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

[alsa-devel] implement delay in ioplug alsa plugin
by Stefan Schoenleitner 19 Aug '09

19 Aug '09

Hi, I managed to write an ioplug plugin which dumps a raw PCM stream to a file (similar to the way the file or tee plugins do, but without a slave device). However, instead of immediately writing all the samples to the file, I would like to add some delay so that the samples are written to the file in real time at the frequency of the sampling rate. Thus if there is a single channel stream with 16bits being played at a sampling rate of 8kHz, each 1/8000sec exactly 16 bits should be written to the file. The reason for this is that I have a DSP attached over SPI which needs exactly one audio sample per interval. In order to talk to the DSP there is a SPI device-file which my ioplugin would use to write the samples to. What is the best way to implement such a delay ? I'm pretty sure that just adding some usleep calls() won't work very well, since the write operations to the device file probably don't take the same amount of time every time. Maybe it would work with non-blocking writes to the device file and a high precision timer ? cheers, Stefan

2 1

[alsa-devel] [PATCH] hda: enable speaker output for Compaq 6530s/6531s
by Wu Fengguang 19 Aug '09

19 Aug '09

There are 3 causes to the silence of Compaq 6530s' internal speaker: - HP pin 0x11 always return 0xffff,ffff (pin_presence=1), which automutes the speaker at module loading time. - HP pin 0x11 won't emit unsol events on jack plug/unplug, so the speaker have no chance to be unmuted. - pin 0x1c must be set to PIN_OUT mode to make the speaker work this is the most weird part - line-in pin 0x1c and speaker pin 0x16 seem to be unrelated. Fix the problems by - add "Speaker" and "Headphone" mixers to allow hand tuning by user - add the 0x1c PIN_OUT quirk Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_analog.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) --- sound-2.6.orig/sound/pci/hda/patch_analog.c 2009-08-05 09:40:13.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_analog.c 2009-08-16 16:49:39.000000000 +0800 @@ -3717,6 +3717,8 @@ static struct snd_kcontrol_new ad1884a_l HDA_CODEC_VOLUME("Master Playback Volume", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0x21, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Dock Playback Switch", 0x12, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x11, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x16, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("PCM Playback Volume", 0x20, 0x5, HDA_INPUT), HDA_CODEC_MUTE("PCM Playback Switch", 0x20, 0x5, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x20, 0x00, HDA_INPUT), @@ -3837,8 +3839,10 @@ static struct hda_verb ad1884a_laptop_ve {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, {0x0b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, /* Port-F pin */ - {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + /* required for compaq 6530s speaker output */ + {0x1c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Port-C pin - internal mic-in */ {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, 0x7002}, /* raise mic as default */ @@ -4003,6 +4007,8 @@ static struct snd_pci_quirk ad1884a_cfg_ SND_PCI_QUIRK(0x103c, 0x3030, "HP", AD1884A_MOBILE), SND_PCI_QUIRK(0x103c, 0x3037, "HP 2230s", AD1884A_LAPTOP), SND_PCI_QUIRK(0x103c, 0x3056, "HP", AD1884A_MOBILE), + SND_PCI_QUIRK(0x103c, 0x30e8, "HP Compaq 6530s", AD1884A_LAPTOP), + SND_PCI_QUIRK(0x103c, 0x30e9, "HP Compaq 6531s", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x3070, "HP", AD1884A_MOBILE), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30d0, "HP laptop", AD1884A_LAPTOP), SND_PCI_QUIRK_MASK(0x103c, 0xfff0, 0x30e0, "HP laptop", AD1884A_LAPTOP),

2 15

[alsa-devel] [PATCH] hda: add model for Intel DG45ID/DG45FC boards
by Wu Fengguang 19 Aug '09

19 Aug '09

The BIOS pin configs are in fact correct and shall not be overwritten. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/patch_sigmatel.c | 6 ++++++ 1 file changed, 6 insertions(+) --- sound-2.6.orig/sound/pci/hda/patch_sigmatel.c 2009-08-19 16:59:50.000000000 +0800 +++ sound-2.6/sound/pci/hda/patch_sigmatel.c 2009-08-19 16:59:51.000000000 +0800 @@ -78,6 +78,7 @@ enum { STAC_92HD73XX_AUTO, STAC_92HD73XX_NO_JD, /* no jack-detection */ STAC_92HD73XX_REF, + STAC_92HD73XX_INTEL, STAC_DELL_M6_AMIC, STAC_DELL_M6_DMIC, STAC_DELL_M6_BOTH, @@ -1775,6 +1776,7 @@ static const char *stac92hd73xx_models[S [STAC_92HD73XX_AUTO] = "auto", [STAC_92HD73XX_NO_JD] = "no-jd", [STAC_92HD73XX_REF] = "ref", + [STAC_92HD73XX_INTEL] = "intel", [STAC_DELL_M6_AMIC] = "dell-m6-amic", [STAC_DELL_M6_DMIC] = "dell-m6-dmic", [STAC_DELL_M6_BOTH] = "dell-m6", @@ -1787,6 +1789,10 @@ static struct snd_pci_quirk stac92hd73xx "DFI LanParty", STAC_92HD73XX_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DFI, 0x3101, "DFI LanParty", STAC_92HD73XX_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x5002, + "Intel DG45ID", STAC_92HD73XX_INTEL), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x5003, + "Intel DG45FC", STAC_92HD73XX_INTEL), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0254, "Dell Studio 1535", STAC_DELL_M6_DMIC), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0255,

2 1

Re: [alsa-devel] Old photos
by tglx＠linutronix.de 18 Aug '09

18 Aug '09

Greetings from france, your friend.

1 0

[alsa-devel] SB Audigy 4 No Front speakers, only Rear / SigmaTel STAC9750, 51 codec
by David P 18 Aug '09

18 Aug '09

Hi, after some time trying to trouble shoot this sound issue, I was encuraged to post a message explaining my situation in hopes maybe I would get some help. I have a SB Audigy 4. Surround Speakers work, Mic works. I do not have sound from front speakers. I'm not trying to do 5.1 surround or anything fancy. Just have headphones and a pair of speakers. I find it strange I have sound coming from the Rear and not the Front. All the searching I have done for this card seems people are having the oposite issue. I have checked all connections, they are all fine. I enabled all mixer controls and turned all the volumes up. Running various command line speaker-test's always reports no sound from Fronts, only Rears. So here I am asking for help. Here is a copy of my Alsa-info I hope this helps . upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.57 !!################################ !!Script ran on: Tue Aug 18 15:42:27 UTC 2009 !!Linux Distribution !!------------------ Welcome to openSUSE 11.1 - Kernel \r (\l). openSUSE 11.1 (i586) LSB_VERSION="core-2.0-noarch:core-3.2-noarch:core-2.0-ia32:core-3.2-ia32" !!DMI Information !!--------------- Manufacturer: Gigabyte Technology Co., Ltd. Product Name: P31-S3G !!Kernel Information !!------------------ Kernel release: 2.6.27.25-0.1-pae Operating System: GNU/Linux Architecture: i686 Processor: i686 SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.17 Library version: Utilities version: 1.0.20 !!Loaded ALSA modules !!------------------- snd_emu10k1 !!Sound Servers on this system !!---------------------------- ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No aRts: Installed - Yes (/opt/kde3/bin/artsd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [Audigy2 ]: Audigy2 - Audigy 4 [SB0610] Audigy 4 [SB0610] (rev.0, serial:0x10211102) at 0xd000, irq 18 !!PCI Soundcards installed in the system !!-------------------------------------- 04:02.0 Multimedia audio controller: Creative Labs SB0400 Audigy2 Value !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 04:02.0 0401: 1102:0008 Subsystem: 1102:1021 !!Modprobe options (Sound related) !!-------------------------------- snd-emu10k1: extin=0x3fff extout=0x7fff !!Loaded sound module options !!-------------------------- !!Module: snd_emu10k1 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_ir : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N extin : 16383,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 extout : 32767,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 max_buffer_size : 128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128,128 max_synth_voices : 64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64,64 seq_ports : 4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4,4 subsystem : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 !!AC97 Codec information !!--------------------------- --startcollapse-- 0-0/0: SigmaTel STAC9750,51 PCI Subsys Vendor: 0x0000 PCI Subsys Device: 0x0000 Capabilities : -headphone out- DAC resolution : 20-bit ADC resolution : 20-bit 3D enhancement : SigmaTel 3D Enhancement Current setup Mic gain : +20dB [+20dB] POP path : pre 3D Sim. stereo : off 3D enhancement : off Loudness : off Mono output : MIX Mic select : Mic1 ADC/DAC loopback : off Extended ID : codec=0 rev=1 AMAP DSA=0 SPDIF VRA Extended status : SPCV SPDIF=7/8 PCM front DAC : 48000Hz PCM ADC : 48000Hz SPDIF Control : Consumer PCM Copyright Category=0x0 Generation=0 Rate=48kHz 0:00 = 6a90 0:02 = 0505 0:04 = 9f1f 0:06 = 801f 0:08 = 0000 0:0a = 801e 0:0c = 8009 0:0e = 0047 0:10 = 0a0a 0:12 = 8808 0:14 = 8808 0:16 = 8b0b 0:18 = 9f1f 0:1a = 0000 0:1c = 0000 0:1e = 0000 0:20 = 0000 0:22 = 0000 0:24 = 0000 0:26 = 000f 0:28 = 0605 0:2a = 0410 0:2c = bb80 0:2e = 0000 0:30 = 0000 0:32 = bb80 0:34 = 0000 0:36 = 0000 0:38 = 0000 0:3a = 2000 0:3c = 0000 0:3e = 0100 0:40 = 0000 0:42 = 0000 0:44 = 0000 0:46 = 0000 0:48 = 0000 0:4a = 0000 0:4c = 0003 0:4e = ffff 0:50 = 0000 0:52 = 0000 0:54 = 0000 0:56 = 0000 0:58 = 0000 0:5a = 0000 0:5c = 0000 0:5e = 0000 0:60 = 0000 0:62 = 0000 0:64 = 0000 0:66 = 0000 0:68 = 0000 0:6a = 0000 0:6c = 0002 0:6e = 1000 0:70 = 0000 0:72 = 0000 0:74 = 0800 0:76 = 0000 0:78 = 0000 0:7a = 0000 0:7c = 8384 0:7e = 7650 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 12 Aug 18 05:04 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 3 Aug 18 05:04 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 14 Aug 18 10:04 /dev/snd/hwC0D2 crw-rw----+ 1 root audio 116, 5 Aug 18 05:04 /dev/snd/midiC0D0 crw-rw----+ 1 root audio 116, 4 Aug 18 05:04 /dev/snd/midiC0D1 crw-rw----+ 1 root audio 116, 15 Aug 18 10:04 /dev/snd/midiC0D2 crw-rw----+ 1 root audio 116, 16 Aug 18 10:04 /dev/snd/midiC0D3 crw-rw----+ 1 root audio 116, 11 Aug 18 05:04 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 10 Aug 18 10:20 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 9 Aug 18 05:04 /dev/snd/pcmC0D1c crw-rw----+ 1 root audio 116, 8 Aug 18 05:04 /dev/snd/pcmC0D2c crw-rw----+ 1 root audio 116, 7 Aug 18 05:04 /dev/snd/pcmC0D2p crw-rw----+ 1 root audio 116, 6 Aug 18 05:04 /dev/snd/pcmC0D3p crw-rw----+ 1 root audio 116, 13 Aug 18 10:04 /dev/snd/seq crw-rw----+ 1 root audio 116, 2 Aug 18 05:04 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Audigy2 [Audigy 4 [SB0610]], device 0: emu10k1 [ADC Capture/Standard PCM Playback] Subdevices: 31/32 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 Subdevice #8: subdevice #8 Subdevice #9: subdevice #9 Subdevice #10: subdevice #10 Subdevice #11: subdevice #11 Subdevice #12: subdevice #12 Subdevice #13: subdevice #13 Subdevice #14: subdevice #14 Subdevice #15: subdevice #15 Subdevice #16: subdevice #16 Subdevice #17: subdevice #17 Subdevice #18: subdevice #18 Subdevice #19: subdevice #19 Subdevice #20: subdevice #20 Subdevice #21: subdevice #21 Subdevice #22: subdevice #22 Subdevice #23: subdevice #23 Subdevice #24: subdevice #24 Subdevice #25: subdevice #25 Subdevice #26: subdevice #26 Subdevice #27: subdevice #27 Subdevice #28: subdevice #28 Subdevice #29: subdevice #29 Subdevice #30: subdevice #30 Subdevice #31: subdevice #31 card 0: Audigy2 [Audigy 4 [SB0610]], device 2: emu10k1 efx [Multichannel Capture/PT Playback] Subdevices: 8/8 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 Subdevice #4: subdevice #4 Subdevice #5: subdevice #5 Subdevice #6: subdevice #6 Subdevice #7: subdevice #7 card 0: Audigy2 [Audigy 4 [SB0610]], device 3: emu10k1 [Multichannel Playback] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Audigy2 [Audigy 4 [SB0610]], device 0: emu10k1 [ADC Capture/Standard PCM Playback] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Audigy2 [Audigy 4 [SB0610]], device 1: emu10k1 mic [Mic Capture] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Audigy2 [Audigy 4 [SB0610]], device 2: emu10k1 efx [Multichannel Capture/PT Playback] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Audigy2] Card hw:0 'Audigy2'/'Audigy 4 [SB0610] (rev.0, serial:0x10211102) at 0xd000, irq 18' Mixer name : 'SigmaTel STAC9750,51' Components : 'AC97a:83847650' Controls : 201 Simple ctrls : 36 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined cvolume cswitch cswitch-joined Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 100 Capture 0 - 31 Mono: Playback 68 [68%] [-12.80dB] Front Left: Capture 26 [84%] [-7.50dB] [on] Front Right: Capture 26 [84%] [-7.50dB] [on] Simple mixer control 'Tone',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'Bass',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 40 Front Left: 27 [68%] Front Right: 27 [68%] Simple mixer control 'Treble',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 40 Front Left: 27 [68%] Front Right: 27 [68%] Simple mixer control 'PCM',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 100 Front Left: Capture 65 [65%] [-14.00dB] Front Right: Capture 65 [65%] [-14.00dB] Simple mixer control 'PCM Center',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 100 Mono: Playback 75 [75%] [-10.00dB] Simple mixer control 'PCM Front',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 100 Mono: Front Left: Playback 75 [75%] [-10.00dB] Front Right: Playback 75 [75%] [-10.00dB] Simple mixer control 'PCM LFE',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 100 Mono: Playback 75 [75%] [-10.00dB] Simple mixer control 'PCM Out Path & Mute',0 Capabilities: enum Items: 'pre 3D' 'post 3D' Item0: 'pre 3D' Simple mixer control 'PCM Side',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 100 Mono: Front Left: Playback 72 [72%] [-11.20dB] Front Right: Playback 72 [72%] [-11.20dB] Simple mixer control 'PCM Surround',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 100 Mono: Front Left: Playback 75 [75%] [-10.00dB] Front Right: Playback 75 [75%] [-10.00dB] Simple mixer control 'Front',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 100 Mono: Front Left: Playback 82 [82%] [-7.20dB] Front Right: Playback 82 [82%] [-7.20dB] Simple mixer control 'Surround',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 100 Mono: Front Left: Playback 84 [84%] [-6.40dB] Front Right: Playback 84 [84%] [-6.40dB] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 100 Mono: Playback 70 [70%] [-12.00dB] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined Playback channels: Mono Limits: Playback 0 - 100 Mono: Playback 64 [64%] [-14.40dB] Simple mixer control 'Side',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 100 Mono: Front Left: Playback 82 [82%] [-7.20dB] Front Right: Playback 82 [82%] [-7.20dB] Simple mixer control 'Synth',0 Capabilities: pvolume cvolume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 100 Capture 0 - 100 Front Left: Playback 81 [81%] [-7.60dB] Capture 100 [100%] [0.00dB] Front Right: Playback 81 [81%] [-7.60dB] Capture 100 [100%] [0.00dB] Simple mixer control 'Wave',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 100 Mono: Front Left: Playback 82 [82%] [-7.20dB] Front Right: Playback 82 [82%] [-7.20dB] Simple mixer control 'Line',0 Capabilities: cvolume cswitch cswitch-joined Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 21 [68%] [-3.00dB] [on] Front Right: Capture 21 [68%] [-3.00dB] [on] Simple mixer control 'CD',0 Capabilities: cvolume cswitch cswitch-joined Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 23 [74%] [0.00dB] [off] Front Right: Capture 23 [74%] [0.00dB] [off] Simple mixer control 'Mic',0 Capabilities: cvolume cvolume-joined cswitch cswitch-joined Capture channels: Mono Limits: Capture 0 - 31 Mono: Capture 24 [77%] [1.50dB] [on] Simple mixer control 'Mic Boost (+20dB)',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Mic Select',0 Capabilities: enum Items: 'Mic1' 'Mic2' Item0: 'Mic1' Simple mixer control 'Video',0 Capabilities: cvolume cswitch cswitch-joined Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 23 [74%] [0.00dB] [off] Front Right: Capture 23 [74%] [0.00dB] [off] Simple mixer control 'Phone',0 Capabilities: cvolume cvolume-joined cswitch cswitch-joined Capture channels: Mono Limits: Capture 0 - 31 Mono: Capture 22 [71%] [-1.50dB] [off] Simple mixer control 'IEC958 Optical',0 Capabilities: pvolume cvolume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 100 Capture 0 - 100 Front Left: Playback 75 [75%] [-10.00dB] Capture 73 [73%] [-10.80dB] Front Right: Playback 75 [75%] [-10.00dB] Capture 73 [73%] [-10.80dB] Simple mixer control 'IEC958 Optical Raw',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'PC Speaker',0 Capabilities: cvolume cvolume-joined cswitch cswitch-joined Capture channels: Mono Limits: Capture 0 - 15 Mono: Capture 0 [0%] [-45.00dB] [off] Simple mixer control 'Aux',0 Capabilities: cvolume cswitch cswitch-joined Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 20 [65%] [-4.50dB] [off] Front Right: Capture 20 [65%] [-4.50dB] [off] Simple mixer control 'Aux2',0 Capabilities: pvolume cvolume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 100 Capture 0 - 100 Front Left: Playback 0 [0%] [-99999.99dB] Capture 0 [0%] [-99999.99dB] Front Right: Playback 0 [0%] [-99999.99dB] Capture 0 [0%] [-99999.99dB] Simple mixer control 'Mono Output Select',0 Capabilities: enum Items: 'Mix' 'Mic' Item0: 'Mix' Simple mixer control 'AMic',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 100 Mono: Front Left: Playback 0 [0%] [-99999.99dB] Front Right: Playback 0 [0%] [-99999.99dB] Simple mixer control 'Analog Mix',0 Capabilities: pvolume cvolume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 100 Capture 0 - 100 Front Left: Playback 0 [0%] [-99999.99dB] Capture 0 [0%] [-99999.99dB] Front Right: Playback 0 [0%] [-99999.99dB] Capture 0 [0%] [-99999.99dB] Simple mixer control 'Audigy Analog/Digital Output Jack',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Audigy CD',0 Capabilities: pvolume cvolume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 100 Capture 0 - 100 Front Left: Playback 76 [76%] [-9.60dB] Capture 74 [74%] [-10.40dB] Front Right: Playback 76 [76%] [-9.60dB] Capture 74 [74%] [-10.40dB] Simple mixer control 'External Amplifier',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] !!Alsactl output !!------------- --startcollapse-- state.Audigy2 { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'PCM Front Playback Volume' value.0 75 value.1 75 } control.2 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'PCM Surround Playback Volume' value.0 75 value.1 75 } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'PCM Side Playback Volume' value.0 72 value.1 72 } control.4 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'PCM Center Playback Volume' value 75 } control.5 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'PCM LFE Playback Volume' value 75 } control.6 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Wave Playback Volume' value.0 82 value.1 82 } control.7 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Synth Playback Volume' value.0 81 value.1 81 } control.8 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'PCM Capture Volume' value.0 65 value.1 65 } control.9 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Synth Capture Volume' value.0 100 value.1 100 } control.10 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'AMic Playback Volume' value.0 0 value.1 0 } control.12 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Audigy CD Playback Volume' value.0 76 value.1 76 } control.13 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Audigy CD Capture Volume' value.0 74 value.1 74 } control.14 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'IEC958 Optical Playback Volume' value.0 75 value.1 75 } control.15 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'IEC958 Optical Capture Volume' value.0 73 value.1 73 } control.18 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Analog Mix Playback Volume' value.0 0 value.1 0 } control.19 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Analog Mix Capture Volume' value.0 0 value.1 0 } control.20 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Aux2 Playback Volume' value.0 0 value.1 0 } control.21 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Aux2 Capture Volume' value.0 0 value.1 0 } control.22 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Front Playback Volume' value.0 82 value.1 82 } control.23 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Surround Playback Volume' value.0 84 value.1 84 } control.24 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Center Playback Volume' value 70 } control.25 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'LFE Playback Volume' value 64 } control.26 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Side Playback Volume' value.0 82 value.1 82 } control.27 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 40' iface MIXER name 'Tone Control - Bass' value.0 27 value.1 27 } control.28 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 40' iface MIXER name 'Tone Control - Treble' value.0 27 value.1 27 } control.29 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Tone Control - Switch' value.0 true value.1 true } control.30 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 100' comment.dbmin -4000 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value 68 } control.31 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'IEC958 Optical Raw Playback Switch' value.0 true value.1 true } control.32 { comment.access 'read write' comment.type BOOLEAN comment.count 64 iface PCM device 2 name 'Captured FX8010 Outputs' value.0 false value.1 false value.2 false value.3 false value.4 false value.5 false value.6 false value.7 false value.8 false value.9 false value.10 false value.11 false value.12 false value.13 false value.14 false value.15 false value.16 false value.17 false value.18 false value.19 false value.20 false value.21 false value.22 false value.23 false value.24 false value.25 false value.26 false value.27 false value.28 false value.29 false value.30 false value.31 false value.32 true value.33 true value.34 true value.35 true value.36 true value.37 true value.38 true value.39 true value.40 true value.41 true value.42 true value.43 true value.44 true value.45 true value.46 true value.47 true value.48 false value.49 false value.50 false value.51 false value.52 false value.53 false value.54 false value.55 false value.56 false value.57 false value.58 false value.59 false value.60 false value.61 false value.62 false value.63 false } control.33 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Capture Switch' value true } control.34 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -4650 comment.dbmax 0 iface MIXER name 'Master Capture Volume' value.0 26 value.1 26 } control.39 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'PC Speaker Capture Switch' value false } control.40 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 15' comment.dbmin -4500 comment.dbmax 0 iface MIXER name 'PC Speaker Capture Volume' value 0 } control.41 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Phone Capture Switch' value false } control.42 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Phone Capture Volume' value 22 } control.43 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Mic Capture Switch' value true } control.44 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Mic Capture Volume' value 24 } control.45 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Mic Boost (+20dB)' value true } control.46 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Line Capture Switch' value true } control.47 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Line Capture Volume' value.0 21 value.1 21 } control.48 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'CD Capture Switch' value false } control.49 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'CD Capture Volume' value.0 23 value.1 23 } control.50 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Video Capture Switch' value false } control.51 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Video Capture Volume' value.0 23 value.1 23 } control.52 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Aux Capture Switch' value false } control.53 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Aux Capture Volume' value.0 20 value.1 20 } control.59 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 'pre 3D' comment.item.1 'post 3D' iface MIXER name 'PCM Out Path & Mute' value 'pre 3D' } control.61 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mix comment.item.1 Mic iface MIXER name 'Mono Output Select' value Mix } control.62 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic1 comment.item.1 Mic2 iface MIXER name 'Mic Select' value Mic1 } control.65 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'External Amplifier' value true } control.66 { comment.access 'read write' comment.type INTEGER comment.count 24 comment.range '0 - 63' iface PCM name 'EMU10K1 PCM Send Routing' value.0 0 value.1 1 value.2 2 value.3 3 value.4 0 value.5 0 value.6 0 value.7 0 value.8 0 value.9 1 value.10 2 value.11 3 value.12 0 value.13 0 value.14 0 value.15 0 value.16 0 value.17 1 value.18 2 value.19 3 value.20 0 value.21 0 value.22 0 value.23 0 } control.98 { comment.access 'read write' comment.type INTEGER comment.count 24 comment.range '0 - 255' iface PCM name 'EMU10K1 PCM Send Volume' value.0 255 value.1 255 value.2 0 value.3 0 value.4 0 value.5 0 value.6 0 value.7 0 value.8 255 value.9 0 value.10 0 value.11 0 value.12 0 value.13 0 value.14 0 value.15 0 value.16 0 value.17 255 value.18 0 value.19 0 value.20 0 value.21 0 value.22 0 value.23 0 } control.130 { comment.access 'read write' comment.type INTEGER comment.count 3 comment.range '0 - 65535' iface PCM name 'EMU10K1 PCM Volume' value.0 65535 value.1 65535 value.2 65535 } control.210 { comment.access read comment.type IEC958 comment.count 1 iface PCM name 'IEC958 Playback Mask' value ffffffff00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000 } control.211 { comment.access read comment.type IEC958 comment.count 1 iface PCM name 'IEC958 Playback Mask' index 1 value ffffffff00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000 } control.212 { comment.access read comment.type IEC958 comment.count 1 iface PCM name 'IEC958 Playback Mask' index 2 value ffffffff00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000 } control.213 { comment.access 'read write' comment.type IEC958 comment.count 1 iface PCM name 'IEC958 Playback Default' value '0482000200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.214 { comment.access 'read write' comment.type IEC958 comment.count 1 iface PCM name 'IEC958 Playback Default' index 1 value '0492100200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.215 { comment.access 'read write' comment.type IEC958 comment.count 1 iface PCM name 'IEC958 Playback Default' index 2 value '0492100200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.216 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Audigy Analog/Digital Output Jack' value true } } --endcollapse-- !!All Loaded Modules !!------------------ Module ip6t_LOG xt_tcpudp xt_pkttype ipt_LOG xt_limit binfmt_misc snd_pcm_oss snd_mixer_oss snd_seq_midi snd_emu10k1_synth snd_emux_synth snd_seq_virmidi snd_seq_midi_event snd_seq_midi_emul snd_seq af_packet ip6t_REJECT nf_conntrack_ipv6 ip6table_raw xt_NOTRACK ipt_REJECT xt_state iptable_raw iptable_filter ip6table_mangle nf_conntrack_netbios_ns nf_conntrack_ipv4 nf_conntrack ip_tables ip6table_filter ip6_tables x_tables cpufreq_conservative ipv6 cpufreq_userspace cpufreq_powersave acpi_cpufreq speedstep_lib fuse loop dm_mod snd_emu10k1 snd_rawmidi snd_ac97_codec ac97_bus snd_pcm snd_seq_device snd_timer snd_page_alloc snd_util_mem snd_hwdep snd iTCO_wdt nvidia iTCO_vendor_support rtc_cmos r8169 ppdev i2c_i801 sr_mod intel_agp soundcore rtc_core cdrom mii pcspkr rtc_lib i2c_core parport_pc button joydev parport agpgart floppy sg usbhid hid ff_memless uhci_hcd ehci_hcd sd_mod crc_t10dif usbcore edd ext3 mbcache jbd fan ide_pci_generic ide_core ata_generic ata_piix libata scsi_mod dock thermal processor thermal_sys hwmon !!ALSA/HDA dmesg !!------------------ EMU10K1_Audigy 0000:04:02.0: PCI INT A -> GSI 18 (level, low) -> IRQ 18 ALSA sound/pci/emu10k1/emu10k1_main.c:219: Audigy2 value: Special config. Adding 2104472k swap on /dev/sda5. Priority:-1 extents:1 across:2104472

