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[alsa-devel] [PATCH] ASoC: wm0010: Enable 44.1kHz support
by Mark Brown 25 Aug '12

25 Aug '12

With appropriate clocking configuration the WM0010 driver supports 44.1kHz audio; enable that. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index e1315e0..f820b57 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -722,7 +722,7 @@ static struct snd_soc_codec_driver soc_codec_dev_wm0010 = { .num_dapm_routes = ARRAY_SIZE(wm0010_dapm_routes), }; -#define WM0010_RATES (SNDRV_PCM_RATE_48000) +#define WM0010_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) #define WM0010_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE |\ SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE |\ SNDRV_PCM_FMTBIT_S32_LE) -- 1.7.10.4

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Re: [alsa-devel] Logitech USB headset not working in 3.6-rc3
by Daniel Mack 25 Aug '12

25 Aug '12

On 25.08.2012 14:07, Bruno Wolff III wrote: > On Sat, Aug 25, 2012 at 14:02:51 +0200, > Daniel Mack <zonque(a)gmail.com> wrote: >> >> Can you revert commit e9ba389c5 ("ALSA: usb-audio: Fix >> scheduling-while-atomic bug in PCM capture stream") and see if that > > I can try that, but it takes a long time to build a new kernel on my > old hardware. > >> helps? If not, can you summarize again which kernels still work for you >> and which don't? > > The latest kernel that works is 3.6.0-0.rc2.git1.2.fc18. The earliest that > doesn't work is 3.6.0-0.rc2.git2.1.fc18. > The report you sent doesn't look like it could be caused by e9ba389c5. It fixes a kernel Ooops. But as it is the only relevant patch in that area, it would be interesting if reverting it fixes anything. Btw - thanks a lot for testing -rc kernels, much appreciated! Daniel

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Re: [alsa-devel] Logitech USB headset not working in 3.6-rc3
by Daniel Mack 25 Aug '12

25 Aug '12

On 25.08.2012 13:54, Bruno Wolff III wrote: > On Sat, Aug 25, 2012 at 06:17:49 -0500, > Bruno Wolff III <bruno(a)wolff.to> wrote: >> >> I see this behavior on three different computers. Rebooting into an old >> enough kernel gets things working again. The older kernels also have >> older initramfs images. I could rebuild those to see if there is some >> issue there instead of the kernels themselves? > > I tested one of the older kernels (where the headset still worked) with a > rebuilt initramfs and the headset still worked. So it likely isn't an issue > with something in the initramfs images. > Can you revert commit e9ba389c5 ("ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream") and see if that helps? If not, can you summarize again which kernels still work for you and which don't? Daniel

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[alsa-devel] multistream singlechanel DACs
by Rafal Powierski 25 Aug '12

25 Aug '12

Developers, how to enable multiple playback streaming?: I am manipulating bios verb pin table, for special analog hardware, using ALC886 and kubuntu / 64bit. I need a number of separate stereo (only stereo) i/o's I have correctly configured inputs: 4. single stereo adc port-C 1Ah 5. single spdif input 1Fh I have configured 3 independent stereo outputs, but ALSA is joining them into one pcmC1D0p device. 1.port-A,15h 2.port-E,1Bh 3.port-D,14h I have tried this on win7 and I got all associations as a separate devices. in my kubuntu 12.4LTS I have only one playback device. Mixer is showing even more then needed number of volume controls, But there is only one playback device. it seams, all 3 outputs are connected to single DAC. I have tried to buildup more input devices - this is working good. I can have up to 3 (pcmC1D0c, pcmC1D1c pcmC1D2c) I can also have 2 outputs if I enable spdif-out. But there is no separate DAC assignment to separate devices resulting from my config. Is it possible to get independent DACs devices under ALSA? or else is no such configuration possible and I have to use single multiple channel device?

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Re: [alsa-devel] Logitech USB headset not working in 3.6-rc3
by Daniel Mack 25 Aug '12

25 Aug '12

Hi, On 24.08.2012 21:08, Josh Boyer wrote: > We've had a report[1] that the Logitech USB headset 0003:046D:0A0C isn't > working with 3.6-rc3. It seems the last working kernel was based on > commit 10c63c9, and it first stopped working with a kernel based on > commit 23dcfa6. There are only a few ALSA commits between those > revisions, so hopefully this is something that is fairly easy to > identify. The only commit to USB audio in that set is: [...] > !!ALSA/HDA dmesg > !!------------------ > > [ 38.190306] SELinux: initialized (dev configfs, type configfs), uses genfs_contexts > [ 38.229616] snd_hda_intel 0000:00:1b.0: irq 66 for MSI/MSI-X > [ 38.270699] ALSA sound/usb/mixer.c:866 6:0: cannot get min/max values for control 2 (id 6) > [ 38.274097] ALSA sound/usb/mixer.c:866 1:0: cannot get min/max values for control 2 (id 1) > [ 38.276753] ALSA sound/usb/mixer.c:866 2:0: cannot get min/max values for control 2 (id 2) > [ 38.279322] ALSA sound/pci/hda/hda_auto_parser.c:322 autoconfig: line_outs=1 (0xe/0x0/0x0/0x0/0x0) type:line > [ 38.279326] ALSA sound/pci/hda/hda_auto_parser.c:326 speaker_outs=1 (0x11/0x0/0x0/0x0/0x0) > [ 38.279329] ALSA sound/pci/hda/hda_auto_parser.c:330 hp_outs=1 (0xd/0x0/0x0/0x0/0x0) > [ 38.279331] ALSA sound/pci/hda/hda_auto_parser.c:331 mono: mono_out=0x0 > [ 38.279333] ALSA sound/pci/hda/hda_auto_parser.c:335 inputs: > [ 38.279336] ALSA sound/pci/hda/hda_auto_parser.c:339 Mic=0x10 > [ 38.279342] ALSA sound/pci/hda/hda_auto_parser.c:339 Line=0xf > [ 38.285983] usbcore: registered new interface driver snd-usb-audio Also, according to this dmesg, the device is probed just fine (which the commit you mentioned wouldn't change anything about though). So what does "isn't working anymore" refer to precisely? Are there any more dmesg entries generated once the stream is started? Daniel

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Re: [alsa-devel] Logitech USB headset not working in 3.6-rc3
by Daniel Mack 24 Aug '12

24 Aug '12

On Fri, Aug 24, 2012 at 9:08 PM, Josh Boyer <jwboyer(a)redhat.com> wrote: > Hi All, > > We've had a report[1] that the Logitech USB headset 0003:046D:0A0C isn't > working with 3.6-rc3. It seems the last working kernel was based on > commit 10c63c9, and it first stopped working with a kernel based on > commit 23dcfa6. There are only a few ALSA commits between those > revisions, so hopefully this is something that is fairly easy to > identify. The only commit to USB audio in that set is: > > commit e9ba389c5ffc4dd29dfe17e00e48877302111135 > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Wed Aug 15 12:32:00 2012 +0200 > > ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM capture stream > > > I've CC'd the reporter and attached the alsa-info is below. Does it work again once you revert that commit? Daniel > [1] https://bugzilla.redhat.com/show_bug.cgi?id=851619 > > upload=true&script=true&cardinfo= > !!################################ > !!ALSA Information Script v 0.4.60 > !!################################ > > !!Script ran on: Fri Aug 24 18:35:42 UTC 2012 > > > !!Linux Distribution > !!------------------ > > Fedora release 18 (Rawhide) Fedora release 18 (Rawhide) NAME=Fedora ID=fedora PRETTY_NAME="Fedora 18 (Rawhide)" CPE_NAME="cpe:/o:fedoraproject:fedora:18" Fedora release 18 (Rawhide) Fedora release 18 (Rawhide) > > > !!DMI Information > !!--------------- > > Manufacturer: Dell Inc. > Product Name: Precision WorkStation 690 > Product Version: > > > !!Kernel Information > !!------------------ > > Kernel release: 3.6.0-0.rc3.git0.1.fc18.x86_64 > Operating System: GNU/Linux > Architecture: x86_64 > Processor: x86_64 > SMP Enabled: Yes > > > !!ALSA Version > !!------------ > > Driver version: 1.0.25 > Library version: 1.0.25 > Utilities version: 1.0.25 > > > !!Loaded ALSA modules > !!------------------- > > snd_hda_intel > snd_usb_audio > > > !!Sound Servers on this system > !!---------------------------- > > Pulseaudio: > Installed - Yes (/usr/bin/pulseaudio) > Running - Yes > > aRts: > Installed - Yes (/usr/bin/artsd) > Running - No > > Jack: > Installed - Yes (/usr/bin/jackd) > Running - No > > > !!Soundcards recognised by ALSA > !!----------------------------- > > 0 [Intel ]: HDA-Intel - HDA Intel > HDA Intel at 0xfcffc000 irq 66 > 1 [Headset ]: USB-Audio - Logitech USB Headset > Logitech Logitech USB Headset at usb-0000:00:1d.7-6.4, full speed > > > !!PCI Soundcards installed in the system > !!-------------------------------------- > > 00:1b.0 Audio device: Intel Corporation 631xESB/632xESB High Definition Audio Controller (rev 09) > > > !!Advanced information - PCI Vendor/Device/Subsystem ID's > !!-------------------------------------------------------- > > 00:1b.0 0403: 8086:269a (rev 09) > Subsystem: 1028:01c0 > > > !!Loaded sound module options > !!-------------------------- > > !!Module: snd_hda_intel > align_buffer_size : -1 > bdl_pos_adj : 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 > beep_mode : 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 > 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_msi : -1 > 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 > model : (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) > patch : (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) > position_fix : 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 > power_save : 0 > power_save_controller : Y > probe_mask : -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 > probe_only : 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 > single_cmd : N > snoop : Y > > !!Module: snd_usb_audio > async_unlink : Y > device_setup : 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 > 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 > 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) > ignore_ctl_error : N > 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 > nrpacks : 8 > pid : -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 > vid : -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 > > > !!HDA-Intel Codec information > !!--------------------------- > --startcollapse-- > > Codec: SigmaTel STAC9200 > Address: 0 > AFG Function Id: 0x1 (unsol 1) > Vendor Id: 0x83847690 > Subsystem Id: 0x102801c0 > Revision Id: 0x102201 > 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: N/A > Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 > GPIO: io=4, 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 > Node 0x02 [Audio Output] wcaps 0xd0401: Stereo > Device: name="STAC92xx Analog", type="Audio", device=0 > Converter: stream=8, channel=0 > Power states: > Power: setting=D0, actual=D0 > Delay: 13 samples > Node 0x03 [Audio Input] wcaps 0x1d0541: Stereo > Device: name="STAC92xx Analog", type="Audio", device=0 > Converter: stream=4, channel=0 > SDI-Select: 0 > Power states: > Power: setting=D0, actual=D0 > Delay: 13 samples > Connection: 1 > 0x0a > Processing caps: benign=0, ncoeff=0 > Node 0x04 [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 > 0x08 > Node 0x05 [Audio Output] wcaps 0x40211: Stereo Digital > Converter: stream=0, channel=0 > Digital: > Digital category: 0x0 > PCM: > rates [0x1e0]: 44100 48000 88200 96000 > bits [0xe]: 16 20 24 > formats [0x5]: PCM AC3 > Delay: 4 samples > Node 0x06 [Vendor Defined Widget] wcaps 0xf30201: Stereo Digital > Delay: 3 samples > Node 0x07 [Audio Selector] wcaps 0x300901: Stereo R/L > Connection: 3 > 0x02* 0x08 0x0a > Node 0x08 [Pin Complex] wcaps 0x430681: Stereo Digital > Pincap 0x00010024: IN EAPD Detect > EAPD 0x0: > Pin Default 0x400001f0: [N/A] Line Out at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Unsolicited: tag=00, enabled=0 > Power states: > Power: setting=D0, actual=D0 > Delay: 3 samples > Node 0x09 [Pin Complex] wcaps 0x400301: Stereo Digital > Pincap 0x00000010: OUT > Pin Default 0x400001f1: [N/A] Line Out at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x1 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Connection: 2 > 0x05* 0x0a > Node 0x0a [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L > Control: name="Capture Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Control: name="Capture Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 > Amp-Out vals: [0x07 0x07] > Connection: 1 > 0x0c > Node 0x0b [Audio Selector] wcaps 0x300105: Stereo Amp-Out > Control: name="PCM Playback Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Control: name="PCM Playback Switch", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: N/A > Amp-Out vals: [0x9f 0x9f] > Connection: 1 > 0x07 > Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out > Control: name="Mux Capture Volume", index=0, device=0 > ControlAmp: chs=3, dir=Out, idx=0, ofs=0 > Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 > Amp-Out vals: [0x00 0x00] > Connection: 5 > 0x10* 0x0f 0x0e 0x0d 0x12 > Node 0x0d [Pin Complex] wcaps 0x400181: Stereo > Control: name="Front Headphone Jack", index=0, device=0 > Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense > Pin Default 0x0221401f: [Jack] HP Out at Ext Front > Conn = 1/8, Color = Green > DefAssociation = 0x1, Sequence = 0xf > Pin-ctls: 0xc0: OUT HP > Unsolicited: tag=01, enabled=1 > Connection: 1 > 0x0b > Node 0x0e [Pin Complex] wcaps 0x400181: Stereo > Control: name="Line Out Jack", index=0, device=0 > Pincap 0x0000003f: IN OUT HP Detect Trigger ImpSense > Pin Default 0x01014010: [Jack] Line Out at Ext Rear > Conn = 1/8, Color = Green > DefAssociation = 0x1, Sequence = 0x0 > Pin-ctls: 0x40: OUT > Unsolicited: tag=02, enabled=1 > Connection: 1 > 0x0b > Node 0x0f [Pin Complex] wcaps 0x400181: Stereo > Control: name="Line Jack", index=0, device=0 > Pincap 0x00000037: IN OUT Detect Trigger ImpSense > Pin Default 0x01813020: [Jack] Line In at Ext Rear > Conn = 1/8, Color = Blue > DefAssociation = 0x2, Sequence = 0x0 > Pin-ctls: 0x20: IN > Unsolicited: tag=04, enabled=1 > Connection: 1 > 0x0b > Node 0x10 [Pin Complex] wcaps 0x400181: Stereo > Control: name="Mic Jack", index=0, device=0 > Pincap 0x00001737: IN OUT Detect Trigger ImpSense > Vref caps: HIZ 50 GRD 80 > Pin Default 0x02a19021: [Jack] Mic at Ext Front > Conn = 1/8, Color = Pink > DefAssociation = 0x2, Sequence = 0x1 > Pin-ctls: 0x24: IN VREF_80 > Unsolicited: tag=03, enabled=1 > Connection: 1 > 0x0b > Node 0x11 [Pin Complex] wcaps 0x400104: Mono Amp-Out > Control: name="LFE Playback Volume", index=0, device=0 > ControlAmp: chs=1, dir=Out, idx=0, ofs=0 > Control: name="LFE Playback Switch", index=0, device=0 > ControlAmp: chs=1, dir=Out, idx=0, ofs=0 > Control: name="Speaker Phantom Jack", index=0, device=0 > Amp-Out caps: N/A > Amp-Out vals: [0x00] > Pincap 0x00000010: OUT > Pin Default 0x90100140: [Fixed] Speaker at Int N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0x4, Sequence = 0x0 > Misc = NO_PRESENCE > Pin-ctls: 0x00: > Connection: 1 > 0x13 > Node 0x12 [Pin Complex] wcaps 0x400001: Stereo > Pincap 0x00000020: IN > Pin Default 0x400001f2: [N/A] Line Out at Ext N/A > Conn = Unknown, Color = Unknown > DefAssociation = 0xf, Sequence = 0x2 > Misc = NO_PRESENCE > Pin-ctls: 0x20: IN > Node 0x13 [Audio Mixer] wcaps 0x200100: Mono > Connection: 1 > 0x07 > Node 0x14 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out > Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 > Amp-Out vals: [0x00] > --endcollapse-- > > > !!USB Mixer information > !!--------------------------- > --startcollapse-- > > USB Mixer: usb_id=0x046d0a0c, ctrlif=0, ctlerr=0 > Card: Logitech Logitech USB Headset at usb-0000:00:1d.7-6.4, full speed > Unit: 1 > Control: name="Speaker Playback Volume", index=0 > Info: id=1, control=2, cmask=0x3, channels=2, type="S16" > Volume: min=-10496, max=768, dBmin=-4100, dBmax=300 > Unit: 1 > Control: name="Speaker Playback Switch", index=0 > Info: id=1, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" > Volume: min=0, max=1, dBmin=0, dBmax=0 > Unit: 2 > Control: name="Mic Capture Volume", index=0 > Info: id=2, control=2, cmask=0x0, channels=1, type="S16" > Volume: min=4096, max=7424, dBmin=1600, dBmax=2900 > Unit: 2 > Control: name="Mic Capture Switch", index=0 > Info: id=2, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" > Volume: min=0, max=1, dBmin=0, dBmax=0 > Unit: 6 > Control: name="Mic Playback Volume", index=0 > Info: id=6, control=2, cmask=0x0, channels=1, type="S16" > Volume: min=-10496, max=-3072, dBmin=-4100, dBmax=-1200 > Unit: 6 > Control: name="Mic Playback Switch", index=0 > Info: id=6, control=1, cmask=0x0, channels=1, type="INV_BOOLEAN" > Volume: min=0, max=1, dBmin=0, dBmax=0 > --endcollapse-- > > > !!ALSA Device nodes > !!----------------- > > crw-rw----+ 1 root audio 116, 8 Aug 24 12:18 /dev/snd/controlC0 > crw-rw----+ 1 root audio 116, 4 Aug 24 12:18 /dev/snd/controlC1 > crw-rw----+ 1 root audio 116, 7 Aug 24 12:18 /dev/snd/hwC0D0 > crw-rw----+ 1 root audio 116, 6 Aug 24 12:19 /dev/snd/pcmC0D0c > crw-rw----+ 1 root audio 116, 5 Aug 24 12:19 /dev/snd/pcmC0D0p > crw-rw----+ 1 root audio 116, 3 Aug 24 12:19 /dev/snd/pcmC1D0c > crw-rw----+ 1 root audio 116, 2 Aug 24 12:30 /dev/snd/pcmC1D0p > crw-rw----+ 1 root audio 116, 1 Aug 24 12:18 /dev/snd/seq > crw-rw----+ 1 root audio 116, 33 Aug 24 12:18 /dev/snd/timer > > /dev/snd/by-id: > total 0 > drwxr-xr-x. 2 root root 60 Aug 24 12:18 . > drwxr-xr-x. 4 root root 260 Aug 24 12:18 .. > lrwxrwxrwx. 1 root root 12 Aug 24 12:18 usb-Logitech_Logitech_USB_Headset-00 -> ../controlC1 > > /dev/snd/by-path: > total 0 > drwxr-xr-x. 2 root root 80 Aug 24 12:18 . > drwxr-xr-x. 4 root root 260 Aug 24 12:18 .. > lrwxrwxrwx. 1 root root 12 Aug 24 12:18 pci-0000:00:1b.0 -> ../controlC0 > lrwxrwxrwx. 1 root root 12 Aug 24 12:18 pci-0000:00:1d.7-usb-0:6.4:1.0 -> ../controlC1 > > > !!ALSA configuration files > !!------------------------ > > !!System wide config file (/etc/asound.conf) > > # > # Place your global alsa-lib configuration here... > # > > > !!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 > card 1: Headset [Logitech USB Headset], device 0: USB Audio [USB Audio] > Subdevices: 0/1 > 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 > card 1: Headset [Logitech USB Headset], device 0: USB Audio [USB Audio] > Subdevices: 1/1 > Subdevice #0: subdevice #0 > > !!Amixer output > !!------------- > > !!-------Mixer controls for card 0 [Intel] > > Card hw:0 'Intel'/'HDA Intel at 0xfcffc000 irq 66' > Mixer name : 'SigmaTel STAC9200' > Components : 'HDA:83847690,102801c0,00102201' > Controls : 15 > Simple ctrls : 6 > Simple mixer control 'Master',0 > Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum > Playback channels: Mono > Limits: Playback 0 - 31 > Mono: Playback 31 [100%] [0.00dB] [off] > Simple mixer control 'PCM',0 > Capabilities: pvolume pswitch penum > Playback channels: Front Left - Front Right > Limits: Playback 0 - 31 > Mono: > Front Left: Playback 31 [100%] [0.00dB] [off] > Front Right: Playback 31 [100%] [0.00dB] [off] > Simple mixer control 'LFE',0 > Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum > Playback channels: Mono > Limits: Playback 0 - 31 > Mono: Playback 31 [100%] [0.00dB] [off] > Simple mixer control 'Capture',0 > Capabilities: cvolume cswitch penum > Capture channels: Front Left - Front Right > Limits: Capture 0 - 15 > Front Left: Capture 7 [47%] [10.50dB] [on] > Front Right: Capture 7 [47%] [10.50dB] [on] > Simple mixer control 'Input Source',0 > Capabilities: cenum > Items: 'Mic' 'Line' > Item0: 'Mic' > Simple mixer control 'Mux',0 > Capabilities: cvolume penum > Capture channels: Front Left - Front Right > Limits: Capture 0 - 4 > Front Left: Capture 0 [0%] [0.00dB] > Front Right: Capture 0 [0%] [0.00dB] > > !!-------Mixer controls for card 1 [Headset] > > Card hw:1 'Headset'/'Logitech Logitech USB Headset at usb-0000:00:1d.7-6.4, full speed' > Mixer name : 'USB Mixer' > Components : 'USB046d:0a0c' > Controls : 6 > Simple ctrls : 2 > Simple mixer control 'Speaker',0 > Capabilities: pvolume pswitch pswitch-joined penum > Playback channels: Front Left - Front Right > Limits: Playback 0 - 44 > Mono: > Front Left: Playback 35 [80%] [-6.00dB] [on] > Front Right: Playback 35 [80%] [-6.00dB] [on] > Simple mixer control 'Mic',0 > Capabilities: pvolume pvolume-joined cvolume cvolume-joined pswitch pswitch-joined cswitch cswitch-joined penum > Playback channels: Mono > Capture channels: Mono > Limits: Playback 0 - 29 Capture 0 - 13 > Mono: Playback 0 [0%] [-41.00dB] [off] Capture 13 [100%] [29.00dB] [on] > > > !!Alsactl output > !!------------- > > --startcollapse-- > state.Intel { > control.1 { > iface MIXER > name 'PCM Playback Volume' > value.0 31 > value.1 31 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 31' > dbmin -9999999 > dbmax 0 > dbvalue.0 0 > dbvalue.1 0 > } > } > control.2 { > iface MIXER > name 'PCM Playback Switch' > value.0 false > value.1 false > comment { > access 'read write' > type BOOLEAN > count 2 > } > } > control.3 { > iface MIXER > name 'Capture Volume' > value.0 7 > value.1 7 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 15' > dbmin 0 > dbmax 2250 > dbvalue.0 1050 > dbvalue.1 1050 > } > } > control.4 { > iface MIXER > name 'Capture Switch' > value.0 true > value.1 true > comment { > access 'read write' > type BOOLEAN > count 2 > } > } > control.5 { > iface MIXER > name 'LFE Playback Volume' > value 31 > comment { > access 'read write' > type INTEGER > count 1 > range '0 - 31' > dbmin -4650 > dbmax 0 > dbvalue.0 0 > } > } > control.6 { > iface MIXER > name 'LFE Playback Switch' > value false > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.7 { > iface MIXER > name 'Mux Capture Volume' > value.0 0 > value.1 0 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 4' > dbmin 0 > dbmax 4000 > dbvalue.0 0 > dbvalue.1 0 > } > } > control.8 { > iface MIXER > name 'Input Source' > value Mic > comment { > access 'read write' > type ENUMERATED > count 1 > item.0 Mic > item.1 Line > } > } > control.9 { > iface MIXER > name 'Master Playback Volume' > value 31 > comment { > access 'read write' > type INTEGER > count 1 > range '0 - 31' > dbmin -9999999 > dbmax 0 > dbvalue.0 0 > } > } > control.10 { > iface MIXER > name 'Master Playback Switch' > value false > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.11 { > iface CARD > name 'Line Out Jack' > value true > comment { > access read > type BOOLEAN > count 1 > } > } > control.12 { > iface CARD > name 'Front Headphone Jack' > value false > comment { > access read > type BOOLEAN > count 1 > } > } > control.13 { > iface CARD > name 'Speaker Phantom Jack' > value true > comment { > access read > type BOOLEAN > count 1 > } > } > control.14 { > iface CARD > name 'Mic Jack' > value false > comment { > access read > type BOOLEAN > count 1 > } > } > control.15 { > iface CARD > name 'Line Jack' > value false > comment { > access read > type BOOLEAN > count 1 > } > } > } > state.Headset { > control.1 { > iface MIXER > name 'Mic Playback Switch' > value false > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.2 { > iface MIXER > name 'Mic Playback Volume' > value 0 > comment { > access 'read write' > type INTEGER > count 1 > range '0 - 29' > dbmin -4100 > dbmax -1200 > dbvalue.0 -4100 > } > } > control.3 { > iface MIXER > name 'Speaker Playback Switch' > value true > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.4 { > iface MIXER > name 'Speaker Playback Volume' > value.0 35 > value.1 35 > comment { > access 'read write' > type INTEGER > count 2 > range '0 - 44' > dbmin -4100 > dbmax 300 > dbvalue.0 -600 > dbvalue.1 -600 > } > } > control.5 { > iface MIXER > name 'Mic Capture Switch' > value true > comment { > access 'read write' > type BOOLEAN > count 1 > } > } > control.6 { > iface MIXER > name 'Mic Capture Volume' > value 13 > comment { > access 'read write' > type INTEGER > count 1 > range '0 - 13' > dbmin 1600 > dbmax 2900 > dbvalue.0 2900 > } > } > } > --endcollapse-- > > > !!All Loaded Modules > !!------------------ > > Module > fuse > nf_conntrack_ipv4 > nf_defrag_ipv4 > xt_state > nf_conntrack > snd_hda_codec_idt > snd_usb_audio > snd_hda_intel > snd_usbmidi_lib > snd_hda_codec > snd_rawmidi > snd_hwdep > iTCO_wdt > iTCO_vendor_support > lpc_ich > i5000_edac > snd_seq > snd_seq_device > edac_core > snd_pcm > mfd_core > serio_raw > ppdev > snd_page_alloc > snd_timer > coretemp > snd > i5k_amb > soundcore > parport_pc > dcdbas > microcode > shpchp > parport > i2c_i801 > vhost_net > tun > macvtap > macvlan > kvm_intel > kvm > uinput > binfmt_misc > xts > gf128mul > dm_crypt > raid1 > radeon > i2c_algo_bit > drm_kms_helper > ttm > drm > tg3 > mptsas > mptscsih > mptbase > i2c_core > scsi_transport_sas > sunrpc > be2iscsi > bnx2i > cnic > uio > cxgb4i > cxgb4 > cxgb3 > mdio > libcxgbi > libiscsi_tcp > qla4xxx > iscsi_boot_sysfs > libiscsi > scsi_transport_iscsi > > > !!Sysfs Files > !!----------- > > /sys/class/sound/hwC0D0/init_pin_configs: > 0x08 0x40f000f0 > 0x09 0x40f000f1 > 0x0d 0x0221101f > 0x0e 0x01014010 > 0x0f 0x0181302e > 0x10 0x02a11020 > 0x11 0x40f000f2 > 0x12 0x40f000f3 > > /sys/class/sound/hwC0D0/driver_pin_configs: > 0x08 0x400001f0 > 0x09 0x400001f1 > 0x0d 0x0221401f > 0x0e 0x01014010 > 0x0f 0x01813020 > 0x10 0x02a19021 > 0x11 0x90100140 > 0x12 0x400001f2 > > /sys/class/sound/hwC0D0/user_pin_configs: > > /sys/class/sound/hwC0D0/init_verbs: > > > !!ALSA/HDA dmesg > !!------------------ > > [ 38.190306] SELinux: initialized (dev configfs, type configfs), uses genfs_contexts > [ 38.229616] snd_hda_intel 0000:00:1b.0: irq 66 for MSI/MSI-X > [ 38.270699] ALSA sound/usb/mixer.c:866 6:0: cannot get min/max values for control 2 (id 6) > [ 38.274097] ALSA sound/usb/mixer.c:866 1:0: cannot get min/max values for control 2 (id 1) > [ 38.276753] ALSA sound/usb/mixer.c:866 2:0: cannot get min/max values for control 2 (id 2) > [ 38.279322] ALSA sound/pci/hda/hda_auto_parser.c:322 autoconfig: line_outs=1 (0xe/0x0/0x0/0x0/0x0) type:line > [ 38.279326] ALSA sound/pci/hda/hda_auto_parser.c:326 speaker_outs=1 (0x11/0x0/0x0/0x0/0x0) > [ 38.279329] ALSA sound/pci/hda/hda_auto_parser.c:330 hp_outs=1 (0xd/0x0/0x0/0x0/0x0) > [ 38.279331] ALSA sound/pci/hda/hda_auto_parser.c:331 mono: mono_out=0x0 > [ 38.279333] ALSA sound/pci/hda/hda_auto_parser.c:335 inputs: > [ 38.279336] ALSA sound/pci/hda/hda_auto_parser.c:339 Mic=0x10 > [ 38.279342] ALSA sound/pci/hda/hda_auto_parser.c:339 Line=0xf > [ 38.285983] usbcore: registered new interface driver snd-usb-audio > [ 38.290348] input: HDA Intel Line as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 > [ 38.294975] input: HDA Intel Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 > [ 38.297406] input: HDA Intel Front Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input10 > [ 38.301369] input: HDA Intel Line Out as /devices/pci0000:00/0000:00:1b.0/sound/card0/input11 > [ 39.377352] EXT4-fs: Warning: mounting with data=journal disables delayed allocation and O_DIRECT support! > > > -- > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/

