January 2025 - Alsa-devel - mailman.alsa-project.org

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

2 1

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

1 0

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

1 0

[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

3 9

[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

1 5

[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

1 5

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

1 0

[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

2 1

speaker-test.1: Some remarks and a patch with editorial changes for this man page
by Bjarni Ingi Gislason 11 Jan '25

11 Jan '25

Package: alsa-utils Version: 1.2.13-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with a new version test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -rCHECKSTYLE=10 -ww -z < "man page" [Use "groff -e ' $' <file>" to find trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? an.tmac:<stdin>:183: misuse, warning: .BR is for at least 2 arguments, got 1 Use macro '.B' for one argument or split argument. * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: trixie/sid APT prefers testing APT policy: (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.12.6-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 33+20240816-2 ii libasound2t64 1.2.13-1+b1 ii libatopology2t64 1.2.13-1+b1 ii libc6 2.40-4 ii libfftw3-single3 3.3.10-2+b1 ii libncursesw6 6.5-2+b1 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5-2+b1 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

1 0

[GIT PULL] ASoC fixes for v6.13-rc6
by Mark Brown 10 Jan '25

10 Jan '25

The following changes since commit 32c9c06adb5b157ef259233775a063a43746d699: ASoC: mediatek: disable buffer pre-allocation (2024-12-19 11:15:09 +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-rc6 for you to fetch changes up to 139fa599cea0fd9d38e00246ea9f79af6c59acbd: ASoC: rsnd: check rsnd_adg_clk_enable() return value (2025-01-09 13:17:02 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.13 A collection of device specific fixes that came in over the holidays, plus a MAINTAINERS update and some documentation to help users debug problems with some of the Cirrus CODECs found in modern laptops. ---------------------------------------------------------------- Charles Keepax (3): ASoC: wm8994: Add depends on MFD core ASoC: samsung: Add missing selects for MFD_WM8994 ASoC: samsung: Add missing depends on I2C Kuninori Morimoto (2): MAINTAINERS: add missing maintainers for Simple Audio Card ASoC: rsnd: check rsnd_adg_clk_enable() return value Maciej Strozek (1): ASoC: cs42l43: Add codec force suspend/resume ops Marian Postevca (1): ASoC: codecs: es8316: Fix HW rate calculation for 48Mhz MCLK Richard Fitzgerald (2): ALSA: doc: cs35l56: Add information about Cirrus Logic CS35L54/56/57 ALSA: doc: Add codecs/index.rst to top-level index Shenghao Ding (1): ASoC: tas2781: Fix occasional calibration failture Zhang Yi (1): ASoC: codecs: ES8326: Adjust ANA_MICBIAS to reduce pop noise Documentation/sound/codecs/cs35l56.rst | 292 +++++++++++++++++++++++++++++++++ Documentation/sound/codecs/index.rst | 9 + Documentation/sound/index.rst | 1 + MAINTAINERS | 2 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/cs42l43.c | 1 + sound/soc/codecs/es8316.c | 10 +- sound/soc/codecs/es8326.c | 23 ++- sound/soc/codecs/tas2781-i2c.c | 2 +- sound/soc/renesas/rcar/adg.c | 28 +++- sound/soc/renesas/rcar/core.c | 4 +- sound/soc/renesas/rcar/rsnd.h | 2 +- sound/soc/samsung/Kconfig | 6 +- 13 files changed, 364 insertions(+), 17 deletions(-) create mode 100644 Documentation/sound/codecs/cs35l56.rst create mode 100644 Documentation/sound/codecs/index.rst

2 1

alsactl info fails with exit code 25
by GitHub issues - opened 09 Jan '25

09 Jan '25

alsa-project/alsa-utils issue #287 was opened from z-s-e: Here is the output of running alsactl info on my machine: ``` $ alsactl info 0 # # Sound card # - card: 0 id: Generic name: HD-Audio Generic longname: HD-Audio Generic at 0xb88c8000 irq 87 driver_name: HDA-Intel mixer_name: ATI R6xx HDMI components: HDA:1002aa01,00aa0100,00100800 controls_count: 15 pcm: - stream: PLAYBACK devices: - device: 3 id: HDMI 0 name: G246HYL subdevices: - subdevice: 0 name: subdevice #0 - device: 7 id: HDMI 1 name: HDMI 1 subdevices: - subdevice: 0 name: subdevice #0 alsactl: rawmidi_device_list:105: snd_ctl_rawmidi_next_device $ echo $? 25 ``` Apparently it fails when trying to enumerate raw midi devices. I don't think this is supposed to happen, right? alsa-info for reference: [alsa-info.txt](https://github.com/user-attachments/files/18362369/alsa-info… Issue URL : https://github.com/alsa-project/alsa-utils/issues/287 Repository URL: https://github.com/alsa-project/alsa-utils

