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[PATCH v2 0/3] soundwire: cadence: Fix oversized FIFO size define
by Richard Fitzgerald 09 Jan '23

09 Jan '23

As determined by experimentation and asking a hardware person, the FIFO in the Cadence IP is actually only 8 entries long, not 32. This is fixed in patch #1. As a bonus, patches #2 and #3 fix two other things I noticed while debugging this. Changes since v1: - Rewrite commit message of patch #1 - Only reduce response_buf to 34 (32 + 2) - Trim RX_FIFO_AVAIL to length of response_buf instead of expected FIFO size Richard Fitzgerald (3): soundwire: cadence: Don't overflow the command FIFOs soundwire: cadence: Remove wasted space in response_buf soundwire: cadence: Drain the RX FIFO after an IO timeout drivers/soundwire/cadence_master.c | 46 +++++++++++++++++++----------- drivers/soundwire/cadence_master.h | 13 ++++++++- 2 files changed, 41 insertions(+), 18 deletions(-) -- 2.30.2

3 5

[PATCH 0/3] ALSA: firewire-lib: keep history to process isochronous packet
by Takashi Sakamoto 09 Jan '23

09 Jan '23

Hi, This patchset is preparation for computation of extra delay in runtime of PCM substream. Current implementation uses list of packet descriptor to process isochronous packets. The packet descriptors are overwritten every time to process packets, while the history of packet descriptors is useful when computing gap between current isochronous cycle and the latest isochronous cycle in which isochronous packet is processed or scheduled. Conveniently, circular linked list was added to Linux kernel v5.19 at a commit 2fbdf45d7d26 ("list: Add list_next_entry_circular() and list_prev_entry_circular()"). This patchset changes data structure from list to the circular linked list for the packet descriptors. Takashi Sakamoto (3): ALSA: firewire-lib: use circular linked list to enumerate packet descriptors ALSA: firewire-lib: use circular linked list for context payload processing layer ALSA: firewire-lib: store history to process isochronous packet sound/firewire/amdtp-am824.c | 18 +++--- sound/firewire/amdtp-stream.c | 61 ++++++++++++------- sound/firewire/amdtp-stream.h | 17 +++++- sound/firewire/digi00x/amdtp-dot.c | 18 +++--- sound/firewire/fireface/amdtp-ff.c | 18 +++--- sound/firewire/motu/amdtp-motu.c | 48 ++++++++------- .../motu/motu-command-dsp-message-parser.c | 11 ++-- .../motu/motu-register-dsp-message-parser.c | 11 ++-- sound/firewire/motu/motu.h | 8 +-- sound/firewire/tascam/amdtp-tascam.c | 18 +++--- 10 files changed, 139 insertions(+), 89 deletions(-) -- 2.37.2

2 4

[PATCH v3] ALSA: control-led: use strscpy in set_led_id()
by Jaroslav Kysela 09 Jan '23

09 Jan '23

The use of strncpy() in the set_led_id() was incorrect. The len variable should use 'min(sizeof(buf2) - 1, count)' expression. Use strscpy() function to simplify things and handle the error gracefully. Fixes: a135dfb5de15 ("ALSA: led control - add sysfs kcontrol LED marking layer") Reported-by: yang.yang29(a)zte.com.cn Link: https://lore.kernel.org/alsa-devel/202301091945513559977@zte.com.cn/ Cc: <stable(a)vger.kernel.org> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> v1..v2: - remove min() expression which strips the last char v2..v3: - BugLink -> Link - add Fixes tag --- sound/core/control_led.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index f975cc85772b..3cadd40100f3 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -530,12 +530,11 @@ static ssize_t set_led_id(struct snd_ctl_led_card *led_card, const char *buf, si bool attach) { char buf2[256], *s, *os; - size_t len = max(sizeof(s) - 1, count); struct snd_ctl_elem_id id; int err; - strncpy(buf2, buf, len); - buf2[len] = '\0'; + if (strscpy(buf2, buf, sizeof(buf2)) < 0) + return -E2BIG; memset(&id, 0, sizeof(id)); id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; s = buf2; -- 2.39.0

