May 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Error patching file pcm_native.c
by Delusion's Master 20 Nov '24

20 Nov '24

Hi, first of all I apologize in case you receive multiple copies of this message, but I forgot to specify a subject in the original mail. I have a problem compiling the latest alsa driver snapshots. Specifically, it's a problem when patching pcm_native.c: [..] copying file alsa-kernel/core/info.c patching file info.c copying file alsa-kernel/core/pcm.c patching file pcm.c Hunk #3 succeeded at 739 (offset 2 lines). Hunk #5 succeeded at 954 (offset 2 lines). Hunk #7 succeeded at 1045 (offset 2 lines). copying file alsa-kernel/core/pcm_native.c patching file pcm_native.c Hunk #4 succeeded at 1544 (offset -25 lines). Hunk #5 succeeded at 2827 (offset 12 lines). Hunk #6 succeeded at 2878 (offset 10 lines). Hunk #7 FAILED at 2906. Hunk #8 FAILED at 2922. Hunk #9 succeeded at 2915 (offset -23 lines). Hunk #10 succeeded at 3036 (offset 10 lines). Hunk #11 succeeded at 3027 (offset -23 lines). Hunk #12 succeeded at 3074 (offset 10 lines). Hunk #13 succeeded at 3098 (offset -23 lines). Hunk #14 succeeded at 3143 (offset 10 lines). Hunk #15 succeeded at 3124 (offset -23 lines). Hunk #16 succeeded at 3214 (offset 10 lines). Hunk #17 succeeded at 3208 (offset -23 lines). Hunk #18 succeeded at 3271 (offset 10 lines). Hunk #19 succeeded at 3304 (offset -23 lines). Hunk #20 succeeded at 3386 (offset 10 lines). Hunk #21 succeeded at 3411 (offset -23 lines). Hunk #22 succeeded at 3456 (offset 10 lines). Hunk #23 succeeded at 3450 (offset -23 lines). Hunk #24 succeeded at 3525 (offset 10 lines). Hunk #25 succeeded at 3644 (offset -23 lines). 2 out of 25 hunks FAILED -- saving rejects to file pcm_native.c.rej make[2]: *** [pcm_native.c] Error 1 It's some days I get this, but I waited to report this since I thought it could be a temporary problem. My OS is Fedora 11 x86_64. Previous snapshots didn't give me this problem, but I can't say which one broke the compilation, since I usually try updated snapshots only at kernel updates. I searched for this error and the only relevant report I could find is this one: http://mailman.alsa-project.org/pipermail/alsa-devel/2009-April/016420.html a report from last April. What could the problem be? Thanks _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

3 10

[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

4 4

[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

3 2

[alsa-devel] icon utrack pro usb audio interface
by Adrien Plisson 30 Aug '14

30 Aug '14

hello, i just bought an "icon utrack pro" usb audio interface <http://www.icon-global.com/ShowPro.aspx?ID=24>. as expected, this interface is not supported under linux: it is not usb audio class compliant, even the midi port is not class compliant... first, i would like to know if anybody ever managed to use this interface under linux: is there an existing driver ? a hack or patch that would allow the kernel to recognize this interface ? a specific module dedicated to this kind of interface ? i can't find any reference to this interface anywhere on the net... anyway, while searching, i found that its firewire counterpart, the "firexon", has an experimental support on ffado. it seems that the firewire interface uses the bridgeco BeBoB chipset. given that some versions of BeBoB supports both firewire and usb, i suppose (but i may be wrong) that the "utrack pro" is also based on BeBoB. is there any other BeBoB based usb interface supported under linux ? finally, i would proudly help the development of a specific driver for this interface: if anyone wants me to test some piece of code, or wants some information on this interface, don't hesitate to contact me. i am also a software developper, so i could help writing a module, but i first need to learn more about the alsa code base (actually, i don't know where to start...). thanks a lot, and have a nice day ! -- rien

3 3

[alsa-devel] wm8974 and sam9g45 atmel
by Michele Da Rold 11 Jul '14

11 Jul '14

Hello everybody, I'm starting a new development with a Atmel SAM9G45 and a WM8974. I'm using kernel 2.6.33.2 I have connected the SSC1 pin TD,TF,TK and the RD pin, so I want to use WM8974 like a slave. I wrote a interface driver based on sam9263_wm8731.c The driver seem to load ok, I recognize the codec but when I start a mp3 playing (I use mpg123) the playback not start. Putting some dedug messagge the driver not call the function "set_bias_level", and call the mute function with mute=0. Testing the MCLK signal I not see any clock on pin because I never call clk_enable!! There is a way to identify the problem? Someone could help me? Thank you in advance Michele Da Rold I put the code of driver (don't care about printk I used it for debug purpose!!) /* * sam9g45_wm8974 -- SoC audio for AT91SAM9G20-based * ATMEL AT91SAM9G20ek board. * * Copyright (C) 2005 SAN People * Copyright (C) 2008 Atmel * * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> * * Based on sam9263_wm8731.c. * Frank Mandarino <fmandarino(a)endrelia.com> * Copyright 2006 Endrelia Technologies Inc. * Based on corgi.c by: * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/kernel.h> #include <linux/clk.h> #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> #include <linux/i2c.h> #include <linux/atmel-ssc.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> #include "../codecs/wm8974.h" #include "atmel-pcm.h" #include "atmel_ssc_dai.h" #define MCLK_RATE 12000000 /* * As shipped the board does not have inputs. However, it is relatively * straightforward to modify the board to hook them up so support is left * in the driver. */ #undef ENABLE_MIC_INPUT #define ENABLE_MIC_INPUT 1 static struct clk *mclk; static int at91sam9g45ek_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; printk(KERN_DEBUG "at91sam9g45_wm8974 " ": at91sam9g45ek_hw_params() called\n"); /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS/*SND_SOC_DAIFMT_CBM_CFM*/); if (ret < 0) return ret; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS/*SND_SOC_DAIFMT_CBM_CFM*/); if (ret < 0) return ret; return 0; } static struct snd_soc_ops at91sam9g45ek_ops = { .hw_params = at91sam9g45ek_hw_params, }; static int at91sam9g45ek_set_bias_level(struct snd_soc_card *card, enum snd_soc_bias_level level) { static int mclk_on; int ret = 0; printk(KERN_DEBUG "at91sam9g45_wm8974 " ": at91sam9g45ek_set_bias_level() called %d\n",level); switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: if (!mclk_on) ret = clk_enable(mclk); if (ret == 0) mclk_on = 1; break; case SND_SOC_BIAS_OFF: case SND_SOC_BIAS_STANDBY: if (mclk_on) clk_disable(mclk); mclk_on = 0; break; } return ret; } #if 0 static const struct snd_soc_dapm_widget at91sam9g45ek_dapm_widgets[] = { SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), }; static const struct snd_soc_dapm_route intercon[] = { /* Speaker output mixer */ {"Speaker Mixer", "PCM Playback Switch", "DAC"}, {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, /* speaker connected to SPKOUTP */ {"SpkN Out", NULL, "Speaker Mixer"}, {"SpkP Out", NULL, "Speaker Mixer"}, {"SPKOUTN", NULL, "SpkN Out"}, {"SPKOUTP", NULL, "SpkP Out"}, /* Input PGA */ {"Input PGA", "Aux Switch", "Aux Input"}, {"Input PGA", "MicN Switch", "MICN"}, {"Input PGA", "MicP Switch", "MICP"}, }; /* * Logic for a wm974 as connected on a at91sam9g45ek board. */ static int at91sam9g45ek_wm8974_init(struct snd_soc_codec *codec) { struct snd_soc_dai *codec_dai = &codec->dai[0]; int ret; printk(KERN_DEBUG "at91sam9g45_wm8974 " ": at91sam9g45_wm8974_init() called\n"); /* ret = snd_soc_dai_set_sysclk(codec_dai, 0, MCLK_RATE, SND_SOC_CLOCK_IN); if (ret < 0) { printk(KERN_ERR "Failed to set WM8974 SYSCLK: %d\n", ret); return ret; } */ /* Add specific widgets */ snd_soc_dapm_new_controls(codec, at91sam9g45ek_dapm_widgets, ARRAY_SIZE(at91sam9g45ek_dapm_widgets)); /* Set up specific audio path interconnects */ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); /* not connected */ snd_soc_dapm_nc_pin(codec, "MONOOUT"); snd_soc_dapm_nc_pin(codec, "AUX"); #ifdef ENABLE_MIC_INPUT snd_soc_dapm_enable_pin(codec, "Int Mic"); #else snd_soc_dapm_nc_pin(codec, "Int Mic"); #endif /* always connected */ snd_soc_dapm_enable_pin(codec, "Ext Spk"); snd_soc_dapm_sync(codec); return 0; } #endif static struct snd_soc_dai_link at91sam9g45ek_dai = { .name = "WM8974", .stream_name = "WM8974 PCM", .cpu_dai = &atmel_ssc_dai[1], .codec_dai = &wm8974_dai, #if 0 .init = at91sam9g45ek_wm8974_init, #endif .ops = &at91sam9g45ek_ops, }; static struct snd_soc_card snd_soc_at91sam9g45ek = { .name = "AT91SAMG45", .platform = &atmel_soc_platform, .dai_link = &at91sam9g45ek_dai, .num_links = 1, .set_bias_level = at91sam9g45ek_set_bias_level, }; static struct snd_soc_device at91sam9g45ek_snd_devdata = { .card = &snd_soc_at91sam9g45ek, .codec_dev = &soc_codec_dev_wm8974, }; static struct platform_device *at91sam9g45ek_snd_device; static int __init at91sam9g45ek_init(void) { struct atmel_ssc_info *ssc_p = at91sam9g45ek_dai.cpu_dai->private_data; struct ssc_device *ssc = NULL; struct clk *main; int ret; at91_set_A_periph(AT91_PIN_PD27, 0); /* * Codec MCLK is supplied by PCK1 - set it up. */ mclk = clk_get(NULL, "pck1"); if (IS_ERR(mclk)) { printk(KERN_ERR "ASoC: Failed to get MCLK\n"); ret = PTR_ERR(mclk); goto err; } main = clk_get(NULL, "main"); if (IS_ERR(main)) { printk(KERN_ERR "ASoC: Failed to get PLLB\n"); ret = PTR_ERR(mclk); goto err_mclk; } ret = clk_set_parent(mclk, main); clk_put(main); if (ret != 0) { printk(KERN_ERR "ASoC: Failed to set MCLK parent\n"); goto err_mclk; } clk_set_rate(mclk, MCLK_RATE); /* * Request SSC device */ ssc = ssc_request(1); if (IS_ERR(ssc)) { printk(KERN_ERR "ASoC: Failed to request SSC 1\n"); ret = PTR_ERR(ssc); ssc = NULL; goto err_ssc; } ssc_p->ssc = ssc; at91sam9g45ek_snd_device = platform_device_alloc("soc-audio", -1); if (!at91sam9g45ek_snd_device) { printk(KERN_ERR "ASoC: Platform device allocation failed\n"); ret = -ENOMEM; } platform_set_drvdata(at91sam9g45ek_snd_device, &at91sam9g45ek_snd_devdata); at91sam9g45ek_snd_devdata.dev = &at91sam9g45ek_snd_device->dev; ret = platform_device_add(at91sam9g45ek_snd_device); if (ret) { printk(KERN_ERR "ASoC: Platform device allocation failed\n"); platform_device_put(at91sam9g45ek_snd_device); } return ret; err_ssc: ssc_free(ssc); ssc_p->ssc = NULL; err_mclk: clk_put(mclk); mclk = NULL; err: return ret; } static void __exit at91sam9g45ek_exit(void) { struct atmel_ssc_info *ssc_p = at91sam9g45ek_dai.cpu_dai->private_data; struct ssc_device *ssc; if (ssc_p != NULL) { ssc = ssc_p->ssc; if (ssc != NULL) ssc_free(ssc); ssc_p->ssc = NULL; } platform_device_unregister(at91sam9g45ek_snd_device); at91sam9g45ek_snd_device = NULL; clk_put(mclk); mclk = NULL; } module_init(at91sam9g45ek_init); module_exit(at91sam9g45ek_exit); /* Module information */ MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); MODULE_DESCRIPTION("ALSA SoC AT91SAM9G45_WM8974"); MODULE_LICENSE("GPL");