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[PATCH v1] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 09 Jan '25

09 Jan '25

Due to internal re-org, Kevin has no longer maintained audio driver. With the job responsibility change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- MAINTAINERS | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 30cbc3d44cd5..4c806ad2fb16 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -23172,25 +23172,42 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: linux-sound(a)vger.kernel.org -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml F: Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) and Touch Screen DRIVERS +L: linux-sound(a)vger.kernel.org +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: include/sound/tas2*.h +F: drivers/input/touchscreen/tsc2*.* F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -23200,9 +23217,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> -- 2.34.1

1 0

Topology: NHLT: Intel: Fix mono DMIC configure for MTL platform
by GitHub pull_request - opened 08 Jan '25

08 Jan '25

alsa-project/alsa-utils pull request #286 was opened from singalsu: This change fixes the blob generator for mono microphone configuration. As difference to previous platforms the FIFO packer mono/stereo mode set up in OUTCONTROLx IPM_SOURCE_MODE bit-field. Request URL : https://github.com/alsa-project/alsa-utils/pull/286 Patch URL : https://github.com/alsa-project/alsa-utils/pull/286.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aseqdump version 1.2.11, There are small bugs
by GitHub issues - opened 07 Jan '25

07 Jan '25

alsa-project/alsa-utils issue #285 was opened from masahirokakishita: Hello, I found small bugs in aseqdump.c static void dump_ump_midi2_event(const unsigned int *ump) ... case SND_UMP_MSG_PITCHBEND: printf("Channel pressure %2d, value %s", channel, midi2_pitchbend(m->channel_pressure.data)); break; It is not "Channel pressure", but "Pitch Bend". case SND_UMP_MSG_NOTE_ON: printf("Note on %2d, note %d, velocity %s, attr type = %d, data = 0x%x", channel, m->note_off.note, midi2_velocity(m->note_off.velocity), m->note_off.attr_type, m->note_off.attr_data); break; "note->on" is correct. This bug does not affect the display. Issue URL : https://github.com/alsa-project/alsa-utils/issues/285 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: rt5682: Support the ALC5682I-VE codec
by derek.fang＠realtek.com 07 Jan '25

07 Jan '25

From: Derek Fang <derek.fang(a)realtek.com> The ALC5682I-VD and ALC5682I-VE use the same I2C codec driver with different calibration settings. This patch aims to handle their differences. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index ff9e14fad0cd..a8820435d1e0 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -186,6 +186,12 @@ static int rt5682_i2c_probe(struct i2c_client *i2c) return -ENODEV; } + regmap_read(rt5682->regmap, RT5682_INT_DEVICE_ID, &val); + if (val == 0x6956) { + dev_dbg(&i2c->dev, "ALC5682I-VE device\n"); + rt5682->ve_ic = true; + } + mutex_init(&rt5682->calibrate_mutex); rt5682_calibrate(rt5682); diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index aa163ec40862..b4d72fc4a44d 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -395,6 +395,7 @@ bool rt5682_volatile_register(struct device *dev, unsigned int reg) case RT5682_4BTN_IL_CMD_1: case RT5682_AJD1_CTRL: case RT5682_HP_CALIB_CTRL_1: + case RT5682_INT_DEVICE_ID: case RT5682_DEVICE_ID: case RT5682_I2C_MODE: case RT5682_HP_CALIB_CTRL_10: @@ -419,6 +420,7 @@ bool rt5682_readable_register(struct device *dev, unsigned int reg) { switch (reg) { case RT5682_RESET: + case RT5682_INT_DEVICE_ID: case RT5682_VERSION_ID: case RT5682_VENDOR_ID: case RT5682_DEVICE_ID: @@ -3139,7 +3141,10 @@ void rt5682_calibrate(struct rt5682_priv *rt5682) regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0100); regmap_write(rt5682->regmap, RT5682_HP_IMP_SENS_CTRL_19, 0x3800); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000); - regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005); + if (rt5682->ve_ic) + regmap_write(rt5682->regmap, RT5682_CHOP_ADC, 0x7005); + else + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005); regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c); regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321); @@ -3168,7 +3173,10 @@ void rt5682_calibrate(struct rt5682_priv *rt5682) regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x0000); regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000); - regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); + if (rt5682->ve_ic) + regmap_write(rt5682->regmap, RT5682_CHOP_ADC, 0x2005); + else + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0c0c); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index b2d9e87af259..de43a5d99403 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -22,6 +22,7 @@ /* Info */ #define RT5682_RESET 0x0000 +#define RT5682_INT_DEVICE_ID 0x00f9 #define RT5682_VERSION_ID 0x00fd #define RT5682_VENDOR_ID 0x00fe #define RT5682_DEVICE_ID 0x00ff @@ -1446,6 +1447,7 @@ struct rt5682_priv { bool hw_init; bool first_hw_init; bool is_sdw; + bool ve_ic; #ifdef CONFIG_COMMON_CLK struct clk_hw dai_clks_hw[RT5682_DAI_NUM_CLKS]; -- 2.34.1

