July 2011 - 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] icon utrack pro usb audio interface
by Adrien Plisson 30 Aug '14

30 Aug '14

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

3 3

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

11 Jul '14

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

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[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

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[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

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[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; }

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

15 Apr '13

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

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

23 Dec '12

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

4 26

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

19 Oct '12

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

2 2

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

09 Oct '12

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

3 2

[alsa-devel] [PATCH] snd-usb-us122l: Fix missing NULL checks
by Karsten Wiese 27 Apr '12

27 Apr '12

Fix missing NULL checks in usb_stream_hwdep_poll() and usb_stream_hwdep_ioctl(). Wake up poll waiters before returning from usb_stream_hwdep_ioctl(). Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> Cc: stable(a)kernel.org --- sound/usb/usx2y/us122l.c | 41 ++++++++++++++++++++--------------------- 1 files changed, 20 insertions(+), 21 deletions(-) diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index 91bb296..7abda8d 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -269,29 +269,26 @@ static unsigned int usb_stream_hwdep_poll(struct snd_hwdep *hw, struct file *file, poll_table *wait) { struct us122l *us122l = hw->private_data; - struct usb_stream *s = us122l->sk.s; unsigned *polled; unsigned int mask; poll_wait(file, &us122l->sk.sleep, wait); - switch (s->state) { - case usb_stream_ready: - if (us122l->first == file) - polled = &s->periods_polled; - else - polled = &us122l->second_periods_polled; - if (*polled != s->periods_done) { - *polled = s->periods_done; - mask = POLLIN | POLLOUT | POLLWRNORM; - break; + mask = POLLIN | POLLOUT | POLLWRNORM | POLLERR; + if (mutex_trylock(&us122l->mutex)) { + struct usb_stream *s = us122l->sk.s; + if (s && s->state == usb_stream_ready) { + if (us122l->first == file) + polled = &s->periods_polled; + else + polled = &us122l->second_periods_polled; + if (*polled != s->periods_done) { + *polled = s->periods_done; + mask = POLLIN | POLLOUT | POLLWRNORM; + } else + mask = 0; } - /* Fall through */ - mask = 0; - break; - default: - mask = POLLIN | POLLOUT | POLLWRNORM | POLLERR; - break; + mutex_unlock(&us122l->mutex); } return mask; } @@ -377,6 +374,7 @@ static int usb_stream_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, { struct usb_stream_config *cfg; struct us122l *us122l = hw->private_data; + struct usb_stream *s; unsigned min_period_frames; int err = 0; bool high_speed; @@ -422,18 +420,18 @@ static int usb_stream_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, snd_power_wait(hw->card, SNDRV_CTL_POWER_D0); mutex_lock(&us122l->mutex); + s = us122l->sk.s; if (!us122l->master) us122l->master = file; else if (us122l->master != file) { - if (memcmp(cfg, &us122l->sk.s->cfg, sizeof(*cfg))) { + if (!s || memcmp(cfg, &s->cfg, sizeof(*cfg))) { err = -EIO; goto unlock; } us122l->slave = file; } - if (!us122l->sk.s || - memcmp(cfg, &us122l->sk.s->cfg, sizeof(*cfg)) || - us122l->sk.s->state == usb_stream_xrun) { + if (!s || memcmp(cfg, &s->cfg, sizeof(*cfg)) || + s->state == usb_stream_xrun) { us122l_stop(us122l); if (!us122l_start(us122l, cfg->sample_rate, cfg->period_frames)) err = -EIO; @@ -444,6 +442,7 @@ unlock: mutex_unlock(&us122l->mutex); free: kfree(cfg); + wake_up_all(&us122l->sk.sleep); return err; } -- 1.7.2.3

2 5

[alsa-devel] [PATCH] ALSA: HDA: Remove unconnected PCM devices for Intel HDMI
by David Henningsson 01 Feb '12

01 Feb '12

Since I'm a little new in HDMI land and the patch is non-trivial, I wouldn't mind some comments on this one. Also thanks to Jaroslav who cleared a few things out about the logical and physical devices which helped me figure out what I should do about it :-) Anyway, here's the patch. Some newer chips have more than one HDMI output, but usually not all of them are exposed as physical jacks. Removing the unused PCM devices (as indicated by BIOS in the pin config default) will reduce user confusion as they currently have to choose between several HDMI devices, some of them not working anyway. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

4 14

Re: [alsa-devel] [patch][saa7134] do not change mute state for capturing audio
by Mauro Carvalho Chehab 03 Dec '11

03 Dec '11

Em 24-07-2011 14:45, Stas Sergeev escreveu: > 23.07.2011 19:09, Mauro Carvalho Chehab wrote: >> > In this case, it will not be autounmuted after tuning. >> Hard to tell about your solution without seeing a patch. > OK, it turns out the automute code is already there, > but it doesn't work. The driver for some reasons > starts the scan on initialization, finds the carrier: > --- > saa7134[0]/audio: found PAL main sound carrier @ 6.000 MHz [3969/324] > --- > and, because of that, disables the automute. If the > real mute is not enabled at that point, you get the > white noise right away. The automute code works fine. Maybe you have a strong interference at the default tuning frequency, leading into saa7134 miss-detection. For saa7134 driver: 1) There is an audio carrier; 2) an application wants to listen audio; 3) the device is at automute mode. So, when there's an audio carrier, it will unmute the audio at streaming start. The driver is doing the _right_ thing by letting the audio to flow to PA, when it starts the capture. Unfortunately, on your specific device, starting audio capture also enables audio at the audio output pin. So, you're noticing a problem. People with a saa7134 device without alsa stream won't notice it (old devices). People with alsa stream without anything connected to the LINE OUT people aren't noticing it, if PA is not copying the saa7134 PCM IN stream to the sound card PCM out device (the default, for PA). So, only people that has saa7134 with alsa stream that opted to wire the saa7134 device to the sound card, and with a strong interference at the default tunning frequency (400 MHz) would notice a problem. > Since I have no idea why it finds some carrier, I can't > fix that in any way. Or, maybe, not to call the scan > on driver init? What will that break? Analog tuners need to be tuned at the device init on a high frequency according with their datasheets, otherwise the PLL may fail. > Anyway, as long as the automute code is broken, > we should either start fixing it, or fix PA, or fix mplayer... As I always said, the fix should be at PA, as it makes no sense to start capturing at saa7134 without first configuring it. So, or PA should be converted into a V4L2-aware application, in order to properly init the device (with seems to be the wrong approach) or it _SHOULD_NOT_ automatically enable capture on those devices, as this may cause undesired side effects, on the devices that have the capture pin directly wired to the output pin, witch seems to be the case of your device. PS.: it seems that you've removed Lennart/alsa people from the c/c. I'm re-adding them, as I'm expecting a fix from their side. > Dunno. I wonder how come so many bugs left unfixed > for so long, resulting in a white noise to people... You're the first one that reported it, and the code is there for _years_. So, this is not a commonly noticed problem at all. Mauro.

2 15

Re: [alsa-devel] max channels of 3 jacks 6 channels motherboard using model=auto realtek codec
by Takashi Iwai 20 Oct '11

20 Oct '11

At Sun, 10 Jul 2011 08:47:46 +0200, Takashi Iwai wrote: > > At Sun, 10 Jul 2011 11:31:44 +0800, > Raymond Yau wrote: > > > > 2011/7/6 Takashi Iwai <tiwai(a)suse.de>: > > > At Wed, 6 Jul 2011 11:53:47 +0800, > > > Raymond Yau wrote: > > >> > > >> Refer to commit ce764ab22e40a046065c9417ee5f04ad2a816ac1 > > >> > > >> ALSA: hda - Add channel-mode support to Realtek auto-parser > > >> > > >> It seem that max channel is six instead of two when the channel mode > > >> is 2ch for those 3 jacks motherboard > > >> > > >> using hda-emu and alsa-info from > > >> http://thread.gmane.org/gmane.linux.alsa.devel/86816 > > >> > > >> > PCM 0 p 44100 2 16 > > >> Open PCM ALC887 Analog for play > > >> Available PCM parameters: > > >> channels: 2/6 > > >> formats: S16_LE S32_LE > > >> rates: 44100 48000 96000 192000 > > >> Prepare PCM, rate=44100, channels=2, format=16 bits > > >> > > >> > > >> because > > >> > > >> spec->multiout.max_channels = spec->multiout.num_dacs * 2; > > >> > > >> is assigned after alc_auto_add_multi_channel_mode() increase > > >> spec->multiout.num_dacs from 1 to 3 > > >> > > >> spec->multiout.num_dacs = num_pins + 1; > > > > > > Right, and this is intentional. Since there should be enough amount > > > of DACs, the driver keeps using the surround DACs statically. > > > In that way, apps can keep opening the device in the surround mode > > > while user can switch the outputs on the fly via channel-mode mixer > > > enum. > > > > > > > > > > How can the driver support 6 channels when the input jacks have not > > been retasked as output ? > > Even if the pins aren't retasked, DACs can receive data. And then > you can change the pins and routing on the fly. > > You can open a two-channel PCM stream even if the master or (any > other) volume is muted. A similar reason. > > > This behaviour seem different from model=3stack-6ch or > > model=3stack-6ch-dig or those via codec with smart51 "OFF" The smart51 case is a bit different (we prefer DAC unused because of independent HP control), but the 3stack-6ch case is a regression, indeed. This was just because of dropped code snippet. Now I fixed with the patch below. thanks, Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Re-add need_dac_fix check for multi-io jacks of Realtek codecs During the rewrite, the check of spec->need_dac_fix and the corresponding num_dacs change was dropped from the channel-mode control. This patch re-adds it, and also enables need_dac_fix for ALC880 as default, as this feature was originally introduced to fix h/w bugs of this chip. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 124c63f..52ce075 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3196,6 +3196,8 @@ static int alc_auto_ch_mode_put(struct snd_kcontrol *kcontrol, for (i = 0; i < spec->multi_ios; i++) alc_set_multi_io(codec, i, i < ch); spec->multiout.max_channels = spec->ext_channel_count; + if (spec->need_dac_fix && !spec->const_channel_count) + spec->multiout.num_dacs = spec->multiout.max_channels / 2; return 1; } @@ -3642,6 +3644,7 @@ static int patch_alc880(struct hda_codec *codec) codec->spec = spec; spec->mixer_nid = 0x0b; + spec->need_dac_fix = 1; board_config = alc_board_config(codec, ALC880_MODEL_LAST, alc880_models, alc880_cfg_tbl); -- 1.7.6

2 1

[alsa-devel] [PATCH] ASoC: WM8903: Implement DMIC support
by Stephen Warren 14 Oct '11

