April 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

4 4

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

27 Oct '23

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

3 2

[alsa-devel] 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 0/5] 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] 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] 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] hdmi output (switching sample format/rates)
by Sander Jansen 25 May '13

25 May '13

In my audio application when I sometimes need to switch to a different sample format and/or rate, I usually leave the audio device open and merely reconfigure the device to the new parameters. This works with all the regular pcm devices, but it fails when I use the "hdmi" device. It seemed to just silently fail and no audio output is heard. As far as I know, no error condition is triggered (but I'll admit, my knowledge of the alsa api is rather limited). My question, Is this a bug or is it supposed to work this way and if so are there ways to detect whether a devices needs to be reopened or not. Thanks, Sander

2 2

[alsa-devel] [PATCH] ALSA: ASoC: add codec driver for TI TAS5086
by Daniel Mack 22 May '13

22 May '13

This patch adds a driver for TI's TA5086 6-channel PWM processor. This chip has a very unusual register layout, specifically because the registers are of unequal size, and multi-byte registers require bulk writes to take effect. Regmap does not support these kind of mappings. Currently, the driver does not touch any of the registers >= 0x20, so it doesn't matter, because the register map is mapped to an 8-bit array. In case more features will be added in the future that require access to higher registers, the entire regmap H/W I/O routines have to be open-coded. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- .../devicetree/bindings/sound/ti,tas5086.txt | 32 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5086.c | 587 +++++++++++++++++++++ 4 files changed, 625 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tas5086.txt create mode 100644 sound/soc/codecs/tas5086.c diff --git a/Documentation/devicetree/bindings/sound/ti,tas5086.txt b/Documentation/devicetree/bindings/sound/ti,tas5086.txt new file mode 100644 index 0000000..8ea4f5b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas5086.txt @@ -0,0 +1,32 @@ +Texas Instruments TAS5086 6-channel PWM Processor + +Required properties: + + - compatible: Should contain "ti,tas5086". + - reg: The i2c address. Should contain <0x1b>. + +Optional properties: + + - reset-gpio: A GPIO spec to define which pin is connected to the + chip's !RESET pin. If specified, the driver will + assert a hardware reset at probe time. + + - ti,charge-period: This property should contain the time in microseconds + that closely matches the external single-ended + split-capacitor charge period. The hardware chip + waits for this period of time before starting the + PWM signals. This helps reduce pops and clicks. + + When not specified, the hardware default of 1300ms + is retained. + +Examples: + + i2c_bus { + tas5086@1b { + compatible = "ti,tas5086"; + reg = <0x1b>; + reset-gpio = <&gpio 23 0>; + ti,charge-period = <156000>; + }; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 45b7256..86b3524 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -63,6 +63,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STA32X if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS + select SND_SOC_TAS5086 if I2C select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC32X4 if I2C @@ -320,6 +321,9 @@ config SND_SOC_STA529 config SND_SOC_STAC9766 tristate +config SND_SOC_TAS5086 + tristate + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6a3b3c3..8077bc2 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o +snd-soc-tas5086-objs := tas5086.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -177,6 +178,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o +obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c new file mode 100644 index 0000000..cd3c267 --- /dev/null +++ b/sound/soc/codecs/tas5086.c @@ -0,0 +1,587 @@ +/* + * TAS5086 ASoC codec driver + * + * Copyright (c) 2013 Daniel Mack <zonque(a)gmail.com> + * + * 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. + * + * TODO: + * - implement DAPM and input muxing + * - implement modulation limit + * - implement non-default PWM start + * + * Note that this chip has a very unusual register layout, specifically + * because the registers are of unequal size, and multi-byte registers + * require bulk writes to take effect. Regmap does not support that kind + * of devices. + * + * Currently, the driver does not touch any of the registers >= 0x20, so + * it doesn't matter because the entire map can be accessed as 8-bit + * array. In case more features will be added in the future + * that require access to higher registers, the entire regmap H/W I/O + * routines have to be open-coded. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/delay.h> +#include <linux/gpio.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/spi/spi.h> +#include <linux/of_device.h> +#include <linux/of_gpio.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#define TAS5086_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE) + +#define TAS5086_PCM_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +/* + * TAS5086 registers + */ +#define TAS5086_CLOCK_CONTROL 0x00 /* Clock control register */ +#define TAS5086_CLOCK_RATE(val) (val << 5) +#define TAS5086_CLOCK_RATE_MASK (0x7 << 5) +#define TAS5086_CLOCK_RATIO(val) (val << 2) +#define TAS5086_CLOCK_RATIO_MASK (0x7 << 2) +#define TAS5086_CLOCK_SCLK_RATIO_48 (1 << 1) +#define TAS5086_CLOCK_VALID (1 << 0) + +#define TAS5086_DEV_ID 0x01 /* Device ID register */ +#define TAS5086_ERROR_STATUS 0x02 /* Error status register */ +#define TAS5086_SYS_CONTROL_1 0x03 /* System control register 1 */ +#define TAS5086_SERIAL_DATA_IF 0x04 /* Serial data interface register */ +#define TAS5086_SYS_CONTROL_2 0x05 /* System control register 2 */ +#define TAS5086_SOFT_MUTE 0x06 /* Soft mute register */ +#define TAS5086_MASTER_VOL 0x07 /* Master volume */ +#define TAS5086_CHANNEL_VOL(X) (0x08 + (X)) /* Channel 1-6 volume */ +#define TAS5086_VOLUME_CONTROL 0x09 /* Volume control register */ +#define TAS5086_MOD_LIMIT 0x10 /* Modulation limit register */ +#define TAS5086_PWM_START 0x18 /* PWM start register */ +#define TAS5086_SURROUND 0x19 /* Surround register */ +#define TAS5086_SPLIT_CAP_CHARGE 0x1a /* Split cap charge period register */ +#define TAS5086_OSC_TRIM 0x1b /* Oscillator trim register */ +#define TAS5086_BKNDERR 0x1c + +#define TAS5086_DEEMPH_MASK 0x03 + +/* + * Default TAS5086 power-up configuration + */ +static const struct reg_default tas5086_reg_defaults[] = { + { 0x00, 0x6c }, + { 0x01, 0x03 }, + { 0x02, 0x00 }, + { 0x03, 0xa0 }, + { 0x04, 0x05 }, + { 0x05, 0x60 }, + { 0x06, 0x00 }, + { 0x07, 0xff }, + { 0x08, 0x30 }, + { 0x09, 0x30 }, + { 0x0a, 0x30 }, + { 0x0b, 0x30 }, + { 0x0c, 0x30 }, + { 0x0d, 0x30 }, + { 0x0e, 0xb1 }, + { 0x0f, 0x00 }, + { 0x10, 0x02 }, + { 0x11, 0x00 }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, + { 0x14, 0x00 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, + { 0x18, 0x3f }, + { 0x19, 0x00 }, + { 0x1a, 0x18 }, + { 0x1b, 0x82 }, + { 0x1c, 0x05 }, +}; + +static bool tas5086_accessible_reg(struct device *dev, unsigned int reg) +{ + return !((reg == 0x0f) || (reg >= 0x11 && reg <= 0x17)); +} + +static bool tas5086_volatile_reg(struct device *dev, unsigned int reg) +{ + return reg == TAS5086_DEV_ID || + reg == TAS5086_ERROR_STATUS; +} + +static bool tas5086_writeable_reg(struct device *dev, unsigned int reg) +{ + return tas5086_accessible_reg(dev, reg) && (reg != TAS5086_DEV_ID); +} + +struct tas5086_private { + struct regmap *regmap; + unsigned int mclk, sclk; + unsigned int format; + bool deemph; + /* Current sample rate for de-emphasis control */ + int rate; + /* GPIO driving Reset pin, if any */ + int gpio_nreset; +}; + +static int tas5086_deemph[] = { 0, 32000, 44100, 48000 }; + +static int tas5086_set_deemph(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int i, val = 0; + + if (priv->deemph) + for (i = 0; i < ARRAY_SIZE(tas5086_deemph); i++) + if (tas5086_deemph[i] == priv->rate) + val = i; + + return regmap_update_bits(priv->regmap, TAS5086_SYS_CONTROL_1, + TAS5086_DEEMPH_MASK, val); +} + +static int tas5086_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = priv->deemph; + + return 0; +} + +static int tas5086_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->deemph = ucontrol->value.enumerated.item[0]; + + return tas5086_set_deemph(codec); +} + + +static int tas5086_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case 0: /* MCLK */ + priv->mclk = freq; + break; + case 1: /* SCLK */ + priv->sclk = freq; + break; + } + + return 0; +} + +static int tas5086_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + /* The TAS5086 can only be slave to all clocks */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Invalid clocking mode\n"); + return -EINVAL; + } + + /* we need to refer to the data format from hw_params() */ + priv->format = format; + + return 0; +} + +static const int tas5086_sample_rates[] = { + 32000, 38000, 44100, 48000, 88200, 96000, 176400, 192000 +}; + +static const int tas5086_ratios[] = { + 64, 128, 192, 256, 384, 512 +}; + +static int index_in_array(const int *array, int len, int needle) +{ + int i; + + for (i = 0; i < len; i++) + if (array[i] == needle) + return i; + + return -ENOENT; +} + +static int tas5086_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int ret; + + priv->rate = params_rate(params); + + /* Look up the sample rate and refer to the offset in the list */ + val = index_in_array(tas5086_sample_rates, + ARRAY_SIZE(tas5086_sample_rates), priv->rate); + + if (val < 0) { + dev_err(codec->dev, "Invalid sample rate\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_RATE_MASK, + TAS5086_CLOCK_RATE(val)); + if (ret < 0) + return ret; + + /* MCLK / Fs ratio */ + val = index_in_array(tas5086_ratios, ARRAY_SIZE(tas5086_ratios), + priv->mclk / priv->rate); + if (val < 0) { + dev_err(codec->dev, "Inavlid MCLK / Fs ratio\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_RATIO_MASK, + TAS5086_CLOCK_RATIO(val)); + if (ret < 0) + return ret; + + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_SCLK_RATIO_48, + (priv->sclk == 48 * priv->rate) ? + TAS5086_CLOCK_SCLK_RATIO_48 : 0); + if (ret < 0) + return ret; + + /* + * The chip has a very unituitive register mapping and muxes information + * about data format and sample depth into the same register, but not on + * a logical bit-boundary. Hence, we have to refer to the format passed + * in the set_dai_fmt() callback and set up everything from here. + * + * First, determine the 'base' value, using the format ... + */ + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + val = 0x00; + break; + case SND_SOC_DAIFMT_I2S: + val = 0x03; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = 0x06; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + /* ... then add the offset for the sample bit depth. */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + val += 0; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val += 1; + break; + case SNDRV_PCM_FORMAT_S24_3LE: + val += 2; + break; + default: + dev_err(codec->dev, "Invalid bit width\n"); + return -EINVAL; + }; + + ret = regmap_write(priv->regmap, TAS5086_SERIAL_DATA_IF, val); + if (ret < 0) + return ret; + + /* clock is considered valid now */ + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_VALID, TAS5086_CLOCK_VALID); + if (ret < 0) + return ret; + + return tas5086_set_deemph(codec); +} + +static int tas5086_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + return regmap_write(priv->regmap, TAS5086_SOFT_MUTE, + mute ? 0x3f : 0x00); +} + +/* TAS5086 controls */ +static const DECLARE_TLV_DB_SCALE(tas5086_dac_tlv, -10350, 50, 1); + +static const struct snd_kcontrol_new tas5086_controls[] = { + SOC_SINGLE_TLV("Master Playback Volume", TAS5086_MASTER_VOL, + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 1/2 Playback Volume", + TAS5086_CHANNEL_VOL(0), TAS5086_CHANNEL_VOL(1), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 3/4 Playback Volume", + TAS5086_CHANNEL_VOL(2), TAS5086_CHANNEL_VOL(3), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 5/6 Playback Volume", + TAS5086_CHANNEL_VOL(4), TAS5086_CHANNEL_VOL(5), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0, + tas5086_get_deemph, tas5086_put_deemph), +}; + +static const struct snd_soc_dai_ops tas5086_dai_ops = { + .hw_params = tas5086_hw_params, + .set_sysclk = tas5086_set_dai_sysclk, + .set_fmt = tas5086_set_dai_fmt, + .digital_mute = tas5086_digital_mute, +}; + +static struct snd_soc_dai_driver tas5086_dai = { + .name = "tas5086-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 6, + .rates = TAS5086_PCM_RATES, + .formats = TAS5086_PCM_FORMATS, + }, + .ops = &tas5086_dai_ops, +}; + +#ifdef CONFIG_PM +static int tas5086_soc_suspend(struct snd_soc_codec *codec) +{ + return 0; +} + +static int tas5086_soc_resume(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + /* Restore codec state */ + return regcache_sync(priv->regmap); +} +#else +#define tas5086_soc_suspend NULL +#define tas5086_soc_resume NULL +#endif /* CONFIG_PM */ + +#ifdef CONFIG_OF +static const struct of_device_id tas5086_dt_ids[] = { + { .compatible = "ti,tas5086", }, + { } +}; +MODULE_DEVICE_TABLE(of, tas5086_dt_ids); +#endif + +/* charge period values in microseconds */ +static const int tas5086_charge_period[] = { + 13000, 16900, 23400, 31200, 41600, 54600, 72800, 96200, + 130000, 156000, 234000, 312000, 416000, 546000, 728000, 962000, + 1300000, 169000, 2340000, 3120000, 4160000, 5460000, 7280000, 9620000, +}; + +static int tas5086_probe(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int charge_period = 1300000; /* hardware default is 1300 ms */ + int gpio_nreset = -EINVAL; + int i, ret; + + if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) { + struct device_node *of_node = codec->dev->of_node; + + gpio_nreset = of_get_named_gpio(of_node, "reset-gpio", 0); + of_property_read_u32(of_node, "ti,charge-period", &charge_period); + } + + if (gpio_nreset >= 0) + if (devm_gpio_request(codec->dev, gpio_nreset, "TAS5086 Reset")) + gpio_nreset = -EINVAL; + if (gpio_nreset >= 0) { + /* Reset codec - minimum assertion time is 400ns */ + gpio_direction_output(gpio_nreset, 0); + udelay(1); + gpio_set_value(gpio_nreset, 1); + + /* Codec needs ~15ms to wake up */ + msleep(15); + } +msleep(300); + priv->gpio_nreset = gpio_nreset; + + /* The TAS5086 always returns 0x03 in its TAS5086_DEV_ID register */ + ret = regmap_read(priv->regmap, TAS5086_DEV_ID, &i); + if (ret < 0) + return ret; + + if (i != 0x3) { + dev_err(codec->dev, + "Failed to identify TAS5086 codec (got %02x)\n", i); + return -ENODEV; + } + + /* lookup and set split-capacitor charge period */ + if (charge_period == 0) { + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); + } else { + i = index_in_array(tas5086_charge_period, + ARRAY_SIZE(tas5086_charge_period), + charge_period); + if (i >= 0) + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, + i + 0x08); + else + dev_warn(codec->dev, + "Invalid split-cap charge period of %d ns.\n", + charge_period); + } + + /* enable factory trim */ + ret = regmap_write(priv->regmap, TAS5086_OSC_TRIM, 0x00); + if (ret < 0) + return ret; + + /* start all channels */ + ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x20); + if (ret < 0) + return ret; + + /* set master volume to 0 dB */ + ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30); + if (ret < 0) + return ret; + + return 0; +} + +static int tas5086_remove(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(priv->gpio_nreset)) + /* Set codec to the reset state */ + gpio_set_value(priv->gpio_nreset, 0); + + return 0; +}; + +static struct snd_soc_codec_driver soc_codec_dev_tas5086 = { + .probe = tas5086_probe, + .remove = tas5086_remove, + .suspend = tas5086_soc_suspend, + .resume = tas5086_soc_resume, + .controls = tas5086_controls, + .num_controls = ARRAY_SIZE(tas5086_controls), +}; + +static const struct i2c_device_id tas5086_i2c_id[] = { + { "tas5086", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas5086_i2c_id); + +static const struct regmap_config tas5086_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(tas5086_reg_defaults), + .reg_defaults = tas5086_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas5086_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .volatile_reg = tas5086_volatile_reg, + .writeable_reg = tas5086_writeable_reg, + .readable_reg = tas5086_accessible_reg, +}; + +static int tas5086_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct tas5086_private *priv; + int ret; + + priv = devm_kzalloc(&i2c->dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(i2c, &tas5086_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&i2c->dev, "Failed to create regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(i2c, priv); + + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086, + &tas5086_dai, 1); +} + +static int tas5086_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +static struct i2c_driver tas5086_i2c_driver = { + .driver = { + .name = "tas5086", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tas5086_dt_ids), + }, + .id_table = tas5086_i2c_id, + .probe = tas5086_i2c_probe, + .remove = tas5086_i2c_remove, +}; + +static int __init tas5086_modinit(void) +{ + return i2c_add_driver(&tas5086_i2c_driver); +} +module_init(tas5086_modinit); + +static void __exit tas5086_modexit(void) +{ + i2c_del_driver(&tas5086_i2c_driver); +} +module_exit(tas5086_modexit); + +MODULE_AUTHOR("Daniel Mack <zonque(a)gmail.com>"); +MODULE_DESCRIPTION("Texas Instruments TAS5086 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); -- 1.8.1.4