1 0

[alsa-devel] [PATCH] ASoC: samsung dma - prepare dma in start/resume trigger path
by Dylan Reid 24 Aug '12

24 Aug '12

On suspend all dma transactions are terminated (by the stop/suspend trigger calling ops->stop), they need to be set up again during resume. Move the prepare op to the start/resume/unpause trigger path so that dma and audio will restart on resume/unpause. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- sound/soc/samsung/dma.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index f3ebc38..5772762 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -250,6 +250,7 @@ static int dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + dma_prepare(substream); prtd->state |= ST_RUNNING; prtd->params->ops->trigger(prtd->params->ch); break; @@ -352,7 +353,6 @@ static struct snd_pcm_ops dma_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = dma_hw_params, .hw_free = dma_hw_free, - .prepare = dma_prepare, .trigger = dma_trigger, .pointer = dma_pointer, .mmap = dma_mmap, -- 1.7.7.3

3 6

Re: [alsa-devel] linux-next: Tree for Aug 23 (sound/isa/cmi)
by Randy Dunlap 24 Aug '12

24 Aug '12

On 08/22/2012 10:22 PM, Stephen Rothwell wrote: > Hi all, > > Changes since 20120822: > on i386: CC sound/isa/cmi8328.o sound/isa/cmi8328.c: In function 'snd_cmi8328_remove': sound/isa/cmi8328.c:416:24: error: 'cmi' undeclared (first use in this function) sound/isa/cmi8328.c:416:24: note: each undeclared identifier is reported only once for each function it appears in make[3]: *** [sound/isa/cmi8328.o] Error 1 Full randconfig file is attached. -- ~Randy

2 1

[alsa-devel] [PATCH v2 1/1] [ALSA] hda - bug fix on references without checking CONFIG_SND_HDA_POWER_SAVE
by mengdong.lin＠intel.com 24 Aug '12

24 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> The patch to support runtime PM introduced a bug: Module parameter 'power_save_controller', and the codec flag 'd3_stop_clk' 'd3_stop_clk_ok' are defined only when HDA power save is enabled in config. But there are references to them without checking macro CONFIG_SND_HDA_POWER_SAVE. This patch is to fix the bug. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_codec.c | 6 ++++++ sound/pci/hda/hda_intel.c | 2 ++ 2 files changed, 8 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index d6b3a6d..e618394 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1320,11 +1320,13 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, AC_VERB_GET_SUBSYSTEM_ID, 0); } +#ifdef CONFIG_SND_HDA_POWER_SAVE codec->d3_stop_clk = snd_hda_codec_get_supported_ps(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_CLKSTOP); if (!codec->d3_stop_clk) bus->power_keep_link_on = 1; +#endif /* power-up all before initialization */ hda_set_power_state(codec, @@ -3523,7 +3525,9 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, int count; unsigned int state; +#ifdef CONFIG_SND_HDA_POWER_SAVE codec->d3_stop_clk_ok = 0; +#endif if (codec->patch_ops.set_power_state) { codec->patch_ops.set_power_state(codec, fg, power_state); @@ -3548,9 +3552,11 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, break; } +#ifdef CONFIG_SND_HDA_POWER_SAVE if ((power_state == AC_PWRST_D3) && codec->d3_stop_clk && (state & AC_PWRST_CLK_STOP_OK)) codec->d3_stop_clk_ok = 1; +#endif } #ifdef CONFIG_SND_HDA_HWDEP diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2baa3ee..97307df 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2515,8 +2515,10 @@ static int azx_runtime_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; +#ifdef CONFIG_SND_HDA_POWER_SAVE if (!power_save_controller) return -EAGAIN; +#endif azx_stop_chip(chip); azx_clear_irq_pending(chip); -- 1.7.9.5

2 1

Re: [alsa-devel] [RFC] ASoC: snd_soc_jack for HDMI audio: does it make sense?
by Ricardo Neri 24 Aug '12

24 Aug '12

On 08/21/2012 07:39 AM, Clark, Rob wrote: > On Tue, Aug 21, 2012 at 1:01 AM, Tomi Valkeinen <tomi.valkeinen(a)ti.com> wrote: >> On Mon, 2012-08-20 at 20:47 -0500, Ricardo Neri wrote: >>> Hello! >>> >>> I have been working on prototypes for the ASoC OMAP HDMI audio driver to >>> propagate events from the HDMI output (e.g., display getting >>> enabled/disabled/suspended). This for the users of the driver to react >>> to such events. For instance, if the display is disabled or disconected, >>> audio could be stopped, rerouted or whatever other decision the user >>> makes. This is needed because, if, for instance, the HDMI IP goes off, >>> audio will stall and the audio users will only see a "playback write >>> error (DMA or IRQ trouble?)" >>> >>> In my prototypes I have used snd_soc_jack for this purpose and I have >>> some questions: >>> >>> *I see snd_soc_jack is used mostly for headsets and microphones with >>> actual external mechanical connections. Strictly, in my case I propagate >>> events originated by the OMAP display driver (changes in the power >>> state), and not from external events. Some of these events are generated >>> from an actual HDMI cable connection/disconnection, though. >>> >>> *Maybe the event should be propagated by omapdss/omapdrm/drm and the >>> entity in charge of the audio policy should listen those events instead. >>> >>> *I do see SND_JACK_VIDEOOUT and SND_JACK_AVOUT types so maybe it is >>> feasible for an audio driver to report events from an AV output. >>> >>> I was wondering about how much sense does it make to you guys use a >>> snd_soc_jack in this case? >> >> How does DRM handle audio? I made a quick grep, but I see the drm >> drivers only enabling the audio in the HW, nothing else. > > I confess to not knowing too much about audio/alsa, but what does > audio driver need from hdmi? Just hotplug events? At least for the case of the ASoC HDMI audio driver (but hopefully for other audio drivers as well), it needs to detect whether an HDMI output is available, if the display's current configuration supports audio (e.g., a 1080p display configured as VGA should be reported as not supporting audio). It also needs interfaces to configure/prepare/start/stop audio. Also, of course, needs to know if the display is off/on/suspended and when transitions occur. For OMAP, omapdss provide an interface for this functionality for ALSA or any other interested user. > > From a quick look, it seems most of what the existing drm drivers are > doing is detecting if the display supports audio, and then turning > on/off the hw.. (and some infoframe stuff in some cases). Yes, it seems to me that every driver makes its own audio implementation, mainly focused on configuration. I could not find any audio common interface so that users like ALSA can take advantage of. Also, I could not see any ALSA driver using functionality provided by a drm driver. Maybe the lack of audio support in drm is because the audio users should not talk to drm directly but to a lower level component (omapdrm, omapdss?). However, today there exists video technology supports audio as well, such as DisplayPort or HDMI. Could it make more sense now to provide audio support? > > Does ASoC support 'hotplug' of audio devices? If so, maybe it makes > some sense to have some support in drm core. At least all the edid > parsing stuff to determine if the display supports audio should be > generic and not driver specific. drm already supports generic edid parsing to check for audio support. This is a good example of how an audio driver could just use that functionality to probe audio support. Ricardo > > BR, > -R > >> If there's a common generic way to handle this, we should obviously use >> that. But if we need to choose between doing something custom or doing >> it in omapdrm driver, I think we should go for drm the only solution and >> forget about audio with omapfb. >> >> Tomi >>

3 4

[alsa-devel] [PATCH] [ALSA] hda - bug fix on references without checking CONFIG_SND_HDA_POWER_SAVE
by mengdong.lin＠intel.com 24 Aug '12

24 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> The patch to support runtime PM introduced a bug: Module parameter 'power_save_controller', and the codec flag 'd3_stop_clk' 'd3_stop_clk_ok' are defined only when HDA power save is enabled in config. But there are references to them without checking macro CONFIG_SND_HDA_POWER_SAVE. This patch is to fix the bug. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_intel.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2baa3ee..97307df 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2515,8 +2515,10 @@ static int azx_runtime_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; +#ifdef CONFIG_SND_HDA_POWER_SAVE if (!power_save_controller) return -EAGAIN; +#endif azx_stop_chip(chip); azx_clear_irq_pending(chip); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: dapm: Don't force card bias level to be updated
by Mark Brown 23 Aug '12

23 Aug '12

Commit 412312 (ASoC: dapm: Make sure all dapm contexts are updated) means that any DAPM context being updated will have the bias level automatically set, including the card. We can't safely do this as the card callbacks are called for each device context and so the management of the card bias is more complex. Several multi-component cards rely on this behaviour. Skip updates during the asynchronous run entirely. We should really do them in the synchronous section but it's not 100% clear which values to pick as the different DAPM contexts may have different bias levels. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c1fb6a6..53f69b5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -293,8 +293,9 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, level); else dapm->bias_level = level; - } else + } else if (!card || dapm != &card->dapm) { dapm->bias_level = level; + } if (ret != 0) goto out; -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: dapm: Don't force card bias level to be updated
by Mark Brown 23 Aug '12

23 Aug '12

Commit 412312 (ASoC: dapm: Make sure all dapm contexts are updated) means that any DAPM context being updated will have the bias level automatically set, including the card. We can't safely do this as the card callbacks are called for each device context and so the management of the card bias is more complex. Several multi-component cards rely on this behaviour. Skip updates during the asynchronous run entirely. We should really do them in the synchronous section but it's not 100% clear which values to pick as the different DAPM contexts may have different bias levels. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c1fb6a6..53f69b5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -293,8 +293,9 @@ static int snd_soc_dapm_set_bias_level(struct snd_soc_dapm_context *dapm, level); else dapm->bias_level = level; - } else + } else if (!card || dapm != &card->dapm) { dapm->bias_level = level; + } if (ret != 0) goto out; -- 1.7.10.4

1 0

[alsa-devel] Undelivered Mail Returned to Sender
by MAILER-DAEMON＠opensource.wolfsonmicro.com 23 Aug '12

23 Aug '12

This is the mail system at host opensource.wolfsonmicro.com. I'm sorry to have to inform you that your message could not be delivered to one or more recipients. It's attached below. For further assistance, please send mail to postmaster. If you do so, please include this problem report. You can delete your own text from the attached returned message. The mail system <alsa-devel(a)alsa-projects.org>: Host or domain name not found. Name service error for name=alsa-projects.org type=AAAA: Host not found

1 0

[alsa-devel] [PATCH] ASoC: wm0010: Fix passthrough routing
by Mark Brown 23 Aug '12

23 Aug '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm0010.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 8e0b6d6..e1315e0 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -119,7 +119,8 @@ struct wm0010_spi_msg { }; static const struct snd_soc_dapm_route wm0010_dapm_routes[] = { - { "SDI2 Playback", NULL, "SDI1 Playback" }, + { "SDI2 Capture", NULL, "SDI1 Playback" }, + { "SDI1 Capture", NULL, "SDI2 Playback" }, }; static const char *wm0010_state_to_str(enum wm0010_state state) -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: wm0010: Add initial wm0010 DSP driver
by Dimitris Papastamos 23 Aug '12

23 Aug '12

The WM0010 is a compact digital signal processor that has been highly optimised for low-power audio applications. Extensive memory resources and core optimisation allow the device to manage all audio processing algorithms efficiently and autonomously, while the host processor sleeps or performs other tasks. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/wm0010.h | 27 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wm0010.c | 930 ++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 963 insertions(+) create mode 100644 include/sound/wm0010.h create mode 100644 sound/soc/codecs/wm0010.c diff --git a/include/sound/wm0010.h b/include/sound/wm0010.h new file mode 100644 index 0000000..3261e90 --- /dev/null +++ b/include/sound/wm0010.h @@ -0,0 +1,27 @@ +/* + * wm0010.h -- Platform data for WM0010 DSP Driver + * + * Copyright 2012 Wolfson Microelectronics PLC. + * + * Author: Dimitris Papastamos <dp(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 as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef WM0010_PDATA_H +#define WM0010_PDATA_H + +struct wm0010_pdata { + int gpio_reset; + + /* Set if there is an inverter between the GPIO controlling + * the reset signal and the device. + */ + int reset_active_high; + int irq_flags; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 6e7b795..5149a46 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -66,6 +66,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_UDA134X select SND_SOC_UDA1380 if I2C select SND_SOC_WL1273 if MFD_WL1273_CORE + select SND_SOC_WM0010 if SPI_MASTER select SND_SOC_WM1250_EV1 if I2C select SND_SOC_WM2000 if I2C select SND_SOC_WM2200 if I2C @@ -319,6 +320,9 @@ config SND_SOC_UDA1380 config SND_SOC_WL1273 tristate +config SND_SOC_WM0010 + tristate + config SND_SOC_WM1250_EV1 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index a0cf18c..e9dd804 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -57,6 +57,7 @@ snd-soc-twl6040-objs := twl6040.o snd-soc-uda134x-objs := uda134x.o snd-soc-uda1380-objs := uda1380.o snd-soc-wl1273-objs := wl1273.o +snd-soc-wm0010-objs := wm0010.o snd-soc-wm1250-ev1-objs := wm1250-ev1.o snd-soc-wm2000-objs := wm2000.o snd-soc-wm2200-objs := wm2200.o @@ -175,6 +176,7 @@ obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o obj-$(CONFIG_SND_SOC_UDA134X) += snd-soc-uda134x.o obj-$(CONFIG_SND_SOC_UDA1380) += snd-soc-uda1380.o obj-$(CONFIG_SND_SOC_WL1273) += snd-soc-wl1273.o +obj-$(CONFIG_SND_SOC_WM0010) += snd-soc-wm0010.o obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o Obj-$(CONFIG_SND_SOC_WM2200) += snd-soc-wm2200.o diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c new file mode 100644 index 0000000..8e0b6d6 --- /dev/null +++ b/sound/soc/codecs/wm0010.c @@ -0,0 +1,930 @@ +/* + * wm0010.c -- WM0010 DSP Driver + * + * Copyright 2012 Wolfson Microelectronics PLC. + * + * Authors: Mark Brown <broonie(a)opensource.wolfsonmicro.com> + * Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> + * Scott Ling <sl(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. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/irqreturn.h> +#include <linux/init.h> +#include <linux/spi/spi.h> +#include <linux/firmware.h> +#include <linux/delay.h> +#include <linux/fs.h> +#include <linux/miscdevice.h> +#include <linux/gpio.h> +#include <linux/regulator/consumer.h> +#include <linux/mutex.h> +#include <linux/workqueue.h> + +#include <sound/soc.h> +#include <sound/wm0010.h> + +#define DEVICE_ID_WM0010 10 + +enum dfw_cmd { + DFW_CMD_FUSE = 0x01, + DFW_CMD_CODE_HDR, + DFW_CMD_CODE_DATA, + DFW_CMD_PLL, + DFW_CMD_INFO = 0xff +}; + +struct dfw_binrec { + u8 command; + u32 length:24; + u32 address; + uint8_t data[0]; +} __packed; + +struct dfw_pllrec { + u8 command; + u32 length:24; + u32 address; + u32 clkctrl1; + u32 clkctrl2; + u32 clkctrl3; + u32 ldetctrl; + u32 uart_div; + u32 spi_div; +} __packed; + +static struct pll_clock_map { + int max_sysclk; + int max_pll_spi_speed; + u32 pll_clkctrl1; +} pll_clock_map[] = { /* Dividers */ + { 22000000, 26000000, 0x00201f11 }, /* 2,32,2 */ + { 18000000, 26000000, 0x00203f21 }, /* 2,64,4 */ + { 14000000, 26000000, 0x00202620 }, /* 1,39,4 */ + { 10000000, 22000000, 0x00203120 }, /* 1,50,4 */ + { 6500000, 22000000, 0x00204520 }, /* 1,70,4 */ + { 5500000, 22000000, 0x00103f10 }, /* 1,64,2 */ +}; + +enum wm0010_state { + WM0010_POWER_OFF, + WM0010_OUT_OF_RESET, + WM0010_BOOTROM, + WM0010_STAGE2, + WM0010_FIRMWARE, +}; + +struct wm0010_priv { + struct snd_soc_codec *codec; + + struct mutex lock; + struct device *dev; + + struct wm0010_pdata pdata; + + int gpio_reset; + int gpio_reset_value; + + struct regulator_bulk_data core_supplies[2]; + struct regulator *dbvdd; + + int sysclk; + + enum wm0010_state state; + bool boot_failed; + int boot_done; + bool ready; + bool pll_running; + int max_spi_freq; + int board_max_spi_speed; + u32 pll_clkctrl1; + + spinlock_t irq_lock; + int irq; + + struct completion boot_completion; +}; + +struct wm0010_spi_msg { + struct spi_message m; + struct spi_transfer t; + u8 *tx_buf; + u8 *rx_buf; + size_t len; +}; + +static const struct snd_soc_dapm_route wm0010_dapm_routes[] = { + { "SDI2 Playback", NULL, "SDI1 Playback" }, +}; + +static const char *wm0010_state_to_str(enum wm0010_state state) +{ + const char *state_to_str[] = { + "Power off", + "Out of reset", + "Bootrom", + "Stage2", + "Firmware" + }; + + if (state < 0 || state >= ARRAY_SIZE(state_to_str)) + return "null"; + return state_to_str[state]; +} + +/* Called with wm0010->lock held */ +static void wm0010_halt(struct snd_soc_codec *codec) +{ + struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); + unsigned long flags; + enum wm0010_state state; + + /* Fetch the wm0010 state */ + spin_lock_irqsave(&wm0010->irq_lock, flags); + state = wm0010->state; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + switch (state) { + case WM0010_POWER_OFF: + /* If there's nothing to do, bail out */ + return; + case WM0010_OUT_OF_RESET: + case WM0010_BOOTROM: + case WM0010_STAGE2: + case WM0010_FIRMWARE: + /* Remember to put chip back into reset */ + gpio_set_value(wm0010->gpio_reset, wm0010->gpio_reset_value); + /* Disable the regulators */ + regulator_disable(wm0010->dbvdd); + regulator_bulk_disable(ARRAY_SIZE(wm0010->core_supplies), + wm0010->core_supplies); + break; + } + + spin_lock_irqsave(&wm0010->irq_lock, flags); + wm0010->state = WM0010_POWER_OFF; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); +} + +struct wm0010_boot_xfer { + struct list_head list; + struct snd_soc_codec *codec; + struct completion *done; + struct spi_message m; + struct spi_transfer t; +}; + +/* Called with wm0010->lock held */ +static void wm0010_mark_boot_failure(struct wm0010_priv *wm0010) +{ + enum wm0010_state state; + unsigned long flags; + + spin_lock_irqsave(&wm0010->irq_lock, flags); + state = wm0010->state; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + dev_err(wm0010->dev, "Failed to transition from `%s' state to `%s' state\n", + wm0010_state_to_str(state), wm0010_state_to_str(state + 1)); + + wm0010->boot_failed = true; +} + +static void wm0010_boot_xfer_complete(void *data) +{ + struct wm0010_boot_xfer *xfer = data; + struct snd_soc_codec *codec = xfer->codec; + struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); + u32 *out32 = xfer->t.rx_buf; + int i; + + if (xfer->m.status != 0) { + dev_err(codec->dev, "SPI transfer failed: %d\n", + xfer->m.status); + wm0010_mark_boot_failure(wm0010); + if (xfer->done) + complete(xfer->done); + return; + } + + for (i = 0; i < xfer->t.len / 4; i++) { + dev_dbg(codec->dev, "%d: %04x\n", i, out32[i]); + + switch (be32_to_cpu(out32[i])) { + case 0xe0e0e0e0: + dev_err(codec->dev, + "%d: ROM error reported in stage 2\n", i); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x55555555: + if (wm0010->boot_done == 0) + break; + dev_err(codec->dev, + "%d: ROM bootloader running in stage 2\n", i); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0000: + dev_dbg(codec->dev, "Stage2 loader running\n"); + break; + + case 0x0fed0007: + dev_dbg(codec->dev, "CODE_HDR packet received\n"); + break; + + case 0x0fed0008: + dev_dbg(codec->dev, "CODE_DATA packet received\n"); + break; + + case 0x0fed0009: + dev_dbg(codec->dev, "Download complete\n"); + break; + + case 0x0fed000c: + dev_dbg(codec->dev, "Application start\n"); + break; + + case 0x0fed000e: + dev_dbg(codec->dev, "PLL packet received\n"); + wm0010->pll_running = true; + break; + + case 0x0fed0025: + dev_err(codec->dev, "Device reports image too long\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed002c: + dev_err(codec->dev, "Device reports bad SPI packet\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0031: + dev_err(codec->dev, "Device reports SPI read overflow\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0032: + dev_err(codec->dev, "Device reports SPI underclock\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0033: + dev_err(codec->dev, "Device reports bad header packet\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0034: + dev_err(codec->dev, "Device reports invalid packet type\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0035: + dev_err(codec->dev, "Device reports data before header error\n"); + wm0010_mark_boot_failure(wm0010); + break; + + case 0x0fed0038: + dev_err(codec->dev, "Device reports invalid PLL packet\n"); + break; + + case 0x0fed003a: + dev_err(codec->dev, "Device reports packet alignment error\n"); + wm0010_mark_boot_failure(wm0010); + break; + + default: + dev_err(codec->dev, "Unrecognised return 0x%x\n", + be32_to_cpu(out32[i])); + wm0010_mark_boot_failure(wm0010); + break; + } + + if (wm0010->boot_failed) + break; + } + + wm0010->boot_done++; + if (xfer->done) + complete(xfer->done); +} + +static void byte_swap_64(u64 *data_in, u64 *data_out, u32 len) +{ + int i; + + for (i = 0; i < len / 8; i++) + data_out[i] = cpu_to_be64(le64_to_cpu(data_in[i])); +} + +static int wm0010_boot(struct snd_soc_codec *codec) +{ + struct spi_device *spi = to_spi_device(codec->dev); + struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); + unsigned long flags; + struct list_head xfer_list; + struct wm0010_boot_xfer *xfer; + int ret; + struct completion done; + const struct firmware *fw; + const struct dfw_binrec *rec; + struct spi_message m; + struct spi_transfer t; + struct dfw_pllrec pll_rec; + u32 *img, *p; + u64 *img_swap; + u8 *out; + u32 len, offset; + int i; + + spin_lock_irqsave(&wm0010->irq_lock, flags); + if (wm0010->state != WM0010_POWER_OFF) + dev_warn(wm0010->dev, "DSP already powered up!\n"); + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + if (wm0010->sysclk > 26000000) { + dev_err(codec->dev, "Max DSP clock frequency is 26MHz\n"); + ret = -ECANCELED; + goto err; + } + + INIT_LIST_HEAD(&xfer_list); + + mutex_lock(&wm0010->lock); + wm0010->pll_running = false; + + dev_dbg(codec->dev, "max_spi_freq: %d\n", wm0010->max_spi_freq); + + ret = regulator_bulk_enable(ARRAY_SIZE(wm0010->core_supplies), + wm0010->core_supplies); + if (ret != 0) { + dev_err(&spi->dev, "Failed to enable core supplies: %d\n", + ret); + mutex_unlock(&wm0010->lock); + goto err; + } + + ret = regulator_enable(wm0010->dbvdd); + if (ret != 0) { + dev_err(&spi->dev, "Failed to enable DBVDD: %d\n", ret); + goto err_core; + } + + /* Release reset */ + gpio_set_value(wm0010->gpio_reset, !wm0010->gpio_reset_value); + spin_lock_irqsave(&wm0010->irq_lock, flags); + wm0010->state = WM0010_OUT_OF_RESET; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + /* First the bootloader */ + ret = request_firmware(&fw, "wm0010_stage2.bin", codec->dev); + if (ret != 0) { + dev_err(codec->dev, "Failed to request stage2 loader: %d\n", + ret); + goto abort; + } + + if (!wait_for_completion_timeout(&wm0010->boot_completion, + msecs_to_jiffies(10))) + dev_err(codec->dev, "Failed to get interrupt from DSP\n"); + + spin_lock_irqsave(&wm0010->irq_lock, flags); + wm0010->state = WM0010_BOOTROM; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + dev_dbg(codec->dev, "Downloading %d byte stage 2 loader\n", fw->size); + + /* Copy to local buffer first as vmalloc causes problems for dma */ + img = kzalloc(fw->size, GFP_KERNEL); + if (!img) { + dev_err(codec->dev, "Failed to allocate image buffer\n"); + goto abort; + } + + out = kzalloc(fw->size, GFP_KERNEL); + if (!out) { + dev_err(codec->dev, "Failed to allocate output buffer\n"); + goto abort; + } + + memcpy(img, &fw->data[0], fw->size); + + spi_message_init(&m); + memset(&t, 0, sizeof(t)); + t.rx_buf = out; + t.tx_buf = img; + t.len = fw->size; + t.bits_per_word = 8; + t.speed_hz = wm0010->sysclk / 10; + spi_message_add_tail(&t, &m); + + dev_dbg(codec->dev, "Starting initial download at %dHz\n", + t.speed_hz); + + ret = spi_sync(spi, &m); + if (ret != 0) { + dev_err(codec->dev, "Initial download failed: %d\n", ret); + goto abort; + } + + /* Look for errors from the boot ROM */ + for (i = 0; i < fw->size; i++) { + if (out[i] != 0x55) { + ret = -EBUSY; + dev_err(codec->dev, "Boot ROM error: %x in %d\n", + out[i], i); + wm0010_mark_boot_failure(wm0010); + goto abort; + } + } + + release_firmware(fw); + kfree(img); + kfree(out); + + if (!wait_for_completion_timeout(&wm0010->boot_completion, + msecs_to_jiffies(10))) + dev_err(codec->dev, "Failed to get interrupt from DSP loader.\n"); + + spin_lock_irqsave(&wm0010->irq_lock, flags); + wm0010->state = WM0010_STAGE2; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + /* Only initialise PLL if max_spi_freq initialised */ + if (wm0010->max_spi_freq) { + + /* Initialise a PLL record */ + memset(&pll_rec, 0, sizeof(pll_rec)); + pll_rec.command = DFW_CMD_PLL; + pll_rec.length = (sizeof(pll_rec) - 8); + + /* On wm0010 only the CLKCTRL1 value is used */ + pll_rec.clkctrl1 = wm0010->pll_clkctrl1; + + len = pll_rec.length + 8; + out = kzalloc(len, GFP_KERNEL); + if (!out) { + dev_err(codec->dev, + "Failed to allocate RX buffer\n"); + goto abort; + } + + img_swap = kzalloc(len, GFP_KERNEL); + if (!img_swap) { + dev_err(codec->dev, + "Failed to allocate image buffer\n"); + goto abort; + } + + /* We need to re-order for 0010 */ + byte_swap_64((u64 *)&pll_rec, img_swap, len); + + spi_message_init(&m); + memset(&t, 0, sizeof(t)); + t.rx_buf = out; + t.tx_buf = img_swap; + t.len = len; + t.bits_per_word = 8; + t.speed_hz = wm0010->sysclk / 6; + spi_message_add_tail(&t, &m); + + ret = spi_sync(spi, &m); + if (ret != 0) { + dev_err(codec->dev, "First PLL write failed: %d\n", ret); + goto abort; + } + + /* Use a second send of the message to get the return status */ + ret = spi_sync(spi, &m); + if (ret != 0) { + dev_err(codec->dev, "Second PLL write failed: %d\n", ret); + goto abort; + } + + p = (u32 *)out; + + /* Look for PLL active code from the DSP */ + for (i = 0; i < len / 4; i++) { + if (*p == 0x0e00ed0f) { + dev_dbg(codec->dev, "PLL packet received\n"); + wm0010->pll_running = true; + break; + } + p++; + } + + kfree(img_swap); + kfree(out); + } else + dev_dbg(codec->dev, "Not enabling DSP PLL."); + + ret = request_firmware(&fw, "wm0010.dfw", codec->dev); + if (ret != 0) { + dev_err(codec->dev, "Failed to request application: %d\n", + ret); + goto abort; + } + + rec = (const struct dfw_binrec *)fw->data; + offset = 0; + wm0010->boot_done = 0; + wm0010->boot_failed = false; + BUG_ON(!list_empty(&xfer_list)); + init_completion(&done); + + /* First record should be INFO */ + if (rec->command != DFW_CMD_INFO) { + dev_err(codec->dev, "First record not INFO\r\n"); + goto abort; + } + + /* Check it's a 0010 file */ + if (rec->data[0] != DEVICE_ID_WM0010) { + dev_err(codec->dev, "Not a WM0010 firmware file.\r\n"); + goto abort; + } + + /* Skip the info record as we don't need to send it */ + offset += ((rec->length) + 8); + rec = (void *)&rec->data[rec->length]; + + while (offset < fw->size) { + dev_dbg(codec->dev, + "Packet: command %d, data length = 0x%x\r\n", + rec->command, rec->length); + len = rec->length + 8; + + out = kzalloc(len, GFP_KERNEL); + if (!out) { + dev_err(codec->dev, + "Failed to allocate RX buffer\n"); + goto abort; + } + + img_swap = kzalloc(len, GFP_KERNEL); + if (!img_swap) { + dev_err(codec->dev, + "Failed to allocate image buffer\n"); + goto abort; + } + + /* We need to re-order for 0010 */ + byte_swap_64((u64 *)&rec->command, img_swap, len); + + xfer = kzalloc(sizeof(*xfer), GFP_KERNEL); + if (!xfer) { + dev_err(codec->dev, "Failed to allocate xfer\n"); + goto abort; + } + + xfer->codec = codec; + list_add_tail(&xfer->list, &xfer_list); + + spi_message_init(&xfer->m); + xfer->m.complete = wm0010_boot_xfer_complete; + xfer->m.context = xfer; + xfer->t.tx_buf = img_swap; + xfer->t.rx_buf = out; + xfer->t.len = len; + xfer->t.bits_per_word = 8; + + if (!wm0010->pll_running) { + xfer->t.speed_hz = wm0010->sysclk / 6; + } else { + xfer->t.speed_hz = wm0010->max_spi_freq; + + if (wm0010->board_max_spi_speed && + (wm0010->board_max_spi_speed < wm0010->max_spi_freq)) + xfer->t.speed_hz = wm0010->board_max_spi_speed; + } + + /* Store max usable spi frequency for later use */ + wm0010->max_spi_freq = xfer->t.speed_hz; + + spi_message_add_tail(&xfer->t, &xfer->m); + + offset += ((rec->length) + 8); + rec = (void *)&rec->data[rec->length]; + + if (offset >= fw->size) { + dev_dbg(codec->dev, "All transfers scheduled\n"); + xfer->done = &done; + } + + ret = spi_async(spi, &xfer->m); + if (ret != 0) { + dev_err(codec->dev, "Write failed: %d\n", ret); + goto abort; + } + + if (wm0010->boot_failed) + goto abort; + } + + wait_for_completion(&done); + + spin_lock_irqsave(&wm0010->irq_lock, flags); + wm0010->state = WM0010_FIRMWARE; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + mutex_unlock(&wm0010->lock); + + release_firmware(fw); + + while (!list_empty(&xfer_list)) { + xfer = list_first_entry(&xfer_list, struct wm0010_boot_xfer, + list); + kfree(xfer->t.rx_buf); + kfree(xfer->t.tx_buf); + list_del(&xfer->list); + kfree(xfer); + } + + return 0; + +abort: + /* Put the chip back into reset */ + wm0010_halt(codec); + mutex_unlock(&wm0010->lock); + return ret; +err_core: + regulator_bulk_disable(ARRAY_SIZE(wm0010->core_supplies), + wm0010->core_supplies); +err: + return ret; +} + +static int wm0010_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) + wm0010_boot(codec); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_PREPARE) { + mutex_lock(&wm0010->lock); + wm0010_halt(codec); + mutex_unlock(&wm0010->lock); + } + break; + case SND_SOC_BIAS_OFF: + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +static int wm0010_set_sysclk(struct snd_soc_codec *codec, int source, + int clk_id, unsigned int freq, int dir) +{ + struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); + unsigned int i; + + wm0010->sysclk = freq; + + if (freq < pll_clock_map[ARRAY_SIZE(pll_clock_map)-1].max_sysclk) { + wm0010->max_spi_freq = 0; + } else { + for (i = 0; i < ARRAY_SIZE(pll_clock_map); i++) + if (freq >= pll_clock_map[i].max_sysclk) + break; + + wm0010->max_spi_freq = pll_clock_map[i].max_pll_spi_speed; + wm0010->pll_clkctrl1 = pll_clock_map[i].pll_clkctrl1; + } + + return 0; +} + +static int wm0010_probe(struct snd_soc_codec *codec); + +static struct snd_soc_codec_driver soc_codec_dev_wm0010 = { + .probe = wm0010_probe, + .set_bias_level = wm0010_set_bias_level, + .set_sysclk = wm0010_set_sysclk, + + .dapm_routes = wm0010_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm0010_dapm_routes), +}; + +#define WM0010_RATES (SNDRV_PCM_RATE_48000) +#define WM0010_FORMATS (SNDRV_PCM_FMTBIT_S8 | 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_driver wm0010_dai[] = { + { + .name = "wm0010-sdi1", + .playback = { + .stream_name = "SDI1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM0010_RATES, + .formats = WM0010_FORMATS, + }, + .capture = { + .stream_name = "SDI1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM0010_RATES, + .formats = WM0010_FORMATS, + }, + }, + { + .name = "wm0010-sdi2", + .playback = { + .stream_name = "SDI2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM0010_RATES, + .formats = WM0010_FORMATS, + }, + .capture = { + .stream_name = "SDI2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM0010_RATES, + .formats = WM0010_FORMATS, + }, + }, +}; + +static irqreturn_t wm0010_irq(int irq, void *data) +{ + struct wm0010_priv *wm0010 = data; + + switch (wm0010->state) { + case WM0010_POWER_OFF: + case WM0010_OUT_OF_RESET: + case WM0010_BOOTROM: + case WM0010_STAGE2: + spin_lock(&wm0010->irq_lock); + complete(&wm0010->boot_completion); + spin_unlock(&wm0010->irq_lock); + return IRQ_HANDLED; + default: + return IRQ_NONE; + } + + return IRQ_NONE; +} + +static int wm0010_probe(struct snd_soc_codec *codec) +{ + struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); + struct spi_device *spi = to_spi_device(wm0010->dev); + unsigned long flags; + unsigned long gpio_flags; + int ret; + int trigger; + int irq; + + wm0010->codec = codec; + + init_completion(&wm0010->boot_completion); + + wm0010->core_supplies[0].supply = "AVDD"; + wm0010->core_supplies[1].supply = "DCVDD"; + ret = devm_regulator_bulk_get(wm0010->dev, ARRAY_SIZE(wm0010->core_supplies), + wm0010->core_supplies); + if (ret != 0) { + dev_err(wm0010->dev, "Failed to obtain core supplies: %d\n", + ret); + return ret; + } + + wm0010->dbvdd = devm_regulator_get(wm0010->dev, "DBVDD"); + if (IS_ERR(wm0010->dbvdd)) { + ret = PTR_ERR(wm0010->dbvdd); + dev_err(wm0010->dev, "Failed to obtain DBVDD: %d\n", ret); + return ret; + } + + if (wm0010->pdata.gpio_reset) { + wm0010->gpio_reset = wm0010->pdata.gpio_reset; + + if (wm0010->pdata.reset_active_high) + wm0010->gpio_reset_value = 1; + else + wm0010->gpio_reset_value = 0; + + if (wm0010->gpio_reset_value) + gpio_flags = GPIOF_OUT_INIT_HIGH; + else + gpio_flags = GPIOF_OUT_INIT_LOW; + + ret = devm_gpio_request_one(wm0010->dev, wm0010->gpio_reset, + gpio_flags, "wm0010 reset"); + if (ret < 0) { + dev_err(wm0010->dev, + "Failed to request GPIO for DSP reset: %d\n", + ret); + return ret; + } + } else { + dev_err(wm0010->dev, "No reset GPIO configured\n"); + return ret; + } + + irq = spi->irq; + if (wm0010->pdata.irq_flags) + trigger = wm0010->pdata.irq_flags; + else + trigger = IRQF_TRIGGER_FALLING; + trigger |= IRQF_ONESHOT; + + ret = request_threaded_irq(irq, NULL, wm0010_irq, trigger, + "wm0010", wm0010); + if (ret) + dev_err(wm0010->dev, "Failed to request IRQ %d: %d\n", + irq, ret); + wm0010->irq = irq; + + if (spi->max_speed_hz) + wm0010->board_max_spi_speed = spi->max_speed_hz; + else + wm0010->board_max_spi_speed = 0; + + spin_lock_irqsave(&wm0010->irq_lock, flags); + wm0010->state = WM0010_POWER_OFF; + spin_unlock_irqrestore(&wm0010->irq_lock, flags); + + return 0; +} + +static int __devinit wm0010_spi_probe(struct spi_device *spi) +{ + struct wm0010_priv *wm0010; + int ret; + + wm0010 = devm_kzalloc(&spi->dev, sizeof(*wm0010), + GFP_KERNEL); + if (!wm0010) + return -ENOMEM; + + mutex_init(&wm0010->lock); + spin_lock_init(&wm0010->irq_lock); + + spi_set_drvdata(spi, wm0010); + wm0010->dev = &spi->dev; + + if (dev_get_platdata(&spi->dev)) + memcpy(&wm0010->pdata, dev_get_platdata(&spi->dev), + sizeof(wm0010->pdata)); + + ret = snd_soc_register_codec(&spi->dev, + &soc_codec_dev_wm0010, wm0010_dai, + ARRAY_SIZE(wm0010_dai)); + if (ret < 0) + return ret; + + return 0; +} + +static int __devexit wm0010_spi_remove(struct spi_device *spi) +{ + struct wm0010_priv *wm0010 = spi_get_drvdata(spi); + + snd_soc_unregister_codec(&spi->dev); + + if (wm0010->gpio_reset) { + /* Remember to put chip back into reset */ + gpio_set_value(wm0010->gpio_reset, wm0010->gpio_reset_value); + gpio_free(wm0010->gpio_reset); + } + + if (wm0010->irq) + free_irq(wm0010->irq, wm0010); + + return 0; +} + +static struct spi_driver wm0010_spi_driver = { + .driver = { + .name = "wm0010", + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = wm0010_spi_probe, + .remove = __devexit_p(wm0010_spi_remove), +}; + +module_spi_driver(wm0010_spi_driver); + +MODULE_DESCRIPTION("ASoC WM0010 driver"); +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); -- 1.7.12