2 1

[PATCH] soundwire: intel: remove DAI startup/shutdown
by Bard Liao 09 Jan '23

09 Jan '23

From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> The only thing these DAI startup/shutdown callbacks do is play with pm_runtime reference counts. This is not wrong, but it's not necessary at all. At the ASoC core level, only the component matters for pm_runtime. The ASoC core already calls pm_runtime_get_sync() in snd_soc_pcm_component_pm_runtime_get(), before the DAI startup callback is invoked. None of the SoundWire codec drivers rely on pm_runtime helpers in their DAI startup/shutdown either. This adds to the evidence that only the component, or more precisely the device specified when registering a component, should deal with pm_runtime transitions. Beyond the code cleanup, this move prepares for the addition of link power management in the auxiliary device startup/resume/suspend callbacks. The DAI callbacks can by-design assume that the device is already pm_runtime active. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Reviewed-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Reviewed-by: Rander Wang <rander.wang(a)intel.com> Reviewed-by: Péter Ujfalusi <peter.ujfalusi(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/intel.c | 27 --------------------------- 1 file changed, 27 deletions(-) diff --git a/drivers/soundwire/intel.c b/drivers/soundwire/intel.c index bc9c50bacc49..2651767272c7 100644 --- a/drivers/soundwire/intel.c +++ b/drivers/soundwire/intel.c @@ -778,22 +778,6 @@ static int intel_post_bank_switch(struct sdw_intel *sdw) * DAI routines */ -static int intel_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sdw_cdns *cdns = snd_soc_dai_get_drvdata(dai); - int ret; - - ret = pm_runtime_resume_and_get(cdns->dev); - if (ret < 0 && ret != -EACCES) { - dev_err_ratelimited(cdns->dev, - "pm_runtime_resume_and_get failed in %s, ret %d\n", - __func__, ret); - return ret; - } - return 0; -} - static int intel_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -954,15 +938,6 @@ intel_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) return 0; } -static void intel_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sdw_cdns *cdns = snd_soc_dai_get_drvdata(dai); - - pm_runtime_mark_last_busy(cdns->dev); - pm_runtime_put_autosuspend(cdns->dev); -} - static int intel_pcm_set_sdw_stream(struct snd_soc_dai *dai, void *stream, int direction) { @@ -1088,12 +1063,10 @@ static int intel_component_dais_suspend(struct snd_soc_component *component) } static const struct snd_soc_dai_ops intel_pcm_dai_ops = { - .startup = intel_startup, .hw_params = intel_hw_params, .prepare = intel_prepare, .hw_free = intel_hw_free, .trigger = intel_trigger, - .shutdown = intel_shutdown, .set_stream = intel_pcm_set_sdw_stream, .get_stream = intel_get_sdw_stream, }; -- 2.25.1

2 1

[PATCH v2] ALSA: control-led: use strscpy in set_led_id()
by Jaroslav Kysela 09 Jan '23

09 Jan '23

The use of strncpy() in the set_led_id() was incorrect. The len variable should use 'min(sizeof(buf2) - 1, count)' expression. Use strscpy() function to simplify things and handle the error gracefully. Reported-by: yang.yang29(a)zte.com.cn BugLink: https://lore.kernel.org/alsa-devel/202301091945513559977@zte.com.cn/ Cc: <stable(a)vger.kernel.org> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> v1..v2: - remove min() expression which strips the last char --- sound/core/control_led.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index f975cc85772b..3cadd40100f3 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -530,12 +530,11 @@ static ssize_t set_led_id(struct snd_ctl_led_card *led_card, const char *buf, si bool attach) { char buf2[256], *s, *os; - size_t len = max(sizeof(s) - 1, count); struct snd_ctl_elem_id id; int err; - strncpy(buf2, buf, len); - buf2[len] = '\0'; + if (strscpy(buf2, buf, sizeof(buf2)) < 0) + return -E2BIG; memset(&id, 0, sizeof(id)); id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; s = buf2; -- 2.39.0

2 2

[PATCH] ALSA: usb-audio: Always initialize fixed_rate in snd_usb_find_implicit_fb_sync_format()
by Jaroslav Kysela 09 Jan '23

