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[PATCH] ALSA: aloop: Fix random zeros in capture data when using jiffies timer
by Pattara Teerapong 02 Sep '22

02 Sep '22

In loopback_jiffies_timer_pos_update(), we are getting jiffies twice. First time for playback, second time for capture. Jiffies can be updated between these two calls and if the capture jiffies is larger, extra zeros will be filled in the capture buffer. Change to get jiffies once and use it for both playback and capture. Signed-off-by: Pattara Teerapong <pteerapong(a)chromium.org> --- sound/drivers/aloop.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 9b4a7cdb103a..12f12a294df5 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -605,17 +605,18 @@ static unsigned int loopback_jiffies_timer_pos_update cable->streams[SNDRV_PCM_STREAM_PLAYBACK]; struct loopback_pcm *dpcm_capt = cable->streams[SNDRV_PCM_STREAM_CAPTURE]; - unsigned long delta_play = 0, delta_capt = 0; + unsigned long delta_play = 0, delta_capt = 0, cur_jiffies; unsigned int running, count1, count2; + cur_jiffies = jiffies; running = cable->running ^ cable->pause; if (running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) { - delta_play = jiffies - dpcm_play->last_jiffies; + delta_play = cur_jiffies - dpcm_play->last_jiffies; dpcm_play->last_jiffies += delta_play; } if (running & (1 << SNDRV_PCM_STREAM_CAPTURE)) { - delta_capt = jiffies - dpcm_capt->last_jiffies; + delta_capt = cur_jiffies - dpcm_capt->last_jiffies; dpcm_capt->last_jiffies += delta_capt; } -- 2.37.2.789.g6183377224-goog

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[PATCH] ASoC: rsnd: Add check for rsnd_mod_power_on
by Jiasheng Jiang 02 Sep '22

02 Sep '22

As rsnd_mod_power_on() can return negative numbers, it should be better to check the return value and deal with the exception. Fixes: e7d850dd10f4 ("ASoC: rsnd: use mod base common method on SSI-parent") Signed-off-by: Jiasheng Jiang <jiasheng(a)iscas.ac.cn> --- sound/soc/sh/rcar/ssi.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 43c5e27dc5c8..7ade6c5ed96f 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -480,7 +480,9 @@ static int rsnd_ssi_init(struct rsnd_mod *mod, ssi->usrcnt++; - rsnd_mod_power_on(mod); + ret = rsnd_mod_power_on(mod); + if (ret < 0) + return ret; rsnd_ssi_config_init(mod, io); -- 2.25.1

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[PATCH] ALSA: usb-audio: Clean up endpoint setups at PCM prepare
by Takashi Iwai 01 Sep '22

01 Sep '22

This patch cleans up the superfluous checks and calls for setting up the endpoints at PCM prepare callback: - Drop stop_endpoints() and sync_pending_stops() calls; the stream is guaranteed to have been already stopped and synced at each PCM prepare call by ALSA PCM core - Call snd_usb_endpoint_prepare() unconditionally; the check for endpoint->need_setup is done in snd_pcm_hw_endpoint_prepare() itself - Apply snd_usb_set_format_quirk() only when the endpoint is actually set up (i.e. the return code from snd_usb_endpoint_prepare() > 0) - Move a few lines back into snd_usb_pcm_prepare(); it's even easier to follow than a small useless function Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 40 ++++++++++------------------------------ 1 file changed, 10 insertions(+), 30 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index b604f7e95e82..4ed53a3dc922 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -433,35 +433,6 @@ static void close_endpoints(struct snd_usb_audio *chip, } } -static int configure_endpoints(struct snd_usb_audio *chip, - struct snd_usb_substream *subs) -{ - int err; - - if (subs->data_endpoint->need_setup) { - /* stop any running stream beforehand */ - if (stop_endpoints(subs, false)) - sync_pending_stops(subs); - if (subs->sync_endpoint) { - err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); - if (err < 0) - return err; - } - err = snd_usb_endpoint_prepare(chip, subs->data_endpoint); - if (err < 0) - return err; - snd_usb_set_format_quirk(subs, subs->cur_audiofmt); - } else { - if (subs->sync_endpoint) { - err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); - if (err < 0) - return err; - } - } - - return 0; -} - /* * hw_params callback * @@ -640,9 +611,18 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } - ret = configure_endpoints(chip, subs); + if (subs->sync_endpoint) { + ret = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); + if (ret < 0) + goto unlock; + } + + ret = snd_usb_endpoint_prepare(chip, subs->data_endpoint); if (ret < 0) goto unlock; + else if (ret > 0) + snd_usb_set_format_quirk(subs, subs->cur_audiofmt); + ret = 0; /* reset the pointer */ subs->buffer_bytes = frames_to_bytes(runtime, runtime->buffer_size); -- 2.35.3

