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[PATCH v2 1/2] ASoC: ops: Correct bounds check for second channel on SX controls
by Charles Keepax 28 Nov '22

28 Nov '22

Currently the check against the max value for the control is being applied after the value has had the minimum applied and been masked. But the max value simply indicates the number of volume levels on an SX control, and as such should just be applied on the raw value. Fixes: 97eea946b939 ("ASoC: ops: Check bounds for second channel in snd_soc_put_volsw_sx()") Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Changes since v1: - Correct fixes SHA. sound/soc/soc-ops.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 1970bda074d8a..55b009d3c6815 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -464,16 +464,15 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, ret = err; if (snd_soc_volsw_is_stereo(mc)) { - unsigned int val2; - - val_mask = mask << rshift; - val2 = (ucontrol->value.integer.value[1] + min) & mask; + unsigned int val2 = ucontrol->value.integer.value[1]; if (mc->platform_max && val2 > mc->platform_max) return -EINVAL; if (val2 > max) return -EINVAL; + val_mask = mask << rshift; + val2 = (val2 + min) & mask; val2 = val2 << rshift; err = snd_soc_component_update_bits(component, reg2, val_mask, -- 2.30.2

2 5

[PATCH v5] ASoC: SOF: Fix deadlock when shutdown a frozen userspace
by Ricardo Ribalda 28 Nov '22

28 Nov '22

During kexec(), the userspace is frozen. Therefore we cannot wait for it to complete. Avoid running snd_sof_machine_unregister during shutdown. This fixes: [ 84.943749] Freezing user space processes ... (elapsed 0.111 seconds) done. [ 246.784446] INFO: task kexec-lite:5123 blocked for more than 122 seconds. [ 246.819035] Call Trace: [ 246.821782] <TASK> [ 246.824186] __schedule+0x5f9/0x1263 [ 246.828231] schedule+0x87/0xc5 [ 246.831779] snd_card_disconnect_sync+0xb5/0x127 ... [ 246.889249] snd_sof_device_shutdown+0xb4/0x150 [ 246.899317] pci_device_shutdown+0x37/0x61 [ 246.903990] device_shutdown+0x14c/0x1d6 [ 246.908391] kernel_kexec+0x45/0xb9 And: [ 246.893222] INFO: task kexec-lite:4891 blocked for more than 122 seconds. [ 246.927709] Call Trace: [ 246.930461] <TASK> [ 246.932819] __schedule+0x5f9/0x1263 [ 246.936855] ? fsnotify_grab_connector+0x5c/0x70 [ 246.942045] schedule+0x87/0xc5 [ 246.945567] schedule_timeout+0x49/0xf3 [ 246.949877] wait_for_completion+0x86/0xe8 [ 246.954463] snd_card_free+0x68/0x89 ... [ 247.001080] platform_device_unregister+0x12/0x35 Cc: stable(a)vger.kernel.org Fixes: 83bfc7e793b5 ("ASoC: SOF: core: unregister clients and machine drivers in .shutdown") Signed-off-by: Ricardo Ribalda <ribalda(a)chromium.org> --- To: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> To: Liam Girdwood <lgirdwood(a)gmail.com> To: Peter Ujfalusi <peter.ujfalusi(a)linux.intel.com> To: Bard Liao <yung-chuan.liao(a)linux.intel.com> To: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> To: Kai Vehmanen <kai.vehmanen(a)linux.intel.com> To: Daniel Baluta <daniel.baluta(a)nxp.com> To: Mark Brown <broonie(a)kernel.org> To: Jaroslav Kysela <perex(a)perex.cz> To: Takashi Iwai <tiwai(a)suse.com> Cc: sound-open-firmware(a)alsa-project.org Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- Changes in v5: - Edit subject prefix - Link to v4: https://lore.kernel.org/r/20221127-snd-freeze-v4-0-51ca64b7f2ab@chromium.org Changes in v4: - Do not call snd_sof_machine_unregister from shutdown. - Link to v3: https://lore.kernel.org/r/20221127-snd-freeze-v3-0-a2eda731ca14@chromium.org Changes in v3: - Wrap pm_freezing in a function - Link to v2: https://lore.kernel.org/r/20221127-snd-freeze-v2-0-d8a425ea9663@chromium.org Changes in v2: - Only use pm_freezing if CONFIG_FREEZER - Link to v1: https://lore.kernel.org/r/20221127-snd-freeze-v1-0-57461a366ec2@chromium.org --- sound/soc/sof/core.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c index 3e6141d03770..9616ba607ded 100644 --- a/sound/soc/sof/core.c +++ b/sound/soc/sof/core.c @@ -475,19 +475,16 @@ EXPORT_SYMBOL(snd_sof_device_remove); int snd_sof_device_shutdown(struct device *dev) { struct snd_sof_dev *sdev = dev_get_drvdata(dev); - struct snd_sof_pdata *pdata = sdev->pdata; if (IS_ENABLED(CONFIG_SND_SOC_SOF_PROBE_WORK_QUEUE)) cancel_work_sync(&sdev->probe_work); /* - * make sure clients and machine driver(s) are unregistered to force - * all userspace devices to be closed prior to the DSP shutdown sequence + * make sure clients are unregistered prior to the DSP shutdown + * sequence. */ sof_unregister_clients(sdev); - snd_sof_machine_unregister(sdev, pdata); - if (sdev->fw_state == SOF_FW_BOOT_COMPLETE) return snd_sof_shutdown(sdev); --- base-commit: 4312098baf37ee17a8350725e6e0d0e8590252d4 change-id: 20221127-snd-freeze-1ee143228326 Best regards, -- Ricardo Ribalda <ribalda(a)chromium.org>

