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

[PATCH v2] ASoC: AMD Renoir - add DMI entry for Lenovo ThinkPad X395
by Jaroslav Kysela 28 Dec '20

28 Dec '20

The ThinkPad X395 latop does not have the internal digital microphone connected to the AMD's ACP bridge, but it's advertised via BIOS. The internal microphone is connected to the HDA codec. Use DMI to block the microphone PCM device for this platform. BugLink: https://bugzilla.redhat.com/show_bug.cgi?id=1892115 Cc: <stable(a)kernel.org> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> --- sound/soc/amd/renoir/rn-pci-acp3x.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/amd/renoir/rn-pci-acp3x.c b/sound/soc/amd/renoir/rn-pci-acp3x.c index 17ec35be73ac..deca8c7a0e87 100644 --- a/sound/soc/amd/renoir/rn-pci-acp3x.c +++ b/sound/soc/amd/renoir/rn-pci-acp3x.c @@ -178,6 +178,13 @@ static const struct dmi_system_id rn_acp_quirk_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "20T6CTO1WW"), } }, + { + /* Lenovo ThinkPad X395 */ + .matches = { + DMI_EXACT_MATCH(DMI_BOARD_VENDOR, "LENOVO"), + DMI_EXACT_MATCH(DMI_BOARD_NAME, "20NLCTO1WW"), + } + }, {} }; -- 2.29.2

2 2

[PATCH] ASoC: wm_adsp: Improve handling of raw byte streams
by Richard Fitzgerald 28 Dec '20

28 Dec '20

As the register map is 16-bit or 32-bit big-endian, the 24-bit DSP words appear padded and with the bytes swapped. When reading a raw stream of bytes, the pad bytes must be removed and the data bytes swapped back to their original order. The previous implementation of this assumed that the be32_to_cpu() in wm_adsp_read_data_block() would swap back to little-endian. But this is obviously only true on a little-endian CPU. It also made two walks through the data, once to endian-swap and again to strip the pad bytes. This patch re-works the code so that the endian-swap and unpad are done together in a single walk, and it is not dependent on the endianness of the CPU. The data_word_size argument to wm_adsp_remove_padding() has been dropped because currently this is always 3. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 55 +++++++++++++++++++++----------------- 1 file changed, 30 insertions(+), 25 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index bcf18bf15a02..7c52ada10af3 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3662,12 +3662,12 @@ int wm_adsp_compr_get_caps(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(wm_adsp_compr_get_caps); -static int wm_adsp_read_data_block(struct wm_adsp *dsp, int mem_type, - unsigned int mem_addr, - unsigned int num_words, u32 *data) +static int wm_adsp_read_raw_data_block(struct wm_adsp *dsp, int mem_type, + unsigned int mem_addr, + unsigned int num_words, __be32 *data) { struct