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[alsa-devel] [PATCH 0/5] ASoC: fsl: imx-pcm driver cleanup
by Shawn Guo 15 May '13

15 May '13

The series removes the use of imx-pcm-audio/imx-fiq-pcm-audio platform device, and deletes imx-pcm driver by moving the functions needed by imx-pcm-fiq driver into imx-pcm-fiq.c. This can be done because imx-pcm-dma has moved to generic dmaengine pcm driver and does not use any helper from imx-pcm driver now. Shawn Guo (5): ASoC: fsl: remove use of imx-pcm-audio from fsl_ssi ASoC: fsl: remove use of imx-pcm-audio from imx-ssi ASoC: fsl: create function imx_pcm_fiq_exit() ASoC: fsl: remove use of imx-fiq-pcm-audio from imx-ssi ASoC: fsl: remove imx-pcm driver sound/soc/fsl/Kconfig | 5 -- sound/soc/fsl/Makefile | 11 +-- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl_ssi.c | 13 ++-- sound/soc/fsl/imx-mc13783.c | 2 +- sound/soc/fsl/imx-pcm-dma.c | 2 + sound/soc/fsl/imx-pcm-fiq.c | 92 +++++++++++++++++++++++++ sound/soc/fsl/imx-pcm.c | 145 --------------------------------------- sound/soc/fsl/imx-pcm.h | 10 +-- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/imx-ssi.c | 44 +++--------- sound/soc/fsl/imx-ssi.h | 3 - sound/soc/fsl/mx27vis-aic32x4.c | 2 +- sound/soc/fsl/phycore-ac97.c | 2 +- sound/soc/fsl/wm1133-ev1.c | 2 +- 15 files changed, 122 insertions(+), 215 deletions(-) delete mode 100644 sound/soc/fsl/imx-pcm.c -- 1.7.9.5

2 13

[alsa-devel] [PATCH 1/8] ASoC: generic-dmaengine-pcm: Add support for half-duplex
by Lars-Peter Clausen 13 May '13

13 May '13

Some platforms which are half-duplex share the same DMA channel between the playback and capture stream. Add support for this to the generic dmaengine PCM driver. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/dmaengine_pcm.h | 5 ++++ sound/soc/soc-generic-dmaengine-pcm.c | 43 ++++++++++++++++++++++++----------- 2 files changed, 35 insertions(+), 13 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index b1d1150..f11c35c 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -91,6 +91,11 @@ void snd_dmaengine_pcm_set_config_from_dai_data( * bytes that are still left to transfer. */ #define SND_DMAENGINE_PCM_FLAG_NO_RESIDUE BIT(2) +/* + * The PCM is half duplex and the DMA channel is shared between capture and + * playback. + */ +#define SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX BIT(3) /** * struct snd_dmaengine_pcm_config - Configuration data for dmaengine based PCM diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index ae0c37e..5fd5ed4 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -29,7 +29,7 @@ struct dmaengine_pcm { struct dma_chan *chan[SNDRV_PCM_STREAM_CAPTURE + 1]; const struct snd_dmaengine_pcm_config *config; struct snd_soc_platform platform; - bool compat; + unsigned int flags; }; static struct dmaengine_pcm *soc_platform_to_pcm(struct snd_soc_platform *p) @@ -128,6 +128,9 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) + return pcm->chan[0]; + if (pcm->config->compat_request_channel) return pcm->config->compat_request_channel(rtd, substream); @@ -148,7 +151,7 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) if (!substream) continue; - if (!pcm->chan[i] && pcm->compat) { + if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd, substream); } @@ -215,6 +218,25 @@ static const char * const dmaengine_pcm_dma_channel_names[] = { [SNDRV_PCM_STREAM_CAPTURE] = "rx", }; +static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, + struct device_node *of_node) +{ + unsigned int i; + + if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !of_node) + return; + + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { + pcm->chan[0] = of_dma_request_slave_channel(of_node, "tx_rx"); + pcm->chan[1] = pcm->chan[0]; + } else { + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { + pcm->chan[i] = of_dma_request_slave_channel(of_node, + dmaengine_pcm_dma_channel_names[i]); + } + } +} + /** * snd_dmaengine_pcm_register - Register a dmaengine based PCM device * @dev: The parent device for the PCM device @@ -225,23 +247,15 @@ int snd_dmaengine_pcm_register(struct device *dev, const struct snd_dmaengine_pcm_config *config, unsigned int flags) { struct dmaengine_pcm *pcm; - unsigned int i; pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); if (!pcm) return -ENOMEM; pcm->config = config; + pcm->flags = flags; - if (flags & SND_DMAENGINE_PCM_FLAG_COMPAT) - pcm->compat = true; - - if (!(flags & SND_DMAENGINE_PCM_FLAG_NO_DT) && dev->of_node) { - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - pcm->chan[i] = of_dma_request_slave_channel(dev->of_node, - dmaengine_pcm_dma_channel_names[i]); - } - } + dmaengine_pcm_request_chan_of(pcm, dev->of_node); if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) return snd_soc_add_platform(dev, &pcm->platform, @@ -272,8 +286,11 @@ void snd_dmaengine_pcm_unregister(struct device *dev) pcm = soc_platform_to_pcm(platform); for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - if (pcm->chan[i]) + if (pcm->chan[i]) { dma_release_channel(pcm->chan[i]); + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) + break; + } } snd_soc_remove_platform(platform); -- 1.8.0

3 19

[alsa-devel] [PATCH 1/2] ASoC: mxs-sgtl5000: Remove unneeded 'ret' variable
by Fabio Estevam 06 May '13

