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

[PATCH] ASoC: hdmi-codec: fix channel info for compressed formats
by Matthias Reichl 29 Jun '23

29 Jun '23

According to CTA 861 the channel/speaker allocation info in the audio infoframe only applies to uncompressed (PCM) audio streams. The channel count info should indicate the number of channels in the transmitted audio, which usually won't match the number of channels used to transmit the compressed bitstream. Some devices (eg some Sony TVs) will refuse to decode compressed audio if these values are not set correctly. To fix this we can simply set the channel count to 0 (which means "refer to stream header") and set the channel/speaker allocation to 0 as well (which would mean stereo FL/FR for PCM, a safe value all sinks will support) when transmitting compressed audio. Signed-off-by: Matthias Reichl <hias(a)horus.com> --- sound/soc/codecs/hdmi-codec.c | 36 +++++++++++++++++++++++------------ 1 file changed, 24 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/hdmi-codec.c b/sound/soc/codecs/hdmi-codec.c index 6d980fbc42077..d21f69f053422 100644 --- a/sound/soc/codecs/hdmi-codec.c +++ b/sound/soc/codecs/hdmi-codec.c @@ -495,31 +495,43 @@ static int hdmi_codec_fill_codec_params(struct snd_soc_dai *dai, struct hdmi_codec_params *hp) { struct hdmi_codec_priv *hcp = snd_soc_dai_get_drvdata(dai); - int idx; - - /* Select a channel allocation that matches with ELD and pcm channels */ - idx = hdmi_codec_get_ch_alloc_table_idx(hcp, channels); - if (idx < 0) { - dev_err(dai->dev, "Not able to map channels to speakers (%d)\n", - idx); - hcp->chmap_idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN; - return idx; + int idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN; + u8 ca_id = 0; + bool pcm_audio = !(hcp->iec_status[0] & IEC958_AES0_NONAUDIO); + + if (pcm_audio) { + /* Select a channel allocation that matches with ELD and pcm channels */ + idx = hdmi_codec_get_ch_alloc_table_idx(hcp, channels); + + if (idx < 0) { + dev_err(dai->dev, "Not able to map channels to speakers (%d)\n", + idx); + hcp->chmap_idx = HDMI_CODEC_CHMAP_IDX_UNKNOWN; + return idx; + } + + ca_id = hdmi_codec_channel_alloc[idx].ca_id; } memset(hp, 0, sizeof(*hp)); hdmi_audio_infoframe_init(&hp->cea); - hp->cea.channels = channels; + + if (pcm_audio) + hp->cea.channels = channels; + else + hp->cea.channels = 0; + hp->cea.coding_type = HDMI_AUDIO_CODING_TYPE_STREAM; hp->cea.sample_size = HDMI_AUDIO_SAMPLE_SIZE_STREAM; hp->cea.sample_frequency = HDMI_AUDIO_SAMPLE_FREQUENCY_STREAM; - hp->cea.channel_allocation = hdmi_codec_channel_alloc[idx].ca_id; + hp->cea.channel_allocation = ca_id; hp->sample_width = sample_width; hp->sample_rate = sample_rate; hp->channels = channels; - hcp->chmap_idx = hdmi_codec_channel_alloc[idx].ca_id; + hcp->chmap_idx = idx; return 0; } -- 2.39.2

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[PATCH 1/6] ASoC: amd: ps: add comments for DMA irq bits mapping
by Vijendar Mukunda 29 Jun '23