3 8

[alsa-devel] how to create alsa nodes?
by loody 12 Dec '13

12 Dec '13

Dear all: In destop x86 system, the udev will automatically create the nodes in snd directory, such as pcmC0D0c, pcmC0D1c, etc. I try to let the usb sound device work on mips platform, but I have no idea what is the proper way to create these nodes. Should I 1. create them one by one by hand from copying the attributes in my PC (if I have 2 or more usb sound devices, I have to first plug them in pc and monitor the nodes) 2. smarter way to do so? -- Regards, miloody

7 19

[alsa-devel] alsa/asoundlib.h file not found
by Appalayagari Sreedhar 05 Dec '13

05 Dec '13

Hi, I have integrated the codec driver and the kernel log shows the sound card is detected. now i want to test the codec. i tried the sample test code for capture and playback. i am getting a compilation errors due to the asoundlib.h. please find the error log for my application. main.c:2:23: error: asoundlib.h: No such file or directory main.c: In function 'main': main.c:7: error: 'snd_pcm_t' undeclared (first use in this function) main.c:7: error: (Each undeclared identifier is reported only once main.c:7: error: for each function it appears in.) main.c:7: error: 'pcm_handle' undeclared (first use in this function) main.c:10: error: 'snd_pcm_stream_t' undeclared (first use in this function) main.c:10: error: expected ';' before 'stream' main.c:15: error: 'snd_pcm_hw_params_t' undeclared (first use in this function) main.c:15: error: 'hwparams' undeclared (first use in this function) main.c:22: warning: 'return' with a value, in function returning void main.c:34: error: 'snd_pcm_uframes_t' undeclared (first use in this function) main.c:34: error: expected ';' before 'periodsize' main.c:46: error: 'SND_PCM_ACCESS_RW_INTERLEAVED' undeclared (first use in this function) main.c:48: warning: 'return' with a value, in function returning void main.c:52: error: 'SND_PCM_FORMAT_S16_LE' undeclared (first use in this function) main.c:54: warning: 'return' with a value, in function returning void main.c:62: warning: 'return' with a value, in function returning void main.c:65:19: warning: missing terminating " character main.c:65: error: missing terminating " character main.c:66: error: expected expression before '==' token main.c:66: error: stray '\' in program main.c:66:31: warning: missing terminating " character main.c:66: error: missing terminating " character main.c:67: error: expected ';' before '}' token main.c:72: warning: 'return' with a value, in function returning void main.c:78: warning: 'return' with a value, in function returning void main.c:86: error: 'periodsize' undeclared (first use in this function) main.c:88: warning: 'return' with a value, in function returning void main.c:99: warning: 'return' with a value, in function returning void main.c:112: warning: incompatible implicit declaration of built-in function 'malloc' main.c:115: error: 'num_frames' undeclared (first use in this function) main.c:5: warning: return type of 'main' is not 'int' please suggest what needs to be done to resolve this issue asap. Thank you, Sreedhar.

5 4

[alsa-devel] Fwd: Re: Realtek ALC889: HDA Intel and kernel 3.1 gives choppy sound (again)
by Colomban Wendling 18 Sep '13

18 Sep '13

Here's my initial mail, but with gzipped outputs. Regards, Colomban -------- Message original -------- Sujet: Re: [alsa-devel] Realtek ALC889: HDA Intel and kernel 3.1 gives choppy sound (again) Date : Thu, 10 Nov 2011 02:11:17 +0100 De : Colomban Wendling <lists.ban(a)herbesfolles.org> Pour : Takashi Iwai <tiwai(a)suse.de> Copie à : alsa-devel(a)alsa-project.org Hi again, and sorry for the delay. Le 08/11/2011 07:32, Takashi Iwai a écrit : > At Mon, 07 Nov 2011 23:45:40 +0100, > Colomban Wendling wrote: >> >> Hi, >> >> Back to 3.0, the sound on my soundcard [1] started to be choppy, I >> reported it [2] and it got fixed (thanks to Takashi Iwai!). >> >> However, the story repeated with 3.1 (and probably 3.0.8 or before [3]): >> I've got similar choppy sound again. >> >> I bisected the few commits that happened on >> sound/pci/hda/patch_realtek.c, and finally found the villain: >> >> "8974bd51 ALSA: hda/realtek - Fix auto-mute with HP+LO configuration" >> >> Reverting it from v3.1 fixes the problem. > > Does it? Very weird. This patch has nothing to do with the HD-audio > controller side but purely a codec change. Yep, I just check once again and it really fixes the problem here. >> master (94956ee) also suffers of the problem, but I couldn't test >> without that commit because reverting it fails, too much changes happened. > > Did you try to pass position_fix=1 (or 2) option? > Also, passing enable_msi=0 (or 1) may change anything? I just tried with the 3 kernels: * vanilla v3.1 * vanilla 3.2-rc1 (master at 1ea6b8f) * v3.1 with 8974bd51 reverted and none of the option changed anything: both vanilla always failed, the third always succeeded. Though, I haven't combined the options, just tried each at once. > In anyway, please give alsa-info.sh output again with the fresh > kernel. Run it with --no-upload option and attach the output. Here it is, for the 3 kernels cited above. Regards, Colomban

