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No audio on Acer Swift SF315-52
by GitHub issues - opened 18 Jan '25

18 Jan '25

alsa-project/alsa-ucm-conf issue #497 was opened from pavethem: [Alsa info](http://alsa-project.org/db/?f=8e6dc59510d7aaa42899f81a8f29a740032e9ce… Ever since kernel 6.12 dropped for Fedora, my laptop only shows a dummy output. This is probably due to the kernel removing legacy sound drivers and replacing them with avs, which doesn’t seem work for the kaby lake architecture present in this model. ([phoronix](https://www.phoronix.com/news/Linux-6.12-Sound-Drivers)). I am unsure whether or not I am filing this bug in the right place. Since I am able to get the microphone to work by copying a file: `cp /usr/share/alsa/ucm2/conf.d/avs_dmic/Acer-Lars-1.0.conf /usr/share/alsa/ucm2/conf.d/avs_dmic/avs_dmic.conf` I am guessing this is a case of a missing ucm file for this particular hardware. If this is incorrect, I would appreciate if someone could point me in the right direction and tell me if this is rather a bug with alsa-firmware or the kernel itself. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/497 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[GIT PULL] ASoC fixes for v6.13-rc7
by Mark Brown 18 Jan '25

18 Jan '25

The following changes since commit 139fa599cea0fd9d38e00246ea9f79af6c59acbd: ASoC: rsnd: check rsnd_adg_clk_enable() return value (2025-01-09 13:17:02 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.13-rc7 for you to fetch changes up to 75a7f7d6e7e10b9052d287be37f965103e1abae1: ASoC: Intel: sof_sdw: Fix DMI match entries for a (2025-01-15 19:50:01 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.13 A few final driver specific fixes - a couple of x86 ID field changes, plus bug fixes for simple-card-utils and nau8824. ---------------------------------------------------------------- Laurentiu Mihalcea (1): ASoC: simple-card-utils: fix priv->dai_props indexing Mark Brown (1): ASoC: Intel: sof_sdw: Fix DMI match entries for a Maxim Kochetkov (1): ASoC: codecs: nau8824: fix max volume for Speaker output Simon Trimmer (2): ASoC: Intel: sof_sdw: Fix DMI match for Lenovo 83LC ASoC: Intel: sof_sdw: Fix DMI match for Lenovo 83JX, 83MC and 83NM include/sound/simple_card_utils.h | 7 +++++++ sound/soc/codecs/nau8824.c | 8 ++++---- sound/soc/generic/simple-card-utils.c | 10 +++++----- sound/soc/intel/boards/sof_sdw.c | 23 +++++++++++++++++++---- 4 files changed, 35 insertions(+), 13 deletions(-)

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RE: [REGRESSION] Distorted sound on Acer Aspire A115-31 laptop
by Kailang 17 Jan '25

17 Jan '25

Attached. > -----Original Message----- > From: Takashi Iwai <tiwai(a)suse.de> > Sent: Thursday, January 16, 2025 10:27 PM > To: Evgeny Kapun <abacabadabacaba(a)gmail.com> > Cc: Kailang <kailang(a)realtek.com>; Takashi Iwai <tiwai(a)suse.de>; Linux > Sound Mailing List <linux-sound(a)vger.kernel.org>; Linux Kernel Mailing List > <linux-kernel(a)vger.kernel.org>; Linux Regressions Mailing List > <regressions(a)lists.linux.dev>; Linux Stable Mailing List > <stable(a)vger.kernel.org> > Subject: Re: [REGRESSION] Distorted sound on Acer Aspire A115-31 laptop > > > External mail. > > > > On Sat, 11 Jan 2025 16:00:33 +0100, > Evgeny Kapun wrote: > > > > On 12/24/24 04:54, Kailang wrote: > > > Please test attach patch. > > > > This patch, when applied to kernel version 6.12.8, appears to fix the > > issue. There are no distortions, and the left and the right channel > > can be controlled independently. > > Good to hear. > > Kailang, care to submit a proper patch for merging? > > > thanks, > > Takashi

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Topology: NHLT: Intel: Fix DMA slots config in SSP blob
by GitHub pull_request - opened 16 Jan '25

