May 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] AT32AP7000 (AVR32) + WM8510
by Geoffrey Wossum 28 May '08

28 May '08

Hi all, I'm attempting to bring the sound up on a board using an Atmel AT32AP7000 (AVR32) microcontroller with a Wolfson WM8510 mono CODEC. I'm using the Linux 2.6.24.3.atmel.3 kernel, the latest available for the AVR32. I've patched this kernel with the ASoC subsystem and WM8510 driver from git://opensource.wolfsonmicro.com/linux-2.6-asoc, as it appeared when v2.6.24 was tagged. I've also written the platform driver for the AT32, which is very close to the AT91 driver, and a machine driver for my specific hardware. I'm letting the WM8510 provide the I2S clocking from its internal PLL. I'm having issues getting the sound to work. I found what I believe are several bugs in the wm8510.c. I also had to lie about the WM8510 supporting 2 channels. Otherwise, the upper ALSA seem to include only 1 channel in the data stream, and the WM8510 seems to still expect 2 channels in the I2S stream. A patch is found a the bottom of this email. With this patch, the registers on the WM8510 seem to be getting configured properly (all the clock rates and such look correct), but I get no signal at the speaker outputs or the mono output, no matter what the mixer settings are. They just sit VMID after the initial power up of the chip. Is anyone else using the WM8510? Any hints or suggestions? TIA, --- Geoffrey Index: linux-2.6.24.3.atmel.3/sound/soc/codecs/wm8510.c =================================================================== --- linux-2.6.24.3.atmel.3.orig/sound/soc/codecs/wm8510.c +++ linux-2.6.24.3.atmel.3/sound/soc/codecs/wm8510.c @@ -186,7 +186,7 @@ SOC_SINGLE("Speaker Playback Volume", WM SOC_SINGLE("Speaker Boost", WM8510_OUTPUT, 2, 1, 0), SOC_SINGLE("Capture Boost(+20dB)", WM8510_ADCBOOST, 8, 1, 0), -SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 0), +SOC_SINGLE("Mono Playback Switch", WM8510_MONOMIX, 6, 1, 1), }; /* add non dapm controls */ @@ -209,14 +209,14 @@ static int wm8510_add_controls(struct sn static const struct snd_kcontrol_new wm8510_speaker_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_SPKMIX, 1, 1, 0), SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_SPKMIX, 5, 1, 0), -SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_SPKMIX, 0, 1, 1), +SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_SPKMIX, 0, 1, 0), }; /* Mono Output Mixer */ static const struct snd_kcontrol_new wm8510_mono_mixer_controls[] = { SOC_DAPM_SINGLE("Line Bypass Switch", WM8510_MONOMIX, 1, 1, 0), SOC_DAPM_SINGLE("Aux Playback Switch", WM8510_MONOMIX, 2, 1, 0), -SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 1), +SOC_DAPM_SINGLE("PCM Playback Switch", WM8510_MONOMIX, 0, 1, 0), }; /* AUX Input boost vol */ @@ -251,7 +251,7 @@ SND_SOC_DAPM_MIXER("Mono Mixer", WM8510_ &wm8510_mono_mixer_controls[0], ARRAY_SIZE(wm8510_mono_mixer_controls)), SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8510_POWER3, 0, 0), -SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8510_POWER3, 0, 0), +SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8510_POWER2, 0, 0), SND_SOC_DAPM_PGA("Aux Input", WM8510_POWER1, 6, 0, NULL, 0), SND_SOC_DAPM_PGA("SpkN Out", WM8510_POWER3, 5, 0, NULL, 0), SND_SOC_DAPM_PGA("SpkP Out", WM8510_POWER3, 6, 0, NULL, 0), @@ -393,8 +393,8 @@ static int wm8510_set_dai_pll(struct snd wm8510_write(codec, WM8510_PLLN, (pll_div.pre_div << 4) | pll_div.n); wm8510_write(codec, WM8510_PLLK1, pll_div.k >> 18); - wm8510_write(codec, WM8510_PLLK1, (pll_div.k >> 9) && 0x1ff); - wm8510_write(codec, WM8510_PLLK1, pll_div.k && 0x1ff); + wm8510_write(codec, WM8510_PLLK2, (pll_div.k >> 9) & 0x1ff); + wm8510_write(codec, WM8510_PLLK3, pll_div.k & 0x1ff); reg = wm8510_read_reg_cache(codec, WM8510_POWER1); wm8510_write(codec, WM8510_POWER1, reg | 0x020); return 0; @@ -412,23 +412,23 @@ static int wm8510_set_dai_clkdiv(struct switch (div_id) { case WM8510_OPCLKDIV: - reg = wm8510_read_reg_cache(codec, WM8510_GPIO & 0x1cf); + reg = wm8510_read_reg_cache(codec, WM8510_GPIO) & 0x1cf; wm8510_write(codec, WM8510_GPIO, reg | div); break; case WM8510_MCLKDIV: - reg = wm8510_read_reg_cache(codec, WM8510_CLOCK & 0x1f); + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1f; wm8510_write(codec, WM8510_CLOCK, reg | div); break; case WM8510_ADCCLK: - reg = wm8510_read_reg_cache(codec, WM8510_ADC & 0x1f7); + reg = wm8510_read_reg_cache(codec, WM8510_ADC) & 0x1f7; wm8510_write(codec, WM8510_ADC, reg | div); break; case WM8510_DACCLK: - reg = wm8510_read_reg_cache(codec, WM8510_DAC & 0x1f7); + reg = wm8510_read_reg_cache(codec, WM8510_DAC) & 0x1f7; wm8510_write(codec, WM8510_DAC, reg | div); break; case WM8510_BCLKDIV: - reg = wm8510_read_reg_cache(codec, WM8510_CLOCK & 0x1e3); + reg = wm8510_read_reg_cache(codec, WM8510_CLOCK) & 0x1e3; wm8510_write(codec, WM8510_CLOCK, reg | div); break; default: @@ -597,14 +597,14 @@ struct snd_soc_codec_dai wm8510_dai = { .name = "WM8510 HiFi", .playback = { .stream_name = "Playback", - .channels_min = 1, - .channels_max = 1, + .channels_min = 2, + .channels_max = 2, .rates = WM8510_RATES, .formats = WM8510_FORMATS,}, .capture = { .stream_name = "Capture", - .channels_min = 1, - .channels_max = 1, + .channels_min = 2, + .channels_max = 2, .rates = WM8510_RATES, .formats = WM8510_FORMATS,}, .ops = {