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[PATCH 1/2] ASoC: apple: mca: Unselect COMMON_CLK in Kconfig
by Martin Povišer 01 Sep '22

01 Sep '22

The MCA driver selects COMMON_CLK, which leads to misconfiguration on platforms with HAVE_LEGACY_CLK (under compile test). Kconfig catches the conflict with the following warning: WARNING: unmet direct dependencies detected for COMMON_CLK Depends on [n]: !HAVE_LEGACY_CLK [=y] Selected by [m]: - SND_SOC_APPLE_MCA [=m] && SOUND [=m] && !UML && SND [=m] && SND_SOC [=m] && (ARCH_APPLE || COMPILE_TEST [=y]) Eventually the build fails with errors like: >> drivers/clk/clk.c:867:6: error: redefinition of 'clk_unprepare' 867 | void clk_unprepare(struct clk *clk) | ^~~~~~~~~~~~~ In file included from drivers/clk/clk.c:9: include/linux/clk.h:303:20: note: previous definition of 'clk_unprepare' with type 'void(struct clk *)' 303 | static inline void clk_unprepare(struct clk *clk) | ^~~~~~~~~~~~~ which appears to be because COMMON_CLK is selected but HAVE_CLK_PREPARE is not. In the end it seems we had no business selecting COMMON_CLK from an unrelated driver like that, so remove the selection. The linux/clk.h API is there anyway. Fixes: 3df5d0d97289 ("ASoC: apple: mca: Start new platform driver") Reported-by: kernel test robot <lkp(a)intel.com> Signed-off-by: Martin Povišer <povik+lin(a)cutebit.org> --- sound/soc/apple/Kconfig | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/apple/Kconfig b/sound/soc/apple/Kconfig index 0ba955657e98..793f7782e0d7 100644 --- a/sound/soc/apple/Kconfig +++ b/sound/soc/apple/Kconfig @@ -2,7 +2,6 @@ config SND_SOC_APPLE_MCA tristate "Apple Silicon MCA driver" depends on ARCH_APPLE || COMPILE_TEST select SND_DMAENGINE_PCM - select COMMON_CLK default ARCH_APPLE help This option enables an ASoC platform driver for MCA peripherals found -- 2.33.0

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[PATCH] ALSA: usb-audio: Split endpoint setups for hw_params and prepare
by Takashi Iwai 01 Sep '22