2 1

[PATCH v1] ASoC: tas2781: Fix occasional calibration failture
by Shenghao Ding 07 Jan '25

07 Jan '25

The root cause is that TAS2781 internal register is not unlocked before writing data into TAS2781_PRM_TEST_57_REG. Fixes: 49e2e353fb0d ("ASoC: tas2781: Add Calibration Kcontrols for Chromebook") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index fb8cd2284fe8..728bf78ae71f 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -78,7 +78,7 @@ static const struct bulk_reg_val tas2781_cali_start_reg[] = { X2781_CL_STT_VAL(TAS2781_PRM_INT_MASK_REG, 0xfe, false), X2781_CL_STT_VAL(TAS2781_PRM_CLK_CFG_REG, 0xdd, false), X2781_CL_STT_VAL(TAS2781_PRM_RSVD_REG, 0x20, false), - X2781_CL_STT_VAL(TAS2781_PRM_TEST_57_REG, 0x14, false), + X2781_CL_STT_VAL(TAS2781_PRM_TEST_57_REG, 0x14, true), X2781_CL_STT_VAL(TAS2781_PRM_TEST_62_REG, 0x45, true), X2781_CL_STT_VAL(TAS2781_PRM_PVDD_UVLO_REG, 0x03, false), X2781_CL_STT_VAL(TAS2781_PRM_CHNL_0_REG, 0xa8, false), -- 2.34.1

2 1

[PATCH RESEND] ASoC: codecs: es8316: Fix HW rate calculation for 48Mhz MCLK
by Marian Postevca 07 Jan '25

07 Jan '25

For 48Mhz MCLK systems the calculation of the HW rate is broken, and will not produce even one sane rate. Since es83xx supports the option to halve MCLK, calculate also rates with MCLK/2. Signed-off-by: Marian Postevca <posteuca(a)mutex.one> --- sound/soc/codecs/es8316.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 61729e5b50a8..f508df01145b 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -39,7 +39,9 @@ struct es8316_priv { struct snd_soc_jack *jack; int irq; unsigned int sysclk; - unsigned int allowed_rates[ARRAY_SIZE(supported_mclk_lrck_ratios)]; + /* ES83xx supports halving the MCLK so it supports twice as many rates + */ + unsigned int allowed_rates[ARRAY_SIZE(supported_mclk_lrck_ratios) * 2]; struct snd_pcm_hw_constraint_list sysclk_constraints; bool jd_inverted; }; @@ -386,6 +388,12 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, if (freq % ratio == 0) es8316->allowed_rates[count++] = freq / ratio; + + /* We also check if the halved MCLK produces a valid rate + * since the codec supports halving the MCLK. + */ + if ((freq / ratio) % 2 == 0) + es8316->allowed_rates[count++] = freq / ratio / 2; } if (count) { -- 2.45.2

2 1

[PATCH] ASoC: codecs: ES8326: Adjust ANA_MICBIAS to reduce pop noise
by Zhang Yi 07 Jan '25

