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

[PATCH] ASoC: Switch two more i2c drivers back to use .probe()
by Uwe Kleine-König 12 Jun '23

12 Jun '23

The previous conversion back to .probe() applied in commit 9abcd24002bf ("ASoC: Switch i2c drivers back to use .probe()") was created based on v6.3. Since then two more drivers were added which need to be convert back in the same way before eventually .probe_new() can be dropped from struct i2c_driver. Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- Hello, cs35l56 was already in v6.4-rc1, ssm3515 is only in next. This patch is based on 4ac690bbae02 ("ASoC: ssm3515: Add new amp driver") which is the oldest commit in next that has both drivers. Best regards Uwe sound/soc/codecs/cs35l56-i2c.c | 2 +- sound/soc/codecs/ssm3515.c | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs35l56-i2c.c b/sound/soc/codecs/cs35l56-i2c.c index 295caad26224..ed2a41943d97 100644 --- a/sound/soc/codecs/cs35l56-i2c.c +++ b/sound/soc/codecs/cs35l56-i2c.c @@ -68,7 +68,7 @@ static struct i2c_driver cs35l56_i2c_driver = { .pm = &cs35l56_pm_ops_i2c_spi, }, .id_table = cs35l56_id_i2c, - .probe_new = cs35l56_i2c_probe, + .probe = cs35l56_i2c_probe, .remove = cs35l56_i2c_remove, }; diff --git a/sound/soc/codecs/ssm3515.c b/sound/soc/codecs/ssm3515.c index 784e890031a4..008cb3eb5758 100644 --- a/sound/soc/codecs/ssm3515.c +++ b/sound/soc/codecs/ssm3515.c @@ -439,7 +439,7 @@ static struct i2c_driver ssm3515_i2c_driver = { .name = "ssm3515", .of_match_table = of_match_ptr(ssm3515_of_match), }, - .probe_new = ssm3515_i2c_probe, + .probe = ssm3515_i2c_probe, }; module_i2c_driver(ssm3515_i2c_driver); base-commit: 4ac690bbae02e26e36e133becd86babb657126ef -- 2.39.2

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[PATCH] ASoC: dt-bindings: cirrus,cs35l45: drop unneeded quotes
by Krzysztof Kozlowski 12 Jun '23

12 Jun '23

Cleanup bindings dropping unneeded quotes. Once all these are fixed, checking for this can be enabled in yamllint. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- .../devicetree/bindings/sound/cirrus,cs35l45.yaml | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml b/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml index 2ab74f995685..4c9acb8d4c4c 100644 --- a/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml +++ b/Documentation/devicetree/bindings/sound/cirrus,cs35l45.yaml @@ -62,7 +62,7 @@ patternProperties: GPIO pin direction. Valid only when 'gpio-ctrl' is 1 0 = Output 1 = Input - $ref: "/schemas/types.yaml#/definitions/uint32" + $ref: /schemas/types.yaml#/definitions/uint32 minimum: 0 maximum: 1 default: 1 @@ -71,7 +71,7 @@ patternProperties: GPIO level. Valid only when 'gpio-ctrl' is 1 and 'gpio-dir' is 0 0 = Low 1 = High - $ref: "/schemas/types.yaml#/definitions/uint32" + $ref: /schemas/types.yaml#/definitions/uint32 minimum: 0 maximum: 1 default: 0 @@ -80,7 +80,7 @@ patternProperties: GPIO level. Valid only when 'gpio-ctrl' is 1 and 'gpio-dir' is 0 0 = CMOS 1 = Open Drain - $ref: "/schemas/types.yaml#/definitions/uint32" + $ref: /schemas/types.yaml#/definitions/uint32 minimum: 0 maximum: 1 default: 0 @@ -90,7 +90,7 @@ patternProperties: and 'gpio-dir' is 0 0 = Non-inverted, Active High 1 = Inverted, Active Low - $ref: "/schemas/types.yaml#/definitions/uint32" + $ref: /schemas/types.yaml#/definitions/uint32 minimum: 0 maximum: 1 default: 0 @@ -114,7 +114,7 @@ patternProperties: 0 = High impedance input 1 = Pin acts as a GPIO, direction controlled by 'gpio-dir' 2-7 = Reserved - $ref: "/schemas/types.yaml#/definitions/uint32" + $ref: /schemas/types.yaml#/definitions/uint32 minimum: 0 maximum: 7 default: 0 -- 2.34.1