14 Oct '11

In addition to the currently supported analog capture path, the WM8903 also supports digital mics. The analog and digital capture paths are exclusive; a mux is present to select the capture source. Logically, the mux exists to select the decimator's input, from either the ADC or DMIC block outputs. However, the ADC power domain also includes the DMIC interface. Consequently, this change represents the mux as existing immediately before the ADC, and selecting between the Input PGA and DMIC block outputs. An alternative might be to represent the mux in its correct location, and associate the ADC power enable controls with both the real ADC, and a fake ADC for the DMIC? Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/codecs/wm8903.c | 23 +++++++++++++++++++++-- 1 files changed, 21 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index d53f206..f3cab84 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -634,6 +634,13 @@ static const struct soc_enum lsidetone_enum = static const struct soc_enum rsidetone_enum = SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text); +static const char *adcinput_text[] = { + "ADC", "DMIC" +}; + +static const struct soc_enum adcinput_enum = + SOC_ENUM_SINGLE(WM8903_CLOCK_RATE_TEST_4, 9, 2, adcinput_text); + static const char *aif_text[] = { "Left", "Right" }; @@ -767,6 +774,9 @@ static const struct snd_kcontrol_new lsidetone_mux = static const struct snd_kcontrol_new rsidetone_mux = SOC_DAPM_ENUM("DACR Sidetone Mux", rsidetone_enum); +static const struct snd_kcontrol_new adcinput_mux = + SOC_DAPM_ENUM("ADC Input", adcinput_enum); + static const struct snd_kcontrol_new lcapture_mux = SOC_DAPM_ENUM("Left Capture Mux", lcapture_enum); @@ -817,6 +827,7 @@ SND_SOC_DAPM_INPUT("IN2L"), SND_SOC_DAPM_INPUT("IN2R"), SND_SOC_DAPM_INPUT("IN3L"), SND_SOC_DAPM_INPUT("IN3R"), +SND_SOC_DAPM_INPUT("DMICDAT"), SND_SOC_DAPM_OUTPUT("HPOUTL"), SND_SOC_DAPM_OUTPUT("HPOUTR"), @@ -842,6 +853,9 @@ SND_SOC_DAPM_MUX("Right Input Mode Mux", SND_SOC_NOPM, 0, 0, &rinput_mode_mux), SND_SOC_DAPM_PGA("Left Input PGA", WM8903_POWER_MANAGEMENT_0, 1, 0, NULL, 0), SND_SOC_DAPM_PGA("Right Input PGA", WM8903_POWER_MANAGEMENT_0, 0, 0, NULL, 0), +SND_SOC_DAPM_MUX("Left ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux), +SND_SOC_DAPM_MUX("Right ADC Input", SND_SOC_NOPM, 0, 0, &adcinput_mux), + SND_SOC_DAPM_ADC("ADCL", NULL, WM8903_POWER_MANAGEMENT_6, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, WM8903_POWER_MANAGEMENT_6, 0, 0), @@ -979,6 +993,11 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = { { "Left Input PGA", NULL, "Left Input Mode Mux" }, { "Right Input PGA", NULL, "Right Input Mode Mux" }, + { "Left ADC Input", "ADC", "Left Input PGA" }, + { "Left ADC Input", "DMIC", "DMICDAT" }, + { "Right ADC Input", "ADC", "Right Input PGA" }, + { "Right ADC Input", "DMIC", "DMICDAT" }, + { "Left Capture Mux", "Left", "ADCL" }, { "Left Capture Mux", "Right", "ADCR" }, @@ -988,9 +1007,9 @@ static const struct snd_soc_dapm_route wm8903_intercon[] = { { "AIFTXL", NULL, "Left Capture Mux" }, { "AIFTXR", NULL, "Right Capture Mux" }, - { "ADCL", NULL, "Left Input PGA" }, + { "ADCL", NULL, "Left ADC Input" }, { "ADCL", NULL, "CLK_DSP" }, - { "ADCR", NULL, "Right Input PGA" }, + { "ADCR", NULL, "Right ADC Input" }, { "ADCR", NULL, "CLK_DSP" }, { "Left Playback Mux", "Left", "AIFRXL" }, -- 1.7.1

4 4

[alsa-devel] [PATCH] ASoC: sdp4430: Fix string for FM input name
by Peter Ujfalusi 26 Sep '11

26 Sep '11

The name contains invalid valid character (/), which causes problems when trying to create the debugfs directory structure: ASoC: Failed to create Aux/FM Stereo In debugfs file Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/omap/sdp4430.c | 6 +++--- 1 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index a951774..d462f1a 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -88,7 +88,7 @@ static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), SND_SOC_DAPM_SPK("Earphone Spk", NULL), - SND_SOC_DAPM_INPUT("Aux/FM Stereo In"), + SND_SOC_DAPM_INPUT("FM Stereo In"), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -113,8 +113,8 @@ static const struct snd_soc_dapm_route audio_map[] = { {"Earphone Spk", NULL, "EP"}, /* Aux/FM Stereo In: AFML, AFMR */ - {"AFML", NULL, "Aux/FM Stereo In"}, - {"AFMR", NULL, "Aux/FM Stereo In"}, + {"AFML", NULL, "FM Stereo In"}, + {"AFMR", NULL, "FM Stereo In"}, }; static int sdp4430_twl6040_init(struct snd_soc_pcm_runtime *rtd) -- 1.7.6

3 4

[alsa-devel] [PATCH 0/3] Fallback mechanism for pulse plugin
by Takashi Iwai 25 Sep '11

25 Sep '11

Hi, the following are experimental patches for implementing the fallback option of PCM / control pulse plugin. When the connection to PA server fails, the plugin tries to open the fallback name. For achieving this, I added the new standard definition "sysdefault", which is equal with the normal "default" PCM / control definitions. The difference is only the name, i.e. it won't be overridden by other setups. Then two new API functions for opening a fallback PCM / control, and finally a patch for pulse-plugin will follow. All changes are relatively small and easy. Let me know if you have any suggestions or a better idea. thanks, Takashi

5 23

[alsa-devel] [PATCHv2] ASoC: omap-mcbsp: Do not attempt to change DAI sysclk if stream is active
by Jarkko Nikula 23 Sep '11

23 Sep '11

Attempt to change McBSP CLKS source while another stream is active is not safe after commit d135865 ("OMAP: McBSP: implement functional clock switching via clock framework") in 2.6.37. CLKS parent clock switching using clock framework have to idle the McBSP before switching and then activate it again. This short break can cause a DMA transaction error to already running stream which halts and recovers only by closing and restarting the stream. This goes more fatal after commit e2fa61d ("OMAP3: l3: Introduce l3-interconnect error handling driver") in 2.6.39 where l3 driver detects a severe timeout error and does BUG_ON(). Fix this by not changing any configuration in omap_mcbsp_dai_set_dai_sysclk if the McBSP is already active. This test should have been here just from the beginning anyway. Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: stable(a)kernel.org --- v2: - Returns -EBUSY from omap_mcbsp_dai_set_dai_sysclk if the input clock rate doesn't match with another stream's configuration and 0 otherwise. For 2.6.37 and onward. This issue is valid to those machine drivers that use the McBSP as a DAI link master and are configuring the CLKS source (like sound/soc/omap/omap3pandora.c in mainline). --- sound/soc/omap/omap-mcbsp.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 07b7723..4b82290 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -516,6 +516,12 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int err = 0; + if (mcbsp_data->active) + if (freq == mcbsp_data->in_freq) + return 0; + else + return -EBUSY; + /* The McBSP signal muxing functions are only available on McBSP1 */ if (clk_id == OMAP_MCBSP_CLKR_SRC_CLKR || clk_id == OMAP_MCBSP_CLKR_SRC_CLKX || -- 1.7.4.4

4 5

[alsa-devel] Noise on snd-hda-intel
by Sune Mølgaard 18 Sep '11

18 Sep '11

Plugging almost any headset into the front headset jacks on my machine results in noise in the headphones as well as in Skype conversations and/or recording. Said noise is, however not head in the regular speakers (except when playing back recordings, of course. lspci -vvnn gives 00:1b.0 Audio device [0403]: Intel Corporation 82801JI (ICH10 Family) HD Audio Controller [8086:3a3e] Subsystem: ASUSTeK Computer Inc. Device [1043:82ea] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx+ Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 256 bytes Interrupt: pin A routed to IRQ 70 Region 0: Memory at f7ff8000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel and aplay -l gives **** List of PLAYBACK Hardware Devices **** card 0: Device [USB Multimedia Audio Device], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 I have tried with no modules options as well as options snd_hda_intel model=6stack and options snd_hda_intel model=6stack-dig Best regards, Sune Mølgaard -- You buy Windows, and you put it on your machine-you can't do a whole lot with it. - Bob Muglia, Microsoft Senior Vice President

4 14

[alsa-devel] Dell XPS 13 (Codec: IDT 92HD73C1X5) and 3 stereo jack output
by Ruben Van Boxem 14 Sep '11

14 Sep '11

Hi, I have been trying to get my Dell XPS 13 to output 5.1 sound through the stereo jacks (because the NVIDIA HDMI only does 2.1). Attached is alsa-info output. I can't seem to be able to set the Line/Mic In to output. I have tried several things, going from fiddling with Kmix to fiddling with the modprobe "model=" parameter, and even looking through the hda_analyzer.py output. This last one lifted a tiny corner of the solution, but I don't think a vanilla Alsa install will get me through this. In Windows, I can use the three jacks to output 2 channels each, so I am sure the hardware is capable of it. In HDA analyzer I can see that there are several "Line Out" entries, but since they all have the node cap "Unsolicited Capabilities", I'm pretty sure the problem is alsa not knowing about the, well, capabilities. I searched the web and documentation for anything sane to try, but nothing worked. The default modprobe "model=" is the best (I can get 4.0 output from the jacks, and everything works, except the third jack). If you need more input or information, please ask, I'm very willing to get this solved! Thanks, Ruben

2 3

[alsa-devel] asoc: no valid backend routes for PCM: SDP4430 Media
by hong zhang 19 Aug '11

19 Aug '11

List, After capturing data from DMIC on OMAP4 blaze board with kernel 2.6.38 and save wav data to 38.wav. But aplay complains "no valid backend routes". See following, $aplay -f dat 38.wav [ 157.167724] asoc: no valid backend routes for PCM: SDP4430 Media aplay: main:654: audio open error: Invalid argument $ uname -a Linux ubuntu 2.6.38.4+ #58 SMP PREEMPT Fri Jun 3 09:44:19 PDT 2011 armv7l GNU/Linux Does ALSA add new functionality of backend? Thanks. ---henry

3 10

Re: [alsa-devel] HDAudio period/buffer sizes
by Takashi Iwai 15 Aug '11

15 Aug '11

At Tue, 26 Jul 2011 14:23:40 -0500, Pierre-Louis Bossart wrote: > > that the buffer itself was already aligned to 128 bytes from the > > very beginning as it's allocated via dma_alloc_coherent(). > > The fact that the commit fixes the real problem means that the buffer > > size must be aligned, too, at least for some hardware. > > I checked internally and the reply was "There is no requirement the buffer > must be 128 byte multiples. As long as it is integer number of samples, we > are good." > > What's the best way to deal with it? Add a driver option? A compilation > option? I'd say a driver option since this must be dynamically adjustable. Care to make a patch? For 3.2 (after 3.1 merge window), we can disable 128-bytes limit as default again for testing. If someone hits the problem, he can switch easily and we can investigate more at this time. thanks, Takashi

3 3

[alsa-devel] [PATCH] ASoC: Revert "ASoC: SAMSUNG: Add I2S0 internal dma driver"
by Mark Brown 12 Aug '11