wm_adsp_region const *mem = wm_adsp_find_region(dsp, mem_type); - unsigned int i, reg; + unsigned int reg; int ret; if (!mem) @@ -3680,16 +3680,22 @@ static int wm_adsp_read_data_block(struct wm_adsp *dsp, int mem_type, if (ret < 0) return ret; - for (i = 0; i < num_words; ++i) - data[i] = be32_to_cpu(data[i]) & 0x00ffffffu; - return 0; } static inline int wm_adsp_read_data_word(struct wm_adsp *dsp, int mem_type, unsigned int mem_addr, u32 *data) { - return wm_adsp_read_data_block(dsp, mem_type, mem_addr, 1, data); + __be32 raw; + int ret; + + ret = wm_adsp_read_raw_data_block(dsp, mem_type, mem_addr, 1, &raw); + if (ret) + return ret; + + *data = be32_to_cpu(raw) & 0x00ffffffu; + + return 0; } static int wm_adsp_write_data_word(struct wm_adsp *dsp, int mem_type, @@ -3722,18 +3728,22 @@ static inline int wm_adsp_buffer_write(struct wm_adsp_compr_buf *buf, buf->host_buf_ptr + field_offset, data); } -static void wm_adsp_remove_padding(u32 *buf, int nwords, int data_word_size) +static void wm_adsp_remove_padding(u32 *buf, int nwords) { - u8 *pack_in = (u8 *)buf; + const __be32 *pack_in = (__be32 *)buf; u8 *pack_out = (u8 *)buf; - int i, j; + int i; - /* Remove the padding bytes from the data read from the DSP */ + /* + * DSP words from the register map have pad bytes and the data bytes + * are in swapped order. This swaps back to the original little-endian + * order and strips the pad bytes. + */ for (i = 0; i < nwords; i++) { - for (j = 0; j < data_word_size; j++) - *pack_out++ = *pack_in++; - - pack_in += sizeof(*buf) - data_word_size; + u32 word = be32_to_cpu(*pack_in++); + *pack_out++ = (u8)word; + *pack_out++ = (u8)(word >> 8); + *pack_out++ = (u8)(word >> 16); } } @@ -3917,12 +3927,7 @@ static int wm_adsp_buffer_parse_coeff(struct wm_coeff_ctl *ctl) return -EINVAL; } - for (i = 0; i < ARRAY_SIZE(coeff_v1.name); i++) - coeff_v1.name[i] = be32_to_cpu(coeff_v1.name[i]); - - wm_adsp_remove_padding((u32 *)&coeff_v1.name, - ARRAY_SIZE(coeff_v1.name), - WM_ADSP_DATA_WORD_SIZE); + wm_adsp_remove_padding((u32 *)&coeff_v1.name, ARRAY_SIZE(coeff_v1.name)); buf->name = kasprintf(GFP_KERNEL, "%s-dsp-%s", ctl->dsp->part, (char *)&coeff_v1.name); @@ -4261,12 +4266,12 @@ static int wm_adsp_buffer_capture_block(struct wm_adsp_compr *compr, int target) return 0; /* Read data from DSP */ - ret = wm_adsp_read_data_block(buf->dsp, mem_type, adsp_addr, - nwords, compr->raw_buf); + ret = wm_adsp_read_raw_data_block(buf->dsp, mem_type, adsp_addr, + nwords, compr->raw_buf); if (ret < 0) return ret; - wm_adsp_remove_padding(compr->raw_buf, nwords, WM_ADSP_DATA_WORD_SIZE); + wm_adsp_remove_padding(compr->raw_buf, nwords); /* update read index to account for words read */ buf->read_index += nwords; -- 2.20.1