29 Jun '23

Add comments for DMA stream id and IRQ bit mapping in ACP_EXTERNAL_CNTL & ACP_EXTERNAL_CNTL1 registers for SDW0 and SDW1 manager instances. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/ps/acp63.h | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/amd/ps/acp63.h b/sound/soc/amd/ps/acp63.h index e96e6dc9d90f..733a16e23d32 100644 --- a/sound/soc/amd/ps/acp63.h +++ b/sound/soc/amd/ps/acp63.h @@ -116,8 +116,28 @@ #define ACP63_SDW0_DMA_MAX_STREAMS 6 #define ACP63_SDW1_DMA_MAX_STREAMS 2 #define ACP_P1_AUDIO_TX_THRESHOLD 6 + +/* + * Below entries describes SDW0 instance DMA stream id and DMA irq bit mapping + * in ACP_EXTENAL_INTR_CNTL register. + * Stream id IRQ Bit + * 0 (SDW0_AUDIO0_TX) 28 + * 1 (SDW0_AUDIO1_TX) 26 + * 2 (SDW0_AUDIO2_TX) 24 + * 3 (SDW0_AUDIO0_RX) 27 + * 4 (SDW0_AUDIO1_RX) 25 + * 5 (SDW0_AUDIO2_RX) 23 + */ #define SDW0_DMA_TX_IRQ_MASK(i) (ACP_AUDIO0_TX_THRESHOLD - (2 * (i))) #define SDW0_DMA_RX_IRQ_MASK(i) (ACP_AUDIO0_RX_THRESHOLD - (2 * (i))) + +/* + * Below entries describes SDW1 instance DMA stream id and DMA irq bit mapping + * in ACP_EXTENAL_INTR_CNTL1 register. + * Stream id IRQ Bit + * 0 (SDW1_AUDIO1_TX) 6 + * 1 (SDW1_AUDIO1_RX) 5 + */ #define SDW1_DMA_IRQ_MASK(i) (ACP_P1_AUDIO_TX_THRESHOLD - (i)) #define ACP_DELAY_US 5 -- 2.34.1

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[PATCH] ALSA: hda/realtek: Add quirk for Clevo NPx0SNx
by Werner Sembach 29 Jun '23

29 Jun '23

This applies a SND_PCI_QUIRK(...) to the Clevo NPx0SNx barebones fixing the microphone not being detected on the headset combo port. Signed-off-by: Werner Sembach <wse(a)tuxedocomputers.com> Cc: <stable(a)vger.kernel.org> --- 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 f595a250fab33..d2ac56398d554 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9642,6 +9642,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x971d, "Clevo N970T[CDF]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa500, "Clevo NL5[03]RU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa600, "Clevo NL50NU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0xa650, "Clevo NP[567]0SN[CD]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xa671, "Clevo NP70SN[CDE]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb018, "Clevo NP50D[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0xb019, "Clevo NH77D[BE]Q", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), -- 2.34.1

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Fix symver build error on non-ELF platforms
by GitHub pull_request - opened 29 Jun '23

29 Jun '23

alsa-project/alsa-lib pull request #334 was opened from bkuhls: The following error is observed on Microblaze [1] build: error: symver is only supported on ELF platforms due to using __attribute__((symver)) on non-ELF platform. [1] http://autobuild.buildroot.net/results/1e9/1e965d83d75615f35308440c5db04431… ac_check_attribute_symver.m4 was downloaded from https://github.com/smuellerDD/libkcapi/blob/master/m4/ac_check_attribute_sy… Request URL : https://github.com/alsa-project/alsa-lib/pull/334 Patch URL : https://github.com/alsa-project/alsa-lib/pull/334.patch Repository URL: https://github.com/alsa-project/alsa-lib

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MIC Capture Stops due to Playback
by Arjav Parikh 29 Jun '23

29 Jun '23

Hi I am using alsa api's in my C Program to implement two-way talk i.e capturing data from mic and playing sound on speaker received from Cloud. I am using a Microsemi ZL38050 Audio DSP connected to an Ambarella CV25 SOC via I2S/SPI. Randomly I have observed that If I stop and resume receiving data from Cloud to play it on speaker using snd_pcm_writei() then my capture path stops. I do not hear any sound captured from MIC. Can someone help me to debug this issue because when this issue occurs there are no errors reported from Alsa nor do I see anything in dmesg as well. Thanks & Regards, Arjav Parikh

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configure: Add large file support via CFLAGS
by GitHub pull_request - opened 29 Jun '23

29 Jun '23

alsa-project/alsa-lib pull request #333 was opened from kendase3: Currently scanelf (pax-utils) detects missing LFS support in the following files/calls: fopen,__open_2,mmap,lseek,open /usr/lib/libasound.so.2.0.0 fopen,open /usr/lib/libatopology.so.2.0.0 Note that this may cause problems on systems with a 32-bit kernel, but I've tested playing audio on a more recent 32-bit system with a 64-bit kernel. Notably AC_SYS_LARGEFILE (currently used in alsa-utils for example) did not solve the issue for my case, whereas FILE_OFFSET_BITS did. Let me know if you'd prefer an alternative approach! Request URL : https://github.com/alsa-project/alsa-lib/pull/333 Patch URL : https://github.com/alsa-project/alsa-lib/pull/333.patch Repository URL: https://github.com/alsa-project/alsa-lib