4 14

[alsa-devel] Support new HDA codec
by Jorge Fernandez Monteagudo 26 Aug '13

26 Aug '13

Hi all, This is my first email to the devel mailing list. I'm asking the same question in the user mailing list but I suspect that I'll have to add some code in order to get this hda codec from Wolfson to work. Attached is the log I get from alsa-info.sh. This codec has three output paths: 1) Node 0x06, DAC1, and Node 0x11, Port A, the stereo headphone playback, which is the default one and it seems it works because I can see the stream set to 5 when a sound is played. 2) Node 0x04, SPDIF Tx1, and Node 0x0f, SPDIF OUT, the SPDIF transmit functionality provides an IEC-60958-3 compatible S/PIDF output which I would like to enable but without luck yet. I can enable the node 0x04, with hda-verb, and the spdif amplifier receives spdif frames but without audio info. 3) Node 0x07, DAC2, and Node 0x12, Port G, the stereo line playback with a differential stereo output wich I would like to enable but without luck again. I just can mute/unmute the node but without effect in the outputs. I would like to know if I have to develop a patch for this device or with the attached log info I can expect to use it with the current code in kernel. Now, with alsamixer and other config tools I can't see any control related to the outputs for 2) and 3)... Any hint is welcome! Thanks Jorge --- upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.60 !!################################ !!Script ran on: Tue Feb 21 09:44:41 UTC 2012 !!Linux Distribution !!------------------ Ubuntu 11.10 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 11.10" !!DMI Information !!--------------- Manufacturer: nVidia Product Name: MMP9-ION Product Version: To Be Filled By O.E.M. !!Kernel Information !!------------------ Kernel release: 3.0.0-12-generic Operating System: GNU/Linux Architecture: i686 Processor: i686 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.24 Library version: 1.0.24.1 Utilities version: 1.0.24.2 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfae78000 irq 20 !!PCI Soundcards installed in the system !!-------------------------------------- 00:08.0 Audio device: nVidia Corporation MCP79 High Definition Audio (rev b1) !!Advanced information - PCI Vendor/Device/Subsystem ID's !!-------------------------------------------------------- 00:08.0 0403: 10de:0ac0 (rev b1) Subsystem: 10de:cb84 !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-caiaq: index=-2 snd-usb-ua101: index=-2 snd-usb-us122l: index=-2 snd-usb-usx2y: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd-pcsp: index=-2 snd-usb-audio: index=-2 !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 32,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 beep_mode : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) patch : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null) position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Wolfson Microelectronics ID 8800 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x1aec8800 Subsystem Id: 0x00000100 Revision Id: 0x100301 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x01, mute=1 GPIO: io=2, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=1, unsol=0 Node 0x02 [Audio Input] wcaps 0x1005d1: Stereo Device: name="HDA Generic", type="Audio", device=0 Converter: stream=1, channel=0 SDI-Select: 0 PCM: rates [0x1ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=02, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x0a Processing caps: benign=0, ncoeff=0 Node 0x03 [Audio Input] wcaps 0x1005d1: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=03, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Connection: 1 0x09 Processing caps: benign=0, ncoeff=0 Node 0x04 [Audio Output] wcaps 0x2d1: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=04, enabled=0 Processing caps: benign=0, ncoeff=0 Node 0x05 [Audio Input] wcaps 0x1003d1: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Copyright Digital category: 0x0 PCM: rates [0x1f0]: 32000 44100 48000 88200 96000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=05, enabled=0 Connection: 1 0x10 Processing caps: benign=0, ncoeff=0 Node 0x06 [Audio Output] wcaps 0x495: Stereo Amp-Out Control: name="Master Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Master Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="HDA Generic", type="Audio", device=0 Amp-Out caps: N/A Amp-Out vals: [0x57 0x57] Converter: stream=5, channel=0 PCM: rates [0x7ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=06, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x07 [Audio Output] wcaps 0x495: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 PCM: rates [0x7ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=07, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x08 [Audio Output] wcaps 0x495: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 PCM: rates [0x7ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=08, enabled=0 Power states: D0 D1 D2 D3 Power: setting=D0, actual=D0 Node 0x09 [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x0b* 0x0e Node 0x0a [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x18, nsteps=0x30, stepsize=0x01, mute=1 Amp-Out vals: [0x18 0x18] Connection: 1 0x0c Node 0x0b [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x18, nsteps=0x30, stepsize=0x01, mute=1 Amp-Out vals: [0x98 0x98] Connection: 1 0x0d Node 0x0c [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00000064: IN Detect Balanced Pin Default 0x018130f0: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=0c, enabled=0 Node 0x0d [Pin Complex] wcaps 0x400083: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x00001767: IN Detect Balanced Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x02a190f0: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=0d, enabled=0 Node 0x0e [Pin Complex] wcaps 0x40020b: Stereo Digital Amp-In Amp-In caps: ofs=0x18, nsteps=0x58, stepsize=0x01, mute=1 Amp-In vals: [0x18 0x18] Pincap 0x00000020: IN Pin Default 0x10a600f0: [Jack] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Node 0x0f [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x014190f0: [Jack] SPDIF Out at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=0f, enabled=0 Connection: 1 0x04 Node 0x10 [Pin Complex] wcaps 0x400281: Stereo Digital Pincap 0x00000020: IN Pin Default 0x01c520f0: [Jack] SPDIF In at Ext Rear Conn = Optical, Color = Grey DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=10, enabled=0 Node 0x11 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000001c: OUT HP Detect Pin Default 0x01014020: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=11, enabled=0 Connection: 1 0x06 Node 0x12 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01016121: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x2, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=12, enabled=0 Connection: 1 0x07 Node 0x13 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x01011022: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x2 Pin-ctls: 0x40: OUT Unsolicited: tag=13, enabled=0 Connection: 1 0x08 Node 0x14 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x04, nsteps=0x04, stepsize=0x17, mute=0 Amp-Out vals: [0x04] Node 0x15 [Audio Input] wcaps 0x1003d1: Stereo Digital Converter: stream=0, channel=2 Digital: Digital category: 0x0 PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=15, enabled=0 Connection: 1 0x16 Processing caps: benign=0, ncoeff=0 Node 0x16 [Pin Complex] wcaps 0x40030b: Stereo Digital Amp-In Amp-In caps: ofs=0x18, nsteps=0x58, stepsize=0x01, mute=1 Amp-In vals: [0x18 0x18] Pincap 0x00000030: IN OUT Pin Default 0x10a601f0: [Jack] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Connection: 1 0x17 Node 0x17 [Audio Output] wcaps 0x291: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x7f0]: 32000 44100 48000 88200 96000 176400 192000 bits [0x1e]: 16 20 24 32 formats [0x3]: PCM FLOAT Unsolicited: tag=17, enabled=0 --endcollapse-- !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: HDA Generic [HDA Generic] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: HDA Generic [HDA Generic] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [NVidia] Card hw:0 'NVidia'/'HDA NVidia at 0xfae78000 irq 20' Mixer name : 'Wolfson Microelectronics ID 8800' Components : 'HDA:1aec8800,00000100,00100301' Controls : 3 Simple ctrls : 2 Simple mixer control 'Master',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 87 [69%] [-20.00dB] [on] Front Right: Playback 87 [69%] [-20.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume penum Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] !!Alsactl output !!------------- --startcollapse-- state.NVidia { control.1 { iface MIXER name 'Master Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } control.2 { iface MIXER name 'Master Playback Volume' value.0 87 value.1 87 comment { access 'read write' type INTEGER count 2 range '0 - 127' dbmin -6350 dbmax 0 dbvalue.0 -2000 dbvalue.1 -2000 } } control.3 { iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } } --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 5 Feb 21 09:38 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 4 Feb 21 09:38 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 3 Feb 21 09:38 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 2 Feb 21 09:39 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 1 Feb 21 09:37 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Feb 21 09:37 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Feb 21 09:38 . drwxr-xr-x 3 root root 180 Feb 21 09:38 .. lrwxrwxrwx 1 root root 12 Feb 21 09:38 pci-0000:00:08.0 -> ../controlC0 Este mensaje se dirige exclusivamente a su destinatario y puede contener información privilegiada o CONFIDENCIAL. Si no es vd. el destinatario indicado, queda notificado de que la utilización, divulgación y/o copia sin autorización está prohibida en virtud de la legislación vigente. Si ha recibido este mensaje por error, le rogamos que nos lo comunique inmediatamente por esta misma vía y proceda a su destrucción. This message is intended exclusively for its addressee and may contain information that is CONFIDENTIAL and protected by professional privilege. If you are not the intended recipient you are hereby notified that any dissemination, copy or disclosure of this communication is strictly prohibited by law. If this message has been received in error, please immediately notify us via e-mail and delete it.

