->txclk[rate])); + clk_get_rate(spdif_priv->txclk[clk])); /* set fs field in consumer channel status */ spdif_set_cstatus(ctrl, IEC958_AES3_CON_FS, csfs); @@ -1376,12 +1376,10 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, struct device *dev = &pdev->dev; u64 savesub = 100000, ret; struct clk *clk; - char tmp[16]; int i; for (i = 0; i < STC_TXCLK_SRC_MAX; i++) { - sprintf(tmp, "rxtx%d", i); - clk = devm_clk_get(dev, tmp); + clk = spdif_priv->txclk[i]; if (IS_ERR(clk)) { dev_err(dev, "no rxtx%d clock in devicetree\n", i); return PTR_ERR(clk); @@ -1395,7 +1393,6 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, continue; savesub = ret; - spdif_priv->txclk[index] = clk; spdif_priv->txclk_src[index] = i; /* To quick catch a divisor, we allow a 0.1% deviation */ @@ -1407,7 +1404,7 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, spdif_priv->txclk_src[index], rate[index]); dev_dbg(dev, "use txclk df %d for %dHz sample rate\n", spdif_priv->txclk_df[index], rate[index]); - if (clk_is_match(spdif_priv->txclk[index], spdif_priv->sysclk)) + if (clk_is_match(spdif_priv->txclk[spdif_priv->txclk_src[index]], spdif_priv->sysclk)) dev_dbg(dev, "use sysclk df %d for %dHz sample rate\n", spdif_priv->sysclk_df[index], rate[index]); dev_dbg(dev, "the best rate for %dHz sample rate is %dHz\n", @@ -1423,6 +1420,7 @@ static int fsl_spdif_probe(struct platform_device *pdev) struct resource *res; void __iomem *regs; int irq, ret, i; + char tmp[16]; spdif_priv = devm_kzalloc(&pdev->dev, sizeof(*spdif_priv), GFP_KERNEL); if (!spdif_priv) @@ -1462,8 +1460,17 @@ static int fsl_spdif_probe(struct platform_device *pdev) } } + for (i = 0; i < STC_TXCLK_SRC_MAX; i++) { + sprintf(tmp, "rxtx%d", i); + spdif_priv->txclk[i] = devm_clk_get(&pdev->dev, tmp); + if (IS_ERR(spdif_priv->txclk[i])) { + dev_err(&pdev->dev, "no rxtx%d clock in devicetree\n", i); + return PTR_ERR(spdif_priv->txclk[i]); + } + } + /* Get system clock for rx clock rate calculation */ - spdif_priv->sysclk = devm_clk_get(&pdev->dev, "rxtx5"); + spdif_priv->sysclk = spdif_priv->txclk[5]; if (IS_ERR(spdif_priv->sysclk)) { dev_err(&pdev->dev, "no sys clock (rxtx5) in devicetree\n"); return PTR_ERR(spdif_priv->sysclk); @@ -1481,7 +1488,7 @@ static int fsl_spdif_probe(struct platform_device *pdev) dev_warn(&pdev->dev, "no spba clock in devicetree\n"); /* Select clock source for rx/tx clock */ - spdif_priv->rxclk = devm_clk_get(&pdev->dev, "rxtx1"); + spdif_priv->rxclk = spdif_priv->txclk[1]; if (IS_ERR(spdif_priv->rxclk)) { dev_err(&pdev->dev, "no rxtx1 clock in devicetree\n"); return PTR_ERR(spdif_priv->rxclk); @@ -1562,9 +1569,7 @@ static int fsl_spdif_runtime_suspend(struct device *dev) &spdif_priv->regcache_srpc); regcache_cache_only(spdif_priv->regmap, true); - clk_disable_unprepare(spdif_priv->rxclk); - - for (i = 0; i < SPDIF_TXRATE_MAX; i++) + for (i = 0; i < STC_TXCLK_SRC_MAX; i++) clk_disable_unprepare(spdif_priv->txclk[i]); if (!IS_ERR(spdif_priv->spbaclk)) @@ -1594,16 +1599,12 @@ static int fsl_spdif_runtime_resume(struct device *dev) } } - for (i = 0; i < SPDIF_TXRATE_MAX; i++) { + for (i = 0; i < STC_TXCLK_SRC_MAX; i++) { ret = clk_prepare_enable(spdif_priv->txclk[i]); if (ret) goto disable_tx_clk; } - ret = clk_prepare_enable(spdif_priv->rxclk); - if (ret) - goto disable_tx_clk; - regcache_cache_only(spdif_priv->regmap, false); regcache_mark_dirty(spdif_priv->regmap); @@ -1613,12 +1614,10 @@ static int fsl_spdif_runtime_resume(struct device *dev) ret = regcache_sync(spdif_priv->regmap); if (ret) - goto disable_rx_clk; + goto disable_tx_clk; return 0; -disable_rx_clk: - clk_disable_unprepare(spdif_priv->rxclk); disable_tx_clk: for (i--; i >= 0; i--) clk_disable_unprepare(spdif_priv->txclk[i]); -- 2.17.1

2 1

[PATCH] ASoC: fsl_rpmsg: Remove SET_SYSTEM_SLEEP_PM_OPS callback
by Shengjiu Wang 10 Mar '22

10 Mar '22

For sound need to be continuously output at suspend with rpmsg sound card, so need to keep the clock always on at suspend, then suspend & resume callback is not needed. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_rpmsg.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/fsl/fsl_rpmsg.c b/sound/soc/fsl/fsl_rpmsg.c index 8508bc7f239d..19fd31250883 100644 --- a/sound/soc/fsl/fsl_rpmsg.c +++ b/sound/soc/fsl/fsl_rpmsg.c @@ -297,8 +297,6 @@ static const struct dev_pm_ops fsl_rpmsg_pm_ops = { SET_RUNTIME_PM_OPS(fsl_rpmsg_runtime_suspend, fsl_rpmsg_runtime_resume, NULL) - SET_SYSTEM_SLEEP_PM_OPS(pm_runtime_force_suspend, - pm_runtime_force_resume) }; static struct platform_driver fsl_rpmsg_driver = { -- 2.17.1

2 1

[PATCH] ASoC: qcom: fix Kconfig for SC7280
by Srinivas Kandagatla 10 Mar '22

10 Mar '22

select would force the symbol to value without checking the dependencies. In this case selecting TX and RX MACROs directly without checking its dependency on COMMON_CLK would break builds on platform which do no set COMMON_CLK. ex: WARNING: unmet direct dependencies detected for SND_SOC_LPASS_RX_MACRO Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && COMMON_CLK [=n] Selected by [m]: - SND_SOC_SC7280 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_SOC_QCOM [=m] && I2C [=y] && SOUNDWIRE [=m] move select to imply which should enforce symbol to be set to 'n' if any dependencies are not resolved. Fixes: 57350bd41c3a ("ASoC: qcom: SC7280: Add machine driver") Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index ae99833b3fdb..593fd85e03ad 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -194,8 +194,8 @@ config SND_SOC_SC7280 select SND_SOC_LPASS_SC7280 select SND_SOC_MAX98357A select SND_SOC_WCD938X - select SND_SOC_LPASS_RX_MACRO - select SND_SOC_LPASS_TX_MACRO + imply SND_SOC_LPASS_RX_MACRO + imply SND_SOC_LPASS_TX_MACRO help Add support for audio on Qualcomm Technologies Inc. SC7280 SoC-based systems. -- 2.21.0

2 2

[PATCH] ASoC: SOF: ipc-msg-injector: Use devm_kzalloc() for the rx_buffer
by Peter Ujfalusi 10 Mar '22

10 Mar '22

The rx_buffer is cleared before sending an IPC to make sure that when the /sys/kernel/debug/sof/ipc_msg_inject file is read we will have correct information in the buffer (no random or stale data). But if the user reads the file before sending any message the buffer might contain garbage which should not be interpreted. To prevent this, clear the rx_buffer on allocation. Fixes: cac0b0887e53 ("ASoC: SOF: Convert the generic IPC message injector into SOF client") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Reviewed-by: Bard Liao <yung-chuan.liao(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> --- sound/soc/sof/sof-client-ipc-msg-injector.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sof/sof-client-ipc-msg-injector.c b/sound/soc/sof/sof-client-ipc-msg-injector.c index bce103da4c49..dba6cfd7db09 100644 --- a/sound/soc/sof/sof-client-ipc-msg-injector.c +++ b/sound/soc/sof/sof-client-ipc-msg-injector.c @@ -137,7 +137,7 @@ static int sof_msg_inject_probe(struct auxiliary_device *auxdev, return -ENOMEM; priv->tx_buffer = devm_kmalloc(dev, SOF_IPC_MSG_MAX_SIZE, GFP_KERNEL); - priv->rx_buffer = devm_kmalloc(dev, SOF_IPC_MSG_MAX_SIZE, GFP_KERNEL); + priv->rx_buffer = devm_kzalloc(dev, SOF_IPC_MSG_MAX_SIZE, GFP_KERNEL); if (!priv->tx_buffer || !priv->rx_buffer) return -ENOMEM; -- 2.35.1

2 1

[PATCH v2 00/18] Clean ups and preparation for IPC abstraction in the SOF driver
by Ranjani Sridharan 10 Mar '22

10 Mar '22

In preparation for adding support for the new IPC version that has been introduced in the SOF firmware, this patch set includes some clean ups and necessary modifications to commonly used functions that will be re-used across different IPC-specific code. Changes in v2: - Use guid_t instead of u8 array for uuid and the helper guid_is_null() Pierre-Louis Bossart (1): ASoC: SOF: sof-audio: removed unused function Rander Wang (1): ASoC: SOF: move definition of snd_sof_ipc to header file Ranjani Sridharan (16): ASoC: SOF: remove snd_sof_pipeline_find() ASoC: SOF: simplify snd_sof_device_remove() ASoC: SOF: set swidget's core for scheduler widget ASoC: SOF: topology: remove redundant code ASoC: SOF: topology: remove redundant code in sof_link_afe_load() ASoC: SOF: topology: Drop the size parameter from struct sof_topology_token ASoC: SOF: topology: Modify the get_token op for string tokens ASoC: SOF: topology: expose some get_token ops ASoC: SOF: change comp_dai to a pointer in struct snd_sof_dai ASoC: SOF: make struct snd_sof_widget IPC agnostic ASoC: SOF: topology: make sof_route_load() IPC agnostic ASoC: SOF: Add a tuples array to struct snd_sof_widget ASoC: SOF: topology: Modify signature for token parsing functions ASoC: SOF: topology: Rename arguments in sof_parse_token_sets() ASoC: SOF: topology: Rename arguments in sof_parse_tokens() ASoC: SOF: make struct snd_sof_dai IPC agnostic include/sound/sof/dai.h | 5 + include/sound/sof/topology.h | 8 - sound/soc/sof/core.c | 10 +- sound/soc/sof/intel/hda-dai.c | 27 ++- sound/soc/sof/intel/hda.c | 39 +++- sound/soc/sof/ipc.c | 12 -- sound/soc/sof/pcm.c | 77 +++---- sound/soc/sof/sof-audio.c | 108 +++------- sound/soc/sof/sof-audio.h | 30 ++- sound/soc/sof/sof-priv.h | 12 ++ sound/soc/sof/topology.c | 392 +++++++++++++++++----------------- 11 files changed, 354 insertions(+), 366 deletions(-) -- 2.25.1

2 22

[PATCH] ASoC: Intel: boards: fix randconfig issue
by Pierre-Louis Bossart 10 Mar '22

10 Mar '22

on x86_64: ERROR: modpost: "sof_dai_get_bclk" [sound/soc/intel/boards/snd-soc-intel-sof-cirrus-common.ko] undefined! ERROR: modpost: "sof_dai_get_mclk" [sound/soc/intel/boards/snd-soc-intel-sof-realtek-common.ko] undefined! This comes from a missing dependency on at least ONE SOF platform being selected. This dependency exists for all other machine drivers, this was missed in the earlier reviews. Fixes: 2fe14ff61bd6 ("ASoC: Intel: sof_ssp_amp: rename driver and support cs35l41 amplifier") Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/Kconfig | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index a62785893bec..f3873b5bea87 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -616,6 +616,8 @@ config SND_SOC_INTEL_SOF_DA7219_MAX98373_MACH endif ## SND_SOC_SOF_JASPERLAKE +if SND_SOC_SOF_HDA_LINK + config SND_SOC_INTEL_SOF_SSP_AMP_MACH tristate "SOF with amplifiers in I2S Mode" depends on I2C && ACPI @@ -632,6 +634,7 @@ config SND_SOC_INTEL_SOF_SSP_AMP_MACH with RT1308/CS35L41 I2S audio codec. Say Y if you have such a device. If unsure select "N". +endif ## SND_SOC_SOF_HDA_LINK if SND_SOC_SOF_ELKHARTLAKE -- 2.30.2

4 3

[PATCH] ASoC: wm_adsp: Compressed stream DSP memory structs should be __packed
by Simon Trimmer 10 Mar '22

10 Mar '22

The compressed stream code has a bunch of structs that are used to represent DSP memory but have not been marked __packed. This isn't safe, they could get padded on a 64-bit build. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 0582585236a2..5a57bb04a0ae 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -97,13 +97,13 @@ struct wm_adsp_system_config_xm_hdr { __be32 wdma[8]; __be32 build_job_name[3]; __be32 build_job_number; -}; +} __packed; struct wm_halo_system_config_xm_hdr { __be32 halo_heartbeat; __be32 build_job_name[3]; __be32 build_job_number; -}; +} __packed; struct wm_adsp_alg_xm_struct { __be32 magic; @@ -114,13 +114,13 @@ struct wm_adsp_alg_xm_struct { __be32 high_water_mark; __be32 low_water_mark; __be64 smoothed_power; -}; +} __packed; struct wm_adsp_host_buf_coeff_v1 { __be32 host_buf_ptr; /* Host buffer pointer */ __be32 versions; /* Version numbers */ __be32 name[4]; /* The buffer name */ -}; +} __packed; struct wm_adsp_buffer { __be32 buf1_base; /* Base addr of first buffer area */ @@ -141,7 +141,7 @@ struct wm_adsp_buffer { __be32 min_free; /* min free space since stream start */ __be32 blocks_written[2]; /* total blocks written (64 bit) */ __be32 words_written[2]; /* total words written (64 bit) */ -}; +} __packed; struct wm_adsp_compr; -- 2.34.1

3 2

RE: [PATCH alsa-lib v2 3/3] pcm: direct: Check xrun/suspend before the slave hwptr update
by S.J. Wang 10 Mar '22

10 Mar '22

Hi > > The xrun/suspend may happen at any time and we should check it right after > the slave hwptr update (but before the actual sync_ptr update in direct pcm > side). Otherwise the hwptr value may be screwed and get unexpected large > read/write. > > Reported-by: S.J. Wang <shengjiu.wang(a)nxp.com> > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Acked-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> > --- > src/pcm/pcm_dmix.c | 8 +++++--- > src/pcm/pcm_dshare.c | 8 +++++--- > src/pcm/pcm_dsnoop.c | 6 +++--- > 3 files changed, 13 insertions(+), 9 deletions(-) > > diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index > d00d53bef604..c6cb47f0f840 100644 > --- a/src/pcm/pcm_dmix.c > +++ b/src/pcm/pcm_dmix.c > @@ -424,15 +424,17 @@ static int snd_pcm_dmix_sync_ptr0(snd_pcm_t > *pcm, snd_pcm_uframes_t slave_hw_ptr static int > snd_pcm_dmix_sync_ptr(snd_pcm_t *pcm) { > snd_pcm_direct_t *dmix = pcm->private_data; > + snd_pcm_uframes_t slave_hw_ptr; > int err; > > + if (dmix->slowptr) > + snd_pcm_hwsync(dmix->spcm); > + slave_hw_ptr = *dmix->spcm->hw.ptr; > err = snd_pcm_direct_check_xrun(dmix, pcm); > if (err < 0) > return err; > - if (dmix->slowptr) > - snd_pcm_hwsync(dmix->spcm); > > - return snd_pcm_dmix_sync_ptr0(pcm, *dmix->spcm->hw.ptr); > + return snd_pcm_dmix_sync_ptr0(pcm, slave_hw_ptr); > } > > /* > diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index > 0ff43a90d270..461adafc77f8 100644 > --- a/src/pcm/pcm_dshare.c > +++ b/src/pcm/pcm_dshare.c > @@ -199,15 +199,17 @@ static int snd_pcm_dshare_sync_ptr0(snd_pcm_t > *pcm, snd_pcm_uframes_t slave_hw_p static int > snd_pcm_dshare_sync_ptr(snd_pcm_t *pcm) { > snd_pcm_direct_t *dshare = pcm->private_data; > + snd_pcm_uframes_t slave_hw_ptr; > int err; > > + if (dshare->slowptr) > + snd_pcm_hwsync(dshare->spcm); > + slave_hw_ptr = *dshare->spcm->hw.ptr; > err = snd_pcm_direct_check_xrun(dshare, pcm); > if (err < 0) > return err; > - if (dshare->slowptr) > - snd_pcm_hwsync(dshare->spcm); > > - return snd_pcm_dshare_sync_ptr0(pcm, *dshare->spcm->hw.ptr); > + return snd_pcm_dshare_sync_ptr0(pcm, slave_hw_ptr); > } > > /* > diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index > 729ff447b41f..9abbbef2c1b6 100644 > --- a/src/pcm/pcm_dsnoop.c > +++ b/src/pcm/pcm_dsnoop.c > @@ -134,14 +134,14 @@ static int snd_pcm_dsnoop_sync_ptr(snd_pcm_t > *pcm) > snd_pcm_sframes_t diff; > int err; > > - err = snd_pcm_direct_check_xrun(dsnoop, pcm); > - if (err < 0) > - return err; > if (dsnoop->slowptr) > snd_pcm_hwsync(dsnoop->spcm); > old_slave_hw_ptr = dsnoop->slave_hw_ptr; > snoop_timestamp(pcm); > slave_hw_ptr = dsnoop->slave_hw_ptr; > + err = snd_pcm_direct_check_xrun(dsnoop, pcm); > + if (err < 0) > + return err; > diff = pcm_frame_diff(slave_hw_ptr, old_slave_hw_ptr, dsnoop- > >slave_boundary); > if (diff == 0) /* fast path */ > return 0; > -- > 2.34.1

1 0

[PATCH alsa-lib v2 1/3] pcm: direct: Improved suspend/resume support
by Takashi Iwai 10 Mar '22

10 Mar '22

The current resume handling in PCM direct plugins don't treat multiple clients properly: once after the slave PCM gets resumed by one client, the access from others at a later point is seen as already running although the internal state isn't updated and becomes inconsistent. This may end up a negative size, which eventually hangs up. This patch is an attempt to improve the handling for resume. Now the suspended state is treated similarly like XRUN; namely, we keep the slave PCM "recoveries" count that is modified at each time the slave PCM XRUN happens, so that we can check the inconsistency against the client's state. As a differentiation to XRUN, we set the highest bit of recoveries count when the slave stream hits SUSPENDED state. This bit is referred at comparing with clients, and the client's state is updated to either XRUN or SUSPENDED depending on this bit. Along with this change, the actual resume is done in snd_pcm_direct_slave_recover(), and snd_pcm_direct_resume() rather calls this internally. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.c | 72 ++++++++++++++++++++++++++------------------ src/pcm/pcm_dmix.c | 6 ++-- src/pcm/pcm_dshare.c | 6 ++-- src/pcm/pcm_dsnoop.c | 6 ++-- 4 files changed, 52 insertions(+), 38 deletions(-) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 193dc3e76d49..1ca3b76306b1 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -560,8 +560,11 @@ int snd_pcm_direct_timer_stop(snd_pcm_direct_t *dmix) return 0; } +#define RECOVERIES_FLAG_SUSPENDED (1U << 31) +#define RECOVERIES_MASK ((1U << 31) - 1) + /* - * Recover slave on XRUN. + * Recover slave on XRUN or SUSPENDED. * Even if direct plugins disable xrun detection, there might be an xrun * raised directly by some drivers. * The first client recovers slave pcm. @@ -569,6 +572,8 @@ int snd_pcm_direct_timer_stop(snd_pcm_direct_t *dmix) */ int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) { + unsigned int recoveries; + int state; int ret; int semerr; @@ -579,7 +584,8 @@ int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) return semerr; } - if (snd_pcm_state(direct->spcm) != SND_PCM_STATE_XRUN) { + state = snd_pcm_state(direct->spcm); + if (state != SND_PCM_STATE_XRUN && state != SND_PCM_STATE_SUSPENDED) { /* ignore... someone else already did recovery */ semerr = snd_pcm_direct_semaphore_up(direct, DIRECT_IPC_SEM_CLIENT); @@ -590,6 +596,24 @@ int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) return 0; } + recoveries = direct->shmptr->s.recoveries; + recoveries = (recoveries + 1) & RECOVERIES_MASK; + if (state == SND_PCM_STATE_SUSPENDED) + recoveries |= RECOVERIES_FLAG_SUSPENDED; + direct->shmptr->s.recoveries = recoveries; + + /* some buggy drivers require the device resumed before prepared; + * when a device has RESUME flag and is in SUSPENDED state, resume + * here but immediately drop to bring it to a sane active state. + */ + if (state == SND_PCM_STATE_SUSPENDED && + (direct->spcm->info & SND_PCM_INFO_RESUME)) { + snd_pcm_resume(direct->spcm); + snd_pcm_drop(direct->spcm); + snd_pcm_direct_timer_stop(direct); + snd_pcm_direct_clear_timer_queue(direct); + } + ret = snd_pcm_prepare(direct->spcm); if (ret < 0) { SNDERR("recover: unable to prepare slave"); @@ -621,7 +645,6 @@ int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) } return ret; } - direct->shmptr->s.recoveries++; semerr = snd_pcm_direct_semaphore_up(direct, DIRECT_IPC_SEM_CLIENT); if (semerr < 0) { @@ -632,11 +655,15 @@ int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) } /* - * enter xrun state, if slave xrun occurred - * @return: 0 for no xrun or a negative error code for xrun + * enter xrun or suspended state, if slave xrun occurred or suspended + * @return: 0 for no xrun/suspend or a negative error code for xrun/suspend */ int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm) { + if (direct->state == SND_PCM_STATE_XRUN) + return -EPIPE; + else if (direct->state == SND_PCM_STATE_SUSPENDED) + return -ESTRPIPE; if (direct->shmptr->s.recoveries != direct->recoveries) { /* no matter how many xruns we missed - * so don't increment but just update to actual counter @@ -649,10 +676,14 @@ int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm) * if slave already entered xrun again the event is lost. * snd_pcm_direct_clear_timer_queue(direct); */ - direct->state = SND_PCM_STATE_XRUN; + if (direct->recoveries & RECOVERIES_FLAG_SUSPENDED) { + direct->state = SND_PCM_STATE_SUSPENDED; + return -ESTRPIPE; + } else { + direct->state = SND_PCM_STATE_XRUN; + return -EPIPE; + } } - if (direct->state == SND_PCM_STATE_XRUN) - return -EPIPE; return 0; } @@ -721,13 +752,13 @@ timer_changed: } switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: /* recover slave and update client state to xrun * before returning POLLERR */ snd_pcm_direct_slave_recover(dmix); snd_pcm_direct_client_chk_xrun(dmix, pcm); /* fallthrough */ - case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_SETUP: events |= POLLERR; break; @@ -1074,27 +1105,10 @@ int snd_pcm_direct_prepare(snd_pcm_t *pcm) int snd_pcm_direct_resume(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; - snd_pcm_t *spcm = dmix->spcm; + int err; - snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT); - /* some buggy drivers require the device resumed before prepared; - * when a device has RESUME flag and is in SUSPENDED state, resume - * here but immediately drop to bring it to a sane active state. - */ - if ((spcm->info & SND_PCM_INFO_RESUME) && - snd_pcm_state(spcm) == SND_PCM_STATE_SUSPENDED) { - snd_pcm_resume(spcm); - snd_pcm_drop(spcm); - snd_pcm_direct_timer_stop(dmix); - snd_pcm_direct_clear_timer_queue(dmix); - snd_pcm_areas_silence(snd_pcm_mmap_areas(spcm), 0, - spcm->channels, spcm->buffer_size, - spcm->format); - snd_pcm_prepare(spcm); - snd_pcm_start(spcm); - } - snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT); - return -ENOSYS; + err = snd_pcm_direct_slave_recover(dmix); + return err < 0 ? err : -ENOSYS; } #define COPY_SLAVE(field) (dmix->shmptr->s.field = spcm->field) diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index dfff18b992c5..e3bf49269f0c 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -431,6 +431,7 @@ static int snd_pcm_dmix_sync_ptr(snd_pcm_t *pcm) dmix->state = SND_PCM_STATE_DISCONNECTED; return -ENODEV; case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: if ((err = snd_pcm_direct_slave_recover(dmix)) < 0) return err; break; @@ -457,11 +458,11 @@ static snd_pcm_state_t snd_pcm_dmix_state(snd_pcm_t *pcm) snd_pcm_state_t state; state = snd_pcm_state(dmix->spcm); switch (state) { - case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dmix->state = state; return state; case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: if ((err = snd_pcm_direct_slave_recover(dmix)) < 0) return err; break; @@ -833,11 +834,10 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: if ((err = snd_pcm_direct_slave_recover(dmix)) < 0) return err; break; - case SND_PCM_STATE_SUSPENDED: - return -ESTRPIPE; default: break; } diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index d63c86bd4044..5a52ae9173be 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -206,6 +206,7 @@ static int snd_pcm_dshare_sync_ptr(snd_pcm_t *pcm) dshare->state = SNDRV_PCM_STATE_DISCONNECTED; return -ENODEV; case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: if ((err = snd_pcm_direct_slave_recover(dshare)) < 0) return err; break; @@ -260,11 +261,11 @@ static snd_pcm_state_t snd_pcm_dshare_state(snd_pcm_t *pcm) snd_pcm_state_t state; state = snd_pcm_state(dshare->spcm); switch (state) { - case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dshare->state = state; return state; case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: if ((err = snd_pcm_direct_slave_recover(dshare)) < 0) return err; break; @@ -532,11 +533,10 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dshare->spcm)) { case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: if ((err = snd_pcm_direct_slave_recover(dshare)) < 0) return err; break; - case SND_PCM_STATE_SUSPENDED: - return -ESTRPIPE; default: break; } diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 77ffbada931f..1653f6fba86c 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -139,6 +139,7 @@ static int snd_pcm_dsnoop_sync_ptr(snd_pcm_t *pcm) dsnoop->state = SNDRV_PCM_STATE_DISCONNECTED; return -ENODEV; case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: if ((err = snd_pcm_direct_slave_recover(dsnoop)) < 0) return err; break; @@ -211,11 +212,11 @@ static snd_pcm_state_t snd_pcm_dsnoop_state(snd_pcm_t *pcm) snd_pcm_state_t state; state = snd_pcm_state(dsnoop->spcm); switch (state) { - case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dsnoop->state = state; return state; case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: if ((err = snd_pcm_direct_slave_recover(dsnoop)) < 0) return err; break; @@ -423,11 +424,10 @@ static snd_pcm_sframes_t snd_pcm_dsnoop_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dsnoop->spcm)) { case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: if ((err = snd_pcm_direct_slave_recover(dsnoop)) < 0) return err; break; - case SND_PCM_STATE_SUSPENDED: - return -ESTRPIPE; default: break; } -- 2.34.1

1 2

RE: [PATCH alsa-lib 3/3] pcm: direct: Check xrun/suspend before the slave hwptr update
by S.J. Wang 10 Mar '22

10 Mar '22

Thanks. > > The xrun/suspend may happen at any time and we should check it right > before the slave hwptr update. Otherwise the hwptr value may be screwed I think should be "after the slave hwptr update". If hwptr is screwed, means Suspend happened then check_xrun() can return. > and get unexpected large read/write. > > Reported-by: S.J. Wang <shengjiu.wang(a)nxp.com> > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > --- > src/pcm/pcm_dmix.c | 4 ++-- > src/pcm/pcm_dshare.c | 4 ++-- > src/pcm/pcm_dsnoop.c | 6 +++--- > 3 files changed, 7 insertions(+), 7 deletions(-) > > diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index > d00d53bef604..111fea157228 100644 > --- a/src/pcm/pcm_dmix.c > +++ b/src/pcm/pcm_dmix.c > @@ -426,11 +426,11 @@ static int snd_pcm_dmix_sync_ptr(snd_pcm_t > *pcm) > snd_pcm_direct_t *dmix = pcm->private_data; > int err; > > + if (dmix->slowptr) > + snd_pcm_hwsync(dmix->spcm); > err = snd_pcm_direct_check_xrun(dmix, pcm); > if (err < 0) > return err; > - if (dmix->slowptr) > - snd_pcm_hwsync(dmix->spcm); > > return snd_pcm_dmix_sync_ptr0(pcm, *dmix->spcm->hw.ptr); } diff -- Better to get slave_hw_ptr before check_xrun(), like this: --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -424,15 +424,17 @@ static int snd_pcm_dmix_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_ptr static int snd_pcm_dmix_sync_ptr(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; + snd_pcm_uframes_t slave_hw_ptr; int err; if (dmix->slowptr) snd_pcm_hwsync(dmix->spcm); + slave_hw_ptr = *dmix->spcm->hw.ptr; err = snd_pcm_direct_check_xrun(dmix, pcm); if (err < 0) return err; - return snd_pcm_dmix_sync_ptr0(pcm, *dmix->spcm->hw.ptr); + return snd_pcm_dmix_sync_ptr0(pcm, slave_hw_ptr); } > git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index > 0ff43a90d270..c46e8d6a53da 100644 > --- a/src/pcm/pcm_dshare.c > +++ b/src/pcm/pcm_dshare.c > @@ -201,11 +201,11 @@ static int snd_pcm_dshare_sync_ptr(snd_pcm_t > *pcm) > snd_pcm_direct_t *dshare = pcm->private_data; > int err; > > + if (dshare->slowptr) > + snd_pcm_hwsync(dshare->spcm); > err = snd_pcm_direct_check_xrun(dshare, pcm); > if (err < 0) > return err; > - if (dshare->slowptr) > - snd_pcm_hwsync(dshare->spcm); > > return snd_pcm_dshare_sync_ptr0(pcm, *dshare->spcm->hw.ptr); } diff --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -199,15 +199,17 @@ static int snd_pcm_dshare_sync_ptr0(snd_pcm_t *pcm, snd_pcm_uframes_t slave_hw_p static int snd_pcm_dshare_sync_ptr(snd_pcm_t *pcm) { snd_pcm_direct_t *dshare = pcm->private_data; + snd_pcm_uframes_t slave_hw_ptr; int err; if (dshare->slowptr) snd_pcm_hwsync(dshare->spcm); + slave_hw_ptr = *dshare->spcm->hw.ptr; err = snd_pcm_direct_check_xrun(dshare, pcm); if (err < 0) return err; - return snd_pcm_dshare_sync_ptr0(pcm, *dshare->spcm->hw.ptr); + return snd_pcm_dshare_sync_ptr0(pcm, slave_hw_ptr); > --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index > 729ff447b41f..9abbbef2c1b6 100644 > --- a/src/pcm/pcm_dsnoop.c > +++ b/src/pcm/pcm_dsnoop.c > @@ -134,14 +134,14 @@ static int snd_pcm_dsnoop_sync_ptr(snd_pcm_t > *pcm) > snd_pcm_sframes_t diff; > int err; > > - err = snd_pcm_direct_check_xrun(dsnoop, pcm); > - if (err < 0) > - return err; > if (dsnoop->slowptr) > snd_pcm_hwsync(dsnoop->spcm); > old_slave_hw_ptr = dsnoop->slave_hw_ptr; > snoop_timestamp(pcm); > slave_hw_ptr = dsnoop->slave_hw_ptr; > + err = snd_pcm_direct_check_xrun(dsnoop, pcm); > + if (err < 0) > + return err; > diff = pcm_frame_diff(slave_hw_ptr, old_slave_hw_ptr, dsnoop- > >slave_boundary); > if (diff == 0) /* fast path */ > return 0; > -- > 2.34.1

2 1

[PATCH alsa-lib 1/3] pcm: direct: Improved suspend/resume support
by Takashi Iwai 10 Mar '22

10 Mar '22

The current resume handling in PCM direct plugins don't treat multiple clients properly: once after the slave PCM gets resumed by one client, the access from others at a later point is seen as already running although the internal state isn't updated and becomes inconsistent. This may end up a negative size, which eventually hangs up. This patch is an attempt to improve the handling for resume. Now the suspended state is treated similarly like XRUN; namely, we keep the slave PCM "recoveries" count that is modified at each time the slave PCM XRUN happens, so that we can check the inconsistency against the client's state. As a differentiation to XRUN, we set the highest bit of recoveries count when the slave stream hits SUSPENDED state. This bit is referred at comparing with clients, and the client's state is updated to either XRUN or SUSPENDED depending on this bit. Along with this change, the actual resume is done in snd_pcm_direct_slave_recover(), and snd_pcm_direct_resume() rather calls this internally. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/pcm/pcm_direct.c | 72 ++++++++++++++++++++++++++------------------ src/pcm/pcm_dmix.c | 6 ++-- src/pcm/pcm_dshare.c | 6 ++-- src/pcm/pcm_dsnoop.c | 6 ++-- 4 files changed, 52 insertions(+), 38 deletions(-) diff --git a/src/pcm/pcm_direct.c b/src/pcm/pcm_direct.c index 193dc3e76d49..1ca3b76306b1 100644 --- a/src/pcm/pcm_direct.c +++ b/src/pcm/pcm_direct.c @@ -560,8 +560,11 @@ int snd_pcm_direct_timer_stop(snd_pcm_direct_t *dmix) return 0; } +#define RECOVERIES_FLAG_SUSPENDED (1U << 31) +#define RECOVERIES_MASK ((1U << 31) - 1) + /* - * Recover slave on XRUN. + * Recover slave on XRUN or SUSPENDED. * Even if direct plugins disable xrun detection, there might be an xrun * raised directly by some drivers. * The first client recovers slave pcm. @@ -569,6 +572,8 @@ int snd_pcm_direct_timer_stop(snd_pcm_direct_t *dmix) */ int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) { + unsigned int recoveries; + int state; int ret; int semerr; @@ -579,7 +584,8 @@ int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) return semerr; } - if (snd_pcm_state(direct->spcm) != SND_PCM_STATE_XRUN) { + state = snd_pcm_state(direct->spcm); + if (state != SND_PCM_STATE_XRUN && state != SND_PCM_STATE_SUSPENDED) { /* ignore... someone else already did recovery */ semerr = snd_pcm_direct_semaphore_up(direct, DIRECT_IPC_SEM_CLIENT); @@ -590,6 +596,24 @@ int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) return 0; } + recoveries = direct->shmptr->s.recoveries; + recoveries = (recoveries + 1) & RECOVERIES_MASK; + if (state == SND_PCM_STATE_SUSPENDED) + recoveries |= RECOVERIES_FLAG_SUSPENDED; + direct->shmptr->s.recoveries = recoveries; + + /* some buggy drivers require the device resumed before prepared; + * when a device has RESUME flag and is in SUSPENDED state, resume + * here but immediately drop to bring it to a sane active state. + */ + if (state == SND_PCM_STATE_SUSPENDED && + (direct->spcm->info & SND_PCM_INFO_RESUME)) { + snd_pcm_resume(direct->spcm); + snd_pcm_drop(direct->spcm); + snd_pcm_direct_timer_stop(direct); + snd_pcm_direct_clear_timer_queue(direct); + } + ret = snd_pcm_prepare(direct->spcm); if (ret < 0) { SNDERR("recover: unable to prepare slave"); @@ -621,7 +645,6 @@ int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) } return ret; } - direct->shmptr->s.recoveries++; semerr = snd_pcm_direct_semaphore_up(direct, DIRECT_IPC_SEM_CLIENT); if (semerr < 0) { @@ -632,11 +655,15 @@ int snd_pcm_direct_slave_recover(snd_pcm_direct_t *direct) } /* - * enter xrun state, if slave xrun occurred - * @return: 0 for no xrun or a negative error code for xrun + * enter xrun or suspended state, if slave xrun occurred or suspended + * @return: 0 for no xrun/suspend or a negative error code for xrun/suspend */ int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm) { + if (direct->state == SND_PCM_STATE_XRUN) + return -EPIPE; + else if (direct->state == SND_PCM_STATE_SUSPENDED) + return -ESTRPIPE; if (direct->shmptr->s.recoveries != direct->recoveries) { /* no matter how many xruns we missed - * so don't increment but just update to actual counter @@ -649,10 +676,14 @@ int snd_pcm_direct_client_chk_xrun(snd_pcm_direct_t *direct, snd_pcm_t *pcm) * if slave already entered xrun again the event is lost. * snd_pcm_direct_clear_timer_queue(direct); */ - direct->state = SND_PCM_STATE_XRUN; + if (direct->recoveries & RECOVERIES_FLAG_SUSPENDED) { + direct->state = SND_PCM_STATE_SUSPENDED; + return -ESTRPIPE; + } else { + direct->state = SND_PCM_STATE_XRUN; + return -EPIPE; + } } - if (direct->state == SND_PCM_STATE_XRUN) - return -EPIPE; return 0; } @@ -721,13 +752,13 @@ timer_changed: } switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: /* recover slave and update client state to xrun * before returning POLLERR */ snd_pcm_direct_slave_recover(dmix); snd_pcm_direct_client_chk_xrun(dmix, pcm); /* fallthrough */ - case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_SETUP: events |= POLLERR; break; @@ -1074,27 +1105,10 @@ int snd_pcm_direct_prepare(snd_pcm_t *pcm) int snd_pcm_direct_resume(snd_pcm_t *pcm) { snd_pcm_direct_t *dmix = pcm->private_data; - snd_pcm_t *spcm = dmix->spcm; + int err; - snd_pcm_direct_semaphore_down(dmix, DIRECT_IPC_SEM_CLIENT); - /* some buggy drivers require the device resumed before prepared; - * when a device has RESUME flag and is in SUSPENDED state, resume - * here but immediately drop to bring it to a sane active state. - */ - if ((spcm->info & SND_PCM_INFO_RESUME) && - snd_pcm_state(spcm) == SND_PCM_STATE_SUSPENDED) { - snd_pcm_resume(spcm); - snd_pcm_drop(spcm); - snd_pcm_direct_timer_stop(dmix); - snd_pcm_direct_clear_timer_queue(dmix); - snd_pcm_areas_silence(snd_pcm_mmap_areas(spcm), 0, - spcm->channels, spcm->buffer_size, - spcm->format); - snd_pcm_prepare(spcm); - snd_pcm_start(spcm); - } - snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT); - return -ENOSYS; + err = snd_pcm_direct_slave_recover(dmix); + return err < 0 ? err : -ENOSYS; } #define COPY_SLAVE(field) (dmix->shmptr->s.field = spcm->field) diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index dfff18b992c5..e3bf49269f0c 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -431,6 +431,7 @@ static int snd_pcm_dmix_sync_ptr(snd_pcm_t *pcm) dmix->state = SND_PCM_STATE_DISCONNECTED; return -ENODEV; case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: if ((err = snd_pcm_direct_slave_recover(dmix)) < 0) return err; break; @@ -457,11 +458,11 @@ static snd_pcm_state_t snd_pcm_dmix_state(snd_pcm_t *pcm) snd_pcm_state_t state; state = snd_pcm_state(dmix->spcm); switch (state) { - case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dmix->state = state; return state; case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: if ((err = snd_pcm_direct_slave_recover(dmix)) < 0) return err; break; @@ -833,11 +834,10 @@ static snd_pcm_sframes_t snd_pcm_dmix_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dmix->spcm)) { case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: if ((err = snd_pcm_direct_slave_recover(dmix)) < 0) return err; break; - case SND_PCM_STATE_SUSPENDED: - return -ESTRPIPE; default: break; } diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index d63c86bd4044..5a52ae9173be 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -206,6 +206,7 @@ static int snd_pcm_dshare_sync_ptr(snd_pcm_t *pcm) dshare->state = SNDRV_PCM_STATE_DISCONNECTED; return -ENODEV; case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: if ((err = snd_pcm_direct_slave_recover(dshare)) < 0) return err; break; @@ -260,11 +261,11 @@ static snd_pcm_state_t snd_pcm_dshare_state(snd_pcm_t *pcm) snd_pcm_state_t state; state = snd_pcm_state(dshare->spcm); switch (state) { - case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dshare->state = state; return state; case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: if ((err = snd_pcm_direct_slave_recover(dshare)) < 0) return err; break; @@ -532,11 +533,10 @@ static snd_pcm_sframes_t snd_pcm_dshare_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dshare->spcm)) { case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: if ((err = snd_pcm_direct_slave_recover(dshare)) < 0) return err; break; - case SND_PCM_STATE_SUSPENDED: - return -ESTRPIPE; default: break; } diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 77ffbada931f..1653f6fba86c 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -139,6 +139,7 @@ static int snd_pcm_dsnoop_sync_ptr(snd_pcm_t *pcm) dsnoop->state = SNDRV_PCM_STATE_DISCONNECTED; return -ENODEV; case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: if ((err = snd_pcm_direct_slave_recover(dsnoop)) < 0) return err; break; @@ -211,11 +212,11 @@ static snd_pcm_state_t snd_pcm_dsnoop_state(snd_pcm_t *pcm) snd_pcm_state_t state; state = snd_pcm_state(dsnoop->spcm); switch (state) { - case SND_PCM_STATE_SUSPENDED: case SND_PCM_STATE_DISCONNECTED: dsnoop->state = state; return state; case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: if ((err = snd_pcm_direct_slave_recover(dsnoop)) < 0) return err; break; @@ -423,11 +424,10 @@ static snd_pcm_sframes_t snd_pcm_dsnoop_mmap_commit(snd_pcm_t *pcm, switch (snd_pcm_state(dsnoop->spcm)) { case SND_PCM_STATE_XRUN: + case SND_PCM_STATE_SUSPENDED: if ((err = snd_pcm_direct_slave_recover(dsnoop)) < 0) return err; break; - case SND_PCM_STATE_SUSPENDED: - return -ESTRPIPE; default: break; } -- 2.34.1

1 2

Re: Issue in pcm_dsnoop.c in alsa-lib
by S.J. Wang 10 Mar '22

10 Mar '22

Hi > > > > > > > > > > Hi Takashi Iwai, Jaroslav Kysela > > > > > > > > We encountered an issue in the pcm_dsnoop use case, could you > > > > please help to have a look? > > > > > > > > *Issue description:* > > > > With two instances for dsnoop type device running in parallel, > > > > after suspend/resume, one of the instances will be hung in memcpy > > > > because the very large copy size is obtained. > > > > > > > > #3 0x0000ffffa78d5098 in snd_pcm_dsnoop_sync_ptr > > > (pcm=0xaaab06563da0) > > > > at pcm_dsnoop.c:158 dsnoop = 0xaaab06563c20 slave_hw_ptr = 64 > > > > old_slave_hw_ptr = 533120 avail = *187651522444320* > > > > > > > > * Reason analysis: * > > > > The root cause that I analysis is that after suspend/resume, > > > > one instance will get the SND_PCM_STATE_SUSPENDED state from slave > > > > pcm > > > device, > > > > then it will do snd_pcm_prepare() and snd_pcm_start(), which > > > > will reset the dsnoop->slave_hw_ptr and the hw_ptr of slave pcm > > > > device, then the state of this instance is correct. But another > > > > instance may not get the SND_PCM_STATE_SUSPENDED state from > slave > > > > pcm device because slave device may have been recovered by first > > > > instance, so the dsnoop->slave_hw_ptr is not reset. but because > > > > hw_ptr of slave pcm device has been reset, so there will be a very large > "avail" size. > > > > > > > > *Solution:* > > > > I didn't come up with a fix for this issue, seems there is no > > > > easy way to let another instance know this case and reset the > > > > dsnoop->slave_hw_ptr, could you please help? > > > > > > Could you try topic/pcm-direct-resume branch on > > > > > > https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgi > > > thub > > > .com%2Ftiwai%2Falsa- > > > > lib&data=04%7C01%7Cshengjiu.wang%40nxp.com%7C95f97de3f2c840d > > > > 9853508d9fd2e79ea%7C686ea1d3bc2b4c6fa92cd99c5c301635%7C0%7C0%7C > > > > 637819198319430045%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwM > > > > DAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000&sdat > > > > a=WWX1ZlcQhJF3pHJdHPIH%2B0xG9o%2FjQnHG5fHDbKXwQwE%3D&r > > > eserved=0 > > > > > > > Thanks, I push my test result in > > https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgith > > ub.com%2Falsa-project%2Falsa- > lib%2Fissues%2F213&data=04%7C01%7Cshe > > > ngjiu.wang%40nxp.com%7Cf71e70640d1b40b66be508d9fdbb2ac2%7C686ea > 1d3bc2b > > > 4c6fa92cd99c5c301635%7C0%7C0%7C637819802581943763%7CUnknown%7 > CTWFpbGZs > > > b3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn > 0%3D > > %7C3000&sdata=fZ2ogNj2RDTv4DV8vgB71M2m0XtU8UhMiXEV1%2Bl > wUrQ%3D&amp > > ;reserved=0 > > Could you please review? > > Please keep the discussion on ML. > I saw you have update the origin/topic/pcm-direct-resume branch, I test your latest change, it is more stable than before, but still meet once of the issue after overnight test, it it very very low possibility. So I suggest if we need to do below change, shall we? diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 729ff447b41f..cc333b3f4384 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -134,14 +134,21 @@ static int snd_pcm_dsnoop_sync_ptr(snd_pcm_t *pcm) snd_pcm_sframes_t diff; int err; - err = snd_pcm_direct_check_xrun(dsnoop, pcm); - if (err < 0) - return err; if (dsnoop->slowptr) snd_pcm_hwsync(dsnoop->spcm); old_slave_hw_ptr = dsnoop->slave_hw_ptr; snoop_timestamp(pcm); slave_hw_ptr = dsnoop->slave_hw_ptr; + /* + * FIXME: Move snd_pcm_direct_client_chk_xrun after getting the + * dsnoop->spcm->hw.ptr. If the snd_pcm_direct_slave_recover() + * of another instance happening before dsnoop->spcm->hw.ptr + * is got, then a wrong spcm->hw.ptr is got which cause a wrong + * 'diff' data later. + */ + err = snd_pcm_direct_check_xrun(dsnoop, pcm); + if (err < 0) + return err; diff = pcm_frame_diff(slave_hw_ptr, old_slave_hw_ptr, dsnoop->slave_boundary); best regards wang shengjiu

2 1

[PATCH] ASoC: atmel: mchp-pdmc: Remove unnecessary print function dev_err()
by Jiapeng Chong 10 Mar '22

10 Mar '22

The print function dev_err() is redundant because platform_get_irq() already prints an error. Eliminate the follow coccicheck warning: ./sound/soc/atmel/mchp-pdmc.c:991:2-9: line 991 is redundant because platform_get_irq() already prints an error. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/soc/atmel/mchp-pdmc.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/atmel/mchp-pdmc.c b/sound/soc/atmel/mchp-pdmc.c index c44636f6207d..2b48bd3434c7 100644 --- a/sound/soc/atmel/mchp-pdmc.c +++ b/sound/soc/atmel/mchp-pdmc.c @@ -988,7 +988,6 @@ static int mchp_pdmc_probe(struct platform_device *pdev) irq = platform_get_irq(pdev, 0); if (irq < 0) { - dev_err(dev, "failed to get irq: %d\n", irq); return irq; } -- 2.20.1.7.g153144c

2 1

[PATCH V0 0/1] ASoC: msm: fix integer overflow for long duration compress offload playback
by Raghu Bankapur 10 Mar '22

10 Mar '22

32 bit variable is used for storing number of bytes copied to DSP, which can overflow when playback duration goes beyond 24 hours. Change data type for this variable to uint64_t to prevent overflow and related playback anomaly. below are the steps used to reproduce Steps:- 1. play all clips from music app with Repeat All Songs enabled 2. one of clip is 1hr duration - 5.1_16bit_192khz_1hr.m4a 3. ACTUAL BEHAVIOUR:- there is no audio playback & progress bar was at end of stream, but time stamp continues to update till 31hrs for 1hr clip Raghu Bankapur (1): ASoC: msm: fix integer overflow for long duration offload playback include/uapi/sound/compress_offload.h | 2 +- sound/core/compress_offload.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) -- 2.17.1

4 5

[PATCH v3 00/17] ASoC: Intel: AVS - Audio DSP for cAVS
by Cezary Rojewski 09 Mar '22

09 Mar '22

A continuation of cleanup work of Intel SST solutions found in sound/soc/intel/. With two major chapters released last year catpt [1] and removal of haswell solution [2], time has come for Skylake-driver. Througout 2019, 2020 and 2021 Skylake-driver has had many fixes applied and even attempts of refactors as seen in fundamental overhaul [3], IPC flow adjustments [4] and LARGE_CONFIG overhaul [5] series. Unfortunately, story repeats itself - problems are found within the core of a driver. Painting it with different colors does not change the fact that is it still a house of cards. As changes needed to address those issues would make Skylake solution incompatible with its previous revisions, a decision has been made to provide a new solution instead. In time it would deprecate and replace Skylake-driver. That solution has been called AVS - from AudioDSP architecture name: Audio-Voice-Speech. It is meant to provide support for the exact same range of platforms as its predecessor: SKL, KBL, AML and APL. Note: this series is dependent upon HDA-series [6] which exposes several codec-organization functions allowing for reduced code size on avs-driver side. Note: this series does not add fully functional driver as its size would get out of control. Here, focus is put on adding IPC protocol and code loading code. Changes v2 -> v3: - fixed variable initialization errors mentioned by sparse Changes v1 -> v2: Almost all updates here are thanks to feedback from Pierre. - several comments and few kernel-docs have been added in areas which felt more or less unclear - avs_ipc_wait_busy_completion() now spins up to 'repeats_left' number of times before giving up - 'adsp_ba' field of struct avs_dev has been renamed to 'dsp_ba' - 'dops' field of struct avs_spec has been renamed to 'dsp_ops' - IPC abstraction has been simplified: SKL_ADSP_REG_HIP* regs are used directly - fixed allnoconfig with AVS enabled compilation issues - fixed code loading error paths: previously requested firmware is now released before function return to the caller - code and function arguments tied to D0IX support have been removed from this patchset and will be part of followup series adding that feature instead - enriched dev_err() messages in avs_dsp_get_core() and avs_dsp_put_core() - numerous wording fixes used in power/reset/stall DSP operations Changes RFC v1 [7]: -> v1: - separated HDA codec-organization patches, path and topology handling, PCM and complementary features such as recovery from this series to ease the review process - fixed EXPORT_SYMBOL_GPL for exported members of ASoC framework - result of stall() is now checked when sending ROM message - result of snd_hdac_ext_stream_set_spib() is now checked when loading basefw - if basefw is not ready, notification processing is now skipped - documented several topology parsing helpers Changes [internal] RFC v2 -> [public] RFC v1: - dropped any sysfs related changes from this series, moved to follow up one - dropped entire subscription-mechanism found in ipc.c. Handlers that are delegated to service certain firmware notifications are now called directly - fixed kernel doc for snd_soc_dapm_new_dai_widgets() as reported by ikp - prefixed snd_hda_codec_device_init() as suggested by Amadeo - improved comments for d0ix transitions for APL-based platforms as suggested by Pierre - a ton of spelling related fixes in most of the commit messages - fixed remaining warnings pointed by scan-build (variable assigned but not used) - replaced most of 'cAVS X.Y' expression usages with 'platform-based' equivalents as suggested by Pierre e.g.: cAVS 1.5 -> SKL-based Changes [internal] RFC v1 -> [internal] RFC v2: - fixed memleak caused by lack of kfree(vols) if memory allocation fails in avs_peakvol_create() as reported by Curtis - fixed missing 'i' iterator incrementation in avs_widget_ready() causing reference loss as reported by Curtis - replace hardcode: 0x40 usage with snd_hdac_calc_stream_format as suggested by Curtis. In consequence, readability for all code loading (CL) procedues has increased and such approach auto-documents the CL stream preparation - updated behavior of all index-fetching functions found in utils.c: avs_module_entry_index(), avs_module_id_entry_index() and follow ups: avs_get_module_entry(), avs_get_module_id_entry() to better conform to linux-kernel standard when no entry is found (return -ENOENT) rather than C++ standard (return -1, what in kernel case translated to -EPERM) as suggested by Curtis and Peter - several suggestions have been made regarding spacing, and so far, I've agreed and applied with all of them. None proposed seemed out of place or redundant - avs_path_stop() renamed to avs_path_pause() pipeline states are represented by RESET/PAUSED/RUNNING. avs_path_reset() and avs_path_run() were already there and avs_path_stop() just didn't look cohesive - added missing parsers for num_output_pin and num_input_pin which are required for custom modules such as WAVES or DSM - dropped 'priv_param_length' from custom module descriptor as this field is obsolete in firmware - parse_dictionary() has been split into parse_dictionary_header() and parse_dictionary_entries() to drop code duplication present in several parsing function which could not re-use entire parse_dictionary() - added avs_tplg_vendor_array_lookup_next() and avs_tplg_vendor_entry_next() to drop code duplicated present in several parsing functions. This change greatly impacted readability of all parsers - parsing helpers such avs_tplg_vendor_array_lookup() now return offset by updating specified in function argument list u32 *offset variable. This is to address problem when u32 offset would be greater than max int, which is the return type for these functions - AVS_DEFINE_PTR_PARSER() macro has been introduced to drop code duplication for all ptr-parsing users - all struct avs_path_module creators have had their declaration updated: function argument *owner ceased to exist as it could already be accessed by mod->owner - fixed the order of operation for conditional paths (e.g.: echo reference) so these are no longer controlled by "source" path and instead are impacted by state changes of source and sink paths both. Previously only source path e.g. playback sourcing echo reference would trigger RUNNING status for conditional path. Equivalent RUNNING on WoV path which is in this case sink path, would not do so, leading to order-of-operation problems. Behavior has been changed to: both source and sink need to be RUNNING for conditional path to be set to RUNNING too. PAUSED for either source or sink will cause PAUSED transition for conditional path. - to achieve the above, path states are now saved in 'state' i.e. new u32 field for struct avs_path - resigned from fw_filename field usage in favour of newly added tplg_filename for machine board descriptors as suggested by Pierre - platform descriptor fields have had their names update better reflect their purpose as suggested by Pierre - fixed comp_list missing locking when manipulated - all message senders now accept request as pointer as suggeseted by Peter - resigned of AZX_ usage for all ADSP-related registers, leaving them only for HOST memory space related operations - fixed disable path for core DSP operations: power/reset/stall as reported by Peter - safety when locking between received responses (reply vs notification) has been lowered as suggested by Pierre. Most usages are not performed in IRQ context and none is done in hard-IRQ one - s/master/main/ plus AVS_MAIN_CORE_MASK has replaced ->master_mask - several functions have had their logging updated - logs have been moved to lower level functions as suggested by Pierre - hdac_ext_stream usage has been streamlined to estream, hdac_streams are represented by hstream instead - hw_params() are resilient to scenarios when they are called mutliple times as reported by Pierre - avs_dsp_enable() now collapses if any of its steps fails as reported by Pierre and Peter - avs_module_ida_empty() now returns value of type bool as suggested by Bard [1]: https://www.spinics.net/lists/alsa-devel/msg116440.html [2]: https://www.spinics.net/lists/alsa-devel/msg116901.html [3]: https://www.spinics.net/lists/alsa-devel/msg94199.html [4]: https://www.spinics.net/lists/alsa-devel/msg92588.html [5]: https://lore.kernel.org/all/20190808181549.12521-1-cezary.rojewski@intel.co… [6]: https://lore.kernel.org/alsa-devel/20220207114906.3759800-1-cezary.rojewski… [7]: https://lore.kernel.org/all/20211208111301.1817725-1-cezary.rojewski@intel.… Cezary Rojewski (17): ALSA: hda: Add helper macros for DSP capable devices ASoC: Export DAI register and widget ctor and dctor functions ASoC: Intel: Introduce AVS driver ASoC: Intel: avs: Inter process communication ASoC: Intel: avs: Add code loading requests ASoC: Intel: avs: Add pipeline management requests ASoC: Intel: avs: Add module management requests ASoC: Intel: avs: Add power management requests ASoC: Intel: avs: Add ROM requests ASoC: Intel: avs: Add basefw runtime-parameter requests ASoC: Intel: avs: Firmware resources management utilities ASoC: Intel: avs: Declare module configuration types ASoC: Intel: avs: Dynamic firmware resources management ASoC: Intel: avs: General code loading flow ASoC: Intel: avs: Implement CLDMA transfer ASoC: Intel: avs: Code loading over CLDMA ASoC: Intel: avs: Code loading over HDA include/sound/hdaudio.h | 2 + include/sound/hdaudio_ext.h | 50 +++ include/sound/soc-dapm.h | 1 + sound/soc/intel/Kconfig | 12 + sound/soc/intel/Makefile | 1 + sound/soc/intel/avs/Makefile | 6 + sound/soc/intel/avs/avs.h | 248 +++++++++++ sound/soc/intel/avs/cldma.c | 317 ++++++++++++++ sound/soc/intel/avs/cldma.h | 29 ++ sound/soc/intel/avs/core.c | 62 +++ sound/soc/intel/avs/dsp.c | 303 +++++++++++++ sound/soc/intel/avs/ipc.c | 393 +++++++++++++++++ sound/soc/intel/avs/loader.c | 607 +++++++++++++++++++++++++ sound/soc/intel/avs/messages.c | 697 +++++++++++++++++++++++++++++ sound/soc/intel/avs/messages.h | 753 ++++++++++++++++++++++++++++++++ sound/soc/intel/avs/registers.h | 75 ++++ sound/soc/intel/avs/utils.c | 301 +++++++++++++ sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 15 + 19 files changed, 3873 insertions(+) create mode 100644 sound/soc/intel/avs/Makefile create mode 100644 sound/soc/intel/avs/avs.h create mode 100644 sound/soc/intel/avs/cldma.c create mode 100644 sound/soc/intel/avs/cldma.h create mode 100644 sound/soc/intel/avs/core.c create mode 100644 sound/soc/intel/avs/dsp.c create mode 100644 sound/soc/intel/avs/ipc.c create mode 100644 sound/soc/intel/avs/loader.c create mode 100644 sound/soc/intel/avs/messages.c create mode 100644 sound/soc/intel/avs/messages.h create mode 100644 sound/soc/intel/avs/registers.h create mode 100644 sound/soc/intel/avs/utils.c -- 2.25.1

3 64

[PATCH 0/3] ASoC: Intel: boards: cleanups for 5.18
by Pierre-Louis Bossart 09 Mar '22

09 Mar '22

Updates to clean-up the GPIOLIB dependency and a quirk for HP SoundWire devices. Anthony I Gilea (1): ASoC: Intel: sof_sdw: fix quirks for 2022 HP Spectre x360 13" Pierre-Louis Bossart (2): ASoC: Intel: boards: remove explicit dependency on GPIOLIB when DMIC is used" ASoC: Intel: boards: add GPIOLIB dependency where missed drivers/soundwire/dmi-quirks.c | 2 +- sound/soc/intel/boards/Kconfig | 49 +++++++++++++++++++------------- sound/soc/intel/boards/sof_sdw.c | 2 +- 3 files changed, 31 insertions(+), 22 deletions(-) -- 2.30.2

3 5

[PATCH v2] ALSA: hda/i915 - avoid hung task timeout in i915 wait
by Kai Vehmanen 09 Mar '22

09 Mar '22

If kernel is built with hung task detection enabled and CONFIG_DEFAULT_HUNG_TASK_TIMEOUT set to less than 60 seconds, snd_hdac_i915_init() will trigger the hung task timeout in case i915 is not available and taint the kernel. Split the 60sec wait into a loop of smaller waits to avoid this. Reviewed-by: Lucas De Marchi <lucas.demarchi(a)intel.com> Co-developed-by: Ramalingam C <ramalingam.c(a)intel.com> Signed-off-by: Ramalingam C <ramalingam.c(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/hda/hdac_i915.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) Changes V1->V2: - address local variable naming issue raised by Amadeusz and use Takashi's proposal diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 454474ac5716..aa48bed7baf7 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -143,7 +143,7 @@ static bool i915_gfx_present(void) int snd_hdac_i915_init(struct hdac_bus *bus) { struct drm_audio_component *acomp; - int err; + int err, i; if (!i915_gfx_present()) return -ENODEV; @@ -159,9 +159,11 @@ int snd_hdac_i915_init(struct hdac_bus *bus) if (!acomp->ops) { if (!IS_ENABLED(CONFIG_MODULES) || !request_module("i915")) { - /* 60s timeout */ - wait_for_completion_timeout(&acomp->master_bind_complete, - msecs_to_jiffies(60 * 1000)); + /* max 60s timeout */ + for (i = 0; i < 60; i++) + if (wait_for_completion_timeout(&acomp->master_bind_complete, + msecs_to_jiffies(1000))) + break; } } if (!acomp->ops) { base-commit: fd7698cf0858f8c5e659b655109cd93c2f15cdd3 -- 2.35.1