16 Jan '25

alsa-project/alsa-utils pull request #289 was opened from singalsu: This fixes a mistake in function ssp_calculate_intern_v30(). The rx_dir[0] and tx_dir[0] value was hard-coded to 0x3 for typical stereo slots configuration. It should be instead retrieved from topology parameters for the SSP. Request URL : https://github.com/alsa-project/alsa-utils/pull/289 Patch URL : https://github.com/alsa-project/alsa-utils/pull/289.patch Repository URL: https://github.com/alsa-project/alsa-utils

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no internal sound with snd_hda_intel, ALC256/ Lenovo ThinkCentre M90a Pro Gen4
by Ronny Sattler 15 Jan '25

15 Jan '25

Hello, I have a problem with the snd-hda-intel. There is no sound. Only when I plug in headphones do I hear the sound on the headphones. The speakers are onboard (All in One - PC) and always remain mute. I suspect it is because the amplifier is not activated or a PIN assignment is incorrect. Please let me know if there is anything else I can do for you on the PC. Attached are previous log files and general information. Thank you very much for your work :-) Best regards Ronny

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[RFC PATCH] soc: audio-graph-card2: use correct endpoint when getting link parameters
by Ivaylo Dimitrov 15 Jan '25

15 Jan '25

We may have multiple links between ports, with each link having different parameters. Currently, no matter the topology, it is always port endpoint 0 that is used when setting parameters. On a complex sound system, like the one found on Motorola droid4, hifi and voice DAIs require differents formats (i2s vs dsp_a) and curently it is impossible to use DT to set that. Implementing the change leads to partially dropping of at least 0dedbde5062d (ASoC: cpcap: Implement set_tdm_slot for voice call support), as core does most of what is needed to configure voice DAI. We (on Maemo Leste ) use the patch (along with few others) to have voice calls working properly on d4 through UCM. The patch is for linux 6.6, I want to know whether the approach would be accepted before sending a proper patch for current master. the original commit message follows: When link parameters are parsed, it is always endpoint@0 that is used and parameters set to other endpoints are ignored. Fix that by using endpoint that is set in DT when parsing link parameters. Signed-off-by: Ivaylo Dimitrov <ivo.g.dimitrov.75(a)gmail.com> --- sound/soc/generic/audio-graph-card2.c | 59 +++++++++++++-------------- 1 file changed, 28 insertions(+), 31 deletions(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index b1c675c6b6db..163a20c8ffee 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -508,17 +508,16 @@ static int __graph_parse_node(struct asoc_simple_priv *priv, static int graph_parse_node(struct asoc_simple_priv *priv, enum graph_type gtype, - struct device_node *port, + struct device_node *ep, struct link_info *li, int is_cpu) { - struct device_node *ep; int ret = 0; + struct device_node *port = of_get_parent(ep); + bool is_multi = graph_lnk_is_multi(port); - if (graph_lnk_is_multi(port)) { + if (is_multi) { int idx; - of_node_get(port); - for (idx = 0;; idx++) { ep = graph_get_next_multi_ep(&port); if (!ep) @@ -532,9 +531,8 @@ static int graph_parse_node(struct asoc_simple_priv *priv, } } else { /* Single CPU / Codec */ - ep = port_to_endpoint(port); + of_node_put(port); ret = __graph_parse_node(priv, gtype, ep, li, is_cpu, 0); - of_node_put(ep); } return ret; @@ -591,22 +589,20 @@ static void graph_parse_daifmt(struct device_node *node, } static void graph_link_init(struct asoc_simple_priv *priv, - struct device_node *port, + struct device_node *ep, struct link_info *li, int is_cpu_node) { struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); - struct device_node *ep; + struct device_node *port = of_get_parent(ep); + bool is_multi = graph_lnk_is_multi(port); struct device_node *ports; unsigned int