12 Aug '11

This reverts commit d7c3e9525ac8e898f1156a1f3a7c5038f6560186 as it does not currently build due to missing dependencies in the Samsung tree. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/Makefile | 2 - sound/soc/samsung/idma.c | 453 -------------------------------------------- sound/soc/samsung/idma.h | 26 --- 3 files changed, 0 insertions(+), 481 deletions(-) delete mode 100644 sound/soc/samsung/idma.c delete mode 100644 sound/soc/samsung/idma.h diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 8509d3c..9eb3b12 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -1,6 +1,5 @@ # S3c24XX Platform Support snd-soc-s3c24xx-objs := dma.o -snd-soc-idma-objs := idma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-ac97-objs := ac97.o @@ -17,7 +16,6 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o obj-$(CONFIG_SND_SAMSUNG_SPDIF) += snd-soc-samsung-spdif.o obj-$(CONFIG_SND_SAMSUNG_PCM) += snd-soc-pcm.o obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-i2s.o -obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-idma.o # S3C24XX Machine Support snd-soc-jive-wm8750-objs := jive_wm8750.o diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c deleted file mode 100644 index ebde074..0000000 --- a/sound/soc/samsung/idma.c +++ /dev/null @@ -1,453 +0,0 @@ -/* - * sound/soc/samsung/idma.c - * - * Copyright (c) 2011 Samsung Electronics Co., Ltd. - * http://www.samsung.com - * - * I2S0's Internal DMA driver - * - * 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/interrupt.h> -#include <linux/platform_device.h> -#include <linux/dma-mapping.h> -#include <linux/slab.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include "i2s.h" -#include "idma.h" -#include "dma.h" -#include "i2s-regs.h" - -#define ST_RUNNING (1<<0) -#define ST_OPENED (1<<1) - -static const struct snd_pcm_hardware idma_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_PAUSE | - SNDRV_PCM_INFO_RESUME, - .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_U16_LE | - SNDRV_PCM_FMTBIT_S24_LE | - SNDRV_PCM_FMTBIT_U24_LE | - SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S8, - .channels_min = 2, - .channels_max = 2, - .buffer_bytes_max = MAX_IDMA_BUFFER, - .period_bytes_min = 128, - .period_bytes_max = MAX_IDMA_PERIOD, - .periods_min = 1, - .periods_max = 2, -}; - -struct idma_ctrl { - spinlock_t lock; - int state; - dma_addr_t start; - dma_addr_t pos; - dma_addr_t end; - dma_addr_t period; - dma_addr_t periodsz; - void *token; - void (*cb)(void *dt, int bytes_xfer); -}; - -static struct idma_info { - spinlock_t lock; - void __iomem *regs; - dma_addr_t lp_tx_addr; -} idma; - -static void idma_getpos(dma_addr_t *src) -{ - *src = idma.lp_tx_addr + - (readl(idma.regs + I2STRNCNT) & 0xffffff) * 4; -} - -static int idma_enqueue(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct idma_ctrl *prtd = substream->runtime->private_data; - u32 val; - - spin_lock(&prtd->lock); - prtd->token = (void *) substream; - spin_unlock(&prtd->lock); - - /* Internal DMA Level0 Interrupt Address */ - val = idma.lp_tx_addr + prtd->periodsz; - writel(val, idma.regs + I2SLVL0ADDR); - - /* Start address0 of I2S internal DMA operation. */ - val = idma.lp_tx_addr; - writel(val, idma.regs + I2SSTR0); - - /* - * Transfer block size for I2S internal DMA. - * Should decide transfer size before start dma operation - */ - val = readl(idma.regs + I2SSIZE); - val &= ~(I2SSIZE_TRNMSK << I2SSIZE_SHIFT); - val |= (((runtime->dma_bytes >> 2) & - I2SSIZE_TRNMSK) << I2SSIZE_SHIFT); - writel(val, idma.regs + I2SSIZE); - - val = readl(idma.regs + I2SAHB); - val |= AHB_INTENLVL0; - writel(val, idma.regs + I2SAHB); - - return 0; -} - -static void idma_setcallbk(struct snd_pcm_substream *substream, - void (*cb)(void *, int)) -{ - struct idma_ctrl *prtd = substream->runtime->private_data; - - spin_lock(&prtd->lock); - prtd->cb = cb; - spin_unlock(&prtd->lock); -} - -static void idma_control(int op) -{ - u32 val = readl(idma.regs + I2SAHB); - - spin_lock(&idma.lock); - - switch (op) { - case LPAM_DMA_START: - val |= (AHB_INTENLVL0 | AHB_DMAEN); - break; - case LPAM_DMA_STOP: - val &= ~(AHB_INTENLVL0 | AHB_DMAEN); - break; - default: - spin_unlock(&idma.lock); - return; - } - - writel(val, idma.regs + I2SAHB); - spin_unlock(&idma.lock); -} - -static void idma_done(void *id, int bytes_xfer) -{ - struct snd_pcm_substream *substream = id; - struct idma_ctrl *prtd = substream->runtime->private_data; - - if (prtd && (prtd->state & ST_RUNNING)) - snd_pcm_period_elapsed(substream); -} - -static int idma_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct idma_ctrl *prtd = substream->runtime->private_data; - u32 mod = readl(idma.regs + I2SMOD); - u32 ahb = readl(idma.regs + I2SAHB); - - ahb |= (AHB_DMARLD | AHB_INTMASK); - mod |= MOD_TXS_IDMA; - writel(ahb, idma.regs + I2SAHB); - writel(mod, idma.regs + I2SMOD); - - snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); - runtime->dma_bytes = params_buffer_bytes(params); - - prtd->start = prtd->pos = runtime->dma_addr; - prtd->period = params_periods(params); - prtd->periodsz = params_period_bytes(params); - prtd->end = runtime->dma_addr + runtime->dma_bytes; - - idma_setcallbk(substream, idma_done); - - return 0; -} - -static int idma_hw_free(struct snd_pcm_substream *substream) -{ - snd_pcm_set_runtime_buffer(substream, NULL); - - return 0; -} - -static int idma_prepare(struct snd_pcm_substream *substream) -{ - struct idma_ctrl *prtd = substream->runtime->private_data; - - prtd->pos = prtd->start; - - /* flush the DMA channel */ - idma_control(LPAM_DMA_STOP); - idma_enqueue(substream); - - return 0; -} - -static int idma_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct idma_ctrl *prtd = substream->runtime->private_data; - int ret = 0; - - spin_lock(&prtd->lock); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - prtd->state |= ST_RUNNING; - idma_control(LPAM_DMA_START); - break; - - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - prtd->state &= ~ST_RUNNING; - idma_control(LPAM_DMA_STOP); - break; - - default: - ret = -EINVAL; - break; - } - - spin_unlock(&prtd->lock); - - return ret; -} - -static snd_pcm_uframes_t - idma_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct idma_ctrl *prtd = runtime->private_data; - dma_addr_t src; - unsigned long res; - - spin_lock(&prtd->lock); - - idma_getpos(&src); - res = src - prtd->start; - - spin_unlock(&prtd->lock); - - return bytes_to_frames(substream->runtime, res); -} - -static int idma_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long size, offset; - int ret; - - /* From snd_pcm_lib_mmap_iomem */ - vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); - vma->vm_flags |= VM_IO; - size = vma->vm_end - vma->vm_start; - offset = vma->vm_pgoff << PAGE_SHIFT; - ret = io_remap_pfn_range(vma, vma->vm_start, - (runtime->dma_addr + offset) >> PAGE_SHIFT, - size, vma->vm_page_prot); - - return ret; -} - -static irqreturn_t iis_irq(int irqno, void *dev_id) -{ - struct idma_ctrl *prtd = (struct idma_ctrl *)dev_id; - u32 iiscon, iisahb, val, addr; - - iisahb = readl(idma.regs + I2SAHB); - iiscon = readl(idma.regs + I2SCON); - - val = (iisahb & AHB_LVL0INT) ? AHB_CLRLVL0INT : 0; - - if (val) { - iisahb |= val; - writel(iisahb, idma.regs + I2SAHB); - - addr = readl(idma.regs + I2SLVL0ADDR) - idma.lp_tx_addr; - addr += prtd->periodsz; - addr %= (prtd->end - prtd->start); - addr += idma.lp_tx_addr; - - writel(addr, idma.regs + I2SLVL0ADDR); - - if (prtd->cb) - prtd->cb(prtd->token, prtd->period); - } - - return IRQ_HANDLED; -} - -static int idma_open(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct idma_ctrl *prtd; - int ret; - - snd_soc_set_runtime_hwparams(substream, &idma_hardware); - - prtd = kzalloc(sizeof(struct idma_ctrl), GFP_KERNEL); - if (prtd == NULL) - return -ENOMEM; - - ret = request_irq(IRQ_I2S0, iis_irq, 0, "i2s", prtd); - if (ret < 0) { - pr_err("fail to claim i2s irq , ret = %d\n", ret); - kfree(prtd); - return ret; - } - - spin_lock_init(&prtd->lock); - - runtime->private_data = prtd; - - return 0; -} - -static int idma_close(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct idma_ctrl *prtd = runtime->private_data; - - free_irq(IRQ_I2S0, prtd); - - if (!prtd) - pr_err("idma_close called with prtd == NULL\n"); - - kfree(prtd); - - return 0; -} - -static struct snd_pcm_ops idma_ops = { - .open = idma_open, - .close = idma_close, - .ioctl = snd_pcm_lib_ioctl, - .trigger = idma_trigger, - .pointer = idma_pointer, - .mmap = idma_mmap, - .hw_params = idma_hw_params, - .hw_free = idma_hw_free, - .prepare = idma_prepare, -}; - -static void idma_free(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - - substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - if (!substream) - return; - - buf = &substream->dma_buffer; - if (!buf->area) - return; - - iounmap(buf->area); - - buf->area = NULL; - buf->addr = 0; -} - -static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - - /* Assign PCM buffer pointers */ - buf->dev.type = SNDRV_DMA_TYPE_CONTINUOUS; - buf->addr = idma.lp_tx_addr; - buf->bytes = idma_hardware.buffer_bytes_max; - buf->area = (unsigned char *)ioremap(buf->addr, buf->bytes); - - return 0; -} - -static u64 idma_mask = DMA_BIT_MASK(32); - -static int idma_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_soc_dai *dai = rtd->cpu_dai; - struct snd_pcm *pcm = rtd->pcm; - int ret = 0; - - if (!card->dev->dma_mask) - card->dev->dma_mask = &idma_mask; - if (!card->dev->coherent_dma_mask) - card->dev->coherent_dma_mask = DMA_BIT_MASK(32); - - if (dai->driver->playback.channels_min) - ret = preallocate_idma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - - return ret; -} - -void idma_reg_addr_init(void *regs, dma_addr_t addr) -{ - spin_lock_init(&idma.lock); - idma.regs = regs; - idma.lp_tx_addr = addr; -} - -struct snd_soc_platform_driver asoc_idma_platform = { - .ops = &idma_ops, - .pcm_new = idma_new, - .pcm_free = idma_free, -}; - -static int __devinit asoc_idma_platform_probe(struct platform_device *pdev) -{ - return snd_soc_register_platform(&pdev->dev, &asoc_idma_platform); -} - -static int __devexit asoc_idma_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; -} - -static struct platform_driver asoc_idma_driver = { - .driver = { - .name = "samsung-idma", - .owner = THIS_MODULE, - }, - - .probe = asoc_idma_platform_probe, - .remove = __devexit_p(asoc_idma_platform_remove), -}; - -static int __init asoc_idma_init(void) -{ - return platform_driver_register(&asoc_idma_driver); -} -module_init(asoc_idma_init); - -static void __exit asoc_idma_exit(void) -{ - platform_driver_unregister(&asoc_idma_driver); -} -module_exit(asoc_idma_exit); - -MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar(a)gmail.com>"); -MODULE_DESCRIPTION("Samsung ASoC IDMA Driver"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/idma.h b/sound/soc/samsung/idma.h deleted file mode 100644 index 4827321..0000000 --- a/sound/soc/samsung/idma.h +++ /dev/null @@ -1,26 +0,0 @@ -/* - * sound/soc/samsung/idma.h - * - * Copyright (c) 2011 Samsung Electronics Co., Ltd - * http://www.samsung.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. - * - */ - -#ifndef __SND_SOC_SAMSUNG_IDMA_H_ -#define __SND_SOC_SAMSUNG_IDMA_H_ - -extern void idma_reg_addr_init(void *regs, dma_addr_t addr); - -/* dma_state */ -#define LPAM_DMA_STOP 0 -#define LPAM_DMA_START 1 - -#define MAX_IDMA_PERIOD (128 * 1024) -#define MAX_IDMA_BUFFER (160 * 1024) - -#endif /* __SND_SOC_SAMSUNG_IDMA_H_ */ -- 1.7.5.4