07 Jan '25

We modify the value of ES8326_ANA_MICBIAS to reduce the pop noise Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 23 +++++++++++++++++++---- 1 file changed, 19 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index a5603b617688..b06eead7e0f6 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -616,7 +616,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) 0x0F, 0x0F); if (es8326->version > ES8326_VERSION_B) { regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30); } } } else { @@ -631,6 +631,8 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) regmap_write(es8326->regmap, ES8326_HPR_OFFSET_INI, offset_r); es8326->calibrated = true; } + regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0x00); + regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x00); if (direction == SNDRV_PCM_STREAM_PLAYBACK) { regmap_update_bits(es8326->regmap, ES8326_DAC_DSM, 0x01, 0x01); usleep_range(1000, 5000); @@ -645,7 +647,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) } else { msleep(300); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x50); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x70); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x00); } regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE, @@ -676,6 +678,10 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x00); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x20); regmap_update_bits(es8326->regmap, ES8326_RESET, 0x02, 0x00); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30); + } break; case SND_SOC_BIAS_PREPARE: break; @@ -683,6 +689,12 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x3b); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x00); regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, ES8326_IO_INPUT); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); + } + regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0xc0); + regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x80); break; case SND_SOC_BIAS_OFF: clk_disable_unprepare(es8326->mclk); @@ -773,7 +785,10 @@ static void es8326_jack_button_handler(struct work_struct *work) case 0x6f: case 0x4b: /* button volume up */ - cur_button = SND_JACK_BTN_1; + if ((iface == 0x6f) && (es8326->version > ES8326_VERSION_B)) + cur_button = SND_JACK_BTN_0; + else + cur_button = SND_JACK_BTN_1; break; case 0x27: /* button volume down */ @@ -1082,7 +1097,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); es8326_disable_micbias(es8326->component); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x13); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x10); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); } -- 2.17.1

2 1

headers: make more headers self-contained
by GitHub pull_request - opened 06 Jan '25

06 Jan '25

alsa-project/alsa-lib pull request #435 was opened from timblechmann: Some IDEs parse headers in isolation to populate their code model. This exposes some missing headers, mainly for alsa internal struct definitions, ssize_t (requiring sys/types.h), size_t (requiring string.h), va_args (requiring stdargs.h) or FILE (requiring stdio.h) This patch makes the main headers parsable by qtcreator Request URL : https://github.com/alsa-project/alsa-lib/pull/435 Patch URL : https://github.com/alsa-project/alsa-lib/pull/435.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

bug in the asla driver asla-sink-viritio-snd
by GitHub issues - opened 05 Jan '25

05 Jan '25

alsa-project/alsa-lib issue #434 was opened from LoreFiore: <img width="1440" alt="Screenshot 2025-01-05 alle 23 31 08" src="https://github.com/user-attachments/assets/34b279a0-b256-4e7b-82ec-a9df46f9…" /> Issue URL : https://github.com/alsa-project/alsa-lib/issues/434 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ld.lld default version symbol snd_dlopen@@ALSA_1.1.6 must be defined
by GitHub issues - edited 05 Jan '25