3 2

[PATCH] ASoC: meson: Use managed DMA buffer allocation
by Lars-Peter Clausen 28 Dec '20

28 Dec '20

Using a managed buffer will pre-allocate the buffer using snd_pcm_lib_preallocate_pages() and automatically free it when the PCM is destroyed. In addition it will call snd_pcm_lib_malloc_pages() before the driver's hw_params() callback and snd_pcm_lib_free_pages() after the driver's hw_free() callback. This slightly reduces the boilerplate code of the driver. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/meson/aiu-fifo-i2s.c | 1 - sound/soc/meson/aiu-fifo-spdif.c | 1 - sound/soc/meson/aiu-fifo.c | 18 ++---------------- 3 files changed, 2 insertions(+), 18 deletions(-) diff --git a/sound/soc/meson/aiu-fifo-i2s.c b/sound/soc/meson/aiu-fifo-i2s.c index d91b0d874342..2388a2d0b3a6 100644 --- a/sound/soc/meson/aiu-fifo-i2s.c +++ b/sound/soc/meson/aiu-fifo-i2s.c @@ -124,7 +124,6 @@ const struct snd_soc_dai_ops aiu_fifo_i2s_dai_ops = { .trigger = aiu_fifo_i2s_trigger, .prepare = aiu_fifo_i2s_prepare, .hw_params = aiu_fifo_i2s_hw_params, - .hw_free = aiu_fifo_hw_free, .startup = aiu_fifo_startup, .shutdown = aiu_fifo_shutdown, }; diff --git a/sound/soc/meson/aiu-fifo-spdif.c b/sound/soc/meson/aiu-fifo-spdif.c index 44eb6faacf44..2fb30f89bf7a 100644 --- a/sound/soc/meson/aiu-fifo-spdif.c +++ b/sound/soc/meson/aiu-fifo-spdif.c @@ -158,7 +158,6 @@ const struct snd_soc_dai_ops aiu_fifo_spdif_dai_ops = { .trigger = fifo_spdif_trigger, .prepare = fifo_spdif_prepare, .hw_params = fifo_spdif_hw_params, - .hw_free = aiu_fifo_hw_free, .startup = aiu_fifo_startup, .shutdown = aiu_fifo_shutdown, }; diff --git a/sound/soc/meson/aiu-fifo.c b/sound/soc/meson/aiu-fifo.c index aa88aae8e517..4ad23267cace 100644 --- a/sound/soc/meson/aiu-fifo.c +++ b/sound/soc/meson/aiu-fifo.c @@ -99,11 +99,6 @@ int aiu_fifo_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct aiu_fifo *fifo = dai->playback_dma_data; dma_addr_t end; - int ret; - - ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - if (ret < 0) - return ret; /* Setup the fifo boundaries */ end = runtime->dma_addr + runtime->dma_bytes - fifo->fifo_block; @@ -124,12 +119,6 @@ int aiu_fifo_hw_params(struct snd_pcm_substream *substream, return 0; } -int aiu_fifo_hw_free(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - return snd_pcm_lib_free_pages(substream); -} - static irqreturn_t aiu_fifo_isr(int irq, void *dev_id) { struct snd_pcm_substream *playback = dev_id; @@ -187,15 +176,12 @@ void aiu_fifo_shutdown(struct snd_pcm_substream *substream, int aiu_fifo_pcm_new(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai *dai) { - struct snd_pcm_substream *substream = - rtd->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; struct snd_card *card = rtd->card->snd_card; struct aiu_fifo *fifo = dai->playback_dma_data; size_t size = fifo->pcm->buffer_bytes_max; - snd_pcm_lib_preallocate_pages(substream, - SNDRV_DMA_TYPE_DEV, - card->dev, size, size); + snd_pcm_set_managed_buffer_all(rtd->pcm, SNDRV_DMA_TYPE_DEV, + card->dev, size, size); return 0; } -- 2.20.1

3 4

[PATCH v1] catpt: Switch to use list_entry_is_head() helper
by Andy Shevchenko 28 Dec '20

28 Dec '20

Since we got list_entry_is_head() helper in the generic header, we may switch catpt to use it. It removes the need in additional variable. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/intel/catpt/pcm.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c index e5d54bb1c42a..1c7260db3238 100644 --- a/sound/soc/intel/catpt/pcm.c +++ b/sound/soc/intel/catpt/pcm.c @@ -331,10 +331,10 @@ static int catpt_dai_apply_usettings(struct snd_soc_dai *dai, { struct catpt_dev *cdev = dev_get_drvdata(dai->dev); struct snd_soc_component *component = dai->component; - struct snd_kcontrol *pos, *kctl = NULL; const char *name; int ret; u32 id = stream->info.stream_hw_id; + struct snd_kcontrol *pos; /* only selected streams have individual controls */ switch (id) { @@ -356,17 +356,15 @@ static int catpt_dai_apply_usettings(struct snd_soc_dai *dai, list_for_each_entry(pos, &component->card->snd_card->controls, list) { if (pos->private_data == component && - !strncmp(name, pos->id.name, sizeof(pos->id.name))) { - kctl = pos; + !strncmp(name, pos->id.name, sizeof(pos->id.name))) break; - } } - if (!kctl) + if (list_entry_is_head(pos, &component->card->snd_card->controls, list)) return -ENOENT; if (stream->template->type != CATPT_STRM_TYPE_LOOPBACK) - return catpt_set_dspvol(cdev, id, (long *)kctl->private_value); - ret = catpt_ipc_mute_loopback(cdev, id, *(bool *)kctl->private_value); + return catpt_set_dspvol(cdev, id, (long *)pos->private_value); + ret = catpt_ipc_mute_loopback(cdev, id, *(bool *)pos->private_value); if (ret) return CATPT_IPC_ERROR(ret); return 0; -- 2.29.2

3 4

[PATCH v1] ASoC: sprd: Switch to use list_entry_is_head() helper
by Andy Shevchenko 28 Dec '20

