May 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] unsupported variant of snd_hda_intel
by Alexandre 04 Jul '24

04 Jul '24

Hello, I have an Acer Aspire 4736G laptop. The soundcard that is shipped in this laptop is intel-hda based as shown (actually it's a realtek): cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0x9b300000 irq 46 Basic sound output is correct but few feature are missing: - internal mic seems to be found but do not works. - sound output stop working when trying to send it using HDMI output - heads do not work when plugged neither when forcing pulseaudio to use it. I have an ubuntu 10.10 which is shipped with alsa 1.0.23. I have googled around and tried loading the snd-hda-intel with several parameters () as found in HD-Audio-Models.txt to no avail. cat /proc/asound/pcm 00-00: ALC888 Analog : ALC888 Analog : playback 1 : capture 1 00-01: ALC888 Digital : ALC888 Digital : playback 1 00-03: NVIDIA HDMI : NVIDIA HDMI : playback 1 Where can I begin to help implement the support for this sound card? I have very skills in developement in general but am ready to try if it is not to hard (I supose I can use previous code for similar models).. I just need a shepherd. Thanks

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[alsa-devel] echo-cancellation plugin advice?
by Max Klein 27 Oct '23

27 Oct '23

> Well, asym wouldn't be the answer. Basically you'll need to open two > slave PCMs in one plugin. So, it's no simple filter-type plugin. > It's doable, but not easy as it sounds, I guess. Takashi, can you please further explain why asym is not the solution and what you see as a problem? I am also trying to figure out if it would be possible to create such a plugin for ALSA? I am still searching for some good starting points on how to write an ALSA plugin... Jean-Marc Valin (http://lists.linuxaudio.org/pipermail/linux-audio-dev/2005-June/012454.html) already released his work as part of speex but as far I know there is no plugin that has been made out of it? Thanks for any hint! -- NEU: FreePhone 3-fach-Flat mit kostenlosem Smartphone! Jetzt informieren: http://mobile.1und1.de/?ac=OM.PW.PW003K20328T7073a

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[alsa-devel] ALSA 1.0.29 release
by Jaroslav Kysela 19 Feb '20

19 Feb '20

Hello all, all userspace packages (except alsa-oss) were released today. You may download them from the ALSA website http://www.alsa-project.org or directly from the FTP server ftp://ftp.alsa-project.org/pub . Full list of changes: http://www.alsa-project.org/main/index.php/Changes_v1.0.28_v1.0.29 Have fun, Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project; Red Hat, Inc.

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[alsa-devel] [PATCH RFC 00/26] Kill set_fs() in ALSA codes
by Takashi Iwai 27 Mar '19

