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

[PATCH] ASoC: codecs/jz4760: add digital gain controls
by Christophe Branchereau 01 Feb '23

01 Feb '23

Both the DAC and ADC have digital gain controls available for their mixers, which go from -31 to 0db by step of 1db. Signed-off-by: Christophe Branchereau <cbranchereau(a)gmail.com> --- sound/soc/codecs/jz4760.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/jz4760.c b/sound/soc/codecs/jz4760.c index d96a4f6c9183..9df58e23d360 100644 --- a/sound/soc/codecs/jz4760.c +++ b/sound/soc/codecs/jz4760.c @@ -287,6 +287,7 @@ static const DECLARE_TLV_DB_MINMAX_MUTE(dac_tlv, -3100, 100); static const DECLARE_TLV_DB_SCALE(adc_tlv, 0, 100, 0); static const DECLARE_TLV_DB_MINMAX(out_tlv, -2500, 100); static const DECLARE_TLV_DB_SCALE(linein_tlv, -2500, 100, 0); +static const DECLARE_TLV_DB_MINMAX(mixer_tlv, -3100, 0); /* Unconditional controls. */ static const struct snd_kcontrol_new jz4760_codec_snd_controls[] = { @@ -299,6 +300,14 @@ static const struct snd_kcontrol_new jz4760_codec_snd_controls[] = { JZ4760_CODEC_REG_GCR4, JZ4760_CODEC_REG_GCR3, REG_GCR_GAIN_OFFSET, REG_GCR_GAIN_MAX, 1, linein_tlv), + SOC_SINGLE_TLV("Mixer Capture Volume", + JZ4760_CODEC_REG_MIX1, + REG_GCR_GAIN_OFFSET, REG_GCR_GAIN_MAX, 1, mixer_tlv), + + SOC_SINGLE_TLV("Mixer Playback Volume", + JZ4760_CODEC_REG_MIX2, + REG_GCR_GAIN_OFFSET, REG_GCR_GAIN_MAX, 1, mixer_tlv), + SOC_SINGLE("High-Pass Filter Capture Switch", JZ4760_CODEC_REG_CR4, REG_CR4_ADC_HPF_OFFSET, 1, 0), -- 2.39.0

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[PATCH] ALSA: hda/realtek: Fix the speaker output on Samsung Galaxy Book2 Pro 360
by Guillaume Pinot 01 Feb '23

01 Feb '23

Samsung Galaxy Book2 Pro 360 (13" 2022 NP930QED-KA1FR) with codec SSID 144d:ca03 requires the same workaround for enabling the speaker amp like other Samsung models with ALC298 codec. Cc: <stable(a)vger.kernel.org> Signed-off-by: Guillaume Pinot <texitoi(a)texitoi.eu> --- I've tested this fix on my laptop with success. I've took "inspiration" from https://lore.kernel.org/all/20221115170235.18875-1-tiwai@suse.de/ This is my first contribution, so feel free to give me feedbacks! 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 6fab7c8fc19a..c4496206c3e7 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9521,6 +9521,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x144d, 0xc812, "Samsung Notebook Pen S (NT950SBE-X58)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc830, "Samsung Galaxy Book Ion (NT950XCJ-X716A)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x144d, 0xc832, "Samsung Galaxy Book Flex Alpha (NP730QCJ)", ALC256_FIXUP_SAMSUNG_HEADPHONE_VERY_QUIET), + SND_PCI_QUIRK(0x144d, 0xca03, "Samsung Galaxy Book2 Pro 360 (NP930QED)", ALC298_FIXUP_SAMSUNG_AMP), SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x1462, 0xb171, "Cubi N 8GL (MS-B171)", ALC283_FIXUP_HEADSET_MIC), -- 2.30.2

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[PATCH] ASoC: dt-bindings: irondevice, sma1303.yaml: Fix about breaking the checks
by Kiseok Jo 01 Feb '23

01 Feb '23

Fix the bindings checks like syntax error. Signed-off-by: Kiseok Jo <kiseok.jo(a)irondevice.com> Reported-by: Rob Herring <robh(a)kernel.org> --- .../bindings/sound/irondevice,sma1303.yaml | 13 ++++++------- 1 file changed, 6 insertions(+), 7 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml b/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml index eb87374cc812..162c52606635 100644 --- a/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml +++ b/Documentation/devicetree/bindings/sound/irondevice,sma1303.yaml @@ -19,14 +19,13 @@ required: - compatible - reg -additionalProperties: true +additionalProperties: false examples: - | - #include <dt-bindings/gpio/gpio.h> i2c_bus { - sma1303_amp: sma1303@1e { - compatible = "irondevice,sma1303"; - reg = <0x1e>; - }; - }; + amplifier@1e { + compatible = "irondevice,sma1303"; + reg = <0x1e>; + }; + }; base-commit: eeb6b0feb597e8fb88c4e1459323060e17350752 -- 2.20.1

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[PATCH v2] ALSA: pci: lx6464es: fix a debug loop
by Dan Carpenter 01 Feb '23