2 1

[alsa-devel] [PATCH] ASoC: wm8994: Update for new WM1811 variants
by Mark Brown 23 Aug '12

23 Aug '12

There are some new WM1811 variants distinguished by both revision and cust_id which need slightly different handling. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 8d5b65a..c7dbbd4 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3823,14 +3823,20 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.no_cache_dac_hp_direct = true; wm8994->fll_byp = true; - switch (wm8994->revision) { + switch (control->cust_id) { case 0: - case 1: case 2: - case 3: wm8994->hubs.dcs_codes_l = -9; wm8994->hubs.dcs_codes_r = -7; break; + case 1: + case 3: + wm8994->hubs.dcs_codes_l = -8; + wm8994->hubs.dcs_codes_r = -7; + break; + default: + break; + } default: break; } @@ -3919,7 +3925,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (control->type) { case WM1811: - if (wm8994->revision > 1) { + if (control->cust_id > 1 || wm8994->revision > 1) { ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_GPIO(6), wm1811_jackdet_irq, "JACKDET", -- 1.7.10.4

1 0

[alsa-devel] SAIF configuration on imx28
by George Stefan 23 Aug '12

23 Aug '12

Hello, I am using a custom board based on imx28 (running 2.6.35) with the SAIF interface connected to the PCM output of a modem. I have developed a dummy codec for the modem and customized mxs-devb.c to use that codec instead of sgtl5000. Clock master, fixed at 256kHz, Frame sync master, fixed 125us frame duration (8kHz): 32 bits of total data; first 16 bits are valid (while FS is high), following 16 are "don't care" (while FS low) 16-bit linear samples, sent MSB first "Long Frame" sync: transmit and receive occur simultaneously while the common FS line is active (high). TX data with 0-bit delay; start at the rising edge of the clock, while FS is high. RX data with 0-bit delay; sample at the falling edge of the clock, while FS is high. This configuration is fixed so the only setup needed for SAIF is dai_format = SND_SOC_DAIFMT_RIGHT_J | SND_SOC_DAIFMT_IB_NF | SND_SOC_DAIFMT_CBM_CFM; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, dai_format); if (ret < 0) return ret; My problem is, although i am receiving dma irqs when i record ( arecord -f S16_LE -r 8000 -c 1 <file>.wav), when i play the resulting file its just silence. [ 5295.890000] dump @ mxs_saif_trigger [ 5295.890000] scr 101 , 107 [ 5295.890000] stat 80010041 , 80010071 [ 5295.890000] data 0 , ffffffff [ 5295.890000] version 1010000 , 1010000 This is a dump for arecord! Do you have any ideeas? Thanks, George.

3 14

[alsa-devel] [PATCH] ASoC: dapm: Fix return value in snd_soc_dapm_new_control()
by Dimitris Papastamos 23 Aug '12

23 Aug '12

Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 99aac65..90b8fb9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2884,7 +2884,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, ret = PTR_ERR(w->priv); dev_err(dapm->dev, "Failed to request %s: %d\n", w->name, ret); - return ret; + return NULL; } break; default: -- 1.7.12

2 1

[alsa-devel] [PATCH] ARM: OMAP2+: twl-common: Fix compile time error when omap-twl4030 audio is not enabled
by Peter Ujfalusi 23 Aug '12

23 Aug '12

Fixes: CC arch/arm/mach-omap2/twl-common.o arch/arm/mach-omap2/twl-common.c:562: error: conflicting types for ‘omap_twl4030_audio_init’ arch/arm/mach-omap2/twl-common.h:62: error: previous declaration of ‘omap_twl4030_audio_init’ was here make[1]: *** [arch/arm/mach-omap2/twl-common.o] Error 1 make: *** [arch/arm/mach-omap2] Error 2 When building the kernel with !SND_OMAP_SOC_OMAP_TWL4030 Reported-by: Brian Austin <brian.austin(a)cirrus.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- arch/arm/mach-omap2/twl-common.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-omap2/twl-common.c b/arch/arm/mach-omap2/twl-common.c index 1c6045a..2919d7a 100644 --- a/arch/arm/mach-omap2/twl-common.c +++ b/arch/arm/mach-omap2/twl-common.c @@ -544,7 +544,7 @@ void __init omap_twl4030_audio_init(char *card_name) } #else /* SOC_OMAP_TWL4030 */ -void __init omap_twl4030_audio_init(char *card_name, int codec_sysclk) +void __init omap_twl4030_audio_init(char *card_name) { return; } -- 1.7.8.6

3 4

[alsa-devel] [PATCH v5 1/1] ALSA: hda - add runtime PM support
by mengdong.lin＠intel.com 23 Aug '12

23 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Runtime PM can bring more power saving: - When the controller is suspended, its parent device will also have a chance to suspend. - PCI subsystem can choose the lowest power state the controller can signal wake up from. This state can be D3cold on platforms with ACPI PM support. And runtime PM can provide a gerneral sysfs interface for a system policy manager. Runtime PM support is based on current HDA power saving implementation. The user can enable runtime PM on platfroms that provide acceptable latency on transition from D3 to D0. Details: - When both power saving and runtime PM are enabled: -- If a codec supports 'stop-clock' in D3, it will request suspending the controller after it enters D3 and request resuming the controller before back to D0. Thus the controller will be suspended only when all codecs are suspended and support stop-clock in D3. -- User IO operations and HW wakeup signal can resume the controller back to D0. - If runtime PM is disabled, power saving just works as before. - If power saving is disabled, the controller won't be suspended because the power usage counter can never be 0. More about 'stop-clock' feature: If a codec can support targeted pass-through operations in D3 state when there is no BCLK present on the link, it will set CLKSTOP flag in the supported power states and report PS-ClkStopOk when entering D3 state. Please refer to HDA spec section 7.3.3.10 Power state and 7.3.4.12 Supported Power State. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_codec.c | 19 ++++++++++-- sound/pci/hda/hda_codec.h | 5 ++- sound/pci/hda/hda_intel.c | 75 +++++++++++++++++++++++++++++++++++---------- 3 files changed, 80 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f2f5347..d6b3a6d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -1209,6 +1209,9 @@ static void snd_hda_codec_free(struct hda_codec *codec) kfree(codec); } +static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, + hda_nid_t fg, unsigned int power_state); + static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); @@ -1317,6 +1320,12 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, AC_VERB_GET_SUBSYSTEM_ID, 0); } + codec->d3_stop_clk = snd_hda_codec_get_supported_ps(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_CLKSTOP); + if (!codec->d3_stop_clk) + bus->power_keep_link_on = 1; + /* power-up all before initialization */ hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, @@ -3514,6 +3523,8 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, int count; unsigned int state; + codec->d3_stop_clk_ok = 0; + if (codec->patch_ops.set_power_state) { codec->patch_ops.set_power_state(codec, fg, power_state); return; @@ -3536,6 +3547,10 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, if (!(state & AC_PWRST_ERROR)) break; } + + if ((power_state == AC_PWRST_D3) + && codec->d3_stop_clk && (state & AC_PWRST_CLK_STOP_OK)) + codec->d3_stop_clk_ok = 1; } #ifdef CONFIG_SND_HDA_HWDEP @@ -4385,7 +4400,7 @@ static void hda_power_work(struct work_struct *work) hda_call_codec_suspend(codec); if (bus->ops.pm_notify) - bus->ops.pm_notify(bus); + bus->ops.pm_notify(bus, codec); } static void hda_keep_power_on(struct hda_codec *codec) @@ -4440,7 +4455,7 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) spin_unlock(&codec->power_lock); if (bus->ops.pm_notify) - bus->ops.pm_notify(bus); + bus->ops.pm_notify(bus, codec); hda_call_codec_resume(codec); spin_lock(&codec->power_lock); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index fe6206b..b384c721 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -612,7 +612,7 @@ struct hda_bus_ops { void (*bus_reset)(struct hda_bus *bus); #ifdef CONFIG_SND_HDA_POWER_SAVE /* notify power-up/down from codec to controller */ - void (*pm_notify)(struct hda_bus *bus); + void (*pm_notify)(struct hda_bus *bus, struct hda_codec *codec); #endif }; @@ -870,6 +870,9 @@ struct hda_codec { unsigned long power_off_acct; unsigned long power_jiffies; spinlock_t power_lock; + + unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ + unsigned int d3_stop_clk_ok:1; /* BCLK can stop */ #endif /* codec-specific additional proc output */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b6a4ea7..2baa3ee 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,6 +46,7 @@ #include <linux/mutex.h> #include <linux/reboot.h> #include <linux/io.h> +#include <linux/pm_runtime.h> #ifdef CONFIG_X86 /* for snoop control */ #include <asm/pgtable.h> @@ -1022,7 +1023,7 @@ static unsigned int azx_get_response(struct hda_bus *bus, } #ifdef CONFIG_SND_HDA_POWER_SAVE -static void azx_power_notify(struct hda_bus *bus); +static void azx_power_notify(struct hda_bus *bus, struct hda_codec *codec); #endif /* reset codec link */ @@ -1278,6 +1279,9 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) u8 sd_status; int i, ok; + if (chip->pci->dev.power.runtime_status != RPM_ACTIVE) + return IRQ_NONE; + spin_lock(&chip->reg_lock); if (chip->disabled) { @@ -2390,23 +2394,17 @@ static void azx_stop_chip(struct azx *chip) #ifdef CONFIG_SND_HDA_POWER_SAVE /* power-up/down the controller */ -static void azx_power_notify(struct hda_bus *bus) +static void azx_power_notify(struct hda_bus *bus, struct hda_codec *codec) { struct azx *chip = bus->private_data; - struct hda_codec *c; - int power_on = 0; - list_for_each_entry(c, &bus->codec_list, list) { - if (c->power_on) { - power_on = 1; - break; - } - } - if (power_on) - azx_init_chip(chip, 1); - else if (chip->running && power_save_controller && - !bus->power_keep_link_on) - azx_stop_chip(chip); + if (bus->power_keep_link_on || !codec->d3_stop_clk_ok) + return; + + if (codec->power_on) + pm_runtime_get_sync(&chip->pci->dev); + else + pm_runtime_put_sync(&chip->pci->dev); } static DEFINE_MUTEX(card_list_lock); @@ -2511,7 +2509,35 @@ static int azx_resume(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -static SIMPLE_DEV_PM_OPS(azx_pm, azx_suspend, azx_resume); + +static int azx_runtime_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + if (!power_save_controller) + return -EAGAIN; + + azx_stop_chip(chip); + azx_clear_irq_pending(chip); + return 0; +} + +static int azx_runtime_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + azx_init_pci(chip); + azx_init_chip(chip, 1); + return 0; +} + +static const struct dev_pm_ops azx_pm = { + SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) + SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, NULL) +}; + #define AZX_PM_OPS &azx_pm #else #define azx_suspend(dev) @@ -3201,6 +3227,15 @@ static void power_down_all_codecs(struct azx *chip) #endif } +static void rpm_get_all_codecs(struct azx *chip) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &chip->bus->codec_list, list) { + pm_runtime_get_noresume(&chip->pci->dev); + } +} + static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -3238,6 +3273,9 @@ static int __devinit azx_probe(struct pci_dev *pci, pci_set_drvdata(pci, card); + if (pci_dev_run_wake(pci)) + pm_runtime_put_noidle(&pci->dev); + dev++; return 0; @@ -3289,6 +3327,7 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) goto out_free; chip->running = 1; + rpm_get_all_codecs(chip); /* all codecs are active */ power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); @@ -3303,6 +3342,10 @@ out_free: static void __devexit azx_remove(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); + + if (pci_dev_run_wake(pci)) + pm_runtime_get_noresume(&pci->dev); + if (card) snd_card_free(card); pci_set_drvdata(pci, NULL); -- 1.7.9.5

3 3

[alsa-devel] [Cross compile issue] - I met problem when to cross compile alsa-lib and alsa-utils.Could you please help on it?
by Tim Jiang 23 Aug '12

23 Aug '12

Hi experts, #My question: I met problem when to cross compile alsa-lib and alsa-utils.Could you please help on it? Thanks in advance. #details My host PC: ubuntu 11.10 64-bit; Alsa-lib source code version: 1.0.23; Alsa-utils source code version: 1.0.23; My target platform: PandaBoard; My build shell script: #!/bin/bash echo "=======================start build alsa-lib-1.0.23==============" cd ./alsa-lib-1.0.23/ export CC=arm-linux-gnueabi-gcc echo "-------------->`pwd`" ./configure --target=arm-linux --host=i686-linux --prefix=/home/tim/work/dev_board/tools/ALSA/output/alsa_lib --disable-python make clean make make install cd - echo "========================end build alsa-lib-1.0.23===============" echo "=======================start build alsa-utils-1.0.23====================" cd ./alsa-utils-1.0.23/ echo "-------------->`pwd`" ./configure --target=arm-linux --host=i686-linux --with-alsa-inc-prefix=/home/tim/work/dev_board/tools/ALSA/output/alsa_lib/include --with-alsa-prefix=/home/tim/work/dev_board/tools/ALSA/output/alsa_lib/lib --disable-alsamixer --disable-xmlto make I can build Alsa-lib successfully but error happened when to build Alsa-utils. =======================start build alsa-utils-1.0.23==================== -------------->/home/tim/work/dev_board/tools/ALSA/alsa-utils-1.0.23 configure: WARNING: If you wanted to set the --build type, don't use --host. If a cross compiler is detected then cross compile mode will be used. checking for a BSD-compatible install... /usr/bin/install -c checking whether build environment is sane... yes checking for gawk... no checking for mawk... mawk checking whether make sets $(MAKE)... yes checking for i686-linux-strip... no checking for strip... strip checking whether NLS is requested... yes checking for msgfmt... /usr/bin/msgfmt checking for gmsgfmt... /usr/bin/msgfmt checking for xgettext... /usr/bin/xgettext checking for msgmerge... /usr/bin/msgmerge checking for style of include used by make... GNU checking for i686-linux-gcc... arm-linux-gnueabi-gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... yes checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether arm-linux-gnueabi-gcc accepts -g... yes checking for arm-linux-gnueabi-gcc option to accept ISO C89... none needed checking dependency style of arm-linux-gnueabi-gcc... gcc3 checking build system type... x86_64-unknown-linux-gnu checking host system type... i686-pc-linux-gnu checking for ld used by GCC... /usr/arm-linux-gnueabi/bin/ld checking if the linker (/usr/arm-linux-gnueabi/bin/ld) is GNU ld... yes checking for shared library run path origin... done checking for CFPreferencesCopyAppValue... no checking for CFLocaleCopyCurrent... no checking for GNU gettext in libc... yes checking whether to use NLS... yes checking where the gettext function comes from... libc checking for i686-linux-gcc... (cached) arm-linux-gnueabi-gcc checking whether we are using the GNU C compiler... (cached) yes checking whether arm-linux-gnueabi-gcc accepts -g... (cached) yes checking for arm-linux-gnueabi-gcc option to accept ISO C89... (cached) none needed checking dependency style of arm-linux-gnueabi-gcc... (cached) gcc3 checking for a BSD-compatible install... /usr/bin/install -c checking whether ln -s works... yes checking for ALSA CFLAGS... -I/home/tim/work/dev_board/tools/ALSA/output/alsa_lib/include checking for ALSA LDFLAGS... -L/home/tim/work/dev_board/tools/ALSA/output/alsa_lib/lib -lasound -lm -ldl -lpthread checking for libasound headers version >= 1.0.16... found. checking for snd_ctl_open in -lasound... no configure: error: No linkable libasound was found. make: *** No targets specified and no makefile found. Stop. Best Regards, Tim

1 0

[alsa-devel] [PATCH v4 1/1] ALSA: hda - add runtime PM support
by mengdong.lin＠intel.com 23 Aug '12