09 Jan '23

Handle the fallback code path, too. Fixes: fd28941cff1c ("ALSA: usb-audio: Add new quirk FIXED_RATE for JBL Quantum810 Wireless") BugLink: https://lore.kernel.org/alsa-devel/Y7frf3N%2FxzvESEsN@kili/ Reported-by: Dan Carpenter <error27(a)gmail.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/usb/implicit.c | 3 ++- sound/usb/pcm.c | 2 ++ 2 files changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/usb/implicit.c b/sound/usb/implicit.c index 41ac7185b42b..4727043fd745 100644 --- a/sound/usb/implicit.c +++ b/sound/usb/implicit.c @@ -471,7 +471,7 @@ snd_usb_find_implicit_fb_sync_format(struct snd_usb_audio *chip, subs = find_matching_substream(chip, stream, target->sync_ep, target->fmt_type); if (!subs) - return sync_fmt; + goto end; high_score = 0; list_for_each_entry(fp, &subs->fmt_list, list) { @@ -485,6 +485,7 @@ snd_usb_find_implicit_fb_sync_format(struct snd_usb_audio *chip, } } + end: if (fixed_rate) *fixed_rate = snd_usb_pcm_has_fixed_rate(subs); return sync_fmt; diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 99a66d0ef5b2..d2c652aa1385 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -163,6 +163,8 @@ bool snd_usb_pcm_has_fixed_rate(struct snd_usb_substream *subs) struct snd_usb_audio *chip = subs->stream->chip; int rate = -1; + if (!subs) + return false; if (!(chip->quirk_flags & QUIRK_FLAG_FIXED_RATE)) return false; list_for_each_entry(fp, &subs->fmt_list, list) { -- 2.39.0

2 1

[PATCH] ALSA: control-led: use strscpy in set_led_id()
by Jaroslav Kysela 09 Jan '23

09 Jan '23

The use of strncpy() in the set_led_id() was incorrect. The len variable should use 'min(sizeof(buf2) - 1, count)' expression. Use strscpy() function to simplify things and handle the error gracefully. Reported-by: yang.yang29(a)zte.com.cn BugLink: https://lore.kernel.org/alsa-devel/202301091945513559977@zte.com.cn/ Cc: <stable(a)vger.kernel.org> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/core/control_led.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/core/control_led.c b/sound/core/control_led.c index f975cc85772b..45c8eb5700c1 100644 --- a/sound/core/control_led.c +++ b/sound/core/control_led.c @@ -530,12 +530,11 @@ static ssize_t set_led_id(struct snd_ctl_led_card *led_card, const char *buf, si bool attach) { char buf2[256], *s, *os; - size_t len = max(sizeof(s) - 1, count); struct snd_ctl_elem_id id; int err; - strncpy(buf2, buf, len); - buf2[len] = '\0'; + if (strscpy(buf2, buf, min(sizeof(buf2), count)) < 0) + return -E2BIG; memset(&id, 0, sizeof(id)); id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; s = buf2; -- 2.39.0

1 1

[PATCH 1/2] ASoC: dt-bindings: qcom, lpass-wsa-macro: correct clocks on SM8250
by Krzysztof Kozlowski 09 Jan '23