01 Sep '22

One of the former changes for the endpoint management was the more consistent setup of endpoints at hw_params. snd_usb_endpoint_configure() is a single function that does the full setup, and it's called from both PCM hw_params and prepare callbacks. Although the EP setup at the prepare phase is usually skipped (by checking need_setup flag), it may be still effective in some cases like suspend/resume that requires the interface setup again. As it's a full and single setup, the invocation of snd_usb_endpoint_configure() includes not only the USB interface setup but also the buffer release and allocation. OTOH, doing the buffer release and re-allocation at PCM prepare phase is rather superfluous, and better to be done only in the hw_params phase. For those optimizations, this patch splits the endpoint setup to two phases: snd_usb_endpoint_set_params() and snd_usb_endpoint_prepare(), to be called from hw_params and from prepare, respectively. Note that this patch changes the driver operation slightly, effectively moving the USB interface setup again to PCM prepare stage instead of hw_params stage, while the buffer allocation and such initializations are still done at hw_params stage. And, the change of the USB interface setup timing (moving to prepare) gave an interesting "fix", too: it was reported that the recent kernels caused silent output at the beginning on playbacks on some devices on Android, and this change casually fixed the regression. It seems that those devices are picky about the sample rate change (or the interface change?), and don't follow the too immediate rate changes. Meanwhile, Android operates the PCM in the following order: - open, then hw_params with the possibly highest sample rate - close without prepare - re-open, hw_params with the normal sample rate - prepare, and start streaming This procedure ended up the hw_params twice with different rates, and because the recent kernel did set up the sample rate twice one and after, it screwed up the device. OTOH, the earlier kernels didn't set up the USB interface at hw_params, hence this problem didn't appear. Now, with this patch, the USB interface setup is again back to the prepare phase, and it works around the problem automagically. Although we should address the sample rate problem in a more solid way in future, let's keep things working as before for now. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Cc: <stable(a)vger.kernel.org> Reported-by: chihhao chen <chihhao.chen(a)mediatek.com> Link: https://lore.kernel.org/r/87e6d6ae69d68dc588ac9acc8c0f24d6188375c3.camel@me… Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/endpoint.c | 23 +++++++++-------------- sound/usb/endpoint.h | 6 ++++-- sound/usb/pcm.c | 14 ++++++++++---- 3 files changed, 23 insertions(+), 20 deletions(-) diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index 0d7b73bf7945..a42f2ce19455 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -758,7 +758,8 @@ bool snd_usb_endpoint_compatible(struct snd_usb_audio *chip, * The endpoint needs to be closed via snd_usb_endpoint_close() later. * * Note that this function doesn't configure the endpoint. The substream - * needs to set it up later via snd_usb_endpoint_configure(). + * needs to set it up later via snd_usb_endpoint_set_params() and + * snd_usb_endpoint_prepare(). */ struct snd_usb_endpoint * snd_usb_endpoint_open(struct snd_usb_audio *chip, @@ -1290,12 +1291,13 @@ static int sync_ep_set_params(struct snd_usb_endpoint *ep) /* * snd_usb_endpoint_set_params: configure an snd_usb_endpoint * + * It's called either from hw_params callback. * Determine the number of URBs to be used on this endpoint. * An endpoint must be configured before it can be started. * An endpoint that is already running can not be reconfigured. */ -static int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep) +int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) { const struct audioformat *fmt = ep->cur_audiofmt; int err; @@ -1378,18 +1380,18 @@ static int init_sample_rate(struct snd_usb_audio *chip, } /* - * snd_usb_endpoint_configure: Configure the endpoint + * snd_usb_endpoint_prepare: Prepare the endpoint * * This function sets up the EP to be fully usable state. - * It's called either from hw_params or prepare callback. + * It's called either from prepare callback. * The function checks need_setup flag, and performs nothing unless needed, * so it's safe to call this multiple times. * * This returns zero if unchanged, 1 if the configuration has changed, * or a negative error code. */ -int snd_usb_endpoint_configure(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep) +int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep) { bool iface_first; int err = 0; @@ -1410,9 +1412,6 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; } - err = snd_usb_endpoint_set_params(chip, ep); - if (err < 0) - goto unlock; goto done; } @@ -1440,10 +1439,6 @@ int snd_usb_endpoint_configure(struct snd_usb_audio *chip, if (err < 0) goto unlock; - err = snd_usb_endpoint_set_params(chip, ep); - if (err < 0) - goto unlock; - err = snd_usb_select_mode_quirk(chip, ep->cur_audiofmt); if (err < 0) goto unlock; diff --git a/sound/usb/endpoint.h b/sound/usb/endpoint.h index 6a9af04cf175..e67ea28faa54 100644 --- a/sound/usb/endpoint.h +++ b/sound/usb/endpoint.h @@ -17,8 +17,10 @@ snd_usb_endpoint_open(struct snd_usb_audio *chip, bool is_sync_ep); void snd_usb_endpoint_close(struct snd_usb_audio *chip, struct snd_usb_endpoint *ep); -int snd_usb_endpoint_configure(struct snd_usb_audio *chip, - struct snd_usb_endpoint *ep); +int snd_usb_endpoint_set_params(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep); +int snd_usb_endpoint_prepare(struct snd_usb_audio *chip, + struct snd_usb_endpoint *ep); int snd_usb_endpoint_get_clock_rate(struct snd_usb_audio *chip, int clock); bool snd_usb_endpoint_compatible(struct snd_usb_audio *chip, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index d45d1d7e6664..b604f7e95e82 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -443,17 +443,17 @@ static int configure_endpoints(struct snd_usb_audio *chip, if (stop_endpoints(subs, false)) sync_pending_stops(subs); if (subs->sync_endpoint) { - err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); + err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); if (err < 0) return err; } - err = snd_usb_endpoint_configure(chip, subs->data_endpoint); + err = snd_usb_endpoint_prepare(chip, subs->data_endpoint); if (err < 0) return err; snd_usb_set_format_quirk(subs, subs->cur_audiofmt); } else { if (subs->sync_endpoint) { - err = snd_usb_endpoint_configure(chip, subs->sync_endpoint); + err = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); if (err < 0) return err; } @@ -551,7 +551,13 @@ static int snd_usb_hw_params(struct snd_pcm_substream *substream, subs->cur_audiofmt = fmt; mutex_unlock(&chip->mutex); - ret = configure_endpoints(chip, subs); + if (subs->sync_endpoint) { + ret = snd_usb_endpoint_set_params(chip, subs->sync_endpoint); + if (ret < 0) + goto unlock; + } + + ret = snd_usb_endpoint_set_params(chip, subs->data_endpoint); unlock: if (ret < 0) -- 2.35.3