daifmt = 0, daiclk = 0; unsigned int bit_frame = 0; - if (graph_lnk_is_multi(port)) { - of_node_get(port); + if (is_multi) { ep = graph_get_next_multi_ep(&port); port = of_get_parent(ep); - } else { - ep = port_to_endpoint(port); } ports = of_get_parent(port); @@ -642,6 +638,9 @@ static void graph_link_init(struct asoc_simple_priv *priv, dai_link->ops = &graph_ops; if (priv->ops) dai_link->ops = priv->ops; + + of_node_put(port); + of_node_put(ports); } int audio_graph2_link_normal(struct asoc_simple_priv *priv, @@ -650,7 +649,7 @@ int audio_graph2_link_normal(struct asoc_simple_priv *priv, { struct device_node *cpu_port = lnk; struct device_node *cpu_ep = port_to_endpoint(cpu_port); - struct device_node *codec_port = of_graph_get_remote_port(cpu_ep); + struct device_node *codec_ep = of_graph_get_remote_endpoint(cpu_ep); int ret; /* @@ -658,20 +657,20 @@ int audio_graph2_link_normal(struct asoc_simple_priv *priv, * see * __graph_parse_node() :: DAI Naming */ - ret = graph_parse_node(priv, GRAPH_NORMAL, codec_port, li, 0); + ret = graph_parse_node(priv, GRAPH_NORMAL, codec_ep, li, 0); if (ret < 0) goto err; /* * call CPU, and set DAI Name */ - ret = graph_parse_node(priv, GRAPH_NORMAL, cpu_port, li, 1); + ret = graph_parse_node(priv, GRAPH_NORMAL, cpu_ep, li, 1); if (ret < 0) goto err; - graph_link_init(priv, cpu_port, li, 1); + graph_link_init(priv, cpu_ep, li, 1); err: - of_node_put(codec_port); + of_node_put(codec_ep); of_node_put(cpu_ep); return ret; @@ -684,7 +683,6 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, { struct device_node *ep = port_to_endpoint(lnk); struct device_node *rep = of_graph_get_remote_endpoint(ep); - struct device_node *rport = of_graph_get_remote_port(ep); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); int is_cpu = asoc_graph_is_ports0(lnk); @@ -718,7 +716,7 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, dai_link->dynamic = 1; dai_link->dpcm_merged_format = 1; - ret = graph_parse_node(priv, GRAPH_DPCM, rport, li, 1); + ret = graph_parse_node(priv, GRAPH_DPCM, rep, li, 1); if (ret) goto err; } else { @@ -751,7 +749,7 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = asoc_simple_be_hw_params_fixup; - ret = graph_parse_node(priv, GRAPH_DPCM, rport, li, 0); + ret = graph_parse_node(priv, GRAPH_DPCM, rep, li, 0); if (ret < 0) goto err; } @@ -761,11 +759,10 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, snd_soc_dai_link_set_capabilities(dai_link); - graph_link_init(priv, rport, li, is_cpu); + graph_link_init(priv, rep, li, is_cpu); err: of_node_put(ep); of_node_put(rep); - of_node_put(rport); return ret; } @@ -777,7 +774,7 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, { struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct device_node *port0, *port1, *ports; - struct device_node *codec0_port, *codec1_port; + struct device_node *codec0_ep, *codec1_ep; struct device_node *ep0, *ep1; u32 val = 0; int ret = -EINVAL; @@ -834,31 +831,31 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, ep0 = port_to_endpoint(port0); ep1 = port_to_endpoint(port1); - codec0_port = of_graph_get_remote_port(ep0); - codec1_port = of_graph_get_remote_port(ep1); + codec0_ep = of_graph_get_remote_endpoint(ep0); + codec1_ep = of_graph_get_remote_endpoint(ep1); /* * call Codec first. * see * __graph_parse_node() :: DAI Naming */ - ret = graph_parse_node(priv, GRAPH_C2C, codec1_port, li, 0); + ret = graph_parse_node(priv, GRAPH_C2C, codec1_ep, li, 0); if (ret < 0) goto err2; /* * call CPU, and set DAI Name */ - ret = graph_parse_node(priv, GRAPH_C2C, codec0_port, li, 1); + ret = graph_parse_node(priv, GRAPH_C2C, codec0_ep, li, 1); if (ret < 0) goto err2; - graph_link_init(priv, codec0_port, li, 1); + graph_link_init(priv, codec0_ep, li, 1); err2: of_node_put(ep0); of_node_put(ep1); - of_node_put(codec0_port); - of_node_put(codec1_port); + of_node_put(codec0_ep); + of_node_put(codec1_ep); err1: of_node_put(ports); of_node_put(port0); -- 2.30.2