09 Jan '23

SM8250 DTS uses additional "va" clock in WSA macro device node: sm8250-sony-xperia-edo-pdx203.dtb: codec@3240000: clock-names: ['mclk', 'npl', 'macro', 'dcodec', 'va', 'fsgen'] is too long Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- .../bindings/sound/qcom,lpass-wsa-macro.yaml | 58 +++++++++++++++---- 1 file changed, 47 insertions(+), 11 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml index 2bf8d082f8f1..66cbb1f5e31a 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-wsa-macro.yaml @@ -9,9 +9,6 @@ title: LPASS(Low Power Audio Subsystem) VA Macro audio codec maintainers: - Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> -allOf: - - $ref: dai-common.yaml# - properties: compatible: enum: @@ -30,15 +27,12 @@ properties: const: 0 clocks: - maxItems: 5 + minItems: 5 + maxItems: 6 clock-names: - items: - - const: mclk - - const: npl - - const: macro - - const: dcodec - - const: fsgen + minItems: 5 + maxItems: 6 clock-output-names: maxItems: 1 @@ -55,10 +49,51 @@ required: - reg - "#sound-dai-cells" +allOf: + - $ref: dai-common.yaml# + + - if: + properties: + compatible: + enum: + - qcom,sc7280-lpass-wsa-macro + - qcom,sm8450-lpass-wsa-macro + - qcom,sc8280xp-lpass-wsa-macro + then: + properties: + clocks: + maxItems: 5 + clock-names: + items: + - const: mclk + - const: npl + - const: macro + - const: dcodec + - const: fsgen + + - if: + properties: + compatible: + enum: + - qcom,sm8250-lpass-wsa-macro + then: + properties: + clocks: + minItems: 6 + clock-names: + items: + - const: mclk + - const: npl + - const: macro + - const: dcodec + - const: va + - const: fsgen + unevaluatedProperties: false examples: - | + #include <dt-bindings/clock/qcom,sm8250-lpass-aoncc.h> #include <dt-bindings/sound/qcom,q6afe.h> codec@3240000 { compatible = "qcom,sm8250-lpass-wsa-macro"; @@ -69,7 +104,8 @@ examples: <&audiocc 0>, <&q6afecc LPASS_HW_MACRO_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, <&q6afecc LPASS_HW_DCODEC_VOTE LPASS_CLK_ATTRIBUTE_COUPLE_NO>, + <&aoncc LPASS_CDC_VA_MCLK>, <&vamacro>; - clock-names = "mclk", "npl", "macro", "dcodec", "fsgen"; + clock-names = "mclk", "npl", "macro", "dcodec", "va", "fsgen"; clock-output-names = "mclk"; }; -- 2.34.1

3 4

[PATCH] ASoC: qcom: Fix building APQ8016 machine driver without SOUNDWIRE
by Stephan Gerhold 09 Jan '23