4 6

[alsa-devel] hda-intel missing mixer channels
by Orm Finnendahl 26 Aug '13

26 Aug '13

Hi, I sent this to alsa-user a week ago but didn't get any reply. Does anybody know, what is responsible for the missing faders? Did the driver change or is it some module loading mechanism in the distro which prevents the faders from showing up in alsamixer? This is so strange. Please help! -- Orm -------------------------------------------------------------------- Hi everybody, after the last update of Ubuntu (intrepid to jaunty) on my Lenovo Ideapad my microphone input doesn't work anymore (neither internal nor external). I found out that alsamixer lacks quite a lot of faders in the newer kernel/module compared to the older one. Some of the faders now unavailable were crucial for Mic input to work on the older system so I'd very much like to get them back ;-) Some more infos: Soundcard: Intel 82801G (ICH7 Family) Chip: Realtek ALC269 Module: snd_hda_intel (in both kernels) Older kernel: Ubuntu stock 2.6.27-14-generic Newer kernel: Ubuntu stock 2.6.28-13-rt Is there any way to make the missing controls available or did the driver change and I have to compile the older driver for the new kernel? Yours, Orm

2 3

[alsa-devel] Problem with MX27 ASoC audio
by Randolph Chung 11 Jul '13

11 Jul '13

I have a target that uses iMX27(L) + tlv320aic3x. The kernel tree is HEAD of linux-arm from about a week ago. Simple playback and recording (using aplay and arecord) work fine. However, I've run into a strange problem when both playback and capture streams are opened at the same time. Consider the attached test program. The capture device is setup with a period time of 10ms, and the program reads samples in a loop. The expected result is that I get a sample every 10ms. This works fine if the playback device is *not* opened (more specifically, if the hw_params are not set). However if the playback device is opened (#if 1 code) then I get samples every 20ms, which seems very wrong. I have tried using both DMA and FIQ ssi-pcm interfaces and both give the same strange result. With the FIQ driver, if I instrument the timer callback, the timer triggers every 10ms but there is only enough data every 20ms. The codec is being driven with a 13MHz MCLK (output from MX27) and the BCLK and WCLK come from the codec. They seem to have the correct rates. Any ideas? Can anybody else with MX27 hardware verify if my test program also fails on their target? thanks, randolph /* Test program */ #include <alsa/asoundlib.h> #include <stdio.h> #include <stdlib.h> #include <assert.h> #include <sys/time.h> int main(int argc, char **argv) { int err = -1; char buf[4096]; unsigned long period_size; snd_pcm_t *in_handle = NULL, *out_handle; snd_pcm_hw_params_t *hwparams; if (snd_pcm_open(&in_handle, "hw:0,0", SND_PCM_STREAM_CAPTURE, 0) < 0) { printf("Cannot open capture device\n"); return -1; } snd_pcm_hw_params_malloc(&hwparams); snd_pcm_hw_params_any(in_handle, hwparams); snd_pcm_hw_params_set_access(in_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(in_handle, hwparams, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_channels(in_handle, hwparams, 1); snd_pcm_hw_params_set_rate(in_handle, hwparams, 8000, 0); snd_pcm_hw_params_set_period_time(in_handle, hwparams, 10000, 0); snd_pcm_hw_params_get_period_size(hwparams, &period_size, 0); snd_pcm_hw_params(in_handle, hwparams); snd_pcm_hw_params_free(hwparams); if (snd_pcm_open(&out_handle, "hw:0,0", SND_PCM_STREAM_PLAYBACK, 0) < 0) { printf("Cannot open playback device\n"); return -1; } #if 1 snd_pcm_hw_params_malloc(&hwparams); snd_pcm_hw_params_any(out_handle, hwparams); snd_pcm_hw_params_set_access(out_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format(out_handle, hwparams, SND_PCM_FORMAT_S16_LE); snd_pcm_hw_params_set_channels(out_handle, hwparams, 1); snd_pcm_hw_params_set_rate(out_handle, hwparams, 8000, 0); snd_pcm_hw_params_set_period_time(out_handle, hwparams, 10000, 0); snd_pcm_hw_params(out_handle, hwparams); snd_pcm_hw_params_free(hwparams); #endif printf("period size = %u\n", period_size); printf("Config done, start loop\n"); while (1) { struct timeval tv; int ret = snd_pcm_readi(in_handle, buf, period_size); if (ret < 0) snd_pcm_prepare(in_handle); gettimeofday(&tv, NULL); printf("[%u.%06u] snd_pcm_readi() return %d\n", tv.tv_sec, tv.tv_usec, ret); } return 0; }

5 11

[alsa-devel] USB Quirks for Boss JS-8
by Keith A. Milner 15 Apr '13

15 Apr '13

Hi, I'm looking at developing a quirk for the Boss JS-8. The lsusb is as follows: Bus 001 Device 003: ID 0582:0109 Roland Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x0582 Roland Corp. idProduct 0x0109 bcdDevice 0.01 iManufacturer 1 BOSS iProduct 2 JS-8 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 167 bNumInterfaces 3 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 06 24 f1 01 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x05 EP 5 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 44 ac 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x0070 1x 112 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 02 02 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 3 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0040 1x 64 bytes bInterval 1 Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 8 Mass Storage bDeviceSubClass 6 SCSI bDeviceProtocol 80 Bulk (Zip) bMaxPacketSize0 64 bNumConfigurations 0 Device Status: 0x0001 Self Powered I wasn't entirely sure about the quirk syntax as the two interfaces are at EP 4 and EP5. Does this mean I need to ignore the (non-existant) lower interfaces? Here's what I came up with: { /* Boss JS-8 Jam Station */ USB_DEVICE(0x0582, 0x0109), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { /* .vendor_name = "BOSS", */ /* .product_name = "JS-8", */ .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 1, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 2, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 3, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 4, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 5, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = -1 } } } }, regards, -- Keith A. Milner

5 9

[alsa-devel] hdsp rpm support
by Roman Haefeli 25 Jan '13

25 Jan '13

Hi everyone I would like to know current state of the RME HDSP RPM support. There was a thread [1] in 2008 on this list about the issue and things looked promising back then. However, I'm still unable to upload the firmware with alsa-driver from git on my Ubuntu 11.04 (32bit) machine with a 2.6.38 kernel. I posted a detailed description of the issue on the alsa-user list here [2]. Is it supposed to work with Cardbus? If not, is there something I can do to help support it? Any hints appreciated. Roman [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2008-May/008128.html [2] http://article.gmane.org/gmane.linux.alsa.user/36022

5 12

[alsa-devel] sound: usb-audio: full capture/playback/spdif support for Digidesign Mbox 2
by Damien Zammit 23 Dec '12