23 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Runtime PM can bring more power saving: - When the controller is suspended, its parent device will also have a chance to suspend. - PCI subsystem can choose the lowest power state the controller can signal wake up from. This state can be D3cold on platforms with ACPI PM support. And runtime PM can provide a gerneral sysfs interface for a system policy manager. Runtime PM support is based on current HDA power saving implementation. The user can enable runtime PM on platfroms that provide acceptable latency on transition from D3 to D0. Details: - When both power saving and runtime PM are enabled: -- If a codec supports 'stop-clock' in D3, it will request suspending the controller after it enters D3 and request resuming the controller before back to D0. Thus the controller will be suspended only when all codecs are suspended and support stop-clock in D3. -- User IO operations and HW wakeup signal can resume the controller back to D0. - If runtime PM is disabled, power saving just works as before. - If power saving is disabled, the controller won't be suspended because the power usage counter can never be 0. More about 'stop-clock' feature: If a codec can support targeted pass-through operations in D3 state when there is no BCLK present on the link, it will set CLKSTOP flag in the supported power states and report PS-ClkStopOk when entering D3 state. Please refer to HDA spec section 7.3.3.10 Power state and 7.3.4.12 Supported Power State. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_codec.c | 20 ++++++++++-- sound/pci/hda/hda_codec.h | 5 ++- sound/pci/hda/hda_intel.c | 75 +++++++++++++++++++++++++++++++++++---------- 3 files changed, 81 insertions(+), 19 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f2f5347..72e6328 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -26,6 +26,7 @@ #include <linux/pci.h> #include <linux/mutex.h> #include <linux/module.h> +#include <linux/pm_runtime.h> #include <sound/core.h> #include "hda_codec.h" #include <sound/asoundef.h> @@ -1209,6 +1210,9 @@ static void snd_hda_codec_free(struct hda_codec *codec) kfree(codec); } +static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, + hda_nid_t fg, unsigned int power_state); + static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); @@ -1317,6 +1321,12 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, AC_VERB_GET_SUBSYSTEM_ID, 0); } + codec->d3_stop_clk = snd_hda_codec_get_supported_ps(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_CLKSTOP); + if (!codec->d3_stop_clk) + bus->power_keep_link_on = 1; + /* power-up all before initialization */ hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, @@ -3514,6 +3524,8 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, int count; unsigned int state; + codec->d3_stop_clk_ok = 0; + if (codec->patch_ops.set_power_state) { codec->patch_ops.set_power_state(codec, fg, power_state); return; @@ -3536,6 +3548,10 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, if (!(state & AC_PWRST_ERROR)) break; } + + if ((power_state == AC_PWRST_D3) + && codec->d3_stop_clk && (state & AC_PWRST_CLK_STOP_OK)) + codec->d3_stop_clk_ok = 1; } #ifdef CONFIG_SND_HDA_HWDEP @@ -4385,7 +4401,7 @@ static void hda_power_work(struct work_struct *work) hda_call_codec_suspend(codec); if (bus->ops.pm_notify) - bus->ops.pm_notify(bus); + bus->ops.pm_notify(bus, codec); } static void hda_keep_power_on(struct hda_codec *codec) @@ -4440,7 +4456,7 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) spin_unlock(&codec->power_lock); if (bus->ops.pm_notify) - bus->ops.pm_notify(bus); + bus->ops.pm_notify(bus, codec); hda_call_codec_resume(codec); spin_lock(&codec->power_lock); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index fe6206b..b384c721 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -612,7 +612,7 @@ struct hda_bus_ops { void (*bus_reset)(struct hda_bus *bus); #ifdef CONFIG_SND_HDA_POWER_SAVE /* notify power-up/down from codec to controller */ - void (*pm_notify)(struct hda_bus *bus); + void (*pm_notify)(struct hda_bus *bus, struct hda_codec *codec); #endif }; @@ -870,6 +870,9 @@ struct hda_codec { unsigned long power_off_acct; unsigned long power_jiffies; spinlock_t power_lock; + + unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ + unsigned int d3_stop_clk_ok:1; /* BCLK can stop */ #endif /* codec-specific additional proc output */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b6a4ea7..2baa3ee 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,6 +46,7 @@ #include <linux/mutex.h> #include <linux/reboot.h> #include <linux/io.h> +#include <linux/pm_runtime.h> #ifdef CONFIG_X86 /* for snoop control */ #include <asm/pgtable.h> @@ -1022,7 +1023,7 @@ static unsigned int azx_get_response(struct hda_bus *bus, } #ifdef CONFIG_SND_HDA_POWER_SAVE -static void azx_power_notify(struct hda_bus *bus); +static void azx_power_notify(struct hda_bus *bus, struct hda_codec *codec); #endif /* reset codec link */ @@ -1278,6 +1279,9 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) u8 sd_status; int i, ok; + if (chip->pci->dev.power.runtime_status != RPM_ACTIVE) + return IRQ_NONE; + spin_lock(&chip->reg_lock); if (chip->disabled) { @@ -2390,23 +2394,17 @@ static void azx_stop_chip(struct azx *chip) #ifdef CONFIG_SND_HDA_POWER_SAVE /* power-up/down the controller */ -static void azx_power_notify(struct hda_bus *bus) +static void azx_power_notify(struct hda_bus *bus, struct hda_codec *codec) { struct azx *chip = bus->private_data; - struct hda_codec *c; - int power_on = 0; - list_for_each_entry(c, &bus->codec_list, list) { - if (c->power_on) { - power_on = 1; - break; - } - } - if (power_on) - azx_init_chip(chip, 1); - else if (chip->running && power_save_controller && - !bus->power_keep_link_on) - azx_stop_chip(chip); + if (bus->power_keep_link_on || !codec->d3_stop_clk_ok) + return; + + if (codec->power_on) + pm_runtime_get_sync(&chip->pci->dev); + else + pm_runtime_put_sync(&chip->pci->dev); } static DEFINE_MUTEX(card_list_lock); @@ -2511,7 +2509,35 @@ static int azx_resume(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -static SIMPLE_DEV_PM_OPS(azx_pm, azx_suspend, azx_resume); + +static int azx_runtime_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + if (!power_save_controller) + return -EAGAIN; + + azx_stop_chip(chip); + azx_clear_irq_pending(chip); + return 0; +} + +static int azx_runtime_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + azx_init_pci(chip); + azx_init_chip(chip, 1); + return 0; +} + +static const struct dev_pm_ops azx_pm = { + SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) + SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, NULL) +}; + #define AZX_PM_OPS &azx_pm #else #define azx_suspend(dev) @@ -3201,6 +3227,15 @@ static void power_down_all_codecs(struct azx *chip) #endif } +static void rpm_get_all_codecs(struct azx *chip) +{ + struct hda_codec *codec; + + list_for_each_entry(codec, &chip->bus->codec_list, list) { + pm_runtime_get_noresume(&chip->pci->dev); + } +} + static int __devinit azx_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) { @@ -3238,6 +3273,9 @@ static int __devinit azx_probe(struct pci_dev *pci, pci_set_drvdata(pci, card); + if (pci_dev_run_wake(pci)) + pm_runtime_put_noidle(&pci->dev); + dev++; return 0; @@ -3289,6 +3327,7 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) goto out_free; chip->running = 1; + rpm_get_all_codecs(chip); /* all codecs are active */ power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); @@ -3303,6 +3342,10 @@ out_free: static void __devexit azx_remove(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); + + if (pci_dev_run_wake(pci)) + pm_runtime_get_noresume(&pci->dev); + if (card) snd_card_free(card); pci_set_drvdata(pci, NULL); -- 1.7.9.5

2 1

[alsa-devel] [PATCH v3 0/9] ARM/ASoC: OMAP McBSP device tree support
by Peter Ujfalusi 23 Aug '12

23 Aug '12

Hello, Changes since v2: - Replaced the IS_ERR_OR_NULL() with IS_ERR() Changes since v1: - ICLK workaround for the sidetone module has been kept for legacy mode. Intro mail from v2: in order to be able to add DT support for the McBSP driver which is used on all OMAP platforms (OMAP1/2/3/4/5) I needed to make some cleanups to the stack: - Removing callbacks used by the driver for - clock re-parenting (code moved to sound driver) - FSR/CLKR mux configuration (from now on it need to be done at board level) This mux is board specific and depends on the wiring. - Replaced the cpu_is_omap* checks with future proof check. - The FIFO was not enabled on OMAP2430 SoC It has been tested on BeagleBoard (with legacy and DT boot). The series changes McBSP related files mostly. It would be great if the whole series could go via audio tree (if the patches are OK and it is fine by Tony). For DT booted kernel on OMAP3 proper ICLK woraround need to be implemented. At the moment I don't see this as a stopper since we can not properly boot OMAP3 devices (especially rx51 which is using the ST). Other OMAP revisions does not have ST block so the missing workaround is not going to hurt them. I'm working on a solution which can replace the current workaround with a generic implementation to support DT and non DT booted kernels. Regards, Peter --- Peter Ujfalusi (9): ARM/ASoC: omap-mcbsp: Move OMAP2+ clock parenting code to ASoC driver ARM: OMAP: mcbsp: Enable FIFO use for OMAP2430 ARM: OMAP: board-am3517evm: Configure McBSP1 CLKR/FSR signal source ASoC: am3517evm: Do not configure McBSP1 CLKR/FSR signal muxing ARM/ASoC: omap-mcbsp: Remove CLKR/FSR mux configuration code ASoC: omap-mcbsp: Remove unused defines ASoC: omap-mcbsp: Remove cpu_is_omap* checks from the code ARM: OMAP2+: McBSP: Do not create legacy devices when booting with DT data ASoC: omap-mcbsp: Add device tree bindings .../devicetree/bindings/sound/omap-mcbsp.txt | 45 +++++++ arch/arm/mach-omap2/board-am3517evm.c | 13 ++ arch/arm/mach-omap2/mcbsp.c | 126 +------------------ arch/arm/plat-omap/include/plat/mcbsp.h | 2 - sound/soc/omap/am3517evm.c | 20 +--- sound/soc/omap/mcbsp.c | 54 ++++----- sound/soc/omap/mcbsp.h | 3 - sound/soc/omap/omap-mcbsp.c | 106 +++++++++++----- sound/soc/omap/omap-mcbsp.h | 20 --- 9 files changed, 162 insertions(+), 227 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/omap-mcbsp.txt -- 1.7.8.6

5 33

[alsa-devel] [PATCH 0/5] ARM/ASoC: Davinci: Add Device Tree Support
by Hebbar, Gururaja 23 Aug '12

23 Aug '12

This patch-set is a cumulative of below tasks 1. Split asp header into platform and audio specific Davinci driver is shared by few OMAP platforms (like TI81xx, AM335x). Splitting asp header into Davinci platform specific and Audio specific helps to share them across platforms. 2. Add DT support for Davinci Audio Module (McASP, machine, pcm & Codec) This patch was tested on AM18x Board (Tested to check if the commits doesn't break on compile or runtime) and is based on repo at http://git.kernel.org/?p=linux/kernel/git/broonie/sound.git; a=shortlog;h=refs/heads/for-3.7 Note: Matt Porter from TI is working on DMA engine conversion of DaVinci EDMA support and DT conversion of DMA parts will be taken up post that is complete. Since Audio requires DMA to work, this patch set does not yet provide complete functionality required to use sound on DT-based platforms, but is a step towards that direction. Hebbar, Gururaja (5): ASoC/ARM: Davinci: McASP: split asp header into platform and audio specific ASoC: Davinci: McASP: add device tree support for McASP ASoC: Davinci: pcm: Add device tree bindings ASoC: Davinci: machine: Add device tree binding ASoC: tlv320aic3x: Add device tree bindings .../bindings/sound/davinci-evm-audio.txt | 52 ++++++ .../bindings/sound/davinci-mcasp-audio.txt | 43 +++++ .../devicetree/bindings/sound/davinci-pcm.txt | 13 ++ .../devicetree/bindings/sound/tlv320aic3x.txt | 18 ++ arch/arm/mach-davinci/asp.h | 49 +++++ arch/arm/mach-davinci/davinci.h | 3 +- arch/arm/mach-davinci/devices-da8xx.c | 1 + arch/arm/mach-davinci/dm355.c | 2 +- arch/arm/mach-davinci/dm365.c | 2 +- arch/arm/mach-davinci/dm644x.c | 2 +- arch/arm/mach-davinci/dm646x.c | 2 +- arch/arm/mach-davinci/include/mach/da8xx.h | 2 +- .../linux/platform_data/davinci_asp.h | 73 ++------ sound/soc/codecs/tlv320aic3x.c | 30 +++ sound/soc/davinci/davinci-evm.c | 190 +++++++++++++++++--- sound/soc/davinci/davinci-i2s.c | 3 +- sound/soc/davinci/davinci-mcasp.c | 113 ++++++++++++- sound/soc/davinci/davinci-mcasp.h | 3 +- sound/soc/davinci/davinci-pcm.c | 12 ++- sound/soc/davinci/davinci-pcm.h | 3 +- 20 files changed, 527 insertions(+), 89 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/davinci-evm-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcasp-audio.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-pcm.txt create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic3x.txt create mode 100644 arch/arm/mach-davinci/asp.h rename arch/arm/mach-davinci/include/mach/asp.h => include/linux/platform_data/davinci_asp.h (56%)

2 17

[alsa-devel] [PATCH v3 2/2] [ALSA] hda - suspend the controller only if all codecs support stop-clk in D3
by mengdong.lin＠intel.com 23 Aug '12

23 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> If a codec can support targeted pass-through operations in D3 state when there is no BCLK present on the link, it will set CLKSTOP flag in the supported power states and report PS-ClkStopOk when entering D3 state. So only if a codec supports this stop-clock feature, it will request suspending the controller after it enters D3 and requst resuming the controller before back to D0. Thus the controller will be suspended only when all codecs are suspended and support stop-clock in D3. Please refer to HDA spec section 7.3.3.10 Power state and 7.3.4.12 Supported Power States. Details: - azx_power_notify() add 'codec' as 2nd parameter to check its stop-clock flag. - each codec count 1 on the controller's power usage counter, and inc/dec the count independently from each other. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_codec.c | 23 +++++++++++++++++++++-- sound/pci/hda/hda_codec.h | 5 ++++- sound/pci/hda/hda_intel.c | 25 +++++++++++-------------- 3 files changed, 36 insertions(+), 17 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f2f5347..5d09662 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -26,6 +26,7 @@ #include <linux/pci.h> #include <linux/mutex.h> #include <linux/module.h> +#include <linux/pm_runtime.h> #include <sound/core.h> #include "hda_codec.h" #include <sound/asoundef.h> @@ -1206,9 +1207,14 @@ static void snd_hda_codec_free(struct hda_codec *codec) kfree(codec->chip_name); kfree(codec->modelname); kfree(codec->wcaps); + /* undo get in snd_hda_codec_new() */ + pm_runtime_put_noidle(&codec->bus->pci->dev); kfree(codec); } +static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, + hda_nid_t fg, unsigned int power_state); + static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, unsigned int power_state); @@ -1317,10 +1323,17 @@ int /*__devinit*/ snd_hda_codec_new(struct hda_bus *bus, AC_VERB_GET_SUBSYSTEM_ID, 0); } + codec->d3_stop_clk = snd_hda_codec_get_supported_ps(codec, + codec->afg ? codec->afg : codec->mfg, + AC_PWRST_CLKSTOP); + if (!codec->d3_stop_clk) + bus->power_keep_link_on = 1; + /* power-up all before initialization */ hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_D0); + pm_runtime_get_noresume(&bus->pci->dev); /* active by default */ snd_hda_codec_proc_new(codec); @@ -3514,6 +3527,8 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, int count; unsigned int state; + codec->d3_stop_clk_ok = 0; + if (codec->patch_ops.set_power_state) { codec->patch_ops.set_power_state(codec, fg, power_state); return; @@ -3536,6 +3551,10 @@ static void hda_set_power_state(struct hda_codec *codec, hda_nid_t fg, if (!(state & AC_PWRST_ERROR)) break; } + + if ((power_state == AC_PWRST_D3) + && codec->d3_stop_clk && (state & AC_PWRST_CLK_STOP_OK)) + codec->d3_stop_clk_ok = 1; } #ifdef CONFIG_SND_HDA_HWDEP @@ -4385,7 +4404,7 @@ static void hda_power_work(struct work_struct *work) hda_call_codec_suspend(codec); if (bus->ops.pm_notify) - bus->ops.pm_notify(bus); + bus->ops.pm_notify(bus, codec); } static void hda_keep_power_on(struct hda_codec *codec) @@ -4440,7 +4459,7 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) spin_unlock(&codec->power_lock); if (bus->ops.pm_notify) - bus->ops.pm_notify(bus); + bus->ops.pm_notify(bus, codec); hda_call_codec_resume(codec); spin_lock(&codec->power_lock); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index fe6206b..b384c721 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -612,7 +612,7 @@ struct hda_bus_ops { void (*bus_reset)(struct hda_bus *bus); #ifdef CONFIG_SND_HDA_POWER_SAVE /* notify power-up/down from codec to controller */ - void (*pm_notify)(struct hda_bus *bus); + void (*pm_notify)(struct hda_bus *bus, struct hda_codec *codec); #endif }; @@ -870,6 +870,9 @@ struct hda_codec { unsigned long power_off_acct; unsigned long power_jiffies; spinlock_t power_lock; + + unsigned int d3_stop_clk:1; /* support D3 operation without BCLK */ + unsigned int d3_stop_clk_ok:1; /* BCLK can stop */ #endif /* codec-specific additional proc output */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 4c30fd7..16f037c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1023,7 +1023,7 @@ static unsigned int azx_get_response(struct hda_bus *bus, } #ifdef CONFIG_SND_HDA_POWER_SAVE -static void azx_power_notify(struct hda_bus *bus); +static void azx_power_notify(struct hda_bus *bus, struct hda_codec *codec); #endif /* reset codec link */ @@ -2394,7 +2394,7 @@ static void azx_stop_chip(struct azx *chip) #ifdef CONFIG_SND_HDA_POWER_SAVE /* power-up/down the controller */ -static void azx_power_notify(struct hda_bus *bus) +static void azx_power_notify(struct hda_bus *bus, struct hda_codec *codec) { struct azx *chip = bus->private_data; struct hda_codec *c; @@ -2406,19 +2406,20 @@ static void azx_power_notify(struct hda_bus *bus) break; } } - if (power_on) { - if (!chip->initialized) - pm_runtime_get_sync(&chip->pci->dev); + + if (!bus->power_keep_link_on + && codec->power_on && codec->d3_stop_clk_ok) + pm_runtime_get_sync(&chip->pci->dev); + + if (power_on) azx_init_chip(chip, 1); - } else if (chip->running && power_save_controller && - !bus->power_keep_link_on) { + !bus->power_keep_link_on) azx_stop_chip(chip); - /* TODO: Suspend controller only if all codec support - stop-clock in D3, for wakeup consideration */ + if (!bus->power_keep_link_on + && !codec->power_on && codec->d3_stop_clk_ok) pm_runtime_put_sync(&chip->pci->dev); - } } static DEFINE_MUTEX(card_list_lock); @@ -3327,7 +3328,6 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) goto out_free; chip->running = 1; - pm_runtime_get_noresume(&chip->pci->dev); /* active by default */ power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); @@ -3343,9 +3343,6 @@ static void __devexit azx_remove(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); - /* undo 'get' in azx_probe_continue() */ - pm_runtime_put_noidle(&pci->dev); - if (pci_dev_run_wake(pci)) pm_runtime_get_noresume(&pci->dev); -- 1.7.9.5

3 4

[alsa-devel] [PATCH] ASoC: omap-mcbsp: Device tree binding documentation update
by Peter Ujfalusi 22 Aug '12

22 Aug '12

To reflect the final devicetree node structure of McBSPs. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hello, the initial OMAP McBSP DT structure was not able to describe the IP (and it's versions) correctly. The main issue was the sidetone block of McBSP2/3 on OMAP3. With this change in the DT description the OS can get the needed information about the IP. The sidetone is still not supported when the Linux kernel is booted with DT since we still depend on hwmod to fill the resources. This patch depend on the McBSP DT support series going via audio tree. Regards, Peter .../devicetree/bindings/sound/omap-mcbsp.txt | 28 +++++++------------ 1 files changed, 10 insertions(+), 18 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/omap-mcbsp.txt b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt index 447cb13..17cce44 100644 --- a/Documentation/devicetree/bindings/sound/omap-mcbsp.txt +++ b/Documentation/devicetree/bindings/sound/omap-mcbsp.txt @@ -8,38 +8,30 @@ Required properties: - reg: Register location and size, for OMAP4+ as an array: <MPU access base address, size>, <L3 interconnect address, size>; +- reg-names: Array of strings associated with the address space - interrupts: Interrupt numbers for the McBSP port, as an array in case the McBSP IP have more interrupt lines: <OCP compliant irq>, <TX irq>, <RX irq>; +- interrupt-names: Array of strings associated with the interrupt numbers - interrupt-parent: The parent interrupt controller - ti,buffer-size: Size of the FIFO on the port (OMAP2430 and newer SoC) - ti,hwmods: Name of the hwmod associated to the McBSP port -Sidetone support for OMAP3 McBSP2 and 3 ports: -- sidetone { }: Within this section the following parameters are required: -- reg: Register location and size for the ST block -- interrupts: The interrupt number for the ST block -- interrupt-parent: The parent interrupt controller for the ST block - Example: mcbsp2: mcbsp@49022000 { compatible = "ti,omap3-mcbsp"; - #address-cells = <1>; - #size-cells = <1>; - reg = <0x49022000 0xff>; - interrupts = <0 17 0x4>, /* OCP compliant interrup */ - <0 62 0x4>, /* TX interrup */ - <0 63 0x4>; /* RX interrup */ + reg = <0x49022000 0xff>, + <0x49028000 0xff>; + reg-names = "mpu", "sidetone"; + interrupts = <0 17 0x4>, /* OCP compliant interrupt */ + <0 62 0x4>, /* TX interrupt */ + <0 63 0x4>, /* RX interrupt */ + <0 4 0x4>; /* Sidetone */ + interrupt-names = "common", "tx", "rx", "sidetone"; interrupt-parent = <&intc>; ti,buffer-size = <1280>; ti,hwmods = "mcbsp2"; - - sidetone { - reg = <0x49028000 0xff>; - interrupts = <0 4 0x4>; - interrupt-parent = <&intc>; - }; }; -- 1.7.8.6

2 1

[alsa-devel] [PATCH] ALSA: hda - fix broken HDMI jack detection after S3
by David Henningsson 22 Aug '12

22 Aug '12

The HDMI codec (an NVIDIA one in this case) forgot that its pins were unsol enabled, while it was suspended. Therefore jack detection was broken after S3. With this patch, we reenable the unsol events on resume, and also do an extra check afterwards, to see if the HDMI monitor was plugged/unplugged while in S3. Cc: stable(a)kernel.org (3.3+) BugLink: https://bugs.launchpad.net/bugs/1040030 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 8f23374..6a3ac05 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1315,6 +1315,16 @@ static int generic_hdmi_init(struct hda_codec *codec) return 0; } +#ifdef CONFIG_PM +static int generic_hdmi_resume(struct hda_codec *codec) +{ + snd_hda_codec_resume_cache(codec); + snd_hda_jack_set_dirty_all(codec); + snd_hda_jack_report_sync(codec); + return 0; +} +#endif + static void generic_hdmi_free(struct hda_codec *codec) { struct hdmi_spec *spec = codec->spec; @@ -1338,6 +1348,9 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { .build_pcms = generic_hdmi_build_pcms, .build_controls = generic_hdmi_build_controls, .unsol_event = hdmi_unsol_event, +#ifdef CONFIG_PM + .resume = generic_hdmi_resume, +#endif }; static int patch_generic_hdmi(struct hda_codec *codec) -- 1.7.9.5

2 8

[alsa-devel] [PATCH] ASoC: wm9712: Provide TLV information for capture boost controls
by Mark Brown 22 Aug '12

22 Aug '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9712.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index f6baf9c..1992a62 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -132,8 +132,9 @@ SOC_SINGLE("Aux Playback Phone Volume", AC97_CD, 4, 7, 1), SOC_SINGLE("Phone Volume", AC97_PHONE, 0, 15, 1), SOC_DOUBLE("Line Capture Volume", AC97_LINE, 8, 0, 31, 1), -SOC_SINGLE("Capture 20dB Boost Switch", AC97_REC_SEL, 14, 1, 0), -SOC_SINGLE("Capture to Phone 20dB Boost Switch", AC97_REC_SEL, 11, 1, 1), +SOC_SINGLE_TLV("Capture Boost Switch", AC97_REC_SEL, 14, 1, 0, boost_tlv), +SOC_SINGLE_TLV("Capture to Phone Boost Switch", AC97_REC_SEL, 11, 1, 1, + boost_tlv), SOC_SINGLE("3D Upper Cut-off Switch", AC97_3D_CONTROL, 5, 1, 1), SOC_SINGLE("3D Lower Cut-off Switch", AC97_3D_CONTROL, 4, 1, 1), -- 1.7.10.4

1 0

Re: [alsa-devel] ARM: OMAP: twl-common: Add helper function to register the omap-twl4030 audio driver
by Brian Austin 22 Aug '12

22 Aug '12

Hey guys, This commit 552d9dc316cfedd134ee882cd1ae24ca6934ffc0 breaks the build for OMAP3 Beagelboard. with the following error. CC arch/arm/mach-omap2/twl-common.o arch/arm/mach-omap2/twl-common.c:562: error: conflicting types for ‘omap_twl4030_audio_init’ arch/arm/mach-omap2/twl-common.h:62: error: previous declaration of ‘omap_twl4030_audio_init’ was here make[1]: *** [arch/arm/mach-omap2/twl-common.o] Error 1 make: *** [arch/arm/mach-omap2] Error 2 You added a redifinition of omap_twl4030_audio_init in twl-common.c with the extra parameter of sysclk, but the header file has the definition with just the card_name. I'm not sure which way ya'll wanted to go on this so I didnt submit a patch. Just letting it be known. This is off of for-next. Regards, Brian

1 0

[alsa-devel] [PATCH] ALSA: hda - Do not set GPIOs for speakers on IDT if there are no speakers
by David Henningsson 22 Aug '12

22 Aug '12

This fixes an issue with a machine where there were no speakers, but GPIO0 had to be data=1 for the headphone to be functioning. I'm not sure if we need a more advanced patch to solve all possible cases, but if so, this patch would still provide a minor optimisation. BugLink: https://bugs.launchpad.net/bugs/1040077 Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 file changed, 3 insertions(+) The alsa-info for this machine can not be released yet, sorry. diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index ea5775a..3edd73c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4543,6 +4543,9 @@ static void stac92xx_line_out_detect(struct hda_codec *codec, struct auto_pin_cfg *cfg = &spec->autocfg; int i; + if (cfg->speaker_outs == 0) + return; + for (i = 0; i < cfg->line_outs; i++) { if (presence) break; -- 1.7.9.5

2 1

[alsa-devel] [PATCHv2.1-CA0132 HDA Codec 2/2] Add DSP loader code to CA0132 codec
by Ian Minett 22 Aug '12