09 Jan '23

Older Qualcomm platforms like APQ8016 do not have hardware support for SoundWire, so kernel configurations made specifically for those platforms will usually not have CONFIG_SOUNDWIRE enabled. Unfortunately commit 8d89cf6ff229 ("ASoC: qcom: cleanup and fix dependency of QCOM_COMMON") breaks those kernel configurations, because SOUNDWIRE is now a required dependency for SND_SOC_QCOM_COMMON (and in turn also SND_SOC_APQ8016_SBC). Trying to migrate such a kernel config silently disables SND_SOC_APQ8016_SBC and breaks audio functionality. The soundwire helpers in common.c are only used by two of the Qualcomm audio machine drivers, so building and requiring CONFIG_SOUNDWIRE for all platforms is unnecessary. There is no need to stuff all common code into a single module. Fix the issue by moving the soundwire helpers to a separate SND_SOC_QCOM_SDW module/option that is selected only by the machine drivers that make use of them. This also allows reverting the imply/depends changes from the previous fix because both SM8250 and SC8280XP already depend on SOUNDWIRE, so the soundwire helpers will be only built if SOUNDWIRE is really enabled. Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Fixes: 8d89cf6ff229 ("ASoC: qcom: cleanup and fix dependency of QCOM_COMMON") Signed-off-by: Stephan Gerhold <stephan(a)gerhold.net> --- sound/soc/qcom/Kconfig | 21 ++++--- sound/soc/qcom/Makefile | 2 + sound/soc/qcom/common.c | 114 ----------------------------------- sound/soc/qcom/common.h | 10 ---- sound/soc/qcom/sc8280xp.c | 1 + sound/soc/qcom/sdw.c | 123 ++++++++++++++++++++++++++++++++++++++ sound/soc/qcom/sdw.h | 18 ++++++ sound/soc/qcom/sm8250.c | 1 + 8 files changed, 157 insertions(+), 133 deletions(-) create mode 100644 sound/soc/qcom/sdw.c create mode 100644 sound/soc/qcom/sdw.h diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 96a6d4731e6f..e7b00d1d9e99 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -2,7 +2,6 @@ menuconfig SND_SOC_QCOM tristate "ASoC support for QCOM platforms" depends on ARCH_QCOM || COMPILE_TEST - imply SND_SOC_QCOM_COMMON help Say Y or M if you want to add support to use audio devices in Qualcomm Technologies SOC-based platforms. @@ -60,14 +59,16 @@ config SND_SOC_STORM config SND_SOC_APQ8016_SBC tristate "SoC Audio support for APQ8016 SBC platforms" select SND_SOC_LPASS_APQ8016 - depends on SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_COMMON help Support for Qualcomm Technologies LPASS audio block in APQ8016 SOC-based systems. Say Y if you want to use audio devices on MI2S. config SND_SOC_QCOM_COMMON - depends on SOUNDWIRE + tristate + +config SND_SOC_QCOM_SDW tristate config SND_SOC_QDSP6_COMMON @@ -144,7 +145,7 @@ config SND_SOC_MSM8996 depends on QCOM_APR depends on COMMON_CLK select SND_SOC_QDSP6 - depends on SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_COMMON help Support for Qualcomm Technologies LPASS audio block in APQ8096 SoC-based systems. @@ -155,7 +156,7 @@ config SND_SOC_SDM845 depends on QCOM_APR && I2C && SOUNDWIRE depends on COMMON_CLK select SND_SOC_QDSP6 - depends on SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_COMMON select SND_SOC_RT5663 select SND_SOC_MAX98927 imply SND_SOC_CROS_EC_CODEC @@ -169,7 +170,8 @@ config SND_SOC_SM8250 depends on QCOM_APR && SOUNDWIRE depends on COMMON_CLK select SND_SOC_QDSP6 - depends on SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_SDW help To add support for audio on Qualcomm Technologies Inc. SM8250 SoC-based systems. @@ -180,7 +182,8 @@ config SND_SOC_SC8280XP depends on QCOM_APR && SOUNDWIRE depends on COMMON_CLK select SND_SOC_QDSP6 - depends on SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_SDW help To add support for audio on Qualcomm Technologies Inc. SC8280XP SoC-based systems. @@ -190,7 +193,7 @@ config SND_SOC_SC7180 tristate "SoC Machine driver for SC7180 boards" depends on I2C && GPIOLIB depends on SOUNDWIRE || SOUNDWIRE=n - depends on SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_COMMON select SND_SOC_LPASS_SC7180 select SND_SOC_MAX98357A select SND_SOC_RT5682_I2C @@ -204,7 +207,7 @@ config SND_SOC_SC7180 config SND_SOC_SC7280 tristate "SoC Machine driver for SC7280 boards" depends on I2C && SOUNDWIRE - depends on SND_SOC_QCOM_COMMON + select SND_SOC_QCOM_COMMON select SND_SOC_LPASS_SC7280 select SND_SOC_MAX98357A select SND_SOC_WCD938X_SDW diff --git a/sound/soc/qcom/Makefile b/sound/soc/qcom/Makefile index 