November 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/2] Make cb a required parameter of buffer-cb
by Nuno Sá 28 Nov '20

28 Nov '20

When going through the code of the buffer-cb interface and all it's users, I realized that the stm32_adfsdm driver is calling `iio_channel_get_all_cb()` with NULL for the cb. After going a bit trough the stm drivers, it looks like this is actually intentional. However, it is clear that we have a clear/direct route here for a NULL pointer dereference. This change makes cb a required parameter of the API. The first patch makes the necessary changes to the stm32_adfsdm driver so that it does not break. Nuno Sá (1): iio: buffer: Return error if no callback is given Olivier Moysan (1): ASoC: stm32: dfsdm: add stm32_adfsdm_dummy_cb() callback drivers/iio/buffer/industrialio-buffer-cb.c | 5 +++++ sound/soc/stm/stm32_adfsdm.c | 12 +++++++++++- 2 files changed, 16 insertions(+), 1 deletion(-) -- 2.17.1

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[PATCH] ALSA: hda/realtek: Add mute LED quirk to yet another HP x360 model
by Takashi Iwai 28 Nov '20

28 Nov '20

HP Spectre x360 Convertible 15" version (SSID 103c:827f) needs the same quirk to make the mute LED working like other models. System Information Manufacturer: HP Product Name: HP Spectre x360 Convertible 15-bl1XX Sound Codec: Codec: Realtek ALC295 Vendor Id: 0x10ec0295 Subsystem Id: 0x103c827f Revision Id: 0x100002 Reported-by: <christoph.plattner(a)gmx.at> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index c7e8680a7190..8616c5624870 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7934,6 +7934,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x820d, "HP Pavilion 15", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x8256, "HP", ALC221_FIXUP_HP_FRONT_MIC), SND_PCI_QUIRK(0x103c, 0x827e, "HP x360", ALC295_FIXUP_HP_X360), + SND_PCI_QUIRK(0x103c, 0x827f, "HP x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), SND_PCI_QUIRK(0x103c, 0x82bf, "HP G3 mini", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x82c0, "HP G3 mini premium", ALC221_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x83b9, "HP Spectre x360", ALC269_FIXUP_HP_MUTE_LED_MIC3), -- 2.26.2

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[PATCH] ALSA: remove trailing semicolon in macro definition
by trix＠redhat.com 28 Nov '20

28 Nov '20

From: Tom Rix <trix(a)redhat.com> The macro use will already have a semicolon. Signed-off-by: Tom Rix <trix(a)redhat.com> --- include/sound/hda_codec.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/hda_codec.h b/include/sound/hda_codec.h index 73827b7d17e0..2e8d51937acd 100644 --- a/include/sound/hda_codec.h +++ b/include/sound/hda_codec.h @@ -344,7 +344,7 @@ snd_hda_get_num_conns(struct hda_codec *codec, hda_nid_t nid) #define snd_hda_get_raw_connections(codec, nid, list, max_conns) \ snd_hdac_get_connections(&(codec)->core, nid, list, max_conns) #define snd_hda_get_num_raw_conns(codec, nid) \ - snd_hdac_get_connections(&(codec)->core, nid, NULL, 0); + snd_hdac_get_connections(&(codec)->core, nid, NULL, 0) int snd_hda_get_conn_list(struct hda_codec *codec, hda_nid_t nid, const hda_nid_t **listp); -- 2.18.4

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[PATCH] ALSA: usb-audio: Add support for Pioneer DJ DDJ-RR controller
by Takashi Iwai 28 Nov '20

