May 2013 - 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…

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[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

3 2

[alsa-devel] Trouble about Roland UA-11 UA-33 UA-55 and snd-usb-audio module
by FirebUntU 07 Mar '16

07 Mar '16

Hi guys, I write you about support of snd-usb-audio module and a relatively-new gamma of devices from Roland/Edirol: DUO CAPTURE, TRI CAPTURE and QUAD CAPTURE respectively: UA-11, UA-33, UA-55 These aren't not listed in the Alsa db. I have searched a solution for a week and I haven't found a trick to make UA-11 to work under Alsa 1.25.0 then I downloaded the driver for try to patch it and I found these devices is not listed in the tricks table In despair I have played with the switch button called "EXT * **" in the bottom of the unit of UA-11 and magically the device ID is transformed from * 0582:012b to ** 0582:0137 Image: https://dl.dropbox.com/u/32759898/output.jpg (Rear of the UA-11 unit, from the manual) *Before* (Recognized from udev - Unknown from alsa): http://www.alsa-project.org/db/?f=502ee68cf31882ba9ef23525fd66452c34d329e1 *After* (Recognized from udev - Recognized from alsa): http://www.alsa-project.org/db/?f=b2ae3e4331fb3fe613ba71e7aba1ad2e0e6dcd29 In the tricks table header of ALSA the 0582:0137 stay for PM-1000 in /TODO/ statement. I understand the EXT switch, undiscovered in the manual of the DUO CAPTURE is - as the fact - the Advanced Mode. Now the peripherical running both in playback and recording in 16 and 24 bit and without any mixer options in alsamixer. Can you update future releases with this gamma of devices? Awaiting for your positive response :) Thanks. Best regards, Leonardo

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[alsa-devel] Usb audio - unsuported sampling rates
by Gordon Dilan 12 Feb '16

12 Feb '16

Hello, I'm using an usb mic which(according to it's producer should support up to 48kHz sampling rate). The problem is ALSA allows me to use only 16 kHz for this device. I do not want to resample the data, I want to sample it at 48kHz in order to be able to record some high pitch sounds 18-20kHz. I would want to try to do any modifications necessary in order to try support 48kHz, but I'm not sure where to start. Some information: cat /proc/asound/modules ... 1 snd_usb_audio arecord --list-devices **** List of CAPTURE Hardware Devices **** .... card 1: CameraB409241 [USB Camera-B4.09.24.1], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 when I try to use: arecord -v -D hw:1,0 -f S16_LE -c 4 -r 48000 gigi.wav I get this: Recording WAVE 'gigi.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Channels 4 Warning: rate is not accurate (requested = 48000Hz, got = 16000Hz) please, try the plug plugin Hardware PCM card 1 'USB Camera-B4.09.24.1' device 0 subdevice 0 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 4 rate : 16000 exact rate : 16000 (16000/1) msbits : 16 buffer_size : 8000 period_size : 2000 period_time : 125000 tstamp_mode : NONE period_step : 1 avail_min : 2000 period_event : 0 start_threshold : 1 stop_threshold : 8000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 appl_ptr : 0 hw_ptr : 0 Any sampling rate I would try higher or lower than 16kHz, I will always get 16000 kHz. Some more information: lsusb Bus 001 Device 007: ID 1415:2000 Nam Tai E&E Products Ltd. or OmniVision Technologies, Inc. Sony Playstation Eye lsusb -t /: Bus 01.Port 1: Dev 1, Class=root_hub, Driver=ehci_hcd/2p, 480M |__ Port 1: Dev 2, If 0, Class=hub, Driver=hub/6p, 480M |__ Port 2: Dev 7, If 0, Class=vend., Driver=ov534, 480M |__ Port 2: Dev 7, If 1, Class=audio, Driver=snd-usb-audio, 480M |__ Port 2: Dev 7, If 2, Class=audio, Driver=snd-usb-audio, 480M At the moment I'm looking inside ALSA's source code to try to see from where this limitation of 16000kHz for my device appears, but I have to admit that I'm kind of lost. Any pointers on what direction should I go to try make my desired rate supported would be greatly appreciated! Thank you and best regards, Gordon

3 6

[alsa-devel] [PATCH 0/4] ASoC: fix saif record unwork issue
by Dong Aisheng 16 Jan '16

16 Jan '16

This patch series mainly fixes the saif record unwork issue. The root cause consists of: 1) clk_prepare and clk_enable un-balanced 2) missed setting a proper initial clock rate for saif in dt 3) sgtl5000 dapm widget kcontrol does not work So can not test record on mx28evk. 4) missed setting clkmux mode for saif in dt With this patch series it can fix sgtl5000 widget kcontrol unwork and saif record unwork issue. BTW, just let people konw, i still met a few overruns during record, that's another issue and will be fixed in separte patches later. Patch based on sound-2.6/for-3.6. Dong Aisheng (4): ASoC: mxs-saif: fix clock prepare and enable unbalance issue ASoC: mxs-saif: set a base clock rate for EXTMASTER mode work ASoC: sgtl5000: remove unneeded snd_soc_dapm_new_widgets in probe ARM: mx28evk: add missed clkmux select for saif arch/arm/mach-mxs/mach-mxs.c | 3 +++ sound/soc/codecs/sgtl5000.c | 2 -- sound/soc/mxs/mxs-saif.c | 24 ++++++++++++++++++++++++ 3 files changed, 27 insertions(+), 2 deletions(-)

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[alsa-devel] [PATCH - ALSA: snd-virtuoso 1/1] Xonar DSX support
by sgiurgiu11＠gmail.com 04 Jan '15

04 Jan '15

From: Sergiu Giurgiu <sgiurgiu11(a)gmail.com> This patch adds support for ASUS - Xonar DSX sound cards. Tested on openSUSE 12.2 with kernel: Linux 3.4.6-2.10-desktop #1 SMP PREEMPT Thu Jul 26 09:36:26 UTC 2012 (641c197) x86_64 x86_64 x86_64 GNU/Linux Works: - play sounds - adjust volume on master channel. - mute . Since Xonar DS uses the same chip, everything that works for DS should work for DSX as well. Thank you, Sergiu Giurgiu. Signed-off-by: Sergiu Giurgiu <sgiurgiu11(a)gmail.com> diff --git a/mirror/sound/pci/oxygen/virtuoso.c b/mirror/sound/pci/oxygen/virtuoso.c index 3d71423..8104eab 100644 --- a/mirror/sound/pci/oxygen/virtuoso.c +++ b/mirror/sound/pci/oxygen/virtuoso.c @@ -52,6 +52,7 @@ static DEFINE_PCI_DEVICE_TABLE(xonar_ids) = { { OXYGEN_PCI_SUBID(0x1043, 0x835d) }, { OXYGEN_PCI_SUBID(0x1043, 0x835e) }, { OXYGEN_PCI_SUBID(0x1043, 0x838e) }, + { OXYGEN_PCI_SUBID(0x1043, 0x8522) }, { OXYGEN_PCI_SUBID_BROKEN_EEPROM }, { } }; diff --git a/mirror/sound/pci/oxygen/xonar_wm87x6.c b/mirror/sound/pci/oxygen/xonar_wm87x6.c index 63cff90..b555b62 100644 --- a/mirror/sound/pci/oxygen/xonar_wm87x6.c +++ b/mirror/sound/pci/oxygen/xonar_wm87x6.c @@ -1255,7 +1255,6 @@ static void dump_wm87x6_registers(struct oxygen *chip, } static const struct oxygen_model model_xonar_ds = { - .shortname = "Xonar DS", .longname = "Asus Virtuoso 66", .chip = "AV200", .init = xonar_ds_init, @@ -1327,6 +1326,11 @@ int __devinit get_xonar_wm87x6_model(struct oxygen *chip, switch (id->subdevice) { case 0x838e: chip->model = model_xonar_ds; + chip->model.shortname = "Xonar DS"; + break; + case 0x8522: + chip->model = model_xonar_ds; + chip->model.shortname = "Xonar DSX"; break; case 0x835e: chip->model = model_xonar_hdav_slim; -- 1.7.10.4

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[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

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[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");

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[alsa-devel] Segfaults in snd_pcm_area_silence() when using softvol
by Tanu Kaskinen 18 May '14

18 May '14

Hi, A bug was reported to pulseaudio about a segfault in alsa-lib code when using softvol: https://bugs.freedesktop.org/show_bug.cgi?id=51060 . It has been observed by several people, and this mail was prompted by the last bug comment, which rightly recommends getting in touch with the alsa developers: > I've had this problem too and the problem only occurs with > -ftree-vectorize. > > It looks to me as if this is a gcc bug, as it produces a "movdqa" or > "vmovdqa" instruction which is used with an unaligned pointer. > > Now, I have no clue why gcc assumes the pointer would be an aligned > one, maybe it is even right and something else is wrong, but I haven't > spent too much time digging into this. > > (note that it still happens with gcc 4.7 and the current 4.8 > development one) > > Also, this bug should probably be brought up with the alsa guys as > well, as I don't think pulseaudio is at fault here. Here's the stack trace for your convenience (see the linked bug for more details) #0 snd_pcm_area_silence (dst_area=0x7fffe875bb00, dst_offset=<optimized out>, samples=2, format=SND_PCM_FORMAT_S16_LE) at pcm.c:2549 #1 0x00007fffe87b4398 in snd_pcm_areas_silence (dst_areas=0x5555557fa100, dst_offset=891, channels=2, frames=439, format=SND_PCM_FORMAT_S16_LE) at pcm.c:2662 #2 0x00007fffe87f9bdb in softvol_convert_stereo_vol (svol=0x555555813cf0, dst_areas=0x5555557fa0e0, dst_offset=891, src_areas=0x5555557fa0e0, src_offset=<optimized out>, channels=2, frames=439) at pcm_softvol.c:286 #3 0x00007fffe87fae58 in snd_pcm_softvol_write_areas (pcm=0x55555581db60, areas=0x5555557fa0e0, offset=891, size=439, slave_areas=0x5555557fa0e0, slave_offset=891, slave_sizep=0x7fffe875bcd0) at pcm_softvol.c:607 #4 0x00007fffe87c6fc7 in snd_pcm_plugin_mmap_commit (pcm=0x55555581db60, offset=<optimized out>, size=439) at pcm_plugin.c:424 #5 0x00007fffe87b9676 in snd_pcm_mmap_commit (pcm=<optimized out>, offset=<optimized out>, frames=<optimized out>) at pcm.c:6584 #6 0x00007fffe8a74876 in mmap_write (on_timeout=false, polled=false, sleep_usec=<synthetic pointer>, u=0x5555557e6d40) at modules/alsa/alsa-sink.c:636 #7 thread_func (userdata=0x5555557e6d40) at modules/alsa/alsa-sink.c:1694 #8 0x00007ffff722c039 in internal_thread_func (userdata=0x55555587ef80) at pulsecore/thread-posix.c:83 #9 0x00007ffff4a24e2c in ?? () from /lib64/libpthread.so.0 #10 0x00007ffff3ccc1dd in clone () from /lib64/libc.so.6 -- Tanu

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Re: [alsa-devel] snd-usb: "delay: estimated 0, actual 352"
by Takashi Iwai 08 Jan '14

08 Jan '14

At Thu, 6 Sep 2012 15:17:58 +0200, Markus Trippelsdorf wrote: > > On 2012.09.06 at 15:08 +0200, Takashi Iwai wrote: > > At Thu, 6 Sep 2012 11:43:48 +0200, > > Markus Trippelsdorf wrote: > > > > > > On 2012.09.06 at 10:21 +0200, Takashi Iwai wrote: > > > > At Thu, 06 Sep 2012 09:35:26 +0200, > > > > Takashi Iwai wrote: > > > > > > > > In short, a patch like below may fix the issue (note: completely > > > > untested!) > > > > > > No it doesn't, unfortunately... > > > > OK, I start tracking down the problem a bit more deeply now. > > > > The issue happens when the first two URBs are passed to > > retire_playback_urb(). These are URBs filled before start_endpoints() > > are set, so they contain actually zero size. Even though these are > > a sort of dummy packets, the driver still tries to check with the > > queued delay account, and gives bogus errors. > > > > So, essentially the messages are harmless and nothing to worry too > > much, but surely it doesn't look sexy. > > > > The patch below should fix the problem. Please give it a try. > > Yes, your patch finally fixes the problem. > Thank you Takashi-san. Thanks for your quick test! If Daniel has no objection with that patch, I'm going to merge it. Takashi

3 2

[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. 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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

Re: [alsa-devel] usb-audio: Reloop Play support (TI TUSB3200AC)
by Daniel Mack 23 Aug '13