8b97172cf990..254350d9dc06 100644 --- a/sound/soc/qcom/Makefile +++ b/sound/soc/qcom/Makefile @@ -28,6 +28,7 @@ snd-soc-sdm845-objs := sdm845.o snd-soc-sm8250-objs := sm8250.o snd-soc-sc8280xp-objs := sc8280xp.o snd-soc-qcom-common-objs := common.o +snd-soc-qcom-sdw-objs := sdw.o obj-$(CONFIG_SND_SOC_STORM) += snd-soc-storm.o obj-$(CONFIG_SND_SOC_APQ8016_SBC) += snd-soc-apq8016-sbc.o @@ -38,6 +39,7 @@ obj-$(CONFIG_SND_SOC_SC8280XP) += snd-soc-sc8280xp.o obj-$(CONFIG_SND_SOC_SDM845) += snd-soc-sdm845.o obj-$(CONFIG_SND_SOC_SM8250) += snd-soc-sm8250.o obj-$(CONFIG_SND_SOC_QCOM_COMMON) += snd-soc-qcom-common.o +obj-$(CONFIG_SND_SOC_QCOM_SDW) += snd-soc-qcom-sdw.o #DSP lib obj-$(CONFIG_SND_SOC_QDSP6) += qdsp6/ diff --git a/sound/soc/qcom/common.c b/sound/soc/qcom/common.c index 49c74c1662a3..96fe80241fb4 100644 --- a/sound/soc/qcom/common.c +++ b/sound/soc/qcom/common.c @@ -180,120 +180,6 @@ int qcom_snd_parse_of(struct snd_soc_card *card) } EXPORT_SYMBOL_GPL(qcom_snd_parse_of); -int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream, - struct sdw_stream_runtime *sruntime, - bool *stream_prepared) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - int ret; - - if (!sruntime) - return 0; - - switch (cpu_dai->id) { - case WSA_CODEC_DMA_RX_0: - case WSA_CODEC_DMA_RX_1: - case RX_CODEC_DMA_RX_0: - case RX_CODEC_DMA_RX_1: - case TX_CODEC_DMA_TX_0: - case TX_CODEC_DMA_TX_1: - case TX_CODEC_DMA_TX_2: - case TX_CODEC_DMA_TX_3: - break; - default: - return 0; - } - - if (*stream_prepared) { - sdw_disable_stream(sruntime); - sdw_deprepare_stream(sruntime); - *stream_prepared = false; - } - - ret = sdw_prepare_stream(sruntime); - if (ret) - return ret; - - /** - * NOTE: there is a strict hw requirement about the ordering of port - * enables and actual WSA881x PA enable. PA enable should only happen - * after soundwire ports are enabled if not DC on the line is - * accumulated resulting in Click/Pop Noise - * PA enable/mute are handled as part of codec DAPM and digital mute. - */ - - ret = sdw_enable_stream(sruntime); - if (ret) { - sdw_deprepare_stream(sruntime); - return ret; - } - *stream_prepared = true; - - return ret; -} -EXPORT_SYMBOL_GPL(qcom_snd_sdw_prepare); - -int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct sdw_stream_runtime **psruntime) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai; - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - struct sdw_stream_runtime *sruntime; - int i; - - switch (cpu_dai->id) { - case WSA_CODEC_DMA_RX_0: - case RX_CODEC_DMA_RX_0: - case RX_CODEC_DMA_RX_1: - case TX_CODEC_DMA_TX_0: - case TX_CODEC_DMA_TX_1: - case TX_CODEC_DMA_TX_2: - case TX_CODEC_DMA_TX_3: - for_each_rtd_codec_dais(rtd, i, codec_dai) { - sruntime = snd_soc_dai_get_stream(codec_dai, substream->stream); - if (sruntime != ERR_PTR(-ENOTSUPP)) - *psruntime = sruntime; - } - break; - } - - return 0; - -} -EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_params); - -int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream, - struct sdw_stream_runtime *sruntime, bool *stream_prepared) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); - - switch (cpu_dai->id) { - case WSA_CODEC_DMA_RX_0: - case WSA_CODEC_DMA_RX_1: - case RX_CODEC_DMA_RX_0: - case RX_CODEC_DMA_RX_1: - case TX_CODEC_DMA_TX_0: - case TX_CODEC_DMA_TX_1: - case TX_CODEC_DMA_TX_2: - case TX_CODEC_DMA_TX_3: - if (sruntime && *stream_prepared) { - sdw_disable_stream(sruntime); - sdw_deprepare_stream(sruntime); - *stream_prepared = false; - } - break; - default: - break; - } - - return 0; -} -EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_free); - int qcom_snd_wcd_jack_setup(struct snd_soc_pcm_runtime *rtd, struct snd_soc_jack *jack, bool *jack_setup) { diff --git a/sound/soc/qcom/common.h b/sound/soc/qcom/common.h index 3ef5bb6d12df..d7f80ee5ae26 100644 --- a/sound/soc/qcom/common.h +++ b/sound/soc/qcom/common.h @@ -5,19 +5,9 @@ #define __QCOM_SND_COMMON_H__ #include <sound/soc.h> -#include <linux/soundwire/sdw.h> int qcom_snd_parse_of(struct snd_soc_card *card); int qcom_snd_wcd_jack_setup(struct snd_soc_pcm_runtime *rtd, struct snd_soc_jack *jack, bool *jack_setup); -int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream, - struct sdw_stream_runtime *runtime, - bool *stream_prepared); -int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct sdw_stream_runtime **psruntime); -int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream, - struct sdw_stream_runtime *sruntime, - bool *stream_prepared); #endif diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c index ade44ad7c585..14d9fea33d16 100644 --- a/sound/soc/qcom/sc8280xp.c +++ b/sound/soc/qcom/sc8280xp.c @@ -12,6 +12,7 @@ #include <linux/input-event-codes.h> #include "qdsp6/q6afe.h" #include "common.h" +#include "sdw.h" #define DRIVER_NAME "sc8280xp" diff --git a/sound/soc/qcom/sdw.c b/sound/soc/qcom/sdw.c new file mode 100644 index 000000000000..10249519a39e --- /dev/null +++ b/sound/soc/qcom/sdw.c @@ -0,0 +1,123 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright (c) 2018, Linaro Limited. +// Copyright (c) 2018, The Linux Foundation. All rights reserved. + +#include <linux/module.h> +#include <sound/soc.h> +#include "qdsp6/q6afe.h" +#include "sdw.h" + +int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream, + struct sdw_stream_runtime *sruntime, + bool *stream_prepared) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + int ret; + + if (!sruntime) + return 0; + + switch (cpu_dai->id) { + case WSA_CODEC_DMA_RX_0: + case WSA_CODEC_DMA_RX_1: + case RX_CODEC_DMA_RX_0: + case RX_CODEC_DMA_RX_1: + case TX_CODEC_DMA_TX_0: + case TX_CODEC_DMA_TX_1: + case TX_CODEC_DMA_TX_2: + case TX_CODEC_DMA_TX_3: + break; + default: + return 0; + } + + if (*stream_prepared) { + sdw_disable_stream(sruntime); + sdw_deprepare_stream(sruntime); + *stream_prepared = false; + } + + ret = sdw_prepare_stream(sruntime); + if (ret) + return ret; + + /** + * NOTE: there is a strict hw requirement about the ordering of port + * enables and actual WSA881x PA enable. PA enable should only happen + * after soundwire ports are enabled if not DC on the line is + * accumulated resulting in Click/Pop Noise + * PA enable/mute are handled as part of codec DAPM and digital mute. + */ + + ret = sdw_enable_stream(sruntime); + if (ret) { + sdw_deprepare_stream(sruntime); + return ret; + } + *stream_prepared = true; + + return ret; +} +EXPORT_SYMBOL_GPL(qcom_snd_sdw_prepare); + +int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct sdw_stream_runtime **psruntime) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + struct sdw_stream_runtime *sruntime; + int i; + + switch (cpu_dai->id) { + case WSA_CODEC_DMA_RX_0: + case RX_CODEC_DMA_RX_0: + case RX_CODEC_DMA_RX_1: + case TX_CODEC_DMA_TX_0: + case TX_CODEC_DMA_TX_1: + case TX_CODEC_DMA_TX_2: + case TX_CODEC_DMA_TX_3: + for_each_rtd_codec_dais(rtd, i, codec_dai) { + sruntime = snd_soc_dai_get_stream(codec_dai, substream->stream); + if (sruntime != ERR_PTR(-ENOTSUPP)) + *psruntime = sruntime; + } + break; + } + + return 0; + +} +EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_params); + +int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream, + struct sdw_stream_runtime *sruntime, bool *stream_prepared) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0); + + switch (cpu_dai->id) { + case WSA_CODEC_DMA_RX_0: + case WSA_CODEC_DMA_RX_1: + case RX_CODEC_DMA_RX_0: + case RX_CODEC_DMA_RX_1: + case TX_CODEC_DMA_TX_0: + case TX_CODEC_DMA_TX_1: + case TX_CODEC_DMA_TX_2: + case TX_CODEC_DMA_TX_3: + if (sruntime && *stream_prepared) { + sdw_disable_stream(sruntime); + sdw_deprepare_stream(sruntime); + *stream_prepared = false; + } + break; + default: + break; + } + + return 0; +} +EXPORT_SYMBOL_GPL(qcom_snd_sdw_hw_free); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/qcom/sdw.h b/sound/soc/qcom/sdw.h new file mode 100644 index 000000000000..d74cbb84da13 --- /dev/null +++ b/sound/soc/qcom/sdw.h @@ -0,0 +1,18 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// Copyright (c) 2018, The Linux Foundation. All rights reserved. + +#ifndef __QCOM_SND_SDW_H__ +#define __QCOM_SND_SDW_H__ + +#include <linux/soundwire/sdw.h> + +int qcom_snd_sdw_prepare(struct snd_pcm_substream *substream, + struct sdw_stream_runtime *runtime, + bool *stream_prepared); +int qcom_snd_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct sdw_stream_runtime **psruntime); +int qcom_snd_sdw_hw_free(struct snd_pcm_substream *substream, + struct sdw_stream_runtime *sruntime, + bool *stream_prepared); +#endif diff --git a/sound/soc/qcom/sm8250.c b/sound/soc/qcom/sm8250.c index 8dbe9ef41b1c..9626a9ef78c2 100644 --- a/sound/soc/qcom/sm8250.c +++ b/sound/soc/qcom/sm8250.c @@ -12,6 +12,7 @@ #include <linux/input-event-codes.h> #include "qdsp6/q6afe.h" #include "common.h" +#include "sdw.h" #define DRIVER_NAME "sm8250" #define MI2S_BCLK_RATE 1536000 -- 2.39.0