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[PATCH] ALSA: asihpi - Remove useless code in hpi_meter_get_peak()
by Valentina Goncharenko 01 Sep '22

01 Sep '22

The hpi_meter_get_peak() function contains the expression "hm.obj_index = hm.obj_index", which does not carry any semantic load. Found by Linux Verification Center (linuxtesting.org) with SVACE. Fixes: 719f82d3987a ("ALSA: Add support of AudioScience ASI boards") Signed-off-by: Valentina Goncharenko <goncharenko.vp(a)ispras.ru> --- sound/pci/asihpi/hpifunc.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/pci/asihpi/hpifunc.c b/sound/pci/asihpi/hpifunc.c index 1de05383126a..24047fafef51 100644 --- a/sound/pci/asihpi/hpifunc.c +++ b/sound/pci/asihpi/hpifunc.c @@ -2020,7 +2020,6 @@ u16 hpi_meter_get_peak(u32 h_control, short an_peakdB[HPI_MAX_CHANNELS] HPI_CONTROL_GET_STATE); if (hpi_handle_indexes(h_control, &hm.adapter_index, &hm.obj_index)) return HPI_ERROR_INVALID_HANDLE; - hm.obj_index = hm.obj_index; hm.u.c.attribute = HPI_METER_PEAK; hpi_send_recv(&hm, &hr); -- 2.25.1

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[PATCH 1/2] ASoC: codecs: wcd934x: add Slimbus dependency
by Krzysztof Kozlowski 01 Sep '22

01 Sep '22

The WCD934X codec is a Slimbus driver, so it must depend on SLIMBUS, also for compile tests: ERROR: modpost: "slim_stream_prepare" [sound/soc/codecs/snd-soc-wcd934x.ko] undefined! Reported-by: kernel test robot <lkp(a)intel.com> Fixes: 5b7f4e5de61b ("ASoC: codecs: allow compile testing without MFD drivers") Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 968d0701f2e8..2fc37f4e061c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -1744,6 +1744,7 @@ config SND_SOC_WCD_MBHC config SND_SOC_WCD934X tristate "WCD9340/WCD9341 Codec" depends on COMMON_CLK + depends on SLIMBUS select SND_SOC_WCD_MBHC depends on MFD_WCD934X || COMPILE_TEST help -- 2.34.1

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mt8195-demo: redefine soundcard initialization
by GitHub pull_request - opened 01 Sep '22