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[PATCH 0/3] ASoC: qcom: get tplg firmware-name from device tree
by Srinivas Kandagatla 28 Jun '23

28 Jun '23

With the existing setup the firmware file is looked up in a fixed location based on soundcard name, however this is not really scalable and not inline with where linux-firmware stores board specific firmwares. This patchset adds a new binding to q6apm to allow board specific Device Tree to pass firmware-name, this also makes it inline with how other firmwares are specified in qcom boards. Srinivas Kandagatla (3): ASoC: dt-bindings: q6apm: add firmware-name bindings ASoC: qcom: q6apm: add support for reading firmware name from DT arm64: dts: qcom: sc8280xp-x13s: Add audio topology firmware path .../devicetree/bindings/sound/qcom,q6apm.yaml | 5 +++++ .../dts/qcom/sc8280xp-lenovo-thinkpad-x13s.dts | 5 +++++ sound/soc/qcom/qdsp6/topology.c | 14 ++++++-------- 3 files changed, 16 insertions(+), 8 deletions(-) -- 2.21.0

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ucm2: USB-Audio: Add support for ALC4080 Gigabyte Z590 AORUS ULTRA
by GitHub pull_request - reopened 28 Jun '23

28 Jun '23

alsa-project/alsa-ucm-conf pull request #310 was reopened from BearsPunch: When installed Manjaro I had an issue my audio card was not recognized. USB ID is 0414:a00d and as I see you already have it under if.realtek-alc1220-vb condition, but I have ALC4080. So I manually changed USB-Audio.conf under If.realtek-alc4080 condition from (0414:a0(0e|1[0124])) to (0414:a0(0[de]|1[0124])) and now it works perfectly. But Ubuntu (maybe from 22.04) doesn't have this issue. Does Canonical update alsa-ucm-conf by themselves without pull request to main branch? Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/310 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/310.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH] ALSA: ump: Correct wrong byte size at converting a UMP System message
by Takashi Iwai 28 Jun '23

28 Jun '23

A wrong size for UMP_SYSTEM_STATUS_MIDI_TIME_CODE and case UMP_SYSTEM_STATUS_SONG_SELECT was reported at converting to the legacy MIDI 1.0 stream. This patch corrects the value. Fixes: 0b5288f5fe63 ("ALSA: ump: Add legacy raw MIDI support") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/ump_convert.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/ump_convert.c b/sound/core/ump_convert.c index fb61df424a87..de04799fdb69 100644 --- a/sound/core/ump_convert.c +++ b/sound/core/ump_convert.c @@ -73,7 +73,7 @@ static int cvt_ump_system_to_legacy(u32 data, unsigned char *buf) case UMP_SYSTEM_STATUS_MIDI_TIME_CODE: case UMP_SYSTEM_STATUS_SONG_SELECT: buf[1] = (data >> 8) & 0x7f; - return 1; + return 2; case UMP_SYSTEM_STATUS_SONG_POSITION: buf[1] = (data >> 8) & 0x7f; buf[2] = data & 0x7f; -- 2.35.3

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[PATCH] ALSA: fireface: make read-only const array for model names static
by Takashi Sakamoto 28 Jun '23

28 Jun '23

It is preferable not to populate the constant array for constant strings on the stack. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/fireface/ff.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/firewire/fireface/ff.c b/sound/firewire/fireface/ff.c index 82241058ea14..6e84e4787259 100644 --- a/sound/firewire/fireface/ff.c +++ b/sound/firewire/fireface/ff.c @@ -16,7 +16,7 @@ MODULE_LICENSE("GPL"); static void name_card(struct snd_ff *ff) { struct fw_device *fw_dev = fw_parent_device(ff->unit); - const char *const names[] = { + static const char *const names[] = { [SND_FF_UNIT_VERSION_FF800] = "Fireface800", [SND_FF_UNIT_VERSION_FF400] = "Fireface400", [SND_FF_UNIT_VERSION_UFX] = "FirefaceUFX", -- 2.39.2

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