3 8

[PATCH] ASoC: tas5805m: fix pdn polarity
by John Keeping 09 Mar '22

09 Mar '22

The binding defines the GPIO as "pdn-gpios" so when the GPIO is active the expectation is that the power down signal is asserted, but the driver swaps this to represent the inverted logic of the electrical signal. The GPIO_ACTIVE_HIGH/LOW flags should be used to identify the inverted logic here with the driver treating power down as active when the mapped GPIO value is 1. Fixes: ec45268467f4 ("ASoC: add support for TAS5805M digital amplifier") Signed-off-by: John Keeping <john(a)metanate.com> --- sound/soc/codecs/tas5805m.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tas5805m.c b/sound/soc/codecs/tas5805m.c index fa0e81ec875a..12146a860ef8 100644 --- a/sound/soc/codecs/tas5805m.c +++ b/sound/soc/codecs/tas5805m.c @@ -155,7 +155,7 @@ static const uint32_t tas5805m_volume[] = { struct tas5805m_priv { struct regulator *pvdd; - struct gpio_desc *gpio_pdn_n; + struct gpio_desc *gpio_pdn; uint8_t *dsp_cfg_data; int dsp_cfg_len; @@ -444,11 +444,11 @@ static int tas5805m_i2c_probe(struct i2c_client *i2c) dev_set_drvdata(dev, tas5805m); tas5805m->regmap = regmap; - tas5805m->gpio_pdn_n = devm_gpiod_get(dev, "pdn", GPIOD_OUT_LOW); - if (IS_ERR(tas5805m->gpio_pdn_n)) { + tas5805m->gpio_pdn = devm_gpiod_get(dev, "pdn", GPIOD_OUT_HIGH); + if (IS_ERR(tas5805m->gpio_pdn)) { dev_err(dev, "error requesting PDN gpio: %ld\n", - PTR_ERR(tas5805m->gpio_pdn_n)); - return PTR_ERR(tas5805m->gpio_pdn_n); + PTR_ERR(tas5805m->gpio_pdn)); + return PTR_ERR(tas5805m->gpio_pdn); } /* This configuration must be generated by PPC3. The file loaded @@ -505,7 +505,7 @@ static int tas5805m_i2c_probe(struct i2c_client *i2c) } usleep_range(100000, 150000); - gpiod_set_value(tas5805m->gpio_pdn_n, 1); + gpiod_set_value(tas5805m->gpio_pdn, 0); usleep_range(10000, 15000); /* Don't register through devm. We need to be able to unregister @@ -515,7 +515,7 @@ static int tas5805m_i2c_probe(struct i2c_client *i2c) &tas5805m_dai, 1); if (ret < 0) { dev_err(dev, "unable to register codec: %d\n", ret); - gpiod_set_value(tas5805m->gpio_pdn_n, 0); + gpiod_set_value(tas5805m->gpio_pdn, 1); regulator_disable(tas5805m->pvdd); return ret; } @@ -529,7 +529,7 @@ static int tas5805m_i2c_remove(struct i2c_client *i2c) struct tas5805m_priv *tas5805m = dev_get_drvdata(dev); snd_soc_unregister_component(dev); - gpiod_set_value(tas5805m->gpio_pdn_n, 0); + gpiod_set_value(tas5805m->gpio_pdn, 1); usleep_range(10000, 15000); regulator_disable(tas5805m->pvdd); return 0; -- 2.35.1

2 3

[PATCH v3] arm64: dts: imx8mn-bsh-smm-s2pro: Add tlv320aic31xx audio card node
by Ariel D'Alessandro 09 Mar '22

09 Mar '22

BSH SystemMaster (SMM) S2 PRO board comes with an audio card based on tlv320aic31xx family codec. The audio card exposes two playback devices, one of them using the EASRC (Enhanced Asynchronous Sample Rate Converter) module. Note that this would require SDMA and EASRC firmware in order to work. Signed-off-by: Ariel D'Alessandro <ariel.dalessandro(a)collabora.com> Signed-off-by: Michael Trimarchi <michael(a)amarulasolutions.com> --- .../dts/freescale/imx8mn-bsh-smm-s2pro.dts | 94 +++++++++++++++++++ 1 file changed, 94 insertions(+) diff --git a/arch/arm64/boot/dts/freescale/imx8mn-bsh-smm-s2pro.dts b/arch/arm64/boot/dts/freescale/imx8mn-bsh-smm-s2pro.dts index c6a8ed6745c1..44e2635d3096 100644 --- a/arch/arm64/boot/dts/freescale/imx8mn-bsh-smm-s2pro.dts +++ b/arch/arm64/boot/dts/freescale/imx8mn-bsh-smm-s2pro.dts @@ -7,6 +7,7 @@ /dts-v1/; #include "imx8mn-bsh-smm-s2-common.dtsi" +#include <dt-bindings/sound/tlv320aic31xx.h> / { model = "BSH SMM S2 PRO"; @@ -16,6 +17,69 @@ memory@40000000 { device_type = "memory"; reg = <0x0 0x40000000 0x0 0x20000000>; }; + + sound-tlv320aic31xx { + compatible = "fsl,imx-audio-tlv320aic31xx"; + model = "tlv320aic31xx-hifi"; + audio-cpu = <&sai3>; + audio-codec = <&codec>; + audio-asrc = <&easrc>; + audio-routing = + "Ext Spk", "SPL", + "Ext Spk", "SPR"; + mclk-id = <PLL_CLKIN_BCLK>; + }; + + vdd_input: vdd_input { + compatible = "regulator-fixed"; + regulator-name = "vdd_input"; + regulator-min-microvolt = <5000000>; + regulator-max-microvolt = <5000000>; + }; +}; + +&easrc { + fsl,asrc-rate = <48000>; + fsl,asrc-format = <10>; + status = "okay"; +}; + +&i2c2 { + clock-frequency = <400000>; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_i2c2>; + status = "okay"; + + tlv320dac3101: audio-codec@18 { + #sound-dai-cells = <0>; + compatible = "ti,tlv320dac3101"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_dac_rst>; + reg = <0x18>; + + ai31xx-micbias-vg = <MICBIAS_AVDDV>; + + HPVDD-supply = <&buck4_reg>; + SPRVDD-supply = <&vdd_input>; + SPLVDD-supply = <&vdd_input>; + AVDD-supply = <&buck4_reg>; + IOVDD-supply = <&buck4_reg>; + DVDD-supply = <&buck5_reg>; + reset-gpios = <&gpio1 6 GPIO_ACTIVE_LOW>; + + clocks = <&clk IMX8MN_CLK_SAI3_ROOT>; + clock-names = "mclk"; + }; +}; + +&sai3 { + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_sai3>; + assigned-clocks = <&clk IMX8MN_CLK_SAI3>; + assigned-clock-parents = <&clk IMX8MN_AUDIO_PLL1_OUT>; + assigned-clock-rates = <24576000>; + fsl,sai-mclk-direction-output; + status = "okay"; }; /* eMMC */ @@ -30,6 +94,36 @@ &usdhc1 { }; &iomuxc { + pinctrl_dac_rst: dacrstgrp { + fsl,pins = < + MX8MN_IOMUXC_GPIO1_IO06_GPIO1_IO6 0x19 /* DAC_RST */ + >; + }; + + pinctrl_espi2: espi2grp { + fsl,pins = < + MX8MN_IOMUXC_ECSPI2_SCLK_ECSPI2_SCLK 0x082 + MX8MN_IOMUXC_ECSPI2_MOSI_ECSPI2_MOSI 0x082 + MX8MN_IOMUXC_ECSPI2_MISO_ECSPI2_MISO 0x082 + MX8MN_IOMUXC_ECSPI2_SS0_ECSPI2_SS0 0x040 + >; + }; + + pinctrl_i2c2: i2c2grp { + fsl,pins = < + MX8MN_IOMUXC_I2C2_SCL_I2C2_SCL 0x400000c3 + MX8MN_IOMUXC_I2C2_SDA_I2C2_SDA 0x400000c3 + >; + }; + + pinctrl_sai3: sai3grp { + fsl,pins = < + MX8MN_IOMUXC_SAI3_TXFS_SAI3_TX_SYNC 0xd6 + MX8MN_IOMUXC_SAI3_TXC_SAI3_TX_BCLK 0xd6 + MX8MN_IOMUXC_SAI3_TXD_SAI3_TX_DATA0 0xd6 + >; + }; + pinctrl_usdhc1: usdhc1grp { fsl,pins = < MX8MN_IOMUXC_SD1_CLK_USDHC1_CLK 0x40000090 -- 2.34.1

2 2

[PATCH] ALSA: hda: Add AlderLake-PS variant PCI ID
by Kai Vehmanen 09 Mar '22

09 Mar '22

Add HD Audio PCI ID for a variant of Intel AlderLake-P. Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 5af7a1e18013..0a83eb6b88b1 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2499,6 +2499,8 @@ static const struct pci_device_id azx_ids[] = { /* Alderlake-P */ { PCI_DEVICE(0x8086, 0x51c8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + { PCI_DEVICE(0x8086, 0x51c9), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, { PCI_DEVICE(0x8086, 0x51cd), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Alderlake-M */ base-commit: fd7698cf0858f8c5e659b655109cd93c2f15cdd3 prerequisite-patch-id: 0259e96fec7d4c6138df07807ca8fbc6f73e0c31 -- 2.35.1

2 1

[PATCH v7 0/2] Add support for SoundWire1.6 audio cgcr register control
by Srinivasa Rao Mandadapu 09 Mar '22

09 Mar '22

This patch series is to add v1.6.0 compatible name for qcom soundwire driver and corresponding dt bindings. Changes Since V6: -- Update commit message -- Update Example with reset control in dt bindings. Changes Since V5: -- Constify static struct qcom_swrm_data global variables. -- Remove redundant swrm_v1_6_data variable. Changes Since V4: -- Dropped audio cgcr control patch due to dependency on clock patches. -- Update dt-bindings as per new reset control properties. Changes Since V3: -- Add v1.6.0 compatible name and soundwire data structure. -- Change macro define name properly. -- Update bindings for new property. -- Change commit message description. -- Change signedoff by sequence. Changes since v2: -- Update error check after ioremap. Changes since v1: -- Add const name to mask value. Srinivasa Rao Mandadapu (2): soundwire: qcom: Add compatible name for v1.6.0 dt-bindings: soundwire: qcom: Add bindings for audio clock reset control property Documentation/devicetree/bindings/soundwire/qcom,sdw.txt | 14 ++++++++++++++ drivers/soundwire/qcom.c | 1 + 2 files changed, 15 insertions(+) -- 2.7.4

3 5

[PATCH] ASoC: mxs: Fix error handling in mxs_sgtl5000_probe
by Miaoqian Lin 08 Mar '22

08 Mar '22

This function only calls of_node_put() in the regular path. And it will cause refcount leak in error paths. For example, when codec_np is NULL, saif_np[0] and saif_np[1] are not NULL, it will cause leaks. of_node_put() will check if the node pointer is NULL, so we can call it directly to release the refcount of regular pointers. Fixes: e968194b45c4 ("ASoC: mxs: add device tree support for mxs-sgtl5000") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/mxs/mxs-sgtl5000.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 2412dc7e65d4..746f40938675 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -118,6 +118,9 @@ static int mxs_sgtl5000_probe(struct platform_device *pdev) codec_np = of_parse_phandle(np, "audio-codec", 0); if (!saif_np[0] || !saif_np[1] || !codec_np) { dev_err(&pdev->dev, "phandle missing or invalid\n"); + of_node_put(codec_np); + of_node_put(saif_np[0]); + of_node_put(saif_np[1]); return -EINVAL; } -- 2.17.1

2 1

[PATCH] ASoC: rk817: Fix missing clk_disable_unprepare() in rk817_platform_probe
by Miaoqian Lin 08 Mar '22

08 Mar '22

Fix the missing clk_disable_unprepare() before return from rk817_platform_probe() in the error handling case. Fixes: 0d6a04da9b25 ("ASoC: Add Rockchip rk817 audio CODEC support") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/codecs/rk817_codec.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 03f24edfe4f6..8fffe378618d 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -508,12 +508,14 @@ static int rk817_platform_probe(struct platform_device *pdev) if (ret < 0) { dev_err(&pdev->dev, "%s() register codec error %d\n", __func__, ret); - goto err_; + goto err_clk; } return 0; -err_: +err_clk: + clk_disable_unprepare(rk817_codec_data->mclk); +err_: return ret; } -- 2.17.1

3 2

[PATCH] ASoC: SOF: Add missing of_node_put() in imx8m_probe
by Miaoqian Lin 08 Mar '22

08 Mar '22

The device_node pointer is returned by of_parse_phandle() with refcount incremented. We should use of_node_put() on it when done. Fixes: afb93d716533 ("ASoC: SOF: imx: Add i.MX8M HW support") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/sof/imx/imx8m.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c index 788e77bcb603..60251486b24b 100644 --- a/sound/soc/sof/imx/imx8m.c +++ b/sound/soc/sof/imx/imx8m.c @@ -224,6 +224,7 @@ static int imx8m_probe(struct snd_sof_dev *sdev) } ret = of_address_to_resource(res_node, 0, &res); + of_node_put(res_node); if (ret) { dev_err(&pdev->dev, "failed to get reserved region address\n"); goto exit_pdev_unregister; -- 2.17.1

3 2

[PATCH] ASoC: mediatek: mt8192-mt6359: Fix error handling in mt8192_mt6359_dev_probe
by Miaoqian Lin 08 Mar '22

08 Mar '22

The device_node pointer is returned by of_parse_phandle() with refcount incremented. We should use of_node_put() on it when done. This function only calls of_node_put() in the regular path. And it will cause refcount leak in error paths. Fix this by calling of_node_put() in error handling too. Fixes: 4e28491a7a19 ("ASoC: mediatek: mt8192-mt6359: fix device_node leak") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 18 ++++++++++++------ 1 file changed, 12 insertions(+), 6 deletions(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index f7daad1bfe1e..ee91569c0911 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -1116,8 +1116,10 @@ static int mt8192_mt6359_dev_probe(struct platform_device *pdev) } card = (struct snd_soc_card *)of_device_get_match_data(&pdev->dev); - if (!card) - return -EINVAL; + if (!card) { + ret = -EINVAL; + goto put_platform_node; + } card->dev = &pdev->dev; hdmi_codec = of_parse_phandle(pdev->dev.of_node, @@ -1159,20 +1161,24 @@ static int mt8192_mt6359_dev_probe(struct platform_device *pdev) } priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL); - if (!priv) - return -ENOMEM; + if (!priv) { + ret = -ENOMEM; + goto put_hdmi_codec; + } snd_soc_card_set_drvdata(card, priv); ret = mt8192_afe_gpio_init(&pdev->dev); if (ret) { dev_err(&pdev->dev, "init gpio error %d\n", ret); - return ret; + goto put_hdmi_codec; } ret = devm_snd_soc_register_card(&pdev->dev, card); - of_node_put(platform_node); +put_hdmi_codec: of_node_put(hdmi_codec); +put_platform_node: + of_node_put(platform_node); return ret; } -- 2.17.1

3 2

[PATCH] ASoC: atmel: Fix error handling in snd_proto_probe
by Miaoqian Lin 08 Mar '22

08 Mar '22

The device_node pointer is returned by of_parse_phandle() with refcount incremented. We should use of_node_put() on it when done. This function only calls of_node_put() in the regular path. And it will cause refcount leak in error paths. Fix this by calling of_node_put() in error handling too. Fixes: a45f8853a5f9 ("ASoC: Add driver for PROTO Audio CODEC (with a WM8731)") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/atmel/mikroe-proto.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/soc/atmel/mikroe-proto.c b/sound/soc/atmel/mikroe-proto.c index 627564c18c27..9ed634ee9dfc 100644 --- a/sound/soc/atmel/mikroe-proto.c +++ b/sound/soc/atmel/mikroe-proto.c @@ -115,7 +115,8 @@ static int snd_proto_probe(struct platform_device *pdev) cpu_np = of_parse_phandle(np, "i2s-controller", 0); if (!cpu_np) { dev_err(&pdev->dev, "i2s-controller missing\n"); - return -EINVAL; + ret = -EINVAL; + goto put_codec_node; } dai->cpus->of_node = cpu_np; dai->platforms->of_node = cpu_np; @@ -125,7 +126,8 @@ static int snd_proto_probe(struct platform_device *pdev) &bitclkmaster, &framemaster); if (bitclkmaster != framemaster) { dev_err(&pdev->dev, "Must be the same bitclock and frame master\n"); - return -EINVAL; + ret = -EINVAL; + goto put_cpu_node; } if (bitclkmaster) { if (codec_np == bitclkmaster) @@ -136,18 +138,21 @@ static int snd_proto_probe(struct platform_device *pdev) dai_fmt |= snd_soc_daifmt_parse_clock_provider_as_flag(np, NULL); } - of_node_put(bitclkmaster); - of_node_put(framemaster); - dai->dai_fmt = dai_fmt; - of_node_put(codec_np); - of_node_put(cpu_np); + dai->dai_fmt = dai_fmt; ret = snd_soc_register_card(&snd_proto); if (ret) dev_err_probe(&pdev->dev, ret, "snd_soc_register_card() failed\n"); + +put_cpu_node: + of_node_put(bitclkmaster); + of_node_put(framemaster); + of_node_put(cpu_np); +put_codec_node: + of_node_put(codec_np); return ret; } -- 2.17.1

3 3

[v2] ASoC: mediatek: mt8183: support wb bt audio
by Jiaxin Yu 08 Mar '22

08 Mar '22

Use "bt-sco-pcm-wb" codec dai driver for wb bt audio. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> Reviewed-by: Tzung-Bi Shih <tzungbi(a)kernel.org> --- changed from v1: - describe changes in imperative mood - remove description of unnecessary dependenceies sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index b0ec5ebd4f2d..889f9e4a96aa 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -260,7 +260,7 @@ SND_SOC_DAILINK_DEFS(pcm2, SND_SOC_DAILINK_DEFS(i2s0, DAILINK_COMP_ARRAY(COMP_CPU("I2S0")), - DAILINK_COMP_ARRAY(COMP_CODEC("bt-sco", "bt-sco-pcm")), + DAILINK_COMP_ARRAY(COMP_CODEC("bt-sco", "bt-sco-pcm-wb")), DAILINK_COMP_ARRAY(COMP_EMPTY())); SND_SOC_DAILINK_DEFS(i2s1, @@ -291,7 +291,7 @@ SND_SOC_DAILINK_DEFS(i2s3_rt1015p, SND_SOC_DAILINK_DEFS(i2s5, DAILINK_COMP_ARRAY(COMP_CPU("I2S5")), - DAILINK_COMP_ARRAY(COMP_CODEC("bt-sco", "bt-sco-pcm")), + DAILINK_COMP_ARRAY(COMP_CODEC("bt-sco", "bt-sco-pcm-wb")), DAILINK_COMP_ARRAY(COMP_EMPTY())); SND_SOC_DAILINK_DEFS(tdm, @@ -508,7 +508,6 @@ static struct snd_soc_dai_link mt8183_mt6358_ts3a227_dai_links[] = { .no_pcm = 1, .dpcm_capture = 1, .ignore_suspend = 1, - .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, .ops = &mt8183_mt6358_i2s_ops, SND_SOC_DAILINK_REG(i2s0), }, @@ -541,7 +540,6 @@ static struct snd_soc_dai_link mt8183_mt6358_ts3a227_dai_links[] = { .no_pcm = 1, .dpcm_playback = 1, .ignore_suspend = 1, - .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, .ops = &mt8183_mt6358_i2s_ops, SND_SOC_DAILINK_REG(i2s5), }, -- 2.25.1

2 1

[PATCH 00/18] Clean ups and preparation for IPC abstraction in the SOF driver
by Ranjani Sridharan 08 Mar '22

08 Mar '22

In preparation for adding support for the new IPC version that has been introduced in the SOF firmware, this patch set includes some clean ups and necessary modifications to commonly used functions that will be re-used across different IPC-specific code. Pierre-Louis Bossart (1): ASoC: SOF: sof-audio: removed unused function Rander Wang (1): ASoC: SOF: move definition of snd_sof_ipc to header file Ranjani Sridharan (16): ASoC: SOF: remove snd_sof_pipeline_find() ASoC: SOF: simplify snd_sof_device_remove() ASoC: SOF: set swidget's core for scheduler widget ASoC: SOF: topology: remove redundant code ASoC: SOF: topology: remove redundant code in sof_link_afe_load() ASoC: SOF: topology: Drop the size parameter from struct sof_topology_token ASoC: SOF: topology: Modify the get_token op for string tokens ASoC: SOF: topology: expose some get_token ops ASoC: SOF: change comp_dai to a pointer in struct snd_sof_dai ASoC: SOF: make struct snd_sof_widget IPC agnostic ASoC: SOF: topology: make sof_route_load() IPC agnostic ASoC: SOF: Add a tuples array to struct snd_sof_widget ASoC: SOF: topology: Modify signature for token parsing functions ASoC: SOF: topology: Rename arguments in sof_parse_token_sets() ASoC: SOF: topology: Rename arguments in sof_parse_tokens() ASoC: SOF: make struct snd_sof_dai IPC agnostic include/sound/sof/dai.h | 5 + include/sound/sof/topology.h | 5 - sound/soc/sof/core.c | 10 +- sound/soc/sof/intel/hda-dai.c | 27 ++- sound/soc/sof/intel/hda.c | 39 +++- sound/soc/sof/ipc.c | 12 -- sound/soc/sof/pcm.c | 77 +++---- sound/soc/sof/sof-audio.c | 108 +++------- sound/soc/sof/sof-audio.h | 30 ++- sound/soc/sof/sof-priv.h | 12 ++ sound/soc/sof/topology.c | 391 +++++++++++++++++----------------- 11 files changed, 354 insertions(+), 362 deletions(-) -- 2.25.1

3 21

[PATCH v2] ASoC: Intel: catpt: use asoc_substream_to_rtd()
by Lianjie Zhang 08 Mar '22

08 Mar '22

Now we can use asoc_substream_to_rtd() macro, let's use it. Signed-off-by: Lianjie Zhang <zhanglianjie(a)uniontech.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c index 939a9b801dec..a26000cd5ceb 100644 --- a/sound/soc/intel/catpt/pcm.c +++ b/sound/soc/intel/catpt/pcm.c @@ -74,7 +74,7 @@ static struct catpt_stream_template *catpt_topology[] = { static struct catpt_stream_template * catpt_get_stream_template(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtm = substream->private_data; + struct snd_soc_pcm_runtime *rtm = asoc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtm, 0); enum catpt_stream_type type; @@ -593,7 +593,7 @@ static int catpt_component_pcm_construct(struct snd_soc_component *component, static int catpt_component_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtm = substream->private_data; + struct snd_soc_pcm_runtime *rtm = asoc_substream_to_rtd(substream); if (!rtm->dai_link->no_pcm) snd_soc_set_runtime_hwparams(substream, &catpt_pcm_hardware); @@ -604,7 +604,7 @@ static snd_pcm_uframes_t catpt_component_pointer(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtm = substream->private_data; + struct snd_soc_pcm_runtime *rtm = asoc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtm, 0); struct catpt_stream_runtime *stream; struct catpt_dev *cdev = dev_get_drvdata(component->dev); -- 2.20.1

3 3

[PATCH] ASoC: rockchip: i2s: Fix missing clk_disable_unprepare() in rockchip_i2s_probe
by Miaoqian Lin 08 Mar '22

08 Mar '22

Fix the missing clk_disable_unprepare() before return from rockchip_i2s_probe() in the error handling case. Fixes: 01605ad12875 ("ASoC: rockchip-i2s: enable "hclk" for rockchip I2S controller") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/rockchip/rockchip_i2s.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) diff --git a/sound/soc/rockchip/rockchip_i2s.c b/sound/soc/rockchip/rockchip_i2s.c index a6d7656c206e..4ce5d2579387 100644 --- a/sound/soc/rockchip/rockchip_i2s.c +++ b/sound/soc/rockchip/rockchip_i2s.c @@ -716,19 +716,23 @@ static int rockchip_i2s_probe(struct platform_device *pdev) i2s->mclk = devm_clk_get(&pdev->dev, "i2s_clk"); if (IS_ERR(i2s->mclk)) { dev_err(&pdev->dev, "Can't retrieve i2s master clock\n"); - return PTR_ERR(i2s->mclk); + ret = PTR_ERR(i2s->mclk); + goto err_clk; } regs = devm_platform_get_and_ioremap_resource(pdev, 0, &res); - if (IS_ERR(regs)) - return PTR_ERR(regs); + if (IS_ERR(regs)) { + ret = PTR_ERR(regs); + goto err_clk; + } i2s->regmap = devm_regmap_init_mmio(&pdev->dev, regs, &rockchip_i2s_regmap_config); if (IS_ERR(i2s->regmap)) { dev_err(&pdev->dev, "Failed to initialise managed register map\n"); - return PTR_ERR(i2s->regmap); + ret = PTR_ERR(i2s->regmap); + goto err_clk; } i2s->bclk_ratio = 64; @@ -768,7 +772,8 @@ static int rockchip_i2s_probe(struct platform_device *pdev) i2s_runtime_suspend(&pdev->dev); err_pm_disable: pm_runtime_disable(&pdev->dev); - +err_clk: + clk_disable_unprepare(i2s->hclk); return ret; } -- 2.17.1

2 1

[PATCH -next] ASoC: amd: acp5x-pcm-dma: Fix signedness bug
by YueHaibing 08 Mar '22

08 Mar '22

In acp5x_audio_probe() platform_get_irq() may return error, but i2s_irq now is unsigned int so the error handling is never triggered. Fixes: 87d71a128771 ("ASoC: amd: pcm-dma: Use platform_get_irq() to get the interrupt") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/amd/vangogh/acp5x.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/vangogh/acp5x.h b/sound/soc/amd/vangogh/acp5x.h index b85d3ee369a3..bd9f1c5684d1 100644 --- a/sound/soc/amd/vangogh/acp5x.h +++ b/sound/soc/amd/vangogh/acp5x.h @@ -85,7 +85,7 @@ struct i2s_dev_data { bool tdm_mode; bool master_mode; - unsigned int i2s_irq; + int i2s_irq; u16 i2s_instance; u32 tdm_fmt; void __iomem *acp5x_base; -- 2.17.1

2 1

[PATCH -next] ASoC: amd: acp3x: Fix signedness bug in acp3x
by YueHaibing 08 Mar '22

08 Mar '22

In acp3x_audio_probe() platform_get_irq() may return error, but i2s_irq now is unsigned int so the error handling is never triggered. Fixes: 87d71a128771 ("ASoC: amd: pcm-dma: Use platform_get_irq() to get the interrupt") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/amd/raven/acp3x.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/raven/acp3x.h b/sound/soc/amd/raven/acp3x.h index c3f0c8b7545d..7702f628ecd6 100644 --- a/sound/soc/amd/raven/acp3x.h +++ b/sound/soc/amd/raven/acp3x.h @@ -87,7 +87,7 @@ struct acp3x_platform_info { struct i2s_dev_data { bool tdm_mode; - unsigned int i2s_irq; + int i2s_irq; u16 i2s_instance; u32 tdm_fmt; u32 substream_type; -- 2.17.1

2 1

[PATCH] ALSA: hda/i915 - avoid hung task timeout in i915 wait
by Kai Vehmanen 08 Mar '22

08 Mar '22

If kernel is built with hung task detection enabled and CONFIG_DEFAULT_HUNG_TASK_TIMEOUT set to less than 60 seconds, snd_hdac_i915_init() will trigger the hung task timeout in case i915 is not available and taint the kernel. Split the 60sec wait into a loop of smaller waits to avoid this. Cc: Lucas De Marchi <lucas.demarchi(a)intel.com> Co-developed-by: Ramalingam C <ramalingam.c(a)intel.com> Signed-off-by: Ramalingam C <ramalingam.c(a)intel.com> Signed-off-by: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> --- sound/hda/hdac_i915.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 454474ac5716..6219de739b56 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -143,7 +143,8 @@ static bool i915_gfx_present(void) int snd_hdac_i915_init(struct hdac_bus *bus) { struct drm_audio_component *acomp; - int err; + unsigned long wait = 0; + int err, i; if (!i915_gfx_present()) return -ENODEV; @@ -159,9 +160,10 @@ int snd_hdac_i915_init(struct hdac_bus *bus) if (!acomp->ops) { if (!IS_ENABLED(CONFIG_MODULES) || !request_module("i915")) { - /* 60s timeout */ - wait_for_completion_timeout(&acomp->master_bind_complete, - msecs_to_jiffies(60 * 1000)); + /* max 60s timeout */ + for (i = 0; !wait && i < 60; i++) + wait = wait_for_completion_timeout(&acomp->master_bind_complete, + msecs_to_jiffies(1000)); } } if (!acomp->ops) { base-commit: fd7698cf0858f8c5e659b655109cd93c2f15cdd3 -- 2.35.1

3 2

[PATCH v3] ASoC: Intel: catpt: use asoc_substream_to_rtd()
by Lianjie Zhang 08 Mar '22

08 Mar '22

Now we can use asoc_substream_to_rtd() macro, let's use it. Signed-off-by: Lianjie Zhang <zhanglianjie(a)uniontech.com> diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c index 939a9b801dec..a26000cd5ceb 100644 --- a/sound/soc/intel/catpt/pcm.c +++ b/sound/soc/intel/catpt/pcm.c @@ -74,7 +74,7 @@ static struct catpt_stream_template *catpt_topology[] = { static struct catpt_stream_template * catpt_get_stream_template(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtm = substream->private_data; + struct snd_soc_pcm_runtime *rtm = asoc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtm, 0); enum catpt_stream_type type; @@ -593,7 +593,7 @@ static int catpt_component_pcm_construct(struct snd_soc_component *component, static int catpt_component_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtm = substream->private_data; + struct snd_soc_pcm_runtime *rtm = asoc_substream_to_rtd(substream); if (!rtm->dai_link->no_pcm) snd_soc_set_runtime_hwparams(substream, &catpt_pcm_hardware); @@ -604,7 +604,7 @@ static snd_pcm_uframes_t catpt_component_pointer(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtm = substream->private_data; + struct snd_soc_pcm_runtime *rtm = asoc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtm, 0); struct catpt_stream_runtime *stream; struct catpt_dev *cdev = dev_get_drvdata(component->dev); -- 2.20.1

1 0

[PATCH 0/2] ALSA: scarlett2: Add "Standalone" switch
by Geoffrey D. Bennett 08 Mar '22

08 Mar '22

Hi Takashi, I discovered an internal "standalone" switch on all the Scarlett Gen 2/3 interfaces with internal mixers. After enabling this switch, the interface will act as a standalone mixer (according to its previous configuration) when not connected to a USB host. The interfaces come from the factory with the switch off, and the vendor driver enables it without question. This patch adds a new ALSA control to set the switch on or off. This is the first configuration item that is common between the Gen 2 and 3 interfaces but with a different offset, so the patch is in two parts. The first patch allows for the same configuration item to have a different offset between Gen 2/3 and the second patch adds the new switch. Regards, Geoffrey. Geoffrey D. Bennett (2): ALSA: scarlett2: Split scarlett2_config_items[] into 3 sections ALSA: scarlett2: Add support for the internal "standalone" switch sound/usb/mixer_scarlett_gen2.c | 171 ++++++++++++++++++++++++++------ 1 file changed, 141 insertions(+), 30 deletions(-) -- 2.35.1

2 3

[PATCH] ALSA: hda/realtek: Add quirk for Clevo NP50PNJ
by Tim Crawford 08 Mar '22

08 Mar '22

Fixes headset detection on Clevo NP50PNJ. Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7d85989f585b..595f492c0268 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9144,6 +9144,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x8561, "Clevo NH[57][0-9][ER][ACDH]Q", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[57][0-9]RZ[Q]", ALC269_FIXUP_DMIC), SND_PCI_QUIRK(0x1558, 0x8668, "Clevo NP50B[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x866d, "Clevo NP5[05]PN[HJK]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x867d, "Clevo NP7[01]PN[HJK]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8680, "Clevo NJ50LU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC256_FIXUP_MIC_NO_PRESENCE_AND_RESUME), -- 2.35.1

2 1

Re: mmotm 2022-03-06-20-33 uploaded (sound/soc/intel/boards/)
by Randy Dunlap 08 Mar '22

08 Mar '22

On 3/6/22 20:34, Andrew Morton wrote: > The mm-of-the-moment snapshot 2022-03-06-20-33 has been uploaded to > > https://www.ozlabs.org/~akpm/mmotm/ > > mmotm-readme.txt says > > README for mm-of-the-moment: > > https://www.ozlabs.org/~akpm/mmotm/ > > This is a snapshot of my -mm patch queue. Uploaded at random hopefully > more than once a week. > > You will need quilt to apply these patches to the latest Linus release (5.x > or 5.x-rcY). The series file is in broken-out.tar.gz and is duplicated in > https://ozlabs.org/~akpm/mmotm/series on x86_64: ERROR: modpost: "sof_dai_get_bclk" [sound/soc/intel/boards/snd-soc-intel-sof-cirrus-common.ko] undefined! ERROR: modpost: "sof_dai_get_mclk" [sound/soc/intel/boards/snd-soc-intel-sof-realtek-common.ko] undefined! Full randconfig file is attached. -- ~Randy

2 2

Re: mmotm 2022-03-06-20-33 uploaded (sound/soc/codecs/snd-soc-lpass-[rt]x-macro.ko)
by Randy Dunlap 07 Mar '22

07 Mar '22

On 3/6/22 20:34, Andrew Morton wrote: > The mm-of-the-moment snapshot 2022-03-06-20-33 has been uploaded to > > https://www.ozlabs.org/~akpm/mmotm/ > > mmotm-readme.txt says > > README for mm-of-the-moment: > > https://www.ozlabs.org/~akpm/mmotm/ > > This is a snapshot of my -mm patch queue. Uploaded at random hopefully > more than once a week. > > You will need quilt to apply these patches to the latest Linus release (5.x > or 5.x-rcY). The series file is in broken-out.tar.gz and is duplicated in > https://ozlabs.org/~akpm/mmotm/series on x86_64: WARNING: unmet direct dependencies detected for SND_SOC_LPASS_RX_MACRO Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && COMMON_CLK [=n] Selected by [m]: - SND_SOC_SC7280 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_SOC_QCOM [=m] && I2C [=y] && SOUNDWIRE [=m] WARNING: unmet direct dependencies detected for SND_SOC_LPASS_TX_MACRO Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && COMMON_CLK [=n] Selected by [m]: - SND_SOC_SC7280 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_SOC_QCOM [=m] && I2C [=y] && SOUNDWIRE [=m] WARNING: unmet direct dependencies detected for SND_SOC_LPASS_RX_MACRO Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && COMMON_CLK [=n] Selected by [m]: - SND_SOC_SC7280 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_SOC_QCOM [=m] && I2C [=y] && SOUNDWIRE [=m] WARNING: unmet direct dependencies detected for SND_SOC_LPASS_TX_MACRO Depends on [n]: SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && COMMON_CLK [=n] Selected by [m]: - SND_SOC_SC7280 [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && SND_SOC_QCOM [=m] && I2C [=y] && SOUNDWIRE [=m] and then: ERROR: modpost: "of_clk_hw_simple_get" [sound/soc/codecs/snd-soc-lpass-tx-macro.ko] undefined! ERROR: modpost: "devm_clk_hw_register" [sound/soc/codecs/snd-soc-lpass-tx-macro.ko] undefined! ERROR: modpost: "devm_of_clk_add_hw_provider" [sound/soc/codecs/snd-soc-lpass-tx-macro.ko] undefined! ERROR: modpost: "__clk_get_name" [sound/soc/codecs/snd-soc-lpass-tx-macro.ko] undefined! ERROR: modpost: "of_clk_hw_simple_get" [sound/soc/codecs/snd-soc-lpass-rx-macro.ko] undefined! ERROR: modpost: "devm_clk_hw_register" [sound/soc/codecs/snd-soc-lpass-rx-macro.ko] undefined! ERROR: modpost: "devm_of_clk_add_hw_provider" [sound/soc/codecs/snd-soc-lpass-rx-macro.ko] undefined! ERROR: modpost: "__clk_get_name" [sound/soc/codecs/snd-soc-lpass-rx-macro.ko] undefined! Full randconfig file is attached. -- ~Randy

1 0

[PATCH] ASoC: qcom: select correct WCD938X config for SC7280
by Srinivas Kandagatla 07 Mar '22

07 Mar '22

SC7280 config selected WCD938X instead of WCD938X_SDW Soundwire codecs. WCD938X_SDW actually selects WCD938X, so directly selecting WCD938X results in unmet dependencies and below warning WARNING: unmet direct dependencies detected for SND_SOC_WCD938X Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=m] && SND_SOC_WCD938X_SDW [=n] && (SOUNDWIRE [=n] || !SOUNDWIRE [=n]) Selected by [m]: - SND_SOC_SC7280 [=m] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=m] && SND_SOC_QCOM [=m] && (I2C [=y] && SOUNDWIRE [=n] || COMPILE_TEST [=y]) Fix this issue by selecting WCD SoundWire codecs instead of component driver. Fixes: 57350bd41c3a ("ASoC: qcom: SC7280: Add machine driver") Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 0cd0dae5c545..82f5eafb2f6c 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -193,7 +193,7 @@ config SND_SOC_SC7280 select SND_SOC_QCOM_COMMON select SND_SOC_LPASS_SC7280 select SND_SOC_MAX98357A - select SND_SOC_WCD938X + select SND_SOC_WCD938X_SDW select SND_SOC_LPASS_MACRO_COMMON select SND_SOC_LPASS_RX_MACRO select SND_SOC_LPASS_TX_MACRO -- 2.21.0

2 1

[PATCH] ASoC: rt5682s: Stabilize the combo jack detection
by derek.fang＠realtek.com 07 Mar '22

07 Mar '22

From: Derek Fang <derek.fang(a)realtek.com> Changes: 1. Revise rt5682s_sar_power_mode and rt5682s_headset_detect to be more rational. 2. Manually set to the jack-unplugging state via rt5682s_headset_detect during going to suspend. Close unnecessary powers and prepare for re-detecting the CBJ during resuming. 3. Simplize rt5682s_resume. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682s.c | 47 +++++++++++++++++++++----------------- 1 file changed, 26 insertions(+), 21 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 92b8753f1267..1cba8ec7cedb 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -644,8 +644,7 @@ enum { SAR_PWR_SAVING, }; -static void rt5682s_sar_power_mode(struct snd_soc_component *component, - int mode, int jd_step) +static void rt5682s_sar_power_mode(struct snd_soc_component *component, int mode) { struct rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); @@ -675,16 +674,17 @@ static void rt5682s_sar_power_mode(struct snd_soc_component *component, snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, RT5682S_MB1_PATH_MASK | RT5682S_MB2_PATH_MASK, RT5682S_CTRL_MB1_FSM | RT5682S_CTRL_MB2_FSM); - if (!jd_step) { - snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, - RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_SEL_MB1_2_AUTO); - usleep_range(5000, 5500); - snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, - RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK, - RT5682S_SAR_BUTDET_EN | RT5682S_SAR_BUTDET_POW_NORM); - } + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_SEL_MB1_2_AUTO); + usleep_range(5000, 5500); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK, + RT5682S_SAR_BUTDET_EN | RT5682S_SAR_BUTDET_POW_NORM); break; case SAR_PWR_OFF: + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, + RT5682S_MB1_PATH_MASK | RT5682S_MB2_PATH_MASK, + RT5682S_CTRL_MB1_FSM | RT5682S_CTRL_MB2_FSM); snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK | RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_BUTDET_DIS | @@ -702,6 +702,10 @@ static void rt5682s_enable_push_button_irq(struct snd_soc_component *component) { snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_13, RT5682S_SAR_SOUR_MASK, RT5682S_SAR_SOUR_BTN); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK | + RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_BUTDET_EN | + RT5682S_SAR_BUTDET_POW_NORM | RT5682S_SAR_SEL_MB1_2_AUTO); snd_soc_component_write(component, RT5682S_IL_CMD_1, 0x0040); snd_soc_component_update_bits(component, RT5682S_4BTN_IL_CMD_2, RT5682S_4BTN_IL_MASK | RT5682S_4BTN_IL_RST_MASK, @@ -718,6 +722,10 @@ static void rt5682s_disable_push_button_irq(struct snd_soc_component *component) RT5682S_4BTN_IL_MASK, RT5682S_4BTN_IL_DIS); snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_13, RT5682S_SAR_SOUR_MASK, RT5682S_SAR_SOUR_TYPE); + snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, + RT5682S_SAR_BUTDET_MASK | RT5682S_SAR_BUTDET_POW_MASK | + RT5682S_SAR_SEL_MB1_2_CTL_MASK, RT5682S_SAR_BUTDET_DIS | + RT5682S_SAR_BUTDET_POW_SAV | RT5682S_SAR_SEL_MB1_2_MANU); } /** @@ -753,7 +761,8 @@ static int rt5682s_headset_detect(struct snd_soc_component *component, int jack_ RT5682S_OSW_L_DIS | RT5682S_OSW_R_DIS); snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_13, RT5682S_SAR_SOUR_MASK, RT5682S_SAR_SOUR_TYPE); - rt5682s_sar_power_mode(component, SAR_PWR_NORMAL, 1); + snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_3, + RT5682S_CBJ_IN_BUF_MASK, RT5682S_CBJ_IN_BUF_EN); snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, RT5682S_TRIG_JD_MASK, RT5682S_TRIG_JD_LOW); usleep_range(45000, 50000); @@ -779,9 +788,8 @@ static int rt5682s_headset_detect(struct snd_soc_component *component, int jack_ RT5682S_FAST_OFF_MASK, RT5682S_FAST_OFF_EN); snd_soc_component_update_bits(component, RT5682S_SAR_IL_CMD_1, RT5682S_SAR_SEL_MB1_2_MASK, val << RT5682S_SAR_SEL_MB1_2_SFT); - if (!snd_soc_dapm_get_pin_status(&component->dapm, "SAR")) - rt5682s_sar_power_mode(component, SAR_PWR_SAVING, 1); rt5682s_enable_push_button_irq(component); + rt5682s_sar_power_mode(component, SAR_PWR_SAVING); break; default: jack_type = SND_JACK_HEADPHONE; @@ -792,7 +800,7 @@ static int rt5682s_headset_detect(struct snd_soc_component *component, int jack_ RT5682S_OSW_L_EN | RT5682S_OSW_R_EN); usleep_range(35000, 40000); } else { - rt5682s_sar_power_mode(component, SAR_PWR_OFF, 1); + rt5682s_sar_power_mode(component, SAR_PWR_OFF); rt5682s_disable_push_button_irq(component); snd_soc_component_update_bits(component, RT5682S_CBJ_CTRL_1, RT5682S_TRIG_JD_MASK, RT5682S_TRIG_JD_LOW); @@ -1398,10 +1406,10 @@ static int sar_power_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: - rt5682s_sar_power_mode(component, SAR_PWR_NORMAL, 0); + rt5682s_sar_power_mode(component, SAR_PWR_NORMAL); break; case SND_SOC_DAPM_POST_PMD: - rt5682s_sar_power_mode(component, SAR_PWR_SAVING, 0); + rt5682s_sar_power_mode(component, SAR_PWR_SAVING); break; } @@ -2830,9 +2838,8 @@ static int rt5682s_suspend(struct snd_soc_component *component) cancel_delayed_work_sync(&rt5682s->jack_detect_work); cancel_delayed_work_sync(&rt5682s->jd_check_work); - if (rt5682s->hs_jack && rt5682s->jack_type == SND_JACK_HEADSET) - snd_soc_component_update_bits(component, RT5682S_4BTN_IL_CMD_2, - RT5682S_4BTN_IL_MASK, RT5682S_4BTN_IL_DIS); + if (rt5682s->hs_jack) + rt5682s->jack_type = rt5682s_headset_detect(component, 0); regcache_cache_only(rt5682s->regmap, true); regcache_mark_dirty(rt5682s->regmap); @@ -2848,8 +2855,6 @@ static int rt5682s_resume(struct snd_soc_component *component) regcache_sync(rt5682s->regmap); if (rt5682s->hs_jack) { - rt5682s->jack_type = 0; - rt5682s_sar_power_mode(component, SAR_PWR_NORMAL, 0); mod_delayed_work(system_power_efficient_wq, &rt5682s->jack_detect_work, msecs_to_jiffies(0)); } -- 2.17.1

2 1

[PATCH] ASoC: atmel: Add missing of_node_put() in at91sam9g20ek_audio_probe
by Miaoqian Lin 07 Mar '22

07 Mar '22

This node pointer is returned by of_parse_phandle() with refcount incremented in this function. Calling of_node_put() to avoid the refcount leak. Fixes: 531f67e41dcd ("ASoC: at91sam9g20ek-wm8731: convert to dt support") Signed-off-by: Miaoqian Lin <linmq006(a)gmail.com> --- sound/soc/atmel/sam9g20_wm8731.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 915da92e1ec8..33e43013ff77 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -214,6 +214,7 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) cpu_np = of_parse_phandle(np, "atmel,ssc-controller", 0); if (!cpu_np) { dev_err(&pdev->dev, "dai and pcm info missing\n"); + of_node_put(codec_np); return -EINVAL; } at91sam9g20ek_dai.cpus->of_node = cpu_np; -- 2.17.1

3 2

[PATCH 00/10] ASoC: SOF: updates for 5.18
by Pierre-Louis Bossart 07 Mar '22

07 Mar '22

A couple of updates for Intel and AMD hardware, along with minor cleanups Ajit Kumar Pandey (4): ASoC: SOF: amd: Flush cache after ATU_BASE_ADDR_GRP register update ASoC: SOF: amd: Use semaphore register to synchronize ipc's irq ASoC: SOF: amd: Move group register configuration to acp-loader ASoC: SOF: amd: Increase ACP_HW_SEM_RETRY_COUNT value Curtis Malainey (1): ASoC: SOF: fix 32 signed bit overflow Gongjun Song (1): ASoC: SOF: Intel: pci-tgl: add RPL-S support Peter Ujfalusi (2): ASoC: SOF: amd: acp-pcm: Take buffer information directly from runtime ASoC: SOF: amd: Do not set ipc_pcm_params ops as it is optional Pierre-Louis Bossart (2): ASoC: SOF: debug: clarify operator precedence ASoC: SOF: Intel: hda: clarify operator precedence include/sound/sof/header.h | 2 +- include/uapi/sound/sof/abi.h | 2 +- sound/soc/sof/amd/acp-dsp-offset.h | 1 + sound/soc/sof/amd/acp-ipc.c | 22 ++++++++++++++-------- sound/soc/sof/amd/acp-loader.c | 9 +++++++++ sound/soc/sof/amd/acp-pcm.c | 7 ++++--- sound/soc/sof/amd/acp-stream.c | 3 +++ sound/soc/sof/amd/acp.c | 29 ++++++++++++++--------------- sound/soc/sof/amd/acp.h | 3 +-- sound/soc/sof/amd/renoir.c | 1 - sound/soc/sof/debug.c | 2 +- sound/soc/sof/intel/hda.c | 2 +- sound/soc/sof/intel/pci-tgl.c | 2 ++ 13 files changed, 52 insertions(+), 33 deletions(-) -- 2.30.2

2 11

[PATCH 1/4] ASoC: amd: vg: fix signedness bug in acp5x_audio_probe()
by Dan Carpenter 07 Mar '22

07 Mar '22

The "adata->i2s_irq" variable is unsigned so the error handling will not work. Fixes: 87d71a128771 ("ASoC: amd: pcm-dma: Use platform_get_irq() to get the interrupt") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/amd/vangogh/acp5x-pcm-dma.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/vangogh/acp5x-pcm-dma.c b/sound/soc/amd/vangogh/acp5x-pcm-dma.c index e4e668593b3d..31fa166df98a 100644 --- a/sound/soc/amd/vangogh/acp5x-pcm-dma.c +++ b/sound/soc/amd/vangogh/acp5x-pcm-dma.c @@ -388,9 +388,10 @@ static int acp5x_audio_probe(struct platform_device *pdev) if (!adata->acp5x_base) return -ENOMEM; - adata->i2s_irq = platform_get_irq(pdev, 0); - if (adata->i2s_irq < 0) - return -ENODEV; + status = platform_get_irq(pdev, 0); + if (status < 0) + return status; + adata->i2s_irq = status; dev_set_drvdata(&pdev->dev, adata); status = devm_snd_soc_register_component(&pdev->dev, -- 2.20.1

2 7

[PATCH] ASoC: cs42l42: Add warnings about DETECT_MODE and PLL_START
by Richard Fitzgerald 07 Mar '22

07 Mar '22

DETECT_MODE and PLL_START must be zero while HP_PDN and ADC_PDN are both 1. If this condition is broken it can discharge FILT+ and it can then take up to 1 second for FILT+ to recharge. There is no workaround required for this, simply avoiding settings and sequences that would break the requirement. The driver already meets the requirement. But it is not obvious from reading the code that this requirement exists, or what is ensuring it is met. So it would not currently be obvious to someone changing the code that there is certain special behaviour that must be maintained. To avoid accidental breakage in the future: - Add comments into the register definitions to warn about this so that anyone changing the code around DETECT_MODE and PLL_START is aware of this requirement. - Add a comment where PLL_START is written to 1 to highlight the requirement and why it is satisfied. - Add a comment in cs42l42_setup_hs_type_detect() when DETECT_MODE is initialized. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/cs42l42.c | 13 ++++++++++++- sound/soc/codecs/cs42l42.h | 9 ++++++++- 2 files changed, 20 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42l42.c b/sound/soc/codecs/cs42l42.c index db6ef6cdce15..c8409d50e934 100644 --- a/sound/soc/codecs/cs42l42.c +++ b/sound/soc/codecs/cs42l42.c @@ -1012,7 +1012,14 @@ static int cs42l42_mute_stream(struct snd_soc_dai *dai, int mute, int stream) } } else { if (!cs42l42->stream_use) { - /* SCLK must be running before codec unmute */ + /* SCLK must be running before codec unmute. + * + * PLL must not be started with ADC and HP both off + * otherwise the FILT+ supply will not charge properly. + * DAPM widgets power-up before stream unmute so at least + * one of the "DAC" or "ADC" widgets will already have + * powered-up. + */ if (pll_ratio_table[cs42l42->pll_config].mclk_src_sel) { snd_soc_component_update_bits(component, CS42L42_PLL_CTL1, CS42L42_PLL_START_MASK, 1); @@ -1830,6 +1837,10 @@ static void cs42l42_setup_hs_type_detect(struct cs42l42_private *cs42l42) cs42l42->hs_type = CS42L42_PLUG_INVALID; + /* + * DETECT_MODE must always be 0 with ADC and HP both off otherwise the + * FILT+ supply will not charge properly. + */ regmap_update_bits(cs42l42->regmap, CS42L42_MISC_DET_CTL, CS42L42_DETECT_MODE_MASK, 0); diff --git a/sound/soc/codecs/cs42l42.h b/sound/soc/codecs/cs42l42.h index 244b24d1f5e9..60d3bdf5d7c9 100644 --- a/sound/soc/codecs/cs42l42.h +++ b/sound/soc/codecs/cs42l42.h @@ -491,7 +491,10 @@ #define CS42L42_TS_UNPLUG 0 #define CS42L42_TS_TRANS 1 -/* Page 0x15 Fractional-N PLL Registers */ +/* + * NOTE: PLL_START must be 0 while both ADC_PDN=1 and HP_PDN=1. + * Otherwise it will prevent FILT+ from charging properly. + */ #define CS42L42_PLL_CTL1 (CS42L42_PAGE_15 + 0x01) #define CS42L42_PLL_START_SHIFT 0 #define CS42L42_PLL_START_MASK (1 << CS42L42_PLL_START_SHIFT) @@ -574,6 +577,10 @@ #define CS42L42_TIP_SENSE_CTRL_MASK (3 << \ CS42L42_TIP_SENSE_CTRL_SHIFT) +/* + * NOTE: DETECT_MODE must be 0 while both ADC_PDN=1 and HP_PDN=1. + * Otherwise it will prevent FILT+ from charging properly. + */ #define CS42L42_MISC_DET_CTL (CS42L42_PAGE_1B + 0x74) #define CS42L42_PDN_MIC_LVL_DET_SHIFT 0 #define CS42L42_PDN_MIC_LVL_DET_MASK (1 << CS42L42_PDN_MIC_LVL_DET_SHIFT) -- 2.30.2

2 1

[PATCH v2] ASoC: wm8350: Handle error for wm8350_register_irq
by Jiasheng Jiang 07 Mar '22

07 Mar '22

As the potential failure of the wm8350_register_irq(), it should be better to check it and return error if fails. Also, use 'free_' in order to avoid the same code. Fixes: a6ba2b2dabb5 ("ASoC: Implement WM8350 headphone jack detection") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- Changelog: v1 -> v2 * Change 1. Change to non-caps labels and remove the revert to 'return ret;'. --- sound/soc/codecs/wm8350.c | 28 ++++++++++++++++++++++++---- 1 file changed, 24 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 15d42ce3b21d..41504ce2a682 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1537,18 +1537,38 @@ static int wm8350_component_probe(struct snd_soc_component *component) wm8350_clear_bits(wm8350, WM8350_JACK_DETECT, WM8350_JDL_ENA | WM8350_JDR_ENA); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, wm8350_hpl_jack_handler, 0, "Left jack detect", priv); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, + if (ret != 0) + goto err; + + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, wm8350_hpr_jack_handler, 0, "Right jack detect", priv); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, + if (ret != 0) + goto free_jck_det_l; + + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, wm8350_mic_handler, 0, "Microphone short", priv); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD, + if (ret != 0) + goto free_jck_det_r; + + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD, wm8350_mic_handler, 0, "Microphone detect", priv); + if (ret != 0) + goto free_micscd; return 0; + +free_micscd: + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, priv); +free_jck_det_r: + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, priv); +free_jck_det_l: + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, priv); +err: + return ret; } static void wm8350_component_remove(struct snd_soc_component *component) -- 2.25.1

3 2

[PATCH v2 0/2] ASoC: codecs: Add Awinic AW8738 audio amplifier driver
by Stephan Gerhold 07 Mar '22

07 Mar '22

This series adds a simple driver and DT schema for the Awinic AW8738 audio amplifier. It's fairly simple - the main difference to simple-amplifier is that there is a "one-wire pulse control" that allows configuring the amplifier to one of a few pre-defined modes. This can be used to configure the speaker-guard function (primarily the power limit for the amplifier). Jonathan Albrieux (1): ASoC: codecs: Add Awinic AW8738 audio amplifier driver Stephan Gerhold (1): ASoC: dt-bindings: Add schema for "awinic,aw8738" .../bindings/sound/awinic,aw8738.yaml | 54 +++++++++ sound/soc/codecs/Kconfig | 10 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/aw8738.c | 104 ++++++++++++++++++ 4 files changed, 170 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/awinic,aw8738.yaml create mode 100644 sound/soc/codecs/aw8738.c -- 2.35.1

2 3

[PATCH] ASoC: wm_adsp: Expand firmware loading search options
by Simon Trimmer 07 Mar '22

07 Mar '22

The parts supported by this driver can have product-specific firmware and tunings files. Typically these have been used on embedded systems where the manufacturer is responsible for installing the correct product-specific firmware files into /lib/firmware. However, the linux-firmware repository places all available firmwares into /lib/firmware and it is up to the driver to select the correct product-specific firmware from that directory. For example a product containing four smart amplifiers may provide firmware specific for that product and each of the amplifiers may have coefficient files containing tunings for their placement in the mechanical design. This change extends firmware (wmfw) and coefficient (bin) filenames to be of the general form: <cirrus/>part-dspN-fwtype<-system_name<-asoc_component_prefix>>.type Where the cirrus subdirectory, system_name and asoc_component_prefix are optional. New files will be placed in the cirrus subdirectory to avoid polluting the main /lib/firmware/ location. The generic name must be searched in /lib/firmware before /lib/firmware/cirrus so that a generic file in the new location does not override existing product-specific files in the legacy location. The search order for firmware files is: - cirrus/part-dspN-fwtype-system_name-asoc_component_prefix.wmfw - cirrus/part-dspN-fwtype-system_name.wmfw - part-dspN-fwtype.wmfw - cirrus/part-dspN-fwtype.wmfw - Qualifications are added to the filename so that rightwards is more specific. - The system_name is provided by the codec driver. - The asoc_component_prefix is used to identify tunings for individual parts because it would already exist to disambiguate the controls and it makes it obvious which firmware file applies to which device. The optional coefficient file must have the same filename construction as the discovered wmfw except: - where the wmfw has only system_name then the bin file can optionally include the asoc_component_prefix. This is to allow a common wmfw for all amps but separate tunings per amp. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 99 +++++++++++++++++++++++++++++++++----- sound/soc/codecs/wm_adsp.h | 1 + 2 files changed, 88 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 0582585236a2..5ba88c5272d0 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -744,21 +744,48 @@ static void wm_adsp_release_firmware_files(struct wm_adsp *dsp, } static int wm_adsp_request_firmware_file(struct wm_adsp *dsp, - const struct firmware **firmware, - char **filename, - char *suffix) + const struct firmware **firmware, char **filename, + const char *dir, const char *system_name, + const char *asoc_component_prefix, + const char *filetype) { struct cs_dsp *cs_dsp = &dsp->cs_dsp; + char *s, c; int ret = 0; - *filename = kasprintf(GFP_KERNEL, "%s-%s-%s.%s", dsp->part, dsp->fwf_name, - wm_adsp_fw[dsp->fw].file, suffix); + if (system_name && asoc_component_prefix) + *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s-%s.%s", dir, dsp->part, + dsp->fwf_name, wm_adsp_fw[dsp->fw].file, system_name, + asoc_component_prefix, filetype); + else if (system_name) + *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s-%s.%s", dir, dsp->part, + dsp->fwf_name, wm_adsp_fw[dsp->fw].file, system_name, + filetype); + else + *filename = kasprintf(GFP_KERNEL, "%s%s-%s-%s.%s", dir, dsp->part, dsp->fwf_name, + wm_adsp_fw[dsp->fw].file, filetype); + if (*filename == NULL) return -ENOMEM; - ret = request_firmware(firmware, *filename, cs_dsp->dev); + /* + * Make sure that filename is lower-case and any non alpha-numeric + * characters except full stop and forward slash are replaced with + * hyphens. + */ + s = *filename; + while (*s) { + c = *s; + if (isalnum(c)) + *s = tolower(c); + else if ((c != '.') && (c != '/')) + *s = '-'; + s++; + } + + ret = firmware_request_nowarn(firmware, *filename, cs_dsp->dev); if (ret != 0) { - adsp_err(dsp, "Failed to request '%s'\n", *filename); + adsp_dbg(dsp, "Failed to request '%s'\n", *filename); kfree(*filename); *filename = NULL; } @@ -766,21 +793,69 @@ static int wm_adsp_request_firmware_file(struct wm_adsp *dsp, return ret; } +static const char *cirrus_dir = "cirrus/"; static int wm_adsp_request_firmware_files(struct wm_adsp *dsp, const struct firmware **wmfw_firmware, char **wmfw_filename, const struct firmware **coeff_firmware, char **coeff_filename) { + const char *system_name = dsp->system_name; + const char *asoc_component_prefix = dsp->component->name_prefix; int ret = 0; - ret = wm_adsp_request_firmware_file(dsp, wmfw_firmware, wmfw_filename, "wmfw"); - if (ret != 0) - return ret; + if (system_name && asoc_component_prefix) { + if (!wm_adsp_request_firmware_file(dsp, wmfw_firmware, wmfw_filename, + cirrus_dir, system_name, + asoc_component_prefix, "wmfw")) { + adsp_dbg(dsp, "Found '%s'\n", *wmfw_filename); + wm_adsp_request_firmware_file(dsp, coeff_firmware, coeff_filename, + cirrus_dir, system_name, + asoc_component_prefix, "bin"); + return 0; + } + } - wm_adsp_request_firmware_file(dsp, coeff_firmware, coeff_filename, "bin"); + if (system_name) { + if (!wm_adsp_request_firmware_file(dsp, wmfw_firmware, wmfw_filename, + cirrus_dir, system_name, + NULL, "wmfw")) { + adsp_dbg(dsp, "Found '%s'\n", *wmfw_filename); + if (asoc_component_prefix) + wm_adsp_request_firmware_file(dsp, coeff_firmware, coeff_filename, + cirrus_dir, system_name, + asoc_component_prefix, "bin"); + + if (!*coeff_firmware) + wm_adsp_request_firmware_file(dsp, coeff_firmware, coeff_filename, + cirrus_dir, system_name, + NULL, "bin"); + return 0; + } + } - return 0; + if (!wm_adsp_request_firmware_file(dsp, wmfw_firmware, wmfw_filename, + "", NULL, NULL, "wmfw")) { + adsp_dbg(dsp, "Found '%s'\n", *wmfw_filename); + wm_adsp_request_firmware_file(dsp, coeff_firmware, coeff_filename, + "", NULL, NULL, "bin"); + return 0; + } + + ret = wm_adsp_request_firmware_file(dsp, wmfw_firmware, wmfw_filename, + cirrus_dir, NULL, NULL, "wmfw"); + if (!ret) { + adsp_dbg(dsp, "Found '%s'\n", *wmfw_filename); + wm_adsp_request_firmware_file(dsp, coeff_firmware, coeff_filename, + cirrus_dir, NULL, NULL, "bin"); + return 0; + } + + adsp_err(dsp, "Failed to request firmware <%s>%s-%s-%s<-%s<%s>>.wmfw\n", + cirrus_dir, dsp->part, dsp->fwf_name, wm_adsp_fw[dsp->fw].file, + system_name, asoc_component_prefix); + + return -ENOENT; } static int wm_adsp_common_init(struct wm_adsp *dsp) diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index 7f4fabbc6ad3..375009a65828 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -28,6 +28,7 @@ struct wm_adsp { struct cs_dsp cs_dsp; const char *part; const char *fwf_name; + const char *system_name; struct snd_soc_component *component; unsigned int sys_config_size; -- 2.34.1

