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

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[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

[alsa-devel] [PATCH v2 2/2] ASoC: DaVinci: Add a DAI format to McASP driver
by Chaithrika U S 11 Aug '09

11 Aug '09

The patch adds a DAI format: Codec bit clock master and frame sync slave, to the driver. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- This patch has been separated out from the first patch of this series. In the previous version, this was part of the McASP driver FIFO support 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-mcasp.c | 10 ++++++++++ 1 files changed, 10 insertions(+), 0 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index e672f43..eca22d7 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -443,6 +443,16 @@ 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_CFS: + /* codec is clock master and frame slave */ + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x2d << 26)); + break; case SND_SOC_DAIFMT_CBM_CFM: /* codec is clock and frame master */ mcasp_clr_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); -- 1.5.6

1 0

[alsa-devel] hda-emu codec information files
by Ozan Çağlayan 11 Aug '09

11 Aug '09

Hi, I'm able to collect a lot of alsa-info from a bunch of netbook/laptop and desktop boards. I can send them as a patch series if it will help debug problems Thanks Ozan

2 1

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 11 Aug '09

11 Aug '09

The following changes since commit e0c48a18f76415d3ec1ac28e5f7a0df10e2f3d8c: Mark Brown (1): ASoC: Drop unneeded declaration of removed wm8731 SPI write function 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 Daniel Ribeiro (1): ASoC: change set_tdm_slot api to allow slot_width override. Mark Brown (2): Merge commit 'a5479e389e989acfeca9c32eeb0083d086202280' into for-2.6.32 ASoC: Update WM9081 for tdm_slot() API change Randy Dunlap (1): ASoC: fix I2C build errors include/sound/soc-dai.h | 5 ++- sound/soc/codecs/wm9081.c | 63 ++++++++++++++++++++++++++------------------ sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/pxa-ssp.c | 27 ++++++++++++++----- sound/soc/soc-cache.c | 4 +- sound/soc/soc-core.c | 9 ++++-- 6 files changed, 69 insertions(+), 41 deletions(-)

2 1

[alsa-devel] hard freezes with hdsp with cardbus/pci adapter card
by Tim Blechmann 11 Aug '09

11 Aug '09

hi all, i have some issues with playback on one of my machines with different kernels, having my rme hammerfall hdsp connected via a cardbus/pci adapter card. after some time (between a few seconds and several minutes) of flawless playback, the system completely freezes, magic sysrq keys are not functional either ... the freeze occurs with rt and non-rt kernels, with jack running with and without realtime scheduling. i have the suspicion, that it is related to the combination of the hdsp with the cardbus controller of the adapter card ... the interesting part of lspci is: 06:01.0 CardBus bridge: ENE Technology Inc CB1410 Cardbus Controller (rev 01) Subsystem: ENE Technology Inc CB1410 Cardbus Controller Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 168, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 17 Region 0: Memory at fb600000 (32-bit, non-prefetchable) [size=4K] Bus: primary=06, secondary=07, subordinate=0a, sec-latency=176 Memory window 0: c4000000-c7fff000 (prefetchable) Memory window 1: c8000000-cbfff000 I/O window 0: 0000e000-0000e0ff I/O window 1: 0000e400-0000e4ff BridgeCtl: Parity+ SERR+ ISA- VGA- MAbort- >Reset- 16bInt- PostWrite+ 16-bit legacy interface ports at 0001 Kernel driver in use: yenta_cardbus Kernel modules: yenta_socket 07:00.0 Multimedia audio controller: Xilinx Corporation RME Hammerfall DSP (rev 0f) Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 255 Interrupt: pin A routed to IRQ 17 Region 0: Memory at c8000000 (32-bit, non-prefetchable) [size=64K] Kernel driver in use: RME Hammerfall DSP Kernel modules: snd-hdsp the irq 17 is used by these two devices only. i am not sure, how to proceed debugging this, since i cannot get any useful debugging information ... thnx, tim -- tim(a)klingt.org http://tim.klingt.org A paranoid is a man who knows a little of what's going on. William S. Burroughs

3 7

[alsa-devel] (no subject)
by Chaithrika U S 11 Aug '09

11 Aug '09

Define resources for McASP used on DA850/OMAP-L138 EVM, add platform device defintion and Pin Mux configurations. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- Applies on DaVinci GIT tree. This patch is dependent on the following patch which I have submitted earlier: [PATCH v2] ARM: DaVinci: Audio support for DA830 EVM arch/arm/mach-davinci/board-da850-evm.c | 29 ++++++++++++++++++++++ arch/arm/mach-davinci/da850.c | 39 ++++++++++++++++++++++++++++++ arch/arm/mach-davinci/devices-da8xx.c | 34 +++++++++++++++++++++++++- arch/arm/mach-davinci/include/mach/asp.h | 7 +++++ arch/arm/mach-davinci/include/mach/mux.h | 25 +++++++++++++++++++ 5 files changed, 133 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-davinci/board-da850-evm.c b/arch/arm/mach-davinci/board-da850-evm.c index d989346..52bfe4c 100644 --- a/arch/arm/mach-davinci/board-da850-evm.c +++ b/arch/arm/mach-davinci/board-da850-evm.c @@ -38,6 +38,28 @@ static struct davinci_uart_config da850_evm_uart_config __initdata = { .enabled_uarts = 0x7, }; +/* davinci da850 evm audio machine driver */ +static u8 da850_iis_serializer_direction[] = { + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, + INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, TX_MODE, + RX_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, +}; + +static struct snd_platform_data da850_evm_snd_data = { + .tx_dma_offset = 0x2000, + .rx_dma_offset = 0x2000, + .op_mode = DAVINCI_MCASP_IIS_MODE, + .num_serializer = ARRAY_SIZE(da850_iis_serializer_direction), + .tdm_slots = 2, + .serial_dir = da850_iis_serializer_direction, + .eventq_no = EVENTQ_1, + .version = MCASP_VERSION_2, + .txnumevt = 1, + .rxnumevt = 1, +}; + + static __init void da850_evm_init(void) { struct davinci_soc_info *soc_info = &davinci_soc_info; @@ -86,6 +108,13 @@ static __init void da850_evm_init(void) */ __raw_writel(0, IO_ADDRESS(DA8XX_UART1_BASE) + 0x30); __raw_writel(0, IO_ADDRESS(DA8XX_UART0_BASE) + 0x30); + + ret = da8xx_pinmux_setup(da850_mcasp_pins); + if (ret) + pr_warning("da850_evm_init: mcasp mux setup failed: %d\n", + ret); + + da8xx_init_mcasp(0, &da850_evm_snd_data); } #ifdef CONFIG_SERIAL_8250_CONSOLE diff --git a/arch/arm/mach-davinci/da850.c b/arch/arm/mach-davinci/da850.c index 4a43ae2..e33b226 100644 --- a/arch/arm/mach-davinci/da850.c +++ b/arch/arm/mach-davinci/da850.c @@ -289,6 +289,13 @@ static struct clk emac_clk = { .lpsc = DA8XX_LPSC1_CPGMAC, }; +static struct clk mcasp_clk = { + .name = "mcasp", + .parent = &pll0_sysclk2, + .lpsc = DA8XX_LPSC1_McASP0, + .psc_ctlr = 1, +}; + static struct davinci_clk da850_clks[] = { CLK(NULL, "ref", &ref_clk), CLK(NULL, "pll0", &pll0_clk), @@ -326,6 +333,7 @@ static struct davinci_clk da850_clks[] = { CLK(NULL, "arm", &arm_clk), CLK(NULL, "rmii", &rmii_clk), CLK("davinci_emac.1", NULL, &emac_clk), + CLK("davinci-mcasp.0", NULL, &mcasp_clk), CLK(NULL, NULL, NULL), }; @@ -370,6 +378,30 @@ static const struct mux_config da850_pins[] = { MUX_CFG(DA850, MII_RXD_2, 3, 20, 15, 8, false) MUX_CFG(DA850, MII_RXD_1, 3, 24, 15, 8, false) MUX_CFG(DA850, MII_RXD_0, 3, 28, 15, 8, false) + /* McASP function */ + MUX_CFG(DA850, ACLKR, 0, 0, 15, 1, false) + MUX_CFG(DA850, ACLKX, 0, 4, 15, 1, false) + MUX_CFG(DA850, AFSR, 0, 8, 15, 1, false) + MUX_CFG(DA850, AFSX, 0, 12, 15, 1, false) + MUX_CFG(DA850, AHCLKR, 0, 16, 15, 1, false) + MUX_CFG(DA850, AHCLKX, 0, 20, 15, 1, false) + MUX_CFG(DA850, AMUTE, 0, 24, 15, 1, false) + MUX_CFG(DA850, AXR_15, 1, 0, 15, 1, false) + MUX_CFG(DA850, AXR_14, 1, 4, 15, 1, false) + MUX_CFG(DA850, AXR_13, 1, 8, 15, 1, false) + MUX_CFG(DA850, AXR_12, 1, 12, 15, 1, false) + MUX_CFG(DA850, AXR_11, 1, 16, 15, 1, false) + MUX_CFG(DA850, AXR_10, 1, 20, 15, 1, false) + MUX_CFG(DA850, AXR_9, 1, 24, 15, 1, false) + MUX_CFG(DA850, AXR_8, 1, 28, 15, 1, false) + MUX_CFG(DA850, AXR_7, 2, 0, 15, 1, false) + MUX_CFG(DA850, AXR_6, 2, 4, 15, 1, false) + MUX_CFG(DA850, AXR_5, 2, 8, 15, 1, false) + MUX_CFG(DA850, AXR_4, 2, 12, 15, 1, false) + MUX_CFG(DA850, AXR_3, 2, 16, 15, 1, false) + MUX_CFG(DA850, AXR_2, 2, 20, 15, 1, false) + MUX_CFG(DA850, AXR_1, 2, 24, 15, 1, false) + MUX_CFG(DA850, AXR_0, 2, 28, 15, 1, false) #endif }; @@ -406,6 +438,13 @@ const short da850_cpgmac_pins[] __initdata = { -1 }; +const short da850_mcasp_pins[] __initdata = { + DA850_AHCLKX, DA850_ACLKX, DA850_AFSX, + DA850_AHCLKR, DA850_ACLKR, DA850_AFSR, DA850_AMUTE, + DA850_AXR_11, DA850_AXR_12, + -1 +}; + /* FIQ are pri 0-1; otherwise 2-7, with 7 lowest priority */ static u8 da850_default_priorities[DA850_N_CP_INTC_IRQ] = { [IRQ_DA8XX_COMMTX] = 7, diff --git a/arch/arm/mach-davinci/devices-da8xx.c b/arch/arm/mach-davinci/devices-da8xx.c index fe0baaf..1882eef 100644 --- a/arch/arm/mach-davinci/devices-da8xx.c +++ b/arch/arm/mach-davinci/devices-da8xx.c @@ -309,6 +309,34 @@ static struct platform_device da830_mcasp1_device = { .resource = da830_mcasp1_resources, }; +static struct resource da850_mcasp_resources[] = { + { + .name = "mcasp", + .start = DAVINCI_DA8XX_MCASP0_REG_BASE, + .end = DAVINCI_DA8XX_MCASP0_REG_BASE + (SZ_1K * 12) - 1, + .flags = IORESOURCE_MEM, + }, + /* TX event */ + { + .start = DAVINCI_DA8XX_DMA_MCASP0_AXEVT, + .end = DAVINCI_DA8XX_DMA_MCASP0_AXEVT, + .flags = IORESOURCE_DMA, + }, + /* RX event */ + { + .start = DAVINCI_DA8XX_DMA_MCASP0_AREVT, + .end = DAVINCI_DA8XX_DMA_MCASP0_AREVT, + .flags = IORESOURCE_DMA, + }, +}; + +static struct platform_device da850_mcasp_device = { + .name = "davinci-mcasp", + .id = 0, + .num_resources = ARRAY_SIZE(da850_mcasp_resources), + .resource = da850_mcasp_resources, +}; + int __init da8xx_register_emac(void) { return platform_device_register(&da8xx_emac_device); @@ -316,8 +344,12 @@ int __init da8xx_register_emac(void) void __init da8xx_init_mcasp(int id, struct snd_platform_data *pdata) { - if (id == 1) { /* DA830/OMAP-L137 has 3 instances of McASP */ + /* DA830/OMAP-L137 has 3 instances of McASP */ + if (cpu_is_davinci_da830() && id == 1) { da830_mcasp1_device.dev.platform_data = pdata; platform_device_register(&da830_mcasp1_device); + } else if (cpu_is_davinci_da850()) { + da850_mcasp_device.dev.platform_data = pdata; + platform_device_register(&da850_mcasp_device); } } diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index f3c97ac..18e4ce3 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 da850/da830 McASP0 register banks */ +#define DAVINCI_DA8XX_MCASP0_REG_BASE 0x01D00000 + /* Bases of da830 McASP1 register banks */ #define DAVINCI_DA830_MCASP1_REG_BASE 0x01D04000 @@ -29,6 +32,10 @@ #define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9 #define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12 +/* EDMA channels of da850/da830 McASP0 */ +#define DAVINCI_DA8XX_DMA_MCASP0_AREVT 0 +#define DAVINCI_DA8XX_DMA_MCASP0_AXEVT 1 + /* EDMA channels of da830 McASP1 */ #define DAVINCI_DA830_DMA_MCASP1_AREVT 2 #define DAVINCI_DA830_DMA_MCASP1_AXEVT 3 diff --git a/arch/arm/mach-davinci/include/mach/mux.h b/arch/arm/mach-davinci/include/mach/mux.h index a676b2f..f50f5b2 100644 --- a/arch/arm/mach-davinci/include/mach/mux.h +++ b/arch/arm/mach-davinci/include/mach/mux.h @@ -748,6 +748,31 @@ enum davinci_da850_index { DA850_MII_RXD_2, DA850_MII_RXD_1, DA850_MII_RXD_0, + + /* McASP function */ + DA850_ACLKR, + DA850_ACLKX, + DA850_AFSR, + DA850_AFSX, + DA850_AHCLKR, + DA850_AHCLKX, + DA850_AMUTE, + DA850_AXR_15, + DA850_AXR_14, + DA850_AXR_13, + DA850_AXR_12, + DA850_AXR_11, + DA850_AXR_10, + DA850_AXR_9, + DA850_AXR_8, + DA850_AXR_7, + DA850_AXR_6, + DA850_AXR_5, + DA850_AXR_4, + DA850_AXR_3, + DA850_AXR_2, + DA850_AXR_1, + DA850_AXR_0, }; #ifdef CONFIG_DAVINCI_MUX -- 1.5.6

2 1

[alsa-devel] [PATCH 1/5] ASoC: DaVinci: i2s: don't bounce through rtd to get dai
by Troy Kisky 11 Aug '09

11 Aug '09

dai is a parameter to the functions, so use it instead of looking it up. Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- sound/soc/davinci/davinci-i2s.c | 14 +++++--------- 1 files changed, 5 insertions(+), 9 deletions(-) diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index e5cd97b..2a56fb7 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -353,9 +353,8 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; - struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct davinci_pcm_dma_params *dma_params = dai->dma_data; + struct davinci_mcbsp_dev *dev = dai->private_data; struct snd_interval *i = NULL; int mcbsp_word_length; unsigned int rcr, xcr, srgr; @@ -425,8 +424,7 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, static int davinci_i2s_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct davinci_mcbsp_dev *dev = dai->private_data; int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); davinci_mcbsp_stop(dev, playback); if ((dev->pcr & DAVINCI_MCBSP_PCR_FSXM) == 0) { @@ -439,8 +437,7 @@ static int davinci_i2s_prepare(struct snd_pcm_substream *substream, static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct davinci_mcbsp_dev *dev = dai->private_data; int ret = 0; int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); if ((dev->pcr & DAVINCI_MCBSP_PCR_FSXM) == 0) @@ -466,8 +463,7 @@ static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd, static void davinci_i2s_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct davinci_mcbsp_dev *dev = dai->private_data; int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); davinci_mcbsp_stop(dev, playback); } -- 1.5.6.3

2 11

[alsa-devel] Automatic mic-selection on Realtek and IDT/STAC codecs
by Takashi Iwai 11 Aug '09

11 Aug '09

Hi, recently I added the code for HD-audio Realtek and IDT/Sigmatel codecs to support the automatic capture-source selection via mic jack plugging. This feature is enabled for machines with one external and one internal mic, and only for auto-BIOS parser mode in the case of Realtek codecs. So, don't be surprised if you see neither "Capture Source" nor "Input Source" mixer elements after updates. Just plugging the mic jack will switch the input source from the internal to the external mic. If you encounter any problems regarding this change, please let me know. thanks, Takashi

1 0

[alsa-devel] NULL pointer dereference in ALSA triggered through saa7134-alsa
by Ozan Çağlayan 10 Aug '09