01 Sep '22

alsa-project/alsa-ucm-conf pull request #211 was opened from fchiby: The previous initialization in Bootsequence set all Switch to off after boot. Whithout alsactl, alsa-ucm-conf does't work. So remove the setting off of the Switch in the BootSequence and define Enable/DisableSequence in each device section for a proper setting. Signed-off-by: Fadwa Chiby <fchiby(a)baylibre.com> Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/211 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/211.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v2] soundwire: qcom: fix device status array range
by Srinivas Kandagatla 01 Sep '22

01 Sep '22

This patch updates device status array range from 11 to 12 as we will be reading status from device number 0 to device number 11 inclusive. Without this patch we can potentially access status array out of range during auto-enumeration. Fixes: aa1262ca6695 ("soundwire: qcom: Check device status before reading devid") Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/qcom.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/soundwire/qcom.c b/drivers/soundwire/qcom.c index 9df970eeca45..16a4a51978ca 100644 --- a/drivers/soundwire/qcom.c +++ b/drivers/soundwire/qcom.c @@ -169,7 +169,7 @@ struct qcom_swrm_ctrl { u8 wcmd_id; struct qcom_swrm_port_config pconfig[QCOM_SDW_MAX_PORTS]; struct sdw_stream_runtime *sruntime[SWRM_MAX_DAIS]; - enum sdw_slave_status status[SDW_MAX_DEVICES]; + enum sdw_slave_status status[SDW_MAX_DEVICES + 1]; int (*reg_read)(struct qcom_swrm_ctrl *ctrl, int reg, u32 *val); int (*reg_write)(struct qcom_swrm_ctrl *ctrl, int reg, int val); u32 slave_status; @@ -420,7 +420,7 @@ static int qcom_swrm_get_alert_slave_dev_num(struct qcom_swrm_ctrl *ctrl) ctrl->reg_read(ctrl, SWRM_MCP_SLV_STATUS, &val); - for (dev_num = 0; dev_num < SDW_MAX_DEVICES; dev_num++) { + for (dev_num = 0; dev_num <= SDW_MAX_DEVICES; dev_num++) { status = (val >> (dev_num * SWRM_MCP_SLV_STATUS_SZ)); if ((status & SWRM_MCP_SLV_STATUS_MASK) == SDW_SLAVE_ALERT) { @@ -440,7 +440,7 @@ static void qcom_swrm_get_device_status(struct qcom_swrm_ctrl *ctrl) ctrl->reg_read(ctrl, SWRM_MCP_SLV_STATUS, &val); ctrl->slave_status = val; - for (i = 0; i < SDW_MAX_DEVICES; i++) { + for (i = 0; i <= SDW_MAX_DEVICES; i++) { u32 s; s = (val >> (i * 2)); -- 2.25.1

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[PATCH 0/6] ASoC: codecs: lpass: add support fro sm8450 and sc8280xp
by Srinivas Kandagatla 01 Sep '22

01 Sep '22

THis patchset adds support for SM8450 and SC8280XP SoC. Tested SmartSpeakers and Headset on SM8450 MTP and Lenovo Thinkpad X13s. THanks, Srini Srinivas Kandagatla (6): ASoC: qcom: dt-bindings: add sm8450 and sc8280xp compatibles ASoC: codecs: wsa-macro: add support for sm8450 and sc8280xp ASoC: codecs: tx-macro: add support for sm8450 and sc8280xp ASoC: codecs: rx-macro: add support for sm8450 and sc8280xp ASoC: codecs: va-macro: clear the frame sync counter before enabling ASoC: codecs: tx-macro: add support for sm8450 and sc8280xp .../bindings/sound/qcom,lpass-rx-macro.yaml | 2 + .../bindings/sound/qcom,lpass-tx-macro.yaml | 2 + .../bindings/sound/qcom,lpass-va-macro.yaml | 2 + .../bindings/sound/qcom,lpass-wsa-macro.yaml | 2 + sound/soc/codecs/lpass-rx-macro.c | 2 + sound/soc/codecs/lpass-tx-macro.c | 2 + sound/soc/codecs/lpass-va-macro.c | 72 +++++++++++++++++-- sound/soc/codecs/lpass-wsa-macro.c | 2 + 8 files changed, 82 insertions(+), 4 deletions(-) -- 2.21.0

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