3 2

[PATCH v3] ASoC: fsi: Add check for clk_enable
by Jiasheng Jiang 07 Mar '22

07 Mar '22

As the potential failure of the clk_enable(), it should be better to check it and return error if fails. Fixes: ab6f6d85210c ("ASoC: fsi: add master clock control functions") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- Changelog: v1 -> v2 * Change 1. Seperate the error handler. v2 -> v3 * Change 1. Revert disable_xck and disable_ick. --- sound/soc/sh/fsi.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index cdf3b7f69ba7..e9a1eb6bdf66 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -816,14 +816,27 @@ static int fsi_clk_enable(struct device *dev, return ret; } - clk_enable(clock->xck); - clk_enable(clock->ick); - clk_enable(clock->div); + ret = clk_enable(clock->xck); + if (ret) + goto err; + ret = clk_enable(clock->ick); + if (ret) + goto disable_xck; + ret = clk_enable(clock->div); + if (ret) + goto disable_ick; clock->count++; } return ret; + +disable_ick: + clk_disable(clock->ick); +disable_xck: + clk_disable(clock->xck); +err: + return ret; } static int fsi_clk_disable(struct device *dev, -- 2.25.1

2 1

[PATCH 0/7] ASoC: fsl_sai: Cleanups and 1:1 bclk:mclk ratio support
by Sascha Hauer 07 Mar '22

07 Mar '22

This series has some updates for the fsl_sai driver: Some general cleanup patches, a bugfix in the ip revision checking and finally the mclk setting is made more accurate and support for 1:1 bclk:mclk setting is added. Ahmad Fatoum (2): ASoC: fsl_sai: simplify register poking in fsl_sai_set_bclk ASoC: fsl_sai: implement 1:1 bclk:mclk ratio support Sascha Hauer (5): ASoC: fsl_sai: Drop unnecessary defines ASoC: fsl_sai: simplify irq return value ASoC: fsl_sai: store full version instead of major/minor ASoC: fsl_sai: Use better variable names ASoC: fsl_sai: use DIV_ROUND_CLOSEST() to calculate divider sound/soc/fsl/fsl_sai.c | 100 ++++++++++++++++++++-------------------- sound/soc/fsl/fsl_sai.h | 9 +--- 2 files changed, 52 insertions(+), 57 deletions(-) -- 2.30.2

3 9

[PATCH V3 0/2] ASoC: audio_graph_card2: Support variable slot widths
by Richard Fitzgerald 07 Mar '22

07 Mar '22

This adds support for I2S/TDM links where the slot width varies depending on the sample width, in a way that cannot be guessed by component hw_params(). A typical example is: - 16-bit samples use 16-bit slots - 24-bit samples use 32-bit slots There is no way for a component hw_params() to deduce from the information it is passed that 24-bit samples will be in 32-bit slots. Some audio hardware cannot support a fixed slot width or a slot width equal to the sample width in all cases. This is usually due either to limitations of the audio serial port or system clocking restrictions. These two patches add support for defining a mapping between sample widths and sample slots. This allows audio_graph_card2 to be used in these situations instead of having to write a custom machine driver. Changes since V2: - Change property type to uint32-matrix. - Put min/max ranges on cells. - Use per-item description to document the tuple entries. - Add a slot count to the tuple - it's unlikely that this would change for different sample rates but it's trivial to implement support for it. - Minor tidying of the code in asoc_simple_parse_tdm_width_map() and asoc_simple_set_tdm(). Richard Fitzgerald (2): ASoC: dt-bindings: audio-graph-port: Add dai-tdm-slot-width-map ASoC: audio_graph_card2: Add support for variable slot widths .../bindings/sound/audio-graph-port.yaml | 20 ++++ include/sound/simple_card_utils.h | 11 +++ sound/soc/generic/audio-graph-card2.c | 4 + sound/soc/generic/simple-card-utils.c | 97 +++++++++++++++++++ 4 files changed, 132 insertions(+) -- 2.30.2

2 3

[PATCH V2 0/2] ASoC: audio_graph_card2: Support variable slot widths
by Richard Fitzgerald 07 Mar '22

07 Mar '22

This adds support for I2S/TDM links where the slot width varies depending on the sample width, in a way that cannot be guessed by component hw_params(). A typical example is: - 16-bit samples use 16-bit slots - 24-bit samples use 32-bit slots There is no way for a component hw_params() to deduce from the information it is passed that 24-bit samples will be in 32-bit slots. Some audio hardware cannot support a fixed slot width or a slot width equal to the sample width in all cases. This is usually due either to limitations of the audio serial port or system clocking restrictions. These two patches add support for defining a mapping between sample widths and sample slots. This allows audio_graph_card2 to be used in these situations instead of having to write a custom machine driver. Richard Fitzgerald (2): ASoC: dt-bindings: audio-graph-port: Add dai-tdm-slot-width-map ASoC: audio_graph_card2: Add support for variable slot widths .../bindings/sound/audio-graph-port.yaml | 7 ++ include/sound/simple_card_utils.h | 10 ++ sound/soc/generic/audio-graph-card2.c | 4 + sound/soc/generic/simple-card-utils.c | 93 +++++++++++++++++++ 4 files changed, 114 insertions(+) -- 2.30.2

3 6

[PATCH v2 00/20] Support external boost at CS35l41 ASoC driver
by Lucas Tanure 07 Mar '22

07 Mar '22

Move the support for CS35L41 external boost to its shared library for ASoC use. This move resulted in cs35l41_hda_reg_sequence being removed, and its steps were broken down into regmap writes or functions from the library. And hardware configuration struct was unified for its use in the shared lib. While at it, some minor bugs were found and fixed it. v2 changelog: - Instead of removing the log, playback actions will log the last regmap access. - Documentation patch with the correct subject line and fixed bug reported by Rob Herring on the provided example. Previous versions: v1: https://lkml.org/lkml/2022/3/3/759 David Rhodes (2): ASoC: cs35l41: Fix GPIO2 configuration ASoC: cs35l41: Document CS35l41 External Boost Lucas Tanure (18): ASoC: cs35l41: Fix max number of TX channels ASoC: cs35l41: Fix DSP mbox start command and global enable order ASoC: cs35l41: Remove unnecessary param sound: cs35l41: Unify hardware configuration sound: cs35l41: Check hw_config before using it sound: cs35l41: Move cs35l41_gpio_config to shared lib hda: cs35l41: Fix I2S params comments hda: cs35l41: Always configure the DAI hda: cs35l41: Add Boost type flag hda: cs35l41: Put the device into safe mode for external boost hda: cs35l41: Mute the device before shutdown sound: cs35l41: Enable Internal Boost in shared lib hda: cs35l41: Move boost config to initialization code hda: cs35l41: Remove cs35l41_hda_reg_sequence struct hda: cs35l41: Reorganize log for playback actions hda: cs35l41: Handle all external boost setups the same way hda: cs35l41: Move external boost handling to lib for ASoC use ASoC: cs35l41: Support external boost .../bindings/sound/cirrus,cs35l41.yaml | 44 ++- include/sound/cs35l41.h | 53 +++- sound/pci/hda/cs35l41_hda.c | 295 ++++++------------ sound/pci/hda/cs35l41_hda.h | 27 +- sound/soc/codecs/cs35l41-i2c.c | 4 +- sound/soc/codecs/cs35l41-lib.c | 190 ++++++++++- sound/soc/codecs/cs35l41-spi.c | 4 +- sound/soc/codecs/cs35l41.c | 174 +++++------ sound/soc/codecs/cs35l41.h | 5 +- 9 files changed, 440 insertions(+), 356 deletions(-) -- 2.35.1

5 25

[PATCH 00/20] Support external boost at CS35l41 ASoC driver
by Lucas Tanure 07 Mar '22

07 Mar '22

Move the support for CS35L41 external boost to its shared library for ASoC use. This move resulted in cs35l41_hda_reg_sequence being removed, and its steps were broken down into regmap writes or functions from the library. And hardware configuration struct was unified for its use in the shared lib. While at it, some minor bugs were found and fixed it. David Rhodes (2): ASoC: cs35l41: Fix GPIO2 configuration Documentation: devicetree: CS35l41 External Boost Lucas Tanure (18): ASoC: cs35l41: Fix max number of TX channels ASoC: cs35l41: Fix DSP mbox start command and global enable order ASoC: cs35l41: Remove unnecessary param sound: cs35l41: Unify hardware configuration sound: cs35l41: Check hw_config before using it sound: cs35l41: Move cs35l41_gpio_config to shared lib hda: cs35l41: Fix I2S params comments hda: cs35l41: Always configure the DAI hda: cs35l41: Add Boost type flag hda: cs35l41: Put the device into safe mode for external boost hda: cs35l41: Mute the device before shutdown sound: cs35l41: Enable Internal Boost in shared lib hda: cs35l41: Move boost config to initialization code hda: cs35l41: Remove unnecessary log hda: cs35l41: Remove cs35l41_hda_reg_sequence struct hda: cs35l41: Handle all external boost setups the same way hda: cs35l41: Move external boost handling to lib for ASoC use ASoC: cs35l41: Support external boost .../bindings/sound/cirrus,cs35l41.yaml | 42 ++- include/sound/cs35l41.h | 53 +++- sound/pci/hda/cs35l41_hda.c | 296 +++++------------- sound/pci/hda/cs35l41_hda.h | 27 +- sound/soc/codecs/cs35l41-i2c.c | 4 +- sound/soc/codecs/cs35l41-lib.c | 185 ++++++++++- sound/soc/codecs/cs35l41-spi.c | 4 +- sound/soc/codecs/cs35l41.c | 174 +++++----- sound/soc/codecs/cs35l41.h | 5 +- 9 files changed, 431 insertions(+), 359 deletions(-) -- 2.35.1

4 33

[PATCH 0/1] ALSA: virtio: add support for audio controls
by Anton Yakovlev 07 Mar '22

07 Mar '22

Implementation of support for audio controls in accordance with the proposed extension of the virtio sound device specification [1]. Design　of virtual audio controls is based on and derived from ALSA audio controls. It allows the driver to perform all standard operations, such as reading and writing audio control value, as well as working with metadata (represented in the TLV form). The driver part was tested on top of the Linux 5.10 kernel. As a device part was used OpenSynergy proprietary implementation. [1] https://lists.oasis-open.org/archives/virtio-comment/202104/msg00013.html Anton Yakovlev (1): ALSA: virtio: add support for audio controls include/uapi/linux/virtio_snd.h | 154 +++++++++++ sound/virtio/Makefile | 1 + sound/virtio/virtio_card.c | 21 ++ sound/virtio/virtio_card.h | 22 ++ sound/virtio/virtio_kctl.c | 464 ++++++++++++++++++++++++++++++++ 5 files changed, 662 insertions(+) create mode 100644 sound/virtio/virtio_kctl.c -- 2.35.1

1 1

[PATCH 0/6] Remove usage of list iterator past the loop body
by Jakob Koschel 07 Mar '22

07 Mar '22

This is the first patch removing several categories of use cases of the list iterator variable past the loop. This is follow up to the discussion in: https://lore.kernel.org/all/20220217184829.1991035-1-jakobkoschel@gmail.com/ As concluded in: https://lore.kernel.org/all/YhdfEIwI4EdtHdym@kroah.com/ the correct use should be using a separate variable after the loop and using a 'tmp' variable as the list iterator. The list iterator will not point to a valid structure after the loop if no break/goto was hit. Invalid uses of the list iterator variable can be avoided altogether by simply using a separate pointer to iterate the list. Linus and Greg agreed on the following pattern: - struct gr_request *req; + struct gr_request *req = NULL; + struct gr_request *tmp; struct gr_ep *ep; int ret = 0; - list_for_each_entry(req, &ep->queue, queue) { - if (&req->req == _req) + list_for_each_entry(tmp, &ep->queue, queue) { + if (&tmp->req == _req) { + req = tmp; break; + } } - if (&req->req != _req) { + if (!req) { ret = -EINVAL; goto out; } With gnu89 the list iterator variable cannot yet be declared within the for loop of the list iterator. Moving to a more modern version of C would allow defining the list iterator variable within the macro, limiting the scope to the loop. This avoids any incorrect usage past the loop altogether. This are around 30% of the cases where the iterator variable is used past the loop (identified with a slightly modified version of use_after_iter.cocci). I've decided to split it into at a few patches separated by similar use cases. Because the output of get_maintainer.pl was too big, I included all the found lists and everyone from the previous discussion. Jakob Koschel (6): drivers: usb: remove usage of list iterator past the loop body treewide: remove using list iterator after loop body as a ptr treewide: fix incorrect use to determine if list is empty drivers: remove unnecessary use of list iterator variable treewide: remove dereference of list iterator after loop body treewide: remove check of list iterator against head past the loop body arch/arm/mach-mmp/sram.c | 9 ++-- arch/arm/plat-pxa/ssp.c | 28 +++++------- arch/powerpc/sysdev/fsl_gtm.c | 4 +- arch/x86/kernel/cpu/sgx/encl.c | 6 ++- drivers/block/drbd/drbd_req.c | 45 ++++++++++++------- drivers/counter/counter-chrdev.c | 26 ++++++----- drivers/crypto/cavium/nitrox/nitrox_main.c | 11 +++-- drivers/dma/dw-edma/dw-edma-core.c | 4 +- drivers/dma/ppc4xx/adma.c | 11 +++-- drivers/firewire/core-transaction.c | 32 +++++++------ drivers/firewire/sbp2.c | 14 +++--- drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c | 19 +++++--- drivers/gpu/drm/drm_memory.c | 15 ++++--- drivers/gpu/drm/drm_mm.c | 17 ++++--- drivers/gpu/drm/drm_vm.c | 13 +++--- drivers/gpu/drm/gma500/oaktrail_lvds.c | 9 ++-- drivers/gpu/drm/i915/gem/i915_gem_context.c | 14 +++--- .../gpu/drm/i915/gem/i915_gem_execbuffer.c | 15 ++++--- drivers/gpu/drm/i915/gt/intel_ring.c | 15 ++++--- .../gpu/drm/nouveau/nvkm/subdev/clk/base.c | 11 +++-- .../gpu/drm/nouveau/nvkm/subdev/fb/ramgk104.c | 13 +++--- drivers/gpu/drm/scheduler/sched_main.c | 14 +++--- drivers/gpu/drm/ttm/ttm_bo.c | 19 ++++---- drivers/gpu/drm/vmwgfx/vmwgfx_kms.c | 22 +++++---- drivers/infiniband/hw/hfi1/tid_rdma.c | 16 ++++--- drivers/infiniband/hw/mlx4/main.c | 12 ++--- drivers/media/dvb-frontends/mxl5xx.c | 11 +++-- drivers/media/pci/saa7134/saa7134-alsa.c | 4 +- drivers/media/v4l2-core/v4l2-ctrls-api.c | 31 +++++++------ drivers/misc/mei/interrupt.c | 12 ++--- .../net/ethernet/intel/i40e/i40e_ethtool.c | 3 +- .../net/ethernet/qlogic/qede/qede_filter.c | 11 +++-- drivers/net/wireless/ath/ath6kl/htc_mbox.c | 2 +- .../net/wireless/intel/ipw2x00/libipw_rx.c | 15 ++++--- drivers/perf/xgene_pmu.c | 13 +++--- drivers/power/supply/cpcap-battery.c | 11 +++-- drivers/scsi/lpfc/lpfc_bsg.c | 16 ++++--- drivers/scsi/scsi_transport_sas.c | 17 ++++--- drivers/scsi/wd719x.c | 12 +++-- drivers/staging/rtl8192e/rtl819x_TSProc.c | 17 +++---- drivers/staging/rtl8192e/rtllib_rx.c | 17 ++++--- .../staging/rtl8192u/ieee80211/ieee80211_rx.c | 15 ++++--- .../rtl8192u/ieee80211/rtl819x_TSProc.c | 19 ++++---- drivers/thermal/thermal_core.c | 38 ++++++++++------ drivers/usb/gadget/composite.c | 9 ++-- drivers/usb/gadget/configfs.c | 22 +++++---- drivers/usb/gadget/udc/aspeed-vhub/epn.c | 11 +++-- drivers/usb/gadget/udc/at91_udc.c | 26 ++++++----- drivers/usb/gadget/udc/atmel_usba_udc.c | 11 +++-- drivers/usb/gadget/udc/bdc/bdc_ep.c | 11 +++-- drivers/usb/gadget/udc/fsl_qe_udc.c | 11 +++-- drivers/usb/gadget/udc/fsl_udc_core.c | 11 +++-- drivers/usb/gadget/udc/goku_udc.c | 11 +++-- drivers/usb/gadget/udc/gr_udc.c | 11 +++-- drivers/usb/gadget/udc/lpc32xx_udc.c | 11 +++-- drivers/usb/gadget/udc/max3420_udc.c | 11 +++-- drivers/usb/gadget/udc/mv_u3d_core.c | 11 +++-- drivers/usb/gadget/udc/mv_udc_core.c | 11 +++-- drivers/usb/gadget/udc/net2272.c | 12 ++--- drivers/usb/gadget/udc/net2280.c | 11 +++-- drivers/usb/gadget/udc/omap_udc.c | 11 +++-- drivers/usb/gadget/udc/pxa25x_udc.c | 11 +++-- drivers/usb/gadget/udc/s3c-hsudc.c | 11 +++-- drivers/usb/gadget/udc/tegra-xudc.c | 11 +++-- drivers/usb/gadget/udc/udc-xilinx.c | 11 +++-- drivers/usb/mtu3/mtu3_gadget.c | 11 +++-- drivers/usb/musb/musb_gadget.c | 11 +++-- drivers/vfio/mdev/mdev_core.c | 11 +++-- fs/cifs/smb2misc.c | 10 +++-- fs/f2fs/segment.c | 9 ++-- fs/proc/kcore.c | 13 +++--- kernel/debug/kdb/kdb_main.c | 36 +++++++++------ kernel/power/snapshot.c | 10 +++-- kernel/trace/ftrace.c | 22 +++++---- kernel/trace/trace_eprobe.c | 15 ++++--- kernel/trace/trace_events.c | 11 ++--- net/9p/trans_xen.c | 11 +++-- net/ipv4/udp_tunnel_nic.c | 10 +++-- net/tipc/name_table.c | 11 +++-- net/tipc/socket.c | 11 +++-- net/xfrm/xfrm_ipcomp.c | 11 +++-- sound/soc/intel/catpt/pcm.c | 13 +++--- sound/soc/sprd/sprd-mcdt.c | 13 +++--- 83 files changed, 708 insertions(+), 465 deletions(-) base-commit: 7ee022567bf9e2e0b3cd92461a2f4986ecc99673 -- 2.25.1

9 23

[PATCH v2] dt-bindings: mfd: Fix pinctrl node name warnings
by Rob Herring 07 Mar '22

07 Mar '22

The recent addition pinctrl.yaml in commit c09acbc499e8 ("dt-bindings: pinctrl: use pinctrl.yaml") resulted in some node name warnings: Documentation/devicetree/bindings/mfd/cirrus,lochnagar.example.dt.yaml: \ lochnagar-pinctrl: $nodename:0: 'lochnagar-pinctrl' does not match '^(pinctrl|pinmux)(@[0-9a-f]+)?$' Documentation/devicetree/bindings/mfd/cirrus,madera.example.dt.yaml: \ codec@1a: $nodename:0: 'codec@1a' does not match '^(pinctrl|pinmux)(@[0-9a-f]+)?$' Documentation/devicetree/bindings/mfd/brcm,cru.example.dt.yaml: \ pin-controller@1c0: $nodename:0: 'pin-controller@1c0' does not match '^(pinctrl|pinmux)(@[0-9a-f]+)?$' Fix the node names to the preferred 'pinctrl'. For cirrus,madera, nothing from pinctrl.yaml schema is used, so just drop the reference. Fixes: c09acbc499e8 ("dt-bindings: pinctrl: use pinctrl.yaml") Cc: Rafał Miłecki <rafal(a)milecki.pl> Signed-off-by: Rob Herring <robh(a)kernel.org> --- v2: - Fix lochnagar-pinctrl nodename in example Documentation/devicetree/bindings/mfd/brcm,cru.yaml | 4 ++-- Documentation/devicetree/bindings/mfd/cirrus,lochnagar.yaml | 4 ++-- Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml | 3 --- 3 files changed, 4 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/mfd/brcm,cru.yaml b/Documentation/devicetree/bindings/mfd/brcm,cru.yaml index be4a2df71c25..b85819fbb07c 100644 --- a/Documentation/devicetree/bindings/mfd/brcm,cru.yaml +++ b/Documentation/devicetree/bindings/mfd/brcm,cru.yaml @@ -39,7 +39,7 @@ patternProperties: '^phy@[a-f0-9]+$': $ref: ../phy/bcm-ns-usb2-phy.yaml - '^pin-controller@[a-f0-9]+$': + '^pinctrl@[a-f0-9]+$': $ref: ../pinctrl/brcm,ns-pinmux.yaml '^syscon@[a-f0-9]+$': @@ -94,7 +94,7 @@ examples: reg = <0x180 0x4>; }; - pin-controller@1c0 { + pinctrl@1c0 { compatible = "brcm,bcm4708-pinmux"; reg = <0x1c0 0x24>; reg-names = "cru_gpio_control"; diff --git a/Documentation/devicetree/bindings/mfd/cirrus,lochnagar.yaml b/Documentation/devicetree/bindings/mfd/cirrus,lochnagar.yaml index c00ad3e21c21..a04570335b76 100644 --- a/Documentation/devicetree/bindings/mfd/cirrus,lochnagar.yaml +++ b/Documentation/devicetree/bindings/mfd/cirrus,lochnagar.yaml @@ -126,7 +126,7 @@ properties: clock-frequency: const: 12288000 - lochnagar-pinctrl: + pinctrl: type: object $ref: /schemas/pinctrl/cirrus,lochnagar.yaml# @@ -293,7 +293,7 @@ examples: clock-frequency = <32768>; }; - lochnagar-pinctrl { + pinctrl { compatible = "cirrus,lochnagar-pinctrl"; gpio-controller; diff --git a/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml b/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml index c85f759ae5a3..8a90d8273767 100644 --- a/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml +++ b/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml @@ -107,9 +107,6 @@ properties: additionalProperties: false -allOf: - - $ref: "pinctrl.yaml#" - required: - pinctrl-0 - pinctrl-names -- 2.32.0

2 1

[PATCH] ASoC: intel: use asoc_substream_to_rtd()
by zhanglianjie 07 Mar '22

07 Mar '22

Now we can use asoc_substream_to_rtd() macro, let's use it. Signed-off-by: zhanglianjie <zhanglianjie(a)uniontech.com> diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c index 939a9b801dec..a26000cd5ceb 100644 --- a/sound/soc/intel/catpt/pcm.c +++ b/sound/soc/intel/catpt/pcm.c @@ -74,7 +74,7 @@ static struct catpt_stream_template *catpt_topology[] = { static struct catpt_stream_template * catpt_get_stream_template(struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtm = substream->private_data; + struct snd_soc_pcm_runtime *rtm = asoc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtm, 0); enum catpt_stream_type type; @@ -593,7 +593,7 @@ static int catpt_component_pcm_construct(struct snd_soc_component *component, static int catpt_component_open(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtm = substream->private_data; + struct snd_soc_pcm_runtime *rtm = asoc_substream_to_rtd(substream); if (!rtm->dai_link->no_pcm) snd_soc_set_runtime_hwparams(substream, &catpt_pcm_hardware); @@ -604,7 +604,7 @@ static snd_pcm_uframes_t catpt_component_pointer(struct snd_soc_component *component, struct snd_pcm_substream *substream) { - struct snd_soc_pcm_runtime *rtm = substream->private_data; + struct snd_soc_pcm_runtime *rtm = asoc_substream_to_rtd(substream); struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtm, 0); struct catpt_stream_runtime *stream; struct catpt_dev *cdev = dev_get_drvdata(component->dev); -- 2.20.1

3 2

[PATCH] ASoC: simple-card: add bindings for amplifier support
by Alifer Moraes 07 Mar '22

07 Mar '22

From: Pierluigi Passaro <pierluigi.p(a)variscite.com> Add an optional property 'pa-gpios' for controlling external amplifier, which depends on DAPM widgets and routing. Signed-off-by: Pierluigi Passaro <pierluigi.p(a)variscite.com> Signed-off-by: Alifer Moraes <alifer.m(a)variscite.com> --- sound/soc/generic/simple-card.c | 37 +++++++++++++++++++++++++++++++++ 1 file changed, 37 insertions(+) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index a89d1cfdda32..d042be556f1b 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -10,6 +10,7 @@ #include <linux/module.h> #include <linux/of.h> #include <linux/of_device.h> +#include <linux/of_gpio.h> #include <linux/platform_device.h> #include <linux/string.h> #include <sound/simple_card.h> @@ -22,6 +23,33 @@ #define CELL "#sound-dai-cells" #define PREFIX "simple-audio-card," +static int simple_outdrv_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct asoc_simple_priv *priv = snd_soc_card_get_drvdata(dapm->card); + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + gpiod_set_value_cansleep(priv->pa_gpio, 1); + break; + case SND_SOC_DAPM_PRE_PMD: + gpiod_set_value_cansleep(priv->pa_gpio, 0); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const struct snd_soc_dapm_widget simple_dapm_widgets[] = { + SND_SOC_DAPM_OUT_DRV_E("Amplifier", SND_SOC_NOPM, + 0, 0, NULL, 0, simple_outdrv_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +}; + static const struct snd_soc_ops simple_ops = { .startup = asoc_simple_startup, .shutdown = asoc_simple_shutdown, @@ -620,6 +648,8 @@ static int asoc_simple_probe(struct platform_device *pdev) card = simple_priv_to_card(priv); card->owner = THIS_MODULE; card->dev = dev; + card->dapm_widgets = simple_dapm_widgets; + card->num_dapm_widgets = ARRAY_SIZE(simple_dapm_widgets); card->probe = simple_soc_probe; card->driver_name = "simple-card"; @@ -638,6 +668,13 @@ static int asoc_simple_probe(struct platform_device *pdev) if (ret < 0) return ret; + priv->pa_gpio = devm_gpiod_get_optional(dev, "pa", GPIOD_OUT_LOW); + if (IS_ERR(priv->pa_gpio)) { + ret = PTR_ERR(priv->pa_gpio); + dev_err(dev, "failed to get amplifier gpio: %d\n", ret); + return ret; + } + if (np && of_device_is_available(np)) { ret = simple_parse_of(priv, li); -- 2.25.1

2 1

[PATCH v2 0/6] Add driver for SAMA7G5's PDMC
by Codrin Ciubotariu 07 Mar '22

07 Mar '22

This patch series adds support for Pulse Density Microphone Controller (PDMC), present on Microchip's SAMA7G5. The PDMC interfaces up to 4 digital microphones having Pulse Density Modulated (PDM) outputs. It generates a single clock line and samples 1 or 2 data lines. The signal path includes an audio grade programmable decimation filter and outputs 24-bit audio words. The source of each channel can be independently defined as PDMC_DS0 or PDMC_DS1, sampled at the rising or falling edge of PDMC_CLK. The patch series starts with a fix on the ASoC DMA engine support. Then continues with the bindings and the driver of PDMC. It is followed by the DT nodes for SAMA7G5 and SAMA7G5-EK. In the end, the drivers for PDMC and PDM microphones are enabled in sama7_defconfig. Changes in v2: - addressed comments from Krzysztof Kozlowski Codrin Ciubotariu (6): ASoC: dmaengine: do not use a NULL prepare_slave_config() callback ASoC: dt-bindings: Document Microchip's PDMC ASoC: atmel: mchp-pdmc: add PDMC driver ARM: dts: at91: sama7g5: add nodes for PDMC ARM: dts: at91: sama7g5ek: add node for PDMC0 ARM: configs: at91: sama7_defconfig: add MCHP PDMC and DMIC drivers .../bindings/sound/microchip,pdmc.yaml | 99 ++ arch/arm/boot/dts/at91-sama7g5ek.dts | 21 +- arch/arm/boot/dts/sama7g5.dtsi | 24 + arch/arm/configs/sama7_defconfig | 2 + include/dt-bindings/sound/microchip,pdmc.h | 13 + sound/soc/atmel/Kconfig | 16 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/mchp-pdmc.c | 1084 +++++++++++++++++ sound/soc/soc-generic-dmaengine-pcm.c | 6 +- 9 files changed, 1263 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/microchip,pdmc.yaml create mode 100644 include/dt-bindings/sound/microchip,pdmc.h create mode 100644 sound/soc/atmel/mchp-pdmc.c -- 2.32.0

3 9

Re: [syzbot] possible deadlock in snd_timer_interrupt (2)
by syzbot 07 Mar '22

07 Mar '22

syzbot has found a reproducer for the following issue on: HEAD commit: 38f80f42147f MAINTAINERS: Remove dead patchwork link git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=161736d6700000 kernel config: https://syzkaller.appspot.com/x/.config?x=4e9585407e09f75f dashboard link: https://syzkaller.appspot.com/bug?extid=1ee0910eca9c94f71f25 compiler: gcc (Debian 10.2.1-6) 10.2.1 20210110, GNU ld (GNU Binutils for Debian) 2.35.2 syz repro: https://syzkaller.appspot.com/x/repro.syz?x=11b6ad85700000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=12527ef1700000 IMPORTANT: if you fix the issue, please add the following tag to the commit: Reported-by: syzbot+1ee0910eca9c94f71f25(a)syzkaller.appspotmail.com ======================================================== WARNING: possible irq lock inversion dependency detected 5.17.0-rc6-syzkaller-00184-g38f80f42147f #0 Not tainted -------------------------------------------------------- syz-executor301/4486 just changed the state of lock: ffff88814a78a948 (&timer->lock){..-.}-{2:2}, at: snd_timer_interrupt.part.0+0x34/0xcf0 sound/core/timer.c:856 but this lock took another, SOFTIRQ-READ-unsafe lock in the past: (tasklist_lock){.+.+}-{2:2} and interrupts could create inverse lock ordering between them. other info that might help us debug this: Chain exists of: &timer->lock --> &new->fa_lock --> tasklist_lock Possible interrupt unsafe locking scenario: CPU0 CPU1 ---- ---- lock(tasklist_lock); local_irq_disable(); lock(&timer->lock); lock(&new->fa_lock); <Interrupt> lock(&timer->lock); *** DEADLOCK *** 1 lock held by syz-executor301/4486: #0: ffffc90000007d70 ((&priv->tlist)){+.-.}-{0:0}, at: lockdep_copy_map include/linux/lockdep.h:35 [inline] #0: ffffc90000007d70 ((&priv->tlist)){+.-.}-{0:0}, at: call_timer_fn+0xd5/0x6b0 kernel/time/timer.c:1411 the shortest dependencies between 2nd lock and 1st lock: -> (tasklist_lock){.+.+}-{2:2} { HARDIRQ-ON-R at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_read_lock include/linux/rwlock_api_smp.h:150 [inline] _raw_read_lock+0x5b/0x70 kernel/locking/spinlock.c:228 do_wait+0x284/0xce0 kernel/exit.c:1518 kernel_wait+0x9c/0x150 kernel/exit.c:1708 call_usermodehelper_exec_sync kernel/umh.c:139 [inline] call_usermodehelper_exec_work+0xf5/0x180 kernel/umh.c:166 process_one_work+0x9ac/0x1650 kernel/workqueue.c:2307 worker_thread+0x657/0x1110 kernel/workqueue.c:2454 kthread+0x2e9/0x3a0 kernel/kthread.c:377 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:295 SOFTIRQ-ON-R at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_read_lock include/linux/rwlock_api_smp.h:150 [inline] _raw_read_lock+0x5b/0x70 kernel/locking/spinlock.c:228 do_wait+0x284/0xce0 kernel/exit.c:1518 kernel_wait+0x9c/0x150 kernel/exit.c:1708 call_usermodehelper_exec_sync kernel/umh.c:139 [inline] call_usermodehelper_exec_work+0xf5/0x180 kernel/umh.c:166 process_one_work+0x9ac/0x1650 kernel/workqueue.c:2307 worker_thread+0x657/0x1110 kernel/workqueue.c:2454 kthread+0x2e9/0x3a0 kernel/kthread.c:377 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:295 INITIAL USE at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_write_lock_irq include/linux/rwlock_api_smp.h:195 [inline] _raw_write_lock_irq+0x32/0x50 kernel/locking/spinlock.c:326 copy_process+0x486a/0x7250 kernel/fork.c:2295 kernel_clone+0xe7/0xab0 kernel/fork.c:2565 kernel_thread+0xb5/0xf0 kernel/fork.c:2617 rest_init+0x23/0x3e0 init/main.c:690 start_kernel+0x47a/0x49b init/main.c:1138 secondary_startup_64_no_verify+0xc3/0xcb INITIAL READ USE at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_read_lock include/linux/rwlock_api_smp.h:150 [inline] _raw_read_lock+0x5b/0x70 kernel/locking/spinlock.c:228 do_wait+0x284/0xce0 kernel/exit.c:1518 kernel_wait+0x9c/0x150 kernel/exit.c:1708 call_usermodehelper_exec_sync kernel/umh.c:139 [inline] call_usermodehelper_exec_work+0xf5/0x180 kernel/umh.c:166 process_one_work+0x9ac/0x1650 kernel/workqueue.c:2307 worker_thread+0x657/0x1110 kernel/workqueue.c:2454 kthread+0x2e9/0x3a0 kernel/kthread.c:377 ret_from_fork+0x1f/0x30 arch/x86/entry/entry_64.S:295 } ... key at: [<ffffffff8b80a098>] tasklist_lock+0x18/0x40 ... acquired at: __raw_read_lock include/linux/rwlock_api_smp.h:150 [inline] _raw_read_lock+0x5b/0x70 kernel/locking/spinlock.c:228 send_sigio+0xab/0x380 fs/fcntl.c:810 kill_fasync_rcu fs/fcntl.c:1021 [inline] kill_fasync fs/fcntl.c:1035 [inline] kill_fasync+0x1f8/0x470 fs/fcntl.c:1028 snd_timer_user_ccallback+0x298/0x330 sound/core/timer.c:1386 snd_timer_notify1+0x11c/0x3b0 sound/core/timer.c:516 snd_timer_stop1+0x496/0x860 sound/core/timer.c:657 snd_timer_stop sound/core/timer.c:710 [inline] snd_timer_close_locked+0x20f/0xbb0 sound/core/timer.c:408 snd_timer_close+0x87/0xf0 sound/core/timer.c:463 __snd_timer_user_ioctl.isra.0+0x10e2/0x2490 sound/core/timer.c:1762 snd_timer_user_ioctl+0x77/0xb0 sound/core/timer.c:2128 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:874 [inline] __se_sys_ioctl fs/ioctl.c:860 [inline] __x64_sys_ioctl+0x193/0x200 fs/ioctl.c:860 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae -> (&f->f_owner.lock){....}-{2:2} { INITIAL USE at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_write_lock_irq include/linux/rwlock_api_smp.h:195 [inline] _raw_write_lock_irq+0x32/0x50 kernel/locking/spinlock.c:326 f_modown+0x2a/0x390 fs/fcntl.c:91 __f_setown fs/fcntl.c:110 [inline] f_setown+0xd7/0x230 fs/fcntl.c:138 do_fcntl+0x749/0x1210 fs/fcntl.c:393 __do_sys_fcntl fs/fcntl.c:472 [inline] __se_sys_fcntl fs/fcntl.c:457 [inline] __x64_sys_fcntl+0x165/0x1e0 fs/fcntl.c:457 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae INITIAL READ USE at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_read_lock_irqsave include/linux/rwlock_api_smp.h:160 [inline] _raw_read_lock_irqsave+0x70/0x90 kernel/locking/spinlock.c:236 send_sigio+0x24/0x380 fs/fcntl.c:796 kill_fasync_rcu fs/fcntl.c:1021 [inline] kill_fasync fs/fcntl.c:1035 [inline] kill_fasync+0x1f8/0x470 fs/fcntl.c:1028 snd_timer_user_ccallback+0x298/0x330 sound/core/timer.c:1386 snd_timer_notify1+0x11c/0x3b0 sound/core/timer.c:516 snd_timer_start1+0x4d4/0x800 sound/core/timer.c:578 snd_timer_start sound/core/timer.c:696 [inline] snd_timer_start sound/core/timer.c:689 [inline] snd_timer_user_start.isra.0+0x1e3/0x260 sound/core/timer.c:1984 __snd_timer_user_ioctl.isra.0+0xda8/0x2490 sound/core/timer.c:2107 snd_timer_user_ioctl+0x77/0xb0 sound/core/timer.c:2128 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:874 [inline] __se_sys_ioctl fs/ioctl.c:860 [inline] __x64_sys_ioctl+0x193/0x200 fs/ioctl.c:860 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae } ... key at: [<ffffffff90585a40>] __key.5+0x0/0x40 ... acquired at: __raw_read_lock_irqsave include/linux/rwlock_api_smp.h:160 [inline] _raw_read_lock_irqsave+0x70/0x90 kernel/locking/spinlock.c:236 send_sigio+0x24/0x380 fs/fcntl.c:796 kill_fasync_rcu fs/fcntl.c:1021 [inline] kill_fasync fs/fcntl.c:1035 [inline] kill_fasync+0x1f8/0x470 fs/fcntl.c:1028 snd_timer_user_ccallback+0x298/0x330 sound/core/timer.c:1386 snd_timer_notify1+0x11c/0x3b0 sound/core/timer.c:516 snd_timer_start1+0x4d4/0x800 sound/core/timer.c:578 snd_timer_start sound/core/timer.c:696 [inline] snd_timer_start sound/core/timer.c:689 [inline] snd_timer_user_start.isra.0+0x1e3/0x260 sound/core/timer.c:1984 __snd_timer_user_ioctl.isra.0+0xda8/0x2490 sound/core/timer.c:2107 snd_timer_user_ioctl+0x77/0xb0 sound/core/timer.c:2128 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:874 [inline] __se_sys_ioctl fs/ioctl.c:860 [inline] __x64_sys_ioctl+0x193/0x200 fs/ioctl.c:860 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae -> (&new->fa_lock){....}-{2:2} { INITIAL USE at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_write_lock_irq include/linux/rwlock_api_smp.h:195 [inline] _raw_write_lock_irq+0x32/0x50 kernel/locking/spinlock.c:326 fasync_remove_entry+0xb6/0x1f0 fs/fcntl.c:891 fasync_helper+0x9e/0xb0 fs/fcntl.c:994 __fput+0x846/0x9f0 fs/file_table.c:314 task_work_run+0xdd/0x1a0 kernel/task_work.c:164 exit_task_work include/linux/task_work.h:32 [inline] do_exit+0xb29/0x2a30 kernel/exit.c:806 do_group_exit+0xd2/0x2f0 kernel/exit.c:935 __do_sys_exit_group kernel/exit.c:946 [inline] __se_sys_exit_group kernel/exit.c:944 [inline] __x64_sys_exit_group+0x3a/0x50 kernel/exit.c:944 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae INITIAL READ USE at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_read_lock_irqsave include/linux/rwlock_api_smp.h:160 [inline] _raw_read_lock_irqsave+0x70/0x90 kernel/locking/spinlock.c:236 kill_fasync_rcu fs/fcntl.c:1014 [inline] kill_fasync fs/fcntl.c:1035 [inline] kill_fasync+0x136/0x470 fs/fcntl.c:1028 snd_timer_user_ccallback+0x298/0x330 sound/core/timer.c:1386 snd_timer_notify1+0x11c/0x3b0 sound/core/timer.c:516 snd_timer_start1+0x4d4/0x800 sound/core/timer.c:578 snd_timer_start sound/core/timer.c:696 [inline] snd_timer_start sound/core/timer.c:689 [inline] snd_timer_user_start.isra.0+0x1e3/0x260 sound/core/timer.c:1984 __snd_timer_user_ioctl.isra.0+0xda8/0x2490 sound/core/timer.c:2107 snd_timer_user_ioctl+0x77/0xb0 sound/core/timer.c:2128 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:874 [inline] __se_sys_ioctl fs/ioctl.c:860 [inline] __x64_sys_ioctl+0x193/0x200 fs/ioctl.c:860 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae } ... key at: [<ffffffff90586820>] __key.0+0x0/0x40 ... acquired at: __raw_read_lock_irqsave include/linux/rwlock_api_smp.h:160 [inline] _raw_read_lock_irqsave+0x70/0x90 kernel/locking/spinlock.c:236 kill_fasync_rcu fs/fcntl.c:1014 [inline] kill_fasync fs/fcntl.c:1035 [inline] kill_fasync+0x136/0x470 fs/fcntl.c:1028 snd_timer_user_ccallback+0x298/0x330 sound/core/timer.c:1386 snd_timer_notify1+0x11c/0x3b0 sound/core/timer.c:516 snd_timer_start1+0x4d4/0x800 sound/core/timer.c:578 snd_timer_start sound/core/timer.c:696 [inline] snd_timer_start sound/core/timer.c:689 [inline] snd_timer_user_start.isra.0+0x1e3/0x260 sound/core/timer.c:1984 __snd_timer_user_ioctl.isra.0+0xda8/0x2490 sound/core/timer.c:2107 snd_timer_user_ioctl+0x77/0xb0 sound/core/timer.c:2128 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:874 [inline] __se_sys_ioctl fs/ioctl.c:860 [inline] __x64_sys_ioctl+0x193/0x200 fs/ioctl.c:860 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae -> (&timer->lock){..-.}-{2:2} { IN-SOFTIRQ-W at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:110 [inline] _raw_spin_lock_irqsave+0x39/0x50 kernel/locking/spinlock.c:162 snd_timer_interrupt.part.0+0x34/0xcf0 sound/core/timer.c:856 snd_timer_interrupt sound/core/timer.c:1154 [inline] snd_timer_s_function+0x14b/0x200 sound/core/timer.c:1154 call_timer_fn+0x1a5/0x6b0 kernel/time/timer.c:1421 expire_timers kernel/time/timer.c:1466 [inline] __run_timers.part.0+0x67c/0xa30 kernel/time/timer.c:1734 __run_timers kernel/time/timer.c:1715 [inline] run_timer_softirq+0xb3/0x1d0 kernel/time/timer.c:1747 __do_softirq+0x29b/0x9c2 kernel/softirq.c:558 invoke_softirq kernel/softirq.c:432 [inline] __irq_exit_rcu+0x123/0x180 kernel/softirq.c:637 irq_exit_rcu+0x5/0x20 kernel/softirq.c:649 sysvec_apic_timer_interrupt+0x93/0xc0 arch/x86/kernel/apic/apic.c:1097 asm_sysvec_apic_timer_interrupt+0x12/0x20 arch/x86/include/asm/idtentry.h:638 bytes_is_nonzero mm/kasan/generic.c:85 [inline] memory_is_nonzero mm/kasan/generic.c:102 [inline] memory_is_poisoned_n mm/kasan/generic.c:128 [inline] memory_is_poisoned mm/kasan/generic.c:159 [inline] check_region_inline mm/kasan/generic.c:180 [inline] kasan_check_range+0xde/0x180 mm/kasan/generic.c:189 instrument_atomic_read include/linux/instrumented.h:71 [inline] atomic_long_read include/linux/atomic/atomic-instrumented.h:1265 [inline] filp_close+0x22/0x170 fs/open.c:1318 close_files fs/file.c:403 [inline] put_files_struct fs/file.c:418 [inline] put_files_struct+0x1d0/0x350 fs/file.c:415 exit_files+0x7e/0xa0 fs/file.c:435 do_exit+0xaf2/0x2a30 kernel/exit.c:801 do_group_exit+0xd2/0x2f0 kernel/exit.c:935 __do_sys_exit_group kernel/exit.c:946 [inline] __se_sys_exit_group kernel/exit.c:944 [inline] __x64_sys_exit_group+0x3a/0x50 kernel/exit.c:944 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae INITIAL USE at: lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:110 [inline] _raw_spin_lock_irqsave+0x39/0x50 kernel/locking/spinlock.c:162 snd_timer_stop1+0x5c/0x860 sound/core/timer.c:626 snd_timer_stop sound/core/timer.c:710 [inline] snd_timer_user_start.isra.0+0x8b/0x260 sound/core/timer.c:1981 __snd_timer_user_ioctl.isra.0+0xda8/0x2490 sound/core/timer.c:2107 snd_timer_user_ioctl+0x77/0xb0 sound/core/timer.c:2128 vfs_ioctl fs/ioctl.c:51 [inline] __do_sys_ioctl fs/ioctl.c:874 [inline] __se_sys_ioctl fs/ioctl.c:860 [inline] __x64_sys_ioctl+0x193/0x200 fs/ioctl.c:860 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae } ... key at: [<ffffffff908843c0>] __key.12+0x0/0x40 ... acquired at: mark_lock kernel/locking/lockdep.c:4569 [inline] mark_usage kernel/locking/lockdep.c:4500 [inline] __lock_acquire+0x11e3/0x56c0 kernel/locking/lockdep.c:4981 lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:110 [inline] _raw_spin_lock_irqsave+0x39/0x50 kernel/locking/spinlock.c:162 snd_timer_interrupt.part.0+0x34/0xcf0 sound/core/timer.c:856 snd_timer_interrupt sound/core/timer.c:1154 [inline] snd_timer_s_function+0x14b/0x200 sound/core/timer.c:1154 call_timer_fn+0x1a5/0x6b0 kernel/time/timer.c:1421 expire_timers kernel/time/timer.c:1466 [inline] __run_timers.part.0+0x67c/0xa30 kernel/time/timer.c:1734 __run_timers kernel/time/timer.c:1715 [inline] run_timer_softirq+0xb3/0x1d0 kernel/time/timer.c:1747 __do_softirq+0x29b/0x9c2 kernel/softirq.c:558 invoke_softirq kernel/softirq.c:432 [inline] __irq_exit_rcu+0x123/0x180 kernel/softirq.c:637 irq_exit_rcu+0x5/0x20 kernel/softirq.c:649 sysvec_apic_timer_interrupt+0x93/0xc0 arch/x86/kernel/apic/apic.c:1097 asm_sysvec_apic_timer_interrupt+0x12/0x20 arch/x86/include/asm/idtentry.h:638 bytes_is_nonzero mm/kasan/generic.c:85 [inline] memory_is_nonzero mm/kasan/generic.c:102 [inline] memory_is_poisoned_n mm/kasan/generic.c:128 [inline] memory_is_poisoned mm/kasan/generic.c:159 [inline] check_region_inline mm/kasan/generic.c:180 [inline] kasan_check_range+0xde/0x180 mm/kasan/generic.c:189 instrument_atomic_read include/linux/instrumented.h:71 [inline] atomic_long_read include/linux/atomic/atomic-instrumented.h:1265 [inline] filp_close+0x22/0x170 fs/open.c:1318 close_files fs/file.c:403 [inline] put_files_struct fs/file.c:418 [inline] put_files_struct+0x1d0/0x350 fs/file.c:415 exit_files+0x7e/0xa0 fs/file.c:435 do_exit+0xaf2/0x2a30 kernel/exit.c:801 do_group_exit+0xd2/0x2f0 kernel/exit.c:935 __do_sys_exit_group kernel/exit.c:946 [inline] __se_sys_exit_group kernel/exit.c:944 [inline] __x64_sys_exit_group+0x3a/0x50 kernel/exit.c:944 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae stack backtrace: CPU: 0 PID: 4486 Comm: syz-executor301 Not tainted 5.17.0-rc6-syzkaller-00184-g38f80f42147f #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: <IRQ> __dump_stack lib/dump_stack.c:88 [inline] dump_stack_lvl+0xcd/0x134 lib/dump_stack.c:106 print_irq_inversion_bug kernel/locking/lockdep.c:194 [inline] check_usage_forwards kernel/locking/lockdep.c:4043 [inline] mark_lock_irq kernel/locking/lockdep.c:4175 [inline] mark_lock.part.0.cold+0x86/0xd8 kernel/locking/lockdep.c:4605 mark_lock kernel/locking/lockdep.c:4569 [inline] mark_usage kernel/locking/lockdep.c:4500 [inline] __lock_acquire+0x11e3/0x56c0 kernel/locking/lockdep.c:4981 lock_acquire kernel/locking/lockdep.c:5639 [inline] lock_acquire+0x1ab/0x510 kernel/locking/lockdep.c:5604 __raw_spin_lock_irqsave include/linux/spinlock_api_smp.h:110 [inline] _raw_spin_lock_irqsave+0x39/0x50 kernel/locking/spinlock.c:162 snd_timer_interrupt.part.0+0x34/0xcf0 sound/core/timer.c:856 snd_timer_interrupt sound/core/timer.c:1154 [inline] snd_timer_s_function+0x14b/0x200 sound/core/timer.c:1154 call_timer_fn+0x1a5/0x6b0 kernel/time/timer.c:1421 expire_timers kernel/time/timer.c:1466 [inline] __run_timers.part.0+0x67c/0xa30 kernel/time/timer.c:1734 __run_timers kernel/time/timer.c:1715 [inline] run_timer_softirq+0xb3/0x1d0 kernel/time/timer.c:1747 __do_softirq+0x29b/0x9c2 kernel/softirq.c:558 invoke_softirq kernel/softirq.c:432 [inline] __irq_exit_rcu+0x123/0x180 kernel/softirq.c:637 irq_exit_rcu+0x5/0x20 kernel/softirq.c:649 sysvec_apic_timer_interrupt+0x93/0xc0 arch/x86/kernel/apic/apic.c:1097 </IRQ> <TASK> asm_sysvec_apic_timer_interrupt+0x12/0x20 arch/x86/include/asm/idtentry.h:638 RIP: 0010:bytes_is_nonzero mm/kasan/generic.c:85 [inline] RIP: 0010:memory_is_nonzero mm/kasan/generic.c:102 [inline] RIP: 0010:memory_is_poisoned_n mm/kasan/generic.c:128 [inline] RIP: 0010:memory_is_poisoned mm/kasan/generic.c:159 [inline] RIP: 0010:check_region_inline mm/kasan/generic.c:180 [inline] RIP: 0010:kasan_check_range+0xde/0x180 mm/kasan/generic.c:189 Code: 74 f2 48 89 c2 b8 01 00 00 00 48 85 d2 75 56 5b 5d 41 5c c3 48 85 d2 74 5e 48 01 ea eb 09 48 83 c0 01 48 39 d0 74 50 80 38 00 <74> f2 eb d4 41 bc 08 00 00 00 48 89 ea 45 29 dc 4d 8d 1c 2c eb 0c RSP: 0018:ffffc9000376fd30 EFLAGS: 00000246 RAX: ffffed100f1d3d9f RBX: ffffed100f1d3da0 RCX: ffffffff81cb24b2 RDX: ffffed100f1d3da0 RSI: 0000000000000008 RDI: ffff888078e9ecf8 RBP: ffffed100f1d3d9f R08: 0000000000000000 R09: ffff888078e9ecff R10: ffffed100f1d3d9f R11: 0000000000000000 R12: ffff88807ee70f00 R13: ffff888078e9ecf8 R14: ffff88807ee70f00 R15: ffff88807ee71028 instrument_atomic_read include/linux/instrumented.h:71 [inline] atomic_long_read include/linux/atomic/atomic-instrumented.h:1265 [inline] filp_close+0x22/0x170 fs/open.c:1318 close_files fs/file.c:403 [inline] put_files_struct fs/file.c:418 [inline] put_files_struct+0x1d0/0x350 fs/file.c:415 exit_files+0x7e/0xa0 fs/file.c:435 do_exit+0xaf2/0x2a30 kernel/exit.c:801 do_group_exit+0xd2/0x2f0 kernel/exit.c:935 __do_sys_exit_group kernel/exit.c:946 [inline] __se_sys_exit_group kernel/exit.c:944 [inline] __x64_sys_exit_group+0x3a/0x50 kernel/exit.c:944 do_syscall_x64 arch/x86/entry/common.c:50 [inline] do_syscall_64+0x35/0xb0 arch/x86/entry/common.c:80 entry_SYSCALL_64_after_hwframe+0x44/0xae RIP: 0033:0x7ff95f986c99 Code: Unable to access opcode bytes at RIP 0x7ff95f986c6f. RSP: 002b:00007ffecd4a5e98 EFLAGS: 00000246 ORIG_RAX: 00000000000000e7 RAX: ffffffffffffffda RBX: 00007ff95f9fb330 RCX: 00007ff95f986c99 RDX: 000000000000003c RSI: 00000000000000e7 RDI: 0000000000000000 RBP: 0000000000000000 R08: ffffffffffffffc0 R09: 0000000000000001 R10: 0000000000000001 R11: 0000000000000246 R12: 00007ff95f9fb330 R13: 0000000000000001 R14: 0000000000000000 R15: 0000000000000001 </TASK> ---------------- Code disassembly (best guess): 0: 74 f2 je 0xfffffff4 2: 48 89 c2 mov %rax,%rdx 5: b8 01 00 00 00 mov $0x1,%eax a: 48 85 d2 test %rdx,%rdx d: 75 56 jne 0x65 f: 5b pop %rbx 10: 5d pop %rbp 11: 41 5c pop %r12 13: c3 retq 14: 48 85 d2 test %rdx,%rdx 17: 74 5e je 0x77 19: 48 01 ea add %rbp,%rdx 1c: eb 09 jmp 0x27 1e: 48 83 c0 01 add $0x1,%rax 22: 48 39 d0 cmp %rdx,%rax 25: 74 50 je 0x77 27: 80 38 00 cmpb $0x0,(%rax) * 2a: 74 f2 je 0x1e <-- trapping instruction 2c: eb d4 jmp 0x2 2e: 41 bc 08 00 00 00 mov $0x8,%r12d 34: 48 89 ea mov %rbp,%rdx 37: 45 29 dc sub %r11d,%r12d 3a: 4d 8d 1c 2c lea (%r12,%rbp,1),%r11 3e: eb 0c jmp 0x4c

3 5

[PATCH 0/6] Add driver for SAMA7G5's PDMC
by Codrin Ciubotariu 07 Mar '22

07 Mar '22

This patch series adds support for Pulse Density Microphone Controller (PDMC), present on Microchip's SAMA7G5. The PDMC interfaces up to 4 digital microphones having Pulse Density Modulated (PDM) outputs. It generates a single clock line and samples 1 or 2 data lines. The signal path includes an audio grade programmable decimation filter and outputs 24-bit audio words. The source of each channel can be independently defined as PDMC_DS0 or PDMC_DS1, sampled at the rising or falling edge of PDMC_CLK. The patch series starts with a fix on the ASoC DMA engine support. Then continues with the bindings and the driver of PDMC. It is followed by the DT nodes for SAMA7G5 and SAMA7G5-EK. In the end, the drivers for PDMC and PDM microphones are enabled in sama7_defconfig. Codrin Ciubotariu (6): ASoC: dmaengine: do not use a NULL prepare_slave_config() callback ASoC: add DT bindings for Microchip PDMC ASoC: atmel: mchp-pdmc: add PDMC driver ARM: dts: at91: sama7g5: add nodes for PDMC ARM: dts: at91: sama7g5ek: add node for PDMC0 ARM: configs: at91: sama7_defconfig: add MCHP PDMC and DMIC drivers .../devicetree/bindings/sound/mchp,pdmc.yaml | 99 ++ arch/arm/boot/dts/at91-sama7g5ek.dts | 21 +- arch/arm/boot/dts/sama7g5.dtsi | 24 + arch/arm/configs/sama7_defconfig | 2 + include/dt-bindings/sound/mchp,pdmc.h | 13 + sound/soc/atmel/Kconfig | 16 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/mchp-pdmc.c | 1084 +++++++++++++++++ sound/soc/soc-generic-dmaengine-pcm.c | 6 +- 9 files changed, 1263 insertions(+), 4 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mchp,pdmc.yaml create mode 100644 include/dt-bindings/sound/mchp,pdmc.h create mode 100644 sound/soc/atmel/mchp-pdmc.c -- 2.32.0

3 9

[v2 00/17] ASoC: mediatek: Add support for MT8186 SoC
by Jiaxin Yu 07 Mar '22

07 Mar '22

This series of patches adds support for Mediatek AFE of MT8186 Soc. Patches are based on broonie tree "for-next" branch. Changes since v1: [v2 01/17] - add a new ID to the existing mt6358 codec driver [v2 03/17] - modify log level in DAPM events - use standard numeric control with name ending in Switch - return 1 when the value changed in mixer control's .get callback [v2 05/17] - ending in Switch to the standard on/off controls - change to "HW Gain 1 Volume" and "HW Gain 2 Volume" [v2 09/17] - return an error in the default case rather than just picking one of the behaviours when do .set_fmt - use the new defines that are _PROVIDER_MASK, _DAIFMT_CBP_CFP and _DAIFMT_CBC_CFC [v2 10/17] [v2 11/17] - the clock and gpio are aplit out into separate patches The source file's GPL comment use c++ style, and the header fils's GPL comment use c style. We have added "Switch" after the names of all the controls that just are simple on/off. Jiaxin Yu (17): ASoC: mediatek: mt6366: add codec driver ASoC: mediatek: mt8186: support audsys clock control ASoC: mediatek: mt8186: support adda in platform driver ASoC: mediatek: mt8186: support hostless in platform driver ASoC: mediatek: mt8186: support hw gain in platform driver ASoC: mediatek: mt8186: support i2s in platform driver ASoC: mediatek: mt8186: support pcm in platform driver ASoC: mediatek: mt8186: support src in platform driver ASoC: mediatek: mt8186: support tdm in platform driver ASoC: mediatek: mt8186: support audio clock control in platform driver ASoC: mediatek: mt8186: support gpio control in platform driver ASoC: mediatek: mt8186: add platform driver dt-bindings: mediatek: mt8186: add audio afe document ASoC: mediatek: mt8186: add machine driver with mt6366, da7219 and max98357 dt-bindings: mediatek: mt8186: add mt8186-mt6366-da7219-max98357 document ASoC: mediatek: mt8186: add machine driver with mt6366, rt1019 and rt5682s dt-bindings: mediatek: mt8186: add mt8186-mt6366-rt1019-rt5682s document .../bindings/sound/mt8186-afe-pcm.yaml | 175 + .../sound/mt8186-mt6366-da7219-max98357.yaml | 47 + .../sound/mt8186-mt6366-rt1019-rt5682s.yaml | 47 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 1 + sound/soc/mediatek/Kconfig | 44 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/mt8186/Makefile | 21 + sound/soc/mediatek/mt8186/mt8186-afe-clk.c | 719 ++++ sound/soc/mediatek/mt8186/mt8186-afe-clk.h | 210 + sound/soc/mediatek/mt8186/mt8186-afe-common.h | 245 ++ .../soc/mediatek/mt8186/mt8186-afe-control.c | 261 ++ sound/soc/mediatek/mt8186/mt8186-afe-gpio.c | 210 + sound/soc/mediatek/mt8186/mt8186-afe-gpio.h | 19 + sound/soc/mediatek/mt8186/mt8186-afe-pcm.c | 3029 +++++++++++++++ sound/soc/mediatek/mt8186/mt8186-audsys-clk.c | 150 + sound/soc/mediatek/mt8186/mt8186-audsys-clk.h | 15 + .../soc/mediatek/mt8186/mt8186-audsys-clkid.h | 45 + sound/soc/mediatek/mt8186/mt8186-dai-adda.c | 891 +++++ .../soc/mediatek/mt8186/mt8186-dai-hostless.c | 295 ++ .../soc/mediatek/mt8186/mt8186-dai-hw-gain.c | 245 ++ sound/soc/mediatek/mt8186/mt8186-dai-i2s.c | 1371 +++++++ sound/soc/mediatek/mt8186/mt8186-dai-pcm.c | 432 +++ sound/soc/mediatek/mt8186/mt8186-dai-src.c | 758 ++++ sound/soc/mediatek/mt8186/mt8186-dai-tdm.c | 713 ++++ .../mediatek/mt8186/mt8186-interconnection.h | 69 + .../soc/mediatek/mt8186/mt8186-misc-control.c | 1728 +++++++++ .../mt8186/mt8186-mt6366-da7219-max98357.c | 910 +++++ .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 894 +++++ sound/soc/mediatek/mt8186/mt8186-reg.h | 3433 +++++++++++++++++ 30 files changed, 16986 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/mt8186-afe-pcm.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-da7219-max98357.yaml create mode 100644 Documentation/devicetree/bindings/sound/mt8186-mt6366-rt1019-rt5682s.yaml create mode 100644 sound/soc/mediatek/mt8186/Makefile create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-clk.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-clk.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-common.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-gpio.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-gpio.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clk.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clk.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-audsys-clkid.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-adda.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-hostless.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-hw-gain.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-i2s.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-pcm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-src.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-dai-tdm.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-interconnection.h create mode 100644 sound/soc/mediatek/mt8186/mt8186-misc-control.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-da7219-max98357.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-mt6366-rt1019-rt5682s.c create mode 100644 sound/soc/mediatek/mt8186/mt8186-reg.h -- 2.18.0