23 Aug '13

[Please do not drop the mailing list address] On 19.10.2012 15:45, Didier 'Ptitjes' Villevalois wrote: > On ven., 2012-10-19 at 15:20 +0200, Daniel Mack wrote: >> On 19.10.2012 15:09, Didier 'Ptitjes' Villevalois wrote: >>> Hello, >>> >>> I've bought a Reloop Play usb soundcard and I'd like to make it work >>> with ALSA. >>> >>> It is a 4-channels USB soundcard. It is currently correctly recognized >>> by ALSA as a 4-channels soundcard and I can correctly open the channels >>> and feed them with audio. However, no audio is output by the box. >> >> You're most probably affected by a regression in the driver that was >> fixes in 3.6.0 and 3.5.5 (and all later kernels). Could you please try >> one of these and tell us if that works? > > Sure, I'll do that now (I'm currently using 3.5.4 + mbp8,3 patches). > I guess 3.5.7 is OK ? I'm building it now and will reboot on that. Any kernel >= 3.6.0 and 3.5.5 should be ok. >>> It seems to contain a Texas Instrument TUSB3200AC chipset (same chipset >>> as the Maya44 USB and Hercules DJ console). I guess the problem is a >>> firmware problem but I don't know what to do about it. >>> >>> I've got a MacOS X driver that works correctly, if that of any help. (I >>> suppose the Windows driver works too but I don't have any Windows at >>> hand.) >> >> Did you install any vendor driver on OS X or does it just work with the >> one that ships with the OS? > > Yep, I had to install the Vendor-provided driver. What was the effect without that special driver? Could you see the device listed in the AudioMIDISetup? Could you stream audio? IOW, is the effect you see there comparable to how your Linux box behaves? You could try and remove that driver again and see what happens. > But it seems the > built-in USB soundcard support on MacOS is very poor. No it's not. In fact, it's actually very good. Vendors who keep their hardware compliant to the USB audio spec don't have to ship any driver at all. And this is how it should be on Linux, too. > Also the mpkg install file (inside the provided install dmg file) was > called something like "firmware0_and_driver.mpkg". This is why I > suspected a firmware problem. However I tried to look at the package > content and did not find any .bin file or whatever could be a standalone > firmware file. Maybe it is embedded in the driver install executable. Maybe, but they could also write the firmware permanently to the device once with that tool. That would be perfect for Linux users. Daniel

3 10

[alsa-devel] [PATCH 0/4 V7] Power-well API implementation for Haswell
by Wang Xingchao 25 Jul '13

25 Jul '13

Hi all, This is V7 and here're some changes notes: change from V6-->V7: - rename variable - use HAS_POWER_WELL instead of IS_HASWELL - put structure inside drm_i915_private - use WARN_ON for global pointer check change from V5-->V6: - Remove duplication code in new introduced probe work - move duplication code in azx_probe_continue - remove unused #ifdef - replace request_module with symbol_request - replace spin_lock_irq with spin_lock_irqsave in gfx side - other typo fixes (review by Takashi) change from V4-->V5: - fix reference count bug - new patch on general runtime pm support for audio pci device - new patch to avoid request_module() deadlock change between V3-->V4: - add new structure i915_power_well - initialize drm_device pointer at module init time - change function name change between V2-->V3: - make SND_HDA_I915 selectable - use snd_printdd to output message - add return error code check - use symbol_request to replace symbol_get - release power_well at azx_free - some typo fixes changes between V1-->V2: - use reference count to track power-well usage - remove external module, compiled into snd-hda-intel instead - manage symbols and module loading properly - remove IS_HSW macro, use flag instead - remove audio callback for gfx driver to avoid dependency - split whole patch into two pieces for easy review - more typo fixes Takashi Iwai (1): ALSA: hda - Move azx_first_init() into azx_probe_continue() Wang Xingchao (3): ALSA: hda - Fix runtime PM check ALSA: hda - Add power-welll support for haswell HDA i915/drm: Add private api for power well usage drivers/gpu/drm/i915/i915_dma.c | 6 +++ drivers/gpu/drm/i915/i915_drv.h | 12 ++++++ drivers/gpu/drm/i915/intel_drv.h | 4 ++ drivers/gpu/drm/i915/intel_pm.c | 81 ++++++++++++++++++++++++++++++++--- include/drm/i915_powerwell.h | 36 ++++++++++++++++ sound/pci/hda/Kconfig | 10 +++++ sound/pci/hda/Makefile | 2 + sound/pci/hda/hda_i915.c | 75 ++++++++++++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 35 +++++++++++++++ sound/pci/hda/hda_intel.c | 87 ++++++++++++++++++++++++++++++-------- 10 files changed, 324 insertions(+), 24 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.7.9.5

7 40

[alsa-devel] idea: a reserve alsa plugin
by David Henningsson 17 Jul '13

17 Jul '13

Just had an idea which I'll write down here before I forget it again...and I'm not saying I'll implement this anytime soon either, but here goes: There is a device reserve protocol between PulseAudio and JACK2 - when JACK needs the sound card, it'll send a dbus message to PulseAudio and grab a name in D-Bus. However, there are plenty of applications who like to access ALSA directly, without going through JACK2 or PulseAudio. By making a "reserve" plugin, we could have this functionality for those apps too. In practice, if the app usually opens "plughw:0" or "hw:0", it could instead open "reserve:plughw:0" or "reserve:hw:0" to also reserve the device from PulseAudio usage while the device is open. Meanwhile, PulseAudio is free to use other audio devices (which is not the case when using e g pasuspender). How does that sound? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

6 12

[alsa-devel] [PATCH 0/5] ASoC: add simple-card DT support
by Kuninori Morimoto 12 Jul '13

12 Jul '13

Hi Mark, Simon These patches add DT support on simple-card driver. #1 patch adds new snd_soc_of_parse_daifmt() function on soc-core.c and #5 patch will use it. #2 - #5 patches are based on mark/topic/fsi branch >> Simon #2, #3 patches modify sh-arm platform settings. please check it Kuninori Morimoto (5): ASoC: add snd_soc_of_parse_daifmt() ASoC: simple-card: use struct device pointer for dev_xxx() ASoC: simple-card: add asoc_simple_dai_set for initializing ASoC: simple-card: remove pointless struct asoc_simple_dai_init_info ASoC: simple-card: add Device Tree support .../devicetree/bindings/sound/simple-card.txt | 53 ++++++++ arch/arm/mach-shmobile/board-ap4evb.c | 19 +-- arch/arm/mach-shmobile/board-armadillo800eva.c | 18 +-- arch/arm/mach-shmobile/board-kzm9g.c | 12 +- arch/arm/mach-shmobile/board-mackerel.c | 19 +-- arch/sh/boards/mach-ecovec24/setup.c | 11 +- arch/sh/boards/mach-se/7724/setup.c | 13 +- include/sound/simple_card.h | 11 +- include/sound/soc.h | 2 + sound/soc/generic/simple-card.c | 132 ++++++++++++++++---- sound/soc/soc-core.c | 115 +++++++++++++++++ 11 files changed, 307 insertions(+), 98 deletions(-) Best regards --- Kuninori Morimoto

6 110

[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] [PATCH alsa-lib 1/2] pcm_hw: Remove unused fields in struct
by David Henningsson 26 Jun '13

26 Jun '13

These fields are not used, and their name similarity to other fields are quite confusing when trying to debug alsa-lib. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- src/pcm/pcm_hw.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 0f3226d..196393d 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -95,8 +95,6 @@ typedef struct { volatile struct snd_pcm_mmap_status * mmap_status; struct snd_pcm_mmap_control *mmap_control; struct snd_pcm_sync_ptr *sync_ptr; - snd_pcm_uframes_t hw_ptr; - snd_pcm_uframes_t appl_ptr; int period_event; snd_timer_t *period_timer; struct pollfd period_timer_pfd; -- 1.7.9.5

3 4

[alsa-devel] PulseAudio and softvol
by Arun Raghavan 26 Jun '13

26 Jun '13

Hello, A number of users have intermittently(?) been hitting a crash in alsa-lib 1.0.27 [1, 2] related to the softvol plugin. I'm not able to reproduce this reliably, so can't find an easy way to debug/fix. However, this raises a tangential question - why do we need softvol to be plugged for 'front' at all? David explained to me that this is to guarantee the existence of a PCM control. Perhaps I don't fully understand this, because I'm unconvinced by the reason. Could someone explain/refute? This is especially bad for us, from PulseAudio's perspective, because we aren't getting a zero-copy path. Cheers, Arun [1] https://bugzilla.redhat.com/show_bug.cgi?id=953352 [2] https://bugs.freedesktop.org/show_bug.cgi?id=64299

5 25

[alsa-devel] [PATCH] Add M2Tech hiFace USB-SPDIF driver
by Antonio Ospite 24 Jun '13