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Re: [alsa-devel] Sound BoF in LinuxCon Europe?
by Mark Brown 05 Aug '11

05 Aug '11

On Sat, Jul 09, 2011 at 09:43:32AM +0200, Takashi Iwai wrote: > Mark Brown wrote: > > Yes, I don't think it's good for covering at KS as it's far too > > detailed. If we're going to do something probably LinuxCon/ELC Europe > > make most sense. > Well, KS day 1 is basically an open-day for workshops, AFAIK. > (The days 2 and 3 are different.) > So, it can be suitable for covering subsystem(s) specific problems. > Though, I also think LinuxCon/ELC would be a better choice, too. If nothing else we can always arrange something in the hallway/beer tracks.

2 3

[alsa-devel] Suspend/Resume failure of Alsa but OSS emulation still works
by Shane Volpe 05 Aug '11

05 Aug '11

My embedded system's audio is not resuming after the system resumes from suspend to RAM, below is a detailed discription of the problem and what I have done, to date, to identify the source of the problem. I have tested the following scenario with several audio applications such as Alsa aplay and ogg123: If I have an audio player (aplay or ogg123) using Alsa and I suspend my system to RAM when the system wakes up (resumes) there is no sound. If I kill the audio player and start it back up the sound returns so it seems the act of opening and closing Alsa fixes the failure. During the "no sound state" I also opened alsamixer and adjusted (successfully) several alsa settings including volume but the sound still did not resume until I restarted the player. I also configured one player to use the OSS (emulated by Alsa). When I used the emulated OSS audio driver (/dev/dsp) the sound did resume after the system woke up (resumed) from memory so everything WORKED! I then configured the player back to use the native Alsa driver and again there was no sound after resume. Has anyone seen this issue before or does anyone have additional tests or debugging ideas they can recommend to further narrow down the issue. I'm very comfortable with modifying and compiling the Kernel so please do not limit your suggestions to user-space tweaks. Embedded System Information: Processor: ARM/Xscale (PXA270) Audio Codec: AC97 (UCB1400) alsa lib version: 1.0.24.1 Linux Kernel Version: 2.6.39.2 Regards, Shane

1 2

[alsa-devel] memory leak in mmap ops
by Scott Jiang 04 Aug '11

04 Aug '11

Hi all, I found there is a memory leak about 128K every time in linux-2.6.34, but it doesn't exist in linux-3.0. If I use mmap ops in driver the bug will come out, while use copy ops no memory leak. This driver is based on asoc. I tested with aplay to play a wav file. Because there is a lot of changes in ALSA between 2.6.34 and 3.0, so can anyone who familiar with the change history tell me which patch has solved this bug? Thanks in advance. Scott

2 2

[alsa-devel] Timer interface
by Nenad Sljivic 02 Aug '11

02 Aug '11

Hi, I am extending my ALSA driver in order to support software mixing. Driver currently supports both MMAP and RW access modes. However, I am still not able to use dmix since it requires timer interface to be implemented. My hardware does not generate interrupts at the period boundary but issues timer interrupts at a fixed timer rate. Could you please point me to the document or sample driver that can help me to fully understand timer interface concept (especially for High frequency timer interrupts)? Thanks, Nenad

2 4

[alsa-devel] [PATCH] ALSA: asihpi - Clarify adapter index validity check.
by linux＠audioscience.com 01 Aug '11

01 Aug '11

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Avoids assigning possibly invalid address to pa, even if it is never dereferenced. Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- sound/pci/asihpi/hpioctl.c | 19 ++++++++++++------- 1 files changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/pci/asihpi/hpioctl.c b/sound/pci/asihpi/hpioctl.c index 9683f84..a32502e 100644 --- a/sound/pci/asihpi/hpioctl.c +++ b/sound/pci/asihpi/hpioctl.c @@ -177,16 +177,21 @@ long asihpi_hpi_ioctl(struct file *file, unsigned int cmd, unsigned long arg) } else { u16 __user *ptr = NULL; u32 size = 0; - + u32 adapter_present; /* -1=no data 0=read from user mem, 1=write to user mem */ int wrflag = -1; - u32 adapter = hm->h.adapter_index; - struct hpi_adapter *pa = &adapters[adapter]; + struct hpi_adapter *pa; + + if (hm->h.adapter_index < HPI_MAX_ADAPTERS) { + pa = &adapters[hm->h.adapter_index]; + adapter_present = pa->type; + } else { + adapter_present = 0; + } - if ((adapter >= HPI_MAX_ADAPTERS) || (!pa->type)) { - hpi_init_response(&hr->r0, HPI_OBJ_ADAPTER, - HPI_ADAPTER_OPEN, - HPI_ERROR_BAD_ADAPTER_NUMBER); + if (!adapter_present) { + hpi_init_response(&hr->r0, hm->h.object, + hm->h.function, HPI_ERROR_BAD_ADAPTER_NUMBER); uncopied_bytes = copy_to_user(puhr, hr, sizeof(hr->h)); -- 1.7.0.4

3 4

[alsa-devel] [PATCH -next] sound: rtctimer.c needs module.h
by Randy Dunlap 30 Jul '11

30 Jul '11

From: Randy Dunlap <rdunlap(a)xenotime.net> rtctimer.c uses interfaces from linux/module.h, so it should include that file. This fixes build errors. Signed-off-by: Randy Dunlap <rdunlap(a)xenotime.net> --- sound/core/rtctimer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- linux-next-20110729.orig/sound/core/rtctimer.c +++ linux-next-20110729/sound/core/rtctimer.c @@ -22,7 +22,7 @@ #include <linux/init.h> #include <linux/interrupt.h> -#include <linux/moduleparam.h> +#include <linux/module.h> #include <linux/log2.h> #include <sound/core.h> #include <sound/timer.h>

2 1

Re: [alsa-devel] [RFC PATCH] ARM ASoC: add sound driver for imx27pdk using mc13783 codec
by Jürgen Lambrecht 29 Jul '11

29 Jul '11

2011/7/27 Philippe Rétornaz <philippe.retornaz(a)epfl.ch>: > Hi > > Le vendredi 8 juillet 2011 16:19:22, Lambrecht Jürgen a écrit : >> indeed, doesn't work at all. I know it used to work on 2.6.34. >> We also have a fsl linux 2.6.22 were it works. >> I must say: when i use aplay to play a mono file, then there is no >> error, it just never ends, and no sound also. >> For a stereo file, it gives the error. >> Mark also that when I use madplay (v0.15.2) (or mp3play) to play a mono >> or stereo mp3, it allocates 3 DMA channels, but only frees one after the >> error. >> > > Did you made any progress on this ? Yes. Sascha fixed it, and will commit it. Regards, Jürgen > > Regards, > > Philippe > > >

1 0

[alsa-devel] [PATCH 0/2] hdsploader/hdspmixer: Fix RPM constant
by Adrian Knoth 29 Jul '11

29 Jul '11

As noticed by Takashi, the RPM constant needs to be 4, not 5. For hdspmixer, this doesn't matter, for hdsploader, it's essential. Adrian Knoth (2): hdsploader: Fix temporary RPM constant hdspmixer: Fix temporary RPM constant hdsploader/hdsploader.c | 2 +- hdspmixer/src/HDSPMixerCard.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) -- 1.7.5.4

2 3

[alsa-devel] [PATCH] ASoC: sgtl5000: add device tree probe support
by Shawn Guo 29 Jul '11

29 Jul '11

It adds device tree probe support for sgtl5000 driver. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> Cc: Grant Likely <grant.likely(a)secretlab.ca> Cc: Zeng Zhaoming <zhaoming.zeng(a)freescale.com> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- .../bindings/sound/soc/codecs/fsl-sgtl5000.txt | 11 +++++++++++ sound/soc/codecs/sgtl5000.c | 8 ++++++++ 2 files changed, 19 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt diff --git a/Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt b/Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt new file mode 100644 index 0000000..2c3cd41 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/soc/codecs/fsl-sgtl5000.txt @@ -0,0 +1,11 @@ +* Freescale SGTL5000 Stereo Codec + +Required properties: +- compatible : "fsl,sgtl5000". + +Example: + +codec: sgtl5000@0a { + compatible = "fsl,sgtl5000"; + reg = <0x0a>; +}; diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 76258f2..cf6eea8 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -20,6 +20,7 @@ #include <linux/regulator/driver.h> #include <linux/regulator/machine.h> #include <linux/regulator/consumer.h> +#include <linux/of_device.h> #include <sound/core.h> #include <sound/tlv.h> #include <sound/pcm.h> @@ -1494,10 +1495,17 @@ static const struct i2c_device_id sgtl5000_id[] = { MODULE_DEVICE_TABLE(i2c, sgtl5000_id); +static const struct of_device_id sgtl5000_dt_ids[] = { + { .compatible = "fsl,sgtl5000", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(i2c, sgtl5000_dt_ids); + static struct i2c_driver sgtl5000_i2c_driver = { .driver = { .name = "sgtl5000", .owner = THIS_MODULE, + .of_match_table = sgtl5000_dt_ids, }, .probe = sgtl5000_i2c_probe, .remove = __devexit_p(sgtl5000_i2c_remove), -- 1.7.4.1

4 5

Re: [alsa-devel] Copy ALSA buffer to the buffer allocated in my driver.
by Clemens Ladisch 29 Jul '11

29 Jul '11

`/ET!-i1S!-i wrote: > Actually i need to stream the audio data to other system through kernel > socket. Why aren't you using PulseAudio? Regards, Clemens

1 0

[alsa-devel] [PATCH 0/3] [RME - HDSP] Userlevel support for RPM
by Adrian Knoth 29 Jul '11