3 45

How to judge USB Audio device has no digital in/out.
by Chengyi Zhao 07 Mar '22

07 Mar '22

Hi guys, I found that in the following configuration file "alsa-lib/src/conf/cards/USB-Audio.conf" there are many kinds of USB Audio devices that need to turn off digital in/out. However, how can I judge that this USB Audio device has no digital in/out, thanks. USB-Audio.pcm.iec958_device { # The below don't have digital in/out, so prevent them from being opened. "Andrea PureAudio USB-SA Headset" 999 "Blue Snowball" 999 "C-Media USB Headphone Set" 999 "Cmedia Audio" 999 "DELL PROFESSIONAL SOUND BAR AE5" 999 "HP Digital Stereo Headset" 999 "GN 9330" 999 "Logitech Speaker Lapdesk N700" 999 "Logitech G35 Headset" 999 "Logitech USB Headset" 999 "Logitech USB Headset H540" 999 "Logitech Wireless Headset" 999 "Plantronics Blackwire 3220 Seri" 999 "Plantronics GameCom 780" 999 "Plantronics USB Headset" 999 "Plantronics Wireless Audio" 999 "SB WoW Headset" 999 "Scarlett 2i2 USB" 999 "Scarlett 2i4 USB" 999 "Sennheiser USB headset" 999 "SWTOR Gaming Headset by Razer" 999 "ThinkStation P620 Main" 999 "ThinkStation P620 Rear" 999 "Thunderbolt Dock Audio Headset" 999 "Thunderbolt Dock Audio Module" 999 "USB Device 0x46d_0x821" 999 "USB Device 0x46d_0x992" 999 "WD15 Dock" 999 "WD19 Dock" 999 } Best Regards, Chengyi

1 0

[PATCH 1/2] sound: usb: Add vendor's hooking interface
by Oh Eomji 07 Mar '22

07 Mar '22

In mobile, a co-processor can be used with USB audio to improve power consumption. To support this type of hardware, hooks need to be added to the USB audio subsystem to be able to call into the hardware when needed. The main operation of the call-backs are: - Initialize the co-processor by transmitting data when initializing. - Change the co-processor setting value through the interface function. - Configure sampling rate - pcm open/close - other housekeeping Known issues: - This only supports one set of callback hooks, meaning that this only works if there is one type of USB controller in the system. This should be changed to be a per-host-controller interface instead of one global set of callbacks. Signed-off-by: JaeHun Jung <jh0801.jung(a)samsung.com> Signed-off-by: Oh Eomji <eomji.oh(a)samsung.com> --- sound/usb/card.c | 120 +++++++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/card.h | 21 +++++++++ sound/usb/usbaudio.h | 46 ++++++++++++++++++++ 3 files changed, 187 insertions(+) diff --git a/sound/usb/card.c b/sound/usb/card.c index 3769622..ac81c5f 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -117,6 +117,118 @@ MODULE_PARM_DESC(skip_validation, "Skip unit descriptor validation (default: no) static DEFINE_MUTEX(register_mutex); static struct snd_usb_audio *usb_chip[SNDRV_CARDS]; static struct usb_driver usb_audio_driver; +static struct snd_usb_audio_vendor_ops *usb_vendor_ops; + +int snd_vendor_set_ops(struct snd_usb_audio_vendor_ops *ops) +{ + if ((!ops->connect) || + (!ops->disconnect) || + (!ops->set_interface) || + (!ops->set_rate) || + (!ops->set_pcm_buf) || + (!ops->set_pcm_intf) || + (!ops->set_pcm_connection) || + (!ops->set_pcm_binterval) || + (!ops->usb_add_ctls)) + return -EINVAL; + + usb_vendor_ops = ops; + return 0; +} +EXPORT_SYMBOL_GPL(snd_vendor_set_ops); + +struct snd_usb_audio_vendor_ops *snd_vendor_get_ops(void) +{ + return usb_vendor_ops; +} + +static int snd_vendor_connect(struct usb_interface *intf) +{ + struct snd_usb_audio_vendor_ops *ops = snd_vendor_get_ops(); + + if (ops) + return ops->connect(intf); + return 0; +} + +static void snd_vendor_disconnect(struct usb_interface *intf) +{ + struct snd_usb_audio_vendor_ops *ops = snd_vendor_get_ops(); + + if (ops) + ops->disconnect(intf); +} + +int snd_vendor_set_interface(struct usb_device *udev, + struct usb_host_interface *intf, + int iface, int alt) +{ + struct snd_usb_audio_vendor_ops *ops = snd_vendor_get_ops(); + + if (ops) + return ops->set_interface(udev, intf, iface, alt); + return 0; +} + +int snd_vendor_set_rate(struct usb_interface *intf, int iface, int rate, + int alt) +{ + struct snd_usb_audio_vendor_ops *ops = snd_vendor_get_ops(); + + if (ops) + return ops->set_rate(intf, iface, rate, alt); + return 0; +} + +int snd_vendor_set_pcm_buf(struct usb_device *udev, int iface) +{ + struct snd_usb_audio_vendor_ops *ops = snd_vendor_get_ops(); + + if (ops) + ops->set_pcm_buf(udev, iface); + return 0; +} + +int snd_vendor_set_pcm_intf(struct usb_interface *intf, int iface, int alt, + int direction) +{ + struct snd_usb_audio_vendor_ops *ops = snd_vendor_get_ops(); + + if (ops) + return ops->set_pcm_intf(intf, iface, alt, direction); + return 0; +} + +int snd_vendor_set_pcm_connection(struct usb_device *udev, + enum snd_vendor_pcm_open_close onoff, + int direction) +{ + struct snd_usb_audio_vendor_ops *ops = snd_vendor_get_ops(); + + if (ops) + return ops->set_pcm_connection(udev, onoff, direction); + return 0; +} + +int snd_vendor_set_pcm_binterval(const struct audioformat *fp, + const struct audioformat *found, + int *cur_attr, int *attr) +{ + struct snd_usb_audio_vendor_ops *ops = snd_vendor_get_ops(); + + if (ops) + return ops->set_pcm_binterval(fp, found, cur_attr, attr); + return 0; +} + +static int snd_vendor_usb_add_ctls(struct snd_usb_audio *chip) +{ + struct snd_usb_audio_vendor_ops *ops = snd_vendor_get_ops(); + + if (ops) + return ops->usb_add_ctls(chip); + return 0; +} /* * disconnect streams @@ -753,6 +865,10 @@ static int usb_audio_probe(struct usb_interface *intf, if (err < 0) return err; + err = snd_vendor_connect(intf); + if (err) + return err; + /* * found a config. now register to ALSA */ @@ -820,6 +936,8 @@ static int usb_audio_probe(struct usb_interface *intf, if (chip->quirk_flags & QUIRK_FLAG_DISABLE_AUTOSUSPEND) usb_disable_autosuspend(interface_to_usbdev(intf)); + snd_vendor_usb_add_ctls(chip); + /* * For devices with more than one control interface, we assume the * first contains the audio controls. We might need a more specific @@ -907,6 +1025,8 @@ static void usb_audio_disconnect(struct usb_interface *intf) card = chip->card; + snd_vendor_disconnect(intf); + mutex_lock(&register_mutex); if (atomic_inc_return(&chip->shutdown) == 1) { struct snd_usb_stream *as; diff --git a/sound/usb/card.h b/sound/usb/card.h index 87f042d..2b686e4 100644 --- a/sound/usb/card.h +++ b/sound/usb/card.h @@ -204,4 +204,25 @@ struct snd_usb_stream { struct list_head list; }; +struct snd_usb_substream *find_snd_usb_substream(unsigned int card_num, + unsigned int pcm_idx, unsigned int direction, struct snd_usb_audio + **uchip, void (*disconnect_cb)(struct snd_usb_audio *chip)); + +int snd_vendor_set_ops(struct snd_usb_audio_vendor_ops *vendor_ops); +struct snd_usb_audio_vendor_ops *snd_vendor_get_ops(void); +int snd_vendor_set_interface(struct usb_device *udev, + struct usb_host_interface *alts, + int iface, int alt); +int snd_vendor_set_rate(struct usb_interface *intf, int iface, int rate, + int alt); +int snd_vendor_set_pcm_buf(struct usb_device *udev, int iface); +int snd_vendor_set_pcm_intf(struct usb_interface *intf, int iface, int alt, + int direction); +int snd_vendor_set_pcm_connection(struct usb_device *udev, + enum snd_vendor_pcm_open_close onoff, + int direction); +int snd_vendor_set_pcm_binterval(const struct audioformat *fp, + const struct audioformat *found, + int *cur_attr, int *attr); + #endif /* __USBAUDIO_CARD_H */ diff --git a/sound/usb/usbaudio.h b/sound/usb/usbaudio.h index 1678341..edcb5a3 100644 --- a/sound/usb/usbaudio.h +++ b/sound/usb/usbaudio.h @@ -184,4 +184,50 @@ extern bool snd_usb_skip_validation; #define QUIRK_FLAG_DSD_RAW (1U << 15) #define QUIRK_FLAG_SET_IFACE_FIRST (1U << 16) +struct audioformat; + +enum snd_vendor_pcm_open_close { + SOUND_PCM_CLOSE = 0, + SOUND_PCM_OPEN, +}; + +/** + * struct snd_usb_audio_vendor_ops - function callbacks for USB audio accelerators + * @connect: called when a new interface is found + * @disconnect: called when an interface is removed + * @set_interface: called when an interface is initialized + * @set_rate: called when the rate is set + * @set_pcm_buf: called when the pcm buffer is set + * @set_pcm_intf: called when the pcm interface is set + * @set_pcm_connection: called when pcm is opened/closed + * @set_pcm_binterval: called when the pcm binterval is set + * @usb_add_ctls: called when USB controls are added + * + * Set of callbacks for some accelerated USB audio streaming hardware. + * + * TODO: make this USB host-controller specific, right now this only works for + * one USB controller in the system at a time, which is only realistic for + * self-contained systems like phones. + */ +struct snd_usb_audio_vendor_ops { + int (*connect)(struct usb_interface *intf); + void (*disconnect)(struct usb_interface *intf); + + int (*set_interface)(struct usb_device *udev, + struct usb_host_interface *alts, + int iface, int alt); + int (*set_rate)(struct usb_interface *intf, int iface, int rate, + int alt); + int (*set_pcm_buf)(struct usb_device *udev, int iface); + int (*set_pcm_intf)(struct usb_interface *intf, int iface, int alt, + int direction); + int (*set_pcm_connection)(struct usb_device *udev, + enum snd_vendor_pcm_open_close onoff, + int direction); + int (*set_pcm_binterval)(const struct audioformat *fp, + const struct audioformat *found, + int *cur_attr, int *attr); + int (*usb_add_ctls)(struct snd_usb_audio *chip); +}; + #endif /* __USBAUDIO_H */ -- 2.7.4

2 3

[PATCH -next] ASoC: amd: acp: signedness bug in acp-renoir
by YueHaibing 05 Mar '22

05 Mar '22

In renoir_audio_probe() platform_get_irq_byname() may return error, but i2s_irq now is unsigned int so the error handling is never triggered. Fixes: 3304a242f45a ("ASoC: amd: Use platform_get_irq_byname() to get the interrupt") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/amd/acp/amd.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/amd.h b/sound/soc/amd/acp/amd.h index 8fd38bf4d3bd..acb50d9fca58 100644 --- a/sound/soc/amd/acp/amd.h +++ b/sound/soc/amd/acp/amd.h @@ -96,7 +96,7 @@ struct acp_dev_data { char *name; struct device *dev; void __iomem *acp_base; - unsigned int i2s_irq; + int i2s_irq; /* SOC specific dais */ struct snd_soc_dai_driver *dai_driver; -- 2.17.1

2 1

[PATCH] ALSA: x86: Use standard mmap helper for Intel HDMI LPE audio
by Takashi Iwai 05 Mar '22

05 Mar '22

Intel HDMI LPE audio driver has its own mmap callback that mimics with the noncached page attributes, but this is rather superfluous and can be replaced with the standard helper, as the device is only for playback and the write-cache should suffice. This patch drops the own code and just uses the standard helper. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/x86/intel_hdmi_audio.c | 13 ------------- 1 file changed, 13 deletions(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 4a3ff6468aa7..b00634663346 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1253,18 +1253,6 @@ static snd_pcm_uframes_t had_pcm_pointer(struct snd_pcm_substream *substream) return len; } -/* - * ALSA PCM mmap callback - */ -static int had_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); - return remap_pfn_range(vma, vma->vm_start, - substream->runtime->dma_addr >> PAGE_SHIFT, - vma->vm_end - vma->vm_start, vma->vm_page_prot); -} - /* * ALSA PCM ops */ @@ -1276,7 +1264,6 @@ static const struct snd_pcm_ops had_pcm_ops = { .trigger = had_pcm_trigger, .sync_stop = had_pcm_sync_stop, .pointer = had_pcm_pointer, - .mmap = had_pcm_mmap, }; /* process mode change of the running stream; called in mutex */ -- 2.31.1

1 0

[PATCH] ALSA: usb-audio: add mapping for new Corsair Virtuoso SE
by Reza Jahanbakhshi 05 Mar '22

05 Mar '22

New device id for Corsair Virtuoso SE RGB Wireless that currently is not in the mixer_map. This entry in the mixer_map is necessary in order to label its mixer appropriately and allow userspace to pick the correct volume controls. For instance, my own Corsair Virtuoso SE RGB Wireless headset has this new ID and consequently, the sidetone and volume are not working correctly without this change. > sudo lsusb -v | grep -i corsair Bus 007 Device 011: ID 1b1c:0a40 Corsair CORSAIR VIRTUOSO SE Wireless Gam idVendor 0x1b1c Corsair iManufacturer 1 Corsair iProduct 2 CORSAIR VIRTUOSO SE Wireless Gaming Headset Signed-off-by: Reza Jahanbakhshi <reza.jahanbakhshi(a)gmail.com> --- sound/usb/mixer_maps.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 96991ddf5055..64f5544d0a0a 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -542,6 +542,16 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x05a7, 0x40fa), .map = bose_soundlink_map, }, + { + /* Corsair Virtuoso SE Latest (wired mode) */ + .id = USB_ID(0x1b1c, 0x0a3f), + .map = corsair_virtuoso_map, + }, + { + /* Corsair Virtuoso SE Latest (wireless mode) */ + .id = USB_ID(0x1b1c, 0x0a40), + .map = corsair_virtuoso_map, + }, { /* Corsair Virtuoso SE (wired mode) */ .id = USB_ID(0x1b1c, 0x0a3d), -- 2.35.1

2 1

[PATCH] ALSA: hda/realtek: Add quirk for Clevo NP70PNJ
by Tim Crawford 05 Mar '22

05 Mar '22

Fixes headset detection on Clevo NP70PNJ. Signed-off-by: Tim Crawford <tcrawford(a)system76.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0ef70158f88c..7d85989f585b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9144,6 +9144,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x8561, "Clevo NH[57][0-9][ER][ACDH]Q", ALC269_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[57][0-9]RZ[Q]", ALC269_FIXUP_DMIC), SND_PCI_QUIRK(0x1558, 0x8668, "Clevo NP50B[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x867d, "Clevo NP7[01]PN[HJK]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8680, "Clevo NJ50LU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC256_FIXUP_MIC_NO_PRESENCE_AND_RESUME), SND_PCI_QUIRK(0x1558, 0x8a20, "Clevo NH55DCQ-Y", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.35.1

2 1

[PATCH v18 0/2] mtk: add ADSP mailbox controller for MT8195
by Allen-KH Cheng 05 Mar '22

05 Mar '22

Mediatek ADSP IPC is used to send notification or short message between processors with dsp. It will place the message to the share buffer and will access the ADSP mailbox registers to kick dsp. Two mailboxes used to send notification or short message between processors with dsp. This patchset was tested and confirmed on MT8195 cherry board. Based on tag: next-20220223, linux-next/master changes since v17: - remove description and add maxItems for reg and interrupts in mtk,adsp-mbox.yaml changes since v16: - fix WARNING: modpost: missing MODULE_LICENSE() in drivers/mailbox /mtk-adsp-mailbox.o. Add MODULE_LICENSE in the last line. - Due to WARNING: Missing or malformed SPDX-License-Identifier tag in line 1 in checkpatch, we don't remove SPDX-License in line 1. changes since v15: - add paragraphs to describe the config symbols fully changes since v14: - add inline for get_mtk_adsp_mbox_priv changes since v13: - rebase on v5.16-rc8 - add back ptr check from of_device_get_match_data changes since v12: - remove of_device_get_match_data ptr check: of_device_get_match_data(dev) will never going to return NULL. driver probe with compatible mediatek,mt8195-adsp-mbox. changes since v11: - remove useless MODULE_LICENSE changes since v10: - split up v9 into two separate submissions changes since v9: - rename adsp_mbox_chan_ops to mtk_adsp_mbox_chan_ops changes since v8: - remove struct adsp_mbox_ch_info - add get_mtk_adsp_mbox_priv - use mtk_adsp_mbox_priv va_mboxreg address in adsp mbox driver - add struct mtk_adsp_mbox_cfg for DSP mbox register offset - remove adsp mbox register offset hard code define - remove mtk-adsp-ipc.h reference in adsp mbox driver changes since v7: - add mtk prefix for adsp ipc functions - rename adsp_mbox_ch_info to mtk_adsp_mbox_ch_info - remove incorrect reviewers in commit message changes since v6: - dt-bindings: change example dtsi node. - rename config MTK_ADSP_IPC_MBOX to MTK_ADSP_MBOX - remove unused variable - add reviewers changes since v5: - remove some redundant code changes since v4: - use switch ... case in adsp_ipc_recv - add error handling path for chan_name pointer - refine some code to be concise changes since v3: - reorder MTK_ADSP_IPC_MBOX config - remove some redundant code - remove lock in mtk-adsp-mailbox changes since v2: - separate adsp_mailbox into two instances changes since v1: - fix dt_binding_check error Allen-KH Cheng (2): dt-bindings: mailbox: mtk,adsp-mbox: add mtk adsp-mbox document mailbox: mediatek: add support for adsp mailbox controller .../bindings/mailbox/mtk,adsp-mbox.yaml | 50 +++++ drivers/mailbox/Kconfig | 9 + drivers/mailbox/Makefile | 2 + drivers/mailbox/mtk-adsp-mailbox.c | 176 ++++++++++++++++++ 4 files changed, 237 insertions(+) create mode 100644 Documentation/devicetree/bindings/mailbox/mtk,adsp-mbox.yaml create mode 100644 drivers/mailbox/mtk-adsp-mailbox.c -- 2.18.0

2 3

Re: mmotm 2022-03-03-21-24 uploaded (sound/soc/intel/boards/)
by Randy Dunlap 04 Mar '22

04 Mar '22

On 3/3/22 21:24, Andrew Morton wrote: > The mm-of-the-moment snapshot 2022-03-03-21-24 has been uploaded to > > https://www.ozlabs.org/~akpm/mmotm/ > > mmotm-readme.txt says > > README for mm-of-the-moment: > > https://www.ozlabs.org/~akpm/mmotm/ > > This is a snapshot of my -mm patch queue. Uploaded at random hopefully > more than once a week. > > You will need quilt to apply these patches to the latest Linus release (5.x > or 5.x-rcY). The series file is in broken-out.tar.gz and is duplicated in > https://ozlabs.org/~akpm/mmotm/series on i386: ERROR: modpost: "sof_dai_get_bclk" [sound/soc/intel/boards/snd-soc-intel-sof-cirrus-common.ko] undefined! ERROR: modpost: "sof_dai_get_mclk" [sound/soc/intel/boards/snd-soc-intel-sof-realtek-common.ko] undefined! Full randconfig file is attached. -- ~Randy

1 0

[PATCH v2] ALSA: firewire-lib: fix uninitialized flag for AV/C deferred transaction
by Takashi Sakamoto 04 Mar '22

04 Mar '22

AV/C deferred transaction was supported at a commit 00a7bb81c20f ("ALSA: firewire-lib: Add support for deferred transaction") while 'deferrable' flag can be uninitialized for non-control/notify AV/C transactions. UBSAN reports it: kernel: ================================================================================ kernel: UBSAN: invalid-load in /build/linux-aa0B4d/linux-5.15.0/sound/firewire/fcp.c:363:9 kernel: load of value 158 is not a valid value for type '_Bool' kernel: CPU: 3 PID: 182227 Comm: irq/35-firewire Tainted: P OE 5.15.0-18-generic #18-Ubuntu kernel: Hardware name: Gigabyte Technology Co., Ltd. AX370-Gaming 5/AX370-Gaming 5, BIOS F42b 08/01/2019 kernel: Call Trace: kernel: <IRQ> kernel: show_stack+0x52/0x58 kernel: dump_stack_lvl+0x4a/0x5f kernel: dump_stack+0x10/0x12 kernel: ubsan_epilogue+0x9/0x45 kernel: __ubsan_handle_load_invalid_value.cold+0x44/0x49 kernel: fcp_response.part.0.cold+0x1a/0x2b [snd_firewire_lib] kernel: fcp_response+0x28/0x30 [snd_firewire_lib] kernel: fw_core_handle_request+0x230/0x3d0 [firewire_core] kernel: handle_ar_packet+0x1d9/0x200 [firewire_ohci] kernel: ? handle_ar_packet+0x1d9/0x200 [firewire_ohci] kernel: ? transmit_complete_callback+0x9f/0x120 [firewire_core] kernel: ar_context_tasklet+0xa8/0x2e0 [firewire_ohci] kernel: tasklet_action_common.constprop.0+0xea/0xf0 kernel: tasklet_action+0x22/0x30 kernel: __do_softirq+0xd9/0x2e3 kernel: ? irq_finalize_oneshot.part.0+0xf0/0xf0 kernel: do_softirq+0x75/0xa0 kernel: </IRQ> kernel: <TASK> kernel: __local_bh_enable_ip+0x50/0x60 kernel: irq_forced_thread_fn+0x7e/0x90 kernel: irq_thread+0xba/0x190 kernel: ? irq_thread_fn+0x60/0x60 kernel: kthread+0x11e/0x140 kernel: ? irq_thread_check_affinity+0xf0/0xf0 kernel: ? set_kthread_struct+0x50/0x50 kernel: ret_from_fork+0x22/0x30 kernel: </TASK> kernel: ================================================================================ This commit fixes the bug. The bug has no disadvantage for the non- control/notify AV/C transactions since the flag has an effect for AV/C response with INTERIM (0x0f) status which is not used for the transactions in AV/C general specification. Fixes: 00a7bb81c20f ("ALSA: firewire-lib: Add support for deferred transaction") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fcp.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/firewire/fcp.c b/sound/firewire/fcp.c index bbfbebf4affb..df44dd5dc4b2 100644 --- a/sound/firewire/fcp.c +++ b/sound/firewire/fcp.c @@ -240,9 +240,7 @@ int fcp_avc_transaction(struct fw_unit *unit, t.response_match_bytes = response_match_bytes; t.state = STATE_PENDING; init_waitqueue_head(&t.wait); - - if (*(const u8 *)command == 0x00 || *(const u8 *)command == 0x03) - t.deferrable = true; + t.deferrable = (*(const u8 *)command == 0x00 || *(const u8 *)command == 0x03); spin_lock_irq(&transactions_lock); list_add_tail(&t.list, &transactions); -- 2.34.1

2 1

[PATCH] dt-bindings: mfd: Fix pinctrl node name warnings
by Rob Herring 04 Mar '22

04 Mar '22

The recent addition pinctrl.yaml in commit c09acbc499e8 ("dt-bindings: pinctrl: use pinctrl.yaml") resulted in some node name warnings: Documentation/devicetree/bindings/mfd/cirrus,lochnagar.example.dt.yaml: \ lochnagar-pinctrl: $nodename:0: 'lochnagar-pinctrl' does not match '^(pinctrl|pinmux)(@[0-9a-f]+)?$' Documentation/devicetree/bindings/mfd/cirrus,madera.example.dt.yaml: \ codec@1a: $nodename:0: 'codec@1a' does not match '^(pinctrl|pinmux)(@[0-9a-f]+)?$' Documentation/devicetree/bindings/mfd/brcm,cru.example.dt.yaml: \ pin-controller@1c0: $nodename:0: 'pin-controller@1c0' does not match '^(pinctrl|pinmux)(@[0-9a-f]+)?$' Fix the node names to the preferred 'pinctrl'. For cirrus,madera, nothing from pinctrl.yaml schema is used, so just drop the reference. Fixes: c09acbc499e8 ("dt-bindings: pinctrl: use pinctrl.yaml") Cc: Rafał Miłecki <rafal(a)milecki.pl> Signed-off-by: Rob Herring <robh(a)kernel.org> --- Documentation/devicetree/bindings/mfd/brcm,cru.yaml | 4 ++-- Documentation/devicetree/bindings/mfd/cirrus,lochnagar.yaml | 2 +- Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml | 3 --- 3 files changed, 3 insertions(+), 6 deletions(-) diff --git a/Documentation/devicetree/bindings/mfd/brcm,cru.yaml b/Documentation/devicetree/bindings/mfd/brcm,cru.yaml index be4a2df71c25..b85819fbb07c 100644 --- a/Documentation/devicetree/bindings/mfd/brcm,cru.yaml +++ b/Documentation/devicetree/bindings/mfd/brcm,cru.yaml @@ -39,7 +39,7 @@ patternProperties: '^phy@[a-f0-9]+$': $ref: ../phy/bcm-ns-usb2-phy.yaml - '^pin-controller@[a-f0-9]+$': + '^pinctrl@[a-f0-9]+$': $ref: ../pinctrl/brcm,ns-pinmux.yaml '^syscon@[a-f0-9]+$': @@ -94,7 +94,7 @@ examples: reg = <0x180 0x4>; }; - pin-controller@1c0 { + pinctrl@1c0 { compatible = "brcm,bcm4708-pinmux"; reg = <0x1c0 0x24>; reg-names = "cru_gpio_control"; diff --git a/Documentation/devicetree/bindings/mfd/cirrus,lochnagar.yaml b/Documentation/devicetree/bindings/mfd/cirrus,lochnagar.yaml index c00ad3e21c21..975a46f3c46f 100644 --- a/Documentation/devicetree/bindings/mfd/cirrus,lochnagar.yaml +++ b/Documentation/devicetree/bindings/mfd/cirrus,lochnagar.yaml @@ -126,7 +126,7 @@ properties: clock-frequency: const: 12288000 - lochnagar-pinctrl: + pinctrl: type: object $ref: /schemas/pinctrl/cirrus,lochnagar.yaml# diff --git a/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml b/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml index c85f759ae5a3..8a90d8273767 100644 --- a/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml +++ b/Documentation/devicetree/bindings/pinctrl/cirrus,madera.yaml @@ -107,9 +107,6 @@ properties: additionalProperties: false -allOf: - - $ref: "pinctrl.yaml#" - required: - pinctrl-0 - pinctrl-names -- 2.32.0

2 2

[PATCH] ALSA: firewire-lib: fix uninitialized flag for AV/C deferred transaction
by Takashi Sakamoto 04 Mar '22

04 Mar '22

AV/C deferred transaction was supported at a commit 00a7bb81c20f ("ALSA: firewire-lib: Add support for deferred transaction") while 'deferrable' flag can be uninitialized for non-control/notify AV/C transactions. UBSAN reports it: kernel: ================================================================================ kernel: UBSAN: invalid-load in /build/linux-aa0B4d/linux-5.15.0/sound/firewire/fcp.c:363:9 kernel: load of value 158 is not a valid value for type '_Bool' kernel: CPU: 3 PID: 182227 Comm: irq/35-firewire Tainted: P OE 5.15.0-18-generic #18-Ubuntu kernel: Hardware name: Gigabyte Technology Co., Ltd. AX370-Gaming 5/AX370-Gaming 5, BIOS F42b 08/01/2019 kernel: Call Trace: kernel: <IRQ> kernel: show_stack+0x52/0x58 kernel: dump_stack_lvl+0x4a/0x5f kernel: dump_stack+0x10/0x12 kernel: ubsan_epilogue+0x9/0x45 kernel: __ubsan_handle_load_invalid_value.cold+0x44/0x49 kernel: fcp_response.part.0.cold+0x1a/0x2b [snd_firewire_lib] kernel: fcp_response+0x28/0x30 [snd_firewire_lib] kernel: fw_core_handle_request+0x230/0x3d0 [firewire_core] kernel: handle_ar_packet+0x1d9/0x200 [firewire_ohci] kernel: ? handle_ar_packet+0x1d9/0x200 [firewire_ohci] kernel: ? transmit_complete_callback+0x9f/0x120 [firewire_core] kernel: ar_context_tasklet+0xa8/0x2e0 [firewire_ohci] kernel: tasklet_action_common.constprop.0+0xea/0xf0 kernel: tasklet_action+0x22/0x30 kernel: __do_softirq+0xd9/0x2e3 kernel: ? irq_finalize_oneshot.part.0+0xf0/0xf0 kernel: do_softirq+0x75/0xa0 kernel: </IRQ> kernel: <TASK> kernel: __local_bh_enable_ip+0x50/0x60 kernel: irq_forced_thread_fn+0x7e/0x90 kernel: irq_thread+0xba/0x190 kernel: ? irq_thread_fn+0x60/0x60 kernel: kthread+0x11e/0x140 kernel: ? irq_thread_check_affinity+0xf0/0xf0 kernel: ? set_kthread_struct+0x50/0x50 kernel: ret_from_fork+0x22/0x30 kernel: </TASK> kernel: ================================================================================ This commit fixes the bug. The bug has no disadvantage for the non- control/notify AV/C transactions since the flag has an effect for AV/C response with INTERIM (0x0f) status which is not used for the transactions in AV/C general specification. Fixes: 00a7bb81c20f ("ALSA: firewire-lib: Add support for deferred transaction") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fcp.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/firewire/fcp.c b/sound/firewire/fcp.c index bbfbebf4affb..564607e083ca 100644 --- a/sound/firewire/fcp.c +++ b/sound/firewire/fcp.c @@ -240,9 +240,7 @@ int fcp_avc_transaction(struct fw_unit *unit, t.response_match_bytes = response_match_bytes; t.state = STATE_PENDING; init_waitqueue_head(&t.wait); - - if (*(const u8 *)command == 0x00 || *(const u8 *)command == 0x03) - t.deferrable = true; + t.deferrable = !!(*(const u8 *)command == 0x00 || *(const u8 *)command == 0x03); spin_lock_irq(&transactions_lock); list_add_tail(&t.list, &transactions); -- 2.34.1

2 1

[PATCH] ALSA: lola: add a check for the return of vmalloc()
by xkernel.wang＠foxmail.com 04 Mar '22

04 Mar '22

From: Xiaoke Wang <xkernel.wang(a)foxmail.com> vmalloc() is a memory allocation function which can return NULL when some internal memory errors happen. So it is better to check the return of it to catch the error in time. Signed-off-by: Xiaoke Wang <xkernel.wang(a)foxmail.com> --- sound/pci/lola/lola_mixer.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/lola/lola_mixer.c b/sound/pci/lola/lola_mixer.c index e2c8f14..6b16248 100644 --- a/sound/pci/lola/lola_mixer.c +++ b/sound/pci/lola/lola_mixer.c @@ -121,6 +121,8 @@ int lola_init_mixer_widget(struct lola *chip, int nid) /* reserve memory to copy mixer data for sleep mode transitions */ chip->mixer.array_saved = vmalloc(sizeof(struct lola_mixer_array)); + if (!chip->mixer.array_saved) + return -ENOMEM; /* mixer matrix sources are physical input data and play streams */ chip->mixer.src_stream_outs = chip->pcm[PLAY].num_streams; --

2 1

[PATCH] ALSA: echoaudio: remove redundant assignment to variable bytes
by Colin Ian King 04 Mar '22

04 Mar '22

The variable bytes is being assigned a value that is never read, it is being re-assigned inside a following if block. The assignment is redundant and can be removed. Cleans up clang scan build warning: sound/pci/echoaudio/midi.c:211:9: warning: Although the value stored to 'bytes' is used in the enclosing expression, the value is never actually read from 'bytes' [deadcode.DeadStores] Signed-off-by: Colin Ian King <colin.i.king(a)gmail.com> --- sound/pci/echoaudio/midi.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/echoaudio/midi.c b/sound/pci/echoaudio/midi.c index cb72d27e809e..7be5c3327b16 100644 --- a/sound/pci/echoaudio/midi.c +++ b/sound/pci/echoaudio/midi.c @@ -208,7 +208,7 @@ static void snd_echo_midi_output_write(struct timer_list *t) /* No interrupts are involved: we have to check at regular intervals if the card's output buffer has room for new data. */ - sent = bytes = 0; + sent = 0; spin_lock_irqsave(&chip->lock, flags); chip->midi_full = 0; if (!snd_rawmidi_transmit_empty(chip->midi_out)) { -- 2.34.1

2 1

Re: Issue in pcm_dsnoop.c in alsa-lib
by S.J. Wang 04 Mar '22

04 Mar '22

> > > > Hi Takashi Iwai, Jaroslav Kysela > > > > We encountered an issue in the pcm_dsnoop use case, could you > > please help to have a look? > > > > *Issue description:* > > With two instances for dsnoop type device running in parallel, > > after suspend/resume, one of the instances will be hung in memcpy > > because the very large copy size is obtained. > > > > #3 0x0000ffffa78d5098 in snd_pcm_dsnoop_sync_ptr > (pcm=0xaaab06563da0) > > at pcm_dsnoop.c:158 dsnoop = 0xaaab06563c20 slave_hw_ptr = 64 > > old_slave_hw_ptr = 533120 avail = *187651522444320* > > > > * Reason analysis: * > > The root cause that I analysis is that after suspend/resume, one > > instance will get the SND_PCM_STATE_SUSPENDED state from slave pcm > device, > > then it will do snd_pcm_prepare() and snd_pcm_start(), which will > > reset the dsnoop->slave_hw_ptr and the hw_ptr of slave pcm device, > > then the state of this instance is correct. But another instance may > > not get the SND_PCM_STATE_SUSPENDED state from slave pcm device > > because slave device may have been recovered by first instance, so > > the dsnoop->slave_hw_ptr is not reset. but because hw_ptr of slave > > pcm device has been reset, so there will be a very large "avail" size. > > > > *Solution:* > > I didn't come up with a fix for this issue, seems there is no easy > > way to let another instance know this case and reset the > > dsnoop->slave_hw_ptr, could you please help? > > Could you try topic/pcm-direct-resume branch on > > https://eur01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fgithub > .com%2Ftiwai%2Falsa- > lib&data=04%7C01%7Cshengjiu.wang%40nxp.com%7C95f97de3f2c840d > 9853508d9fd2e79ea%7C686ea1d3bc2b4c6fa92cd99c5c301635%7C0%7C0%7C > 637819198319430045%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwM > DAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000&sdat > a=WWX1ZlcQhJF3pHJdHPIH%2B0xG9o%2FjQnHG5fHDbKXwQwE%3D&r > eserved=0 > Thanks, I push my test result in https://github.com/alsa-project/alsa-lib/issues/213 Could you please review? Best regards Wang shengjiu

2 1

Re: [PATCH 2/6] treewide: remove using list iterator after loop body as a ptr
by Linus Torvalds 04 Mar '22

04 Mar '22

On Mon, Feb 28, 2022 at 11:56 AM Linus Torvalds <torvalds(a)linux-foundation.org> wrote: > > I do wish we could actually poison the 'pos' value after the loop > somehow - but clearly the "might be uninitialized" I was hoping for > isn't the way to do it. Side note: we do need *some* way to do it. Because if we make that change, and only set it to another pointer when not the head, then we very much change the semantics of "list_for_each_head()". The "list was empty" case would now exit with 'pos' not set at all (or set to NULL if we add that). Or it would be set to the last entry. And regardless, that means that all the if (pos == head) kinds of checks after the loop would be fundamentally broken. Darn. I really hoped for (and naively expected) that we could actually have the compiler warn about the use-after-loop case. That whole "compiler will now complain about bad use" was integral to my clever plan to use the C99 feature of declaring the iterator inside the loop. But my "clever plan" was apparently some ACME-level Wile E. Coyote sh*t. Darn. Linus

11 34

[PATCH] ASoC: rt5665: Don't block workqueue if card is unbound
by Zhen Ni 04 Mar '22

04 Mar '22

The current rt5665_jack_detect_handler() assumes the component and card will always show up and implements an infinite usleep loop waiting for them to show up. This does not hold true if a codec interrupt (or other event) occurs when the card is unbound. The codec driver's remove or shutdown functions cannot cancel the workqueue due to the wait loop. As a result, code can either end up blocking the workqueue, or hit a kernel oops when the card is freed. Fix the issue by rescheduling the jack detect handler in case the card is not ready. In case card never shows up, the shutdown/remove/suspend calls can now cancel the detect task. Signed-off-by: Zhen Ni <nizhen(a)uniontech.com> --- sound/soc/codecs/rt5665.c | 20 +++++++------------- 1 file changed, 7 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 33e889802ff8..cdfd7085df2a 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -1293,19 +1293,13 @@ static void rt5665_jack_detect_handler(struct work_struct *work) container_of(work, struct rt5665_priv, jack_detect_work.work); int val, btn_type; - while (!rt5665->component) { - pr_debug("%s codec = null\n", __func__); - usleep_range(10000, 15000); - } - - while (!rt5665->component->card->instantiated) { - pr_debug("%s\n", __func__); - usleep_range(10000, 15000); - } - - while (!rt5665->calibration_done) { - pr_debug("%s calibration not ready\n", __func__); - usleep_range(10000, 15000); + if (!rt5665->component || !rt5665->component->card->instantiated || + !rt5665->calibration_done) { + pr_debug("%s card not yet ready\n", __func__); + /* try later */ + mod_delayed_work(system_power_efficient_wq, + &rt5665->jack_detect_work, msecs_to_jiffies(15)); + return; } mutex_lock(&rt5665->calibrate_mutex); -- 2.20.1

1 0

Re: Re: [PATCH] ASoC: wm8350: Handle error for wm8350_register_irq
by Jiasheng Jiang 04 Mar '22

04 Mar '22

On Thu, Mar 03, 2022 at 09:38:34PM +0800, Charles Keepax wrote: >> if (ret != 0) >> - return ret; >> + goto err; > > I would probably just leave this as a return, nothing is gained > changing it to a goto. > >> >> + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, >> wm8350_hpr_jack_handler, 0, "Right jack detect", >> priv); >> - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, >> + if (ret != 0) >> + goto free_JCK_DET_L; > > Probably better to use non-caps here, having caps in a label is a > little unusual. Thansk, I have submitted v2 to solve the problems. Jiang

1 0

Re: mmotm 2022-02-23-21-20 uploaded (Kconfig warning: SND_SOC_WCD938X)
by Randy Dunlap 04 Mar '22

04 Mar '22

On 2/23/22 21:21, Andrew Morton wrote: > The mm-of-the-moment snapshot 2022-02-23-21-20 has been uploaded to > > https://www.ozlabs.org/~akpm/mmotm/ > > mmotm-readme.txt says > > README for mm-of-the-moment: > > https://www.ozlabs.org/~akpm/mmotm/ > > This is a snapshot of my -mm patch queue. Uploaded at random hopefully > more than once a week. on x86_64: WARNING: unmet direct dependencies detected for SND_SOC_WCD938X Depends on [n]: SOUND [=y] && !UML && SND [=y] && SND_SOC [=m] && SND_SOC_WCD938X_SDW [=n] && (SOUNDWIRE [=n] || !SOUNDWIRE [=n]) Selected by [m]: - SND_SOC_SC7280 [=m] && SOUND [=y] && !UML && SND [=y] && SND_SOC [=m] && SND_SOC_QCOM [=m] && (I2C [=y] && SOUNDWIRE [=n] || COMPILE_TEST [=y]) Full randconfig file is attached. -- ~Randy

1 1

[PATCH] soundwire: qcom: fix build failures if CONFIG_PM=n
by Srinivas Kandagatla 03 Mar '22

03 Mar '22

Fix below build failure drivers/soundwire/qcom.c:1460:12: error: 'swrm_runtime_resume' defined but not used [-Werror=unused-function] by marking the resume swrm_runtime_resume() with __maybe_unused attribute. Reported-by: Geert Uytterhoeven <geert(a)linux-m68k.org> Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/qcom.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index e893aee1b057..6575695cd400 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1459,7 +1459,7 @@ static bool swrm_wait_for_frame_gen_enabled(struct qcom_swrm_ctrl *swrm) return false; } -static int swrm_runtime_resume(struct device *dev) +static int __maybe_unused swrm_runtime_resume(struct device *dev) { struct qcom_swrm_ctrl *ctrl = dev_get_drvdata(dev); int ret; -- 2.21.0

2 2

Issue in pcm_dsnoop.c in alsa-lib
by Shengjiu Wang 03 Mar '22

03 Mar '22

Hi Takashi Iwai, Jaroslav Kysela We encountered an issue in the pcm_dsnoop use case, could you please help to have a look? *Issue description:* With two instances for dsnoop type device running in parallel, after suspend/resume, one of the instances will be hung in memcpy because the very large copy size is obtained. #3 0x0000ffffa78d5098 in snd_pcm_dsnoop_sync_ptr (pcm=0xaaab06563da0) at pcm_dsnoop.c:158 dsnoop = 0xaaab06563c20 slave_hw_ptr = 64 old_slave_hw_ptr = 533120 avail = *187651522444320* * Reason analysis: * The root cause that I analysis is that after suspend/resume, one instance will get the SND_PCM_STATE_SUSPENDED state from slave pcm device, then it will do snd_pcm_prepare() and snd_pcm_start(), which will reset the dsnoop->slave_hw_ptr and the hw_ptr of slave pcm device, then the state of this instance is correct. But another instance may not get the SND_PCM_STATE_SUSPENDED state from slave pcm device because slave device may have been recovered by first instance, so the dsnoop->slave_hw_ptr is not reset. but because hw_ptr of slave pcm device has been reset, so there will be a very large "avail" size. *Solution:* I didn't come up with a fix for this issue, seems there is no easy way to let another instance know this case and reset the dsnoop->slave_hw_ptr, could you please help? Best regards Wang shengjiu

2 1

[PATCH] ASoC: wm8350: Handle error for wm8350_register_irq
by Jiasheng Jiang 03 Mar '22

03 Mar '22

As the potential failure of the wm8350_register_irq(), it should be better to check it and return error if fails. Also, use 'free_' in order to avoid the same code. Fixes: a6ba2b2dabb5 ("ASoC: Implement WM8350 headphone jack detection") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/soc/codecs/wm8350.c | 30 +++++++++++++++++++++++++----- 1 file changed, 25 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 15d42ce3b21d..0c70bbfbedb5 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1483,7 +1483,7 @@ static int wm8350_component_probe(struct snd_soc_component *component) ret = devm_regulator_bulk_get(wm8350->dev, ARRAY_SIZE(priv->supplies), priv->supplies); if (ret != 0) - return ret; + goto err; /* Put the codec into reset if it wasn't already */ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); @@ -1537,18 +1537,38 @@ static int wm8350_component_probe(struct snd_soc_component *component) wm8350_clear_bits(wm8350, WM8350_JACK_DETECT, WM8350_JDL_ENA | WM8350_JDR_ENA); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, wm8350_hpl_jack_handler, 0, "Left jack detect", priv); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, + if (ret != 0) + goto err; + + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, wm8350_hpr_jack_handler, 0, "Right jack detect", priv); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, + if (ret != 0) + goto free_JCK_DET_L; + + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, wm8350_mic_handler, 0, "Microphone short", priv); - wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD, + if (ret != 0) + goto free_JCK_DET_R; + + ret = wm8350_register_irq(wm8350, WM8350_IRQ_CODEC_MICD, wm8350_mic_handler, 0, "Microphone detect", priv); + if (ret != 0) + goto free_MICSCD; return 0; + +free_MICSCD: + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_MICSCD, priv); +free_JCK_DET_R: + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_R, priv); +free_JCK_DET_L: + wm8350_free_irq(wm8350, WM8350_IRQ_CODEC_JCK_DET_L, priv); +err: + return ret; } static void wm8350_component_remove(struct snd_soc_component *component) -- 2.25.1

2 1

[PATCH] ASoC: hdac_hda: Avoid unexpected match when pcm_name is "Analog"
by Meng Tang 03 Mar '22

03 Mar '22

pcm name can be "Analog" and "Alt Analog", cpcm->name can be "Analog Codec DAI" and "Alt Analog Codec DAI". When pcm_name is "Analog", "Analog Codec DAI" and "Alt Analog Codec DAI" are both satisfy the 'if (strstr(cpcm->name, pcm_name))' condition, which may cause the returned cpcm to be "Alt Analog Codec DAI". Even if we get the pcm name by id, and "Analog Codec DAI" goes into the loop before "Alt Analog Codec DAI", but I still think we'd better have multiple insurances against unexpected return values. After, we can correctly return the expected result even if other relevant places are changed. Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/soc/codecs/hdac_hda.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 667f3df239c7..a9f61c7e44ee 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -363,8 +363,13 @@ static struct hda_pcm *snd_soc_find_pcm_from_dai(struct hdac_hda_priv *hda_pvt, } list_for_each_entry(cpcm, &hcodec->pcm_list_head, list) { - if (strstr(cpcm->name, pcm_name)) + if (strstr(cpcm->name, pcm_name)) { + if (strcmp(pcm_name, "Analog") == 0) { + if (strstr(cpcm->name, "Alt Analog")) + continue; + } return cpcm; + } } dev_err(&hcodec->core.dev, "didn't find PCM for DAI %s\n", dai->name); -- 2.20.1

2 1

[v2] ASoC: bt-sco: fix bt-sco-pcm-wb dai widget don't connect to the endpoint
by Jiaxin Yu 03 Mar '22

03 Mar '22

This patch fix the second dai driver's dai widget can't connect to the endpoint. Because "bt-sco-pcm" and "bt-sco-pcm-wb" dai driver have the same stream_name, so it will cause they have the same widget name. Therefor it will just create only one route when do snd_soc_dapm_add_route that only find the widget through the widget name. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- Change from v1: fix build error Hi maintainer, Need your comments. The patch is the one that I think it makes the most sense. Maybe we can define the new stream_name for "bt-sco-pcm-wb" and add the new route. sound/soc/codecs/bt-sco.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c index 4d286844e3c8..cf17b9741bd8 100644 --- a/sound/soc/codecs/bt-sco.c +++ b/sound/soc/codecs/bt-sco.c @@ -13,11 +13,15 @@ static const struct snd_soc_dapm_widget bt_sco_widgets[] = { SND_SOC_DAPM_INPUT("RX"), SND_SOC_DAPM_OUTPUT("TX"), + SND_SOC_DAPM_AIF_IN("BT_SCO_RX", "Playback", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("BT_SCO_TX", "Capture", 0, + SND_SOC_NOPM, 0, 0), }; static const struct snd_soc_dapm_route bt_sco_routes[] = { - { "Capture", NULL, "RX" }, - { "TX", NULL, "Playback" }, + { "BT_SCO_TX", NULL, "RX" }, + { "TX", NULL, "BT_SCO_RX" }, }; static struct snd_soc_dai_driver bt_sco_dai[] = { -- 2.25.1

2 1

Re: [PATCH] wm8962: add a simple DMIC enable control
by Charles Keepax 03 Mar '22

03 Mar '22

On Wed, Feb 02, 2022 at 12:55:04PM +0100, Martin Kepplinger wrote: > Am Mittwoch, dem 02.02.2022 um 10:46 +0000 schrieb Charles Keepax: > > On Wed, Feb 02, 2022 at 11:17:34AM +0100, Martin Kepplinger wrote: > > > Am Mittwoch, dem 02.02.2022 um 09:53 +0000 schrieb Charles Keepax: > > > > On Tue, Feb 01, 2022 at 04:01:13PM +0100, Martin Kepplinger > > > > wrote: > > > Do you have a code example from a different codec that has roughly > > > what > > > is missing here? (the sound subsystem is new to me) > > > > Full disclosure this is complete untested, but it should be > > pretty close. Let me know if it does the trick and I will send a > > proper patch to the list. I do have a Librem 5 in a draw > > somewhere so can pull that out if we get really stuck, but that > > might have to wait until the weekend :-). > > > > I don't know if you guys are using the analogue bypass paths > > around the digital core on the chip. I think those will still > > work with the mics set to digital, so I have left the routes as > > is, but that might require some checking at some point. > > ok that's great and seems to work! that's luxury. > Excellent glad that is working for you, I will prep up a proper patch and send it to the list. Should get that done tomorrow morning, if I don't manage it this afternoon. > Volume / sensitivity of Analog input is too low, I saw that before. > What would you try to change that? > Hmm... you say you saw this before? I assume the input volume is always low, not just low sometimes? I would probably start by checking the voltage you have on the micbias, make sure that is as expected. Does the signal coming into the IN3R pin look low on a scope or is it just the level after it has been through the ADC on the chip that seems low? The input routing on this chip is pretty byzantine, the output of just "amixer" showing all the controls in the relevant use-case would probably be helpful to look over. I suspect there is a reasonable chance something around the input PGA is not configured to match the hardware, although I am not the most familiar with this part so hard to guess at exactly what off the top of my head. Finally, do you know how much the amplitude is off by? > you can do all of our tasks if you want to :) > Ha! Not sure about that, but happy to help out where I can. Thanks, Charles

2 13

Re: [PATCH 2/6] treewide: remove using list iterator after loop body as a ptr
by Christian König 03 Mar '22

03 Mar '22

Am 28.02.22 um 20:56 schrieb Linus Torvalds: > On Mon, Feb 28, 2022 at 4:19 AM Christian König > <christian.koenig(a)amd.com> wrote: >> I don't think that using the extra variable makes the code in any way >> more reliable or easier to read. > So I think the next step is to do the attached patch (which requires > that "-std=gnu11" that was discussed in the original thread). > > That will guarantee that the 'pos' parameter of list_for_each_entry() > is only updated INSIDE the for_each_list_entry() loop, and can never > point to the (wrongly typed) head entry. > > And I would actually hope that it should actually cause compiler > warnings about possibly uninitialized variables if people then use the > 'pos' pointer outside the loop. Except > > (a) that code in sgx/encl.c currently initializes 'tmp' to NULL for > inexplicable reasons - possibly because it already expected this > behavior > > (b) when I remove that NULL initializer, I still don't get a warning, > because we've disabled -Wno-maybe-uninitialized since it results in so > many false positives. > > Oh well. > > Anyway, give this patch a look, and at least if it's expanded to do > "(pos) = NULL" in the entry statement for the for-loop, it will avoid > the HEAD type confusion that Jakob is working on. And I think in a > cleaner way than the horrid games he plays. > > (But it won't avoid possible CPU speculation of such type confusion. > That, in my opinion, is a completely different issue) Yes, completely agree. > I do wish we could actually poison the 'pos' value after the loop > somehow - but clearly the "might be uninitialized" I was hoping for > isn't the way to do it. > > Anybody have any ideas? I think we should look at the use cases why code is touching (pos) after the loop. Just from skimming over the patches to change this and experience with the drivers/subsystems I help to maintain I think the primary pattern looks something like this: list_for_each_entry(entry, head, member) { if (some_condition_checking(entry)) break; } do_something_with(entry); So the solution should probably not be to change all those use cases to use more temporary variables, but rather to add a list_find_entry(..., condition) macro and consistently use that one instead. Regards, Christian. > > Linus

12 23

[PATCH] ASoC: mediatek: mt8183: support wb bt audio
by Jiaxin Yu 03 Mar '22

03 Mar '22

This patch use "bt-sco-pcm-wb" codec dai driver for wb bt audio. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- This patch depends on: https://patchwork.kernel.org/project/alsa-devel/patch/20220302013533.29068-… sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index b0ec5ebd4f2d..889f9e4a96aa 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -260,7 +260,7 @@ SND_SOC_DAILINK_DEFS(pcm2, SND_SOC_DAILINK_DEFS(i2s0, DAILINK_COMP_ARRAY(COMP_CPU("I2S0")), - DAILINK_COMP_ARRAY(COMP_CODEC("bt-sco", "bt-sco-pcm")), + DAILINK_COMP_ARRAY(COMP_CODEC("bt-sco", "bt-sco-pcm-wb")), DAILINK_COMP_ARRAY(COMP_EMPTY())); SND_SOC_DAILINK_DEFS(i2s1, @@ -291,7 +291,7 @@ SND_SOC_DAILINK_DEFS(i2s3_rt1015p, SND_SOC_DAILINK_DEFS(i2s5, DAILINK_COMP_ARRAY(COMP_CPU("I2S5")), - DAILINK_COMP_ARRAY(COMP_CODEC("bt-sco", "bt-sco-pcm")), + DAILINK_COMP_ARRAY(COMP_CODEC("bt-sco", "bt-sco-pcm-wb")), DAILINK_COMP_ARRAY(COMP_EMPTY())); SND_SOC_DAILINK_DEFS(tdm, @@ -508,7 +508,6 @@ static struct snd_soc_dai_link mt8183_mt6358_ts3a227_dai_links[] = { .no_pcm = 1, .dpcm_capture = 1, .ignore_suspend = 1, - .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, .ops = &mt8183_mt6358_i2s_ops, SND_SOC_DAILINK_REG(i2s0), }, @@ -541,7 +540,6 @@ static struct snd_soc_dai_link mt8183_mt6358_ts3a227_dai_links[] = { .no_pcm = 1, .dpcm_playback = 1, .ignore_suspend = 1, - .be_hw_params_fixup = mt8183_i2s_hw_params_fixup, .ops = &mt8183_mt6358_i2s_ops, SND_SOC_DAILINK_REG(i2s5), }, -- 2.25.1

2 1

[PATCH 0/7 resend] ASoC: fsl_sai: Cleanups and 1:1 bclk:mclk ratio support
by Sascha Hauer 03 Mar '22

03 Mar '22

Resending this with the Cc: list expanded as Nicolin said he doesn't work on FSL platforms at the moment. This series has some updates for the fsl_sai driver: Some general cleanup patches, a bugfix in the ip revision checking and finally the mclk setting is made more accurate and support for 1:1 bclk:mclk setting is added. Ahmad Fatoum (2): ASoC: fsl_sai: simplify register poking in fsl_sai_set_bclk ASoC: fsl_sai: implement 1:1 bclk:mclk ratio support Sascha Hauer (5): ASoC: fsl_sai: Drop unnecessary defines ASoC: fsl_sai: simplify irq return value ASoC: fsl_sai: store full version instead of major/minor ASoC: fsl_sai: Use better variable names ASoC: fsl_sai: use DIV_ROUND_CLOSEST() to calculate divider sound/soc/fsl/fsl_sai.c | 100 ++++++++++++++++++++-------------------- sound/soc/fsl/fsl_sai.h | 9 +--- 2 files changed, 52 insertions(+), 57 deletions(-) -- 2.30.2

1 7

[PATCH 0/2] ASoC: mediatek: mt8192-mt6359: support machine
by Jiaxin Yu 03 Mar '22

03 Mar '22

The series reuses mt8192-mt6359-rt1015-rt5682.c for supporting machine driver with rt1015p speaker amplifier and rt5682s headset codec. Jiaxin Yu (2): ASoC: dt-bindings: mt8192-mt6359: add new compatible for using rt1015p and rt5682 ASoC: mediatek: mt8192: support rt1015p_rt5682s .../sound/mt8192-mt6359-rt1015-rt5682.yaml | 1 + sound/soc/mediatek/Kconfig | 1 + .../mt8192/mt8192-mt6359-rt1015-rt5682.c | 87 +++++++++++++++++-- 3 files changed, 84 insertions(+), 5 deletions(-) -- 2.18.0

2 5

[PATCH v6 0/3] Add support for SoundWire1.6 audio cgcr register control
by Srinivasa Rao Mandadapu 03 Mar '22

03 Mar '22

This patch series is to add v1.6.0 compatible name for qcom soundwire driver and corresponding dt bindings. Changes Since V4: -- Constify static struct qcom_swrm_data global variables. -- Remove redundant swrm_v1_6_data variable. Changes Since V4: -- Dropped audio cgcr control patch due to dependency on clock patches. -- Update dt-bindings as per new reset control properties. Changes Since V3: -- Add v1.6.0 compatible name and soundwire data structure. -- Change macro define name properly. -- Update bindings for new property. -- Change commit message description. -- Change signedoff by sequence. Changes since v2: -- Update error check after ioremap. Changes since v1: -- Add const name to mask value. Srinivasa Rao Mandadapu (3): soundwire: qcom: Add compatible name for v1.6.0 soundwire: qcom: constify static struct qcom_swrm_data global variables dt-bindings: soundwire: qcom: Add bindings for audio CSR reset control property Documentation/devicetree/bindings/soundwire/qcom,sdw.txt | 12 ++++++++++++ drivers/soundwire/qcom.c | 5 +++-- 2 files changed, 15 insertions(+), 2 deletions(-) -- 2.7.4

3 9

Issue in pcm_dsnoop.c in alsa-lib
by GitHub issues - edited 02 Mar '22

02 Mar '22

alsa-project/alsa-lib issue #213 was edited from TE-N-ShengjiuWang: Hi Takashi Iwai, Jaroslav Kysela We encountered an issue in the pcm_dsnoop use case, could you please help to have a look? Issue description: With two instances for dsnoop type device running in parallel, after suspend/resume, one of the instances will be hung in memcpy because the very large copy size is obtained. ``` #3 0x0000ffffa78d5098 in snd_pcm_dsnoop_sync_ptr (pcm=0xaaab06563da0) at pcm_dsnoop.c:158 dsnoop = 0xaaab06563c20 slave_hw_ptr = 64 old_slave_hw_ptr = 533120 avail = 187651522444320 ``` Reason analysis: The root cause that I analysis is that after suspend/resume, one instance will get the SND_PCM_STATE_SUSPENDED state from slave pcm device, then it will do snd_pcm_prepare() and snd_pcm_start(), which will reset the dsnoop->slave_hw_ptr and the hw_ptr of slave pcm device, then the state of this instance is correct. But another instance may not get the SND_PCM_STATE_SUSPENDED state from slave pcm device because slave device may have been recovered by first instance, so the dsnoop->slave_hw_ptr is not reset. but because hw_ptr of slave pcm device has been reset, so there will be a very large "avail" size. Solution: I didn't come up with a fix for this issue, seems there is no easy way to let another instance know this case and reset the dsnoop->slave_hw_ptr, could you please help? Issue URL : https://github.com/alsa-project/alsa-lib/issues/213 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 0/2] ASoC: codecs: Add Awinic AW8738 audio amplifier driver
by Stephan Gerhold 02 Mar '22