05 Jan '25

alsa-project/alsa-lib issue #433 was edited from trcrsired: ``` clang: warning: argument unused during compilation: '--unwindlib=libunwind' [-Wunused-command-line-argument] CCLD libasound.la ld.lld: error: seq/.libs/libseq.a(seq.o at 50494) <inline asm>:1:1: default version symbol snd_dlopen@@ALSA_1.1.6 must be defined .symver __snd_dlopen, snd_dlopen@@ALSA_1.1.6 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:1:1: default version symbol snd_pcm_hw_params_set_format_first@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_format_first, snd_pcm_hw_params_set_format_first@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:2:1: default version symbol snd_pcm_hw_params_set_channels_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_channels_near, snd_pcm_hw_params_set_channels_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:3:1: default version symbol snd_pcm_hw_params_set_rate_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_rate_near, snd_pcm_hw_params_set_rate_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:4:1: default version symbol snd_pcm_hw_params_set_period_time_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_period_time_near, snd_pcm_hw_params_set_period_time_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:5:1: default version symbol snd_pcm_hw_params_set_period_size_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_period_size_near, snd_pcm_hw_params_set_period_size_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:6:1: default version symbol snd_pcm_hw_params_set_periods_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_periods_near, snd_pcm_hw_params_set_periods_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:7:1: default version symbol snd_pcm_hw_params_set_buffer_time_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_buffer_time_near, snd_pcm_hw_params_set_buffer_time_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:8:1: default version symbol snd_pcm_hw_params_set_buffer_size_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_buffer_size_near, snd_pcm_hw_params_set_buffer_size_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:9:1: default version symbol snd_timer_params_set_filter@@ALSA_0.9.0 must be defined .symver __snd_timer_params_set_filter, snd_timer_params_set_filter@@ALSA_0.9.0 ^ ld.lld: error: .libs/conf.o <inline asm>:1:1: default version symbol snd_dlopen@@ALSA_1.1.6 must be defined .symver __snd_dlopen, snd_dlopen@@ALSA_1.1.6 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_hw.o at 720480) <inline asm>:1:1: default version symbol snd_timer_params_set_filter@@ALSA_0.9.0 must be defined .symver __snd_timer_params_set_filter, snd_timer_params_set_filter@@ALSA_0.9.0 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:1:1: default version symbol snd_pcm_hw_params_get_access@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_access, snd_pcm_hw_params_get_access@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:2:1: default version symbol snd_pcm_hw_params_get_format@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_format, snd_pcm_hw_params_get_format@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:3:1: default version symbol snd_pcm_hw_params_get_subformat@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_subformat, snd_pcm_hw_params_get_subformat@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:4:1: default version symbol snd_pcm_hw_params_get_channels@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_channels, snd_pcm_hw_params_get_channels@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:5:1: default version symbol snd_pcm_hw_params_get_rate@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_rate, snd_pcm_hw_params_get_rate@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:6:1: default version symbol snd_pcm_hw_params_get_period_time@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_period_time, snd_pcm_hw_params_get_period_time@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:7:1: default version symbol snd_pcm_hw_params_get_period_size@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_period_size, snd_pcm_hw_params_get_period_size@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:8:1: default version symbol snd_pcm_hw_params_get_buffer_size@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_buffer_size, snd_pcm_hw_params_get_buffer_size@@ALSA_0.9.0rc4 ^ ld.lld: error: too many errors emitted, stopping now (use --error-limit=0 to see all errors) clang: error: linker command failed with exit code 1 (use -v to see invocation) make[3]: *** [Makefile:506: libasound.la] Error 1 make[3]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make[2]: *** [Makefile:567: all-recursive] Error 1 make[2]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make[1]: *** [Makefile:436: all] Error 2 make[1]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make: *** [Makefile:414: all-recursive] Error 1 alsa-lib build failure ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/433 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

default version symbol snd_dlopen@@ALSA_1.1.6 must be defined
by GitHub issues - opened 05 Jan '25

05 Jan '25

alsa-project/alsa-lib issue #433 was opened from trcrsired: ``` clang: warning: argument unused during compilation: '--unwindlib=libunwind' [-Wunused-command-line-argument] CCLD libasound.la ld.lld: error: seq/.libs/libseq.a(seq.o at 50494) <inline asm>:1:1: default version symbol snd_dlopen@@ALSA_1.1.6 must be defined .symver __snd_dlopen, snd_dlopen@@ALSA_1.1.6 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:1:1: default version symbol snd_pcm_hw_params_set_format_first@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_format_first, snd_pcm_hw_params_set_format_first@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:2:1: default version symbol snd_pcm_hw_params_set_channels_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_channels_near, snd_pcm_hw_params_set_channels_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:3:1: default version symbol snd_pcm_hw_params_set_rate_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_rate_near, snd_pcm_hw_params_set_rate_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:4:1: default version symbol snd_pcm_hw_params_set_period_time_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_period_time_near, snd_pcm_hw_params_set_period_time_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:5:1: default version symbol snd_pcm_hw_params_set_period_size_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_period_size_near, snd_pcm_hw_params_set_period_size_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:6:1: default version symbol snd_pcm_hw_params_set_periods_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_periods_near, snd_pcm_hw_params_set_periods_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:7:1: default version symbol snd_pcm_hw_params_set_buffer_time_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_buffer_time_near, snd_pcm_hw_params_set_buffer_time_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:8:1: default version symbol snd_pcm_hw_params_set_buffer_size_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_buffer_size_near, snd_pcm_hw_params_set_buffer_size_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:9:1: default version symbol snd_timer_params_set_filter@@ALSA_0.9.0 must be defined .symver __snd_timer_params_set_filter, snd_timer_params_set_filter@@ALSA_0.9.0 ^ ld.lld: error: .libs/conf.o <inline asm>:1:1: default version symbol snd_dlopen@@ALSA_1.1.6 must be defined .symver __snd_dlopen, snd_dlopen@@ALSA_1.1.6 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_hw.o at 720480) <inline asm>:1:1: default version symbol snd_timer_params_set_filter@@ALSA_0.9.0 must be defined .symver __snd_timer_params_set_filter, snd_timer_params_set_filter@@ALSA_0.9.0 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:1:1: default version symbol snd_pcm_hw_params_get_access@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_access, snd_pcm_hw_params_get_access@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:2:1: default version symbol snd_pcm_hw_params_get_format@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_format, snd_pcm_hw_params_get_format@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:3:1: default version symbol snd_pcm_hw_params_get_subformat@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_subformat, snd_pcm_hw_params_get_subformat@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:4:1: default version symbol snd_pcm_hw_params_get_channels@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_channels, snd_pcm_hw_params_get_channels@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:5:1: default version symbol snd_pcm_hw_params_get_rate@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_rate, snd_pcm_hw_params_get_rate@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:6:1: default version symbol snd_pcm_hw_params_get_period_time@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_period_time, snd_pcm_hw_params_get_period_time@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:7:1: default version symbol snd_pcm_hw_params_get_period_size@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_period_size, snd_pcm_hw_params_get_period_size@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:8:1: default version symbol snd_pcm_hw_params_get_buffer_size@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_buffer_size, snd_pcm_hw_params_get_buffer_size@@ALSA_0.9.0rc4 ^ ld.lld: error: too many errors emitted, stopping now (use --error-limit=0 to see all errors) clang: error: linker command failed with exit code 1 (use -v to see invocation) make[3]: *** [Makefile:506: libasound.la] Error 1 make[3]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make[2]: *** [Makefile:567: all-recursive] Error 1 make[2]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make[1]: *** [Makefile:436: all] Error 2 make[1]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make: *** [Makefile:414: all-recursive] Error 1 alsa-lib build failure ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/433 Repository URL: https://github.com/alsa-project/alsa-lib