27 Mar '19

Hi, this is a slightly lengthy patchset, basically targeted for 4.13, thus it's still in RFC. The main purpose of this patchset is to get rid of get_fs() / set_fs() usages in ALSA codes. The patchset starts from a patch that looks fairly irrelevant with the goal (the conversion of HD-audio bind-mute code), but it's a step for the long trail. The biggest change in the patchset is the conversion of copy & silence PCM ops into the new copy_silence ops. It helped for reducing the code size and for handling the in-kernel buffer copies. The current patches are found in topic/kill-set_fs branch of my sound.git tree. thanks, Takashi === Takashi Iwai (26): ALSA: hda - Simplify bound-beep mute control for ALC268 ALSA: hda - Move bind-mixer switch codes to generic parser ALSA: hda - Remove the generic bind ctl helpers ALSA: hda - Remove the use of set_fs() ALSA: hda - Fix a typo in comment ALSA: hda - Remove superfluous header inclusions ALSA: opl3: Kill unused set_fs() ALSA: emu10k1: Get rid of set_fs() usage ALSA: pcm: Remove set_fs() in PCM core code ALSA: pcm: Introduce copy_silence PCM ops ALSA: Update document about copy_silence PCM ops ALSA: dummy: Convert to copy_silence ops ALSA: es1938: Convert to copy_silence ops ALSA: korg1212: Convert to copy_silence ops ALSA: nm256: Convert to copy_silence ops ALSA: rme32: Convert to copy_silence ops ALSA: rme96: Convert to copy_silence ops ALSA: rme9652: Convert to copy_silence ops ALSA: hdsp: Convert to copy_silence ops ALSA: gus: Convert to copy_silence ops ALSA: sb: Convert to copy_silence ops ALSA: sh: Convert to copy_silence ops ASoC: blackfin: Convert to copy_silence ops [media] solo6x10: Convert to copy_silence ops ALSA: pcm: Drop the old copy and silence ops ALSA: pcm: Kill set_fs() usage in OSS layer and USB gadget .../sound/kernel-api/writing-an-alsa-driver.rst | 110 +++++---- drivers/media/pci/solo6x10/solo6x10-g723.c | 11 +- drivers/usb/gadget/function/u_uac1.c | 8 +- include/sound/pcm.h | 19 +- sound/core/oss/pcm_oss.c | 62 ++--- sound/core/pcm_compat.c | 22 +- sound/core/pcm_lib.c | 174 +++++++++----- sound/core/pcm_native.c | 122 ++++++---- sound/drivers/dummy.c | 13 +- sound/drivers/opl3/opl3_oss.c | 14 -- sound/isa/gus/gus_pcm.c | 43 +--- sound/isa/sb/emu8000_pcm.c | 99 +++----- sound/pci/emu10k1/emufx.c | 127 ++++++----- sound/pci/es1938.c | 11 +- sound/pci/hda/hda_codec.c | 251 +++------------------ sound/pci/hda/hda_generic.c | 46 +++- sound/pci/hda/hda_local.h | 61 ----- sound/pci/hda/patch_realtek.c | 37 ++- sound/pci/korg1212/korg1212.c | 128 +++-------- sound/pci/nm256/nm256.c | 35 ++- sound/pci/rme32.c | 49 ++-- sound/pci/rme96.c | 52 ++--- sound/pci/rme9652/hdsp.c | 44 ++-- sound/pci/rme9652/rme9652.c | 46 ++-- sound/sh/sh_dac_audio.c | 40 +--- sound/soc/blackfin/bf5xx-ac97-pcm.c | 6 +- sound/soc/blackfin/bf5xx-ac97.c | 18 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 46 ++-- sound/soc/soc-pcm.c | 3 +- 29 files changed, 696 insertions(+), 1001 deletions(-) -- 2.12.2

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[alsa-devel] [PATCH 4/6] alsa-lib:pcm: fix the hw_ptr update until the boundary available.
by sutar.mounesh＠gmail.com 19 Jan '18