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Re: [alsa-devel] bandwidth issue with usb-audio in alsa
by Kunal Singh 27 May '08

27 May '08

Hi, I have some update on the issue which I had reported earlier. In my last e-mail, I had not mentioned that iPod is connected to the target through a USB (HS) Hub. When I connect the same hub to my Desktop and read its configuration using "usbview" - I see that "interval" for this hub is mentioned as "256 msec". (I have two hubs and I get all the results same with both the hubs). With 36 URBs, the gap between two set of transactions is appx 220 msec (220+36 => 256). So, I am wondering if the problem which I am seeing is because the external hub. I can not connect iPod directly to my target board, because this board does not support enough current (as needed by iPod). Can some one suggest, where is the problem - is it with the host controller driver? Regards, Kunal Singh ----------------------------- Kunal Singh wrote: Hi, My Platform ------------- I am trying to stream audio data from a USB-device (an iPod) to my arm-based embedded device (running kernel 2.6.23). The USB-device confirms to usb digital audio (subclass - streaming) class. The endpoint buffer size 196 bytes and transfer period is 1 msec. So, I should be reading data every msec. Problem ----------- I am facing some bandwidth issues. The data is not being captured fast enough. I notice that once one set of URBs are captured (one set equal to MAX_URBS), there is a significant delay in capturing the next set of URBs. I have set MAX_URBS equal to 32. When I connect a USB-Analyzer and analyze the timings, I can see that 32 URBs are captured at a interval of 1 msec each. After that there is a break of around 250 msec. And then again 32 URBs captured in sequence. If I change the MAX_URBs to any other value N, N URBs get captured at interval of 1 msec each, and then there is a break. Can some one suggest a way to improve this problem. Is there some configuration or parameter which I can change, so that URBs get submitted fast enough? May be as soon as the old one has retired. Thanks and Regards, Kunal www.singhkunal.com

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Re: [alsa-devel] [Alsa-user] ICH8 HDA not detected after upgrade from Ubuntu 7.10 to 8.04
by Takashi Iwai 27 May '08

27 May '08

At Tue, 27 May 2008 15:17:40 +0200, Janto Schraiber wrote: > > Takashi Iwai wrote: > > You should have more messages than that. It's just the end result, > > not the cause... > > I'm not sure what's caused this, but I guess this is the whole part: > > [ 41.554112] ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ > 21 > [ 41.554119] PCI: Unable to reserve mem region #1:4000@febf8000 for device > 0000:00:1b.0 So, this is no sound-driver but rather a deeper kernel problem. Please report this on kernel bugzilla, or let distro bug master to point to there. Takashi