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ucm2: USB-Audio: add Presonus Revelator IO 44 (USB194f:0424)
by GitHub pull_request - edited 04 Jan '25

04 Jan '25

alsa-project/alsa-ucm-conf pull request #495 was edited from markulovi: [alsa-info.txt](https://github.com/user-attachments/files/18306150/alsa-info… This adds support for [Presonus Revelator io44](https://eu.presonus.com/products/revelator-io44). This device has 3 virtual stereo channels out and 4 virtual stereo channels in as follows: ![out](https://github.com/user-attachments/assets/f4cacf14-e75e-4d33-ad64-cca4ca7ee18e) ![in](https://github.com/user-attachments/assets/60515091-51a5-4299-97c7-a12d3c091b8a) Microphone inputs have been separated into two mono channels for convenience. As far as I can tell, the [Universal Control](https://www.presonus.com/pages/support-documents-downloads?product… software is required to configure levels and effects of these channels, but the settings can be saved to the device and operation without the software is possible outside Windows. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/495 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/495.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: USB-Audio: add Presonus Revelator IO 44 (USB194f:0424)
by GitHub pull_request - opened 04 Jan '25

04 Jan '25

alsa-project/alsa-ucm-conf pull request #495 was opened from markulovi: [alsa-info.txt](https://github.com/user-attachments/files/18306150/alsa-info… This adds support for [Presonus Revelator io44](https://eu.presonus.com/products/revelator-io44). This device has 3 virtual stereo channels out and 4 virtual stereo channels in as follows: ![out](https://github.com/user-attachments/assets/f4cacf14-e75e-4d33-ad64-cca4ca7ee18e) ![in](https://github.com/user-attachments/assets/60515091-51a5-4299-97c7-a12d3c091b8a) Microphone inputs have been separated to two mono channels for convenience. As far as I can tell, the [Universal Control](https://www.presonus.com/pages/support-documents-downloads?product… software is required to configure levels and effects of these channels, but the settings can be saved to the device and operation without the software is possible outside Windows. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/495 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/495.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/5] soundwire: amd: code improvements and new platform
by Vijendar Mukunda 02 Jan '25

02 Jan '25

This patch series consists of code improvements and new platform support for ACP7.0 & ACP7.1 platforms. Vijendar Mukunda (5): soundwire: amd: change the soundwire wake enable/disable sequence soundwire: amd: add debug log for soundwire wake event soundwire: amd: add support for ACP7.0 & ACP7.1 platforms soundwire: amd: add soundwire host wake interrupt enable/disable sequence soundwire: amd: clear soundwire pme wake status drivers/soundwire/amd_manager.c | 102 ++++++++++++++++++++++++++++++ drivers/soundwire/amd_manager.h | 27 ++++++++ include/linux/soundwire/sdw_amd.h | 2 + 3 files changed, 131 insertions(+) -- 2.34.1

2 7