19 Jan '18

From: mahendran.k <mahendran.kuppusamy(a)in.bosch.com> For long time test case, the slave_hw_ptr will exceed the boundary and wraparound the slave_hw_ptr. This slave boundary wraparound will cause the rate->hw_ptr to wraparound irrespective of the rate->boundary availability and due to that the available size goes wrong. Hence, To get the correct available size, - Its necessary to increment the rate->hw_ptr upto the rate->boundary and then wraparound the rate->hw_ptr. - While handling fraction part of slave period, rounded value will be introduced by input_frames(). To eliminate rounding issue on rate->hw_ptr, subtract last rounded value from rate->hw_ptr and add new rounded value of present slave_hw_ptr fraction part to rate->hw_ptr. Signed-off-by: mahendran.k <mahendran.kuppusamy(a)in.bosch.com> Signed-off-by: Mounesh Sutar <mounesh_sutar(a)mentor.com> diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 1f830dd..5bee77c 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -50,7 +50,7 @@ typedef struct _snd_pcm_rate snd_pcm_rate_t; struct _snd_pcm_rate { snd_pcm_generic_t gen; - snd_pcm_uframes_t appl_ptr, hw_ptr; + snd_pcm_uframes_t appl_ptr, hw_ptr, last_slave_hw_ptr; snd_pcm_uframes_t last_commit_ptr; snd_pcm_uframes_t orig_avail_min; snd_pcm_sw_params_t sw_params; @@ -563,14 +563,28 @@ static inline void snd_pcm_rate_sync_hwptr0(snd_pcm_t *pcm, snd_pcm_uframes_t sl { snd_pcm_rate_t *rate = pcm->private_data; + snd_pcm_sframes_t slave_hw_ptr_diff = slave_hw_ptr - rate->last_slave_hw_ptr; + snd_pcm_sframes_t last_slave_hw_ptr_frac; + if (pcm->stream != SND_PCM_STREAM_PLAYBACK) return; - /* FIXME: boundary overlap of slave hw_ptr isn't evaluated here! - * e.g. if slave rate is small... - */ - rate->hw_ptr = - (slave_hw_ptr / rate->gen.slave->period_size) * pcm->period_size + - rate->ops.input_frames(rate->obj, slave_hw_ptr % rate->gen.slave->period_size); + + if (slave_hw_ptr_diff < 0) + slave_hw_ptr_diff += rate->gen.slave->boundary; /* slave boundary wraparound */ + else if (slave_hw_ptr_diff == 0) + return; + last_slave_hw_ptr_frac = rate->last_slave_hw_ptr % rate->gen.slave->period_size; + /* While handling fraction part fo slave period, rounded value will be introduced by input_frames(). + * To eliminate rounding issue on rate->hw_ptr, subtract last rounded value from rate->hw_ptr and + * add new rounded value of present slave_hw_ptr fraction part to rate->hw_ptr. Hence, + * rate->hw_ptr += [ (no. of updated slave periods * pcm rate period size) - + * fractional part of last_slave_hw_ptr rounded value + + * fractional part of updated slave hw ptr's rounded value ] */ + rate->hw_ptr += ( + (((last_slave_hw_ptr_frac + slave_hw_ptr_diff) / rate->gen.slave->period_size) * pcm->period_size) - + rate->ops.input_frames(rate->obj, last_slave_hw_ptr_frac) + + rate->ops.input_frames(rate->obj, (last_slave_hw_ptr_frac + slave_hw_ptr_diff) % rate->gen.slave->period_size)); + rate->last_slave_hw_ptr = slave_hw_ptr; rate->hw_ptr %= pcm->boundary; } @@ -635,6 +649,7 @@ static int snd_pcm_rate_prepare(snd_pcm_t *pcm) return err; *pcm->hw.ptr = 0; *pcm->appl.ptr = 0; + rate->last_slave_hw_ptr = 0; err = snd_pcm_rate_init(pcm); if (err < 0) return err; @@ -650,6 +665,7 @@ static int snd_pcm_rate_reset(snd_pcm_t *pcm) return err; *pcm->hw.ptr = 0; *pcm->appl.ptr = 0; + rate->last_slave_hw_ptr = 0; err = snd_pcm_rate_init(pcm); if (err < 0) return err; -- 1.7.9.5

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[alsa-devel] [PATCH 0/7] ALSA: Fine-tuning for several function implementations
by SF Markus Elfring 14 Nov '17

14 Nov '17

From: Markus Elfring <elfring(a)users.sourceforge.net> Date: Tue, 24 Jan 2017 11:02:34 +0100 Some update suggestions were taken into account from static source code analysis. Markus Elfring (7): seq: Delete unnecessary checks in snd_seq_ioctl_query_subs() seq: Delete unnecessary checks in snd_seq_ioctl_get_subscription() seq: Delete unnecessary checks in snd_seq_ioctl_unsubscribe_port() seq: Delete unnecessary checks in snd_seq_ioctl_subscribe_port() seq: Adjust 16 function calls together with a variable assignment dmasound_core: Move two assignments for the variable "ret" in state_open() dmasound_core: Adjust six function calls together with a variable assignment sound/core/seq/seq_clientmgr.c | 163 ++++++++++++++++++++----------------- sound/oss/dmasound/dmasound_core.c | 33 +++++--- 2 files changed, 109 insertions(+), 87 deletions(-) -- 2.11.0

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[alsa-devel] [PATCH 1/1] ASoC: bcm2835-i2s: Support 384k sample rate
by Clive Messer 10 Nov '17

10 Nov '17

Add support for sample rates up to 384k to bcm2835-i2s driver, with SNDRV_PCM_RATE_CONTINUOUS and specifying 8k min and 384k max, for both playback and capture. Signed-off-by: Clive Messer <clive.messer(a)digitaldreamtime.co.uk> --- sound/soc/bcm/bcm2835-i2s.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index 6ba2049..4a67db2 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -578,7 +578,9 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .playback = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE @@ -586,7 +588,9 @@ static struct snd_soc_dai_driver bcm2835_i2s_dai = { .capture = { .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, + .rate_max = 384000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE -- 2.5.5