02 Mar '22

This series adds a simple driver and DT schema for the Awinic AW8738 audio amplifier. It's fairly simple - the main difference to simple-amplifier is that there is a "one-wire pulse control" that allows configuring the amplifier to one of a few pre-defined modes. Jonathan Albrieux (1): ASoC: codecs: Add Awinic AW8738 audio amplifier driver Stephan Gerhold (1): ASoC: dt-bindings: Add schema for "awinic,aw8738" .../bindings/sound/awinic,aw8738.yaml | 48 ++++++++ sound/soc/codecs/Kconfig | 10 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/aw8738.c | 104 ++++++++++++++++++ 4 files changed, 164 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/awinic,aw8738.yaml create mode 100644 sound/soc/codecs/aw8738.c -- 2.35.1

3 11

[PATCH 0/8] ASoC: Intel: machine driver updates for 5.18
by Pierre-Louis Bossart 02 Mar '22

02 Mar '22

Two cleanups to remove an unused filename and typos, and one addition of an ACPI matching table for a Dell SoundWire SKU without local microphones. The main change is the addition of a common 'sof-ssp-amp' machine driver for devices with amplifiers only (no headset codec) and different connections using I2S links (Bluetooth offload, HDMI receiver). It's likely that the amplifier will be swapped out by OEMs, this machine driver provides a relatively generic solution to avoid copy-paste of similar solutions. Brent Lu (4): ASoC: Intel: boards: create sof-realtek-common module ASoC: Intel: sof_rt1308: move rt1308 code to common module ASoC: Intel: cirrus-common: support cs35l41 amplifier ASoC: Intel: sof_ssp_amp: rename driver and support cs35l41 amplifier Libin Yang (1): ASoC: Intel: soc-acpi: add entries in ADL match table Pierre-Louis Bossart (2): ASoC: soc-acpi: remove sof_fw_filename ASoC: Intel: boards: fix spelling in comments balamurugan.c (1): ASoC: Intel: add RT1308 I2S machine driver and HDMI-in capture via I2S support. include/sound/soc-acpi.h | 2 - sound/soc/intel/boards/Kconfig | 24 + sound/soc/intel/boards/Makefile | 12 +- 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_cx2072x.c | 2 +- sound/soc/intel/boards/bytcht_da7213.c | 2 +- sound/soc/intel/boards/bytcht_es8316.c | 2 +- sound/soc/intel/boards/bytcr_rt5640.c | 2 +- sound/soc/intel/boards/bytcr_rt5651.c | 2 +- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 4 +- sound/soc/intel/boards/cht_bsw_nau8824.c | 4 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 +- sound/soc/intel/boards/cht_bsw_rt5672.c | 2 +- sound/soc/intel/boards/glk_rt5682_max98357a.c | 2 +- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/sof_cirrus_common.c | 163 ++++++ sound/soc/intel/boards/sof_cirrus_common.h | 25 + sound/soc/intel/boards/sof_realtek_common.c | 107 ++++ sound/soc/intel/boards/sof_realtek_common.h | 4 + sound/soc/intel/boards/sof_rt5682.c | 1 + sound/soc/intel/boards/sof_ssp_amp.c | 483 ++++++++++++++++++ .../intel/common/soc-acpi-intel-adl-match.c | 44 +- .../intel/common/soc-acpi-intel-bxt-match.c | 6 - .../intel/common/soc-acpi-intel-byt-match.c | 11 - .../intel/common/soc-acpi-intel-cht-match.c | 12 - .../intel/common/soc-acpi-intel-cml-match.c | 11 - .../intel/common/soc-acpi-intel-cnl-match.c | 4 - .../intel/common/soc-acpi-intel-ehl-match.c | 1 - .../intel/common/soc-acpi-intel-glk-match.c | 6 - .../intel/common/soc-acpi-intel-hda-match.c | 2 - .../common/soc-acpi-intel-hsw-bdw-match.c | 5 - .../intel/common/soc-acpi-intel-icl-match.c | 5 - .../intel/common/soc-acpi-intel-jsl-match.c | 7 - .../intel/common/soc-acpi-intel-tgl-match.c | 19 +- sound/soc/sof/intel/hda.c | 5 +- sound/soc/sof/intel/pci-tng.c | 1 - 40 files changed, 879 insertions(+), 117 deletions(-) create mode 100644 sound/soc/intel/boards/sof_cirrus_common.c create mode 100644 sound/soc/intel/boards/sof_cirrus_common.h create mode 100644 sound/soc/intel/boards/sof_ssp_amp.c -- 2.25.1

3 15

[PATCH] ASoC: amd: use asoc_substream_to_rtd()
by Zhen Ni 02 Mar '22

02 Mar '22

Uses asoc_substream_to_rtd() helper. Signed-off-by: Zhen Ni <nizhen(a)uniontech.com> --- sound/soc/amd/vangogh/acp5x-mach.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/vangogh/acp5x-mach.c b/sound/soc/amd/vangogh/acp5x-mach.c index 14cf325e4b23..f2c32e5ed24f 100644 --- a/sound/soc/amd/vangogh/acp5x-mach.c +++ b/sound/soc/amd/vangogh/acp5x-mach.c @@ -101,7 +101,7 @@ static const struct snd_pcm_hw_constraint_list constraints_channels = { static int acp5x_8821_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_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct acp5x_platform_info *machine = snd_soc_card_get_drvdata(card); @@ -119,7 +119,7 @@ static int acp5x_8821_startup(struct snd_pcm_substream *substream) static int acp5x_nau8821_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_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, @@ -141,7 +141,7 @@ static int acp5x_nau8821_hw_params(struct snd_pcm_substream *substream, static int acp5x_cs35l41_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_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct acp5x_platform_info *machine = snd_soc_card_get_drvdata(card); @@ -158,7 +158,7 @@ static int acp5x_cs35l41_startup(struct snd_pcm_substream *substream) static int acp5x_cs35l41_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_pcm_runtime *rtd = asoc_substream_to_rtd(substream); struct snd_soc_card *card = rtd->card; struct snd_soc_dai *codec_dai; int ret, i; -- 2.20.1

2 1

[PATCH] ASoC: atmel_ssc_dai: Handle errors for clk_enable
by Jiasheng Jiang 02 Mar '22

02 Mar '22

As the potential failure of the clk_enable(), it should be better to check it and return error if fals. Fixes: cbaadf0f90d6 ("ASoC: atmel_ssc_dai: refactor the startup and shutdown") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/soc/atmel/atmel_ssc_dai.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 26e2bc690d86..c1dea8d62416 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -280,7 +280,10 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, /* Enable PMC peripheral clock for this SSC */ pr_debug("atmel_ssc_dai: Starting clock\n"); - clk_enable(ssc_p->ssc->clk); + ret = clk_enable(ssc_p->ssc->clk); + if (ret) + return ret; + ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk); /* Reset the SSC unless initialized to keep it in a clean state */ -- 2.25.1

3 3

[PATCH v2] ASoC: soc-compress: prevent the potentially use of null pointer
by Jiasheng Jiang 02 Mar '22

02 Mar '22

There is one call trace that snd_soc_register_card() ->snd_soc_bind_card()->soc_init_pcm_runtime() ->snd_soc_dai_compress_new()->snd_soc_new_compress(). In the trace the 'codec_dai' transfers from card->dai_link, and we can see from the snd_soc_add_pcm_runtime() in snd_soc_bind_card() that, if value of card->dai_link->num_codecs is 0, then 'codec_dai' could be null pointer caused by index out of bound in 'asoc_rtd_to_codec(rtd, 0)'. And snd_soc_register_card() is called by various platforms. Therefore, it is better to add the check in the case of misusing. And because 'cpu_dai' has already checked in soc_init_pcm_runtime(), there is no need to check again. Adding the check as follow, then if 'codec_dai' is null, snd_soc_new_compress() will not pass through the check 'if (playback + capture != 1)', avoiding the leftover use of 'codec_dai'. Fixes: 467fece ("ASoC: soc-dai: move snd_soc_dai_stream_valid() to soc-dai.c") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/soc-compress.c | 14 ++++++++------ 1 file changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index b4f5935..67c3df1 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -535,12 +535,14 @@ int snd_soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) } /* check client and interface hw capabilities */ - if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && - snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_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_CAPTURE)) - capture = 1; + if (codec_dai) { + if (snd_soc_dai_stream_valid(codec_dai, SNDRV_PCM_STREAM_PLAYBACK) && + snd_soc_dai_stream_valid(cpu_dai, SNDRV_PCM_STREAM_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_CAPTURE)) + capture = 1; + } /* * Compress devices are unidirectional so only one of the directions -- 2.7.4

2 1

[PATCH] ASoC: dwc-i2s: Handle errors for clk_enable
by Jiasheng Jiang 02 Mar '22

02 Mar '22

As the potential failure of the clk_enable(), it should be better to check it, as same as clk_prepare_enable(). Fixes: c9afc1834e81 ("ASoC: dwc: Disallow building designware_pcm as a module") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/soc/dwc/dwc-i2s.c | 17 ++++++++++++----- 1 file changed, 12 insertions(+), 5 deletions(-) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index 5cb58929090d..1edac3e10f34 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -403,9 +403,13 @@ static int dw_i2s_runtime_suspend(struct device *dev) static int dw_i2s_runtime_resume(struct device *dev) { struct dw_i2s_dev *dw_dev = dev_get_drvdata(dev); + int ret; - if (dw_dev->capability & DW_I2S_MASTER) - clk_enable(dw_dev->clk); + if (dw_dev->capability & DW_I2S_MASTER) { + ret = clk_enable(dw_dev->clk); + if (ret) + return ret; + } return 0; } @@ -422,10 +426,13 @@ static int dw_i2s_resume(struct snd_soc_component *component) { struct dw_i2s_dev *dev = snd_soc_component_get_drvdata(component); struct snd_soc_dai *dai; - int stream; + int stream, ret; - if (dev->capability & DW_I2S_MASTER) - clk_enable(dev->clk); + if (dev->capability & DW_I2S_MASTER) { + ret = clk_enable(dev->clk); + if (ret) + return ret; + } for_each_component_dais(component, dai) { for_each_pcm_streams(stream) -- 2.25.1

2 1

[PATCH] ASoC: mxs-saif: Handle errors for clk_enable
by Jiasheng Jiang 02 Mar '22

02 Mar '22

As the potential failure of the clk_enable(), it should be better to check it, like mxs_saif_trigger(). Fixes: d0ba4c014934 ("ASoC: mxs-saif: set a base clock rate for EXTMASTER mode work") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/soc/mxs/mxs-saif.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index 6a2d24d48964..879c1221a809 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -455,7 +455,10 @@ static int mxs_saif_hw_params(struct snd_pcm_substream *substream, * basic clock which should be fast enough for the internal * logic. */ - clk_enable(saif->clk); + ret = clk_enable(saif->clk); + if (ret) + return ret; + ret = clk_set_rate(saif->clk, 24000000); clk_disable(saif->clk); if (ret) -- 2.25.1

2 1

[PATCH] ASoC: qcom: Fix error code in lpass_platform_copy()
by Dan Carpenter 02 Mar '22

02 Mar '22

The copy_to/from_user() functions return the number of bytes remaining to be copied. This function needs to return negative error codes because snd_soc_pcm_component_copy_user() treats positive returns as success in soc_component_ret(). Fixes: 7d7209557b67 ("ASoC: qcom: Add support for codec dma driver") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- sound/soc/qcom/lpass-platform.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index bf180a594c19..620312529c2f 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -1228,15 +1228,19 @@ static int lpass_platform_copy(struct snd_soc_component *component, channel * (rt->dma_bytes / rt->channels)); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (is_cdc_dma_port(dai_id)) + if (is_cdc_dma_port(dai_id)) { ret = copy_from_user_toio(dma_buf, buf, bytes); - else - ret = copy_from_user((void __force *)dma_buf, buf, bytes); + } else { + if (copy_from_user((void __force *)dma_buf, buf, bytes)) + ret = -EFAULT; + } } else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { - if (is_cdc_dma_port(dai_id)) + if (is_cdc_dma_port(dai_id)) { ret = copy_to_user_fromio(buf, dma_buf, bytes); - else - ret = copy_to_user(buf, (void __force *)dma_buf, bytes); + } else { + if (copy_to_user(buf, (void __force *)dma_buf, bytes)) + ret = -EFAULT; + } } return ret; -- 2.20.1

2 1

[PATCH] ASoC: soc-generic-dmaengine-pcm: set period_bytes_min based on maxburst
by Sascha Hauer 02 Mar '22

02 Mar '22

In dmaengine_pcm_set_runtime_hwparams() period_bytes_min is hardcoded to 256. For some applications that may be too big. This patch changes that to calculate the value based on dma_data->maxburst. The correct value would be maxburst multiplied by the address width of the hardware FIFO. Unfortunately the address width is dynamically calculated based on the stream parameters and is not known at open time, so the worst case is chosen here which is 8 bytes, the maximum that is supported by dmaengine drivers. Not all drivers may set a maxburst value, so we fall back to the previously used hardcoded value of 256 bytes. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/soc-generic-dmaengine-pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 99253de29a74e..734d46b9783f7 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -122,7 +122,9 @@ dmaengine_pcm_set_runtime_hwparams(struct snd_soc_component *component, SNDRV_PCM_INFO_INTERLEAVED; hw.periods_min = 2; hw.periods_max = UINT_MAX; - hw.period_bytes_min = 256; + hw.period_bytes_min = dma_data->maxburst * DMA_SLAVE_BUSWIDTH_8_BYTES; + if (!hw.period_bytes_min) + hw.period_bytes_min = 256; hw.period_bytes_max = dma_get_max_seg_size(dma_dev); hw.buffer_bytes_max = SIZE_MAX; hw.fifo_size = dma_data->fifo_size; -- 2.30.2

2 1

[PATCH -next] ASoC: amd: Fix an ignored error return from platform_get_irq_byname()
by Yang Li 02 Mar '22

02 Mar '22

The return from the call to platform_get_irq_byname() is int, it can be a negative error code, however this is being assigned to an unsigned int variable 'adata->i2s_irq', so assign the value to 'ret' concurrently to solve this problem without affecting other functions. Eliminate the following coccicheck warning: ./sound/soc/amd/acp/acp-renoir.c:286:5-19: WARNING: Unsigned expression compared with zero: adata -> i2s_irq < 0 Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Fixes: 3304a242f45a ("ASoC: amd: Use platform_get_irq_byname() to get the interrupt") Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/amd/acp/acp-renoir.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp/acp-renoir.c b/sound/soc/amd/acp/acp-renoir.c index 738cf2e2b973..64e824161091 100644 --- a/sound/soc/amd/acp/acp-renoir.c +++ b/sound/soc/amd/acp/acp-renoir.c @@ -282,8 +282,8 @@ static int renoir_audio_probe(struct platform_device *pdev) if (!adata->acp_base) return -ENOMEM; - adata->i2s_irq = platform_get_irq_byname(pdev, "acp_dai_irq"); - if (adata->i2s_irq < 0) + adata->i2s_irq = ret = platform_get_irq_byname(pdev, "acp_dai_irq"); + if (ret < 0) return -ENODEV; adata->dev = dev; -- 2.20.1.7.g153144c

3 3

[PATCH] ASoC: rockchip: i2s_tdm: fix runtime pm imbalance on error
by Zhen Ni 02 Mar '22

02 Mar '22

pm_runtime_get_sync() increments the runtime PM usage counter even the call returns an error code. Thus a pairing decrement is needed on the error handling path to keep the counter balanced. Signed-off-by: Zhen Ni <nizhen(a)uniontech.com> --- sound/soc/rockchip/rockchip_i2s_tdm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/rockchip/rockchip_i2s_tdm.c b/sound/soc/rockchip/rockchip_i2s_tdm.c index 5f9cb5c4c7f0..4691a76b499d 100644 --- a/sound/soc/rockchip/rockchip_i2s_tdm.c +++ b/sound/soc/rockchip/rockchip_i2s_tdm.c @@ -1739,8 +1739,10 @@ static int __maybe_unused rockchip_i2s_tdm_resume(struct device *dev) int ret; ret = pm_runtime_get_sync(dev); - if (ret < 0) + if (ret < 0) { + pm_runtime_put(dev); return ret; + } ret = regcache_sync(i2s_tdm->regmap); pm_runtime_put(dev); -- 2.20.1

2 1

[PATCH] Documentation: sound: fix typo in control-names.rst
by SPeak Shen 02 Mar '22

02 Mar '22

From: Sunrisepeak <speakshen(a)163.com> change 'cannel' to 'channel' Signed-off-by: Sunrisepeak <speakshen(a)163.com> --- Documentation/sound/designs/control-names.rst | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/sound/designs/control-names.rst b/Documentation/sound/designs/control-names.rst index 7fedd0f33..765ff9b5b 100644 --- a/Documentation/sound/designs/control-names.rst +++ b/Documentation/sound/designs/control-names.rst @@ -34,7 +34,7 @@ CHANNEL Front front left/right channels Surround rear left/right in 4.0/5.1 surround CLFE C/LFE channels -Center center cannel +Center center channel LFE LFE channel Side side left/right for 7.1 surround ============ ================================================== -- 2.17.1

2 1

[PATCH] ALSA: intel_hdmi: Fix reference to PCM buffer address
by Zhen Ni 02 Mar '22

02 Mar '22

PCM buffers might be allocated dynamically when the buffer preallocation failed or a larger buffer is requested, and it's not guaranteed that substream->dma_buffer points to the actually used buffer. The driver needs to refer to substream->runtime->dma_addr instead for the buffer address. Signed-off-by: Zhen Ni <nizhen(a)uniontech.com> --- sound/x86/intel_hdmi_audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/x86/intel_hdmi_audio.c b/sound/x86/intel_hdmi_audio.c index 1c94eaff1931..4a3ff6468aa7 100644 --- a/sound/x86/intel_hdmi_audio.c +++ b/sound/x86/intel_hdmi_audio.c @@ -1261,7 +1261,7 @@ static int had_pcm_mmap(struct snd_pcm_substream *substream, { vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); return remap_pfn_range(vma, vma->vm_start, - substream->dma_buffer.addr >> PAGE_SHIFT, + substream->runtime->dma_addr >> PAGE_SHIFT, vma->vm_end - vma->vm_start, vma->vm_page_prot); } -- 2.20.1

2 1

Re: [PATCH 5/5] spi: make remove callback a void function
by Pavel Machek 02 Mar '22

02 Mar '22

Hi! > The value returned by an spi driver's remove function is mostly ignored. > (Only an error message is printed if the value is non-zero that the > error is ignored.) > > So change the prototype of the remove function to return no value. This > way driver authors are not tempted to assume that passing an error to > the upper layer is a good idea. All drivers are adapted accordingly. > There is no intended change of behaviour, all callbacks were prepared to > return 0 before. Acked-by: Pavel Machek <pavel(a)ucw.cz> Pavel -- http://www.livejournal.com/~pavelmachek

1 0

Re: Re: [PATCH v2] ASoC: fsi: Add check for clk_enable
by Jiasheng Jiang 02 Mar '22

02 Mar '22

On Wed, Mar 02, 2022 at 12:35:02PM +0800, Kuninori Morimoto wrote: >> +disable_xck: >> + clk_disable(clock->xck); >> +disable_ick: >> + clk_disable(clock->ick); >> +err: >> + return ret; >> } > > I think disable_ick() / disable_xck() order are inverted ? Thanks, I will submit a v3 to solve it. Jiang

1 0

[PATCH v2] ASoC: fsi: Add check for clk_enable
by Jiasheng Jiang 02 Mar '22

02 Mar '22

As the potential failure of the clk_enable(), it should be better to check it and return error if fails. Fixes: ab6f6d85210c ("ASoC: fsi: add master clock control functions") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- Changelog: v1 -> v2 * Change 1. Seperate the error handler. --- sound/soc/sh/fsi.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index cdf3b7f69ba7..91050478844a 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -816,14 +816,27 @@ static int fsi_clk_enable(struct device *dev, return ret; } - clk_enable(clock->xck); - clk_enable(clock->ick); - clk_enable(clock->div); + ret = clk_enable(clock->xck); + if (ret) + goto err; + ret = clk_enable(clock->ick); + if (ret) + goto disable_xck; + ret = clk_enable(clock->div); + if (ret) + goto disable_ick; clock->count++; } return ret; + +disable_xck: + clk_disable(clock->xck); +disable_ick: + clk_disable(clock->ick); +err: + return ret; } static int fsi_clk_disable(struct device *dev, -- 2.25.1

2 1

Re: Re: [PATCH] ASoC: fsi: Add check for clk_enable
by Jiasheng Jiang 02 Mar '22

02 Mar '22

On Tue, Mar 01, 2022 at 08:44:12PM +0800, Mark Brown wrote: >> +err: >> + clk_disable(clock->xck); >> + clk_disable(clock->ick); >> + clk_disable(clock->div); > > You need separate labels for each enable so that we don't end up > disabling clocks we didn't enable, that would also be a bug. Thanks, I have submitted a v2 to fix it. Jiang

1 0

[PATCH] dt-bindings: Another pass removing cases of 'allOf' containing a '$ref'
by Rob Herring 02 Mar '22

02 Mar '22

Another pass at removing unnecessary use of 'allOf' with a '$ref'. json-schema versions draft7 and earlier have a weird behavior in that any keywords combined with a '$ref' are ignored (silently). The correct form was to put a '$ref' under an 'allOf'. This behavior is now changed in the 2019-09 json-schema spec and '$ref' can be mixed with other keywords. Cc: Krzysztof Kozlowski <krzysztof.kozlowski(a)canonical.com> Cc: Thierry Reding <thierry.reding(a)gmail.com> Cc: Sam Ravnborg <sam(a)ravnborg.org> Cc: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> Cc: Pavel Machek <pavel(a)ucw.cz> Cc: Lee Jones <lee.jones(a)linaro.org> Cc: Guenter Roeck <groeck(a)chromium.org> Cc: Miquel Raynal <miquel.raynal(a)bootlin.com> Cc: Richard Weinberger <richard(a)nod.at> Cc: Vignesh Raghavendra <vigneshr(a)ti.com> Cc: "David S. Miller" <davem(a)davemloft.net> Cc: Jakub Kicinski <kuba(a)kernel.org> Cc: Kishon Vijay Abraham I <kishon(a)ti.com> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Sebastian Reichel <sre(a)kernel.org> Cc: Bjorn Andersson <bjorn.andersson(a)linaro.org> Cc: Mathieu Poirier <mathieu.poirier(a)linaro.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Laurent Pinchart <laurent.pinchart(a)ideasonboard.com> Cc: dri-devel(a)lists.freedesktop.org Cc: linux-arm-kernel(a)lists.infradead.org Cc: linux-input(a)vger.kernel.org Cc: linux-leds(a)vger.kernel.org Cc: linux-mtd(a)lists.infradead.org Cc: netdev(a)vger.kernel.org Cc: linux-phy(a)lists.infradead.org Cc: linux-pm(a)vger.kernel.org Cc: linux-remoteproc(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Cc: linux-spi(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../bindings/connector/usb-connector.yaml | 3 +-- .../bindings/display/brcm,bcm2711-hdmi.yaml | 3 +-- .../bindings/display/bridge/adi,adv7511.yaml | 5 ++--- .../bindings/display/bridge/synopsys,dw-hdmi.yaml | 5 ++--- .../bindings/display/panel/display-timings.yaml | 3 +-- .../devicetree/bindings/display/ste,mcde.yaml | 4 ++-- .../devicetree/bindings/input/adc-joystick.yaml | 9 ++++----- .../bindings/leds/cznic,turris-omnia-leds.yaml | 3 +-- .../devicetree/bindings/leds/leds-lp50xx.yaml | 3 +-- .../devicetree/bindings/mfd/google,cros-ec.yaml | 12 ++++-------- .../devicetree/bindings/mtd/nand-controller.yaml | 8 +++----- .../bindings/mtd/rockchip,nand-controller.yaml | 3 +-- .../devicetree/bindings/net/ti,cpsw-switch.yaml | 3 +-- .../bindings/phy/phy-stm32-usbphyc.yaml | 3 +-- .../bindings/power/supply/sbs,sbs-manager.yaml | 4 +--- .../bindings/remoteproc/ti,k3-r5f-rproc.yaml | 3 +-- .../devicetree/bindings/soc/ti/ti,pruss.yaml | 15 +++------------ .../devicetree/bindings/sound/st,stm32-sai.yaml | 3 +-- .../devicetree/bindings/sound/tlv320adcx140.yaml | 13 ++++++------- .../devicetree/bindings/spi/spi-controller.yaml | 4 +--- .../devicetree/bindings/usb/st,stusb160x.yaml | 4 +--- 21 files changed, 39 insertions(+), 74 deletions(-) diff --git a/Documentation/devicetree/bindings/connector/usb-connector.yaml b/Documentation/devicetree/bindings/connector/usb-connector.yaml index 7eb8659fa610..0420fa563532 100644 --- a/Documentation/devicetree/bindings/connector/usb-connector.yaml +++ b/Documentation/devicetree/bindings/connector/usb-connector.yaml @@ -104,8 +104,7 @@ properties: - "1.5A" and "3.0A", 5V 1.5A and 5V 3.0A respectively, as defined in USB Type-C Cable and Connector specification, when Power Delivery is not supported. - allOf: - - $ref: /schemas/types.yaml#/definitions/string + $ref: /schemas/types.yaml#/definitions/string enum: - default - 1.5A diff --git a/Documentation/devicetree/bindings/display/brcm,bcm2711-hdmi.yaml b/Documentation/devicetree/bindings/display/brcm,bcm2711-hdmi.yaml index a1d5a32660e0..a9d34dd7bbc5 100644 --- a/Documentation/devicetree/bindings/display/brcm,bcm2711-hdmi.yaml +++ b/Documentation/devicetree/bindings/display/brcm,bcm2711-hdmi.yaml @@ -72,8 +72,7 @@ properties: - const: hpd-removed ddc: - allOf: - - $ref: /schemas/types.yaml#/definitions/phandle + $ref: /schemas/types.yaml#/definitions/phandle description: > Phandle of the I2C controller used for DDC EDID probing diff --git a/Documentation/devicetree/bindings/display/bridge/adi,adv7511.yaml b/Documentation/devicetree/bindings/display/bridge/adi,adv7511.yaml index d3dd7a79b909..f08a01dfedf3 100644 --- a/Documentation/devicetree/bindings/display/bridge/adi,adv7511.yaml +++ b/Documentation/devicetree/bindings/display/bridge/adi,adv7511.yaml @@ -76,9 +76,8 @@ properties: adi,input-depth: description: Number of bits per color component at the input. - allOf: - - $ref: /schemas/types.yaml#/definitions/uint32 - - enum: [ 8, 10, 12 ] + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [ 8, 10, 12 ] adi,input-colorspace: description: Input color space. diff --git a/Documentation/devicetree/bindings/display/bridge/synopsys,dw-hdmi.yaml b/Documentation/devicetree/bindings/display/bridge/synopsys,dw-hdmi.yaml index 9be44a682e67..b00246faea57 100644 --- a/Documentation/devicetree/bindings/display/bridge/synopsys,dw-hdmi.yaml +++ b/Documentation/devicetree/bindings/display/bridge/synopsys,dw-hdmi.yaml @@ -26,9 +26,8 @@ properties: reg-io-width: description: Width (in bytes) of the registers specified by the reg property. - allOf: - - $ref: /schemas/types.yaml#/definitions/uint32 - - enum: [1, 4] + $ref: /schemas/types.yaml#/definitions/uint32 + enum: [1, 4] default: 1 clocks: diff --git a/Documentation/devicetree/bindings/display/panel/display-timings.yaml b/Documentation/devicetree/bindings/display/panel/display-timings.yaml index 56903ded005e..6d30575819d3 100644 --- a/Documentation/devicetree/bindings/display/panel/display-timings.yaml +++ b/Documentation/devicetree/bindings/display/panel/display-timings.yaml @@ -31,8 +31,7 @@ properties: patternProperties: "^timing": type: object - allOf: - - $ref: panel-timing.yaml# + $ref: panel-timing.yaml# additionalProperties: false diff --git a/Documentation/devicetree/bindings/display/ste,mcde.yaml b/Documentation/devicetree/bindings/display/ste,mcde.yaml index de0c678b3c29..564ea845c82e 100644 --- a/Documentation/devicetree/bindings/display/ste,mcde.yaml +++ b/Documentation/devicetree/bindings/display/ste,mcde.yaml @@ -58,8 +58,8 @@ patternProperties: "^dsi@[0-9a-f]+$": description: subnodes for the three DSI host adapters type: object - allOf: - - $ref: dsi-controller.yaml# + $ref: dsi-controller.yaml# + properties: compatible: const: ste,mcde-dsi diff --git a/Documentation/devicetree/bindings/input/adc-joystick.yaml b/Documentation/devicetree/bindings/input/adc-joystick.yaml index 721878d5b7af..2ee04e03bc22 100644 --- a/Documentation/devicetree/bindings/input/adc-joystick.yaml +++ b/Documentation/devicetree/bindings/input/adc-joystick.yaml @@ -61,11 +61,10 @@ patternProperties: description: EV_ABS specific event code generated by the axis. abs-range: - allOf: - - $ref: /schemas/types.yaml#/definitions/uint32-array - - items: - - description: minimum value - - description: maximum value + $ref: /schemas/types.yaml#/definitions/uint32-array + items: + - description: minimum value + - description: maximum value description: > Minimum and maximum values produced by the axis. For an ABS_X axis this will be the left-most and right-most diff --git a/Documentation/devicetree/bindings/leds/cznic,turris-omnia-leds.yaml b/Documentation/devicetree/bindings/leds/cznic,turris-omnia-leds.yaml index c7ed2871da06..9362b1ef9e88 100644 --- a/Documentation/devicetree/bindings/leds/cznic,turris-omnia-leds.yaml +++ b/Documentation/devicetree/bindings/leds/cznic,turris-omnia-leds.yaml @@ -32,8 +32,7 @@ properties: patternProperties: "^multi-led@[0-9a-b]$": type: object - allOf: - - $ref: leds-class-multicolor.yaml# + $ref: leds-class-multicolor.yaml# description: This node represents one of the RGB LED devices on Turris Omnia. No subnodes need to be added for subchannels since this controller only diff --git a/Documentation/devicetree/bindings/leds/leds-lp50xx.yaml b/Documentation/devicetree/bindings/leds/leds-lp50xx.yaml index c192b5feadc7..f12fe5b53f30 100644 --- a/Documentation/devicetree/bindings/leds/leds-lp50xx.yaml +++ b/Documentation/devicetree/bindings/leds/leds-lp50xx.yaml @@ -55,8 +55,7 @@ properties: patternProperties: '^multi-led@[0-9a-f]$': type: object - allOf: - - $ref: leds-class-multicolor.yaml# + $ref: leds-class-multicolor.yaml# properties: reg: minItems: 1 diff --git a/Documentation/devicetree/bindings/mfd/google,cros-ec.yaml b/Documentation/devicetree/bindings/mfd/google,cros-ec.yaml index d1f53bd449f7..a52f3a82283c 100644 --- a/Documentation/devicetree/bindings/mfd/google,cros-ec.yaml +++ b/Documentation/devicetree/bindings/mfd/google,cros-ec.yaml @@ -38,18 +38,14 @@ properties: description: This property specifies the delay in usecs between the assertion of the CS and the first clock pulse. - allOf: - - $ref: /schemas/types.yaml#/definitions/uint32 - - default: 0 - - minimum: 0 + $ref: /schemas/types.yaml#/definitions/uint32 + default: 0 google,cros-ec-spi-msg-delay: description: This property specifies the delay in usecs between messages. - allOf: - - $ref: /schemas/types.yaml#/definitions/uint32 - - default: 0 - - minimum: 0 + $ref: /schemas/types.yaml#/definitions/uint32 + default: 0 google,has-vbc-nvram: description: diff --git a/Documentation/devicetree/bindings/mtd/nand-controller.yaml b/Documentation/devicetree/bindings/mtd/nand-controller.yaml index bd217e6f5018..0714f614550d 100644 --- a/Documentation/devicetree/bindings/mtd/nand-controller.yaml +++ b/Documentation/devicetree/bindings/mtd/nand-controller.yaml @@ -58,8 +58,7 @@ patternProperties: Contains the native Ready/Busy IDs. nand-ecc-engine: - allOf: - - $ref: /schemas/types.yaml#/definitions/phandle + $ref: /schemas/types.yaml#/definitions/phandle description: | A phandle on the hardware ECC engine if any. There are basically three possibilities: @@ -79,9 +78,8 @@ patternProperties: description: Do not use any ECC correction. nand-ecc-placement: - allOf: - - $ref: /schemas/types.yaml#/definitions/string - - enum: [ oob, interleaved ] + $ref: /schemas/types.yaml#/definitions/string + enum: [ oob, interleaved ] description: Location of the ECC bytes. This location is unknown by default but can be explicitly set to "oob", if all ECC bytes are diff --git a/Documentation/devicetree/bindings/mtd/rockchip,nand-controller.yaml b/Documentation/devicetree/bindings/mtd/rockchip,nand-controller.yaml index 0922536b1811..d681a4676f06 100644 --- a/Documentation/devicetree/bindings/mtd/rockchip,nand-controller.yaml +++ b/Documentation/devicetree/bindings/mtd/rockchip,nand-controller.yaml @@ -96,8 +96,7 @@ patternProperties: rockchip,boot-ecc-strength: enum: [16, 24, 40, 60, 70] - allOf: - - $ref: /schemas/types.yaml#/definitions/uint32 + $ref: /schemas/types.yaml#/definitions/uint32 description: | If specified it indicates that a different BCH/ECC setting is supported by the boot ROM. diff --git a/Documentation/devicetree/bindings/net/ti,cpsw-switch.yaml b/Documentation/devicetree/bindings/net/ti,cpsw-switch.yaml index 07a00f53adbf..31bf825c6598 100644 --- a/Documentation/devicetree/bindings/net/ti,cpsw-switch.yaml +++ b/Documentation/devicetree/bindings/net/ti,cpsw-switch.yaml @@ -88,8 +88,7 @@ properties: type: object description: CPSW external ports - allOf: - - $ref: ethernet-controller.yaml# + $ref: ethernet-controller.yaml# properties: reg: diff --git a/Documentation/devicetree/bindings/phy/phy-stm32-usbphyc.yaml b/Documentation/devicetree/bindings/phy/phy-stm32-usbphyc.yaml index 267b695215b6..dc287d428e49 100644 --- a/Documentation/devicetree/bindings/phy/phy-stm32-usbphyc.yaml +++ b/Documentation/devicetree/bindings/phy/phy-stm32-usbphyc.yaml @@ -76,8 +76,7 @@ patternProperties: connector: type: object - allOf: - - $ref: ../connector/usb-connector.yaml + $ref: /schemas/connector/usb-connector.yaml properties: vbus-supply: true diff --git a/Documentation/devicetree/bindings/power/supply/sbs,sbs-manager.yaml b/Documentation/devicetree/bindings/power/supply/sbs,sbs-manager.yaml index 72e8f274c791..99f506d6b0a0 100644 --- a/Documentation/devicetree/bindings/power/supply/sbs,sbs-manager.yaml +++ b/Documentation/devicetree/bindings/power/supply/sbs,sbs-manager.yaml @@ -46,9 +46,7 @@ additionalProperties: false patternProperties: "^i2c@[1-4]$": type: object - - allOf: - - $ref: /schemas/i2c/i2c-controller.yaml# + $ref: /schemas/i2c/i2c-controller.yaml# examples: - | diff --git a/Documentation/devicetree/bindings/remoteproc/ti,k3-r5f-rproc.yaml b/Documentation/devicetree/bindings/remoteproc/ti,k3-r5f-rproc.yaml index 0868154a0364..fb9605f0655b 100644 --- a/Documentation/devicetree/bindings/remoteproc/ti,k3-r5f-rproc.yaml +++ b/Documentation/devicetree/bindings/remoteproc/ti,k3-r5f-rproc.yaml @@ -95,8 +95,7 @@ patternProperties: addresses. Cache and memory access settings are provided through a Memory Protection Unit (MPU), programmable only from the R5Fs. - allOf: - - $ref: /schemas/arm/keystone/ti,k3-sci-common.yaml# + $ref: /schemas/arm/keystone/ti,k3-sci-common.yaml# properties: compatible: diff --git a/Documentation/devicetree/bindings/soc/ti/ti,pruss.yaml b/Documentation/devicetree/bindings/soc/ti/ti,pruss.yaml index 9d128b9e7deb..64461d432004 100644 --- a/Documentation/devicetree/bindings/soc/ti/ti,pruss.yaml +++ b/Documentation/devicetree/bindings/soc/ti/ti,pruss.yaml @@ -281,10 +281,7 @@ patternProperties: PRUSS INTC Node. Each PRUSS has a single interrupt controller instance that is common to all the PRU cores. This should be represented as an interrupt-controller node. - - allOf: - - $ref: /schemas/interrupt-controller/ti,pruss-intc.yaml# - + $ref: /schemas/interrupt-controller/ti,pruss-intc.yaml# type: object mdio@[a-f0-9]+$: @@ -292,10 +289,7 @@ patternProperties: MDIO Node. Each PRUSS has an MDIO module that can be used to control external PHYs. The MDIO module used within the PRU-ICSS is an instance of the MDIO Controller used in TI Davinci SoCs. - - allOf: - - $ref: /schemas/net/ti,davinci-mdio.yaml# - + $ref: /schemas/net/ti,davinci-mdio.yaml# type: object "^(pru|rtu|txpru)@[0-9a-f]+$": @@ -305,10 +299,7 @@ patternProperties: inactive by using the standard DT string property, "status". The ICSSG IP present on K3 SoCs have additional auxiliary PRU cores with slightly different IP integration. - - allOf: - - $ref: /schemas/remoteproc/ti,pru-rproc.yaml# - + $ref: /schemas/remoteproc/ti,pru-rproc.yaml# type: object required: diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml index d4fc8fdcb72f..b3dbcba33e41 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.yaml @@ -115,8 +115,7 @@ patternProperties: 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 + $ref: /schemas/types.yaml#/definitions/flag "#clock-cells": description: Configure the SAI device as master clock provider. diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml index d77c8283526d..2ad17b361db0 100644 --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -176,13 +176,12 @@ patternProperties: 4 - Drive weak low and active high 5 - Drive Hi-Z and active high - allOf: - - $ref: /schemas/types.yaml#/definitions/uint32-array - - minItems: 2 - maxItems: 2 - items: - maximum: 15 - default: [2, 2] + $ref: /schemas/types.yaml#/definitions/uint32-array + minItems: 2 + maxItems: 2 + items: + maximum: 15 + default: [2, 2] required: - compatible diff --git a/Documentation/devicetree/bindings/spi/spi-controller.yaml b/Documentation/devicetree/bindings/spi/spi-controller.yaml index 36b72518f565..6d0511a8727f 100644 --- a/Documentation/devicetree/bindings/spi/spi-controller.yaml +++ b/Documentation/devicetree/bindings/spi/spi-controller.yaml @@ -93,9 +93,7 @@ properties: patternProperties: "^.*@[0-9a-f]+$": type: object - - allOf: - - $ref: spi-peripheral-props.yaml + $ref: spi-peripheral-props.yaml required: - compatible diff --git a/Documentation/devicetree/bindings/usb/st,stusb160x.yaml b/Documentation/devicetree/bindings/usb/st,stusb160x.yaml index ead1571e0e43..b5a8c9814dd3 100644 --- a/Documentation/devicetree/bindings/usb/st,stusb160x.yaml +++ b/Documentation/devicetree/bindings/usb/st,stusb160x.yaml @@ -32,9 +32,7 @@ properties: connector: type: object - - allOf: - - $ref: ../connector/usb-connector.yaml + $ref: /schemas/connector/usb-connector.yaml# properties: compatible: -- 2.32.0

10 9

[PATCH] ASoC: bt-sco: fix bt-sco-pcm-wb dai widget don't connect to the endpoint
by Jiaxin Yu 01 Mar '22

01 Mar '22

This patch fix the second dai driver's dai widget can't connect to the endpoint. Because "bt-sco-pcm" and "bt-sco-pcm-wb" dai driver have the same stream_name, so it will cause they have the same widget name. Therefor it will just create only one route when do snd_soc_dapm_add_route that only find the widget through the widget name. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- Hi maintainter, Need your comments. The patch is the one that I think it makes the mose sense. Maybe we can define the new stream_name for "bt-sco-pcm-wb" and add the new route. sound/soc/codecs/bt-sco.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/bt-sco.c b/sound/soc/codecs/bt-sco.c index 4d286844e3c8..578f9df2a25e 100644 --- a/sound/soc/codecs/bt-sco.c +++ b/sound/soc/codecs/bt-sco.c @@ -13,11 +13,16 @@ static const struct snd_soc_dapm_widget bt_sco_widgets[] = { SND_SOC_DAPM_INPUT("RX"), SND_SOC_DAPM_OUTPUT("TX"), + SND_SOC_DAPM_AIF_IN("BT_SCO_RX", "Playback", 0, + SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("BT_SCO_TX", "Capture", 0, + SND_SOC_NOPM, 0, 0), }; static const struct snd_soc_dapm_route bt_sco_routes[] = { - { "Capture", NULL, "RX" }, - { "TX", NULL, "Playback" }, + { "BT_SCO_TX", NULL, "RX" }, + { "TX", NULL, "BT_SCO_RX" }, +}; }; static struct snd_soc_dai_driver bt_sco_dai[] = { -- 2.25.1

2 2

[PATCH v2 00/17] ASoC: Intel: AVS - Audio DSP for cAVS
by Cezary Rojewski 01 Mar '22

01 Mar '22

A continuation of cleanup work of Intel SST solutions found in sound/soc/intel/. With two major chapters released last year catpt [1] and removal of haswell solution [2], time has come for Skylake-driver. Througout 2019, 2020 and 2021 Skylake-driver has had many fixes applied and even attempts of refactors as seen in fundamental overhaul [3], IPC flow adjustments [4] and LARGE_CONFIG overhaul [5] series. Unfortunately, story repeats itself - problems are found within the core of a driver. Painting it with different colors does not change the fact that is it still a house of cards. As changes needed to address those issues would make Skylake solution incompatible with its previous revisions, a decision has been made to provide a new solution instead. In time it would deprecate and replace Skylake-driver. That solution has been called AVS - from AudioDSP architecture name: Audio-Voice-Speech. It is meant to provide support for the exact same range of platforms as its predecessor: SKL, KBL, AML and APL. Note: this series is dependent upon HDA-series [6] which exposes several codec-organization functions allowing for reduced code size on avs-driver side. Note: this series does not add fully functional driver as its size would get out of control. Here, focus is put on adding IPC protocol and code loading code. Changes v2 -> v1: Almost all updates here are thanks to feedback from Pierre. - several comments and few kernel-docs have been added in areas which felt more or less unclear - avs_ipc_wait_busy_completion() now spins up to 'repeats_left' number of times before giving up - 'adsp_ba' field of struct avs_dev has been renamed to 'dsp_ba' - 'dops' field of struct avs_spec has been renamed to 'dsp_ops' - IPC abstraction has been simplified: SKL_ADSP_REG_HIP* regs are used directly - fixed allnoconfig with AVS enabled compilation issues - fixed code loading error paths: previously requested firmware is now released before function return to the caller - code and function arguments tied to D0IX support have been removed from this patchset and will be part of followup series adding that feature instead - enriched dev_err() messages in avs_dsp_get_core() and avs_dsp_put_core() - numerous wording fixes used in power/reset/stall DSP operations Changes RFC v1 [7]: -> v1: - separated HDA codec-organization patches, path and topology handling, PCM and complementary features such as recovery from this series to ease the review process - fixed EXPORT_SYMBOL_GPL for exported members of ASoC framework - result of stall() is now checked when sending ROM message - result of snd_hdac_ext_stream_set_spib() is now checked when loading basefw - if basefw is not ready, notification processing is now skipped - documented several topology parsing helpers Changes [internal] RFC v2 -> [public] RFC v1: - dropped any sysfs related changes from this series, moved to follow up one - dropped entire subscription-mechanism found in ipc.c. Handlers that are delegated to service certain firmware notifications are now called directly - fixed kernel doc for snd_soc_dapm_new_dai_widgets() as reported by ikp - prefixed snd_hda_codec_device_init() as suggested by Amadeo - improved comments for d0ix transitions for APL-based platforms as suggested by Pierre - a ton of spelling related fixes in most of the commit messages - fixed remaining warnings pointed by scan-build (variable assigned but not used) - replaced most of 'cAVS X.Y' expression usages with 'platform-based' equivalents as suggested by Pierre e.g.: cAVS 1.5 -> SKL-based Changes [internal] RFC v1 -> [internal] RFC v2: - fixed memleak caused by lack of kfree(vols) if memory allocation fails in avs_peakvol_create() as reported by Curtis - fixed missing 'i' iterator incrementation in avs_widget_ready() causing reference loss as reported by Curtis - replace hardcode: 0x40 usage with snd_hdac_calc_stream_format as suggested by Curtis. In consequence, readability for all code loading (CL) procedues has increased and such approach auto-documents the CL stream preparation - updated behavior of all index-fetching functions found in utils.c: avs_module_entry_index(), avs_module_id_entry_index() and follow ups: avs_get_module_entry(), avs_get_module_id_entry() to better conform to linux-kernel standard when no entry is found (return -ENOENT) rather than C++ standard (return -1, what in kernel case translated to -EPERM) as suggested by Curtis and Peter - several suggestions have been made regarding spacing, and so far, I've agreed and applied with all of them. None proposed seemed out of place or redundant - avs_path_stop() renamed to avs_path_pause() pipeline states are represented by RESET/PAUSED/RUNNING. avs_path_reset() and avs_path_run() were already there and avs_path_stop() just didn't look cohesive - added missing parsers for num_output_pin and num_input_pin which are required for custom modules such as WAVES or DSM - dropped 'priv_param_length' from custom module descriptor as this field is obsolete in firmware - parse_dictionary() has been split into parse_dictionary_header() and parse_dictionary_entries() to drop code duplication present in several parsing function which could not re-use entire parse_dictionary() - added avs_tplg_vendor_array_lookup_next() and avs_tplg_vendor_entry_next() to drop code duplicated present in several parsing functions. This change greatly impacted readability of all parsers - parsing helpers such avs_tplg_vendor_array_lookup() now return offset by updating specified in function argument list u32 *offset variable. This is to address problem when u32 offset would be greater than max int, which is the return type for these functions - AVS_DEFINE_PTR_PARSER() macro has been introduced to drop code duplication for all ptr-parsing users - all struct avs_path_module creators have had their declaration updated: function argument *owner ceased to exist as it could already be accessed by mod->owner - fixed the order of operation for conditional paths (e.g.: echo reference) so these are no longer controlled by "source" path and instead are impacted by state changes of source and sink paths both. Previously only source path e.g. playback sourcing echo reference would trigger RUNNING status for conditional path. Equivalent RUNNING on WoV path which is in this case sink path, would not do so, leading to order-of-operation problems. Behavior has been changed to: both source and sink need to be RUNNING for conditional path to be set to RUNNING too. PAUSED for either source or sink will cause PAUSED transition for conditional path. - to achieve the above, path states are now saved in 'state' i.e. new u32 field for struct avs_path - resigned from fw_filename field usage in favour of newly added tplg_filename for machine board descriptors as suggested by Pierre - platform descriptor fields have had their names update better reflect their purpose as suggested by Pierre - fixed comp_list missing locking when manipulated - all message senders now accept request as pointer as suggeseted by Peter - resigned of AZX_ usage for all ADSP-related registers, leaving them only for HOST memory space related operations - fixed disable path for core DSP operations: power/reset/stall as reported by Peter - safety when locking between received responses (reply vs notification) has been lowered as suggested by Pierre. Most usages are not performed in IRQ context and none is done in hard-IRQ one - s/master/main/ plus AVS_MAIN_CORE_MASK has replaced ->master_mask - several functions have had their logging updated - logs have been moved to lower level functions as suggested by Pierre - hdac_ext_stream usage has been streamlined to estream, hdac_streams are represented by hstream instead - hw_params() are resilient to scenarios when they are called mutliple times as reported by Pierre - avs_dsp_enable() now collapses if any of its steps fails as reported by Pierre and Peter - avs_module_ida_empty() now returns value of type bool as suggested by Bard [1]: https://www.spinics.net/lists/alsa-devel/msg116440.html [2]: https://www.spinics.net/lists/alsa-devel/msg116901.html [3]: https://www.spinics.net/lists/alsa-devel/msg94199.html [4]: https://www.spinics.net/lists/alsa-devel/msg92588.html [5]: https://lore.kernel.org/all/20190808181549.12521-1-cezary.rojewski@intel.co… [6]: https://lore.kernel.org/alsa-devel/20220207114906.3759800-1-cezary.rojewski… [7]: https://lore.kernel.org/all/20211208111301.1817725-1-cezary.rojewski@intel.… Cezary Rojewski (17): ALSA: hda: Add helper macros for DSP capable devices ASoC: Export DAI register and widget ctor and dctor functions ASoC: Intel: Introduce AVS driver ASoC: Intel: avs: Inter process communication ASoC: Intel: avs: Add code loading requests ASoC: Intel: avs: Add pipeline management requests ASoC: Intel: avs: Add module management requests ASoC: Intel: avs: Add power management requests ASoC: Intel: avs: Add ROM requests ASoC: Intel: avs: Add basefw runtime-parameter requests ASoC: Intel: avs: Firmware resources management utilities ASoC: Intel: avs: Declare module configuration types ASoC: Intel: avs: Dynamic firmware resources management ASoC: Intel: avs: General code loading flow ASoC: Intel: avs: Implement CLDMA transfer ASoC: Intel: avs: Code loading over CLDMA ASoC: Intel: avs: Code loading over HDA include/sound/hdaudio.h | 2 + include/sound/hdaudio_ext.h | 50 +++ include/sound/soc-dapm.h | 1 + sound/soc/intel/Kconfig | 12 + sound/soc/intel/Makefile | 1 + sound/soc/intel/avs/Makefile | 6 + sound/soc/intel/avs/avs.h | 248 +++++++++++ sound/soc/intel/avs/cldma.c | 317 ++++++++++++++ sound/soc/intel/avs/cldma.h | 29 ++ sound/soc/intel/avs/core.c | 62 +++ sound/soc/intel/avs/dsp.c | 303 +++++++++++++ sound/soc/intel/avs/ipc.c | 393 +++++++++++++++++ sound/soc/intel/avs/loader.c | 604 +++++++++++++++++++++++++ sound/soc/intel/avs/messages.c | 697 +++++++++++++++++++++++++++++ sound/soc/intel/avs/messages.h | 753 ++++++++++++++++++++++++++++++++ sound/soc/intel/avs/registers.h | 75 ++++ sound/soc/intel/avs/utils.c | 301 +++++++++++++ sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 15 + 19 files changed, 3870 insertions(+) create mode 100644 sound/soc/intel/avs/Makefile create mode 100644 sound/soc/intel/avs/avs.h create mode 100644 sound/soc/intel/avs/cldma.c create mode 100644 sound/soc/intel/avs/cldma.h create mode 100644 sound/soc/intel/avs/core.c create mode 100644 sound/soc/intel/avs/dsp.c create mode 100644 sound/soc/intel/avs/ipc.c create mode 100644 sound/soc/intel/avs/loader.c create mode 100644 sound/soc/intel/avs/messages.c create mode 100644 sound/soc/intel/avs/messages.h create mode 100644 sound/soc/intel/avs/registers.h create mode 100644 sound/soc/intel/avs/utils.c -- 2.25.1

2 20

[PATCH v5 0/2] Add support for SoundWire1.6 audio cgcr register control
by Srinivasa Rao Mandadapu 01 Mar '22

01 Mar '22

This patch series is to add v1.6.0 compatible name for qcom soundwire driver and corresponding dt bindings. Changes Since V4: -- Dropped audio cgcr control patch due to dependency on clock patches. -- Update dt-bindings as per new reset control properties. Changes Since V3: -- Add v1.6.0 compatible name and soundwire data structure. -- Change macro define name properly. -- Update bindings for new property. -- Change commit message description. -- Change signedoff by sequence. Changes since v2: -- Update error check after ioremap. Changes since v1: -- Add const name to mask value. Srinivasa Rao Mandadapu (2): soundwire: qcom: Add compatible name for v1.6.0 dt-bindings: soundwire: qcom: Add bindings for audio CSR reset control property Documentation/devicetree/bindings/soundwire/qcom,sdw.txt | 12 ++++++++++++ drivers/soundwire/qcom.c | 6 ++++++ 2 files changed, 18 insertions(+) -- 2.7.4

2 6

[PATCH] Asoc: au1x: Use devm_snd_soc_register_component()
by Zhen Ni 01 Mar '22

01 Mar '22

Using devm_snd_soc_register_component() instead of snd_soc_register_component() Signed-off-by: Zhen Ni <nizhen(a)uniontech.com> --- sound/soc/au1x/ac97c.c | 2 +- sound/soc/au1x/i2sc.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index 3b1700e665f5..795bba756585 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -275,7 +275,7 @@ static int au1xac97c_drvprobe(struct platform_device *pdev) if (ret) return ret; - ret = snd_soc_register_component(&pdev->dev, &au1xac97c_component, + ret = devm_snd_soc_register_component(&pdev->dev, &au1xac97c_component, &au1xac97c_dai_driver, 1); if (ret) return ret; diff --git a/sound/soc/au1x/i2sc.c b/sound/soc/au1x/i2sc.c index 740d4e052e4d..a3c9d52ef337 100644 --- a/sound/soc/au1x/i2sc.c +++ b/sound/soc/au1x/i2sc.c @@ -265,7 +265,7 @@ static int au1xi2s_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, ctx); - return snd_soc_register_component(&pdev->dev, &au1xi2s_component, + return devm_snd_soc_register_component(&pdev->dev, &au1xi2s_component, &au1xi2s_dai_driver, 1); } diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 05eb36991f14..3f60313822e7 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -411,7 +411,7 @@ static int au1xpsc_ac97_drvprobe(struct platform_device *pdev) if (ret) return ret; - ret = snd_soc_register_component(&pdev->dev, &au1xpsc_ac97_component, + ret = devm_snd_soc_register_component(&pdev->dev, &au1xpsc_ac97_component, &wd->dai_drv, 1); if (ret) return ret; -- 2.20.1

2 1

[PATCH] ASoC: fsi: Add check for clk_enable
by Jiasheng Jiang 01 Mar '22

01 Mar '22

As the potential failure of the clk_enable(), it should be better to check it and return error if fails. Fixes: ab6f6d85210c ("ASoC: fsi: add master clock control functions") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/soc/sh/fsi.c | 18 +++++++++++++++--- 1 file changed, 15 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index cdf3b7f69ba7..93cfa77b8b54 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -816,14 +816,26 @@ static int fsi_clk_enable(struct device *dev, return ret; } - clk_enable(clock->xck); - clk_enable(clock->ick); - clk_enable(clock->div); + ret = clk_enable(clock->xck); + if (ret) + goto err; + ret = clk_enable(clock->ick); + if (ret) + goto err; + ret = clk_enable(clock->div); + if (ret) + goto err; clock->count++; } return ret; + +err: + clk_disable(clock->xck); + clk_disable(clock->ick); + clk_disable(clock->div); + return ret; } static int fsi_clk_disable(struct device *dev, -- 2.25.1

2 1

[PATCH] ASoC: soc-generic-dmaengine-pcm: Use default config when none is given
by Sascha Hauer 01 Mar '22

01 Mar '22

snd_dmaengine_pcm_register() can be passed a NULL pointer for the config which means that the we have to test for pcm->config being non NULL before accessing it. Make the code more straight forward by providing a default config when none is passed. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/soc-generic-dmaengine-pcm.c | 55 ++++++++++++--------------- 1 file changed, 24 insertions(+), 31 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 0f6c9c55a22d6..734d46b9783f7 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -79,29 +79,19 @@ static int dmaengine_pcm_hw_params(struct snd_soc_component *component, { struct dmaengine_pcm *pcm = soc_component_to_pcm(component); struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); - int (*prepare_slave_config)(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct dma_slave_config *slave_config); struct dma_slave_config slave_config; + int ret; - memset(&slave_config, 0, sizeof(slave_config)); - - if (!pcm->config) - prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config; - else - prepare_slave_config = pcm->config->prepare_slave_config; + if (!pcm->config->prepare_slave_config) + return 0; - if (prepare_slave_config) { - int ret = prepare_slave_config(substream, params, &slave_config); - if (ret) - return ret; + memset(&slave_config, 0, sizeof(slave_config)); - ret = dmaengine_slave_config(chan, &slave_config); - if (ret) - return ret; - } + ret = pcm->config->prepare_slave_config(substream, params, &slave_config); + if (ret) + return ret; - return 0; + return dmaengine_slave_config(chan, &slave_config); } static int @@ -121,7 +111,7 @@ dmaengine_pcm_set_runtime_hwparams(struct snd_soc_component *component, return -EINVAL; } - if (pcm->config && pcm->config->pcm_hardware) + if (pcm->config->pcm_hardware) return snd_soc_set_runtime_hwparams(substream, pcm->config->pcm_hardware); @@ -188,7 +178,6 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( { struct dmaengine_pcm *pcm = soc_component_to_pcm(component); struct snd_dmaengine_dai_dma_data *dma_data; - dma_filter_fn fn = NULL; if (rtd->num_cpus > 1) { dev_err(rtd->dev, @@ -201,13 +190,11 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) return pcm->chan[0]; - if (pcm->config && pcm->config->compat_request_channel) + if (pcm->config->compat_request_channel) return pcm->config->compat_request_channel(rtd, substream); - if (pcm->config) - fn = pcm->config->compat_filter_fn; - - return snd_dmaengine_pcm_request_channel(fn, dma_data->filter_data); + return snd_dmaengine_pcm_request_channel(pcm->config->compat_filter_fn, + dma_data->filter_data); } static bool dmaengine_pcm_can_report_residue(struct device *dev, @@ -239,7 +226,7 @@ static int dmaengine_pcm_new(struct snd_soc_component *component, size_t max_buffer_size; unsigned int i; - if (config && config->prealloc_buffer_size) { + if (config->prealloc_buffer_size) { prealloc_buffer_size = config->prealloc_buffer_size; max_buffer_size = config->pcm_hardware->buffer_bytes_max; } else { @@ -252,7 +239,7 @@ static int dmaengine_pcm_new(struct snd_soc_component *component, if (!substream) continue; - if (!pcm->chan[i] && config && config->chan_names[i]) + if (!pcm->chan[i] && config->chan_names[i]) pcm->chan[i] = dma_request_slave_channel(dev, config->chan_names[i]); @@ -365,10 +352,10 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, struct dma_chan *chan; if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || (!dev->of_node && - !(config && config->dma_dev && config->dma_dev->of_node))) + !(config->dma_dev && config->dma_dev->of_node))) return 0; - if (config && config->dma_dev) { + if (config->dma_dev) { /* * If this warning is seen, it probably means that your Linux * device structure does not match your HW device structure. @@ -385,7 +372,7 @@ static int dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, name = "rx-tx"; else name = dmaengine_pcm_dma_channel_names[i]; - if (config && config->chan_names[i]) + if (config->chan_names[i]) name = config->chan_names[i]; chan = dma_request_chan(dev, name); if (IS_ERR(chan)) { @@ -423,6 +410,10 @@ static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm) } } +static const struct snd_dmaengine_pcm_config snd_dmaengine_pcm_default_config = { + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, +}; + /** * snd_dmaengine_pcm_register - Register a dmaengine based PCM device * @dev: The parent device for the PCM device @@ -443,6 +434,8 @@ int snd_dmaengine_pcm_register(struct device *dev, #ifdef CONFIG_DEBUG_FS pcm->component.debugfs_prefix = "dma"; #endif + if (!config) + config = &snd_dmaengine_pcm_default_config; pcm->config = config; pcm->flags = flags; @@ -450,7 +443,7 @@ int snd_dmaengine_pcm_register(struct device *dev, if (ret) goto err_free_dma; - if (config && config->process) + if (config->process) driver = &dmaengine_pcm_component_process; else driver = &dmaengine_pcm_component; -- 2.30.2