24 Jun '13

Add driver for M2Tech hiFace USB-SPDIF interface and compatible devices. The supported products are: * M2Tech Young * M2Tech hiFace * M2Tech North Star * M2Tech W4S Young * M2Tech Corrson * M2Tech AUDIA * M2Tech SL Audio * M2Tech Empirical * M2Tech Rockna * M2Tech Pathos * M2Tech Metronome * M2Tech CAD * M2Tech Audio Esclusive * M2Tech Rotel * M2Tech Eeaudio * The Chord Company CHORD * AVA Group A/S Vitus Signed-off-by: Antonio Ospite <ao2(a)amarulasolutions.com> --- Hi, this is my first ALSA driver I still have some doubts and I marked them in the comments below with a XXX prefix, could someone please take a look at them? I plan on submitting a version 2 of the driver after I get some feedback, especially on those doubts. I also noticed that many drivers under sound/usb/ have text in the Kconfig entry stating "Say Y here to include support for ..." but I only see "[N/m/?]" as options when I run "make oldconfig" maybe because of some dependency, I am not sure, should such text be removed in order to avoid confusion? Thanks, Antonio sound/usb/Kconfig | 33 +++ sound/usb/Makefile | 2 +- sound/usb/hiface/Makefile | 2 + sound/usb/hiface/chip.c | 331 ++++++++++++++++++++++ sound/usb/hiface/chip.h | 34 +++ sound/usb/hiface/pcm.c | 666 +++++++++++++++++++++++++++++++++++++++++++++ sound/usb/hiface/pcm.h | 24 ++ 7 files changed, 1091 insertions(+), 1 deletion(-) create mode 100644 sound/usb/hiface/Makefile create mode 100644 sound/usb/hiface/chip.c create mode 100644 sound/usb/hiface/chip.h create mode 100644 sound/usb/hiface/pcm.c create mode 100644 sound/usb/hiface/pcm.h diff --git a/sound/usb/Kconfig b/sound/usb/Kconfig index 225dfd7..2b206e8 100644 --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -115,5 +115,38 @@ config SND_USB_6FIRE and further help can be found at http://sixfireusb.sourceforge.net +config SND_USB_HIFACE + tristate "M2Tech hiFace USB-SPDIF driver" + select SND_PCM + help + Say Y here to include support for M2Tech hiFace USB-SPDIF interface. + + M2Tech hiFace and compatible devices offer a Hi-End S/PDIF Output + Interface, see http://www.m2tech.biz/hiface.html + + This driver supports the original M2Tech hiFace and some other + compatible devices. The supported products are: + + * M2Tech Young + * M2Tech hiFace + * M2Tech North Star + * M2Tech W4S Young + * M2Tech Corrson + * M2Tech AUDIA + * M2Tech SL Audio + * M2Tech Empirical + * M2Tech Rockna + * M2Tech Pathos + * M2Tech Metronome + * M2Tech CAD + * M2Tech Audio Esclusive + * M2Tech Rotel + * M2Tech Eeaudio + * The Chord Company CHORD + * AVA Group A/S Vitus + + To compile this driver as a module, choose M here: the module + will be called snd-usb-hiface. + endif # SND_USB diff --git a/sound/usb/Makefile b/sound/usb/Makefile index ac256dc..abe668f 100644 --- a/sound/usb/Makefile +++ b/sound/usb/Makefile @@ -23,4 +23,4 @@ obj-$(CONFIG_SND_USB_UA101) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_USX2Y) += snd-usbmidi-lib.o obj-$(CONFIG_SND_USB_US122L) += snd-usbmidi-lib.o -obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ +obj-$(CONFIG_SND) += misc/ usx2y/ caiaq/ 6fire/ hiface/ diff --git a/sound/usb/hiface/Makefile b/sound/usb/hiface/Makefile new file mode 100644 index 0000000..463b136 --- /dev/null +++ b/sound/usb/hiface/Makefile @@ -0,0 +1,2 @@ +snd-usb-hiface-objs := chip.o pcm.o +obj-$(CONFIG_SND_USB_HIFACE) += snd-usb-hiface.o diff --git a/sound/usb/hiface/chip.c b/sound/usb/hiface/chip.c new file mode 100644 index 0000000..dce74e9 --- /dev/null +++ b/sound/usb/hiface/chip.c @@ -0,0 +1,331 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * 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. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <sound/initval.h> + +#include "chip.h" +#include "pcm.h" + +MODULE_AUTHOR("Michael Trimarchi <michael(a)amarulasolutions.com>"); +MODULE_AUTHOR("Antonio Ospite <ao2(a)amarulasolutions.com>"); +MODULE_DESCRIPTION("M2Tech HiFace USB audio driver"); +MODULE_LICENSE("GPL v2"); +MODULE_SUPPORTED_DEVICE("{{HiFace, Evo}}"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-max */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* Id for card */ +static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ + +#define DRIVER_NAME "snd-usb-hiface" +#define CARD_NAME "HiFace" + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); + +static struct hiface_chip *chips[SNDRV_CARDS] = SNDRV_DEFAULT_PTR; + +static DEFINE_MUTEX(register_mutex); + +struct hiface_vendor_quirk { + const char *device_name; + u8 extra_freq; +}; + +static int hiface_dev_free(struct snd_device *device) +{ + struct hiface_chip *chip = device->device_data; + kfree(chip); + return 0; +} + +static int hiface_chip_create(struct usb_device *device, int idx, + const struct hiface_vendor_quirk *quirk, + struct hiface_chip **rchip) +{ + struct snd_card *card = NULL; + struct hiface_chip *chip; + int ret; + static struct snd_device_ops ops = { + .dev_free = hiface_dev_free, + }; + + *rchip = NULL; + + /* if we are here, card can be registered in alsa. */ + ret = snd_card_create(index[idx], id[idx], THIS_MODULE, 0, &card); + if (ret < 0) { + snd_printk(KERN_ERR "cannot create alsa card.\n"); + return ret; + } + + strcpy(card->driver, DRIVER_NAME); + + if (quirk && quirk->device_name) + strcpy(card->shortname, quirk->device_name); + else + strcpy(card->shortname, "M2Tech generic audio"); + + sprintf(card->longname, "%s at %d:%d", card->shortname, + device->bus->busnum, device->devnum); + + chip = kzalloc(sizeof(*chip), GFP_KERNEL); + if (!chip) { + snd_card_free(card); + return -ENOMEM; + } + + chip->dev = device; + chip->index = idx; + chip->card = card; + + ret = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (ret < 0) { + kfree(chip); + snd_card_free(card); + return ret; + } + + *rchip = chip; + return 0; +} + +static int hiface_chip_probe(struct usb_interface *intf, + const struct usb_device_id *usb_id) +{ + const struct hiface_vendor_quirk *quirk = (struct hiface_vendor_quirk *)usb_id->driver_info; + int ret; + int i; + struct hiface_chip *chip; + struct usb_device *device = interface_to_usbdev(intf); + + pr_info("Probe " DRIVER_NAME " driver.\n"); + + ret = usb_set_interface(device, 0, 0); + if (ret != 0) { + snd_printk(KERN_ERR "can't set first interface for " CARD_NAME " device.\n"); + return -EIO; + } + + /* check whether the card is already registered */ + chip = NULL; + mutex_lock(&register_mutex); + for (i = 0; i < SNDRV_CARDS; i++) { + if (chips[i] && chips[i]->dev == device) { + if (chips[i]->shutdown) { + snd_printk(KERN_ERR CARD_NAME " device is in the shutdown state, cannot create a card instance\n"); + ret = -ENODEV; + goto err; + } + chip = chips[i]; + break; + } + } + if (!chip) { + /* it's a fresh one. + * now look for an empty slot and create a new card instance + */ + for (i = 0; i < SNDRV_CARDS; i++) + if (enable[i] && !chips[i]) { + ret = hiface_chip_create(device, i, quirk, + &chip); + if (ret < 0) + goto err; + + snd_card_set_dev(chip->card, &intf->dev); + break; + } + if (!chip) { + snd_printk(KERN_ERR "no available " CARD_NAME " audio device\n"); + ret = -ENODEV; + goto err; + } + } + + ret = hiface_pcm_init(chip, quirk ? quirk->extra_freq : 0); + if (ret < 0) + goto err_chip_destroy; + + ret = snd_card_register(chip->card); + if (ret < 0) { + snd_printk(KERN_ERR "cannot register " CARD_NAME " card\n"); + goto err_chip_destroy; + } + + chips[chip->index] = chip; + chip->intf_count++; + + mutex_unlock(&register_mutex); + + usb_set_intfdata(intf, chip); + return 0; + +err_chip_destroy: + snd_card_free(chip->card); +err: + mutex_unlock(&register_mutex); + return ret; +} + +static void hiface_chip_disconnect(struct usb_interface *intf) +{ + struct hiface_chip *chip; + struct snd_card *card; + + pr_debug("%s: called.\n", __func__); + + chip = usb_get_intfdata(intf); + if (!chip) + return; + + card = chip->card; + chip->intf_count--; + if (chip->intf_count <= 0) { + /* Make sure that the userspace cannot create new request */ + snd_card_disconnect(card); + + mutex_lock(&register_mutex); + chips[chip->index] = NULL; + mutex_unlock(&register_mutex); + + chip->shutdown = true; + hiface_pcm_abort(chip); + snd_card_free_when_closed(card); + } +} + +static struct usb_device_id device_table[] = { + { + USB_DEVICE(0x04b4, 0x0384), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Young", + .extra_freq = 1, + } + }, + { + USB_DEVICE(0x04b4, 0x930b), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "hiFace", + } + }, + { + USB_DEVICE(0x04b4, 0x931b), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "North Star", + } + }, + { + USB_DEVICE(0x04b4, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "W4S Young", + } + }, + { + USB_DEVICE(0x04b4, 0x931d), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Corrson", + } + }, + { + USB_DEVICE(0x04b4, 0x931e), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "AUDIA", + } + }, + { + USB_DEVICE(0x04b4, 0x931f), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "SL Audio", + } + }, + { + USB_DEVICE(0x04b4, 0x9320), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Empirical", + } + }, + { + USB_DEVICE(0x04b4, 0x9321), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Rockna", + } + }, + { + USB_DEVICE(0x249c, 0x9001), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Pathos", + } + }, + { + USB_DEVICE(0x249c, 0x9002), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Metronome", + } + }, + { + USB_DEVICE(0x249c, 0x9006), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "CAD", + } + }, + { + USB_DEVICE(0x249c, 0x9008), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Audio Esclusive", + } + }, + { + USB_DEVICE(0x249c, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Rotel", + } + }, + { + USB_DEVICE(0x249c, 0x932c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Eeaudio", + } + }, + { + USB_DEVICE(0x245f, 0x931c), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "CHORD", + } + }, + { + USB_DEVICE(0x25c6, 0x9002), + .driver_info = (unsigned long)&(const struct hiface_vendor_quirk) { + .device_name = "Vitus", + } + }, + {} +}; + +MODULE_DEVICE_TABLE(usb, device_table); + +static struct usb_driver hiface_usb_driver = { + .name = DRIVER_NAME, + .probe = hiface_chip_probe, + .disconnect = hiface_chip_disconnect, + .id_table = device_table, +}; + +module_usb_driver(hiface_usb_driver); diff --git a/sound/usb/hiface/chip.h b/sound/usb/hiface/chip.h new file mode 100644 index 0000000..c2e9fda --- /dev/null +++ b/sound/usb/hiface/chip.h @@ -0,0 +1,34 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * 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. + */ + +#ifndef HIFACE_CHIP_H +#define HIFACE_CHIP_H + +#include <linux/usb.h> +#include <sound/core.h> + +struct pcm_runtime; + +struct hiface_chip { + struct usb_device *dev; + struct snd_card *card; + int intf_count; /* number of registered interfaces */ + int index; /* index in module parameter arrays */ + bool shutdown; + + struct pcm_runtime *pcm; +}; +#endif /* HIFACE_CHIP_H */ diff --git a/sound/usb/hiface/pcm.c b/sound/usb/hiface/pcm.c new file mode 100644 index 0000000..34fe0ed --- /dev/null +++ b/sound/usb/hiface/pcm.c @@ -0,0 +1,666 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * 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. + */ + +#include <linux/slab.h> +#include <sound/pcm.h> + +#include "pcm.h" +#include "chip.h" + +#define OUT_EP 0x2 +#define PCM_N_URBS 8 +#define PCM_MAX_PACKET_SIZE 4096 +#define MAX_BUFSIZE (2 * PCM_N_URBS * PCM_MAX_PACKET_SIZE) + +struct pcm_urb { + struct hiface_chip *chip; + + struct urb instance; + struct usb_anchor submitted; + u8 *buffer; +}; + +struct pcm_substream { + spinlock_t lock; + struct snd_pcm_substream *instance; + + bool active; + snd_pcm_uframes_t dma_off; /* current position in alsa dma_area */ + snd_pcm_uframes_t period_off; /* current position in current period */ +}; + +enum { /* pcm streaming states */ + STREAM_DISABLED, /* no pcm streaming */ + STREAM_STARTING, /* pcm streaming requested, waiting to become ready */ + STREAM_RUNNING, /* pcm streaming running */ + STREAM_STOPPING +}; + +struct pcm_runtime { + struct hiface_chip *chip; + struct snd_pcm *instance; + + struct pcm_substream playback; + bool panic; /* if set driver won't do anymore pcm on device */ + + struct pcm_urb out_urbs[PCM_N_URBS]; + + struct mutex stream_mutex; + u8 stream_state; /* one of STREAM_XXX */ + u8 rate; /* one of PCM_RATE_XXX */ + u8 extra_freq; + wait_queue_head_t stream_wait_queue; + bool stream_wait_cond; +}; + +static const unsigned int rates[] = { 44100, 48000, 88200, 96000, 176400, 192000, + 352800, 384000 }; +static struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates) - 2, /* by default rates up to 192000 are supported */ + .list = rates, + .mask = 0, +}; + +static const int rates_alsaid[] = { + SNDRV_PCM_RATE_44100, SNDRV_PCM_RATE_48000, + SNDRV_PCM_RATE_88200, SNDRV_PCM_RATE_96000, + SNDRV_PCM_RATE_176400, SNDRV_PCM_RATE_192000, + SNDRV_PCM_RATE_KNOT, SNDRV_PCM_RATE_KNOT }; + +static const struct snd_pcm_hardware pcm_hw = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_BATCH, + + .formats = SNDRV_PCM_FMTBIT_S32_LE, + + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_176400 | + SNDRV_PCM_RATE_192000 | + SNDRV_PCM_RATE_KNOT, + + .rate_min = 44100, + .rate_max = 192000, /* changes in hiface_pcm_open to support extra rates */ + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = MAX_BUFSIZE, + .period_bytes_min = PCM_MAX_PACKET_SIZE, + .period_bytes_max = MAX_BUFSIZE, + .periods_min = 2, + .periods_max = 1024 +}; + +static int hiface_pcm_set_rate(struct pcm_runtime *rt, unsigned int rate) +{ + u8 rate_value[] = { 0x43, 0x4b, 0x42, 0x4a, 0x40, 0x48, 0x58, 0x68 }; + int ret; + struct usb_device *device = rt->chip->dev; + + /* We are already sure that the rate is supported here thanks to the + * contraints set in hiface_pcm_open(), but we have to retrieve the + * index to access rate_value. + */ + for (rt->rate = 0; rt->rate < ARRAY_SIZE(rates); rt->rate++) + if (rate == rates[rt->rate]) + break; + + if (unlikely(rt->rate == ARRAY_SIZE(rates))) { + pr_err("Unsupported rate %d\n", rate); + return -EINVAL; + } + + /* + * USBIO: Vendor 0xb0(wValue=0x0043, wIndex=0x0000) + * 43 b0 43 00 00 00 00 00 + * USBIO: Vendor 0xb0(wValue=0x004b, wIndex=0x0000) + * 43 b0 4b 00 00 00 00 00 + * This control message doesn't have any ack from the + * other side + */ + ret = usb_control_msg(device, usb_sndctrlpipe(device, 0), + 0xb0, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_OTHER, + rate_value[rt->rate], 0, NULL, 0, 100); + if (ret < 0) { + snd_printk(KERN_ERR "Error setting samplerate %d.