23 Dec '12

Hi folks, I have put together full support for Digidesign Mbox 2, but it needs someone to help me merge it into alsa. I haven't seen the latest snd-usb-audio code since last year, it seems to have branched into multiple files. I have worked out how the Digidesign Mbox 2 needs to be configured, it supports SPDIF i/o simultaneously with analogue but can be toggled on and off by using different altsettings. Also the sample clock rate must be configured via a special method. Therefore I have provided for this in a device_setup parameter with 4 settings: 48Khz Spdif+analog = 0 44.1Khz Spdif+analog = 1 48Khz Analog only = 2 44.1Khz Analog only = 3 ###### audio /* Digidesign Mbox 2 workaround: * supports S24_3BE BIG ENDIAN in an unusual way */ if (chip->usb_id == USB_ID(0x0dba, 0x3000)) pcm_format = SNDRV_PCM_FORMAT_S24_3BE; static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno); static int parse_audio_endpoints(struct snd_usb_audio *chip, int iface_no) { ..... ..... /* Digidesign Mbox 2: skip altsets incompatible with device_setup */ if (chip->usb_id == USB_ID(0x0dba, 0x3000) && mbox2_skip_setting_quirk(chip, iface_no, altno)) continue; ..... } #define MBOX2_SET_48K_SPDIF 0 #define MBOX2_SET_44K_SPDIF 1 #define MBOX2_SET_48K_ANALOG 2 #define MBOX2_SET_44K_ANALOG 3 static int mbox2_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno) { u8 srate[3]; /* Reset ifaces 2-5 to 0 altsetting. */ usb_set_interface(chip->dev, iface, 0); if (device_setup[chip->index] == MBOX2_SET_48K_SPDIF || device_setup[chip->index] == MBOX2_SET_48K_ANALOG) { //setup 48k srate[0]=0x80; srate[1]=0xbb; srate[2]=0x00; /* Send the magic! */ snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x01, 0x22, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003, 1000); snd_printk(KERN_INFO "Digidesign Mbox 2: 24 Bit 48000Hz"); if (device_setup[chip->index] == MBOX2_SET_48K_SPDIF) { //setup spdif snd_printk(KERN_INFO ", SPDIF\n"); if (altno != 2) return 1; } else { //setup analog snd_printk(KERN_INFO ", ANALOGUE\n"); if (altno != 3) return 1; } } else { //setup 44k srate[0]=0x44; srate[1]=0xac; srate[2]=0x00; /* Send the magic! */ snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x01, 0x22, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0085, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003, 1000); snd_usb_ctl_msg(chip->dev, usb_sndctrlpipe(chip->dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003, 1000); snd_printk(KERN_INFO "Digidesign Mbox 2: 24 Bit 44100Hz"); if (device_setup[chip->index] == MBOX2_SET_44K_SPDIF || device_setup[chip->index] == MBOX2_SET_44K_ANALOG) { //setup spdif snd_printk(KERN_INFO ", SPDIF\n"); if (altno != 2) return 1; } else { //setup analog snd_printk(KERN_INFO ", ANALOGUE\n"); if (altno != 3) return 1; } } return 0; } #define MBOX2_FIRMWARE_SIZE 646 #define MBOX2_BOOT_LOADING 0x01 /* Hard coded into the device */ #define MBOX2_BOOT_READY 0x02 /* Hard coded into the device */ static int snd_usb_mbox2_boot_quirk(struct usb_device *dev) { struct usb_host_config *config = dev->actconfig; int err; u8 enablemagic[3]; u8 bootresponse; int fwsize; if ((fwsize = le16_to_cpu(get_cfg_desc(config)->wTotalLength)) == MBOX2_FIRMWARE_SIZE) { snd_printdd("Sending Digidesign Mbox 2 boot sequence...\n"); /* From USB Snoop, * * SetupPacket = RT RQ VHVL INDX SIZE * RT = 0xRT Request Type * RQ = 0xRQ Request * VHVL = 0xVLVH Value in reverse byte order * INDX = 0xDXIN Index in reverse byte order * SIZE = 0xZESI Size in reverse byte order * * Magic boot code setup packet: c0 85 01 00 00 00 12 00 * RQ RT VLVH DXIN ZESI * becomes 0x85, 0xc0, 0x0001, 0x0000, &RETURNDATA, 0x0012, TIMEOUT * for snd_usb_ctl_msg() */ mbox2_reboot: snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), 0x85, 0xc0, 0x0001, 0x0000, &bootresponse, 0x0012, 1000); if (bootresponse == MBOX2_BOOT_LOADING) { snd_printdd("device not ready, resending boot sequence...\n"); goto mbox2_reboot; } if (bootresponse == MBOX2_BOOT_READY) { snd_printdd("device initialised!\n"); err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, &dev->descriptor, sizeof(dev->descriptor)); config = dev->actconfig; if (err < 0) snd_printdd("error usb_get_descriptor: %d\n", err); err = usb_reset_configuration(dev); if (err < 0) snd_printdd("error usb_reset_configuration: %d\n", err); snd_printdd("mbox2_boot: new boot length = %d\n", le16_to_cpu(get_cfg_desc(config)->wTotalLength)); /* Successful boot, (should reset all altsettings to 0 like windows does, but works without it) * but now we will try sending capture/playback enable magic * * 80 bb 00 = 24bit mode - S24_3BE * 44 ac 00 = 16bit mode? * */ /* We want 48khz mode for now */ enablemagic[0]=0x80; enablemagic[1]=0xbb; enablemagic[2]=0x00; /* Send the magic! */ snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x01, 0x22, 0x0100, 0x0085, &enablemagic, 0x0003, 1000); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0085, &enablemagic, 0x0003, 1000); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0086, &enablemagic, 0x0003, 1000); snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), 0x81, 0xa2, 0x0100, 0x0003, &enablemagic, 0x0003, 1000); return 0; /* Succesful boot */ } snd_printdd("unknown bootresponse, ignoring device: %d\n",bootresponse); return -ENODEV; } snd_printk("Invalid firmware size: %d\n",fwsize); return -ENODEV; } ####### quirk: { USB_DEVICE(0x0dba, 0x3000), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Digidesign", .product_name = "Mbox 2", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 1, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 2, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 3, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 4, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 5, .type = QUIRK_AUDIO_STANDARD_INTERFACE }, { .ifnum = 6, .type = QUIRK_MIDI_MBOX2, .data = & (const struct snd_usb_midi_endpoint_info) { .out_ep = 0x02, .out_cables = 0x0001, .in_ep = 0x81, .in_interval = 0x01, .in_cables = 0x0001 } }, { .ifnum = -1 } } } }, ##### midi case QUIRK_MIDI_MBOX2: /* Digidesign Mbox 2 uses MIDIMAN MIDI protocol */ umidi->usb_protocol_ops = &snd_usbmidi_midiman_ops; /* * We have to make sure that the USB core looks * again at interface 6 by calling usb_set_interface() on it. */ usb_set_interface(umidi->chip->dev, 6, 0); memcpy(&endpoints[0], quirk->data, sizeof(struct snd_usb_midi_endpoint_info)); err = 0; break; Damien

4 26

Re: [alsa-devel] Help requested: new HSS1394 MIDI back-end
by Clemens Ladisch 12 Nov '12

12 Nov '12

Sean M. Pappalardo - D.J. Pegasus wrote: > On 05/30/2012 06:51 AM, Clemens Ladisch wrote: >> Sean M. Pappalardo - D.J. Pegasus wrote: >>> four-byte platter messages that start with >>> 0xF9 and the remaining three can range from 0x00 to 0xFF. >> >> This is not MIDI. Into what MIDI messages should >> the driver convert this? > > It shouldn't convert it at all. Just deliver it as-is in a single > four-byte message, if possible. The problem is the communication between the driver and userspace; _that_ must be valid MIDI. > 0xF9 is a valid, if undefined, MIDI status. Undefined means invalid. Furthermore, data bytes must not have the eighth bit set. > I will also need the ability to use the HSS1394 SendUserControl > function to set the device's timers, and that is definitely not MIDI > at all. > I wonder if it would be a good idea to define a special SYSEX message > that the SCS1x driver would interpret and convert to a call to that > function? As it happens, the actual SysEx commands use the wrong manufacturer ID ("00 01 02" is Crystal Semiconductor); I could just use the real ID (Stanton is "00 01 60") to escape non-MIDI HSS1394 messages. Let's add "HSS" to identify this, and to allow the full byte range, each HSS1394 byte is split into two nibbles. So: 00 F9 xx yy zz -> F0 00 01 60 48 53 53 00 00 0F 09 0x 0x 0y 0y 0z 0z F7 13 xx yy zz <- F0 00 01 60 48 53 53 01 03 0x 0x 0y 0y 0z 0z F7 14 xx yy zz <- F0 00 01 60 48 53 53 01 04 0x 0x 0y 0y 0z 0z F7 (0x13 = 0x10 + 3 = kUserTagBase + uUserTag) Regards, Clemens

3 21

[alsa-devel] Status of Zoom R16
by Evan Ta 08 Nov '12

08 Nov '12

Hi, as subj says: can you tell me how the Zoom R16 is doing? The hardware looks nice, but the latest info I can find is a seemingly unresolved failure to work with Alsa as of feb 2010. Thanks, Evan