22 Aug '12

From: Ian Minett <ian_minett(a)creativelabs.com> Draft patch to add new DSP loader code, with calls to load_dsp_xxxx() functions. Signed-off-by: Ian Minett <ian_minett(a)creativelabs.com> diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c index 825f5b4..45dce9c 100644 --- a/sound/pci/hda/patch_ca0132.c +++ b/sound/pci/hda/patch_ca0132.c @@ -1952,22 +1952,13 @@ struct dma_engine { unsigned short m_converter_format; void *m_buffer_addr; unsigned int m_buffer_size; - unsigned int m_req_size; - struct snd_pcm_substream *substream; }; enum dma_state { - DMA_STATE_RESET = 0, - DMA_STATE_STOP = 1, - DMA_STATE_RUN = 2 + DMA_STATE_STOP = 0, + DMA_STATE_RUN = 1 }; -#define azx_pcm_open(a) (a->ops->open(a)) -#define azx_pcm_close(a) (a->ops->close(a)) -#define azx_pcm_prepare(a) (a->ops->prepare(a)) -#define azx_pcm_trigger(a, b) (a->ops->trigger(a, b)) -#define azx_pcm_hw_free(a) (a->ops->hw_free(a)) - static int dma_convert_to_hda_format( struct hda_stream_format *stream_format, unsigned short *hda_format) @@ -1986,113 +1977,6 @@ static int dma_convert_to_hda_format( return 0; } -static int dma_init( - struct hda_codec *codec, - struct dma_engine **pp_dma, - struct hda_stream_format *stream_format, - unsigned short *format, - unsigned int req_size) -{ - struct dma_engine *dma; - struct snd_pcm_substream *substream; - struct snd_pcm *pcm; - struct snd_pcm_runtime *runtime; - unsigned int bits; - snd_pcm_uframes_t frames; - - *pp_dma = NULL; - dma = kzalloc(sizeof(*dma), GFP_KERNEL); - memset((void *)dma, 0, sizeof(*dma)); - - dma_convert_to_hda_format(stream_format, format); - dma->m_converter_format = *format; - - dma->substream = NULL; - pcm = codec->pcm_info->pcm; - for (substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; - substream; substream = substream->next) { - if (codec->pcm_info->pcm_type == HDA_PCM_TYPE_SPDIF) - continue; - - if (!SUBSTREAM_BUSY(substream)) { - dma->substream = substream; - break; - } - } - - if (NULL == dma->substream) { - kfree(dma); - return -1; - } - - runtime = kzalloc(sizeof(*runtime), GFP_KERNEL); - memset((void *)runtime, 0, sizeof(*runtime)); - dma->substream->runtime = runtime; - dma->substream->private_data = pcm->private_data; - - azx_pcm_open(dma->substream); - req_size = req_size * 2; - snd_pcm_lib_malloc_pages(dma->substream, req_size); - - runtime->rate = stream_format->sample_rate; - runtime->channels = stream_format->number_channels; - runtime->format = SNDRV_PCM_FORMAT_S32_LE; - runtime->no_period_wakeup = 1; - - bits = snd_pcm_format_physical_width(runtime->format); - runtime->sample_bits = bits; - bits *= runtime->channels; - runtime->frame_bits = bits; - frames = 1; - while (bits % 8 != 0) { - bits *= 2; - frames *= 2; - } - runtime->byte_align = bits / 8; - runtime->min_align = frames; - runtime->buffer_size = bytes_to_frames(runtime, req_size); - runtime->period_size = runtime->buffer_size; - dma->m_req_size = req_size; - dma->codec = codec; - - *pp_dma = dma; - CA0132_LOG("dma_init: succeeded.\n"); - return 0; -} - -static int dma_prepare(struct dma_engine *dma) -{ - struct snd_pcm_runtime *runtime; - int err; - - CA0132_LOG("dma_prepare: begin\n"); - runtime = dma->substream->runtime; - - err = azx_pcm_prepare(dma->substream); - if (err < 0) - return -1; - - dma->m_buffer_size = snd_pcm_lib_buffer_bytes(dma->substream); - dma->m_buffer_addr = runtime->dma_area; - - return 0; -} - -static int dma_reset(struct dma_engine *dma) -{ - struct snd_pcm_runtime *runtime = dma->substream->runtime; - - CA0132_LOG("dma_reset: begin\n"); - azx_pcm_hw_free(dma->substream); - snd_pcm_lib_malloc_pages(dma->substream, dma->m_req_size); - - azx_pcm_prepare(dma->substream); - dma->m_buffer_size = snd_pcm_lib_buffer_bytes(dma->substream); - dma->m_buffer_addr = runtime->dma_area; - - return 0; -} - static int dma_set_state(struct dma_engine *dma, enum dma_state state) { int cmd; @@ -2100,9 +1984,6 @@ static int dma_set_state(struct dma_engine *dma, enum dma_state state) CA0132_LOG("dma_set_state state=%d\n", state); switch (state) { - case DMA_STATE_RESET: - dma_reset(dma); - return 0; case DMA_STATE_STOP: cmd = SNDRV_PCM_TRIGGER_STOP; break; @@ -2113,9 +1994,7 @@ static int dma_set_state(struct dma_engine *dma, enum dma_state state) return 0; } - azx_pcm_trigger(dma->substream, cmd); - - return 0; + return load_dsp_trigger(dma->codec->bus, cmd); } static unsigned int dma_get_buffer_size(struct dma_engine *dma) @@ -2128,15 +2007,6 @@ static unsigned int *dma_get_buffer_addr(struct dma_engine *dma) return dma->m_buffer_addr; } -static int dma_free_buffer(struct dma_engine *dma) -{ - azx_pcm_hw_free(dma->substream); - azx_pcm_close(dma->substream); - kfree(dma->substream->runtime); - dma->substream->runtime = NULL; - return 0; -} - static int dma_xfer(struct dma_engine *dma, const unsigned int *data, unsigned int count) @@ -2160,13 +2030,6 @@ static unsigned int dma_get_stream_id(struct dma_engine *dma) return spec->dsp_stream_id; } -static int dma_exit(struct dma_engine *dma) -{ - CA0132_LOG("dma_exit\n"); - kfree(dma); - return 0; -} - /* * CA0132 chip DSP image segment stuffs */ @@ -2384,7 +2247,6 @@ static int dspxfr_one_seg(struct hda_codec *codec, } CA0132_DSP_LOG("+++++ DMA complete"); dma_set_state(dma_engine, DMA_STATE_STOP); - dma_set_state(dma_engine, DMA_STATE_RESET); } CTASSERT(run_size_words <= words_to_write); @@ -2413,17 +2275,29 @@ static int dspxfr_image(struct hda_codec *codec, struct dma_engine *dma_engine; unsigned int dma_chan; unsigned int port_map_mask; - unsigned int buf_size; CTASSERT(fls_data != NULL); if (fls_data == NULL) return -1; - buf_size = ovly ? DSP_DMA_WRITE_BUFLEN_OVLY : DSP_DMA_WRITE_BUFLEN_INIT; - status = dma_init(codec, &dma_engine, format, &hda_format, buf_size); - - if (FAILED(status)) - return -1; + dma_engine = kzalloc(sizeof(*dma_engine), GFP_KERNEL); + if (!dma_engine) { + status = -ENOMEM; + goto exit; + } + memset((void*)dma_engine, 0, sizeof(*dma_engine)); + + dma_engine->codec = codec; + dma_convert_to_hda_format(format, &hda_format); + dma_engine->m_converter_format = hda_format; + dma_engine->m_buffer_size = + ovly ? DSP_DMA_WRITE_BUFLEN_OVLY : DSP_DMA_WRITE_BUFLEN_INIT; + dma_engine->m_buffer_addr = kzalloc(dma_engine->m_buffer_size, + GFP_KERNEL); + if (!dma_engine->m_buffer_addr) { + status = -ENOMEM; + goto exit; + } dma_chan = 0; do { @@ -2434,7 +2308,10 @@ static int dspxfr_image(struct hda_codec *codec, break; } - status = dma_prepare(dma_engine); + status = load_dsp_prepare(codec->bus, + dma_engine->m_converter_format, + dma_engine->m_buffer_addr, + dma_engine->m_buffer_size); if (FAILED(status)) break; @@ -2495,15 +2372,9 @@ static int dspxfr_image(struct hda_codec *codec, if (ovly && (dma_chan != INVALID_DMA_CHANNEL)) status = dspio_free_dma_chan(codec, dma_chan); - status = dma_set_state(dma_engine, DMA_STATE_RESET); - if (FAILED(status)) - status = FAIL_MSG(status, "dma set state Reset fail"); - - status = dma_free_buffer(dma_engine); - if (FAILED(status)) - status = FAIL_MSG(status, "dma free buffer fail"); - - dma_exit(dma_engine); +exit: + kfree(dma_engine->m_buffer_addr); + kfree(dma_engine); return status; } -- 1.7.4.1

2 6

Re: [alsa-devel] [pulseaudio-discuss] [PATCH v0 07/20] Consider unknown availability in module-switch-on-port-available
by Raymond Yau 22 Aug '12

22 Aug '12

2012-8-22 下午5:13 於 "David Henningsson" <david.henningsson(a)canonical.com> 寫道： > > On 08/22/2012 10:08 AM, Mikel Astiz wrote: >> >> From: Mikel Astiz <mikel.astiz(a)bmw-carit.de> >> >> A port with the availability state flag set to unknown should not be >> considered an active port. >> --- >> src/modules/module-switch-on-port-available.c | 2 +- >> 1 files changed, 1 insertions(+), 1 deletions(-) >> >> diff --git a/src/modules/module-switch-on-port-available.c b/src/modules/module-switch-on-port-available.c >> index 1da612d..8c16821 100644 >> --- a/src/modules/module-switch-on-port-available.c >> +++ b/src/modules/module-switch-on-port-available.c >> @@ -85,7 +85,7 @@ static pa_bool_t try_to_switch_profile(pa_card *card, pa_device_port *port) { >> PA_IDXSET_FOREACH(sink, card->sinks, state2) { >> if (!sink->active_port) >> continue; >> - if (sink->active_port->available != PA_PORT_AVAILABLE_NO) >> + if (sink->active_port->available == PA_PORT_AVAILABLE_YES) >> found_active_port = TRUE; >> } >> >> > > I think this will break the following use case (on a normal laptop) : > > Assume we plug in HDMI. With this patch we will get inconsistent behaviour: if headphones are plugged in (available = yes), we won't switch to HDMI, but if not (speakers are available = unknown), we will switch to HDMI. > > if you look at those bug reports of no sound for those on-board ac97 codecs with ubuntu 12.04 https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1033996 how does pulseaudio determine which controls are effective in the headphone path and speaker path of those ac97 codecs ? the analog-output-stereo profile for the line out jack is missing ( 0.056| 0.000) D: [pulseaudio] alsa-mixer.c: Removing path 'analog-output' as it is a subset of 'analog-output-headphone is there any solution for those snd-intel8x0 and those onboard ac97 codecs with headphone playback volume control ( no ctljack control and no phantom port)? http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa/a… http://git.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa/al…

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[alsa-devel] [PATCH v3 1/2] ALSA: hda - add runtime PM support
by mengdong.lin＠intel.com 22 Aug '12

22 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Runtime PM can bring more power saving: - When the controller is suspended, its parent device will also have a chance to suspend. - PCI subsystem can choose the lowest power state the controller can signal wake up from. This state can be D3cold on platforms with ACPI PM support. And runtime PM can provide a gernel sysfs interface for a system policy manager. Runtime PM support is based on current HDA power saving implementation. The user can enable runtime PM on platfroms that provide acceptable latency on transition from D3 to D0. Details: - When both power saving and runtime PM are enabled, and if all codecs are suspended (in D3), the HDA controller can also be suspended to a low power state by decreasing the power usage counter. And the user IO operation can resume the controller back to D0. - If runtime PM is disabled, power saving just works as before. - If power saving is disabled, the controller won't be suspended because the power usage counter can never be 0. TODO: Suspend the controller only if all codecs support CLKSTOP in D3, for pass-through operations and wakeup. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_intel.c | 52 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 49 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b6a4ea7..4c30fd7 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,6 +46,7 @@ #include <linux/mutex.h> #include <linux/reboot.h> #include <linux/io.h> +#include <linux/pm_runtime.h> #ifdef CONFIG_X86 /* for snoop control */ #include <asm/pgtable.h> @@ -1278,6 +1279,9 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) u8 sd_status; int i, ok; + if (chip->pci->dev.power.runtime_status != RPM_ACTIVE) + return IRQ_NONE; + spin_lock(&chip->reg_lock); if (chip->disabled) { @@ -2402,11 +2406,19 @@ static void azx_power_notify(struct hda_bus *bus) break; } } - if (power_on) + if (power_on) { + if (!chip->initialized) + pm_runtime_get_sync(&chip->pci->dev); azx_init_chip(chip, 1); + } else if (chip->running && power_save_controller && - !bus->power_keep_link_on) + !bus->power_keep_link_on) { azx_stop_chip(chip); + + /* TODO: Suspend controller only if all codec support + stop-clock in D3, for wakeup consideration */ + pm_runtime_put_sync(&chip->pci->dev); + } } static DEFINE_MUTEX(card_list_lock); @@ -2511,7 +2523,30 @@ static int azx_resume(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -static SIMPLE_DEV_PM_OPS(azx_pm, azx_suspend, azx_resume); + +static int azx_runtime_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + azx_clear_irq_pending(chip); + return 0; +} + +static int azx_runtime_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + azx_init_pci(chip); + return 0; +} + +static const struct dev_pm_ops azx_pm = { + SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) + SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, NULL) +}; + #define AZX_PM_OPS &azx_pm #else #define azx_suspend(dev) @@ -3238,6 +3273,9 @@ static int __devinit azx_probe(struct pci_dev *pci, pci_set_drvdata(pci, card); + if (pci_dev_run_wake(pci)) + pm_runtime_put_noidle(&pci->dev); + dev++; return 0; @@ -3289,6 +3327,7 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) goto out_free; chip->running = 1; + pm_runtime_get_noresume(&chip->pci->dev); /* active by default */ power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); @@ -3303,6 +3342,13 @@ out_free: static void __devexit azx_remove(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); + + /* undo 'get' in azx_probe_continue() */ + pm_runtime_put_noidle(&pci->dev); + + if (pci_dev_run_wake(pci)) + pm_runtime_get_noresume(&pci->dev); + if (card) snd_card_free(card); pci_set_drvdata(pci, NULL); -- 1.7.9.5

1 0

Re: [alsa-devel] [PATCH 1/5] ASoC/ARM: Davinci: McASP: split asp header into platform and audio specific
by Hebbar, Gururaja 22 Aug '12

22 Aug '12

On Fri, Aug 17, 2012 at 23:43:44, Nori, Sekhar wrote: > Hi Gururaja, > > On 8/14/2012 6:12 PM, Hebbar, Gururaja wrote: > > Davinci McASP header & driver are shared by few OMAP platforms (like > > TI81xx, AM335x). Splitting asp header into Davinci platform specific > > header and Audio specific header helps to share them across platforms. > > > > Audio specific defines is moved to to common > > <linux/platform_data/davinci_asp.h> so that the header can be > > accessed by all related platforms. > > > > While here, correct the header usage (remove multiple header > > re-definitions and unused headers) and remove platform names from > > structures comments and enum > > > > Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> > > Acked-by: Vaibhav Bedia <vaibhav.bedia(a)ti.com> > > This patch looks good to me. See a minor nit below. > > Acked-by: Sekhar Nori <nsekhar(a)ti.com> > > > diff --git a/arch/arm/mach-davinci/asp.h b/arch/arm/mach-davinci/asp.h > > new file mode 100644 > > index 0000000..8fd5238 > > --- /dev/null > > +++ b/arch/arm/mach-davinci/asp.h > > @@ -0,0 +1,49 @@ > > +/* > > + * TI DaVinci Audio definitions > > + */ > > +#ifndef __ASM_ARCH_DAVINCI_ASP_H > > +#define __ASM_ARCH_DAVINCI_ASP_H > > + > > +/* Bases of dm644x and dm355 register banks */ > > +#define DAVINCI_ASP0_BASE 0x01E02000 > > +#define DAVINCI_ASP1_BASE 0x01E04000 > > + > > +/* Bases of dm365 register banks */ > > +#define DAVINCI_DM365_ASP0_BASE 0x01D02000 > > + > > +/* Bases of dm646x register banks */ > > +#define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000 > > +#define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800 > > Can you remove usage of tabs after 'define'. I understand this is coming > from existing code. Ok. I have got few more comments from Mark. Will add this into it. > > Thanks, > Sekhar > Regards, Gururaja

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[alsa-devel] [RFC] Channel mapping API
by Takashi Iwai 21 Aug '12

21 Aug '12

Hi, this is a progress report of my longstanding TODO, the channel map API implementation. I'm going to cover this at Plumbers audio uconf, so we can discuss details there, too. The channel mapping API provides a method for user-space to query, get and set the channel map of a PCM stream. It's required for assigning channels properly for multi-channel streams. * KERNEL IMPLEMENTATION In my latest attempt, I implemented with control elements. A control element is created for each PCM substream with the corresponding device and substream index. Then it gives a TLV for querying maps, a read op for obtaining the current map, and optionally a write op for setting the map. The obvious merit by this way is that no extra kernel ABI is required. A couple of new helper functions are provided for assigning standard channel maps. Currently, HD-audio and AC97 drivers has some implementation. * ALSA-LIB IMPLEMENTATION The additional alsa-lib API functions look like: int **snd_pcm_query_chmaps(snd_pcm_t *pcm); int *snd_pcm_get_chmap(snd_pcm_t *pcm); int snd_pcm_set_chmap(snd_pcm_t *pcm, const int *map); snd_pcm_query_chmaps() returns the list of channel maps. A channel map is represented by an integer array, beginning with the channel map type, followed by the number of channels, and the position of each channel, e.g. { SND_CHMAP_FIXED, 4, SND_CHMAP_FL, SND_CHMAP_FR, SND_CHMAP_RL, SND_CHMAP_RR } snd_pcm_get_chmap() returns the currently assigned channel map for the given PCM stream. If the PCM is before prepared, it fills UNKNOWN. When a driver allows user to change the channel map, user can call snd_pcm_set_chmap(). For example, HDMI allows you to choose whether it's 4.0 or 3.1 output. * RESOURCES The latest kernel patchset is found in sound-unstable tree topic/tlv-chmap branch. git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git topic/tlv-chmap The alsa-lib portion is found in a tree on github topic/chmap branch. git://github.com/tiwai/alsa-lib.git topic/chmap thanks, Takashi

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[alsa-devel] [PATCH] ASoC: wm_hubs: Allow configuration of MICBIAS power up delay via pdata
by Mark Brown 21 Aug '12

21 Aug '12

Sometimes the analogue circuitry connected to the microphone needs some time to settle after power up. Allow systems to configure this delay in the platform data, the driver will then insert the required delay during power up of paths that involve the microphone. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/linux/mfd/wm8994/pdata.h | 4 ++++ include/sound/wm8993.h | 4 ++++ sound/soc/codecs/wm8993.c | 2 ++ sound/soc/codecs/wm8994.c | 2 ++ sound/soc/codecs/wm_hubs.c | 35 +++++++++++++++++++++++++++++++---- sound/soc/codecs/wm_hubs.h | 4 ++++ 6 files changed, 47 insertions(+), 4 deletions(-) diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index f0361c0..fc87be4 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -164,6 +164,10 @@ struct wm8994_pdata { int num_micd_rates; struct wm8958_micd_rate *micd_rates; + /* Power up delays to add after microphone bias power up (ms) */ + int micb1_delay; + int micb2_delay; + /* LINEOUT can be differential or single ended */ unsigned int lineout1_diff:1; unsigned int lineout2_diff:1; diff --git a/include/sound/wm8993.h b/include/sound/wm8993.h index eee19f6..8016fd8 100644 --- a/include/sound/wm8993.h +++ b/include/sound/wm8993.h @@ -32,6 +32,10 @@ struct wm8993_platform_data { unsigned int lineout1fb:1; unsigned int lineout2fb:1; + /* Delay to add for microphones to stabalise after power up */ + int micbias1_delay; + int micbias2_delay; + /* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */ unsigned int micbias1_lvl:1; unsigned int micbias2_lvl:1; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 9fd80d6..94737a3 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1520,6 +1520,8 @@ static int wm8993_probe(struct snd_soc_codec *codec) wm8993->pdata.lineout2fb, wm8993->pdata.jd_scthr, wm8993->pdata.jd_thr, + wm8993->pdata.micbias1_delay, + wm8993->pdata.micbias2_delay, wm8993->pdata.micbias1_lvl, wm8993->pdata.micbias2_lvl); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 2d60ed3..a51431c 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3233,6 +3233,8 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) pdata->lineout2fb, pdata->jd_scthr, pdata->jd_thr, + pdata->micb1_delay, + pdata->micb2_delay, pdata->micbias1_lvl, pdata->micbias2_lvl); diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 8dc4e89..c72eaef 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -859,6 +859,28 @@ static int lineout_event(struct snd_soc_dapm_widget *w, return 0; } +static int micbias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); + + switch (w->shift) { + case WM8993_MICB1_ENA_SHIFT: + if (hubs->micb1_delay) + msleep(hubs->micb1_delay); + break; + case WM8993_MICB2_ENA_SHIFT: + if (hubs->micb2_delay) + msleep(hubs->micb2_delay); + break; + default: + return -EINVAL; + } + + return 0; +} + void wm_hubs_update_class_w(struct snd_soc_codec *codec) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); @@ -1049,8 +1071,10 @@ SND_SOC_DAPM_INPUT("IN1RP"), SND_SOC_DAPM_INPUT("IN2RN"), SND_SOC_DAPM_INPUT("IN2RP:VXRP"), -SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", WM8993_POWER_MANAGEMENT_1, 5, 0, + micbias_event, SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_SUPPLY("MICBIAS1", WM8993_POWER_MANAGEMENT_1, 4, 0, + micbias_event, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_MIXER("IN1L PGA", WM8993_POWER_MANAGEMENT_2, 6, 0, in1l_pga, ARRAY_SIZE(in1l_pga)), @@ -1388,13 +1412,16 @@ EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_routes); int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *codec, int lineout1_diff, int lineout2_diff, int lineout1fb, int lineout2fb, - int jd_scthr, int jd_thr, int micbias1_lvl, - int micbias2_lvl) + int jd_scthr, int jd_thr, + int micbias1_delay, int micbias2_delay, + int micbias1_lvl, int micbias2_lvl) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); hubs->lineout1_se = !lineout1_diff; hubs->lineout2_se = !lineout2_diff; + hubs->micb1_delay = micbias1_delay; + hubs->micb2_delay = micbias2_delay; if (!lineout1_diff) snd_soc_update_bits(codec, WM8993_LINE_MIXER1, diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 14927f6..704e53f 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -55,6 +55,9 @@ struct wm_hubs_data { struct list_head dcs_cache; bool (*check_class_w_digital)(struct snd_soc_codec *); + int micb1_delay; + int micb2_delay; + bool lineout1_se; bool lineout1n_ena; bool lineout1p_ena; @@ -75,6 +78,7 @@ extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *, int lineout1_diff, int lineout2_diff, int lineout1fb, int lineout2fb, int jd_scthr, int jd_thr, + int micbias1_dly, int micbias2_dly, int micbias1_lvl, int micbias2_lvl); extern irqreturn_t wm_hubs_dcs_done(int irq, void *data); -- 1.7.10.4

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Re: [alsa-devel] mx27_3ds : Sound doesn't work anymore on linux-next
by Gaëtan Carlier 21 Aug '12

21 Aug '12

Just some more informations, process enter in D+ state and this is so impossible to kill them. On 08/17/2012 05:16 PM, Gaëtan Carlier wrote: > Hi, > It is several days that I tried to make sound work on mx27_3ds platform > from linux-next (daily cloned/pulled) but each time I try start start > aplay or alsamixer, the process freeze and CTRL+C has not effect. I can > open another SSH session and execute other commands but alsa commands > always freeze. > The sound card is correctly installer at boot : > > imx_mc13783 imx_mc13783.0: mc13783-hifi <-> imx-ssi.0 mapping ok > ALSA device list: > #0: imx_mc13783 > > I only apply the patch "[PATCH 1/2] ARM: mx27pdk: Add audio support". I > attach my config file. > > I try to apply patch on a previous commit when I told that this patch > was working but it didn't succeed the is still freezing. > I try to make it working on another mx27-based platform but same > problem. I try to update Alsa from version 1.0.23 to 1.0.25 but same > behaviour. > What is wrong ? > > Regards, > Gaëtan Carlier. Regards, Gaëtan.

1 1

Re: [alsa-devel] [PATCH] ASoC: Remove unused variable
by Mark Brown 21 Aug '12

21 Aug '12

On Tue, Aug 21, 2012 at 04:51:59PM +0200, Emil Goode wrote: > This patch removes a unused struct snd_soc_dai. > > Sparse is warning: > sound/soc/soc-compress.c:156:22: warning: > unused variable ‘codec_dai’ [-Wunused-variable] Already fixed by someone else.

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[alsa-devel] [PATCH 0/3] ASoC: omap-mcbsp: Sidetone related changes
by Peter Ujfalusi 21 Aug '12

21 Aug '12

Hello, When we boot with DT blob the sidetone is not supported at the moment. Prepare the McBSP code for the time when we are going to be able to support ST with DT booted kernel. Meanwhile do not block the audio card if the st_data is missing from the port, but print a warning about it. This will allow us to debug the DT booted kernels with audio. Mark: This series as such does not depend on the McBSP DT support, but it can be attached to that branch. Regards, Peter --- Peter Ujfalusi (3): ASoC: omap-mcbsp: Check mcbsp->id instead of cpu_dai->id when adding ST controls ASoC: omap-mcbsp: Only print warning if the st_data is missing for the port ASoC: omap-mcbsp: Use macro to create the McBSP2/3 ST controls sound/soc/omap/omap-mcbsp.c | 47 ++++++++++++++++++------------------------ 1 files changed, 20 insertions(+), 27 deletions(-) -- 1.7.8.6

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Re: [alsa-devel] [PATCH] ASoC: soc-compress: Remove unused variable
by Mark Brown 21 Aug '12

21 Aug '12

On Tue, Aug 21, 2012 at 12:20:22PM +0530, Sachin Kamat wrote: > codec_dai is not used in the function. > > sound/soc/soc-compress.c: In function ‘soc_compr_set_params’: > sound/soc/soc-compress.c:156:22: warning: > unused variable ‘codec_dai’ [-Wunused-variable] Applied, thanks.

1 0

[alsa-devel] [PATCH] ASoC: compress - fix code alignment
by Vinod Koul 21 Aug '12

21 Aug '12

Reported-by: Fengguang Wu <wfg(a)linux.intel.com> Signed-off-by: Namarta Kohli <namartax.kohli(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)linux.intel.com> --- sound/soc/soc-compress.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 88d85ba..cc0562d 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -104,7 +104,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) if (platform->driver->compr_ops && platform->driver->compr_ops->free) platform->driver->compr_ops->free(cstream); - cpu_dai->runtime = NULL; + cpu_dai->runtime = NULL; if (cstream->direction == SND_COMPRESS_PLAYBACK) { if (!rtd->pmdown_time || codec->ignore_pmdown_time || -- 1.7.0.4

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Re: [alsa-devel] [PATCH] ASoC: soc-compress: Remove unused variable
by Mark Brown 21 Aug '12

21 Aug '12

On Tue, Aug 21, 2012 at 10:48:30AM -0300, Fabio Estevam wrote: > Remove unused variable ‘codec_dai’ and fix the following warning: > > sound/soc/soc-compress.c: In function ‘soc_compr_set_params’: > sound/soc/soc-compress.c:156:22: warning: unused variable ‘codec_dai’ [-Wunused-variable] Someone already sent a patch for this.

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[alsa-devel] [PATCH 08/22] ASoC: codecs: Enable AB8500 CODEC for Device Tree
by Lee Jones 21 Aug '12

21 Aug '12

We continue to allow the AB8500 CODEC to be registered via the AB8500 Multi Functional Device API, only this time we extract its configuration from the Device Tree binary. CC: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- include/linux/mfd/abx500/ab8500-codec.h | 6 ++- sound/soc/codecs/ab8500-codec.c | 81 +++++++++++++++++++++++++++++++ 2 files changed, 85 insertions(+), 2 deletions(-) diff --git a/include/linux/mfd/abx500/ab8500-codec.h b/include/linux/mfd/abx500/ab8500-codec.h index dc65292..d707941 100644 --- a/include/linux/mfd/abx500/ab8500-codec.h +++ b/include/linux/mfd/abx500/ab8500-codec.h @@ -23,7 +23,8 @@ enum amic_type { /* Mic-biases */ enum amic_micbias { AMIC_MICBIAS_VAMIC1, - AMIC_MICBIAS_VAMIC2 + AMIC_MICBIAS_VAMIC2, + AMIC_MICBIAS_UNKNOWN }; /* Bias-voltage */ @@ -31,7 +32,8 @@ enum ear_cm_voltage { EAR_CMV_0_95V, EAR_CMV_1_10V, EAR_CMV_1_27V, - EAR_CMV_1_58V + EAR_CMV_1_58V, + EAR_CMV_UNKNOWN }; /* Analog microphone settings */ diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 23b4018..fa80961 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -34,6 +34,7 @@ #include <linux/mfd/abx500/ab8500-sysctrl.h> #include <linux/mfd/abx500/ab8500-codec.h> #include <linux/regulator/consumer.h> +#include <linux/of.h> #include <sound/core.h> #include <sound/pcm.h> @@ -2394,9 +2395,65 @@ struct snd_soc_dai_driver ab8500_codec_dai[] = { } }; +static void ab8500_codec_of_probe(struct device *dev, struct device_node *np, + struct ab8500_codec_platform_data *codec) +{ + u32 value; + + if (of_get_property(np, "stericsson,amic1-type-single-ended", NULL)) + codec->amics.mic1_type = AMIC_TYPE_SINGLE_ENDED; + else + codec->amics.mic1_type = AMIC_TYPE_DIFFERENTIAL; + + if (of_get_property(np, "stericsson,amic2-type-single-ended", NULL)) + codec->amics.mic2_type = AMIC_TYPE_SINGLE_ENDED; + else + codec->amics.mic2_type = AMIC_TYPE_DIFFERENTIAL; + + /* Has a non-standard Vamic been requested? */ + if (of_get_property(np, "stericsson,amic1a-bias-vamic2", NULL)) + codec->amics.mic1a_micbias = AMIC_MICBIAS_VAMIC2; + else + codec->amics.mic1a_micbias = AMIC_MICBIAS_VAMIC1; + + if (of_get_property(np, "stericsson,amic1b-bias-vamic2", NULL)) + codec->amics.mic1b_micbias = AMIC_MICBIAS_VAMIC2; + else + codec->amics.mic1b_micbias = AMIC_MICBIAS_VAMIC1; + + if (of_get_property(np, "stericsson,amic2-bias-vamic1", NULL)) + codec->amics.mic2_micbias = AMIC_MICBIAS_VAMIC1; + else + codec->amics.mic2_micbias = AMIC_MICBIAS_VAMIC2; + + if (!of_property_read_u32(np, "stericsson,earpeice-cmv", &value)) { + switch (value) { + case 950 : + codec->ear_cmv = EAR_CMV_0_95V; + break; + case 1100 : + codec->ear_cmv = EAR_CMV_1_10V; + break; + case 1270 : + codec->ear_cmv = EAR_CMV_1_27V; + break; + case 1580 : + codec->ear_cmv = EAR_CMV_1_58V; + break; + default : + codec->ear_cmv = EAR_CMV_UNKNOWN; + dev_err(dev, "Unsuitable earpiece voltage found in DT\n"); + } + } else { + dev_warn(dev, "No earpiece voltage found in DT - using default\n"); + codec->ear_cmv = EAR_CMV_0_95V; + } +} + static int ab8500_codec_probe(struct snd_soc_codec *codec) { struct device *dev = codec->dev; + struct device_node *np = dev->of_node; struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev); struct ab8500_platform_data *pdata; struct filter_control *fc; @@ -2407,6 +2464,30 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) /* Setup AB8500 according to board-settings */ pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent); + if (np) { + if (!pdata) + pdata = devm_kzalloc(dev, + sizeof(struct ab8500_platform_data), + GFP_KERNEL); + + if (!pdata->codec) + pdata->codec + = devm_kzalloc(dev, + sizeof(struct ab8500_codec_platform_data), + GFP_KERNEL); + + if (!(pdata && pdata->codec)) + return -ENOMEM; + + ab8500_codec_of_probe(dev, np, pdata->codec); + + } else { + if (!(pdata && pdata->codec)) { + dev_err(dev, "No codec platform data or DT found\n"); + return -EINVAL; + } + } + /* Inform SoC Core that we have our own I/O arrangements. */ codec->control_data = (void *)true; -- 1.7.9.5

4 8

Re: [alsa-devel] [RFC] ASoC: snd_soc_jack for HDMI audio: does it make sense?
by Mark Brown 21 Aug '12

21 Aug '12

On Tue, Aug 21, 2012 at 07:39:55AM -0500, Clark, Rob wrote: > Does ASoC support 'hotplug' of audio devices? If so, maybe it makes > some sense to have some support in drm core. At least all the edid > parsing stuff to determine if the display supports audio should be > generic and not driver specific. Not really (except on a card level), and it'd probably confuse most userspaces. What I'd expect to happen currently is that the current state of the connector would get reported via extcon.

