April 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Backported sbxfi driver (UNTESTED!)
by Takashi Iwai 12 Jun '09

12 Jun '09

Hi, $SUBJECT is now on my sound-unstable git tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git together with other experimental patches. If you're using 2.6.27-rc* git tree, pull the master branch of the tree above into yours, and run make oldconfig. That is, % cd /your/git-tree % git pull git://...../sound-unstable-2.6.git master If you are not using 2.6.27-rc*, or not familiar with git, you can try alsa-driver-unstable snapshot tarball available at ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ Run configure, make and make install as usual ALSA-driver tarball. (I didn't check whether the tarball correctly includes the sbxfi stuff. It should have been generated automatically. If not, wait for a while. If it still doesn't include sbxfi code, please report. I'll fix it tomorrow morning.) The driver is built only for 2.6.26 or later. If you have an older kernel, edit alsa-driver*/kconfig-vers and change the version of CONFIG_SND_SBXFI to 2.6.24 or whatever you want. Then run ./gitcompile, instead of configure in this case to update the configure script. **NOTE** The driver is totally untested. It's just compiled without errors, but not reviewed after a quick writing. So, don't expect it ever runs at the first try. A crash is highly possible. There are some build conditions found in sound/pci/sbxfi/sbxfi.c, starting with XXX_*. You can change it if you want. As default, it's for non-fullduplex but accept different rates. Not sure whether this works at all. Any test- (and better debugging-) reports are appreciated. thanks, Takashi

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[alsa-devel] Roland/Edirol M-16DX
by Lasse Kärkkäinen 01 Jun '09

01 Jun '09

This device doesn't seem to be supported yet. Does Roland make the specifications available, etc? The device is not compatible with USB Audio, but rather uses Vendor Specific Class. lsusb -vvv is attached below and if you need more information, I can debug the device on Linux and on Vista. Bus 007 Device 025: ID 0582:00c4 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 0x00c4 bcdDevice 0.00 iManufacturer 1 EDIROL iProduct 2 M-16DX 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 1 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 80 bb 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0038 1x 56 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 2 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 12 04 18 01 80 bb 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x01f8 1x 504 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 3 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 01 01 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x04 EP 4 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 0x83 EP 3 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 3 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x04 EP 4 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 0x83 EP 3 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1

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[alsa-devel] skype and old alsa RC4 symbols
by Jon Smirl 27 May '09

27 May '09

I would like to run pulseaudio 9.15 contained in this ubuntu repository since it fixes my problems with S/PDIF https://launchpad.net/~themuso/+archive/ppa However, the 32b compatibility library in this build don't work with skype since it is missing some RC4 obsolete symbols. jonsmirl@terra:/home/apps/alsa-lib-1.0.19/debian/lib32asound2/usr/lib32$ readelf -W -s libasound.so.2.0.0 | grep snd_pcm_hw_params_get_channels_min 174: 0004a970 58 FUNC WEAK DEFAULT 12 snd_pcm_hw_params_get_channels_min@@ALSA_0.9 454: 0004a970 58 FUNC GLOBAL DEFAULT 12 __snd_pcm_hw_params_get_channels_min@@ALSA_0.9 The 64b library has the missing RC4 symbols. jonsmirl@terra:/home/apps/alsa-lib-1.0.19/debian/libasound2/usr/lib$ readelf -W -s libasound.so.2.0.0 | grep snd_pcm_hw_params_get_channels_min 173: 000000000004ff10 15 FUNC GLOBAL DEFAULT 12 snd_pcm_hw_params_get_channels_min@@ALSA_0.9.0rc4 177: 000000000004ff20 31 FUNC GLOBAL DEFAULT 12 snd_pcm_hw_params_get_channels_min(a)ALSA_0.9 462: 000000000004ff10 15 FUNC GLOBAL DEFAULT 12 __snd_pcm_hw_params_get_channels_min@@ALSA_0.9 jonsmirl@terra:/home/apps/alsa-lib-1.0.19/debian/libasound2/usr/lib$ How do I get the RC4 symbols into the 32 library? I can pull down the source for this and build without problem. I just can't identify what needs to be changed to make the symbols appear in the 32b compat lib. -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] Alsamixer isn't working properly with HDA VIA VT82xx with Realtek ALC888 Codec
by Stephan Jänecke 25 May '09

25 May '09

Dear developers, alsamixer doesn't work on my fedora machine. When I type in 'alsamixer -c 0', I get "hda_codec: num-steps=0 for NID=0xf (ctl=Side Playback Volume)" "alsamixer: function snd_mixer_load failed: Invalid argument" Although Pulseaudio and OSS Mixers seem to be present, I don't get any sound. Please have a look at http://www.alsa- project.org/db/?f=15c9f665eb141dee755a533d804456e81442f270 or check the attachment. I hope you can help me solving my problem. Regards, Stephan Jänecke PS: I had the same mixer problem with Mandriva 2009.0 before I installed Fedora 10 to check whether it's an distribution specific bug. Whereas alsamixer didn't work properly as well, I could enjoy working sound. -- Powered by openSUSE 11.1 and KMail 1.10.3