10 Aug '09

Hi, I've finally succesfully compiled and linked saa7134-alsa driver using an external alsa-driver and its Module.symvers file. Everything seems okay, no undefined symbol or something else: - An installed 2.6.30.4 kernel which only builds and brings soundcore and sound_firmware, - Latest alsa-driver built externally and installed, - Latest saa7134-alsa, cx88-alsa, etc. code from linus-2.6 (seen that they don't affected by some API/ABI changes) patched on top of the alsa-driver tarball, I'm copying Module.symvers file generated after building the alsa-driver into each directory for getting rid of undefined snd_*() symbols. When I plug a saa7134 PCI card and reboot, the -alsa driver is automatically probed but it immediately oopses. I wonder if this could be related to my weird-but-apparently-successful compilation experience above? [ 6.876120] Linux video capture interface: v2.00 [ 7.089550] saa7130/34: v4l2 driver version 0.2.15 loaded [ 7.089612] saa7134 0000:03:05.0: PCI INT A -> GSI 20 (level, low) -> IRQ 20 [ 7.089622] saa7133[0]: found at 0000:03:05.0, rev: 209, irq: 20, latency: 64, mmio: 0xfdbff000 [ 7.089631] saa7133[0]: subsystem: 1461:f11d, board: Avermedia PCI pure analog (M135A) [card=149,autodetected] [ 7.089659] saa7133[0]: board init: gpio is 40000 [ 7.089744] input: saa7134 IR (Avermedia PCI pure as /devices/pci0000:00/0000:00:14.4/0000:03:05.0/input/input6 [ 7.089789] IRQ 20/saa7133[0]: IRQF_DISABLED is not guaranteed on shared IRQs [ 7.230265] saa7133[0]: i2c eeprom 00: 61 14 1d f1 54 20 1c 00 43 43 a9 1c 55 d2 b2 92 [ 7.230274] saa7133[0]: i2c eeprom 10: ff ff ff ff ff 20 ff ff ff ff ff ff ff ff ff ff [ 7.230282] saa7133[0]: i2c eeprom 20: 01 40 01 32 32 01 01 43 88 ff 00 56 ff ff ff ff [ 7.230289] saa7133[0]: i2c eeprom 30: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230296] saa7133[0]: i2c eeprom 40: ff 22 00 c0 96 ff 03 30 15 00 ff ff ff ff ff ff [ 7.230302] saa7133[0]: i2c eeprom 50: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230309] saa7133[0]: i2c eeprom 60: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230316] saa7133[0]: i2c eeprom 70: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230323] saa7133[0]: i2c eeprom 80: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230330] saa7133[0]: i2c eeprom 90: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230337] saa7133[0]: i2c eeprom a0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230344] saa7133[0]: i2c eeprom b0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230350] saa7133[0]: i2c eeprom c0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230357] saa7133[0]: i2c eeprom d0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230364] saa7133[0]: i2c eeprom e0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.230371] saa7133[0]: i2c eeprom f0: ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff [ 7.412095] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 [ 7.512683] hda_codec: ALC883: BIOS auto-probing. [ 7.512886] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:14.2/input/input7 [ 7.516862] HDA Intel 0000:01:05.2: PCI INT B -> GSI 19 (level, low) -> IRQ 19 [ 7.689107] tuner 1-004b: chip found @ 0x96 (saa7133[0]) [ 7.724034] tda829x 1-004b: setting tuner address to 60 [ 7.800011] tda829x 1-004b: type set to tda8290+75a [ 8.294455] usblp0: USB Bidirectional printer dev 2 if 0 alt 0 proto 2 vid 0x04E8 pid 0x325B [ 8.294500] usbcore: registered new interface driver usblp [ 10.814379] saa7133[0]: dsp access error [ 10.845083] saa7133[0]: registered device video0 [v4l2] [ 10.845111] saa7133[0]: registered device vbi0 [ 10.845130] saa7133[0]: registered device radio0 [ 10.950359] EXT3 FS on sda5, internal journal [ 10.996203] saa7134 ALSA driver for DMA sound loaded [ 10.996217] IRQ 20/saa7133[0]: IRQF_DISABLED is not guaranteed on shared IRQs [ 10.996247] saa7133[0]/alsa: saa7133[0] at 0xfdbff000 irq 20 registered as card -1 <-- /* That's interesting? */ [ 10.996349] BUG: unable to handle kernel NULL pointer dereference at 00000002 <-- ********** 1st null ptr deref [ 10.996962] IP: [<f8834cd4>] snd_pcm_timer_init+0x26/0xd1 [snd_pcm] [ 10.997255] *pde = 00000000· [ 10.997255] Oops: 0000 [#1] SMP· [ 10.997255] last sysfs file: /sys/module/saa7134/initstate [ 10.997255] Modules linked in: saa7134_alsa(+) usblp tda827x tda8290 snd_hda_codec_atihdmi tuner snd_hda_codec_realtek snd_hda_intel snd_hda_codec snd_seq_dummy snd_hwdep snd_seq_oss saa7134 snd_seq_m idi_event snd_seq snd_seq_device ir_common snd_pcm_oss snd_mixer_oss v4l2_common videodev v4l1_compat videobuf_dma_sg videobuf_core snd_pcm tveeprom snd_timer snd i2c_piix4 shpchp k8temp soundcore snd_pa ge_alloc pcspkr r8169 ati_agp agpgart mii brd ata_generic pata_acpi ahci pata_atiixp libata [ 10.997255]· [ 10.997255] Pid: 1141, comm: modprobe.bin Not tainted (2.6.30.4-126 #3) Unknow [ 10.997255] EIP: 0060:[<f8834cd4>] EFLAGS: 00010246 CPU: 1 [ 10.997255] EIP is at snd_pcm_timer_init+0x26/0xd1 [snd_pcm] [ 10.997255] EAX: 00000002 EBX: f65c0e0c ECX: f6b2fe98 EDX: 00000000 [ 10.997255] ESI: f65c0c00 EDI: 00000018 EBP: f6b2febc ESP: f6b2fe98 [ 10.997255] DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068 [ 10.997255] Process modprobe.bin (pid: 1141, ti=f6b2e000 task=f6a2caa0 task.ti=f6b2e000) [ 10.997255] Stack: [ 10.997255] 00000003 00000000 c030c2e9 f6b2feb8 f87ca221 00000000 f65c0e0c f6921400 [ 10.997255] 00000018 f6b2feec f882da8f 00000001 f6921438 f88358e4 436d6370 63304432 [ 10.997255] f65c0c00 f87cf9fe f69b42a0 f65c0d70 f65c0c00 f6b2fefc f87ce0bb f65c0e0c [ 10.997255] Call Trace: [ 10.997255] [<c030c2e9>] ? device_create_file+0x13/0x15 [ 10.997255] [<f87ca221>] ? snd_add_device_sysfs_file+0x67/0x6e [snd] [ 10.997255] [<f882da8f>] ? snd_pcm_dev_register+0x176/0x1de [snd_pcm] [ 10.997255] [<f87ce0bb>] ? snd_device_register_all+0x27/0x44 [snd] [ 10.997255] [<f87cadce>] ? snd_card_register+0x4c/0x131 [snd] [ 10.997255] [<f8d17687>] ? alsa_device_init+0x200/0x24b [saa7134_alsa] [ 10.997255] [<f8d1773d>] ? saa7134_alsa_init+0x6b/0x93 [saa7134_alsa] [ 10.997255] [<c0101137>] ? do_one_initcall+0x4a/0x115 [ 10.997255] [<f8d176d2>] ? saa7134_alsa_init+0x0/0x93 [saa7134_alsa] [ 10.997255] [<c0145205>] ? __blocking_notifier_call_chain+0x40/0x4c [ 10.997255] [<c015302d>] ? sys_init_module+0x87/0x18b [ 10.997255] [<c01031b4>] ? sysenter_do_call+0x12/0x28 [ 11.066012] Code: 00 00 5b 5d c3 55 89 e5 57 56 53 89 c3 83 ec 18 8b 53 30 c7 45 e0 00 00 00 00 8d 4d dc c7 45 dc 03 00 00 00 8b 30 83 e2 01 8b 06 <8b> 00 89 45 e4 8b 46 0c 89 45 e8 8b 43 0c 01 c0 09 c2 8d 45 f0· [ 11.066012] EIP: [<f8834cd4>] snd_pcm_timer_init+0x26/0xd1 [snd_pcm] SS:ESP 0068:f6b2fe98 [ 11.066012] CR2: 0000000000000002 [ 11.066318] ---[ end trace 000b75f3b1903e42 ]--- [ 14.079154] r8169: eth0: link up [ 14.095472] r8169: eth0: link up [ 14.335980] r8169: eth0: link up [ 17.074440] NET: Registered protocol family 10 [ 17.074879] lo: Disabled Privacy Extensions [ 20.520592] BUG: unable to handle kernel NULL pointer dereference at 00000002 <-- ********* 2nd nullptr deref [ 20.535353] IP: [<f8830abf>] snd_pcm_info+0x2e/0xe0 [snd_pcm] [ 20.535353] *pde = 00000000· [ 20.559932] Oops: 0000 [#2] SMP· [ 20.559932] last sysfs file: /sys/devices/pci0000:00/0000:00:14.4/0000:03:05.0/sound/card2/pcmC2D0c/pcm_class [ 20.586971] Modules linked in: ipv6 saa7134_alsa(+) usblp tda827x tda8290 snd_hda_codec_atihdmi tuner snd_hda_codec_realtek snd_hda_intel snd_hda_codec snd_seq_dummy snd_hwdep snd_seq_oss saa7134 snd_ seq_midi_event snd_seq snd_seq_device ir_common snd_pcm_oss snd_mixer_oss v4l2_common videodev v4l1_compat videobuf_dma_sg videobuf_core snd_pcm tveeprom snd_timer snd i2c_piix4 shpchp k8temp soundcore s nd_page_alloc pcspkr r8169 ati_agp agpgart mii brd ata_generic pata_acpi ahci pata_atiixp libata [ 20.606741]· [ 20.606741] Pid: 1367, comm: hald Tainted: G D (2.6.30.4-126 #3) Unknow[ 20.704561] Process hald (pid: 1367, ti=f660e000 task=f6420c70 task.ti=f660e000) [ 20.704561] Stack: [ 20.704561] f65c0e0c f65c0c00 f6ada3f0 f642fe00 f6919ac0 f660fe34 f882e15a f65c0e0c [ 20.704561] 00000200 f6ada3f0 f6ada3c0 f6667000 f660fe3c f882e254 f660fe70 f87cb9e9 [ 20.704561] f680db00 0160fe70 ffffffff f64cb870 f6919ac0 00001000 00000020 00000020 [ 20.704561] Call Trace: [ 20.704561] [<f882e15a>] ? snd_pcm_proc_info_read+0x80/0x16c [snd_pcm] [ 20.704561] [<f882e254>] ? snd_pcm_stream_proc_info_read+0xe/0x10 [snd_pcm] [ 20.704561] [<f87cb9e9>] ? snd_info_entry_open+0x26b/0x2de [snd] [ 20.704561] [<c01df157>] ? proc_reg_open+0xb6/0x122 [ 20.704561] [<f87cb6ca>] ? snd_info_entry_release+0x0/0xa7 [snd] [ 20.704561] [<c01ac441>] ? __dentry_open+0x119/0x207 [ 20.704561] [<c01ac5c9>] ? nameidata_to_filp+0x2c/0x43 [ 20.704561] [<c01df0a1>] ? proc_reg_open+0x0/0x122 [ 20.704561] [<c01b656b>] ? do_filp_open+0x3ae/0x6c8 [ 20.704561] [<c019ff47>] ? free_pages_and_swap_cache+0x44/0x57 [ 20.704561] [<c01b6937>] ? getname+0x20/0xb7 [ 20.704561] [<c01be23c>] ? alloc_fd+0x55/0xbe [ 20.704561] [<c01ac253>] ? do_sys_open+0x44/0xba [ 20.704561] [<c01ac30b>] ? sys_open+0x1e/0x26 [ 20.704561] [<c01031b4>] ? sysenter_do_call+0x12/0x28 [ 20.704561] Code: 00 00 00 89 e5 57 56 53 89 d3 83 ec 08 89 df 89 45 ec 8b 00 8b 55 ec 89 45 f0 31 c0 8b 72 04 f3 ab 8b 55 f0 b9 40 00 00 00 8b 02 <8b> 00 89 43 0c 8b 42 0c 8b 55 ec 89 03 8b 42 30 89 43 08 8b 42· [ 20.704561] EIP: [<f8830abf>] snd_pcm_info+0x2e/0xe0 [snd_pcm] SS:ESP 0068:f660fe04 [ 21.199052] CR2: 0000000000000002 [ 21.222000] ---[ end trace 000b75f3b1903e43 ]--- [ 20.606741] EIP: 0060:[<f8830abf>] EFLAGS: 00010246 CPU: 1 [ 20.704561] EIP is at snd_pcm_info+0x2e/0xe0 [snd_pcm] [ 20.704561] EAX: 00000002 EBX: f642fe00 ECX: 00000040 EDX: f65c0c00 [ 20.704561] ESI: 00000000 EDI: f642ff20 EBP: f660fe18 ESP: f660fe04 [ 20.704561] DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068 The first dereference is happening at: (gdb) list *snd_pcm_timer_init+0x26 0x7cf8 is in snd_pcm_timer_init (/var/pisi/module-alsa-driver-1.0.20_20090808-49/work/alsa-driver/acore/../alsa-kernel/core/pcm_timer.c:131). 126 struct snd_timer_id tid; 127 struct snd_timer *timer; 128 129 tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE; 130 tid.dev_class = SNDRV_TIMER_CLASS_PCM; 131 tid.card = substream->pcm->card->number; <---------- 132 tid.device = substream->pcm->device; 133 tid.subdevice = (substream->number << 1) | (substream->stream & 1); 134 if (snd_timer_new(substream->pcm->card, "PCM", &tid, &timer) < 0) 135 return;at the following place: Can it be related to a race condition between the sound card and the saa7134-alsa interface?

2 7

[alsa-devel] issue at 44.1 Khz capturing
by Harsha, Priya 10 Aug '09

10 Aug '09

Hi, I am using arecord for capturing data. When I try to record at 8 or 48 KHz, it's perfectly fine. But when I try to record at 44.1 Khz, there are spikes in the recorded data for every buffer. When I tried to tap the data captured at the firmware/hardware end, it's smooth but the same data when is checked from the user space after being recorded, it has spikes. On further analysis, looks like for every buffer recorded there is either 1 sample recorded extra or 1 sample dropped. But when I printed out the values passed from the pointer function, they are exactly at the intervals of 5513 bytes. Can anyone give me any clue as to what is going wrong? I am using indirect_* functions in the driver code. Is there any buffer size round off that is giving this error? Can I handle something in the driver to avoid this? Thanks, Harsha.

3 4

[alsa-devel] Implementing sync start
by Jon Smirl 10 Aug '09

10 Aug '09

I'm implementing sync start as Takashi described... --------------- If the hardware supports the simultaneous triggering of multiple streams, add the sync support to the driver. Namely, - Add SNDRV_PCM_INFO_SYNC_START flag to PCM info field - Call snd_pcm_set_sync(substream) at each open callback - In the trigger callback, check all bundled streams via snd_pcm_group_for_each_entry(). Mark all the streams but the current one via snd_pcm_trigger_done(). Then start all streams at once. Ditto for stop and pause. ------------- Is there a simple way to test this with something like a mono sine wav file? A aplay mono.wav and then use and ALSA conf to copy the sine to all channels and then sync start them. ----------- brutefit has this routine for sync start, does this look correct? int bfio_synch_start(void) { snd_pcm_status_t *status; int err, n; if (base_handle == NULL) { return 0; } /* FIXME: the SND_PCM_STATE_RUNNING code would not be needed if the bfio_write autostart hack was not there */ snd_pcm_status_alloca(&status); if (link_handles) { if ((err = snd_pcm_status(base_handle, status)) < 0) { fprintf(stderr, "ALSA I/O: Could not get status: %s.\n", snd_strerror(err)); return -1; } if (snd_pcm_status_get_state(status) == SND_PCM_STATE_RUNNING) { return 0; } if ((err = snd_pcm_start(base_handle)) < 0) { fprintf(stderr, "ALSA I/O: Could not start audio: %s.\n", snd_strerror(err)); return -1; } return 0; } FOR_IN_AND_OUT { for (n = 0; n < n_handles[IO]; n++) { if ((err = snd_pcm_status(handles[IO][n], status)) < 0) { fprintf(stderr, "ALSA I/O: Could not get status: %s.\n", snd_strerror(err)); return -1; } if (snd_pcm_status_get_state(status) == SND_PCM_STATE_RUNNING) { continue; } if ((err = snd_pcm_start(handles[IO][n])) < 0) { fprintf(stderr, "ALSA I/O: Could not start audio: %s.\n", snd_strerror(err)); return -1; } } } return 0; } -- Jon Smirl jonsmirl(a)gmail.com

3 12

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 10 Aug '09

10 Aug '09

The following changes since commit b7b8f9bf0cd73b90561e8123fd5ec28f4539c419: Janusz Krzysztofik (1): TTY/ASoC: Rename N_AMSDELTA line discipline to N_V253 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 Chaithrika U S (1): ASoC: DaVinci: Support Audio on DA830 EVM Mark Brown (3): ASoC: Convert WM8776 to use factored out register cache code ASoC: Define more formats for the AC97 CODECs ASoC: Drop unneeded declaration of removed wm8731 SPI write function Uwe Kleine-König (1): ASoC: s3c2443-ac97: convert semaphore to mutex include/sound/soc-dai.h | 8 ++++- sound/soc/codecs/wm8731.c | 3 -- sound/soc/codecs/wm8776.c | 59 +++++++------------------------------ sound/soc/davinci/Kconfig | 11 +++++++ sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 33 +++++++++++++++++++++ sound/soc/s3c24xx/s3c2443-ac97.c | 10 +++--- 7 files changed, 68 insertions(+), 57 deletions(-)

2 1

[alsa-devel] load_for_all_cards bug
by Kevin Goodsell 10 Aug '09

10 Aug '09

(I originally tried to send this via the moderator without subscribing to the list, but it never went through and I never received a moderator response. Apparently this list is set up as moderated, but nobody is actually moderating it.) Summary: load_for_all_cards fails with existing configuration elements In snd_config_hook_load_for_all_cards, the first call to snd_config_search attempts to locate an existing configuration node with the name of the driver. Typically none is found, and everything is good. However, if such a node is located, the next line assumes it is a leaf node with type 'string' and calls snd_config_get_string to fetch the string value. If this fails, the entire hook is abandoned. Because of this, setting something like the following in asoundrc: cards.<driver name>.foo 0 is sufficient to disable the entire card-specific configuration. As a concrete example, I have a HDA-Intel sound card. dmix.conf includes a way to set period_size, period_time, and periods by using configuration elements of the form cards.<driver name>.pcm.dmix.<var>. In ~/.asoundrc I add cards.HDA-Intel.pcm.dmix.period_size 1024 This will cause HDA-Intel.conf to fail to load, and the pcm defined in default.conf will fail to find the device-specific pcm cards.HDA-Intel.pcm.default, and fall back on the default pcm using plughw. By attempting to configure dmix, I have disabled it. Fixing this may just be a matter of changing the line in snd_config_hook_load_for_all_cards from: if (snd_config_search(root, fdriver, &n) >= 0) { to: if (snd_config_search(root, fdriver, &n) >= 0 && n->type != SND_CONFIG_TYPE_COMPOUND) { I have not tested this. -Kevin

2 1

[alsa-devel] [RFC][PATCH 0/3] Enhance ASoC on OMAP1510 by implementing DMA channel linking
by Janusz Krzysztofik 10 Aug '09

10 Aug '09

Current implementation of ASoC OMAP PCM makes use of DMA channel self linking feature for all machines but OMAP1510. This is caused by channel linking not supported on OMAP1510 in current OMAP DMA implementation. Instead, DMA interrputs are counted as periods, and after all periods in a buffer has been detected, DMA transfer is started again for a new buffer content. The period counter is also used as a software replacement for a hardware DMA progress counter that has been found unusable on OMAP1510 in case of playback. In my attempts to debug random PCM hangups, I was wondering what could happen if a single interrupt was missing, ie. the period index was not incremented for a period. It seems that the transfer would just hang after the current buffer processed, as DMA would not be started again, waiting for a missing period. Furthermore, software calculated PCM pointer accuracy can suffer from missing DMA interrupts. Even if OMAP1510 DMA hardware does not support channel linking in a way like newer models do, it has some other features that could be used for the purpose. Interrupt status flags, passed to OMAP PCM interrupt handler routine, can be used to get more accurate DMA progress counter software calculation. Even if these enhancements did not help me in eliminating random PCM hangups completely, I find them worth of sharing. There are three patches in the series: [RFC][PATCH 1/3] ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510 [RFC][PATCH 2/3] ASoC: OMAP: Make use of DMA channel self linking on OMAP1510 [RFC][PATCH 3/3] ASoC: OMAP: Enhance OMAP1510 DMA progress software counter Created against linux-2.6.31-rc5. Tested on Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl>

1 0

[alsa-devel] alsamixer -a basic
by Elimar Riesebieter 10 Aug '09

10 Aug '09

alsa-lib 1.0.20 running alsamixer -a basic gives: ALSA lib python.c:991:(alsa_mixer_simple_finit) Unable to find python module '/usr/lib/alsa-lib/smixer/python/main.py' How do I build smixer/python/main.py ? Elimar -- On the keyboard of life you have always to keep a finger at the escape key;-)

2 1

Re: [alsa-devel] [Pkg-alsa-devel] Bug#539454: libasound2: Division by zero in pcm_rate.c when period_size->min == 0
by Elimar Riesebieter 10 Aug '09

10 Aug '09

forwarded 539454 alsa-devel(a)alsa-project.org tags 539454 + patch thanks Hi, this is Debian bug 539454. It would be better to fix it in general. Patching the Debian Package isn't a reasonable solution. Thanks Elimar * John Lindgren [090731 22:29 -0400] > Package: libasound2 > Version: 1.0.20-2 > Severity: important > Tags: patch > > > System: Dell Inspiron 1200 > Card: Intel ICH6 > Chip: SigmaTel STAC9752,53 > > > Steps to reproduce: > > 1. Disable dmix: mv /usr/share/alsa/cards/ICH4.conf /usr/share/alsa/cards/ICH4.conf.0 > 2. Restart ALSA: /etc/rcS.d/S50alsa-utils restart > 3. Clear Audacity configuration: rm -r ~/.audacity* > 4. Run Audacity 1.3.7-3 in Valgrind: valgrind --tool=none audacity > 5. Choose "OK" in the "Audacity First Run" dialog. > > > Valgrind output: > > ==26237== Process terminating with default action of signal 8 (SIGFPE) > ==26237== Integer divide by zero at address 0x64D1CD56 > ==26237== at 0x4CAD600: snd_pcm_rate_hw_refine_cchange (pcm_rate.c:189) > ==26237== by 0x4C97B2E: snd1_pcm_hw_refine_slave (pcm_params.c:2182) > ==26237== by 0x4CAD7D5: snd_pcm_rate_hw_refine (pcm_rate.c:222) > ==26237== by 0x4C96015: snd_pcm_hw_refine (pcm_params.c:2264) > ==26237== by 0x4CA116B: snd1_pcm_generic_hw_refine (pcm_generic.c:98) > ==26237== by 0x4C97B0D: snd1_pcm_hw_refine_slave (pcm_params.c:2158) > ==26237== by 0x4CC0E25: snd_pcm_lfloat_hw_refine (pcm_lfloat.c:261) > ==26237== by 0x4C96015: snd_pcm_hw_refine (pcm_params.c:2264) > ==26237== by 0x4C99068: sndrv_pcm_hw_params (pcm_params.c:2290) > ==26237== by 0x4CB209C: snd_pcm_plug_hw_params (pcm_plug.c:1045) > ==26237== by 0x4C9921D: sndrv_pcm_hw_params (pcm_params.c:2299) > ==26237== by 0x4C947B0: snd_pcm_hw_params (pcm.c:822) > > > Diagnosis: > > pcm_rate.c:189 reads: > > if ((buffer_size->min / period_size->min) * period_size->min == buffer_size->min) { > > When using the ICH4 driver without dmix, period_size->min == 0. Handle > this by changing pcm_rate.c:189 to: > > if (period_size->min > 0 && (buffer_size->min / period_size->min) * period_size->min == buffer_size->min) { > > With this change, Audacity starts without problems. Patch against native 1.0.20: diff -Naurd a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c --- a/src/pcm/pcm_rate.c 2009-05-06 09:07:23.000000000 +0200 +++ b/src/pcm/pcm_rate.c 2009-08-06 21:34:23.000000000 +0200 @@ -186,7 +186,7 @@ if (!snd_interval_checkempty(period_size) && period_size->openmin && period_size->openmax && period_size->min + 1 == period_size->max) { - if ((buffer_size->min / period_size->min) * period_size->min == buffer_size->min) { + if (period_size->min > 0 && (buffer_size->min / period_size->min) * period_size->min == buffer_size->min) { snd_interval_set_value(period_size, period_size->min); } else if ((buffer_size->max / period_size->max) * period_size->max == buffer_size->max) { snd_interval_set_value(period_size, period_size->max); -- Experience is something you don't get until just after you need it!

2 1

[alsa-devel] [BUG] NULL pointer dereference in patch_sigmatel.c
by Ozan Çağlayan 10 Aug '09