1 0

[PATCH] ALSA: n64: Use platform_get_irq() to get the interrupt
by Zhen Ni 01 Mar '22

01 Mar '22

platform_get_resource(pdev, IORESOURCE_IRQ, ..) relies on static allocation of IRQ resources in DT core code, this causes an issue when using hierarchical interrupt domains using "interrupts" property in the node as this bypassed the hierarchical setup and messed up the irq chaining. In preparation for removal of static setup of IRQ resource from DT core code use platform_get_irq(). Signed-off-by: Zhen Ni <nizhen(a)uniontech.com> --- sound/mips/snd-n64.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/mips/snd-n64.c b/sound/mips/snd-n64.c index 463a6fe589eb..19365aab15ba 100644 --- a/sound/mips/snd-n64.c +++ b/sound/mips/snd-n64.c @@ -289,7 +289,7 @@ static int __init n64audio_probe(struct platform_device *pdev) struct snd_card *card; struct snd_pcm *pcm; struct n64audio *priv; - struct resource *res; + int irq; int err; err = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, @@ -337,12 +337,12 @@ static int __init n64audio_probe(struct platform_device *pdev) strcpy(card->shortname, "N64 Audio"); strcpy(card->longname, "N64 Audio"); - res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!res) { + irq = platform_get_irq(pdev, 0); + if (irq < 0) { err = -EINVAL; goto fail_dma_alloc; } - if (devm_request_irq(&pdev->dev, res->start, n64audio_isr, + if (devm_request_irq(&pdev->dev, irq, n64audio_isr, IRQF_SHARED, "N64 Audio", priv)) { err = -EBUSY; goto fail_dma_alloc; -- 2.20.1

2 1

Issue in pcm_dsnoop.c in alsa-lib
by GitHub issues - opened 01 Mar '22

01 Mar '22

alsa-project/alsa-lib issue #213 was opened from TE-N-ShengjiuWang: Hi Takashi Iwai, Jaroslav Kysela We encountered an issue in the pcm_dsnoop use case, could you please help to have a look? Issue description: With two instances for dsnoop type device running in parallel, after suspend/resume, one of the instances will be hung in memcpy because the very large copy size is obtained. #3 0x0000ffffa78d5098 in snd_pcm_dsnoop_sync_ptr (pcm=0xaaab06563da0) at pcm_dsnoop.c:158 dsnoop = 0xaaab06563c20 slave_hw_ptr = 64 old_slave_hw_ptr = 533120 avail = 187651522444320 Reason analysis: The root cause that I analysis is that after suspend/resume, one instance will get the SND_PCM_STATE_SUSPENDED state from slave pcm device, then it will do snd_pcm_prepare() and snd_pcm_start(), which will reset the dsnoop->slave_hw_ptr and the hw_ptr of slave pcm device, then the state of this instance is correct. But another instance may not get the SND_PCM_STATE_SUSPENDED state from slave pcm device because slave device may have been recovered by first instance, so the dsnoop->slave_hw_ptr is not reset. but because hw_ptr of slave pcm device has been reset, so there will be a very large "avail" size. Solution: I didn't come up with a fix for this issue, seems there is no easy way to let another instance know this case and reset the dsnoop->slave_hw_ptr, could you please help? Issue URL : https://github.com/alsa-project/alsa-lib/issues/213 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

chancel.liu@nxp.com, shengjiu.wang@nxp.com
by Shengjiu Wang 01 Mar '22

01 Mar '22

Hi Takashi Iwai, Jaroslav Kysela We encountered an issue in the pcm_dsnoop use case, could you please help to have a look? *Issue description:* With two instances for dsnoop type device running in parallel, after suspend/resume, one of the instances will be hung in memcpy because the very large copy size is obtained. #3 0x0000ffffa78d5098 in snd_pcm_dsnoop_sync_ptr (pcm=0xaaab06563da0) at pcm_dsnoop.c:158 dsnoop = 0xaaab06563c20 slave_hw_ptr = 64 old_slave_hw_ptr = 533120 avail = *187651522444320* * Reason analysis: * The root cause that I analysis is that after suspend/resume, one instance will get the SND_PCM_STATE_SUSPENDED state from slave pcm device, then it will do snd_pcm_prepare() and snd_pcm_start(), which will reset the dsnoop->slave_hw_ptr and the hw_ptr of slave pcm device, then the state of this instance is correct. But another instance may not get the SND_PCM_STATE_SUSPENDED state from slave pcm device because slave device may have been recovered by first instance, so the dsnoop->slave_hw_ptr is not reset. but because hw_ptr of slave pcm device has been reset, so there will be a very large "avail" size. *Solution:* I didn't come up with a fix for this issue, seems there is no easy way to let another instance know this case and reset the dsnoop->slave_hw_ptr, could you please help? Best regards Wang shengjiu

1 0

RE: [PATCH 2/6] treewide: remove using list iterator after loop body as a ptr
by David Laight 01 Mar '22

01 Mar '22

From: Linus Torvalds > Sent: 28 February 2022 19:56 > On Mon, Feb 28, 2022 at 4:19 AM Christian König > <christian.koenig(a)amd.com> wrote: > > > > I don't think that using the extra variable makes the code in any way > > more reliable or easier to read. > > So I think the next step is to do the attached patch (which requires > that "-std=gnu11" that was discussed in the original thread). > > That will guarantee that the 'pos' parameter of list_for_each_entry() > is only updated INSIDE the for_each_list_entry() loop, and can never > point to the (wrongly typed) head entry. > > And I would actually hope that it should actually cause compiler > warnings about possibly uninitialized variables if people then use the > 'pos' pointer outside the loop. Except > > (a) that code in sgx/encl.c currently initializes 'tmp' to NULL for > inexplicable reasons - possibly because it already expected this > behavior > > (b) when I remove that NULL initializer, I still don't get a warning, > because we've disabled -Wno-maybe-uninitialized since it results in so > many false positives. > > Oh well. > > Anyway, give this patch a look, and at least if it's expanded to do > "(pos) = NULL" in the entry statement for the for-loop, it will avoid > the HEAD type confusion that Jakob is working on. And I think in a > cleaner way than the horrid games he plays. > > (But it won't avoid possible CPU speculation of such type confusion. > That, in my opinion, is a completely different issue) > > I do wish we could actually poison the 'pos' value after the loop > somehow - but clearly the "might be uninitialized" I was hoping for > isn't the way to do it. > > Anybody have any ideas? > > Linus diff --git a/include/linux/list.h b/include/linux/list.h index dd6c2041d09c..bab995596aaa 100644 --- a/include/linux/list.h +++ b/include/linux/list.h @@ -634,10 +634,10 @@ static inline void list_splice_tail_init(struct list_head *list, * @head: the head for your list. * @member: the name of the list_head within the struct. */ -#define list_for_each_entry(pos, head, member) \ - for (pos = list_first_entry(head, typeof(*pos), member); \ - !list_entry_is_head(pos, head, member); \ - pos = list_next_entry(pos, member)) +#define list_for_each_entry(pos, head, member) \ + for (typeof(pos) __iter = list_first_entry(head, typeof(*pos), member); \ + !list_entry_is_head(__iter, head, member) && (((pos)=__iter),1); \ + __iter = list_next_entry(__iter, member)) /** * list_for_each_entry_reverse - iterate backwards over list of given type. I think you actually want: !list_entry_is_head(__iter, head, member) ? (((pos)=__iter),1) : (((pos) = NULL),0); Which can be done in the original by: !list_entry_is_head(pos, head, member) ? 1 : (((pos) = NULL), 0); Although it would be safer to have (without looking up the actual name): for (item *__item = head; \ __item ? (((pos) = list_item(__item, member)), 1) : (((pos) = NULL), 0); __item = (pos)->member) The local does need 'fixing' to avoid shadowing for nested loops. David - Registered Address Lakeside, Bramley Road, Mount Farm, Milton Keynes, MK1 1PT, UK Registration No: 1397386 (Wales)

1 0

[PATCH v3] ASoC: codecs: wsa881x: add runtime pm support
by Srinivas Kandagatla 01 Mar '22

01 Mar '22

WSA SoundWire Controller does not support Clock stop and performs a soft reset on suspend resume path. Its recommended that WSA881x codecs connected to this are also reset using a hard reset during suspend resume. So this codec driver performs a hard reset during suspend resume cycle. --- Hi Mark, Looks like the runtime pm series was partly applied, so am sending just this leftover patch with requested changes. Changes since v2: removed setting up simple_clk_stop_capable flag update log srini sound/soc/codecs/wsa881x.c | 53 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 53 insertions(+) diff --git a/sound/soc/codecs/wsa881x.c b/sound/soc/codecs/wsa881x.c index 0222370ff95d..616b26c70c3b 100644 --- a/sound/soc/codecs/wsa881x.c +++ b/sound/soc/codecs/wsa881x.c @@ -11,6 +11,7 @@ #include <linux/of_gpio.h> #include <linux/regmap.h> #include <linux/slab.h> +#include <linux/pm_runtime.h> #include <linux/soundwire/sdw.h> #include <linux/soundwire/sdw_registers.h> #include <linux/soundwire/sdw_type.h> @@ -198,6 +199,7 @@ #define WSA881X_OCP_CTL_TIMER_SEC 2 #define WSA881X_OCP_CTL_TEMP_CELSIUS 25 #define WSA881X_OCP_CTL_POLL_TIMER_SEC 60 +#define WSA881X_PROBE_TIMEOUT 1000 #define WSA881X_PA_GAIN_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -747,6 +749,12 @@ static int wsa881x_put_pa_gain(struct snd_kcontrol *kc, unsigned int mask = (1 << fls(max)) - 1; int val, ret, min_gain, max_gain; + ret = pm_runtime_get_sync(comp->dev); + if (ret < 0 && ret != -EACCES) { + pm_runtime_put_noidle(comp->dev); + return ret; + } + max_gain = (max - ucontrol->value.integer.value[0]) & mask; /* * Gain has to set incrementally in 4 steps @@ -773,6 +781,9 @@ static int wsa881x_put_pa_gain(struct snd_kcontrol *kc, usleep_range(1000, 1010); } + pm_runtime_mark_last_busy(comp->dev); + pm_runtime_put_autosuspend(comp->dev); + return 1; } @@ -1101,6 +1112,7 @@ static int wsa881x_probe(struct sdw_slave *pdev, const struct sdw_device_id *id) { struct wsa881x_priv *wsa881x; + struct device *dev = &pdev->dev; wsa881x = devm_kzalloc(&pdev->dev, sizeof(*wsa881x), GFP_KERNEL); if (!wsa881x) @@ -1132,12 +1144,52 @@ static int wsa881x_probe(struct sdw_slave *pdev, return PTR_ERR(wsa881x->regmap); } + pm_runtime_set_autosuspend_delay(dev, 3000); + pm_runtime_use_autosuspend(dev); + pm_runtime_mark_last_busy(dev); + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + return devm_snd_soc_register_component(&pdev->dev, &wsa881x_component_drv, wsa881x_dais, ARRAY_SIZE(wsa881x_dais)); } +static int __maybe_unused wsa881x_runtime_suspend(struct device *dev) +{ + struct regmap *regmap = dev_get_regmap(dev, NULL); + struct wsa881x_priv *wsa881x = dev_get_drvdata(dev); + + gpiod_direction_output(wsa881x->sd_n, 0); + + regcache_cache_only(regmap, true); + regcache_mark_dirty(regmap); + + return 0; +} + +static int __maybe_unused wsa881x_runtime_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct regmap *regmap = dev_get_regmap(dev, NULL); + struct wsa881x_priv *wsa881x = dev_get_drvdata(dev); + + gpiod_direction_output(wsa881x->sd_n, 1); + + wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(WSA881X_PROBE_TIMEOUT)); + + regcache_cache_only(regmap, false); + regcache_sync(regmap); + + return 0; +} + +static const struct dev_pm_ops wsa881x_pm_ops = { + SET_RUNTIME_PM_OPS(wsa881x_runtime_suspend, wsa881x_runtime_resume, NULL) +}; + static const struct sdw_device_id wsa881x_slave_id[] = { SDW_SLAVE_ENTRY(0x0217, 0x2010, 0), SDW_SLAVE_ENTRY(0x0217, 0x2110, 0), @@ -1151,6 +1203,7 @@ static struct sdw_driver wsa881x_codec_driver = { .id_table = wsa881x_slave_id, .driver = { .name = "wsa881x-codec", + .pm = &wsa881x_pm_ops, } }; module_sdw_driver(wsa881x_codec_driver); -- 2.21.0

2 3

[PATCH v16 0/9] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 28 Feb '22

28 Feb '22

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. Changes Since V15: -- Bisect patches to avoid build failure in other architectures. -- Remove redundant variables lpass variant structure. Changes Since V14: -- Split common wrapper function to separate wrapper for each handle in platform driver. -- Update cdc dma buffer handling with memremap with ioremap. -- Remove redundant error prints. -- Update irq flag. Changes Since V13: -- Change bulk cdc clock voting to individual clock voting. -- Remove redundant code, conditional check and prints. -- Fix typo errors. Changes Since V12: -- Fix arguments type mismatch. Changes Since V11: -- Fix kernel robot issue on arguments type mismatch. Changes Since V10: -- Split bulk clock voting to individual clock voting as per use case in cdc-dma driver. -- Add missing codec dma clocks. -- Update rxtx lpm buffer size. Changes Since V9: -- Change individual clock voting to bulk clock voting of lpass-sc7280 platform driver. -- Remove redundant clocks in lpass variant structure. -- Add mclk for MI2S based headset path. -- Remove unused lpass variant structure members in lpass header. Changes Since V8: -- Fix errors in sc7280 lpass cpu dt-bindings. -- Move to quicinc domain email id's. Changes Since V7: -- Fix indentation errors. -- Bisect patches to avoid interdependency. Changes Since V6: -- Split cdc dma regmap config macros. -- Add write dma reg fields for i2s path. -- Add helper function to distinguish rxtx and va dma ports. -- Optimizing clock and reg name in cpu dt-bindings. -- Update buffer management for cdc dma path. -- Remove Kconfig fields of machine driver. Changes Since V5: -- Include MI2S primary node to snd_soc_dai_driver in lpass-sc7280 platform driver. -- Move dependency patch list to corresponding patch. -- Add support for missing cdc-dma ports. -- Change if/else conditional statements to switch cases. -- Add missing error handlings. -- Typo errors fix. Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (9): ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add regmap config support for codec dma driver ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: dt-bindings: Add SC7280 lpass cpu bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 75 ++- sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 301 ++++++++++ sound/soc/qcom/lpass-cpu.c | 253 ++++++++- sound/soc/qcom/lpass-lpaif-reg.h | 127 ++++- sound/soc/qcom/lpass-platform.c | 628 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 438 ++++++++++++++ sound/soc/qcom/lpass.h | 141 +++++ 9 files changed, 1890 insertions(+), 88 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- 2.7.4

3 14

[PATCH v5 0/2] Add power domains support for digital macro codecs
by Srinivasa Rao Mandadapu 28 Feb '22

28 Feb '22

This patch set is to add power domains support for RX, TX and VA macros. Changes since v4: -- Fix uninitialized variable return error Changes since v3: -- Add dt-bindings support. Changes since v2: -- Remove redundant local variable. -- Update pds error handling sequence. -- Update module description. -- Clean up pds init function. -- Remove redundant arguments. Changes since v1: -- Add missing macros in Kconfig. Srinivasa Rao Mandadapu (2): ASoC: codecs: Add power domains support in digital macro codecs ASoC: qcom: dt-bindings: Add bindings for power domains in lpass digital codecs .../bindings/sound/qcom,lpass-rx-macro.yaml | 8 +++ .../bindings/sound/qcom,lpass-tx-macro.yaml | 8 +++ .../bindings/sound/qcom,lpass-va-macro.yaml | 8 +++ sound/soc/codecs/Kconfig | 7 +++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/lpass-macro-common.c | 67 ++++++++++++++++++++++ sound/soc/codecs/lpass-macro-common.h | 17 ++++++ sound/soc/codecs/lpass-rx-macro.c | 10 +++- sound/soc/codecs/lpass-tx-macro.c | 9 +++ sound/soc/codecs/lpass-va-macro.c | 10 ++++ sound/soc/qcom/Kconfig | 1 + 11 files changed, 146 insertions(+), 1 deletion(-) create mode 100644 sound/soc/codecs/lpass-macro-common.c create mode 100644 sound/soc/codecs/lpass-macro-common.h -- 2.7.4

2 3

[PATCH V2 1/4] ASoC: amd: vg: fix for pm resume callback sequence
by Vijendar Mukunda 28 Feb '22

28 Feb '22

The previous condition is used to cross check only the active stream status for I2S HS instance playback and capture use cases. Modified logic to invoke sequence for two i2s controller instances. This also fixes warnings reported by kernel robot: "warning: variable 'frmt_val' set but not used" "warning: variable 'reg_val' set but not used" Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- v1 -> v2: fixed kernel rebot warnings sound/soc/amd/vangogh/acp5x-pcm-dma.c | 66 +++++++++++++-------------- 1 file changed, 33 insertions(+), 33 deletions(-) diff --git a/sound/soc/amd/vangogh/acp5x-pcm-dma.c b/sound/soc/amd/vangogh/acp5x-pcm-dma.c index f10de38976cb..3aa2d9a36880 100644 --- a/sound/soc/amd/vangogh/acp5x-pcm-dma.c +++ b/sound/soc/amd/vangogh/acp5x-pcm-dma.c @@ -426,51 +426,51 @@ static int acp5x_audio_remove(struct platform_device *pdev) static int __maybe_unused acp5x_pcm_resume(struct device *dev) { struct i2s_dev_data *adata; - u32 val, reg_val, frmt_val; + struct i2s_stream_instance *rtd; + u32 val; - reg_val = 0; - frmt_val = 0; adata = dev_get_drvdata(dev); if (adata->play_stream && adata->play_stream->runtime) { - struct i2s_stream_instance *rtd = - adata->play_stream->runtime->private_data; + rtd = adata->play_stream->runtime->private_data; config_acp5x_dma(rtd, SNDRV_PCM_STREAM_PLAYBACK); - switch (rtd->i2s_instance) { - case I2S_HS_INSTANCE: - reg_val = ACP_HSTDM_ITER; - frmt_val = ACP_HSTDM_TXFRMT; - break; - case I2S_SP_INSTANCE: - default: - reg_val = ACP_I2STDM_ITER; - frmt_val = ACP_I2STDM_TXFRMT; + acp_writel((rtd->xfer_resolution << 3), rtd->acp5x_base + ACP_HSTDM_ITER); + if (adata->tdm_mode == TDM_ENABLE) { + acp_writel(adata->tdm_fmt, adata->acp5x_base + ACP_HSTDM_TXFRMT); + val = acp_readl(adata->acp5x_base + ACP_HSTDM_ITER); + acp_writel(val | 0x2, adata->acp5x_base + ACP_HSTDM_ITER); + } + } + if (adata->i2ssp_play_stream && adata->i2ssp_play_stream->runtime) { + rtd = adata->i2ssp_play_stream->runtime->private_data; + config_acp5x_dma(rtd, SNDRV_PCM_STREAM_PLAYBACK); + acp_writel((rtd->xfer_resolution << 3), rtd->acp5x_base + ACP_I2STDM_ITER); + if (adata->tdm_mode == TDM_ENABLE) { + acp_writel(adata->tdm_fmt, adata->acp5x_base + ACP_I2STDM_TXFRMT); + val = acp_readl(adata->acp5x_base + ACP_I2STDM_ITER); + acp_writel(val | 0x2, adata->acp5x_base + ACP_I2STDM_ITER); } - acp_writel((rtd->xfer_resolution << 3), - rtd->acp5x_base + reg_val); } if (adata->capture_stream && adata->capture_stream->runtime) { - struct i2s_stream_instance *rtd = - adata->capture_stream->runtime->private_data; + rtd = adata->capture_stream->runtime->private_data; config_acp5x_dma(rtd, SNDRV_PCM_STREAM_CAPTURE); - switch (rtd->i2s_instance) { - case I2S_HS_INSTANCE: - reg_val = ACP_HSTDM_IRER; - frmt_val = ACP_HSTDM_RXFRMT; - break; - case I2S_SP_INSTANCE: - default: - reg_val = ACP_I2STDM_IRER; - frmt_val = ACP_I2STDM_RXFRMT; + acp_writel((rtd->xfer_resolution << 3), rtd->acp5x_base + ACP_HSTDM_IRER); + if (adata->tdm_mode == TDM_ENABLE) { + acp_writel(adata->tdm_fmt, adata->acp5x_base + ACP_HSTDM_RXFRMT); + val = acp_readl(adata->acp5x_base + ACP_HSTDM_IRER); + acp_writel(val | 0x2, adata->acp5x_base + ACP_HSTDM_IRER); } - acp_writel((rtd->xfer_resolution << 3), - rtd->acp5x_base + reg_val); } - if (adata->tdm_mode == TDM_ENABLE) { - acp_writel(adata->tdm_fmt, adata->acp5x_base + frmt_val); - val = acp_readl(adata->acp5x_base + reg_val); - acp_writel(val | 0x2, adata->acp5x_base + reg_val); + if (adata->i2ssp_capture_stream && adata->i2ssp_capture_stream->runtime) { + rtd = adata->i2ssp_capture_stream->runtime->private_data; + config_acp5x_dma(rtd, SNDRV_PCM_STREAM_CAPTURE); + acp_writel((rtd->xfer_resolution << 3), rtd->acp5x_base + ACP_I2STDM_IRER); + if (adata->tdm_mode == TDM_ENABLE) { + acp_writel(adata->tdm_fmt, adata->acp5x_base + ACP_I2STDM_RXFRMT); + val = acp_readl(adata->acp5x_base + ACP_I2STDM_IRER); + acp_writel(val | 0x2, adata->acp5x_base + ACP_I2STDM_IRER); + } } acp_writel(1, adata->acp5x_base + ACP_EXTERNAL_INTR_ENB); return 0; -- 2.17.1

2 4

[PATCH] ASoC: dt-bindings: renesas,rz-ssi: Document RZ/V2L SoC
by Lad Prabhakar 28 Feb '22

28 Feb '22

Document RZ/V2L SSI bindings. RZ/V2L SSI is identical to one found on the RZ/G2L SoC. No driver changes are required as generic compatible string "renesas,rz-ssi" will be used as a fallback. Signed-off-by: Lad Prabhakar <prabhakar.mahadev-lad.rj(a)bp.renesas.com> Reviewed-by: Biju Das <biju.das.jz(a)bp.renesas.com> --- DTSI changes have been posted as part of series [0]. [0] https://patchwork.kernel.org/project/linux-renesas-soc/ patch/20220227203744.18355-10-prabhakar.mahadev-lad.rj(a)bp.renesas.com/ --- Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml index 414ff8035a4e..7e8d252f7bca 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml +++ b/Documentation/devicetree/bindings/sound/renesas,rz-ssi.yaml @@ -4,7 +4,7 @@ $id: http://devicetree.org/schemas/sound/renesas,rz-ssi.yaml# $schema: http://devicetree.org/meta-schemas/core.yaml# -title: Renesas RZ/G2L ASoC Sound Serial Interface (SSIF-2) +title: Renesas RZ/{G2L,V2L} ASoC Sound Serial Interface (SSIF-2) maintainers: - Biju Das <biju.das.jz(a)bp.renesas.com> @@ -14,6 +14,7 @@ properties: items: - enum: - renesas,r9a07g044-ssi # RZ/G2{L,LC} + - renesas,r9a07g054-ssi # RZ/V2L - const: renesas,rz-ssi reg: -- 2.17.1

4 3

[PATCH] ASoC: twl4030: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the twl4030 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/twl4030.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e059711ff293..0ba3546ef870 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1840,13 +1840,12 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) old_format = twl4030_read(component, TWL4030_REG_AUDIO_IF); format = old_format; - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: format &= ~(TWL4030_AIF_SLAVE_EN); format &= ~(TWL4030_CLK256FS_EN); break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: format |= TWL4030_AIF_SLAVE_EN; format |= TWL4030_CLK256FS_EN; break; @@ -2038,9 +2037,8 @@ static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, old_format = twl4030_read(component, TWL4030_REG_VOICE_IF); format = old_format; - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: format &= ~(TWL4030_VIF_SLAVE_EN); break; case SND_SOC_DAIFMT_CBS_CFS: -- 2.30.2

2 2

[PATCH] ASoC: ti: davinci-i2s: Add check for clk_enable()
by Jiasheng Jiang 28 Feb '22

28 Feb '22

As the potential failure of the clk_enable(), it should be better to check it and return error if fails. Fixes: 5f9a50c3e55e ("ASoC: Davinci: McBSP: add device tree support for McBSP") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/soc/ti/davinci-i2s.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c index 6dca51862dd7..0363a088d2e0 100644 --- a/sound/soc/ti/davinci-i2s.c +++ b/sound/soc/ti/davinci-i2s.c @@ -708,7 +708,9 @@ static int davinci_i2s_probe(struct platform_device *pdev) dev->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(dev->clk)) return -ENODEV; - clk_enable(dev->clk); + ret = clk_enable(dev->clk); + if (ret) + goto err_put_clk; dev->dev = &pdev->dev; dev_set_drvdata(&pdev->dev, dev); @@ -730,6 +732,7 @@ static int davinci_i2s_probe(struct platform_device *pdev) snd_soc_unregister_component(&pdev->dev); err_release_clk: clk_disable(dev->clk); +err_put_clk: clk_put(dev->clk); return ret; } -- 2.25.1

3 2

[PATCH] ASoC: amd: pcm-dma: Use platform_get_irq() to get the interrupt
by Meng Tang 28 Feb '22

28 Feb '22

platform_get_resource(pdev, IORESOURCE_IRQ, ..) relies on static allocation of IRQ resources in DT core code, this causes an issue when using hierarchical interrupt domains using "interrupts" property in the node as this bypassed the hierarchical setup and messed up the irq chaining. In preparation for removal of static setup of IRQ resource from DT core code use platform_get_irq(). Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/soc/amd/acp-pcm-dma.c | 11 ++++------- sound/soc/amd/raven/acp3x-pcm-dma.c | 8 ++------ sound/soc/amd/renoir/acp3x-pdm-dma.c | 7 ++----- sound/soc/amd/vangogh/acp5x-pcm-dma.c | 7 ++----- 4 files changed, 10 insertions(+), 23 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 8fa2e2fde4f1..1cd2e70a57df 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1217,9 +1217,8 @@ static const struct snd_soc_component_driver acp_asoc_platform = { static int acp_audio_probe(struct platform_device *pdev) { - int status; + int status, irq; struct audio_drv_data *audio_drv_data; - struct resource *res; const u32 *pdata = pdev->dev.platform_data; if (!pdata) { @@ -1249,13 +1248,11 @@ static int acp_audio_probe(struct platform_device *pdev) audio_drv_data->asic_type = *pdata; - res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!res) { - dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + irq = platform_get_irq(pdev, 0); + if (irq < 0) return -ENODEV; - } - status = devm_request_irq(&pdev->dev, res->start, dma_irq_handler, + status = devm_request_irq(&pdev->dev, irq, dma_irq_handler, 0, "ACP_IRQ", &pdev->dev); if (status) { dev_err(&pdev->dev, "ACP IRQ request failed\n"); diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 75c06697fa09..e4f8dbf0d11d 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -394,13 +394,9 @@ static int acp3x_audio_probe(struct platform_device *pdev) if (!adata->acp3x_base) return -ENOMEM; - res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!res) { - dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + adata->i2s_irq = platform_get_irq(pdev, 0); + if (adata->i2s_irq < 0) return -ENODEV; - } - - adata->i2s_irq = res->start; dev_set_drvdata(&pdev->dev, adata); status = devm_snd_soc_register_component(&pdev->dev, diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index 9dd22a2fa2e5..88a242538461 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -399,13 +399,10 @@ static int acp_pdm_audio_probe(struct platform_device *pdev) if (!adata->acp_base) return -ENOMEM; - res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!res) { - dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + adata->pdm_irq = platform_get_irq(pdev, 0); + if (adata->pdm_irq < 0) return -ENODEV; - } - adata->pdm_irq = res->start; adata->capture_stream = NULL; dev_set_drvdata(&pdev->dev, adata); diff --git a/sound/soc/amd/vangogh/acp5x-pcm-dma.c b/sound/soc/amd/vangogh/acp5x-pcm-dma.c index f10de38976cb..c8cd1777a63c 100644 --- a/sound/soc/amd/vangogh/acp5x-pcm-dma.c +++ b/sound/soc/amd/vangogh/acp5x-pcm-dma.c @@ -388,13 +388,10 @@ static int acp5x_audio_probe(struct platform_device *pdev) if (!adata->acp5x_base) return -ENOMEM; - res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!res) { - dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + adata->i2s_irq = platform_get_irq(pdev, 0); + if (adata->i2s_irq < 0) return -ENODEV; - } - adata->i2s_irq = res->start; dev_set_drvdata(&pdev->dev, adata); status = devm_snd_soc_register_component(&pdev->dev, &acp5x_i2s_component, -- 2.20.1

2 1

[PATCH v4 0/2] Add power domains support for digital macro codecs
by Srinivasa Rao Mandadapu 28 Feb '22

28 Feb '22

This patch set is to add power domains support for RX, TX and VA macros. Changes since v3: -- Add dt-bindings support. Changes since v2: -- Remove redundant local variable. -- Update pds error handling sequence. -- Update module description. -- Clean up pds init function. -- Remove redundant arguments. Changes since v1: -- Add missing macros in Kconfig. Srinivasa Rao Mandadapu (2): ASoC: codecs: Add power domains support in digital macro codecs ASoC: qcom: dt-bindings: Add bindings for power domains in lpass digital codecs .../bindings/sound/qcom,lpass-rx-macro.yaml | 8 +++ .../bindings/sound/qcom,lpass-tx-macro.yaml | 8 +++ .../bindings/sound/qcom,lpass-va-macro.yaml | 8 +++ sound/soc/codecs/Kconfig | 7 +++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/lpass-macro-common.c | 67 ++++++++++++++++++++++ sound/soc/codecs/lpass-macro-common.h | 17 ++++++ sound/soc/codecs/lpass-rx-macro.c | 13 ++++- sound/soc/codecs/lpass-tx-macro.c | 10 ++++ sound/soc/codecs/lpass-va-macro.c | 11 +++- sound/soc/qcom/Kconfig | 1 + 11 files changed, 150 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/lpass-macro-common.c create mode 100644 sound/soc/codecs/lpass-macro-common.h -- 2.7.4

2 4

[PATCH] ASoC: fsl: Drop unused argument from imx_pcm_dma_init()
by Sascha Hauer 28 Feb '22

28 Feb '22

Since 70d435ba1cd ("ASoC: imx-pcm-dma: simplify pcm_config") the size argument to imx_pcm_dma_init() is unused, so drop it. Also remove the now unused defines that the users of imx_pcm_dma_init() used to pass the size argument Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- sound/soc/fsl/fsl_aud2htx.c | 2 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/fsl/fsl_spdif.c | 2 +- sound/soc/fsl/fsl_ssi.c | 2 +- sound/soc/fsl/imx-pcm-dma.c | 2 +- sound/soc/fsl/imx-pcm.h | 7 ++----- 7 files changed, 8 insertions(+), 11 deletions(-) diff --git a/sound/soc/fsl/fsl_aud2htx.c b/sound/soc/fsl/fsl_aud2htx.c index 99ab7f0241cf3..422922146f2a5 100644 --- a/sound/soc/fsl/fsl_aud2htx.c +++ b/sound/soc/fsl/fsl_aud2htx.c @@ -241,7 +241,7 @@ static int fsl_aud2htx_probe(struct platform_device *pdev) return ret; } - ret = imx_pcm_dma_init(pdev, IMX_DEFAULT_DMABUF_SIZE); + ret = imx_pcm_dma_init(pdev); if (ret) dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 3a9e2df4e16f6..ed444e8f1d6bf 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -1077,7 +1077,7 @@ static int fsl_esai_probe(struct platform_device *pdev) * Register platform component before registering cpu dai for there * is not defer probe for platform component in snd_soc_add_pcm_runtime(). */ - ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE); + ret = imx_pcm_dma_init(pdev); if (ret) { dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret); goto err_pm_get_sync; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 10544fa27dc0b..d7b990c6429ab 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -1157,7 +1157,7 @@ static int fsl_sai_probe(struct platform_device *pdev) * is not defer probe for platform component in snd_soc_add_pcm_runtime(). */ if (sai->soc_data->use_imx_pcm) { - ret = imx_pcm_dma_init(pdev, IMX_SAI_DMABUF_SIZE); + ret = imx_pcm_dma_init(pdev); if (ret) goto err_pm_get_sync; } else { diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index d178b479c8bd4..57c41b2f7d177 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1522,7 +1522,7 @@ static int fsl_spdif_probe(struct platform_device *pdev) * Register platform component before registering cpu dai for there * is not defer probe for platform component in snd_soc_add_pcm_runtime(). */ - ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE); + ret = imx_pcm_dma_init(pdev); if (ret) { dev_err_probe(&pdev->dev, ret, "imx_pcm_dma_init failed\n"); goto err_pm_disable; diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 1169d1104b9e8..ca30a4ede076e 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -1372,7 +1372,7 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, if (ret) goto error_pcm; } else { - ret = imx_pcm_dma_init(pdev, IMX_SSI_DMABUF_SIZE); + ret = imx_pcm_dma_init(pdev); if (ret) goto error_pcm; } diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 04a9bc749016d..14e94270911cf 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -34,7 +34,7 @@ static const struct snd_dmaengine_pcm_config imx_dmaengine_pcm_config = { .compat_filter_fn = filter, }; -int imx_pcm_dma_init(struct platform_device *pdev, size_t size) +int imx_pcm_dma_init(struct platform_device *pdev) { struct snd_dmaengine_pcm_config *config; diff --git a/sound/soc/fsl/imx-pcm.h b/sound/soc/fsl/imx-pcm.h index 5dd406774d3ea..5c6cf1ca8c8ab 100644 --- a/sound/soc/fsl/imx-pcm.h +++ b/sound/soc/fsl/imx-pcm.h @@ -17,9 +17,6 @@ #define IMX_SSI_DMABUF_SIZE (64 * 1024) #define IMX_DEFAULT_DMABUF_SIZE (64 * 1024) -#define IMX_SAI_DMABUF_SIZE (64 * 1024) -#define IMX_SPDIF_DMABUF_SIZE (64 * 1024) -#define IMX_ESAI_DMABUF_SIZE (256 * 1024) static inline void imx_pcm_dma_params_init_data(struct imx_dma_data *dma_data, @@ -40,9 +37,9 @@ struct imx_pcm_fiq_params { }; #if IS_ENABLED(CONFIG_SND_SOC_IMX_PCM_DMA) -int imx_pcm_dma_init(struct platform_device *pdev, size_t size); +int imx_pcm_dma_init(struct platform_device *pdev); #else -static inline int imx_pcm_dma_init(struct platform_device *pdev, size_t size) +static inline int imx_pcm_dma_init(struct platform_device *pdev) { return -ENODEV; } -- 2.30.2

2 1

[PATCH -next] ASoC: mediatek: mt8195: Remove unnecessary print function dev_err()
by Yang Li 28 Feb '22

28 Feb '22

The print function dev_err() is redundant because platform_get_irq() already prints an error. Eliminate the follow coccicheck warning: ./sound/soc/mediatek/mt8195/mt8195-afe-pcm.c:3126:2-9: line 3126 is redundant because platform_get_irq() already prints an error Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Signed-off-by: Yang Li <yang.lee(a)linux.alibaba.com> --- sound/soc/mediatek/mt8195/mt8195-afe-pcm.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c index b77c2ba5a629..550636500949 100644 --- a/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c +++ b/sound/soc/mediatek/mt8195/mt8195-afe-pcm.c @@ -3122,10 +3122,8 @@ static int mt8195_afe_pcm_dev_probe(struct platform_device *pdev) /* request irq */ irq_id = platform_get_irq(pdev, 0); - if (irq_id < 0) { - dev_err(dev, "%s no irq found\n", dev->of_node->name); + if (irq_id < 0) return -ENXIO; - } ret = devm_request_irq(dev, irq_id, mt8195_afe_irq_handler, IRQF_TRIGGER_NONE, "asys-isr", (void *)afe); -- 2.20.1.7.g153144c

2 1

[PATCH 1/4] ASoC: pcm1681: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the pcm1681 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/pcm1681.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 5b78e9299c95..9eb65f94fc4d 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -136,8 +136,8 @@ static int pcm1681_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_component *component = codec_dai->component; struct pcm1681_private *priv = snd_soc_component_get_drvdata(component); - /* The PCM1681 can only be slave to all clocks */ - if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + /* The PCM1681 can only be consumer to all clocks */ + if ((format & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) { dev_err(component->dev, "Invalid clocking mode\n"); return -EINVAL; } -- 2.30.2

1 4

[PATCH] ASoC: pcm3060: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the pcm3060 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/pcm3060.c | 14 +++++++------- sound/soc/codecs/pcm3060.h | 2 +- 2 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/pcm3060.c b/sound/soc/codecs/pcm3060.c index b2358069cf9b..4e3bfb9fa444 100644 --- a/sound/soc/codecs/pcm3060.c +++ b/sound/soc/codecs/pcm3060.c @@ -68,15 +68,15 @@ static int pcm3060_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - priv->dai[dai->id].is_master = true; + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: + priv->dai[dai->id].is_provider = true; break; - case SND_SOC_DAIFMT_CBS_CFS: - priv->dai[dai->id].is_master = false; + case SND_SOC_DAIFMT_CBC_CFC: + priv->dai[dai->id].is_provider = false; break; default: - dev_err(comp->dev, "unsupported DAI master mode: 0x%x\n", fmt); + dev_err(comp->dev, "unsupported DAI mode: 0x%x\n", fmt); return -EINVAL; } @@ -116,7 +116,7 @@ static int pcm3060_hw_params(struct snd_pcm_substream *substream, unsigned int reg; unsigned int val; - if (!priv->dai[dai->id].is_master) { + if (!priv->dai[dai->id].is_provider) { val = PCM3060_REG_MS_S; goto val_ready; } diff --git a/sound/soc/codecs/pcm3060.h b/sound/soc/codecs/pcm3060.h index 18d51e5dac2c..5e1185e7b03d 100644 --- a/sound/soc/codecs/pcm3060.h +++ b/sound/soc/codecs/pcm3060.h @@ -23,7 +23,7 @@ extern const struct regmap_config pcm3060_regmap; #define PCM3060_CLK2 2 struct pcm3060_priv_dai { - bool is_master; + bool is_provider; unsigned int sclk_freq; }; -- 2.30.2

1 1

[PATCH] ASoC: wl1273: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the wl1273 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wl1273.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index d8ced4559bf2..02232f64110e 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -414,13 +414,13 @@ int wl1273_get_format(struct snd_soc_component *component, unsigned int *fmt) case WL1273_MODE_FM_TX: *fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM; + SND_SOC_DAIFMT_CBP_CFP; break; case WL1273_MODE_BT: *fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF | - SND_SOC_DAIFMT_CBM_CFM; + SND_SOC_DAIFMT_CBP_CFP; break; default: -- 2.30.2

1 1

[PATCH] ASoC: uda1380: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the uda1380 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/uda1380.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 89f2bfeeb70e..13060a9a2388 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -435,8 +435,8 @@ static int uda1380_set_dai_fmt_both(struct snd_soc_dai *codec_dai, iface |= R01_SFORI_MSB | R01_SFORO_MSB; } - /* DATAI is slave only, so in single-link mode, this has to be slave */ - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + /* DATAI is consumer only */ + if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) return -EINVAL; uda1380_write_reg_cache(component, UDA1380_IFACE, iface); @@ -465,8 +465,8 @@ static int uda1380_set_dai_fmt_playback(struct snd_soc_dai *codec_dai, iface |= R01_SFORI_MSB; } - /* DATAI is slave only, so this has to be slave */ - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + /* DATAI is consumer only */ + if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) return -EINVAL; uda1380_write(component, UDA1380_IFACE, iface); @@ -495,7 +495,7 @@ static int uda1380_set_dai_fmt_capture(struct snd_soc_dai *codec_dai, iface |= R01_SFORO_MSB; } - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM) + if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) == SND_SOC_DAIFMT_CBP_CFP) iface |= R01_SIM; uda1380_write(component, UDA1380_IFACE, iface); -- 2.30.2

1 1

[PATCH] ASoC: uda134x: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the uda134x driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/uda134x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index bf9182cedb82..037833c509f7 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -272,9 +272,9 @@ static int uda134x_set_dai_fmt(struct snd_soc_dai *codec_dai, pr_debug("%s fmt: %08X\n", __func__, fmt); - /* codec supports only full slave mode */ - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { - printk(KERN_ERR "%s unsupported slave mode\n", __func__); + /* codec supports only full consumer mode */ + if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) { + printk(KERN_ERR "%s unsupported clocking mode\n", __func__); return -EINVAL; } -- 2.30.2

1 1

[PATCH] ASoC: uda1334: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the uda1334 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/uda1334.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/uda1334.c b/sound/soc/codecs/uda1334.c index 21ab8c5487ba..8670a2a05a56 100644 --- a/sound/soc/codecs/uda1334.c +++ b/sound/soc/codecs/uda1334.c @@ -172,7 +172,7 @@ static int uda1334_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) SND_SOC_DAIFMT_MASTER_MASK); if (fmt != (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS)) { + SND_SOC_DAIFMT_CBC_CFC)) { dev_err(codec_dai->dev, "Invalid DAI format\n"); return -EINVAL; } -- 2.30.2

1 1

[PATCH 1/2] ASoC: tscs42xx: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the tscs42xx driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tscs42xx.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index fb861baf50e8..5b63e017a43b 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -1197,9 +1197,9 @@ static int tscs42xx_set_dai_fmt(struct snd_soc_dai *codec_dai, struct snd_soc_component *component = codec_dai->component; int ret; - /* Slave mode not supported since it needs always-on frame clock */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + /* Consumer mode not supported since it needs always-on frame clock */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: ret = snd_soc_component_update_bits(component, R_AIC1, RM_AIC1_MS, RV_AIC1_MS_MASTER); if (ret < 0) { -- 2.30.2

1 2

[PATCH] ASoC: sti-sas: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the sti-sas driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/sti-sas.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/sti-sas.c b/sound/soc/codecs/sti-sas.c index 82a24e330065..3be4940e3c77 100644 --- a/sound/soc/codecs/sti-sas.c +++ b/sound/soc/codecs/sti-sas.c @@ -154,10 +154,10 @@ static int sti_sas_init_sas_registers(struct snd_soc_component *component, static int sti_sas_dac_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { /* Sanity check only */ - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) { dev_err(dai->component->dev, - "%s: ERROR: Unsupporter master mask 0x%x\n", - __func__, fmt & SND_SOC_DAIFMT_MASTER_MASK); + "%s: ERROR: Unsupported clocking 0x%x\n", + __func__, fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK); return -EINVAL; } -- 2.30.2

1 1

[PATCH] ASoC: si476x: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the si476x driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/si476x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 8d88db9c11a6..8bd2edf70f13 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -69,7 +69,7 @@ static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, int err; u16 format = 0; - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) + if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) return -EINVAL; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { -- 2.30.2

1 1

[PATCH] ASoC: ml26124: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the ml26124 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ml26124.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 4d7c0be2a4aa..0823527e4a75 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -402,12 +402,11 @@ static int ml26124_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned char mode; struct snd_soc_component *component = codec_dai->component; - /* set master/slave audio interface */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: mode = 1; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: mode = 0; break; default: -- 2.30.2

1 1

[PATCH] ASoC: mc13783: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the mc13783 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/mc13783.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index a21072503cb9..08517547e66c 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -181,15 +181,14 @@ static int mc13783_set_fmt(struct snd_soc_dai *dai, unsigned int fmt, } /* DAI clock master masks */ - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: val |= AUDIO_C_CLK_EN; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: val |= AUDIO_CSM; break; - case SND_SOC_DAIFMT_CBM_CFS: - case SND_SOC_DAIFMT_CBS_CFM: + default: return -EINVAL; } @@ -217,11 +216,11 @@ static int mc13783_set_fmt_sync(struct snd_soc_dai *dai, unsigned int fmt) return ret; /* - * In synchronous mode force the voice codec into slave mode + * In synchronous mode force the voice codec into consumer mode * so that the clock / framesync from the stereo DAC is used */ - fmt &= ~SND_SOC_DAIFMT_MASTER_MASK; - fmt |= SND_SOC_DAIFMT_CBS_CFS; + fmt &= ~SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK; + fmt |= SND_SOC_DAIFMT_CBC_CFC; ret = mc13783_set_fmt(dai, fmt, MC13783_AUDIO_CODEC); return ret; -- 2.30.2

1 1

[PATCH 01/10] ASoC: max98088: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the max98088 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/max98088.c | 24 ++++++++++-------------- 1 file changed, 10 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index f8e49e45ce33..429717d4ac5a 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1156,20 +1156,18 @@ static int max98088_dai1_set_fmt(struct snd_soc_dai *codec_dai, if (fmt != cdata->fmt) { cdata->fmt = fmt; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - /* Slave mode PLL */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: + /* Consumer mode PLL */ snd_soc_component_write(component, M98088_REG_12_DAI1_CLKCFG_HI, 0x80); snd_soc_component_write(component, M98088_REG_13_DAI1_CLKCFG_LO, 0x00); break; - case SND_SOC_DAIFMT_CBM_CFM: - /* Set to master mode */ + case SND_SOC_DAIFMT_CBP_CFP: + /* Set to provider mode */ reg14val |= M98088_DAI_MAS; break; - case SND_SOC_DAIFMT_CBS_CFM: - case SND_SOC_DAIFMT_CBM_CFS: default: dev_err(component->dev, "Clock mode unsupported"); return -EINVAL; @@ -1227,20 +1225,18 @@ static int max98088_dai2_set_fmt(struct snd_soc_dai *codec_dai, if (fmt != cdata->fmt) { cdata->fmt = fmt; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - /* Slave mode PLL */ + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: + /* Consumer mode PLL */ snd_soc_component_write(component, M98088_REG_1A_DAI2_CLKCFG_HI, 0x80); snd_soc_component_write(component, M98088_REG_1B_DAI2_CLKCFG_LO, 0x00); break; - case SND_SOC_DAIFMT_CBM_CFM: - /* Set to master mode */ + case SND_SOC_DAIFMT_CBP_CFP: + /* Set to provider mode */ reg1Cval |= M98088_DAI_MAS; break; - case SND_SOC_DAIFMT_CBS_CFM: - case SND_SOC_DAIFMT_CBM_CFS: default: dev_err(component->dev, "Clock mode unsupported"); return -EINVAL; -- 2.30.2

1 10

[PATCH] ASoC: lm49453: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the lm49453 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/lm49453.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index fb0fb23537e7..973d781b4b6a 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1146,17 +1146,17 @@ static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) int clk_phase = 0; int clk_shift = 0; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: aif_val = 0; break; - case SND_SOC_DAIFMT_CBS_CFM: + case SND_SOC_DAIFMT_CBC_CFP: aif_val = LM49453_AUDIO_PORT1_BASIC_SYNC_MS; break; - case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBP_CFC: aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: aif_val = LM49453_AUDIO_PORT1_BASIC_CLK_MS | LM49453_AUDIO_PORT1_BASIC_SYNC_MS; break; -- 2.30.2

1 1

[PATCH] ASoC: isabelle: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the isabelle driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/isabelle.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 79afced75d76..1d86b6a0eb9d 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -973,11 +973,11 @@ static int isabelle_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct snd_soc_component *component = codec_dai->component; unsigned int aif_val = 0; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: aif_val &= ~ISABELLE_AIF_MS; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: aif_val |= ISABELLE_AIF_MS; break; default: -- 2.30.2

1 1

[PATCH] ASoC: inno_rk3036: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the inno_rk3036 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/inno_rk3036.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/inno_rk3036.c b/sound/soc/codecs/inno_rk3036.c index e05c4f27486e..ca0f4c1911e4 100644 --- a/sound/soc/codecs/inno_rk3036.c +++ b/sound/soc/codecs/inno_rk3036.c @@ -200,12 +200,12 @@ static int rk3036_codec_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) dev_dbg(component->dev, "rk3036_codec dai set fmt : %08x\n", fmt); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: reg01_val |= INNO_R01_PINDIR_IN_SLAVE | INNO_R01_I2SMODE_SLAVE; break; - case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBP_CFP: reg01_val |= INNO_R01_PINDIR_OUT_MASTER | INNO_R01_I2SMODE_MASTER; break; -- 2.30.2

1 1

[PATCH] ASoC: es8328: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the es8328 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/es8328.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/es8328.c b/sound/soc/codecs/es8328.c index 9632afc2d4d6..3f00ead97006 100644 --- a/sound/soc/codecs/es8328.c +++ b/sound/soc/codecs/es8328.c @@ -84,7 +84,7 @@ struct es8328_priv { int mclkdiv2; const struct snd_pcm_hw_constraint_list *sysclk_constraints; const int *mclk_ratios; - bool master; + bool provider; struct regulator_bulk_data supplies[ES8328_SUPPLY_NUM]; }; @@ -462,7 +462,7 @@ static int es8328_startup(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct es8328_priv *es8328 = snd_soc_component_get_drvdata(component); - if (es8328->master && es8328->sysclk_constraints) + if (es8328->provider && es8328->sysclk_constraints) snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, es8328->sysclk_constraints); @@ -486,7 +486,7 @@ static int es8328_hw_params(struct snd_pcm_substream *substream, else reg = ES8328_ADCCONTROL5; - if (es8328->master) { + if (es8328->provider) { if (!es8328->sysclk_constraints) { dev_err(component->dev, "No MCLK configured\n"); return -EINVAL; @@ -590,19 +590,19 @@ static int es8328_set_dai_fmt(struct snd_soc_dai *codec_dai, u8 dac_mode = 0; u8 adc_mode = 0; - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBP_CFP: /* Master serial port mode, with BCLK generated automatically */ snd_soc_component_update_bits(component, ES8328_MASTERMODE, ES8328_MASTERMODE_MSC, ES8328_MASTERMODE_MSC); - es8328->master = true; + es8328->provider = true; break; - case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBC_CFC: /* Slave serial port mode */ snd_soc_component_update_bits(component, ES8328_MASTERMODE, ES8328_MASTERMODE_MSC, 0); - es8328->master = false; + es8328->provider = false; break; default: return -EINVAL; -- 2.30.2

1 1

[PATCH] ASoC: es8316: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the es8316 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/es8316.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 8f30a3ea8bfe..ff33eab6f9de 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -401,8 +401,8 @@ static int es8316_set_dai_fmt(struct snd_soc_dai *codec_dai, u8 clksw; u8 mask; - if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { - dev_err(component->dev, "Codec driver only supports slave mode\n"); + if ((fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) != SND_SOC_DAIFMT_CBC_CFC) { + dev_err(component->dev, "Codec driver only supports consumer mode\n"); return -EINVAL; } -- 2.30.2

1 1

[PATCH] ASoC: es7241: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the es7241 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/es7241.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) diff --git a/sound/soc/codecs/es7241.c b/sound/soc/codecs/es7241.c index 9f20bfb855b3..0baa86241cf9 100644 --- a/sound/soc/codecs/es7241.c +++ b/sound/soc/codecs/es7241.c @@ -29,7 +29,7 @@ struct es7241_data { struct gpio_desc *m1; unsigned int fmt; unsigned int mclk; - bool is_slave; + bool is_consumer; const struct es7241_chip *chip; }; @@ -46,9 +46,9 @@ static void es7241_set_mode(struct es7241_data *priv, int m0, int m1) gpiod_set_value_cansleep(priv->reset, 1); } -static int es7241_set_slave_mode(struct es7241_data *priv, - const struct es7241_clock_mode *mode, - unsigned int mfs) +static int es7241_set_consumer_mode(struct es7241_data *priv, + const struct es7241_clock_mode *mode, + unsigned int mfs) { int j; @@ -67,9 +67,9 @@ static int es7241_set_slave_mode(struct es7241_data *priv, return 0; } -static int es7241_set_master_mode(struct es7241_data *priv, - const struct es7241_clock_mode *mode, - unsigned int mfs) +static int es7241_set_provider_mode(struct es7241_data *priv, + const struct es7241_clock_mode *mode, + unsigned int mfs) { /* * We can't really set clock ratio, if the mclk/lrclk is different @@ -98,10 +98,10 @@ static int es7241_hw_params(struct snd_pcm_substream *substream, if (rate < mode->rate_min || rate >= mode->rate_max) continue; - if (priv->is_slave) - return es7241_set_slave_mode(priv, mode, mfs); + if (priv->is_consumer) + return es7241_set_consumer_mode(priv, mode, mfs); else - return es7241_set_master_mode(priv, mode, mfs); + return es7241_set_provider_mode(priv, mode, mfs); } /* should not happen */ @@ -136,12 +136,12 @@ static int es7241_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBS_CFS: - priv->is_slave = true; + switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) { + case SND_SOC_DAIFMT_CBC_CFC: + priv->is_consumer = true; break; - case SND_SOC_DAIFMT_CBM_CFM: - priv->is_slave = false; + case SND_SOC_DAIFMT_CBP_CFP: + priv->is_consumer = false; break; default: -- 2.30.2

1 1

[PATCH] ASoC: es7134: Use modern ASoC DAI format terminology
by Mark Brown 28 Feb '22

28 Feb '22

As part of moving to remove the old style defines for the bus clocks update the es7134 driver to use more modern terminology for clocking. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/es7134.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/es7134.c b/sound/soc/codecs/es7134.c index e2b79879354b..f443351677df 100644 --- a/sound/soc/codecs/es7134.c +++ b/sound/soc/codecs/es7134.c @@ -94,7 +94,7 @@ static int es7134_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) SND_SOC_DAIFMT_MASTER_MASK); if (fmt != (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS)) { + SND_SOC_DAIFMT_CBC_CFC)) { dev_err(codec_dai->dev, "Invalid DAI format\n"); return -EINVAL; } -- 2.30.2

1 1

[PATCH 00/17] ASoC: Intel: AVS - Audio DSP for cAVS
by Cezary Rojewski 28 Feb '22

28 Feb '22

A continuation of cleanup work of Intel SST solutions found in sound/soc/intel/. With two major chapters released last year catpt [1] and removal of haswell solution [2], time has come for Skylake-driver. Througout 2019, 2020 and 2021 Skylake-driver has had many fixes applied and even attempts of refactors as seen in fundamental overhaul [3], IPC flow adjustments [4] and LARGE_CONFIG overhaul [5] series. Unfortunately, story repeats itself - problems are found within the core of a driver. Painting it with different colors does not change the fact that is it still a house of cards. As changes needed to address those issues would make Skylake solution incompatible with its previous revisions, a decision has been made to provide a new solution instead. In time it would deprecate and replace Skylake-driver. That solution has been called AVS - from AudioDSP architecture name: Audio-Voice-Speech. It is meant to provide support for the exact same range of platforms as its predecessor: SKL, KBL, AML and APL. Note: this series is dependent upon HDA-series [6] which exposes several codec-organization functions allowing for reduced code size on avs-driver side. Note: this series does not add fully functional driver as its size would get out of control. Here, focus is put on adding IPC protocol and code loading code. Changes RFC v1 [7]: -> v1: - separated HDA codec-organization patches, path and topology handling, PCM and complementary features such as recovery from this series to ease the review process - fixed EXPORT_SYMBOL_GPL for exported members of ASoC framework - result of stall() is now checked when sending ROM message - result of snd_hdac_ext_stream_set_spib() is now checked when loading basefw - if basefw is not ready, notification processing is now skipped - documented several topology parsing helpers Changes [internal] RFC v2 -> [public] RFC v1: - dropped any sysfs related changes from this series, moved to follow up one - dropped entire subscription-mechanism found in ipc.c. Handlers that are delegated to service certain firmware notifications are now called directly - fixed kernel doc for snd_soc_dapm_new_dai_widgets() as reported by ikp - prefixed snd_hda_codec_device_init() as suggested by Amadeo - improved comments for d0ix transitions for APL-based platforms as suggested by Pierre - a ton of spelling related fixes in most of the commit messages - fixed remaining warnings pointed by scan-build (variable assigned but not used) - replaced most of 'cAVS X.Y' expression usages with 'platform-based' equivalents as suggested by Pierre e.g.: cAVS 1.5 -> SKL-based Changes [internal] RFC v1 -> [internal] RFC v2: - fixed memleak caused by lack of kfree(vols) if memory allocation fails in avs_peakvol_create() as reported by Curtis - fixed missing 'i' iterator incrementation in avs_widget_ready() causing reference loss as reported by Curtis - replace hardcode: 0x40 usage with snd_hdac_calc_stream_format as suggested by Curtis. In consequence, readability for all code loading (CL) procedues has increased and such approach auto-documents the CL stream preparation - updated behavior of all index-fetching functions found in utils.c: avs_module_entry_index(), avs_module_id_entry_index() and follow ups: avs_get_module_entry(), avs_get_module_id_entry() to better conform to linux-kernel standard when no entry is found (return -ENOENT) rather than C++ standard (return -1, what in kernel case translated to -EPERM) as suggested by Curtis and Peter - several suggestions have been made regarding spacing, and so far, I've agreed and applied with all of them. None proposed seemed out of place or redundant - avs_path_stop() renamed to avs_path_pause() pipeline states are represented by RESET/PAUSED/RUNNING. avs_path_reset() and avs_path_run() were already there and avs_path_stop() just didn't look cohesive - added missing parsers for num_output_pin and num_input_pin which are required for custom modules such as WAVES or DSM - dropped 'priv_param_length' from custom module descriptor as this field is obsolete in firmware - parse_dictionary() has been split into parse_dictionary_header() and parse_dictionary_entries() to drop code duplication present in several parsing function which could not re-use entire parse_dictionary() - added avs_tplg_vendor_array_lookup_next() and avs_tplg_vendor_entry_next() to drop code duplicated present in several parsing functions. This change greatly impacted readability of all parsers - parsing helpers such avs_tplg_vendor_array_lookup() now return offset by updating specified in function argument list u32 *offset variable. This is to address problem when u32 offset would be greater than max int, which is the return type for these functions - AVS_DEFINE_PTR_PARSER() macro has been introduced to drop code duplication for all ptr-parsing users - all struct avs_path_module creators have had their declaration updated: function argument *owner ceased to exist as it could already be accessed by mod->owner - fixed the order of operation for conditional paths (e.g.: echo reference) so these are no longer controlled by "source" path and instead are impacted by state changes of source and sink paths both. Previously only source path e.g. playback sourcing echo reference would trigger RUNNING status for conditional path. Equivalent RUNNING on WoV path which is in this case sink path, would not do so, leading to order-of-operation problems. Behavior has been changed to: both source and sink need to be RUNNING for conditional path to be set to RUNNING too. PAUSED for either source or sink will cause PAUSED transition for conditional path. - to achieve the above, path states are now saved in 'state' i.e. new u32 field for struct avs_path - resigned from fw_filename field usage in favour of newly added tplg_filename for machine board descriptors as suggested by Pierre - platform descriptor fields have had their names update better reflect their purpose as suggested by Pierre - fixed comp_list missing locking when manipulated - all message senders now accept request as pointer as suggeseted by Peter - resigned of AZX_ usage for all ADSP-related registers, leaving them only for HOST memory space related operations - fixed disable path for core DSP operations: power/reset/stall as reported by Peter - safety when locking between received responses (reply vs notification) has been lowered as suggested by Pierre. Most usages are not performed in IRQ context and none is done in hard-IRQ one - s/master/main/ plus AVS_MAIN_CORE_MASK has replaced ->master_mask - several functions have had their logging updated - logs have been moved to lower level functions as suggested by Pierre - hdac_ext_stream usage has been streamlined to estream, hdac_streams are represented by hstream instead - hw_params() are resilient to scenarios when they are called mutliple times as reported by Pierre - avs_dsp_enable() now collapses if any of its steps fails as reported by Pierre and Peter - avs_module_ida_empty() now returns value of type bool as suggested by Bard [1]: https://www.spinics.net/lists/alsa-devel/msg116440.html [2]: https://www.spinics.net/lists/alsa-devel/msg116901.html [3]: https://www.spinics.net/lists/alsa-devel/msg94199.html [4]: https://www.spinics.net/lists/alsa-devel/msg92588.html [5]: https://lore.kernel.org/all/20190808181549.12521-1-cezary.rojewski@intel.co… [6]: https://lore.kernel.org/alsa-devel/20220207114906.3759800-1-cezary.rojewski… [7]: https://lore.kernel.org/all/20211208111301.1817725-1-cezary.rojewski@intel.… Cezary Rojewski (17): ALSA: hda: Add helper macros for DSP capable devices ASoC: Export DAI register and widget ctor and dctor functions ASoC: Intel: Introduce AVS driver ASoC: Intel: avs: Inter process communication ASoC: Intel: avs: Add code loading requests ASoC: Intel: avs: Add pipeline management requests ASoC: Intel: avs: Add module management requests ASoC: Intel: avs: Add power management requests ASoC: Intel: avs: Add ROM requests ASoC: Intel: avs: Add basefw runtime-parameter requests ASoC: Intel: avs: Firmware resources management utilities ASoC: Intel: avs: Declare module configuration types ASoC: Intel: avs: Dynamic firmware resources management ASoC: Intel: avs: General code loading flow ASoC: Intel: avs: Implement CLDMA transfer ASoC: Intel: avs: Code loading over CLDMA ASoC: Intel: avs: Code loading over HDA include/sound/hdaudio.h | 2 + include/sound/hdaudio_ext.h | 49 ++ include/sound/soc-dapm.h | 1 + sound/soc/intel/Kconfig | 15 + sound/soc/intel/Makefile | 1 + sound/soc/intel/avs/Makefile | 6 + sound/soc/intel/avs/avs.h | 226 ++++++++++ sound/soc/intel/avs/cldma.c | 328 ++++++++++++++ sound/soc/intel/avs/cldma.h | 29 ++ sound/soc/intel/avs/core.c | 62 +++ sound/soc/intel/avs/dsp.c | 303 +++++++++++++ sound/soc/intel/avs/ipc.c | 410 +++++++++++++++++ sound/soc/intel/avs/loader.c | 594 +++++++++++++++++++++++++ sound/soc/intel/avs/messages.c | 642 +++++++++++++++++++++++++++ sound/soc/intel/avs/messages.h | 762 ++++++++++++++++++++++++++++++++ sound/soc/intel/avs/registers.h | 75 ++++ sound/soc/intel/avs/utils.c | 282 ++++++++++++ sound/soc/soc-core.c | 1 + sound/soc/soc-dapm.c | 15 + 19 files changed, 3803 insertions(+) create mode 100644 sound/soc/intel/avs/Makefile create mode 100644 sound/soc/intel/avs/avs.h create mode 100644 sound/soc/intel/avs/cldma.c create mode 100644 sound/soc/intel/avs/cldma.h create mode 100644 sound/soc/intel/avs/core.c create mode 100644 sound/soc/intel/avs/dsp.c create mode 100644 sound/soc/intel/avs/ipc.c create mode 100644 sound/soc/intel/avs/loader.c create mode 100644 sound/soc/intel/avs/messages.c create mode 100644 sound/soc/intel/avs/messages.h create mode 100644 sound/soc/intel/avs/registers.h create mode 100644 sound/soc/intel/avs/utils.c -- 2.25.1