28 Dec '20

Since we got list_entry_is_head() helper in the generic header, we may switch driver to use it. This eliminates the need in additional variable. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> --- sound/soc/sprd/sprd-mcdt.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/sprd/sprd-mcdt.c b/sound/soc/sprd/sprd-mcdt.c index f439e5503a3c..34b2ce733b54 100644 --- a/sound/soc/sprd/sprd-mcdt.c +++ b/sound/soc/sprd/sprd-mcdt.c @@ -866,23 +866,23 @@ EXPORT_SYMBOL_GPL(sprd_mcdt_chan_dma_disable); struct sprd_mcdt_chan *sprd_mcdt_request_chan(u8 channel, enum sprd_mcdt_channel_type type) { - struct sprd_mcdt_chan *temp, *chan = NULL; + struct sprd_mcdt_chan *temp; mutex_lock(&sprd_mcdt_list_mutex); list_for_each_entry(temp, &sprd_mcdt_chan_list, list) { if (temp->type == type && temp->id == channel) { - chan = temp; + list_del_init(&temp->list); break; } } - if (chan) - list_del(&chan->list); + if (list_entry_is_head(temp, &sprd_mcdt_chan_list, list)) + temp = NULL; mutex_unlock(&sprd_mcdt_list_mutex); - return chan; + return temp; } EXPORT_SYMBOL_GPL(sprd_mcdt_request_chan); -- 2.29.2

2 1

[PATCH v2] catpt: Switch to use list_entry_is_head() helper
by Andy Shevchenko 28 Dec '20

28 Dec '20

Since we got list_entry_is_head() helper in the generic header, we may switch catpt to use it. It removes the need in additional variable. Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Reviewed-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- v2: left the structure definition at its initial location (Cezary) sound/soc/intel/catpt/pcm.c | 12 +++++------- 1 file changed, 5 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c index e5d54bb1c42a..9d304b3a817f 100644 --- a/sound/soc/intel/catpt/pcm.c +++ b/sound/soc/intel/catpt/pcm.c @@ -331,7 +331,7 @@ static int catpt_dai_apply_usettings(struct snd_soc_dai *dai, { struct catpt_dev *cdev = dev_get_drvdata(dai->dev); struct snd_soc_component *component = dai->component; - struct snd_kcontrol *pos, *kctl = NULL; + struct snd_kcontrol *pos; const char *name; int ret; u32 id = stream->info.stream_hw_id; @@ -356,17 +356,15 @@ static int catpt_dai_apply_usettings(struct snd_soc_dai *dai, list_for_each_entry(pos, &component->card->snd_card->controls, list) { if (pos->private_data == component && - !strncmp(name, pos->id.name, sizeof(pos->id.name))) { - kctl = pos; + !strncmp(name, pos->id.name, sizeof(pos->id.name))) break; - } } - if (!kctl) + if (list_entry_is_head(pos, &component->card->snd_card->controls, list)) return -ENOENT; if (stream->template->type != CATPT_STRM_TYPE_LOOPBACK) - return catpt_set_dspvol(cdev, id, (long *)kctl->private_value); - ret = catpt_ipc_mute_loopback(cdev, id, *(bool *)kctl->private_value); + return catpt_set_dspvol(cdev, id, (long *)pos->private_value); + ret = catpt_ipc_mute_loopback(cdev, id, *(bool *)pos->private_value); if (ret) return CATPT_IPC_ERROR(ret); return 0; -- 2.29.2

2 2

cross-compilation broken since cf09b6742a507555bb712101aff9e919009f0683
by GitHub issues - opened 28 Dec '20