2 2

[PATCH] ASoC: rsnd: adg: use __clk_get_name() instead of local clk_name[]
by Kuninori Morimoto 28 Nov '22

28 Nov '22

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current rsnd_adg_clk_dbg_info() is using locak clk_name[] to ndicating clk name, but we don't want to use local clk_name[] everywhere when we support R-Car Gen4 sound to handling non compatible clk naming. This patch uses __clk_get_name() instead of local clk_name[] for it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 5f1e72edfee0..ca1e1281cefa 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -648,8 +648,8 @@ void rsnd_adg_clk_dbg_info(struct rsnd_priv *priv, struct seq_file *m) int i; for_each_rsnd_clk(clk, adg, i) - dbg_msg(dev, m, "%s : %pa : %ld\n", - clk_name[i], clk, clk_get_rate(clk)); + dbg_msg(dev, m, "%-18s : %pa : %ld\n", + __clk_get_name(clk), clk, clk_get_rate(clk)); dbg_msg(dev, m, "BRGCKR = 0x%08x, BRRA/BRRB = 0x%x/0x%x\n", adg->ckr, adg->rbga, adg->rbgb); -- 2.25.1

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[PATCH] ASoC: rt5640: Fix Jack work after system suspend
by Oder Chiou 28 Nov '22

28 Nov '22

We found an corner case in RT5640 codec driver which schedules jack work after system suspend as IRQ was enabled. Due to this, hitting the error as register access happening after suspend as jack worker thread getting scheduled in irq handler. The patch disables the irq during the suspend to prevent the corner case happening. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> Reported-by: Mohan Kumar D <mkumard(a)nvidia.com> --- sound/soc/codecs/rt5640.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index 0f8e6dd214b..708e55b7431 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -2792,6 +2792,11 @@ static int rt5640_suspend(struct snd_soc_component *component) { struct rt5640_priv *rt5640 = snd_soc_component_get_drvdata(component); + if (rt5640->irq) { + /* disable jack interrupts during system suspend */ + disable_irq(rt5640->irq); + } + rt5640_cancel_work(rt5640); snd_soc_component_force_bias_level(component, SND_SOC_BIAS_OFF); rt5640_reset(component); @@ -2815,6 +2820,9 @@ static int rt5640_resume(struct snd_soc_component *component) regcache_cache_only(rt5640->regmap, false); regcache_sync(rt5640->regmap); + if (rt5640->irq) + enable_irq(rt5640->irq); + if (rt5640->jack) { if (rt5640->jd_src == RT5640_JD_SRC_HDA_HEADER) { snd_soc_component_update_bits(component, -- 2.34.1

2 1

[PATCH v3 1/9] sound: sdw: Add hw_params to SoundWire config helper function
by Charles Keepax 28 Nov '22