06 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Variable 'ret' is not needed here, so just remove it. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index b1d9b5e..4f74b05 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -116,7 +116,7 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct device_node *saif_np[2], *codec_np; - int i, ret = 0; + int i; if (!np) return 1; /* no device tree */ @@ -142,7 +142,7 @@ static int mxs_sgtl5000_probe_dt(struct platform_device *pdev) of_node_put(saif_np[0]); of_node_put(saif_np[1]); - return ret; + return 0; } static int mxs_sgtl5000_probe(struct platform_device *pdev) -- 1.7.9.5

3 5

[alsa-devel] [PATCH] ALSA: hda - delay resume haswell hdmi codec in system resume
by mengdong.lin＠intel.com 06 May '13

06 May '13

From: Mengdong Lin <mengdong.lin(a)intel.com> In system resume, Haswell codec cannot be programmed to D0 before Gfx driver initializes the display pipeline and audio, which will trigger an unsol event on the pin with HDMI/DP cable connected. Otherwise, the connected pin will stay in D3 with right channel muted and thus no sound can be heard. This patch - adds a codec flag to delay resuming a codec. System resume will skip the codecs if this flag is set, and these codecs will be resumed on later codec access. - adds a set_power_state ops for Haswell HDMI codec. In a delayed resume, this ops will enable and wait for the unsol event, and then resume the codec. A 300ms timeout is set in case unsol event is lost. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 04b5738..bcb7205 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -5508,11 +5508,15 @@ int snd_hda_resume(struct hda_bus *bus) struct hda_codec *codec; list_for_each_entry(codec, &bus->codec_list, list) { - hda_call_codec_resume(codec); + if (codec->support_delay_resume) + codec->resume_delayed = 1; + else + hda_call_codec_resume(codec); } return 0; } EXPORT_SYMBOL_HDA(snd_hda_resume); + #endif /* CONFIG_PM */ /* diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 23ca172..5b5e5f4 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -886,6 +886,8 @@ struct hda_codec { unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ unsigned int pm_down_notified:1; /* PM notified to controller */ unsigned int in_pm:1; /* suspend/resume being performed */ + unsigned int support_delay_resume:1; /* codec support delay resume */ + unsigned int resume_delayed:1; /* resume delayed by PM */ int power_transition; /* power-state in transition */ int power_count; /* current (global) power refcount */ struct delayed_work power_work; /* delayed task for powerdown */ diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 78e1827..d116908 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -98,6 +98,14 @@ struct hdmi_spec { */ struct hda_multi_out multiout; struct hda_pcm_stream pcm_playback; + +#ifdef CONFIG_PM + /* + * Non-generic Intel Haswell specific + */ + unsigned int ready_to_resume:1; + wait_queue_head_t resume_wq; +#endif }; @@ -977,6 +985,13 @@ static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) if (pin_idx < 0) return; +#ifdef CONFIG_PM + if (codec->resume_delayed) { + spec->ready_to_resume = 1; + wake_up(&spec->resume_wq); + } +#endif + hdmi_present_sense(&spec->pins[pin_idx], 1); snd_hda_jack_report_sync(codec); } @@ -1846,6 +1861,63 @@ static const struct hda_fixup hdmi_fixups[] = { }; +#ifdef CONFIG_PM +static void intel_haswell_wait_ready_to_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + int pin_idx; + + spec->ready_to_resume = 0; + + for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { + struct hdmi_spec_per_pin *per_pin; + hda_nid_t pin_nid; + struct hda_jack_tbl *jack; + + per_pin = &spec->pins[pin_idx]; + pin_nid = per_pin->pin_nid; + jack = snd_hda_jack_tbl_get(codec, pin_nid); + if (jack) + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | jack->tag); + } + + wait_event_timeout(spec->resume_wq, + spec->ready_to_resume, msecs_to_jiffies(300)); + if (!spec->ready_to_resume) + snd_printd(KERN_WARNING "HDMI: Haswell not ready to resume\n"); +} + +static void intel_haswell_set_power_state(struct hda_codec *codec, hda_nid_t fg, + unsigned int power_state) +{ + if (codec->resume_delayed && power_state == AC_PWRST_D0) { + intel_haswell_wait_ready_to_resume(codec); + codec->resume_delayed = 0; + } + + snd_hda_codec_read(codec, fg, 0, + AC_VERB_SET_POWER_STATE, + power_state); + + snd_hda_codec_set_power_to_all(codec, fg, power_state); +} + +static inline void intel_haswell_allow_delay_resume(struct hda_codec *codec) +{ + struct hdmi_spec *spec = codec->spec; + + init_waitqueue_head(&spec->resume_wq); + codec->support_delay_resume = 1; + + codec->patch_ops.set_power_state = + intel_haswell_set_power_state; +} +#else +define intel_haswell_allow_delay_resume NULL +#endif + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1868,6 +1940,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -EINVAL; } codec->patch_ops = generic_hdmi_patch_ops; + + if (codec->vendor_id == 0x80862807) + intel_haswell_allow_delay_resume(codec); + generic_hdmi_init_per_pins(codec); init_channel_allocations(); -- 1.7.10.4

4 51

[alsa-devel] [RFC] Sound/HID: wiimote: add speaker support
by David Herrmann 05 May '13