10 Aug '09

Hi, One of our users is having a NULL ptr dereference upon loading the snd_hda_intel module with 20090624's snapshot. There's only one commit after that date in patch_sigmatel.c so I didn't tell him to try with the latest snapshot but if you think that the bug may be related to another part of the ALSA codebase, I can make him try the latest snapshot. I'm attaching the alsa-info output on a working system with 1.0.18a. The kernel version in subject is 2.6.25.20. Thanks, Ozan ------------------------ First the BUG's trace: ------------------------ BUG: unable to handle kernel NULL pointer dereference at 00000074 IP: [<f8cdb83a>] :snd_hda_codec_idt:stac92xx_add_jack+0x84/0x9e *pde = 00000000 Oops: 0002 [#1] SMP Modules linked in: snd_hda_codec_idt snd_hda_intel(+) snd_hda_codec aes_i586 aes_generic ipv6 af_packet bridge bnep rfcomm l2cap microcode acpi_cpufreq cpufreq_powersave cpufreq_userspace cpufreq_conservative ndiswrapper vboxdrv pcmcia snd_hwdep rtc_cmos rtc_core yenta_socket tpm_infineon snd_seq_dummy rtc_lib tpm snd_seq_oss rsrc_nonstatic tpm_bios tifm_7xx1 iTCO_wdt snd_seq_midi_event tifm_core i2c_i801 snd_seq iTCO_vendor_support pcmcia_core r5u870 arc4 joydev snd_seq_device snd_pcm_oss nvidia(P) battery sg usbcam snd_mixer_oss ecb ac videobuf_dma_sg videobuf_core sony_laptop snd_pcm uvcvideo iwl4965 video compat_ioctl32 snd_timer output hci_usb iwlcore videodev thermal bluetooth snd processor v4l1_compat rfkill button led_class soundcore firmware_class snd_page_alloc mac80211 intel_agp i2c_core sky2 cfg80211 agpgart ext3 jbd mbcache sr_mod cdrom sd_mod ata_piix uhci_hcd pata_acpi ehci_hcd usbcore ohci1394 ieee1394 ata_generic libata scsi_mod dock Pid: 1977, comm: modprobe Tainted: P (2.6.25.20-114 #1) EIP: 0060:[<f8cdb83a>] EFLAGS: 00210206 CPU: 0 EIP is at stac92xx_add_jack+0x84/0x9e [snd_hda_codec_idt] EAX: 00000000 EBX: f8c7973e ECX: 00000004 EDX: f8cdeb1a ESI: 03211020 EDI: f675c600 EBP: f68d7d20 ESP: f68d7cd4 DS: 007b ES: 007b FS: 00d8 GS: 0033 SS: 0068 Process modprobe (pid: 1977, ti=f68d6000 task=f69ece60 task.ti=f68d6000) Stack: f68d7cf4 f8cdeb0a f8c796d5 f8c7973e f8c79793 00000001 000a69e0 f8c79793 4f205048 61207475 78452074 654c2074 4a207466 006b6361 f6898800 f6898800 00000000 f6898800 f6898800 f68d7d4c f8cdbac0 f65ef000 f6970000 00000001 Call Trace: [<f8cdbac0>] ? stac92xx_build_controls+0x26c/0x2d0 [snd_hda_codec_idt] [<f8c73aa9>] ? snd_hda_codec_build_controls+0x31/0x3d [snd_hda_codec] [<f8c750d3>] ? snd_hda_build_controls+0x18/0x67 [snd_hda_codec] [<f8c69138>] ? azx_probe+0x79e/0x840 [snd_hda_intel] [<f8c68857>] ? azx_send_cmd+0x0/0x101 [snd_hda_intel] [<f8c686a6>] ? azx_get_response+0x0/0x1b1 [snd_hda_intel] [<f8c67ece>] ? azx_attach_pcm_stream+0x0/0x15c [snd_hda_intel] [<f8c67b84>] ? azx_bus_reset+0x0/0x56 [snd_hda_intel] [<f8c67a40>] ? azx_power_notify+0x0/0x57 [snd_hda_intel] [<c01e7a37>] ? pci_device_probe+0x39/0x59 [<c024395f>] ? driver_probe_device+0xa0/0x136 [<c0243a50>] ? __driver_attach+0x5b/0x91 [<c024333c>] ? bus_for_each_dev+0x3b/0x63 [<c0243804>] ? driver_attach+0x14/0x16 [<c02439f5>] ? __driver_attach+0x0/0x91 [<c0242d3a>] ? bus_add_driver+0x9d/0x1ba [<c0243bc4>] ? driver_register+0x47/0xa7 [<c0168681>] ? __vunmap+0x93/0x9b [<c01e7bec>] ? __pci_register_driver+0x35/0x61 [<f8afd017>] ? alsa_card_azx_init+0x17/0x19 [snd_hda_intel] [<c0141f9c>] ? sys_init_module+0x18ad/0x19ca [<c0109c77>] ? do_syscall_trace+0x138/0x17f [<c0104a2e>] ? syscall_call+0x7/0xb ======================= Code: f9 ff 89 45 d0 89 f0 e8 29 68 f9 ff 89 c3 89 f0 e8 32 68 f9 ff ff 75 d0 53 50 68 0a eb cd f8 8d 45 d4 50 e8 45 25 50 c7 8b 47 08 <89> 78 74 8b 47 08 83 c4 14 c7 40 78 7f a2 cd f8 31 c0 8d 65 f4 EIP: [<f8cdb83a>] stac92xx_add_jack+0x84/0x9e [snd_hda_codec_idt] SS:ESP 0068:f68d7cd4 ---[ end trace 4a628a5b4fc302e7 ]--- alsa-info output: ------------------- upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.52 !!################################ !!Script ran on: Sun Feb 8 19:15:27 EET 2009 !!Linux Distribution !!------------------ Pardus 2008.2 Canis aureus !!Kernel Information !!------------------ Kernel release: 2.6.25.20-114 Operating System: GNU/Linux Architecture: i686 Processor: Intel(R) Core(TM)2 Duo CPU T8100 @ 2.10GHz SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.18a Library version: 1.0.18 Utilities version: 1.0.18 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Soundcards recognised by ALSA !!----------------------------- 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfc300000 irq 21 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) 09:04.2 Mass storage controller: Texas Instruments 5-in-1 Multimedia Card Reader (SD/MMC/MS/MS PRO/xD) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 104d:9008 !!Modprobe options (Sound related) !!-------------------------------- snd-hda-intel: model=vaio !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 1,-1,-1,-1,-1,-1,-1,-1 enable : Y,Y,Y,Y,Y,Y,Y,Y enable_msi : 0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1 model : vaio,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1 single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: SigmaTel STAC9872AK Address: 0 Vendor Id: 0x83847662 Subsystem Id: 0x104d1e00 Revision Id: 0x100201 No Modem Function Group found Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 GPIO: io=5, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x7f 0x7f] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x03 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x04 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x05 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x7f 0x7f] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x06 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x07 Processing caps: benign=0, ncoeff=0 Node 0x07 [Audio Selector] wcaps 0x300903: Stereo Amp-In R/L Amp-In caps: N/A Amp-In vals: [0x00 0x00] Connection: 1 0x0e Node 0x08 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x09 Processing caps: benign=0, ncoeff=0 Node 0x09 [Audio Selector] wcaps 0x300903: Stereo Amp-In R/L Amp-In caps: N/A Amp-In vals: [0x0d 0x0d] Connection: 1 0x15 Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x03211020: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x80: HP VREF_HIZ Unsolicited: tag=04, enabled=1 Connection: 1 0x02 Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x04 Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x03 Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x03a15030: [Jack] Mic at Ext Left Conn = 1/8, Color = Red DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 1 0x02 Node 0x0e [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00000024: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x05 Node 0x10 [Audio Output] wcaps 0x40211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x3e0]: 44100 48000 88200 96000 176400 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x11 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Connection: 2 0x10* 0x09 Node 0x12 [Audio Input] wcaps 0x140311: Stereo Digital Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Connection: 1 0x13 Node 0x13 [Pin Complex] wcaps 0x440381: Stereo Digital Pincap 0x00000034: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Delay: 4 samples Connection: 1 0x18 Node 0x14 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x90a7013e: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x3, Sequence = 0xe Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x15 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=1 Amp-Out vals: [0x00 0x00] Connection: 4 0x0a 0x0d 0x14* 0x02 Node 0x16 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] Node 0x17 [Volume Knob Widget] wcaps 0x600000: Mono Volume-Knob: delta=1, steps=127, direct=0, val=127 Connection: 4 0x02* 0x03 0x04 0x05 Node 0x18 [Audio Output] wcaps 0x40201: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 Delay: 4 samples Codec: Conexant ID 2c06 Address: 1 Vendor Id: 0x14f12c06 Subsystem Id: 0x104d1700 Revision Id: 0x100000 Modem Function Group: 0x2 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 0 Şub 8 2009 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 4 Şub 8 2009 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 5 Şub 8 2009 /dev/snd/hwC0D1 crw-rw----+ 1 root audio 116, 24 Şub 8 19:14 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 16 Şub 8 19:15 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 1 Şub 8 2009 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Şub 8 2009 /dev/snd/timer !!ALSA configuration files !!------------------------ !!System wide config file (/etc/asound.conf) # PulseAudio plugin configuration # Let's create a virtual device "pulse" for mixer and PCM pcm.pulse { type pulse hint { description "PulseAudio Sound Server" } } ctl.pulse { type pulse hint { description "PulseAudio Sound Server" } } # Let's make it the default! pcm.!default { type pulse hint { description "Default" } } ctl.!default { type pulse hint { description "Default" } } !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Intel] Card hw:0 'Intel'/'HDA Intel at 0xfc300000 irq 21' Mixer name : 'SigmaTel STAC9872AK' Components : 'HDA:83847662,104d1e00,00100201 HDA:14f12c06,104d1700,00100000' Controls : 10 Simple ctrls : 7 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 127 Mono: Playback 127 [100%] [0.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 127 [100%] [0.00dB] [on] Front Right: Playback 127 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Mono Limits: Playback 0 - 255 Mono: Capture [off] Front Left: Playback 0 [0%] [-51.00dB] Front Right: Playback 0 [0%] [-51.00dB] Simple mixer control 'Mic Jack',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 15 Front Left: Capture 13 [87%] [19.50dB] [on] Front Right: Capture 13 [87%] [19.50dB] [on] Simple mixer control 'Internal Mic',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [on] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 127 [100%] [0.00dB] [on] Front Right: Playback 127 [100%] [0.00dB] [on] !!Alsactl output !!------------- --startcollapse-- state.Intel { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 127' comment.dbmin -9525 comment.dbmax 0 iface MIXER name 'Headphone Playback Volume' value.0 127 value.1 127 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 127' comment.dbmin -9525 comment.dbmax 0 iface MIXER name 'Speaker Playback Volume' value.0 127 value.1 127 } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Speaker Playback Switch' value.0 true value.1 true } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 15' comment.dbmin 0 comment.dbmax 2250 iface MIXER name 'Capture Volume' value.0 13 value.1 13 } control.6 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 true value.1 true } control.7 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 'Mic Jack' comment.item.1 'Internal Mic' comment.item.2 PCM iface MIXER name 'Capture Source' value 'Internal Mic' } control.8 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 127' comment.dbmin -9525 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value 127 } control.9 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Playback Switch' value true } control.10 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' comment.dbmin -5100 comment.dbmax 0 iface MIXER name 'PCM Playback Volume' value.0 0 value.1 0 } } --endcollapse-- !!All Loaded Modules !!------------------ Module aes_i586 aes_generic ipv6 bridge bnep af_packet rfcomm l2cap microcode acpi_cpufreq cpufreq_powersave cpufreq_userspace ndiswrapper r5u870 usbcam videobuf_dma_sg snd_hda_intel snd_seq_dummy snd_seq_oss snd_seq_midi_event videobuf_core nvidia snd_seq uvcvideo snd_seq_device snd_pcm_oss arc4 compat_ioctl32 ecb snd_mixer_oss intel_agp snd_pcm iwl4965 snd_timer snd_page_alloc snd_hwdep snd pcmcia hci_usb rtc_cmos bluetooth yenta_socket rsrc_nonstatic tpm_infineon pcmcia_core tifm_7xx1 sky2 battery videodev iTCO_wdt agpgart ac thermal processor iwlcore rfkill led_class firmware_class i2c_i801 mac80211 video output tpm tpm_bios rtc_core rtc_lib joydev iTCO_vendor_support tifm_core sony_laptop v4l1_compat i2c_core sg button soundcore cfg80211 ext3 jbd mbcache sr_mod cdrom sd_mod ata_piix ohci1394 ieee1394 uhci_hcd pata_acpi ata_generic libata scsi_mod dock ehci_hcd usbcore

3 24

[alsa-devel] [PATCH 1/2] ASoC: DaVinci: McASP driver enhacements
by Chaithrika U S 10 Aug '09

10 Aug '09

On DA830/OMAPL137 SoC, the McASP peripheral has internal FIFO support. It provides additional data buffering. It also provides tolerance to vairiation 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 inturn 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 member has a version field which differentiates the McASP on different SoCs. The patch also adds another DAI format to the driver. Signed-off-by: Chaithrika U S <chaithrika(a)ti.com> --- 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 + 4 files changed, 115 insertions(+), 8 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index f0c0347..eca22d7 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 +443,16 @@ 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_CFS: + /* codec is clock master and frame slave */ + mcasp_set_bits(base + DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE); + mcasp_set_bits(base + DAVINCI_MCASP_TXFMCTL_REG, AFSXE); + mcasp_set_bits(base + DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE); + mcasp_set_bits(base + DAVINCI_MCASP_RXFMCTL_REG, AFSRE); + + mcasp_set_bits(base + DAVINCI_MCASP_PDIR_REG, (0x2d << 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 +556,8 @@ 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 +578,37 @@ 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 +621,8 @@ 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 +702,16 @@ 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 +738,13 @@ 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 +873,9 @@ 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 +905,9 @@ 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 +925,13 @@ 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 8fd0c3c..1e2dfd0 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

4 4

[alsa-devel] ASoC: PCM formats for AC97
by Jon Smirl 09 Aug '09

09 Aug '09

Tell me if this is right and I'll do a patch.... ----------------------------------------------------------- The current mpc5200 code has: static const struct snd_pcm_hardware psc_dma_hardware = { .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, struct snd_soc_dai psc_ac97_dai[] = { { .playback = { .formats = SNDRV_PCM_FMTBIT_S32_BE, I think I have this backwards, the ac97 driver can take 8/16/24/32 it's the Bestcomm program that doesn't know about anything except 32b. So if we wrote new Bestcomm programs that expanded bytes from memory out into 32b DMA register writes, we could support 8/16b formats. The correct formats would be: static const struct snd_pcm_hardware psc_dma_hardware = { .formats = SNDRV_PCM_FMTBIT_S32_BE, struct snd_soc_dai psc_ac97_dai[] = { { .playback = { .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, After implementing new Bestcomm programs.... static const struct snd_pcm_hardware psc_dma_hardware = { .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE, ----------------------------------------------------------- A related issue in soc-dai.h: #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S32_LE |\ SNDRV_PCM_FMTBIT_S32_BE) shouldn't this be #define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16 |\ SNDRV_PCM_FMTBIT_S32) The first form says the AC97 hardware can take both endians, but it doesn't take both endians it takes the endian matching the machine it is attached to. Don't the standard AC97 formats also include 8 and 24b? -- Jon Smirl jonsmirl(a)gmail.com

3 6

[alsa-devel] Testers wanted: New SB X-Fi driver
by Frashman 09 Aug '09

09 Aug '09

Hi Takashi, first of all, thank you for developing this driver. I tested it today. It works, so far. But two annoying things. As you mentioned, I got problems, if the onboard sound modules are loaded. Then i get no sound out auf my xfi. If I disable the loading of snd-hda-intel modules while booting, then it works. The second problem is, that SIDE and SURROUND connectors are interchanged. I'm absolutly sure, it works fine under windows and the connector colors are also correct. If I run speaker-test -c8 the SIDE Speakers sounds come out of my REAR (SURROUND) Speakers, while nothing is coming out of my rear speakers, if i'm using speaker-test -c6 Here lspci: $ lspci 00:00.0 Host bridge: ATI Technologies Inc RD780 Northbridge only dual slot PCI-e_GFX and HT1 K8 part 00:02.0 PCI bridge: ATI Technologies Inc RD790 PCI to PCI bridge (external gfx0 port A) 00:04.0 PCI bridge: ATI Technologies Inc RD790 PCI to PCI bridge (PCI express gpp port A) 00:0a.0 PCI bridge: ATI Technologies Inc RD790 PCI to PCI bridge (PCI express gpp port F) 00:12.0 SATA controller: ATI Technologies Inc SB600 Non-Raid-5 SATA 00:13.0 USB Controller: ATI Technologies Inc SB600 USB (OHCI0) 00:13.1 USB Controller: ATI Technologies Inc SB600 USB (OHCI1) 00:13.2 USB Controller: ATI Technologies Inc SB600 USB (OHCI2) 00:13.3 USB Controller: ATI Technologies Inc SB600 USB (OHCI3) 00:13.4 USB Controller: ATI Technologies Inc SB600 USB (OHCI4) 00:13.5 USB Controller: ATI Technologies Inc SB600 USB Controller (EHCI) 00:14.0 SMBus: ATI Technologies Inc SBx00 SMBus Controller (rev 14) 00:14.1 IDE interface: ATI Technologies Inc SB600 IDE 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) 00:14.3 ISA bridge: ATI Technologies Inc SB600 PCI to LPC Bridge 00:14.4 PCI bridge: ATI Technologies Inc SBx00 PCI to PCI Bridge 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control 01:00.0 VGA compatible controller: ATI Technologies Inc RV770 [Radeon HD 4850] 01:00.1 Audio device: ATI Technologies Inc HD48x0 audio 02:00.0 Audio device: Creative Labs X-Fi Titanium series [EMU20k2] (rev 03) 03:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168B PCI Express Gigabit Ethernet controller (rev 01) 04:06.0 Multimedia video controller: Brooktree Corporation Bt878 Video Capture (rev 11) 04:06.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 11) 04:0e.0 FireWire (IEEE 1394): Texas Instruments TSB43AB23 IEEE-1394a-2000 Controller (PHY/Link) Thank you so far, Frashman

7 33

[alsa-devel] Auzentech X-Fi Prelude with alsa drivers
by Adrian Patzak 09 Aug '09

09 Aug '09

Dear sirs, I've got a Auzentech X-Fi Prelude and I installed the alsa unstable snapshot drivers. Stereo sound for my headphones and ditital output are working fine but my microphone isn't recording anything. I was checking the mixer settings but i wasn't able to find something wrong. Are the x-fi drivers just not designed to fit with the Auzentech card or do i have a local configuration problem? The output of alsa-info.sh script is attached. Thanking you in anticipation for any answer. upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.57 !!################################ !!Script ran on: Fri Aug 7 10:48:57 UTC 2009 !!Linux Distribution !!------------------ Ubuntu 9.04 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 9.04" !!DMI Information !!--------------- Manufacturer: FUJITSU SIEMENS Product Name: AMD690VM-FMH !!Kernel Information !!------------------ Kernel release: 2.6.28-14-generic Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.20 Library version: 1.0.18 Utilities version: 1.0.18 !!Loaded ALSA modules !!------------------- snd_ctxfi !!Sound Servers on this system !!---------------------------- Pulseaudio: Installed - Yes (/usr/bin/pulseaudio) Running - Yes ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [XFi ]: SB-XFi - Creative X-Fi Creative X-Fi 20K1 Unknown !!PCI Soundcards installed in the system !!-------------------------------------- 03:01.0 Multimedia audio controller: Creative Labs SB X-Fi !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 03:01.0 0401: 1102:0005 Subsystem: 415a:0034 !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-us122l: index=-2 snd-usb-usx2y: index=-2 snd-usb-caiaq: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd-pcsp: index=-2 !!Loaded sound module options !!-------------------------- !!Module: snd_ctxfi enable : Y,Y,Y,Y,Y,Y,Y,Y id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1 multiple : 2 reference_rate : 48000 use_system_timer : N !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 0 Aug 7 08:13 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 24 Aug 7 12:02 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 16 Aug 7 12:02 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 17 Aug 7 08:13 /dev/snd/pcmC0D1p crw-rw----+ 1 root audio 116, 18 Aug 7 08:13 /dev/snd/pcmC0D2p crw-rw----+ 1 root audio 116, 19 Aug 7 08:13 /dev/snd/pcmC0D3p crw-rw----+ 1 root audio 116, 20 Aug 7 08:13 /dev/snd/pcmC0D4p crw-rw----+ 1 root audio 116, 33 Aug 7 08:13 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY E: core-util.c: Home directory /home/adrian not ours. **** List of PLAYBACK Hardware Devices **** card 0: XFi [Creative X-Fi], device 0: ctxfi [Front/WaveIn] 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: XFi [Creative X-Fi], device 1: ctxfi [Surround] 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: XFi [Creative X-Fi], device 2: ctxfi [Center/LFE] 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: XFi [Creative X-Fi], device 3: ctxfi [Side] 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: XFi [Creative X-Fi], device 4: ctxfi [IEC958 Non-audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD E: core-util.c: Home directory /home/adrian not ours. **** List of CAPTURE Hardware Devices **** card 0: XFi [Creative X-Fi], device 0: ctxfi [Front/WaveIn] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [XFi] Card hw:0 'XFi'/'Creative X-Fi 20K1 Unknown' Mixer name : '20K1' Components : '' Controls : 30 Simple ctrls : 11 Simple mixer control 'Master',0 Capabilities: pvolume cvolume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 256 Capture 0 - 256 Front Left: Playback 256 [100%] [0.00dB] Capture 256 [100%] [0.00dB] Front Right: Playback 256 [100%] [0.00dB] Capture 256 [100%] [0.00dB] Simple mixer control 'PCM',0 Capabilities: pvolume cvolume cswitch cswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 256 Capture 0 - 256 Front Left: Playback 256 [100%] [0.00dB] Capture 256 [100%] [0.00dB] [off] Front Right: Playback 256 [100%] [0.00dB] Capture 256 [100%] [0.00dB] [off] Simple mixer control 'Front',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 256 Mono: Front Left: Playback 256 [100%] [0.00dB] [on] Front Right: Playback 256 [100%] [0.00dB] [on] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 256 Mono: Front Left: Playback 256 [100%] [0.00dB] [off] Front Right: Playback 256 [100%] [0.00dB] [off] Simple mixer control 'Center/LFE',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 256 Mono: Front Left: Playback 256 [100%] [0.00dB] [off] Front Right: Playback 256 [100%] [0.00dB] [off] Simple mixer control 'Side',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 256 Mono: Front Left: Playback 256 [100%] [0.00dB] [off] Front Right: Playback 256 [100%] [0.00dB] [off] Simple mixer control 'Line-in',0 Capabilities: pvolume cvolume pswitch pswitch-joined cswitch cswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 256 Capture 0 - 256 Front Left: Playback 256 [100%] [0.00dB] [off] Capture 256 [100%] [0.00dB] [off] Front Right: Playback 256 [100%] [0.00dB] [off] Capture 256 [100%] [0.00dB] [off] Simple mixer control 'Mic',0 Capabilities: pvolume cvolume cswitch cswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 256 Capture 0 - 256 Front Left: Playback 256 [100%] [0.00dB] Capture 256 [100%] [0.00dB] [on] Front Right: Playback 256 [100%] [0.00dB] Capture 256 [100%] [0.00dB] [on] Simple mixer control 'Digit-IO',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'S/PDIF-in',0 Capabilities: pvolume cvolume pswitch pswitch-joined cswitch cswitch-joined Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 256 Capture 0 - 256 Front Left: Playback 256 [100%] [0.00dB] [off] Capture 256 [100%] [0.00dB] [off] Front Right: Playback 256 [100%] [0.00dB] [off] Capture 256 [100%] [0.00dB] [off] Simple mixer control 'S/PDIF-out',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 256 Mono: Front Left: Playback 256 [100%] [0.00dB] [on] Front Right: Playback 256 [100%] [0.00dB] [on] !!Alsactl output !!------------- --startcollapse-- state.XFi { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value.0 256 value.1 256 } control.2 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'PCM Playback Volume' value.0 256 value.1 256 } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Line-in Playback Volume' value.0 256 value.1 256 } control.4 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Mic Playback Volume' value.0 256 value.1 256 } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'S/PDIF-in Playback Volume' value.0 256 value.1 256 } control.6 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'S/PDIF-out Playback Volume' value.0 256 value.1 256 } control.7 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Front Playback Volume' value.0 256 value.1 256 } control.8 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Surround Playback Volume' value.0 256 value.1 256 } control.9 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Center/LFE Playback Volume' value.0 256 value.1 256 } control.10 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Side Playback Volume' value.0 256 value.1 256 } control.11 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Master Capture Volume' value.0 256 value.1 256 } control.12 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'PCM Capture Volume' value.0 256 value.1 256 } control.13 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Line-in Capture Volume' value.0 256 value.1 256 } control.14 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Mic Capture Volume' value.0 256 value.1 256 } control.15 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 256' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'S/PDIF-in Capture Volume' value.0 256 value.1 256 } control.16 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'PCM Capture Switch' value false } control.17 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Line-in Capture Switch' value false } control.18 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Mic Capture Switch' value true } control.19 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'S/PDIF-in Capture Switch' value false } control.20 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Line-in Playback Switch' value false } control.21 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'S/PDIF-out Playback Switch' value true } control.22 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'S/PDIF-in Playback Switch' value false } control.23 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Front Playback Switch' value true } control.24 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Surround Playback Switch' value false } control.25 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Center/LFE Playback Switch' value false } control.26 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Side Playback Switch' value false } control.27 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Digit-IO Playback Switch' value true } control.28 { comment.access read comment.type IEC958 comment.count 1 iface PCM device 4 name 'IEC958 Playback Mask' value ffffffff00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000 } control.29 { comment.access 'read write' comment.type IEC958 comment.count 1 iface PCM device 4 name 'IEC958 Playback Default' value '0082000200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.30 { comment.access 'read write' comment.type IEC958 comment.count 1 iface PCM device 4 name 'IEC958 Playback PCM Stream' value '0000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } } --endcollapse-- !!All Loaded Modules !!------------------ Module nls_iso8859_1 nls_cp437 vfat fat binfmt_misc ppdev bridge stp bnep video output input_polldev sbp2 lp parport joydev snd_ctxfi snd_pcm snd_timer snd usbhid i2c_piix4 soundcore snd_page_alloc psmouse serio_raw usblp pcspkr nvidia usb_storage ohci1394 ieee1394 r8169 mii fbcon tileblit font bitblit softcursor !!ALSA/HDA dmesg !!------------------