28 Nov '20

From: Daniel Martin <dmanlfc(a)gmail.com> This patch allows the Pioneer DJ DDJ-RR to be seen as a USB audio device under Linux and therefore usable in such applications as Mixxx. Tested Master Audio out, headphones (both output jacks) and microphone input. All work perfectly. Signed-off-by: Daniel Martin <dmanlfc(a)gmail.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/quirks-table.h | 56 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 56 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 85b99c6d3c61..7454011f6006 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3523,6 +3523,62 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), } }, +{ + /* + * PIONEER DJ DDJ-RR + * PCM is 6 channels out & 4 channels in @ 44.1 fixed + */ + USB_DEVICE_VENDOR_SPEC(0x2b73, 0x000d), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 6, //Master, Headphones & Booth + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x01, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 4, //2x RCA inputs (CH1 & CH2) + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC| + USB_ENDPOINT_SYNC_ASYNC| + USB_ENDPOINT_USAGE_IMPLICIT_FB, + .rates = SNDRV_PCM_RATE_44100, + .rate_min = 44100, + .rate_max = 44100, + .nr_rates = 1, + .rate_table = (unsigned int[]) { 44100 } + } + }, + { + .ifnum = -1 + } + } + } +}, + { /* * Pioneer DJ DJM-900NXS2 -- 2.26.2

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[PATCH 0/3] clk/sunxi/media: Fix builds with COMMON_CLK and HAVE_LEGACY_CLK
by Krzysztof Kozlowski 27 Nov '20

27 Nov '20

Hi, Multiple configurations create unbuildable config by selecting COMMON_CLK and HAVE_LEGACY_CLK. The first simply should not be selected. The patches 2/3 and 3/3 address this specific problem. I performed few compile tests and I am still building other configurations, therefore they were marked as RFC. Best regards, Krzysztof Krzysztof Kozlowski (3): clk: fix redefinition of clk_prepare on MIPS with HAVE_LEGACY_CLK ARM: sunxi: do not select COMMON_CLK to fix builds media: atomisp: do not select COMMON_CLK to fix builds arch/arm/mach-sunxi/Kconfig | 1 + drivers/clk/clk.c | 4 ++++ drivers/staging/media/atomisp/Kconfig | 2 +- sound/soc/sunxi/Kconfig | 2 +- 4 files changed, 7 insertions(+), 2 deletions(-) -- 2.25.1

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[PATCH v3 1/1] ALSA: compress: allow pause and resume during draining
by Gyeongtaek Lee 27 Nov '20

