July 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH v2 0/4] soundwire: qcom: stablity fixes
by Srinivas Kandagatla 10 Jul '23

10 Jul '23

During x13s audio testing we hit few corner cases due to issues in codec drivers and some obvious code bugs. Here are the fixes for those issues, mostly the issues are around devices loosing the sync in between runtime pm suspend resume path. With codec fixes along with these fixes, audio on x13s is pretty stable. Thanks, Srini Changes since v1: - dropped runtime pm changes patch as unable to reproduced it anymore - fixed clk stop flag as suggested by Pierre - rebased on top of linux-next Srinivas Kandagatla (4): soundwire: qcom: update status correctly with mask soundwire: qcom: wait for fifo to be empty before suspend soundwire: qcom: add software workaround for bus clash interrupt assertion soundwire: qcom: set clk stop need reset flag at runtime drivers/soundwire/qcom.c | 100 ++++++++++++++++++++++++++------------- 1 file changed, 68 insertions(+), 32 deletions(-) -- 2.21.0

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on (not) breaking user space
by Oswald Buddenhagen 09 Jul '23

09 Jul '23

hi linus, the alsa maintainers seem to be applying a reading of "we don't break user space" that i find counter-productive. so i would appreciate you clarifying your position on that matter (once more ...) to be clear, we're not arguing over changing any abi, but simply exercising one somewhat more than before. the claim is that this would likely have unacceptable consequences. i disagree. context: > Due to hardware constraints, changing the clock multiplier (CM) > changes the available audio ports and the number of available > channels. [...] Therefore, we do dynamic reconfiguration (DR) of the > mixer in response to changing the CM. arguments: > DR is somewhat controversial, as it has the potential to crash poorly > programmed applications. > But that in itself isn't a very convincing argument against it, as by > that logic we'd have to ban all hot-plugging. (1) > Such crashes would also not really qualify as regressions, as the > D.A.S. mode is a new opt-in feature, and therefore no previously > stable setups would be impacted. (2) the full text is at https://lore.kernel.org/all/20230630144542.664190-7-oswald.buddenhagen@gmx.… the preceding discussion is at https://lore.kernel.org/all/87v8fren1k.wl-tiwai@suse.de/T/#u regards, ossi

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[Feature Request] New hdajackretask ncurses TUI
by GitHub issues - opened 09 Jul '23

09 Jul '23

alsa-project/alsa-tools issue #16 was opened from HalanoSiblee: Hi Why i need gtk GUI when I do such a simple retaskable jack input Please make hdajackretask with ncurses like alsamixer to work in tty and terminal's at this point it will be so useful less bloat ,and blazingly fast Thanks. Issue URL : https://github.com/alsa-project/alsa-tools/issues/16 Repository URL: https://github.com/alsa-project/alsa-tools

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[PATCH 1/2] ALSA: pcmtest: Convert to platform remove callback returning void
by Uwe Kleine-König 07 Jul '23

07 Jul '23

The .remove() callback for a platform driver returns an int which makes many driver authors wrongly assume it's possible to do error handling by returning an error code. However the value returned is ignored (apart from emitting a warning) and this typically results in resource leaks. To improve here there is a quest to make the remove callback return void. In the first step of this quest all drivers are converted to .remove_new() which already returns void. Eventually after all drivers are converted, .remove_new() is renamed to .remove(). Trivially convert this driver from always returning zero in the remove callback to the void returning variant. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/drivers/pcmtest.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/drivers/pcmtest.c b/sound/drivers/pcmtest.c index 2ae912a64d16..1fff85feaf82 100644 --- a/sound/drivers/pcmtest.c +++ b/sound/drivers/pcmtest.c @@ -576,10 +576,9 @@ static int pcmtst_probe(struct platform_device *pdev) return 0; } -static int pdev_remove(struct platform_device *dev) +static void pdev_remove(struct platform_device *dev) { snd_pcmtst_free(pcmtst); - return 0; } static struct platform_device pcmtst_pdev = { @@ -589,7 +588,7 @@ static struct platform_device pcmtst_pdev = { static struct platform_driver pcmtst_pdrv = { .probe = pcmtst_probe, - .remove = pdev_remove, + .remove_new = pdev_remove, .driver = { .name = "pcmtest", }, base-commit: 5133c9e51de41bfa902153888e11add3342ede18 -- 2.39.2

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[PATCH] dt-bindings: sound: Update maintainer email id
by Rohit kumar 07 Jul '23