05 May '13

Classic Wii Remotes provide a speaker on the device. We can stream PCM data, mute and change volume via special protocol requests and remote audio registers. The device supports several different formats, but fully understood are only signed 8-bit PCM and something that looks like 4-bit Yamaha ADPCM. Theoretically, we can set data-rates from 183Hz up to 12MHz, but a realistic range for Bluetooth l2cap is 1500-4000 Hz. Data is streamed as 20bytes blocks and must be sent in a constant rate. There is no way to read the current position. It would be way too slow, anyway. Signed-off-by: David Herrmann <dh.herrmann(a)gmail.com> --- Hi I'm reworking large parts of the HID Nintendo Wii Remote driver and wanted to add speaker support. I have never worked with the sound subsystem until now and need some help. The Wii Remote is a Bluetooth gamepad that features a small speaker on the remote. All information was gathered via reverse-engineering so I cannot tell you any specifics about the hardware, sorry. What I figured out so far is how to set up the speaker, stream data, set volume and mute the speaker. Supported formats are: signed 8-bit PCM 4-bit Yamaha ADPCM I implemented only the 8-bit PCM as I couldn't figure out whether the kernel supports the other format? The wiiproto_* helpers in hid-wiimote-core.c are used to send the requests. Data is streamed as 20-bytes packets and must be sent at a constant rate. I would really appreciate if someone can have a look at hid-wiimote-speaker.c and help me fill the gaps. I figured out how to write the basic snd_card management and PCM setup, but I am stuck with buffer management now. I have a function called wiiproto_req_audio() which sends up to 20bytes of PCM data to the device. However, I cannot figure out how to integrate that into the snd_pcm object. The problem is that all other drivers I found have large DMA buffers which are filled whenever an interrupt signals more data. However, I don't have this setup but instead I need a timer (does the sound-core provide that?) which causes the "copy" callback to be called with 20bytes of data at a constant rate. Any help welcome! And if this whole idea is stupid, please tell me! ;) Thanks David drivers/hid/Kconfig | 3 + drivers/hid/Makefile | 3 + drivers/hid/hid-wiimote-core.c | 54 +++- drivers/hid/hid-wiimote-modules.c | 5 + drivers/hid/hid-wiimote-speaker.c | 539 ++++++++++++++++++++++++++++++++++++++ drivers/hid/hid-wiimote.h | 17 ++ 6 files changed, 619 insertions(+), 2 deletions(-) create mode 100644 drivers/hid/hid-wiimote-speaker.c diff --git a/drivers/hid/Kconfig b/drivers/hid/Kconfig index e8ef86c..0b4d7e2 100644 --- a/drivers/hid/Kconfig +++ b/drivers/hid/Kconfig @@ -715,6 +715,9 @@ config HID_WIIMOTE the Wii U Gamepad) might be supported in the future. But currently support is limited to Bluetooth based devices. + If Alsa sound support (CONFIG_SND) is enabled, this driver also provides + a sound card interface for Wii Remote speakers. + If unsure, say N. To compile this driver as a module, choose M here: the diff --git a/drivers/hid/Makefile b/drivers/hid/Makefile index 8b7106b..d149e10 100644 --- a/drivers/hid/Makefile +++ b/drivers/hid/Makefile @@ -32,6 +32,9 @@ hid-wiimote-y := hid-wiimote-core.o hid-wiimote-modules.o ifdef CONFIG_DEBUG_FS hid-wiimote-y += hid-wiimote-debug.o endif +ifdef CONFIG_SND + hid-wiimote-y += hid-wiimote-speaker.o +endif obj-$(CONFIG_HID_A4TECH) += hid-a4tech.o obj-$(CONFIG_HID_ACRUX) += hid-axff.o diff --git a/drivers/hid/hid-wiimote-core.c b/drivers/hid/hid-wiimote-core.c index 906c146..53f1e88 100644 --- a/drivers/hid/hid-wiimote-core.c +++ b/drivers/hid/hid-wiimote-core.c @@ -17,6 +17,7 @@ #include <linux/module.h> #include <linux/mutex.h> #include <linux/spinlock.h> +#include <linux/uaccess.h> #include "hid-ids.h" #include "hid-wiimote.h" @@ -309,8 +310,8 @@ void wiiproto_req_ir2(struct wiimote_data *wdata, __u8 flags) #define wiiproto_req_weeprom(wdata, os, buf, sz) \ wiiproto_req_wmem((wdata), true, (os), (buf), (sz)) -static void wiiproto_req_wmem(struct wiimote_data *wdata, bool eeprom, - __u32 offset, const __u8 *buf, __u8 size) +void wiiproto_req_wmem(struct wiimote_data *wdata, bool eeprom, + __u32 offset, const __u8 *buf, __u8 size) { __u8 cmd[22]; @@ -359,6 +360,52 @@ void wiiproto_req_rmem(struct wiimote_data *wdata, bool eeprom, __u32 offset, wiimote_queue(wdata, cmd, sizeof(cmd)); } +void wiiproto_req_speaker(struct wiimote_data *wdata, bool on) +{ + __u8 cmd[2]; + + cmd[0] = WIIPROTO_REQ_SPEAKER; + cmd[1] = on ? 0x04 : 0x00; + + wiiproto_keep_rumble(wdata, &cmd[1]); + wiimote_queue(wdata, cmd, sizeof(cmd)); +} + +void wiiproto_req_mute(struct wiimote_data *wdata, bool on) +{ + __u8 cmd[2]; + + cmd[0] = WIIPROTO_REQ_MUTE; + cmd[1] = on ? 0x04 : 0x00; + + wiiproto_keep_rumble(wdata, &cmd[1]); + wiimote_queue(wdata, cmd, sizeof(cmd)); +} + +/* must not be called from atomic contexts */ +int wiiproto_req_audio_user(struct wiimote_data *wdata, + void __user *buf, size_t len) +{ + unsigned long flags; + __u8 cmd[22]; + + if (len > 20) + len = 20; + + cmd[0] = WIIPROTO_REQ_AUDIO; + cmd[1] = (len & 0xff) << 3; + + if (copy_from_user(&cmd[2], buf, len)) + return -EFAULT; + + spin_lock_irqsave(&wdata->state.lock, flags); + wiiproto_keep_rumble(wdata, &cmd[1]); + wiimote_queue(wdata, cmd, sizeof(cmd)); + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return 0; +} + /* requries the cmd-mutex to be held */ int wiimote_cmd_write(struct wiimote_data *wdata, __u32 offset, const __u8 *wmem, __u8 size) @@ -570,6 +617,7 @@ static const __u8 * const wiimote_devtype_mods[WIIMOTE_DEV_NUM] = { WIIMOD_LED4, WIIMOD_ACCEL, WIIMOD_IR, + WIIMOD_SPEAKER, WIIMOD_NULL, }, [WIIMOTE_DEV_GEN10] = (const __u8[]){ @@ -582,6 +630,7 @@ static const __u8 * const wiimote_devtype_mods[WIIMOTE_DEV_NUM] = { WIIMOD_LED4, WIIMOD_ACCEL, WIIMOD_IR, + WIIMOD_SPEAKER, WIIMOD_NULL, }, [WIIMOTE_DEV_GEN20] = (const __u8[]){ @@ -595,6 +644,7 @@ static const __u8 * const wiimote_devtype_mods[WIIMOTE_DEV_NUM] = { WIIMOD_ACCEL, WIIMOD_IR, WIIMOD_BUILTIN_MP, + WIIMOD_SPEAKER, WIIMOD_NULL, }, [WIIMOTE_DEV_BALANCE_BOARD] = (const __u8[]) { diff --git a/drivers/hid/hid-wiimote-modules.c b/drivers/hid/hid-wiimote-modules.c index 4ce0e02..7acf721 100644 --- a/drivers/hid/hid-wiimote-modules.c +++ b/drivers/hid/hid-wiimote-modules.c @@ -2072,6 +2072,11 @@ const struct wiimod_ops *wiimod_table[WIIMOD_NUM] = { [WIIMOD_IR] = &wiimod_ir, [WIIMOD_BUILTIN_MP] = &wiimod_builtin_mp, [WIIMOD_NO_MP] = &wiimod_no_mp, +#ifdef CONFIG_SND + [WIIMOD_SPEAKER] = &wiimod_speaker, +#else + [WIIMOD_SPEAKER] = &wiimod_dummy, +#endif }; const struct wiimod_ops *wiimod_ext_table[WIIMOTE_EXT_NUM] = { diff --git a/drivers/hid/hid-wiimote-speaker.c b/drivers/hid/hid-wiimote-speaker.c new file mode 100644 index 0000000..ecec5d9 --- /dev/null +++ b/drivers/hid/hid-wiimote-speaker.c @@ -0,0 +1,539 @@ +/* + * Driver for audio speakers of Nintendo Wii / Wii U peripherals + * Copyright (c) 2012-2013 David Herrmann <dh.herrmann(a)gmail.com> + */ + +/* + * 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. + */ + +/* + * Audio Speakers + * Some Wii peripherals provide an audio speaker that supports 8bit PCM and + * some other mostly unknown formats. Not all setup options are known, but we + * know how to setup an 8bit PCM or 4bit ADPCM stream and adjust volume. Data + * is sent as 20bytes chunks and needs to be streamed at a constant rate. + */ + +#include <linux/device.h> +#include <linux/hid.h> +#include <linux/mutex.h> +#include <linux/spinlock.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include "hid-wiimote.h" + +struct wiimote_speaker { + struct snd_card *card; + unsigned int online : 1; + unsigned int mute : 1; + __u8 volume; +}; + +enum wiimod_speaker_mode { + WIIMOD_SPEAKER_MODE_PCM8, + WIIMOD_SPEAKER_MODE_ADPCM4, +}; + +static int wiimod_speaker_enable(struct wiimote_data *wdata) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (!speaker->online) { + speaker->online = 1; + wiiproto_req_speaker(wdata, true); + wiiproto_req_mute(wdata, speaker->mute); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return 0; +} + +static void wiimod_speaker_disable(struct wiimote_data *wdata) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (speaker->online) { + speaker->online = 0; + wiiproto_req_speaker(wdata, false); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); +} + +static void wiimod_speaker_set_mute(struct wiimote_data *wdata, bool mute) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (speaker->mute != mute) { + speaker->mute = mute; + if (speaker->online) + wiiproto_req_mute(wdata, mute); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); +} + +static void wiimod_speaker_set_volume(struct wiimote_data *wdata, __u8 volume) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + + if (speaker->volume != volume) { + speaker->volume = volume; + if (speaker->online) + wiiproto_req_wmem(wdata, false, 0xa20005, &volume, + sizeof(volume)); + } + + spin_unlock_irqrestore(&wdata->state.lock, flags); +} + +/* Change speaker configuration. \mode can be one of WIIMOD_SPEAKER_MODE_*, + * \rate is the PCM sample rate and \volume is the requested volume. */ +static int wiimod_speaker_setup(struct wiimote_data *wdata, + __u8 mode, __u16 rate, __s16 volume) +{ + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + __u8 config[7], m, wmem; + __u16 r; + int ret; + + if (!rate) + return -EINVAL; + + switch (mode) { + case WIIMOD_SPEAKER_MODE_PCM8: + r = 12000000ULL / rate; + m = 0x40; + break; + case WIIMOD_SPEAKER_MODE_ADPCM4: + r = 6000000ULL / rate; + m = 0x00; + break; + default: + return -EINVAL; + } + + config[0] = 0x00; + config[1] = m; + config[2] = r & 0x00ff; + config[3] = (r & 0xff00) >> 8; + config[4] = volume & 0xff; + config[5] = 0x00; + config[6] = 0x00; + + wiimote_cmd_acquire_noint(wdata); + + /* mute speaker during setup and read/write volume field */ + spin_lock_irqsave(&wdata->state.lock, flags); + + wiiproto_req_mute(wdata, true); + if (volume < 0) + config[4] = speaker->volume; + else + speaker->volume = volume; + + spin_unlock_irqrestore(&wdata->state.lock, flags); + + /* power speaker */ + wmem = 0x01; + ret = wiimote_cmd_write(wdata, 0xa20009, &wmem, sizeof(wmem)); + if (ret) + goto out_unlock; + + /* prepare setup */ + wmem = 0x08; + ret = wiimote_cmd_write(wdata, 0xa20001, &wmem, sizeof(wmem)); + if (ret) + goto out_unlock; + + /* write configuration */ + ret = wiimote_cmd_write(wdata, 0xa20001, config, sizeof(config)); + if (ret) + goto out_unlock; + + /* enable speaker */ + wmem = 0x01; + ret = wiimote_cmd_write(wdata, 0xa20008, &wmem, sizeof(wmem)); + if (ret) + goto out_unlock; + + /* unmute speaker after setup if not muted */ + spin_lock_irqsave(&wdata->state.lock, flags); + if (!speaker->mute) + wiiproto_req_mute(wdata, false); + spin_unlock_irqrestore(&wdata->state.lock, flags); + +out_unlock: + wiimote_cmd_release(wdata); + return ret; +} + +/* PCM layer */ + +static const struct snd_pcm_hardware wiimod_speaker_playback_hw = { + .info = SNDRV_PCM_INFO_NONINTERLEAVED, + .formats = SNDRV_PCM_FMTBIT_S8, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 1500, + .rate_max = 4000, + .channels_min = 1, + .channels_max = 1, + .buffer_bytes_max = 20, + .period_bytes_min = 20, + .period_bytes_max = 20, + .periods_min = 1, + .periods_max = 1, +}; + +static int wiimod_speaker_playback_open(struct snd_pcm_substream *substream) +{ + struct wiimote_data *wdata = snd_pcm_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw = wiimod_speaker_playback_hw; + runtime->private_data = wdata; + + return wiimod_speaker_enable(wdata); +} + +static int wiimod_speaker_playback_close(struct snd_pcm_substream *substream) +{ + struct wiimote_data *wdata = snd_pcm_chip(substream); + + wiimod_speaker_disable(wdata); + + return 0; +} + +static int wiimod_speaker_playback_hw_params(struct snd_pcm_substream *subs, + struct snd_pcm_hw_params *hw) +{ + /* TODO: anything to do here? */ + + return 0; +} + +static int wiimod_speaker_playback_hw_free(struct snd_pcm_substream *subs) +{ + return 0; +} + +static int wiimod_speaker_playback_prepare(struct snd_pcm_substream *subs) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + struct snd_pcm_runtime *runtime = subs->runtime; + size_t buf_size, period_size, periods; + int ret; + + /* TODO: If I have set the hw-params to a fixed 20 bytes buffer, do + * I actually need to do any computations here? */ + buf_size = frames_to_bytes(runtime, runtime->buffer_size); + period_size = frames_to_bytes(runtime, runtime->period_size); + periods = runtime->periods; + + switch (runtime->format) { + case SNDRV_PCM_FMTBIT_S8: + ret = wiimod_speaker_setup(wdata, WIIMOD_SPEAKER_MODE_PCM8, + runtime->rate, -1); + break; + default: + return -EINVAL; + } + + return ret; +} + +static int wiimod_speaker_playback_trigger(struct snd_pcm_substream *subs, + int cmd) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_START: + /* unmute device on start if not muted by user-space */ + spin_lock_irqsave(&wdata->state.lock, flags); + if (!speaker->mute) + wiiproto_req_mute(wdata, false); + spin_unlock_irqrestore(&wdata->state.lock, flags); + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + /* mute device when stopping transmission */ + spin_lock_irqsave(&wdata->state.lock, flags); + wiiproto_req_mute(wdata, true); + spin_unlock_irqrestore(&wdata->state.lock, flags); + break; + } + + return 0; +} + +static snd_pcm_uframes_t wiimod_speaker_playback_pointer(struct snd_pcm_substream *subs) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + unsigned long pos; + + /* There is no way to read the current position. Even if there was a + * way to do that, it would be so slow that it would be useless. + * TODO: Should we interpolate via a timer? Is this function actually + * needed? When is it called? */ + pos = 0; + + pos %= 20; + + return bytes_to_frames(subs->runtime, pos); +} + +static int wiimod_speaker_playback_copy(struct snd_pcm_substream *subs, + int channel, snd_pcm_uframes_t pos, + void __user *buf, + snd_pcm_uframes_t count) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + struct snd_pcm_runtime *runtime = subs->runtime; + + count = frames_to_bytes(runtime, count); + pos = frames_to_bytes(runtime, pos); + + /* TODO: copy from "buf" "count" bytes to "protobuf + pos" + * With the given hw-params, can pos be non-zero? Can count != 20? + * If not, should I simply send a 20byte frame to the device here? */ + + return 0; +} + +static int wiimod_speaker_playback_silence(struct snd_pcm_substream *subs, + int channel, snd_pcm_uframes_t pos, + snd_pcm_uframes_t count) +{ + struct wiimote_data *wdata = snd_pcm_chip(subs); + struct snd_pcm_runtime *runtime = subs->runtime; + + count = frames_to_bytes(runtime, count); + pos = frames_to_bytes(runtime, pos); + + /* TODO: set "count" bytes of "protobuf + pos" to zero + * Can "pos" actually be non-zero with the given hw-params? If not, + * can I simply send a 20byte zeroed frame to the remote device? */ + + return 0; +} + +/* TODO: is there a reason this cannot be "const"? */ +static struct snd_pcm_ops wiimod_speaker_playback_ops = { + .open = wiimod_speaker_playback_open, + .close = wiimod_speaker_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = wiimod_speaker_playback_hw_params, + .hw_free = wiimod_speaker_playback_hw_free, + .prepare = wiimod_speaker_playback_prepare, + .trigger = wiimod_speaker_playback_trigger, + .pointer = wiimod_speaker_playback_pointer, + .copy = wiimod_speaker_playback_copy, + .silence = wiimod_speaker_playback_silence, +}; + +/* volume control */ + +static int wiimod_speaker_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *info) +{ + info->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + info->count = 1; + info->value.integer.min = 0; + info->value.integer.max = 0xff; + + return 0; +} + +static int wiimod_speaker_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_data *wdata = snd_kcontrol_chip(kcontrol); + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + val->value.integer.value[0] = speaker->volume; + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return 0; +} + +static int wiimod_speaker_volume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_data *wdata = snd_kcontrol_chip(kcontrol); + unsigned long value; + + value = val->value.integer.value[0]; + if (value > 0xff) + value = 0xff; + + wiimod_speaker_set_volume(wdata, value); + + return 0; +} + +static const struct snd_kcontrol_new wiimod_speaker_volume = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .name = "PCM Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = wiimod_speaker_volume_info, + .get = wiimod_speaker_volume_get, + .put = wiimod_speaker_volume_put, +}; + +/* mute control */ + +static int wiimod_speaker_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_data *wdata = snd_kcontrol_chip(kcontrol); + struct wiimote_speaker *speaker = wdata->speaker; + unsigned long flags; + + spin_lock_irqsave(&wdata->state.lock, flags); + val->value.integer.value[0] = !!speaker->mute; + spin_unlock_irqrestore(&wdata->state.lock, flags); + + return 0; +} + +static int wiimod_speaker_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *val) +{ + struct wiimote_data *wdata = snd_kcontrol_chip(kcontrol); + + wiimod_speaker_set_mute(wdata, val->value.integer.value[0]); + + return 0; +} + +/* TODO: Is *_IFACE_CARD the right interface? */ +static const struct snd_kcontrol_new wiimod_speaker_mute = { + .iface = SNDRV_CTL_ELEM_IFACE_CARD, + .name = "PCM Playback Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_ctl_boolean_mono_info, + .get = wiimod_speaker_mute_get, + .put = wiimod_speaker_mute_put, +}; + +/* initialization and setup */ + +static int wiimod_speaker_probe(const struct wiimod_ops *ops, + struct wiimote_data *wdata) +{ + int ret; + struct wiimote_speaker *speaker; + struct snd_card *card; + struct snd_kcontrol *kcontrol; + struct snd_pcm *pcm; + + /* create sound card device */ + ret = snd_card_create(-1, NULL, THIS_MODULE, + sizeof(struct wiimote_speaker), &card); + if (ret) + return ret; + speaker = card->private_data; + + wdata->speaker = speaker; + speaker->card = card; + speaker->mute = 1; + speaker->volume = 0xff; + strcpy(card->driver, "hid-wiimote"); + strcpy(card->shortname, "wiimote"); + strcpy(card->longname, "Nintendo Wii Remote speaker"); + + /* create volume control */ + kcontrol = snd_ctl_new1(&wiimod_speaker_volume, wdata); + if (!kcontrol) { + ret = -ENOMEM; + goto err_free; + } + + ret = snd_ctl_add(card, kcontrol); + if (ret) { + snd_ctl_free_one(kcontrol); + goto err_free; + } + + /* create mute control */ + kcontrol = snd_ctl_new1(&wiimod_speaker_mute, wdata); + if (!kcontrol) { + ret = -ENOMEM; + goto err_free; + } + + ret = snd_ctl_add(card, kcontrol); + if (ret) { + snd_ctl_free_one(kcontrol); + goto err_free; + } + + /* create PCM sub-device for playback */ + ret = snd_pcm_new(card, "Speaker", 0, 1, 0, &pcm); + if (ret) + goto err_free; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &wiimod_speaker_playback_ops); + pcm->private_data = wdata; + + /* register sound card */ + snd_card_set_dev(card, &wdata->hdev->dev); + ret = snd_card_register(card); + if (ret) + goto err_free; + + return 0; + +err_free: + snd_card_free(card); + wdata->speaker = NULL; + return ret; +} + +static void wiimod_speaker_remove(const struct wiimod_ops *ops, + struct wiimote_data *wdata) +{ + struct wiimote_speaker *speaker = wdata->speaker; + + if (!speaker) + return; + + snd_card_free_when_closed(speaker->card); + wdata->speaker = NULL; +} + +const struct wiimod_ops wiimod_speaker = { + .flags = 0, + .arg = 0, + .probe = wiimod_speaker_probe, + .remove = wiimod_speaker_remove, +}; diff --git a/drivers/hid/hid-wiimote.h b/drivers/hid/hid-wiimote.h index 30caef5..4907bc7 100644 --- a/drivers/hid/hid-wiimote.h +++ b/drivers/hid/hid-wiimote.h @@ -148,6 +148,7 @@ struct wiimote_data { struct timer_list timer; struct wiimote_ext *ext; struct wiimote_debug *debug; + struct wiimote_speaker *speaker; union { struct input_dev *input; @@ -172,6 +173,7 @@ enum wiimod_module { WIIMOD_IR, WIIMOD_BUILTIN_MP, WIIMOD_NO_MP, + WIIMOD_SPEAKER, WIIMOD_NUM, WIIMOD_NULL = WIIMOD_NUM, }; @@ -208,9 +210,12 @@ enum wiiproto_reqs { WIIPROTO_REQ_LED = 0x11, WIIPROTO_REQ_DRM = 0x12, WIIPROTO_REQ_IR1 = 0x13, + WIIPROTO_REQ_SPEAKER = 0x14, WIIPROTO_REQ_SREQ = 0x15, WIIPROTO_REQ_WMEM = 0x16, WIIPROTO_REQ_RMEM = 0x17, + WIIPROTO_REQ_AUDIO = 0x18, + WIIPROTO_REQ_MUTE = 0x19, WIIPROTO_REQ_IR2 = 0x1a, WIIPROTO_REQ_STATUS = 0x20, WIIPROTO_REQ_DATA = 0x21, @@ -241,6 +246,12 @@ extern void wiiproto_req_status(struct wiimote_data *wdata); extern void wiiproto_req_accel(struct wiimote_data *wdata, __u8 accel); extern void wiiproto_req_ir1(struct wiimote_data *wdata, __u8 flags); extern void wiiproto_req_ir2(struct wiimote_data *wdata, __u8 flags); +extern void wiiproto_req_wmem(struct wiimote_data *wdata, bool eeprom, + __u32 offset, const __u8 *buf, __u8 size); +extern void wiiproto_req_speaker(struct wiimote_data *wdata, bool on); +extern void wiiproto_req_mute(struct wiimote_data *wdata, bool on); +extern int wiiproto_req_audio_user(struct wiimote_data *wdata, + void __user *buf, size_t len); extern int wiimote_cmd_write(struct wiimote_data *wdata, __u32 offset, const __u8 *wmem, __u8 size); extern ssize_t wiimote_cmd_read(struct wiimote_data *wdata, __u32 offset, @@ -283,6 +294,12 @@ static inline void wiidebug_deinit(void *u) { } #endif +#ifdef CONFIG_SND + +extern const struct wiimod_ops wiimod_speaker; + +#endif + /* requires the state.lock spinlock to be held */ static inline bool wiimote_cmd_pending(struct wiimote_data *wdata, int cmd, __u32 opt) -- 1.8.2.1