1 0

[alsa-devel] [PATCH 0/5] hda-emu: Add playback/capture test to test suite
by David Henningsson 21 Aug '12

21 Aug '12

When I added those tests, I did not discover anything that needed to be fixed in the driver code, but there were some additional things in hda-emu to fix up. David Henningsson (5): hda-emu: Store pointers to pcm streams instead of their content hda-emu: Add support for get/set converter channel count hda-emu: Add CX20585 to modem whitelist hda-emu: Add playback/capture test to test suite hda-emu: improve test suite summary script hda-emu.c | 24 ++++++++++++------------ hda-int.c | 15 +++++++++++++++ hda-parse.c | 1 + include/hda-types.h | 1 + tester/runall.sh | 3 --- tester/runner.py | 25 ++++++++++++++++++++++--- tester/summary.py | 9 +++++++++ 7 files changed, 60 insertions(+), 18 deletions(-) delete mode 100755 tester/runall.sh -- 1.7.9.5

2 6

[alsa-devel] Can module snd-aloop (virtual looback device) be made clock slave of a real audio device ?
by James Warden 21 Aug '12

21 Aug '12

Hello alsa-devel's, A while back, I had written a rather hackish method to hook up the ALSA virtual loopback device to jack (jackaudio.org) via Jack clients alsa_in / alsa_out so that non jack-aware applications that can however use ALSA devices "directly" (via alsa-lib) can have their audio stream redirected to jack (see my rather lengthy prose at http://alsa.opensrc.org/Jack_and_Loopback_device_as_Alsa-to-Jack_bridge). There are a few advantages in this method (depends on your use-cases of course). But it could be improved (or so I think) by removing the following drawbacks: The alsa_in/out jack clients are resampling the audio stream to synchronize with the device that jack is using. The resampling, done by libsamplerate, takes some CPU power. Since many users are using my recipe on laptops, this is not always welcome. Because snd-aloop is a virtual device, couldn't it be made a clock slave of the h/w device that jack is operating on ? This would in principle remove the need for resampling. I could imagine that the module could be passed an option at loading time (e.g. modprobe snd-aloop clock-master=<real h/w device index> or something like that) Since I have no clue as to the kernel code required to implement / extend snd-aloop, I would like some opinions about this. Thanks for any hints. J.

2 3

[alsa-devel] Query on PCM lib: wait_for_avail() API
by Sarawadi, Ravishankar 21 Aug '12

21 Aug '12

Greetings, I recently started working with ALSA architecture. I am very excited to be part of ALSA group and this is my first query as ALSA developer. This is in regards to function wait_for_avail() defined in pcm_lib.c If audio path is broken the wait_time inside this function is 10 seconds. wait_time = msecs_to_jiffies(wait_time * 1000); I have a use case where I need to force exit out of this wait when audio hardware path is broken. Waiting for 10 seconds is too long and causing synchronization issues. My Use-case: 1 PCM Audio playback is going on. PCM write is a blocking call. Waiting on DSP WRITE-DONE event. 2 Audio hardware path failure 3 PCM driver notified of hardware path failure. 4 Meanwhile DSP ring buffer gets full as buffers are not getting consumed. 5 Next WRITE is blocked on wait_for_avail() buffer call. 6 PCM close from driver as part process to handle audio path failure is blocked on Step 5 I am trying to figure out a way to exit out of this wait forcefully by some external trigger. Another option is to reduce this wait time to less than 5 seconds. Could someone please recommend the best way to handle this use case? Thanks, RaviShankar (Ravi)

2 1

[alsa-devel] [PATCH] introduce snd-cmi8328: C-Media CMI8328 driver
by Ondrej Zary 21 Aug '12

21 Aug '12

Introduce snd-cmi8328 driver for C-Media CMI8328-based sound cards, such as AudioExcel AV500. It supports PCM playback and capture (full-duplex) through wss_lib, gameport, OPL3 and MPU401. The AV500 card has onboard Dream wavetable synth connected to the MPU401 port and Aux 1 input internally which works too. The CDROM interface is not supported (as the drivers for these CDROMs were removed from the kernel some time ago). A separate driver is needed because CMI8328 is completely different chip to CMI8329/CMI8330. It's configured by magic registers (there's no PnP). Sound is provided by a real WSS codec (CS4231A) and the SB part is just a SB Pro emulation (for DOS games, useless for Linux). When SB is enabled, the CMI8328 chip disables access to the WSS codec, emulates SoundBlaster on one side and outputs sound data to the codec - so SB and WSS can't work together with this card. The WSS codec can do full duplex by itself so there's no need for crazy things like snd-cmi8330 does (combining SB and WSS parts into one driver). Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> commit ebd26fe68c73eb979ace14b51a4fbdd899233952 Author: Ondrej Zary <linux(a)rainbow-software.org> Date: Thu Aug 2 09:17:57 2012 +0200 introduce snd-cmi8328: C-Media CMI8328 driver Introduce snd-cmi8328 driver for C-Media CMI8328-based sound cards, such as AudioExcel AV500. It supports PCM playback and capture (full-duplex) through wss_lib, gameport, OPL3 and MPU401. The AV500 card has onboard Dream wavetable synth connected to the MPU401 port and Aux 1 input internally which works too. The CDROM interface is not supported (as the drivers for these CDROMs were removed from the kernel some time ago). diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 52064cf..a38d964 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -117,6 +117,18 @@ config SND_AZT2320 To compile this driver as a module, choose M here: the module will be called snd-azt2320. +config SND_CMI8328 + tristate "C-Media CMI8328" + select SND_WSS_LIB + select SND_OPL3_LIB + select SND_MPU401_UART + help + Say Y here to include support for soundcards based on the + C-Media CMI8328 chip. + + To compile this driver as a module, choose M here: the module + will be called snd-cmi8328. + config SND_CMI8330 tristate "C-Media CMI8330" select SND_WSS_LIB diff --git a/sound/isa/Makefile b/sound/isa/Makefile index 8d781e4..9a15f14 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -6,6 +6,7 @@ snd-adlib-objs := adlib.o snd-als100-objs := als100.o snd-azt2320-objs := azt2320.o +snd-cmi8328-objs := cmi8328.o snd-cmi8330-objs := cmi8330.o snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o @@ -16,6 +17,7 @@ snd-sscape-objs := sscape.o obj-$(CONFIG_SND_ADLIB) += snd-adlib.o obj-$(CONFIG_SND_ALS100) += snd-als100.o obj-$(CONFIG_SND_AZT2320) += snd-azt2320.o +obj-$(CONFIG_SND_CMI8328) += snd-cmi8328.o obj-$(CONFIG_SND_CMI8330) += snd-cmi8330.o obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o diff --git a/sound/isa/cmi8328.c b/sound/isa/cmi8328.c new file mode 100644 index 0000000..aefafff --- /dev/null +++ b/sound/isa/cmi8328.c @@ -0,0 +1,482 @@ +/* + * Driver for C-Media CMI8328-based soundcards, such as AudioExcel AV500 + * Copyright (c) 2012 Ondrej Zary + * + * AudioExcel AV500 card consists of: + * - CMI8328 - main chip (SB Pro emulation, gameport, OPL3, MPU401, CD-ROM) + * - CS4231A - WSS codec + * - Dream SAM9233+GMS950400+RAM+ROM: Wavetable MIDI, connected to MPU401 + */ + +#include <linux/init.h> +#include <linux/isa.h> +#include <linux/module.h> +#include <linux/gameport.h> +#include <asm/dma.h> +#include <sound/core.h> +#include <sound/wss.h> +#include <sound/opl3.h> +#include <sound/mpu401.h> +#define SNDRV_LEGACY_FIND_FREE_IOPORT +#define SNDRV_LEGACY_FIND_FREE_IRQ +#define SNDRV_LEGACY_FIND_FREE_DMA +#include <sound/initval.h> + +MODULE_AUTHOR("Ondrej Zary <linux(a)rainbow-software.org>"); +MODULE_DESCRIPTION("C-Media CMI8328"); +MODULE_LICENSE("GPL"); + +#if defined(CONFIG_GAMEPORT) || defined(CONFIG_GAMEPORT_MODULE) +#define SUPPORT_JOYSTICK 1 +#endif + +/* I/O port is configured by jumpers on the card to one of these */ +static int cmi8328_ports[] = { 0x530, 0xe80, 0xf40, 0x604 }; +#define CMI8328_MAX ARRAY_SIZE(cmi8328_ports) + +static int index[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = -1}; +static char *id[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = NULL}; +static long port[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_PORT}; +static int irq[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_IRQ}; +static int dma1[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_DMA}; +static int dma2[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_DMA}; +static long mpuport[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_PORT}; +static int mpuirq[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = SNDRV_AUTO_IRQ}; +#ifdef SUPPORT_JOYSTICK +static bool gameport[CMI8328_MAX] = {[0 ... (CMI8328_MAX-1)] = true}; +#endif + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for CMI8328 soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for CMI8328 soundcard."); + +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for CMI8328 driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for CMI8328 driver."); +module_param_array(dma1, int, NULL, 0444); +MODULE_PARM_DESC(dma1, "DMA1 for CMI8328 driver."); +module_param_array(dma2, int, NULL, 0444); +MODULE_PARM_DESC(dma2, "DMA2 for CMI8328 driver."); + +module_param_array(mpuport, long, NULL, 0444); +MODULE_PARM_DESC(mpuport, "MPU-401 port # for CMI8328 driver."); +module_param_array(mpuirq, int, NULL, 0444); +MODULE_PARM_DESC(mpuirq, "IRQ # for CMI8328 MPU-401 port."); +#ifdef SUPPORT_JOYSTICK +module_param_array(gameport, bool, NULL, 0444); +MODULE_PARM_DESC(gameport, "Enable gameport."); +#endif + +struct snd_cmi8328 { + u16 port; + u8 cfg[3]; + u8 wss_cfg; + struct snd_card *card; + struct snd_wss *wss; +#ifdef SUPPORT_JOYSTICK + struct gameport *gameport; +#endif +}; + +/* CMI8328 configuration registers */ +#define CFG1 0x61 +#define CFG1_SB_DISABLE (1 << 0) +#define CFG1_GAMEPORT (1 << 1) +/* + * bit 0: SB: 0=enabled, 1=disabled + * bit 1: gameport: 0=disabled, 1=enabled + * bits 2-4: SB IRQ: 001=3, 010=5, 011=7, 100=9, 101=10, 110=11 + * bits 5-6: SB DMA: 00=disabled (when SB disabled), 01=DMA0, 10=DMA1, 11=DMA3 + * bit 7: SB port: 0=0x220, 1=0x240 + */ +#define CFG2 0x62 +#define CFG2_MPU_ENABLE (1 << 2) +/* + * bits 0-1: CD-ROM mode: 00=disabled, 01=Panasonic, 10=Sony/Mitsumi/Wearnes, + 11=IDE + * bit 2: MPU401: 0=disabled, 1=enabled + * bits 3-4: MPU401 IRQ: 00=3, 01=5, 10=7, 11=9, + * bits 5-7: MPU401 port: 000=0x300, 001=0x310, 010=0x320, 011=0x330, 100=0x332, + 101=0x334, 110=0x336 + */ +#define CFG3 0x63 +/* + * bits 0-2: CD-ROM IRQ: 000=disabled, 001=3, 010=5, 011=7, 100=9, 101=10, + 110=11 + * bits 3-4: CD-ROM DMA: 00=disabled, 01=DMA0, 10=DMA1, 11=DMA3 + * bits 5-7: CD-ROM port: 000=0x300, 001=0x310, 010=0x320, 011=0x330, 100=0x340, + 101=0x350, 110=0x360, 111=0x370 + */ + +static u8 snd_cmi8328_cfg_read(u16 port, u8 reg) +{ + outb(0x43, port + 3); + outb(0x21, port + 3); + outb(reg, port + 3); + return inb(port); +} + +static void snd_cmi8328_cfg_write(u16 port, u8 reg, u8 val) +{ + outb(0x43, port + 3); + outb(0x21, port + 3); + outb(reg, port + 3); + outb(val, port + 3); /* yes, value goes to the same port as index */ +} + +static void snd_cmi8328_cfg_save(u16 port, u8 cfg[]) +{ + cfg[0] = snd_cmi8328_cfg_read(port, CFG1); + cfg[1] = snd_cmi8328_cfg_read(port, CFG2); + cfg[2] = snd_cmi8328_cfg_read(port, CFG3); +} + +static void snd_cmi8328_cfg_restore(u16 port, u8 cfg[]) +{ + snd_cmi8328_cfg_write(port, CFG1, cfg[0]); + snd_cmi8328_cfg_write(port, CFG2, cfg[1]); + snd_cmi8328_cfg_write(port, CFG3, cfg[2]); +} + +static int __devinit snd_cmi8328_mixer(struct snd_wss *chip) +{ + struct snd_card *card; + struct snd_ctl_elem_id id1, id2; + int err; + + card = chip->card; + + memset(&id1, 0, sizeof(id1)); + memset(&id2, 0, sizeof(id2)); + id1.iface = id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + /* rename AUX0 switch to CD */ + strcpy(id1.name, "Aux Playback Switch"); + strcpy(id2.name, "CD Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "error renaming control\n"); + return err; + } + /* rename AUX0 volume to CD */ + strcpy(id1.name, "Aux Playback Volume"); + strcpy(id2.name, "CD Playback Volume"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "error renaming control\n"); + return err; + } + /* rename AUX1 switch to Synth */ + strcpy(id1.name, "Aux Playback Switch"); + id1.index = 1; + strcpy(id2.name, "Synth Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "error renaming control\n"); + return err; + } + /* rename AUX1 volume to Synth */ + strcpy(id1.name, "Aux Playback Volume"); + id1.index = 1; + strcpy(id2.name, "Synth Playback Volume"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) { + snd_printk(KERN_ERR "error renaming control\n"); + return err; + } + + return 0; +} + +/* find index of an item in "-1"-ended array */ +int array_find(int array[], int item) +{ + int i; + + for (i = 0; array[i] != -1; i++) + if (array[i] == item) + return i; + + return -1; +} +/* the same for long */ +int array_find_l(long array[], long item) +{ + int i; + + for (i = 0; array[i] != -1; i++) + if (array[i] == item) + return i; + + return -1; +} + +static int __devinit snd_cmi8328_probe(struct device *pdev, unsigned int ndev) +{ + struct snd_card *card; + struct snd_opl3 *opl3; + struct snd_cmi8328 *cmi; +#ifdef SUPPORT_JOYSTICK + struct resource *res; +#endif + int err, pos; + static long mpu_ports[] = { 0x330, 0x300, 0x310, 0x320, 0x332, 0x334, + 0x336, -1 }; + static u8 mpu_port_bits[] = { 3, 0, 1, 2, 4, 5, 6 }; + static int mpu_irqs[] = { 9, 7, 5, 3, -1 }; + static u8 mpu_irq_bits[] = { 3, 2, 1, 0 }; + static int irqs[] = { 9, 10, 11, 7, -1 }; + static u8 irq_bits[] = { 2, 3, 4, 1 }; + static int dma1s[] = { 3, 1, 0, -1 }; + static u8 dma_bits[] = { 3, 2, 1 }; + static int dma2s[][2] = { {1, -1}, {0, -1}, {-1, -1}, {0, -1} }; + u16 port = cmi8328_ports[ndev]; + u8 val; + + /* 0xff is invalid configuration (but settable - hope it isn't set) */ + if (snd_cmi8328_cfg_read(port, CFG1) == 0xff) + return -ENODEV; + /* the SB disable bit must NEVER EVER be cleared or the WSS dies */ + snd_cmi8328_cfg_write(port, CFG1, CFG1_SB_DISABLE); + if (snd_cmi8328_cfg_read(port, CFG1) != CFG1_SB_DISABLE) + return -ENODEV; + /* disable everything first */ + snd_cmi8328_cfg_write(port, CFG2, 0); /* disable CDROM and MPU401 */ + snd_cmi8328_cfg_write(port, CFG3, 0); /* disable CDROM IRQ and DMA */ + + if (irq[ndev] == SNDRV_AUTO_IRQ) { + irq[ndev] = snd_legacy_find_free_irq(irqs); + if (irq[ndev] < 0) { + snd_printk(KERN_ERR "unable to find a free IRQ\n"); + return -EBUSY; + } + } + if (dma1[ndev] == SNDRV_AUTO_DMA) { + dma1[ndev] = snd_legacy_find_free_dma(dma1s); + if (dma1[ndev] < 0) { + snd_printk(KERN_ERR "unable to find a free DMA1\n"); + return -EBUSY; + } + } + if (dma2[ndev] == SNDRV_AUTO_DMA) { + dma2[ndev] = snd_legacy_find_free_dma(dma2s[dma1[ndev] % 4]); + if (dma2[ndev] < 0) { + snd_printk(KERN_WARNING "unable to find a free DMA2, full-duplex will not work\n"); + dma2[ndev] = -1; + } + } + /* configure WSS IRQ... */ + pos = array_find(irqs, irq[ndev]); + if (pos < 0) { + snd_printk(KERN_ERR "invalid IRQ %d\n", irq[ndev]); + return -EINVAL; + } + val = irq_bits[pos] << 3; + /* ...and DMA... */ + pos = array_find(dma1s, dma1[ndev]); + if (pos < 0) { + snd_printk(KERN_ERR "invalid DMA1 %d\n", dma1[ndev]); + return -EINVAL; + } + val |= dma_bits[pos]; + /* ...and DMA2 */ + if (dma2[ndev] >= 0 && dma1[ndev] != dma2[ndev]) { + pos = array_find(dma2s[dma1[ndev]], dma2[ndev]); + if (pos < 0) { + snd_printk(KERN_ERR "invalid DMA2 %d\n", dma2[ndev]); + return -EINVAL; + } + val |= 0x04; /* enable separate capture DMA */ + } + outb(val, port); + + err = snd_card_create(index[ndev], id[ndev], THIS_MODULE, + sizeof(struct snd_cmi8328), &card); + if (err < 0) + return err; + cmi = card->private_data; + cmi->card = card; + cmi->port = port; + cmi->wss_cfg = val; + snd_card_set_dev(card, pdev); + + err = snd_wss_create(card, port + 4, -1, irq[ndev], dma1[ndev], + dma2[ndev], WSS_HW_DETECT, 0, &cmi->wss); + if (err < 0) + goto error; + + err = snd_wss_pcm(cmi->wss, 0, NULL); + if (err < 0) + goto error; + + err = snd_wss_mixer(cmi->wss); + if (err < 0) + goto error; + err = snd_cmi8328_mixer(cmi->wss); + if (err < 0) + goto error; + + if (snd_wss_timer(cmi->wss, 0, NULL) < 0) + snd_printk(KERN_WARNING "error initializing WSS timer\n"); + + if (mpuport[ndev] == SNDRV_AUTO_PORT) { + mpuport[ndev] = snd_legacy_find_free_ioport(mpu_ports, 2); + if (mpuport[ndev] < 0) + snd_printk(KERN_ERR "unable to find a free MPU401 port\n"); + } + if (mpuirq[ndev] == SNDRV_AUTO_IRQ) { + mpuirq[ndev] = snd_legacy_find_free_irq(mpu_irqs); + if (mpuirq[ndev] < 0) + snd_printk(KERN_ERR "unable to find a free MPU401 IRQ\n"); + } + /* enable and configure MPU401 */ + if (mpuport[ndev] > 0 && mpuirq[ndev] > 0) { + val = CFG2_MPU_ENABLE; + pos = array_find_l(mpu_ports, mpuport[ndev]); + if (pos < 0) + snd_printk(KERN_WARNING "invalid MPU401 port 0x%lx\n", + mpuport[ndev]); + else { + val |= mpu_port_bits[pos] << 5; + pos = array_find(mpu_irqs, mpuirq[ndev]); + if (pos < 0) + snd_printk(KERN_WARNING "invalid MPU401 IRQ %d\n", + mpuirq[ndev]); + else { + val |= mpu_irq_bits[pos] << 3; + snd_cmi8328_cfg_write(port, CFG2, val); + if (snd_mpu401_uart_new(card, 0, + MPU401_HW_MPU401, mpuport[ndev], + 0, mpuirq[ndev], NULL) < 0) + snd_printk(KERN_ERR "error initializing MPU401\n"); + } + } + } + /* OPL3 is hardwired to 0x388 and cannot be disabled */ + if (snd_opl3_create(card, 0x388, 0x38a, OPL3_HW_AUTO, 0, &opl3) < 0) + snd_printk(KERN_ERR "error initializing OPL3\n"); + else + if (snd_opl3_hwdep_new(opl3, 0, 1, NULL) < 0) + snd_printk(KERN_WARNING "error initializing OPL3 hwdep\n"); + + strcpy(card->driver, "CMI8328"); + strcpy(card->shortname, "C-Media CMI8328"); + sprintf(card->longname, "%s at 0x%lx, irq %d, dma %d,%d", + card->shortname, cmi->wss->port, irq[ndev], dma1[ndev], + (dma2[ndev] >= 0) ? dma2[ndev] : dma1[ndev]); + + dev_set_drvdata(pdev, card); + err = snd_card_register(card); + if (err < 0) + goto error; +#ifdef SUPPORT_JOYSTICK + if (!gameport[ndev]) + return 0; + /* gameport is hardwired to 0x200 */ + res = request_region(0x200, 8, "CMI8328 gameport"); + if (!res) + snd_printk(KERN_WARNING "unable to allocate gameport I/O port\n"); + else { + struct gameport *gp = cmi->gameport = gameport_allocate_port(); + if (!cmi->gameport) + release_and_free_resource(res); + else { + gameport_set_name(gp, "CMI8328 Gameport"); + gameport_set_phys(gp, "%s/gameport0", dev_name(pdev)); + gameport_set_dev_parent(gp, pdev); + gp->io = 0x200; + gameport_set_port_data(gp, res); + /* Enable gameport */ + snd_cmi8328_cfg_write(port, CFG1, + CFG1_SB_DISABLE | CFG1_GAMEPORT); + gameport_register_port(gp); + } + } +#endif + return 0; +error: + snd_card_free(card); + + return err; +} + +static int __devexit snd_cmi8328_remove(struct device *pdev, unsigned int dev) +{ + struct snd_card *card = dev_get_drvdata(pdev); +#ifdef SUPPORT_JOYSTICK + struct snd_cmi8328 *cmi = card->private_data; + if (cmi->gameport) { + struct resource *res = gameport_get_port_data(cmi->gameport); + gameport_unregister_port(cmi->gameport); + release_and_free_resource(res); + } +#endif + /* disable everything */ + snd_cmi8328_cfg_write(cmi->port, CFG1, CFG1_SB_DISABLE); + snd_cmi8328_cfg_write(cmi->port, CFG2, 0); + snd_cmi8328_cfg_write(cmi->port, CFG3, 0); + snd_card_free(card); + dev_set_drvdata(pdev, NULL); + return 0; +} + +#ifdef CONFIG_PM +static int snd_cmi8328_suspend(struct device *pdev, unsigned int n, + pm_message_t state) +{ + struct snd_card *card = dev_get_drvdata(pdev); + struct snd_cmi8328 *cmi; + + if (!card) /* ignore absent devices */ + return 0; + cmi = card->private_data; + snd_cmi8328_cfg_save(cmi->port, cmi->cfg); + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(cmi->wss->pcm); + cmi->wss->suspend(cmi->wss); + + return 0; +} + +static int snd_cmi8328_resume(struct device *pdev, unsigned int n) +{ + struct snd_card *card = dev_get_drvdata(pdev); + struct snd_cmi8328 *cmi; + + if (!card) /* ignore absent devices */ + return 0; + cmi = card->private_data; + snd_cmi8328_cfg_restore(cmi->port, cmi->cfg); + outb(cmi->wss_cfg, cmi->port); + cmi->wss->resume(cmi->wss); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + + return 0; +} +#endif + +static struct isa_driver snd_cmi8328_driver = { + .probe = snd_cmi8328_probe, + .remove = __devexit_p(snd_cmi8328_remove), +#ifdef CONFIG_PM + .suspend = snd_cmi8328_suspend, + .resume = snd_cmi8328_resume, +#endif + .driver = { + .name = "cmi8328" + }, +}; + +static int __init alsa_card_cmi8328_init(void) +{ + return isa_register_driver(&snd_cmi8328_driver, CMI8328_MAX); +} + +static void __exit alsa_card_cmi8328_exit(void) +{ + isa_unregister_driver(&snd_cmi8328_driver); +} + +module_init(alsa_card_cmi8328_init) +module_exit(alsa_card_cmi8328_exit) -- Ondrej Zary

2 1

[alsa-devel] [PATCH] snd-als100: fix suspend/resume
by Ondrej Zary 21 Aug '12

21 Aug '12

snd_card_als100_probe() does not set pcm field in struct snd_sb. As a result, PCM is not suspended and applications don't know that they need to resume the playback. Tested with Labway A381-F20 card (ALS120). Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- a/sound/isa/als100.c +++ b/sound/isa/als100.c @@ -233,7 +233,7 @@ static int __devinit snd_card_als100_probe(int dev, irq[dev], dma8[dev], dma16[dev]); } - if ((error = snd_sb16dsp_pcm(chip, 0, NULL)) < 0) { + if ((error = snd_sb16dsp_pcm(chip, 0, &chip->pcm)) < 0) { snd_card_free(card); return error; } -- Ondrej Zary

2 1

[alsa-devel] [PATCH v2] ALSA: hda - add runtime PM support
by mengdong.lin＠intel.com 21 Aug '12

21 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> Runtime PM can bring more power saving: - When the controller is suspended, its parent device will also have a chance to suspend. - PCI subsystem can choose the lowest power state the controller can signal wake up from. This state can be D3cold on platforms with ACPI PM support. And runtime PM can provide a gernel sysfs interface for a system policy manager. Runtime PM support is based on current HDA power saving implementation. The user can enable runtime PM on platfroms that provide acceptable latency on transition from D3 to D0. Details: - When both power saving and runtime PM are enabled, and if all codecs are suspended (in D3), the HDA controller can also be suspended to a low power state by decreasing the power usage counter. And the user IO operation can resume the controller back to D0. - If runtime PM is disabled, power saving just works as before. - If power saving is disabled, the controller won't be suspended because the power usage counter can never be 0. TODO: Suspend the controller only if all codecs support CLKSTOP in D3, for pass-through operations and wakeup. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_intel.c | 55 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 52 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b6a4ea7..cd02971 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,6 +46,7 @@ #include <linux/mutex.h> #include <linux/reboot.h> #include <linux/io.h> +#include <linux/pm_runtime.h> #ifdef CONFIG_X86 /* for snoop control */ #include <asm/pgtable.h> @@ -1278,6 +1279,9 @@ static irqreturn_t azx_interrupt(int irq, void *dev_id) u8 sd_status; int i, ok; + if (chip->pci->dev.power.runtime_status != RPM_ACTIVE) + return IRQ_NONE; + spin_lock(&chip->reg_lock); if (chip->disabled) { @@ -2402,11 +2406,19 @@ static void azx_power_notify(struct hda_bus *bus) break; } } - if (power_on) + if (power_on) { + if (!chip->initialized) + pm_runtime_get_sync(&chip->pci->dev); azx_init_chip(chip, 1); + } else if (chip->running && power_save_controller && - !bus->power_keep_link_on) + !bus->power_keep_link_on) { azx_stop_chip(chip); + + /* TODO: Suspend controller only if all codec support + stop-clock in D3, for wakeup consideration */ + pm_runtime_put_sync(&chip->pci->dev); + } } static DEFINE_MUTEX(card_list_lock); @@ -2511,7 +2523,33 @@ static int azx_resume(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -static SIMPLE_DEV_PM_OPS(azx_pm, azx_suspend, azx_resume); + +static int azx_runtime_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + azx_clear_irq_pending(chip); + return 0; +} + +static int azx_runtime_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + azx_init_pci(chip); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} + +static const struct dev_pm_ops azx_pm = { + SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) + SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, NULL) +}; + #define AZX_PM_OPS &azx_pm #else #define azx_suspend(dev) @@ -3238,6 +3276,9 @@ static int __devinit azx_probe(struct pci_dev *pci, pci_set_drvdata(pci, card); + if (pci_dev_run_wake(pci)) + pm_runtime_put_noidle(&pci->dev); + dev++; return 0; @@ -3289,6 +3330,7 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) goto out_free; chip->running = 1; + pm_runtime_get_noresume(&chip->pci->dev); /* active by default */ power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); @@ -3303,6 +3345,13 @@ out_free: static void __devexit azx_remove(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); + + /* undo 'get' in azx_probe_continue() */ + pm_runtime_put_noidle(&pci->dev); + + if (pci_dev_run_wake(pci)) + pm_runtime_get_noresume(&pci->dev); + if (card) snd_card_free(card); pci_set_drvdata(pci, NULL); -- 1.7.9.5