29 Jul '11

Hi! Here's a series of recent work on RME RPM support. Confirmed to be working by our one and only user so far. ;) As soon as alsa-lib supports the RPM (it's a single define), I'll drop the temporary define in both, hdsploader and hdspmixer. RPM users: note that the current kernel code seems to be broken and does not correctly detect your RPM box. More magic could be added to hdsploader to provide a temporary workaround. Cheers Adrian Knoth (3): hdspmixer: Add support for RME RPM hdsploader: Add support for RME RPM hdsploader: Temporary define for RPM hdsploader/hdsploader.c | 7 +++++++ hdspmixer/src/HDSPMixerCard.cxx | 11 +++++++++++ hdspmixer/src/HDSPMixerCard.h | 5 +++++ hdspmixer/src/HDSPMixerOutput.cxx | 15 +++++++++++++++ hdspmixer/src/HDSPMixerSelector.cxx | 7 +++++++ hdspmixer/src/HDSPMixerWindow.cxx | 4 ++++ hdspmixer/src/channelmap.cxx | 12 ++++++++++++ hdspmixer/src/channelmap.h | 5 +++++ hdspmixer/src/hdspmixer.cxx | 4 ++++ 9 files changed, 70 insertions(+), 0 deletions(-) -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Parse board ID/revision information from WM1250-EV1 board
by Mark Brown 29 Jul '11

29 Jul '11

The WM1250-EV1 board has an ID chip on it, check the board ID and display the board revision during startup. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm1250-ev1.c | 20 +++++++++++++++++++- 1 files changed, 19 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index bcc2089..bbcf9ec 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -56,8 +56,26 @@ static struct snd_soc_codec_driver soc_codec_dev_wm1250_ev1 = { }; static int __devinit wm1250_ev1_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) + const struct i2c_device_id *i2c_id) { + int ret, id, board, rev; + + board = i2c_smbus_read_byte_data(i2c, 0); + if (board < 0) { + dev_err(&i2c->dev, "Failed to read ID: %d\n", ret); + return ret; + } + + id = (board & 0xfe) >> 2; + rev = board & 0x3; + + if (id != 1) { + dev_err(&i2c->dev, "Unknown board ID %d\n", id); + return -ENODEV; + } + + dev_info(&i2c->dev, "revision %d\n", rev); + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm1250_ev1, &wm1250_ev1_dai, 1); } -- 1.7.5.4

1 0

[alsa-devel] Copy ALSA buffer to the buffer allocated in my driver.
by `/ET!-i1S!-i 29 Jul '11

29 Jul '11

Hi everyone. I'm writing a virtual ALSA driver for a playback device, with the help of dummy driver available at the linux kernel source code. I need to manually copy data from the ALSA buffer to my buffer. I have no idea how to do this, since dummy driver has completely discarded handling of buffers. So can anyone please provide a sample code to copy the audio data to the my buffer allocated in the driver. Is it ok to use the hrtimer interrupts (which is used in dummy driver) or mutex / spinlocks for my driver. Best regards, cherry

2 1

[alsa-devel] mmap ops in snd_pcm_ops
by Scott Jiang 29 Jul '11

29 Jul '11

Hi all, I found there is a memory leak about 128K every time in linux-2.6.34, but it doesn't exist in linux-3.0. If I use mmap ops in driver the bug will come out, while use copy ops no memory leak. I tested with aplay to play a wav file. Because there is a lot of changes in ALSA between 2.6.34 and 3.0, so can anyone familiar with the change history tell me which patch has solved this bug? Thanks in advance. Scott

1 0

[alsa-devel] [PATCH] ASoC: Disable wm_hubs periodic DC servo update
by Mark Brown 29 Jul '11

29 Jul '11

This does not function correctly in all circumstances so disable the periodic updates unconditionally for stable; a future patch will reenable where appropriate. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 3 +-- 1 files changed, 1 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 4de6794..4416a76 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -440,9 +440,8 @@ static int hp_event(struct snd_soc_dapm_widget *w, reg |= WM8993_HPOUT1L_DLY | WM8993_HPOUT1R_DLY; snd_soc_write(codec, WM8993_ANALOGUE_HP_0, reg); - /* Smallest supported update interval */ snd_soc_update_bits(codec, WM8993_DC_SERVO_1, - WM8993_DCS_TIMER_PERIOD_01_MASK, 1); + WM8993_DCS_TIMER_PERIOD_01_MASK, 0); calibrate_dc_servo(codec); -- 1.7.5.4

2 1

[alsa-devel] [PATCH] SOUND: Fix txx9aclc.c build
by Ralf Baechle 29 Jul '11

29 Jul '11

552d1ef6b5a98d7b95959d5b139071e3c90cebf1 [ASoC: core - Optimise and refactor pcm_new() to pass only rtd] breaks compilation of txx9aclc.c: CC [M] sound/soc/txx9/txx9aclc.o /home/ralf/src/linux/linux-mips/sound/soc/txx9/txx9aclc.c: In function 'txx9aclc_pcm_new': /home/ralf/src/linux/linux-mips/sound/soc/txx9/txx9aclc.c:318:3: error: 'card' undeclared (first use in this function) /home/ralf/src/linux/linux-mips/sound/soc/txx9/txx9aclc.c:318:3: note: each undeclared identifier is reported only once for each function it appears in make[5]: *** [sound/soc/txx9/txx9aclc.o] Error 1 Fixed by providing a definition for card. Signed-off-by: Ralf Baechle <ralf(a)linux-mips.org> sound/soc/txx9/txx9aclc.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index 34aa972..3de99af 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -290,6 +290,7 @@ static void txx9aclc_pcm_free_dma_buffers(struct snd_pcm *pcm) static int txx9aclc_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; struct snd_soc_dai *dai = rtd->cpu_dai; struct snd_pcm *pcm = rtd->pcm; struct platform_device *pdev = to_platform_device(dai->platform->dev);

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[alsa-devel] [PATCH 0/3] [RME - HDSPM/MADI] Speed mode selector
by Adrian Knoth 29 Jul '11

29 Jul '11

Hi! Here's another small series of patches for the RME MADI cards. Most prominently, we now provide a control element to the user to explicitly select a frequency multiplier for the MADI link. This way, one can run applications at 64kHz and above (up to 192kHz) even when the card is not the clock master (read: is in slave mode). Cheers Adrian Knoth (3): ALSA: hdspm - Provide MADI speed mode selector on RME MADI and MADIface ALSA: hdspm - Fix reported external sample rate on RME MADI and MADIface ALSA: hdspm - Add firmware revision 0xcc for RME MADI sound/pci/rme9652/hdspm.c | 109 ++++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 107 insertions(+), 2 deletions(-) -- 1.7.5.4

2 4

[alsa-devel] [PATCH alsa-lib 1/1] Add RME RPM to HDSP_IO_Type.
by Adrian Knoth 29 Jul '11

29 Jul '11

The hdsp.c driver supports the RME RPM devices, but we also need this information in alsa-lib for tools like hdsploader and hdspmixer. Signed-off-by: Adrian Knoth <adi(a)drcomp.erfurt.thur.de> diff --git a/include/sound/hdsp.h b/include/sound/hdsp.h index 0fc9e98..5adaf7b 100644 --- a/include/sound/hdsp.h +++ b/include/sound/hdsp.h @@ -28,6 +28,7 @@ typedef enum { Multiface, H9652, H9632, + RPM, Undefined, } HDSP_IO_Type; -- 1.7.5.4

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Re: [alsa-devel] [PATCH] Sound: AudioScience HPI: Don't leak firmware if mem alloc fails
by Takashi Iwai 29 Jul '11

29 Jul '11

At Thu, 28 Jul 2011 21:48:27 +0200 (CEST), Jesper Juhl wrote: > > We leak the memory allocated to 'firmware' when we fail to > release_firmware() after a kmalloc() failure in hpi_dsp_code_open(). > This patch should take care of the leak. > > Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> > --- > sound/pci/asihpi/hpidspcd.c | 4 +++- > 1 files changed, 3 insertions(+), 1 deletions(-) > > Build tested only. > > diff --git a/sound/pci/asihpi/hpidspcd.c b/sound/pci/asihpi/hpidspcd.c > index 3a7afa3..8d261ef 100644 > --- a/sound/pci/asihpi/hpidspcd.c > +++ b/sound/pci/asihpi/hpidspcd.c > @@ -85,8 +85,10 @@ short hpi_dsp_code_open(u32 adapter, void *os_data, struct dsp_code *dsp_code, > > HPI_DEBUG_LOG(DEBUG, "dsp code %s opened\n", fw_name); > dsp_code->pvt = kmalloc(sizeof(*dsp_code->pvt), GFP_KERNEL); > - if (!dsp_code->pvt) > + if (!dsp_code->pvt) { > + release_firmware(firmware); > return HPI_ERROR_MEMORY_ALLOC; > + } IMO, it's better to use the same error handling like others, i.e. "goto error2". But you want to pass a different error code, so we'd need to introduce an error code variable there... Takashi

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Re: [alsa-devel] [PATCH] ASoC: Add device tree binding for WM8731
by Grant Likely 29 Jul '11

29 Jul '11

On Tue, Jul 26, 2011 at 09:02:11PM +0100, Mark Brown wrote: > Tested with the famous "hey, look! this compiles" test plan. > > Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > --- > Documentation/devicetree/bindings/sound/wm8731.txt | 18 ++++++++++++++++++ > sound/soc/codecs/wm8731.c | 12 ++++++++++++ > 2 files changed, 30 insertions(+), 0 deletions(-) > create mode 100644 Documentation/devicetree/bindings/sound/wm8731.txt > > diff --git a/Documentation/devicetree/bindings/sound/wm8731.txt b/Documentation/devicetree/bindings/sound/wm8731.txt > new file mode 100644 > index 0000000..15f7004 > --- /dev/null > +++ b/Documentation/devicetree/bindings/sound/wm8731.txt > @@ -0,0 +1,18 @@ > +WM8731 audio CODEC > + > +This device supports both I2C and SPI (configured with pin strapping > +on the board). > + > +Required properties: > + > + - compatible : "wlf,wm8731" > + > + - reg : the I2C address of the device for I2C, the chip select > + number for SPI. > + > +Example: > + > +codec: wm8731@1a { > + compatible = "wlf,wm8731"; > + reg = <0x1a>; > +}; Acked by: Grant Likely <grant.likely(a)secretlab.ca> Although I'm sure some actual testing would be valuable before it is actually merged. :-) And there is one comment below. > diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c > index 76b4361..5ecc05a 100644 > --- a/sound/soc/codecs/wm8731.c > +++ b/sound/soc/codecs/wm8731.c > @@ -22,6 +22,7 @@ > #include <linux/platform_device.h> > #include <linux/regulator/consumer.h> > #include <linux/spi/spi.h> > +#include <linux/of_device.h> > #include <sound/core.h> > #include <sound/pcm.h> > #include <sound/pcm_params.h> > @@ -607,6 +608,11 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8731 = { > .num_dapm_routes = ARRAY_SIZE(wm8731_intercon), > }; > > +static const struct of_device_id wm8731_of_match[] = { > + { .compatible = "wlf,wm8731", }, > + { } > +}; > + > #if defined(CONFIG_SPI_MASTER) > static int __devinit wm8731_spi_probe(struct spi_device *spi) > { > @@ -634,10 +640,13 @@ static int __devexit wm8731_spi_remove(struct spi_device *spi) > return 0; > } > > +MODULE_DEVICE_TABLE(spi, wm8731_of_match); MODULE_DEVICE_TABLE(of, wm8731_of_match); > + > static struct spi_driver wm8731_spi_driver = { > .driver = { > .name = "wm8731", > .owner = THIS_MODULE, > + .of_match_table = wm8731_of_match, > }, > .probe = wm8731_spi_probe, > .remove = __devexit_p(wm8731_spi_remove), > @@ -678,10 +687,13 @@ static const struct i2c_device_id wm8731_i2c_id[] = { > }; > MODULE_DEVICE_TABLE(i2c, wm8731_i2c_id); > > +MODULE_DEVICE_TABLE(i2c, wm8731_of_match); > + And this line can be dropped. i2c module info is separate from i2c and spi tables (although there may be some breakage here that need to be looked at. Right now I don't think the OF device ids get exported to udev for non-platform_devices. It needs some investigation.) > static struct i2c_driver wm8731_i2c_driver = { > .driver = { > .name = "wm8731", > .owner = THIS_MODULE, > + .of_match_table = wm8731_of_match, > }, > .probe = wm8731_i2c_probe, > .remove = __devexit_p(wm8731_i2c_remove), > -- > 1.7.5.4 > > _______________________________________________ > devicetree-discuss mailing list > devicetree-discuss(a)lists.ozlabs.org > https://lists.ozlabs.org/listinfo/devicetree-discuss