4 6

[alsa-devel] HDSPM - Problems with SyncIn
by Andre Schramm 19 Oct '12

19 Oct '12

Hello, I did some tests with the SyncIn-Feature of the HDSPe cards. [root@localhost ~]# lspci |grep Mult 01:00.0 Multimedia audio controller: Xilinx Corporation RME Hammerfall DSP MADI (rev d2) 02:00.0 Multimedia audio controller: Xilinx Corporation RME Hammerfall DSP MADI (rev d2) The SyncOut of card 2:00 was plugged to the SyncIn of 1:00 and 1:00 was put to AutoSync-mode automatically upon boot. Problem is that the amixer output seems to be inconsistent and differs from the proc output as well, so I'm not sure which one to trust. [root@localhost ~]# amixer -c0 contents numid=5,iface=MIXER,name='AutoSync Reference' ; type=ENUMERATED,access=r-------,values=1,items=5 ; Item #0 'Word Clock' ; Item #1 'MADI' ; Item #2 'TCO' ; Item #3 'Sync In' ; Item #4 'None' : values=3 numid=2,iface=MIXER,name='Internal Clock' ; type=ENUMERATED,access=rw------,values=1,items=9 ; Item #0 '32 kHz' ; Item #1 '44.1 kHz' ; Item #2 '48 kHz' ; Item #3 '64 kHz' ; Item #4 '88.2 kHz' ; Item #5 '96 kHz' ; Item #6 '128 kHz' ; Item #7 '176.4 kHz' ; Item #8 '192 kHz' : values=2 numid=4,iface=MIXER,name='Preferred Sync Reference' ; type=ENUMERATED,access=rw------,values=1,items=3 ; Item #0 'Word Clock' ; Item #1 'MADI' ; Item #2 'Sync In' : values=2 numid=10,iface=MIXER,name='SYNC IN SyncCheck' ; type=ENUMERATED,access=r-------,values=1,items=4 ; Item #0 'No Lock' ; Item #1 'Lock' ; Item #2 'Sync' ; Item #3 'N/A' : values=0 numid=3,iface=MIXER,name='System Clock Mode' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Master' ; Item #1 'AutoSync' : values=1 numid=6,iface=MIXER,name='System Sample Rate' ; type=INTEGER,access=r-------,values=1,min=27000,max=207000,step=1 : values=47999 'Preferred Sync Reference' is SyncIn, 'SYNC IN SyncCheck' says 'No Lock', meaning no signal at all, but the 'System Sample Rate' coming from the master card is detected with 1Hz offset. When putting the master to 32000Hz, 31999Hz are shown and so on. [root@localhost ~]# cat /proc/asound/card0/hdspm RME MADI (Card #1) Rev.d2 Status2first3bits: 1 HW Serial: 0x0000019196d4 IRQ: 16 Registers bus: 0xfe9f0000 VM: 0xf8920000 --- System --- IRQ Pending: Audio=0, MIDI0=0, MIDI1=0, IRQcount=0 HW pointer: id = 0, rawptr = 0 (0->16320) estimated= 0 (bytes) MIDI FIFO: Out1=0x0, Out2=0x0, In1=0x0, In2=0x0 MIDIoverMADI FIFO: In=0x0, Out=0x0 Register: ctrl1=0x10344ca, ctrl2=0x0, status1=0xc30004, status2=0xc0020361 No TCO module detected. --- Settings --- Size (Latency): 2048 samples (2 periods of 8192 bytes) Line out: on ClearTrackMarker = off, Transmit in 64 Channel Mode, Auto Input off AutoSync Reference: AutoSync Preferred Sync Reference: TCO System Clock Frequency: 48000 --- Status: Inputs MADI=NoLock, WordClock=NoLock AutoSync: Reference= Sync In, Freq=0 (MADI = 3, Word = 3) Input: Coax, Mode=56 channels Now the proc-output reports 'No TCO module detected', which is correct, as no expansion modules where plugged to the cards. But it also says 'Preferred Sync Reference: TCO' and 'Freq=0', where amixer said SyncIn and 47999? Output of the master card is below. Regards, Andre [root@localhost ~]# amixer -c1 contents numid=5,iface=MIXER,name='AutoSync Reference' ; type=ENUMERATED,access=r-------,values=1,items=5 ; Item #0 'Word Clock' ; Item #1 'MADI' ; Item #2 'TCO' ; Item #3 'Sync In' ; Item #4 'None' : values=4 numid=15,iface=MIXER,name='Input Select' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'optical' ; Item #1 'coaxial' : values=0 numid=2,iface=MIXER,name='Internal Clock' ; type=ENUMERATED,access=rw------,values=1,items=9 ; Item #0 '32 kHz' ; Item #1 '44.1 kHz' ; Item #2 '48 kHz' ; Item #3 '64 kHz' ; Item #4 '88.2 kHz' ; Item #5 '96 kHz' ; Item #6 '128 kHz' ; Item #7 '176.4 kHz' ; Item #8 '192 kHz' : values=2 numid=4,iface=MIXER,name='Preferred Sync Reference' ; type=ENUMERATED,access=rw------,values=1,items=3 ; Item #0 'Word Clock' ; Item #1 'MADI' ; Item #2 'Sync In' : values=0 numid=10,iface=MIXER,name='SYNC IN SyncCheck' ; type=ENUMERATED,access=r-------,values=1,items=4 ; Item #0 'No Lock' ; Item #1 'Lock' ; Item #2 'Sync' ; Item #3 'N/A' : values=0 numid=3,iface=MIXER,name='System Clock Mode' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Master' ; Item #1 'AutoSync' : values=0 numid=6,iface=MIXER,name='System Sample Rate' ; type=INTEGER,access=r-------,values=1,min=27000,max=207000,step=1 : values=48000 [root@localhost ~]# cat /proc/asound/card1/hdspm RME MADI (Card #2) Rev.d2 Status2first3bits: 1 HW Serial: 0x000001b1c606 IRQ: 17 Registers bus: 0xfeaf0000 VM: 0xf9280000 --- System --- IRQ Pending: Audio=0, MIDI0=0, MIDI1=0, IRQcount=0 HW pointer: id = 0, rawptr = 0 (0->16320) estimated= 0 (bytes) MIDI FIFO: Out1=0x0, Out2=0x0, In1=0x0, In2=0x0 MIDIoverMADI FIFO: In=0x0, Out=0x0 Register: ctrl1=0x10004da, ctrl2=0x0, status1=0x2800000, status2=0xc0020701 No TCO module detected. --- Settings --- Size (Latency): 2048 samples (2 periods of 8192 bytes) Line out: on ClearTrackMarker = off, Transmit in 64 Channel Mode, Auto Input off AutoSync Reference: Master Preferred Sync Reference: Word Clock System Clock Frequency: 48000 --- Status: Inputs MADI=NoLock, WordClock=NoLock AutoSync: Reference= Input not valid, Freq=0 (MADI = 10, Word = 0) Input: Optical, Mode=56 channels

2 2

[alsa-devel] snd-usb-audio probe problem (Teac UD-H01)
by bEN 13 Oct '12

13 Oct '12

Hi all, the new USB DAC from Teac seems detected as 'Human Interface Device' instead of 'Audio'. Despite the fact it is based on a standard USB Audio class v2 controller (Tenor 8802). Below is dmesg and lsusb outputs. Does it need to apply a patch in /sound/usb/quirks-table.h ? If yes, someone could help me to write this patch, please? Your help would be very much appreciated, Thanks in advance. bEN (from Ubuntu 11.04 with Linux 2.6.38-13-generic + Alsa driver 1.0.23) $ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.23. $ dmesg (...) [34035.000078] usb 2-4: new high speed USB device using ehci_hcd and address 3 [34035.151753] usb 2-4: config 1 has an invalid interface number: 3 but max is 2 [34035.151763] usb 2-4: config 1 has an invalid interface number: 3 but max is 2 [34035.151770] usb 2-4: config 1 has an invalid interface number: 3 but max is 2 [34035.151777] usb 2-4: config 1 has no interface number 1 [34035.155108] input: TEAC TEAC USB AUDIO DEVICE as /devices/pci0000:00/0000:00:1d.7/usb2/2-4/2-4:1.0/input/input13 [34035.155328] generic-usb 0003:0644:8038.0002: input,hidraw1: USB HID v1.00 Device [TEAC TEAC USB AUDIO DEVICE] on usb-0000:00:1d.7-4/input0 [34035.202368] Audio class v2 interfaces need an interface association [34035.202387] snd-usb-audio: probe of 2-4:1.2 failed with error -5 [34035.202426] usbcore: registered new interface driver snd-usb-audio $ lsusb -v --> see attachment.

5 6

Re: [alsa-devel] [PATCH 2.6.39-rc3] hda_intel: ALSA HD Audio patch for Intel Panther Point DeviceIDs
by Takashi Iwai 09 Oct '12

09 Oct '12

At Wed, 20 Apr 2011 10:59:57 -0700, Seth Heasley wrote: > > This patch adds the HD Audio Controller DeviceIDs for the Intel Panther Point PCH. > > Signed-off-by: Seth Heasley <seth.heasley(a)intel.com> Applied now. Thanks. Takashi > --- > --- linux-2.6.39-rc3/sound/pci/hda/hda_intel.c.orig 2011-04-11 17:21:51.000000000 -0700 > +++ linux-2.6.39-rc3/sound/pci/hda/hda_intel.c 2011-04-14 11:44:40.000000000 -0700 > @@ -126,6 +126,7 @@ > "{Intel, ICH10}," > "{Intel, PCH}," > "{Intel, CPT}," > + "{Intel, PPT}," > "{Intel, PBG}," > "{Intel, SCH}," > "{ATI, SB450}," > @@ -2759,6 +2760,8 @@ > { PCI_DEVICE(0x8086, 0x1c20), .driver_data = AZX_DRIVER_PCH }, > /* PBG */ > { PCI_DEVICE(0x8086, 0x1d20), .driver_data = AZX_DRIVER_PCH }, > + /* Panther Point */ > + { PCI_DEVICE(0x8086, 0x1e20), .driver_data = AZX_DRIVER_PCH }, > /* SCH */ > { PCI_DEVICE(0x8086, 0x811b), .driver_data = AZX_DRIVER_SCH }, > /* Generic Intel */ >