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[PATCH v2 0/2] ASoC: mediatek: mt8173: Fix error paths
by Ricardo Ribalda Delgado 12 Jun '23

12 Jun '23

ASoC: mediatek: mt8173, presented a couple of error paths errors, lets fix them. Signed-off-by: Ricardo Ribalda Delgado <ribalda(a)chromium.org> --- Changes in v2: - Fix other error paths bugs, Thanks Dan! - Link to v1: https://lore.kernel.org/r/20230612-mt8173-fixup-v1-1-81f5c90316d1@chromium.… --- Ricardo Ribalda Delgado (2): ASoC: mediatek: mt8173: Fix snd_soc_component_initialize error path ASoC: mediatek: mt8173: Fix irq error path sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) --- base-commit: 858fd168a95c5b9669aac8db6c14a9aeab446375 change-id: 20230612-mt8173-fixup-bc602a96dbad Best regards, -- Ricardo Ribalda Delgado <ribalda(a)chromium.org>

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[PATCH] ASoC: SOF: ipc4/intel: Add missing mutex_unlock()
by Jyri Sarha 12 Jun '23

12 Jun '23

There was a missing mutex_unlock() in sof_ipc4_widget_free() use_chain_dma if-branch that caused a static analysis error. The branch should not be used in a normal working configuration and if its used its an indication of a bad topology. Add missing mutex_unlock() and a warning print if the if-branch is taken, and another warning print to a symmetric place in sof_ipc4_widget_setup(). Fixes: ca5ce0caa67fa9 ("ASoC: SOF: ipc4/intel: Add support for chained DMA") Reported-by: kernel test robot <lkp(a)intel.com> Reported-by: Dan Carpenter <error27(a)gmail.com> Link: https://lore.kernel.org/r/202303222050.dCw0fPCW-lkp@intel.com/ Signed-off-by: Jyri Sarha <jyri.sarha(a)linux.intel.com> --- sound/soc/sof/ipc4-topology.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/ipc4-topology.c b/sound/soc/sof/ipc4-topology.c index f1e1aed94da4..12775fcb6b54 100644 --- a/sound/soc/sof/ipc4-topology.c +++ b/sound/soc/sof/ipc4-topology.c @@ -1989,8 +1989,11 @@ static int sof_ipc4_widget_setup(struct snd_sof_dev *sdev, struct snd_sof_widget case snd_soc_dapm_scheduler: pipeline = swidget->private; - if (pipeline->use_chain_dma) + if (pipeline->use_chain_dma) { + dev_warn(sdev->dev, "use_chain_dma set for schduler %s", + swidget->widget->name); return 0; + } dev_dbg(sdev->dev, "pipeline: %d memory pages: %d\n", swidget->pipeline_id, pipeline->mem_usage); @@ -2145,8 +2148,12 @@ static int sof_ipc4_widget_free(struct snd_sof_dev *sdev, struct snd_sof_widget struct sof_ipc4_msg msg = {{ 0 }}; u32 header; - if (pipeline->use_chain_dma) + if (pipeline->use_chain_dma) { + dev_warn(sdev->dev, "use_chain_dma set for schduler %s", + swidget->widget->name); + mutex_unlock(&ipc4_data->pipeline_state_mutex); return 0; + } header = SOF_IPC4_GLB_PIPE_INSTANCE_ID(swidget->instance_id); header |= SOF_IPC4_MSG_TYPE_SET(SOF_IPC4_GLB_DELETE_PIPELINE); -- 2.34.1

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[PATCH] ALSA: pcm: fix wait_time calculations
by Oswald Buddenhagen 12 Jun '23