1 0

[alsa-devel] [PATCH] ASoC: Fix data format configuration for S3C64XX IISv2
by Joonyoung Shim 18 Aug '09

18 Aug '09

The data format configuration for S3C64xx IISv2 was hardcoded for IISMOD register. This patch changes to the defined values it. And instead of bits 9 and 10 of IISMOD we should clear bits 13 and 14. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h | 5 +++++ sound/soc/s3c24xx/s3c-i2s-v2.c | 6 +++--- 2 files changed, 8 insertions(+), 3 deletions(-) diff --git a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h index 0fad757..07659da 100644 --- a/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h +++ b/arch/arm/plat-s3c/include/plat/regs-s3c2412-iis.h @@ -33,6 +33,11 @@ #define S3C2412_IISCON_RXDMA_ACTIVE (1 << 1) #define S3C2412_IISCON_IIS_ACTIVE (1 << 0) +#define S3C64XX_IISMOD_BLC_16BIT (0 << 13) +#define S3C64XX_IISMOD_BLC_8BIT (1 << 13) +#define S3C64XX_IISMOD_BLC_24BIT (2 << 13) +#define S3C64XX_IISMOD_BLC_MASK (3 << 13) + #define S3C64XX_IISMOD_IMS_PCLK (0 << 10) #define S3C64XX_IISMOD_IMS_SYSMUX (1 << 10) diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c index 1a28317..ebfb2f6 100644 --- a/sound/soc/s3c24xx/s3c-i2s-v2.c +++ b/sound/soc/s3c24xx/s3c-i2s-v2.c @@ -357,19 +357,19 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, #endif #ifdef CONFIG_PLAT_S3C64XX - iismod &= ~0x606; + iismod &= ~(S3C64XX_IISMOD_BLC_MASK | S3C2412_IISMOD_BCLK_MASK); /* Sample size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: /* 8 bit sample, 16fs BCLK */ - iismod |= 0x2004; + iismod |= (S3C64XX_IISMOD_BLC_8BIT | S3C2412_IISMOD_BCLK_16FS); break; case SNDRV_PCM_FORMAT_S16_LE: /* 16 bit sample, 32fs BCLK */ break; case SNDRV_PCM_FORMAT_S24_LE: /* 24 bit sample, 48fs BCLK */ - iismod |= 0x4002; + iismod |= (S3C64XX_IISMOD_BLC_24BIT | S3C2412_IISMOD_BCLK_48FS); break; } #endif -- 1.6.0.4

2 4

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 18 Aug '09

18 Aug '09

The following changes since commit d2a382143b35a486b8089861790d6b3ec4e46932: Mark Brown (1): ASoC: Update AD1938 for new TDM slot API are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Barry Song (3): ASoC: add output/input widgets in ad1938 to make dac/adc dynamic PM work new ad1836 codec driver based on asoc ASoC: delete -spi suffix in ad1938 and free private data while registers fail Chaithrika U S (3): ASoC: DaVinci: McASP driver enhacements ASoC: DaVinci: Add a DAI format to McASP driver ASoC: DaVinci: Add audio support fot DA850/OMAP-L138 EVM Janusz Krzysztofik (2): ASoC: OMAP: Make use of DMA channel self linking on OMAP1510 ASoC: OMAP: Enhance OMAP1510 DMA progress software counter Marek Vasut (1): ALSA: Allow passing platform_data for pxa2xx-ac97 Mark Brown (20): ASoC: Add WM8974 CODEC driver ASoC: WM8974 checkpatch cleanups ASoC: Update WM8974 to use standard I2C device probe methods ASoC: Remove unreferenced wm8974_add_controls() ASoC: Refresh WM8974 bias configuration ASoC: Declare 2 channels for WM8974 ASoC: Refresh WM8974 PLL configuration ASoC: Add WM8974 TLV information ASoC: Use symmetric rates for WM8974 ASoC: WM8974 cosmetic cleanups ASoC: WM8974 DAPM cleanups ASoC: Minor cleanups to AD1938 driver ASoC: Factor out shared code from WM8993 Merge branch 'for-2.6.32' into mxc ASoC: Remove unneeded i.MX dependency on SND ASoC: Staticise unexported variables ASoC: Hook i.MX into build Merge branch 'wm8974-upstream' into for-2.6.32 ASoC: Factor out cache I/O from WM8974 ASoC: Fix handling of bias levels for non-DAPM codecs Peter Ujfalusi (2): ASoC: TWL4030: Introduce PGAs for outputs ASoC: TWL4030: Fix for capture mixer strings Shine Liu (1): ASoC: UDA134X: Fix mistaken mute/unmute code javier Martin (5): ASoC: Correct a bug with "ADC Inversion Switch" in wm8974 codec. ASoC: add DMA platform driver for MX1x and MX2x ASoC: add DAI platform ssi driver for MXC ASoC: add machine driver for i.mx27_visstrim_m10 board ASoC: Fix review issues in i.MX2x PCM driver arch/arm/mach-pxa/include/mach/audio.h | 3 + include/sound/ac97_codec.h | 3 + sound/arm/pxa2xx-ac97.c | 10 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/ad1836.c | 446 ++++++++++++++++ sound/soc/codecs/ad1836.h | 64 +++ sound/soc/codecs/ad1938.c | 16 +- sound/soc/codecs/twl4030.c | 155 ++++--- sound/soc/codecs/uda134x.c | 2 +- sound/soc/codecs/wm8974.c | 808 +++++++++++++++++++++++++++++ sound/soc/codecs/wm8974.h | 99 ++++ sound/soc/codecs/wm8993.c | 716 +------------------------- sound/soc/codecs/wm_hubs.c | 758 ++++++++++++++++++++++++++++ sound/soc/codecs/wm_hubs.h | 24 + sound/soc/davinci/Kconfig | 13 +- sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 29 +- sound/soc/davinci/davinci-i2s.c | 1 + sound/soc/davinci/davinci-mcasp.c | 113 ++++- sound/soc/davinci/davinci-mcasp.h | 5 + sound/soc/davinci/davinci-pcm.c | 4 +- sound/soc/davinci/davinci-pcm.h | 1 + sound/soc/imx/Kconfig | 21 + sound/soc/imx/Makefile | 10 + sound/soc/imx/mx1_mx2-pcm.c | 488 ++++++++++++++++++ sound/soc/imx/mx1_mx2-pcm.h | 26 + sound/soc/imx/mx27vis_wm8974.c | 317 ++++++++++++ sound/soc/imx/mxc-ssi.c | 868 ++++++++++++++++++++++++++++++++ sound/soc/imx/mxc-ssi.h | 238 +++++++++ sound/soc/omap/omap-pcm.c | 37 +- sound/soc/pxa/pxa2xx-ac97.c | 12 +- sound/soc/soc-dapm.c | 16 + 35 files changed, 4532 insertions(+), 792 deletions(-) create mode 100644 sound/soc/codecs/ad1836.c create mode 100644 sound/soc/codecs/ad1836.h create mode 100644 sound/soc/codecs/wm8974.c create mode 100644 sound/soc/codecs/wm8974.h create mode 100644 sound/soc/codecs/wm_hubs.c create mode 100644 sound/soc/codecs/wm_hubs.h create mode 100644 sound/soc/imx/Kconfig create mode 100644 sound/soc/imx/Makefile create mode 100644 sound/soc/imx/mx1_mx2-pcm.c create mode 100644 sound/soc/imx/mx1_mx2-pcm.h create mode 100644 sound/soc/imx/mx27vis_wm8974.c create mode 100644 sound/soc/imx/mxc-ssi.c create mode 100644 sound/soc/imx/mxc-ssi.h

2 1

[alsa-devel] [PATCH] ASoC: SDP3430: Fix TWL GPIO6 pin mux request
by Candelaria Villareal, Jorge 18 Aug '09

18 Aug '09

Fix the write to PMBR1 register through I2C. Also, the constant which holds the value to write is now called TWL4030_GPIO6_PWM0_MUTE. This name is based on TRM to avoid confusion. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/sdp3430.c | 15 +++++++++++---- 1 files changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index f7e5b74..2eb1a69 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -40,8 +40,10 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" -#define TWL4030_INTBR_PMBR1 0x0D -#define EXTMUTE(value) (value << 2) +/* TWL4030 PMBR1 Register */ +#define TWL4030_INTBR_PMBR1 0x0D +/* TWL4030 PMBR1 Register GPIO6 mux bit */ +#define TWL4030_GPIO6_PWM0_MUTE(value) (value << 2) static struct snd_soc_card snd_soc_sdp3430; @@ -299,6 +301,7 @@ static struct platform_device *sdp3430_snd_device; static int __init sdp3430_soc_init(void) { int ret; + u8 pin_mux; if (!machine_is_omap_3430sdp()) { pr_debug("Not SDP3430!\n"); @@ -318,8 +321,12 @@ static int __init sdp3430_soc_init(void) *(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */ /* Set TWL4030 GPIO6 as EXTMUTE signal */ - twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, EXTMUTE(0x02), - TWL4030_MODULE_INTBR); + twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, &pin_mux, + TWL4030_INTBR_PMBR1); + pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x3); + pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x2); + twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux, + TWL4030_INTBR_PMBR1); ret = platform_device_add(sdp3430_snd_device); if (ret) -- 1.6.0.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: change set_tdm_slot api to allow slot_width override.
by Mark Brown 17 Aug '09

17 Aug '09

From: Daniel Ribeiro <drwyrm(a)gmail.com> Extend set_tdm_slot to allow the user to arbitrarily set the frame width and active TX/RX slots. Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c still doesn't handle the slot_width override. While being there, correct an incorrect use of SlotsPerFrm(7) use in bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) ). (this series is meant for Mark's for-2.6.32 branch) Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc-dai.h | 5 +++-- sound/soc/codecs/wm9081.c | 5 +++-- sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/pxa-ssp.c | 27 ++++++++++++++++++++------- sound/soc/soc-core.c | 9 ++++++--- 5 files changed, 33 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 25d62ac..2d3fa29 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -106,7 +106,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); @@ -140,7 +140,8 @@ struct snd_soc_dai_ops { */ int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); int (*set_tdm_slot)(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 156f2a4..c965323 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1214,8 +1214,9 @@ static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai, return 0; } +/* FIXME: Needs to handle slot_width */ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_codec *codec = dai->codec; unsigned int aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1); @@ -1227,7 +1228,7 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, aif1 |= (slots - 1) << WM9081_AIFDAC_TDM_MODE_SHIFT; - switch (mask) { + switch (rx_mask) { case 1: break; case 2: diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index 8889cd3..9f7c61e 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -190,7 +190,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width); if (ret < 0) return ret; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index e22c5ce..5b9ed64 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -375,21 +375,34 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, * Set the active slots in TDM/Network mode */ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; u32 sscr0; - sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7); + sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(8) | SSCR0_EDSS | SSCR0_DSS); + + /* set slot width */ + if (slot_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(slot_width - 16); + else + sscr0 |= SSCR0_DataSize(slot_width); - /* set number of active slots */ - sscr0 |= SSCR0_SlotsPerFrm(slots); + if (slots > 1) { + /* enable network mode */ + sscr0 |= SSCR0_MOD; + + /* set number of active slots */ + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* set active slot mask */ + ssp_write_reg(ssp, SSTSA, tx_mask); + ssp_write_reg(ssp, SSRSA, rx_mask); + } ssp_write_reg(ssp, SSCR0, sscr0); - /* set active slot mask */ - ssp_write_reg(ssp, SSTSA, mask); - ssp_write_reg(ssp, SSRSA, mask); return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fb8d7a7..e984a17 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2223,17 +2223,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); /** * snd_soc_dai_set_tdm_slot - configure DAI TDM. * @dai: DAI - * @mask: DAI specific mask representing used slots. + * @tx_mask: bitmask representing active TX slots. + * @rx_mask: bitmask representing active RX slots. * @slots: Number of slots in use. + * @slot_width: Width in bits for each slot. * * Configures a DAI for TDM operation. Both mask and slots are codec and DAI * specific. */ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { if (dai->ops && dai->ops->set_tdm_slot) - return dai->ops->set_tdm_slot(dai, mask, slots); + return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask, + slots, slot_width); else return -EINVAL; } -- 1.6.3.3

3 12

[alsa-devel] Strange div-by-zero in alsa-lib 1.0.20's snd_pcm_mmap_begin()
by Daniel Chen 17 Aug '09

17 Aug '09

Hi, In https://bugs.launchpad.net/ubuntu/+source/alsa-lib/+bug/412677 we're debugging an issue where snd_pcm_mmap_begin(), at line 6409 with *offset = *pcm->appl.ptr % pcm->buffer_size; appears to have pcm->buffer_size == 0. What's the correct approach in handling this corner case? Thanks, Dan

3 4

[alsa-devel] How to use hda-verb to detect
by Ozan Çağlayan 17 Aug '09

17 Aug '09

Hi, I'm playing with an HP DV6(which produces sound only when model=hp-dv5, will send a patch for it after further testing) on which HP automuting is not working. The laptop has apparently two HP outputs which seems to behave completely parallel. One thing that I really don't get with hda-verb, is the last parameter. I used hda-emu for finding out an example call: Front Playback Volume:0 MIN/MAX: 0/64, VAL: [39] [39] dB min/max: -48.00/0.00, [-18.-75dB] [-18.-75dB] > set 1 64 64 send: NID=0x10, VERB=0x3a0(set_amp_gain_mute,O:L#0), PARM=0x7f receive: 0x0 send: NID=0x10, VERB=0x390(set_amp_gain_mute,O:R#0), PARM=0x7f receive: 0x0 > set 1 63 63 send: NID=0x10, VERB=0x3a0(set_amp_gain_mute,O:L#0), PARM=0x7e receive: 0x0 send: NID=0x10, VERB=0x390(set_amp_gain_mute,O:R#0), PARM=0x7e receive: 0x0 I can easily understand that I'm calling set_amp_gain_mute VERB for NID 0x10 with a level value 0x7f and 0x7e. Now when I run hda-verb without parameter, it gives me a bunch of possible values for PARM: VENDOR_ID, SUBSYSTEM_ID, REV_ID, NODE_COUNT, FUNCTION_TYPE AUDIO_FG_CAP, AUDIO_WIDGET_CAP, PCM, STREAM, PIN_CAP, AMP_IN_CAP CONNLIST_LEN, POWER_STATE, PROC_CAP, GPIO_CAP, AMP_OUT_CAP VOL_KNB_CA I can't easily understand which one I have to choose for example to send the verb SET_UNSOLICITED_ENABLE or SET_PIN_SENSE. I've found the widget below in HP DV6 (OUT HP Detect grabbed my attention and I've seen that the unsolicited events has something to do with HP sense/automute, just playling blindly), Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000001c: OUT HP Detect ** Pin Default 0x40f100f0: [N/A] Other at Ext N/A Conn = 1/8, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=0c, enabled=0 Connection: 3 0x10 0x11 0x17* and say that I want to send SET_UNSOLICITED_ENABLE to 0x0a, which PARM should I choose? I'm also attaching alsa-info output, Thanks! Ozan

2 7

[alsa-devel] aplay: set_params:984: Channels count non available
by prashant chopade 17 Aug '09

17 Aug '09

Hello, I am using alsa library 1.0.20. I am using linux-2.6.27. But I have shown certain useful messages: # cat /etc/asound.conf pcm.!default { type hw card 0 } ctl.!default { type hw card 0 } # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: AC97 [Atmel AC97], device 0: Atmel AC97 [Atmel AC97] Subdevices: 1/1 Subdevice #0: subdevice #0 # aplay /home/Front_Center.wav -c 1 Playing WAVE '/home/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono aplay: set_params:984: Channels count non available To remove this error which channel i have to used or set and how? Thanks & Regards, Prashant C. Looking for local information? Find it on Yahoo! Local http://in.local.yahoo.com/

1 0

[alsa-devel] [PATCH] ASoC: UDA134X Codec: Fix mute/unmute code mistake and add ADC/DAC power control support
by Shine Liu 17 Aug '09