01 Feb '23

This loop accidentally reuses the "i" iterator for both the inside and the outside loop. The value of MAX_STREAM_BUFFER is 5. I believe that chip->rmh.stat_len is in the 2-12 range. If the value of .stat_len is 4 or more then it will loop exactly one time, but if it's less then it is a forever loop. It looks like it was supposed to combined into one loop where conditions are checked. Fixes: 8e6320064c33 ("ALSA: lx_core: Remove useless #if 0 .. #endif") Signed-off-by: Dan Carpenter <error27(a)gmail.com> --- v2: In the first version I just deleted the outside loop but that was not the correct fix. Combine the conditions instead. sound/pci/lx6464es/lx_core.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index d3f58a3d17fb..b5b0d43bb8dc 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -493,12 +493,11 @@ int lx_buffer_ask(struct lx6464es *chip, u32 pipe, int is_capture, dev_dbg(chip->card->dev, "CMD_08_ASK_BUFFERS: needed %d, freed %d\n", *r_needed, *r_freed); - for (i = 0; i < MAX_STREAM_BUFFER; ++i) { - for (i = 0; i != chip->rmh.stat_len; ++i) - dev_dbg(chip->card->dev, - " stat[%d]: %x, %x\n", i, - chip->rmh.stat[i], - chip->rmh.stat[i] & MASK_DATA_SIZE); + for (i = 0; i < MAX_STREAM_BUFFER && i < chip->rmh.stat_len; + ++i) { + dev_dbg(chip->card->dev, " stat[%d]: %x, %x\n", i, + chip->rmh.stat[i], + chip->rmh.stat[i] & MASK_DATA_SIZE); } } -- 2.35.1

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[PATCH] ALSA: emu10k1: clean up some inconsistent indenting
by Jiapeng Chong 01 Feb '23

01 Feb '23

No functional modification involved. sound/pci/emu10k1/memory.c:446 snd_emu10k1_synth_free() warn: inconsistent indenting. Reported-by: Abaci Robot <abaci(a)linux.alibaba.com> Link: https://bugzilla.openanolis.cn/show_bug.cgi?id=3903 Signed-off-by: Jiapeng Chong <jiapeng.chong(a)linux.alibaba.com> --- sound/pci/emu10k1/memory.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index edb3f1763719..6a7406014f81 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -443,7 +443,7 @@ snd_emu10k1_synth_free(struct snd_emu10k1 *emu, struct snd_util_memblk *memblk) unmap_memblk(emu, blk); spin_unlock_irqrestore(&emu->memblk_lock, flags); synth_free_pages(emu, blk); - __snd_util_mem_free(hdr, memblk); + __snd_util_mem_free(hdr, memblk); mutex_unlock(&hdr->block_mutex); return 0; } -- 2.20.1.7.g153144c

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[PATCH] ASoC: cs42l42: use helper function
by Kuninori Morimoto 01 Feb '23

01 Feb '23

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current ASoC has many helper function. Link: https://lore.kernel.org/r/20230201111415.67bd4b7e@canb.auug.org.au Reported-by: Stephen Rothwell <sfr(a)canb.auug.org.au> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/cs42l42-sdw.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs42l42-sdw.c b/sound/soc/codecs/cs42l42-sdw.c index 79023268d4c1..7b539ee55499 100644 --- a/sound/soc/codecs/cs42l42-sdw.c +++ b/sound/soc/codecs/cs42l42-sdw.c @@ -155,10 +155,7 @@ static int cs42l42_sdw_dai_set_sdw_stream(struct snd_soc_dai *dai, void *sdw_str if (!sdw_stream) return 0; - if (direction == SNDRV_PCM_STREAM_PLAYBACK) - dai->playback_dma_data = sdw_stream; - else - dai->capture_dma_data = sdw_stream; + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); return 0; } -- 2.25.1

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[PATCH] ASoC: qcom: apq8096: set driver name correctly
by Yassine Oudjana 01 Feb '23

01 Feb '23

From: Yassine Oudjana <y.oudjana(a)protonmail.com> Set driver name to allow matching different UCM2 configurations for the multiple devices sharing the same APQ8096 ASoC. Signed-off-by: Yassine Oudjana <y.oudjana(a)protonmail.com> --- sound/soc/qcom/apq8096.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/qcom/apq8096.c b/sound/soc/qcom/apq8096.c index c7b7d0864d1a..5d07b38f6d72 100644 --- a/sound/soc/qcom/apq8096.c +++ b/sound/soc/qcom/apq8096.c @@ -113,6 +113,7 @@ static int apq8096_platform_probe(struct platform_device *pdev) if (!card) return -ENOMEM; + card->driver_name = "apq8096"; card->dev = dev; card->owner = THIS_MODULE; dev_set_drvdata(dev, card); -- 2.36.1

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Bug? Strange crash with snd_hdspm and RME RayDAT on MIDI-input
by AiO 01 Feb '23

01 Feb '23

Hello there, I've tried to find some insights in a weird problem I have with snd_hdspm. I guess it is an error-report/bug-posting. Audio-wise everything is super-duper, so don't worry. However... MIDI only works outbound from the on-board MIDI interface. MIDI-out always works perfectly from the card during normal operations. Whenever I connect anything to the MIDI-in and start feeding some MIDI-data, it seems ALSA crashes on me. here's a more detailed description: * I'm running linux-rt 6.0.5.14.realtime1-3-rt #1 SMP PREEMPT_RT (Arch) * When i give it a stream of MIDI data on the MIDI-in while playing some audio that application (e.g. mplayer or live inputs) logs: ALSA: poll time out, polled for 1999005 usecs, Retrying with a recovery, retry cnt = 1 (2, 3, 4, 5, and so on) And ALSA seem stuck in looping the last buffer over and over. * Nothing in journal, nothing in dmesg tells anything more about it What did I run in to?! :) Kind regards, AiO

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