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[alsa-devel] [PATCH] ASoC: TWL4030: change DAPMs for analog microphone selection
by Joonyoung Shim 12 May '09

12 May '09

The mixer DAPM is suitable than the mux DAPM for the analog microphone registers(0x05, 0x06) because each bit for path is switchable. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 52 ++++++++++++++++--------------------------- 1 files changed, 19 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index e23c20c..a7875c2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -397,36 +397,18 @@ static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control = SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum); /* Left analog microphone selection */ -static const char *twl4030_analoglmic_texts[] = - {"Off", "Main mic", "Headset mic", "AUXL", "Carkit mic"}; - -static const unsigned int twl4030_analoglmic_values[] = - {0x0, 0x1, 0x2, 0x4, 0x8}; - -static const struct soc_enum twl4030_analoglmic_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICL, 0, 0xf, - ARRAY_SIZE(twl4030_analoglmic_texts), - twl4030_analoglmic_texts, - twl4030_analoglmic_values); - -static const struct snd_kcontrol_new twl4030_dapm_analoglmic_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_analoglmic_enum); +static const struct snd_kcontrol_new twl4030_dapm_analoglmic_controls[] = { + SOC_DAPM_SINGLE("Main mic", TWL4030_REG_ANAMICL, 0, 1, 0), + SOC_DAPM_SINGLE("Headset mic", TWL4030_REG_ANAMICL, 1, 1, 0), + SOC_DAPM_SINGLE("AUXL", TWL4030_REG_ANAMICL, 2, 1, 0), + SOC_DAPM_SINGLE("Carkit mic", TWL4030_REG_ANAMICL, 3, 1, 0), +}; /* Right analog microphone selection */ -static const char *twl4030_analogrmic_texts[] = - {"Off", "Sub mic", "AUXR"}; - -static const unsigned int twl4030_analogrmic_values[] = - {0x0, 0x1, 0x4}; - -static const struct soc_enum twl4030_analogrmic_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICR, 0, 0x5, - ARRAY_SIZE(twl4030_analogrmic_texts), - twl4030_analogrmic_texts, - twl4030_analogrmic_values); - -static const struct snd_kcontrol_new twl4030_dapm_analogrmic_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_analogrmic_enum); +static const struct snd_kcontrol_new twl4030_dapm_analogrmic_controls[] = { + SOC_DAPM_SINGLE("Sub mic", TWL4030_REG_ANAMICR, 0, 1, 0), + SOC_DAPM_SINGLE("AUXR", TWL4030_REG_ANAMICR, 1, 1, 0), +}; /* TX1 L/R Analog/Digital microphone selection */ static const char *twl4030_micpathtx1_texts[] = @@ -1034,11 +1016,15 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD| SND_SOC_DAPM_POST_REG), - /* Analog input muxes with switch for the capture amplifiers */ - SND_SOC_DAPM_VALUE_MUX("Analog Left Capture Route", - TWL4030_REG_ANAMICL, 4, 0, &twl4030_dapm_analoglmic_control), - SND_SOC_DAPM_VALUE_MUX("Analog Right Capture Route", - TWL4030_REG_ANAMICR, 4, 0, &twl4030_dapm_analogrmic_control), + /* Analog input mixers for the capture amplifiers */ + SND_SOC_DAPM_MIXER("Analog Left Capture Route", + TWL4030_REG_ANAMICL, 4, 0, + &twl4030_dapm_analoglmic_controls[0], + ARRAY_SIZE(twl4030_dapm_analoglmic_controls)), + SND_SOC_DAPM_MIXER("Analog Right Capture Route", + TWL4030_REG_ANAMICR, 4, 0, + &twl4030_dapm_analogrmic_controls[0], + ARRAY_SIZE(twl4030_dapm_analogrmic_controls)), SND_SOC_DAPM_PGA("ADC Physical Left", TWL4030_REG_AVADC_CTL, 3, 0, NULL, 0), -- 1.6.0.4

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[alsa-devel] spdif/iec958 outputs at VT1708S not recognized
by Peter Niemayer 11 May '09

11 May '09

Hi, I recently installed an Asus M4N78 mainboard, that uses a nVidia GeForce 8300 chipset and a VT1708S soundchip. Alas, the code in pci/hda/patch_via.c did not recognize the 3 existing iec958/spdif outputs, and presented only the analog devices. I had to short circuit the pin definition in vt1708S_parse_auto_config() to get the digital outputs to work: spec->multiout.max_channels = spec->multiout.num_dacs * 2; - if (spec->autocfg.dig_out_pin) + // test fails for Asus M4N78 if (spec->autocfg.dig_out_pin) spec->multiout.dig_out_nid = VT1708S_DIGOUT_NID; spec->extra_dig_out_nid = 0x15; The digital sound output works just fine for me with this change, but I guess there should be a cleaner way to make this work? Several people in forums on the InterNet reported similar problems with getting their digital outputs on VT1708S to work under Linux. Regards, Peter Niemayer