28 Nov '22

The vast majority of the current users of the SoundWire framework have almost identical code for converting from hw_params to SoundWire configuration. Whilst complex devices might require more, it is very likely that most new devices will follow the same pattern. Save a little code by factoring this out into a helper function. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- I was a little bit two minds about whether to make this an inline or not, so any thoughts on that would be super welcome. The function does very little, especially given that SNDRV_PCM_STREAM_PLAYBACK == SDW_DATA_DIR_RX so the if is also really just an assignment. Thanks, Charles Changes since v2: - No difference Changes since v1: - Correct spelling error in commit message include/sound/sdw.h | 49 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 49 insertions(+) create mode 100644 include/sound/sdw.h diff --git a/include/sound/sdw.h b/include/sound/sdw.h new file mode 100644 index 0000000000000..6dcdb3228dba6 --- /dev/null +++ b/include/sound/sdw.h @@ -0,0 +1,49 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * linux/sound/sdw.h -- SoundWire helpers for ALSA/ASoC + * + * Copyright (c) 2022 Cirrus Logic Inc. + * + * Author: Charles Keepax <ckeepax(a)opensource.cirrus.com> + */ + +#include <linux/soundwire/sdw.h> +#include <sound/asound.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#ifndef __INCLUDE_SOUND_SDW_H +#define __INCLUDE_SOUND_SDW_H + +/** + * snd_sdw_params_to_config() - Conversion from hw_params to SoundWire config + * + * @substream: Pointer to the PCM substream structure + * @params: Pointer to the hardware params structure + * @stream_config: Stream configuration for the SoundWire audio stream + * @port_config: Port configuration for the SoundWire audio stream + * + * This function provides a basic conversion from the hw_params structure to + * SoundWire configuration structures. The user will at a minimum need to also + * set the port number in the port config, but may also override more of the + * setup, or in the case of a complex user, not use this helper at all and + * open-code everything. + */ +static inline void snd_sdw_params_to_config(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct sdw_stream_config *stream_config, + struct sdw_port_config *port_config) +{ + stream_config->frame_rate = params_rate(params); + stream_config->ch_count = params_channels(params); + stream_config->bps = snd_pcm_format_width(params_format(params)); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + stream_config->direction = SDW_DATA_DIR_RX; + else + stream_config->direction = SDW_DATA_DIR_TX; + + port_config->ch_mask = GENMASK(stream_config->ch_count - 1, 0); +} + +#endif -- 2.30.2

2 9

[PATCH v2 00/11] ASoC: dt-bindings: Rework Qualcomm APR/GPR Sound nodes for SM8450
by Krzysztof Kozlowski 28 Nov '22