28 Dec '20

alsa-project/alsa-firmware issue #3 was opened from flokli: I was excited to see c9e5b369919a956cf56d50a6f0be076e364f7820 getting merged, but realized this broke immediately afterwards in cf09b6742a507555bb712101aff9e919009f0683: ``` git bisect start # good: [c9e5b369919a956cf56d50a6f0be076e364f7820] Get cross compilation to work git bisect good c9e5b369919a956cf56d50a6f0be076e364f7820 # bad: [3ab9711ae106851b4185da42d5873702a3008744] Release v1.2.4 git bisect bad 3ab9711ae106851b4185da42d5873702a3008744 # bad: [41008541de9bd2b184058ec2204f172e3c9f64fe] introduce Makefile.hotplug git bisect bad 41008541de9bd2b184058ec2204f172e3c9f64fe # bad: [546aa5b30694889b00b424a6e9de2c5f34ce5da2] introduce Makefile.fw_writer git bisect bad 546aa5b30694889b00b424a6e9de2c5f34ce5da2 # bad: [cf09b6742a507555bb712101aff9e919009f0683] upgrade ax_prog_cc_for_build.m4 to serial 18 git bisect bad cf09b6742a507555bb712101aff9e919009f0683 # first bad commit: [cf09b6742a507555bb712101aff9e919009f0683] upgrade ax_prog_cc_for_build.m4 to serial 18 ``` cc @perexg Issue URL : https://github.com/alsa-project/alsa-firmware/issues/3 Repository URL: https://github.com/alsa-project/alsa-firmware

1 0

[PATCH] dt-bindings: Drop redundant maxItems/items
by Rob Herring 28 Dec '20

28 Dec '20

'maxItems' equal to the 'items' list length is redundant. 'maxItems' is preferred for a single entry while greater than 1 should have an 'items' list. A meta-schema check for this is pending once these existing cases are fixed. Cc: Laurent Pinchart <laurent.pinchart(a)ideasonboard.com> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Jassi Brar <jaswinder.singh(a)linaro.org> Cc: dri-devel(a)lists.freedesktop.org Cc: dmaengine(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Cc: linux-usb(a)vger.kernel.org Signed-off-by: Rob Herring <robh(a)kernel.org> --- .../devicetree/bindings/display/xlnx/xlnx,zynqmp-dpsub.yaml | 1 - Documentation/devicetree/bindings/dma/renesas,rcar-dmac.yaml | 1 - Documentation/devicetree/bindings/mailbox/arm,mhu.yaml | 1 - .../devicetree/bindings/sound/nvidia,tegra30-hda.yaml | 2 -- Documentation/devicetree/bindings/usb/renesas,usb-xhci.yaml | 1 - Documentation/devicetree/bindings/usb/renesas,usbhs.yaml | 3 --- 6 files changed, 9 deletions(-) diff --git a/Documentation/devicetree/bindings/display/xlnx/xlnx,zynqmp-dpsub.yaml b/Documentation/devicetree/bindings/display/xlnx/xlnx,zynqmp-dpsub.yaml index 7b9d468c3e52..403d57977ee7 100644 --- a/Documentation/devicetree/bindings/display/xlnx/xlnx,zynqmp-dpsub.yaml +++ b/Documentation/devicetree/bindings/display/xlnx/xlnx,zynqmp-dpsub.yaml @@ -98,7 +98,6 @@ properties: maxItems: 1 dmas: - maxItems: 4 items: - description: Video layer, plane 0 (RGB or luma) - description: Video layer, plane 1 (U/V or U) diff --git a/Documentation/devicetree/bindings/dma/renesas,rcar-dmac.yaml b/Documentation/devicetree/bindings/dma/renesas,rcar-dmac.yaml index b548e4723936..c07eb6f2fc8d 100644 --- a/Documentation/devicetree/bindings/dma/renesas,rcar-dmac.yaml +++ b/Documentation/devicetree/bindings/dma/renesas,rcar-dmac.yaml @@ -73,7 +73,6 @@ properties: maxItems: 1 clock-names: - maxItems: 1 items: - const: fck diff --git a/Documentation/devicetree/bindings/mailbox/arm,mhu.yaml b/Documentation/devicetree/bindings/mailbox/arm,mhu.yaml index d43791a2dde7..d07eb00b97c8 100644 --- a/Documentation/devicetree/bindings/mailbox/arm,mhu.yaml +++ b/Documentation/devicetree/bindings/mailbox/arm,mhu.yaml @@ -61,7 +61,6 @@ properties: - description: low-priority non-secure - description: high-priority non-secure - description: Secure - maxItems: 3 clocks: maxItems: 1 diff --git a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.yaml b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.yaml index e543a6123792..b55775e21de6 100644 --- a/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.yaml +++ b/Documentation/devicetree/bindings/sound/nvidia,tegra30-hda.yaml @@ -44,7 +44,6 @@ properties: maxItems: 3 clock-names: - maxItems: 3 items: - const: hda - const: hda2hdmi @@ -54,7 +53,6 @@ properties: maxItems: 3 reset-names: - maxItems: 3 items: - const: hda - const: hda2hdmi diff --git a/Documentation/devicetree/bindings/usb/renesas,usb-xhci.yaml b/Documentation/devicetree/bindings/usb/renesas,usb-xhci.yaml index 0f078bd0a3e5..22603256ddf8 100644 --- a/Documentation/devicetree/bindings/usb/renesas,usb-xhci.yaml +++ b/Documentation/devicetree/bindings/usb/renesas,usb-xhci.yaml @@ -51,7 +51,6 @@ properties: maxItems: 1 phy-names: - maxItems: 1 items: - const: usb diff --git a/Documentation/devicetree/bindings/usb/renesas,usbhs.yaml b/Documentation/devicetree/bindings/usb/renesas,usbhs.yaml index 737c1f47b7de..54c361d4a7af 100644 --- a/Documentation/devicetree/bindings/usb/renesas,usbhs.yaml +++ b/Documentation/devicetree/bindings/usb/renesas,usbhs.yaml @@ -74,11 +74,8 @@ properties: phys: maxItems: 1 - items: - - description: phandle + phy specifier pair. phy-names: - maxItems: 1 items: - const: usb -- 2.27.0