3 60

[PATCH v4 1/2] ALSA: core: remove initialise static variables to 0
by Meng Tang 28 Feb '22

28 Feb '22

Initializing the static variable to 0 causes the following error when exec checkpatch: ERROR: do not initialise statics to 0 FILE: sound/sound_core.c:142: static int preclaim_oss = 0; In addition, considering the following way of writing 139: #ifdef config_sound_oss_core_preclaim 140: Static int preclaim_oss = 1; 141: #ELSE 142: Static int preclaim_oss = 0; 143: #ENDIF We can optimize it by IS_ENABLED(CONFIG_SOUND_OSS_CORE_PRECLAIM), so modified it to static int preclaim_oss = IS_ENABLED(CONFIG_SOUND_OSS_CORE_PRECLAIM); Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> Signed-off-by: Joe Perches <joe(a)perches.com> --- sound/sound_core.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 90d118cd9164..aa4a57e488e5 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -136,11 +136,7 @@ struct sound_unit * All these clutters are scheduled to be removed along with * sound-slot/service-* module aliases. */ -#ifdef CONFIG_SOUND_OSS_CORE_PRECLAIM -static int preclaim_oss = 1; -#else -static int preclaim_oss = 0; -#endif +static int preclaim_oss = IS_ENABLED(CONFIG_SOUND_OSS_CORE_PRECLAIM); module_param(preclaim_oss, int, 0444); -- 2.20.1

4 6

[PATCH] ALSA: spi: Add check for clk_enable()
by Jiasheng Jiang 28 Feb '22

28 Feb '22

As the potential failure of the clk_enable(), it should be better to check it and return error if fails. Fixes: 3568459a5113 ("ALSA: at73c213: manage SSC clock") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/spi/at73c213.c | 27 +++++++++++++++++++++------ 1 file changed, 21 insertions(+), 6 deletions(-) diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 76c0e37a838c..8a2da6b1012e 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -218,7 +218,9 @@ static int snd_at73c213_pcm_open(struct snd_pcm_substream *substream) runtime->hw = snd_at73c213_playback_hw; chip->substream = substream; - clk_enable(chip->ssc->clk); + err = clk_enable(chip->ssc->clk); + if (err) + return err; return 0; } @@ -776,7 +778,9 @@ static int snd_at73c213_chip_init(struct snd_at73c213 *chip) goto out; /* Enable DAC master clock. */ - clk_enable(chip->board->dac_clk); + retval = clk_enable(chip->board->dac_clk); + if (retval) + goto out; /* Initialize at73c213 on SPI bus. */ retval = snd_at73c213_write_reg(chip, DAC_RST, 0x04); @@ -889,7 +893,9 @@ static int snd_at73c213_dev_init(struct snd_card *card, chip->card = card; chip->irq = -1; - clk_enable(chip->ssc->clk); + retval = clk_enable(chip->ssc->clk); + if (retval) + return retval; retval = request_irq(irq, snd_at73c213_interrupt, 0, "at73c213", chip); if (retval) { @@ -1008,7 +1014,9 @@ static int snd_at73c213_remove(struct spi_device *spi) int retval; /* Stop playback. */ - clk_enable(chip->ssc->clk); + retval = clk_enable(chip->ssc->clk); + if (retval) + goto out; ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); clk_disable(chip->ssc->clk); @@ -1088,9 +1096,16 @@ static int snd_at73c213_resume(struct device *dev) { struct snd_card *card = dev_get_drvdata(dev); struct snd_at73c213 *chip = card->private_data; + int retval; - clk_enable(chip->board->dac_clk); - clk_enable(chip->ssc->clk); + retval = clk_enable(chip->board->dac_clk); + if (retval) + return retval; + retval = clk_enable(chip->ssc->clk); + if (retval) { + clk_disable(chip->board->dac_clk); + return retval; + } ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXEN)); return 0; -- 2.25.1

2 1

[PATCH] sound/mips: Use platform_get_irq() to get the interrupt
by Meng Tang 28 Feb '22

28 Feb '22

platform_get_resource(pdev, IORESOURCE_IRQ, ..) relies on static allocation of IRQ resources in DT core code, this causes an issue when using hierarchical interrupt domains using "interrupts" property in the node as this bypassed the hierarchical setup and messed up the irq chaining. In preparation for removal of static setup of IRQ resource from DT core code use platform_get_irq(). Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/mips/snd-n64.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/mips/snd-n64.c b/sound/mips/snd-n64.c index 463a6fe589eb..bff6d85b8fe2 100644 --- a/sound/mips/snd-n64.c +++ b/sound/mips/snd-n64.c @@ -289,8 +289,7 @@ static int __init n64audio_probe(struct platform_device *pdev) struct snd_card *card; struct snd_pcm *pcm; struct n64audio *priv; - struct resource *res; - int err; + int err, irq; err = snd_card_new(&pdev->dev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1, @@ -337,12 +336,12 @@ static int __init n64audio_probe(struct platform_device *pdev) strcpy(card->shortname, "N64 Audio"); strcpy(card->longname, "N64 Audio"); - res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!res) { + irq = platform_get_irq(pdev, 0); + if (irq < 0) { err = -EINVAL; goto fail_dma_alloc; } - if (devm_request_irq(&pdev->dev, res->start, n64audio_isr, + if (devm_request_irq(&pdev->dev, irq, n64audio_isr, IRQF_SHARED, "N64 Audio", priv)) { err = -EBUSY; goto fail_dma_alloc; -- 2.20.1

2 1

[PATCH V0 0/1] ALSA: pcm: fix blocking while loop in
by Raghu Bankapur 28 Feb '22

28 Feb '22

When period interrupts are disabled, while loop in snd_pcm_update_hw_ptr0() results in the machine locking up if runtime->hw_ptr_buffer_jiffies is 0. Validate runtime->hw_ptr_buffer_jiffies value before while loop to avoid delta check. when runtime->hw_ptr_buffer_jiffies is 0 then update of while condition hdelta > xrun_threshold will be always true because hdelta will not change. This will lead to infinite loop causing lock-up This issue found during stability test, below log explains CPU lock-up on CPU1 [ 922.249028] sched: RT throttling activated for rt_rq ffffffc0f9970778 (cpu 1) [ 922.249028] potential CPU hogs: [ 922.249028] **FastCapture (11426) [ 922.249105] ------------[ cut here ]--------- To fix this lock-up issue, added new condition to check if runtime-> hw_ptr_buffer_jiffies is ZERO or negative. Raghu Bankapur (1): ALSA: pcm: fix blocking while loop in snd_pcm_update_hw_ptr0() sound/core/pcm_lib.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) -- 2.17.1

4 5

[PATCH] ASoC: qcom: soundwire: Add support for controlling audio CGCR from HLOS
by Srinivasa Rao Mandadapu 28 Feb '22

28 Feb '22

Add support for controlling soundwire audio CGCR interface using clock framework to make hclk ungating with software. As per new hardware changes, software has to always ungate hclk if soundwire is operational and keep it running. This requirement is for latest LPASS chipsets for RX, TX and WSA path to work. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Co-developed-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> Signed-off-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> --- This patch set depends on: -- Clock driver patches with reset control support, yet to be posted. drivers/soundwire/qcom.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index fb183bd..440aad8 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -12,6 +12,7 @@ #include <linux/of_irq.h> #include <linux/of_device.h> #include <linux/regmap.h> +#include <linux/reset.h> #include <linux/slab.h> #include <linux/slimbus.h> #include <linux/soundwire/sdw.h> @@ -129,6 +130,7 @@ struct qcom_swrm_ctrl { struct device *dev; struct regmap *regmap; void __iomem *mmio; + struct reset_control *audio_cgcr; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs; #endif @@ -618,6 +620,8 @@ static int qcom_swrm_init(struct qcom_swrm_ctrl *ctrl) val = FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_ROW_CTRL_BMSK, ctrl->rows_index); val |= FIELD_PREP(SWRM_MCP_FRAME_CTRL_BANK_COL_CTRL_BMSK, ctrl->cols_index); + reset_control_reset(ctrl->audio_cgcr); + ctrl->reg_write(ctrl, SWRM_MCP_FRAME_CTRL_BANK_ADDR(0), val); /* Enable Auto enumeration */ @@ -1269,6 +1273,9 @@ static int qcom_swrm_probe(struct platform_device *pdev) ctrl->bus.ops = &qcom_swrm_ops; ctrl->bus.port_ops = &qcom_swrm_port_ops; ctrl->bus.compute_params = &qcom_swrm_compute_params; + ctrl->audio_cgcr = devm_reset_control_get_exclusive(dev, "swr_audio_cgcr"); + if (IS_ERR(ctrl->audio_cgcr)) + dev_err(dev, "Failed to get audio_cgcr reset required for soundwire-v1.6.0\n"); ret = qcom_swrm_get_port_config(ctrl); if (ret) -- 2.7.4

2 3

Add UCM for PinePhone
by GitHub pull_request - edited 28 Feb '22

28 Feb '22

alsa-project/alsa-ucm-conf pull request #134 was edited from tomfitzhenry: Fixes https://github.com/alsa-project/alsa-ucm-conf/issues/124. The device-tree configuration has had a unique audio card name since 5.14: https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/arc… This UCM config is copied from https://gitlab.com/pine64-org/pine64-alsa-ucm/, which was developed by many developers ( https://gitlab.com/pine64-org/pine64-alsa-ucm/-/merge_requests/1 ) and is BSD-3-Clause licensed. I have tested that with this PR, the PinePhone has the expected HiFi and VoiceCall profiles.This testing found that the `File` directive in `SectionUseCase` needs to be absolute (for the symlink to work). I've fixed this in this PR: https://github.com/alsa-project/alsa-ucm-conf/pull/134/commits/5059e57a2197…. See also: * `alsa-info`: https://alsa-project.org/db/?f=5e331f5ef10bd167b3bd7640bfcf0e879100f80d * `pa-info`: [pa-info.txt](https://github.com/alsa-project/alsa-ucm-conf/files/8015083/pa… * https://gitlab.com/pine64-org/pine64-alsa-ucm/-/issues/3 Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/134.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 0/3] soundwire: qcom: add pm runtime support
by Srinivas Kandagatla 28 Feb '22

28 Feb '22

This patchset adds pm runtime support to Qualcomm SounWire Controller using SoundWire Clock Stop and Wake up using Headset events on supported instances and instances like WSA which do not support clock stop a soft reset of controller along with full rest of slaves is done to resume from a low power state. Tested it on SM8250 MTP and Dragon Board DB845c --srini Changes since v1: - updated wake irq to not deal with slave pm runtime directly. - added delay after soft reset of SoundWire controller where clock stop is not supported Srinivas Kandagatla (3): soundwire: qcom: add runtime pm support dt-bindings: soundwire: qcom: document optional wake irq soundwire: qcom: add in-band wake up interrupt support .../bindings/soundwire/qcom,sdw.txt | 2 +- drivers/soundwire/qcom.c | 202 +++++++++++++++++- 2 files changed, 202 insertions(+), 2 deletions(-) -- 2.21.0

3 8

ucm2: add additional symlink for Pinebook Pro
by GitHub pull_request - opened 28 Feb '22

28 Feb '22

alsa-project/alsa-ucm-conf pull request #140 was opened from kmcopper: since 907f0a305186 ASoC: simple-card: Fill in driver name of the linux kernel, the base name is simple-card instead of rockchip_es8316 Fixes: 85eea19783cf ("ucm2: Add UCM support for rockchip_es8316 on Pinebook Pro") Signed-off-by: Kyle Copperfield <kmcopper(a)danwin1210.me> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/140 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/140.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v3 1/2] ALSA: core: remove initialise static variables to 0
by Meng Tang 28 Feb '22

28 Feb '22

Initializing the static variable to 0 causes the following error when exec checkpatch: ERROR: do not initialise statics to 0 FILE: sound/sound_core.c:142: static int preclaim_oss = 0; In addition, considering the following way of writing 139: #ifdef config_sound_oss_core_preclaim 140: Static int preclaim_oss = 1; 141: #ELSE 142: Static int preclaim_oss = 0; 143: #ENDIF We can optimize it by IS_ENABLED(CONFIG_SOUND_OSS_CORE_PRECLAIM), so modified it to static int preclaim_oss = IS_ENABLED(CONFIG_SOUND_OSS_CORE_PRECLAIM); Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> Signed-off-by: JOE Perches <joe(a)perches.com> --- sound/sound_core.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 90d118cd9164..aa4a57e488e5 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -136,11 +136,7 @@ struct sound_unit * All these clutters are scheduled to be removed along with * sound-slot/service-* module aliases. */ -#ifdef CONFIG_SOUND_OSS_CORE_PRECLAIM -static int preclaim_oss = 1; -#else -static int preclaim_oss = 0; -#endif +static int preclaim_oss = IS_ENABLED(CONFIG_SOUND_OSS_CORE_PRECLAIM); module_param(preclaim_oss, int, 0444); -- 2.20.1

2 2

[PATCH v2 1/2] ALSA: core: remove initialise static variables to 0
by Meng Tang 28 Feb '22

28 Feb '22

Initializing the static variable to 0 causes the following error when exec checkpatch: ERROR: do not initialise statics to 0 FILE: sound/sound_core.c:142: static int preclaim_oss = 0; In addition, considering the following way of writing 139: #ifdef config_sound_oss_core_preclaim 140: Static int preclaim_oss = 1; 141: #ELSE 142: Static int preclaim_oss = 0; 143: #ENDIF We can optimize it by IS_ENABLED(CONFIG_SOUND_OSS_CORE_PRECLAIM), so modified it to static int preclaim_oss = IS_ENABLED(CONFIG_SOUND_OSS_CORE_PRECLAIM); Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/sound_core.c | 6 +----- 1 file changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 90d118cd9164..aa4a57e488e5 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -136,11 +136,7 @@ struct sound_unit * All these clutters are scheduled to be removed along with * sound-slot/service-* module aliases. */ -#ifdef CONFIG_SOUND_OSS_CORE_PRECLAIM -static int preclaim_oss = 1; -#else -static int preclaim_oss = 0; -#endif +static int preclaim_oss = IS_ENABLED(CONFIG_SOUND_OSS_CORE_PRECLAIM); module_param(preclaim_oss, int, 0444); -- 2.20.1

3 3

[PATCH 1/2] ALSA: core: remove initialise static variables to 0
by Meng Tang 27 Feb '22

27 Feb '22

Initializing the static variable to 0 causes the following error when exec checkpatch: ERROR: do not initialise statics to 0 FILE: sound/sound_core.c:142: static int preclaim_oss = 0; Static variable does not need to be initialised to 0, because compiler will initialise all uninitialised statics to 0. Thus, remove the unneeded initializations. Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/sound_core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 90d118cd9164..2ee54070b2b5 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -139,7 +139,7 @@ struct sound_unit #ifdef CONFIG_SOUND_OSS_CORE_PRECLAIM static int preclaim_oss = 1; #else -static int preclaim_oss = 0; +static int preclaim_oss; #endif module_param(preclaim_oss, int, 0444); -- 2.20.1

2 2

[PATCH v2] soundwire: qcom: remove redundant wait for completion
by Srinivasa Rao Mandadapu 26 Feb '22

26 Feb '22

Remove wait_for_completion_timeout from soundwire probe as it seems unnecessary and device enumeration is anyway not happening here, hence this api is blocking till it completes max wait time. Also, as device enumeration event is dependent on wcd938x probe to be completed, its of no use waiting here. Waiting here increasing the boot time almost 4 seconds and impacting other modules like touch screen. Fixes: 06dd96738d618 ("soundwire: qcom: wait for enumeration to be complete in probe") Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Co-developed-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> Signed-off-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> --- drivers/soundwire/qcom.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 5481341..9a32a24 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1309,8 +1309,6 @@ static int qcom_swrm_probe(struct platform_device *pdev) } qcom_swrm_init(ctrl); - wait_for_completion_timeout(&ctrl->enumeration, - msecs_to_jiffies(TIMEOUT_MS)); ret = qcom_swrm_register_dais(ctrl); if (ret) goto err_master_add; -- 2.7.4

3 5

[PATCH] "soundwire: qcom: revert wait for enumeration to be complete in probe"
by Srinivasa Rao Mandadapu 26 Feb '22

26 Feb '22

Revert wait for enumeration from soundwire probe as it seems unnecessary and device enumeration is anyway not happening here, hence this api is blocking till it completes max wait time. Also, as device enumeration event is dependent on wcd938x probe to be completed, its of no use waiting here. Waiting here increasing the boot time almost 4 seconds and impacting other modules like touch screen. Revert: commit 06dd96738d618 ("soundwire: qcom: wait for enumeration to be complete in probe") Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Co-developed-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> Signed-off-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> --- drivers/soundwire/qcom.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 5481341..9eb31ba 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -133,7 +133,6 @@ struct qcom_swrm_ctrl { struct dentry *debugfs; #endif struct completion broadcast; - struct completion enumeration; struct work_struct slave_work; /* Port alloc/free lock */ struct mutex port_lock; @@ -486,7 +485,6 @@ static int qcom_swrm_enumerate(struct sdw_bus *bus) } } - complete(&ctrl->enumeration); return 0; } @@ -1262,7 +1260,6 @@ static int qcom_swrm_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, ctrl); mutex_init(&ctrl->port_lock); init_completion(&ctrl->broadcast); - init_completion(&ctrl->enumeration); ctrl->bus.ops = &qcom_swrm_ops; ctrl->bus.port_ops = &qcom_swrm_port_ops; @@ -1309,8 +1306,6 @@ static int qcom_swrm_probe(struct platform_device *pdev) } qcom_swrm_init(ctrl); - wait_for_completion_timeout(&ctrl->enumeration, - msecs_to_jiffies(TIMEOUT_MS)); ret = qcom_swrm_register_dais(ctrl); if (ret) goto err_master_add; -- 2.7.4

1 0

[RESEND v4 0/2] Add power domains support for digital macro codecs
by Srinivasa Rao Mandadapu 26 Feb '22

26 Feb '22

This patch set is to add power domains support for RX, TX and VA macros. Changes since v3: -- Add dt-bindings support. Changes since v2: -- Remove redundant local variable. -- Update pds error handling sequence. -- Update module description. -- Clean up pds init function. -- Remove redundant arguments. Changes since v1: -- Add missing macros in Kconfig. Srinivasa Rao Mandadapu (2): ASoC: codecs: Add power domains support in digital macro codecs ASoC: qcom: dt-bindings: Add bindings for power domains in lpass digital codecs .../bindings/sound/qcom,lpass-rx-macro.yaml | 8 +++ .../bindings/sound/qcom,lpass-tx-macro.yaml | 8 +++ .../bindings/sound/qcom,lpass-va-macro.yaml | 8 +++ sound/soc/codecs/Kconfig | 7 +++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/lpass-macro-common.c | 67 ++++++++++++++++++++++ sound/soc/codecs/lpass-macro-common.h | 17 ++++++ sound/soc/codecs/lpass-rx-macro.c | 12 +++- sound/soc/codecs/lpass-tx-macro.c | 10 ++++ sound/soc/codecs/lpass-va-macro.c | 12 ++++ sound/soc/qcom/Kconfig | 1 + 11 files changed, 151 insertions(+), 1 deletion(-) create mode 100644 sound/soc/codecs/lpass-macro-common.c create mode 100644 sound/soc/codecs/lpass-macro-common.h -- 2.7.4

1 2

Need help understanding DPCM
by Aidan MacDonald 25 Feb '22

25 Feb '22

Hi all. I want to extend an existing ASoC driver (jz4740-i2s) so it can be used with an external (discrete) codec; currently it is hardcoded to use a SoC internal codec. I'd like to be able to switch between the two codecs at runtime. Switching between the internal and external codecs is done by toggling a bit in a register. The switch determines where the I2S signals get routed: to the internal codec or the external one. Because of this only one of the codecs can be used at a time. It sounds like I need DPCM to handle this sort of thing - I have a single FIFO on the CPU for audio data (an FE DAI?) and it can connect to one of two codecs (the BE DAIs?) via a switch. But most of the drivers I see using DPCM look way more complicated: big DSP chips, and CPUs with a lot of DAIs. I am having a hard time separating out the essential bits from the overall "mess." The documentation is not very helpful either. Could someone helpfully explain what a (presumably) very basic DPCM setup for this situation would look like? Or is DPCM overkill - should I be doing something else entirely? I want to do this the "right" way so I can send patches upstream. If possible, I'd like to use one of the 'generic' audio card drivers like simple-audio-card. BTW, what is the difference between simple-audio-card, audio-graph-card, and audio-graph-card2? They all seem to do very similar things and share a lot of code. Thanks! Best regards, Aidan

1 0

[PATCH 1/9] ASoC: amd: vg: fix for pm resume callback sequence
by Vijendar Mukunda 25 Feb '22

25 Feb '22

The previous condition is used to cross check only the active stream status for I2S HS instance playback and capture use cases. Modified logic to invoke sequence for two i2s controller instances. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/vangogh/acp5x-pcm-dma.c | 62 ++++++++++++++------------- 1 file changed, 32 insertions(+), 30 deletions(-) diff --git a/sound/soc/amd/vangogh/acp5x-pcm-dma.c b/sound/soc/amd/vangogh/acp5x-pcm-dma.c index f10de38976cb..17853d8d56d3 100644 --- a/sound/soc/amd/vangogh/acp5x-pcm-dma.c +++ b/sound/soc/amd/vangogh/acp5x-pcm-dma.c @@ -426,6 +426,7 @@ static int acp5x_audio_remove(struct platform_device *pdev) static int __maybe_unused acp5x_pcm_resume(struct device *dev) { struct i2s_dev_data *adata; + struct i2s_stream_instance *rtd; u32 val, reg_val, frmt_val; reg_val = 0; @@ -433,44 +434,45 @@ static int __maybe_unused acp5x_pcm_resume(struct device *dev) adata = dev_get_drvdata(dev); if (adata->play_stream && adata->play_stream->runtime) { - struct i2s_stream_instance *rtd = - adata->play_stream->runtime->private_data; + rtd = adata->play_stream->runtime->private_data; config_acp5x_dma(rtd, SNDRV_PCM_STREAM_PLAYBACK); - switch (rtd->i2s_instance) { - case I2S_HS_INSTANCE: - reg_val = ACP_HSTDM_ITER; - frmt_val = ACP_HSTDM_TXFRMT; - break; - case I2S_SP_INSTANCE: - default: - reg_val = ACP_I2STDM_ITER; - frmt_val = ACP_I2STDM_TXFRMT; + acp_writel((rtd->xfer_resolution << 3), rtd->acp5x_base + ACP_HSTDM_ITER); + if (adata->tdm_mode == TDM_ENABLE) { + acp_writel(adata->tdm_fmt, adata->acp5x_base + ACP_HSTDM_TXFRMT); + val = acp_readl(adata->acp5x_base + ACP_HSTDM_ITER); + acp_writel(val | 0x2, adata->acp5x_base + ACP_HSTDM_ITER); + } + } + if (adata->i2ssp_play_stream && adata->i2ssp_play_stream->runtime) { + rtd = adata->i2ssp_play_stream->runtime->private_data; + config_acp5x_dma(rtd, SNDRV_PCM_STREAM_PLAYBACK); + acp_writel((rtd->xfer_resolution << 3), rtd->acp5x_base + ACP_I2STDM_ITER); + if (adata->tdm_mode == TDM_ENABLE) { + acp_writel(adata->tdm_fmt, adata->acp5x_base + ACP_I2STDM_TXFRMT); + val = acp_readl(adata->acp5x_base + ACP_I2STDM_ITER); + acp_writel(val | 0x2, adata->acp5x_base + ACP_I2STDM_ITER); } - acp_writel((rtd->xfer_resolution << 3), - rtd->acp5x_base + reg_val); } if (adata->capture_stream && adata->capture_stream->runtime) { - struct i2s_stream_instance *rtd = - adata->capture_stream->runtime->private_data; + rtd = adata->capture_stream->runtime->private_data; config_acp5x_dma(rtd, SNDRV_PCM_STREAM_CAPTURE); - switch (rtd->i2s_instance) { - case I2S_HS_INSTANCE: - reg_val = ACP_HSTDM_IRER; - frmt_val = ACP_HSTDM_RXFRMT; - break; - case I2S_SP_INSTANCE: - default: - reg_val = ACP_I2STDM_IRER; - frmt_val = ACP_I2STDM_RXFRMT; + acp_writel((rtd->xfer_resolution << 3), rtd->acp5x_base + ACP_HSTDM_IRER); + if (adata->tdm_mode == TDM_ENABLE) { + acp_writel(adata->tdm_fmt, adata->acp5x_base + ACP_HSTDM_RXFRMT); + val = acp_readl(adata->acp5x_base + ACP_HSTDM_IRER); + acp_writel(val | 0x2, adata->acp5x_base + ACP_HSTDM_IRER); } - acp_writel((rtd->xfer_resolution << 3), - rtd->acp5x_base + reg_val); } - if (adata->tdm_mode == TDM_ENABLE) { - acp_writel(adata->tdm_fmt, adata->acp5x_base + frmt_val); - val = acp_readl(adata->acp5x_base + reg_val); - acp_writel(val | 0x2, adata->acp5x_base + reg_val); + if (adata->i2ssp_capture_stream && adata->i2ssp_capture_stream->runtime) { + rtd = adata->i2ssp_capture_stream->runtime->private_data; + config_acp5x_dma(rtd, SNDRV_PCM_STREAM_CAPTURE); + acp_writel((rtd->xfer_resolution << 3), rtd->acp5x_base + ACP_I2STDM_IRER); + if (adata->tdm_mode == TDM_ENABLE) { + acp_writel(adata->tdm_fmt, adata->acp5x_base + ACP_I2STDM_RXFRMT); + val = acp_readl(adata->acp5x_base + ACP_I2STDM_IRER); + acp_writel(val | 0x2, adata->acp5x_base + ACP_I2STDM_IRER); + } } acp_writel(1, adata->acp5x_base + ACP_EXTERNAL_INTR_ENB); return 0; -- 2.17.1

4 14

[PATCH v2 00/16] ASoC: codecs: add pm runtime support for Qualcomm codecs
by Srinivas Kandagatla 25 Feb '22

25 Feb '22

This patchset adds support for runtime pm on tx/rx/wsa/wcd lpass macro, wsa881x and wcd938x codecs that are wired up on SoundWire bus. During pm testing it was also found that soundwire clks enabled by lpass macros are not enabling all the required clocks correctly, so last 3 patches corrects them. Tested this on SM8250 MTP along SoundWire In band Headset Button wakeup interrupts. Changes since v1: - move to handing clks individually from clk bluk apis - tidy up the clk provider side code by using devm variants - added error handling. Srinivas Kandagatla (16): ASoC: codecs: va-macro: move to individual clks from bulk ASoC: codecs: rx-macro: move clk provider to managed variants ASoC: codecs: tx-macro: move clk provider to managed variants ASoC: codecs: rx-macro: move to individual clks from bulk ASoC: codecs: tx-macro: move to individual clks from bulk ASoC: codecs: wsa-macro: move to individual clks from bulk ASoC: codecs: wsa-macro: setup soundwire clks correctly ASoC: codecs: tx-macro: setup soundwire clks correctly ASoC: codecs: rx-macro: setup soundwire clks correctly ASoC: codecs: va-macro: add runtime pm support ASoC: codecs: wsa-macro: add runtime pm support ASoC: codecs: rx-macro: add runtime pm support ASoC: codecs: tx-macro: add runtime pm support ASoC: codecs: wsa881x: add runtime pm support ASoC: codecs: wcd938x: add simple clk stop support ASoC: codecs: wcd-mbhc: add runtime pm support sound/soc/codecs/lpass-rx-macro.c | 168 +++++++++++++++++++++++----- sound/soc/codecs/lpass-tx-macro.c | 169 ++++++++++++++++++++++++----- sound/soc/codecs/lpass-va-macro.c | 106 ++++++++++++++---- sound/soc/codecs/lpass-wsa-macro.c | 157 +++++++++++++++++++++++---- sound/soc/codecs/wcd-mbhc-v2.c | 26 +++++ sound/soc/codecs/wcd938x-sdw.c | 1 + sound/soc/codecs/wsa881x.c | 54 +++++++++ 7 files changed, 580 insertions(+), 101 deletions(-) -- 2.21.0

3 19

[PATCH] ALSA: acp: check the return value of devm_kzalloc() in acp_legacy_dai_links_create()
by Jia-Ju Bai 25 Feb '22

25 Feb '22

The function devm_kzalloc() in acp_legacy_dai_links_create() can fail, so its return value should be checked. Fixes: d4c750f2c7d4 ("ASoC: amd: acp: Add generic machine driver support for ACP cards") Reported-by: TOTE Robot <oslab(a)tsinghua.edu.cn> Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/soc/amd/acp/acp-mach-common.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/amd/acp/acp-mach-common.c b/sound/soc/amd/acp/acp-mach-common.c index cd05ee2802c9..5247015e8b31 100644 --- a/sound/soc/amd/acp/acp-mach-common.c +++ b/sound/soc/amd/acp/acp-mach-common.c @@ -556,6 +556,8 @@ int acp_legacy_dai_links_create(struct snd_soc_card *card) num_links++; links = devm_kzalloc(dev, sizeof(struct snd_soc_dai_link) * num_links, GFP_KERNEL); + if (!links) + return -ENOMEM; if (drv_data->hs_cpu_id == I2S_SP) { links[i].name = "acp-headset-codec"; -- 2.17.1

2 1

[PATCH] ASoC: SOF: sof-priv: Drop duplicate sof_compressed_ops declaration
by Peter Ujfalusi 25 Feb '22

25 Feb '22

Other commit added the declaration of the sof_compressed_ops, drop the instance which added it as Platform specific ops, which the sof_compressed_ops is not. 76cdd90b27b4e ("ASoC: SOF: pcm: Add compress_ops for SOF platform component driver") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/sof-priv.h | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 1cc6dce17582..2e19ac619ad5 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -561,11 +561,6 @@ int snd_sof_debugfs_add_region_item_iomem(struct snd_sof_dev *sdev, enum snd_sof_fw_blk_type blk_type, u32 offset, size_t size, const char *name, enum sof_debugfs_access_type access_type); -/* - * Platform specific ops. - */ -extern struct snd_compress_ops sof_compressed_ops; - /* * DSP Architectures. */ -- 2.35.1

3 2

[PATCH] ASoC: qcom: lpass-platform: Update warning print to control excess logging
by Srinivasa Rao Mandadapu 25 Feb '22

25 Feb '22

Update dev_warn to dev_warn_ratelimit to control excess xrun logging in lpass platform driver. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Co-developed-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> Signed-off-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> --- sound/soc/qcom/lpass-platform.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index b3af971..1ce0878 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -986,7 +986,8 @@ static irqreturn_t lpass_dma_interrupt_handler( "error writing to irqclear reg: %d\n", rv); return IRQ_NONE; } - dev_warn(soc_runtime->dev, "xrun warning\n"); + dev_warn_ratelimited(soc_runtime->dev, "xrun warning\n"); + snd_pcm_stop_xrun(substream); ret = IRQ_HANDLED; } -- 2.7.4

2 1

[PATCH] ALSA: rt5663: check the return value of devm_kzalloc() in rt5663_parse_dp()
by Jia-Ju Bai 25 Feb '22

25 Feb '22

The function devm_kzalloc() in rt5663_parse_dp() can fail, so its return value should be checked. Fixes: 457c25efc592 ("ASoC: rt5663: Add the function of impedance sensing") Reported-by: TOTE Robot <oslab(a)tsinghua.edu.cn> Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/soc/codecs/rt5663.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/rt5663.c b/sound/soc/codecs/rt5663.c index 2138f62e6af5..3a8fba101b20 100644 --- a/sound/soc/codecs/rt5663.c +++ b/sound/soc/codecs/rt5663.c @@ -3478,6 +3478,8 @@ static int rt5663_parse_dp(struct rt5663_priv *rt5663, struct device *dev) table_size = sizeof(struct impedance_mapping_table) * rt5663->pdata.impedance_sensing_num; rt5663->imp_table = devm_kzalloc(dev, table_size, GFP_KERNEL); + if (!rt5663->imp_table) + return -ENOMEM; ret = device_property_read_u32_array(dev, "realtek,impedance_sensing_table", (u32 *)rt5663->imp_table, table_size); -- 2.17.1

2 1

[PATCH] sound/soc/amd: Use platform_get_irq_byname() to get the interrupt
by Meng Tang 25 Feb '22

25 Feb '22

platform_get_resource_byname(pdev, IORESOURCE_IRQ, ..) relies on static allocation of IRQ resources in DT core code, this causes an issue when using hierarchical interrupt domains using "interrupts" property in the node as this bypasses the hierarchical setup and messes up the irq chaining. In preparation for removal of static setup of IRQ resource from DT core code use platform_get_irq_byname(). Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/soc/amd/acp/acp-renoir.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/amd/acp/acp-renoir.c b/sound/soc/amd/acp/acp-renoir.c index d06ad5ce7fec..d4deac821300 100644 --- a/sound/soc/amd/acp/acp-renoir.c +++ b/sound/soc/amd/acp/acp-renoir.c @@ -265,13 +265,10 @@ static int renoir_audio_probe(struct platform_device *pdev) if (!adata->acp_base) return -ENOMEM; - res = platform_get_resource_byname(pdev, IORESOURCE_IRQ, "acp_dai_irq"); - if (!res) { - dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + adata->i2s_irq = platform_get_irq_byname(pdev, "acp_dai_irq"); + if (adata->i2s_irq < 0) return -ENODEV; - } - adata->i2s_irq = res->start; adata->dev = dev; adata->dai_driver = acp_renoir_dai; adata->num_dai = ARRAY_SIZE(acp_renoir_dai); -- 2.20.1

2 2

[PATCH v2 00/11] Fix broken usage of driver_override (and kfree of static memory)
by Krzysztof Kozlowski 25 Feb '22

25 Feb '22

Hi, This is a continuation of my old patchset from 2019. [1] Back then, few drivers set driver_override wrong. I fixed Exynos in a different way after discussions. QCOM NGD was not fixed and a new user appeared - IMX SCU. It seems "char *" in driver_override looks too consty, so we tend to make a mistake of storing there string literals. Changes of latest since v1 (not the old 2019 solution): ======================================================= https://lore.kernel.org/all/708eabb1-7b35-d525-d4c3-451d4a3de84f@rasmusvill… 1. Add helper for setting driver_override. 2. Use the helper. Dependencies (and stable): ========================== 1. All patches, including last three fixes, depend on first patch introducing the helper. 2. The last three commits - fixes - are probably not backportable directly, because of this dependency. I don't know how to express it here, since stable-kernel-rules.rst mentions only commits as possible dependencies. [1] https://lore.kernel.org/all/1550484960-2392-3-git-send-email-krzk@kernel.or… Best regards, Krzysztof Krzysztof Kozlowski (11): driver: platform: add and use helper for safer setting of driver_override amba: use helper for safer setting of driver_override fsl-mc: use helper for safer setting of driver_override hv: vmbus: use helper for safer setting of driver_override pci: use helper for safer setting of driver_override s390: cio: use helper for safer setting of driver_override spi: use helper for safer setting of driver_override vdpa: use helper for safer setting of driver_override clk: imx: scu: fix kfree() of static memory on setting driver_override slimbus: qcom-ngd: fix kfree() of static memory on setting driver_override rpmsg: fix kfree() of static memory on setting driver_override drivers/amba/bus.c | 24 +++--------------- drivers/base/driver.c | 44 +++++++++++++++++++++++++++++++++ drivers/base/platform.c | 24 +++--------------- drivers/bus/fsl-mc/fsl-mc-bus.c | 22 +++-------------- drivers/clk/imx/clk-scu.c | 7 +++++- drivers/hv/vmbus_drv.c | 24 +++--------------- drivers/pci/pci-sysfs.c | 24 +++--------------- drivers/rpmsg/rpmsg_internal.h | 13 ++++++++-- drivers/rpmsg/rpmsg_ns.c | 14 +++++++++-- drivers/s390/cio/css.c | 24 +++--------------- drivers/slimbus/qcom-ngd-ctrl.c | 12 ++++++++- drivers/spi/spi.c | 20 +++------------ drivers/vdpa/vdpa.c | 25 +++---------------- include/linux/device/driver.h | 1 + include/linux/platform_device.h | 6 ++++- include/linux/spi/spi.h | 2 +- 16 files changed, 123 insertions(+), 163 deletions(-) -- 2.32.0

4 24

[PATCH] soundwire: qcom: remove redundant wait for completion
by Srinivasa Rao Mandadapu 25 Feb '22

25 Feb '22

Remove wait_for_completion_timeout from soundwire probe as it seems unnecessary and device enumeration is anyway not happening here. Also, as device enumeration event is dependent on wcd938x probe to be completed, its of no use waiting here. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Co-developed-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> Signed-off-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> --- drivers/soundwire/qcom.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 5481341..9a32a24 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -1309,8 +1309,6 @@ static int qcom_swrm_probe(struct platform_device *pdev) } qcom_swrm_init(ctrl); - wait_for_completion_timeout(&ctrl->enumeration, - msecs_to_jiffies(TIMEOUT_MS)); ret = qcom_swrm_register_dais(ctrl); if (ret) goto err_master_add; -- 2.7.4

2 2

[RESEND v17 0/2] mtk: add ADSP mailbox controller for MT8195
by Allen-KH Cheng 25 Feb '22

25 Feb '22

Mediatek ADSP IPC is used to send notification or short message between processors with dsp. It will place the message to the share buffer and will access the ADSP mailbox registers to kick dsp. Two mailboxes used to send notification or short message between processors with dsp. This patchset was tested and confirmed on MT8195 cherry board. Based on tag: next-20220223, linux-next/master changes since v16: - fix WARNING: modpost: missing MODULE_LICENSE() in drivers/mailbox /mtk-adsp-mailbox.o. Add MODULE_LICENSE in the last line. - Due to WARNING: Missing or malformed SPDX-License-Identifier tag in line 1 in checkpatch, we don't remove SPDX-License in line 1. changes since v15: - add paragraphs to describe the config symbols fully changes since v14: - add inline for get_mtk_adsp_mbox_priv changes since v13: - rebase on v5.16-rc8 - add back ptr check from of_device_get_match_data changes since v12: - remove of_device_get_match_data ptr check: of_device_get_match_data(dev) will never going to return NULL. driver probe with compatible mediatek,mt8195-adsp-mbox. changes since v11: - remove useless MODULE_LICENSE changes since v10: - split up v9 into two separate submissions changes since v9: - rename adsp_mbox_chan_ops to mtk_adsp_mbox_chan_ops changes since v8: - remove struct adsp_mbox_ch_info - add get_mtk_adsp_mbox_priv - use mtk_adsp_mbox_priv va_mboxreg address in adsp mbox driver - add struct mtk_adsp_mbox_cfg for DSP mbox register offset - remove adsp mbox register offset hard code define - remove mtk-adsp-ipc.h reference in adsp mbox driver changes since v7: - add mtk prefix for adsp ipc functions - rename adsp_mbox_ch_info to mtk_adsp_mbox_ch_info - remove incorrect reviewers in commit message changes since v6: - dt-bindings: change example dtsi node. - rename config MTK_ADSP_IPC_MBOX to MTK_ADSP_MBOX - remove unused variable - add reviewers changes since v5: - remove some redundant code changes since v4: - use switch ... case in adsp_ipc_recv - add error handling path for chan_name pointer - refine some code to be concise changes since v3: - reorder MTK_ADSP_IPC_MBOX config - remove some redundant code - remove lock in mtk-adsp-mailbox changes since v2: - separate adsp_mailbox into two instances changes since v1: - fix dt_binding_check error Allen-KH Cheng (2): dt-bindings: mailbox: mtk,adsp-mbox: add mtk adsp-mbox document mailbox: mediatek: add support for adsp mailbox controller .../bindings/mailbox/mtk,adsp-mbox.yaml | 52 ++++++ drivers/mailbox/Kconfig | 9 + drivers/mailbox/Makefile | 2 + drivers/mailbox/mtk-adsp-mailbox.c | 176 ++++++++++++++++++ 4 files changed, 239 insertions(+) create mode 100644 Documentation/devicetree/bindings/mailbox/mtk,adsp-mbox.yaml create mode 100644 drivers/mailbox/mtk-adsp-mailbox.c -- 2.18.0

3 4

[PATCH v3 0/6] Add Tegra234 HDA support
by Mohan Kumar 25 Feb '22

25 Feb '22

This series add the support for TEGRA234 HDA driver support - Patch v3 removed conflict on top of linux-next tree Mohan Kumar (6): ALSA: hda/tegra: Add Tegra234 hda driver support ALSA: hda/tegra: Hardcode GCAP ISS value on T234 ALSA: hda/tegra: Update scratch reg. communication dt-bindings: Add HDA support for Tegra234 dt-bindings: Document Tegra234 HDA support arm64: tegra: Add hda dts node for Tegra234 .../bindings/sound/nvidia,tegra30-hda.yaml | 3 + .../nvidia/tegra234-p3737-0000+p3701-0000.dts | 6 + arch/arm64/boot/dts/nvidia/tegra234.dtsi | 18 +++ include/dt-bindings/clock/tegra234-clock.h | 4 + include/dt-bindings/memory/tegra234-mc.h | 6 + .../dt-bindings/power/tegra234-powergate.h | 9 ++ include/dt-bindings/reset/tegra234-reset.h | 2 + sound/pci/hda/hda_tegra.c | 33 ++++- sound/pci/hda/patch_hdmi.c | 118 ++++++++++++++---- 9 files changed, 175 insertions(+), 24 deletions(-) create mode 100644 include/dt-bindings/power/tegra234-powergate.h -- 2.17.1

6 14

[PATCH] sound/soc/amd: Use platform_get_irq() to get the interrupt
by Meng Tang 25 Feb '22

25 Feb '22

platform_get_resource(pdev, IORESOURCE_IRQ, ..) relies on static allocation of IRQ resources in DT core code, this causes an issue when using hierarchical interrupt domains using "interrupts" property in the node as this bypassed the hierarchical setup and messed up the irq chaining. In preparation for removal of static setup of IRQ resource from DT core code use platform_get_irq(). Signed-off-by: Meng Tang <tangmeng(a)uniontech.com> --- sound/soc/amd/acp-pcm-dma.c | 11 ++++------- sound/soc/amd/raven/acp3x-pcm-dma.c | 8 ++------ sound/soc/amd/renoir/acp3x-pdm-dma.c | 7 ++----- sound/soc/amd/vangogh/acp5x-pcm-dma.c | 7 ++----- 4 files changed, 10 insertions(+), 23 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 8fa2e2fde4f1..1cd2e70a57df 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1217,9 +1217,8 @@ static const struct snd_soc_component_driver acp_asoc_platform = { static int acp_audio_probe(struct platform_device *pdev) { - int status; + int status, irq; struct audio_drv_data *audio_drv_data; - struct resource *res; const u32 *pdata = pdev->dev.platform_data; if (!pdata) { @@ -1249,13 +1248,11 @@ static int acp_audio_probe(struct platform_device *pdev) audio_drv_data->asic_type = *pdata; - res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!res) { - dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + irq = platform_get_irq(pdev, 0); + if (irq < 0) return -ENODEV; - } - status = devm_request_irq(&pdev->dev, res->start, dma_irq_handler, + status = devm_request_irq(&pdev->dev, irq, dma_irq_handler, 0, "ACP_IRQ", &pdev->dev); if (status) { dev_err(&pdev->dev, "ACP IRQ request failed\n"); diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 75c06697fa09..e4f8dbf0d11d 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -394,13 +394,9 @@ static int acp3x_audio_probe(struct platform_device *pdev) if (!adata->acp3x_base) return -ENOMEM; - res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!res) { - dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + adata->i2s_irq = platform_get_irq(pdev, 0); + if (adata->i2s_irq < 0) return -ENODEV; - } - - adata->i2s_irq = res->start; dev_set_drvdata(&pdev->dev, adata); status = devm_snd_soc_register_component(&pdev->dev, diff --git a/sound/soc/amd/renoir/acp3x-pdm-dma.c b/sound/soc/amd/renoir/acp3x-pdm-dma.c index 9dd22a2fa2e5..88a242538461 100644 --- a/sound/soc/amd/renoir/acp3x-pdm-dma.c +++ b/sound/soc/amd/renoir/acp3x-pdm-dma.c @@ -399,13 +399,10 @@ static int acp_pdm_audio_probe(struct platform_device *pdev) if (!adata->acp_base) return -ENOMEM; - res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!res) { - dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + adata->pdm_irq = platform_get_irq(pdev, 0); + if (adata->pdm_irq < 0) return -ENODEV; - } - adata->pdm_irq = res->start; adata->capture_stream = NULL; dev_set_drvdata(&pdev->dev, adata); diff --git a/sound/soc/amd/vangogh/acp5x-pcm-dma.c b/sound/soc/amd/vangogh/acp5x-pcm-dma.c index f10de38976cb..c8cd1777a63c 100644 --- a/sound/soc/amd/vangogh/acp5x-pcm-dma.c +++ b/sound/soc/amd/vangogh/acp5x-pcm-dma.c @@ -388,13 +388,10 @@ static int acp5x_audio_probe(struct platform_device *pdev) if (!adata->acp5x_base) return -ENOMEM; - res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!res) { - dev_err(&pdev->dev, "IORESOURCE_IRQ FAILED\n"); + adata->i2s_irq = platform_get_irq(pdev, 0); + if (adata->i2s_irq < 0) return -ENODEV; - } - adata->i2s_irq = res->start; dev_set_drvdata(&pdev->dev, adata); status = devm_snd_soc_register_component(&pdev->dev, &acp5x_i2s_component, -- 2.20.1

1 0

[PATCH v3] ASoC: codecs: Add power domains support in digital macro codecs
by Srinivasa Rao Mandadapu 25 Feb '22

25 Feb '22

Add support for enabling required power domains in digital macro codecs. macro and dcodec power domains are being requested as clocks by HLOS in ADSP based architectures and ADSP internally handling as powerdomains. In ADSP bypass case need to handle them as power domains explicitly. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Co-developed-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> Signed-off-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> Reported-by: kernel test robot <lkp(a)intel.com> --- Changes since v2: -- Remove redundant local variable. -- Update pds error handling sequence. -- Update module description. -- Clean up pds init function. -- Remove redundant arguments. Changes since v1: -- Add missing macros in Kconfig. sound/soc/codecs/Kconfig | 7 ++++ sound/soc/codecs/Makefile | 2 ++ sound/soc/codecs/lpass-macro-common.c | 67 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/lpass-macro-common.h | 17 +++++++++ sound/soc/codecs/lpass-rx-macro.c | 13 ++++++- sound/soc/codecs/lpass-tx-macro.c | 10 ++++++ sound/soc/codecs/lpass-va-macro.c | 11 +++++- sound/soc/qcom/Kconfig | 1 + 8 files changed, 126 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/lpass-macro-common.c create mode 100644 sound/soc/codecs/lpass-macro-common.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c2627f7..4de029a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -244,6 +244,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_WCD9335 imply SND_SOC_WCD934X imply SND_SOC_WCD938X_SDW + imply SND_SOC_LPASS_MACRO_COMMON imply SND_SOC_LPASS_RX_MACRO imply SND_SOC_LPASS_TX_MACRO imply SND_SOC_WL1273 @@ -2008,6 +2009,9 @@ config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C +config SND_SOC_LPASS_MACRO_COMMON + tristate + config SND_SOC_LPASS_WSA_MACRO depends on COMMON_CLK select REGMAP_MMIO @@ -2016,16 +2020,19 @@ config SND_SOC_LPASS_WSA_MACRO config SND_SOC_LPASS_VA_MACRO depends on COMMON_CLK select REGMAP_MMIO + select SND_SOC_LPASS_MACRO_COMMON tristate "Qualcomm VA Macro in LPASS(Low Power Audio SubSystem)" config SND_SOC_LPASS_RX_MACRO depends on COMMON_CLK select REGMAP_MMIO + select SND_SOC_LPASS_MACRO_COMMON tristate "Qualcomm RX Macro in LPASS(Low Power Audio SubSystem)" config SND_SOC_LPASS_TX_MACRO depends on COMMON_CLK select REGMAP_MMIO + select SND_SOC_LPASS_MACRO_COMMON tristate "Qualcomm TX Macro in LPASS(Low Power Audio SubSystem)" endmenu diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b4e11c3..c3c6059 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -112,6 +112,7 @@ snd-soc-l3-objs := l3.o snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o snd-soc-lochnagar-sc-objs := lochnagar-sc.o +snd-soc-lpass-macro-common-objs := lpass-macro-common.o snd-soc-lpass-rx-macro-objs := lpass-rx-macro.o snd-soc-lpass-tx-macro-objs := lpass-tx-macro.o snd-soc-lpass-wsa-macro-objs := lpass-wsa-macro.o @@ -676,6 +677,7 @@ obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_MAX98504) += snd-soc-max98504.o obj-$(CONFIG_SND_SOC_SIMPLE_AMPLIFIER) += snd-soc-simple-amplifier.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o +obj-$(CONFIG_SND_SOC_LPASS_MACRO_COMMON) += snd-soc-lpass-macro-common.o obj-$(CONFIG_SND_SOC_LPASS_WSA_MACRO) += snd-soc-lpass-wsa-macro.o obj-$(CONFIG_SND_SOC_LPASS_VA_MACRO) += snd-soc-lpass-va-macro.o obj-$(CONFIG_SND_SOC_LPASS_RX_MACRO) += snd-soc-lpass-rx-macro.o diff --git a/sound/soc/codecs/lpass-macro-common.c b/sound/soc/codecs/lpass-macro-common.c new file mode 100644 index 0000000..6cede75 --- /dev/null +++ b/sound/soc/codecs/lpass-macro-common.c @@ -0,0 +1,67 @@ +// SPDX-License-Identifier: GPL-2.0-only +// Copyright (c) 2022, The Linux Foundation. All rights reserved. + +#include <linux/export.h> +#include <linux/module.h> +#include <linux/init.h> +#include <linux/of_platform.h> +#include <linux/platform_device.h> +#include <linux/pm_domain.h> +#include <linux/pm_runtime.h> + +#include "lpass-macro-common.h" + +struct lpass_macro *lpass_macro_pds_init(struct device *dev) +{ + struct lpass_macro *l_pds; + int ret; + + if (!of_find_property(dev->of_node, "power-domains", NULL)) + return NULL; + + l_pds = devm_kzalloc(dev, sizeof(*l_pds), GFP_KERNEL); + if (!l_pds) + return ERR_PTR(-ENOMEM); + + l_pds->macro_pd = dev_pm_domain_attach_by_name(dev, "macro"); + if (IS_ERR_OR_NULL(l_pds->macro_pd)) + return NULL; + + ret = pm_runtime_get_sync(l_pds->macro_pd); + if (ret < 0) { + pm_runtime_put_noidle(l_pds->macro_pd); + goto macro_err; + } + + l_pds->dcodec_pd = dev_pm_domain_attach_by_name(dev, "dcodec"); + if (IS_ERR_OR_NULL(l_pds->dcodec_pd)) + goto dcodec_err; + + ret = pm_runtime_get_sync(l_pds->dcodec_pd); + if (ret < 0) { + pm_runtime_put_noidle(l_pds->dcodec_pd); + goto dcodec_sync_err; + } + return l_pds; + +dcodec_sync_err: + dev_pm_domain_detach(l_pds->dcodec_pd, false); +dcodec_err: + pm_runtime_put(l_pds->macro_pd); +macro_err: + dev_pm_domain_detach(l_pds->macro_pd, false); + return ERR_PTR(ret); +} +EXPORT_SYMBOL_GPL(lpass_macro_pds_init); + +void lpass_macro_pds_exit(struct lpass_macro *pds) +{ + pm_runtime_put(pds->macro_pd); + dev_pm_domain_detach(pds->macro_pd, false); + pm_runtime_put(pds->dcodec_pd); + dev_pm_domain_detach(pds->dcodec_pd, false); +} +EXPORT_SYMBOL_GPL(lpass_macro_pds_exit); + +MODULE_DESCRIPTION("Common macro driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/lpass-macro-common.h b/sound/soc/codecs/lpass-macro-common.h new file mode 100644 index 0000000..f2cbf9f --- /dev/null +++ b/sound/soc/codecs/lpass-macro-common.h @@ -0,0 +1,17 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Copyright (c) 2022, The Linux Foundation. All rights reserved. + */ + +#ifndef __LPASS_MACRO_COMMON_H__ +#define __LPASS_MACRO_COMMON_H__ + +struct lpass_macro { + struct device *macro_pd; + struct device *dcodec_pd; +}; + +struct lpass_macro *lpass_macro_pds_init(struct device *dev); +void lpass_macro_pds_exit(struct lpass_macro *pds); + +#endif /* __LPASS_MACRO_COMMON_H__ */ diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 29d214f..4ba5876 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -14,6 +14,8 @@ #include <linux/of_clk.h> #include <linux/clk-provider.h> +#include "lpass-macro-common.h" + #define CDC_RX_TOP_TOP_CFG0 (0x0000) #define CDC_RX_TOP_SWR_CTRL (0x0008) #define CDC_RX_TOP_DEBUG (0x000C) @@ -606,7 +608,7 @@ struct rx_macro { int is_softclip_on; int is_aux_hpf_on; int softclip_clk_users; - + struct lpass_macro *pds; struct regmap *regmap; struct clk_bulk_data clks[RX_NUM_CLKS_MAX]; struct clk_hw hw; @@ -3537,6 +3539,12 @@ static int rx_macro_probe(struct platform_device *pdev) return ret; } + rx->pds = lpass_macro_pds_init(dev); + if (!rx->pds) { + dev_err(dev, "Enabling power domains failed in %s\n", __func__); + return ret; + } + base = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(base)) return PTR_ERR(base); @@ -3575,6 +3583,9 @@ static int rx_macro_remove(struct platform_device *pdev) of_clk_del_provider(pdev->dev.of_node); clk_bulk_disable_unprepare(RX_NUM_CLKS_MAX, rx->clks); + + lpass_macro_pds_exit(rx->pds); + return 0; } diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index 9c96ab1..7938bf3 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -13,6 +13,8 @@ #include <linux/of_clk.h> #include <linux/clk-provider.h> +#include "lpass-macro-common.h" + #define CDC_TX_CLK_RST_CTRL_MCLK_CONTROL (0x0000) #define CDC_TX_MCLK_EN_MASK BIT(0) #define CDC_TX_MCLK_ENABLE BIT(0) @@ -266,6 +268,7 @@ struct tx_macro { u16 dmic_clk_div; bool bcs_enable; int dec_mode[NUM_DECIMATORS]; + struct lpass_macro *pds; bool bcs_clk_en; }; #define to_tx_macro(_hw) container_of(_hw, struct tx_macro, hw) @@ -1802,6 +1805,11 @@ static int tx_macro_probe(struct platform_device *pdev) return ret; } + tx->pds = lpass_macro_pds_init(dev); + if (!tx->pds) { + dev_err(dev, "Enabling power domains failed in %s\n", __func__); + return ret; + } base = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(base)) return PTR_ERR(base); @@ -1859,6 +1867,8 @@ static int tx_macro_remove(struct platform_device *pdev) clk_bulk_disable_unprepare(TX_NUM_CLKS_MAX, tx->clks); + lpass_macro_pds_exit(tx->pds); + return 0; } diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 11147e3..748ef44 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -15,6 +15,8 @@ #include <sound/soc-dapm.h> #include <sound/tlv.h> +#include "lpass-macro-common.h" + /* VA macro registers */ #define CDC_VA_CLK_RST_CTRL_MCLK_CONTROL (0x0000) #define CDC_VA_MCLK_CONTROL_EN BIT(0) @@ -195,6 +197,7 @@ struct va_macro { struct regmap *regmap; struct clk_bulk_data clks[VA_NUM_CLKS_MAX]; struct clk_hw hw; + struct lpass_macro *pds; s32 dmic_0_1_clk_cnt; s32 dmic_2_3_clk_cnt; @@ -1413,7 +1416,11 @@ static int va_macro_probe(struct platform_device *pdev) dev_err(dev, "Error getting VA Clocks (%d)\n", ret); return ret; } - + va->pds = lpass_macro_pds_init(dev); + if (!va->pds) { + dev_err(dev, "Enabling power domains failed %s\n", __func__); + return ret; + } ret = of_property_read_u32(dev->of_node, "qcom,dmic-sample-rate", &sample_rate); if (ret) { @@ -1468,6 +1475,8 @@ static int va_macro_remove(struct platform_device *pdev) clk_bulk_disable_unprepare(VA_NUM_CLKS_MAX, va->clks); + lpass_macro_pds_exit(va->pds); + return 0; } diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 52db003..6ffd51a 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -194,6 +194,7 @@ config SND_SOC_SC7280 select SND_SOC_LPASS_SC7280 select SND_SOC_MAX98357A select SND_SOC_WCD938X + select SND_SOC_LPASS_MACRO_COMMON select SND_SOC_LPASS_RX_MACRO select SND_SOC_LPASS_TX_MACRO help -- 2.7.4

2 2

[PATCH] ASoC: SOF: Declare sof_compress_ops in sof-priv.h
by Peter Ujfalusi 25 Feb '22

25 Feb '22

Fix the following sparse error: sound/soc/sof/compress.c:310:25: error: symbol 'sof_compressed_ops' was not declared. Should it be static? Do not enable set the pd->compress_ops yet as it is not a valid assumption that real compress support really works when CONFIG_SND_SOC_SOF_COMPRESS is set as the HDA Probes support also selects it, but compressed audio is not supported (yet) on Intel platforms. Fixes: 6324cf901e14c ("ASoC: SOF: compr: Add compress ops implementation") Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> --- sound/soc/sof/sof-priv.h | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 2c8e556cd5cc..2e19ac619ad5 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -497,6 +497,11 @@ void snd_sof_complete(struct device *dev); void snd_sof_new_platform_drv(struct snd_sof_dev *sdev); +/* + * Compress support + */ +extern struct snd_compress_ops sof_compressed_ops; + /* * Firmware loading. */ -- 2.35.1