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[alsa-devel] [RME - hdsp] Multiface/Digiface/RPM detection
by Adrian Knoth 28 Jul '11

28 Jul '11

Hi! I've recently added support for the RME RPM to hdspmixer. Unfortunately, I don't have one, it's been done blindly with user feedback. This very user now reports that he needs to upload the device firmware from windows. I've checked hdsploader, and of course, it needs patching. I'll take care in a second. More surprisingly, though, the kernel wasn't able to upload the firmware itself, because it fails to detect the RPM and hence tries to upload a multiface firmware. After reading the kernel source, I think the code in hdsp.c is wrong: if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT)) { hdsp_write(hdsp, HDSP_control2Reg, HDSP_VERSION_BIT); hdsp_write(hdsp, HDSP_control2Reg, HDSP_S_LOAD); if (hdsp_fifo_wait(hdsp, 0, HDSP_SHORT_WAIT)) hdsp->io_type = RPM; else hdsp->io_type = Multiface; } else { hdsp->io_type = Digiface; } Who here owns a Digiface and can confirm or deny that the kernel correctly detects it as Digiface? Same for Multiface, though I guess since it's more or less the default, users wouldn't notice it. What's wrong with the code above? I think all occurrences of HDSP_control2Reg in hdsp_check_for_iobox need to be changed to HDSP_controlRegister and the second hdsp_fifo_wait needs to be inverted. But this is pure guesswork. If I come up with a patch, who here has a RPM, Digiface or Multiface to test it? TIA

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Re: [alsa-devel] Build regressions/improvements in v3.0-rc7 (sound/isa/)
by Randy Dunlap 28 Jul '11

28 Jul '11

On Thu, 28 Jul 2011 14:24:49 -0400 Arnaud Lacombe wrote: > Hi, > > On Thu, Jul 28, 2011 at 2:11 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Thu, 28 Jul 2011 13:06:19 -0400, > > Arnaud Lacombe wrote: > >> > >> Hi, > >> > >> On Thu, Jul 28, 2011 at 12:44 PM, Randy Dunlap <rdunlap(a)xenotime.net> wrote: > >> > On Thu, 28 Jul 2011 12:36:27 -0400 Arnaud Lacombe wrote: > >> > > >> >> Hi, > >> >> > >> >> On Thu, Jul 28, 2011 at 12:21 PM, Randy Dunlap <rdunlap(a)xenotime.net> wrote: > >> >> > On Thu, 28 Jul 2011 11:17:09 +1000 Stephen Rothwell wrote: > >> >> > > >> >> >> Hi Randy, > >> >> >> > >> >> >> On Wed, 27 Jul 2011 12:13:20 -0700 Randy Dunlap <rdunlap(a)xenotime.net> wrote: > >> >> >> > > >> >> >> > On Mon, 25 Jul 2011 09:10:03 -0700 Randy Dunlap wrote: > >> >> >> > > >> >> >> > > On Mon, 25 Jul 2011 09:37:53 +0200 Takashi Iwai wrote: > >> >> >> > > > >> >> >> > > > At Sun, 24 Jul 2011 12:26:24 -0700, > >> >> >> > > > Randy Dunlap wrote: > >> >> >> > > > > > >> >> >> > > > > On Wed, 13 Jul 2011 21:33:30 +0200 Geert Uytterhoeven wrote: > >> >> >> > > > > > >> >> >> > > > > > + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit declaration of function 'snd_dma_disable': => 544 > >> >> >> > > > > > + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit declaration of function 'snd_dma_pointer': => 302 > >> >> >> > > > > > + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit declaration of function 'snd_dma_program': => 244 > >> >> >> > > > > > + src/sound/isa/es1688/es1688_lib.c: error: implicit declaration of function 'snd_dma_pointer': => 509 > >> >> >> > > > > > + src/sound/isa/es1688/es1688_lib.c: error: implicit declaration of function 'snd_dma_program': => 417 > >> >> >> > > > > > + src/sound/isa/gus/gus_dma.c: error: implicit declaration of function 'snd_dma_disable': => 177 > >> >> >> > > > > > + src/sound/isa/gus/gus_dma.c: error: implicit declaration of function 'snd_dma_program': => 79 > >> >> >> > > > > > + src/sound/isa/gus/gus_pcm.c: error: implicit declaration of function 'snd_dma_pointer': => 619 > >> >> >> > > > > > + src/sound/isa/gus/gus_pcm.c: error: implicit declaration of function 'snd_dma_program': => 591 > >> >> >> > > > > > + src/sound/isa/sb/sb16_main.c: error: implicit declaration of function 'snd_dma_pointer': => 456 > >> >> >> > > > > > + src/sound/isa/sb/sb16_main.c: error: implicit declaration of function 'snd_dma_program': => 276 > >> >> >> > > > > > + src/sound/isa/sb/sb8_main.c: error: implicit declaration of function 'snd_dma_pointer': => 425 > >> >> >> > > > > > + src/sound/isa/sb/sb8_main.c: error: implicit declaration of function 'snd_dma_program': => 172 > >> >> >> > > > > > + src/sound/isa/sscape.c: error: implicit declaration of function 'snd_dma_program': => 481 > >> >> >> > > > > > + src/sound/isa/wss/wss_lib.c: error: implicit declaration of function 'snd_dma_disable': => 1695 > >> >> >> > > > > > + src/sound/isa/wss/wss_lib.c: error: implicit declaration of function 'snd_dma_pointer': => 1160 > >> >> >> > > > > > + src/sound/isa/wss/wss_lib.c: error: implicit declaration of function 'snd_dma_program': => 1025 > >> >> >> > > > > > >> >> >> > > > > All of these source files do #include <sound/core.h>, > >> >> >> > > > > but these missing/implicit functions are conditional: > >> >> >> > > > > > >> >> >> > > > > #ifdef CONFIG_ISA_DMA_API > >> >> >> > > > > #define DMA_MODE_NO_ENABLE 0x0100 > >> >> >> > > > > > >> >> >> > > > > void snd_dma_program(unsigned long dma, unsigned long addr, unsigned int size, unsigned short mode); > >> >> >> > > > > void snd_dma_disable(unsigned long dma); > >> >> >> > > > > unsigned int snd_dma_pointer(unsigned long dma, unsigned int size); > >> >> >> > > > > #endif > >> >> >> > > > > > >> >> >> > > > > so should these drivers all depend on ISA_DMA_API ? > >> >> >> > > > > >> >> >> > > > Yes, these are ISA sound cards, so they must have CONFIG_ISA and > >> >> >> > > > CONFIG_ISA_DMA. Although CONFIG_SND_SB8_DSP, CONFIG_SND_SB16_DSP > >> >> >> > > > and CONFIG_SND_WSS_LIB have no direct dependency, but the drivers > >> >> >> > > > selecting these are all in "if SND_ISA" in sound/isa/Kconfig. > >> >> >> > > > >> >> >> > > Hm, yes, they are. > >> >> >> > > > >> >> >> > > > > or should only pieces of each driver depend on ISA_DMA_API ? > >> >> >> > > > > >> >> >> > > > I wonder which configuration makes it possible. > >> >> >> > > > Did you see a similar problem with randconfig? > >> >> >> > > > >> >> >> > > No. This parisc allmodconfig file from linux-next builds has > >> >> >> > > CONFIG_SND_ISA=y but it does not have CONFIG_ISA_DMA_API enabled. Weird. > >> >> >> > > > >> >> >> > > Geert, do you know how this happens? > >> >> >> > > > >> >> >> > > parisc config: http://kisskb.ellerman.id.au/kisskb/buildresult/4323458/config/ > >> >> >> > > from this failed build: http://kisskb.ellerman.id.au/kisskb/buildresult/4323458/ > >> >> >> > > >> >> >> > How is CONFIG_ISA_DMA_API enabled in parisc allmodconfig? > >> >> >> > I don't see how that happens, and it's causing build errors... > >> >> >> > >> >> >> Sorry, but I don't understand the question. The problam is that > >> >> >> CONFIG_ISA_DMA_API is not enabled, right? On parisc, it is just (from > >> >> >> arch/parisc/Kconfig) > >> >> >> > >> >> >> config ISA_DMA_API > >> >> >> bool > >> >> >> > >> >> >> and nothing selects it. However, I would have expected an allmodconfig > >> >> >> to set it to y. Clearly I don't understan how allmodconfig works any > >> >> >> more :-( > >> >> >> > >> >> >> Michal, is my understanding correct? Is this a bug in allmodconfig? > >> >> > > >> >> > When I do: > >> >> > $ make ARCH=parisc O=PAR allmodconfig > >> >> > > >> >> > I don't get CONFIG_ISA_DMA_API enabled, so the question is simply > >> >> > how is the linux-next build system ending up with CONFIG_ISA_DMA_API enabled? > >> >> > > >> >> Where do you see that "linux-next build system ending up with > >> >> CONFIG_ISA_DMA_API enabled" ? > >> >> > >> >> From what I understood of the problem, the issue is that > >> >> CONFIG_ISA_DMA_API does _not_ ends up enabled. > >> > > >> > Thanks for catching me. > >> > > >> > The problem (second try) is that ISA SND drivers are being built (on parisc) > >> > even though ISA_DMA_API is not enabled -- and should not be enabled, > >> > so the ISA SND drivers should not be built. I think I have it right this time. > >> > > >> > From the parisc .config file listed above: > >> > > >> > CONFIG_SND_ISA=y > >> > CONFIG_SND_ADLIB=m > >> > CONFIG_SND_AD1816A=m > >> > CONFIG_SND_AD1848=m > >> > CONFIG_SND_ALS100=m > >> > CONFIG_SND_AZT1605=m > >> > CONFIG_SND_AZT2316=m > >> > CONFIG_SND_AZT2320=m > >> > CONFIG_SND_CMI8330=m > >> > > >> > > >> > etc. But none of these should be enabled. AFAIK. > >> > So how does this happen? > >> > > >> warning: (RADIO_MIROPCM20) selects SND_ISA which has unmet direct > >> dependencies (SOUND && !M68K && SND && ISA && ISA_DMA_API) > > > > Hm, I thought this was fixed by > > commit 4756fc64ecd628efa6f8296711d7dbcefabf25dd > > Author: Ralf Baechle <ralf(a)linux-mips.org> > > Date: Fri Jun 24 10:30:09 2011 -0300 > > > > [media] MEDIA: Fix non-ISA_DMA_API link failure of sound code > > > > OK, it landed after 3.0-rc7, so this problem must be fixed now? > % git checkout v3.0-rc7 > % make ARCH=parisc allmodconfig > [...] > warning: (RADIO_WL128X) selects TI_ST which has unmet direct > dependencies (MISC_DEVICES && NET && GPIOLIB) > warning: (RADIO_MIROPCM20) selects SND_ISA which has unmet direct > dependencies (SOUND && !M68K && SND && ISA && ISA_DMA_API) > warning: (LOCKDEP && FAULT_INJECTION_STACKTRACE_FILTER && LATENCYTOP > && FUNCTION_TRACER && KMEMCHECK) selects FRAME_POINTER which has unmet > direct dependencies (DEBUG_KERNEL && (CRIS || M68K || FRV || UML || > AVR32 || SUPERH || BLACKFIN || MN10300) || ARCH_WANT_FRAME_POINTERS) > # > # configuration written to .config > # > % grep -c SND_ISA .config > 1 > > % git checkout v3.0 > Previous HEAD position was 620917d... Linux 3.0-rc7 > HEAD is now at 02f8c6a... Linux 3.0 > % make ARCH=parisc allmodconfig > [...] > warning: (RADIO_WL128X) selects TI_ST which has unmet direct > dependencies (MISC_DEVICES && NET && GPIOLIB) > warning: (LOCKDEP && FAULT_INJECTION_STACKTRACE_FILTER && LATENCYTOP > && FUNCTION_TRACER && KMEMCHECK) selects FRAME_POINTER which has unmet > direct dependencies (DEBUG_KERNEL && (CRIS || M68K || FRV || UML || > AVR32 || SUPERH || BLACKFIN || MN10300) || ARCH_WANT_FRAME_POINTERS) > # > # configuration written to .config > # > % grep -c SND_ISA .config > 0 > > Look good to me. Great, thanks everyone. Yes, I was working on old data. Sorry about that. Anyway, lots of build errors now fixed by Ralf. --- ~Randy *** Remember to use Documentation/SubmitChecklist when testing your code ***