\n", + rates[rt->rate]); + return ret; + } + + return 0; +} + +static struct pcm_substream *hiface_pcm_get_substream( + struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + + if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) + return &rt->playback; + + pr_debug("Error getting pcm substream slot.\n"); + return NULL; +} + +/* call with stream_mutex locked */ +static void hiface_pcm_stream_stop(struct pcm_runtime *rt) +{ + int i, time; + + if (rt->stream_state != STREAM_DISABLED) { + rt->stream_state = STREAM_STOPPING; + + for (i = 0; i < PCM_N_URBS; i++) { + time = usb_wait_anchor_empty_timeout( + &rt->out_urbs[i].submitted, 100); + if (!time) + usb_kill_anchored_urbs( + &rt->out_urbs[i].submitted); + usb_kill_urb(&rt->out_urbs[i].instance); + } + + rt->stream_state = STREAM_DISABLED; + } +} + +/* call with stream_mutex locked */ +static int hiface_pcm_stream_start(struct pcm_runtime *rt) +{ + int ret = 0, i; + + if (rt->stream_state == STREAM_DISABLED) { + /* submit our out urbs zero init */ + rt->stream_state = STREAM_STARTING; + for (i = 0; i < PCM_N_URBS; i++) { + memset(rt->out_urbs[i].buffer, 0, PCM_MAX_PACKET_SIZE); + usb_anchor_urb(&rt->out_urbs[i].instance, + &rt->out_urbs[i].submitted); + ret = usb_submit_urb(&rt->out_urbs[i].instance, + GFP_ATOMIC); + if (ret) { + hiface_pcm_stream_stop(rt); + return ret; + } + } + + /* wait for first out urb to return (sent in in urb handler) */ + wait_event_timeout(rt->stream_wait_queue, rt->stream_wait_cond, + HZ); + if (rt->stream_wait_cond) { + pr_debug("%s: Stream is running wakeup event\n", + __func__); + rt->stream_state = STREAM_RUNNING; + } else { + hiface_pcm_stream_stop(rt); + return -EIO; + } + } + return ret; +} + + +/* The hardware wants word-swapped 32-bit values */ +void memcpy_swahw32(u8 *dest, u8 *src, unsigned int n) +{ + unsigned int i; + + for (i = 0; i < n / 4; i++) + ((u32 *)dest)[i] = swahw32(((u32 *)src)[i]); +} + +/* call with substream locked */ +static int hiface_pcm_playback(struct pcm_substream *sub, + struct pcm_urb *urb) +{ + struct snd_pcm_runtime *alsa_rt = sub->instance->runtime; + u8 *source; + unsigned int pcm_buffer_size; + + /* XXX Can an invalid format make it to here? + * Isn't ALSA checking pcm_hw.formats ? + */ + if (alsa_rt->format != SNDRV_PCM_FORMAT_S32_LE) { + pr_err("Unsupported sample format\n"); + return -EINVAL; + } + + pcm_buffer_size = snd_pcm_lib_buffer_bytes(sub->instance); + + if (sub->dma_off + PCM_MAX_PACKET_SIZE <= pcm_buffer_size) { + pr_debug("%s: (1) buffer_size %#x dma_offset %#x\n", __func__, + (unsigned int) pcm_buffer_size, + (unsigned int) sub->dma_off); + + source = alsa_rt->dma_area + sub->dma_off; + memcpy_swahw32(urb->buffer, source, PCM_MAX_PACKET_SIZE); + } else { + /* wrap around at end of ring buffer */ + unsigned int len; + + pr_debug("%s: (2) buffer_size %#x dma_offset %#x\n", __func__, + (unsigned int) pcm_buffer_size, + (unsigned int) sub->dma_off); + + len = pcm_buffer_size - sub->dma_off; + + source = alsa_rt->dma_area + sub->dma_off; + memcpy_swahw32(urb->buffer, source, len); + + source = alsa_rt->dma_area; + memcpy_swahw32(urb->buffer + len, source, + PCM_MAX_PACKET_SIZE - len); + } + sub->dma_off += PCM_MAX_PACKET_SIZE; + if (sub->dma_off >= pcm_buffer_size) + sub->dma_off -= pcm_buffer_size; + + sub->period_off += PCM_MAX_PACKET_SIZE; + + return 0; +} + +static void hiface_pcm_out_urb_handler(struct urb *usb_urb) +{ + struct pcm_urb *out_urb = usb_urb->context; + struct pcm_runtime *rt = out_urb->chip->pcm; + struct pcm_substream *sub; + unsigned long flags; + + pr_debug("%s: called.\n", __func__); + + if (usb_urb->status || rt->panic || rt->stream_state == STREAM_STOPPING) + return; + + if (rt->stream_state == STREAM_STARTING) { + rt->stream_wait_cond = true; + wake_up(&rt->stream_wait_queue); + } + + /* now send our playback data (if a free out urb was found) */ + sub = &rt->playback; + spin_lock_irqsave(&sub->lock, flags); + if (sub->active) { + int ret; + + ret = hiface_pcm_playback(sub, out_urb); + if (ret < 0) { + spin_unlock_irqrestore(&sub->lock, flags); + goto out_fail; + } + if (sub->period_off >= sub->instance->runtime->period_size) { + sub->period_off %= sub->instance->runtime->period_size; + spin_unlock_irqrestore(&sub->lock, flags); + snd_pcm_period_elapsed(sub->instance); + } else { + spin_unlock_irqrestore(&sub->lock, flags); + } + } else { + memset(out_urb->buffer, 0, PCM_MAX_PACKET_SIZE); + spin_unlock_irqrestore(&sub->lock, flags); + } +out_fail: + usb_submit_urb(&out_urb->instance, GFP_ATOMIC); +} + +static int hiface_pcm_open(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = NULL; + struct snd_pcm_runtime *alsa_rt = alsa_sub->runtime; + int ret; + + pr_debug("%s: called.\n", __func__); + + if (rt->panic) + return -EPIPE; + + mutex_lock(&rt->stream_mutex); + alsa_rt->hw = pcm_hw; + + if (alsa_sub->stream == SNDRV_PCM_STREAM_PLAYBACK) { + + /* XXX can we avoid setting hw.rates below? + * + * FWIU setting alsa_rt->hw.rates in the open callback can be + * useful in two cases: + * + * - different hardware revisions have different + * capabilities + * + * - different stream types must have the same capabilities + * + * but in our case it should not be needed as we only have one + * playback stream and we define the capabilities via + * constraints already to avoid invalid rates to be used. + * + * Moreover, following the code stolen from 6fire it seems + * impossible to get to setting alsa_rt->hw.rates here: + * + * hiface_pcm_init: + * rt->rate = ARRAY_SIZE(rates); + * + * hiface_pcm_open: + * if (rt->rate < ARRAY_SIZE(rates)) + * alsa_rt->hw.rates = rates_alsaid[rt->rate]; + * // So it does not get here on the first open + * + * hiface_pcm_prepare: + * hiface_pcm_set_rate: + * for (rt->rate = 0; rt->rate < ARRAY_SIZE(rates); rt->rate++) + * if (rate == rates[rt->rate]) + * break; + * + * hiface_pcm_close: + * rt->rate = ARRAY_SIZE(rates); + * // So it won't set alsa_rt->hw.rates at the next open either + * + * So the only way to set alsa_rt->hw.rates is to have two + * calls to the open callback, is that even possible? ALSA + * does not allow sharing PCM streams AFAIK and we use only + * one playback stream. + * + * Maybe the 6fire driver was setting alsa_rt->hw.rates in + * order to have the playback stream and the capture stream + * use the same rate? I.e. the first open,prepare,set_rate() + * sequence on one type of stream will decide the rate and the + * second open on the other type would restrict the supported + * rate to the one already in use, but I am not sure. + * + * If the rationale makes sense then we don't need to set + * alsa_rt->hw.rates and we can remove this and related code. + */ + if (rt->rate < ARRAY_SIZE(rates)) + alsa_rt->hw.rates = rates_alsaid[rt->rate]; + + sub = &rt->playback; + } + + if (!sub) { + mutex_unlock(&rt->stream_mutex); + pr_err("Invalid stream type\n"); + return -EINVAL; + } + + if (rt->extra_freq) { + alsa_rt->hw.rate_max = 384000; + constraints_rates.count = ARRAY_SIZE(rates); + } + + ret = snd_pcm_hw_constraint_list(alsa_sub->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_rates); + if (ret < 0) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + + sub->instance = alsa_sub; + sub->active = false; + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_close(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + unsigned long flags; + + if (rt->panic) + return 0; + + pr_debug("%s: called.\n", __func__); + + mutex_lock(&rt->stream_mutex); + if (sub) { + /* deactivate substream */ + spin_lock_irqsave(&sub->lock, flags); + sub->instance = NULL; + sub->active = false; + spin_unlock_irqrestore(&sub->lock, flags); + + /* all substreams closed? if so, stop streaming */ + if (!rt->playback.instance) { + hiface_pcm_stream_stop(rt); + rt->rate = ARRAY_SIZE(rates); + } + } + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_hw_params(struct snd_pcm_substream *alsa_sub, + struct snd_pcm_hw_params *hw_params) +{ + pr_debug("%s: called.\n", __func__); + return snd_pcm_lib_malloc_pages(alsa_sub, + params_buffer_bytes(hw_params)); +} + +static int hiface_pcm_hw_free(struct snd_pcm_substream *alsa_sub) +{ + pr_debug("%s: called.\n", __func__); + return snd_pcm_lib_free_pages(alsa_sub); +} + +static int hiface_pcm_prepare(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct snd_pcm_runtime *alsa_rt = alsa_sub->runtime; + int ret; + + pr_debug("%s: called.\n", __func__); + + if (rt->panic) + return -EPIPE; + if (!sub) + return -ENODEV; + + mutex_lock(&rt->stream_mutex); + + sub->dma_off = 0; + sub->period_off = 0; + + if (rt->stream_state == STREAM_DISABLED) { + + ret = hiface_pcm_set_rate(rt, alsa_rt->rate); + if (ret) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + ret = hiface_pcm_stream_start(rt); + if (ret) { + mutex_unlock(&rt->stream_mutex); + return ret; + } + } + mutex_unlock(&rt->stream_mutex); + return 0; +} + +static int hiface_pcm_trigger(struct snd_pcm_substream *alsa_sub, int cmd) +{ + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + unsigned long flags; + + pr_debug("%s: called.\n", __func__); + + if (rt->panic) + return -EPIPE; + if (!sub) + return -ENODEV; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + spin_lock_irqsave(&sub->lock, flags); + sub->active = true; + spin_unlock_irqrestore(&sub->lock, flags); + return 0; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + spin_lock_irqsave(&sub->lock, flags); + sub->active = false; + spin_unlock_irqrestore(&sub->lock, flags); + return 0; + + default: + return -EINVAL; + } +} + +static snd_pcm_uframes_t hiface_pcm_pointer(struct snd_pcm_substream *alsa_sub) +{ + struct pcm_substream *sub = hiface_pcm_get_substream(alsa_sub); + struct pcm_runtime *rt = snd_pcm_substream_chip(alsa_sub); + unsigned long flags; + snd_pcm_uframes_t dma_offset; + + if (rt->panic || !sub) + return SNDRV_PCM_STATE_XRUN; + + spin_lock_irqsave(&sub->lock, flags); + dma_offset = sub->dma_off; + spin_unlock_irqrestore(&sub->lock, flags); + return bytes_to_frames(alsa_sub->runtime, dma_offset); +} + +static struct snd_pcm_ops pcm_ops = { + .open = hiface_pcm_open, + .close = hiface_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = hiface_pcm_hw_params, + .hw_free = hiface_pcm_hw_free, + .prepare = hiface_pcm_prepare, + .trigger = hiface_pcm_trigger, + .pointer = hiface_pcm_pointer, +}; + +static int hiface_pcm_init_urb(struct pcm_urb *urb, + struct hiface_chip *chip, + unsigned int ep, + void (*handler)(struct urb *)) +{ + urb->chip = chip; + usb_init_urb(&urb->instance); + + urb->buffer = kzalloc(PCM_MAX_PACKET_SIZE, GFP_KERNEL); + if (!urb->buffer) + return -ENOMEM; + + usb_fill_bulk_urb(&urb->instance, chip->dev, + usb_sndbulkpipe(chip->dev, ep), (void *)urb->buffer, + PCM_MAX_PACKET_SIZE, handler, urb); + init_usb_anchor(&urb->submitted); + + return 0; +} + +void hiface_pcm_abort(struct hiface_chip *chip) +{ + struct pcm_runtime *rt = chip->pcm; + + if (rt) { + rt->panic = true; + + if (rt->playback.instance) { + snd_pcm_stop(rt->playback.instance, + SNDRV_PCM_STATE_XRUN); + } + mutex_lock(&rt->stream_mutex); + hiface_pcm_stream_stop(rt); + mutex_unlock(&rt->stream_mutex); + } +} + +static void hiface_pcm_destroy(struct hiface_chip *chip) +{ + struct pcm_runtime *rt = chip->pcm; + int i; + + for (i = 0; i < PCM_N_URBS; i++) + kfree(rt->out_urbs[i].buffer); + + kfree(chip->pcm); + chip->pcm = NULL; +} + +static void hiface_pcm_free(struct snd_pcm *pcm) +{ + struct pcm_runtime *rt = pcm->private_data; + + pr_debug("%s: called.\n", __func__); + + if (rt) + hiface_pcm_destroy(rt->chip); +} + +int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq) +{ + int i; + int ret; + struct snd_pcm *pcm; + struct pcm_runtime *rt; + + rt = kzalloc(sizeof(*rt), GFP_KERNEL); + if (!rt) + return -ENOMEM; + + rt->chip = chip; + rt->stream_state = STREAM_DISABLED; + rt->rate = ARRAY_SIZE(rates); + if (extra_freq) + rt->extra_freq = 1; + + init_waitqueue_head(&rt->stream_wait_queue); + mutex_init(&rt->stream_mutex); + spin_lock_init(&rt->playback.lock); + + for (i = 0; i < PCM_N_URBS; i++) + hiface_pcm_init_urb(&rt->out_urbs[i], chip, OUT_EP, + hiface_pcm_out_urb_handler); + + ret = snd_pcm_new(chip->card, "USB-SPDIF Audio", 0, 1, 0, &pcm); + if (ret < 0) { + kfree(rt); + pr_err("Cannot create pcm instance\n"); + return ret; + } + + pcm->private_data = rt; + pcm->private_free = hiface_pcm_free; + + strcpy(pcm->name, "USB-SPDIF Audio"); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &pcm_ops); + + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + MAX_BUFSIZE, MAX_BUFSIZE); + rt->instance = pcm; + + chip->pcm = rt; + return 0; +} diff --git a/sound/usb/hiface/pcm.h b/sound/usb/hiface/pcm.h new file mode 100644 index 0000000..ddc9fe7 --- /dev/null +++ b/sound/usb/hiface/pcm.h @@ -0,0 +1,24 @@ +/* + * Linux driver for M2Tech HiFace compatible devices + * + * Copyright 2012-2013 (C) M2TECH S.r.l and Amarula Solutions B.V. + * + * Authors: Michael Trimarchi <michael(a)amarulasolutions.com> + * Antonio Ospite <ao2(a)amarulasolutions.com> + * + * The driver is based on the work done in TerraTec DMX 6Fire USB + * + * 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. + */ + +#ifndef HIFACE_PCM_H +#define HIFACE_PCM_H + +struct hiface_chip; + +int hiface_pcm_init(struct hiface_chip *chip, u8 extra_freq); +void hiface_pcm_abort(struct hiface_chip *chip); +#endif /* HIFACE_PCM_H */ -- 1.7.10.4