1 0

[alsa-devel] Problems with gain controls
by James Courtier-Dutton 09 Aug '09

09 Aug '09

Hi, Does anyone know how to disable pulse from making any changes to the alsa hardware mixer controls? I would prefer pulse to use software gain controls, and not touch the alsamixer levels. pulse seems to do a very bizarre thing. The single master gain control within pulse changes both the alsamixer PCM and Master controls. Kind Regards James

2 3

[alsa-devel] Typo in soc/atmel/sam9g20_wm8731.c
by Pedro I. Sanchez 08 Aug '09

08 Aug '09

Hello, This is a patch to fix what I believe is a typo in this file. --- sound/soc/atmel/sam9g20_wm8731.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 173a239..baa42f3 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -341,7 +341,7 @@ static int __init at91sam9g20ek_init(void) } pllb = clk_get(NULL, "pllb"); - if (IS_ERR(mclk)) { + if (IS_ERR(pllb)) { printk(KERN_ERR "ASoC: Failed to get PLLB\n"); ret = PTR_ERR(mclk); goto err_mclk;

1 0

[alsa-devel] [PATCH 2/2] ASoC: DaVinci: Support Audio on DA830 EVM
by Chaithrika U S 08 Aug '09

08 Aug '09

Add support for audio on DA830 EVM- here McASP1 is interfaced to TLV320AIC3106 codec. 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 | 11 +++++++++++ sound/soc/davinci/Makefile | 1 + sound/soc/davinci/davinci-evm.c | 33 +++++++++++++++++++++++++++++++++ 3 files changed, 45 insertions(+), 0 deletions(-) diff --git a/sound/soc/davinci/Kconfig b/sound/soc/davinci/Kconfig index 6802dd5..677a538 100644 --- a/sound/soc/davinci/Kconfig +++ b/sound/soc/davinci/Kconfig @@ -41,3 +41,14 @@ config SND_DAVINCI_SOC_SFFSDR help Say Y if you want to add support for SoC audio on Lyrtech SFFSDR board. + +config SND_DA830_SOC_EVM + tristate "SoC Audio support for DA830/OMAPL137 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 + diff --git a/sound/soc/davinci/Makefile b/sound/soc/davinci/Makefile index 67be54f..5e2195f 100644 --- a/sound/soc/davinci/Makefile +++ b/sound/soc/davinci/Makefile @@ -13,4 +13,5 @@ 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_DAVINCI_SOC_SFFSDR) += snd-soc-sffsdr.o diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index f3bb6f6..a5a9212 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -54,6 +54,9 @@ 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()) + sysclk = 24576000; + else return -EINVAL; @@ -162,6 +165,14 @@ static struct snd_soc_dai_link dm6467_evm_dai[] = { .ops = &evm_ops, }, }; +static struct snd_soc_dai_link da830_evm_dai = { + .name = "TLV320AIC3X", + .stream_name = "AIC3X", + .cpu_dai = &davinci_mcasp_dai[DAVINCI_MCASP_I2S_DAI], + .codec_dai = &aic3x_dai, + .init = evm_aic3x_init, + .ops = &evm_ops, +}; /* davinci-evm audio machine driver */ static struct snd_soc_card snd_soc_card_evm = { @@ -179,6 +190,13 @@ static struct snd_soc_card dm6467_snd_soc_card_evm = { .num_links = ARRAY_SIZE(dm6467_evm_dai), }; +static struct snd_soc_card da830_snd_soc_card = { + .name = "DA830 EVM", + .dai_link = &da830_evm_dai, + .platform = &davinci_soc_platform, + .num_links = 1, +}; + /* evm audio private data */ static struct aic3x_setup_data evm_aic3x_setup = { .i2c_bus = 1, @@ -191,6 +209,11 @@ static struct aic3x_setup_data dm6467_evm_aic3x_setup = { .i2c_address = 0x18, }; +static struct aic3x_setup_data da830_evm_aic3x_setup = { + .i2c_bus = 1, + .i2c_address = 0x18, +}; + /* evm audio subsystem */ static struct snd_soc_device evm_snd_devdata = { .card = &snd_soc_card_evm, @@ -205,6 +228,13 @@ static struct snd_soc_device dm6467_evm_snd_devdata = { .codec_data = &dm6467_evm_aic3x_setup, }; +/* evm audio subsystem */ +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, +}; + static struct platform_device *evm_snd_device; static int __init evm_init(void) @@ -222,6 +252,9 @@ static int __init evm_init(void) } else if (machine_is_davinci_dm6467_evm()) { evm_snd_dev_data = &dm6467_evm_snd_devdata; index = 0; + } else if (machine_is_davinci_da830_evm()) { + evm_snd_dev_data = &da830_evm_snd_devdata; + index = 1; } else return -EINVAL; -- 1.5.6

2 1

[alsa-devel] [PATCH 2/6] s3c2443-ac97: convert semaphore to mutex
by Uwe Kleine-König 08 Aug '09

08 Aug '09

This fixes a build failure for 2.6.31-rc4-rt1 (ARCH=arm, s3c2410_defconfig): CC [M] sound/soc/s3c24xx/s3c2443-ac97.o sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: type defaults to 'int' in declaration of 'DECLARE_MUTEX' sound/soc/s3c24xx/s3c2443-ac97.c:50: warning: parameter names (without types) in function declaration sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_read': sound/soc/s3c24xx/s3c2443-ac97.c:59: error: 'ac97_mutex' undeclared (first use in this function) sound/soc/s3c24xx/s3c2443-ac97.c:59: error: (Each undeclared identifier is reported only once sound/soc/s3c24xx/s3c2443-ac97.c:59: error: for each function it appears in.) sound/soc/s3c24xx/s3c2443-ac97.c: In function 's3c2443_ac97_write': sound/soc/s3c24xx/s3c2443-ac97.c:93: error: 'ac97_mutex' undeclared (first use in this function) Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Liam Girdwood <lrg(a)slimlogic.co.uk> Cc: Ben Dooks <ben-linux(a)fluff.org> Cc: Timur Tabi <timur(a)freescale.com> Cc: Eric Miao <eric.miao(a)marvell.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/soc/s3c24xx/s3c2443-ac97.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 3f03d5d..bf16f20 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -47,7 +47,7 @@ static struct s3c24xx_ac97_info s3c24xx_ac97; static DECLARE_COMPLETION(ac97_completion); static u32 codec_ready; -static DECLARE_MUTEX(ac97_mutex); +static DEFINE_MUTEX(ac97_mutex); static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, unsigned short reg) @@ -56,7 +56,7 @@ static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, u32 ac_codec_cmd; u32 stat, addr, data; - down(&ac97_mutex); + mutex_lock(&ac97_mutex); codec_ready = S3C_AC97_GLBSTAT_CODECREADY; ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); @@ -79,7 +79,7 @@ static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, printk(KERN_ERR "s3c24xx-ac97: req addr = %02x," " rep addr = %02x\n", reg, addr); - up(&ac97_mutex); + mutex_unlock(&ac97_mutex); return (unsigned short)data; } @@ -90,7 +90,7 @@ static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg, u32 ac_glbctrl; u32 ac_codec_cmd; - down(&ac97_mutex); + mutex_lock(&ac97_mutex); codec_ready = S3C_AC97_GLBSTAT_CODECREADY; ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); @@ -109,7 +109,7 @@ static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg, ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); - up(&ac97_mutex); + mutex_unlock(&ac97_mutex); } -- 1.6.3.3

2 1

[alsa-devel] [PATCH] ASoC: SDP3430: Add support for EXTMUTE using TWL GPIO6
by Candelaria Villareal, Jorge 07 Aug '09

07 Aug '09

Board sdp3430 has hardware support for EXTMUTE using TWL4030 GPIO6 line, controlled by register INTBR_PMBR1. Machine driver takes care of enabling gpio line through i2c and codec driver manipulates the line during headset ramp up/down sequence. Signed-off-by: Jorge Eduardo Candelaria <x0107209(a)ti.com> --- sound/soc/omap/sdp3430.c | 9 +++++++++ 1 files changed, 9 insertions(+), 0 deletions(-) diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index c51594d..f7e5b74 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -24,6 +24,7 @@ #include <linux/clk.h> #include <linux/platform_device.h> +#include <linux/i2c/twl4030.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/soc.h> @@ -39,6 +40,9 @@ #include "omap-pcm.h" #include "../codecs/twl4030.h" +#define TWL4030_INTBR_PMBR1 0x0D +#define EXTMUTE(value) (value << 2) + static struct snd_soc_card snd_soc_sdp3430; static int sdp3430_hw_params(struct snd_pcm_substream *substream, @@ -280,6 +284,7 @@ static struct snd_soc_card snd_soc_sdp3430 = { static struct twl4030_setup_data twl4030_setup = { .ramp_delay_value = 3, .sysclk = 26000, + .hs_extmute = 1, }; /* Audio subsystem */ @@ -312,6 +317,10 @@ static int __init sdp3430_soc_init(void) *(unsigned int *)sdp3430_dai[0].cpu_dai->private_data = 1; /* McBSP2 */ *(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); + ret = platform_device_add(sdp3430_snd_device); if (ret) goto err1; -- 1.6.0.4

3 6

[alsa-devel] [PATCH] ASoC: MPC5200: Support for buffer wrap around
by John Bonesio 07 Aug '09

07 Aug '09

We've encountered strange behavior in the alsamixer settings using the wm9712 codec. If we unmute the headphone output and then unmute the PCM output, the headphone output gets reset to mute in the hardware register. At this point the hardware register does not match the value in the register cache. I've spent some time debugging this, and the headphone setting is set outside of any code path that would call the ac97_write() routine. As best as I can tell, there is something strange going on in hardware. I've provided this patch that works around the problem. Have any of you seen this before? Is this patch the right approach? - John The code in psc_dma_bcom_enqueue_tx() didn't account for the fact that s->runtime->control->appl_ptr can wrap around to the beginning of the buffer. This change fixes this problem. Signed-off-by: John Bonesio <bones(a)secretlab.ca> --- sound/soc/fsl/mpc5200_dma.c | 17 +++++++++++++++++ 1 files changed, 17 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index cfe0ea4..2551c58 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -70,6 +70,23 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s) { + if (s->appl_ptr > s->runtime->control->appl_ptr) { + /* + * In this case s->runtime->control->appl_ptr has wrapped around. + * Play the data to the end of the boundary, then wrap our own + * appl_ptr back around. + */ + while (s->appl_ptr < s->runtime->boundary) { + if (bcom_queue_full(s->bcom_task)) + return; + + s->appl_ptr += s->period_size; + + psc_dma_bcom_enqueue_next_buffer(s); + } + s->appl_ptr -= s->runtime->boundary; + } + while (s->appl_ptr < s->runtime->control->appl_ptr) { if (bcom_queue_full(s->bcom_task))

5 14

[alsa-devel] Fixed module builds of soc/fsl/*.c
by Takashi Iwai 07 Aug '09

07 Aug '09

JFYI, I merged the patch below now. Takashi --- From afc5e65245255a268ab22a20477ed2c9f2cdfcd3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai <tiwai(a)suse.de> Date: Fri, 7 Aug 2009 16:33:53 +0200 Subject: ASoC: Add missing DRV_NAME definitions for fsl/* drivers Module builds are broken due to missing DRV_NAME for efika-audio-fabric and pcm030-audio-fabric. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/soc/fsl/efika-audio-fabric.c | 2 ++ sound/soc/fsl/pcm030-audio-fabric.c | 2 ++ 2 files changed, 4 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/efika-audio-fabric.c b/sound/soc/fsl/efika-audio-fabric.c index 85b0e75..3326e2a 100644 --- a/sound/soc/fsl/efika-audio-fabric.c +++ b/sound/soc/fsl/efika-audio-fabric.c @@ -30,6 +30,8 @@ #include "mpc5200_psc_ac97.h" #include "../codecs/stac9766.h" +#define DRV_NAME "efika-audio-fabric" + static struct snd_soc_device device; static struct snd_soc_card card; diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index 8766f7a..b928ef7 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -30,6 +30,8 @@ #include "mpc5200_psc_ac97.h" #include "../codecs/wm9712.h" +#define DRV_NAME "pcm030-audio-fabric" + static struct snd_soc_device device; static struct snd_soc_card card; -- 1.6.3.3

3 2

[alsa-devel] [PATCH] ARM: DaVinci: Audio support for DA830 EVM
by Chaithrika U S 07 Aug '09

07 Aug '09

Define resources for McASP1 used on DA830 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 arch/arm/mach-davinci/board-da830-evm.c | 22 +++++++++++++ arch/arm/mach-davinci/da830.c | 48 ++++++++++++++++++++++++++-- arch/arm/mach-davinci/include/mach/asp.h | 15 +++++++++ arch/arm/mach-davinci/include/mach/da8xx.h | 2 + 4 files changed, 84 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..4643795 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 +113,8 @@ 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)); + + da830_init_mcasp1(&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..a160a7f 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), @@ -1198,6 +1199,47 @@ static struct davinci_soc_info davinci_soc_info_da830 = { .emac_pdata = &da8xx_emac_pdata, }; +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, +}; + +void __init da830_init_mcasp1(struct snd_platform_data *pdata) +{ + int ret; + + ret = da8xx_pinmux_setup(da830_mcasp1_pins); + if (ret) + pr_warning("da830_evm_init: mcasp1 mux setup failed: %d\n", + ret); + + da830_mcasp1_device.dev.platform_data = pdata; + platform_device_register(&da830_mcasp1_device); +} + void __init da830_init(void) { davinci_common_init(&davinci_soc_info_da830); diff --git a/arch/arm/mach-davinci/include/mach/asp.h b/arch/arm/mach-davinci/include/mach/asp.h index cdf1f44..4df3839 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 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 */ +#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..c2aabf1 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 da830_init_mcasp1(struct snd_platform_data *pdata); extern struct platform_device da8xx_serial_device; extern struct emac_platform_data da8xx_emac_pdata; -- 1.5.6

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[alsa-devel] suspicious code in aplay.c
by Dmitri Vorobiev 07 Aug '09

07 Aug '09

Hello, The following code in alsa-utils/aplay/aplay.c looks suspicious to me: 1215 if (!quiet_mode) 1216 fprintf(stderr, _("Suspended. Trying resume. ")); fflush(stderr); Is it really necessary to flush the stderr stream even if there was no call to fprintf()? Perhaps these lines have to be like that: if (!quiet_mode) { fprintf(stderr, _("Suspended. Trying resume. ")); fflush(stderr); } Thanks, Dmitri P.S. I'm not subscribed to the mailing list, so please Cc: me on reply.

1 0

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 07 Aug '09

07 Aug '09

The following changes since commit 6d7f68a1eab3d5b3c0a6a5cf9434c77cd3a8c6ac: Janusz Krzysztofik (1): ASoC: add support for Amstrad E3 (Delta) machine 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 There's a merge up of a for-2.6.31 change in here which I've not pushed yet since the change is relatively invasive for so late in the release cycle (the OMAP simultaneous audio change). I didn't redo the branch in case it does get pushed; worst case it's just an extra merge commit which isn't the end of the world. Daniel Mack (1): ALSA: ASoC: cs4270: move power management hooks to snd_soc_codec_device Janusz Krzysztofik (3): ASoC: CX20442: add some debugging ASoC: CX20442: simplify codec controller usage TTY/ASoC: Rename N_AMSDELTA line discipline to N_V253 Jarkko Nikula (1): ARM: OMAP: McBSP: Fix ASoC on OMAP1510 by fixing API of omap_mcbsp_start/stop John Bonesio (1): ASoC: MPC5200: Support for buffer wrap around Mark Brown (12): ASoC: Begin to factor out register cache I/O functions ASoC: Remove use of hw_read from TLV320AIC3x driver ASoC: Factor out WM8580 register cache code Merge branch 'tlv320aic3x' into reg-cache ASoC: Add I/O control bus information to factored out cache setup ASoC: Factor out I2C 8 bit address 16 bit data I/O ASoC: Factor out I/O for Wolfson 8 bit data 16 bit register CODECs ASoC: Factor out 7 bit register 9 bit data SPI write ASoC: Existing S3C24xx AC97 drivers should depend on S3C24xx ASoC: Add WM8776 CODEC driver Merge branch 'for-2.6.31' into for-2.6.32 Merge branch 'reg-cache' into for-2.6.32 Troy Kisky (2): ASoC: DaVinci: i2s: don't bounce through rtd to get dai ASoC: DaVinci: pcm, constrain buffer size to multiple of period arch/arm/plat-omap/include/mach/mcbsp.h | 4 +- arch/arm/plat-omap/mcbsp.c | 50 ++- include/linux/tty.h | 2 +- include/sound/soc.h | 11 +- sound/soc/Makefile | 2 +- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4270.c | 27 +- sound/soc/codecs/cx20442.c | 3 +- sound/soc/codecs/tlv320aic3x.c | 11 +- sound/soc/codecs/wm8510.c | 174 +++----- sound/soc/codecs/wm8523.c | 98 +--- sound/soc/codecs/wm8580.c | 154 ++----- sound/soc/codecs/wm8728.c | 111 +---- sound/soc/codecs/wm8731.c | 141 ++----- sound/soc/codecs/wm8750.c | 154 ++----- sound/soc/codecs/wm8776.c | 781 +++++++++++++++++++++++++++++++ sound/soc/codecs/wm8776.h | 51 ++ sound/soc/codecs/wm8900.c | 319 +++++-------- sound/soc/codecs/wm8903.c | 244 ++++------- sound/soc/codecs/wm8940.c | 143 ++---- sound/soc/codecs/wm8960.c | 200 +++----- sound/soc/codecs/wm8961.c | 231 ++++------ sound/soc/codecs/wm8971.c | 127 ++---- sound/soc/codecs/wm8988.c | 138 ++---- sound/soc/codecs/wm8990.c | 194 +++----- sound/soc/codecs/wm9081.c | 229 ++++------ sound/soc/davinci/davinci-i2s.c | 14 +- sound/soc/davinci/davinci-pcm.c | 5 + sound/soc/fsl/mpc5200_dma.c | 17 + sound/soc/omap/ams-delta.c | 6 +- sound/soc/omap/omap-mcbsp.c | 10 +- sound/soc/s3c24xx/Kconfig | 4 +- sound/soc/soc-cache.c | 218 +++++++++ 34 files changed, 2048 insertions(+), 1831 deletions(-) create mode 100644 sound/soc/codecs/wm8776.c create mode 100644 sound/soc/codecs/wm8776.h create mode 100644 sound/soc/soc-cache.c

2 1

[alsa-devel] [PATCH 0/2] Give the N_AMSDELTA line discipline more generic name
by Janusz Krzysztofik 07 Aug '09

07 Aug '09

This series of patches renames the N_AMSDELTA line discipline, used by the Amstrad Delta ASoC machine driver and the CX20442 ASoC codec driver for controlling the CX20442 voice codec over the CX81801 voice modem, giving it a more generic name: N_V253. The name has been chosen after the voice modem standard codename V.253. I think that even if the code of the line discipline is currently implemented inside those two specific drivers, the idea, if not the code, could be reused with other similar hardware configurations, if necessary. As the line discipline is conceived to be registered from a machine driver, there should be no conflicts, even if used by several different drivers that happen to be be built-in side-by-side. [PATCH 1/2] TTY: Rename N_AMSDELTA line discipline to N_V253 [PATCH 2/2] ASoC: AMS_DELTA: Use updated line discipline name Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl>

2 1

[alsa-devel] Audio support for Simtec Hermes
by Ben Dooks 07 Aug '09

07 Aug '09

Add support for the TLV320AIC33 codec on the Simtec Hermes board, using the core work done for the TLV320AIC23 codecs. Signed-off-by: Ben Dooks <ben(a)simtec.co.uk> --- sound/soc/s3c24xx/Kconfig | 7 + sound/soc/s3c24xx/Makefile | 3 sound/soc/s3c24xx/s3c24xx_simtec_hermes.c | 151 ++++++++++++++++++++++++++++++ 3 files changed, 161 insertions(+) Index: b/sound/soc/s3c24xx/Kconfig =================================================================== --- a/sound/soc/s3c24xx/Kconfig 2009-08-04 12:27:06.000000000 +0100 +++ b/sound/soc/s3c24xx/Kconfig 2009-08-04 12:29:22.000000000 +0100 @@ -127,3 +127,10 @@ config SND_S3C24XX_SOC_SIMTEC_TLV320AIC2 select SND_S3C24XX_SOC_I2S select SND_SOC_TLV320AIC23 select SND_S3C24XX_SOC_SIMTEC + +config SND_S3C24XX_SOC_SIMTEC_HERMES + tristate "SoC I2S Audio support for Simtec Hermes board" + depends on SND_S3C24XX_SOC + select SND_S3C24XX_SOC_I2S + select SND_SOC_TLV320AIC3X + select SND_S3C24XX_SOC_SIMTEC Index: b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c 2009-08-04 12:42:00.000000000 +0100 @@ -0,0 +1,151 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec_hermes.c + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +#include "../codecs/tlv320aic3x.h" + +static const struct snd_soc_dapm_widget dapm_widgets[] = { + SND_SOC_DAPM_LINE("GSM Out", NULL), + SND_SOC_DAPM_LINE("GSM In", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_LINE("ZV", NULL), + SND_SOC_DAPM_MIC("Mic", NULL), +}; + +static const struct snd_soc_dapm_route base_map[] = { + /* Headphone connected to HP{L,R}OUT and HP{L,R}COM */ + + { "Headphone Jack", NULL, "HPLOUT" }, + { "Headphone Jack", NULL, "HPLCOM" }, + { "Headphone Jack", NULL, "HPROUT" }, + { "Headphone Jack", NULL, "HPRCOM" }, + + /* ZV connected to Line1 */ + + { "LINE1L", NULL, "ZV" }, + { "LINE1R", NULL, "ZV" }, + + /* Line In connected to Line2 */ + + { "LINE2L", NULL, "Line In" }, + { "LINE2R", NULL, "Line In" }, + + /* Microphone connected to MIC3R and MIC_BIAS */ + + { "MIC3L", NULL, "Mic" }, + + /* GSM connected to MONO_LOUT and MIC3L (in) */ + + { "GSM Out", NULL, "MONO_LOUT" }, + { "MIC3L", NULL, "GSM In" }, + + /* Speaker is connected to LINEOUT{LN,LP,RN,RP} */ +}; + +/** + * simtec_hermes_init - initialise and add controls + * @codec; The codec instance to attach to. + * + * Attach our controls and configure the necessary codec + * mappings for our sound card instance. +*/ +static int simtec_hermes_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dapm_widgets, + ARRAY_SIZE(dapm_widgets)); + + snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + simtec_audio_init(codec); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct aic3x_setup_data codec_setup = { + .i2c_bus = 0, + .i2c_already = 1, +}; + +static struct snd_soc_dai_link simtec_dai_aic33 = { + .name = "tlv320aic33", + .stream_name = "TLV320AIC33", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &aic3x_dai, + .init = simtec_hermes_init, +}; + +/* simtec audio machine driver */ +static struct snd_soc_card snd_soc_machine_simtec_aic33 = { + .name = "Simtec-Hermes", + .platform = &s3c24xx_soc_platform, + .dai_link = &simtec_dai_aic33, + .num_links = 1, +}; + +/* simtec audio subsystem */ +static struct snd_soc_device simtec_snd_devdata_aic33 = { + .card = &snd_soc_machine_simtec_aic33, + .codec_dev = &soc_codec_dev_aic3x, + .codec_data = &codec_setup, +}; + +static int __devinit simtec_audio_hermes_probe(struct platform_device *pd) +{ + dev_info(&pd->dev, "probing....\n"); + return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic33); +} + +static struct platform_driver simtec_audio_hermes_platdrv = { + .driver = { + .owner = THIS_MODULE, + .name = "s3c24xx-simtec-hermes-snd", + .pm = simtec_audio_pm, + }, + .probe = simtec_audio_hermes_probe, + .remove = __devexit_p(simtec_audio_remove), +}; + +MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd"); + +static int __init simtec_hermes_modinit(void) +{ + return platform_driver_register(&simtec_audio_hermes_platdrv); +} + +static void __exit simtec_hermes_modexit(void) +{ + platform_driver_unregister(&simtec_audio_hermes_platdrv); +} + +module_init(simtec_hermes_modinit); +module_exit(simtec_hermes_modexit); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); +MODULE_LICENSE("GPL"); Index: b/sound/soc/s3c24xx/Makefile =================================================================== --- a/sound/soc/s3c24xx/Makefile 2009-08-04 12:27:06.000000000 +0100 +++ b/sound/soc/s3c24xx/Makefile 2009-08-04 12:29:22.000000000 +0100 @@ -24,6 +24,7 @@ snd-soc-ln2440sbc-alc650-objs := ln2440s snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o +snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o @@ -36,4 +37,6 @@ obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_A obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o + -- Ben (ben(a)fluff.org, http://www.fluff.org/) 'a smiley only costs 4 bytes'