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Re: [alsa-devel] Build regressions/improvements in v3.0-rc7 (sound/isa/)
by Takashi Iwai 28 Jul '11

28 Jul '11

At Thu, 28 Jul 2011 13:06:19 -0400, Arnaud Lacombe wrote: > > Hi, > > On Thu, Jul 28, 2011 at 12:44 PM, Randy Dunlap <rdunlap(a)xenotime.net> wrote: > > On Thu, 28 Jul 2011 12:36:27 -0400 Arnaud Lacombe wrote: > > > >> Hi, > >> > >> On Thu, Jul 28, 2011 at 12:21 PM, Randy Dunlap <rdunlap(a)xenotime.net> wrote: > >> > On Thu, 28 Jul 2011 11:17:09 +1000 Stephen Rothwell wrote: > >> > > >> >> Hi Randy, > >> >> > >> >> On Wed, 27 Jul 2011 12:13:20 -0700 Randy Dunlap <rdunlap(a)xenotime.net> wrote: > >> >> > > >> >> > On Mon, 25 Jul 2011 09:10:03 -0700 Randy Dunlap wrote: > >> >> > > >> >> > > On Mon, 25 Jul 2011 09:37:53 +0200 Takashi Iwai wrote: > >> >> > > > >> >> > > > At Sun, 24 Jul 2011 12:26:24 -0700, > >> >> > > > Randy Dunlap wrote: > >> >> > > > > > >> >> > > > > On Wed, 13 Jul 2011 21:33:30 +0200 Geert Uytterhoeven wrote: > >> >> > > > > > >> >> > > > > > + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit declaration of function 'snd_dma_disable': => 544 > >> >> > > > > > + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit declaration of function 'snd_dma_pointer': => 302 > >> >> > > > > > + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit declaration of function 'snd_dma_program': => 244 > >> >> > > > > > + src/sound/isa/es1688/es1688_lib.c: error: implicit declaration of function 'snd_dma_pointer': => 509 > >> >> > > > > > + src/sound/isa/es1688/es1688_lib.c: error: implicit declaration of function 'snd_dma_program': => 417 > >> >> > > > > > + src/sound/isa/gus/gus_dma.c: error: implicit declaration of function 'snd_dma_disable': => 177 > >> >> > > > > > + src/sound/isa/gus/gus_dma.c: error: implicit declaration of function 'snd_dma_program': => 79 > >> >> > > > > > + src/sound/isa/gus/gus_pcm.c: error: implicit declaration of function 'snd_dma_pointer': => 619 > >> >> > > > > > + src/sound/isa/gus/gus_pcm.c: error: implicit declaration of function 'snd_dma_program': => 591 > >> >> > > > > > + src/sound/isa/sb/sb16_main.c: error: implicit declaration of function 'snd_dma_pointer': => 456 > >> >> > > > > > + src/sound/isa/sb/sb16_main.c: error: implicit declaration of function 'snd_dma_program': => 276 > >> >> > > > > > + src/sound/isa/sb/sb8_main.c: error: implicit declaration of function 'snd_dma_pointer': => 425 > >> >> > > > > > + src/sound/isa/sb/sb8_main.c: error: implicit declaration of function 'snd_dma_program': => 172 > >> >> > > > > > + src/sound/isa/sscape.c: error: implicit declaration of function 'snd_dma_program': => 481 > >> >> > > > > > + src/sound/isa/wss/wss_lib.c: error: implicit declaration of function 'snd_dma_disable': => 1695 > >> >> > > > > > + src/sound/isa/wss/wss_lib.c: error: implicit declaration of function 'snd_dma_pointer': => 1160 > >> >> > > > > > + src/sound/isa/wss/wss_lib.c: error: implicit declaration of function 'snd_dma_program': => 1025 > >> >> > > > > > >> >> > > > > All of these source files do #include <sound/core.h>, > >> >> > > > > but these missing/implicit functions are conditional: > >> >> > > > > > >> >> > > > > #ifdef CONFIG_ISA_DMA_API > >> >> > > > > #define DMA_MODE_NO_ENABLE 0x0100 > >> >> > > > > > >> >> > > > > void snd_dma_program(unsigned long dma, unsigned long addr, unsigned int size, unsigned short mode); > >> >> > > > > void snd_dma_disable(unsigned long dma); > >> >> > > > > unsigned int snd_dma_pointer(unsigned long dma, unsigned int size); > >> >> > > > > #endif > >> >> > > > > > >> >> > > > > so should these drivers all depend on ISA_DMA_API ? > >> >> > > > > >> >> > > > Yes, these are ISA sound cards, so they must have CONFIG_ISA and > >> >> > > > CONFIG_ISA_DMA. Although CONFIG_SND_SB8_DSP, CONFIG_SND_SB16_DSP > >> >> > > > and CONFIG_SND_WSS_LIB have no direct dependency, but the drivers > >> >> > > > selecting these are all in "if SND_ISA" in sound/isa/Kconfig. > >> >> > > > >> >> > > Hm, yes, they are. > >> >> > > > >> >> > > > > or should only pieces of each driver depend on ISA_DMA_API ? > >> >> > > > > >> >> > > > I wonder which configuration makes it possible. > >> >> > > > Did you see a similar problem with randconfig? > >> >> > > > >> >> > > No. This parisc allmodconfig file from linux-next builds has > >> >> > > CONFIG_SND_ISA=y but it does not have CONFIG_ISA_DMA_API enabled. Weird. > >> >> > > > >> >> > > Geert, do you know how this happens? > >> >> > > > >> >> > > parisc config: http://kisskb.ellerman.id.au/kisskb/buildresult/4323458/config/ > >> >> > > from this failed build: http://kisskb.ellerman.id.au/kisskb/buildresult/4323458/ > >> >> > > >> >> > How is CONFIG_ISA_DMA_API enabled in parisc allmodconfig? > >> >> > I don't see how that happens, and it's causing build errors... > >> >> > >> >> Sorry, but I don't understand the question. The problam is that > >> >> CONFIG_ISA_DMA_API is not enabled, right? On parisc, it is just (from > >> >> arch/parisc/Kconfig) > >> >> > >> >> config ISA_DMA_API > >> >> bool > >> >> > >> >> and nothing selects it. However, I would have expected an allmodconfig > >> >> to set it to y. Clearly I don't understan how allmodconfig works any > >> >> more :-( > >> >> > >> >> Michal, is my understanding correct? Is this a bug in allmodconfig? > >> > > >> > When I do: > >> > $ make ARCH=parisc O=PAR allmodconfig > >> > > >> > I don't get CONFIG_ISA_DMA_API enabled, so the question is simply > >> > how is the linux-next build system ending up with CONFIG_ISA_DMA_API enabled? > >> > > >> Where do you see that "linux-next build system ending up with > >> CONFIG_ISA_DMA_API enabled" ? > >> > >> From what I understood of the problem, the issue is that > >> CONFIG_ISA_DMA_API does _not_ ends up enabled. > > > > Thanks for catching me. > > > > The problem (second try) is that ISA SND drivers are being built (on parisc) > > even though ISA_DMA_API is not enabled -- and should not be enabled, > > so the ISA SND drivers should not be built. I think I have it right this time. > > > > From the parisc .config file listed above: > > > > CONFIG_SND_ISA=y > > CONFIG_SND_ADLIB=m > > CONFIG_SND_AD1816A=m > > CONFIG_SND_AD1848=m > > CONFIG_SND_ALS100=m > > CONFIG_SND_AZT1605=m > > CONFIG_SND_AZT2316=m > > CONFIG_SND_AZT2320=m > > CONFIG_SND_CMI8330=m > > > > > > etc. But none of these should be enabled. AFAIK. > > So how does this happen? > > > warning: (RADIO_MIROPCM20) selects SND_ISA which has unmet direct > dependencies (SOUND && !M68K && SND && ISA && ISA_DMA_API) Hm, I thought this was fixed by commit 4756fc64ecd628efa6f8296711d7dbcefabf25dd Author: Ralf Baechle <ralf(a)linux-mips.org> Date: Fri Jun 24 10:30:09 2011 -0300 [media] MEDIA: Fix non-ISA_DMA_API link failure of sound code OK, it landed after 3.0-rc7, so this problem must be fixed now? I'm not sure at which point this thread is focused on. Takashi