17 Aug '09

There is a mistake in current uda134x_mute function: mute_reg has been changed in line 162 or line 164, so uda134x_write should write "mute_reg" but not "mute_reg & ~(1<<2)" to UDA134X_DATA010. Besides, because there is no DAPM configuration for uda134x, when system starts up, snd_soc_int_card calls snd_soc_dapm_new_widgets, and snd_soc_dapm_new_widgets calls dapm_power_widgets. In function dapm_power_widgets, codec->dapm_widgets has no list entry, so sys_power retains it's original value zero. Then snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_PREPARE) and snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_STANDBY) are called sequentially. Finally, the uda134x codec goes to STANDBY mode, so the ADC/DAC power control bits of UDA134X_STATUS1 register keeps the 0 value. UDA134X has no trigger function currently, and the ADC/DAC power control bits of UDA134X_STATUS1 register are not exported in uda1341_snd_controls, so there's no way to enable the ADC/DAC power control bits when you want to use the codec. When playing with aplay or recording with arecord, there is no sound output or no wave input. I have added the uda134x_trigger function, which turns on/off the ADC/DAC power control bits according the substream type and the command. I also exported the ADC/DAC power control bits to uda1341_snd_controls, so we can also turn these bits on/off as we need via a mixer tool like alsamixer. The patch created against linux-2.6.31-rc3. Tested on a s3c2440 development board with UDA1341TS codec. Signed-off-by: Shine Liu <shinel(a)foxmail.com> --- sound/soc/codecs/uda134x.c.orig 2009-07-14 09:18:52.000000000 +0800 +++ sound/soc/codecs/uda134x.c 2009-08-17 13:46:57.000000000 +0800 @@ -163,7 +163,7 @@ else mute_reg &= ~(1<<2); - uda134x_write(codec, UDA134X_DATA010, mute_reg & ~(1<<2)); + uda134x_write(codec, UDA134X_DATA010, mute_reg); return 0; } @@ -339,6 +339,38 @@ return 0; } +static int uda134x_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + int stream = substream->stream; + struct snd_soc_codec *codec = dai->codec; + u8 power_ctrl_reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); + + pr_debug("%s stream: %d, cmd: %d\n", __func__, stream, cmd); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + power_ctrl_reg |= (1<<0); + else + power_ctrl_reg |= (1<<1); + break; + + case SNDRV_PCM_TRIGGER_STOP: + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + power_ctrl_reg &= ~(1<<0); + else + power_ctrl_reg &= ~(1<<1); + break; + + default: + return 0; + } + + uda134x_write(codec, UDA134X_STATUS1, power_ctrl_reg); + return 0; +} + static int uda134x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -416,6 +448,8 @@ SOC_SINGLE("ADC Polarity Switch", UDA134X_STATUS1, 4, 1, 0), SOC_SINGLE("DAC Polarity Switch", UDA134X_STATUS1, 3, 1, 0), SOC_SINGLE("Double Speed Playback Switch", UDA134X_STATUS1, 2, 1, 0), +SOC_SINGLE("ADC Power Switch", UDA134X_STATUS1, 1, 1, 0), +SOC_SINGLE("DAC Power Switch", UDA134X_STATUS1, 0, 1, 0), SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), }; @@ -438,6 +472,7 @@ .digital_mute = uda134x_mute, .set_sysclk = uda134x_set_dai_sysclk, .set_fmt = uda134x_set_dai_fmt, + .trigger = uda134x_trigger, }; struct snd_soc_dai uda134x_dai = {

2 8

[alsa-devel] [PATCHv3 00/20] OMAP ASoC changes in DMA utilization
by Eduardo Valentin 17 Aug '09

17 Aug '09

Hello again guys, Here is version 3 of these changes. I've changed just 2 things: - all mcbsp instances are in element mode by default - moved all mcbsp related code to omap-mcbsp.c (although a single callback is still needed) Jarkko, I think we can add your patch for reading strings for dma op mode later on. BR, Eduardo Valentin (11): OMAP: McBSP: Add IRQEN, IRQSTATUS, THRESHOLD2 and THRESHOLD1 registers. OMAP: McBSP: Use appropriate value for startup delay OMAP: McBSP: Add transmit/receive threshold handler OMAP: McBSP: Create and export max_(r|t)x_thres property OMAP: McBSP: Rename thres sysfs symbols OMAP: McBSP: Change wakeup signals OMAP: McBSP: Configure NO IDLE mode for DMA mode different of threshold ASoC: OMAP: Make DMA 64 aligned ASoC: OMAP: Enable DMA burst mode ASoC: OMAP: Use McBSP threshold to playback and capture ASoC: OMAP: Use DMA operating mode of McBSP Eero Nurkkala (7): OMAP: McBSP: Provide functions for ASoC frame syncronization OMAP: McBSP: Wakeups utilized OMAP: McBSP: Retain McBSP FCLK clockactivity OMAP: McBSP: Do not enable wakeups for no-idle mode OMAP: McBSP: Let element DMA mode hit retention also ASoC: Add runtime check for RFIG and XFIG ASoC: Always syncronize audio transfers on frames Peter Ujfalusi (2): OMAP3: McBSP: Lower the maximum buffersize for McBSP1,3,4,5 OMAP: McBSP: Add link DMA mode selection arch/arm/mach-omap2/mcbsp.c | 5 + arch/arm/plat-omap/include/mach/mcbsp.h | 49 ++++ arch/arm/plat-omap/mcbsp.c | 377 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 77 ++++++- sound/soc/omap/omap-pcm.c | 14 +- sound/soc/omap/omap-pcm.h | 2 + 6 files changed, 511 insertions(+), 13 deletions(-) -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

4 25

[alsa-devel] custom ioplug plugin does not work
by Stefan Schoenleitner 17 Aug '09

17 Aug '09

Hi, I'm trying to write an ioplug plugin which dumps the raw PCM stream to a device file in real time. With real time I mean that a device available through a device file should receive the PCM samples exactly at the sampling rate. Thus for example with a sampling rate of 8kHz and 16bit linear samples, each 1/8000 = 125us the device should receive exactly one 16bit sample (if we assume that only one audio channel is used). This should all happen on an ARM9 target with the device being a fast SPI device that can handle the transfer speeds. In contrast to the existing file or tee plugins, this should also work if no slave device is specified. With the mentioned two existing plugins all PCM data is immediately written to the output file if the null device is specified. For this reason these plugins do not seem to be usable for my application. To solve the mentioned problem it was suggested to me that I should write an ALSA plugin which does the job. At the moment I already have some partially working code which is available at http://pastebin.com/m3666868b My problem with the code is that when playing audio over the plugin, it transfer some data chunks in the beginning and then it seems to be blocking forever. What I'm doing in my plugin code is basically the following: 1. open output file for writing (file descriptor is filedump->file_fd) 2. filedump->io.poll_fd = filedump->file_fd; filedump->io.poll_events = POLLOUT; filedump->io.mmap_rw = 0; 3. snd_pcm_ioplug_create() I have callbacks for start, stop, close, pointer, prepare and transfer: In the start and stop callbacks I don't do anything at the moment. In the close callback I close the output file. In the pointer callback I just return the current hw_ptr. In the prepare callback the hw_ptr is initialized with 0. However, the most imporant callback seems to be the transfer callback. At the moment it just prints a hexdump of the PCM buffer and returns the size of the buffer to indicate that the whole buffer has been transfered. * Is this correct so far ? * Why is the plugin blocking forever after some parts of the buffer have been transfered and the start callback is called ? The output of the plugin looks like this: ------------------------------------------------------------------------------------------ _snd_pcm_filedump_open:311 25740 SND_PCM_PLUGIN_DEFINE_FUNC(): done, err=0 Playing WAVE '/usr/share/sounds/card_shuffle.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo snd_pcm_filedump_prepare:134 25740 Preparing with io->period_size=5512 io->buffer_size=22050 snd_pcm_filedump_transfer:99 25740 snd_pcm_filedump_transfer(): areas->step=32 areas->first=0 offset=0, size=5512 io->nonblock=0 <hexdump ...> snd_pcm_filedump_transfer:99 25740 snd_pcm_filedump_transfer(): areas->step=32 areas->first=0 offset=0, size=5512 io->nonblock=0 <hexdump ...> snd_pcm_filedump_transfer:99 25740 snd_pcm_filedump_transfer(): areas->step=32 areas->first=0 offset=0, size=5512 io->nonblock=0 <hexdump ...> snd_pcm_filedump_transfer:99 25740 snd_pcm_filedump_transfer(): areas->step=32 areas->first=0 offset=0, size=5512 io->nonblock=0 <hexdump ...> snd_pcm_filedump_transfer:99 25740 snd_pcm_filedump_transfer(): areas->step=32 areas->first=0 offset=0, size=2 io->nonblock=0 <hexdump ...> snd_pcm_filedump_start:70 25740 snd_pcm_filedump_start() ^CAborted by signal Interrupt... snd_pcm_filedump_stop:76 25740 snd_pcm_filedump_stop() snd_pcm_filedump_close:59 25740 snd_pcm_filedump_close() ------------------------------------------------------------------------------------------ As you can see, after calling snd_pcm_filedump_start() the plugin seems to be blocking forever. * Why is data transfered at all before start is called ? Unfortunately the ALSA documentation seems to be really bad. Reading (and partially copying) code from other plugins brought me this far, but not further at the moment. I would appreciate any help to get rid of the problem mentioned above. cheers, stefan

1 0

[alsa-devel] [RFC][PATCH 2/3] ASoC: OMAP: Make use of DMA channel self linking on OMAP1510
by Janusz Krzysztofik 17 Aug '09

17 Aug '09

Use newly implemented DMA channel self linking on OMAP1510 like on other OMAP models. Remove unnecessary DMA transfer restart from interrupt handler routine. The interrupt routine used to maintain a period index, originally needed for counting up periods up to a full buffer in order to restart the DMA transfer. For some time, this counter is also used as a replacement for hardware DMA progress counter that has been found unusable on OMAP1510 in case of playback. Thus, the period index calculation cannot be omitted completely. However, the accuracy of this counter can still suffer from missing DMA interrupts. In order to work correctly, it requires patch 1 from this series also applied: [RFC][PATCH 1/3] ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510 Created against linux-2.6.31-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- linux-2.6.31-rc5/sound/soc/omap/omap-pcm.c.orig 2009-08-07 13:38:23.000000000 +0200 +++ linux-2.6.31-rc5/sound/soc/omap/omap-pcm.c 2009-08-10 08:31:08.000000000 +0200 @@ -59,16 +59,18 @@ static void omap_pcm_dma_irq(int ch, u16 struct omap_runtime_data *prtd = runtime->private_data; unsigned long flags; - if (cpu_is_omap1510()) { + if ((cpu_is_omap1510()) && + (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) { /* - * OMAP1510 doesn't support DMA chaining so have to restart - * the transfer after all periods are transferred + * OMAP1510 doesn't fully support DMA progress counter + * and there is no software emulation implemented yet, + * so have to maintain our own playback progress counter + * that can be used by omap_pcm_pointer() instead. */ spin_lock_irqsave(&prtd->lock, flags); if (prtd->period_index >= 0) { if (++prtd->period_index == runtime->periods) { prtd->period_index = 0; - omap_start_dma(prtd->dma_ch); } } spin_unlock_irqrestore(&prtd->lock, flags); @@ -100,7 +102,7 @@ static int omap_pcm_hw_params(struct snd prtd->dma_data = dma_data; err = omap_request_dma(dma_data->dma_req, dma_data->name, omap_pcm_dma_irq, substream, &prtd->dma_ch); - if (!err && !cpu_is_omap1510()) { + if (!err) { /* * Link channel with itself so DMA doesn't need any * reprogramming while looping the buffer @@ -119,8 +121,7 @@ static int omap_pcm_hw_free(struct snd_p if (prtd->dma_data == NULL) return 0; - if (!cpu_is_omap1510()) - omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch); + omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch); omap_free_dma(prtd->dma_ch); prtd->dma_data = NULL;

2 1

[alsa-devel] [PATCH] ASoC: UDA134X Codec: Fix mute/unmute code mistake and add ADC/DAC power control support
by Shine Liu 17 Aug '09

17 Aug '09

There is a mistake in current uda134x_mute function: mute_reg has been changed in line 162 or line 164, so uda134x_write should write "mute_reg" but not "mute_reg & ~(1<<2)" to UDA134X_DATA010. Besides, because there is no DAPM configuration for uda134x, when system starts up, snd_soc_int_card calls snd_soc_dapm_new_widgets, and snd_soc_dapm_new_widgets calls dapm_power_widgets. In function dapm_power_widgets, codec->dapm_widgets has no list entry, so sys_power retains it's original value zero. Then snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_PREPARE) and snd_soc_dapm_set_bias_level(socdev, SND_SOC_BIAS_STANDBY) are called sequentially. Finally, the uda134x codec goes to STANDBY mode, so the ADC/DAC power control bits of UDA134X_STATUS1 register keeps the 0 value. UDA134X has no trigger function currently, and the ADC/DAC power control bits of UDA134X_STATUS1 register are not exported in uda1341_snd_controls, so there's no way to enable the ADC/DAC power control bits when you want to use the codec. When playing with aplay or recording with arecord, there is no sound output or no wave input. I have added the uda134x_trigger function, which turns on/off the ADC/DAC power control bits according the substream type and the command. I also exported the ADC/DAC power control bits to uda1341_snd_controls, so we can also turn these bits on/off as we need via a mixer tool like alsamixer. The patch created against linux-2.6.31-rc3. Tested on a s3c2440 development board with UDA1341TS codec. Signed-off-by: Shine Liu <shinel(a)foxmail.com> --- sound/soc/codecs/uda134x.c.orig 2009-07-14 09:18:52.000000000 +0800 +++ sound/soc/codecs/uda134x.c 2009-08-17 13:46:57.000000000 +0800 @@ -163,7 +163,7 @@ else mute_reg &= ~(1<<2); - uda134x_write(codec, UDA134X_DATA010, mute_reg & ~(1<<2)); + uda134x_write(codec, UDA134X_DATA010, mute_reg); return 0; } @@ -339,6 +339,38 @@ return 0; } +static int uda134x_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + int stream = substream->stream; + struct snd_soc_codec *codec = dai->codec; + u8 power_ctrl_reg = uda134x_read_reg_cache(codec, UDA134X_STATUS1); + + pr_debug("%s stream: %d, cmd: %d\n", __func__, stream, cmd); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + power_ctrl_reg |= (1<<0); + else + power_ctrl_reg |= (1<<1); + break; + + case SNDRV_PCM_TRIGGER_STOP: + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + power_ctrl_reg &= ~(1<<0); + else + power_ctrl_reg &= ~(1<<1); + break; + + default: + return 0; + } + + uda134x_write(codec, UDA134X_STATUS1, power_ctrl_reg); + return 0; +} + static int uda134x_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -416,6 +448,8 @@ SOC_SINGLE("ADC Polarity Switch", UDA134X_STATUS1, 4, 1, 0), SOC_SINGLE("DAC Polarity Switch", UDA134X_STATUS1, 3, 1, 0), SOC_SINGLE("Double Speed Playback Switch", UDA134X_STATUS1, 2, 1, 0), +SOC_SINGLE("ADC Power Switch", UDA134X_STATUS1, 1, 1, 0), +SOC_SINGLE("DAC Power Switch", UDA134X_STATUS1, 0, 1, 0), SOC_SINGLE("DC Filter Enable Switch", UDA134X_STATUS0, 0, 1, 0), }; @@ -438,6 +472,7 @@ .digital_mute = uda134x_mute, .set_sysclk = uda134x_set_dai_sysclk, .set_fmt = uda134x_set_dai_fmt, + .trigger = uda134x_trigger, }; struct snd_soc_dai uda134x_dai = {

1 0

[alsa-devel] Digidesign Mbox2 - usb support
by Damien Zammit 17 Aug '09

17 Aug '09

I compiled the latest stable ALSA and kernel, and all the utilities. !!################################ !!ALSA Information Script v 0.4.57 !!################################ !!Script ran on: Mon Aug 17 06:39:10 UTC 2009 !!Linux Distribution !!------------------ Ubuntu 8.04 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 8.04" !!DMI Information !!--------------- Manufacturer: Dell Computer Corporation Product Name: Latitude D600 !!Kernel Information !!------------------ Kernel release: 2.6.30.4 Operating System: GNU/Linux Architecture: i686 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.20 Library version: 1.0.20 Utilities version: 1.0.20 !!Loaded ALSA modules !!------------------- snd_usb_audio !!Sound Servers on this system !!---------------------------- ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No aRts: Installed - Yes (/usr/bin/artsd) Running - No Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 1 [M2 ]: USB-Audio - Mbox 2 Digidesign Mbox 2 at usb-0000:00:1d.1-1, full speed !!PCI Soundcards installed in the system !!-------------------------------------- 00:1f.5 Multimedia audio controller: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller (rev 01) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1f.5 0401: 8086:24c5 (rev 01) Subsystem: 1028:011d !!Modprobe options (Sound related) !!-------------------------------- snd-bt87x: index=-2 snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=1 snd-bt87x: index=-2 snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 !!Loaded sound module options !!-------------------------- !!Module: snd_usb_audio async_unlink : Y device_setup : 0,0,0,0,0,0,0,0 enable : Y,Y,Y,Y,Y,Y,Y,Y id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> ignore_ctl_error : N index : 1,-1,-1,-1,-1,-1,-1,-1 nrpacks : 8 pid : -1,-1,-1,-1,-1,-1,-1,-1 vid : -1,-1,-1,-1,-1,-1,-1,-1 !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 32 Aug 17 15:08 /dev/snd/controlC1 crw-rw----+ 1 root audio 116, 40 Aug 17 15:08 /dev/snd/midiC1D0 crw-rw----+ 1 root audio 116, 56 Aug 17 16:36 /dev/snd/pcmC1D0c crw-rw----+ 1 root audio 116, 48 Aug 17 16:36 /dev/snd/pcmC1D0p crw-rw----+ 1 root audio 116, 1 Aug 17 14:50 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Aug 17 14:50 /dev/snd/timer !!ALSA configuration files !!------------------------ !!User specific config file (~/.asoundrc) # ALSA library configuration file # Include settings that are under the control of asoundconf(1). # (To disable these settings, comment out this line.) </home/dczammit/.asoundrc.asoundconf> !!asoundconf-generated config file # ALSA library configuration file managed by asoundconf(1). # # MANUAL CHANGES TO THIS FILE WILL BE OVERWRITTEN! # # Manual changes to the ALSA library configuration should be implemented # by editing the ~/.asoundrc file, not by editing this file. defaults.rawmidi.card 0 defaults.rawmidi.device 0 defaults.rawmidi.subdevice -1 defaults.hwdep.card 0 defaults.hwdep.device 0 defaults.timer.class 2 defaults.timer.sclass 0 defaults.timer.card 0 defaults.timer.device 0 defaults.timer.subdevice 0 defaults.namehint.showall off defaults.namehint.basic on defaults.namehint.extended off !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 1: M2 [Mbox 2], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 1: M2 [Mbox 2], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 1 [M2] Card hw:1 'M2'/'Digidesign Mbox 2 at usb-0000:00:1d.1-1, full speed' Mixer name : '' Components : 'USB0dba:3000' Controls : 0 Simple ctrls : 0 !!Alsactl output !!------------- --startcollapse-- state.M2 { control { } } --endcollapse-- jackd 0.116.2 Copyright 2001-2005 Paul Davis and others. jackd comes with ABSOLUTELY NO WARRANTY This is free software, and you are welcome to redistribute it under certain conditions; see the file COPYING for details JACK compiled with System V SHM support. loading driver .. apparent rate = 48000 creating alsa driver ... hw:1|hw:1|256|2|48000|2|2|nomon|swmeter|-|32bit control device hw:1 configuring for 48000Hz, period = 256 frames (5.3 ms), buffer = 2 periods ALSA: final selected sample format for capture: 24bit little-endian ALSA: use 2 periods for capture ALSA: cannot set hardware parameters for capture ALSA: cannot configure capture channel cannot load driver module alsa [ 6387.449014] WARNING: at sound/usb/usbaudio.c:1314 snd_usb_hw_params+0xbc8/0xc10 [snd_usb_audio]() [ 6387.449056] Hardware name: Latitude D600 [ 6387.449059] Modules linked in: snd_usb_audio snd_pcm snd_page_alloc snd_usb_lib snd_hwdep binfmt_misc radeon drm i2c_core rfcomm l2cap bluetooth ppdev sbs sbshc container acpi_cpufreq cpufreq_userspace cpufreq_stats cpufreq_powersave cpufreq_ondemand freq_table cpufreq_conservative ipv6 dm_crypt snd_seq_midi snd_rawmidi snd_seq_midi_event snd_seq snd_timer snd_seq_device snd soundcore lp af_packet joydev pcmcia yenta_socket rsrc_nonstatic video output pcmcia_core parport_pc parport psmouse serio_raw button battery ac processor pcspkr intel_agp shpchp agpgart pci_hotplug iTCO_wdt iTCO_vendor_support dcdbas evdev sr_mod sg cdrom sd_mod usbhid hid ata_piix pata_acpi ata_generic tg3 firmware_class libphy libata uhci_hcd ehci_hcd usbcore scsi_mod raid10 raid456 raid6_pq async_xor async_memcpy async_tx xor raid1 raid0 multipath linear md_mod dm_mirror dm_region_hash dm_log dm_snapshot dm_mod thermal fan thermal_sys fuse [ 6387.449148] Pid: 25866, comm: jackd Tainted: G W 2.6.30.4 #1 [ 6387.449151] Call Trace: [ 6387.449165] [<f81121d8>] ? snd_usb_hw_params+0xbc8/0xc10 [snd_usb_audio] [ 6387.449175] [<f81121d8>] ? snd_usb_hw_params+0xbc8/0xc10 [snd_usb_audio] [ 6387.449191] [<c012c03f>] ? warn_slowpath_common+0x7f/0xb0 [ 6387.449201] [<f81121d8>] ? snd_usb_hw_params+0xbc8/0xc10 [snd_usb_audio] [ 6387.449206] [<c012c083>] ? warn_slowpath_null+0x13/0x20 [ 6387.449216] [<f81121d8>] ? snd_usb_hw_params+0xbc8/0xc10 [snd_usb_audio] [ 6387.449237] [<f80eae90>] ? snd_pcm_hw_params_choose+0x30/0xa0 [snd_pcm] [ 6387.449248] [<f80e6fd0>] ? snd_pcm_hw_params+0xe0/0x320 [snd_pcm] [ 6387.449254] [<c0271dce>] ? copy_from_user+0x2e/0x70 [ 6387.449265] [<f80e7346>] ? snd_pcm_hw_params_user+0x36/0x80 [snd_pcm] [ 6387.449276] [<f80e7c06>] ? snd_pcm_common_ioctl1+0x7f6/0x1140 [snd_pcm] [ 6387.449281] [<c0129385>] ? check_preempt_wakeup+0xd5/0x160 [ 6387.449286] [<c01242e9>] ? try_to_wake_up+0xa9/0x200 [ 6387.449292] [<c01b80b5>] ? pollwake+0x45/0x50 [ 6387.449296] [<c0124440>] ? default_wake_function+0x0/0x10 [ 6387.449306] [<c011f0fb>] ? __wake_up_common+0x4b/0x80 [ 6387.449310] [<c01201f5>] ? __wake_up_sync_key+0x45/0x80 [ 6387.449314] [<c01bf8cd>] ? mnt_drop_write+0x1d/0xe0 [ 6387.449319] [<c01bbebb>] ? file_update_time+0x7b/0xc0 [ 6387.449325] [<c01b06ee>] ? pipe_write+0x32e/0x450 [ 6387.449337] [<f80e85de>] ? snd_pcm_capture_ioctl1+0x8e/0x400 [snd_pcm] [ 6387.449348] [<f80e8993>] ? snd_pcm_capture_ioctl+0x43/0x50 [snd_pcm] [ 6387.449359] [<f80e8950>] ? snd_pcm_capture_ioctl+0x0/0x50 [snd_pcm] [ 6387.449364] [<c01b629b>] ? vfs_ioctl+0x2b/0x90 [ 6387.449367] [<c01b65d0>] ? do_vfs_ioctl+0x2d0/0x510 [ 6387.449374] [<c0399075>] ? __schedule+0x235/0x730 [ 6387.449378] [<c01a9b0d>] ? vfs_write+0xfd/0x160 [ 6387.449382] [<c01b6866>] ? sys_ioctl+0x56/0x70 [ 6387.449387] [<c0102dd4>] ? sysenter_do_call+0x12/0x28 [ 6387.449391] ---[ end trace 8fe22d9a0585e3c1 ]---