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Re: [alsa-devel] ASoC: pxa2xx-i2s
by Mark Brown 09 May '09

09 May '09

On Wed, Jan 21, 2009 at 10:46:17AM -0600, Brian G Rhodes wrote: > Mark Brown wrote: > >What is the clock master in your system? > The master is the i2s controller sysclk. The dac2x is in slave mode. > Though the i2s controller isn't directly supplying the bit clock to the > codec in this case. We're using the PLL on the dac2x. Just to confirm, the codec is the clock master on the I2S bus itself?

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Re: [alsa-devel] 2.6.29 regression: left audio channel broken after resume from suspend with Intel HDA
by Takashi Iwai 08 May '09

08 May '09

At Tue, 21 Apr 2009 20:39:36 +0200, Tino Keitel wrote: > > On Tue, Apr 21, 2009 at 09:39:13 +0200, Takashi Iwai wrote: > > At Tue, 21 Apr 2009 09:32:19 +0200, > > Tino Keitel wrote: > > > > > > On Tue, Apr 21, 2009 at 07:24:40 +0200, Takashi Iwai wrote: > > > > The commit is fcad94a4c71c36a05f4d5c6dcb174534b4e0b136: > > > > ALSA: hda - Fix the cmd cache keys for amp verbs > > > > > > Thanks for the reply. However, the patch doesn't work. I just got > > > distorted sound again. > > > > Then please run alsa-info.sh (with --no-upload option) before and > > after suspend, and attach the generated files. Also, check the latest > > Linus tree whether the problem still appears. > > My first impression with suspend seems to be wrong. It also sometimes > happens just after a while, event without suspend. > > Attached are 2 alsa-info.sh outputs. One was taken after I noticed the > distorted sound, one after I reloaded snd-hda-intel to fix the > distortion. The mixer seems to be quite different after module reload. The codec communication looks a bit unstable on this device... Does this still happen on the latest 2.6.30 version? Takashi

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[alsa-devel] [PATCH 0/2] ASoC updates
by Mark Brown 07 May '09

07 May '09

The following changes since commit 32c8dabc97d436582298ebd0e33af041c69f5a4b: Mark Brown (1): ASoC: Remove obsolete declaration of struct snd_soc_clock_info are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Daniel Mack (1): ASoC: tlv320aic3x: control additions and cleanups Grazvydas Ignotas (1): ASoC: Add support for OMAP3 Pandora sound/soc/codecs/tlv320aic3x.c | 18 ++- sound/soc/omap/Kconfig | 8 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3pandora.c | 311 ++++++++++++++++++++++++++++++++++++++++ 4 files changed, 334 insertions(+), 5 deletions(-) create mode 100644 sound/soc/omap/omap3pandora.c

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[alsa-devel] dmix and mmap
by Gabriel Johnson 06 May '09

06 May '09

I am currently working on two issues. First, I am trying to add mmaping support to the pcm driver I am using so that I can use the dmix plugin. My driver is a soc driver for the powerpc mpc5121. The pcm_new function preallocates the dma buffers with the following code: static int mpc512x_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { struct snd_pcm_substream *substream = pcm->streams[stream].substream; struct snd_dma_buffer *buf = &substream->dma_buffer; size_t size = mpc512x_pcm_hardware.buffer_bytes_max; buf->dev.type = SNDRV_DMA_TYPE_DEV; buf->dev.dev = pcm->card->dev; buf->private_data = NULL; buf->area = dma_alloc_coherent(pcm->card->dev, size, &buf->addr, GFP_KERNEL); if (!buf->area) return -ENOMEM; buf->bytes = size; return 0; } Given the way the dma buffers are allocated if I just add SNDRV_PCM_INFO_MMAP and SNDRV_PCM_INFO_MMAP_VALID to my snd_pcm_hardware struct with the alsa framework use a generic mmap function or do I still need to implement one? If I need to implement a mmap callback what is the fastest and most simple way to add that support? Second, I need to test dmix with my driver. My codec only supports S16_BE. When I try to use `aplay -Dplug:dmix -f SE16_BE mywave.raw` I get the error `requested or auto-format is not available`. After looking in pcm_direct.c it appears that when I use -Dplug:dmx it ignores -f S16_BE and is defaulting to some format that my codec doesn't support. I don't have a /etc/alsa.conf or ~/.asoundrc file and /proc/asound/modules doesn't exists. Where is aplay getting the default setting for plug:dmix? How can I override those defaults? I am using alsa-lib-1.0.11rc2. Thanks, Gabe

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