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[alsa-devel] [PATCH v2 0/2] ASoC: add support for CX2072X CODEC
by simon.ho.cnxt＠gmail.com 13 Oct '17

13 Oct '17

From: Simon Ho <simon.ho(a)conexant.com> Hi Takashi, Mark I corrected the driver based on your input as below. Please help review the code. Mark: - Corrected the coding style. - EQ/DRC settings can be configured from userspace via mixer controls. - Wrongly cast void pointers. - Print I2C error code. - Use memcmp() to check if the new EQ value is different. - Removed unused TDM code. - Removed noisy "out of memory" message. - Corrected ACPI and I2C match table. - Removed unnessary CONFIG_ACPI define. Takashi: - Fixed printf int argument. - Fixed unexpected error code at no MCLK fallback. Thanks, Simon Simon Ho (2): ASoC: cx2072x: Add DT bingings documentation for CX2072X CODEC ASoC: add support for Conexant CX2072X CODEC .../devicetree/bindings/sound/cx2072x.txt | 36 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cx2072x.c | 2253 ++++++++++++++++++++ sound/soc/codecs/cx2072x.h | 319 +++ 5 files changed, 2615 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cx2072x.txt create mode 100644 sound/soc/codecs/cx2072x.c create mode 100644 sound/soc/codecs/cx2072x.h -- 2.7.4

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[alsa-devel] ASoC Sound Driver I2S over LPE, Intel Baytrail SoC. MinnowBoard Max
by Michael Pogrebinsky 02 Oct '17

02 Oct '17

Hi all, I have the MinnowBoard Max with Baytrail Atom (ValleyView) SoC. I'm attempting to write an Alsa (ASoC) driver for my sound card to work with this board. PCM should work through I2S for this sound card I have. >From my understanding Baytrail's I2S goes over LPE. I read this thread:https://www.mail-archive.com/elinux-minnowboard@lists.elinux.org/msg… So I understand that I need the 3 components for my ASoC driver. However I am having trouble at understanding how to communicate with the LPE to get the I2S working. Here is what I have done so far: I am running Ubuntu 14.04 LTS, kernel : 4.3.0-040300rc4 First of all I downloaded the firmware fw_sst_0f28.bin-48kHz_i2s_master fromhttps://kernel.googlesource.com/pub/scm/linux/kernel/git/firmware/linux… And put it /lib/firmware/intel/. Later I went to the Bios and enabled LPE in PCI mode. I do see the LPE controller now, but it shows as UNCLAIMED. Later I looked at the sound/soc/inte/baytrail/ and from my understanding in order for me to get the I2S working I need to load this module so I did modprobe snd-soc-sst-baytrail-pcm. Later I wrote a simple codec and machine driver (looking at the example of sound/soc/intel/boards/byt_max98090) however everytime I attempt to register the sound card with snd_soc_register_card, I get the following error message in dmesg : "ASoC: CPU DAI baytrail-pcm-audio not registered". >From this I deduct that although I loaded the snd-soc-sst-baytrail-pcm module, the driver has not actually been loaded. I'm clueless on how to proceed from here. I would highly appreciate your help.

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[alsa-devel] [PATCH] ASoC: Intel: Atom: Remove unneeded linux/miscdevice.h include
by Corentin Labbe 19 Sep '17

19 Sep '17

No file in sound/soc/intel/ use any miscdevice. This patch remove this uncessary include. Signed-off-by: Corentin Labbe <clabbe.montjoie(a)gmail.com> --- sound/soc/intel/atom/sst/sst_acpi.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/intel/atom/sst/sst_acpi.c b/sound/soc/intel/atom/sst/sst_acpi.c index f4d92bb..a971a58 100644 --- a/sound/soc/intel/atom/sst/sst_acpi.c +++ b/sound/soc/intel/atom/sst/sst_acpi.c @@ -23,7 +23,6 @@ #include <linux/interrupt.h> #include <linux/slab.h> #include <linux/io.h> -#include <linux/miscdevice.h> #include <linux/platform_device.h> #include <linux/firmware.h> #include <linux/pm_runtime.h> -- 2.10.2

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