12 Jun '23

... in wait_for_avail() and snd_pcm_drain(). t was calculated in seconds, so it would be pretty much always zero, to be subsequently de-facto ignored due to being max(t, 10)'d. And then it (i.e., 10) would be treated as secs, which doesn't seem right. However, fixing it to properly calculate msecs would potentially cause timeouts when using twice the period size for the default timeout (which seems reasonable to me), so instead use the buffer size plus 10 percent to be on the safe side ... but that still seems insufficient, presumably because the hardware typically needs a moment to fire up. To compensate for this, we up the minimal timeout to 100ms, which is still two orders of magnitude less than the bogus minimum. substream->wait_time was also misinterpreted as jiffies, despite being documented as being in msecs. Only the soc/sof driver sets it - to 500, which looks very much like msecs were intended. Speaking of which, shouldn't snd_pcm_drain() also use substream-> wait_time? As a drive-by, make the debug messages on timeout less confusing. Signed-off-by: Oswald Buddenhagen <oswald.buddenhagen(a)gmx.de> --- sound/core/pcm_lib.c | 11 +++++------ sound/core/pcm_native.c | 8 ++++---- 2 files changed, 9 insertions(+), 10 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 02fd65993e7e..af1eb136feb0 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1878,15 +1878,14 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (substream->wait_time) { wait_time = substream->wait_time; } else { - wait_time = 10; + wait_time = 100; if (runtime->rate) { - long t = runtime->period_size * 2 / - runtime->rate; + long t = runtime->buffer_size * 1100 / runtime->rate; wait_time = max(t, wait_time); } - wait_time = msecs_to_jiffies(wait_time * 1000); } + wait_time = msecs_to_jiffies(wait_time); } for (;;) { @@ -1934,8 +1933,8 @@ static int wait_for_avail(struct snd_pcm_substream *substream, } if (!tout) { pcm_dbg(substream->pcm, - "%s write error (DMA or IRQ trouble?)\n", - is_playback ? "playback" : "capture"); + "%s timeout (DMA or IRQ trouble?)\n", + is_playback ? "playback write" : "capture read"); err = -EIO; break; } diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 331380c2438b..94185267a7b9 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -2159,12 +2159,12 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, if (runtime->no_period_wakeup) tout = MAX_SCHEDULE_TIMEOUT; else { - tout = 10; + tout = 100; if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; + long t = runtime->buffer_size * 1100 / runtime->rate; tout = max(t, tout); } - tout = msecs_to_jiffies(tout * 1000); + tout = msecs_to_jiffies(tout); } tout = schedule_timeout(tout); @@ -2187,7 +2187,7 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, result = -ESTRPIPE; else { dev_dbg(substream->pcm->card->dev, - "playback drain error (DMA or IRQ trouble?)\n"); + "playback drain timeout (DMA or IRQ trouble?)\n"); snd_pcm_stop(substream, SNDRV_PCM_STATE_SETUP); result = -EIO; } -- 2.40.0.152.g15d061e6df

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[PATCH v2 1/2] ASoC: dt-bindings: qcom,wsa8840: Add WSA884x family of speakers
by Krzysztof Kozlowski 12 Jun '23

12 Jun '23

Add binding for WSA8840/WSA8845/WSA8845H smart speaker amplifiers used in Qualcomm QRD8550 board with SM8550 SoC. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Changes in v2: 1. Correct compatible (sdw version 1 -> 2). Cc: Patrick Lai <quic_plai(a)quicinc.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- .../bindings/sound/qcom,wsa8840.yaml | 66 +++++++++++++++++++ 1 file changed, 66 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml diff --git a/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml new file mode 100644 index 000000000000..e6723c9e312a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/qcom,wsa8840.yaml @@ -0,0 +1,66 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/qcom,wsa8840.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Qualcomm WSA8840/WSA8845/WSA8845H smart speaker amplifier + +maintainers: + - Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> + - Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> + +description: + WSA884X is a family of Qualcomm Aqstic smart speaker amplifiers using + SoundWire digital audio interface. + +allOf: + - $ref: dai-common.yaml# + +properties: + compatible: + const: sdw20217020400 + + reg: + maxItems: 1 + + powerdown-gpios: + description: Powerdown/Shutdown line to use (pin SD_N) + maxItems: 1 + + '#sound-dai-cells': + const: 0 + + vdd-1p8-supply: true + vdd-io-supply: true + +required: + - compatible + - reg + - powerdown-gpios + - '#sound-dai-cells' + - vdd-1p8-supply + - vdd-io-supply + +unevaluatedProperties: false + +examples: + - | + #include <dt-bindings/gpio/gpio.h> + + soundwire-controller { + #address-cells = <2>; + #size-cells = <0>; + + speaker@0,1 { + compatible = "sdw20217020400"; + reg = <0 1>; + pinctrl-names = "default"; + pinctrl-0 = <&spkr_2_sd_n_active>; + powerdown-gpios = <&lpass_tlmm 18 GPIO_ACTIVE_LOW>; + #sound-dai-cells = <0>; + sound-name-prefix = "SpkrRight"; + vdd-1p8-supply = <&vreg_l15b_1p8>; + vdd-io-supply = <&vreg_l3g_1p2>; + }; + }; -- 2.34.1