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[bug report] ALSA: usb-audio: Add new quirk FIXED_RATE for JBL Quantum810 Wireless
by Dan Carpenter 09 Jan '23

09 Jan '23

Hello Jaroslav Kysela, The patch fd28941cff1c: "ALSA: usb-audio: Add new quirk FIXED_RATE for JBL Quantum810 Wireless" from Dec 15, 2022, leads to the following Smatch static checker warning: sound/usb/pcm.c:543 snd_usb_hw_params() error: uninitialized symbol 'sync_fixed_rate'. sound/usb/pcm.c 471 static int snd_usb_hw_params(struct snd_pcm_substream *substream, 472 struct snd_pcm_hw_params *hw_params) 473 { 474 struct snd_usb_substream *subs = substream->runtime->private_data; 475 struct snd_usb_audio *chip = subs->stream->chip; 476 const struct audioformat *fmt; 477 const struct audioformat *sync_fmt; 478 bool fixed_rate, sync_fixed_rate; 479 int ret; 480 481 ret = snd_media_start_pipeline(subs); 482 if (ret) 483 return ret; 484 485 fixed_rate = snd_usb_pcm_has_fixed_rate(subs); 486 fmt = find_substream_format(subs, hw_params); 487 if (!fmt) { 488 usb_audio_dbg(chip, 489 "cannot find format: format=%s, rate=%d, channels=%d\n", 490 snd_pcm_format_name(params_format(hw_params)), 491 params_rate(hw_params), params_channels(hw_params)); 492 ret = -EINVAL; 493 goto stop_pipeline; 494 } 495 496 if (fmt->implicit_fb) { 497 sync_fmt = snd_usb_find_implicit_fb_sync_format(chip, fmt, 498 hw_params, 499 !substream->stream, 500 &sync_fixed_rate); "sync_fixed_rate" is uninitialized when we "Use the original audioformat as fallback for the shared altset" and find_matching_substream() returns NULL. 501 if (!sync_fmt) { 502 usb_audio_dbg(chip, 503 "cannot find sync format: ep=0x%x, iface=%d:%d, format=%s, rate=%d, channels=%d\n", 504 fmt->sync_ep, fmt->sync_iface, 505 fmt->sync_altsetting, 506 snd_pcm_format_name(params_format(hw_params)), 507 params_rate(hw_params), params_channels(hw_params)); 508 ret = -EINVAL; 509 goto stop_pipeline; 510 } 511 } else { 512 sync_fmt = fmt; 513 sync_fixed_rate = fixed_rate; 514 } 515 516 ret = snd_usb_lock_shutdown(chip); 517 if (ret < 0) 518 goto stop_pipeline; 519 520 ret = snd_usb_pcm_change_state(subs, UAC3_PD_STATE_D0); 521 if (ret < 0) 522 goto unlock; 523 524 if (subs->data_endpoint) { 525 if (snd_usb_endpoint_compatible(chip, subs->data_endpoint, 526 fmt, hw_params)) 527 goto unlock; 528 if (stop_endpoints(subs, false)) 529 sync_pending_stops(subs); 530 close_endpoints(chip, subs); 531 } 532 533 subs->data_endpoint = snd_usb_endpoint_open(chip, fmt, hw_params, false, fixed_rate); 534 if (!subs->data_endpoint) { 535 ret = -EINVAL; 536 goto unlock; 537 } 538 539 if (fmt->sync_ep) { 540 subs->sync_endpoint = snd_usb_endpoint_open(chip, sync_fmt, 541 hw_params, 542 fmt == sync_fmt, --> 543 sync_fixed_rate); 544 if (!subs->sync_endpoint) { 545 ret = -EINVAL; 546 goto unlock; 547 } regards, dan carpenter

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