07 Jul '23

Updated my mail id to latest quicinc id. Signed-off-by: Rohit kumar <quic_rohkumar(a)quicinc.com> --- .../devicetree/bindings/sound/google,sc7180-trogdor.yaml | 2 +- Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml index 666a95ac22c8..ba5b7728cf33 100644 --- a/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml +++ b/Documentation/devicetree/bindings/sound/google,sc7180-trogdor.yaml @@ -7,7 +7,7 @@ $schema: http://devicetree.org/meta-schemas/core.yaml# title: Google SC7180-Trogdor ASoC sound card driver maintainers: - - Rohit kumar <rohitkr(a)codeaurora.org> + - Rohit kumar <quic_rohkumar(a)quicinc.com> - Cheng-Yi Chiang <cychiang(a)chromium.org> description: diff --git a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml index 6cc8f86c7531..3a559bd07a79 100644 --- a/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml +++ b/Documentation/devicetree/bindings/sound/qcom,lpass-cpu.yaml @@ -8,7 +8,7 @@ title: Qualcomm Technologies Inc. LPASS CPU dai driver maintainers: - Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> - - Rohit kumar <rohitkr(a)codeaurora.org> + - Rohit kumar <quic_rohkumar(a)quicinc.com> description: | Qualcomm Technologies Inc. SOC Low-Power Audio SubSystem (LPASS) that consist -- 2.25.1

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[PATCH] ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode
by Shengjiu Wang 07 Jul '23

07 Jul '23

On i.MX8MM, the MCTL_MCLK_EN bit it is not only the gate for MCLK output to PAD, but also the gate bit between root clock and SAI module, So it is need to be enabled for master mode, otherwise there is no bclk generated. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index fa950dde5310..dae16a14f177 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -437,6 +437,12 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) FSL_SAI_CR2_DIV_MASK | FSL_SAI_CR2_BYP, savediv / 2 - 1); + if (sai->soc_data->max_register >= FSL_SAI_MCTL) { + /* SAI is in master mode at this point, so enable MCLK */ + regmap_update_bits(sai->regmap, FSL_SAI_MCTL, + FSL_SAI_MCTL_MCLK_EN, FSL_SAI_MCTL_MCLK_EN); + } + return 0; } -- 2.17.1

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[RFC] ASoC: fsl_sai: Do not set MCTL_MCLK_EN based on the number of registers
by Fabio Estevam 06 Jul '23