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[PATCH] ALSA: usb-audio: Fix broken resume due to UAC3 power state
by Takashi Iwai 12 Jun '23

12 Jun '23

As reported in the bugzilla below, the PM resume of a UAC3 device may fail due to the incomplete power state change, stuck at D1. The reason is that the driver expects the full D0 power state change only at hw_params, while the normal PCM resume procedure doesn't call hw_params. For fixing the bug, we add the same power state update to D0 at the prepare callback, which is certainly called by the resume procedure. Note that, with this change, the power state change in the hw_params becomes almost redundant, since snd_usb_hw_params() doesn't touch the parameters (at least it tires so). But dropping it is still a bit risky (e.g. we have the media-driver binding), so I leave the D0 power state change in snd_usb_hw_params() as is for now. Fixes: a0a4959eb4e9 ("ALSA: usb-audio: Operate UAC3 Power Domains in PCM callbacks") Cc: <stable(a)vger.kernel.org> Link: https://bugzilla.kernel.org/show_bug.cgi?id=217539 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/pcm.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index eec5232f9fb2..08bf535ed163 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -650,6 +650,10 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) goto unlock; } + ret = snd_usb_pcm_change_state(subs, UAC3_PD_STATE_D0); + if (ret < 0) + goto unlock; + again: if (subs->sync_endpoint) { ret = snd_usb_endpoint_prepare(chip, subs->sync_endpoint); -- 2.35.3

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[PATCH] ALSA: seq: oss: Fix racy open/close of MIDI devices
by Takashi Iwai 12 Jun '23

12 Jun '23

Although snd_seq_oss_midi_open() and snd_seq_oss_midi_close() can be called concurrently from different code paths, we have no proper data protection against races. Introduce open_mutex to each seq_oss_midi object for avoiding the races. Reported-by: "Gong, Sishuai" <sishuai(a)purdue.edu> Closes: https://lore.kernel.org/r/7DC9AF71-F481-4ABA-955F-76C535661E33@purdue.edu Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/oss/seq_oss_midi.c | 35 +++++++++++++++++++------------ 1 file changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/core/seq/oss/seq_oss_midi.c b/sound/core/seq/oss/seq_oss_midi.c index 07efb38f58ac..f2940b29595f 100644 --- a/sound/core/seq/oss/seq_oss_midi.c +++ b/sound/core/seq/oss/seq_oss_midi.c @@ -37,6 +37,7 @@ struct seq_oss_midi { struct snd_midi_event *coder; /* MIDI event coder */ struct seq_oss_devinfo *devinfo; /* assigned OSSseq device */ snd_use_lock_t use_lock; + struct mutex open_mutex; }; @@ -172,6 +173,7 @@ snd_seq_oss_midi_check_new_port(struct snd_seq_port_info *pinfo) mdev->flags = pinfo->capability; mdev->opened = 0; snd_use_lock_init(&mdev->use_lock); + mutex_init(&mdev->open_mutex); /* copy and truncate the name of synth device */ strscpy(mdev->name, pinfo->name, sizeof(mdev->name)); @@ -322,15 +324,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) int perm; struct seq_oss_midi *mdev; struct snd_seq_port_subscribe subs; + int err; mdev = get_mididev(dp, dev); if (!mdev) return -ENODEV; + mutex_lock(&mdev->open_mutex); /* already used? */ if (mdev->opened && mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return -EBUSY; + err = -EBUSY; + goto unlock; } perm = 0; @@ -340,14 +344,14 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) perm |= PERM_READ; perm &= mdev->flags; if (perm == 0) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } /* already opened? */ if ((mdev->opened & perm) == perm) { - snd_use_lock_free(&mdev->use_lock); - return 0; + err = 0; + goto unlock; } perm &= ~mdev->opened; @@ -372,13 +376,17 @@ snd_seq_oss_midi_open(struct seq_oss_devinfo *dp, int dev, int fmode) } if (! mdev->opened) { - snd_use_lock_free(&mdev->use_lock); - return -ENXIO; + err = -ENXIO; + goto unlock; } mdev->devinfo = dp; + err = 0; + + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); - return 0; + return err; } /* @@ -393,10 +401,9 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev = get_mididev(dp, dev); if (!mdev) return -ENODEV; - if (! mdev->opened || mdev->devinfo != dp) { - snd_use_lock_free(&mdev->use_lock); - return 0; - } + mutex_lock(&mdev->open_mutex); + if (!mdev->opened || mdev->devinfo != dp) + goto unlock; memset(&subs, 0, sizeof(subs)); if (mdev->opened & PERM_WRITE) { @@ -415,6 +422,8 @@ snd_seq_oss_midi_close(struct seq_oss_devinfo *dp, int dev) mdev->opened = 0; mdev->devinfo = NULL; + unlock: + mutex_unlock(&mdev->open_mutex); snd_use_lock_free(&mdev->use_lock); return 0; } -- 2.35.3