3 2

[alsa-devel] [RFC PATCH] ALSA: hda - add support for runtime power management
by mengdong.lin＠intel.com 21 Aug '12

21 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch is a basic idea about adding runtime PM support to HD audio. Support for runtime PM is based on current HDA power saving implementation. When both power saving and runtime PM are enabled, and if all codecs are suspended (in D3), the HDA controller can also be suspended to a low power state by decreasing the power usage counter and triggering PCI device runtime suspend. And the user IO operation can resume the controller back to D0. The user can enable runtime PM on HDA controller on platfroms that can provide acceptable latency on transition from D3 to D0. If the runtime PM is disabled, power saving just works as without runtime PM before. And if power saving is disabled, the power usage counter will never be 0 so the HDA controller will not be suspended. I've verified this patch on a platfrom on which the HD-A controller can enter D3hot when suspended. And I plan to allow the controller to suspened only if all codecs support "stop-clock" in D3, for wakeup request. Is this idea doable? Any comments are welcome. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_codec.c | 1 + sound/pci/hda/hda_intel.c | 70 +++++++++++++++++++++++++++++++++++++++++++-- 2 files changed, 68 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index f2f5347..c88a95e 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4383,6 +4383,7 @@ static void hda_power_work(struct work_struct *work) } spin_unlock(&codec->power_lock); + snd_printd("hda codec %d suspend\n", codec->addr); hda_call_codec_suspend(codec); if (bus->ops.pm_notify) bus->ops.pm_notify(bus); diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index b6a4ea7..543a5c4 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -46,6 +46,7 @@ #include <linux/mutex.h> #include <linux/reboot.h> #include <linux/io.h> +#include <linux/pm_runtime.h> #ifdef CONFIG_X86 /* for snoop control */ #include <asm/pgtable.h> @@ -2402,11 +2403,22 @@ static void azx_power_notify(struct hda_bus *bus) break; } } - if (power_on) + if (power_on) { + if (!chip->initialized) { + snd_printd(SFX "request resume controller\n"); + pm_runtime_get_sync(&chip->pci->dev); + } azx_init_chip(chip, 1); + } else if (chip->running && power_save_controller && - !bus->power_keep_link_on) + !bus->power_keep_link_on) { azx_stop_chip(chip); + + /* TODO: Suspend controller only if all codec support + stop-clock in D3, for wakeup consideration */ + snd_printd(SFX "request suspend controller\n"); + pm_runtime_put_sync(&chip->pci->dev); + } } static DEFINE_MUTEX(card_list_lock); @@ -2511,7 +2523,48 @@ static int azx_resume(struct device *dev) snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } -static SIMPLE_DEV_PM_OPS(azx_pm, azx_suspend, azx_resume); + +static int azx_runtime_suspend(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + struct azx_pcm *p; + + snd_printd(SFX "controller runtime suspend\n"); + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + azx_clear_irq_pending(chip); + list_for_each_entry(p, &chip->pcm_list, list) + snd_pcm_suspend_all(p->pcm); + + if (chip->irq >= 0) { + free_irq(chip->irq, chip); + chip->irq = -1; + } + return 0; +} + +static int azx_runtime_resume(struct device *dev) +{ + struct snd_card *card = dev_get_drvdata(dev); + struct azx *chip = card->private_data; + + snd_printd(SFX "controller runtime resume\n"); + + if (azx_acquire_irq(chip, 1) < 0) + return -EIO; + + azx_init_pci(chip); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} + +static const struct dev_pm_ops azx_pm = { + SET_SYSTEM_SLEEP_PM_OPS(azx_suspend, azx_resume) + SET_RUNTIME_PM_OPS(azx_runtime_suspend, azx_runtime_resume, NULL) +}; + #define AZX_PM_OPS &azx_pm #else #define azx_suspend(dev) @@ -3238,6 +3291,9 @@ static int __devinit azx_probe(struct pci_dev *pci, pci_set_drvdata(pci, card); + if (pci_dev_run_wake(pci)) + pm_runtime_put_noidle(&pci->dev); + dev++; return 0; @@ -3289,6 +3345,7 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) goto out_free; chip->running = 1; + pm_runtime_get_noresume(&chip->pci->dev); /* active by default */ power_down_all_codecs(chip); azx_notifier_register(chip); azx_add_card_list(chip); @@ -3303,6 +3360,13 @@ out_free: static void __devexit azx_remove(struct pci_dev *pci) { struct snd_card *card = pci_get_drvdata(pci); + + /* undo 'get' in azx_probe_continue() */ + pm_runtime_put_noidle(&pci->dev); + + if (pci_dev_run_wake(pci)) + pm_runtime_get_noresume(&pci->dev); + if (card) snd_card_free(card); pci_set_drvdata(pci, NULL); -- 1.7.9.5

3 7

Re: [alsa-devel] Broken powersave in 3.5.2, codec is always active
by Takashi Iwai 20 Aug '12

20 Aug '12

[Re-added Cc to ML; please don't drop it] At Mon, 20 Aug 2012 22:23:02 +0700, ValdikSS wrote: > > Takashi Iwai <tiwai(a)suse.de> писал(а) в своём письме Mon, 20 Aug 2012 > 16:17:04 +0700: > > > At Sat, 18 Aug 2012 12:44:29 +0700, > > ValdikSS wrote: > >> > >> Hello! > >> I have a Lenovo ThinkPad X220 laptop with a conexant codec and recently > >> (after kernel 3.5.2) it is active all the time after KDE login, as > >> powertop says. This couldn't be fixed by logging of from KDE or playing > >> some music in mplayer from console. The only kernel I tried in 3.5.x is > >> 3.5.2, so I can't say is this bug reproducible in 3.5.0 or 3.5.1, but > >> everything is fine with 3.4.9. This never happens if you don't start KDE > >> after reboot, you can listen to some music in mplayer in console and > >> everything seems to be fine. lsof and fuser /dev/snd/* shows nothing. > > > > Just to be sure, check /sys/module/snd_hda_intel/parameters/power_save > > value after logging into KDE. This value can be changed dynamically. > > > > If the value is still set properly, try to build the kernel with the > > tracing option with the patch below. Then try to trace the power > > up/down sequences in the tracing buffer. Some brief description is > > found in Documentation/sound/alsa/HD-Audio.txt Tracepoints section. > > > > > > thanks, > > > > Takashi > > > > --- > > From: Takashi Iwai <tiwai(a)suse.de> > > Subject: [PATCH] ALSA: hda - Add tracepoints at snd_hda_power_up/down > > entrances. > > > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > --- > > sound/pci/hda/hda_codec.c | 2 ++ > > sound/pci/hda/hda_trace.h | 24 ++++++++++++++++++++++++ > > 2 files changed, 26 insertions(+) > > > > diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c > > index 408babc..ad0101e 100644 > > --- a/sound/pci/hda/hda_codec.c > > +++ b/sound/pci/hda/hda_codec.c > > @@ -4417,6 +4417,7 @@ static void __snd_hda_power_up(struct hda_codec > > *codec, bool wait_power_down) > > spin_lock(&codec->power_lock); > > codec->power_count++; > > + trace_hda_power_count(codec); > > /* Return if power_on or transitioning to power_on, unless currently > > * powering down. */ > > if ((codec->power_on || codec->power_transition > 0) && > > @@ -4496,6 +4497,7 @@ void snd_hda_power_down(struct hda_codec *codec) > > { > > spin_lock(&codec->power_lock); > > --codec->power_count; > > + trace_hda_power_count(codec); > > if (!codec->power_on || codec->power_count || codec->power_transition) > > { > > spin_unlock(&codec->power_lock); > > return; > > diff --git a/sound/pci/hda/hda_trace.h b/sound/pci/hda/hda_trace.h > > index 9884871..9a34b86 100644 > > --- a/sound/pci/hda/hda_trace.h > > +++ b/sound/pci/hda/hda_trace.h > > @@ -87,6 +87,30 @@ DEFINE_EVENT(hda_power, hda_power_up, > > TP_ARGS(codec) > > ); > > +TRACE_EVENT(hda_power_count, > > + TP_PROTO(struct hda_codec *codec), > > + TP_ARGS(codec), > > + TP_STRUCT__entry( > > + __field( unsigned int, card ) > > + __field( unsigned int, addr ) > > + __field( int, power_count ) > > + __field( int, power_on ) > > + __field( int, power_transition ) > > + ), > > + > > + TP_fast_assign( > > + __entry->card = (codec)->bus->card->number; > > + __entry->addr = (codec)->addr; > > + __entry->power_count = (codec)->power_count; > > + __entry->power_on = (codec)->power_on; > > + __entry->power_transition = (codec)->power_transition; > > + ), > > + > > + TP_printk("[%d:%d] power_count=%d, power_on=%d, power_transition=%d", > > + __entry->card, __entry->addr, __entry->power_count, > > + __entry->power_on, __entry->power_transition) > > +); > > + > > TRACE_EVENT(hda_unsol_event, > > TP_PROTO(struct hda_bus *bus, u32 res, u32 res_ex), > > Thanks for the quick reply! I compiled 3.5.2 kernel with this patch, > enabled hda tracing, ran mplayer once in the console and started KDE. Once > KDE is started, card is used 100% of time, which can be seen in powertop. > When I logout from KDE and kill kdm, no events are generated in trace > since KDE start (no sounds are played from KDE start till KDM kill) and > card is still used 100% of time. If you play any sounds in mplayer in > console after kdm kill, the card is still used 100% time. This can be > fixed by laptop suspend. > > The traces are quite huge so I pasted it to pastebin: > 01-trace-just-enabled http://pastebin.com/JNRWXUPy > 02-mplayer-played-and-stopped-card-is-normal http://pastebin.com/r0jZLDey > 03-kde-started-and-stopped-100-percent-card http://pastebin.com/Z7mzbY4c > The last log begins with latest rows from 02 It's still not clear from the log. You can reduce the trace pattern to hda_power_* for reducing the trace log size. In anyway, could you try the patch below? thanks, Takashi --- diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 88a9c20..b94e04d 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4433,6 +4433,8 @@ static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) * then there is no need to go through power up here. */ if (codec->power_on) { + if (codec->power_transition < 0) + codec->power_transition = 0; spin_unlock(&codec->power_lock); return; }

2 2

[alsa-devel] ASoC updates for 3.6
by Mark Brown 20 Aug '12

20 Aug '12

The following changes since commit 0d7614f09c1ebdbaa1599a5aba7593f147bf96ee: Linux 3.6-rc1 (2012-08-02 16:38:10 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.6 for you to fetch changes up to 28c42c28309244d0b15d1b385e33429d59997679: ASoC: wm9712: Fix inverted capture volume (2012-08-17 22:43:18 +0100) ---------------------------------------------------------------- ASoC: Additional updates for 3.6 A batch more bugfixes, all driver-specific and fairly small and unremarkable in a global context. The biggest batch are for the newly added Arizona drivers. ---------------------------------------------------------------- Chris Rattray (1): ASoC: wm8994: Add missing dapm routes for WM8958 rev A Fabio Estevam (2): ASoC: mxs: Fix the name of the SoC family ASoC: imx-ssi: Remove mono support Mark Brown (10): Merge tag 'asoc-3.6' of git://git.kernel.org/.../broonie/sound into for-3.6 ASoC: wm8962: Don't duplicate bias level management in resume ASoC: core: Upgrade the severity of probe deferral errors to dev_err() MAINTAINERS: Add entries for Wolfson Arizona class devices ASoC: wm5102: Add missing input PGA routes ASoC: wm5110: Add missing input PGA routes ASoC: jack: Always notify full jack status ASoC: wm5102: Remove DRC2 ASoC: wm9712: Fix microphone source selection ASoC: wm9712: Fix inverted capture volume Peter Ujfalusi (1): ASoC: omap-mcbsp: Fix 6pin mux configuration Sachin Kamat (1): ASoC: Samsung: Fix build error Scott Jiang (1): ASoC: bfin: fix memory leak in sport3 controller driver Vaibhav Bedia (1): ASoC: Davinci: McASP: Flush the FIFO before enabling MAINTAINERS | 7 ++++++- sound/soc/blackfin/bf6xx-sport.c | 7 +++++++ sound/soc/codecs/ab8500-codec.c | 4 ++++ sound/soc/codecs/ad1980.c | 1 + sound/soc/codecs/mc13783.c | 2 ++ sound/soc/codecs/sgtl5000.c | 3 +-- sound/soc/codecs/stac9766.c | 1 + sound/soc/codecs/wm5102.c | 25 +++++++++---------------- sound/soc/codecs/wm5110.c | 12 ++++++++++++ sound/soc/codecs/wm8962.c | 18 +++--------------- sound/soc/codecs/wm8994.c | 17 ++++++++++++++++- sound/soc/codecs/wm9712.c | 22 +++++++++++++++++++--- sound/soc/codecs/wm9713.c | 1 + sound/soc/davinci/davinci-mcasp.c | 10 ++++++++-- sound/soc/fsl/imx-ssi.c | 5 +++-- sound/soc/mxs/Kconfig | 2 +- sound/soc/mxs/mxs-saif.c | 24 ++++++++++++++++++++++++ sound/soc/omap/mcbsp.c | 2 +- sound/soc/omap/omap-mcbsp.c | 1 + sound/soc/omap/omap-pcm.c | 1 + sound/soc/samsung/pcm.c | 2 +- sound/soc/soc-core.c | 12 +++++++----- sound/soc/soc-jack.c | 2 +- sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 10 +++++----- sound/soc/ux500/ux500_msp_dai.c | 2 +- sound/soc/ux500/ux500_msp_i2s.c | 2 +- sound/soc/ux500/ux500_msp_i2s.h | 2 +- 28 files changed, 139 insertions(+), 60 deletions(-)

2 1

[alsa-devel] [PATCH 0/3] ASoC: add compress stream support
by Vinod Koul 20 Aug '12

20 Aug '12

Here is the much delayed support for compress audio streams in ASoC. The v2 incorprates feedback discussed on v1 patches and this also adds the support in Intel drivers. Namarta Kohli (1): ASoC: add compress stream support Vinod Koul (2): ASoC: add definations for compressed operations ASoC: mid-x86 - add support for compressed streams include/sound/compress_driver.h | 1 + include/sound/soc-dai.h | 3 + include/sound/soc.h | 16 ++- sound/soc/Kconfig | 1 + sound/soc/Makefile | 2 +- sound/soc/mid-x86/mfld_machine.c | 9 ++ sound/soc/mid-x86/sst_dsp.h | 134 +++++++++++++++++ sound/soc/mid-x86/sst_platform.c | 204 ++++++++++++++++++++++++++- sound/soc/mid-x86/sst_platform.h | 26 ++++- sound/soc/soc-compress.c | 295 ++++++++++++++++++++++++++++++++++++++ sound/soc/soc-core.c | 57 +++++--- 11 files changed, 721 insertions(+), 27 deletions(-) create mode 100644 sound/soc/mid-x86/sst_dsp.h create mode 100644 sound/soc/soc-compress.c

2 4

[alsa-devel] [PATCH 0/10] fix error return code
by Julia Lawall 20 Aug '12

20 Aug '12

These patches fix cases where the return code appears to be unintentially nonnegative. The complete semantic match that finds the problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @ok exists@ identifier f,ret,i; expression e; constant c; @@ f(...) { <+... ( return -c@i; | ret = -c@i; ... when != ret = e return ret; | if (ret < 0) { ... return ret; } ) ...+> } @r exists@ identifier ret,l,ok.f; expression e1,e2,e3; statement S; position p1,p2,p3; @@ f(...) { ... when any ( if@p1 ($ret < 0\|ret != 0$) { ... return ret; } | ret@p1 = 0 ) ... when != ret = e1 when != &ret ( if (<+... ret = e3 ...+>) S | if (<+... &ret ...+>) S | if@p2(...) { ... when != ret = e2 when forall return@p3 ret; } ) ... when any } @bad exists@ position r.p1,r.p2; statement S1,S2; identifier r.ret; expression e1; @@ ( if@p1 ($ret < 0\|ret != 0$) S1 | ret@p1 = 0 ) ... when any ret = e1 ... when any if@p2(...) S2 @bad2@ position r.p1,r.p2; identifier r.ret; expression e1; statement S2; @@ ret@p1 = 0 ... when != if (...) { ... ret = e1 ... return ret; } when any if@p2(...) S2 @script:python depends on !bad && !bad2@ p1 << r.p1; p2 << r.p2; p3 << r.p3; @@ cocci.print_main("",p1) cocci.print_secs("",p2) cocci.print_secs("",p3) // </smpl>

4 13

Re: [alsa-devel] [sound:master 88/92] sound/pci/hda/./hda_trace.h:90:1: error: 'struct hda_codec' has no member named 'power_count'
by Takashi Iwai 20 Aug '12

20 Aug '12

At Mon, 20 Aug 2012 23:51:31 +0800, Fengguang Wu wrote: > > Hi Takashi, > > FYI, kernel build failed on > > tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git master > head: b9687805cf4e70e08221856b1bbadfa5e37c3342 > commit: b244d335609f9bbf4c0204a365bf03b7ac53c3c8 [88/92] ALSA: hda - Add tracepoints at snd_hda_power_up/down entrances. > config: i386-randconfig-k767 (attached as .config) > > All related error/warning messages: > > make[4]: *** [sound/pci/hda/hda_codec.o] Error 1 > make[4]: Target `__build' not remade because of errors. > -- > In file included from include/trace/ftrace.h:567:0, > from include/trace/define_trace.h:86, > from sound/pci/hda/hda_trace.h:141, > from sound/pci/hda/hda_codec.c:41: > sound/pci/hda/./hda_trace.h: In function 'ftrace_raw_event_hda_power_count': > sound/pci/hda/./hda_trace.h:90:1: error: 'struct hda_codec' has no member named 'power_count' > sound/pci/hda/./hda_trace.h:90:1: error: 'struct hda_codec' has no member named 'power_on' > sound/pci/hda/./hda_trace.h:90:1: error: 'struct hda_codec' has no member named 'power_transition' > In file included from include/trace/ftrace.h:775:0, > from include/trace/define_trace.h:86, > from sound/pci/hda/hda_trace.h:141, > from sound/pci/hda/hda_codec.c:41: > sound/pci/hda/./hda_trace.h: In function 'perf_trace_hda_power_count': > sound/pci/hda/./hda_trace.h:90:1: error: 'struct hda_codec' has no member named 'power_count' > sound/pci/hda/./hda_trace.h:90:1: error: 'struct hda_codec' has no member named 'power_on' > sound/pci/hda/./hda_trace.h:90:1: error: 'struct hda_codec' has no member named 'power_transition' > > vim +90 sound/pci/hda/./hda_trace.h > 87 TP_ARGS(codec) > 88 ); > 89 > > 90 TRACE_EVENT(hda_power_count, > 91 TP_PROTO(struct hda_codec *codec), > 92 TP_ARGS(codec), > 93 TP_STRUCT__entry( Thanks, looks like the missing ifdef. Fixed now. Takashi

1 0

[alsa-devel] Lockups reading from i.MX51 SSI registers
by Mark Brown 20 Aug '12

20 Aug '12

Hi, I'm trying to use -next to test AC'97 register I/O on an i.MX51 board but I'm seeing the CPU hang during probe at: lreg = (reg & 0x7f) << 12 ; writel(lreg, base + SSI_SACADD); in imx_ssi_ac97_read(). I'm somewhat suspicious this might be because the IP block isn't clocked properly, I do notice the recent conversion to the clock API which looks rather involved but it's possible something else broke. Does anyone have any bright ideas what might be going on here? The board doesn't have the reste functions defined so this is the first interaction with the hardware block AFAICT. Thanks, Mark

5 11

[alsa-devel] [alsa-plugins PATCH] pulse: Support 24-bit formats
by Andrew Eikum 20 Aug '12

20 Aug '12

These formats were introduced in PulseAudio 0.9.15. There aren't any new formats supported since then, so this also brings the PulseAudio plugin in sync with current PulseAudio. Signed-off-by: Andrew Eikum <aeikum(a)codeweavers.com> --- There was a problem once before with a patch I submitted. This is using format-patch and imap-send, so hopefully it goes through better than it did before :) pulse/pcm_pulse.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+)

2 4

[alsa-devel] [PATCH 0/2] Two fixes for ca0132
by David Henningsson 20 Aug '12

20 Aug '12

I came across the following codec: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1038651/+attachment/32… ...and the codec wouldn't load due to mixer errors. I don't know more about the codec than the codec proc shows, and looking at that file it seems a bit broken, but maybe Ian would be able to make it work, with the help of some secret verbs? At least now the codec would load correctly, so that the output side could be used. David Henningsson (2): ALSA: hda - Fix "Line" and "Mic" names for CA0132 ALSA: hda - don't create dysfunctional mixer controls for ca0132 sound/pci/hda/patch_ca0132.c | 12 ++++++++++-- 1 file changed, 10 insertions(+), 2 deletions(-) -- 1.7.9.5

2 6

[alsa-devel] Broken powersave in 3.5.2, codec is always active
by ValdikSS 20 Aug '12

20 Aug '12

Hello! I have a Lenovo ThinkPad X220 laptop with a conexant codec and recently (after kernel 3.5.2) it is active all the time after KDE login, as powertop says. This couldn't be fixed by logging of from KDE or playing some music in mplayer from console. The only kernel I tried in 3.5.x is 3.5.2, so I can't say is this bug reproducible in 3.5.0 or 3.5.1, but everything is fine with 3.4.9. This never happens if you don't start KDE after reboot, you can listen to some music in mplayer in console and everything seems to be fine. lsof and fuser /dev/snd/* shows nothing. Thank you! $ cat /etc/modprobe.d/options.conf options snd-hda-intel power_save=1 power_save_controller=1 $ lspci -nn | grep Audio 00:1b.0 Audio device [0403]: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller [8086:1c20] (rev 04) $ lsmod | grep snd snd_hda_codec_hdmi 23704 1 snd_hda_codec_conexant 47158 1 snd_hda_intel 25700 2 snd_hda_codec 97400 3 snd_hda_codec_hdmi,snd_hda_codec_conexant,snd_hda_intel snd_hwdep 6300 1 snd_hda_codec snd_pcm 74926 3 snd_hda_codec_hdmi,snd_hda_codec,snd_hda_intel snd_page_alloc 7185 2 snd_pcm,snd_hda_intel snd_timer 18966 1 snd_pcm snd 60021 12 snd_hwdep,snd_timer,snd_hda_codec_hdmi,snd_hda_codec_conexant,snd_pcm,snd_hda_codec,snd_hda_intel,thinkpad_acpi soundcore 5442 1 snd

2 1

[alsa-devel] [PATCH 1/2] snd-ad1816a: remove useless struct snd_card_ad1816a
by Ondrej Zary 20 Aug '12

20 Aug '12

struct snd_card_ad1816a is only set but the values are never used then. Removing it allows struct snd_card's private_data to be used for struct snd_ad1816a, simplifying the code. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> diff --git a/include/sound/ad1816a.h b/include/sound/ad1816a.h index d010858..62da41e 100644 --- a/include/sound/ad1816a.h +++ b/include/sound/ad1816a.h @@ -165,7 +165,7 @@ struct snd_ad1816a { extern int snd_ad1816a_create(struct snd_card *card, unsigned long port, int irq, int dma1, int dma2, - struct snd_ad1816a **chip); + struct snd_ad1816a *chip); extern int snd_ad1816a_pcm(struct snd_ad1816a *chip, int device, struct snd_pcm **rpcm); extern int snd_ad1816a_mixer(struct snd_ad1816a *chip); diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 94b83b6..1a374e6 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -63,11 +63,6 @@ MODULE_PARM_DESC(enable, "Enable ad1816a based soundcard."); module_param_array(clockfreq, int, NULL, 0444); MODULE_PARM_DESC(clockfreq, "Clock frequency for ad1816a driver (default = 0)."); -struct snd_card_ad1816a { - struct pnp_dev *dev; - struct pnp_dev *devmpu; -}; - static struct pnp_card_device_id snd_ad1816a_pnpids[] = { /* Analog Devices AD1815 */ { .id = "ADS7150", .devs = { { .id = "ADS7150" }, { .id = "ADS7151" } } }, @@ -99,25 +94,16 @@ MODULE_DEVICE_TABLE(pnp_card, snd_ad1816a_pnpids); #define DRIVER_NAME "snd-card-ad1816a" -static int __devinit snd_card_ad1816a_pnp(int dev, struct snd_card_ad1816a *acard, - struct pnp_card_link *card, +static int __devinit snd_card_ad1816a_pnp(int dev, struct pnp_card_link *card, const struct pnp_card_device_id *id) { struct pnp_dev *pdev; int err; - acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) + pdev = pnp_request_card_device(card, id->devs[0].id, NULL); + if (pdev == NULL) return -EBUSY; - acard->devmpu = pnp_request_card_device(card, id->devs[1].id, NULL); - if (acard->devmpu == NULL) { - mpu_port[dev] = -1; - snd_printk(KERN_WARNING PFX "MPU401 device busy, skipping.\n"); - } - - pdev = acard->dev; - err = pnp_activate_dev(pdev); if (err < 0) { printk(KERN_ERR PFX "AUDIO PnP configure failure\n"); @@ -130,16 +116,17 @@ static int __devinit snd_card_ad1816a_pnp(int dev, struct snd_card_ad1816a *acar dma2[dev] = pnp_dma(pdev, 1); irq[dev] = pnp_irq(pdev, 0); - if (acard->devmpu == NULL) + pdev = pnp_request_card_device(card, id->devs[1].id, NULL); + if (pdev == NULL) { + mpu_port[dev] = -1; + snd_printk(KERN_WARNING PFX "MPU401 device busy, skipping.\n"); return 0; - - pdev = acard->devmpu; + } err = pnp_activate_dev(pdev); if (err < 0) { printk(KERN_ERR PFX "MPU401 PnP configure failure\n"); mpu_port[dev] = -1; - acard->devmpu = NULL; } else { mpu_port[dev] = pnp_port_start(pdev, 0); mpu_irq[dev] = pnp_irq(pdev, 0); @@ -153,18 +140,17 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard { int error; struct snd_card *card; - struct snd_card_ad1816a *acard; struct snd_ad1816a *chip; struct snd_opl3 *opl3; struct snd_timer *timer; error = snd_card_create(index[dev], id[dev], THIS_MODULE, - sizeof(struct snd_card_ad1816a), &card); + sizeof(struct snd_ad1816a), &card); if (error < 0) return error; - acard = card->private_data; + chip = card->private_data; - if ((error = snd_card_ad1816a_pnp(dev, acard, pcard, pid))) { + if ((error = snd_card_ad1816a_pnp(dev, pcard, pid))) { snd_card_free(card); return error; } @@ -174,7 +160,7 @@ static int __devinit snd_card_ad1816a_probe(int dev, struct pnp_card_link *pcard irq[dev], dma1[dev], dma2[dev], - &chip)) < 0) { + chip)) < 0) { snd_card_free(card); return error; } diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index 177eed3..de5cc1c 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -548,7 +548,6 @@ static int snd_ad1816a_free(struct snd_ad1816a *chip) snd_dma_disable(chip->dma2); free_dma(chip->dma2); } - kfree(chip); return 0; } @@ -573,19 +572,13 @@ static const char __devinit *snd_ad1816a_chip_id(struct snd_ad1816a *chip) int __devinit snd_ad1816a_create(struct snd_card *card, unsigned long port, int irq, int dma1, int dma2, - struct snd_ad1816a **rchip) + struct snd_ad1816a *chip) { static struct snd_device_ops ops = { .dev_free = snd_ad1816a_dev_free, }; int error; - struct snd_ad1816a *chip; - *rchip = NULL; - - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) - return -ENOMEM; chip->irq = -1; chip->dma1 = -1; chip->dma2 = -1; @@ -631,7 +624,6 @@ int __devinit snd_ad1816a_create(struct snd_card *card, return error; } - *rchip = chip; return 0; } -- Ondrej Zary