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[PATCH v2 0/5] Updates for rawmidi / sequencer
by Takashi Iwai 14 Jan '25

14 Jan '25

The kernel 6.14 will receive a few small enhancements for rawmidi and sequencer APIs, and this patch set covers those. v1->v2: sync with the latest rawmidi API change to pass 0 to tied_device as default v1: https://lore.kernel.org/20250114085515.11880-1-tiwai@suse.de Takashi === Takashi Iwai (5): Sync UAPI asound.h with 6.14 kernel Sync UAPI asequencer.h with 6.14 kernel rawmidi: Extensions for tied device and substream inactive flag rawmidi: Make rawmidi flag bits doxygen-style comments seq: Define new events for UMP EP/FB change notifications include/local.h | 1 + include/rawmidi.h | 4 +++- include/seq_event.h | 12 ++++++++++++ include/sound/uapi/asequencer.h | 12 +++++++++++- include/sound/uapi/asound.h | 8 ++++++-- src/rawmidi/rawmidi.c | 16 ++++++++++++++++ 6 files changed, 49 insertions(+), 4 deletions(-) -- 2.43.0

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[PATCH alsa-lib 0/5] Updates for rawmidi / sequencer
by Takashi Iwai 14 Jan '25

14 Jan '25

The kernel 6.14 will receive a few small enhancements for rawmidi and sequencer APIs, and this patch set covers those. Takashi === Takashi Iwai (5): Sync UAPI asound.h with 6.14 kernel Sync UAPI asequencer.h with 6.14 kernel rawmidi: Extensions for tied device and substream inactive flag rawmidi: Make rawmidi flag bits doxygen-style comments seq: Define new events for UMP EP/FB change notifications include/local.h | 1 + include/rawmidi.h | 4 +++- include/seq_event.h | 12 ++++++++++++ include/sound/uapi/asequencer.h | 12 +++++++++++- include/sound/uapi/asound.h | 8 ++++++-- src/rawmidi/rawmidi.c | 22 +++++++++++++++++++++- 6 files changed, 54 insertions(+), 5 deletions(-) -- 2.43.0

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rt722: change output volume of headphone to 0dB
by GitHub pull_request - opened 14 Jan '25

14 Jan '25

alsa-project/alsa-ucm-conf pull request #496 was opened from shumingfan: rt722: change output volume of headphone to 0dB Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/496 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/496.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH v2 1/1] ASoC: codecs: nau8824: fix max volume for Speaker output
by Maxim Kochetkov 13 Jan '25

13 Jan '25

There is no audio output if Speaker volume is set above 25. According to datasheet Rev 2.5 maximum allowed value for the Speaker output is 0b11001 (25) 0x6D CLASSD_GAIN_1/ 0x6E CLASSD_GAIN_2: Left/Right Channel Class-D Driver Gain For DAC Left/Right Input (Step size is 1dB.) 00000 = Mute (DEFAULT) 00001 = 0dB 00002 = 1dB ... 11000 = 23dB 11001 = 24dB So adjust this value in accordance with the datasheet. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- sound/soc/codecs/nau8824.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 12540397fd4d..5aaf8c496300 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -368,13 +368,13 @@ static const struct snd_kcontrol_new nau8824_snd_controls[] = { SOC_ENUM("DAC Oversampling Rate", nau8824_dac_oversampl_enum), SOC_SINGLE_TLV("Speaker Right DACR Volume", - NAU8824_REG_CLASSD_GAIN_1, 8, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_1, 8, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Left DACL Volume", - NAU8824_REG_CLASSD_GAIN_2, 0, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_2, 0, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Left DACR Volume", - NAU8824_REG_CLASSD_GAIN_1, 0, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_1, 0, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Right DACL Volume", - NAU8824_REG_CLASSD_GAIN_2, 8, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_2, 8, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Headphone Right DACR Volume", NAU8824_REG_ATT_PORT0, 8, 0x1f, 0, hp_vol_tlv), -- 2.45.2

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