1 0

Re: [alsa-devel] Digidesign Mbox2 - usb support
by Damien Zammit 16 Aug '09

16 Aug '09

Using the following patch to usbquirks.h, I was able to get MIDI in and out working on the Digidesign Mbox2!! The capture/playback PCMs are also configured with alsa, but the status on these are "closed". I am guessing I need to snoop the commands on interface 1 (control device) to enable them. Also, there is some "magic" needed to initialise the device which I have captured. It seems that the device needs to be loaded with altsettings = 0 on interfaces 1-5 first, then some setup codes are transmitted, and finally the following altsettings are used. dczammit@flop:/proc/asound$ ls card0 cards devices hwdep M2 modules pcm seq timers version dczammit@flop:/proc/asound$ cat cards 0 [M2 ]: USB-Audio - Mbox 2 Digidesign Mbox 2 at usb-0000:00:1d.1-1, full speed dczammit@flop:/proc/asound$ cat devices 0: [ 0] : control 1: : sequencer 8: [ 0- 0]: raw midi 16: [ 0- 0]: digital audio playback 24: [ 0- 0]: digital audio capture 33: : timer dczammit@flop:/proc/asound$ cat pcm 00-00: USB Audio : USB Audio : playback 1 : capture 1 dczammit@flop:/proc/asound$ cat timers G0: system timer : 1000.000us (10000000 ticks) G1: RTC timer : 976.562us (100000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE dczammit@flop:/proc/asound/M2$ ls id midi0 pcm0c pcm0p stream0 usbbus usbid dczammit@flop:/proc/asound/M2$ cat midi0 Mbox 2 Output 0 Tx bytes : 212 Input 0 Rx bytes : 314 dczammit@flop:/proc/asound/M2$ cat stream0 Digidesign Mbox 2 at usb-0000:00:1d.1-1, full speed : USB Audio Playback: Status: Stop Interface 3 Altset 3 Format: 0x20 Channels: 2 Endpoint: 4 OUT (ASYNC) Rates: 8000 - 48000 (continuous) Capture: Status: Stop Interface 5 Altset 3 Format: 0x20 Channels: 2 Endpoint: 6 IN (SYNC) Rates: 8000 - 48000 (continuous) dczammit@flop:/proc/asound/seq$ aplay -Ddefault:CARD=M2 -f S16_LE ~/Desktop/tunes/dzjam.wav Playing WAVE '/home/dczammit/Desktop/tunes/dzjam.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Segmentation fault Please find the quirk below and the magic init codes. ################################## { 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 }, { /* Audio control interface - sets the sampling rate? - no idea how this works Possibly also enables the capture/playback using a command? */ .ifnum = 1, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 2, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 3, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 3, .altsetting = 3, /* 0 is also available, but has no endpoints */ .altset_idx = 1, .endpoint = 0x04, /* OUT or 0x84 IN (attr 1) */ .ep_attr = 0x05, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, .rate_max = 48000 } }, { .ifnum = 4, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 5, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 5, .altsetting = 3, .altset_idx = 1, .endpoint = 0x86, /* IN */ .ep_attr = 0x0D, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, .rate_max = 48000 } }, { .ifnum = 6, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = & (const struct snd_usb_midi_endpoint_info) { .out_ep = 2, .out_interval = 0x00, .in_ep = 2, .in_interval = 0x00, .out_cables = 0x0001, .in_cables = 0x0001 } }, { .ifnum = -1 } } } }, ################################## Audio driver magic: [0 ms] UsbSnoop compiled on Jan 18 2003 22:41:32 loading [0 ms] UsbSnoop - DriverEntry(baca2c40) : Windows NT WDM version 1.32 [101 ms] UsbSnoop - AddDevice(baca2f50) : DriverObject 83499768, pdo 83338c10 [102 ms] UsbSnoop - MyDispatchPNP(baca2ee0) : IRP_MJ_PNP (IRP_MN_QUERY_INTERFACE) [102 ms] UsbSnoop - MyDispatchPNP(baca2ee0) : IRP_MJ_PNP (0x00000018) [127 ms] UsbSnoop - MyDispatchPNP(baca2ee0) : IRP_MJ_PNP (IRP_MN_QUERY_RESOURCE_REQUIREMENTS) [127 ms] UsbSnoop - MyDispatchPNP(baca2ee0) : IRP_MJ_PNP (IRP_MN_FILTER_RESOURCE_REQUIREMENTS) [128 ms] UsbSnoop - MyDispatchPNP(baca2ee0) : IRP_MJ_PNP (IRP_MN_START_DEVICE) [128 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=832c7530, IRQL=0 [128 ms] >>> URB 1 going down >>> -- URB_FUNCTION_GET_DESCRIPTOR_FROM_DEVICE: TransferBufferLength = 00000012 TransferBuffer = bacef95c TransferBufferMDL = 00000000 Index = 00000001 DescriptorType = 00000001 (USB_DEVICE_DESCRIPTOR_TYPE) LanguageId = 00000000 [128 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=832c7530, Context=832ff2e8, IRQL=0 [128 ms] <<< URB 1 coming back <<< -- URB_FUNCTION_GET_DESCRIPTOR_FROM_DEVICE: TransferBufferLength = 00000012 TransferBuffer = bacef95c TransferBufferMDL = 00000000 00000000: 12 01 00 01 00 00 00 08 ba 0d 00 30 38 01 01 02 00000010: 00 01 Index = 00000001 DescriptorType = 00000001 (USB_DEVICE_DESCRIPTOR_TYPE) LanguageId = 00000000 [128 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=832c7530, IRQL=0 [128 ms] >>> URB 2 going down >>> -- URB_FUNCTION_GET_DESCRIPTOR_FROM_DEVICE: TransferBufferLength = 00000009 TransferBuffer = bacef9d0 TransferBufferMDL = 00000000 Index = 00000001 DescriptorType = 00000002 (USB_CONFIGURATION_DESCRIPTOR_TYPE) LanguageId = 00000000 [128 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=832c7530, Context=833e2438, IRQL=0 [128 ms] <<< URB 2 coming back <<< -- URB_FUNCTION_GET_DESCRIPTOR_FROM_DEVICE: TransferBufferLength = 00000009 TransferBuffer = bacef9d0 TransferBufferMDL = 00000000 00000000: 09 02 58 02 05 01 00 80 fa Index = 00000001 DescriptorType = 00000002 (USB_CONFIGURATION_DESCRIPTOR_TYPE) LanguageId = 00000000 [128 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=832c7530, IRQL=0 [128 ms] >>> URB 3 going down >>> -- URB_FUNCTION_GET_DESCRIPTOR_FROM_DEVICE: TransferBufferLength = 00000258 TransferBuffer = 83369540 TransferBufferMDL = 00000000 Index = 00000001 DescriptorType = 00000002 (USB_CONFIGURATION_DESCRIPTOR_TYPE) LanguageId = 00000000 [128 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=832c7530, Context=83455a90, IRQL=0 [128 ms] <<< URB 3 coming back <<< -- URB_FUNCTION_GET_DESCRIPTOR_FROM_DEVICE: TransferBufferLength = 00000258 TransferBuffer = 83369540 TransferBufferMDL = 00000000 00000000: 09 02 58 02 05 01 00 80 fa 09 04 01 00 00 01 01 00000010: 00 02 0c 24 01 00 01 60 00 04 02 03 04 05 0c 24 00000020: 02 05 01 01 00 02 03 00 00 00 09 24 03 06 01 06 00000030: 00 05 00 0c 24 02 07 01 01 00 02 03 00 00 00 09 00000040: 24 03 08 01 06 00 07 00 0c 24 02 01 01 06 00 02 00000050: 03 00 00 00 09 24 03 02 01 01 00 01 00 0c 24 02 00000060: 03 01 06 00 02 03 00 00 00 09 24 03 04 01 01 00 00000070: 03 00 09 04 02 00 00 01 02 00 00 09 04 02 02 02 00000080: 01 02 00 00 07 24 01 05 01 01 00 0e 24 02 01 02 00000090: 03 18 00 40 1f 00 80 bb 00 09 05 03 05 28 01 01 000000a0: 00 83 07 25 01 00 00 00 00 09 05 83 01 03 00 01 000000b0: 02 00 09 04 02 03 02 01 02 00 00 07 24 01 05 01 000000c0: 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 000000d0: 09 05 03 05 28 01 01 00 83 07 25 01 00 00 00 00 000000e0: 09 05 83 01 03 00 01 02 00 09 04 03 00 00 01 02 000000f0: 00 00 09 04 03 02 02 01 02 00 00 07 24 01 07 01 00000100: 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 00000110: 09 05 04 05 28 01 01 00 84 07 25 01 00 00 00 00 00000120: 09 05 84 01 03 00 01 02 00 09 04 03 03 02 01 02 00000130: 00 00 07 24 01 07 01 01 00 0e 24 02 01 02 03 18 00000140: 00 40 1f 00 80 bb 00 09 05 04 05 28 01 01 00 84 00000150: 07 25 01 00 00 00 00 09 05 84 01 03 00 01 02 00 00000160: 09 04 03 06 01 01 02 00 00 07 24 01 07 01 01 20 00000170: 0e 24 02 03 02 02 10 00 40 1f 00 80 bb 00 09 05 00000180: 04 0d c0 00 01 00 00 07 25 01 01 00 00 00 09 04 00000190: 04 00 00 01 02 00 00 09 04 04 02 01 01 02 00 00 000001a0: 07 24 01 02 01 01 00 0e 24 02 01 02 03 18 00 40 000001b0: 1f 00 80 bb 00 09 05 85 05 28 01 01 00 00 07 25 000001c0: 01 00 00 00 00 09 04 04 03 01 01 02 00 00 07 24 000001d0: 01 02 01 01 00 0e 24 02 01 02 03 18 00 40 1f 00 000001e0: 80 bb 00 09 05 85 0d 28 01 01 00 00 07 25 01 01 000001f0: 00 00 00 09 04 05 00 00 01 02 00 00 09 04 05 02 00000200: 01 01 02 00 00 07 24 01 04 01 01 00 0e 24 02 01 00000210: 02 03 18 00 40 1f 00 80 bb 00 09 05 86 05 28 01 00000220: 01 00 00 07 25 01 00 00 00 00 09 04 05 03 01 01 00000230: 02 00 00 07 24 01 04 01 01 00 0e 24 02 01 02 03 00000240: 18 00 40 1f 00 80 bb 00 09 05 86 0d 28 01 01 00 00000250: 00 07 25 01 01 00 00 00 Index = 00000001 DescriptorType = 00000002 (USB_CONFIGURATION_DESCRIPTOR_TYPE) LanguageId = 00000000 [128 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=832c7530, IRQL=0 [128 ms] >>> URB 4 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [133 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=832c7530, Context=833573c0, IRQL=2 [133 ms] <<< URB 4 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 83623978 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [433 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=83436508, IRQL=0 [433 ms] >>> URB 5 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [437 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=83436508, Context=83429560, IRQL=2 [437 ms] <<< URB 5 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 834a08e8 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [733 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=83436508, IRQL=0 [733 ms] >>> URB 6 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [738 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=83436508, Context=83625f08, IRQL=2 [738 ms] <<< URB 6 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 834a08e8 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [1034 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=834a1398, IRQL=0 [1034 ms] >>> URB 7 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [1038 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=834a1398, Context=832ff2e8, IRQL=2 [1038 ms] <<< URB 7 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 834a08e8 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [1334 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [1334 ms] >>> URB 8 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [1339 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=833573c0, IRQL=2 [1339 ms] <<< URB 8 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 83621768 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [1635 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [1635 ms] >>> URB 9 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [1639 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=832ff2e8, IRQL=2 [1639 ms] <<< URB 9 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 83621768 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [1935 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [1935 ms] >>> URB 10 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [1940 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=833573c0, IRQL=2 [1940 ms] <<< URB 10 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 83621768 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [2236 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [2236 ms] >>> URB 11 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [2240 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=832ff2e8, IRQL=2 [2240 ms] <<< URB 11 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 83621768 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [2536 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [2536 ms] >>> URB 12 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [2541 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=833573c0, IRQL=2 [2541 ms] <<< URB 12 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 83621768 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [2837 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [2837 ms] >>> URB 13 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [2841 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=832ff2e8, IRQL=2 [2841 ms] <<< URB 13 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 83621768 00000000: 01 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [3137 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [3137 ms] >>> URB 14 going down >>> -- URB_FUNCTION_VENDOR_DEVICE: TransferFlags = 00000001 (USBD_TRANSFER_DIRECTION_IN, ~USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000012 TransferBuffer = bacef8e7 TransferBufferMDL = 00000000 UrbLink = 00000000 RequestTypeReservedBits = 00000000 Request = 00000085 Value = 00000001 Index = 00000000 [3142 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=833573c0, IRQL=2 [3142 ms] <<< URB 14 coming back <<< -- URB_FUNCTION_CONTROL_TRANSFER: PipeHandle = 83323ac0 TransferFlags = 0000000b (USBD_TRANSFER_DIRECTION_IN, USBD_SHORT_TRANSFER_OK) TransferBufferLength = 00000001 TransferBuffer = bacef8e7 TransferBufferMDL = 83621768 00000000: 02 UrbLink = 00000000 SetupPacket = 00000000: c0 85 01 00 00 00 12 00 [3142 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [3142 ms] >>> URB 15 going down >>> -- URB_FUNCTION_SELECT_CONFIGURATION: ConfigurationDescriptor = 0x83369540 (configure) ConfigurationDescriptor : bLength = 9 ConfigurationDescriptor : bDescriptorType = 0x00000002 ConfigurationDescriptor : wTotalLength = 0x00000258 ConfigurationDescriptor : bNumInterfaces = 0x00000005 ConfigurationDescriptor : bConfigurationValue = 0x00000001 ConfigurationDescriptor : iConfiguration = 0x00000000 ConfigurationDescriptor : bmAttributes = 0x00000080 ConfigurationDescriptor : MaxPower = 0x000000fa ConfigurationHandle = 0x00000000 Interface[0]: Length = 16 Interface[0]: InterfaceNumber = 1 Interface[0]: AlternateSetting = 0 Interface[1]: Length = 16 Interface[1]: InterfaceNumber = 2 Interface[1]: AlternateSetting = 0 Interface[2]: Length = 16 Interface[2]: InterfaceNumber = 3 Interface[2]: AlternateSetting = 0 Interface[3]: Length = 16 Interface[3]: InterfaceNumber = 4 Interface[3]: AlternateSetting = 0 Interface[4]: Length = 16 Interface[4]: InterfaceNumber = 5 Interface[4]: AlternateSetting = 0 [3142 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=83429560, IRQL=0 [3142 ms] <<< URB 15 coming back <<< -- URB_FUNCTION_SELECT_CONFIGURATION: ConfigurationDescriptor = 0x83369540 (configure) ConfigurationDescriptor : bLength = 9 ConfigurationDescriptor : bDescriptorType = 0x00000002 ConfigurationDescriptor : wTotalLength = 0x00000258 ConfigurationDescriptor : bNumInterfaces = 0x00000005 ConfigurationDescriptor : bConfigurationValue = 0x00000001 ConfigurationDescriptor : iConfiguration = 0x00000000 ConfigurationDescriptor : bmAttributes = 0x00000080 ConfigurationDescriptor : MaxPower = 0x000000fa ConfigurationHandle = 0x835a8978 Interface[0]: Length = 16 Interface[0]: InterfaceNumber = 1 Interface[0]: AlternateSetting = 0 Interface[0]: Class = 0x00000001 Interface[0]: SubClass = 0x00000001 Interface[0]: Protocol = 0x00000000 Interface[0]: InterfaceHandle = 0x8336b868 Interface[0]: NumberOfPipes = 0 Interface[1]: Length = 16 Interface[1]: InterfaceNumber = 2 Interface[1]: AlternateSetting = 0 Interface[1]: Class = 0x00000000 Interface[1]: SubClass = 0x00000000 Interface[1]: Protocol = 0x00000000 Interface[1]: InterfaceHandle = 0x00000000 Interface[1]: NumberOfPipes = 0 Interface[2]: Length = 16 Interface[2]: InterfaceNumber = 3 Interface[2]: AlternateSetting = 0 Interface[2]: Class = 0x00000000 Interface[2]: SubClass = 0x00000000 Interface[2]: Protocol = 0x00000000 Interface[2]: InterfaceHandle = 0x00000000 Interface[2]: NumberOfPipes = 0 Interface[3]: Length = 16 Interface[3]: InterfaceNumber = 4 Interface[3]: AlternateSetting = 0 Interface[3]: Class = 0x00000000 Interface[3]: SubClass = 0x00000000 Interface[3]: Protocol = 0x00000000 Interface[3]: InterfaceHandle = 0x00000000 Interface[3]: NumberOfPipes = 0 Interface[4]: Length = 16 Interface[4]: InterfaceNumber = 5 Interface[4]: AlternateSetting = 0 Interface[4]: Class = 0x00000000 Interface[4]: SubClass = 0x00000000 Interface[4]: Protocol = 0x00000000 Interface[4]: InterfaceHandle = 0x00000000 Interface[4]: NumberOfPipes = 0 [3142 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [3142 ms] >>> URB 16 going down >>> -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 56 Interface: InterfaceNumber = 2 Interface: AlternateSetting = 0 Interface: Class = 0x00000000 Interface: SubClass = 0x00000000 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 00000000 Interface: NumberOfPipes = 0 [3145 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=832f0090, IRQL=0 [3145 ms] <<< URB 16 coming back <<< -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 16 Interface: InterfaceNumber = 2 Interface: AlternateSetting = 0 Interface: Class = 0x00000001 Interface: SubClass = 0x00000002 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 83621580 Interface: NumberOfPipes = 0 [3145 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [3145 ms] >>> URB 17 going down >>> -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 56 Interface: InterfaceNumber = 2 Interface: AlternateSetting = 3 Interface: Class = 0x00000000 Interface: SubClass = 0x00000000 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 00000000 Interface: NumberOfPipes = 0 [3187 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=83539d80, IRQL=0 [3187 ms] <<< URB 17 coming back <<< -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 56 Interface: InterfaceNumber = 2 Interface: AlternateSetting = 3 Interface: Class = 0x00000001 Interface: SubClass = 0x00000002 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 833b9d68 Interface: NumberOfPipes = 2 Interface: Pipes[0] : MaximumPacketSize = 0x00000128 Interface: Pipes[0] : EndpointAddress = 0x00000003 Interface: Pipes[0] : Interval = 0x00000001 Interface: Pipes[0] : PipeType = 0x00000001 (UsbdPipeTypeIsochronous) Interface: Pipes[0] : PipeHandle = 0x833b9d84 Interface: Pipes[0] : MaxTransferSize = 0x00000000 Interface: Pipes[0] : PipeFlags = 0x00000000 Interface: Pipes[1] : MaximumPacketSize = 0x00000003 Interface: Pipes[1] : EndpointAddress = 0x00000083 Interface: Pipes[1] : Interval = 0x00000001 Interface: Pipes[1] : PipeType = 0x00000001 (UsbdPipeTypeIsochronous) Interface: Pipes[1] : PipeHandle = 0x833b9da4 Interface: Pipes[1] : MaxTransferSize = 0x00000000 Interface: Pipes[1] : PipeFlags = 0x00000000 [3187 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [3187 ms] >>> URB 18 going down >>> -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 76 Interface: InterfaceNumber = 4 Interface: AlternateSetting = 0 Interface: Class = 0x00000000 Interface: SubClass = 0x00000000 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 00000000 Interface: NumberOfPipes = 0 [3190 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=835fced8, IRQL=0 [3190 ms] <<< URB 18 coming back <<< -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 16 Interface: InterfaceNumber = 4 Interface: AlternateSetting = 0 Interface: Class = 0x00000001 Interface: SubClass = 0x00000002 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 83621580 Interface: NumberOfPipes = 0 [3190 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [3190 ms] >>> URB 19 going down >>> -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 76 Interface: InterfaceNumber = 4 Interface: AlternateSetting = 3 Interface: Class = 0x00000000 Interface: SubClass = 0x00000000 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 00000000 Interface: NumberOfPipes = 0 [3207 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=832f0090, IRQL=0 [3207 ms] <<< URB 19 coming back <<< -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 36 Interface: InterfaceNumber = 4 Interface: AlternateSetting = 3 Interface: Class = 0x00000001 Interface: SubClass = 0x00000002 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 834a36c8 Interface: NumberOfPipes = 1 Interface: Pipes[0] : MaximumPacketSize = 0x00000128 Interface: Pipes[0] : EndpointAddress = 0x00000085 Interface: Pipes[0] : Interval = 0x00000001 Interface: Pipes[0] : PipeType = 0x00000001 (UsbdPipeTypeIsochronous) Interface: Pipes[0] : PipeHandle = 0x834a36e4 Interface: Pipes[0] : MaxTransferSize = 0x00000000 Interface: Pipes[0] : PipeFlags = 0x00000000 [3207 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [3207 ms] >>> URB 20 going down >>> -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 76 Interface: InterfaceNumber = 5 Interface: AlternateSetting = 0 Interface: Class = 0x00000000 Interface: SubClass = 0x00000000 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 00000000 Interface: NumberOfPipes = 0 [3210 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=83539d80, IRQL=0 [3210 ms] <<< URB 20 coming back <<< -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 16 Interface: InterfaceNumber = 5 Interface: AlternateSetting = 0 Interface: Class = 0x00000001 Interface: SubClass = 0x00000002 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 83621580 Interface: NumberOfPipes = 0 [3210 ms] UsbSnoop - MyDispatchInternalIOCTL(baca1e80) : fdo=83338c10, Irp=8348a7c0, IRQL=0 [3210 ms] >>> URB 21 going down >>> -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 76 Interface: InterfaceNumber = 5 Interface: AlternateSetting = 3 Interface: Class = 0x00000000 Interface: SubClass = 0x00000000 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 00000000 Interface: NumberOfPipes = 0 [3227 ms] UsbSnoop - MyInternalIOCTLCompletion(baca1db0) : fido=00000000, Irp=8348a7c0, Context=835fced8, IRQL=0 [3227 ms] <<< URB 21 coming back <<< -- URB_FUNCTION_SELECT_INTERFACE: ConfigurationHandle = 0x835a8978 Interface: Length = 36 Interface: InterfaceNumber = 5 Interface: AlternateSetting = 3 Interface: Class = 0x00000001 Interface: SubClass = 0x00000002 Interface: Protocol = 0x00000000 Interface: InterfaceHandle = 83631b58 Interface: NumberOfPipes = 1 Interface: Pipes[0] : MaximumPacketSize = 0x00000128 Interface: Pipes[0] : EndpointAddress = 0x00000086 Interface: Pipes[0] : Interval = 0x00000001 Interface: Pipes[0] : PipeType = 0x00000001 (UsbdPipeTypeIsochronous) Interface: Pipes[0] : PipeHandle = 0x83631b74 Interface: Pipes[0] : MaxTransferSize = 0x00000000 Interface: Pipes[0] : PipeFlags = 0x00000000 [3228 ms] UsbSnoop - MyDispatchPNP(baca2ee0) : IRP_MJ_PNP (IRP_MN_QUERY_CAPABILITIES) [3228 ms] UsbSnoop - MyDispatchPNP(baca2ee0) : IRP_MJ_PNP (IRP_MN_QUERY_PNP_DEVICE_STATE) [3228 ms] UsbSnoop - MyDispatchPNP(baca2ee0) : IRP_MJ_PNP (IRP_MN_QUERY_DEVICE_RELATIONS) [3288 ms] UsbSnoop - MyDispatchPNP(baca2ee0) : IRP_MJ_PNP (IRP_MN_QUERY_DEVICE_RELATIONS) Audio driver "magic" successful response: usb_get_driver_np returned -61 1 get descriptor returned 18, bytes: 12 01 00 01 00 00 00 08 ba 0d 00 30 38 01 01 02 00 01 "...........08....." 2 get descriptor returned 9, bytes: 09 02 86 02 07 01 00 80 fa "........." 3 get descriptor returned 600, bytes: 09 02 86 02 07 01 00 80 fa 09 04 00 00 00 fe 01 00 00 07 21 05 ff ff 08 00 09 04 01 00 00 01 01 00 02 0c 24 01 00 01 60 00 04 02 03 04 05 0c 24 02 05 01 01 00 02 03 00 00 00 09 24 03 06 01 06 00 05 00 0c 24 02 07 01 01 00 02 03 00 00 00 09 24 03 08 01 06 00 07 00 0c 24 02 01 01 06 00 02 03 00 00 00 09 24 03 02 01 01 00 01 00 0c 24 02 03 01 06 00 02 03 00 00 00 09 24 03 04 01 01 00 03 00 09 04 02 00 00 01 02 00 00 09 04 02 02 02 01 02 00 00 07 24 01 05 01 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 09 05 03 05 28 01 01 00 83 07 25 01 00 00 00 00 09 05 83 01 03 00 01 02 00 09 04 02 03 02 01 02 00 00 07 24 01 05 01 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 09 05 03 05 28 01 01 00 83 07 25 01 00 00 00 00 09 05 83 01 03 00 01 02 00 09 04 03 00 00 01 02 00 00 09 04 03 02 02 01 02 00 00 07 24 01 07 01 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 09 05 04 05 28 01 01 00 84 07 25 01 00 00 00 00 09 05 84 01 03 00 01 02 00 09 04 03 03 02 01 02 00 00 07 24 01 07 01 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 09 05 04 05 28 01 01 00 84 07 25 01 00 00 00 00 09 05 84 01 03 00 01 02 00 09 04 03 06 01 01 02 00 00 07 24 01 07 01 01 20 0e 24 02 03 02 02 10 00 40 1f 00 80 bb 00 09 05 04 0d c0 00 01 00 00 07 25 01 01 00 00 00 09 04 04 00 00 01 02 00 00 09 04 04 02 01 01 02 00 00 07 24 01 02 01 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 09 05 85 05 28 01 01 00 00 07 25 01 00 00 00 00 09 04 04 03 01 01 02 00 00 07 24 01 02 01 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 09 05 85 0d 28 01 01 00 00 07 25 01 01 00 00 00 09 04 05 00 00 01 02 00 00 09 04 05 02 01 01 02 00 00 07 24 01 04 01 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 09 05 86 05 28 01 01 00 00 07 25 01 00 00 00 00 09 04 05 03 01 01 02 00 00 07 24 01 04 01 01 00 0e 24 02 01 02 03 18 00 40 1f 00 80 bb 00 "...................!...............$...`.......$...........$........$...........$........$...........$........$...........$..........................$......$......@.........(.....%........................$......$......@.........(.....%.................................$......$......@.........(.....%........................$......$......@.........(.....%........................$.... .$......@...............%........................$......$......@.........(.....%...............$......$......@.........(.....%........................$......$......@.........(.....%...............$......$......@....." 4 control msg returned 1, bytes: 01 "." 5 control msg returned 1, bytes: 01 "." 6 control msg returned 1, bytes: 01 "." 7 control msg returned 1, bytes: 01 "." 8 control msg returned 1, bytes: 01 "." 9 control msg returned 1, bytes: 01 "." 10 control msg returned 1, bytes: 01 "." 11 control msg returned 1, bytes: 01 "." 12 control msg returned 1, bytes: 01 "." 13 control msg returned 1, bytes: 01 "." 14 control msg returned 1, bytes: 02 "." 15 set configuration returned 0 15 set alternate setting returned 0 16 set alternate setting returned 0 17 set alternate setting returned -22 18 set alternate setting returned 0 19 set alternate setting returned -22 20 set alternate setting returned 0 21 set alternate setting returned -22 I need help! How can I enable the capture/playback?