28 Nov '22

Changes since v1 ================ 1. Patch 2: Keep compatibles in qcom,apr.yaml. 2. Patch 3: New patch. Description =========== Adding sound support for Qualcomm SM8450 SoC (and later for SC8280XP) brought some changes to APR/GPR services bindings. These bindings are part of qcom,apr.yaml: apr-or-gpr-device-node <- qcom,apr.yaml apr-gpr-service@[0-9] <- qcom,apr.yaml service-specific-components <- /schemas/sound/qcom,q6*.yaml The schema for services (apr-gpr-service@[0-9]) was not complete and is still quite not specific. It allows several incorrect combinations, like adding a clock-controller to a APM device. Restricting it would complicate the schema even more. Bringing new support for sound on Qualcomm SM8450 and SC8280XP SoC would grow it as well. Refactor the bindings before extending them for Qualcomm SM8450 SoC. DTS was sent here: https://lore.kernel.org/linux-arm-msm/20221115110053.110319-1-krzysztof.koz… Best regards, Krzysztof --- Cc: quic_srivasam(a)quicinc.com Cc: quic_plai(a)quicinc.com Krzysztof Kozlowski (11): ASoC: dt-bindings: qcom,apr: Add GLINK channel name for SM8450 ASoC: dt-bindings: qcom,apr: Split services to shared schema ASoC: dt-bindings: qcom,apr: Correct and extend example ASoC: dt-bindings: qcom,q6afe: Split to separate schema ASoC: dt-bindings: qcom,q6apm: Split to separate schema ASoC: dt-bindings: qcom,q6adm: Split to separate schema ASoC: dt-bindings: qcom,q6asm: Split to separate schema ASoC: dt-bindings: qcom,q6prm: Split to separate schema ASoC: dt-bindings: qcom,q6core: Split to separate schema ASoC: dt-bindings: qcom,q6apm-lpass-dais: Split to separate schema ASoC: dt-bindings: qcom,q6apm: Add SM8450 bedais node .../bindings/soc/qcom/qcom,apr-services.yaml | 54 ++++++ .../bindings/soc/qcom/qcom,apr.yaml | 170 ++++++------------ .../bindings/sound/qcom,q6adm-routing.yaml | 22 +-- .../devicetree/bindings/sound/qcom,q6adm.yaml | 51 ++++++ .../devicetree/bindings/sound/qcom,q6afe.yaml | 68 +++++++ .../bindings/sound/qcom,q6apm-dai.yaml | 19 +- .../bindings/sound/qcom,q6apm-lpass-dais.yaml | 32 ++++ .../devicetree/bindings/sound/qcom,q6apm.yaml | 67 +++++++ .../bindings/sound/qcom,q6asm-dais.yaml | 48 ++--- .../devicetree/bindings/sound/qcom,q6asm.yaml | 68 +++++++ .../bindings/sound/qcom,q6core.yaml | 39 ++++ .../sound/qcom,q6dsp-lpass-clocks.yaml | 40 +---- .../sound/qcom,q6dsp-lpass-ports.yaml | 57 +----- .../devicetree/bindings/sound/qcom,q6prm.yaml | 50 ++++++ MAINTAINERS | 2 +- 15 files changed, 515 insertions(+), 272 deletions(-) create mode 100644 Documentation/devicetree/bindings/soc/qcom/qcom,apr-services.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6adm.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6afe.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm-lpass-dais.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6apm.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6asm.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6core.yaml create mode 100644 Documentation/devicetree/bindings/sound/qcom,q6prm.yaml -- 2.34.1

3 23

[PATCH v1 0/4] ADD support for reading position updates from
by V sujith kumar Reddy 28 Nov '22

28 Nov '22

This series consists of 1.Reading postion update from stream box 2.selecting clock source as external ACLK clock 3.ADD virtual HS and SP DAI 4.Add AMD virtual DAI configuration support in sof core. V sujith kumar Reddy (4): ASoC: SOF: amd: Fix for reading position updates from stream box. ASoC: SOF: amd: Fix for selecting clock source as external clock. ASoC: SOF: amd: ADD HS and SP virtual DAI. ASoC: SOF: Add DAI configuration support for AMD platforms. include/sound/sof/dai-amd.h | 1 + include/sound/sof/dai.h | 2 ++ include/uapi/sound/sof/tokens.h | 5 +++++ sound/soc/sof/amd/acp-common.c | 1 + sound/soc/sof/amd/acp-ipc.c | 30 ++++++++++++++++++++++++++- sound/soc/sof/amd/acp.c | 4 +++- sound/soc/sof/amd/acp.h | 12 +++++++++++ sound/soc/sof/amd/rembrandt.c | 16 +++++++++++++++ sound/soc/sof/amd/renoir.c | 16 +++++++++++++++ sound/soc/sof/ipc3-pcm.c | 2 ++ sound/soc/sof/ipc3-topology.c | 36 +++++++++++++++++++++++++-------- sound/soc/sof/sof-audio.h | 1 + sound/soc/sof/topology.c | 10 +++++++++ 13 files changed, 126 insertions(+), 10 deletions(-) -- 2.25.1