2 1

[alsa-devel] PROC_COEF/PROC_INDEX question
by Wu Fengguang 07 Aug '09

07 Aug '09

Hello, I'm not sure of the usage for GET_PROC_COEF/GET_COEF_INDEX, and found that their return value is unstable (changing all the time) for codec: Codec: Realtek ALC888 Address: 2 Function Id: 0x1 Vendor Id: 0x10ec0888 Subsystem Id: 0x80860023 Revision Id: 0x100202 For example: # hda-verb /dev/snd/hwC0D2 0x20 GET_PROC_COEF 0 nid = 0x20, verb = 0xc00, param = 0x0 value = 0x80 # hda-verb /dev/snd/hwC0D2 0x20 GET_COEF_INDEX 0 nid = 0x20, verb = 0xd00, param = 0x0 value = 0xe # hda-verb /dev/snd/hwC0D2 0x20 GET_PROC_COEF 0 nid = 0x20, verb = 0xc00, param = 0x0 value = 0x9c00 # hda-verb /dev/snd/hwC0D2 0x20 GET_COEF_INDEX 0 nid = 0x20, verb = 0xd00, param = 0x0 value = 0xf I wonder if this is the normal case? Thanks, Fengguang

2 4

[alsa-devel] Audio support for TLV320AIC23 on Simtec S3C24XX products
by Ben Dooks 07 Aug '09

07 Aug '09

Add support for the TLV320AIC23 on most of the current Simtec S3C24XX with an initial core set of routines for use with the other S3C24XX designs that do not use the TLV320AIC23. Signed-off-by: Ben Dooks <ben(a)simtec.co.uk> --- arch/arm/common/sbcd/s3c24xx/dev-audio.c | 146 ++++++--- arch/arm/plat-s3c/include/plat/audio-simtec.h | 37 ++ arch/arm/plat-s3c24xx/audio-simtec.c | 97 +----- sound/soc/s3c24xx/Kconfig | 12 sound/soc/s3c24xx/Makefile | 4 sound/soc/s3c24xx/s3c24xx_simtec.c | 391 +++++++++++++++++++++++++ sound/soc/s3c24xx/s3c24xx_simtec.h | 22 + sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c | 137 ++++++++ 8 files changed, 728 insertions(+), 118 deletions(-) Index: b/sound/soc/s3c24xx/Kconfig =================================================================== --- a/sound/soc/s3c24xx/Kconfig 2009-08-04 12:24:09.000000000 +0100 +++ b/sound/soc/s3c24xx/Kconfig 2009-08-06 11:53:38.000000000 +0100 @@ -115,3 +115,15 @@ config SND_S3C24XX_SOC_S3C24XX_UDA134X select SND_S3C24XX_SOC_I2S select SND_SOC_L3 select SND_SOC_UDA134X + +config SND_S3C24XX_SOC_SIMTEC + tristate + help + Internal node for common S3C24XX/Simtec suppor + +config SND_S3C24XX_SOC_SIMTEC_TLV320AIC23 + tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards" + depends on SND_S3C24XX_SOC + select SND_S3C24XX_SOC_I2S + select SND_SOC_TLV320AIC23 + select SND_S3C24XX_SOC_SIMTEC Index: b/sound/soc/s3c24xx/Makefile =================================================================== --- a/sound/soc/s3c24xx/Makefile 2009-08-04 12:24:09.000000000 +0100 +++ b/sound/soc/s3c24xx/Makefile 2009-08-06 11:53:38.000000000 +0100 @@ -23,6 +23,8 @@ snd-soc-smdk2440-wm8991-objs := smdk2440 snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o +snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o +snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -33,3 +35,5 @@ obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o +obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o Index: b/sound/soc/s3c24xx/s3c24xx_simtec.c =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ b/sound/soc/s3c24xx/s3c24xx_simtec.c 2009-08-04 12:29:02.000000000 +0100 @@ -0,0 +1,391 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec.c + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#define DEBUG + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <linux/i2c.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +static struct s3c24xx_audio_simtec_pdata *pdata; +static struct clk *xtal_clk; + +static int spk_gain; +static int spk_unmute; + +/** + * speaker_gain_get - read the speaker gain setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be updated. + * + * Read the value for the AMP gain control. + */ +static int speaker_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = spk_gain; + return 0; +} + +/** + * speaker_gain_set - set the value of the speaker amp gain + * @value: The value to write. + */ +static void speaker_gain_set(int value) +{ + gpio_set_value_cansleep(pdata->amp_gain[0], value & 1); + gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1); +} + +/** + * speaker_gain_put - set the speaker gain setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be set. + * + * Set the value of the speaker gain from the specified + * @ucontrol setting. + * + * Note, if the speaker amp is muted, then we do not set a gain value + * as at-least one of the ICs that is fitted will try and power up even + * if the main control is set to off. + */ +static int speaker_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + int value = ucontrol->value.integer.value[0]; + + spk_gain = value; + + if (!spk_unmute) + speaker_gain_set(value); + + return 0; +} + +static const struct snd_kcontrol_new amp_gain_controls[] = { + SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0, + speaker_gain_get, speaker_gain_put), +}; + +/** + * spk_unmute_state - set the unmute state of the speaker + * @to: zero to unmute, non-zero to ununmute. + */ +static void spk_unmute_state(int to) +{ + pr_debug("%s: to=%d\n", __func__, to); + + spk_unmute = to; + gpio_set_value(pdata->amp_gpio, to); + + /* if we're umuting, also re-set the gain */ + if (to && pdata->amp_gain[0] > 0) + speaker_gain_set(spk_gain); +} + +/** + * speaker_unmute_get - read the speaker unmute setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be updated. + * + * Read the value for the AMP gain control. + */ +static int speaker_unmute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = spk_unmute; + return 0; +} + +/** + * speaker_unmute_put - set the speaker unmute setting. + * @kcontrol: The control for the speaker gain. + * @ucontrol: The value that needs to be set. + * + * Set the value of the speaker gain from the specified + * @ucontrol setting. + */ +static int speaker_unmute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + spk_unmute_state(ucontrol->value.integer.value[0]); + return 0; +} + +static const struct snd_kcontrol_new amp_unmute_controls[] = { + SOC_SINGLE_EXT("Speaker", 0, 0, 1, 0, + speaker_unmute_get, speaker_unmute_put), +}; + +void simtec_audio_init(struct snd_soc_codec *codec) +{ + if (pdata->amp_gpio > 0) { + pr_debug("%s: adding amp routes\n", __func__); + + snd_soc_add_controls(codec, amp_unmute_controls, + ARRAY_SIZE(amp_unmute_controls)); + } + + if (pdata->amp_gain[0] > 0) { + pr_debug("%s: adding amp controls\n", __func__); + snd_soc_add_controls(codec, amp_gain_controls, + ARRAY_SIZE(amp_gain_controls)); + } +} +EXPORT_SYMBOL_GPL(simtec_audio_init); + +#define CODEC_CLOCK 12000000 + +/** + * simtec_hw_params - update hardware parameters + * @substream: The audio substream instance. + * @params: The parameters requested. + * + * Update the codec data routing and configuration settings + * from the supplied data. + */ +static int simtec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + int ret; + + /* Set the CODEC as the bus clock master, I2S */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set cpu dai format\n", __func__); + return ret; + } + + /* Set the CODEC as the bus clock master */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM); + if (ret) { + pr_err("%s: failed set codec dai format\n", __func__); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, 0, + CODEC_CLOCK, SND_SOC_CLOCK_IN); + if (ret) { + pr_err( "%s: failed setting codec sysclk\n", __func__); + return ret; + } + + if (pdata->use_mpllin) { + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL, + 0, SND_SOC_CLOCK_OUT); + + if (ret) { + pr_err("%s: failed to set MPLLin as clksrc\n", + __func__); + return ret; + } + } + + if (pdata->output_cdclk) { + int cdclk_scale; + + cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK; + cdclk_scale--; + + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + cdclk_scale); + } + + return 0; +} + +static int simtec_call_startup(struct s3c24xx_audio_simtec_pdata *pd) +{ + /* call any board supplied startup code, this currently only + * covers the bast/vr1000 which have a CPLD in the way of the + * LRCLK */ + if (pd->startup) + pd->startup(); + + return 0; +} + +static struct snd_soc_ops simtec_snd_ops = { + .hw_params = simtec_hw_params, +}; + +/** + * attach_gpio_amp - get and configure the necessary gpios + * @dev: The device we're probing. + * @pd: The platform data supplied by the board. + * + * If there is a GPIO based amplifier attached to the board, claim + * the necessary GPIO lines for it, and set default values. + */ +static int attach_gpio_amp(struct device *dev, + struct s3c24xx_audio_simtec_pdata *pd) +{ + int ret; + + /* attach gpio amp gain (if any) */ + if (pdata->amp_gain[0] > 0) { + ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain0"); + if (ret) { + dev_err(dev, "cannot get amp gpio gain0\n"); + return ret; + } + + ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain1"); + if (ret) { + dev_err(dev, "cannot get amp gpio gain1\n"); + gpio_free(pdata->amp_gain[0]); + return ret; + } + + gpio_direction_output(pd->amp_gain[0], 0); + gpio_direction_output(pd->amp_gain[1], 0); + } + + /* note, curently we assume GPA0 isn't valid amp */ + if (pdata->amp_gpio > 0) { + ret = gpio_request(pd->amp_gpio, "gpio-amp"); + if (ret) { + dev_err(dev, "cannot get amp gpio %d (%d)\n", + pd->amp_gpio, ret); + goto err_amp; + } + + /* set the amp off at startup */ + spk_unmute_state(0); + } + + return 0; + +err_amp: + if (pd->amp_gain[0] > 0) { + gpio_free(pd->amp_gain[0]); + gpio_free(pd->amp_gain[1]); + } + + return ret; +} + +static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd) +{ + if (pd->amp_gain[0] > 0) { + gpio_free(pd->amp_gain[0]); + gpio_free(pd->amp_gain[1]); + } + + if (pd->amp_gpio > 0) + gpio_free(pd->amp_gpio); +} + +#ifdef CONFIG_PM +int simtec_audio_resume(struct device *dev) +{ + simtec_call_startup(pdata); + return 0; +} + +struct dev_pm_ops simtec_audio_pmops = { + .resume = simtec_audio_resume, +}; +EXPORT_SYMBOL_GPL(simtec_audio_pmops); +#endif + +int __devinit simtec_audio_core_probe(struct platform_device *pdev, + struct snd_soc_device *socdev) +{ + struct platform_device *snd_dev; + int ret; + + socdev->card->dai_link->ops = &simtec_snd_ops; + + pdata = pdev->dev.platform_data; + if (!pdata) { + dev_err(&pdev->dev, "no platform data supplied\n"); + return -EINVAL; + } + + simtec_call_startup(pdata); + + xtal_clk = clk_get(&pdev->dev, "xtal"); + if (IS_ERR(xtal_clk)) { + dev_err(&pdev->dev, "could not get clkout0\n"); + return -EINVAL; + } + + dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk)); + + ret = attach_gpio_amp(&pdev->dev, pdata); + if (ret) + goto err_clk; + + snd_dev = platform_device_alloc("soc-audio", -1); + if (!snd_dev) { + dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n"); + ret = -ENOMEM; + goto err_gpio; + } + + platform_set_drvdata(snd_dev, socdev); + socdev->dev = &snd_dev->dev; + + ret = platform_device_add(snd_dev); + if (ret) { + dev_err(&pdev->dev, "failed to add soc-audio dev\n"); + goto err_pdev; + } + + platform_set_drvdata(pdev, snd_dev); + return 0; + +err_pdev: + platform_device_put(snd_dev); + +err_gpio: + detach_gpio_amp(pdata); + +err_clk: + clk_put(xtal_clk); + return ret; +} + +int __devexit simtec_audio_remove(struct platform_device *pdev) +{ + struct platform_device *snd_dev = platform_get_drvdata(pdev); + + platform_device_unregister(snd_dev); + + detach_gpio_amp(pdata); + clk_put(xtal_clk); + return 0; +} +EXPORT_SYMBOL_GPL(simtec_audio_remove); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio common support"); +MODULE_LICENSE("GPL"); Index: b/arch/arm/plat-s3c/include/plat/audio-simtec.h =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ b/arch/arm/plat-s3c/include/plat/audio-simtec.h 2009-08-04 12:24:11.000000000 +0100 @@ -0,0 +1,37 @@ +/* arch/arm/plat-s3c/include/plat/audio-simtec.h + * + * Copyright 2008 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * Simtec Audio support. +*/ + +/** + * struct s3c24xx_audio_simtec_pdata - platform data for simtec audio + * @use_mpllin: Select codec clock from MPLLin + * @output_cdclk: Need to output CDCLK to the codec + * @have_mic: Set if we have a MIC socket + * @have_lout: Set if we have a LineOut socket + * @amp_gpio: GPIO pin to enable the AMP + * @amp_gain: Option GPIO to control AMP gain + */ +struct s3c24xx_audio_simtec_pdata { + unsigned int use_mpllin:1; + unsigned int output_cdclk:1; + + unsigned int have_mic:1; + unsigned int have_lout:1; + + int amp_gpio; + int amp_gain[2]; + + void (*startup)(void); +}; + +extern int simtec_audio_add(const char *codec_name, + struct s3c24xx_audio_simtec_pdata *pdata); Index: b/arch/arm/plat-s3c24xx/audio-simtec.c =================================================================== --- a/arch/arm/plat-s3c24xx/audio-simtec.c 2009-08-04 12:05:59.000000000 +0100 +++ b/arch/arm/plat-s3c24xx/audio-simtec.c 2009-08-04 12:24:11.000000000 +0100 @@ -45,7 +45,7 @@ #include <plat/audio.h> #include <plat/regs-iis.h> -#include <asm/plat-s3c24xx/audio-simtec.h> +#include <plat/audio-simtec.h> #include <plat/devs.h> #include <plat/pm.h> @@ -56,7 +56,7 @@ static inline int has_lr_routing(void) return machine_is_bast() || machine_is_vr1000(); } -static int simtec_audio_startup(void *pw) +static void simtec_audio_startup(void) { unsigned int tmp; unsigned long flags; @@ -71,94 +71,40 @@ static int simtec_audio_startup(void *pw } local_irq_restore(flags); - - return 0; -} - -static void simtec_audio_shutdown(void *pw) -{ - unsigned int tmp; - unsigned long flags; - - /* switch off LR clock routing */ - - local_irq_save(flags); - - if (has_lr_routing()) { - tmp = __raw_readb(BAST_VA_CTRL1); - tmp &= ~BAST_CPLD_CTRL1_LRMASK; - tmp |= BAST_CPLD_CTRL1_LRCOFF; - __raw_writeb(tmp, BAST_VA_CTRL1); - } - - local_irq_restore(flags); -} - -static int simtec_audio_matchdev(struct device *dev) -{ - printk(KERN_INFO "%s: dev = %p\n", __func__, dev); - - if (dev->parent == NULL || dev->parent->parent == NULL) - return 0; - - printk(KERN_INFO "%s: %s\n", __func__, dev_name(dev->parent)); - if (strncmp(dev_name(dev->parent), "i2c", strlen("i2c")) != 0) - return 0; - - printk(KERN_INFO "%s: %s\n", __func__, dev_name(dev->parent->parent)); - if (strncmp(dev_name(dev->parent->parent), "s3c24", - strlen("s3c24")) == 0) - return 1; - - return 0; } -/* device registration info */ - -static struct s3c24xx_iis_ops simtec_audio_ops = { - .startup = simtec_audio_startup, - .shutdown = simtec_audio_shutdown, +static struct s3c24xx_audio_simtec_pdata simtec_audio_platdata = { }; -static struct s3c24xx_platdata_iis simtec_audio_platdata = { - .ops = &simtec_audio_ops, - .codec_clk = "clkout0", - .match_dev = simtec_audio_matchdev, -}; +static char our_name[32]; static struct platform_device audio_dev = { - .name = "s3c24xx-tlv320aic23", + .name = our_name, + .id = -1, .dev = { .parent = &s3c_device_iis.dev, .platform_data = &simtec_audio_platdata, }, }; -static struct platform_device audio_dev2 = { - .name = "s3c24xx-tlv320aic33", - .dev = { - .parent = &s3c_device_iis.dev, - .platform_data = &simtec_audio_platdata, - }, -}; - -static void __init simtec_audio_cdclk_mpllin(void) +int __init simtec_audio_add(const char *name, + struct s3c24xx_audio_simtec_pdata *spd) { - simtec_audio_platdata.flags = AUDIO_CDCLK_MPLL | AUDIO_CDCLK_OUTPUT; -} + if (!name) + name = "tlv320aic23"; -int __init simtec_audio_add(unsigned int is_aic33, - unsigned int cdclk_master) -{ - if (cdclk_master) - simtec_audio_cdclk_mpllin(); + snprintf(our_name, sizeof(our_name)-1, "s3c24xx-simtec-%s", name); - platform_device_register(&s3c_device_iis); + if (!spd) { + printk(KERN_ERR "%s: no source platadata\n", __func__); + return -EINVAL; + } - if (is_aic33) - platform_device_register(&audio_dev2); - else - platform_device_register(&audio_dev); + /* copy platform data so the source can be __initdata */ + simtec_audio_platdata = *spd; + + platform_device_register(&s3c_device_iis); + platform_device_register(&audio_dev); return 0; } @@ -172,7 +118,8 @@ static int __init simtec_audio_init(void printk(KERN_INFO "Simtec TLV320AIC23 Audio platform driver\n"); - simtec_audio_shutdown(NULL); + if (has_lr_routing()) + simtec_audio_platdata.startup = simtec_audio_startup; platform_device_register(&s3c_device_iis); return platform_device_register(&audio_dev); Index: b/arch/arm/common/sbcd/s3c24xx/dev-audio.c =================================================================== --- a/arch/arm/common/sbcd/s3c24xx/dev-audio.c 2009-08-04 12:05:59.000000000 +0100 +++ b/arch/arm/common/sbcd/s3c24xx/dev-audio.c 2009-08-04 12:24:11.000000000 +0100 @@ -12,6 +12,9 @@ #include <linux/kernel.h> #include <linux/ioport.h> #include <linux/device.h> +#include <linux/ioport.h> +#include <linux/gpio.h> +#include <linux/io.h> #include "s3c24xx.h" #include <asm/mach/sbcd-utils.h> @@ -24,22 +27,69 @@ #include "../devices/platform.h" #include <plat/devs.h> -#include <asm/plat-s3c24xx/audio-simtec.h> +#include <plat/audio-simtec.h> + +enum amp_type { + AMP_VR2K, + AMP_HERMES, /* GAIN0 = GPD9, GAIN1 = GPD10 */ +}; + +static void __iomem *reg; +static enum amp_type type; + +static void amp_gpio_set(struct gpio_chip *chip, + unsigned offset, int value) +{ + u32 to_write; -static unsigned char is_cdclk_master; + if (type == AMP_VR2K) { + to_write = value ? 0x01 : 0x03; + } else { + to_write = value ? 0x03 : 0x02; + } + + writeb(to_write, reg); +} + +static int amp_gpio_output(struct gpio_chip *chip, + unsigned offset, int value) +{ + amp_gpio_set(chip, offset, value); + return 0; +} + +static struct gpio_chip amp_gpio = { + .label = "amp", + .direction_output = amp_gpio_output, + .set = amp_gpio_set, + .ngpio = 1, + .base = -1, +}; + +static struct s3c24xx_audio_simtec_pdata __initdata pdata = { +}; static __init int s3c24xx_drv_bast_audio_attach(struct sbcd_state *state, struct sbcd_item *parent, struct sbcd_item *drv_it) { - int is_tlv320aic33 = sbcd_board_has_driver("tlv320aic33"); + char *name = NULL; /* some vr2000s do not have the necessary setup in their - * configuration data, so force is_cdclk_master here. */ - if (sbcd_board_has_driver("vr2k-amp")) - is_cdclk_master = 1; + * configuration data, so force cdclk_master here. */ + + if (sbcd_board_has_driver("vr2k-amp") || + sbcd_board_has_driver("hermes-amp")) { + pdata.use_mpllin = 1; + pdata.output_cdclk = 1; + } - return simtec_audio_add(is_tlv320aic33, is_cdclk_master); + if (sbcd_board_has_driver("hermes-amp")) + name = "hermes-snd"; + + pdata.amp_gpio = amp_gpio.base; + + return simtec_audio_add(name, &pdata); } @@ -61,8 +111,10 @@ static __init int s3c24xx_drv_audio_atta sbcd_get_optvals(ptr, &optname, &optval); if (strcmp(optname, "cdclk") == 0 && - strcmp(optval, "master") == 0) - is_cdclk_master = 1; + strcmp(optval, "master") == 0) { + pdata.use_mpllin = 1; + pdata.output_cdclk = 1; + } break; } } @@ -76,49 +128,57 @@ static __init int s3c24xx_drv_audio_atta sbcd_drv_decl2(cpu_audio, "cpu-audio", s3c24xx_drv_audio_attach); sbcd_drv_decl2(bast_audio, "bast-audio", s3c24xx_drv_bast_audio_attach); -static struct resource vr2k_amp_resource[] = { - [0] = { - .start = S3C2410_CS1 + 0xc0, - .end = S3C2410_CS1 + 0xc1, - .flags = IORESOURCE_MEM, - }, -}; - -static struct resource hermes_amp_resource[] = { - [0] = { - .start = S3C2410_CS1 + 0x140, - .end = S3C2410_CS1 + 0x141, - .flags = IORESOURCE_MEM, - }, -}; static __init int s3c24xx_drv_amp_attach(struct sbcd_state *state, - struct sbcd_item *parent, - struct sbcd_item *drv_it) + enum amp_type amp_type) { - struct platform_device *pdev; + unsigned long base; int ret; - ret = sbcd_dev_platform_create(state, parent, drv_it, &pdev); - if (ret) { - printk(KERN_ERR "%s: no platform device\n", __func__); - return -ENODEV; + if (state->pass != SBCD_PASS_GPIO) + return 0; + + type = amp_type; + + if (amp_type == AMP_VR2K) { + base = S3C2410_CS1 + 0xc0; + } else { + base = S3C2410_CS1 + 0x140; + pdata.amp_gain[0] = S3C2410_GPD(9); + pdata.amp_gain[1] = S3C2410_GPD(10); } - if (pdev->num_resources == 0) { - if (strcmp(pdev->name, "vr2k-amp") == 0) { - pdev->resource = vr2k_amp_resource; - pdev->num_resources = ARRAY_SIZE(vr2k_amp_resource); - } else if (strcmp(pdev->name, "hermes-amp") == 0) { - pdev->resource = hermes_amp_resource; - pdev->num_resources = ARRAY_SIZE(hermes_amp_resource); - } + /* note, we don't request the region, other gpio blocks may + * be sharing it. */ + + reg = ioremap(base, 2); + if (!reg) { + printk(KERN_ERR "%s: failed to map regs\n", __func__); + return -ENXIO; + } + + ret = gpiochip_add(&amp_gpio); + if (!ret) { + printk(KERN_ERR "%s: failed to add gpio\n", __func__); + return ret; } - pdev->dev.parent = &s3c_device_iis.dev; - platform_device_register(pdev); return 0; } -sbcd_drv_decl2(vr2k_amp, "vr2k-amp", s3c24xx_drv_amp_attach); -sbcd_drv_decl2(hermes_amp, "hermes-amp", s3c24xx_drv_amp_attach); +static __init int s3c24xx_drv_vr2k_amp_attach(struct sbcd_state *state, + struct sbcd_item *parent, + struct sbcd_item *drv_it) +{ + return s3c24xx_drv_amp_attach(state, AMP_VR2K); +} + +static __init int s3c24xx_drv_hermes_amp_attach(struct sbcd_state *state, + struct sbcd_item *parent, + struct sbcd_item *drv_it) +{ + return s3c24xx_drv_amp_attach(state, AMP_HERMES); +} + +sbcd_drv_decl2_gpio(vr2k_amp, "vr2k-amp", s3c24xx_drv_vr2k_amp_attach); +sbcd_drv_decl2_gpio(hermes_amp, "hermes-amp", s3c24xx_drv_hermes_amp_attach); Index: b/sound/soc/s3c24xx/s3c24xx_simtec.h =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ b/sound/soc/s3c24xx/s3c24xx_simtec.h 2009-08-04 12:24:11.000000000 +0100 @@ -0,0 +1,22 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec.h + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +extern void simtec_audio_init(struct snd_soc_codec *codec); + +extern int simtec_audio_core_probe(struct platform_device *pdev, + struct snd_soc_device *socdev); + +extern int simtec_audio_remove(struct platform_device *pdev); + +#ifdef CONFIG_PM +extern struct dev_pm_ops simtec_audio_pmops; +#define simtec_audio_pm &simtec_audio_pmops +#else +#define simtec_audio_pm NULL +#endif Index: b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c =================================================================== --- /dev/null 1970-01-01 00:00:00.000000000 +0000 +++ b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c 2009-08-04 12:24:11.000000000 +0100 @@ -0,0 +1,137 @@ +/* sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c + * + * Copyright 2009 Simtec Electronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. +*/ + +#include <linux/module.h> +#include <linux/clk.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <plat/audio-simtec.h> + +#include "s3c24xx-pcm.h" +#include "s3c24xx-i2s.h" +#include "s3c24xx_simtec.h" + +#include "../codecs/tlv320aic23.h" + +/* supported machines: + * + * Machine Connections AMP + * ------- ----------- --- + * BAST MIC, HPOUT, LOUT, LIN TPA2001D1 (HPOUTL,R) (gain hardwired) + * VR1000 HPOUT, LIN None + * VR2000 LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) + * DePicture LIN, LOUT, MIC, HP LM4871 (HPOUTL,R) + * Anubis LIN, LOUT, MIC, HP TPA2001D1 (HPOUTL,R) + */ + +static const struct snd_soc_dapm_widget dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), +}; + +static const struct snd_soc_dapm_route base_map[] = { + { "Headphone Jack", NULL, "LHPOUT"}, + { "Headphone Jack", NULL, "RHPOUT"}, + + { "Line Out", NULL, "LOUT" }, + { "Line Out", NULL, "ROUT" }, + + { "LLINEIN", NULL, "Line In"}, + { "RLINEIN", NULL, "Line In"}, + + { "MICIN", NULL, "Mic Jack"}, +}; + +/** + * simtec_tlv320aic23_init - initialise and add controls + * @codec; The codec instance to attach to. + * + * Attach our controls and configure the necessary codec + * mappings for our sound card instance. +*/ +static int simtec_tlv320aic23_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, dapm_widgets, + ARRAY_SIZE(dapm_widgets)); + + snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map)); + + snd_soc_dapm_enable_pin(codec, "Headphone Jack"); + snd_soc_dapm_enable_pin(codec, "Line In"); + snd_soc_dapm_enable_pin(codec, "Line Out"); + snd_soc_dapm_enable_pin(codec, "Mic Jack"); + + simtec_audio_init(codec); + snd_soc_dapm_sync(codec); + + return 0; +} + +static struct snd_soc_dai_link simtec_dai_aic23 = { + .name = "tlv320aic23", + .stream_name = "TLV320AIC23", + .cpu_dai = &s3c24xx_i2s_dai, + .codec_dai = &tlv320aic23_dai, + .init = simtec_tlv320aic23_init, +}; + +/* simtec audio machine driver */ +static struct snd_soc_card snd_soc_machine_simtec_aic23 = { + .name = "Simtec", + .platform = &s3c24xx_soc_platform, + .dai_link = &simtec_dai_aic23, + .num_links = 1, +}; + +/* simtec audio subsystem */ +static struct snd_soc_device simtec_snd_devdata_aic23 = { + .card = &snd_soc_machine_simtec_aic23, + .codec_dev = &soc_codec_dev_tlv320aic23, +}; + +static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd) +{ + return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic23); +} + +static struct platform_driver simtec_audio_tlv320aic23_platdrv = { + .driver = { + .owner = THIS_MODULE, + .name = "s3c24xx-simtec-tlv320aic23", + .pm = simtec_audio_pm, + }, + .probe = simtec_audio_tlv320aic23_probe, + .remove = __devexit_p(simtec_audio_remove), +}; + +MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23"); + +static int __init simtec_tlv320aic23_modinit(void) +{ + return platform_driver_register(&simtec_audio_tlv320aic23_platdrv); +} + +static void __exit simtec_tlv320aic23_modexit(void) +{ + platform_driver_unregister(&simtec_audio_tlv320aic23_platdrv); +} + +module_init(simtec_tlv320aic23_modinit); +module_exit(simtec_tlv320aic23_modexit); + +MODULE_AUTHOR("Ben Dooks <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("ALSA SoC Simtec Audio support"); +MODULE_LICENSE("GPL"); -- Ben (ben(a)fluff.org, http://www.fluff.org/) 'a smiley only costs 4 bytes'