1 0

[alsa-devel] snd_pcm_drain() and O_NONBLOCK
by Jon Smirl 16 Aug '09

16 Aug '09

I'm trying to fix brutefir to properly drain the driver before exiting. This sequence does not work on a running stream. snd_pcm_nonblock() removes the O_NONBLOCK flag from the file handle but the change does not get copied to stream->f_flags. snd_pcm_drain() returns -EAGAIN since the flags have not be copied. snd_pcm_nonblock(handles[IO][n], 0); snd_pcm_drain(handles[IO][n]); snd_pcm_nonblock(handles[IO][n], SND_PCM_NONBLOCK); static int snd_pcm_pre_drain_init(struct snd_pcm_substream *substream, int state) { printk("snd_pcm_pre_drain_init flags %x\n", substream->f_flags); if (substream->f_flags & O_NONBLOCK) return -EAGAIN; printk("snd_pcm_pre_drain_init 1\n"); substream->runtime->trigger_master = substream; return 0; } Searching through the ALSA code the only way f_flags gets copied from the file handle to the stream handle is via snd_pcm_prepare(). But I can't do a snd_pcm_prepare() on a running stream. I can't find any way to change the NONBLOCK status on a running stream. I also tried implementing the wait for drain the the user app. snd_pcm_drain() always returns -EAGAIN of O_NONBLOCK is set. It can't be used to poll for the drain status. So how do I wait for a stream to drain? -- Jon Smirl jonsmirl(a)gmail.com

1 0

Re: [alsa-devel] Digidesign Mbox2 - usb support
by Damien Zammit 16 Aug '09

16 Aug '09

Hi, I wrote an addition to usbquirks.h to provide support for "Digidesign Mbox 2" but I need someone to point me in the right direction as to which interfaces to use. I'm not sure if the MIDI interface is a standard device, and I'm not sure which interfaces to ignore. I am about to compile a new alsa + kernel with my changes to usbquirks.h to test the device. Here is my usbquirks.h addition: /* DIGIDESIGN MBOX 2 addition to usbquirks.h */ { /* Damien Zammit <damien.zammit(a)gmail.com> * 16 August 2009 * * Interface topology of Digidesign Mbox 2, it has 7 interfaces: * 0 - Proprietary firmware updates * 1 - Audio control device * 2 - Double endpoint IN-OUT PCM * 3 - Double endpoint OUT-IN PCM * 4 - Single endpoint IEC1937 AC3 / PCM (selectable by altsetting) * 5 - Single endpoint PCM (supports full 24bit 48000Hz) * 6 - Triple endpoint MIDI (Interrupt + 1 IN + 1 OUT) * * As I am unsure which interfaces to try, I have provided _all_ the settings * I can in this file to hand to the usb-audio developers. * I have no idea what .altset_idx does or where to find it, so I set it to 1 in all cases. * * I am currently testing with just interfaces 5 and 6 (not sure if it will work) */ 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[]) { /* Proprietary firmware updater interface - ignore for now */ { .ifnum = 0, .type = QUIRK_IGNORE_INTERFACE }, /* Audio control interface - altsetting = 0 not sure what to do with this */ { .ifnum = 1, .type = QUIRK_IGNORE_INTERFACE }, { .ifnum = 2, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 2, .altsetting = 3, /* 0 is also available but has no endpoints */ .altset_idx = 1, .endpoint = 0x03, /* IN or 0x83 OUT */ .ep_attr = 0x05, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, .rate_max = 48000 } }, { .ifnum = 3, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 3, .altsetting = 3, /* 0 is also available, but has no endpoints */ .altset_idx = 1, .endpoint = 0x04, /* OUT or 0x84 IN */ .ep_attr = 0x05, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, .rate_max = 48000 } }, { .ifnum = 4, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = <<<<< Type III - IEC1937_AC3 (16 bit) >>>>> .channels = 2, .iface = 3, .altsetting = 6, .altset_idx = 1, .endpoint = 0x04, .ep_attr = 0x05, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, .rate_max = 48000 } }, { .ifnum = 5, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 5, .altsetting = 3, .altset_idx = 1, .endpoint = 0x86, /* Single endpoint only */ .ep_attr = 0x0D, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 8000, .rate_max = 48000 } }, { /* MIDI INTERFACE DESCRIPTION * Is this a standard USB MIDI device? * It has 2 Interface descriptors for the same altsetting... Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 6 bAlternateSetting 0 bNumEndpoints 3 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 6 bAlternateSetting 0 bNumEndpoints 3 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 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 0x0010 1x 16 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 0 */ .ifnum = 6, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = & (const struct snd_usb_midi_endpoint_info) { /* I am not certain of the "cables" flags or how * to find out what they are. */ .out_ep = 2, .out_interval = 0x00, .in_ep = 2, .in_interval = 0x00, .out_cables = 0x0001, .in_cables = 0x0001 } }, { .ifnum = -1 } } } },

1 0

Re: [alsa-devel] [PATCH] ALSA: hda: partially revert ALC268 quirk collapse
by Takashi Iwai 15 Aug '09

15 Aug '09

At Sat, 15 Aug 2009 16:38:46 +0400, Alexey Dobriyan wrote: > > Fix alsamixer inability to start: > > ALSA lib simple_none.c:1491:(simple_add1) helem (MIXER,'Master Playback Switch',0,3,0) appears twice or more > > ALC268 codec shipped with Toshiba Satellite X200 has ids 1179:ff00 and > began to match quirk table. > > If it doesn't match and auto-probe from BIOS is used, alsamixer works: > > hda_codec: Unknown model for ALC268, trying auto-probe from BIOS... > > 8871e5b91518a47284b6bc2603b44dbc79c85446 is first bad commit > commit 8871e5b91518a47284b6bc2603b44dbc79c85446 > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Tue Jun 2 01:02:50 2009 +0200 > > ALSA: hda - Reorder and clean-up ALC268 quirk table > > Rearrange alc268_cfg_tbl[] in the order of vendor id, and group some > entries using SND_PCI_QUIRK_MASK(). > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > :040000 040000 2497bb6fc5305acf3191b8e9b82956ec0ea1aee9 6f1d9ff9fed20550a1b2b398c7d2d5fec705fcd9 M sound > > Signed-off-by: Alexey Dobriyan <adobriyan(a)gmail.com> Well, it's no right fix. The symptom implies that something wrong with the current model=toshiba, so fixing the quirk is the way to go at this moment. Could you give alsa-info.sh output (run with --no-upload option) as vanilla kernel? Run with both model=auto and without model option and attach both results to compare. The former means actually the condition as your patch does. thanks, Takashi

1 0

[alsa-devel] Digidesign Mbox2 - usb support
by Damien Zammit 15 Aug '09

15 Aug '09

I have a Digidesign Mbox2 and I am trying to write an open source alsa driver for it. It is a usb self powered device, and I used usbsnoop on the Windows XP drivers to capture the initialisation of the device. It doesn't seem to require any firmware, as I was able to playback the init code to the device over libusb and all the lights turned on (which hopefully means it's awaiting further instructions to play/record). I have written some C code which sends the init codes to the device using raw URB, but I really need someone who is familiar with snd-usb-audio to take a look at my lsusb output in case it is a standard device before I waste any more time on it. Please find the debugging output for my experimental initialisation driver here: http://cytdesign.com/mbox2/mbox2.tar.bz2 Note: After running my init codes, all the lights appeared on the device just as it does when you boot windows. Please contact me at damien.zammit [at] gmail.com if you are able to help. I'm not up to speed with mailing lists, I would prefer an email to my account if you are replying to me. Thanks, Damien Zammit

1 0

[alsa-devel] snd_pcm_delay, hw buffers and driver api (v2)
by Kai Vehmanen 15 Aug '09

15 Aug '09

Hi, as I got at least one reply, reposting this in a new thread to catch more people. Originally sent to alsa-devel thread: "[PATCH 11/20] OMAP: McBSP: Add link DMA mode selection)" http://kerneltrap.org/mailarchive/alsa-devel/2009/7/30/6272573 --- So not strictly related to the patch that started the original thread, but this touches on, and is a good example of, one question I've been wondering for some time now as an app developer. Could Takashi, Jaroslav, Mark, or others comment on this as well, perhaps? Dropping linux-omap as this is a generic ALSA question. On Wed, 12 Aug 2009, Jarkko Nikula wrote: [i.e. pcm_pointer == hw_ptr] > The threshold based transfer will cause that omap_pcm_pointer will > loose a bit its accuracy. Probably irrelevant but still better to play > safe at least over one kernel release before making it default. [...] > element: > 614 > 669 > 691 [...] > threshold: > 512 > 512 > 1024 > 1024 In both cases, this value (hw_ptr) shows the status of sending data to an separate entity, and one that has its own buffer (multiple ms) before the actual DAC/ADC. So when application developer uses snd_pcm_delay(), the result is not quite what the ALSA API says: alsa-lib.git/src/pcm/pcm.c: --cut-- * For playback the delay is defined as the time that a frame that is written * to the PCM stream shortly after this call will take to be actually * audible. It is as such the overall latency from the write call to the final * DAC. [...] int snd_pcm_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) --cut-- Of course, there are many other cases where the same happens, as the latency of the HW (or where it is connected) is not known. But we also have many cases where we _do_ know more about the latency, and we are just missing the means to expose this info. I think this applies to this OMAP patch as well. E.g. the HW can tell fairly accurately the current status (i.e. which sample is played-out/captured _now_). Now that brings us to my question -- how to expose this info in an ALSA driver. Elsewhere in alsa-lib docs, there is more accurate description of current implementation and behaviour of snd_pcm_delay(): alsa-lib.git/src/pcm/pcm.c: --cut-- The function #snd_pcm_delay() returns the delay in samples. For playback, it means count of samples in the ring buffer before the next sample will be sent to DAC. --cut-- So e.g. the OMAP patch that this thread started from, complies with this definition, and element/threshold modes just differ in accuracy (although threshold implementation should warn the application with the SNDRV_PCM_INFO_BATCH flag). But how would a driver expose both bits of info to apps (in a standard fashion): 1) the hw_ptr for shuffling bits out of the ringbuffer, and 2) the delay to next played-out/captured sample. For application developers, (2) is the piece of info we are mostly interested in (if I write a sample now, how long it will be until it's played out). One obvious solution (that has been used already in other ALSA drivers) is to virtualize the hw_ptr, but this way you lose the accurate info about ringbuffer status. Ideally both pieces of info could be exposed. But as I see it, drivers currently can only implement "pointer" to relay this info. Jaroslav proposed some ideas earlier this year, but then the discussion faded: "Re: Driver code with mpc5200 pointer problem.", 2009-04-28 http://article.gmane.org/gmane.linux.alsa.devel/62170 Takashi made basicly the same point I've tried to make above "Re: Driver code with mpc5200 pointer problem.", 2009-04-28 http://article.gmane.org/gmane.linux.alsa.devel/62172 Mark also commented to the same thread: http://article.gmane.org/gmane.linux.alsa.devel/62176, 2009-04-28 Any update/ideas on this topic? Personally I think adding a new driver callback would make most sense, as that would allow to take full benefit from hardware that allows to query sample-accurate position of playout (i.e. not just support exposing a fixed latency). Of course that's potentially a big change. In alsa-lib, snd_pcm_hwsync() could call this driver callback, and a new variant of snd_pcm_delay() could present the information to the applications.

3 5

[alsa-devel] [PATCH] delete -spi suffix in name and free private data while registers fail
by Barry Song 14 Aug '09

14 Aug '09

Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/codecs/ad1938.c | 4 +++- 1 files changed, 3 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index 051f68c..e62b277 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -456,7 +456,7 @@ static int __devexit ad1938_spi_remove(struct spi_device *spi) static struct spi_driver ad1938_spi_driver = { .driver = { - .name = "ad1938-spi", + .name = "ad1938", .bus = &spi_bus_type, .owner = THIS_MODULE, }, @@ -543,6 +543,7 @@ static int ad1938_register(struct ad1938_priv *ad1938) ret = snd_soc_register_codec(codec); if (ret != 0) { dev_err(codec->dev, "Failed to register codec: %d\n", ret); + kfree(ad1938); return ret; } @@ -550,6 +551,7 @@ static int ad1938_register(struct ad1938_priv *ad1938) if (ret != 0) { dev_err(codec->dev, "Failed to register DAI: %d\n", ret); snd_soc_unregister_codec(codec); + kfree(ad1938); return ret; } -- 1.5.6.3

2 1

[alsa-devel] [PATCH 0/1 - v2] ASoC: TWL4030: Capture mixer string changes
by Peter Ujfalusi 14 Aug '09

14 Aug '09

Hello, Resending with the suggested changes: Before the patch the analog capture path strings ended up like this at the end: "Analog Left Capture Route Main mic Switch" These switches showed up in the playback tab of alsamixer, which was not nice. After the patch the strings will be like this: "Analog Left Main Mic Capture Switch" Which show up correctly in alsamixer as "Analog Left Main mic" under the capture tab. --- Peter Ujfalusi (1): ASoC: TWL4030: Fix for capture mixer strings sound/soc/codecs/twl4030.c | 46 +++++++++++++++++++++++-------------------- 1 files changed, 25 insertions(+), 21 deletions(-)

2 2

[alsa-devel] [PATCH] Allow passing platform_data to devices attached to AC97 bus
by Marek Vasut 14 Aug '09

14 Aug '09

Hi, I managed to put together this patch with huge help from Mark. It allows passing platform_data to devices on ac97 bus and implements this for pxa2xx- ac97. Please consider applying.