2 6

[PATCH] ASoC: max9867: Implement exact integer mode
by Pavel Dobias 28 Nov '22

28 Nov '22

For 8kHz and 16kHz sample rates and certain PCLK values the codec can be programmed to operate in exact integer mode. If available, use it to achieve the exact sample rate. Signed-off-by: Pavel Dobias <dobias(a)2n.com> --- sound/soc/codecs/max9867.c | 33 ++++++++++++++++++++++++++++++--- 1 file changed, 30 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 6d2941a9dbd6..e161ab037bf7 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -323,7 +323,7 @@ static int max9867_startup(struct snd_pcm_substream *substream, static int max9867_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - int value; + int value, freq = 0; unsigned long int rate, ratio; struct snd_soc_component *component = dai->component; struct max9867_priv *max9867 = snd_soc_component_get_drvdata(component); @@ -373,6 +373,35 @@ static int max9867_dai_hw_params(struct snd_pcm_substream *substream, } regmap_update_bits(max9867->regmap, MAX9867_IFC1B, MAX9867_IFC1B_BCLK_MASK, value); + + /* Exact integer mode available for 8kHz and 16kHz sample rates + * and certain PCLK (prescaled MCLK) values. + */ + if (params_rate(params) == 8000 || + params_rate(params) == 16000) { + switch (max9867->pclk) { + case 12000000: + freq = 0x08; + break; + case 13000000: + freq = 0x0A; + break; + case 16000000: + freq = 0x0C; + break; + case 19200000: + freq = 0x0E; + break; + } + } + if (freq && params_rate(params) == 16000) + freq++; + + /* If exact integer mode not available, the freq value + * remains zero, i.e. normal mode is used. + */ + regmap_update_bits(max9867->regmap, MAX9867_SYSCLK, + MAX9867_FREQ_MASK, freq); } else { /* * digital pll locks on to any externally supplied LRCLK signal @@ -428,8 +457,6 @@ static int max9867_set_dai_sysclk(struct snd_soc_dai *codec_dai, freq); max9867->sysclk = freq; value = value << MAX9867_PSCLK_SHIFT; - /* exact integer mode is not supported */ - value &= ~MAX9867_FREQ_MASK; regmap_update_bits(max9867->regmap, MAX9867_SYSCLK, MAX9867_PSCLK_MASK, value); return 0; -- 2.20.1

2 1

[PATCH] ALSA: pcm: fix tracing reason in hw_ptr_error
by John Keeping 28 Nov '22

28 Nov '22

Strings need to be specially marked in trace events to ensure the content is captured, othewise the trace just shows the value of the pointer. Signed-off-by: John Keeping <john(a)metanate.com> --- sound/core/pcm_trace.h | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/pcm_trace.h b/sound/core/pcm_trace.h index f18da2050772..350b40b906ca 100644 --- a/sound/core/pcm_trace.h +++ b/sound/core/pcm_trace.h @@ -88,19 +88,19 @@ TRACE_EVENT(hw_ptr_error, __field( unsigned int, device ) __field( unsigned int, number ) __field( unsigned int, stream ) - __field( const char *, reason ) + __string( reason, why ) ), TP_fast_assign( __entry->card = (substream)->pcm->card->number; __entry->device = (substream)->pcm->device; __entry->number = (substream)->number; __entry->stream = (substream)->stream; - __entry->reason = (why); + __assign_str(reason, why); ), TP_printk("pcmC%dD%d%s/sub%d: ERROR: %s", __entry->card, __entry->device, __entry->stream == SNDRV_PCM_STREAM_PLAYBACK ? "p" : "c", - __entry->number, __entry->reason) + __entry->number, __get_str(reason)) ); TRACE_EVENT(applptr, -- 2.38.1

3 3

[PATCH] ALSA: asihpi: fix missing pci_disable_device()
by Liu Shixin 28 Nov '22

28 Nov '22

pci_disable_device() need be called while module exiting, switch to use pcim_enable(), pci_disable_device() will be called in pcim_release(). Fixes: 3285ea10e9b0 ("ALSA: asihpi - Interrelated HPI tidy up.") Signed-off-by: Liu Shixin <liushixin2(a)huawei.com> --- sound/pci/asihpi/hpioctl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index bb31b7fe867d..477a5b4b50bc 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -361,7 +361,7 @@ int asihpi_adapter_probe(struct pci_dev *pci_dev, pci_dev->device, pci_dev->subsystem_vendor, pci_dev->subsystem_device, pci_dev->devfn); - if (pci_enable_device(pci_dev) < 0) { + if (pcim_enable_device(pci_dev) < 0) { dev_err(&pci_dev->dev, "pci_enable_device failed, disabling device\n"); return -EIO; -- 2.25.1

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