27 Nov '20

With a stream with low bitrate, user can't pause or resume the stream near the end of the stream because current ALSA doesn't allow it. If the stream has very low bitrate enough to store whole stream into the buffer, user can't do anything except stop the stream and then restart it from the first because most of applications call draining after sending last frame to the kernel. If pause, resume are allowed during draining, user experience can be enhanced. To prevent malfunction in HW drivers which don't support pause during draining, pause during draining will only work if HW driver enable this feature explicitly by calling snd_compr_use_pause_in_draining(). Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> Cc: stable(a)vger.kernel.org --- include/sound/compress_driver.h | 16 ++++++++++++++ sound/core/compress_offload.c | 39 ++++++++++++++++++++++++++------- 2 files changed, 47 insertions(+), 8 deletions(-) diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h index 70cbc5095e72..277087f635f3 100644 --- a/include/sound/compress_driver.h +++ b/include/sound/compress_driver.h @@ -67,6 +67,7 @@ struct snd_compr_runtime { * @metadata_set: metadata set flag, true when set * @next_track: has userspace signal next track transition, true when set * @partial_drain: undergoing partial_drain for stream, true when set + * @pause_in_draining: paused during draining state, true when set * @private_data: pointer to DSP private data * @dma_buffer: allocated buffer if any */ @@ -80,6 +81,7 @@ struct snd_compr_stream { bool metadata_set; bool next_track; bool partial_drain; + bool pause_in_draining; void *private_data; struct snd_dma_buffer dma_buffer; }; @@ -142,6 +144,7 @@ struct snd_compr_ops { * @direction: Playback or capture direction * @lock: device lock * @device: device id + * @use_pause_in_draining: allow pause in draining, true when set */ struct snd_compr { const char *name; @@ -152,6 +155,7 @@ struct snd_compr { unsigned int direction; struct mutex lock; int device; + bool use_pause_in_draining; #ifdef CONFIG_SND_VERBOSE_PROCFS /* private: */ char id[64]; @@ -166,6 +170,18 @@ int snd_compress_deregister(struct snd_compr *device); int snd_compress_new(struct snd_card *card, int device, int type, const char *id, struct snd_compr *compr); +/** + * snd_compr_use_pause_in_draining - Allow pause and resume in draining state + * @substream: compress substream to set + * + * Allow pause and resume in draining state. + * Only HW driver supports this transition can call this API. + */ +static inline void snd_compr_use_pause_in_draining(struct snd_compr_stream *substream) +{ + substream->device->use_pause_in_draining = true; +} + /* dsp driver callback apis * For playback: driver should call snd_compress_fragment_elapsed() to let the * framework know that a fragment has been consumed from the ring buffer diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index c1fec932c49d..debc30fcf5b3 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -709,11 +709,22 @@ static int snd_compr_pause(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_RUNNING: + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + break; + case SNDRV_PCM_STATE_DRAINING: + if (!stream->device->use_pause_in_draining) + return -EPERM; + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); + if (!retval) + stream->pause_in_draining = true; + break; + default: return -EPERM; - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_PUSH); - if (!retval) - stream->runtime->state = SNDRV_PCM_STATE_PAUSED; + } return retval; } @@ -721,11 +732,22 @@ static int snd_compr_resume(struct snd_compr_stream *stream) { int retval; - if (stream->runtime->state != SNDRV_PCM_STATE_PAUSED) + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_PAUSED: + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + break; + case SNDRV_PCM_STATE_DRAINING: + if (!stream->pause_in_draining) + return -EPERM; + retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_RELEASE); + if (!retval) + stream->pause_in_draining = false; + break; + default: return -EPERM; - retval = stream->ops->trigger(stream, SNDRV_PCM_TRIGGER_PAUSE_RELEASE); - if (!retval) - stream->runtime->state = SNDRV_PCM_STATE_RUNNING; + } return retval; } @@ -768,6 +790,7 @@ static int snd_compr_stop(struct snd_compr_stream *stream) /* clear flags and stop any drain wait */ stream->partial_drain = false; stream->metadata_set = false; + stream->pause_in_draining = false; snd_compr_drain_notify(stream); stream->runtime->total_bytes_available = 0; stream->runtime->total_bytes_transferred = 0; -- 2.21.0

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[PATCH] Asoc: qcom: Fix for problem in resume with CRAS
by Srinivasa Rao Mandadapu 27 Nov '20

27 Nov '20

To support playback continuation after resume problem in chrome audio server: Prepare device in platform trigger callback. Make I2s and DMA control registers as non volatile. Signed-off-by: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- sound/soc/qcom/lpass-cpu.c | 8 ++------ sound/soc/qcom/lpass-platform.c | 5 +++-- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index af684fd..c99be03 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -454,20 +454,16 @@ static bool lpass_cpu_regmap_volatile(struct device *dev, unsigned int reg) struct lpass_variant *v = drvdata->variant; int i; - for (i = 0; i < v->i2s_ports; ++i) - if (reg == LPAIF_I2SCTL_REG(v, i)) - return true; for (i = 0; i < v->irq_ports; ++i) if (reg == LPAIF_IRQSTAT_REG(v, i)) return true; for (i = 0; i < v->rdma_channels; ++i) - if (reg == LPAIF_RDMACURR_REG(v, i) || reg == LPAIF_RDMACTL_REG(v, i)) + if (reg == LPAIF_RDMACURR_REG(v, i)) return true; for (i = 0; i < v->wrdma_channels; ++i) - if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start) || - reg == LPAIF_WRDMACTL_REG(v, i + v->wrdma_channel_start)) + if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start)) return true; return false; diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 80b09de..2b0a7c1 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -481,8 +481,9 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, return -ENOTRECOVERABLE; } switch (cmd) { - case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + lpass_platform_pcmops_prepare(component, substream); + case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ret = regmap_fields_write(dmactl->enable, id, LPAIF_DMACTL_ENABLE_ON); @@ -592,7 +593,7 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, break; } - return 0; + return ret; } static snd_pcm_uframes_t lpass_platform_pcmops_pointer( -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH v3] ASoC: qcom: Fix playback recover problem in suspend resume
by Srinivasa Rao Mandadapu 27 Nov '20