6 51

Re: [alsa-devel] [PATCHv3 14/20] OMAP: McBSP: Let element DMA mode hit retention also
by Eero Nurkkala 14 Aug '09

14 Aug '09

On Fri, 2009-08-14 at 05:19 +0200, ext Woodruff, Richard wrote: > > From: linux-omap-owner(a)vger.kernel.org [mailto:linux-omap- > > owner(a)vger.kernel.org] On Behalf Of Eduardo Valentin > > > The device no longer hits retention if element DMA > > mode is taken for at least the duration of the > > serial console timeout. Force element DMA mode to > > shut down through smartidle. > > Probably this behaves as you say but it seems possible that the bug is somewhere else in device handling. > For some reason unknown to us, if NO_IDLE mode is taken, and long enough of a arecord | aplay loop was tried (a minute or so), then, and when closing the loop, the device never hit retention anymore =) But after the fix, yes it does so. > The generic hope is you can set smart-idle + wakeups at init time and never need to touch them again. There are a few known cases where the hardware IP does have some bug which force you to dynamically play with bits. > Right. I did try to set them initially at device init (that was one of the first things I tried with McBSP), but things were not really working as expected. (Well, having iclk and fclk on and putting those at device init caused the whole HW device to jam). Another thing is, that if the McBSP is a slave (external clocking), then there will be unnecessary wakeups? If smart-idle + wakeups are always set? (and the slave clocks the McBSP WCLK and BCLK). BTW, are the wakeups active if McBSP is in reset state? > Clockactivity settings are usually static also, but depending on how you configure your device (slave or master clocking) you will need to dynamically set these. > It most you don't need all this dynamic twiddling of the idle protocol handshake bits. Doing it once at pm_init is what should be necessary. > One example which comes to mind is MMC with DMA mode. Early drivers were finding RET was gated once the driver was active. In the first pass port to 3430 the writer toggled smart/no/forced idle and the system went happily back to sleep. However, after some profiling it was seen that this method of putting the device down ended up causing the clock disable to the device to take a very long time. Another level of digging into documents was done and it was found that it was necessary to reset some command line as part of a clean shut down. Doing it this way allowed the system to idle, and the time to shut down clocks was just a few nS instead of uS. As I recall some status was not cleared until the reset of the line. It did happen that toggling through handshakes had a side effect of also clearing it, but not in the intended way. Whether a few uSec:s or nS:s, doesn't really matter in this case? (if that even was the case). > > Regards, > Richard W. -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

[alsa-devel] [PATCH 00/20] OMAP ASoC changes in DMA utilization
by Eduardo Valentin 13 Aug '09

13 Aug '09

Hellos guys, This series provides a way to allow DMA go to idle while performing playback/capture in OMAP3 devices using ASoC driver. The idea behind it is to do not use DMA element mode, but frame mode. In addition to that, it also configures McBSP threshold accordingly to alsa period. Also, McBSP wakeup signals are configured to generate wakeups for threshold events. Besides that, this series also include a way to allow choosing between element mode or the frame mode added here. Patches 1-15 are McBSP changes and, for this reason, are intended to be include in linux-omap tree. Patches 16-20 are OMAP ASoC driver changes, so, they are for alsa-devel. BR, Eduardo Valentin (11): OMAP: McBSP: Change wakeup signals OMAP: McBSP: Add IRQEN, IRQSTATUS, THRESHOLD2 and THRESHOLD1 registers. OMAP: McBSP: Use appropriate value for startup delay OMAP: McBSP: Add transmit/receive threshold handler OMAP: McBSP: Create and export max_(r|t)x_thres property OMAP: McBSP: Rename thres sysfs symbols OMAP: McBSP: Configure NO IDLE mode for DMA mode different of threshold ASoC: OMAP: Use McBSP threshold again ASoC: OMAP: Make DMA 64 aligned ASoC: OMAP: Enable DMA burst mode ASoC: OMAP: Use DMA operating mode of McBSP Eero Nurkkala (7): OMAP: McBSP: Provide functions for ASoC frame syncronization OMAP: McBSP: Wakeups utilized OMAP: McBSP: Retain McBSP FCLK clockactivity OMAP: McBSP: Do not enable wakeups for no-idle mode OMAP: McBSP: Disable all wakeups upon exit OMAP: McBSP: Let element DMA mode hit retention also ASoC: Always syncronize audio transfers on frames Peter Ujfalusi (2): OMAP3: McBSP: Lower the maximum buffersize for McBSP1,3,4,5 OMAP: McBSP: Add link DMA mode selection arch/arm/mach-omap2/mcbsp.c | 5 + arch/arm/plat-omap/include/mach/mcbsp.h | 54 +++++ arch/arm/plat-omap/mcbsp.c | 343 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 16 ++- sound/soc/omap/omap-pcm.c | 57 +++++- 5 files changed, 468 insertions(+), 7 deletions(-) -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

8 66

[alsa-devel] [PATCH v2 1/2] ASoC: DaVinci: McASP driver enhacements
by Chaithrika U S 13 Aug '09

13 Aug '09

On DA830/OMAP-L137 and DA850/OMAP-L138 SoCs, the McASP peripheral has FIFO support. This FIFO provides additional data buffering. It also provides tolerance to variation in host/DMA controller response times. The read and write FIFO sizes are 256 bytes each. If FIFO is enabled, the DMA events from McASP are sent to the FIFO which in turn sends DMA requests to the host CPU according to the thresholds programmed. More details of the FIFO operation can be found at http://focus.ti.com/general/docs/lit/getliterature.tsp?literatureNumber= sprufm1&fileType=pdf This patch adds support for FIFO configuration. The platform data has a version field which differentiates the McASP on different SoCs. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- In this version of the patch, the davinci-i2s driver changes have been done to account acnt member introduced in the dma_params structure. Also, the DAI format addition has been removed from this patch. Applies to ALSA GIT tree on branch topic/asoc at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; h=topic/asoc sound/soc/davinci/davinci-i2s.c | 1 + sound/soc/davinci/davinci-mcasp.c | 105 ++++++++++++++++++++++++++++++++++--- sound/soc/davinci/davinci-mcasp.h | 5 ++ sound/soc/davinci/davinci-pcm.c | 4 +- sound/soc/davinci/davinci-pcm.h | 1 + 5 files changed, 107 insertions(+), 9 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index 2a56fb7..12a6c54 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -406,6 +406,7 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } + dma_params->acnt = dma_params->data_type; rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(1); xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(1); diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index f0c0347..e672f43 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -102,6 +102,11 @@ /* Receive Buffer for Serializer n */ #define DAVINCI_MCASP_RXBUF_REG 0x280 +/* McASP FIFO Registers */ +#define DAVINCI_MCASP_WFIFOCTL (0x1010) +#define DAVINCI_MCASP_WFIFOSTS (0x1014) +#define DAVINCI_MCASP_RFIFOCTL (0x1018) +#define DAVINCI_MCASP_RFIFOSTS (0x101C) /* * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management @@ -276,6 +281,13 @@ */ #define TXDATADMADIS BIT(0) +/* + * DAVINCI_MCASP_W[R]FIFOCTL - Write/Read FIFO Control Register bits + */ +#define FIFO_ENABLE BIT(16) +#define NUMEVT_MASK (0xFF << 8) +#define NUMDMA_MASK (0xFF) + #define DAVINCI_MCASP_NUM_SERIALIZER 16 static inline void mcasp_set_bits(void __iomem *reg, u32 val) @@ -345,6 +357,9 @@ static void mcasp_start_rx(struct davinci_audio_dev *dev) static void mcasp_start_tx(struct davinci_audio_dev *dev) { + u8 offset = 0, i; + u32 cnt; + mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST); mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST); mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR); @@ -353,6 +368,19 @@ static void mcasp_start_tx(struct davinci_audio_dev *dev) mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXSMRST); mcasp_set_ctl_reg(dev->base + DAVINCI_MCASP_GBLCTLX_REG, TXFSRST); mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); + for (i = 0; i < dev->num_serializer; i++) { + if (dev->serial_dir[i] == TX_MODE) { + offset = i; + break; + } + } + + /* wait for TX ready */ + cnt = 0; + while (!(mcasp_get_reg(dev->base + DAVINCI_MCASP_XRSRCTL_REG(offset)) & + TXSTATE) && (cnt < 100000)) + cnt++; + mcasp_set_reg(dev->base + DAVINCI_MCASP_TXBUF_REG, 0); } @@ -362,6 +390,13 @@ static void davinci_mcasp_start(struct davinci_audio_dev *dev, int stream) mcasp_start_tx(dev); else mcasp_start_rx(dev); + + /* enable FIFO */ + if (dev->txnumevt) + mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + + if (dev->rxnumevt) + mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); } static void mcasp_stop_rx(struct davinci_audio_dev *dev) @@ -382,6 +417,13 @@ static void davinci_mcasp_stop(struct davinci_audio_dev *dev, int stream) mcasp_stop_tx(dev); else mcasp_stop_rx(dev); + + /* disable FIFO */ + if (dev->txnumevt) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + + if (dev->rxnumevt) + mcasp_clr_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); } static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, @@ -401,7 +443,6 @@ static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai, mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x7 << 26)); break; - case SND_SOC_DAIFMT_CBM_CFM: /* codec is clock and frame master */ mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); @@ -505,6 +546,8 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) { int i; + u8 tx_ser = 0; + u8 rx_ser = 0; /* Default configuration */ mcasp_set_bits(dev->base + DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT); @@ -525,12 +568,37 @@ static void davinci_hw_common_param(struct davinci_audio_dev *dev, int stream) for (i = 0; i < dev->num_serializer; i++) { mcasp_set_bits(dev->base + DAVINCI_MCASP_XRSRCTL_REG(i), dev->serial_dir[i]); - if (dev->serial_dir[i] == TX_MODE) + if (dev->serial_dir[i] == TX_MODE) { mcasp_set_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); - else if (dev->serial_dir[i] == RX_MODE) + tx_ser++; + } else if (dev->serial_dir[i] == RX_MODE) { mcasp_clr_bits(dev->base + DAVINCI_MCASP_PDIR_REG, AXR(i)); + rx_ser++; + } + } + + if (dev->txnumevt && stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (dev->txnumevt * tx_ser > 64) + dev->txnumevt = 1; + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, tx_ser, + NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, + ((dev->txnumevt * tx_ser) << 8), NUMEVT_MASK); + mcasp_set_bits(dev->base + DAVINCI_MCASP_WFIFOCTL, FIFO_ENABLE); + } + + if (dev->rxnumevt && stream == SNDRV_PCM_STREAM_CAPTURE) { + if (dev->rxnumevt * rx_ser > 64) + dev->rxnumevt = 1; + + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, rx_ser, + NUMDMA_MASK); + mcasp_mod_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, + ((dev->rxnumevt * rx_ser) << 8), NUMEVT_MASK); + mcasp_set_bits(dev->base + DAVINCI_MCASP_RFIFOCTL, FIFO_ENABLE); } } @@ -543,6 +611,8 @@ static void davinci_hw_param(struct davinci_audio_dev *dev, int stream) for (i = 0; i < active_slots; i++) mask |= (1 << i); + mcasp_clr_bits(dev->base + DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC); + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { /* bit stream is MSB first with no delay */ /* DSP_B mode */ @@ -622,8 +692,16 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, struct davinci_pcm_dma_params *dma_params = dev->dma_params[substream->stream]; int word_length; + u8 numevt; davinci_hw_common_param(dev, substream->stream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + numevt = dev->txnumevt; + else + numevt = dev->rxnumevt; + + if (!numevt) + numevt = 1; if (dev->op_mode == DAVINCI_MCASP_DIT_MODE) davinci_hw_dit_param(dev); @@ -650,6 +728,13 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, printk(KERN_WARNING "davinci-mcasp: unsupported PCM format"); return -EINVAL; } + + if (dev->version == MCASP_VERSION_2) { + dma_params->data_type *= numevt; + dma_params->acnt = 4 * numevt; + } else + dma_params->acnt = dma_params->data_type; + davinci_config_channel_size(dev, word_length); return 0; @@ -778,6 +863,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) dev->num_serializer = pdata->num_serializer; dev->serial_dir = pdata->serial_dir; dev->codec_fmt = pdata->codec_fmt; + dev->version = pdata->version; + dev->txnumevt = pdata->txnumevt; + dev->rxnumevt = pdata->rxnumevt; dma_data[count].name = "I2S PCM Stereo out"; dma_data[count].eventq_no = pdata->eventq_no; @@ -807,9 +895,9 @@ static int davinci_mcasp_probe(struct platform_device *pdev) } dma_data[count].channel = res->start; - davinci_mcasp_dai[pdev->id].private_data = dev; - davinci_mcasp_dai[pdev->id].dev = &pdev->dev; - ret = snd_soc_register_dai(&davinci_mcasp_dai[pdev->id]); + davinci_mcasp_dai[pdata->op_mode].private_data = dev; + davinci_mcasp_dai[pdata->op_mode].dev = &pdev->dev; + ret = snd_soc_register_dai(&davinci_mcasp_dai[pdata->op_mode]); if (ret != 0) goto err_release_region; @@ -827,12 +915,13 @@ err_release_dev: static int davinci_mcasp_remove(struct platform_device *pdev) { + struct snd_platform_data *pdata = pdev->dev.platform_data; struct davinci_pcm_dma_params *dma_data; struct davinci_audio_dev *dev; struct resource *mem; - snd_soc_unregister_dai(&davinci_mcasp_dai[pdev->id]); - dev = davinci_mcasp_dai[pdev->id].private_data; + snd_soc_unregister_dai(&davinci_mcasp_dai[pdata->op_mode]); + dev = davinci_mcasp_dai[pdata->op_mode].private_data; clk_disable(dev->clk); clk_put(dev->clk); dev->clk = NULL; diff --git a/sound/soc/davinci/davinci-mcasp.h b/sound/soc/davinci/davinci-mcasp.h index 36b7104..554354c 100644 --- a/sound/soc/davinci/davinci-mcasp.h +++ b/sound/soc/davinci/davinci-mcasp.h @@ -50,6 +50,11 @@ struct davinci_audio_dev { u8 op_mode; u8 num_serializer; u8 *serial_dir; + u8 version; + + /* McASP FIFO related */ + u8 txnumevt; + u8 rxnumevt; }; #endif /* DAVINCI_MCASP_H */ diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index ab43a53..091dacb 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -67,6 +67,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dma_addr_t src, dst; unsigned short src_bidx, dst_bidx; unsigned int data_type; + unsigned short acnt; unsigned int count; period_size = snd_pcm_lib_period_bytes(substream); @@ -91,11 +92,12 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dst_bidx = data_type; } + acnt = prtd->params->acnt; edma_set_src(lch, src, INCR, W8BIT); edma_set_dest(lch, dst, INCR, W8BIT); edma_set_src_index(lch, src_bidx, 0); edma_set_dest_index(lch, dst_bidx, 0); - edma_set_transfer_params(lch, data_type, count, 1, 0, ASYNC); + edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC); prtd->period++; if (unlikely(prtd->period >= runtime->periods)) diff --git a/sound/soc/davinci/davinci-pcm.h b/sound/soc/davinci/davinci-pcm.h index eb4287f..63d9625 100644 --- a/sound/soc/davinci/davinci-pcm.h +++ b/sound/soc/davinci/davinci-pcm.h @@ -19,6 +19,7 @@ struct davinci_pcm_dma_params { char *name; /* stream identifier */ int channel; /* sync dma channel ID */ + unsigned short acnt; dma_addr_t dma_addr; /* device physical address for DMA */ enum dma_event_q eventq_no; /* event queue number */ unsigned char data_type; /* xfer data type */ -- 1.5.6

2 1

[alsa-devel] [RFC] Break up the patch_sigmatel.c
by Maxim Levitsky 13 Aug '09

13 Aug '09

Hi folks, I own an stac9227 and with latest kernel, it doesn't output any sound via front headphones. (this a is desktop) I don't want to bother you with details, etc, I will fix that myself. I have good knowledge of the device. However, the mess that is patch_sigmatel.c seems to be doubled. its huge file, and there are loads on new codecs added. I have a suggestion, to break it up into smaller files. I especially want to move all 'data' from it to header file, and sort it by model. Maybe move pin configs to separate file. Also I know that stac9200 is quite old, and uses many different support functions. I could move these in another file as well. In other words, I want to break it up into several (5 maybe) files, while not touching the code itself (I so that later maybe) Since it is quite dirty and hard work, I want to ask if you agree. Btw, half of static data can be retrieved from codec by querying it (like dac,adc,pin node ids) Best regards, Maxim Levitsky

3 6

[alsa-devel] [PATCH V2 1/3] new ad1836 codec driver based on asoc
by Barry Song 13 Aug '09

13 Aug '09

There has been an ad1836 driver in sound/blackfin based on traditional alsa. The new driver is based on asoc. The architecture of ad1836 codec driver is very much like ad1938. Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/codecs/Kconfig | 3 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ad1836.c | 451 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ad1836.h | 64 +++++++ 4 files changed, 520 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ad1836.c create mode 100644 sound/soc/codecs/ad1836.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 31a6d21..05e2279 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -68,6 +68,9 @@ config SND_SOC_AC97_CODEC tristate select SND_AC97_CODEC +config SND_SOC_AD1836 + tristate + config SND_SOC_AD1938 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 78dce5d..9cbcf8f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,4 +1,5 @@ snd-soc-ac97-objs := ac97.o +snd-soc-ad1836-objs := ad1836.o snd-soc-ad1938-objs := ad1938.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o @@ -45,6 +46,7 @@ snd-soc-wm9713-objs := wm9713.o snd-soc-max9877-objs := max9877.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o +obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD1938) += snd-soc-ad1938.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c new file mode 100644 index 0000000..79b9d19 --- /dev/null +++ b/sound/soc/codecs/ad1836.c @@ -0,0 +1,451 @@ +/* + * File: sound/soc/codecs/ad1836.c + * Author: Barry Song <Barry.Song(a)analog.com> + * + * Created: Aug 04 2009 + * Description: Driver for AD1836 sound chip + * + * Modified: + * Copyright 2009 Analog Devices Inc. + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * 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/init.h> +#include <linux/module.h> +#include <linux/version.h> +#include <linux/kernel.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/soc-dapm.h> +#include <linux/spi/spi.h> +#include "ad1836.h" + +/* codec private data */ +struct ad1836_priv { + struct snd_soc_codec codec; + u16 reg_cache[AD1836_NUM_REGS]; +}; + +static struct snd_soc_codec *ad1836_codec; +struct snd_soc_codec_device soc_codec_dev_ad1836; +static int ad1836_register(struct ad1836_priv *ad1836); +static void ad1836_unregister(struct ad1836_priv *ad1836); + +/* + * AD1836 volume/mute/de-emphasis etc. controls + */ +static const char *ad1836_deemp[] = {"None", "44.1kHz", "32kHz", "48kHz"}; + +static const struct soc_enum ad1836_deemp_enum = + SOC_ENUM_SINGLE(AD1836_DAC_CTRL1, 8, 4, ad1836_deemp); + +static const struct snd_kcontrol_new ad1836_snd_controls[] = { + /* DAC volume control */ + SOC_DOUBLE_R("DAC1 Volume", AD1836_DAC_L1_VOL, + AD1836_DAC_R1_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC2 Volume", AD1836_DAC_L2_VOL, + AD1836_DAC_R2_VOL, 0, 0x3FF, 0), + SOC_DOUBLE_R("DAC3 Volume", AD1836_DAC_L3_VOL, + AD1836_DAC_R3_VOL, 0, 0x3FF, 0), + + /* ADC switch control */ + SOC_DOUBLE("ADC1 Switch", AD1836_ADC_CTRL2, AD1836_ADCL1_MUTE, + AD1836_ADCR1_MUTE, 1, 1), + SOC_DOUBLE("ADC2 Switch", AD1836_ADC_CTRL2, AD1836_ADCL2_MUTE, + AD1836_ADCR2_MUTE, 1, 1), + + /* DAC switch control */ + SOC_DOUBLE("DAC1 Switch", AD1836_DAC_CTRL2, AD1836_DACL1_MUTE, + AD1836_DACR1_MUTE, 1, 1), + SOC_DOUBLE("DAC2 Switch", AD1836_DAC_CTRL2, AD1836_DACL2_MUTE, + AD1836_DACR2_MUTE, 1, 1), + SOC_DOUBLE("DAC3 Switch", AD1836_DAC_CTRL2, AD1836_DACL3_MUTE, + AD1836_DACR3_MUTE, 1, 1), + + /* ADC high-pass filter */ + SOC_SINGLE("ADC High Pass Filter Switch", AD1836_ADC_CTRL1, + AD1836_ADC_HIGHPASS_FILTER, 1, 0), + + /* DAC de-emphasis */ + SOC_ENUM("Playback Deemphasis", ad1836_deemp_enum), +}; + +static const struct snd_soc_dapm_widget ad1836_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", "Playback", AD1836_DAC_CTRL1, + AD1836_DAC_POWERDOWN, 1), + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1836_ADC_CTRL1, + AD1836_ADC_POWERDOWN, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("DAC1OUT"), + SND_SOC_DAPM_OUTPUT("DAC2OUT"), + SND_SOC_DAPM_OUTPUT("DAC3OUT"), + SND_SOC_DAPM_INPUT("ADC1IN"), + SND_SOC_DAPM_INPUT("ADC2IN"), +}; + +static const struct snd_soc_dapm_route audio_paths[] = { + { "DAC", NULL, "ADC_PWR" }, + { "ADC", NULL, "ADC_PWR" }, + { "DAC1OUT", "DAC1 Switch", "DAC" }, + { "DAC2OUT", "DAC2 Switch", "DAC" }, + { "DAC3OUT", "DAC3 Switch", "DAC" }, + { "ADC", "ADC1 Switch", "ADC1IN" }, + { "ADC", "ADC2 Switch", "ADC2IN" }, +}; + +/* + * DAI ops entries + */ + +static int ad1836_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + /* at present, we support adc aux mode to interface with + * blackfin sport tdm mode + */ + case SND_SOC_DAIFMT_DSP_A: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + /* ALCLK,ABCLK are both output, AD1836 can only be master */ + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ad1836_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + int word_len = 0; + + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + word_len = 3; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + word_len = 1; + break; + case SNDRV_PCM_FORMAT_S24_LE: + case SNDRV_PCM_FORMAT_S32_LE: + word_len = 0; + break; + } + + snd_soc_update_bits(codec, AD1836_DAC_CTRL1, + AD1836_DAC_WORD_LEN_MASK, word_len); + + snd_soc_update_bits(codec, AD1836_ADC_CTRL2, + AD1836_ADC_WORD_LEN_MASK, word_len); + + return 0; +} + + +/* + * interface to read/write ad1836 register + */ +#define AD1836_SPI_REG_SHFT 12 +#define AD1836_SPI_READ (1 << 11) +#define AD1836_SPI_VAL_MSK 0x3FF + +/* + * write to the ad1836 register space + */ + +static int ad1836_write_reg(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u16 *reg_cache = codec->reg_cache; + int ret = 0; + + if (value != reg_cache[reg]) { + unsigned short buf; + struct spi_transfer t = { + .tx_buf = &buf, + .len = 2, + }; + struct spi_message m; + + buf = (reg << AD1836_SPI_REG_SHFT) | + (value & AD1836_SPI_VAL_MSK); + spi_message_init(&m); + spi_message_add_tail(&t, &m); + ret = spi_sync(codec->control_data, &m); + if (ret == 0) + reg_cache[reg] = value; + } + + return ret; +} + +/* + * read from the ad1836 register space cache + */ +static unsigned int ad1836_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *reg_cache = codec->reg_cache; + + if (reg >= codec->reg_cache_size) + return -EINVAL; + + return reg_cache[reg]; +} + +static int __devinit ad1836_spi_probe(struct spi_device *spi) +{ + struct snd_soc_codec *codec; + struct ad1836_priv *ad1836; + + ad1836 = kzalloc(sizeof(struct ad1836_priv), GFP_KERNEL); + if (ad1836 == NULL) + return -ENOMEM; + + codec = &ad1836->codec; + codec->control_data = spi; + codec->dev = &spi->dev; + + dev_set_drvdata(&spi->dev, ad1836); + + return ad1836_register(ad1836); +} + +static int __devexit ad1836_spi_remove(struct spi_device *spi) +{ + struct ad1836_priv *ad1836 = dev_get_drvdata(&spi->dev); + + ad1836_unregister(ad1836); + return 0; +} + +static struct spi_driver ad1836_spi_driver = { + .driver = { + .name = "ad1836-spi", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = ad1836_spi_probe, + .remove = __devexit_p(ad1836_spi_remove), +}; + +static struct snd_soc_dai_ops ad1836_dai_ops = { + .hw_params = ad1836_hw_params, + .set_fmt = ad1836_set_dai_fmt, +}; + +/* codec DAI instance */ +struct snd_soc_dai ad1836_dai = { + .name = "AD1836", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 6, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE, + }, + .ops = &ad1836_dai_ops, +}; +EXPORT_SYMBOL_GPL(ad1836_dai); + +static int ad1836_register(struct ad1836_priv *ad1836) +{ + int ret; + struct snd_soc_codec *codec = &ad1836->codec; + + if (ad1836_codec) { + dev_err(codec->dev, "Another ad1836 is registered\n"); + return -EINVAL; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + codec->private_data = ad1836; + codec->reg_cache = ad1836->reg_cache; + codec->reg_cache_size = AD1836_NUM_REGS; + codec->name = "AD1836"; + codec->owner = THIS_MODULE; + codec->dai = &ad1836_dai; + codec->num_dai = 1; + codec->write = ad1836_write_reg; + codec->read = ad1836_read_reg_cache; + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + ad1836_dai.dev = codec->dev; + ad1836_codec = codec; + + /* default setting for ad1836 */ + /* de-emphasis: 48kHz, power-on dac */ + codec->write(codec, AD1836_DAC_CTRL1, 0x300); + /* unmute dac channels */ + codec->write(codec, AD1836_DAC_CTRL2, 0x0); + /* high-pass filter enable, power-on adc */ + codec->write(codec, AD1836_ADC_CTRL1, 0x100); + /* unmute adc channles, adc aux mode */ + codec->write(codec, AD1836_ADC_CTRL2, 0x180); + /* left/right diff:PGA/MUX */ + codec->write(codec, AD1836_ADC_CTRL3, 0x3A); + /* volume */ + codec->write(codec, AD1836_DAC_L1_VOL, 0x3FF); + codec->write(codec, AD1836_DAC_R1_VOL, 0x3FF); + codec->write(codec, AD1836_DAC_L2_VOL, 0x3FF); + codec->write(codec, AD1836_DAC_R2_VOL, 0x3FF); + codec->write(codec, AD1836_DAC_L3_VOL, 0x3FF); + codec->write(codec, AD1836_DAC_R3_VOL, 0x3FF); + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + kfree(ad1836); + return ret; + } + + ret = snd_soc_register_dai(&ad1836_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + snd_soc_unregister_codec(codec); + kfree(ad1836); + return ret; + } + + return 0; +} + +static void ad1836_unregister(struct ad1836_priv *ad1836) +{ + snd_soc_unregister_dai(&ad1836_dai); + snd_soc_unregister_codec(&ad1836->codec); + kfree(ad1836); + ad1836_codec = NULL; +} + +static int ad1836_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (ad1836_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ad1836_codec; + codec = ad1836_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, ad1836_snd_controls, + ARRAY_SIZE(ad1836_snd_controls)); + snd_soc_dapm_new_controls(codec, ad1836_dapm_widgets, + ARRAY_SIZE(ad1836_dapm_widgets)); + snd_soc_dapm_add_routes(codec, audio_paths, ARRAY_SIZE(audio_paths)); + snd_soc_dapm_new_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +/* power down chip */ +static int ad1836_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ad1836 = { + .probe = ad1836_probe, + .remove = ad1836_remove, + /* The power management of ad1836 is very simple. There are + * only adc&dac 2 components to control. Dapm handles them. + */ + .suspend = NULL, + .resume = NULL, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ad1836); + +static int __init ad1836_init(void) +{ + int ret; + + ret = spi_register_driver(&ad1836_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register ad1836 SPI driver: %d\n", + ret); + } + + return ret; +} +module_init(ad1836_init); + +static void __exit ad1836_exit(void) +{ + spi_unregister_driver(&ad1836_spi_driver); +} +module_exit(ad1836_exit); + +MODULE_DESCRIPTION("ASoC ad1836 driver"); +MODULE_AUTHOR("Barry Song <21cnbao(a)gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad1836.h b/sound/soc/codecs/ad1836.h new file mode 100644 index 0000000..7660ee6 --- /dev/null +++ b/sound/soc/codecs/ad1836.h @@ -0,0 +1,64 @@ +/* + * File: sound/soc/codecs/ad1836.h + * Based on: + * Author: Barry Song <Barry.Song(a)analog.com> + * + * Created: Aug 04, 2009 + * Description: definitions for AD1836 registers + * + * Modified: + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * 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 __AD1836_H__ +#define __AD1836_H__ + +#define AD1836_DAC_CTRL1 0 +#define AD1836_DAC_POWERDOWN 2 +#define AD1836_DAC_SERFMT_MASK 0xE0 +#define AD1836_DAC_SERFMT_PCK256 (0x4 << 5) +#define AD1836_DAC_SERFMT_PCK128 (0x5 << 5) +#define AD1836_DAC_WORD_LEN_MASK 0x18 + +#define AD1836_DAC_CTRL2 1 +#define AD1836_DACL1_MUTE 0 +#define AD1836_DACR1_MUTE 1 +#define AD1836_DACL2_MUTE 2 +#define AD1836_DACR2_MUTE 3 +#define AD1836_DACL3_MUTE 4 +#define AD1836_DACR3_MUTE 5 + +#define AD1836_DAC_L1_VOL 2 +#define AD1836_DAC_R1_VOL 3 +#define AD1836_DAC_L2_VOL 4 +#define AD1836_DAC_R2_VOL 5 +#define AD1836_DAC_L3_VOL 6 +#define AD1836_DAC_R3_VOL 7 + +#define AD1836_ADC_CTRL1 12 +#define AD1836_ADC_POWERDOWN 7 +#define AD1836_ADC_HIGHPASS_FILTER 8 + +#define AD1836_ADC_CTRL2 13 +#define AD1836_ADCL1_MUTE 0 +#define AD1836_ADCR1_MUTE 1 +#define AD1836_ADCL2_MUTE 2 +#define AD1836_ADCR2_MUTE 3 +#define AD1836_ADC_WORD_LEN_MASK 0x30 +#define AD1836_ADC_SERFMT_MASK (7 << 6) +#define AD1836_ADC_SERFMT_PCK256 (0x4 << 6) +#define AD1836_ADC_SERFMT_PCK128 (0x5 << 6) + +#define AD1836_ADC_CTRL3 14 + +#define AD1836_NUM_REGS 16 + +extern struct snd_soc_dai ad1836_dai; +extern struct snd_soc_codec_device soc_codec_dev_ad1836; +#endif -- 1.5.6.3