2 1

[alsa-devel] [PATCH][2.6.31-rc5] ARM: OMAP: McBSP: Fix ASoC on OMAP1510 by fixing API of omap_mcbsp_start/stop
by Jarkko Nikula 07 Aug '09

07 Aug '09

Simultaneous audio playback and capture on OMAP1510 can cause that second stream is stalled if there is enough delay between startup of the audio streams. Current implementation of the omap_mcbsp_start is starting both transmitter and receiver at the same time and it is called only for firstly started audio stream from the OMAP McBSP based ASoC DAI driver. Since DMA request lines on OMAP1510 are edge sensitive, the DMA request is missed if there is no DMA transfer set up at that time when the first word after McBSP startup is transmitted. The problem hasn't noted before since later OMAPs are using level sensitive DMA request lines. Fix the problem by changing API of omap_mcbsp_start and omap_mcbsp_stop by allowing to start and stop individually McBSP transmitter and receiver logics. Then call those functions individually for both audio playback and capture streams. This ensures that DMA transfer is setup before transmitter or receiver is started. Thanks to Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> for detailed problem analysis and Peter Ujfalusi <peter.ujfalusi(a)nokia.com> for info about DMA request line behavior differences between the OMAP generations. Reported-and-tested-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> Signed-off-by: Jarkko Nikula <jhnikula(a)gmail.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Tony Lindgren <tony(a)atomide.com> --- arch/arm/plat-omap/include/mach/mcbsp.h | 4 +- arch/arm/plat-omap/mcbsp.c | 50 +++++++++++++++++++----------- sound/soc/omap/omap-mcbsp.c | 10 +++--- 3 files changed, 39 insertions(+), 25 deletions(-) diff --git a/arch/arm/plat-omap/include/mach/mcbsp.h b/arch/arm/plat-omap/include/mach/mcbsp.h index bb154ea..57249bb 100644 --- a/arch/arm/plat-omap/include/mach/mcbsp.h +++ b/arch/arm/plat-omap/include/mach/mcbsp.h @@ -387,8 +387,8 @@ void omap_mcbsp_register_board_cfg(struct omap_mcbsp_platform_data *config, void omap_mcbsp_config(unsigned int id, const struct omap_mcbsp_reg_cfg * config); int omap_mcbsp_request(unsigned int id); void omap_mcbsp_free(unsigned int id); -void omap_mcbsp_start(unsigned int id); -void omap_mcbsp_stop(unsigned int id); +void omap_mcbsp_start(unsigned int id, int tx, int rx); +void omap_mcbsp_stop(unsigned int id, int tx, int rx); void omap_mcbsp_xmit_word(unsigned int id, u32 word); u32 omap_mcbsp_recv_word(unsigned int id); diff --git a/arch/arm/plat-omap/mcbsp.c b/arch/arm/plat-omap/mcbsp.c index efa0e01..a3d2313 100644 --- a/arch/arm/plat-omap/mcbsp.c +++ b/arch/arm/plat-omap/mcbsp.c @@ -328,14 +328,15 @@ void omap_mcbsp_free(unsigned int id) EXPORT_SYMBOL(omap_mcbsp_free); /* - * Here we start the McBSP, by enabling the sample - * generator, both transmitter and receivers, - * and the frame sync. + * Here we start the McBSP, by enabling transmitter, receiver or both. + * If no transmitter or receiver is active prior calling, then sample-rate + * generator and frame sync are started. */ -void omap_mcbsp_start(unsigned int id) +void omap_mcbsp_start(unsigned int id, int tx, int rx) { struct omap_mcbsp *mcbsp; void __iomem *io_base; + int idle; u16 w; if (!omap_mcbsp_check_valid_id(id)) { @@ -348,32 +349,40 @@ void omap_mcbsp_start(unsigned int id) mcbsp->rx_word_length = (OMAP_MCBSP_READ(io_base, RCR1) >> 5) & 0x7; mcbsp->tx_word_length = (OMAP_MCBSP_READ(io_base, XCR1) >> 5) & 0x7; - /* Start the sample generator */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 6)); + idle = !((OMAP_MCBSP_READ(io_base, SPCR2) | + OMAP_MCBSP_READ(io_base, SPCR1)) & 1); + + if (idle) { + /* Start the sample generator */ + w = OMAP_MCBSP_READ(io_base, SPCR2); + OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 6)); + } /* Enable transmitter and receiver */ w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w | 1); + OMAP_MCBSP_WRITE(io_base, SPCR2, w | (tx & 1)); w = OMAP_MCBSP_READ(io_base, SPCR1); - OMAP_MCBSP_WRITE(io_base, SPCR1, w | 1); + OMAP_MCBSP_WRITE(io_base, SPCR1, w | (rx & 1)); udelay(100); - /* Start frame sync */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 7)); + if (idle) { + /* Start frame sync */ + w = OMAP_MCBSP_READ(io_base, SPCR2); + OMAP_MCBSP_WRITE(io_base, SPCR2, w | (1 << 7)); + } /* Dump McBSP Regs */ omap_mcbsp_dump_reg(id); } EXPORT_SYMBOL(omap_mcbsp_start); -void omap_mcbsp_stop(unsigned int id) +void omap_mcbsp_stop(unsigned int id, int tx, int rx) { struct omap_mcbsp *mcbsp; void __iomem *io_base; + int idle; u16 w; if (!omap_mcbsp_check_valid_id(id)) { @@ -386,15 +395,20 @@ void omap_mcbsp_stop(unsigned int id) /* Reset transmitter */ w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(1)); + OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(tx & 1)); /* Reset receiver */ w = OMAP_MCBSP_READ(io_base, SPCR1); - OMAP_MCBSP_WRITE(io_base, SPCR1, w & ~(1)); + OMAP_MCBSP_WRITE(io_base, SPCR1, w & ~(rx & 1)); - /* Reset the sample rate generator */ - w = OMAP_MCBSP_READ(io_base, SPCR2); - OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(1 << 6)); + idle = !((OMAP_MCBSP_READ(io_base, SPCR2) | + OMAP_MCBSP_READ(io_base, SPCR1)) & 1); + + if (idle) { + /* Reset the sample rate generator */ + w = OMAP_MCBSP_READ(io_base, SPCR2); + OMAP_MCBSP_WRITE(io_base, SPCR2, w & ~(1 << 6)); + } } EXPORT_SYMBOL(omap_mcbsp_stop); diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index a5d46a7..6a837ff 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -183,21 +183,21 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); - int err = 0; + int err = 0, play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (!mcbsp_data->active++) - omap_mcbsp_start(mcbsp_data->bus_id); + mcbsp_data->active++; + omap_mcbsp_start(mcbsp_data->bus_id, play, !play); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (!--mcbsp_data->active) - omap_mcbsp_stop(mcbsp_data->bus_id); + omap_mcbsp_stop(mcbsp_data->bus_id, play, !play); + mcbsp_data->active--; break; default: err = -EINVAL; -- 1.6.3.3

4 6

[alsa-devel] tlv320aic3x: Allow driver to use already registered i2c device
by Ben Dooks 07 Aug '09

07 Aug '09

Add the ability for the driver to use an already registered i2c device instead of creating one of them itself. Add the field i2c_already field to the setup data and not set the i2c bus address the driver will then use the device already setup. Signed-off-by: Ben Dooks <ben(a)simtec.co.uk> --- sound/soc/codecs/tlv320aic3x.c | 9 ++++++++- sound/soc/codecs/tlv320aic3x.h | 1 + 2 files changed, 9 insertions(+), 1 deletion(-) Index: b/sound/soc/codecs/tlv320aic3x.c =================================================================== --- a/sound/soc/codecs/tlv320aic3x.c 2009-08-06 11:53:30.000000000 +0100 +++ b/sound/soc/codecs/tlv320aic3x.c 2009-08-06 15:29:45.000000000 +0100 @@ -1310,6 +1310,7 @@ static int aic3x_i2c_remove(struct i2c_c static const struct i2c_device_id aic3x_i2c_id[] = { { "tlv320aic3x", 0 }, + { "tlv320aic33", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, aic3x_i2c_id); @@ -1405,6 +1406,10 @@ static int aic3x_probe(struct platform_d codec->hw_write = (hw_write_t) i2c_master_send; codec->hw_read = (hw_read_t) aic3x_i2c_read; ret = aic3x_add_i2c_device(pdev, setup); + } else if (setup->i2c_already) { + codec->hw_write = (hw_write_t) i2c_master_send; + codec->hw_read = (hw_read_t) aic3x_i2c_read; + i2c_add_driver(&aic3x_i2c_driver); } #else /* Add other interfaces here */ @@ -1421,6 +1426,7 @@ static int aic3x_remove(struct platform_ { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; + struct aic3x_setup_data *setup = socdev->codec_data; /* power down chip */ if (codec->control_data) @@ -1429,7 +1435,8 @@ static int aic3x_remove(struct platform_ snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) - i2c_unregister_device(codec->control_data); + if (setup && !setup->i2c_already) + i2c_unregister_device(codec->control_data); i2c_del_driver(&aic3x_i2c_driver); #endif kfree(codec->private_data); Index: b/sound/soc/codecs/tlv320aic3x.h =================================================================== --- a/sound/soc/codecs/tlv320aic3x.h 2009-08-06 11:53:30.000000000 +0100 +++ b/sound/soc/codecs/tlv320aic3x.h 2009-08-06 12:11:03.000000000 +0100 @@ -283,6 +283,7 @@ int aic3x_button_pressed(struct snd_soc_ struct aic3x_setup_data { int i2c_bus; + unsigned char i2c_already; /* i2c device already registered */ unsigned short i2c_address; unsigned int gpio_func[2]; }; -- Ben (ben(a)fluff.org, http://www.fluff.org/) 'a smiley only costs 4 bytes'

2 1

[alsa-devel] usb audio
by bruce 07 Aug '09

07 Aug '09

Hi I have the TASCAM US-122L USB.I am using ubuntu studio 9.04.The usb audio inputs and outputs are no functional cant fined them anywhere I am using jack connection kit on my studio the midi are recognized but not the audio.I have submitted to launchpad this report.Here are my jack start up log this is with tascam us-122L set as input and output.It will start with just tascam input but there is no audio only midi with that start. jackd 0.116.1 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. server `default' registered 23:53:21.206 JACK was started with PID=8295. registered builtin port type 32 bit float mono audio registered builtin port type 8 bit raw midi clock source = system clock via clock_gettime start poll on 3 fd's loading driver .. new client: alsa_pcm, id = 1 type 1 @ 0x9821d48 fd = -1 apparent rate = 44100 creating alsa driver ... hw:1|hw:1|1024|3|44100|0|0|hwmon|hwmeter|-|32bit control device hw:1 23:53:21.383 ALSA active patchbay scan... 23:53:21.407 jack-imput-output003: 14:0 Midi Through -> 20:0 TASCAM US-122L checked. 23:53:21.416 ALSA connection change. 23:53:21.811 ALSA active patchbay scan... 23:53:21.813 jack-imput-output003: 14:0 Midi Through -> 20:0 TASCAM US-122L checked. ALSA: Cannot open PCM device alsa_pcm for playback. Falling back to capture-only mode cannot load driver module alsa starting server engine shutdown server thread back from poll freeing shared port segments stopping server thread last xrun delay: 0.000 usecs max delay reported by backend: 0.000 usecs freeing engine shared memory max usecs: 0.000, engine deleted cleaning up shared memory cleaning up files unregistering server `default' 23:53:22.586 JACK was stopped successfully. 23:53:22.588 Post-shutdown script... 23:53:22.590 killall jackd jackd: no process killed 23:53:23.015 Post-shutdown script terminated with exit status=256. 23:53:23.727 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info. thank you for any help you mite give :-) Bruce McCandless

1 0

[alsa-devel] [PATCH 1/3] ASoC: change set_tdm_slot api to allow slot_width override.
by Daniel Ribeiro 06 Aug '09

06 Aug '09

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> --- 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 352d7ee..6692410 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 86fc57e..c47d573 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1206,8 +1206,9 @@ 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); @@ -1219,7 +1220,7 @@ 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 c89a3cd..34437d8 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -188,7 +188,7 @@ 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 e1a920c..c72a8e8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2133,17 +2133,20 @@ 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; } -- tg: (c787980..) asoc/set_tdm_slot (depends on: asoc/remove-code-duplication) -- Daniel Ribeiro

6 6

[alsa-devel] [PATCH 2/2] ASoC: AMS_DELTA: Use updated line discipline name
by Janusz Krzysztofik 06 Aug '09

06 Aug '09

This patch updates the tty line discipline name used in the Amstrad Delta machine driver. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- linux-2.6.31-rc5/sound/soc/omap/ams-delta.c.orig 2009-08-05 21:44:35.000000000 +0200 +++ linux-2.6.31-rc5/sound/soc/omap/ams-delta.c 2009-08-05 23:30:53.000000000 +0200 @@ -519,7 +519,7 @@ static int ams_delta_cx20442_init(struct } /* Register optional line discipline for over the modem control */ - ret = tty_register_ldisc(N_AMSDELTA, &cx81801_ops); + ret = tty_register_ldisc(N_V253, &cx81801_ops); if (ret) { dev_warn(card->dev, "Failed to register line discipline, " @@ -644,9 +644,9 @@ static void __exit ams_delta_module_exit } } - if (tty_unregister_ldisc(N_AMSDELTA) != 0) + if (tty_unregister_ldisc(N_V253) != 0) dev_warn(&ams_delta_audio_platform_device->dev, - "failed to unregister AMSDELTA line discipline\n"); + "failed to unregister V253 line discipline\n"); snd_soc_jack_free_gpios(&ams_delta_hook_switch, ARRAY_SIZE(ams_delta_hook_switch_gpios),

1 0

[alsa-devel] [PATCH 1/2] TTY: Rename N_AMSDELTA line discipline to N_V253
by Janusz Krzysztofik 06 Aug '09

06 Aug '09

The patch changes the line discipline name registered in include/linux/tty.h Signed-off-by: Janusz Krzysztofik --- --- linux-2.6.31-rc5/include/linux/tty.h.orig 2009-08-05 21:44:33.000000000 +0200 +++ linux-2.6.31-rc5/include/linux/tty.h 2009-08-05 22:54:12.000000000 +0200 @@ -47,7 +47,7 @@ #define N_SLCAN 17 /* Serial / USB serial CAN Adaptors */ #define N_PPS 18 /* Pulse per Second */ -#define N_AMSDELTA 19 /* codec control over modem, board specific */ +#define N_V253 19 /* Codec control over voice modem */ /* * This character is the same as _POSIX_VDISABLE: it cannot be used as

1 0

[alsa-devel] [PATCH 0/2] ASoC: CX2044: full duplex mode fix
by Janusz Krzysztofik 06 Aug '09

06 Aug '09

This series of patches tries to solve the problem of audio mode changes not working correctly while in full duplex mode. [PATCH 1/2] ASoC: CX20442: add some debugging [PATCH 2/2] ASoC: CX20442: simplify codec controller usage Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl>

2 1

[alsa-devel] [PATCH 2/2] ASoC: CX20442: simplify codec controller usage
by Janusz Krzysztofik 06 Aug '09