7 9

pulse plugin crash when stream has a hole
by GitHub issues - edited 28 Dec '20

28 Dec '20

alsa-project/alsa-plugins issue #16 was edited from i-garrison: Following up on this debian bug report https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=824850 Looks like this crash is caused by missing stream hole handling in pulse plugin. Pulseaudio change was done just before `pulseaudio-v3.0` release and this issue should be reproducible with all current versions: https://gitlab.freedesktop.org/pulseaudio/pulseaudio/-/commit/dfd44036b54d6… quoting from there: ``` ... let the caller decide how the holes should be handled, so in case of holes, pa_stream_peek() will return NULL data pointer and the length of the hole in the nbytes argument ``` Here in pulse plugin, if there is a hole in stream `pa_stream_peek` would return NULL buffer pointer but non-zero `frag_length` and later memcpy is happily trying to copy from NULL source: https://github.com/alsa-project/alsa-plugins/blob/7028580da1db9a2362dd9b063… Instead, pulse plugin should detect there is a hole by checking if `src_buf` is still NULL while `frag_length` is not zero, and adjust transfer result accordingly. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/16 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0

[PATCH v3 0/1] ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger()
by Gyeongtaek Lee 28 Dec '20

28 Dec '20

>Hi Gyeongtaek > >Thank you for your help > >> I tried to find out some change from the history. >> However, I saw that dpcm_lock never be held in dpcm_be_dai_trigger() from the beginning. >> So, I just fixed some miss from the commit comment and resend. >> If you have another suggestion or request, just let me know. > >OK, thanks. >I just thought it was related to recent (especially mine :) patch. > >BYW, your posted v2 still doesn't have alignment on log. >Is it mailer issue ? Sorry. I didn't fully understand what alignment means. I fixed the log by looking at some previous commits. Thank you so much. Gyeongtaek Lee > >> And I have a question. >> This patch can?t be applied to stable branches directly by conflict. >> I found out following commits are needed to apply this patch into the 5.4 stable. >> ASoC: soc-pcm: remove snd_soc_dpcm_be_get/set_state() >> ASoC: soc-pcm: add snd_soc_dpcm_can_be() and remove duplicate code >> In this case, what should I do when stable maintainer requests resolving conflict to me? >> Should I send those patches to the maintainer? >> Or rewrite the patch on the latest of the each stable? > >Mark, please help him > >Thank you for your help !! > >Best regards >--- >Kuninori Morimoto > Gyeongtaek Lee (1): ASoC: dpcm: acquire dpcm_lock in dpcm_do_trigger() sound/soc/soc-pcm.c | 62 ++++++++++++++++++++++++++++++++------------- 1 file changed, 44 insertions(+), 18 deletions(-) base-commit: fa02fcd94b0c8dff6cc65714510cf25ad194b90d -- 2.21.0

1 0