2 1

[alsa-devel] [PATCH 0/2] ASoC: TWL4030: Cosmetic and PM related changes
by Peter Ujfalusi 13 Aug '09

13 Aug '09

Hello, I'm not sure, if the first patch is acceptable by ASoC standards, but: Before the patch the analog capture path strings ended up like this at the end: "Analog Left Capture Route Main mic Switch" These switches showed up in the playback tab of alsamixer, which was not nice. After the patch the strings will be like this: "Analog Left Main mic Capture Route" Which show up correctly in alsamixer as "Analog Left Main mic" under the capture tab. The second patch cleans up the output amplifier mute/un-mute by adding PGA_E to all output pins, which were handled by the twl4030_codec_mute function. Since those gain controls attached to the outputs are not in DAPM domain (yet?) and they have to be muted/turned off when they are not needed. Previously all of these (Earpiece PreDrivL/R, CarkitL/R) muted/un-muted at the same time without checking if the output actually is needed, there could be cases when the output amplifier is enabled but no actual audio data is routed there, which could leak some noise on the output. I know about the changes in core that makes this hassle with the mute unnecesary (the core sets the BIAS_ON also when any bypass path is active), I'll be looking at that next. --- Peter Ujfalusi (2): ASoC: TWL4030: Fix for capture mixer strings ASoC: TWL4030: Introduce PGAs for outputs sound/soc/codecs/twl4030.c | 155 ++++++++++++++++++++++++-------------------- 1 files changed, 85 insertions(+), 70 deletions(-)

2 4

Re: [alsa-devel] Audio support for TLV320AIC23 on Simtec S3C24XX products
by Mark Brown 13 Aug '09

13 Aug '09

On Thu, Aug 13, 2009 at 02:24:30PM +0100, Ben Dooks wrote: > Mark Brown wrote: > >Also, as I've mentioned to you on a number of previous occasions please > >do try to remember to CC subsystem maintainers when submitting patches. > I thought you where the ASoC maintainer? Who else should I be CC'ing? You'd only sent the patches to the list, there wasn't a CC to me. > >So I'm guessing that this is muting through powering off? In that case > >(and probably in general) this would be better implemented as a DAPM > >widget with a SND_SOC_DAPM_PIN_SWITCH() control set up to provide the > >mute functionality. Doing this will allow the speaker to be powered off > >when no audio is being played, saving power and helping avoid any pops > >or clicks being amplified by it. > I tried it that way first, unfortunately the AMP itself makes a POP > when powering up and down, the headphone output of the TLV320AIC23 > doesn't seem to have a problem with pops. But it doesn't pop when people use the mute control? OK, anyway.

2 1

Re: [alsa-devel] tlv320aic3x: Allow driver to use already registered i2c device
by Mark Brown 13 Aug '09

13 Aug '09

On Thu, Aug 13, 2009 at 02:26:38PM +0100, Ben Dooks wrote: > Mark Brown wrote: > >Why would you want to do this? I suspect that perhaps you want to > >convert the driver to use the standard device model instead but without > >understanding your motivation for making this change it is very hard to > >comment in detail. > The device is registered by the code that sets the initial devices > up that have been passed from the bootloader, thus the tlv320aic33 > fails to initialise as there is already an i2c device on the i2c bus > when it comes along. OK, that's what I thought. You need to convert the driver to use the normal device model probe methods rather than using bodges like this. See drivers like wm8731 for an example of how the conversion looks.

1 0

Re: [alsa-devel] Intel card master and PCM channels doesn't work
by Takashi Iwai 13 Aug '09

13 Aug '09

At Thu, 13 Aug 2009 12:41:08 +0300, Mantas Pilipavičius wrote: > > I waited a bit longer and it worked with this command: > speaker-test -c2 > > But again, when I try to play my mp3 files with mp3blaster it doesn't affect > the sound when I use alsamixer. But with speaker-test it does work. It depends how app accesses the sound device. If the application uses "default" PCM, not "hw" or so, then it should work without changing. Or, if app uses OSS emulation over /dev/dsp, softvol doesn't work. Takashi

1 0

[alsa-devel] [PATCH 0/3] patch sequence adds analog devices ad1836 asoc support
by Barry Song 13 Aug '09

13 Aug '09

Hi Mark, The patch sequence adds ad1836 support based on asoc patch 1: new ad1836 codec driver based on asoc patch 2: add set_tdm_slot entry in tdm-dai to define the relationship between audio channel and tdm slot number patch 3: new board driver to connect ad1836 codec and bf5xx patch 3 depends on patch 2. Thanks Barry

3 16

Re: [alsa-devel] Intel card master and PCM channels doesn't work
by Takashi Iwai 13 Aug '09

13 Aug '09

[Please don't drop Cc to alsa-devel ML] At Thu, 13 Aug 2009 11:27:29 +0300, Mantas Pilipavičius wrote: > > So I can play a sound with this command: > speaker-test -Dhw:0,0 -c2 -twav > > I edit my asound.rc: > nano -w ~/.asoundrc and nano -w /etc/asound.conf > > With this: > > pcm.softvol { > type softvol > slave { > pcm "hw:0,0" > } > control { > name "Garsas" > card 0 > } > } > > And the I try: > > speaker-test -Dsoftwol -c2 -twav > > And it doesn't work. Elaborate what doesn't work how. > Sorry to disturb you and consume your precious time, but > I am new to open source and really need this. What about if you run without -Dxxx option? Usually PCM volume should be created as a softvol if not exists. Takashi > On Thu, Aug 13, 2009 at 11:14 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Thu, 13 Aug 2009 10:55:38 +0300, > Mantas Pilipavičius wrote: > > > > Can you explain more detailed? What is data manipulation? > > You need to change all the stream data multiplying the volume > gain/attenuation in software because you hardware gives no volume > control by itself. > > In the case of alsa-lib, it's softvol plugin. Or, if you use PA, > PA does it automatically. > > Takashi > > > On Thu, Aug 13, 2009 at 8:41 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > At Thu, 13 Aug 2009 00:39:48 +0300, > > Mantas Pilipavičius wrote: > > > > > > http://www.alsa-project.org/db/?f= > > afa01dd48ed59d895d18407af62dbd1fa138686c > > > Everything works great, I can play the sound, but PCM and Master > > channels > > > don't work, so I can't increase or decrease volume. Only thing I > can do > > is > > > mute/unmute PCM channel > > > > This hardware has no volume control indeed. > > The volume has to be adjusted in software by data manipulation. > > > > Takashi > > > > > >

1 0

[alsa-devel] ALSA and MP3
by gammay 13 Aug '09

13 Aug '09

Hello, I am looking at making this product use case possible: Audio samples are streamed over wire/wireless from a source device to a sink device. The streaming format is in mp3. On the source device I need to capture locally played mp3 samples and stream to the sink device. The sink device takes care of uncompressing and playback. Can we capture compressed mp3 streams directly using ALSA? It is possible to capture uncompressed PCM samples using file plug or by other means in ALSA. While it is possible to directly read from the mp3 file, I would rather grap the samples that a media player sends out because a) it takes care of timing b) I can control the media player to play/pause the stream and for playlist management. For the user it is transperent - he/she will use only the local media player to control the streaming. I am not sure if this lies in the domain of ALSA at all - as I understand once we are in ALSA we are dealing with only PCM. In such case, please give me some pointers where I can look at to make this possible. I am using mplayer, but can also look at other types (non-GUI, command-line based). Can we do this in a non-media player specific way? Thank you, Girish

2 1

Re: [alsa-devel] Intel card master and PCM channels doesn't work
by Takashi Iwai 13 Aug '09

13 Aug '09

At Thu, 13 Aug 2009 10:55:38 +0300, Mantas Pilipavičius wrote: > > Can you explain more detailed? What is data manipulation? You need to change all the stream data multiplying the volume gain/attenuation in software because you hardware gives no volume control by itself. In the case of alsa-lib, it's softvol plugin. Or, if you use PA, PA does it automatically. Takashi > On Thu, Aug 13, 2009 at 8:41 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Thu, 13 Aug 2009 00:39:48 +0300, > Mantas Pilipavičius wrote: > > > > http://www.alsa-project.org/db/?f= > afa01dd48ed59d895d18407af62dbd1fa138686c > > Everything works great, I can play the sound, but PCM and Master > channels > > don't work, so I can't increase or decrease volume. Only thing I can do > is > > mute/unmute PCM channel > > This hardware has no volume control indeed. > The volume has to be adjusted in software by data manipulation. > > Takashi > >

1 0

[alsa-devel] [PATCH] new board driver to connect bfin-5xx with ad1836 codec
by Barry Song 13 Aug '09

13 Aug '09

Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/blackfin/Kconfig | 8 ++ sound/soc/blackfin/Makefile | 2 + sound/soc/blackfin/bf5xx-ad1836.c | 139 +++++++++++++++++++++++++++++++++++++ 3 files changed, 149 insertions(+), 0 deletions(-) create mode 100644 sound/soc/blackfin/bf5xx-ad1836.c diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 8a4de4d..ac927ff 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -88,6 +88,14 @@ config SND_BF5XX_SOC_AC97 select SND_SOC_AC97_BUS select SND_BF5XX_SOC_SPORT +config SND_BF5XX_SOC_AD1836 + tristate "SoC AD1836 Audio support for BF5xx" + depends on SND_BF5XX_TDM + select SND_BF5XX_SOC_TDM + select SND_SOC_AD1836 + help + Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. + config SND_BF5XX_SOC_AD1980 tristate "SoC AD1980/1 Audio support for BF5xx" depends on SND_BF5XX_AC97 diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index f4d7607..87e3042 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -16,11 +16,13 @@ obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o # Blackfin Machine Support +snd-ad1836-objs := bf5xx-ad1836.o snd-ad1980-objs := bf5xx-ad1980.o snd-ssm2602-objs := bf5xx-ssm2602.o snd-ad73311-objs := bf5xx-ad73311.o snd-ad1938-objs := bf5xx-ad1938.o +obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c new file mode 100644 index 0000000..c0ef965 --- /dev/null +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -0,0 +1,139 @@ +/* + * File: sound/soc/blackfin/bf5xx-ad1836.c + * Author: Barry Song <Barry.Song(a)analog.com> + * + * Created: Aug 4 2009 + * Description: Board driver for ad1836 sound chip + * + * Bugs: Enter bugs at http://blackfin.uclinux.org/ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/pcm_params.h> + +#include <asm/blackfin.h> +#include <asm/cacheflush.h> +#include <asm/irq.h> +#include <asm/dma.h> +#include <asm/portmux.h> + +#include "../codecs/ad1836.h" +#include "bf5xx-sport.h" + +#include "bf5xx-tdm-pcm.h" +#include "bf5xx-tdm.h" + +static struct snd_soc_card bf5xx_ad1836; + +static int bf5xx_ad1836_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + cpu_dai->private_data = sport_handle; + return 0; +} + +static int bf5xx_ad1836_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 *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + int ret = 0; + /* set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_IB_IF | SND_SOC_DAIFMT_CBM_CFM); + if (ret < 0) + return ret; + + /* set cpu DAI slots, 8 channels, mask is defined as slot seq + * 0x73625140 means audio channel 0 uses slot 0 + * channel 1 uses slot 4 + * channel 2 uses slot 1 + * channel 3 uses slot 5 + * ... + */ + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x73625140, 8, 32); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops bf5xx_ad1836_ops = { + .startup = bf5xx_ad1836_startup, + .hw_params = bf5xx_ad1836_hw_params, +}; + +static struct snd_soc_dai_link bf5xx_ad1836_dai = { + .name = "ad1836", + .stream_name = "AD1836", + .cpu_dai = &bf5xx_tdm_dai, + .codec_dai = &ad1836_dai, + .ops = &bf5xx_ad1836_ops, +}; + +static struct snd_soc_card bf5xx_ad1836 = { + .name = "bf5xx_ad1836", + .platform = &bf5xx_tdm_soc_platform, + .dai_link = &bf5xx_ad1836_dai, + .num_links = 1, +}; + +static struct snd_soc_device bf5xx_ad1836_snd_devdata = { + .card = &bf5xx_ad1836, + .codec_dev = &soc_codec_dev_ad1836, +}; + +static struct platform_device *bfxx_ad1836_snd_device; + +static int __init bf5xx_ad1836_init(void) +{ + int ret; + + bfxx_ad1836_snd_device = platform_device_alloc("soc-audio", -1); + if (!bfxx_ad1836_snd_device) + return -ENOMEM; + + platform_set_drvdata(bfxx_ad1836_snd_device, &bf5xx_ad1836_snd_devdata); + bf5xx_ad1836_snd_devdata.dev = &bfxx_ad1836_snd_device->dev; + ret = platform_device_add(bfxx_ad1836_snd_device); + + if (ret) + platform_device_put(bfxx_ad1836_snd_device); + + return ret; +} + +static void __exit bf5xx_ad1836_exit(void) +{ + platform_device_unregister(bfxx_ad1836_snd_device); +} + +module_init(bf5xx_ad1836_init); +module_exit(bf5xx_ad1836_exit); + +/* Module information */ +MODULE_AUTHOR("Barry Song"); +MODULE_DESCRIPTION("ALSA SoC AD1836 board driver"); +MODULE_LICENSE("GPL"); + -- 1.5.6.3

1 0

[alsa-devel] ASoC update for 2.6.32
by Mark Brown 13 Aug '09

13 Aug '09

The following changes since commit e0026beac0f1a6253488478b3f022ff243b012b9: Mark Brown (1): ASoC: Update WM9081 for tdm_slot() API change are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Mark Brown (1): ASoC: Update AD1938 for new TDM slot API sound/soc/codecs/ad1938.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-)

2 1

[alsa-devel] Intel card master and PCM channels doesn't work
by Mantas Pilipavičius 13 Aug '09

13 Aug '09

http://www.alsa-project.org/db/?f=afa01dd48ed59d895d18407af62dbd1fa138686c Everything works great, I can play the sound, but PCM and Master channels don't work, so I can't increase or decrease volume. Only thing I can do is mute/unmute PCM channel

2 1

[alsa-devel] [PATCH v2] ARM: DaVinci: Audio support for DA830 EVM
by Chaithrika U S 12 Aug '09