06 Aug '09

This patch is a workaround for the problem of several subsequent control statements not being applied correctly to the codec controller (modem). In order to follow the hook switch state change from handset to handsfree while in full duplex mode, two consecutive +VLS control commands were sent to the modem. The first one was M1 (microphone only), the seconds one was M1S1 (both microphone and speaker). As there was no real modem handshaking procedure implemented, neither in the codec nor in the machine driver part of the line discipline, the modem was having the second command missed. Since a possibility to switch to microphone only mode (and speaker only mode as well) seams of no value, I have modified the code to issue single M1S1 command only for any of those cases. Tested on my Amstrad Delta. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- linux-2.6.31-rc5/sound/soc/codecs/cx20442.c.orig 2009-08-04 23:41:03.000000000 +0200 +++ linux-2.6.31-rc5/sound/soc/codecs/cx20442.c 2009-08-05 00:56:30.000000000 +0200 @@ -138,9 +138,7 @@ static int cx20442_pm_to_v253_vls(u8 val case 0: return V253_VLS_T; case (1 << CX20442_SPKOUT): - return V253_VLS_S1; case (1 << CX20442_MIC): - return V253_VLS_M1; case (1 << CX20442_SPKOUT) | (1 << CX20442_MIC): return V253_VLS_M1S1; case (1 << CX20442_TELOUT):

1 0

[alsa-devel] [PATCH 1/2] ASoC: CX20442: add some debugging
by Janusz Krzysztofik 06 Aug '09

06 Aug '09

This patch adds debugging statement that can help in tracing how the driver is trying to control the codec device. Signed-off-by: Janusz Krzysztofik <jkrzyszt(a)tis.icnet.pl> --- --- linux-2.6.31-rc5/sound/soc/codecs/cx20442.c.orig 2009-08-04 23:41:03.000000000 +0200 +++ linux-2.6.31-rc5/sound/soc/codecs/cx20442.c 2009-08-05 00:56:30.000000000 +0200 @@ -203,6 +201,7 @@ static int cx20442_write(struct snd_soc_ if (unlikely(len > (ARRAY_SIZE(buf) - 1))) return -ENOMEM; + dev_dbg(codec->dev, "%s: %s\n", __func__, buf); if (codec->hw_write(codec->control_data, buf, len) != len) return -EIO;

1 0

[alsa-devel] [PATCH] ASoC: Add WM8776 CODEC driver
by Mark Brown 06 Aug '09

06 Aug '09

The WM8776 is a high performance, stereo audio CODEC with five channel input selector. The WM8776 is ideal for surround sound processing applications for home hi-fi, DVD-RW and other audio visual equipment. This driver implements support for most WM8776 features - currently the ADC automatic level control/limiter functionality is omitted. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm8776.c | 781 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8776.h | 51 +++ 4 files changed, 838 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/wm8776.c create mode 100644 sound/soc/codecs/wm8776.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index b6c7f7a..8cd24b0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -33,6 +33,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM8731 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8750 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI + select SND_SOC_WM8776 if SND_SOC_I2C_AND_SPI select SND_SOC_WM8900 if I2C select SND_SOC_WM8903 if I2C select SND_SOC_WM8971 if I2C @@ -132,6 +133,9 @@ config SND_SOC_WM8750 config SND_SOC_WM8753 tristate +config SND_SOC_WM8776 + tristate + config SND_SOC_WM8900 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index f265380..e80f164 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -21,6 +21,7 @@ snd-soc-wm8728-objs := wm8728.o snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o +snd-soc-wm8776-objs := wm8776.o snd-soc-wm8900-objs := wm8900.o snd-soc-wm8903-objs := wm8903.o snd-soc-wm8971-objs := wm8971.o @@ -52,6 +53,7 @@ obj-$(CONFIG_SND_SOC_WM8728) += snd-soc-wm8728.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o +obj-$(CONFIG_SND_SOC_WM8776) += snd-soc-wm8776.o obj-$(CONFIG_SND_SOC_WM8900) += snd-soc-wm8900.o obj-$(CONFIG_SND_SOC_WM8903) += snd-soc-wm8903.o obj-$(CONFIG_SND_SOC_WM8971) += snd-soc-wm8971.o diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c new file mode 100644 index 0000000..ee12bf8 --- /dev/null +++ b/sound/soc/codecs/wm8776.c @@ -0,0 +1,781 @@ +/* + * wm8776.c -- WM8776 ALSA SoC Audio driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * TODO: Input ALC/limiter support + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "wm8776.h" + +static struct snd_soc_codec *wm8776_codec; +struct snd_soc_codec_device soc_codec_dev_wm8776; + +/* codec private data */ +struct wm8776_priv { + struct snd_soc_codec codec; + u16 reg_cache[WM8776_CACHEREGNUM]; + int sysclk[2]; +}; + +#ifdef CONFIG_SPI_MASTER +static int wm8776_spi_write(struct spi_device *spi, const char *data, int len); +#endif + +static const u16 wm8776_reg[WM8776_CACHEREGNUM] = { + 0x79, 0x79, 0x79, 0xff, 0xff, /* 4 */ + 0xff, 0x00, 0x90, 0x00, 0x00, /* 9 */ + 0x22, 0x22, 0x22, 0x08, 0xcf, /* 14 */ + 0xcf, 0x7b, 0x00, 0x32, 0x00, /* 19 */ + 0xa6, 0x01, 0x01 +}; + +/* + * read wm8776 register cache + */ +static inline unsigned int wm8776_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u16 *cache = codec->reg_cache; + if (reg >= WM8776_CACHEREGNUM) + return -1; + return cache[reg]; +} + +/* + * write wm8776 register cache + */ +static inline void wm8776_write_reg_cache(struct snd_soc_codec *codec, + u16 reg, unsigned int value) +{ + u16 *cache = codec->reg_cache; + if (reg >= WM8776_CACHEREGNUM) + return; + cache[reg] = value; +} + +/* + * write to the WM8776 register space + */ +static int wm8776_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 data[2]; + + data[0] = (reg << 1) | ((value >> 8) & 0x0001); + data[1] = value & 0x00ff; + + wm8776_write_reg_cache(codec, reg, value); + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static int wm8776_reset(struct snd_soc_codec *codec) +{ + return wm8776_write(codec, WM8776_RESET, 0); +} + +static const DECLARE_TLV_DB_SCALE(hp_tlv, -12100, 100, 1); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -10350, 50, 1); + +static const struct snd_kcontrol_new wm8776_snd_controls[] = { +SOC_DOUBLE_R_TLV("Headphone Playback Volume", WM8776_HPLVOL, WM8776_HPRVOL, + 0, 127, 0, hp_tlv), +SOC_DOUBLE_R_TLV("Digital Playback Volume", WM8776_DACLVOL, WM8776_DACRVOL, + 0, 255, 0, dac_tlv), +SOC_SINGLE("Digital Playback ZC Switch", WM8776_DACCTRL1, 0, 1, 0), + +SOC_SINGLE("Deemphasis Switch", WM8776_DACCTRL2, 0, 1, 0), + +SOC_DOUBLE_R_TLV("Capture Volume", WM8776_ADCLVOL, WM8776_ADCRVOL, + 0, 255, 0, adc_tlv), +SOC_DOUBLE("Capture Switch", WM8776_ADCMUX, 7, 6, 1, 1), +SOC_DOUBLE_R("Capture ZC Switch", WM8776_ADCLVOL, WM8776_ADCRVOL, 8, 1, 0), +SOC_SINGLE("Capture HPF Switch", WM8776_ADCIFCTRL, 8, 1, 1), +}; + +static const struct snd_kcontrol_new inmix_controls[] = { +SOC_DAPM_SINGLE("AIN1 Switch", WM8776_ADCMUX, 0, 1, 0), +SOC_DAPM_SINGLE("AIN2 Switch", WM8776_ADCMUX, 1, 1, 0), +SOC_DAPM_SINGLE("AIN3 Switch", WM8776_ADCMUX, 2, 1, 0), +SOC_DAPM_SINGLE("AIN4 Switch", WM8776_ADCMUX, 3, 1, 0), +SOC_DAPM_SINGLE("AIN5 Switch", WM8776_ADCMUX, 4, 1, 0), +}; + +static const struct snd_kcontrol_new outmix_controls[] = { +SOC_DAPM_SINGLE("DAC Switch", WM8776_OUTMUX, 0, 1, 0), +SOC_DAPM_SINGLE("AUX Switch", WM8776_OUTMUX, 1, 1, 0), +SOC_DAPM_SINGLE("Bypass Switch", WM8776_OUTMUX, 2, 1, 0), +}; + +static const struct snd_soc_dapm_widget wm8776_dapm_widgets[] = { +SND_SOC_DAPM_INPUT("AUX"), +SND_SOC_DAPM_INPUT("AUX"), + +SND_SOC_DAPM_INPUT("AIN1"), +SND_SOC_DAPM_INPUT("AIN2"), +SND_SOC_DAPM_INPUT("AIN3"), +SND_SOC_DAPM_INPUT("AIN4"), +SND_SOC_DAPM_INPUT("AIN5"), + +SND_SOC_DAPM_MIXER("Input Mixer", WM8776_PWRDOWN, 6, 1, + inmix_controls, ARRAY_SIZE(inmix_controls)), + +SND_SOC_DAPM_ADC("ADC", "Capture", WM8776_PWRDOWN, 1, 1), +SND_SOC_DAPM_DAC("DAC", "Playback", WM8776_PWRDOWN, 2, 1), + +SND_SOC_DAPM_MIXER("Output Mixer", SND_SOC_NOPM, 0, 0, + outmix_controls, ARRAY_SIZE(outmix_controls)), + +SND_SOC_DAPM_PGA("Headphone PGA", WM8776_PWRDOWN, 3, 1, NULL, 0), + +SND_SOC_DAPM_OUTPUT("VOUT"), + +SND_SOC_DAPM_OUTPUT("HPOUTL"), +SND_SOC_DAPM_OUTPUT("HPOUTR"), +}; + +static const struct snd_soc_dapm_route routes[] = { + { "Input Mixer", "AIN1 Switch", "AIN1" }, + { "Input Mixer", "AIN2 Switch", "AIN2" }, + { "Input Mixer", "AIN3 Switch", "AIN3" }, + { "Input Mixer", "AIN4 Switch", "AIN4" }, + { "Input Mixer", "AIN5 Switch", "AIN5" }, + + { "ADC", NULL, "Input Mixer" }, + + { "Output Mixer", "DAC Switch", "DAC" }, + { "Output Mixer", "AUX Switch", "AUX" }, + { "Output Mixer", "Bypass Switch", "Input Mixer" }, + + { "VOUT", NULL, "Output Mixer" }, + + { "Headphone PGA", NULL, "Output Mixer" }, + + { "HPOUTL", NULL, "Headphone PGA" }, + { "HPOUTR", NULL, "Headphone PGA" }, +}; + +static int wm8776_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + int reg, iface, master; + + switch (dai->id) { + case WM8776_DAI_DAC: + reg = WM8776_DACIFCTRL; + master = 0x80; + break; + case WM8776_DAI_ADC: + reg = WM8776_ADCIFCTRL; + master = 0x100; + break; + default: + return -EINVAL; + } + + iface = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFS: + master = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + iface |= 0x0002; + break; + case SND_SOC_DAIFMT_RIGHT_J: + break; + case SND_SOC_DAIFMT_LEFT_J: + iface |= 0x0001; + break; + /* FIXME: CHECK A/B */ + case SND_SOC_DAIFMT_DSP_A: + iface |= 0x0003; + break; + case SND_SOC_DAIFMT_DSP_B: + iface |= 0x0007; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + iface |= 0x00c; + break; + case SND_SOC_DAIFMT_IB_NF: + iface |= 0x008; + break; + case SND_SOC_DAIFMT_NB_IF: + iface |= 0x004; + break; + default: + return -EINVAL; + } + + /* Finally, write out the values */ + snd_soc_update_bits(codec, reg, 0xf, iface); + snd_soc_update_bits(codec, WM8776_MSTRCTRL, 0x180, master); + + return 0; +} + +static int mclk_ratios[] = { + 128, + 192, + 256, + 384, + 512, + 768, +}; + +static int wm8776_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8776_priv *wm8776 = codec->private_data; + int iface_reg, iface; + int ratio_shift, master; + int i; + + iface = 0; + + switch (dai->id) { + case WM8776_DAI_DAC: + iface_reg = WM8776_DACIFCTRL; + master = 0x80; + ratio_shift = 4; + break; + case WM8776_DAI_ADC: + iface_reg = WM8776_ADCIFCTRL; + master = 0x100; + ratio_shift = 0; + break; + default: + return -EINVAL; + } + + + /* Set word length */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + break; + case SNDRV_PCM_FORMAT_S20_3LE: + iface |= 0x10; + break; + case SNDRV_PCM_FORMAT_S24_LE: + iface |= 0x20; + break; + case SNDRV_PCM_FORMAT_S32_LE: + iface |= 0x30; + break; + } + + /* Only need to set MCLK/LRCLK ratio if we're master */ + if (snd_soc_read(codec, WM8776_MSTRCTRL) & master) { + for (i = 0; i < ARRAY_SIZE(mclk_ratios); i++) { + if (wm8776->sysclk[dai->id] / params_rate(params) + == mclk_ratios[i]) + break; + } + + if (i == ARRAY_SIZE(mclk_ratios)) { + dev_err(codec->dev, + "Unable to configure MCLK ratio %d/%d\n", + wm8776->sysclk[dai->id], params_rate(params)); + return -EINVAL; + } + + dev_dbg(codec->dev, "MCLK is %dfs\n", mclk_ratios[i]); + + snd_soc_update_bits(codec, WM8776_MSTRCTRL, + 0x7 << ratio_shift, i << ratio_shift); + } else { + dev_dbg(codec->dev, "DAI in slave mode\n"); + } + + snd_soc_update_bits(codec, iface_reg, 0x30, iface); + + return 0; +} + +static int wm8776_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + + return snd_soc_write(codec, WM8776_DACMUTE, !!mute); +} + +static int wm8776_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8776_priv *wm8776 = codec->private_data; + + BUG_ON(dai->id >= ARRAY_SIZE(wm8776->sysclk)); + + wm8776->sysclk[dai->id] = freq; + + return 0; +} + +static int wm8776_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->bias_level == SND_SOC_BIAS_OFF) { + /* Disable the global powerdown; DAPM does the rest */ + snd_soc_update_bits(codec, WM8776_PWRDOWN, 1, 0); + } + + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, WM8776_PWRDOWN, 1, 1); + break; + } + + codec->bias_level = level; + return 0; +} + +#define WM8776_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000) + + +#define WM8776_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops wm8776_dac_ops = { + .digital_mute = wm8776_mute, + .hw_params = wm8776_hw_params, + .set_fmt = wm8776_set_fmt, + .set_sysclk = wm8776_set_sysclk, +}; + +static struct snd_soc_dai_ops wm8776_adc_ops = { + .hw_params = wm8776_hw_params, + .set_fmt = wm8776_set_fmt, + .set_sysclk = wm8776_set_sysclk, +}; + +struct snd_soc_dai wm8776_dai[] = { + { + .name = "WM8776 Playback", + .id = WM8776_DAI_DAC, + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = WM8776_RATES, + .formats = WM8776_FORMATS, + }, + .ops = &wm8776_dac_ops, + }, + { + .name = "WM8776 Capture", + .id = WM8776_DAI_ADC, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = WM8776_RATES, + .formats = WM8776_FORMATS, + }, + .ops = &wm8776_adc_ops, + }, +}; +EXPORT_SYMBOL_GPL(wm8776_dai); + +#ifdef CONFIG_PM +static int wm8776_suspend(struct platform_device *pdev, pm_message_t state) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + + wm8776_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int wm8776_resume(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec = socdev->card->codec; + int i; + u8 data[2]; + u16 *cache = codec->reg_cache; + + /* Sync reg_cache with the hardware */ + for (i = 0; i < ARRAY_SIZE(wm8776_reg); i++) { + data[0] = (i << 1) | ((cache[i] >> 8) & 0x0001); + data[1] = cache[i] & 0x00ff; + codec->hw_write(codec->control_data, data, 2); + } + + wm8776_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} +#else +#define wm8776_suspend NULL +#define wm8776_resume NULL +#endif + +static int wm8776_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (wm8776_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = wm8776_codec; + codec = wm8776_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, wm8776_snd_controls, + ARRAY_SIZE(wm8776_snd_controls)); + snd_soc_dapm_new_controls(codec, wm8776_dapm_widgets, + ARRAY_SIZE(wm8776_dapm_widgets)); + snd_soc_dapm_add_routes(codec, routes, ARRAY_SIZE(routes)); + + 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 wm8776_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_wm8776 = { + .probe = wm8776_probe, + .remove = wm8776_remove, + .suspend = wm8776_suspend, + .resume = wm8776_resume, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_wm8776); + +static int wm8776_register(struct wm8776_priv *wm8776) +{ + int ret, i; + struct snd_soc_codec *codec = &wm8776->codec; + + if (wm8776_codec) { + dev_err(codec->dev, "Another WM8776 is registered\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = wm8776; + codec->name = "WM8776"; + codec->owner = THIS_MODULE; + codec->read = wm8776_read_reg_cache; + codec->write = wm8776_write; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = wm8776_set_bias_level; + codec->dai = wm8776_dai; + codec->num_dai = ARRAY_SIZE(wm8776_dai); + codec->reg_cache_size = WM8776_CACHEREGNUM; + codec->reg_cache = &wm8776->reg_cache; + + memcpy(codec->reg_cache, wm8776_reg, sizeof(wm8776_reg)); + + for (i = 0; i < ARRAY_SIZE(wm8776_dai); i++) + wm8776_dai[i].dev = codec->dev; + + ret = wm8776_reset(codec); + if (ret < 0) { + dev_err(codec->dev, "Failed to issue reset: %d\n", ret); + goto err; + } + + wm8776_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + /* Latch the update bits; right channel only since we always + * update both. */ + snd_soc_update_bits(codec, WM8776_HPRVOL, 0x100, 0x100); + snd_soc_update_bits(codec, WM8776_DACRVOL, 0x100, 0x100); + + wm8776_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dais(wm8776_dai, ARRAY_SIZE(wm8776_dai)); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAIs: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(wm8776); + return ret; +} + +static void wm8776_unregister(struct wm8776_priv *wm8776) +{ + wm8776_set_bias_level(&wm8776->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dais(wm8776_dai, ARRAY_SIZE(wm8776_dai)); + snd_soc_unregister_codec(&wm8776->codec); + kfree(wm8776); + wm8776_codec = NULL; +} + +#if defined(CONFIG_SPI_MASTER) +static int wm8776_spi_write(struct spi_device *spi, const char *data, int len) +{ + struct spi_transfer t; + struct spi_message m; + u8 msg[2]; + + if (len <= 0) + return 0; + + msg[0] = data[0]; + msg[1] = data[1]; + + spi_message_init(&m); + memset(&t, 0, (sizeof t)); + + t.tx_buf = &msg[0]; + t.len = len; + + spi_message_add_tail(&t, &m); + spi_sync(spi, &m); + + return len; +} + +static int __devinit wm8776_spi_probe(struct spi_device *spi) +{ + struct snd_soc_codec *codec; + struct wm8776_priv *wm8776; + + wm8776 = kzalloc(sizeof(struct wm8776_priv), GFP_KERNEL); + if (wm8776 == NULL) + return -ENOMEM; + + codec = &wm8776->codec; + codec->control_data = spi; + codec->hw_write = (hw_write_t)wm8776_spi_write; + codec->dev = &spi->dev; + + dev_set_drvdata(&spi->dev, wm8776); + + return wm8776_register(wm8776); +} + +static int __devexit wm8776_spi_remove(struct spi_device *spi) +{ + struct wm8776_priv *wm8776 = dev_get_drvdata(&spi->dev); + + wm8776_unregister(wm8776); + + return 0; +} + +#ifdef CONFIG_PM +static int wm8776_spi_suspend(struct spi_device *spi, pm_message_t msg) +{ + return snd_soc_suspend_device(&spi->dev); +} + +static int wm8776_spi_resume(struct spi_device *spi) +{ + return snd_soc_resume_device(&spi->dev); +} +#else +#define wm8776_spi_suspend NULL +#define wm8776_spi_resume NULL +#endif + +static struct spi_driver wm8776_spi_driver = { + .driver = { + .name = "wm8776", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm8776_spi_probe, + .suspend = wm8776_spi_suspend, + .resume = wm8776_spi_resume, + .remove = __devexit_p(wm8776_spi_remove), +}; +#endif /* CONFIG_SPI_MASTER */ + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static __devinit int wm8776_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct wm8776_priv *wm8776; + struct snd_soc_codec *codec; + + wm8776 = kzalloc(sizeof(struct wm8776_priv), GFP_KERNEL); + if (wm8776 == NULL) + return -ENOMEM; + + codec = &wm8776->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(i2c, wm8776); + codec->control_data = i2c; + + codec->dev = &i2c->dev; + + return wm8776_register(wm8776); +} + +static __devexit int wm8776_i2c_remove(struct i2c_client *client) +{ + struct wm8776_priv *wm8776 = i2c_get_clientdata(client); + wm8776_unregister(wm8776); + return 0; +} + +#ifdef CONFIG_PM +static int wm8776_i2c_suspend(struct i2c_client *i2c, pm_message_t msg) +{ + return snd_soc_suspend_device(&i2c->dev); +} + +static int wm8776_i2c_resume(struct i2c_client *i2c) +{ + return snd_soc_resume_device(&i2c->dev); +} +#else +#define wm8776_i2c_suspend NULL +#define wm8776_i2c_resume NULL +#endif + +static const struct i2c_device_id wm8776_i2c_id[] = { + { "wm8776", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, wm8776_i2c_id); + +static struct i2c_driver wm8776_i2c_driver = { + .driver = { + .name = "wm8776", + .owner = THIS_MODULE, + }, + .probe = wm8776_i2c_probe, + .remove = __devexit_p(wm8776_i2c_remove), + .suspend = wm8776_i2c_suspend, + .resume = wm8776_i2c_resume, + .id_table = wm8776_i2c_id, +}; +#endif + +static int __init wm8776_modinit(void) +{ + int ret; +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&wm8776_i2c_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8776 I2C driver: %d\n", + ret); + } +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&wm8776_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register WM8776 SPI driver: %d\n", + ret); + } +#endif + return 0; +} +module_init(wm8776_modinit); + +static void __exit wm8776_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&wm8776_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&wm8776_spi_driver); +#endif +} +module_exit(wm8776_exit); + +MODULE_DESCRIPTION("ASoC WM8776 driver"); +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/wm8776.h b/sound/soc/codecs/wm8776.h new file mode 100644 index 0000000..6606d25 --- /dev/null +++ b/sound/soc/codecs/wm8776.h @@ -0,0 +1,51 @@ +/* + * wm8776.h -- WM8776 ASoC driver + * + * Copyright 2009 Wolfson Microelectronics plc + * + * Author: Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM8776_H +#define _WM8776_H + +/* Registers */ + +#define WM8776_HPLVOL 0x00 +#define WM8776_HPRVOL 0x01 +#define WM8776_HPMASTER 0x02 +#define WM8776_DACLVOL 0x03 +#define WM8776_DACRVOL 0x04 +#define WM8776_DACMASTER 0x05 +#define WM8776_PHASESWAP 0x06 +#define WM8776_DACCTRL1 0x07 +#define WM8776_DACMUTE 0x08 +#define WM8776_DACCTRL2 0x09 +#define WM8776_DACIFCTRL 0x0a +#define WM8776_ADCIFCTRL 0x0b +#define WM8776_MSTRCTRL 0x0c +#define WM8776_PWRDOWN 0x0d +#define WM8776_ADCLVOL 0x0e +#define WM8776_ADCRVOL 0x0f +#define WM8776_ALCCTRL1 0x10 +#define WM8776_ALCCTRL2 0x11 +#define WM8776_ALCCTRL3 0x12 +#define WM8776_NOISEGATE 0x13 +#define WM8776_LIMITER 0x14 +#define WM8776_ADCMUX 0x15 +#define WM8776_OUTMUX 0x16 +#define WM8776_RESET 0x17 + +#define WM8776_CACHEREGNUM 0x17 + +#define WM8776_DAI_DAC 0 +#define WM8776_DAI_ADC 1 + +extern struct snd_soc_dai wm8776_dai[]; +extern struct snd_soc_codec_device soc_codec_dev_wm8776; + +#endif -- 1.6.3.3