6 22

[alsa-devel] [PATCH 0/6] Convert JZ4740 to dmaengine
by Lars-Peter Clausen 21 Jun '13

21 Jun '13

Hi, This series replaces the custom JZ4740 DMA API with a dmaengine driver. This is done in 3 steps: 1) Add a dmaengine driver which wraps the custom JZ4740 DMA API 2) Update all users of the JZ4740 DMA API to use dmaengine instead 3) Remove the custom API and move all direct hardware access to the dmaengine driver. The first two patches in the series also make sure that the clock of the DMA core is enabled. Since the patches in this series depend on each other I'd prefer if they could all go through the DMA tree. There are a few minor changes to the patch 3 in v2 of this series, all other patches are identical to v1. - Lars Lars-Peter Clausen (4): dma: Add a jz4740 dmaengine driver MIPS: jz4740: Register jz4740 DMA device ASoC: jz4740: Use the generic dmaengine PCM driver MIPS: jz4740: Remove custom DMA API Maarten ter Huurne (2): MIPS: jz4740: Correct clock gate bit for DMA controller MIPS: jz4740: Acquire and enable DMA controller clock arch/mips/include/asm/mach-jz4740/dma.h | 56 --- arch/mips/include/asm/mach-jz4740/platform.h | 1 + arch/mips/jz4740/Makefile | 2 +- arch/mips/jz4740/board-qi_lb60.c | 1 + arch/mips/jz4740/clock.c | 2 +- arch/mips/jz4740/dma.c | 287 ------------- arch/mips/jz4740/platform.c | 21 + drivers/dma/Kconfig | 6 + drivers/dma/Makefile | 1 + drivers/dma/dma-jz4740.c | 617 +++++++++++++++++++++++++++ sound/soc/jz4740/Kconfig | 1 + sound/soc/jz4740/jz4740-i2s.c | 48 +-- sound/soc/jz4740/jz4740-pcm.c | 310 +------------- sound/soc/jz4740/jz4740-pcm.h | 20 - 14 files changed, 676 insertions(+), 697 deletions(-) delete mode 100644 arch/mips/jz4740/dma.c create mode 100644 drivers/dma/dma-jz4740.c delete mode 100644 sound/soc/jz4740/jz4740-pcm.h -- 1.8.2.1

3 17

[alsa-devel] [stable] usb-audio validation fixes [3.0, 3.2]
by Ben Hutchings 17 Jun '13

17 Jun '13

It looks these fixes are suitable for inclusion in the 3.0.y and 3.2.y stable branches: commit 4fa0e81b83503900be277e6273a79651b375e288 Author: Xi Wang <xi.wang(a)gmail.com> Date: Sun Jan 8 09:02:52 2012 -0500 ALSA: usb-audio: fix possible hang and overflow in parse_uac2_sample_rate_range() commit 8866f405efd4171f9d9c91901d2dd02f01bacb60 Author: Xi Wang <xi.wang(a)gmail.com> Date: Tue Feb 14 05:18:48 2012 -0500 ALSA: usb-audio: avoid integer overflow in create_fixed_stream_quirk() Do you agree? Ben. -- Ben Hutchings If at first you don't succeed, you're doing about average.

2 3

[alsa-devel] Timestamps using I/O Plugin
by Johan Hellman 17 Jun '13

17 Jun '13

I was excited when I found that it was possible to write user space drivers for ALSA, but I can't get my head around how timestamping works when using a I/O plugin instance. I better start explaining what we want to do. Basically we just want to transfer raw audio between the HOST running ALSA and one or more small(tiny) network connected MCUs with audio codecs. We need to synchronize the different network sources with the local HW sources (on the ALSA HOST). Starting looking into how timestamps is handled by ALSA I could not find how it was possible to set the timestamp, there seems to be some differences compared to using a normal kernel driver!? If I understand it correctly timestamp can be obtained by calling snd_pcm_status() but the ioplugs implementation of the .status() newer fills in the ->tstamp only the ->trigger_tstamp, so how can I set the timestamp of the data my plugin delivers to ALSA? Additionally snd_pcm_htimestamp() only seams to return the current monotonic clock at the point the function was called for an ioplug instance!? I saw that there are plugins for pulseaudio and jack doing what we whant to achieve but because of the footprint of the MCU we can't use PA/Jack on these MCU:s. But how have they solved this issue or isn't it an issue for their usecase? /JohanHellman

2 5

[alsa-devel] [PATCH] ASoC: SPEAr spdif_{in, out}: use devm for clk and a few more cleanups
by Uwe Kleine-König 13 Jun '13

13 Jun '13

Drop dev_set_drvdata as this is handled in the core and use devm_request_and_ioremap instead of devm_ioremap to properly register the address range used Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/spear/spdif_in.c | 17 ++--------------- sound/soc/spear/spdif_out.c | 17 +++-------------- 2 files changed, 5 insertions(+), 29 deletions(-) diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c index c7c4b20..d40bb34 100644 --- a/sound/soc/spear/spdif_in.c +++ b/sound/soc/spear/spdif_in.c @@ -231,7 +231,7 @@ static int spdif_in_probe(struct platform_device *pdev) if (host->irq < 0) return -EINVAL; - host->clk = clk_get(&pdev->dev, NULL); + host->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(host->clk)) return PTR_ERR(host->clk); @@ -253,33 +253,20 @@ static int spdif_in_probe(struct platform_device *pdev) ret = devm_request_irq(&pdev->dev, host->irq, spdif_in_irq, 0, "spdif-in", host); if (ret) { - clk_put(host->clk); dev_warn(&pdev->dev, "request_irq failed\n"); return ret; } - ret = snd_soc_register_dai(&pdev->dev, &spdif_in_dai); - if (ret != 0) { - clk_put(host->clk); - return ret; - } - - return 0; + return snd_soc_register_dai(&pdev->dev, &spdif_in_dai); } static int spdif_in_remove(struct platform_device *pdev) { - struct spdif_in_dev *host = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); - - clk_put(host->clk); return 0; } - static struct platform_driver spdif_in_driver = { .probe = spdif_in_probe, .remove = spdif_in_remove, diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c index 5eac4cd..b2488b4 100644 --- a/sound/soc/spear/spdif_out.c +++ b/sound/soc/spear/spdif_out.c @@ -293,14 +293,14 @@ static int spdif_out_probe(struct platform_device *pdev) return -ENOMEM; } - host->io_base = devm_ioremap(&pdev->dev, res->start, + host->io_base = devm_request_and_ioremap(&pdev->dev, res->start, resource_size(res)); if (!host->io_base) { dev_warn(&pdev->dev, "ioremap failed\n"); return -ENOMEM; } - host->clk = clk_get(&pdev->dev, NULL); + host->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(host->clk)) return PTR_ERR(host->clk); @@ -314,23 +314,12 @@ static int spdif_out_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, host); - ret = snd_soc_register_dai(&pdev->dev, &spdif_out_dai); - if (ret != 0) { - clk_put(host->clk); - return ret; - } - - return 0; + return snd_soc_register_dai(&pdev->dev, &spdif_out_dai); } static int spdif_out_remove(struct platform_device *pdev) { - struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev); - snd_soc_unregister_dai(&pdev->dev); - dev_set_drvdata(&pdev->dev, NULL); - - clk_put(host->clk); return 0; } -- 1.8.2.rc2

3 4

[alsa-devel] firewire-kernel-streaming branch into main kernel tree?
by Rolf Anderegg 10 Jun '13

10 Jun '13

Hello Clemens, it's been a long time since I last inquired about the snd_dice module provided by your firewire-kernel-streaming branch (http://git.alsa-project.org/?p=alsa-kprivate.git;a=shortlog;h=refs/heads/fi…), which is a good sign I guess. In the meantime we have been making heavy use of the snd_dice module for over a year with all of our Weiss DICE based products (obviously playback only), and it has proven to be extremely stable (currently 3.4.18-rt29, patched with firewire-kernel-streaming). Therefore I have been meaning to ask you for some time now whether you have any interest in merging this firewire-kernel-streaming branch into the main kernel tree (at least as an experimental option, mind you), or are there any major objections? Given that you don't have unlimited time at hand to further develop the snd_dice module, let alone provide capture support in the near future, AND the fact that DICE is likely to see its EOL soon (and replaced by a sophisticated successor by TCAT), I realize that this is probably not a high priority. However from our perspective it would be sad to see these ALSA Firewire proceedings be lost and not be appreciated by a larger audience. With respect to this: * what would be the customary approach to requesting such a merge? * what are your reservations, resp. are there any existing shortcomings that you are yet unhappy with? * have you received any other (negative/positive) feedback from snd_dice users? I know of the Vortexbox community, which has been offering an optional patch to support Firewire plackback devices; as far as I know, they're happy with it. * if you need any backing or assistance, we'd be happy to help out Keen to hear your feedback on this suggestion. Best regards, Rolf Anderegg (Weiss Engineering Ltd.)