12 Aug '09

Define resources for McASP1 used on DA830/OMAP-L137 EVM, add platform device defintion, initialization function. Additionally, this patch also adds version and FIFO related members to platform data structure. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to DaVinci GIT tree. In this version of the patch, the init fucntion, resource and platform device definitions have neen moved to the devices-da8xx.c file. arch/arm/mach-davinci/board-da830-evm.c | 27 +++++++++++++++++++++ arch/arm/mach-davinci/da830.c | 7 +++-- arch/arm/mach-davinci/devices-da8xx.c | 36 ++++++++++++++++++++++++++++ arch/arm/mach-davinci/include/mach/asp.h | 15 +++++++++++ arch/arm/mach-davinci/include/mach/da8xx.h | 3 ++ 5 files changed, 85 insertions(+), 3 deletions(-) diff --git a/arch/arm/mach-davinci/board-da830-evm.c b/arch/arm/mach-davinci/board-da830-evm.c index 9025669..a45340d 100644 --- a/arch/arm/mach-davinci/board-da830-evm.c +++ b/arch/arm/mach-davinci/board-da830-evm.c @@ -23,6 +23,7 @@ #include <mach/irqs.h> #include <mach/cp_intc.h> #include <mach/da8xx.h> +#include <mach/asp.h> #define DA830_EVM_PHY_MASK 0x0 #define DA830_EVM_MDIO_FREQUENCY 2200000 /* PHY bus frequency */ @@ -51,6 +52,25 @@ static struct davinci_uart_config da830_evm_uart_config __initdata = { .enabled_uarts = 0x7, }; +static u8 da830_iis_serializer_direction[] = { + RX_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, TX_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static struct snd_platform_data da830_evm_snd_data = { + .tx_dma_offset = 0x2000, + .rx_dma_offset = 0x2000, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = ARRAY_SIZE(da830_iis_serializer_direction), + .tdm_slots = 2, + .serial_dir = da830_iis_serializer_direction, + .eventq_no = EVENTQ_0, + .version = MCASP_VERSION_2, + .txnumevt = 1, + .rxnumevt = 1, +}; + static __init void da830_evm_init(void) { struct davinci_soc_info *soc_info = &davinci_soc_info; @@ -93,6 +113,13 @@ static __init void da830_evm_init(void) davinci_serial_init(&da830_evm_uart_config); i2c_register_board_info(1, da830_evm_i2c_devices, ARRAY_SIZE(da830_evm_i2c_devices)); + + ret = da8xx_pinmux_setup(da830_mcasp1_pins); + if (ret) + pr_warning("da830_evm_init: mcasp1 mux setup failed: %d\n", + ret); + + da8xx_init_mcasp(1, &da830_evm_snd_data); } #ifdef CONFIG_SERIAL_8250_CONSOLE diff --git a/arch/arm/mach-davinci/da830.c b/arch/arm/mach-davinci/da830.c index 3a9b634..19b2748 100644 --- a/arch/arm/mach-davinci/da830.c +++ b/arch/arm/mach-davinci/da830.c @@ -23,6 +23,7 @@ #include <mach/common.h> #include <mach/time.h> #include <mach/da8xx.h> +#include <mach/asp.h> #include "clock.h" #include "mux.h" @@ -411,9 +412,9 @@ static struct davinci_clk da830_clks[] = { CLK("eqep.0", NULL, &eqep0_clk), CLK("eqep.1", NULL, &eqep1_clk), CLK("da830_lcdc", NULL, &lcdc_clk), - CLK("soc-audio.0", NULL, &mcasp0_clk), - CLK("soc-audio.1", NULL, &mcasp1_clk), - CLK("soc-audio.2", NULL, &mcasp2_clk), + CLK("davinci-mcasp.0", NULL, &mcasp0_clk), + CLK("davinci-mcasp.1", NULL, &mcasp1_clk), + CLK("davinci-mcasp.2", NULL, &mcasp2_clk), CLK("musb_hdrc", NULL, &usb20_clk), CLK(NULL, "aemif", &aemif_clk), CLK(NULL, "aintc", &aintc_clk), diff --git a/arch/arm/mach-davinci/devices-da8xx.c b/arch/arm/mach-davinci/devices-da8xx.c index 11c0971..fe0baaf 100644 --- a/arch/arm/mach-davinci/devices-da8xx.c +++ b/arch/arm/mach-davinci/devices-da8xx.c @@ -281,7 +281,43 @@ static struct platform_device da8xx_emac_device = { .resource = da8xx_emac_resources, }; +static struct resource da830_mcasp1_resources[] = { + { + .name = "mcasp1", + .start = DAVINCI_DA830_MCASP1_REG_BASE, + .end = DAVINCI_DA830_MCASP1_REG_BASE + (SZ_1K * 12) - 1, + .flags = IORESOURCE_MEM, + }, + /* TX event */ + { + .start = DAVINCI_DA830_DMA_MCASP1_AXEVT, + .end = DAVINCI_DA830_DMA_MCASP1_AXEVT, + .flags = IORESOURCE_DMA, + }, + /* RX event */ + { + .start = DAVINCI_DA830_DMA_MCASP1_AREVT, + .end = DAVINCI_DA830_DMA_MCASP1_AREVT, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device da830_mcasp1_device = { + .name = "davinci-mcasp", + .id = 1, + .num_resources = ARRAY_SIZE(da830_mcasp1_resources), + .resource = da830_mcasp1_resources, +}; + int __init da8xx_register_emac(void) { return platform_device_register(&da8xx_emac_device); } + +void __init da8xx_init_mcasp(int id, struct snd_platform_data *pdata) +{ + if (id == 1) { /* DA830/OMAP-L137 has 3 instances of McASP */ + da830_mcasp1_device.dev.platform_data = pdata; + platform_device_register(&da830_mcasp1_device); + } +} diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index cdf1f44..f3c97ac 100644 --- a/arch/arm/mach-davinci/include/mach/asp.h +++ b/arch/arm/mach-davinci/include/mach/asp.h @@ -15,6 +15,9 @@ #define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 #define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 +/* Bases of da830 McASP1 register banks */ +#define DAVINCI_DA830_MCASP1_REG_BASE 0x01D04000 + /* EDMA channels of dm644x and dm355 */ #define DAVINCI_DMA_ASP0_TX 2 #define DAVINCI_DMA_ASP0_RX 3 @@ -26,6 +29,10 @@ #define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 #define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 +/* EDMA channels of da830 McASP1 */ +#define DAVINCI_DA830_DMA_MCASP1_AREVT 2 +#define DAVINCI_DA830_DMA_MCASP1_AXEVT 3 + /* Interrupts */ #define DAVINCI_ASP0_RX_INT IRQ_MBRINT #define DAVINCI_ASP0_TX_INT IRQ_MBXINT @@ -43,6 +50,14 @@ struct snd_platform_data { u8 op_mode; u8 num_serializer; u8 *serial_dir; + u8 version; + u8 txnumevt; + u8 rxnumevt; +}; + +enum { + MCASP_VERSION_1 = 0, /* DM646x */ + MCASP_VERSION_2, /* DA8xx/OMAPL1x */ }; #define INACTIVE_MODE 0 diff --git a/arch/arm/mach-davinci/include/mach/da8xx.h b/arch/arm/mach-davinci/include/mach/da8xx.h index a8cb570..30c5c40 100644 --- a/arch/arm/mach-davinci/include/mach/da8xx.h +++ b/arch/arm/mach-davinci/include/mach/da8xx.h @@ -15,6 +15,7 @@ #include <mach/edma.h> #include <mach/i2c.h> #include <mach/emac.h> +#include <mach/asp.h> /* * The cp_intc interrupt controller for the da8xx isn't in the same @@ -65,6 +66,7 @@ int da8xx_register_edma(void); int da8xx_register_i2c(int instance, struct davinci_i2c_platform_data *pdata); int da8xx_register_watchdog(void); int da8xx_register_emac(void); +void __init da8xx_init_mcasp(int id, struct snd_platform_data *pdata); extern struct platform_device da8xx_serial_device; extern struct emac_platform_data da8xx_emac_pdata; @@ -100,6 +102,7 @@ extern const short da850_uart2_pins[]; extern const short da850_i2c0_pins[]; extern const short da850_i2c1_pins[]; extern const short da850_cpgmac_pins[]; +extern const short da850_mcasp_pins[]; int da8xx_pinmux_setup(const short pins[]); -- 1.5.6

2 1

[alsa-devel] [PATCH] alsa: hdsp - allow proc reporting with disconnected io box
by Tim Blechmann 12 Aug '09

12 Aug '09

the hdsp driver refuses to report any information via the proc interface, if the io box is not connected. with this patch, the content of the control and status registers is printed before the iobox check. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/rme9652/hdsp.c | 37 ++++++++++++++++++++----------------- 1 files changed, 20 insertions(+), 17 deletions(-)

2 3

[alsa-devel] [PATCH] alsa-info: Use sysfs if available instead of dmidecode
by Ozan Çağlayan 12 Aug '09

12 Aug '09

Using sysfs for acquiring DMI data requires no root privileges. Use it if available instead of dmidecode. Signed-off-by: Ozan Çağlayan <ozan(a)pardus.org.tr> --- utils/alsa-info.sh | 7 ++++++- 1 files changed, 6 insertions(+), 1 deletions(-) diff --git a/utils/alsa-info.sh b/utils/alsa-info.sh index 0ab1ddf..8f401c0 100755 --- a/utils/alsa-info.sh +++ b/utils/alsa-info.sh @@ -417,7 +417,12 @@ ARTSINST=$(which artsd 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) JACKINST=$(which jackd 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) DMIDECODE=$(which dmidecode 2>/dev/null| sed 's|^[^/]*||' 2>/dev/null) -if [ -x $DMIDECODE ]; then +#Check for DMI data +if [ -d /sys/class/dmi/id ]; then + # No root privileges are required when using sysfs method + DMI_SYSTEM_MANUFACTURER=$(cat /sys/class/dmi/id/sys_vendor 2>/dev/null) + DMI_SYSTEM_PRODUCT_NAME=$(cat /sys/class/dmi/id/product_name 2>/dev/null) +elif [ -x $DMIDECODE ]; then DMI_SYSTEM_MANUFACTURER=$($DMIDECODE -s system-manufacturer 2>/dev/null) DMI_SYSTEM_PRODUCT_NAME=$($DMIDECODE -s system-product-name 2>/dev/null) fi -- 1.6.3.4

2 1

[alsa-devel] [PATCH] add output/input widgets in ad1938 to make dac/adc dynamic PM work
by Barry Song 12 Aug '09

12 Aug '09

According to the function dapm_dac_check_power() in sound/soc/soc-dapm.c, dac power can't be on/off stand-alone without any output widget as sink. And according to dapm_adc_check_power(), adc power can't be on/off stand-alone without any input widget as source. So we can't only define some stand-alone SND_SOC_DAPM_DAC/SND_SOC_DAPM_ADC to hope their power can be managed dynamically. Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- sound/soc/codecs/ad1938.c | 12 ++++++++++++ 1 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/ad1938.c b/sound/soc/codecs/ad1938.c index aafda74..9b8cc5c 100644 --- a/sound/soc/codecs/ad1938.c +++ b/sound/soc/codecs/ad1938.c @@ -99,11 +99,23 @@ static const struct snd_soc_dapm_widget ad1938_dapm_widgets[] = { SND_SOC_DAPM_DAC("DAC", "Playback", AD1938_DAC_CTRL0, 0, 1), SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_SUPPLY("ADC_PWR", AD1938_ADC_CTRL0, 0, 1, NULL, 0), + SND_SOC_DAPM_OUTPUT("DAC1OUT"), + SND_SOC_DAPM_OUTPUT("DAC2OUT"), + SND_SOC_DAPM_OUTPUT("DAC3OUT"), + SND_SOC_DAPM_OUTPUT("DAC4OUT"), + SND_SOC_DAPM_INPUT("ADC1IN"), + SND_SOC_DAPM_INPUT("ADC2IN"), }; static const struct snd_soc_dapm_route audio_paths[] = { { "DAC", NULL, "ADC_PWR" }, { "ADC", NULL, "ADC_PWR" }, + { "DAC1OUT", "DAC1 Switch", "DAC" }, + { "DAC2OUT", "DAC2 Switch", "DAC" }, + { "DAC3OUT", "DAC3 Switch", "DAC" }, + { "DAC4OUT", "DAC4 Switch", "DAC" }, + { "ADC", "ADC1 Switch", "ADC1IN" }, + { "ADC", "ADC2 Switch", "ADC2IN" }, }; /* -- 1.5.6.3

2 1

[alsa-devel] [PATCHv2 00/20] OMAP ASoC changes in DMA utilization
by Eduardo Valentin 12 Aug '09

12 Aug '09

Hello guys, Here is version 2 of this series. There were lots of little changes and also one significant change. Here is a changelog: - I tried to avoid additional ifdefs in mcbsp code - The WAKEUPEN_ALL macro was removed - I added a REVISIT prefix for the additional delay on mcbsp start - New added constants are now following mcbsp.h style - While reading DMA operation mode of mcbsp, now it prints also some help text to explain the meaning of each value (even though you still write numbers). Besides, I re-wrote the code for ASoC part, now the threshold and dma op mode usage are un-glued from mcbsp code. I tried to do not leave omap-pcm.c dirty, but, there was still one needed callback for setting the mcbsp threshold, as that needs to be done before DMA is started. In this series I've left THRESHOLD mode as default for mcbsp 2 only. But as I saw in previous comments, this is still an open issue :-) Comments, as usual are welcome. Eduardo Valentin (11): OMAP: McBSP: Add IRQEN, IRQSTATUS, THRESHOLD2 and THRESHOLD1 registers. OMAP: McBSP: Use appropriate value for startup delay OMAP: McBSP: Add transmit/receive threshold handler OMAP: McBSP: Create and export max_(r|t)x_thres property OMAP: McBSP: Rename thres sysfs symbols OMAP: McBSP: Change wakeup signals OMAP: McBSP: Configure NO IDLE mode for DMA mode different of threshold ASoC: OMAP: Use McBSP threshold to playback and capture ASoC: OMAP: Use DMA operating mode of McBSP ASoC: OMAP: Make DMA 64 aligned ASoC: OMAP: Enable DMA burst mode Eero Nurkkala (7): OMAP: McBSP: Provide functions for ASoC frame syncronization OMAP: McBSP: Wakeups utilized OMAP: McBSP: Retain McBSP FCLK clockactivity OMAP: McBSP: Do not enable wakeups for no-idle mode OMAP: McBSP: Let element DMA mode hit retention also ASoC: Add runtime check for RFIG and XFIG ASoC: Always syncronize audio transfers on frames Peter Ujfalusi (2): OMAP3: McBSP: Lower the maximum buffersize for McBSP1,3,4,5 OMAP: McBSP: Add link DMA mode selection arch/arm/mach-omap2/mcbsp.c | 5 + arch/arm/plat-omap/include/mach/mcbsp.h | 49 ++++ arch/arm/plat-omap/mcbsp.c | 375 ++++++++++++++++++++++++++++++- sound/soc/omap/omap-mcbsp.c | 67 +++++- sound/soc/omap/omap-pcm.c | 33 +++- sound/soc/omap/omap-pcm.h | 2 + 6 files changed, 517 insertions(+), 14 deletions(-) -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 23

[alsa-devel] [PATCH] ALSA: hda - fix noise issue when recording from digital mic with alc268
by Herton Ronaldo Krzesinski 12 Aug '09

12 Aug '09

With auto config model of alc268 realtek codec, it allows to select any of possible available digital microphone inputs when only one is available. For example, when only digital mic in nid 0x12 is available, on second input source it will allow you to select unavailable digital mic in nid 0x13. The problem is that selecting unavailable digital mic creates a source of noise when recording (I'm not sure if this happens on all machines with alc268 and only one digital mic input, but testing on a quanta uw1 netbook a lot of noise is introduced in recording from digital mic 0x12/first input source, when you select the unavailable digital mic 0x13 for capture source 0x24 in the second input source in mixer). Then to avoid noise when recording from digital mic with auto model in this case, prevent a digital mic input source to be selected if microphone is not available. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- sound/pci/hda/patch_realtek.c | 62 +++++++++++++++++++++++----------------- 1 files changed, 36 insertions(+), 26 deletions(-) I'm attaching the codec info too here of the machine with the problem. diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ac3207e..7ff2934 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12232,32 +12232,40 @@ static int alc268_auto_create_analog_input_ctls(struct alc_spec *spec, const struct auto_pin_cfg *cfg) { struct hda_input_mux *imux = &spec->private_imux[0]; - int i, idx1; - - for (i = 0; i < AUTO_PIN_LAST; i++) { - switch(cfg->input_pins[i]) { - case 0x18: - idx1 = 0; /* Mic 1 */ - break; - case 0x19: - idx1 = 1; /* Mic 2 */ - break; - case 0x1a: - idx1 = 2; /* Line In */ - break; - case 0x1c: - idx1 = 3; /* CD */ - break; - case 0x12: - case 0x13: - idx1 = 6; /* digital mics */ - break; - default: - continue; + int i, idx1, dmic_nid; + + dmic_nid = 0x12; + while (dmic_nid <= 0x13) { + for (i = 0; i < AUTO_PIN_LAST; i++) { + switch (cfg->input_pins[i]) { + case 0x18: + idx1 = 0; /* Mic 1 */ + break; + case 0x19: + idx1 = 1; /* Mic 2 */ + break; + case 0x1a: + idx1 = 2; /* Line In */ + break; + case 0x1c: + idx1 = 3; /* CD */ + break; + case 0x12: + case 0x13: + if (cfg->input_pins[i] != dmic_nid) + continue; + idx1 = 6; /* digital mics */ + break; + default: + continue; + } + imux->items[imux->num_items].label = + auto_pin_cfg_labels[i]; + imux->items[imux->num_items].index = idx1; + imux->num_items++; } - imux->items[imux->num_items].label = auto_pin_cfg_labels[i]; - imux->items[imux->num_items].index = idx1; - imux->num_items++; + imux++; + dmic_nid++; } return 0; } @@ -12355,7 +12363,7 @@ static int alc268_parse_auto_config(struct hda_codec *codec) add_mixer(spec, alc268_beep_mixer); add_verb(spec, alc268_volume_init_verbs); - spec->num_mux_defs = 1; + spec->num_mux_defs = 2; spec->input_mux = &spec->private_imux[0]; err = alc_auto_add_mic_boost(codec); @@ -12665,6 +12673,8 @@ static int patch_alc268(struct hda_codec *codec) for (i = 0; i < spec->num_adc_nids; i++) snd_hda_codec_write_cache(codec, alc268_capsrc_nids[i], 0, AC_VERB_SET_CONNECT_SEL, + i < spec->num_mux_defs ? + spec->input_mux[i].items[0].index : spec->input_mux->items[0].index); } -- 1.6.4

2 1

[alsa-devel] [PATCH] ASoC: DaVinci: Add audio support fot DA850/OMAP-L138 EVM
by Chaithrika U S 11 Aug '09

11 Aug '09

There is one instance of McASP on DA850/OMAP-L138 SoC. This is connected to TLV320AIC3106 codec for audio playback and capture. This patch adds audio support on this platform. Some of the structure prefix names which are common for DA830/OMAP-L137 EVM and DA850/OMAP-L138 EVM have been renamed to da8xx from da830. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies to ALSA GIT tree on branch topic/asoc at http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=shortlog; h=topic/asoc sound/soc/davinci/Kconfig | 13 +++++++++++-- sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 29 +++++++++++++++++++++++------ 3 files changed, 35 insertions(+), 8 deletions(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 677a538..4dfd4ad 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -43,12 +43,21 @@ config SND_DAVINCI_SOC_SFFSDR Lyrtech SFFSDR board. config SND_DA830_SOC_EVM - tristate "SoC Audio support for DA830/OMAPL137 EVM" + tristate "SoC Audio support for DA830/OMAP-L137 EVM" depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA830_EVM select SND_DAVINCI_SOC_MCASP select SND_SOC_TLV320AIC3X help Say Y if you want to add support for SoC audio on TI - DA830/OMAPL137 EVM + DA830/OMAP-L137 EVM + +config SND_DA850_SOC_EVM + tristate "SoC Audio support for DA850/OMAP-L138 EVM" + depends on SND_DAVINCI_SOC && MACH_DAVINCI_DA850_EVM + select SND_DAVINCI_SOC_MCASP + select SND_SOC_TLV320AIC3X + help + Say Y if you want to add support for SoC audio on TI + DA850/OMAP-L138 EVM diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index 5e2195f..a6939d7 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -14,4 +14,5 @@ snd-soc-sffsdr-objs := davinci-sffsdr.o obj-$(CONFIG_SND_DAVINCI_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DM6467_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DA830_SOC_EVM) += snd-soc-evm.o +obj-$(CONFIG_SND_DA850_SOC_EVM) += snd-soc-evm.o obj-$(CONFIG_SND_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index a5a9212..46c1b0c 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -54,7 +54,8 @@ static int evm_hw_params(struct snd_pcm_substream *substream, else if (machine_is_davinci_evm()) sysclk = 12288000; - else if (machine_is_davinci_da830_evm()) + else if (machine_is_davinci_da830_evm() || + machine_is_davinci_da850_evm()) sysclk = 24576000; else @@ -165,7 +166,7 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .ops = &evm_ops, }, }; -static struct snd_soc_dai_link da830_evm_dai = { +static struct snd_soc_dai_link da8xx_evm_dai = { .name = "TLV320AIC3X", .stream_name = "AIC3X", .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_I2S_DAI], @@ -191,8 +192,15 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = { }; static struct snd_soc_card da830_snd_soc_card = { - .name = "DA830 EVM", - .dai_link = &da830_evm_dai, + .name = "DA830/OMAP-L137 EVM", + .dai_link = &da8xx_evm_dai, + .platform = &davinci_soc_platform, + .num_links = 1, +}; + +static struct snd_soc_card da850_snd_soc_card = { + .name = "DA850/OMAP-L138 EVM", + .dai_link = &da8xx_evm_dai, .platform = &davinci_soc_platform, .num_links = 1, }; @@ -209,7 +217,7 @@ static struct aic3x_setup_data dm6467_evm_aic3x_setup = { .i2c_address = 0x18, }; -static struct aic3x_setup_data da830_evm_aic3x_setup = { +static struct aic3x_setup_data da8xx_evm_aic3x_setup = { .i2c_bus = 1, .i2c_address = 0x18, }; @@ -232,7 +240,13 @@ static struct snd_soc_device dm6467_evm_snd_devdata = { static struct snd_soc_device da830_evm_snd_devdata = { .card = &da830_snd_soc_card, .codec_dev = &soc_codec_dev_aic3x, - .codec_data = &da830_evm_aic3x_setup, + .codec_data = &da8xx_evm_aic3x_setup, +}; + +static struct snd_soc_device da850_evm_snd_devdata = { + .card = &da850_snd_soc_card, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &da8xx_evm_aic3x_setup, }; static struct platform_device *evm_snd_device; @@ -255,6 +269,9 @@ static int __init evm_init(void) } else if (machine_is_davinci_da830_evm()) { evm_snd_dev_data = &da830_evm_snd_devdata; index = 1; + } else if (machine_is_davinci_da850_evm()) { + evm_snd_dev_data = &da850_evm_snd_devdata; + index = 0; } else return -EINVAL; -- 1.5.6

1 0