2 6

[alsa-devel] [PATCH] drm/i915: Add private api for power well usage -- alignment between graphic team and audio team
by Li, Jocelyn 03 May '13

03 May '13

Hi Guys, I apologize that I may have misunderstood on the agreement we have made on power well work. I assumed that audio team is expected to create private gfx driver API to control the power well between gfx and audio for internal releases. That's why I asked Xingchao work on that. So the correct implementation approach between gfx and audio should be, #1, Daniel will work on the existing private API in gfx driver for power well control that can be used by the audio driver to enable/disable the power well. #2, Once #1 is done, Audio team will integrate Daniel's patch into audio driver and add/improve on anything that is missing Please correct me if my understanding is not correct. Thanks, Jocelyn -----Original Message----- From: Daniel Vetter [mailto:daniel.vetter@ffwll.ch] Sent: Thursday, April 25, 2013 3:52 PM To: Wang, Xingchao Cc: Zanoni, Paulo R; ville.syrjala(a)linux.intel.com; Lin, Mengdong; Girdwood, Liam R; Li, Jocelyn; intel-gfx(a)lists.freedesktop.org; alsa-devel(a)alsa-project.org; Wang Xingchao; Takashi Iwai; Barnes, Jesse; Wysocki, Rafael J Subject: Re: [PATCH] drm/i915: Add private api for power well usage On Thu, Apr 25, 2013 at 4:36 AM, Wang, Xingchao <xingchao.wang(a)intel.com> wrote: > Hi Daniel/Paulo, > > Any comments on this? > Add Jesse and Rafael in loop. Well I've figured we want to get the power well integration working before we waste time reviewing an interface which might or might not quite work. But if you want, I can drop two quick comments: - I think the functions needs some prefix like i915_ - Imo the power well should just be reference counted (the current interface name suggests it's generally useable, but actually can only cope with one request from the audio driver). - Is the spin_lock (irq-disabling even) really required for the audio driver to work? power well enabling can take up to 20ms, so this won't work as-is. Can't we just use a mutex? - I'd need to see the audio driver side of this to check how this works for coping with arbitrary module load ordering. Cheers, Daniel > > thanks > --xingchao > > >> -----Original Message----- >> From: Wang, Xingchao >> Sent: Wednesday, April 24, 2013 3:29 PM >> To: daniel.vetter(a)ffwll.ch; Zanoni, Paulo R; 'Takashi Iwai' >> Cc: ville.syrjala(a)linux.intel.com; Lin, Mengdong; Girdwood, Liam R; >> Li, Jocelyn; intel-gfx(a)lists.freedesktop.org; alsa-devel(a)alsa-project.org; 'Wang Xingchao' >> Subject: RE: [PATCH] drm/i915: Add private api for power well usage >> >> Hi Guys, >> >> Sorry I should Add Takashi and alsa maillist in loop. >> >> I tested this patch against Daniel's "drm-intel-next-queued" Branch, >> on Haswell Mobile C3 stepping machines. >> >> Test steps: >> 1. only connect one monitor through HDMI cable 2. echo 1 > >> /sys/module/i915/parameters/disable_power_well ==> I think this could >> be set as 1 by default if applied my patch. >> 3. plug out HDMI cable >> 4. cat /proc/asound/card0/codec#0 >> >> At step 3, i915 will try to disable power well but rejected as >> display audio is using power well. >> Without the patch, display audio would crash at step 4. >> >> I did not list suspend/resume test here, as during suspend sequence, >> i915 will _ENABLE_ power well, that's a BUG? >> >> For me above test case is the only one to reproduce the issue, if you >> have more ideas and need more test, please let me know. >> >> I'm afraid there's some risk because the dependency between alsa >> driver and drm driver. For example, if i915 module was not loaded, >> display audio Will call >> request_power_well() and meet symbol error. Even if i915 module >> loaded after snd-hda-intel module, there's such risk too. Any comment on this? >> >> Another risk is the pointer "power_well_device", it should be set >> NULL when >> i915 module unloaded, I added the caller but have no idea where to >> call it, if anyone has suggestion, that would be fine. >> >> thanks >> --xingchao >> >> >> > -----Original Message----- >> > From: Wang Xingchao [mailto:xingchao.wang@linux.intel.com] >> > Sent: Thursday, April 24, 2014 10:20 PM >> > To: daniel.vetter(a)ffwll.ch; Zanoni, Paulo R >> > Cc: ville.syrjala(a)linux.intel.com; Lin, Mengdong; Girdwood, Liam R; >> > Li, Jocelyn; intel-gfx(a)lists.freedesktop.org; Wang, Xingchao; Wang >> > Xingchao >> > Subject: [PATCH] drm/i915: Add private api for power well usage >> > >> > Display audio depends on power well in graphic side, it should >> > request power well before use it and release power well after use. >> > I915 will not shutdown power well if it detects audio is using. >> > This patch protects display audio crash for Intel Haswell mobile >> > C3 stepping board. >> > >> > Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> >> > --- >> > drivers/gpu/drm/i915/intel_pm.c | 73 >> > +++++++++++++++++++++++++++++++++++---- >> > include/drm/audio_drm.h | 39 +++++++++++++++++++++ >> > sound/pci/hda/hda_intel.c | 8 +++++ >> > 3 files changed, 113 insertions(+), 7 deletions(-) create mode >> > 100644 include/drm/audio_drm.h >> > >> > diff --git a/drivers/gpu/drm/i915/intel_pm.c >> > b/drivers/gpu/drm/i915/intel_pm.c index 2557926..9983f56 100644 >> > --- a/drivers/gpu/drm/i915/intel_pm.c >> > +++ b/drivers/gpu/drm/i915/intel_pm.c >> > @@ -4298,18 +4298,12 @@ bool intel_using_power_well(struct >> > drm_device >> > *dev) >> > return true; >> > } >> > >> > -void intel_set_power_well(struct drm_device *dev, bool enable) >> > +void set_power_well(struct drm_device *dev, bool enable) >> > { >> > struct drm_i915_private *dev_priv = dev->dev_private; >> > bool is_enabled, enable_requested; >> > uint32_t tmp; >> > >> > - if (!HAS_POWER_WELL(dev)) >> > - return; >> > - >> > - if (!i915_disable_power_well && !enable) >> > - return; >> > - >> > tmp = I915_READ(HSW_PWR_WELL_DRIVER); >> > is_enabled = tmp & HSW_PWR_WELL_STATE; >> > enable_requested = tmp & HSW_PWR_WELL_ENABLE; @@ -4332,6 >> > +4326,71 @@ void intel_set_power_well(struct drm_device *dev, bool >> > +enable) >> > } >> > } >> > >> > +/* Global drm_device for display audio drvier usage */ static >> > +struct drm_device *power_well_device; static bool >> > +audio_power_well_using, i915_power_well_using; >> > +/* Lock protecting power well setting */ >> > +DEFINE_SPINLOCK(powerwell_lock); >> > + >> > +void request_power_well(void) >> > +{ >> > + DRM_DEBUG_KMS("Display audio requesting power well\n"); >> > + spin_lock_irq(&powerwell_lock); >> > + if (!power_well_device) >> > + goto out; >> > + if (i915_power_well_using == false) >> > + set_power_well(power_well_device, true); >> > + audio_power_well_using = true; >> > +out: >> > + spin_unlock_irq(&powerwell_lock); >> > + return; >> > +} >> > +EXPORT_SYMBOL_GPL(request_power_well); >> > + >> > +void release_power_well(void) >> > +{ >> > + DRM_DEBUG_KMS("Display audio releasing power well\n"); >> > + spin_lock_irq(&powerwell_lock); >> > + if (!power_well_device) >> > + goto out; >> > + if (i915_power_well_using == false) >> > + set_power_well(power_well_device, false); >> > + audio_power_well_using = false; >> > +out: >> > + spin_unlock_irq(&powerwell_lock); >> > + return; >> > +} >> > +EXPORT_SYMBOL_GPL(release_power_well); >> > + >> > +/* TODO: Call this when i915 module unload */ void >> > +remove_power_well(void) { >> > + spin_lock_irq(&powerwell_lock); >> > + audio_power_well_using = false; >> > + i915_power_well_using = false; >> > + power_well_device = NULL; >> > + spin_unlock_irq(&powerwell_lock); } >> > + >> > +void intel_set_power_well(struct drm_device *dev, bool enable) { >> > + if (!HAS_POWER_WELL(dev)) >> > + return; >> > + >> > + spin_lock_irq(&powerwell_lock); >> > + power_well_device = dev; >> > + i915_power_well_using = enable; >> > + if (!enable && audio_power_well_using) >> > + goto out; >> > + >> > + if (!i915_disable_power_well && !enable) >> > + goto out; >> > + set_power_well(dev, enable); >> > +out: >> > + spin_unlock_irq(&powerwell_lock); >> > + return; >> > +} >> > + >> > /* >> > * Starting with Haswell, we have a "Power Down Well" that can be >> > turned >> off >> > * when not needed anymore. We have 4 registers that can request >> > the power well diff --git a/include/drm/audio_drm.h >> > b/include/drm/audio_drm.h new file mode 100644 index >> > 0000000..4412b36 >> > --- /dev/null >> > +++ b/include/drm/audio_drm.h >> > @@ -0,0 +1,39 @@ >> > >> +/************************************************************** >> > ******** >> > +**** >> > + * >> > + * Copyright 2013 Intel Inc. >> > + * All Rights Reserved. >> > + * >> > + * Permission is hereby granted, free of charge, to any person >> > +obtaining a >> > + * copy of this software and associated documentation files (the >> > + * "Software"), to deal in the Software without restriction, >> > +including >> > + * without limitation the rights to use, copy, modify, merge, >> > +publish, >> > + * distribute, sub license, and/or sell copies of the Software, >> > +and to >> > + * permit persons to whom the Software is furnished to do so, >> > +subject to >> > + * the following conditions: >> > + * >> > + * The above copyright notice and this permission notice >> > +(including the >> > + * next paragraph) shall be included in all copies or substantial >> > +portions >> > + * of the Software. >> > + * >> > + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY >> KIND, >> > +EXPRESS OR >> > + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF >> > +MERCHANTABILITY, >> > + * FITNESS FOR A PARTICULAR PURPOSE AND NON-INFRINGEMENT. IN NO >> > EVENT >> > +SHALL >> > + * THE COPYRIGHT HOLDERS, AUTHORS AND/OR ITS SUPPLIERS BE LIABLE >> > FOR >> > +ANY CLAIM, >> > + * DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, >> > TORT >> > +OR >> > + * OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE >> > SOFTWARE >> > +OR THE >> > + * USE OR OTHER DEALINGS IN THE SOFTWARE. >> > + * >> > + * >> > + >> > >> +*************************************************************** >> > ******** >> > +***/ >> > +/* >> > + * Authors: >> > + * Wang Xingchao <xingchao.wang(a)intel.com> */ >> > + >> > +#ifndef _AUDIO_DRM_H_ >> > +#define _AUDIO_DRM_H_ >> > + >> > +extern void request_power_well(void); extern void >> > +release_power_well(void); >> > + >> > +#endif /* _AUDIO_DRM_H_ */ >> > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c >> > index 418bfc0..59a85de 100644 >> > --- a/sound/pci/hda/hda_intel.c >> > +++ b/sound/pci/hda/hda_intel.c >> > @@ -50,6 +50,7 @@ >> > #include <linux/clocksource.h> >> > #include <linux/time.h> >> > #include <linux/completion.h> >> > +#include <drm/audio_drm.h> >> > >> > #ifdef CONFIG_X86 >> > /* for snoop control */ >> > @@ -2869,6 +2870,7 @@ static int azx_suspend(struct device *dev) >> > pci_disable_device(pci); >> > pci_save_state(pci); >> > pci_set_power_state(pci, PCI_D3hot); >> > + release_power_well(); >> > return 0; >> > } >> > >> > @@ -2881,6 +2883,7 @@ static int azx_resume(struct device *dev) >> > if (chip->disabled) >> > return 0; >> > >> > + request_power_well(); >> > pci_set_power_state(pci, PCI_D0); >> > pci_restore_state(pci); >> > if (pci_enable_device(pci) < 0) { @@ -2913,6 +2916,7 @@ static >> > int azx_runtime_suspend(struct device >> > *dev) >> > >> > azx_stop_chip(chip); >> > azx_clear_irq_pending(chip); >> > + release_power_well(); >> > return 0; >> > } >> > >> > @@ -2921,6 +2925,7 @@ static int azx_runtime_resume(struct device *dev) >> > struct snd_card *card = dev_get_drvdata(dev); >> > struct azx *chip = card->private_data; >> > >> > + request_power_well(); >> > azx_init_pci(chip); >> > azx_init_chip(chip, 1); >> > return 0; >> > @@ -3671,6 +3676,8 @@ static int azx_probe(struct pci_dev *pci, >> > return -ENOENT; >> > } >> > >> > + request_power_well(); >> > + >> > err = snd_card_create(index[dev], id[dev], THIS_MODULE, 0, &card); >> > if (err < 0) { >> > snd_printk(KERN_ERR "hda-intel: Error creating >> > card!\n"); @@ >> > -3806,6 +3813,7 @@ static void azx_remove(struct pci_dev *pci) >> > if (card) >> > snd_card_free(card); >> > pci_set_drvdata(pci, NULL); >> > + release_power_well(); >> > } >> > >> > /* PCI IDs */ >> > -- >> > 1.7.9.5 > -- Daniel Vetter Software Engineer, Intel Corporation +41 (0) 79 365 57 48 - http://blog.ffwll.ch

9 23

[alsa-devel] [PATCH -next 0/3] mop500_ab8500 fixes
by Fabio Baltieri 03 May '13

03 May '13

Hi, these are small fixes to the mop500_ab8500 asoc driver, developed on top of today's ASoC-next. The first patch is actually non ab8500 specific, but as ab8500 seems to be the only codec to use the SND_SOC_DAPM_CLOCK_SUPPLY widget I'm posting it here anyway. The other two patches sort out a mixup of card and codec controls in mop500_ab8500. Thanks, Fabio Fabio Baltieri (3): ASoC: dapm: use clk_prepare_enable and clk_disable_unprepare ASoC: ux500: move clock controls to ab8500-codec ASoC: ux500: register controls to card instead of codec sound/soc/codecs/ab8500-codec.c | 10 ++++++++++ sound/soc/soc-dapm.c | 4 ++-- sound/soc/ux500/mop500_ab8500.c | 20 +++++--------------- 3 files changed, 17 insertions(+), 17 deletions(-) -- 1.8.2

3 11

[alsa-devel] [PATCH] ASoC: Added the condition for the channels_min of cpu_dai
by JongHo Kim 02 May '13

02 May '13

Only playback device, the pcm substream for capture does not need to be generated. If codec dai driver has both playback.channels_min and capture.channels_min, and cpu dai driver has only playback.channels_min, soc_new_pcm() will make the pcm substream for capture although DataIN is not connected. So added the condition for the channels_min of cpu_dai in soc_new_pcm(). Signed-off-by: JongHo Kim <furmuwon(a)gmail.com> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index cf191e6..c405e39 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2018,9 +2018,11 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (cpu_dai->driver->capture.channels_min) capture = 1; } else { - if (codec_dai->driver->playback.channels_min) + if (codec_dai->driver->playback.channels_min && + cpu_dai->driver->playback.channels_min) playback = 1; - if (codec_dai->driver->capture.channels_min) + if (codec_dai->driver->capture.channels_min && + cpu_dai->driver->capture.channels_min) capture = 1; } -- 1.8.1.1

2 4

[alsa-devel] [PATCH v2 1/4] ASoC: imx-sgtl5000: Do not enter the error path on success
by Fabio Estevam 02 May '13