27 Nov '20

To support playback continuation after hard suspend(bypass powerd) and resume: Prepare device in platform trigger callback. Make I2s and DMA control registers as non volatile. Signed-off-by: V Sujith Kumar Reddy <vsujithk(a)codeaurora.org> Signed-off-by: Srinivasa Rao Mandadapu <srivasam(a)codeaurora.org> --- Changes Since v1 and v2: -- Subject lines changed sound/soc/qcom/lpass-cpu.c | 8 ++------ sound/soc/qcom/lpass-platform.c | 5 +++-- 2 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index af684fd..c99be03 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -454,20 +454,16 @@ static bool lpass_cpu_regmap_volatile(struct device *dev, unsigned int reg) struct lpass_variant *v = drvdata->variant; int i; - for (i = 0; i < v->i2s_ports; ++i) - if (reg == LPAIF_I2SCTL_REG(v, i)) - return true; for (i = 0; i < v->irq_ports; ++i) if (reg == LPAIF_IRQSTAT_REG(v, i)) return true; for (i = 0; i < v->rdma_channels; ++i) - if (reg == LPAIF_RDMACURR_REG(v, i) || reg == LPAIF_RDMACTL_REG(v, i)) + if (reg == LPAIF_RDMACURR_REG(v, i)) return true; for (i = 0; i < v->wrdma_channels; ++i) - if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start) || - reg == LPAIF_WRDMACTL_REG(v, i + v->wrdma_channel_start)) + if (reg == LPAIF_WRDMACURR_REG(v, i + v->wrdma_channel_start)) return true; return false; diff --git a/sound/soc/qcom/lpass-platform.c b/sound/soc/qcom/lpass-platform.c index 80b09de..2b0a7c1 100644 --- a/sound/soc/qcom/lpass-platform.c +++ b/sound/soc/qcom/lpass-platform.c @@ -481,8 +481,9 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, return -ENOTRECOVERABLE; } switch (cmd) { - case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + lpass_platform_pcmops_prepare(component, substream); + case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: ret = regmap_fields_write(dmactl->enable, id, LPAIF_DMACTL_ENABLE_ON); @@ -592,7 +593,7 @@ static int lpass_platform_pcmops_trigger(struct snd_soc_component *component, break; } - return 0; + return ret; } static snd_pcm_uframes_t lpass_platform_pcmops_pointer( -- Qualcomm India Private Limited, on behalf of Qualcomm Innovation Center, Inc., is a member of Code Aurora Forum, a Linux Foundation Collaborative Project.

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[PATCH v1 0/1] Fix 32 bit format for adau7002
by xuyuqing 27 Nov '20

27 Nov '20

the microphone is attached to external codec(adau7002) instead of rt5682.We need to always use 32 bit format on sc7180 to meet the clock requirement of adau7002: The ADAU7002 requires a BCLK rate that is a minimum of 64× the LRCLK sample rate xuyuqing (1): ASoC: qcom: sc7180: fix 32 bit format for adau7002 sound/soc/qcom/sc7180.c | 3 +++ 1 file changed, 3 insertions(+) -- 2.25.1

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[PATCH 0/2] ASoC: soc-core: share same logic code
by Kuninori Morimoto 27 Nov '20

27 Nov '20

Hi Mark soc-core is using almost same code at some point. The difference is only used parameter. It is just duplicate code, and we can share it. This patch-set sharing code for mute and for suspend/resume. Kuninori Morimoto (2): ASoC: soc-core: add soc_playback_digital_mute() ASoC: soc-core: add soc_dapm_suspend_resume() sound/soc/soc-core.c | 84 ++++++++++++++++++++------------------------ 1 file changed, 38 insertions(+), 46 deletions(-) -- 2.25.1

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