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[PATCH 0/2] ALSA: firewire: use 'GPL' string for module license
by Takashi Sakamoto 12 Jun '23

12 Jun '23

Hi, Nowadays, the use of "GPL" string in MODULE_LICENSE macro seems to obsolete "GPL v2" string. This series includes two patches to use it for modules in ALSA firewire stack contributed by two developers. This change does not pose any issue for modules contributed by Takashi Sakamoto, the author of this series, while it is advisable to have careful consideration for modules contributed by Clemens Ladisch, since the code related to license will be modified by a developer who is not the copyright holder. A commit bf7fbeeae6db ("module: Cure the MODULE_LICENSE "GPL" vs. "GPL v2" bogosity") adds detail documentation about the macro itself and clarifies the difference between "GPL" and "GPL v2". The macro is designed to provide sufficient information to kernel module loader and user space tools. Both strings have the same meanings, but the latter is legacy. In conclusion, any change made to the usage of the macro neither result in significant alterations to license nor copyright. Takashi Sakamoto (2): ALSA: firewire: use 'GPL' string for module license contributed by Takashi Sakamoto ALSA: firewire: use 'GPL' string for module license contributed by Clemens Ladisch sound/firewire/bebob/bebob.c | 2 +- sound/firewire/dice/dice.c | 2 +- sound/firewire/digi00x/digi00x.c | 2 +- sound/firewire/fireface/ff.c | 2 +- sound/firewire/fireworks/fireworks.c | 2 +- sound/firewire/isight.c | 2 +- sound/firewire/lib.c | 2 +- sound/firewire/motu/motu.c | 2 +- sound/firewire/oxfw/oxfw.c | 2 +- sound/firewire/tascam/tascam.c | 2 +- 10 files changed, 10 insertions(+), 10 deletions(-) -- 2.39.2

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[PATCH v2 00/11] ASoC: qcom: audioreach: add compress offload support
by Srinivas Kandagatla 12 Jun '23

12 Jun '23

This patchset adds compressed offload support to Qualcomm audioreach drivers. Currently it supports AAC, MP3 and FALC along with gapless. Tested this on SM8450 and sc7280. thanks, srini Changes since v1: - removed lots of code duplication - moved ALSA patch out of this series. Mohammad Rafi Shaik (4): ASoC: qcom: SC7280: audioreach: Add sc7280 hardware param fixup callback ASoC: q6dsp: q6apm: add end of stream events ASoC: q6dsp: audioreach: Add support to set compress format params ASoC: q6dsp: audioreach: Add gapless feature support Srinivas Kandagatla (7): ASoC: q6dsp: audioreach: add helper function to set u32 param ASoC: q6dsp: audioreach: Add placeholder decoder for compress playback ASoC: q6dsp: q6apm-dai: Add open/free compress DAI callbacks ASoC: q6dsp: q6apm-dai: Add compress DAI and codec caps get callbacks ASoC: q6dsp: q6apm-dai: Add trigger/pointer compress DAI callbacks ASoC: q6dsp: q6apm-dai: Add compress set params and metadata DAI callbacks ASoC: q6dsp: q6apm-dai: Add mmap and copy compress DAI callbacks sound/soc/qcom/qdsp6/audioreach.c | 248 ++++++++++------- sound/soc/qcom/qdsp6/audioreach.h | 51 ++++ sound/soc/qcom/qdsp6/q6apm-dai.c | 445 ++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6apm.c | 68 +++++ sound/soc/qcom/qdsp6/q6apm.h | 6 + sound/soc/qcom/sc7280.c | 23 +- 6 files changed, 745 insertions(+), 96 deletions(-) -- 2.21.0

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