02 May '13

From: Fabio Estevam <fabio.estevam(a)freescale.com> Return on success instead of entering the error path. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Keep of_node_put in the success path. sound/soc/fsl/imx-sgtl5000.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 9584e78..5a6aaa3 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -174,6 +174,11 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) } platform_set_drvdata(pdev, data); + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; + clk_fail: clk_put(data->codec_clk); fail: -- 1.7.9.5

3 12

[alsa-devel] [PATCH] MAINTAINERS: Add linux-samsung-soc list to all related entries
by Tomasz Figa 01 May '13

01 May '13

Several entries in MAINTAINERS file related to Samsung SoCs do not point to linux-samsung-soc mailing list, which is supposed to collect all Samsung SoC-related threads, to ease following of Samsung SoC-related work. This leads to a problem with people skipping this mailing list in their posts, even though they are related to Samsung SoCs. This patch adds pointers to linux-samsung-soc mailing list to affected entries of MAINTAINERS file. Signed-off-by: Tomasz Figa <tomasz.figa(a)gmail.com> --- MAINTAINERS | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/MAINTAINERS b/MAINTAINERS index 6c0d68b..07cb8da 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1139,6 +1139,7 @@ F: arch/arm/mach-exynos*/ ARM/SAMSUNG MOBILE MACHINE SUPPORT M: Kyungmin Park <kyungmin.park(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: arch/arm/mach-s5pv210/mach-aquila.c F: arch/arm/mach-s5pv210/mach-goni.c @@ -1150,6 +1151,7 @@ M: Kyungmin Park <kyungmin.park(a)samsung.com> M: Kamil Debski <k.debski(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org L: linux-media(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/media/platform/s5p-g2d/ @@ -1158,6 +1160,7 @@ M: Kyungmin Park <kyungmin.park(a)samsung.com> M: Sylwester Nawrocki <s.nawrocki(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org L: linux-media(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: arch/arm/plat-samsung/include/plat/*fimc* F: drivers/media/platform/s5p-fimc/ @@ -1168,6 +1171,7 @@ M: Kamil Debski <k.debski(a)samsung.com> M: Jeongtae Park <jtp.park(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org L: linux-media(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: arch/arm/plat-samsung/s5p-dev-mfc.c F: drivers/media/platform/s5p-mfc/ @@ -1177,6 +1181,7 @@ M: Kyungmin Park <kyungmin.park(a)samsung.com> M: Tomasz Stanislawski <t.stanislaws(a)samsung.com> L: linux-arm-kernel(a)lists.infradead.org L: linux-media(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/media/platform/s5p-tv/ @@ -2679,6 +2684,7 @@ M: Joonyoung Shim <jy0922.shim(a)samsung.com> M: Seung-Woo Kim <sw0312.kim(a)samsung.com> M: Kyungmin Park <kyungmin.park(a)samsung.com> L: dri-devel(a)lists.freedesktop.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) T: git git://git.kernel.org/pub/scm/linux/kernel/git/daeinki/drm-exynos.git S: Supported F: drivers/gpu/drm/exynos @@ -3142,6 +3148,7 @@ F: Documentation/extcon/ EXYNOS DP DRIVER M: Jingoo Han <jg1.han(a)samsung.com> L: linux-fbdev(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/video/exynos/exynos_dp* F: include/video/exynos_dp* @@ -3151,6 +3158,7 @@ M: Inki Dae <inki.dae(a)samsung.com> M: Donghwa Lee <dh09.lee(a)samsung.com> M: Kyungmin Park <kyungmin.park(a)samsung.com> L: linux-fbdev(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/video/exynos/exynos_mipi* F: include/video/exynos_mipi* @@ -6892,12 +6900,14 @@ F: drivers/platform/x86/samsung-laptop.c SAMSUNG AUDIO (ASoC) DRIVERS M: Sangbeom Kim <sbkim73(a)samsung.com> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Supported F: sound/soc/samsung SAMSUNG FRAMEBUFFER DRIVER M: Jingoo Han <jg1.han(a)samsung.com> L: linux-fbdev(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/video/s3c-fb.c @@ -7087,6 +7097,7 @@ F: drivers/mmc/host/sdhci-pltfm.[ch] SECURE DIGITAL HOST CONTROLLER INTERFACE (SDHCI) SAMSUNG DRIVER M: Ben Dooks <ben-linux(a)fluff.org> L: linux-mmc(a)vger.kernel.org +L: linux-samsung-soc(a)vger.kernel.org (moderated for non-subscribers) S: Maintained F: drivers/mmc/host/sdhci-s3c.c -- 1.8.2.1

4 3

[alsa-devel] [PATCH] ASoC: McASP: Add pins output direction for rx clocks when configured in CBS_CFS format.
by Marek Belisko 30 Apr '13

30 Apr '13

When McASP is bit clock and frame clock master enable pin output for rx clocks. Signed-off-by: Marek Belisko <marek.belisko(a)streamunlimited.com> --- sound/soc/davinci/davinci-mcasp.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index d54e0d9..2953c55 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -505,7 +505,10 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); - mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, ACLKX | AFSX); + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, + ACLKX | ACLKR); + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, + AFSX | AFSR); break; case SND_SOC_DAIFMT_CBM_CFS: /* codec is clock master and frame slave */ -- 1.7.9.5

2 1

[alsa-devel] [PATCH v2 0/3] ASoC: Add devicetree support for spdif dummy codecs.
by Marek Belisko 30 Apr '13

30 Apr '13

changes from v1 (for patch 2 and 3): - add prefix linux for compatible property (for both tx and rx) - rename spdif_transceiver.c to spdif_transmitter.c - add bindings documentation Marek Belisko (3): ASoC: spdif_transceiver: Change driver filename to spdif_transmitter.c. ASoC: spdif_transmitter: Add DT support. ASoC: spdif_receiver: Add DT support. .../devicetree/bindings/sound/spdif-receiver.txt | 10 ++++++++++ .../bindings/sound/spdif-transmitter.txt | 10 ++++++++++ sound/soc/codecs/Makefile | 2 +- sound/soc/codecs/spdif_receiver.c | 10 ++++++++++ .../{spdif_transciever.c => spdif_transmitter.c} | 10 ++++++++++ 5 files changed, 41 insertions(+), 1 deletion(-) create mode 100644 Documentation/devicetree/bindings/sound/spdif-receiver.txt create mode 100644 Documentation/devicetree/bindings/sound/spdif-transmitter.txt rename sound/soc/codecs/{spdif_transciever.c => spdif_transmitter.c} (88%) -- 1.7.9.5

4 7

Re: [alsa-devel] linux-next: Tree for Apr 30 (sound)
by Randy Dunlap 30 Apr '13

30 Apr '13

On 04/29/13 23:57, Stephen Rothwell wrote: > Hi all, > > Please do not add any v3.11 destined work to your linux-next included > branches until after v3.10-rc1 is released. > > Changes since 20130429: > when CONFIG_PROC_FS is not enabled: sound/core/memalloc.c: In function 'snd_mem_exit': sound/core/memalloc.c:525:20: error: 'SND_MEM_PROC_FILE' undeclared (first use in this function) -- ~Randy

1 0

[alsa-devel] [patch] ASoC: wm8994: missing break in wm8994_aif3_hw_params()
by Dan Carpenter 30 Apr '13

30 Apr '13

The missing break here means that we always return early and the function is a no-op. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> --- diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 14094f5..1eb152c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2882,6 +2882,7 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream, default: return 0; } + break; default: return 0; }

3 3

Re: [alsa-devel] what is the difference between SNDRV_PCM_FMTBIT_S24_LE and SNDRV_PCM_FMTBIT_S24_3LE
by Clemens Ladisch 30 Apr '13

30 Apr '13

(please do not drop the mailing list from replies.) Zhang wei wrote: > I do not really understand. For example,an codec is wm8985 ... The SNDRV_PCM_* symbols specify the format in which samples are stored in memory. I do not know how ASoC maps these symbols to on-the-wire formats. >>Most devices that support 24 bits use SNDRV_PCM_FORMAT_S32_LE, which >>stores the sample's bits in the upper bits of a 32-bit word. > > But how to distinguish the SNDRV_PCM_FORMAT_S24_LE and the REAL > SNDRV_PCM_FORMAT_S32_LE? If a 24-bit sample with the value 0x123456 is stored in memory, it looks like this: SNDRV_PCM_FORMAT_S24_LE: 56 34 12 00 SNDRV_PCM_FORMAT_S32_LE: 00 56 34 12 Regards, Clemens

3 2

[alsa-devel] [PATCH, for-next] ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
by Eldad Zack 30 Apr '13

30 Apr '13

Current code does this: be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]) Which is effectively (neglecting the index): be16_to_cpu(be16_to_cpu(*((u16 *) buf))) This means the int16 in the buffer is not converted at all. Daniel Mack confirmed that the driver works on little endian CPUs, leading to the conclusion that the device-side structure is actually little endian. This changes the code to use le16_to_cpu(). Caught by sparse. Acked-by: Daniel Mack <zonque(a)gmail.com> Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/caiaq/input.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index efc70ae..4b3fb91 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -488,13 +488,13 @@ static void snd_usb_caiaq_maschine_dispatch(struct snd_usb_caiaqdev *cdev, unsigned int len) { unsigned int i, pad_id; - uint16_t pressure; + __le16 *pressure = (__le16 *) buf; for (i = 0; i < MASCHINE_PADS; i++) { - pressure = be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]); - pad_id = pressure >> 12; - - input_report_abs(cdev->input_dev, MASCHINE_PAD(pad_id), pressure & 0xfff); + pad_id = le16_to_cpu(*pressure) >> 12; + input_report_abs(cdev->input_dev, MASCHINE_PAD(pad_id), + le16_to_cpu(*pressure) & 0xfff); + pressure++; } input_sync(cdev->input_dev); -- 1.8.1.5

3 2

[alsa-devel] ASoC : playback and capture counts in soc_new_pcm
by JongHo Kim 29 Apr '13

29 Apr '13

Hi I am developing the ALSA ASoC part. I has some problem. Now my situation must only use the cpu_dai(AP I2S IP) to half duplex. So, I want to allocate either a capture substream or a playback substream. But ASoC(soc-pcm.c) referred to the playback(capture)_channel_min of codec_dai at the calling snd_pcm_new in soc_new_pcm. I do not want to modify the codec_dai codes. As a result, hope to see below /dev/snd/pcmC0D0p /dev/snd/pcmC0D1c attach the ASCII schematic diagram. Slaves Master +--------------+ +-----------------+ |I2S.0 data_out|---------------|data_in(AIF1) | |I2S.0 bclk |-------+-------|bclk | |I2S.0 lrclk |-------|--+----|lrclk | +--------------+ | | | | +--------------+ | | | AUDIO CODEC | |I2S.1 bclk |-------+ | | | |I2S.1 lrclk |----------+ | | |I2S.1 data_in |---------------|data_out(AIF1) | +--------------+ +-----------------+ attach the i2s and dai_link codes. ----------------------------------------------------------------------------- static struct snd_soc_dai_driver board_i2s_dai_hf_playback_template = { .probe = board_i2s_probe, .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &board_i2s_dai_ops, }; static struct snd_soc_dai_driver board_i2s_dai_hf_capture_template = { .probe = board_i2s_probe, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &board_i2s_dai_ops, }; static struct snd_soc_dai_link board_dai[] = { { .name = "audcodec_l0", .stream_name = "audcodec HiFi", .cpu_dai_name = "board-i2s.0", .codec_dai_name = "audcodec-aif1", .platform_name = "board-pcm-audio", .codec_name = "audcodec-codec", .ops = &board_ops, }, { .name = "audcodec_l1", .stream_name = "audcodec HiFi", .cpu_dai_name = "board-i2s.1", .codec_dai_name = "audcodec-aif1", .platform_name = "board-pcm-audio", .codec_name = "audcodec-codec", .ops = &board_ops, }, }; ----------------------------------------------------------------------------- So, I suggest the next diff. --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2014,9 +2014,11 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (cpu_dai->driver->capture.channels_min) capture = 1; } else { - if (codec_dai->driver->playback.channels_min) + if (codec_dai->driver->playback.channels_min && + cpu_dai->driver->playback.channels_min) playback = 1; - if (codec_dai->driver->capture.channels_min) + if (codec_dai->driver->playback.channels_min && + cpu_dai->driver->playback.channels_min) capture = 1; } Thanks.