2 1

[PATCH v15 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 25 Feb '22

25 Feb '22

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. Changes Since V14: -- Split common wrapper function to separate wrapper for each handle in platform driver. -- Update cdc dma buffer handling with memremap with ioremap. -- Remove redundant error prints. -- Update irq flag. Changes Since V13: -- Change bulk cdc clock voting to individual clock voting. -- Remove redundant code, conditional check and prints. -- Fix typo errors. Changes Since V12: -- Fix arguments type mismatch. Changes Since V11: -- Fix kernel robot issue on arguments type mismatch. Changes Since V10: -- Split bulk clock voting to individual clock voting as per use case in cdc-dma driver. -- Add missing codec dma clocks. -- Update rxtx lpm buffer size. Changes Since V9: -- Change individual clock voting to bulk clock voting of lpass-sc7280 platform driver. -- Remove redundant clocks in lpass variant structure. -- Add mclk for MI2S based headset path. -- Remove unused lpass variant structure members in lpass header. Changes Since V8: -- Fix errors in sc7280 lpass cpu dt-bindings. -- Move to quicinc domain email id's. Changes Since V7: -- Fix indentation errors. -- Bisect patches to avoid interdependency. Changes Since V6: -- Split cdc dma regmap config macros. -- Add write dma reg fields for i2s path. -- Add helper function to distinguish rxtx and va dma ports. -- Optimizing clock and reg name in cpu dt-bindings. -- Update buffer management for cdc dma path. -- Remove Kconfig fields of machine driver. Changes Since V5: -- Include MI2S primary node to snd_soc_dai_driver in lpass-sc7280 platform driver. -- Move dependency patch list to corresponding patch. -- Add support for missing cdc-dma ports. -- Change if/else conditional statements to switch cases. -- Add missing error handlings. -- Typo errors fix. Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: SC7280: Update config for building codec dma drivers ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add regmap config support for codec dma driver ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: dt-bindings: Add SC7280 lpass cpu bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 75 ++- sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 301 ++++++++++ sound/soc/qcom/lpass-cpu.c | 252 ++++++++- sound/soc/qcom/lpass-lpaif-reg.h | 127 ++++- sound/soc/qcom/lpass-platform.c | 628 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 447 +++++++++++++++ sound/soc/qcom/lpass.h | 143 +++++ 9 files changed, 1900 insertions(+), 88 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- 2.7.4

2 15

[PATCH v10 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 25 Feb '22

25 Feb '22

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. Changes Since V9: -- Change individual clock voting to bulk clock voting of lpass-sc7280 platform driver. -- Remove redundant clocks in lpass variant structure. -- Add mclk for MI2S based headset path. -- Remove unused lpass variant structure members in lpass header. Changes Since V8: -- Fix errors in sc7280 lpass cpu dt-bindings. -- Move to quicinc domain email id's. Changes Since V7: -- Fix indentation errors. -- Bisect patches to avoid interdependency. Changes Since V6: -- Split cdc dma regmap config macros. -- Add write dma reg fields for i2s path. -- Add helper function to distinguish rxtx and va dma ports. -- Optimizing clock and reg name in cpu dt-bindings. -- Update buffer management for cdc dma path. -- Remove Kconfig fields of machine driver. Changes Since V5: -- Include MI2S primary node to snd_soc_dai_driver in lpass-sc7280 platform driver. -- Move dependency patch list to corresponding patch. -- Add support for missing cdc-dma ports. -- Change if/else conditional statements to switch cases. -- Add missing error handlings. -- Typo errors fix. Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: SC7280: Update config for building codec dma drivers ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add regmap config support for codec dma driver ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: dt-bindings: Add SC7280 lpass cpu bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 75 ++- sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 275 +++++++++ sound/soc/qcom/lpass-cpu.c | 244 +++++++- sound/soc/qcom/lpass-lpaif-reg.h | 127 ++++- sound/soc/qcom/lpass-platform.c | 617 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 445 +++++++++++++++ sound/soc/qcom/lpass.h | 145 +++++ 9 files changed, 1856 insertions(+), 87 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- 2.7.4

2 12

[RESEND, v9 00/10] Add support for audio on SC7280 based targets
by Srinivasa Rao Mandadapu 25 Feb '22

25 Feb '22

This patch set is to add support for Audio over wcd codec, digital mics, through digital codecs and without ADSP. Changes Since V8: -- Fix errors in sc7280 lpass cpu dt-bindings. -- Move to quicinc domain email id's. Changes Since V7: -- Fix indentation errors. -- Bisect patches to avoid interdependency. Changes Since V6: -- Split cdc dma regmap config macros. -- Add write dma reg fields for i2s path. -- Add helper function to distinguish rxtx and va dma ports. -- Optimizing clock and reg name in cpu dt-bindings. -- Update buffer management for cdc dma path. -- Remove Kconfig fields of machine driver. Changes Since V5: -- Include MI2S primary node to snd_soc_dai_driver in lpass-sc7280 platform driver. -- Move dependency patch list to corresponding patch. -- Add support for missing cdc-dma ports. -- Change if/else conditional statements to switch cases. -- Add missing error handlings. -- Typo errors fix. Changes Since V4: -- Remove unused variable in lpass-sc7280 platform driver. Changes Since V3: -- Remove redundant power domain controls. As power domains can be configured from dtsi. Changes Since V2: -- Split lpass sc7280 cpu driver patch and create regmap config patch. -- Create patches based on latest kernel tip. -- Add helper function to get dma control and lpaif handle. -- Remove unused variables. Changes Since V1: -- Typo errors fix -- CPU driver readable/writable apis optimization. -- Add Missing config patch -- Add Common api for repeated dmactl initialization. Srinivasa Rao Mandadapu (10): ASoC: qcom: SC7280: Update config for building codec dma drivers ASoC: qcom: Move lpass_pcm_data structure to lpass header ASoC: qcom: lpass: Add dma fields for codec dma lpass interface ASoC: qcom: Add helper function to get dma control and lpaif handle ASoC: qcom: Add register definition for codec rddma and wrdma ASoC: qcom: Add regmap config support for codec dma driver ASoC: qcom: Add support for codec dma driver ASoC: qcom: Add lpass CPU driver for codec dma control ASoC: dt-bindings: Add SC7280 lpass cpu bindings ASoC: qcom: lpass-sc7280: Add platform driver for lpass audio .../devicetree/bindings/sound/qcom,lpass-cpu.yaml | 75 ++- sound/soc/qcom/Kconfig | 11 + sound/soc/qcom/Makefile | 4 + sound/soc/qcom/lpass-cdc-dma.c | 275 +++++++++ sound/soc/qcom/lpass-cpu.c | 244 +++++++- sound/soc/qcom/lpass-lpaif-reg.h | 127 ++++- sound/soc/qcom/lpass-platform.c | 617 ++++++++++++++++++--- sound/soc/qcom/lpass-sc7280.c | 441 +++++++++++++++ sound/soc/qcom/lpass.h | 162 ++++++ 9 files changed, 1869 insertions(+), 87 deletions(-) create mode 100644 sound/soc/qcom/lpass-cdc-dma.c create mode 100644 sound/soc/qcom/lpass-sc7280.c -- 2.7.4

3 13

[PATCH] ASoC: SOF: Intel: Fix NULL ptr dereference when ENOMEM
by Ammar Faizi 25 Feb '22

25 Feb '22

From: Ammar Faizi <ammarfaizi2(a)gnuweeb.org> Do not call snd_dma_free_pages() when snd_dma_alloc_pages() returns -ENOMEM because it leads to a NULL pointer dereference bug. The dmesg says: <6>[109482.497835][T138537] usb 1-2: Manufacturer: SIGMACHIP <6>[109482.502506][T138537] input: SIGMACHIP USB Keyboard as /devices/pci0000:00/0000:00:14.0/usb1/1-2/1-2:1.0/0003:1C4F:0002.000D/input/input34 <6>[109482.558976][T138537] hid-generic 0003:1C4F:0002.000D: input,hidraw1: USB HID v1.10 Keyboard [SIGMACHIP USB Keyboard] on usb-0000:00:14.0-2/input0 <6>[109482.561653][T138537] input: SIGMACHIP USB Keyboard Consumer Control as /devices/pci0000:00/0000:00:14.0/usb1/1-2/1-2:1.1/0003:1C4F:0002.000E/input/input35 <6>[109482.615490][T138537] input: SIGMACHIP USB Keyboard System Control as /devices/pci0000:00/0000:00:14.0/usb1/1-2/1-2:1.1/0003:1C4F:0002.000E/input/input36 <6>[109482.615643][T138537] hid-generic 0003:1C4F:0002.000E: input,hidraw2: USB HID v1.10 Device [SIGMACHIP USB Keyboard] on usb-0000:00:14.0-2/input1 <4>[110102.335460][T140985] rtw_8822ce 0000:01:00.0: timed out to flush queue 1 <3>[118575.730928][ T1387] sof-audio-pci-intel-tgl 0000:00:1f.3: error: memory alloc failed: -12 <1>[118575.730948][ T1387] BUG: kernel NULL pointer dereference, address: 0000000000000000 <1>[118575.730951][ T1387] #PF: supervisor read access in kernel mode <1>[118575.730953][ T1387] #PF: error_code(0x0000) - not-present page <6>[118575.730956][ T1387] PGD 0 P4D 0 <4>[118575.730960][ T1387] Oops: 0000 [#1] PREEMPT SMP NOPTI <4>[118575.730965][ T1387] CPU: 6 PID: 1387 Comm: alsa-sink-HDA A Tainted: G W 5.17.0-rc4-superb-owl-00055-g80d47f5de5e3 #3 56590caeed02394520e20ca5a2059907eb2d5079 <4>[118575.730969][ T1387] Hardware name: HP HP Laptop 14s-dq2xxx/87FD, BIOS F.15 09/15/2021 <4>[118575.730972][ T1387] RIP: 0010:dma_free_noncontiguous+0x37/0x80 <4>[118575.730979][ T1387] Code: 8b 87 80 03 00 00 48 85 c0 75 07 48 8b 05 f9 39 2b 02 48 85 c0 74 13 4c 8b 48 28 4d 85 c9 74 0a 48 89 da 44 89 c1 5b 41 ff e1 <48> 8b 0b 48 8b 11 48 8b 49 10 48 83 e2 fc 48 81 c6 ff 0f 00 00 48 <4>[118575.730982][ T1387] RSP: 0000:ffffc90002b87770 EFLAGS: 00010246 <4>[118575.730986][ T1387] RAX: 0000000000000000 RBX: 0000000000000000 RCX: 0000000000000000 <4>[118575.730987][ T1387] RDX: 0000000000000000 RSI: 0000000000000000 RDI: ffff888101db30d0 <4>[118575.730989][ T1387] RBP: 00000000fffffff4 R08: 0000000000000000 R09: 0000000000000000 <4>[118575.730991][ T1387] R10: 0000000000000000 R11: ffffc90002b874d0 R12: 0000000000000001 <4>[118575.730993][ T1387] R13: 0000000000058000 R14: ffff888105260c68 R15: ffff888105260828 <4>[118575.730995][ T1387] FS: 00007f42e2ffd640(0000) GS:ffff888466b80000(0000) knlGS:0000000000000000 <4>[118575.730998][ T1387] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 <4>[118575.731000][ T1387] CR2: 0000000000000000 CR3: 000000014acf0003 CR4: 0000000000770ee0 <4>[118575.731003][ T1387] PKRU: 55555554 <4>[118575.731005][ T1387] Call Trace: <4>[118575.731009][ T1387] <TASK> <4>[118575.731016][ T1387] cl_stream_prepare+0x10a/0x120 [snd_sof_intel_hda_common 146addf995b9279ae7f509621078cccbe4f875e1] <4>[118575.731027][ T1387] hda_dsp_cl_boot_firmware_iccmax+0x50/0xf0 [snd_sof_intel_hda_common 146addf995b9279ae7f509621078cccbe4f875e1] <4>[118575.731034][ T1387] snd_sof_run_firmware+0xb5/0x300 [snd_sof b83a5eaf64712c6f474ef3641062bb71902deae4] <4>[118575.731045][ T1387] sof_resume+0xb1/0x2c0 [snd_sof b83a5eaf64712c6f474ef3641062bb71902deae4] <4>[118575.731054][ T1387] ? pci_pme_active+0x1b4/0x220 <4>[118575.731060][ T1387] pci_pm_runtime_resume+0xaa/0xe0 <4>[118575.731064][ T1387] ? pci_pm_runtime_suspend+0x190/0x190 <4>[118575.731067][ T1387] __rpm_callback+0x9c/0x320 <4>[118575.731072][ T1387] ? pci_pm_runtime_suspend+0x190/0x190 <4>[118575.731075][ T1387] ? pci_pm_runtime_suspend+0x190/0x190 <4>[118575.731077][ T1387] rpm_resume+0x50d/0x690 <4>[118575.731082][ T1387] __pm_runtime_resume+0x69/0x80 <4>[118575.731085][ T1387] snd_soc_pcm_component_pm_runtime_get+0x60/0xe0 [snd_soc_core 6817f963f4c978c6ec831f778545ab4db1a1023f] <4>[118575.731102][ T1387] __soc_pcm_open+0x82/0x530 [snd_soc_core 6817f963f4c978c6ec831f778545ab4db1a1023f] <4>[118575.731114][ T1387] dpcm_be_dai_startup+0x14b/0x260 [snd_soc_core 6817f963f4c978c6ec831f778545ab4db1a1023f] <4>[118575.731128][ T1387] dpcm_fe_dai_startup+0x73/0x930 [snd_soc_core 6817f963f4c978c6ec831f778545ab4db1a1023f] <4>[118575.731139][ T1387] ? dpcm_add_paths+0x109/0x1b0 [snd_soc_core 6817f963f4c978c6ec831f778545ab4db1a1023f] <4>[118575.731152][ T1387] dpcm_fe_dai_open+0x74/0x160 [snd_soc_core 6817f963f4c978c6ec831f778545ab4db1a1023f] <4>[118575.731166][ T1387] snd_pcm_open_substream+0x56f/0x840 [snd_pcm 2213f8e36532d8bc92ec1ec574108b0385b1b13a] <4>[118575.731177][ T1387] snd_pcm_open+0xb3/0x1d0 [snd_pcm 2213f8e36532d8bc92ec1ec574108b0385b1b13a] <4>[118575.731185][ T1387] ? sched_dynamic_update+0x1a0/0x1a0 <4>[118575.731189][ T1387] ? cd_forget+0x80/0x80 <4>[118575.731194][ T1387] snd_pcm_playback_open+0x3c/0x60 [snd_pcm 2213f8e36532d8bc92ec1ec574108b0385b1b13a] <4>[118575.731201][ T1387] chrdev_open+0x1d3/0x200 <4>[118575.731205][ T1387] ? cd_forget+0x80/0x80 <4>[118575.731208][ T1387] do_dentry_open+0x254/0x370 <4>[118575.731213][ T1387] path_openat+0x9e6/0xbc0 <4>[118575.731218][ T1387] ? lock_release+0x230/0x300 <4>[118575.731223][ T1387] ? snd_ctl_ioctl+0x759/0x900 [snd 1eb0a4959d3d3710c16836cd2838d885bf8f75a9] <4>[118575.731232][ T1387] do_filp_open+0x93/0x120 <4>[118575.731237][ T1387] ? alloc_fd+0x147/0x180 <4>[118575.731242][ T1387] ? _raw_spin_unlock+0x29/0x40 <4>[118575.731246][ T1387] ? alloc_fd+0x147/0x180 <4>[118575.731251][ T1387] do_sys_openat2+0x68/0x130 <4>[118575.731255][ T1387] __x64_sys_openat+0x6f/0x80 <4>[118575.731259][ T1387] do_syscall_64+0x3d/0xb0 <4>[118575.731264][ T1387] ? exit_to_user_mode_prepare+0x2c/0x50 <4>[118575.731267][ T1387] entry_SYSCALL_64_after_hwframe+0x44/0xae <4>[118575.731271][ T1387] RIP: 0033:0x7f42ee19a6e4 <4>[118575.731274][ T1387] Code: 24 20 eb 8f 66 90 44 89 54 24 0c e8 16 d2 f7 ff 44 8b 54 24 0c 44 89 e2 48 89 ee 41 89 c0 bf 9c ff ff ff b8 01 01 00 00 0f 05 <48> 3d 00 f0 ff ff 77 34 44 89 c7 89 44 24 0c e8 58 d2 f7 ff 8b 44 <4>[118575.731277][ T1387] RSP: 002b:00007f42e2ffc620 EFLAGS: 00000293 ORIG_RAX: 0000000000000101 <4>[118575.731281][ T1387] RAX: ffffffffffffffda RBX: 0000000000080802 RCX: 00007f42ee19a6e4 <4>[118575.731283][ T1387] RDX: 0000000000080802 RSI: 00007f42e2ffc6c0 RDI: 00000000ffffff9c <4>[118575.731284][ T1387] RBP: 00007f42e2ffc6c0 R08: 0000000000000000 R09: 00007f42e2ffc437 <4>[118575.731286][ T1387] R10: 0000000000000000 R11: 0000000000000293 R12: 0000000000080802 <4>[118575.731289][ T1387] R13: 00007f42e8c3faa0 R14: 00007f42e2ffc6c0 R15: 0000000081204101 <4>[118575.731293][ T1387] </TASK> <4>[118575.731294][ T1387] Modules linked in: ccm xt_CHECKSUM xt_MASQUERADE xt_conntrack ipt_REJECT nf_reject_ipv4 xt_tcpudp nft_compat rfcomm nft_chain_nat nf_nat nf_conntrack nf_defrag_ipv6 nf_defrag_ipv4 nf_tables nfnetlink bridge stp llc snd_soc_skl_hda_dsp snd_soc_hdac_hdmi snd_soc_intel_hda_dsp_common snd_hda_codec_hdmi snd_hda_codec_realtek snd_hda_codec_generic cmac algif_hash algif_skcipher af_alg bnep zram snd_soc_dmic snd_sof_pci_intel_tgl snd_sof_intel_hda_common i915 nls_iso8859_1 snd_soc_hdac_hda snd_sof_intel_hda soundwire_intel soundwire_generic_allocation soundwire_cadence snd_sof_pci snd_sof_xtensa_dsp rtw88_8822ce snd_sof rtw88_8822c rtw88_pci snd_hda_ext_core rtw88_core snd_soc_acpi_intel_match intel_tcc_cooling snd_soc_acpi x86_pkg_temp_thermal soundwire_bus intel_powerclamp coretemp ledtrig_audio snd_soc_core mac80211 snd_compress ac97_bus snd_pcm_dmaengine snd_hda_intel snd_intel_dspcfg snd_intel_sdw_acpi snd_hda_codec ttm snd_hda_core kvm_intel snd_hwdep btusb <4>[118575.731347][ T1387] drm_kms_helper snd_pcm uvcvideo btrtl btmtk btintel snd_seq_midi btbcm snd_seq_midi_event videobuf2_v4l2 snd_rawmidi mei_hdcp intel_rapl_msr videobuf2_vmalloc bluetooth cfg80211 kvm snd_seq cec videobuf2_memops crct10dif_pclmul videobuf2_common rc_core ghash_clmulni_intel aesni_intel snd_seq_device processor_thermal_device_pci_legacy videodev i2c_algo_bit snd_timer processor_thermal_device fb_sys_fops processor_thermal_rfim crypto_simd cryptd processor_thermal_mbox snd sysimgblt mei_me processor_thermal_rapl hp_wmi ecdh_generic syscopyarea mc intel_rapl_common platform_profile input_leds serio_raw sparse_keymap mei ecc sysfillrect wmi_bmof joydev hid_multitouch libarc4 soundcore efi_pstore ee1004 intel_soc_dts_iosf int3400_thermal int3403_thermal mac_hid acpi_thermal_rel int340x_thermal_zone acpi_pad dptf_pch_fivr sch_fq_codel msr drm parport_pc ppdev lp parport ip_tables x_tables autofs4 btrfs xor raid6_pq libcrc32c usbhid nvme nvme_core hid_generic <4>[118575.731410][ T1387] i2c_i801 intel_lpss_pci crc32_pclmul xhci_pci intel_lpss i2c_smbus xhci_pci_renesas idma64 i2c_hid_acpi vmd i2c_hid wmi hid video pinctrl_tigerlake <4>[118575.731425][ T1387] CR2: 0000000000000000 <4>[118575.731428][ T1387] ---[ end trace 0000000000000000 ]--- <4>[118575.867259][ T1387] RIP: 0010:dma_free_noncontiguous+0x37/0x80 <4>[118575.867273][ T1387] Code: 8b 87 80 03 00 00 48 85 c0 75 07 48 8b 05 f9 39 2b 02 48 85 c0 74 13 4c 8b 48 28 4d 85 c9 74 0a 48 89 da 44 89 c1 5b 41 ff e1 <48> 8b 0b 48 8b 11 48 8b 49 10 48 83 e2 fc 48 81 c6 ff 0f 00 00 48 <4>[118575.867277][ T1387] RSP: 0000:ffffc90002b87770 EFLAGS: 00010246 <4>[118575.867282][ T1387] RAX: 0000000000000000 RBX: 0000000000000000 RCX: 0000000000000000 <4>[118575.867284][ T1387] RDX: 0000000000000000 RSI: 0000000000000000 RDI: ffff888101db30d0 <4>[118575.867286][ T1387] RBP: 00000000fffffff4 R08: 0000000000000000 R09: 0000000000000000 <4>[118575.867288][ T1387] R10: 0000000000000000 R11: ffffc90002b874d0 R12: 0000000000000001 <4>[118575.867290][ T1387] R13: 0000000000058000 R14: ffff888105260c68 R15: ffff888105260828 <4>[118575.867292][ T1387] FS: 00007f42e2ffd640(0000) GS:ffff888466b80000(0000) knlGS:0000000000000000 <4>[118575.867295][ T1387] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 <4>[118575.867297][ T1387] CR2: 0000000000000000 CR3: 000000014acf0003 CR4: 0000000000770ee0 <4>[118575.867300][ T1387] PKRU: 55555554 Cc: Daniel Baluta <daniel.baluta(a)nxp.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> Cc: Keyon Jie <yang.jie(a)linux.intel.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Rander Wang <rander.wang(a)intel.com> Cc: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Cc: Takashi Iwai <tiwai(a)suse.com> Fixes: d16046ffa6de040bf580a64d5f4d0aa18258a854 ("ASoC: SOF: Intel: Add Intel specific HDA firmware loader") Cc: stable(a)vger.kernel.org # v5.2+ Cc: sound-open-firmware(a)alsa-project.org Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Ammar Faizi <ammarfaizi2(a)gnuweeb.org> --- sound/soc/sof/intel/hda-loader.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 33306d2023a7..9bbfdab8009d 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -47,7 +47,7 @@ static struct hdac_ext_stream *cl_stream_prepare(struct snd_sof_dev *sdev, unsig ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV_SG, &pci->dev, size, dmab); if (ret < 0) { dev_err(sdev->dev, "error: memory alloc failed: %d\n", ret); - goto error; + goto out_put; } hstream->period_bytes = 0;/* initialize period_bytes */ @@ -58,22 +58,23 @@ static struct hdac_ext_stream *cl_stream_prepare(struct snd_sof_dev *sdev, unsig ret = hda_dsp_iccmax_stream_hw_params(sdev, dsp_stream, dmab, NULL); if (ret < 0) { dev_err(sdev->dev, "error: iccmax stream prepare failed: %d\n", ret); - goto error; + goto out_free; } } else { ret = hda_dsp_stream_hw_params(sdev, dsp_stream, dmab, NULL); if (ret < 0) { dev_err(sdev->dev, "error: hdac prepare failed: %d\n", ret); - goto error; + goto out_free; } hda_dsp_stream_spib_config(sdev, dsp_stream, HDA_DSP_SPIB_ENABLE, size); } return dsp_stream; -error: - hda_dsp_stream_put(sdev, direction, hstream->stream_tag); +out_free: snd_dma_free_pages(dmab); +out_put: + hda_dsp_stream_put(sdev, direction, hstream->stream_tag); return ERR_PTR(ret); } base-commit: 23d04328444a8fa0ca060c5e532220dac8e8bc26 -- 2.32.0

4 12

[PATCH] ASoC: SOF: pcm: Add compress_ops for SOF platform component driver
by Daniel Baluta 24 Feb '22

24 Feb '22

From: Daniel Baluta <daniel.baluta(a)nxp.com> Now that sof_compressed_ops initial implementation was merged we can enable it in SOF platform component driver. This partially reverts commit 8a720724589e ("ASoC: SOF: pcm: Remove non existent CONFIG_SND_SOC_SOF_COMPRESS reference") Reported-by: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/sof/pcm.c | 4 ++++ sound/soc/sof/sof-priv.h | 5 +++++ 2 files changed, 9 insertions(+) diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 137f8ed71677..a312ed855f1a 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -922,6 +922,10 @@ void snd_sof_new_platform_drv(struct snd_sof_dev *sdev) pd->pointer = sof_pcm_pointer; pd->ack = sof_pcm_ack; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_COMPRESS) + pd->compress_ops = &sof_compressed_ops; +#endif + pd->pcm_construct = sof_pcm_new; pd->ignore_machine = drv_name; pd->be_hw_params_fixup = sof_pcm_dai_link_fixup; diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index 2c8e556cd5cc..886787a9997f 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -556,6 +556,11 @@ int snd_sof_debugfs_add_region_item_iomem(struct snd_sof_dev *sdev, enum snd_sof_fw_blk_type blk_type, u32 offset, size_t size, const char *name, enum sof_debugfs_access_type access_type); +/* + * Platform specific ops. + */ +extern struct snd_compress_ops sof_compressed_ops; + /* * DSP Architectures. */ -- 2.27.0

3 2

[PATCH v2 0/3] ASoC: amd: acp: Add new machines and minor tweaks
by Ajit Kumar Pandey 24 Feb '22

24 Feb '22

changes since v1: - Fixed error for undeclared variable. Ajit Kumar Pandey (3): ASoC: amd: acp: Change card name for Guybrush Machine ASoC: amd: acp-legacy: Add legacy card support for new machines ASoC: amd: acp: Add DMIC machine driver ops sound/soc/amd/acp/acp-legacy-mach.c | 34 +++++++++++++++++++++++++++-- sound/soc/amd/acp/acp-mach-common.c | 27 +++++++++++++++++++++++ sound/soc/amd/acp/acp-renoir.c | 19 +++++++++++++++- 3 files changed, 77 insertions(+), 3 deletions(-) -- 2.25.1

2 4

[PATCH] ASoC: tlv320adc3xxx: Fix buggy return value
by Ricard Wanderlof 24 Feb '22

24 Feb '22

snd_soc_component_update_bits returns 1 if the operation was successful and some bits were changed, so we cannot return this value directly as it can be interpreted as an error. Instead, do some minor mangling to avoid inadvertently returning an error. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- sound/soc/codecs/tlv320adc3xxx.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/tlv320adc3xxx.c b/sound/soc/codecs/tlv320adc3xxx.c index 4baf3d881633..f15e3ea8685c 100644 --- a/sound/soc/codecs/tlv320adc3xxx.c +++ b/sound/soc/codecs/tlv320adc3xxx.c @@ -1111,6 +1111,7 @@ static int adc3xxx_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct adc3xxx *adc3xxx = snd_soc_component_get_drvdata(component); u8 clkdir = 0, format = 0; int master = 0; + int ret; /* set master/slave audio interface */ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { @@ -1161,10 +1162,13 @@ static int adc3xxx_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) adc3xxx->master = master; /* set clock direction and format */ - return snd_soc_component_update_bits(component, - ADC3XXX_INTERFACE_CTRL_1, - ADC3XXX_CLKDIR_MASK | ADC3XXX_FORMAT_MASK, - clkdir | format); + ret = snd_soc_component_update_bits(component, + ADC3XXX_INTERFACE_CTRL_1, + ADC3XXX_CLKDIR_MASK | ADC3XXX_FORMAT_MASK, + clkdir | format); + if (ret < 0) + return ret; + return 0; } static const struct snd_soc_dai_ops adc3xxx_dai_ops = { -- 2.20.1 -- Ricard Wolf Wanderlof ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[PATCH V2 0/1] ASoC: compress: propagate the error code from the
by Raghu Bankapur 24 Feb '22

24 Feb '22

Propagate the error code from the compress framework for the timestamp query. This error code will be used by the client to handle the error case scenarios gracefully. This propgated error is used by direct client like Audio HAL through IOCTL -SNDRV_COMPRESS_TSTAMP response, further this response is sent to upper layer, based on this error audio related usecases related to render position and presentation are handled. Changes since v1: - Remove initialization for ret, since it is updated by pointer function - Added new line after ret declaration Raghu Bankapur (1): ASoC: compress: propagate the error code from the compress framework sound/core/compress_offload.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) -- 2.17.1

2 2

[PATCH] ASoC: Intel: soc-acpi-byt: Add new WM5102 ACPI HID
by Hans de Goede 24 Feb '22

24 Feb '22

The Lenovo Yoga Tablet 2 1050F/L tablets use an ACPI HID of "10WM5102" for their wm5102 codec, add this to the list of HIDs for the wm5102 codec. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/common/soc-acpi-intel-byt-match.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c index 142000991813..c532529a3856 100644 --- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c @@ -127,7 +127,7 @@ static const struct snd_soc_acpi_codecs rt5640_comp_ids = { static const struct snd_soc_acpi_codecs wm5102_comp_ids = { .num_codecs = 2, - .codecs = { "WM510204", "WM510205"}, + .codecs = { "10WM5102", "WM510204", "WM510205"}, }; static const struct snd_soc_acpi_codecs da7213_comp_ids = { -- 2.35.1

4 3

[PATCH] ASoC: rt5682s: Fix the wrong jack type detected
by derek.fang＠realtek.com 24 Feb '22

24 Feb '22

From: Derek Fang <derek.fang(a)realtek.com> Some powers were changed during the jack insert detection and clk's enable/disable in CCF. If in parallel, the influence has a chance to detect the wrong jack type. We refer to the below commit of the variant codec (rt5682) to fix this issue. ASoC: rt5682: Fix deadlock on resume 1. Remove rt5682s_headset_detect in rt5682s_jd_check_handler and use jack_detect_work instead of. 2. Use dapm mutex used in CCF to protect most of jack_detect_work. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> --- sound/soc/codecs/rt5682s.c | 26 +++++++++----------------- sound/soc/codecs/rt5682s.h | 1 - 2 files changed, 9 insertions(+), 18 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index efa1016831dd..849fcd7132a8 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -822,6 +822,7 @@ static void rt5682s_jack_detect_handler(struct work_struct *work) { struct rt5682s_priv *rt5682s = container_of(work, struct rt5682s_priv, jack_detect_work.work); + struct snd_soc_dapm_context *dapm; int val, btn_type; while (!rt5682s->component) @@ -830,7 +831,9 @@ static void rt5682s_jack_detect_handler(struct work_struct *work) while (!rt5682s->component->card->instantiated) usleep_range(10000, 15000); - mutex_lock(&rt5682s->jdet_mutex); + dapm = snd_soc_component_get_dapm(rt5682s->component); + + snd_soc_dapm_mutex_lock(dapm); mutex_lock(&rt5682s->calibrate_mutex); val = snd_soc_component_read(rt5682s->component, RT5682S_AJD1_CTRL) @@ -887,6 +890,9 @@ static void rt5682s_jack_detect_handler(struct work_struct *work) rt5682s->irq_work_delay_time = 50; } + mutex_unlock(&rt5682s->calibrate_mutex); + snd_soc_dapm_mutex_unlock(dapm); + snd_soc_jack_report(rt5682s->hs_jack, rt5682s->jack_type, SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3); @@ -896,9 +902,6 @@ static void rt5682s_jack_detect_handler(struct work_struct *work) schedule_delayed_work(&rt5682s->jd_check_work, 0); else cancel_delayed_work_sync(&rt5682s->jd_check_work); - - mutex_unlock(&rt5682s->calibrate_mutex); - mutex_unlock(&rt5682s->jdet_mutex); } static void rt5682s_jd_check_handler(struct work_struct *work) @@ -906,14 +909,9 @@ static void rt5682s_jd_check_handler(struct work_struct *work) struct rt5682s_priv *rt5682s = container_of(work, struct rt5682s_priv, jd_check_work.work); - if (snd_soc_component_read(rt5682s->component, RT5682S_AJD1_CTRL) - & RT5682S_JDH_RS_MASK) { + if (snd_soc_component_read(rt5682s->component, RT5682S_AJD1_CTRL) & RT5682S_JDH_RS_MASK) { /* jack out */ - rt5682s->jack_type = rt5682s_headset_detect(rt5682s->component, 0); - - snd_soc_jack_report(rt5682s->hs_jack, rt5682s->jack_type, - SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | - SND_JACK_BTN_2 | SND_JACK_BTN_3); + schedule_delayed_work(&rt5682s->jack_detect_work, 0); } else { schedule_delayed_work(&rt5682s->jd_check_work, 500); } @@ -1321,7 +1319,6 @@ static int rt5682s_hp_amp_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 rt5682s_priv *rt5682s = snd_soc_component_get_drvdata(component); switch (event) { case SND_SOC_DAPM_POST_PMU: @@ -1337,8 +1334,6 @@ static int rt5682s_hp_amp_event(struct snd_soc_dapm_widget *w, snd_soc_component_write(component, RT5682S_BIAS_CUR_CTRL_11, 0x6666); snd_soc_component_write(component, RT5682S_BIAS_CUR_CTRL_12, 0xa82a); - mutex_lock(&rt5682s->jdet_mutex); - snd_soc_component_update_bits(component, RT5682S_HP_CTRL_2, RT5682S_HPO_L_PATH_MASK | RT5682S_HPO_R_PATH_MASK | RT5682S_HPO_SEL_IP_EN_SW, RT5682S_HPO_L_PATH_EN | @@ -1346,8 +1341,6 @@ static int rt5682s_hp_amp_event(struct snd_soc_dapm_widget *w, usleep_range(5000, 10000); snd_soc_component_update_bits(component, RT5682S_HP_AMP_DET_CTL_1, RT5682S_CP_SW_SIZE_MASK, RT5682S_CP_SW_SIZE_L | RT5682S_CP_SW_SIZE_S); - - mutex_unlock(&rt5682s->jdet_mutex); break; case SND_SOC_DAPM_POST_PMD: @@ -3101,7 +3094,6 @@ static int rt5682s_i2c_probe(struct i2c_client *i2c, mutex_init(&rt5682s->calibrate_mutex); mutex_init(&rt5682s->sar_mutex); - mutex_init(&rt5682s->jdet_mutex); rt5682s_calibrate(rt5682s); regmap_update_bits(rt5682s->regmap, RT5682S_MICBIAS_2, diff --git a/sound/soc/codecs/rt5682s.h b/sound/soc/codecs/rt5682s.h index 1bf2ef7ce578..397a2531b6f6 100644 --- a/sound/soc/codecs/rt5682s.h +++ b/sound/soc/codecs/rt5682s.h @@ -1446,7 +1446,6 @@ struct rt5682s_priv { struct delayed_work jd_check_work; struct mutex calibrate_mutex; struct mutex sar_mutex; - struct mutex jdet_mutex; #ifdef CONFIG_COMMON_CLK struct clk_hw dai_clks_hw[RT5682S_DAI_NUM_CLKS]; -- 2.17.1

2 1

[PATCH V1 0/1] ASoC: compress: propagate the error code from the
by Raghu Bankapur 24 Feb '22

24 Feb '22

Propagate the error code from the compress framework for the timestamp query. This error code will be used by the client to handle the error case scenarios gracefully. This propgated error is used by direct client like Audio HAL through IOCTL -SNDRV_COMPRESS_TSTAMP response, further this response is sent to upper layer, based on this error audio related usecases related to render position and presentation are handled. Raghu Bankapur (1): ASoC: compress: propagate the error code from the compress framework sound/core/compress_offload.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) -- 2.17.1

2 2

[PATCH v2] ASoC: codecs: Add power domains support in digital macro codecs
by Srinivasa Rao Mandadapu 24 Feb '22

24 Feb '22

Add support for enabling required power domains in digital macro codecs. macro and dcodec power domains are being requested as clocks by HLOS in ADSP based architectures and ADSP internally handling as powerdomains. In ADSP bypass case need to handle them as power domains explicitly. Signed-off-by: Srinivasa Rao Mandadapu <quic_srivasam(a)quicinc.com> Co-developed-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> Signed-off-by: Venkata Prasad Potturu <quic_potturu(a)quicinc.com> Reported-by: kernel test robot <lkp(a)intel.com> --- Changes since v1: -- Add missing macros in Kconfig. sound/soc/codecs/Kconfig | 7 ++++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/lpass-macro-common.c | 72 +++++++++++++++++++++++++++++++++++ sound/soc/codecs/lpass-macro-common.h | 18 +++++++++ sound/soc/codecs/lpass-rx-macro.c | 13 ++++++- sound/soc/codecs/lpass-tx-macro.c | 10 +++++ sound/soc/codecs/lpass-va-macro.c | 11 +++++- sound/soc/qcom/Kconfig | 1 + 8 files changed, 132 insertions(+), 2 deletions(-) create mode 100644 sound/soc/codecs/lpass-macro-common.c create mode 100644 sound/soc/codecs/lpass-macro-common.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index c2627f7..4de029a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -244,6 +244,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_WCD9335 imply SND_SOC_WCD934X imply SND_SOC_WCD938X_SDW + imply SND_SOC_LPASS_MACRO_COMMON imply SND_SOC_LPASS_RX_MACRO imply SND_SOC_LPASS_TX_MACRO imply SND_SOC_WL1273 @@ -2008,6 +2009,9 @@ config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C +config SND_SOC_LPASS_MACRO_COMMON + tristate + config SND_SOC_LPASS_WSA_MACRO depends on COMMON_CLK select REGMAP_MMIO @@ -2016,16 +2020,19 @@ config SND_SOC_LPASS_WSA_MACRO config SND_SOC_LPASS_VA_MACRO depends on COMMON_CLK select REGMAP_MMIO + select SND_SOC_LPASS_MACRO_COMMON tristate "Qualcomm VA Macro in LPASS(Low Power Audio SubSystem)" config SND_SOC_LPASS_RX_MACRO depends on COMMON_CLK select REGMAP_MMIO + select SND_SOC_LPASS_MACRO_COMMON tristate "Qualcomm RX Macro in LPASS(Low Power Audio SubSystem)" config SND_SOC_LPASS_TX_MACRO depends on COMMON_CLK select REGMAP_MMIO + select SND_SOC_LPASS_MACRO_COMMON tristate "Qualcomm TX Macro in LPASS(Low Power Audio SubSystem)" endmenu diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b4e11c3..c3c6059 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -112,6 +112,7 @@ snd-soc-l3-objs := l3.o snd-soc-lm4857-objs := lm4857.o snd-soc-lm49453-objs := lm49453.o snd-soc-lochnagar-sc-objs := lochnagar-sc.o +snd-soc-lpass-macro-common-objs := lpass-macro-common.o snd-soc-lpass-rx-macro-objs := lpass-rx-macro.o snd-soc-lpass-tx-macro-objs := lpass-tx-macro.o snd-soc-lpass-wsa-macro-objs := lpass-wsa-macro.o @@ -676,6 +677,7 @@ obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_MAX98504) += snd-soc-max98504.o obj-$(CONFIG_SND_SOC_SIMPLE_AMPLIFIER) += snd-soc-simple-amplifier.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o +obj-$(CONFIG_SND_SOC_LPASS_MACRO_COMMON) += snd-soc-lpass-macro-common.o obj-$(CONFIG_SND_SOC_LPASS_WSA_MACRO) += snd-soc-lpass-wsa-macro.o obj-$(CONFIG_SND_SOC_LPASS_VA_MACRO) += snd-soc-lpass-va-macro.o obj-$(CONFIG_SND_SOC_LPASS_RX_MACRO) += snd-soc-lpass-rx-macro.o diff --git a/sound/soc/codecs/lpass-macro-common.c b/sound/soc/codecs/lpass-macro-common.c new file mode 100644 index 0000000..b8e50e6 --- /dev/null +++ b/sound/soc/codecs/lpass-macro-common.c @@ -0,0 +1,72 @@ +// SPDX-License-Identifier: GPL-2.0-only +// Copyright (c) 2022, The Linux Foundation. All rights reserved. + +#include <linux/export.h> +#include <linux/module.h> +#include <linux/init.h> +#include <linux/of_platform.h> +#include <linux/platform_device.h> +#include <linux/pm_domain.h> +#include <linux/pm_runtime.h> + +#include "lpass-macro-common.h" + +int lpass_macro_pds_init(struct platform_device *pdev, struct lpass_macro **pds) +{ + struct device *dev = &pdev->dev; + struct lpass_macro *l_pds; + int ret; + + const struct property *prop = of_find_property(dev->of_node, "power-domains", NULL); + + if (!prop) + return 0; + + l_pds = devm_kzalloc(dev, sizeof(*l_pds), GFP_KERNEL); + if (!l_pds) + return -ENOMEM; + + l_pds->macro_pd = dev_pm_domain_attach_by_name(dev, "macro"); + if (IS_ERR_OR_NULL(l_pds->macro_pd)) { + ret = PTR_ERR(l_pds->macro_pd) ? : -ENODATA; + return ret; + } + ret = pm_runtime_get_sync(l_pds->macro_pd); + if (ret < 0) { + dev_err(dev, "%s failed for macro_pd, ret %d\n", __func__, ret); + dev_pm_domain_detach(l_pds->macro_pd, false); + pm_runtime_put_noidle(l_pds->macro_pd); + return ret; + } + + l_pds->dcodec_pd = dev_pm_domain_attach_by_name(dev, "dcodec"); + if (IS_ERR_OR_NULL(l_pds->dcodec_pd)) { + ret = PTR_ERR(l_pds->dcodec_pd) ? : -ENODATA; + dev_pm_domain_detach(l_pds->macro_pd, false); + return ret; + } + + ret = pm_runtime_get_sync(l_pds->dcodec_pd); + if (ret < 0) { + dev_err(dev, "%s failed for dcodec_pd, ret %d\n", __func__, ret); + + dev_pm_domain_detach(l_pds->dcodec_pd, false); + pm_runtime_put_noidle(l_pds->dcodec_pd); + return ret; + } + *pds = l_pds; + return ret; +} +EXPORT_SYMBOL_GPL(lpass_macro_pds_init); + +void lpass_macro_pds_exit(struct platform_device *pdev, struct lpass_macro *pds) +{ + pm_runtime_put(pds->macro_pd); + pm_runtime_put(pds->dcodec_pd); + dev_pm_domain_detach(pds->macro_pd, false); + dev_pm_domain_detach(pds->dcodec_pd, false); +} +EXPORT_SYMBOL_GPL(lpass_macro_pds_exit); + +MODULE_DESCRIPTION("QTI SC7280 LPI GPIO pin control driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/lpass-macro-common.h b/sound/soc/codecs/lpass-macro-common.h new file mode 100644 index 0000000..c343f0e --- /dev/null +++ b/sound/soc/codecs/lpass-macro-common.h @@ -0,0 +1,18 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Copyright (c) 2022, The Linux Foundation. All rights reserved. + */ + +#ifndef __LPASS_MACRO_COMMON_H__ +#define __LPASS_MACRO_COMMON_H__ + + +struct lpass_macro { + struct device *macro_pd; + struct device *dcodec_pd; +}; + +int lpass_macro_pds_init(struct platform_device *pdev, struct lpass_macro **pds); +void lpass_macro_pds_exit(struct platform_device *pdev, struct lpass_macro *pds); + +#endif /* __LPASS_MACRO_COMMON_H__ */ diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 29d214f..db32090 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -14,6 +14,8 @@ #include <linux/of_clk.h> #include <linux/clk-provider.h> +#include "lpass-macro-common.h" + #define CDC_RX_TOP_TOP_CFG0 (0x0000) #define CDC_RX_TOP_SWR_CTRL (0x0008) #define CDC_RX_TOP_DEBUG (0x000C) @@ -606,7 +608,7 @@ struct rx_macro { int is_softclip_on; int is_aux_hpf_on; int softclip_clk_users; - + struct lpass_macro *pds; struct regmap *regmap; struct clk_bulk_data clks[RX_NUM_CLKS_MAX]; struct clk_hw hw; @@ -3537,6 +3539,12 @@ static int rx_macro_probe(struct platform_device *pdev) return ret; } + ret = lpass_macro_pds_init(pdev, &rx->pds); + if (ret < 0) { + dev_err(dev, "Enabling power domains failed in %s\n", __func__); + return ret; + } + base = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(base)) return PTR_ERR(base); @@ -3575,6 +3583,9 @@ static int rx_macro_remove(struct platform_device *pdev) of_clk_del_provider(pdev->dev.of_node); clk_bulk_disable_unprepare(RX_NUM_CLKS_MAX, rx->clks); + + lpass_macro_pds_exit(pdev, rx->pds); + return 0; } diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index 9c96ab1..4d1e5ab 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -13,6 +13,8 @@ #include <linux/of_clk.h> #include <linux/clk-provider.h> +#include "lpass-macro-common.h" + #define CDC_TX_CLK_RST_CTRL_MCLK_CONTROL (0x0000) #define CDC_TX_MCLK_EN_MASK BIT(0) #define CDC_TX_MCLK_ENABLE BIT(0) @@ -266,6 +268,7 @@ struct tx_macro { u16 dmic_clk_div; bool bcs_enable; int dec_mode[NUM_DECIMATORS]; + struct lpass_macro *pds; bool bcs_clk_en; }; #define to_tx_macro(_hw) container_of(_hw, struct tx_macro, hw) @@ -1802,6 +1805,11 @@ static int tx_macro_probe(struct platform_device *pdev) return ret; } + ret = lpass_macro_pds_init(pdev, &tx->pds); + if (ret < 0) { + dev_err(dev, "Enabling power domains failed in %s\n", __func__); + return ret; + } base = devm_platform_ioremap_resource(pdev, 0); if (IS_ERR(base)) return PTR_ERR(base); @@ -1859,6 +1867,8 @@ static int tx_macro_remove(struct platform_device *pdev) clk_bulk_disable_unprepare(TX_NUM_CLKS_MAX, tx->clks); + lpass_macro_pds_exit(pdev, tx->pds); + return 0; } diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index 11147e3..b29b9a1 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -15,6 +15,8 @@ #include <sound/soc-dapm.h> #include <sound/tlv.h> +#include "lpass-macro-common.h" + /* VA macro registers */ #define CDC_VA_CLK_RST_CTRL_MCLK_CONTROL (0x0000) #define CDC_VA_MCLK_CONTROL_EN BIT(0) @@ -195,6 +197,7 @@ struct va_macro { struct regmap *regmap; struct clk_bulk_data clks[VA_NUM_CLKS_MAX]; struct clk_hw hw; + struct lpass_macro *pds; s32 dmic_0_1_clk_cnt; s32 dmic_2_3_clk_cnt; @@ -1413,7 +1416,11 @@ static int va_macro_probe(struct platform_device *pdev) dev_err(dev, "Error getting VA Clocks (%d)\n", ret); return ret; } - + ret = lpass_macro_pds_init(pdev, &va->pds); + if (ret < 0) { + dev_err(dev, "Enabling power domains failed %s\n", __func__); + return ret; + } ret = of_property_read_u32(dev->of_node, "qcom,dmic-sample-rate", &sample_rate); if (ret) { @@ -1468,6 +1475,8 @@ static int va_macro_remove(struct platform_device *pdev) clk_bulk_disable_unprepare(VA_NUM_CLKS_MAX, va->clks); + lpass_macro_pds_exit(pdev, va->pds); + return 0; } diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 52db003..6ffd51a 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -194,6 +194,7 @@ config SND_SOC_SC7280 select SND_SOC_LPASS_SC7280 select SND_SOC_MAX98357A select SND_SOC_WCD938X + select SND_SOC_LPASS_MACRO_COMMON select SND_SOC_LPASS_RX_MACRO select SND_SOC_LPASS_TX_MACRO help -- 2.7.4

2 1

[PATCH V1 0/1]ASoC: compress: propagate the error code from the compress framework
by Raghu Bankapur 23 Feb '22

23 Feb '22

Propagate the error code from the compress framework for the timestamp query. This error code will be used by the client to handle the error case scenarios gracefully. This propgated error is used by direct client like Audio HAL through IOCTL -SNDRV_COMPRESS_TSTAMP response, further this response is sent to upper layer, based on this error audio related usecases related to render position and presentation are handled. Raghu Bankapur (1): ASoC: compress: propagate the error code from the compress framework sound/core/compress_offload.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) -- 2.17.1

2 3

[PATCH 0/3] soundwire: qcom: add pm runtime support
by Srinivas Kandagatla 23 Feb '22

23 Feb '22

This patchset adds pm runtime support to Qualcomm SounWire Controller using SoundWire Clock Stop and Wake up using Headset events on supported instances and a bus reset on instances that require full reset. Tested it on SM8250 MTP and Dragon Board DB845c --srini Srinivas Kandagatla (3): soundwire: qcom: add runtime pm support dt-bindings: soundwire: qcom: document optional wake irq soundwire: qcom: add wake up interrupt support .../bindings/soundwire/qcom,sdw.txt | 2 +- drivers/soundwire/qcom.c | 215 +++++++++++++++++- 2 files changed, 215 insertions(+), 2 deletions(-) -- 2.21.0

3 12

[RFT PATCH 0/3] Fix kfree() of const memory on setting driver_override
by Krzysztof Kozlowski 23 Feb '22

23 Feb '22

Hi, Drivers still seem to use driver_override incorrectly. Perhaps my old patch makes sense now? https://lore.kernel.org/all/1550484960-2392-3-git-send-email-krzk@kernel.or… Not tested - please review and test (e.g. by writing to dirver_override sysfs entry with KASAN enabled). Dependencies ============ Patches are independent. Best regards, Krzysztof Krzysztof Kozlowski (3): clk: imx: scu: fix kfree() of const memory on setting driver_override slimbus: qcom-ngd: fix kfree() of const memory on setting driver_override rpmsg: fix kfree() of const memory on setting driver_override drivers/clk/imx/clk-scu.c | 6 +++++- drivers/rpmsg/rpmsg_internal.h | 12 ++++++++++-- drivers/rpmsg/rpmsg_ns.c | 13 +++++++++++-- drivers/slimbus/qcom-ngd-ctrl.c | 9 ++++++++- 4 files changed, 34 insertions(+), 6 deletions(-) -- 2.32.0

3 9

[GIT PULL] ASoC fixes for v5.17-rc5
by Mark Brown 23 Feb '22

23 Feb '22

The following changes since commit a887f9c7a4d37a8e874ba8415a42a92a1b5139fc: ASoC: wm_adsp: Correct control read size when parsing compressed buffer (2022-02-10 17:26:43 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v5.17-rc5 for you to fetch changes up to c5487b9cdea5c1ede38a7ec94db0fc59963c8e86: ASoC: cs4265: Fix the duplicated control name (2022-02-16 16:34:16 +0000) ---------------------------------------------------------------- ASoC: Fixes for v5.17 A few more fixes for v5.17, one followup to the bounds checking fixes handling controls which support negative values internally and a driver specific one. ---------------------------------------------------------------- Fabio Estevam (1): ASoC: cs4265: Fix the duplicated control name Marek Vasut (1): ASoC: ops: Shift tested values in snd_soc_put_volsw() by +min sound/soc/codecs/cs4265.c | 3 +-- sound/soc/soc-ops.c | 4 ++-- 2 files changed, 3 insertions(+), 4 deletions(-)

2 1

[PATCH] pcm: rate: fix drain of partial period at end of buffer
by Alan Young 23 Feb '22

23 Feb '22

In the case that: * the buffer size is not an integer multiple of the period size, and * drain must flush a partial period located before the end of the buffer but without a full period available, where * these conditions may pertain to the source or slave pcm buffer, and * because rate conversion is always done on a full period, it is necessary to check that both a full source period is available before source pcm buffer wrap and a full slave period is available before slave pcm buffer wrap in order to use the simple, single-commit implementation in snd_pcm_rate_commit_area(). The alternative fix would be to change snd_pcm_rate_write_areas1() to take size and slave_size parameters. This would be more disruptive to the code base, tricky to get right, and is unnecessary given that snd_pcm_mmap_commit() only commits the partial period of actually valid converted samples. Fixes: 3047f8fa5a3dce0c9775404a2285fb2cff462d96 Signed-off-by: Alan Young <consult.awy(a)gmail.com> --- src/pcm/pcm_rate.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index c45895a9..ba5364c0 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -781,16 +781,25 @@ static int snd_pcm_rate_commit_area(snd_pcm_t *pcm, snd_pcm_rate_t *rate, snd_pcm_sframes_t result; areas = snd_pcm_mmap_areas(pcm); - if (cont >= size) { + /* + * Because snd_pcm_rate_write_areas1() below will convert a full source period + * then there had better be a full period available in the current buffer. + */ + if (cont >= pcm->period_size) { result = snd_pcm_mmap_begin(rate->gen.slave, &slave_areas, &slave_offset, &slave_frames); if (result < 0) return result; - if (slave_frames < slave_size) { + /* + * Because snd_pcm_rate_write_areas1() below will convert to a full slave period + * then there had better be a full slave period available in the slave buffer. + */ + if (slave_frames < rate->gen.slave->period_size) { snd_pcm_rate_write_areas1(pcm, areas, appl_offset, rate->sareas, 0); goto __partial; } snd_pcm_rate_write_areas1(pcm, areas, appl_offset, slave_areas, slave_offset); + /* Only commit the requested slave_size, even if more was actually converted */ result = snd_pcm_mmap_commit(rate->gen.slave, slave_offset, slave_size); if (result < (snd_pcm_sframes_t)slave_size) { if (result < 0) -- 2.25.1

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[PATCH v9 0/7] Add pin control support for lpass sc7280
by Srinivasa Rao Mandadapu 23 Feb '22

23 Feb '22

This patch series is to split lpass variant common pin control functions and SoC specific functions and to add lpass sc7280 pincontrol support. It also Adds dt-bindings for lpass sc7280 lpass lpi pincontrol. Changes Since V8: -- Remove redundant headers included in v8. Changes Since V7: -- Update optional clock voting with conditional check. -- Add const to lpi_pinctrl_variant_data structure. -- Update required headers and remove redundant. -- Change EXPORT_SYMBOL to EXPORT_SYMBOL_GPL -- Fix typo errors. Changes Since V6: -- Update conditional clock voting to optional clock voting. -- Update Kconfig depends on field with select. -- Fix typo errors. Changes Since V5: -- Create new patch by updating macro name to lpi specific. -- Create new patch by updating lpi pin group structure with core group_desc structure. -- Fix typo errors. -- Sort macros in the make file and configuration file. Changes Since V4: -- Update commit message and description of the chip specific extraction patch. -- Sort macros in kconfig and makefile. -- Update optional clock voting to conditional clock voting. -- Fix typo errors. -- Move to quicinc domain email id's. Changes Since V3: -- Update separate Kconfig fields for sm8250 and sc7280. -- Update module license and description. -- Move static variables to corresponding .c files from header file. Changes Since V2: -- Add new dt-bindings for sc7280 lpi driver. -- Make clock voting change as separate patch. -- Split existing pincontrol driver and make common functions as part of separate file. -- Rename lpass pincontrol lpi dt-bindings to sm8250 specific dt-bindings Changes Since V1: -- Make lpi pinctrl variant data structure as constant -- Add appropriate commit message -- Change signedoff by sequence. Srinivasa Rao Mandadapu (7): dt-bindings: pinctrl: qcom: Update lpass lpi file name to SoC specific dt-bindings: pinctrl: qcom: Add sc7280 lpass lpi pinctrl bindings pinctrl: qcom: Update macro name to LPI specific pinctrl: qcom: Update lpi pin group structure pinctrl: qcom: Extract chip specific LPASS LPI code pinctrl: qcom: Add SC7280 lpass pin configuration pinctrl: qcom: Update clock voting as optional Tested this on SM8250 MTP with WSA and WCD codecs. Tested-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> .../bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml | 133 ----------- .../pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 115 ++++++++++ .../pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml | 133 +++++++++++ drivers/pinctrl/qcom/Kconfig | 16 ++ drivers/pinctrl/qcom/Makefile | 2 + drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 255 ++------------------- drivers/pinctrl/qcom/pinctrl-lpass-lpi.h | 86 +++++++ drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 168 ++++++++++++++ drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c | 165 +++++++++++++ 9 files changed, 703 insertions(+), 370 deletions(-) delete mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml create mode 100644 drivers/pinctrl/qcom/pinctrl-lpass-lpi.h create mode 100644 drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c create mode 100644 drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c -- 2.7.4

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[PATCH v6 0/7] Add pin control support for lpass sc7280
by Srinivasa Rao Mandadapu 23 Feb '22

23 Feb '22

This patch series is to split lpass variant common pin control functions and SoC specific functions and to add lpass sc7280 pincontrol support. It also Adds dt-bindings for lpass sc7280 lpass lpi pincontrol. Changes Since V5: -- Create new patch by updating macro name to lpi specific. -- Create new patch by updating lpi pin group structure with core group_desc structure. -- Fix typo errors. -- Sort macros in the make file and configuration file. Changes Since V4: -- Update commit message and description of the chip specific extraction patch. -- Sort macros in kconfig and makefile. -- Update optional clock voting to conditional clock voting. -- Fix typo errors. -- Move to quicinc domain email id's. Changes Since V3: -- Update separate Kconfig fields for sm8250 and sc7280. -- Update module license and description. -- Move static variables to corresponding .c files from header file. Changes Since V2: -- Add new dt-bindings for sc7280 lpi driver. -- Make clock voting change as separate patch. -- Split existing pincontrol driver and make common functions as part of separate file. -- Rename lpass pincontrol lpi dt-bindings to sm8250 specific dt-bindings Changes Since V1: -- Make lpi pinctrl variant data structure as constant -- Add appropriate commit message -- Change signedoff by sequence. Srinivasa Rao Mandadapu (7): dt-bindings: pinctrl: qcom: Update lpass lpi file name to SoC specific dt-bindings: pinctrl: qcom: Add sc7280 lpass lpi pinctrl bindings pinctrl: qcom: Update macro name to LPI specific pinctrl: qcom: Update lpi pin group structure pinctrl: qcom: Extract chip specific LPASS LPI code pinctrl: qcom: Add SC7280 lpass pin configuration pinctrl: qcom: Update clock voting as optional .../bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml | 133 ----------- .../pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml | 115 +++++++++ .../pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml | 133 +++++++++++ drivers/pinctrl/qcom/Kconfig | 16 ++ drivers/pinctrl/qcom/Makefile | 2 + drivers/pinctrl/qcom/pinctrl-lpass-lpi.c | 257 ++------------------- drivers/pinctrl/qcom/pinctrl-lpass-lpi.h | 87 +++++++ drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c | 170 ++++++++++++++ drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c | 166 +++++++++++++ 9 files changed, 706 insertions(+), 373 deletions(-) delete mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sc7280-lpass-lpi-pinctrl.yaml create mode 100644 Documentation/devicetree/bindings/pinctrl/qcom,sm8250-lpass-lpi-pinctrl.yaml create mode 100644 drivers/pinctrl/qcom/pinctrl-lpass-lpi.h create mode 100644 drivers/pinctrl/qcom/pinctrl-sc7280-lpass-lpi.c create mode 100644 drivers/pinctrl/qcom/pinctrl-sm8250-lpass-lpi.c -- 2.7.4

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[PATCH] ASoC: sc7280: Really depends on SOUNDWIRE
by Mark Brown 23 Feb '22

23 Feb '22

The sc7280 driver really does depend on SOUNDWIRE since it calls various sdw_ functions (eg, sdw_enable_stream(), sdw_prepare_stream()) which do not have stubs when that is disabled so we can't build with COMPILE_TEST. Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/qcom/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index dd5949eb6b15..f09dc0fcff0b 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -178,7 +178,7 @@ config SND_SOC_SC7180 config SND_SOC_SC7280 tristate "SoC Machine driver for SC7280 boards" - depends on I2C && SOUNDWIRE || COMPILE_TEST + depends on I2C && SOUNDWIRE select SND_SOC_QCOM_COMMON select SND_SOC_LPASS_SC7280 select SND_SOC_MAX98357A -- 2.30.2

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