3 2

[alsa-devel] Internal speaker output from Sony Vaio Z (2010 model) lost somewhere between 3.1.0 and 3.2.6
by Adam Williamson 20 Sep '12

20 Sep '12

Welp, pretty much as the topic says. I just noticed today that sound is no longer playing from the internal speakers of my laptop, a 2010 model Sony Vaio Z. Plugging in headphones causes sound to be played fine through those. There are three possible choices for 'Connector' - 'Analog Speakers', 'Analog Output', and 'Analog Headphones' - but none of these seems to make sound come out of the internal speakers. The default is 'Analog Speakers', and when it's set to this, headphone output does work when headphones are plugged in. I've verified that it's broken on Fedora 16 kernels 3.2.6-4, 3.2.8-3 and 3.2.10-1. It's also broken in a very recent Fedora 17 kernel, 3.3.0rc6 or so. It works on a Fedora 16 live image, with kernel 3.1.0-1. Unfortunately the kernels before 3.2.6 have been trashed from Fedora's buildsystem archives, I think, so it's hard to narrow things down any further :/ the alsa-info output is http://www.alsa-project.org/db/?f=368e2359757b537d4daa0f5399830d3942540196 . Thanks! -- Adam Williamson Fedora QA Community Monkey IRC: adamw | Twitter: AdamW_Fedora | identi.ca: adamwfedora http://www.happyassassin.net

3 22

Re: [alsa-devel] M-Audio FastTrack C400
by Grant Diffey 18 Aug '12

18 Aug '12

So this looks reasonably familiar, int0 is DFU int1 is mixer int2 is audio out int3 is audio in int4 is midi int5 is HID Quirk should look like the fast track ultra basically. maybe the mixer is less broken in this device? Should we start with just QUIRK_STANDARD_AUDIO_INTERFACE or copy the whole data section from the Ultra quirk? This has isosyncronous feedback for clock control needs current sound tree usb-audio tree Grant On Wed, Apr 18, 2012 at 8:45 AM, Chris Cavey <chris-alsa(a)rauros.net> wrote: > Will to help with development, may need pointers to get started and > willing to test things. I am mostly concerned with simple playback. > Recording, DSP, and Midi can all come later if at all. > > As requested the output of sudo lsusb -v -d 0763:2030" > > Bus 002 Device 003: ID 0763:2030 Midiman > Device Descriptor: > bLength 18 > bDescriptorType 1 > bcdUSB 2.00 > bDeviceClass 239 Miscellaneous Device > bDeviceSubClass 2 ? > bDeviceProtocol 1 Interface Association > bMaxPacketSize0 64 > idVendor 0x0763 Midiman > idProduct 0x2030 > bcdDevice 1.01 > iManufacturer 1 M-Audio > iProduct 2 Fast Track C400 > iSerial 0 > bNumConfigurations 1 > Configuration Descriptor: > bLength 9 > bDescriptorType 2 > wTotalLength 455 > bNumInterfaces 6 > bConfigurationValue 1 > iConfiguration 0 > bmAttributes 0x80 > (Bus Powered) > MaxPower 500mA > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 0 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 254 Application Specific Interface > bInterfaceSubClass 1 Device Firmware Update > bInterfaceProtocol 0 > iInterface 6 Fast Track C400 DFU > Device Firmware Upgrade Interface Descriptor: > bLength 9 > bDescriptorType 33 > bmAttributes 1 > Will Not Detach > Manifestation Intolerant > Upload Unsupported > Download Supported > wDetachTimeout 2000 milliseconds > wTransferSize 1024 bytes > bcdDFUVersion 1.10 > Interface Association: > bLength 8 > bDescriptorType 11 > bFirstInterface 1 > bInterfaceCount 4 > bFunctionClass 255 Vendor Specific Class > bFunctionSubClass 0 > bFunctionProtocol 32 > iFunction 2 Fast Track C400 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 1 > bInterfaceProtocol 32 > iInterface 2 Fast Track C400 > ** UNRECOGNIZED: 09 24 01 00 02 0a a3 00 00 > ** UNRECOGNIZED: 09 24 0b 80 02 81 82 03 00 > ** UNRECOGNIZED: 08 24 0a 81 03 03 03 00 > ** UNRECOGNIZED: 08 24 0a 82 00 05 03 00 > ** UNRECOGNIZED: 11 24 02 00 01 01 00 80 06 00 00 00 00 00 00 00 00 > ** UNRECOGNIZED: 0c 24 03 01 01 01 00 02 80 00 00 00 > ** UNRECOGNIZED: 11 24 02 02 03 06 00 80 04 00 00 00 00 00 00 00 00 > ** UNRECOGNIZED: 0c 24 03 03 03 06 00 40 80 00 00 00 > ** UNRECOGNIZED: 11 24 09 20 00 00 02 02 00 04 00 00 00 00 00 00 00 > ** UNRECOGNIZED: 15 24 04 40 03 02 00 43 04 00 00 00 00 00 ff ff ff > ff 0f 00 00 > ** UNRECOGNIZED: 10 24 04 42 02 02 00 01 00 00 00 00 00 ff 00 00 > ** UNRECOGNIZED: 10 24 07 43 02 00 42 fd 03 00 00 fd 03 00 00 00 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 32 > iInterface 0 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 1 > bNumEndpoints 1 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 32 > iInterface 4 Line > ** UNRECOGNIZED: 10 24 01 00 00 01 01 00 00 00 06 00 00 00 00 00 > ** UNRECOGNIZED: 06 24 02 01 03 18 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 37 > Transfer Type Isochronous > Synch Type Asynchronous > Usage Type Implicit feedback Data > wMaxPacketSize 0x00ea 1x 234 bytes > bInterval 1 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 2 > bNumEndpoints 1 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 32 > iInterface 4 Line > ** UNRECOGNIZED: 10 24 01 00 00 01 d0 00 00 00 02 00 00 00 00 00 > ** UNRECOGNIZED: 06 24 02 03 02 10 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 37 > Transfer Type Isochronous > Synch Type Asynchronous > Usage Type Implicit feedback Data > wMaxPacketSize 0x004e 1x 78 bytes > bInterval 1 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 3 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 32 > iInterface 4 Line > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 3 > bAlternateSetting 1 > bNumEndpoints 1 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 32 > iInterface 4 Line > ** UNRECOGNIZED: 10 24 01 00 00 01 01 00 00 00 04 00 00 00 00 00 > ** UNRECOGNIZED: 06 24 02 01 03 18 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x81 EP 1 IN > bmAttributes 5 > Transfer Type Isochronous > Synch Type Asynchronous > Usage Type Data > wMaxPacketSize 0x009c 1x 156 bytes > bInterval 1 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 4 > bAlternateSetting 0 > bNumEndpoints 2 > bInterfaceClass 1 Audio > bInterfaceSubClass 3 MIDI Streaming > bInterfaceProtocol 0 > iInterface 0 > MIDIStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (HEADER) > bcdADC 1.00 > wTotalLength 35 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 1 Embedded > bJackID 1 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 2 External > bJackID 2 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 8 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > Warning: Descriptor too short > bJackType 1 Embedded > bJackID 3 > bNrInputPins 1 > baSourceID( 0) 2 > BaSourcePin( 0) 1 > iJack 8 > MIDIStreaming Interface Descriptor: > bLength 8 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > Warning: Descriptor too short > bJackType 2 External > bJackID 4 > bNrInputPins 1 > baSourceID( 0) 1 > BaSourcePin( 0) 1 > iJack 105 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x07 EP 7 OUT > bmAttributes 2 > Transfer Type Bulk > Synch Type None > Usage Type Data > wMaxPacketSize 0x0200 1x 512 bytes > bInterval 0 > MIDIStreaming Endpoint Descriptor: > bLength 5 > bDescriptorType 37 > bDescriptorSubtype 1 (GENERAL) > bNumEmbMIDIJack 1 > baAssocJackID( 0) 1 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x87 EP 7 IN > bmAttributes 2 > Transfer Type Bulk > Synch Type None > Usage Type Data > wMaxPacketSize 0x0200 1x 512 bytes > bInterval 0 > MIDIStreaming Endpoint Descriptor: > bLength 5 > bDescriptorType 37 > bDescriptorSubtype 1 (GENERAL) > bNumEmbMIDIJack 1 > baAssocJackID( 0) 3 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 5 > bAlternateSetting 0 > bNumEndpoints 1 > bInterfaceClass 3 Human Interface Device > bInterfaceSubClass 0 No Subclass > bInterfaceProtocol 1 Keyboard > iInterface 0 > HID Device Descriptor: > bLength 9 > bDescriptorType 33 > bcdHID 1.10 > bCountryCode 0 Not supported > bNumDescriptors 1 > bDescriptorType 34 Report > wDescriptorLength 63 > Report Descriptors: > ** UNAVAILABLE ** > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x86 EP 6 IN > bmAttributes 3 > Transfer Type Interrupt > Synch Type None > Usage Type Data > wMaxPacketSize 0x0008 1x 8 bytes > bInterval 8 > Device Qualifier (for other device speed): > bLength 10 > bDescriptorType 6 > bcdUSB 2.00 > bDeviceClass 239 Miscellaneous Device > bDeviceSubClass 2 ? > bDeviceProtocol 1 Interface Association > bMaxPacketSize0 64 > bNumConfigurations 1 > Device Status: 0x0000 > (Bus Powered) > > > On 04/17/2012 09:15 AM, Grant Diffey wrote: > > start with an lsusb -v -d (vid:pid) pasted here > > On Tue, Apr 17, 2012 at 10:37 PM, Chris Cavey <chris-alsa(a)rauros.net>wrote: > >> I'd love to be able to help ALSA support this USB sound device. I've >> currently got the device working in a VM on my Ubuntu 12.04 workstation. >> I believe I can capture USB packets. Would this be helpful to anyone? >> Any pointers on where to get started? >> _______________________________________________ >> Alsa-devel mailing list >> Alsa-devel(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >> > > >