2 2

[alsa-devel] Sound issues with ALC269VB-based hardware
by Anisse Astier 06 Jun '13

06 Jun '13

Hi, So I have this hardware that has a few issues (all tested with tiwai/sound for-next) Audio out/speakers: - Audio does not work by default. It needs plugging a jack in either headphone or mic. - disabling "Enable Automute" does make it work. But then it doesn't mute when jack is plugged (obviously). Mic: - needs model=laptop-dmic so that working "Internal Mic 1" appears - "Internal Mic" is useless. - no jack sense, need to select manually "Internal Mic 1" or "Mic" (external) as Input Source Please find alsa-info.sh with and without model=laptop-dmic in attachement. Regards, Anisse

3 17

[alsa-devel] [PATCH V2] ASoC: sgtl5000: Fix VAG_POWER enabling/disabling order
by Marek Vasut 04 Jun '13

04 Jun '13

The VAG_POWER must be enabled after all other bits in CHIP_ANA_POWER and disabled before any other bit in CHIP_ANA_POWER. See the SGTL5000 datasheet (Table 31, BIT 7, page 42-43). Failing to follow this order will result in ugly loud "POP" noise at the end of playback. To achieve such order, use the _PRE and _POST DAPM widgets to trigger the power_vag_event, where the event type check has to be fixed accordingly as well. Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Dong Aisheng <dong.aisheng(a)linaro.org> Cc: Eric Nelson <eric.nelson(a)boundarydevices.com> Cc: Fabio Estevam <fabio.estevam(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/sgtl5000.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) V2: Remove that mdelay/msleep regression diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 92bbfec..1994f01 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -151,12 +151,12 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { switch (event) { - case SND_SOC_DAPM_PRE_PMU: + case SND_SOC_DAPM_POST_PMU: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); break; - case SND_SOC_DAPM_POST_PMD: + case SND_SOC_DAPM_PRE_PMD: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, 0); msleep(400); @@ -217,12 +217,11 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { 0, SGTL5000_CHIP_DIG_POWER, 1, 0), - SND_SOC_DAPM_SUPPLY("VAG_POWER", SGTL5000_CHIP_ANA_POWER, 7, 0, - power_vag_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), + + SND_SOC_DAPM_PRE("VAG_POWER_PRE", power_vag_event), + SND_SOC_DAPM_POST("VAG_POWER_POST", power_vag_event), }; /* routes for sgtl5000 */ @@ -230,16 +229,13 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ - {"ADC", NULL, "VAG_POWER"}, {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ - {"DAC", NULL, "VAG_POWER"}, {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ - {"LINE_IN", NULL, "VAG_POWER"}, {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */ {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */ -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: omap-hdmi-codec: make the driver common for other SoCs
by Barry Song 04 Jun '13

04 Jun '13

to support HDMI on CSR SiRFprimaII and atlasVI, we need one more HDMI pseudo codec, rather than add a new driver, we can make omap HDMI codec common for other SoCs as well. then the omap-hdmi codec becomes a generic HDMI pseudo- codec as HDMI audio features depend on HDMI specification not on SoCs. Cc: Ricardo Neri <ricardo.neri(a)ti.com> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- sound/soc/codecs/Kconfig | 4 ++-- sound/soc/codecs/Makefile | 4 ++-- sound/soc/codecs/{omap-hdmi.c => hdmi.c} | 26 +++++++++++++------------- sound/soc/omap/omap-hdmi-card.c | 2 +- 4 files changed, 18 insertions(+), 18 deletions(-) rename sound/soc/codecs/{omap-hdmi.c => hdmi.c} (69%) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00..d8c4f3d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -53,7 +53,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C - select SND_SOC_OMAP_HDMI_CODEC if OMAP4_DSS_HDMI + select SND_SOC_HDMI_CODEC select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_SGTL5000 if I2C @@ -287,7 +287,7 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate -config SND_SOC_OMAP_HDMI_CODEC +config SND_SOC_HDMI_CODEC tristate config SND_SOC_PCM3008 diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b9e41c9..49ff127 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -41,7 +41,7 @@ snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o -snd-soc-omap-hdmi-codec-objs := omap-hdmi.o +snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -168,7 +168,7 @@ obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o -obj-$(CONFIG_SND_SOC_OMAP_HDMI_CODEC) += snd-soc-omap-hdmi-codec.o +obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/omap-hdmi.c b/sound/soc/codecs/hdmi.c similarity index 69% rename from sound/soc/codecs/omap-hdmi.c rename to sound/soc/codecs/hdmi.c index 529d064..2bcae2b 100644 --- a/sound/soc/codecs/omap-hdmi.c +++ b/sound/soc/codecs/hdmi.c @@ -1,5 +1,5 @@ /* - * ALSA SoC codec driver for HDMI audio on OMAP processors. + * ALSA SoC codec driver for HDMI audio codecs. * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/ * Author: Ricardo Neri <ricardo.neri(a)ti.com> * @@ -23,10 +23,10 @@ #define DRV_NAME "hdmi-audio-codec" -static struct snd_soc_codec_driver omap_hdmi_codec; +static struct snd_soc_codec_driver hdmi_codec; -static struct snd_soc_dai_driver omap_hdmi_codec_dai = { - .name = "omap-hdmi-hifi", +static struct snd_soc_dai_driver hdmi_codec_dai = { + .name = "hdmi-hifi", .playback = { .channels_min = 2, .channels_max = 8, @@ -39,31 +39,31 @@ static struct snd_soc_dai_driver omap_hdmi_codec_dai = { }, }; -static int omap_hdmi_codec_probe(struct platform_device *pdev) +static int hdmi_codec_probe(struct platform_device *pdev) { - return snd_soc_register_codec(&pdev->dev, &omap_hdmi_codec, - &omap_hdmi_codec_dai, 1); + return snd_soc_register_codec(&pdev->dev, &hdmi_codec, + &hdmi_codec_dai, 1); } -static int omap_hdmi_codec_remove(struct platform_device *pdev) +static int hdmi_codec_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; } -static struct platform_driver omap_hdmi_codec_driver = { +static struct platform_driver hdmi_codec_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, }, - .probe = omap_hdmi_codec_probe, - .remove = omap_hdmi_codec_remove, + .probe = hdmi_codec_probe, + .remove = hdmi_codec_remove, }; -module_platform_driver(omap_hdmi_codec_driver); +module_platform_driver(hdmi_codec_driver); MODULE_AUTHOR("Ricardo Neri <ricardo.neri(a)ti.com>"); -MODULE_DESCRIPTION("ASoC OMAP HDMI codec driver"); +MODULE_DESCRIPTION("ASoC generic HDMI codec driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/omap/omap-hdmi-card.c b/sound/soc/omap/omap-hdmi-card.c index d4eaa92..7e66e9c 100644 --- a/sound/soc/omap/omap-hdmi-card.c +++ b/sound/soc/omap/omap-hdmi-card.c @@ -35,7 +35,7 @@ static struct snd_soc_dai_link omap_hdmi_dai = { .cpu_dai_name = "omap-hdmi-audio-dai", .platform_name = "omap-pcm-audio", .codec_name = "hdmi-audio-codec", - .codec_dai_name = "omap-hdmi-hifi", + .codec_dai_name = "hdmi-hifi", }; static struct snd_soc_card snd_soc_omap_hdmi = { -- 1.8.2.3

2 1

[alsa-devel] dsnoop memory leak
by Иван Больсунов 04 Jun '13

04 Jun '13

Since alsa bug tracker does not work for me for weeks, I have to report bug here. Sorry if it a wrong place. So, to reproduce memory leak (Alsa lib used v.1.0.27), you have to use next asound.conf: pcm_slave.myslave { pcm "hw:0,0" } pcm.mysnoop { type dsnoop ipc_key 1024 slave myslave } Then you simply do: snd_pcm_open(&pcm_handle, "mysnoop", SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK); snd_pcm_close(pcm_handle); snd_config_update_free_global(); Valgrind will show something like this: ==7889== ==7889== HEAP SUMMARY: ==7889== in use at exit: 15,854 bytes in 871 blocks ==7889== total heap usage: 333,212 allocs, 332,341 frees, 11,695,490 bytes allocated ==7889== ==7889== 15,834 (6,264 direct, 9,570 indirect) bytes in 174 blocks are definitely lost in loss record 6 of 6 ==7889== at 0x4027CBF: calloc (vg_replace_malloc.c:566) ==7889== by 0x406B416: _snd_config_make (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406C69C: snd_config_make (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406C861: snd_config_make_compound (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406D00C: _snd_config_copy (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406F112: snd_config_walk (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406FE04: snd_config_copy (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x4070370: snd_config_expand (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x407107E: snd_config_search_definition (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x40D2C22: _snd_pcm_direct_get_slave_ipc_offset (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x40D68A2: snd1_pcm_direct_parse_open_conf (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x40D28CA: _snd_pcm_dsnoop_open (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x40909C4: snd_pcm_open_conf (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x4091036: snd_pcm_open_noupdate (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x804873D: main (in /home/blah/al/a.out) ==7889== ==7889== LEAK SUMMARY: ==7889== definitely lost: 6,264 bytes in 174 blocks ==7889== indirectly lost: 9,570 bytes in 696 blocks ==7889== possibly lost: 0 bytes in 0 blocks ==7889== still reachable: 20 bytes in 1 blocks ==7889== suppressed: 0 bytes in 0 blocks ==7889== Reachable blocks (those to which a pointer was found) are not shown. ==7889== To see them, rerun with: --leak-check=full --show-reachable=yes ==7889== ==7889== For counts of detected and suppressed errors, rerun with: -v ==7889== ERROR SUMMARY: 1 errors from 1 contexts (suppressed: 0 from 0)

2 1

[alsa-devel] regression: from 3.8 to 3.9: headphones output no sound on Intel HDA, codec VIA VT1802
by Alex Riesen 04 Jun '13

04 Jun '13

Hi, I just noticed (use the headphones rarely) that the headphones on this System76 Lemur Ultra (lemu4) stopped working. There is absolutely no output. I tried reverting all changes to patch_via.c since v3.8, and have the sound back, but there wasn't much to revert: Revert "ALSA: hda - Rearrange INPUT_PIN_ATTR_*" Revert "ALSA: hda - Use generic parser for VIA codec driver" Revert "ALSA: hda - Enable parsing the independent HP mode as default for VIA codecs" Revert "ALSA: hda/via - Fix wrong checks of power state bits" Revert "ALSA: hda - Fix phantom jacks on VT1708" Revert "ALSA: hda - detect jacks on VT1708 even when no streams are active" Revert "ALSA: hda - Use generic array for loopback list management" and one change (which I indeed suspect) is large: the use of generic parser. The problem is also present in 3.10-rc1. The output of alsa-info.sh attached, for 3.8.12 and 3.9.2 respectively. Regards, Alex Riesen

2 30

[alsa-devel] [PATCH 1/1] ASoC: davinci-mcasp: Remove redundant use of of_match_ptr macro
by Sachin Kamat 03 Jun '13

03 Jun '13

'mcasp_dt_ids' is always compiled in. Hence of_match_ptr is not necessary. Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> --- sound/soc/davinci/davinci-mcasp.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 56ecfc7..94f1e46 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1023,7 +1023,7 @@ static struct snd_platform_data *davinci_mcasp_set_pdata_from_of( struct device_node *np = pdev->dev.of_node; struct snd_platform_data *pdata = NULL; const struct of_device_id *match = - of_match_device(of_match_ptr(mcasp_dt_ids), &pdev->dev); + of_match_device(mcasp_dt_ids, &pdev->dev); const u32 *of_serial_dir32; u8 *of_serial_dir; @@ -1256,7 +1256,7 @@ static struct platform_driver davinci_mcasp_driver = { .driver = { .name = "davinci-mcasp", .owner = THIS_MODULE, - .of_match_table = of_match_ptr(mcasp_dt_ids), + .of_match_table = mcasp_dt_ids, }, }; -- 1.7.9.5

2 1

[alsa-devel] No headphone control for ALC892 on Zotac Motherboard D2550ITXS-B-E
by Halim Sahin 03 Jun '13

03 Jun '13

Hi, Attached is output of alsa-info. Updated to Alsa-driver 1.0.25 running with 2.6.34 kernel. Frontpanel of my case has mic and headphone output connectors. front mic works but no headphones. Card works except headphones Can you please help? -- Halim Sahin E-Mail: halim.sahin (at) t-online.de

3 5

[alsa-devel] aplay lists both Playback and Capture devices
by Fabio Estevam 02 Jun '13

02 Jun '13

Hi, When running 'aplay -l' and 'arecord -l' I see the following output on a mx28evk running a 3.10-rc kernel: /$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: mxssgtl5000 [mxs_sgtl5000], device 0: Playback sgtl5000-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: mxssgtl5000 [mxs_sgtl5000], device 1: Capture sgtl5000-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 I would expect to see only the Playback entry in aplay and only the Record entry in arecord. Is there anything I should change in sound/soc/mxs/mxs-sgtl5000.c in order to get the correct outputs? Regards, Fabio Estevam

4 11

[alsa-devel] [PATCH 1/2] dma: Add a PAUSE_RESUME capability
by Lars-Peter Clausen 01 Jun '13

01 Jun '13

Add a capability which allows dmaengine consumers to query whether a dmaengine controller supports the DMA_PAUSE and DMA_RESUME commands. Sometimes a user of the dmaengine API needs to know in advance if a channel will support pause/resume. E.g. in ALSA the information whether it is possible to pause/resume a audio stream or whether it is necessary to abort and restart the stream is communicated to userspace so it can act accordingly. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/linux/dmaengine.h | 1 + 1 file changed, 1 insertion(+) diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h index 96d3e4a..ad98dfb 100644 --- a/include/linux/dmaengine.h +++ b/include/linux/dmaengine.h @@ -74,6 +74,7 @@ enum dma_transaction_type { DMA_SLAVE, DMA_CYCLIC, DMA_INTERLEAVE, + DMA_PAUSE_RESUME, /* last transaction type for creation of the capabilities mask */ DMA_TX_TYPE_END, }; -- 1.8.0

3 3

[alsa-devel] [PATCH 1/2] ASoC: wm8994: fix the codec state machine
by Vinod Koul 01 Jun '13