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Re: [alsa-devel] Build regressions/improvements in v3.0-rc7 (sound/isa/)
by Randy Dunlap 28 Jul '11

28 Jul '11

On Thu, 28 Jul 2011 12:36:27 -0400 Arnaud Lacombe wrote: > Hi, >=20 > On Thu, Jul 28, 2011 at 12:21 PM, Randy Dunlap <rdunlap(a)xenotime.net> wro= te: > > On Thu, 28 Jul 2011 11:17:09 +1000 Stephen Rothwell wrote: > > > >> Hi Randy, > >> > >> On Wed, 27 Jul 2011 12:13:20 -0700 Randy Dunlap <rdunlap(a)xenotime.net>= wrote: > >> > > >> > On Mon, 25 Jul 2011 09:10:03 -0700 Randy Dunlap wrote: > >> > > >> > > On Mon, 25 Jul 2011 09:37:53 +0200 Takashi Iwai wrote: > >> > > > >> > > > At Sun, 24 Jul 2011 12:26:24 -0700, > >> > > > Randy Dunlap wrote: > >> > > > > > >> > > > > On Wed, 13 Jul 2011 21:33:30 +0200 Geert Uytterhoeven wrote: > >> > > > > > >> > > > > > =A0 + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit d= eclaration of function 'snd_dma_disable': =A0=3D> 544 > >> > > > > > =A0 + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit d= eclaration of function 'snd_dma_pointer': =A0=3D> 302 > >> > > > > > =A0 + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit d= eclaration of function 'snd_dma_program': =A0=3D> 244 > >> > > > > > =A0 + src/sound/isa/es1688/es1688_lib.c: error: implicit dec= laration of function 'snd_dma_pointer': =A0=3D> 509 > >> > > > > > =A0 + src/sound/isa/es1688/es1688_lib.c: error: implicit dec= laration of function 'snd_dma_program': =A0=3D> 417 > >> > > > > > =A0 + src/sound/isa/gus/gus_dma.c: error: implicit declarati= on of function 'snd_dma_disable': =A0=3D> 177 > >> > > > > > =A0 + src/sound/isa/gus/gus_dma.c: error: implicit declarati= on of function 'snd_dma_program': =A0=3D> 79 > >> > > > > > =A0 + src/sound/isa/gus/gus_pcm.c: error: implicit declarati= on of function 'snd_dma_pointer': =A0=3D> 619 > >> > > > > > =A0 + src/sound/isa/gus/gus_pcm.c: error: implicit declarati= on of function 'snd_dma_program': =A0=3D> 591 > >> > > > > > =A0 + src/sound/isa/sb/sb16_main.c: error: implicit declarat= ion of function 'snd_dma_pointer': =A0=3D> 456 > >> > > > > > =A0 + src/sound/isa/sb/sb16_main.c: error: implicit declarat= ion of function 'snd_dma_program': =A0=3D> 276 > >> > > > > > =A0 + src/sound/isa/sb/sb8_main.c: error: implicit declarati= on of function 'snd_dma_pointer': =A0=3D> 425 > >> > > > > > =A0 + src/sound/isa/sb/sb8_main.c: error: implicit declarati= on of function 'snd_dma_program': =A0=3D> 172 > >> > > > > > =A0 + src/sound/isa/sscape.c: error: implicit declaration of= function 'snd_dma_program': =A0=3D> 481 > >> > > > > > =A0 + src/sound/isa/wss/wss_lib.c: error: implicit declarati= on of function 'snd_dma_disable': =A0=3D> 1695 > >> > > > > > =A0 + src/sound/isa/wss/wss_lib.c: error: implicit declarati= on of function 'snd_dma_pointer': =A0=3D> 1160 > >> > > > > > =A0 + src/sound/isa/wss/wss_lib.c: error: implicit declarati= on of function 'snd_dma_program': =A0=3D> 1025 > >> > > > > > >> > > > > All of these source files do #include <sound/core.h>, > >> > > > > but these missing/implicit functions are conditional: > >> > > > > > >> > > > > #ifdef CONFIG_ISA_DMA_API > >> > > > > #define DMA_MODE_NO_ENABLE =A0 =A00x0100 > >> > > > > > >> > > > > void snd_dma_program(unsigned long dma, unsigned long addr, un= signed int size, unsigned short mode); > >> > > > > void snd_dma_disable(unsigned long dma); > >> > > > > unsigned int snd_dma_pointer(unsigned long dma, unsigned int s= ize); > >> > > > > #endif > >> > > > > > >> > > > > so should these drivers all depend on ISA_DMA_API ? > >> > > > > >> > > > Yes, these are ISA sound cards, so they must have CONFIG_ISA and > >> > > > CONFIG_ISA_DMA. =A0Although CONFIG_SND_SB8_DSP, CONFIG_SND_SB16_= DSP > >> > > > and CONFIG_SND_WSS_LIB have no direct dependency, but the drivers > >> > > > selecting these are all in "if SND_ISA" in sound/isa/Kconfig. > >> > > > >> > > Hm, yes, they are. > >> > > > >> > > > > or should only pieces of each driver depend on ISA_DMA_API ? > >> > > > > >> > > > I wonder which configuration makes it possible. > >> > > > Did you see a similar problem with randconfig? > >> > > > >> > > No. =A0This parisc allmodconfig file from linux-next builds has > >> > > CONFIG_SND_ISA=3Dy but it does not have CONFIG_ISA_DMA_API enabled= . =A0Weird. > >> > > > >> > > Geert, do you know how this happens? > >> > > > >> > > parisc config: =A0http://kisskb.ellerman.id.au/kisskb/buildresult/= 4323458/config/ > >> > > from this failed build: =A0http://kisskb.ellerman.id.au/kisskb/bui= ldresult/4323458/ > >> > > >> > How is CONFIG_ISA_DMA_API enabled in parisc allmodconfig? > >> > I don't see how that happens, and it's causing build errors... > >> > >> Sorry, but I don't understand the question. =A0The problam is that > >> CONFIG_ISA_DMA_API is not enabled, right? =A0On parisc, it is just (fr= om > >> arch/parisc/Kconfig) > >> > >> config ISA_DMA_API > >> =A0 =A0 =A0 bool > >> > >> and nothing selects it. =A0However, I would have expected an allmodcon= fig > >> to set it to y. =A0Clearly I don't understan how allmodconfig works any > >> more :-( > >> > >> Michal, is my understanding correct? =A0Is this a bug in allmodconfig? > > > > When I do: > > $ make ARCH=3Dparisc O=3DPAR allmodconfig > > > > I don't get CONFIG_ISA_DMA_API enabled, so the question is simply > > how is the linux-next build system ending up with CONFIG_ISA_DMA_API en= abled? > > > Where do you see that "linux-next build system ending up with > CONFIG_ISA_DMA_API enabled" ? >=20 > From what I understood of the problem, the issue is that > CONFIG_ISA_DMA_API does _not_ ends up enabled. Thanks for catching me. The problem (second try) is that ISA SND drivers are being built (on parisc) even though ISA_DMA_API is not enabled -- and should not be enabled, so the ISA SND drivers should not be built. I think I have it right this t= ime. _From the parisc .config file listed above: CONFIG_SND_ISA=3Dy CONFIG_SND_ADLIB=3Dm CONFIG_SND_AD1816A=3Dm CONFIG_SND_AD1848=3Dm CONFIG_SND_ALS100=3Dm CONFIG_SND_AZT1605=3Dm CONFIG_SND_AZT2316=3Dm CONFIG_SND_AZT2320=3Dm CONFIG_SND_CMI8330=3Dm etc. But none of these should be enabled. AFAIK. So how does this happen? > - Arnaud >=20 > > What commands are used to cause this? --- ~Randy *** Remember to use Documentation/SubmitChecklist when testing your code ***

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Re: [alsa-devel] Build regressions/improvements in v3.0-rc7 (sound/isa/)
by Randy Dunlap 28 Jul '11

28 Jul '11

On Wed, 13 Jul 2011 21:33:30 +0200 Geert Uytterhoeven wrote: > + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit declaration of function 'snd_dma_disable': => 544 > + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit declaration of function 'snd_dma_pointer': => 302 > + src/sound/isa/ad1816a/ad1816a_lib.c: error: implicit declaration of function 'snd_dma_program': => 244 > + src/sound/isa/es1688/es1688_lib.c: error: implicit declaration of function 'snd_dma_pointer': => 509 > + src/sound/isa/es1688/es1688_lib.c: error: implicit declaration of function 'snd_dma_program': => 417 > + src/sound/isa/gus/gus_dma.c: error: implicit declaration of function 'snd_dma_disable': => 177 > + src/sound/isa/gus/gus_dma.c: error: implicit declaration of function 'snd_dma_program': => 79 > + src/sound/isa/gus/gus_pcm.c: error: implicit declaration of function 'snd_dma_pointer': => 619 > + src/sound/isa/gus/gus_pcm.c: error: implicit declaration of function 'snd_dma_program': => 591 > + src/sound/isa/sb/sb16_main.c: error: implicit declaration of function 'snd_dma_pointer': => 456 > + src/sound/isa/sb/sb16_main.c: error: implicit declaration of function 'snd_dma_program': => 276 > + src/sound/isa/sb/sb8_main.c: error: implicit declaration of function 'snd_dma_pointer': => 425 > + src/sound/isa/sb/sb8_main.c: error: implicit declaration of function 'snd_dma_program': => 172 > + src/sound/isa/sscape.c: error: implicit declaration of function 'snd_dma_program': => 481 > + src/sound/isa/wss/wss_lib.c: error: implicit declaration of function 'snd_dma_disable': => 1695 > + src/sound/isa/wss/wss_lib.c: error: implicit declaration of function 'snd_dma_pointer': => 1160 > + src/sound/isa/wss/wss_lib.c: error: implicit declaration of function 'snd_dma_program': => 1025 All of these source files do #include <sound/core.h>, but these missing/implicit functions are conditional: #ifdef CONFIG_ISA_DMA_API #define DMA_MODE_NO_ENABLE 0x0100 void snd_dma_program(unsigned long dma, unsigned long addr, unsigned int size, unsigned short mode); void snd_dma_disable(unsigned long dma); unsigned int snd_dma_pointer(unsigned long dma, unsigned int size); #endif so should these drivers all depend on ISA_DMA_API ? or should only pieces of each driver depend on ISA_DMA_API ? --- ~Randy *** Remember to use Documentation/SubmitChecklist when testing your code ***

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[alsa-devel] Hardware audio decoder driver design
by Nikita V. Youshchenko 28 Jul '11

28 Jul '11

Hello. Soon I will work on a linux kernel driver for a custom audio decoder device that is being developed by a company I work for. If not going into details, that devices reads A52-encoded stream from system memory, and writes raw pcm stream to system memory. Simplest thing to do is - implement a character device, where user-space will write encoded stream, and from where user-space will read decoded stream. (driver for similar hardware, located at http://sourceforge.net/projects/vs10xx/, does that). However, perhaps a better architecture (e.g. in-kernel intergation with an audio sink) is possible? I'm looking for any related information - e.g. ideas on what interface to implement, examples of drivers for similar devices, etc. Thanks on any hints. Nikita Youshchenko, embedded linux developer.

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