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Re: [alsa-devel] problem with usb-audio and oss emulation
by Kunal Singh 27 May '08

27 May '08

Hi All, The first issue was resolved by disabling the "CONFIG_SND_PCM_OSS_PLUGINS <http://lxr.linux.no/linux/+code=CONFIG_SND_PCM_OSS_PLUGINS>".i.e. disable the OSS plug-in support under KernelConfigMenu->sound->alsa CAN SOME ONE PLEASE HELP ON THIS? (2) Once I connect the iPod to my target, I can not any more use the default sound card. I read on the oss-emulation tutorial that this could be a problem with "dsp_map". But I am not able to check the current settings. Thanks and Regards, Kunal Date: Mon, 26 May 2008 20:10:11 +0530 From: Kunal Singh <kunal(a)singhkunal.com> Subject: [alsa-devel] problem with usb-audio and oss emulation To: alsa-devel(a)alsa-project.org Message-ID: <483ACBCB.2020800(a)singhkunal.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi, I am working on a ARM based target with 2.6.23 Kernel. My project involves streaming data from a USB-Audio Device (an iPod) and playing it back on the target. I am using OSS-emulation API to communicate with the audio devices. What I am able to do: (1) I have a sound card in my target. This is mapped to card0. I can open the "dsp" device and can playback any audio data on this. (2) I have enabled the usb-audio module support in kernel configuration. When I connect iPod to the target, I can see a new card "card1" being added in the /proc/asound/ directory. (3) I can open the "dsp1" device and can read from it. When I connect a "USB-Analyzer" I can see some valid data being streamed to the target. However, I am stuck with following issues: (1) The data being received on the target (in user mode) is junk. I put some printfs in the function "retire_capture_urb" in file "/linux/sound/usb/usbaudio.c", and I can see that the "data" which I see through usb-analzyer is reaching till this point. However, there seem to be some problem with one of the "snd_pmc_oss_ReadN" functions in file "http://lxr.linux.no/linux/sound/core/oss/pcm_oss.c". I am not able to exactly trace the problem, but I think that the "byte count" and "buffer address" being passed to snd-usb-audio is getting messed up some where. Has any one faced similar problems before? (2) Once I connect the iPod to my target, I can not any more use the default sound card. I read on the oss-emulation tutorial that this could be a problem with "dsp_map". But I am not able to check the current settings. Can some one give me some pointers for further debugging? Thanks and Regards, Kunal Singh www.singhkunal.com

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[alsa-devel] snd_kcontrol questions
by Harsha priya gupta 27 May '08

27 May '08

Hi, Can anyone help me to with the following questions on controls? 1. I want to define a snd_kcontrol that is of type mono. I am basically trying to define a mono output control that is exposed in KMIX and whose 'split channel' does not split into two channels. 2. My input controls give out 9 channels when i say 'split channels' in kmix. can anyone help me how to fix that to 2 channels 3. I would want to get the individual setting of each channel from KMIX into my ALSA driver. How do I get them? will .get and .set functions help in this? -- -Harsha

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[alsa-devel] bandwidth issue with usb-audio in alsa
by kunal＠singhkunal.com 27 May '08

27 May '08

Hi, My Platform ------------- I am trying to stream audio data from a USB-device (an iPod) to my arm-based embedded device (running kernel 2.6.23). The USB-device confirms to usb digital audio (subclass - streaming) class. The endpoint buffer size 196 bytes and transfer period is 1 msec. So, I should be reading data every msec. Problem ----------- I am facing some bandwidth issues. The data is not being captured fast enough. I notice that once one set of URBs are captured (one set equal to MAX_URBS), there is a significant delay in capturing the next set of URBs. I have set MAX_URBS equal to 32. When I connect a USB-Analyzer and analyze the timings, I can see that 32 URBs are captured at a interval of 1 msec each. After that there is a break of around 250 msec. And then again 32 URBs captured in sequence. If I change the MAX_URBs to any other value N, N URBs get captured at interval of 1 msec each, and then there is a break. Can some one suggest a way to improve this problem. Is there some configuration or parameter which I can change, so that URBs get submitted fast enough? May be as soon as the old one has retired. Thanks and Regards, Kunal www.singhkunal.com -------------------------------------------------------------------- mail2web.com What can On Demand Business Solutions do for you? http://link.mail2web.com/Business/SharePoint