06 Jul '23

From: Fabio Estevam <festevam(a)denx.de> Since commit ff87d619ac18 ("ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode") Andreas reports that on an i.MX8MP-based system where MCLK needs to be an input, the MCLK pin is actually an output, despite not having the 'fsl,sai-mclk-direction-output' property present in the devicetree. This commit explains the following: "On i.MX8MM, the MCTL_MCLK_EN bit it is not only the gate for MCLK output to PAD, but also the gate bit between root clock and SAI module, So it is need to be enabled for master mode, otherwise there is no bclk generated." Currently, the decision of using MCTL_MCLK_EN as a clock gate is based on the number of SAI registers present on a platform. This is fragile as it causes issues on i.MX8MP, for example. Fix the problem by introducing a new boolean mclk_en_gates_clock member in the fsl_sai_soc_data structure that indicates that the MCTL_MCLK_EN bit also acts as a gate between the root clock and the SAI block. This allows i.MX8MM to turn on FSL_SAI_MCTL_MCLK_EN as intended and other SoCs such as i.MX8MP can still use MCLK as input. Fixes: ff87d619ac18 ("ASoC: fsl_sai: Enable MCTL_MCLK_EN bit for master mode") Reported-by: Andreas Henriksson <andreas(a)fatal.se> Signed-off-by: Fabio Estevam <festevam(a)denx.de> Signed-off-by: Fabio Estevam <festevam(a)denx.de> --- Hi Shengjiu, Is imx8mm the only SoC that needs mclk_en_gates_clock = true? What about imx8mn and imx93? sound/soc/fsl/fsl_sai.c | 12 +++++++++++- sound/soc/fsl/fsl_sai.h | 1 + 2 files changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5e09f634c61b..e4dc6964f011 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -507,7 +507,7 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) savediv / 2 - 1); } - if (sai->soc_data->max_register >= FSL_SAI_MCTL) { + if (sai->soc_data->mclk_en_gates_clock) { /* SAI is in master mode at this point, so enable MCLK */ regmap_update_bits(sai->regmap, FSL_SAI_MCTL, FSL_SAI_MCTL_MCLK_EN, FSL_SAI_MCTL_MCLK_EN); @@ -1513,6 +1513,7 @@ static const struct fsl_sai_soc_data fsl_sai_vf610_data = { .reg_offset = 0, .mclk0_is_mclk1 = false, .flags = 0, + .mclk_en_gates_clock = false, .max_register = FSL_SAI_RMR, }; @@ -1524,6 +1525,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx6sx_data = { .reg_offset = 0, .mclk0_is_mclk1 = true, .flags = 0, + .mclk_en_gates_clock = false, .max_register = FSL_SAI_RMR, }; @@ -1535,6 +1537,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx7ulp_data = { .reg_offset = 8, .mclk0_is_mclk1 = false, .flags = PMQOS_CPU_LATENCY, + .mclk_en_gates_clock = false, .max_register = FSL_SAI_RMR, }; @@ -1546,6 +1549,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8mq_data = { .reg_offset = 8, .mclk0_is_mclk1 = false, .flags = 0, + .mclk_en_gates_clock = false, .max_register = FSL_SAI_RMR, }; @@ -1557,6 +1561,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8qm_data = { .reg_offset = 0, .mclk0_is_mclk1 = false, .flags = 0, + .mclk_en_gates_clock = false, .max_register = FSL_SAI_RMR, }; @@ -1568,6 +1573,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8mm_data = { .mclk0_is_mclk1 = false, .pins = 8, .flags = 0, + .mclk_en_gates_clock = true, .max_register = FSL_SAI_MCTL, }; @@ -1579,6 +1585,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8mn_data = { .mclk0_is_mclk1 = false, .pins = 8, .flags = 0, + .mclk_en_gates_clock = false, .max_register = FSL_SAI_MDIV, }; @@ -1590,6 +1597,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8mp_data = { .mclk0_is_mclk1 = false, .pins = 8, .flags = 0, + .mclk_en_gates_clock = false, .max_register = FSL_SAI_MDIV, .mclk_with_tere = true, }; @@ -1602,6 +1610,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx8ulp_data = { .mclk0_is_mclk1 = false, .pins = 4, .flags = PMQOS_CPU_LATENCY, + .mclk_en_gates_clock = false, .max_register = FSL_SAI_RTCAP, }; @@ -1613,6 +1622,7 @@ static const struct fsl_sai_soc_data fsl_sai_imx93_data = { .mclk0_is_mclk1 = false, .pins = 4, .flags = 0, + .mclk_en_gates_clock = false, .max_register = FSL_SAI_MCTL, .max_burst = {8, 8}, }; diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 8254c3547b87..473260b0fbea 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -232,6 +232,7 @@ struct fsl_sai_soc_data { bool use_edma; bool mclk0_is_mclk1; bool mclk_with_tere; + bool mclk_en_gates_clock; unsigned int fifo_depth; unsigned int pins; unsigned int reg_offset; -- 2.34.1

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[PATCH] ALSA: jack: Fix mutex call in snd_jack_report()
by Takashi Iwai 06 Jul '23