1 0

[alsa-devel] [RFC] [PATCH] ASoC: OMAP: full duplex mode fix
by Janusz Krzysztofik 06 Aug '09

06 Aug '09

This patch tries to correct the problem of full duplex mode not working over a single McBSP based CPU DAI. 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-mcbsp.c.orig 2009-08-01 02:40:45.000000000 +0200 +++ linux-2.6.31-rc5/sound/soc/omap/omap-mcbsp.c 2009-08-03 03:28:07.000000000 +0200 @@ -183,6 +183,7 @@ static int omap_mcbsp_dai_trigger(struct struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); + u16 buf; int err = 0; switch (cmd) { @@ -191,6 +192,14 @@ static int omap_mcbsp_dai_trigger(struct case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: if (!mcbsp_data->active++) omap_mcbsp_start(mcbsp_data->bus_id); + else if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + /* looks like capture already in progress, + * start playback by taking it out of error condition */ + omap_mcbsp_pollwrite(mcbsp_data->bus_id, 0x0); + else + /* looks like playback already in progress, + * start capture by taking it out of error condition */ + omap_mcbsp_pollread(mcbsp_data->bus_id, &buf); break; case SNDRV_PCM_TRIGGER_STOP:

5 15

[alsa-devel] ASoC: Patches for Freescale MXC SoCs (SSI, DMA and machine).
by javier Martin 05 Aug '09

05 Aug '09

Here comes a series of three patches which currently add audio support for MX1x and MX2x platforms. It has been tested in our custom Visstrim_m10 board based on an i.MX27 processor. All patches are based on stable 2.6.30 kernel and each patch applies over the previous one. [alsa-devel] [PATCH 1/3] ASoC: add DMA platform driver for MX1x and MX2x [alsa-devel] [PATCH 2/3] ASoC: add DAI platform ssi driver for MXC [alsa-devel] [PATCH 3/3] ASoC: add machine driver for i.mx27_visstrim_m10 board -- Javier Martin Vista Silicon S.L. Universidad de Cantabria CDTUC - FASE C - Oficina S-345 Avda de los Castros s/n 39005- Santander. Cantabria. Spain +34 942 25 32 60 www.vista-silicon.com

2 3

[alsa-devel] [PATCH] ALSA: ASoC: cs4270: move power management hooks to snd_soc_codec_device
by Daniel Mack 05 Aug '09

05 Aug '09

Power management for the cs4270 codec is currently implemented as part of the i2c_driver struct. The disadvantage of doing it this way is that the callbacks registered in the snd_soc_card struct are called _before_ the codec's callbacks. That doesn't work, because the snd_soc_card callbacks will most likely switch down the codec's power domains or pull the reset GPIOs, and hence make the i2c communication bail out. Fix this by binding the suspend and resume code to the snd_soc_codec_device driver model. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Timur Tabi <timur(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs4270.c | 55 ++++++++++++++++++++++----------------------- 1 files changed, 27 insertions(+), 28 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index a32b822..69b794b 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -791,6 +791,22 @@ static struct i2c_device_id cs4270_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs4270_id); +/* + * cs4270_i2c_driver - I2C device identification + * + * This structure tells the I2C subsystem how to identify and support a + * given I2C device type. + */ +static struct i2c_driver cs4270_i2c_driver = { + .driver = { + .name = "cs4270", + .owner = THIS_MODULE, + }, + .id_table = cs4270_id, + .probe = cs4270_i2c_probe, + .remove = cs4270_i2c_remove, +}; + #ifdef CONFIG_PM /* This suspend/resume implementation can handle both - a simple standby @@ -802,19 +818,18 @@ MODULE_DEVICE_TABLE(i2c, cs4270_id); * and all registers are written back to the hardware when resuming. */ -static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg) +static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg) { - struct cs4270_private *cs4270 = i2c_get_clientdata(client); - struct snd_soc_codec *codec = &cs4270->codec; + struct snd_soc_codec *codec = cs4270_codec; int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; return snd_soc_write(codec, CS4270_PWRCTL, reg); } -static int cs4270_i2c_resume(struct i2c_client *client) +static int cs4270_soc_resume(struct platform_device *pdev) { - struct cs4270_private *cs4270 = i2c_get_clientdata(client); - struct snd_soc_codec *codec = &cs4270->codec; + struct snd_soc_codec *codec = cs4270_codec; + struct i2c_client *i2c_client = codec->control_data; int reg; /* In case the device was put to hard reset during sleep, we need to @@ -825,7 +840,7 @@ static int cs4270_i2c_resume(struct i2c_client *client) for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) { u8 val = snd_soc_read(codec, reg); - if (i2c_smbus_write_byte_data(client, reg, val)) { + if (i2c_smbus_write_byte_data(i2c_client, reg, val)) { dev_err(codec->dev, "i2c write failed\n"); return -EIO; } @@ -838,29 +853,11 @@ static int cs4270_i2c_resume(struct i2c_client *client) return snd_soc_write(codec, CS4270_PWRCTL, reg); } #else -#define cs4270_i2c_suspend NULL -#define cs4270_i2c_resume NULL +#define cs4270_soc_suspend NULL +#define cs4270_soc_resume NULL #endif /* CONFIG_PM */ /* - * cs4270_i2c_driver - I2C device identification - * - * This structure tells the I2C subsystem how to identify and support a - * given I2C device type. - */ -static struct i2c_driver cs4270_i2c_driver = { - .driver = { - .name = "cs4270", - .owner = THIS_MODULE, - }, - .id_table = cs4270_id, - .probe = cs4270_i2c_probe, - .remove = cs4270_i2c_remove, - .suspend = cs4270_i2c_suspend, - .resume = cs4270_i2c_resume, -}; - -/* * ASoC codec device structure * * Assign this variable to the codec_dev field of the machine driver's @@ -868,7 +865,9 @@ static struct i2c_driver cs4270_i2c_driver = { */ struct snd_soc_codec_device soc_codec_device_cs4270 = { .probe = cs4270_probe, - .remove = cs4270_remove + .remove = cs4270_remove, + .suspend = cs4270_soc_suspend, + .resume = cs4270_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_device_cs4270); -- 1.6.3.3

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[alsa-devel] [PATCH] ASoC: MPC5200: Support for buffer wrap around
by John Bonesio 05 Aug '09

05 Aug '09

Here's another appl_ptr related patch to the alsa soc mpc5200 ac97 driver. This time the patch is to fix a problem with not correctly handling appl_ptr wrapping back around to the beginning of the buffer. I haven't yet seen a symptom related to this. This came up when I was chatting with Jon Smirl about another bug I was examining. Right now there is another problem that is preventing me from playing really long audio streams. Let me know if this is the wrong approach. - John The code in psc_dma_bcom_enqueue_tx() didn't account for the fact that s->runtime->control->appl_ptr can wrap around to the beginning of the buffer. This change fixes this problem. Signed-off-by: John Bonesio <bones(a)secretlab.ca> --- sound/soc/fsl/mpc5200_dma.c | 17 +++++++++++++++++ 1 files changed, 17 insertions(+), 0 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index cfe0ea4..2551c58 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -70,6 +70,23 @@ static void psc_dma_bcom_enqueue_next_buffer(struct psc_dma_stream *s) static void psc_dma_bcom_enqueue_tx(struct psc_dma_stream *s) { + if (s->appl_ptr > s->runtime->control->appl_ptr) { + /* + * In this case s->runtime->control->appl_ptr has wrapped around. + * Play the data to the end of the boundary, then wrap our own + * appl_ptr back around. + */ + while (s->appl_ptr < s->runtime->boundary) { + if (bcom_queue_full(s->bcom_task)) + return; + + s->appl_ptr += s->period_size; + + psc_dma_bcom_enqueue_next_buffer(s); + } + s->appl_ptr -= s->runtime->boundary; + } + while (s->appl_ptr < s->runtime->control->appl_ptr) { if (bcom_queue_full(s->bcom_task))

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[PATCH] ALSA: ASoC: cs4270: move power management hooks to snd_soc_codec_device
by Daniel Mack 05 Aug '09

05 Aug '09

Power management for the cs4270 codec is currently implemented as part of the i2c_driver struct. The disadvantage of doing it this way is that the callbacks registered in the snd_soc_card struct are called _before_ the codec's callbacks. That doesn't work, because the snd_soc_card callbacks will most likely switch down the codec's power domains or pull the reset GPIOs, and hence make the i2c communication bail out. Fix this by binding the suspend and resume code to the snd_soc_codec_device driver model and let the I2C functions only call the SoC core function for resume and suspend, which do nothing currently but will do later. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Timur Tabi <timur(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cs4270.c | 27 +++++++++++++++++++++++---- 1 files changed, 23 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index a32b822..ca1e24a 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -806,15 +806,30 @@ static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg) { struct cs4270_private *cs4270 = i2c_get_clientdata(client); struct snd_soc_codec *codec = &cs4270->codec; - int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; - return snd_soc_write(codec, CS4270_PWRCTL, reg); + return snd_soc_suspend_device(codec->dev); } static int cs4270_i2c_resume(struct i2c_client *client) { struct cs4270_private *cs4270 = i2c_get_clientdata(client); struct snd_soc_codec *codec = &cs4270->codec; + + return snd_soc_resume_device(codec->dev); +} + +static int cs4270_soc_suspend(struct platform_device *pdev, pm_message_t mesg) +{ + struct snd_soc_codec *codec = cs4270_codec; + int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL; + + return snd_soc_write(codec, CS4270_PWRCTL, reg); +} + +static int cs4270_soc_resume(struct platform_device *pdev) +{ + struct snd_soc_codec *codec = cs4270_codec; + struct i2c_client *i2c_client = codec->control_data; int reg; /* In case the device was put to hard reset during sleep, we need to @@ -825,7 +840,7 @@ static int cs4270_i2c_resume(struct i2c_client *client) for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) { u8 val = snd_soc_read(codec, reg); - if (i2c_smbus_write_byte_data(client, reg, val)) { + if (i2c_smbus_write_byte_data(i2c_client, reg, val)) { dev_err(codec->dev, "i2c write failed\n"); return -EIO; } @@ -840,6 +855,8 @@ static int cs4270_i2c_resume(struct i2c_client *client) #else #define cs4270_i2c_suspend NULL #define cs4270_i2c_resume NULL +#define cs4270_soc_suspend NULL +#define cs4270_soc_resume NULL #endif /* CONFIG_PM */ /* @@ -868,7 +885,9 @@ static struct i2c_driver cs4270_i2c_driver = { */ struct snd_soc_codec_device soc_codec_device_cs4270 = { .probe = cs4270_probe, - .remove = cs4270_remove + .remove = cs4270_remove, + .suspend = cs4270_soc_suspend, + .resume = cs4270_soc_resume, }; EXPORT_SYMBOL_GPL(soc_codec_device_cs4270); -- 1.6.3.3

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[alsa-devel] [PATCH - amixer] listen to D-Bus for volume-up, volume-down and mute signals.
by Łukasz Stelmach 05 Aug '09

05 Aug '09

The patch introduces D-Bus support in amixer. With --dbus option given amixer works continiously (like with the --stdin) waiting for org.freedesktop.Hal.Device.Condition with the first argument "ButtonPressed" and the second being one of: "volume-up", "volume-down" and "mute". These are emmited by the HAL input helper. A single amixer process can controll one mixer item. There are options to choose a specific item to controll, an input device to listen and a single adjustment step size. Signed-off-by: Łukasz Stelmach <stlman(a)poczta.fm> --- amixer/Makefile.am | 3 +- amixer/amixer.1 | 18 ++++++ amixer/amixer.c | 171 ++++++++++++++++++++++++++++++++++++++++++++++++++++ configure.in | 12 ++++ 4 files changed, 203 insertions(+), 1 deletions(-) diff --git a/amixer/Makefile.am b/amixer/Makefile.am index fcd0e81..fa951a6 100644 --- a/amixer/Makefile.am +++ b/amixer/Makefile.am @@ -1,5 +1,6 @@ INCLUDES = -I$(top_srcdir)/include -LDADD = -lm +LDADD = -lm @DBUS_LIBS@ +CFLAGS += @DBUS_CFLAGS@ # LDFLAGS = -static # CFLAGS += -g -Wall diff --git a/amixer/amixer.1 b/amixer/amixer.1 index b1ac323..bc160be 100644 --- a/amixer/amixer.1 +++ b/amixer/amixer.1 @@ -88,6 +88,24 @@ Select the card number to control. The device name created from this parameter has syntax 'hw:N' where N is specified card number. .TP +\fI\-\-dbus[=\fISCONTROL\fP] + +Connect to system D-Bus and wait for \fIorg.freedesktop.Hal.Device.Condition\fP +signals with ButtonPressed as the first argument and one of: \fIvolume-up\fP, +\fIvolume-down\fP or \fImute\fP as the second one. Control the selected +\fISCONTROL\fP item. The default item is 'Master,0'. + +.TP +\fI\-\-dbus-path=<\fIUDI\fP> + +Choose a particular input device that emits control events. + +.TP +\fI\-\-dbus-step=<\fIN\fP> + +Set an amount by wich to increase or decrease volume upon event. + +.TP \fI\-D\fP device Select the device name to control. The default control name is 'default'. diff --git a/amixer/amixer.c b/amixer/amixer.c index 9620721..31de674 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -18,6 +18,8 @@ * */ +#include "aconfig.h" + #include <stdio.h> #include <stdlib.h> #include <string.h> @@ -31,6 +33,10 @@ #include <sys/poll.h> #include "amixer.h" +#ifdef HAVE_DBUS +#include <dbus/dbus.h> +#endif /* HAVE_DBUS */ + #define LEVEL_BASIC (1<<0) #define LEVEL_INACTIVE (1<<1) #define LEVEL_ID (1<<2) @@ -43,6 +49,12 @@ static int ignore_error = 0; static struct snd_mixer_selem_regopt smixer_options; static char card[64] = "default"; +#ifdef HAVE_DBUS +static char dbus_sctl[64] = "Master,0"; +static char *dbus_path = NULL; +static char dbus_step[8] = "1"; +#endif /* HAVE_DBUS */ + static void error(const char *fmt,...) { va_list va; @@ -60,6 +72,11 @@ static int help(void) printf("\nAvailable options:\n"); printf(" -h,--help this help\n"); printf(" -c,--card N select the card\n"); +#ifdef HAVE_DBUS + printf(" --dbus[=sID] Connect to the system D-Bus and wait for volume messages\n"); + printf(" --dbus-path UDI Choose a particular input device (udi)\n"); + printf(" --dbus-step N Set the step size (default: 1)\n"); +#endif /* HAVE_DBUS */ printf(" -D,--device N select the device, default '%s'\n", card); printf(" -d,--debug debug mode\n"); printf(" -n,--nocheck do not perform range checking\n"); @@ -1887,15 +1904,147 @@ static int exec_stdin(void) return 0; } +#ifdef HAVE_DBUS + +#define OPT_DBUS 0x81 +#define OPT_DBUS_PATH 0x82 +#define OPT_DBUS_STEP 0x83 + +void* get_dbus_arg(DBusMessageIter* args, int type, void** out) { + if (type != dbus_message_iter_get_arg_type(args)) + return NULL; + dbus_message_iter_get_basic(args, out); + dbus_message_iter_next(args); + return *out; +} + +int exec_dbus(void) +{ + char *sargv[MAX_ARGS]; + int sargc; + const char* s; + int stepl; + DBusError error; + DBusMessage* msg; + DBusConnection *conn; + DBusMessageIter args; + + stepl = strlen(dbus_step); + dbus_step[stepl+1] = '\0'; + + dbus_error_init (&error); + conn = dbus_bus_get (DBUS_BUS_SYSTEM, &error); + + if(!conn) { + fprintf(stderr, "dbus:%s: %s\n", + error.name, error.message); + return 1; + } + + dbus_bus_add_match(conn, + "type='signal'," + "interface='org.freedesktop.Hal.Device'," + "member='Condition'", + &error); + /* The first message is a NameOwnerChanged signal, ignore it. */ + msg = dbus_connection_pop_message(conn); + dbus_message_unref(msg); + + if(dbus_error_is_set(&error)) { + fprintf(stderr, "dbus:match error:%s\n", + error.message); + return 1; + } + + while (1) { + sargv[0] = dbus_sctl; sargc = 1; + + if (!(msg = dbus_connection_pop_message(conn))) { + dbus_connection_read_write(conn, -1); + continue; + } + + s = dbus_message_get_interface(msg); + if(strncmp("org.freedesktop.Hal.Device", s, 26)) { + if(debugflag) + fprintf(stderr, "dbus:unsupported interface:%s\n", s); + goto badmessage; + } + + if(dbus_path && + (s=dbus_message_get_path(msg)) && + strncmp(dbus_path, s, strlen(s))) { + if(debugflag) + fprintf(stderr, "dbus:not the selected udi:%s\n", s); + goto badmessage; + } + + if(!dbus_message_iter_init(msg, &args)) { + if(debugflag) + fprintf(stderr, "dbus:message has no arguments\n"); + goto badmessage; + } + + if (strncmp(get_dbus_arg(&args, DBUS_TYPE_STRING, (void*)&s), "ButtonPressed", 13)) { + if(debugflag) + fprintf(stderr, "dbus:not a ButtonPressed event\n"); + goto badmessage; + } + + if (!get_dbus_arg(&args, DBUS_TYPE_STRING, (void*)&s)) { + if(debugflag) + fprintf(stderr, "dbus:invalid argument type\n"); + goto badmessage; + } + + /* static int sset(unsigned int argc, char *argv[], int roflag, int keep_handle) */ + if (!strncmp(s, "volume-up", 9)) { + if(debugflag) + fprintf(stderr, "dbus:volume-up\n"); + dbus_step[stepl]='+'; + sargv[1] = dbus_step; + } else if (!strncmp(s, "volume-down", 11)) { + if(debugflag) + fprintf(stderr, "dbus:volume-down\n"); + dbus_step[stepl]='-'; + sargv[1] = dbus_step; + } else if (!strncmp(s, "mute", 4)) { + if(debugflag) + fprintf(stderr, "dbus:mute (toggle)\n"); + sargv[1] = "toggle"; + } else { + if(debugflag) + fprintf(stderr, "dbus:invalid argument value\n"); + goto badmessage; + } + sset(++sargc, sargv, 0, 1); +badmessage: + dbus_message_unref(msg); + } + + if(conn) + dbus_connection_unref(conn); + return 0; +} +#endif /* HAVE_DBUS */ int main(int argc, char *argv[]) { int morehelp, level = 0; int read_stdin = 0; +#ifdef HAVE_DBUS + int read_dbus = 0; + int t; +#endif /* HAVE_DBUS */ static const struct option long_option[] = { {"help", 0, NULL, 'h'}, {"card", 1, NULL, 'c'}, +#ifdef HAVE_DBUS + {"dbus", optional_argument, NULL, OPT_DBUS}, + {"dbus-path", 1, NULL, OPT_DBUS_PATH}, + {"dbus-step", 1, NULL, OPT_DBUS_STEP}, +#endif /* HAVE_DBUS */ {"device", 1, NULL, 'D'}, {"quiet", 0, NULL, 'q'}, {"inactive", 0, NULL, 'i'}, @@ -1964,6 +2113,24 @@ int main(int argc, char *argv[]) case 's': read_stdin = 1; break; +#ifdef HAVE_DBUS + case OPT_DBUS: + if(optarg) { + strncpy(dbus_sctl, optarg, sizeof(dbus_sctl)-1); + dbus_sctl[sizeof(dbus_sctl)-1] = '\0'; + } + read_dbus = 1; + break; + case OPT_DBUS_PATH: + dbus_path=optarg; + break; + case OPT_DBUS_STEP: + t=atoi(optarg); + if(t > 0 && t <= 999999) { + snprintf(dbus_step, sizeof(dbus_step), "%d", atoi(optarg)); + } + break; +#endif /* HAVE_DBUS */ default: fprintf(stderr, "Invalid switch or option needs an argument.\n"); morehelp++; @@ -1978,6 +2145,10 @@ int main(int argc, char *argv[]) if (read_stdin) return exec_stdin(); +#ifdef HAVE_DBUS + if(read_dbus) + return exec_dbus(); +#endif /* HAVE_DBUS */ if (argc - optind <= 0) { return selems(LEVEL_BASIC | level) ? 1 : 0; } diff --git a/configure.in b/configure.in index 1349ff3..717d6c5 100644 --- a/configure.in +++ b/configure.in @@ -7,6 +7,9 @@ AM_INIT_AUTOMAKE(alsa-utils, 1.0.20) AM_GNU_GETTEXT([external]) AM_GNU_GETTEXT_VERSION([0.15]) +dnl required versions of other packages +m4_define([dbus_required_version], [1.2.0]) + dnl Checks for programs. dnl try to gues cross-compiler if not set @@ -74,6 +77,15 @@ AC_ARG_ENABLE(alsamixer, no) alsamixer=false ;; *) AC_MSG_ERROR(bad value ${enableval} for --enable-alsamixer) ;; esac],[alsamixer=true]) +AC_ARG_WITH(dbus, [ --with-dbus Enable D-Bus support in amixer]) + if test "x$with_dbus" != xno; then + PKG_CHECK_MODULES(DBUS, dbus-1 >= dbus_required_version, + have_dbus=yes, + have_dbus=no) + fi + if test "x$have_dbus" = xyes; then + AC_DEFINE(HAVE_DBUS, 1, [Define to 1 if amixer is to support D-Bus]) + fi fi AM_CONDITIONAL(ALSAMIXER, test x$alsamixer = xtrue) -- 1.6.3.3 ---------------------------------------------------------------------- Zostan Dziewczyna lub Chlopakiem lata! 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