01 Jun '13

few places the mic_detecting is not updated correctly when state changes, causing jack detection to go wrong Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/codecs/wm8994.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 49b2b00..43f82d2 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3547,6 +3547,7 @@ static void wm8958_mic_id(void *data, u16 status) snd_soc_jack_report(wm8994->micdet[0].jack, 0, wm8994->btn_mask | SND_JACK_HEADSET); + wm8994->mic_detecting = true; return; } @@ -3672,7 +3673,7 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) snd_soc_update_bits(codec, WM1811_JACKDET_CTRL, WM1811_JACKDET_DB, WM1811_JACKDET_DB); - wm8994->mic_detecting = false; + wm8994->mic_detecting = true; wm8994->jack_mic = false; snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, WM8958_MICD_ENA, 0); @@ -3696,6 +3697,7 @@ static irqreturn_t wm1811_jackdet_irq(int irq, void *data) /* Since we only report deltas force an update, ensures we * avoid bootstrapping issues with the core. */ snd_soc_jack_report(wm8994->micdet[0].jack, 0, 0); + wm8994->mic_detecting = true; pm_runtime_put(codec->dev); return IRQ_HANDLED; -- 1.7.0.4

2 3

[alsa-devel] [PATCH] ASoC: sgtl5000: Do not read registers prior to turning on the supplies
by Fabio Estevam 31 May '13

31 May '13

Commit b871f1ad (ASoC: sgtl5000: Read SGTL5000_CHIP_ID in i2c_probe()) placed the code for reading the codec revision prior to turning on the power supplies. Even though this works on some systems that always have the codec power supplies enabled, this is not correct, so revert this commit. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 39 +++++++++++++++------------------------ 1 file changed, 15 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 1c3b20f..788af6c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1275,7 +1275,7 @@ static int sgtl5000_replace_vddd_with_ldo(struct snd_soc_codec *codec) static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) { - int reg; + u16 reg; int ret; int rev; int i; @@ -1303,17 +1303,23 @@ static int sgtl5000_enable_regulators(struct snd_soc_codec *codec) /* wait for all power rails bring up */ udelay(10); - /* - * workaround for revision 0x11 and later, - * roll back to use internal LDO - */ - - ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); - if (ret) + /* read chip information */ + reg = snd_soc_read(codec, SGTL5000_CHIP_ID); + if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != + SGTL5000_PARTID_PART_ID) { + dev_err(codec->dev, + "Device with ID register %x is not a sgtl5000\n", reg); + ret = -ENODEV; goto err_regulator_disable; + } rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; + dev_info(codec->dev, "sgtl5000 revision 0x%x\n", rev); + /* + * workaround for revision 0x11 and later, + * roll back to use internal LDO + */ if (external_vddd && rev >= 0x11) { /* disable all regulator first */ regulator_bulk_disable(ARRAY_SIZE(sgtl5000->supplies), @@ -1497,7 +1503,7 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct sgtl5000_priv *sgtl5000; - int ret, reg, rev; + int ret; sgtl5000 = devm_kzalloc(&client->dev, sizeof(struct sgtl5000_priv), GFP_KERNEL); @@ -1511,21 +1517,6 @@ static int sgtl5000_i2c_probe(struct i2c_client *client, return ret; } - /* read chip information */ - ret = regmap_read(sgtl5000->regmap, SGTL5000_CHIP_ID, &reg); - if (ret) - return ret; - - if (((reg & SGTL5000_PARTID_MASK) >> SGTL5000_PARTID_SHIFT) != - SGTL5000_PARTID_PART_ID) { - dev_err(&client->dev, - "Device with ID register %x is not a sgtl5000\n", reg); - return -ENODEV; - } - - rev = (reg & SGTL5000_REVID_MASK) >> SGTL5000_REVID_SHIFT; - dev_info(&client->dev, "sgtl5000 revision 0x%x\n", rev); - i2c_set_clientdata(client, sgtl5000); /* Ensure sgtl5000 will start with sane register values */ -- 1.8.1.2

5 8

[alsa-devel] Improving or replacing snd_printk()
by Alan Stern 31 May '13

31 May '13

Takashi: Messages produced by snd_printk() aren't as informative as they could be -- especially when CONFIG_SND_VERBOSE_PRINTK isn't enabled, which is the case on most systems. The routine uses printk() rather then dev_dbg(), dev_info(), or dev_err(). As a result, the messages don't say what driver or device is involved. We just get something like: [15919.883604] cannot submit urb (err = -18) That's not much help to anybody. If you didn't know what an urb was, you wouldn't even realize that this referred to a USB device. And if you had multiple USB audio devices, you wouldn't know which one it referred to. Has there been any thought of improving snd_printk(), or even better, changing the call sites to use the dev_*() routines instead? Alan Stern

4 17

[alsa-devel] Fwd: regression: from 3.8 to 3.9: headphones output no sound on Intel HDA, codec VIA VT1802
by Evan 31 May '13

31 May '13

I am having this same issue (with the same hardware). I found a temporary work around for the problem by disabling "Auto-Mute Mode" with alsamixer and then muting my "Speaker" output. If I don't mute the "Speaker" I get audio through both my headphones and my speakers, but when I mute the speakers I get the audio just through the headphones. The output of alsa-info for 3.9.4 is here[1]. [1] http://www.alsa-project.org/db/?f=6120303f74b8e46930e15a544ba11b859de25c04 -E

2 1

[alsa-devel] [PATCH 1/3] ALSA: hdspm - Add support for 128-192kHz WordClock input
by Adrian Knoth 31 May '13

31 May '13

Allow WordClock input rates of 128, 176.4 and 192kHz. Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index ef3cbc0..e070ea8 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -400,8 +400,8 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_wc_freq0 (1<<5) /* input freq detected via autosync */ #define HDSPM_wc_freq1 (1<<6) /* 001=32, 010==44.1, 011=48, */ -#define HDSPM_wc_freq2 (1<<7) /* 100=64, 101=88.2, 110=96, */ -/* missing Bit for 111=128, 1000=176.4, 1001=192 */ +#define HDSPM_wc_freq2 (1<<7) /* 100=64, 101=88.2, 110=96, 111=128 */ +#define HDSPM_wc_freq3 0x800 /* 1000=176.4, 1001=192 */ #define HDSPM_SyncRef0 0x10000 /* Sync Reference */ #define HDSPM_SyncRef1 0x20000 @@ -412,13 +412,17 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); #define HDSPM_wc_valid (HDSPM_wcLock|HDSPM_wcSync) -#define HDSPM_wcFreqMask (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2) +#define HDSPM_wcFreqMask (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2|\ + HDSPM_wc_freq3) #define HDSPM_wcFreq32 (HDSPM_wc_freq0) #define HDSPM_wcFreq44_1 (HDSPM_wc_freq1) #define HDSPM_wcFreq48 (HDSPM_wc_freq0|HDSPM_wc_freq1) #define HDSPM_wcFreq64 (HDSPM_wc_freq2) #define HDSPM_wcFreq88_2 (HDSPM_wc_freq0|HDSPM_wc_freq2) #define HDSPM_wcFreq96 (HDSPM_wc_freq1|HDSPM_wc_freq2) +#define HDSPM_wcFreq128 (HDSPM_wc_freq0|HDSPM_wc_freq1|HDSPM_wc_freq2) +#define HDSPM_wcFreq176_4 (HDSPM_wc_freq3) +#define HDSPM_wcFreq192 (HDSPM_wc_freq0|HDSPM_wc_freq3) #define HDSPM_status1_F_0 0x0400000 #define HDSPM_status1_F_1 0x0800000 @@ -1181,6 +1185,15 @@ static int hdspm_external_sample_rate(struct hdspm *hdspm) case HDSPM_wcFreq96: rate = 96000; break; + case HDSPM_wcFreq128: + rate = 128000; + break; + case HDSPM_wcFreq176_4: + rate = 176400; + break; + case HDSPM_wcFreq192: + rate = 192000; + break; default: rate = 0; break; -- 1.7.10.4

2 3

[alsa-devel] [PATCH] ALSA: hda - Add headset quirk for two Dell machines
by David Henningsson 31 May '13

31 May '13

This quirk is required for the headset mic to work on these two machines. BugLink: https://bugs.launchpad.net/bugs/1186170 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 59d2e91..aa4bf82 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3493,6 +3493,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05f4, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f5, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05f6, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x05f8, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), SND_PCI_QUIRK(0x103c, 0x18e6, "HP", ALC269_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), -- 1.7.9.5

2 1

[alsa-devel] [PATCH] [resend] snd-es1968: Add radio support for MediaForte M56VAP
by Ondrej Zary 31 May '13

31 May '13

Add support for TEA5757 tuner on MediaForte M56VAP sound+modem+radio card. The GPIO connection type is automatically detected (like snd-fm801 driver). Also add a safety subsystem vendor check to skip radio detection if vendor differs from ESS (so we don't touch GPIOs on laptops). Tested with SF64-PCE2 and M56VAP cards. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- sound/pci/es1968.c | 73 +++++++++++++++++++++++++++++++++++---------------- 1 files changed, 50 insertions(+), 23 deletions(-) diff --git a/sound/pci/es1968.c b/sound/pci/es1968.c index a1f32b5..db08641 100644 --- a/sound/pci/es1968.c +++ b/sound/pci/es1968.c @@ -564,6 +564,7 @@ struct es1968 { #ifdef CONFIG_SND_ES1968_RADIO struct v4l2_device v4l2_dev; struct snd_tea575x tea; + unsigned int tea575x_tuner; #endif }; @@ -2557,37 +2558,47 @@ static int snd_es1968_input_register(struct es1968 *chip) bits 1=unmask write to given bit */ #define IO_DIR 8 /* direction register offset from GPIO_DATA bits 0/1=read/write direction */ -/* mask bits for GPIO lines */ -#define STR_DATA 0x0040 /* GPIO6 */ -#define STR_CLK 0x0080 /* GPIO7 */ -#define STR_WREN 0x0100 /* GPIO8 */ -#define STR_MOST 0x0200 /* GPIO9 */ + +/* GPIO to TEA575x maps */ +struct snd_es1968_tea575x_gpio { + u8 data, clk, wren, most; + char *name; +}; + +static struct snd_es1968_tea575x_gpio snd_es1968_tea575x_gpios[] = { + { .data = 6, .clk = 7, .wren = 8, .most = 9, .name = "SF64-PCE2" }, + { .data = 7, .clk = 8, .wren = 6, .most = 10, .name = "M56VAP" }, +}; + +#define get_tea575x_gpio(chip) \ + (&snd_es1968_tea575x_gpios[(chip)->tea575x_tuner]) + static void snd_es1968_tea575x_set_pins(struct snd_tea575x *tea, u8 pins) { struct es1968 *chip = tea->private_data; - unsigned long io = chip->io_port + GPIO_DATA; + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); u16 val = 0; - val |= (pins & TEA575X_DATA) ? STR_DATA : 0; - val |= (pins & TEA575X_CLK) ? STR_CLK : 0; - val |= (pins & TEA575X_WREN) ? STR_WREN : 0; + val |= (pins & TEA575X_DATA) ? (1 << gpio.data) : 0; + val |= (pins & TEA575X_CLK) ? (1 << gpio.clk) : 0; + val |= (pins & TEA575X_WREN) ? (1 << gpio.wren) : 0; - outw(val, io); + outw(val, chip->io_port + GPIO_DATA); } static u8 snd_es1968_tea575x_get_pins(struct snd_tea575x *tea) { struct es1968 *chip = tea->private_data; - unsigned long io = chip->io_port + GPIO_DATA; - u16 val = inw(io); - u8 ret; + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); + u16 val = inw(chip->io_port + GPIO_DATA); + u8 ret = 0; - ret = 0; - if (val & STR_DATA) + if (val & (1 << gpio.data)) ret |= TEA575X_DATA; - if (val & STR_MOST) + if (val & (1 << gpio.most)) ret |= TEA575X_MOST; + return ret; } @@ -2596,13 +2607,18 @@ static void snd_es1968_tea575x_set_direction(struct snd_tea575x *tea, bool outpu struct es1968 *chip = tea->private_data; unsigned long io = chip->io_port + GPIO_DATA; u16 odir = inw(io + IO_DIR); + struct snd_es1968_tea575x_gpio gpio = *get_tea575x_gpio(chip); if (output) { - outw(~(STR_DATA | STR_CLK | STR_WREN), io + IO_MASK); - outw(odir | STR_DATA | STR_CLK | STR_WREN, io + IO_DIR); + outw(~((1 << gpio.data) | (1 << gpio.clk) | (1 << gpio.wren)), + io + IO_MASK); + outw(odir | (1 << gpio.data) | (1 << gpio.clk) | (1 << gpio.wren), + io + IO_DIR); } else { - outw(~(STR_CLK | STR_WREN | STR_DATA | STR_MOST), io + IO_MASK); - outw((odir & ~(STR_DATA | STR_MOST)) | STR_CLK | STR_WREN, io + IO_DIR); + outw(~((1 << gpio.clk) | (1 << gpio.wren) | (1 << gpio.data) | (1 << gpio.most)), + io + IO_MASK); + outw((odir & ~((1 << gpio.data) | (1 << gpio.most))) + | (1 << gpio.clk) | (1 << gpio.wren), io + IO_DIR); } } @@ -2772,6 +2788,9 @@ static int snd_es1968_create(struct snd_card *card, snd_card_set_dev(card, &pci->dev); #ifdef CONFIG_SND_ES1968_RADIO + /* don't play with GPIOs on laptops */ + if (chip->pci->subsystem_vendor != 0x125d) + goto no_radio; err = v4l2_device_register(&pci->dev, &chip->v4l2_dev); if (err < 0) { snd_es1968_free(chip); @@ -2781,10 +2800,18 @@ static int snd_es1968_create(struct snd_card *card, chip->tea.private_data = chip; chip->tea.radio_nr = radio_nr; chip->tea.ops = &snd_es1968_tea_ops; - strlcpy(chip->tea.card, "SF64-PCE2", sizeof(chip->tea.card)); sprintf(chip->tea.bus_info, "PCI:%s", pci_name(pci)); - if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) - printk(KERN_INFO "es1968: detected TEA575x radio\n"); + for (i = 0; i < ARRAY_SIZE(snd_es1968_tea575x_gpios); i++) { + chip->tea575x_tuner = i; + if (!snd_tea575x_init(&chip->tea, THIS_MODULE)) { + snd_printk(KERN_INFO "es1968: detected TEA575x radio type %s\n", + get_tea575x_gpio(chip)->name); + strlcpy(chip->tea.card, get_tea575x_gpio(chip)->name, + sizeof(chip->tea.card)); + break; + } + } +no_radio: #endif *chip_ret = chip; -- Ondrej Zary