06 Jul '23

snd_jack_report() is supposed to be callable from an IRQ context, too, and it's indeed used in that way from virtsnd driver. The fix for input_dev race in commit 1b6a6fc5280e ("ALSA: jack: Access input_dev under mutex"), however, introduced a mutex lock in snd_jack_report(), and this resulted in a potential sleep-in-atomic. For addressing that problem, this patch changes the relevant code to use the object get/put and removes the mutex usage. That is, snd_jack_report(), it takes input_get_device() and leaves with input_put_device() for assuring the input_dev being assigned. Although the whole mutex could be reduced, we keep it because it can be still a protection for potential races between creation and deletion. Fixes: 1b6a6fc5280e ("ALSA: jack: Access input_dev under mutex") Reported-by: Dan Carpenter <dan.carpenter(a)linaro.org> Closes: https://lore.kernel.org/r/cf95f7fe-a748-4990-8378-000491b40329@moroto.mount… Tested-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/jack.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) diff --git a/sound/core/jack.c b/sound/core/jack.c index 88493cc31914..03d155ed362b 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -654,6 +654,7 @@ void snd_jack_report(struct snd_jack *jack, int status) struct snd_jack_kctl *jack_kctl; unsigned int mask_bits = 0; #ifdef CONFIG_SND_JACK_INPUT_DEV + struct input_dev *idev; int i; #endif @@ -670,17 +671,15 @@ void snd_jack_report(struct snd_jack *jack, int status) status & jack_kctl->mask_bits); #ifdef CONFIG_SND_JACK_INPUT_DEV - mutex_lock(&jack->input_dev_lock); - if (!jack->input_dev) { - mutex_unlock(&jack->input_dev_lock); + idev = input_get_device(jack->input_dev); + if (!idev) return; - } for (i = 0; i < ARRAY_SIZE(jack->key); i++) { int testbit = ((SND_JACK_BTN_0 >> i) & ~mask_bits); if (jack->type & testbit) - input_report_key(jack->input_dev, jack->key[i], + input_report_key(idev, jack->key[i], status & testbit); } @@ -688,13 +687,13 @@ void snd_jack_report(struct snd_jack *jack, int status) int testbit = ((1 << i) & ~mask_bits); if (jack->type & testbit) - input_report_switch(jack->input_dev, + input_report_switch(idev, jack_switch_types[i], status & testbit); } - input_sync(jack->input_dev); - mutex_unlock(&jack->input_dev_lock); + input_sync(idev); + input_put_device(idev); #endif /* CONFIG_SND_JACK_INPUT_DEV */ } EXPORT_SYMBOL(snd_jack_report); -- 2.35.3

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[bug report] ALSA: jack: Access input_dev under mutex
by Dan Carpenter 06 Jul '23

06 Jul '23

Hello Amadeusz Sławiński, The patch 1b6a6fc5280e: "ALSA: jack: Access input_dev under mutex" from Apr 12, 2022, leads to the following Smatch static checker warning: sound/core/jack.c:673 snd_jack_report() warn: sleeping in atomic context sound/core/jack.c 663 jack->hw_status_cache = status; 664 665 list_for_each_entry(jack_kctl, &jack->kctl_list, list) 666 if (jack_kctl->sw_inject_enable) 667 mask_bits |= jack_kctl->mask_bits; 668 else 669 snd_kctl_jack_report(jack->card, jack_kctl->kctl, 670 status & jack_kctl->mask_bits); 671 672 #ifdef CONFIG_SND_JACK_INPUT_DEV --> 673 mutex_lock(&jack->input_dev_lock); ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ That patch adds this mutex but we can't take mutex because we're already holding a spinlock. The problematic call trees are: virtsnd_event_notify_cb() <- disables preempt virtsnd_disable_event_vq() <- disables preempt -> virtsnd_event_dispatch() -> virtsnd_jack_event() -> snd_jack_report() The virtsnd_event_notify_cb() and virtsnd_disable_event_vq() functions take the spin_lock_irqsave(&queue->lock, flags); regards, dan carpenter

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[PATCH] ASoC: qcom: q6afe-dai: fix Display Port Playback stream name
by Srinivas Kandagatla 06 Jul '23

06 Jul '23

With recent changes to add more display ports did not change the Stream name in q6afe-dai. This results in below error "ASoC: Failed to add route DISPLAY_PORT_RX -> Display Port Playback(*)" and sound card fails to probe. Fix this by adding correct stream name. Fixes: 90848a2557fe ("ASoC: qcom: q6dsp: add support to more display ports") Reported-by: Amit Pundir <amit.pundir(a)linaro.org> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- sound/soc/qcom/qdsp6/q6afe-dai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c index 31e0bad71e95..dbff55a97162 100644 --- a/sound/soc/qcom/qdsp6/q6afe-dai.c +++ b/sound/soc/qcom/qdsp6/q6afe-dai.c @@ -476,7 +476,7 @@ static int q6afe_mi2s_set_sysclk(struct snd_soc_dai *dai, static const struct snd_soc_dapm_route q6afe_dapm_routes[] = { {"HDMI Playback", NULL, "HDMI_RX"}, - {"Display Port Playback", NULL, "DISPLAY_PORT_RX"}, + {"DISPLAY_PORT_RX_0 Playback", NULL, "DISPLAY_PORT_RX"}, {"Slimbus Playback", NULL, "SLIMBUS_0_RX"}, {"Slimbus1 Playback", NULL, "SLIMBUS_1_RX"}, {"Slimbus2 Playback", NULL, "SLIMBUS_2_RX"}, -- 2.25.1

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