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[alsa-devel] ALSA SOC Driver (mxs-wm8753)
by John Graham 29 Apr '13

29 Apr '13

Hi there, I'm completely new to both kernel/ALSA driver programming (at least at a serious level) and was looking for some help on a project I'm working on - I'm part way there but I've kind of hit a brick wall and don't know where to turn. I'm trying to get a codec working on a new platform (new for the codec - the platform itself is not new). Here are the details: The audio codec is the Wolfson WM8753, which already has a codec driver in the kernel mainline tree. We also have a vendor-supplied (but GPL) patch to make the wm8753 work on a CC9M2443 (based on the Samsung S3C2443). This includes the glue code ("platform" code?) in sound/soc/s3c24xx/cc9m2443js_wm8753.c. We have a product with this combination fully working. The platform is the Karo TX28 (based on the Freescale i.MX28), and there are already drivers in the kernel tree to make the TX28 work with an SGTL5000 codec in sound/soc/mxs/mxs-sgtl5000.c. We also have a product with this combination fully working. Now we want to move the WM8753 over from the CC9M2443 to work on the TX28. To simplify, what I *have* is: CC9M2443 <----> WM8753 TX28 <----> SGTL5000 and what I *want* is: /-> WM8753 TX28 <-/ What I was hoping I could do was just "mash" the existing drivers together - i.e. copy sound/soc/mxs/mxs-sgtl5000.c into sound/soc/mxs/mxs-wm8753.c, change the parts in it that were specific for the SGTL5000 to the parts that drive the wm8753 in sound/soc/s3c24xx/cc9m2443js_wm8753.c, tell the TX28 to probe for it and that it would "magically work" (hey, crazier things have happened...). Obviously, it does not work, but I've got it a little way down the line. The WM8752 is probed for by the TX28 and recognized - the I2C control works, and you can change the mixer settings with amixer. When I try to record (using arecord) I can see with an oscilloscope the I2S clock lines look correct (in that they look the same as when the CC9M2443 talks to the WM8753 at the same audio rate). The problem now is that the I2S data lines don't look right - the line from codec to TX28 looks like a (more-or-less) square wave with the same period as the I2S clock, but only going from ~3.1 - ~3.5 V, as opposed to the 0 - ~3.3 V I expect. Furthermore, it's a pretty regular square wave - it doesn't have all the breaks you see on a working I2S data line. I know I probably haven't provided the right detail to get to the bottom of this problem, but part of my problem is that I'm so new to this that I'm not entirely sure where the problem might lie, or what extra detail to include to ask for more help. Is anyone able to point me in the right direction/suggest some ideas as to what might be wrong/anything else at all? Any help would be much appreciated - many thanks, John Graham

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[alsa-devel] Configure a CODEC which has I2C connected to CPU, but no PCM bus connection
by Craig McQueen 29 Apr '13

29 Apr '13

I have an embedded device with somewhat unusual configuration: there is a second audio CODEC (Wolfson WM8940) which is connected to the PCM interface of another device (a radio module, which is acting as PCM bus master), so that radio device can do analog audio I/O. So there is no CPU<-->CODEC PCM link, but the CODEC _is_ connected to the CPU's I2C, and the CPU needs to configure the CODEC registers to interface to the radio module. See the attached diagram which shows a simplified view of the configuration. This configuration came about because both the CPU and the radio module can _only_ operate as PCM bus masters. I am not sure how I could fit this into the ALSA ASoC architecture. Is the ALSA ASoC driver architecture able to accommodate this scenario? Can I use the WM8940 driver to configure the second CODEC through I2C? Could I simply set up a machine driver that defines a DAI including a pointer to the WM8940 CODEC DAI, but NULL pointer for CPU DAI? Can I set it up so the Wolfson CODEC gains could appear as ALSA mixer widgets? Regards, Craig McQueen

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[alsa-devel] [PATCH] ASoC: wm8962: Convert to devm_input_allocate_device()
by Mark Brown 29 Apr '13

29 Apr '13

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 50b0983..1cee9b3 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3189,7 +3189,7 @@ static void wm8962_init_beep(struct snd_soc_codec *codec) struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); int ret; - wm8962->beep = input_allocate_device(); + wm8962->beep = devm_input_allocate_device(codec->dev); if (!wm8962->beep) { dev_err(codec->dev, "Failed to allocate beep device\n"); return; @@ -3210,7 +3210,6 @@ static void wm8962_init_beep(struct snd_soc_codec *codec) ret = input_register_device(wm8962->beep); if (ret != 0) { - input_free_device(wm8962->beep); wm8962->beep = NULL; dev_err(codec->dev, "Failed to register beep device\n"); } @@ -3227,7 +3226,6 @@ static void wm8962_free_beep(struct snd_soc_codec *codec) struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); device_remove_file(codec->dev, &dev_attr_beep); - input_unregister_device(wm8962->beep); cancel_work_sync(&wm8962->beep_work); wm8962->beep = NULL; -- 1.7.10.4

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[alsa-devel] ALSA: usb-audio: caiaq: endianness bug
by Eldad Zack 29 Apr '13

29 Apr '13

Hi Daniel, I think I found a endian usage issue in the caiaq driver. I have a patch to fix it (below), but it assumes that the driver works currently on little endian CPUs - I don't have a device to test it with. Cheers, Eldad -- >8 -- Subject: [PATCH] ALSA: usb-audio: caiaq: fix endianness bug Current code does this: be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]) Which is effectively: be16_to_cpu(be16_to_cpu(*((u16 *) buf))) This means the int16 in the buffer is not converted at all. Assuming the device-side structure is actually little endian, change the code to use le16_to_cpu(). Signed-off-by: Eldad Zack <eldad(a)fogrefinery.com> --- sound/usb/caiaq/input.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/usb/caiaq/input.c b/sound/usb/caiaq/input.c index efc70ae..2f4ae4d 100644 --- a/sound/usb/caiaq/input.c +++ b/sound/usb/caiaq/input.c @@ -488,13 +488,12 @@ static void snd_usb_caiaq_maschine_dispatch(struct snd_usb_caiaqdev *cdev, unsigned int len) { unsigned int i, pad_id; - uint16_t pressure; + __le16 *pressure = (__le16 *) buf; for (i = 0; i < MASCHINE_PADS; i++) { - pressure = be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1]); - pad_id = pressure >> 12; - - input_report_abs(cdev->input_dev, MASCHINE_PAD(pad_id), pressure & 0xfff); + pad_id = le16_to_cpu(*pressure) >> 12; + input_report_abs(cdev->input_dev, MASCHINE_PAD(pad_id), le16_to_cpu(*pressure) & 0xfff); + pressure++; } input_sync(cdev->input_dev); -- 1.8.1.5

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Re: [alsa-devel] linux-next: Tree for Apr 29 (sound/soc)
by Randy Dunlap 29 Apr '13

29 Apr '13

On 04/29/13 02:17, Stephen Rothwell wrote: > Hi all, > > Changes since 20130426: > on i386: CONFIG_I2C=m sound/built-in.o: In function `snd_soc_codec_set_cache_io': (.text+0x1ebe4): undefined reference to `regmap_init_i2c' make[1]: *** [vmlinux] Error 1 -- ~Randy

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[alsa-devel] [PATCH 0/3] snd-firewire-lib: add handling CMP output connection
by Takashi Sakamoto 29 Apr '13

29 Apr '13

Current implementation of snd-firewire-lib can handle only CMP input connection. This series of patch enable it to handle both CMP output and input connection according to IEC 61883-1. Note that this series of patch still supports only a point-to-point connection without overlaying. The overlaying and any broadcast connection are not supported. Each connections gain its own isochronous resource. Takashi Sakamoto (3): rename macros, variables and functions related to CMP plug Add "direction" member to cmp_connection structure Add handling CMP output connection sound/firewire/cmp.c | 192 +++++++++++++++++++++++++++++++++------------ sound/firewire/cmp.h | 13 ++- sound/firewire/speakers.c | 2 +- 3 files changed, 152 insertions(+), 55 deletions(-) -- 1.7.10.4

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[alsa-devel] Missing patches for 3.10 merge?
by Takashi Iwai 29 Apr '13

29 Apr '13

Hi, if anyone posted fix patches that should be applied to 3.10, but still unapplied, please let me know ASAP. I got a complete home (and system) FS crash on Friday (due to too much experiment with btrfs) and might have lost some incoming mails. thanks, Takashi

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[alsa-devel] [PATCH] ALSA: compress: fix the states to check for allowing read
by Vinod Koul 29 Apr '13

29 Apr '13

for reading compressed data, we need to allow when we are paused, draining or stopped. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Cc: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Cc: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 10 ++++++++-- 1 files changed, 8 insertions(+), 2 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index a0bc47f..5389b9a 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -311,8 +311,14 @@ static ssize_t snd_compr_read(struct file *f, char __user *buf, stream = &data->stream; mutex_lock(&stream->device->lock); - /* read is allowed when stream is running */ - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { + /* read is allowed when stream is running, paused, draining and setup + * (yes setup is state which we transistion to after stop, so if user + * wants to read data after stop we allow that + */ + if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING || + stream->runtime->state != SNDRV_PCM_STATE_DRAINING || + stream->runtime->state != SNDRV_PCM_STATE_PAUSED || + stream->runtime->state != SNDRV_PCM_STATE_SETUP) { retval = -EBADFD; goto out; } -- 1.7.0.4

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[alsa-devel] [PATCH] compress: fix the states to check for allowing read
by Vinod Koul 29 Apr '13

29 Apr '13

for reading compressed data, we need to allow when we are paused, draining or stopped. Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Cc: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> Cc: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> --- sound/core/compress_offload.c | 12 ++++++++++-- 1 files changed, 10 insertions(+), 2 deletions(-) diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index a0bc47f..7b774b7 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -311,8 +311,16 @@ static ssize_t snd_compr_read(struct file *f, char __user *buf, stream = &data->stream; mutex_lock(&stream->device->lock); - /* read is allowed when stream is running */ - if (stream->runtime->state != SNDRV_PCM_STATE_RUNNING) { + /* read is allowed when stream is running, paused, draining and setup + * (yes setup is state which we transition to after stop, so if user + * wants to read data after stop we allow that + */ + switch (stream->runtime->state) { + case SNDRV_PCM_STATE_OPEN: + case SNDRV_PCM_STATE_PREPARED: + case SNDRV_PCM_STATE_XRUN: + case SNDRV_PCM_STATE_SUSPENDED: + case SNDRV_PCM_STATE_DISCONNECTED: retval = -EBADFD; goto out; } -- 1.7.0.4

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