2 1

[alsa-devel] [PATCH 1/1] ALSA: hda - add dock support for Thinkpad T431s
by Ebben Aries 31 May '13

31 May '13

Add a model/fixup string "lenovo-dock", for Thinkpad T431s, to allow sound in docking station. Tested on Lenovo T431s with ThinkPad Mini Dock Plus Series 3 Signed-off-by: Ebben Aries <earies(a)dscp.org> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 59d2e91..9658faf 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3530,6 +3530,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21f3, "Thinkpad T430", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2208, "Thinkpad T431s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), -- 1.7.12.1

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[alsa-devel] [PATCH] ALSA: sis7019: fix error return code in sis_chip_create()
by Wei Yongjun 31 May '13

31 May '13

From: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> Fix to return a negative error code in the pci_set_dma_mask() error handling case instead of 0, as done elsewhere in this function. Signed-off-by: Wei Yongjun <yongjun_wei(a)trendmicro.com.cn> --- sound/pci/sis7019.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index f2639a0..e413b4e 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -1341,7 +1341,8 @@ static int sis_chip_create(struct snd_card *card, if (rc) goto error_out; - if (pci_set_dma_mask(pci, DMA_BIT_MASK(30)) < 0) { + rc = pci_set_dma_mask(pci, DMA_BIT_MASK(30)); + if (rc < 0) { dev_err(&pci->dev, "architecture does not support 30-bit PCI busmaster DMA"); goto error_out_enabled; }

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Re: [alsa-devel] [PATCH] ALSA: hda - Enable runtime pm for Haswell
by Wang, Xingchao 31 May '13

31 May '13

Hi Rafael, > -----Original Message----- > From: Wysocki, Rafael J > Sent: Wednesday, May 22, 2013 7:09 PM > To: Takashi Iwai > Cc: Wang, Xingchao; Girdwood, Liam R; Lin, Mengdong; Li, Jocelyn; > alsa-devel(a)alsa-project.org; Wang Xingchao > Subject: Re: [PATCH] ALSA: hda - Enable runtime pm for Haswell > > On 5/22/2013 7:51 AM, Takashi Iwai wrote: > > At Wed, 22 May 2013 03:56:00 +0000, > > Wang, Xingchao wrote: > >> Hi, > >> > >> Add Rafael in loop. > >> > >>> -----Original Message----- > >>> From: Takashi Iwai [mailto:tiwai@suse.de] > >>> Sent: Thursday, May 16, 2013 4:49 PM > >>> To: Wang Xingchao > >>> Cc: Girdwood, Liam R; Lin, Mengdong; Li, Jocelyn; > >>> alsa-devel(a)alsa-project.org; Wang, Xingchao > >>> Subject: Re: [PATCH] ALSA: hda - Enable runtime pm for Haswell > >>> > >>> At Thu, 16 May 2013 16:29:05 +0800, > >>> Wang Xingchao wrote: > >>>> Haswell doesnot support runtime pm by default. > >>>> This patch let haswell Display HD-A controller enter runtime > >>>> suspend, and bring more power saving whith power-well. > >>>> > >>>> Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> > >>> I don't think it's good to fiddle such a thing in the driver side. > >>> If Haswell can support runtime PM really, it should be set commonly. > >> I wonder whether it's HD-A driver's policy to only support runtime PM if the > device can support signal wakup? > >> According to Rafael, the device can support runtime PM regardless, no > matter it supports PME or not. > >> If so, we should remove the "if" condition check here. > > Well, if the decision is purely a driver issue, then we can get rid of > > PME check. > > Yes, it is. > > > But in that case, it should be simply like: > > > > azx_probe() { > > ... > > if (chip->driver_caps & AZX_DCAPS_PM_RUNTIME) > > pm_runtime_put_noidle(&pci->dev); > > ... > > } > > > > azx_remove() { > > ... > > if (chip->driver_caps & AZX_DCAPS_PM_RUNTIME) > > pm_runtime_get_noresume(&pci->dev); > > ... > > } > > > > > > AFAIU, calling pm_runtime_allow() enables the runtime PM *forcibly*. > > Usually this isn't a good thing. > > > > That's correct. Moreover, calling both pm_runtime_allow() *and* > pm_runtime_put_noidle() together would be a bug IMO. It's interesting something maybe wrong in my test: 1. withtout calling pm_runtime_allow(), azx_runteim_idle/suspend() will not be called. 2. another trick is on my Haswell ULT C stepping board, the runtime PM only work after exit from resume: echo mem > /sys/power/state if you donot let system enter suspend manually, the runtime pm will not be triggered. Is there any dependency between runtime pm suspend and normal suspend? Thanks --xingchao > > Thanks, > Rafael > > > > > >>>> --- > >>>> sound/pci/hda/hda_intel.c | 6 ++++++ > >>>> 1 file changed, 6 insertions(+) > >>>> > >>>> diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > >>>> index bf27693..eb25888 100644 > >>>> --- a/sound/pci/hda/hda_intel.c > >>>> +++ b/sound/pci/hda/hda_intel.c > >>>> @@ -3755,6 +3755,12 @@ static int azx_probe(struct pci_dev *pci, > >>>> > >>>> if (pci_dev_run_wake(pci)) > >>>> pm_runtime_put_noidle(&pci->dev); > >>>> + else if (chip->driver_caps > >>>> + & AZX_DCAPS_I915_POWERWELL) { > >>>> + /* Haswell doesnot support runtime pm by default */ > >>>> + pm_runtime_put_noidle(&pci->dev); > >>>> + pm_runtime_allow(&pci->dev); > >>>> + } > >>>> > >>>> dev++; > >>>> complete_all(&chip->probe_wait); > >>>> -- > >>>> 1.7.9.5 > >>>>

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[alsa-devel] [PATCH v7 00/11] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 31 May '13

31 May '13

Hi, This series adds DT support for phycore-ac97. Beside ac97 support, the series adds imx-pcm-fiq and generic DMA bindings to fsl-ssi. Regards, Markus Changes in v7: - Only one soc_ac97_ops struct for both ssi drivers (fsl-ssi and imx-ssi) to make it possible to compile a kernel with both drivers. - Other small bug/style fixes described in the changelogs of the patches. Changes in v6: - Rebased onto 3.10-rc3 with Shawn's pcm cleanup patches - Setup of reset functions moved to phycore-ac97.c Changes in v5: - Pass of_node via parent device to imx-pcm-dma. Changes in v4: - phycore-ac97 uses the iomux functions based on audmux phandle now. - fsl-ssi imx-pcm-fiq got some new comments about problems with imx-ssi and incompatible codecs Changes in v3: - Rebased onto pcm dma cleanup patches - A lot of cleanups for the fsl-ssi ac97 integration - Some small changes which are listed in the notes of the patches. Markus Pargmann (11): ASoC: imx-pcm-dma: DT support ASoC: imx-pcm-fiq: Introduce pcm-fiq-params ASoC: fsl: Move soc_ac97_ops from imx-ssi to fsl_ssi ASoC: fsl-ssi: Add support for imx-pcm-fiq ASoC: fsl-ssi: Use generic DMA bindings if possible ASoC: fsl-ssi: imx ac97 support ARM: imx: Export ac97 reset functions ASoC: phycore-ac97: Add DT support ASoC: fsl: Kconfig: Use fsl-ssi for phycore-ac97 ASoC: fsl: Move fsl-ssi binding doc to sound/ ASoC: fsl: Update fsl-ssi binding doc .../{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} | 10 + .../bindings/sound/phytec,phycore-ac97.txt | 14 + arch/arm/mach-imx/mach-pca100.c | 7 +- arch/arm/mach-imx/mach-pcm043.c | 7 +- sound/soc/fsl/Kconfig | 4 +- sound/soc/fsl/fsl_ssi.c | 450 +++++++++++++++++---- sound/soc/fsl/fsl_ssi.h | 7 + sound/soc/fsl/imx-pcm-dma.c | 1 - sound/soc/fsl/imx-pcm-fiq.c | 18 +- sound/soc/fsl/imx-pcm.h | 15 +- sound/soc/fsl/imx-ssi.c | 14 +- sound/soc/fsl/imx-ssi.h | 1 + sound/soc/fsl/phycore-ac97.c | 236 +++++++++-- 13 files changed, 656 insertions(+), 128 deletions(-) rename Documentation/devicetree/bindings/{powerpc/fsl/ssi.txt => sound/fsl,ssi.txt} (88%) create mode 100644 Documentation/devicetree/bindings/sound/phytec,phycore-ac97.txt

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[alsa-devel] [PATCH] alsa-utils 1.0.27.1 still ships 90-alsa-restore.rules
by Elimar Riesebieter 30 May '13

30 May '13

Hi all, [0] tells: alsactl: remove 90-alsa-restore.rules from the tar ball but it still ships 90-alsa-restore.rules.in and installs the file as 90-alsa-restore.rules [0] http://www.alsa-project.org/main/index.php/Changes_v1.0.27_v1.0.27.1 Thanks Signed-off-by: Elimar Riesebieter <riesebie(a)lxtec.de> --- alsactl/90-alsa-restore.rules.in | 8 -------- alsactl/Makefile.am | 5 ----- 2 files changed, 13 deletions(-) diff --git a/alsactl/90-alsa-restore.rules.in b/alsactl/90-alsa-restore.rules.in deleted file mode 100644 index 88e12e0..0000000 --- a/alsactl/90-alsa-restore.rules.in +++ /dev/null @@ -1,8 +0,0 @@ -ACTION=="add", SUBSYSTEM=="sound", KERNEL=="controlC*", KERNELS!="card*", GOTO="alsa_restore_go" -GOTO="alsa_restore_end" - -LABEL="alsa_restore_go" -TEST!="@daemonswitch@", RUN+="@sbindir@/alsactl restore $attr{number}" -TEST=="@daemonswitch@", RUN+="@sbindir@/alsactl nrestore $attr{number}" - -LABEL="alsa_restore_end" diff --git a/alsactl/Makefile.am b/alsactl/Makefile.am index bac84eb..3bf8eca 100644 --- a/alsactl/Makefile.am +++ b/alsactl/Makefile.am @@ -11,8 +11,6 @@ alsactl_SOURCES=alsactl.c state.c lock.c utils.c init_parse.c daemon.c alsactl_CFLAGS=$(AM_CFLAGS) -D__USE_GNU -DSYS_ASOUNDRC=\"$(ASOUND_STATE_DIR)/asound.state\" -DSYS_PIDFILE=\"$(ALSACTL_PIDFILE_DIR)/alsactl.pid\" noinst_HEADERS=alsactl.h list.h init_sysdeps.c init_utils_string.c init_utils_run.c init_sysfs.c -udevrules_DATA = \ - 90-alsa-restore.rules if HAVE_SYSTEMD @@ -49,20 +47,17 @@ alsa-restore.service: alsa-restore.service.in alsa-store.service: alsa-store.service.in $(edit) -90-alsa-restore.rules: 90-alsa-restore.rules.in $(edit) EXTRA_DIST += \ alsa-state.service.in \ alsa-restore.service.in \ alsa-store.service.in \ - 90-alsa-restore.rules.in CLEANFILES = \ alsa-state.service \ alsa-restore.service \ alsa-store.service \ - 90-alsa-restore.rules %.7: %.xml xmlto man $?

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