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[alsa-devel] Help requested: new HSS1394 MIDI back-end
by Sean M. Pappalardo - D.J. Pegasus 01 Aug '12

01 Aug '12

Hello everyone. I am a developer for Mixxx, GPL cross-platform digital DJ software: http://www.mixxx.org. My focus is the controller subsystem. I am ready to finally add Linux support for HSS1394 DJ controllers such as Stanton's SCS.1m and SCS.1d. HSS1394 is just MIDI over IEEE1394 (Firewire) at wire speed. The Windows and OSX library source code is LGPL and available on LaunchPad here: https://launchpad.net/hss1394 The Stanton SCS.1m is a sound interface integrated with a two-way control surface that speaks HSS1394. (The sound card uses an Oxford chip set and works fine with FFADO 2.0.1 today up to 96kHz. I use it with Mixxx via JACK.) The Stanton SCS.1d is a two-way control surface (no sound card) that speaks HSS1394 and is capable of sending ALOT of latency-sensitive data: about 4000 messages per platter rotation. (Now imagine if you had four or even two of them on the same network running at +50% speed. This would quickly overwhelm USB, at least at the time these devices were designed.) Read about them here: http://www.stantondj.com/stanton-controllers-systems/system1.html After a lengthy discussion on the FFADO mailing list, we determined that the best (cleanest and most flexible) way to get Linux support for these devices is to write an ALSA kernel driver that speaks HSS1394 with the devices and presents an ALSA MIDI device (with no data rate limits) for each to Linux applications. The advantage of doing this is that the devices will appear to all MIDI applications without them having to have specific support for HSS1394. This means they could be used in DAW applications as well. (FWIW, FFADO trunk has a program called test-scs that does exactly this, but using that is impractical because it depends on various internal parts of the FFADO source so would require bundling all of FFADO with the application, and it's unable to see non-audio devices due to FFADO's intended use (so it sees the SCS.1m but not the .1d.)) So my questions are: where can I find documentation on writing a new MIDI driver for ALSA? And is anyone else interested in helping me do this? Thank you very much for your time and work. Help me take the last step to my dream DJ setup! (Linux, FLAC and SCS.1 controllers.) Sincerely, Sean M. Pappalardo "D.J. Pegasus" Mixxx Developer - Controller Specialist

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[alsa-devel] [PATCH] powerpc: select PPC_CLOCK unconditionally for FSL_SOC
by Shawn Guo 10 Jul '12

10 Jul '12

Freescale PowerPC SoCs share a number of IP blocks with Freescale ARM/IMX SoCs, FlexCAN, SSI, FEC, eSDHC, USB, etc. There are some effort consolidating those drivers to make them work for both architectures. One outstanding difference between two architectures is ARM/IMX will turn off module clocks during platform initialization for power saving and expects drivers manage clocks using clk API, while PowerPC mostly does not do that, and thus does not always build in clk API. Listing all those driver Kconfig options in "select PPC_CLOCK if" seems not scalable for long term maintenance, and could easily introduce Kconfig recursive dependency. This patch chooses to select PPC_CLOCK unconditionally for FSL_SOC to always build clk API for PowerPC in. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- arch/powerpc/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/arch/powerpc/Kconfig b/arch/powerpc/Kconfig index feab3ba..63fa7fb 100644 --- a/arch/powerpc/Kconfig +++ b/arch/powerpc/Kconfig @@ -661,7 +661,7 @@ config SBUS config FSL_SOC bool select HAVE_CAN_FLEXCAN if NET && CAN - select PPC_CLOCK if CAN_FLEXCAN + select PPC_CLOCK config FSL_PCI bool -- 1.7.5.4

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Re: [alsa-devel] emu10k1 driver can't allocate memory
by Takashi Iwai 27 Jun '12

27 Jun '12

At Sun, 20 May 2012 03:04:14 +0200, Mikael Magnusson wrote: > > Hi, (please CC me on replies, not subscribed) > > I recently upgraded to 3.3.6 from 2.6.33.15 (yeah, I know). Since then > I noticed sometimes sounds were sometimes missing, then I noticed > thousands of these in dmesg, > [37280.283484] max memory size is 0x7fffffff (addr = 0x1d5d70071038000)!! I guess you are enabling IOMMU? I guess it translates the address somehow in a way emu10k1 doesn't like. There are really little changes in emu10k1 driver itself, so if the kernel upgrade changes the behavior, it must be in the core part such like IOMMU handling. Try to boot without iommu and see whether it changes the behavior again. Takashi > [37280.283486] emu: failure page = 32 > I also checked asfxload -M output, right after boot it shows all the > synth memory free as you would expect, but after a while of usage, it > is close to 0 (around 50KiB or so). Trying to run a simple program > that writes to /dev/audio fails with > write(3</dev/audio>, > "\32\21\t\3\0\0\4\n\23\34&/6:;94,#\31\17\10\2\0\0\4\v\24\36(07"..., > 4096) = -1 ENOMEM (Cannot allocate memory) > You can also note the output from asfxload -M dropping rapidly while > the program is making sound before it fails. Native alsa apps seem to > fare slightly better as the memory decrease caused by them is actually > returned when they close the device again. > > Loading and unloading soundfonts seem to work as normal, the memory is > returned when the soundfont is unloaded afaict. (I have the patch > applied that makes it possible to load soundfonts (changes the DMA32 > zone to 2GB). I don't think this is the cause of my problem since the > given addresses are way way outside the 4GB range as well.) > > Note that even well before the reported memory is at 0, apps randomly > fail to open the sound output and are silent. Curiously espeak when > successful doesn't cause a decrease in available memory, but when it > fails, it causes about 100KiB drop. > > I can't (don't dare) bisect because this is my production machine, but > I can test patches, or check the major releases between 2.6.33 and > 3.3.6 if you have no idea what causes this. > > I'm not the only one with the problem either, > http://www.google.com/search?nfpr=1&tbs=li:1&q=emu%3A+failure+page+%3D+32+m… > > -- > Mikael Magnusson >

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[alsa-devel] [RFC PATCH] Inverted internal mic
by David Henningsson 25 Jun '12

25 Jun '12

Hi, One of the more common problems on laptop machines, is that the internal mic provides a stereo signal but with one channel phase inverted, or differential output. This means problems for applications recording two channels but later merging them into one, leaving them with zero or near-zero output. There are various ways we can work around this in both the kernel, alsa-lib and PulseAudio layers. It's a matter of picking the right one. I'm leaning towards trying to fix it in the kernel's codec drivers, because 1) we already have quirking infrastructure there 2) we already have some working quirks already in that layer 3) it would benefit other sound applications that use ALSA directly. The downside to that is really that we're silencing out one channel for everyone, leading to no application being able to use both channels, even if they would implement some kind of "auto-detect-and-reverse-one-channel" functionality. For the most part, this has been some Acer Aspire machines running Realtek ALC268 / ALC269 / ALC271X / ALC272X, and for two of these we have proc coefs we can set, but for the other two these proc coefs, if they exist, are unknown to us. Recently I came across a Conexant as well, and I decided to write a patch for it, that would take the approach that the internal mic is forced mono on the kcontrol, and make sure the right channel is always muted. The patch is verified by the reporter to fix this problem. It could use some perfection though - it would make sense to to the same to the internal mic boost as well, and the strcmp('Internal Mic') call could maybe be turned into something more elegant. But before going ahead with that, I'd like to hear your opinion on the matter, if you agree that this is a good approach to the problem? References: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/903853/+attachme… https://bugs.launchpad.net/bugs/903853 -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

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