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[alsa-devel] Recording problem with ALSA: trying to set up some constraints
by Sedji Gaouaou 26 May '08

26 May '08

Hi, I have a problem with my recording driver. I have set up a pcm_hardware structure in which I define the format supported by my hardware: static struct snd_pcm_hardware snd_at91_ac97_capture_hw = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BLOCK_TRANSFER), .formats = (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | SNDRV_PCM_FMTBIT_U20_3BE | SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_U18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | SNDRV_PCM_FMTBIT_U18_3BE), .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, .rate_max = 48000, .channels_min = 1, .channels_max = 2, .buffer_bytes_max = 256*1024, .period_bytes_min = 1024, .period_bytes_max = 4*1024, .periods_min = 1, .periods_max = 64, }; Then in my open function I pass it to the runtime struct. Nevertheless if I try to run arecord with this option: arecord -d 1 test.wav, I will have a kernel oops. The default format is U8, but it is not supported by my Hardware, abd it is not defined in my structure, so I though that it should exit arecord but not crash. Do I miss something when defining my pcm_hardware strucutre, or do I forgot to use a specific function which should return me an error? I have try to set up some constraints as well(for the rate and the format), but still I have the same crash. Regards, Sedji

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[alsa-devel] Surround and SPDIF on STAC9277
by Jon Smirl 26 May '08

26 May '08

How are you supposed to get 5.1 audio out of your PC? When I hook the SPDIF device from my STAC9227/HDA audio up to pulse, pulse says that the SPDIF channel only supports stereo. So I go digging around the in STAC9227 datasheet. http://www.idt.com/?genID=STAC9227 STAC9227 codec only supports 192Khz SPDIF with 4/1*48Khz channels. It doesn't support 384Khz with 8/1*48Khz channels. So there are two problems, first the existing Intel HDA/STAC9227 driver is not exposing the 4 channel SPDIF support. Or I can't figure out how to get to it. Second, the hardware doesn't seem to support any way to get digital 5.1 audio out of the codec and into my Sony amp. STAC9227 does have six channel analog out, but my Sony amp doesn't have 6 channel analog in. Sony amp supports 384Khz 8-channel SPDIF in. -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] Building alsa drivers as part of alsa vs kernel external module
by Ben Stanley 26 May '08

26 May '08

Hi, I'm trying to compile the AudioScience driver as an external kernel module (i.e. using the alsa in the kernel, rather than as a module within the alsa tree). What source code transformations can I expect to have to make? Is there some documentation anywhere on what transformations are performed as standard? Or is there a proper way that the module should be written so that it compiles both ways? Of course, I understand that patches will most likely be required for the module to compile against older kernels in particular. This kind of problem is usually dealt with by autoconf. Is there a standard way of doing this for just compiling a module directly for the kernel? I see that alsa has its own set of autoconf macros, and when I compile directly for the kernel (and its embedded alsa) I will lose the benefit of that. Ben Stanley.

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[alsa-devel] problem with usb-audio and oss emulation
by Kunal Singh 26 May '08

26 May '08

Hi, I am working on a ARM based target with 2.6.23 Kernel. My project involves streaming data from a USB-Audio Device (an iPod) and playing it back on the target. I am using OSS-emulation API to communicate with the audio devices. What I am able to do: (1) I have a sound card in my target. This is mapped to card0. I can open the "dsp" device and can playback any audio data on this. (2) I have enabled the usb-audio module support in kernel configuration. When I connect iPod to the target, I can see a new card "card1" being added in the /proc/asound/ directory. (3) I can open the "dsp1" device and can read from it. When I connect a "USB-Analyzer" I can see some valid data being streamed to the target. However, I am stuck with following issues: (1) The data being received on the target (in user mode) is junk. I put some printfs in the function "retire_capture_urb" in file "/linux/sound/usb/usbaudio.c", and I can see that the "data" which I see through usb-analzyer is reaching till this point. However, there seem to be some problem with one of the "snd_pmc_oss_ReadN" functions in file "http://lxr.linux.no/linux/sound/core/oss/pcm_oss.c". I am not able to exactly trace the problem, but I think that the "byte count" and "buffer address" being passed to snd-usb-audio is getting messed up some where. Has any one faced similar problems before? (2) Once I connect the iPod to my target, I can not any more use the default sound card. I read on the oss-emulation tutorial that this could be a problem with "dsp_map". But I am not able to check the current settings. Can some one give me some pointers for further debugging? Thanks and Regards, Kunal Singh www.singhkunal.com

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