2 1

[alsa-devel] [PATCH] sound: fix pcm.h kernel-doc warning and notation
by Randy Dunlap 20 Aug '12

20 Aug '12

From: Randy Dunlap <rdunlap(a)xenotime.net> Fix kernel-doc warning in <sound/pcm.h> and add function name to make the kernel-doc notation complete. Warning(include/sound/pcm.h:1081): No description found for parameter 'substream' Signed-off-by: Randy Dunlap <rdunlap(a)xenotime.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org (moderated for non-subscribers) --- include/sound/pcm.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) --- lnx-36-rc2.orig/include/sound/pcm.h +++ lnx-36-rc2/include/sound/pcm.h @@ -1075,7 +1075,8 @@ static inline void snd_pcm_limit_isa_dma const char *snd_pcm_format_name(snd_pcm_format_t format); /** - * Get a string naming the direction of a stream + * snd_pcm_stream_str - Get a string naming the direction of a stream + * @substream: the pcm substream instance */ static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream) {

2 1

[alsa-devel] [patch] oss/sb_audio: prevent divide by zero bug
by Dan Carpenter 20 Aug '12

20 Aug '12

Speed comes from get_user() in audio_ioctl(). We use it to set the "s" variable before clamping it to valid values so it could lead to a divide by zero bug. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/oss/sb_audio.c b/sound/oss/sb_audio.c index 733b014..b2b3c01 100644 --- a/sound/oss/sb_audio.c +++ b/sound/oss/sb_audio.c @@ -575,13 +575,15 @@ static int jazz16_audio_set_speed(int dev, int speed) if (speed > 0) { int tmp; - int s = speed * devc->channels; + int s; if (speed < 5000) speed = 5000; if (speed > 44100) speed = 44100; + s = speed * devc->channels; + devc->tconst = (256 - ((1000000 + s / 2) / s)) & 0xff; tmp = 256 - devc->tconst;

2 1

Re: [alsa-devel] Question about Conversion to S24_LE from S24_3LE
by Takashi Iwai 20 Aug '12

20 Aug '12

At Mon, 20 Aug 2012 14:57:51 +0900 (JST), etsuko suzuki wrote: > > Hello, Takashi. > > > Hm, I guess your message didn't reach properly, or was filtered by some spam > filter. > > Better to contact with Jaroslav (Cc'ed), who maintains alsa-devel ML. > > I contacted him in August 17. > However, there is no reply yet. Didn't you read his mail in Aug. 16? You were in Cc. > > [No.1] > > > > http://msdn.microsoft.com/en-us/windows/hardware/gg463006.aspx > > > > > > In this, I found the statement 'left-alinged'. > > > (In 'Details about wValidBitsPerSample') > > > > > > I understood 'left-alinged' was 'LSB padding'. > > > So I thought S24_LE was LSB padding. > > > > No. S24_LE has the MSB padding of 8 bits in 32 bit container. > > That is, does it mean that this document is not correct? > (I don't understand why this document differs from what you say.) Well, Microsoft doesn't conquer the whole world yet. Their definition isn't what we define for our purpose. > http://www-mmsp.ece.mcgill.ca/documents/audioformats/wave/Docs/multichaudP.… > (This PDF is the same as what was written before. > The page is written on this.) > > On the 6 page of this PDF, it is written to like this. > > 'If wValidBitsPerSample is less than wBitsPerSample, > then the actual > PCM data is "left-aligned" within the container. > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ > > The sample itself is justified most significant; > all extra bits are at the least-significant > portion of the container. > All non-valid data bits must be set to 0.' > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ > > In S24_LE, I think 'wBitsPerSample' indicates > 'the container size'(=32) and > 'wValidBitsPerSample' indicates 'the bit of actual valid data'(=24). You _think_ but you utterly misunderstand the terms. ALSA S24 is not what you think. S24 is a special format, 3 bytes LSB in 4 bytes container. This _is_ the definition. There is no such a format in MS's WAV definition. wValidBitsPerSample is rather corresponding to hw_params ms_bits parameter. > > [No.2] > > It was written that Microsoft has defined > the new structure 'WAVE_FORMAT_EXTENSIBLE'(FORMAT_CODE:0xFFFE). > (On the 4 page of this PDF.) > > I think that the new structure should be used > in this(the container size > the bit of actual valid data) case. > > But the structure recording > by arecord was 'WAVE_FORMAT_PCM'(FORMAT_CODE:0x0001). > > > % od -t x4 ubuntu_rec_S24_LE_48000.wav > > 0000000 46464952 003a9824 45564157 20746d66 > > 0000010 00000010 00020001 0000bb80 0005dc00 > ^^^^ 0x0001:WAVE_FORMAT_PCM > > (The 'FORMAT_CODE' is written in this page. > http://www-mmsp.ece.mcgill.ca/documents/audioformats/wave/wave.html) > > > Why does ALSA NOT use the new structure? Because it's new, and no one cared such a thing. > Has ALSA corresponded to the new structure? Please send a fix patch if you'd like to improve :) thanks, Takashi

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[alsa-devel] [PATCH 2/2] snd-ad1816a: Implement suspend/resume
by Ondrej Zary 19 Aug '12

19 Aug '12

Implement suspend/resume support for AD1816 chips. Tested with Terratec SoundSystem Base-1. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> diff --git a/include/sound/ad1816a.h b/include/sound/ad1816a.h index 62da41e..2a89f0d 100644 --- a/include/sound/ad1816a.h +++ b/include/sound/ad1816a.h @@ -147,6 +147,9 @@ struct snd_ad1816a { unsigned int c_dma_size; struct snd_timer *timer; +#ifdef CONFIG_PM + unsigned short image[48]; +#endif }; @@ -171,5 +174,9 @@ extern int snd_ad1816a_pcm(struct snd_ad1816a *chip, int device, struct snd_pcm extern int snd_ad1816a_mixer(struct snd_ad1816a *chip); extern int snd_ad1816a_timer(struct snd_ad1816a *chip, int device, struct snd_timer **rtimer); +#ifdef CONFIG_PM +extern void snd_ad1816a_suspend(struct snd_ad1816a *chip); +extern void snd_ad1816a_resume(struct snd_ad1816a *chip); +#endif #endif /* __SOUND_AD1816A_H */ diff --git a/sound/isa/ad1816a/ad1816a.c b/sound/isa/ad1816a/ad1816a.c index 1a374e6..2c2f829 100644 --- a/sound/isa/ad1816a/ad1816a.c +++ b/sound/isa/ad1816a/ad1816a.c @@ -244,13 +244,37 @@ static void __devexit snd_ad1816a_pnp_remove(struct pnp_card_link * pcard) pnp_set_card_drvdata(pcard, NULL); } +#ifdef CONFIG_PM +static int snd_ad1816a_pnp_suspend(struct pnp_card_link *pcard, + pm_message_t state) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_ad1816a_suspend(card->private_data); + return 0; +} + +static int snd_ad1816a_pnp_resume(struct pnp_card_link *pcard) +{ + struct snd_card *card = pnp_get_card_drvdata(pcard); + + snd_ad1816a_resume(card->private_data); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; +} +#endif + static struct pnp_card_driver ad1816a_pnpc_driver = { .flags = PNP_DRIVER_RES_DISABLE, .name = "ad1816a", .id_table = snd_ad1816a_pnpids, .probe = snd_ad1816a_pnp_detect, .remove = __devexit_p(snd_ad1816a_pnp_remove), - /* FIXME: suspend/resume */ +#ifdef CONFIG_PM + .suspend = snd_ad1816a_pnp_suspend, + .resume = snd_ad1816a_pnp_resume, +#endif }; static int __init alsa_card_ad1816a_init(void) diff --git a/sound/isa/ad1816a/ad1816a_lib.c b/sound/isa/ad1816a/ad1816a_lib.c index de5cc1c..db64df6 100644 --- a/sound/isa/ad1816a/ad1816a_lib.c +++ b/sound/isa/ad1816a/ad1816a_lib.c @@ -491,7 +491,7 @@ static int snd_ad1816a_capture_close(struct snd_pcm_substream *substream) } -static void __devinit snd_ad1816a_init(struct snd_ad1816a *chip) +static void snd_ad1816a_init(struct snd_ad1816a *chip) { unsigned long flags; @@ -511,6 +511,32 @@ static void __devinit snd_ad1816a_init(struct snd_ad1816a *chip) spin_unlock_irqrestore(&chip->lock, flags); } +#ifdef CONFIG_PM +void snd_ad1816a_suspend(struct snd_ad1816a *chip) +{ + int reg; + unsigned long flags; + + snd_pcm_suspend_all(chip->pcm); + spin_lock_irqsave(&chip->lock, flags); + for (reg = 0; reg < 48; reg++) + chip->image[reg] = snd_ad1816a_read(chip, reg); + spin_unlock_irqrestore(&chip->lock, flags); +} + +void snd_ad1816a_resume(struct snd_ad1816a *chip) +{ + int reg; + unsigned long flags; + + snd_ad1816a_init(chip); + spin_lock_irqsave(&chip->lock, flags); + for (reg = 0; reg < 48; reg++) + snd_ad1816a_write(chip, reg, chip->image[reg]); + spin_unlock_irqrestore(&chip->lock, flags); +} +#endif + static int __devinit snd_ad1816a_probe(struct snd_ad1816a *chip) { unsigned long flags; -- Ondrej Zary

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Re: [alsa-devel] M-Audio FastTrack C400
by Grant Diffey 18 Aug '12

18 Aug '12

So this looks reasonably familiar, int0 is DFU int1 is mixer int2 is audio out int3 is audio in int4 is midi int5 is HID Quirk should look like the fast track ultra basically. maybe the mixer is less broken in this device? Should we start with just QUIRK_STANDARD_AUDIO_INTERFACE or copy the whole data section from the Ultra quirk? This has isosyncronous feedback for clock control needs current sound tree usb-audio tree Grant On Wed, Apr 18, 2012 at 8:45 AM, Chris Cavey <chris-alsa(a)rauros.net> wrote: > Will to help with development, may need pointers to get started and > willing to test things. I am mostly concerned with simple playback. > Recording, DSP, and Midi can all come later if at all. > > As requested the output of sudo lsusb -v -d 0763:2030" > > Bus 002 Device 003: ID 0763:2030 Midiman > Device Descriptor: > bLength 18 > bDescriptorType 1 > bcdUSB 2.00 > bDeviceClass 239 Miscellaneous Device > bDeviceSubClass 2 ? > bDeviceProtocol 1 Interface Association > bMaxPacketSize0 64 > idVendor 0x0763 Midiman > idProduct 0x2030 > bcdDevice 1.01 > iManufacturer 1 M-Audio > iProduct 2 Fast Track C400 > iSerial 0 > bNumConfigurations 1 > Configuration Descriptor: > bLength 9 > bDescriptorType 2 > wTotalLength 455 > bNumInterfaces 6 > bConfigurationValue 1 > iConfiguration 0 > bmAttributes 0x80 > (Bus Powered) > MaxPower 500mA > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 0 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 254 Application Specific Interface > bInterfaceSubClass 1 Device Firmware Update > bInterfaceProtocol 0 > iInterface 6 Fast Track C400 DFU > Device Firmware Upgrade Interface Descriptor: > bLength 9 > bDescriptorType 33 > bmAttributes 1 > Will Not Detach > Manifestation Intolerant > Upload Unsupported > Download Supported > wDetachTimeout 2000 milliseconds > wTransferSize 1024 bytes > bcdDFUVersion 1.10 > Interface Association: > bLength 8 > bDescriptorType 11 > bFirstInterface 1 > bInterfaceCount 4 > bFunctionClass 255 Vendor Specific Class > bFunctionSubClass 0 > bFunctionProtocol 32 > iFunction 2 Fast Track C400 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 1 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 1 > bInterfaceProtocol 32 > iInterface 2 Fast Track C400 > ** UNRECOGNIZED: 09 24 01 00 02 0a a3 00 00 > ** UNRECOGNIZED: 09 24 0b 80 02 81 82 03 00 > ** UNRECOGNIZED: 08 24 0a 81 03 03 03 00 > ** UNRECOGNIZED: 08 24 0a 82 00 05 03 00 > ** UNRECOGNIZED: 11 24 02 00 01 01 00 80 06 00 00 00 00 00 00 00 00 > ** UNRECOGNIZED: 0c 24 03 01 01 01 00 02 80 00 00 00 > ** UNRECOGNIZED: 11 24 02 02 03 06 00 80 04 00 00 00 00 00 00 00 00 > ** UNRECOGNIZED: 0c 24 03 03 03 06 00 40 80 00 00 00 > ** UNRECOGNIZED: 11 24 09 20 00 00 02 02 00 04 00 00 00 00 00 00 00 > ** UNRECOGNIZED: 15 24 04 40 03 02 00 43 04 00 00 00 00 00 ff ff ff > ff 0f 00 00 > ** UNRECOGNIZED: 10 24 04 42 02 02 00 01 00 00 00 00 00 ff 00 00 > ** UNRECOGNIZED: 10 24 07 43 02 00 42 fd 03 00 00 fd 03 00 00 00 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 32 > iInterface 0 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 1 > bNumEndpoints 1 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 32 > iInterface 4 Line > ** UNRECOGNIZED: 10 24 01 00 00 01 01 00 00 00 06 00 00 00 00 00 > ** UNRECOGNIZED: 06 24 02 01 03 18 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 37 > Transfer Type Isochronous > Synch Type Asynchronous > Usage Type Implicit feedback Data > wMaxPacketSize 0x00ea 1x 234 bytes > bInterval 1 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 2 > bAlternateSetting 2 > bNumEndpoints 1 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 32 > iInterface 4 Line > ** UNRECOGNIZED: 10 24 01 00 00 01 d0 00 00 00 02 00 00 00 00 00 > ** UNRECOGNIZED: 06 24 02 03 02 10 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x01 EP 1 OUT > bmAttributes 37 > Transfer Type Isochronous > Synch Type Asynchronous > Usage Type Implicit feedback Data > wMaxPacketSize 0x004e 1x 78 bytes > bInterval 1 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 3 > bAlternateSetting 0 > bNumEndpoints 0 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 32 > iInterface 4 Line > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 3 > bAlternateSetting 1 > bNumEndpoints 1 > bInterfaceClass 255 Vendor Specific Class > bInterfaceSubClass 2 > bInterfaceProtocol 32 > iInterface 4 Line > ** UNRECOGNIZED: 10 24 01 00 00 01 01 00 00 00 04 00 00 00 00 00 > ** UNRECOGNIZED: 06 24 02 01 03 18 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x81 EP 1 IN > bmAttributes 5 > Transfer Type Isochronous > Synch Type Asynchronous > Usage Type Data > wMaxPacketSize 0x009c 1x 156 bytes > bInterval 1 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 4 > bAlternateSetting 0 > bNumEndpoints 2 > bInterfaceClass 1 Audio > bInterfaceSubClass 3 MIDI Streaming > bInterfaceProtocol 0 > iInterface 0 > MIDIStreaming Interface Descriptor: > bLength 7 > bDescriptorType 36 > bDescriptorSubtype 1 (HEADER) > bcdADC 1.00 > wTotalLength 35 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 1 Embedded > bJackID 1 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 6 > bDescriptorType 36 > bDescriptorSubtype 2 (MIDI_IN_JACK) > bJackType 2 External > bJackID 2 > iJack 0 > MIDIStreaming Interface Descriptor: > bLength 8 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > Warning: Descriptor too short > bJackType 1 Embedded > bJackID 3 > bNrInputPins 1 > baSourceID( 0) 2 > BaSourcePin( 0) 1 > iJack 8 > MIDIStreaming Interface Descriptor: > bLength 8 > bDescriptorType 36 > bDescriptorSubtype 3 (MIDI_OUT_JACK) > Warning: Descriptor too short > bJackType 2 External > bJackID 4 > bNrInputPins 1 > baSourceID( 0) 1 > BaSourcePin( 0) 1 > iJack 105 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x07 EP 7 OUT > bmAttributes 2 > Transfer Type Bulk > Synch Type None > Usage Type Data > wMaxPacketSize 0x0200 1x 512 bytes > bInterval 0 > MIDIStreaming Endpoint Descriptor: > bLength 5 > bDescriptorType 37 > bDescriptorSubtype 1 (GENERAL) > bNumEmbMIDIJack 1 > baAssocJackID( 0) 1 > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x87 EP 7 IN > bmAttributes 2 > Transfer Type Bulk > Synch Type None > Usage Type Data > wMaxPacketSize 0x0200 1x 512 bytes > bInterval 0 > MIDIStreaming Endpoint Descriptor: > bLength 5 > bDescriptorType 37 > bDescriptorSubtype 1 (GENERAL) > bNumEmbMIDIJack 1 > baAssocJackID( 0) 3 > Interface Descriptor: > bLength 9 > bDescriptorType 4 > bInterfaceNumber 5 > bAlternateSetting 0 > bNumEndpoints 1 > bInterfaceClass 3 Human Interface Device > bInterfaceSubClass 0 No Subclass > bInterfaceProtocol 1 Keyboard > iInterface 0 > HID Device Descriptor: > bLength 9 > bDescriptorType 33 > bcdHID 1.10 > bCountryCode 0 Not supported > bNumDescriptors 1 > bDescriptorType 34 Report > wDescriptorLength 63 > Report Descriptors: > ** UNAVAILABLE ** > Endpoint Descriptor: > bLength 7 > bDescriptorType 5 > bEndpointAddress 0x86 EP 6 IN > bmAttributes 3 > Transfer Type Interrupt > Synch Type None > Usage Type Data > wMaxPacketSize 0x0008 1x 8 bytes > bInterval 8 > Device Qualifier (for other device speed): > bLength 10 > bDescriptorType 6 > bcdUSB 2.00 > bDeviceClass 239 Miscellaneous Device > bDeviceSubClass 2 ? > bDeviceProtocol 1 Interface Association > bMaxPacketSize0 64 > bNumConfigurations 1 > Device Status: 0x0000 > (Bus Powered) > > > On 04/17/2012 09:15 AM, Grant Diffey wrote: > > start with an lsusb -v -d (vid:pid) pasted here > > On Tue, Apr 17, 2012 at 10:37 PM, Chris Cavey <chris-alsa(a)rauros.net>wrote: > >> I'd love to be able to help ALSA support this USB sound device. I've >> currently got the device working in a VM on my Ubuntu 12.04 workstation. >> I believe I can capture USB packets. Would this be helpful to anyone? >> Any pointers on where to get started? >> _______________________________________________ >> Alsa-devel mailing list >> Alsa-devel(a)alsa-project.org >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >> > > >

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[alsa-devel] [PATCH 1/2 v2] ASoC: bf5xx-ad1836: convert to use snd_soc_register_card
by Scott Jiang 18 Aug '12

18 Aug '12

Cpu dai and codec name are passed in through platform data. Signed-off-by: Scott Jiang <scott.jiang.linux(a)gmail.com> --- sound/soc/blackfin/bf5xx-ad1836.c | 73 +++++++++++++++++++------------------ 1 files changed, 37 insertions(+), 36 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index d542d40..16b9c9e 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -59,62 +59,63 @@ static struct snd_soc_ops bf5xx_ad1836_ops = { #define BF5XX_AD1836_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ SND_SOC_DAIFMT_CBM_CFM) -static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { - { - .name = "ad1836", - .stream_name = "AD1836", - .cpu_dai_name = "bfin-tdm.0", - .codec_dai_name = "ad1836-hifi", - .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "spi0.4", - .ops = &bf5xx_ad1836_ops, - .dai_fmt = BF5XX_AD1836_DAIFMT, - }, - { - .name = "ad1836", - .stream_name = "AD1836", - .cpu_dai_name = "bfin-tdm.1", - .codec_dai_name = "ad1836-hifi", - .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "spi0.4", - .ops = &bf5xx_ad1836_ops, - .dai_fmt = BF5XX_AD1836_DAIFMT, - }, +static struct snd_soc_dai_link bf5xx_ad1836_dai = { + .name = "ad1836", + .stream_name = "AD1836", + .codec_dai_name = "ad1836-hifi", + .platform_name = "bfin-tdm-pcm-audio", + .ops = &bf5xx_ad1836_ops, + .dai_fmt = BF5XX_AD1836_DAIFMT, }; static struct snd_soc_card bf5xx_ad1836 = { .name = "bfin-ad1836", .owner = THIS_MODULE, - .dai_link = &bf5xx_ad1836_dai[CONFIG_SND_BF5XX_SPORT_NUM], + .dai_link = &bf5xx_ad1836_dai, .num_links = 1, }; -static struct platform_device *bfxx_ad1836_snd_device; - -static int __init bf5xx_ad1836_init(void) +static __devinit int bf5xx_ad1836_driver_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &bf5xx_ad1836; + const char **link_name; int ret; - bfxx_ad1836_snd_device = platform_device_alloc("soc-audio", -1); - if (!bfxx_ad1836_snd_device) - return -ENOMEM; + link_name = pdev->dev.platform_data; + if (!link_name) { + dev_err(&pdev->dev, "No platform data supplied\n"); + return -EINVAL; + } + bf5xx_ad1836_dai.cpu_dai_name = link_name[0]; + bf5xx_ad1836_dai.codec_name = link_name[1]; - platform_set_drvdata(bfxx_ad1836_snd_device, &bf5xx_ad1836); - ret = platform_device_add(bfxx_ad1836_snd_device); + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + ret = snd_soc_register_card(card); if (ret) - platform_device_put(bfxx_ad1836_snd_device); - + dev_err(&pdev->dev, "Failed to register card\n"); return ret; } -static void __exit bf5xx_ad1836_exit(void) +static int __devexit bf5xx_ad1836_driver_remove(struct platform_device *pdev) { - platform_device_unregister(bfxx_ad1836_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + return 0; } -module_init(bf5xx_ad1836_init); -module_exit(bf5xx_ad1836_exit); +static struct platform_driver bf5xx_ad1836_driver = { + .driver = { + .name = "bfin-snd-ad1836", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = bf5xx_ad1836_driver_probe, + .remove = __devexit_p(bf5xx_ad1836_driver_remove), +}; +module_platform_driver(bf5xx_ad1836_driver); /* Module information */ MODULE_AUTHOR("Barry Song"); -- 1.7.0.4

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Re: [alsa-devel] [PATCH 1/5] ARM: tegra: config: enable dmaengine based APB DMA driver
by Stephen Warren 17 Aug '12

17 Aug '12

On 08/17/2012 12:38 AM, Laxman Dewangan wrote: > On Thursday 16 August 2012 11:23 PM, Stephen Warren wrote: >> On 08/16/2012 08:13 AM, Laxman Dewangan wrote: >>> Enable config for dmaengine based Tegra APB DMA driver and >>> disable the legacy APB DMA driver (SYSTEM_DMA). >> Laxman, if I apply this series to next-20120816 (plus a few patches in >> my local work branch plus the CPU hotplug patches from Joseph, although >> I suspect none of that matters), then audio playback on Tegra20 is >> broken; the pitch is far too high. Audio playback on Tegra30 works as >> expected. > > I run the test again on ventana and did not see any issue. > I play one song and saw similar behavior with/without this series. OK, I'll test some other boards. I tested Whistler, although all the clocking logic is identical between all Tegra20 boards. > With this series: > -------------------- > > ubuntu@tegra-ubuntu:/home$ aplay "1-04 Hungry Like The Wolf.wav" > Playing WAVE '1-04 Hungry Like The Wolf.wav' : Signed 16 bit Little > Endian, Rate > 44100 Hz, Stereo > underrun!!! (at least 5039.388 ms long) Why are you getting all these underruns? I almost never see this, except when something else is hammering the SD card containing the .wav file. If I saw those, I'd count it as a test failure.

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[alsa-devel] How to define card specific pcm devices?
by Tanu Kaskinen 17 Aug '12

17 Aug '12

Hi, I'm writing UCM configuration for RX-51 (Nokia N900), and I have a problem: "hw:RX51" can't be opened in mono mode, but there's only one integrated mono microphone, and its audio is routed to the left input channel. Applications will get only silence in the right channel when they use "hw:RX51". The best solution that I have come up is to define a pcm that converts "hw:RX51" into a mono device. The problem is that I don't know how exactly to do that. I guess the route plugin can be used for that: pcm.rx51_mono_from_left { type route slave.pcm hw:RX51 slave.channels 2 ttable.0.0 1 } I tried putting that definition in /usr/share/alsa/cards/RX-51.conf (and also RX51.conf, since it's not entirely clear to me when the dash should be omitted). I have this in the UCM configuration: Value { CapturePCM "rx51_mono_from_left" CaptureChannels "1" } This is printed to the PulseAudio log: (alsa-lib)pcm.c: Unknown PCM rx51_mono_from_left So the rx51_mono_from_left definition is not loaded. I'm not terribly surprised: I guess the card configuration files are loaded only when necessary, and when PulseAudio tries to open "rx51_mono_from_left", alsa-lib can't really know that it's somehow related to the RX-51 card. How should I solve this? -- Tanu

1 1

[alsa-devel] [PATCH V6 12/15] ALSA: HDA: Make hda sound card usable for Loongson
by Huacai Chen 17 Aug '12

17 Aug '12

Lemote A1004(Laptop) and A1205(All-In-One) use Conexant's hda codec, this patch modify patch_conexant.c to add Lemote specific code. Both A1004 and A1205 use the same pin configurations, but A1004 need to increase the default boost of internal mic. Signed-off-by: Jie Chen <chenj(a)lemote.com> Signed-off-by: Huacai Chen <chenhc(a)lemote.com> Signed-off-by: Hongliang Tao <taohl(a)lemote.com> Signed-off-by: Hua Yan <yanh(a)lemote.com> Cc: alsa-devel(a)alsa-project.org --- sound/pci/hda/patch_conexant.c | 44 ++++++++++++++++++++++++++++++++++++++++ 1 files changed, 44 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5e22a8f..3cc265e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4408,7 +4408,10 @@ static const struct hda_codec_ops cx_auto_patch_ops = { enum { CXT_PINCFG_LENOVO_X200, CXT_PINCFG_LENOVO_TP410, + CXT_PINCFG_LEMOTE_A1004, + CXT_PINCFG_LEMOTE_A1205, CXT_FIXUP_STEREO_DMIC, + CXT_FIXUP_INC_MIC_BOOST, }; static void cxt_fixup_stereo_dmic(struct hda_codec *codec, @@ -4418,6 +4421,19 @@ static void cxt_fixup_stereo_dmic(struct hda_codec *codec, spec->fixup_stereo_dmic = 1; } +static void cxt5066_increase_mic_boost(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action != HDA_FIXUP_ACT_PRE_PROBE) + return; + + snd_hda_override_amp_caps(codec, 0x17, HDA_OUTPUT, + (0x3 << AC_AMPCAP_OFFSET_SHIFT) | + (0x4 << AC_AMPCAP_NUM_STEPS_SHIFT) | + (0x27 << AC_AMPCAP_STEP_SIZE_SHIFT) | + (0 << AC_AMPCAP_MUTE_SHIFT)); +} + /* ThinkPad X200 & co with cxt5051 */ static const struct hda_pintbl cxt_pincfg_lenovo_x200[] = { { 0x16, 0x042140ff }, /* HP (seq# overridden) */ @@ -4435,6 +4451,18 @@ static const struct hda_pintbl cxt_pincfg_lenovo_tp410[] = { {} }; +/* Lemote A1004/A1205 with cxt5066 */ +static const struct hda_pintbl cxt_pincfg_lemote[] = { + { 0x1a, 0x90a10020 }, /* Internal mic */ + { 0x1b, 0x03a11020 }, /* External mic */ + { 0x1d, 0x400101f0 }, /* Not used */ + { 0x1e, 0x40a701f0 }, /* Not used */ + { 0x20, 0x404501f0 }, /* Not used */ + { 0x22, 0x404401f0 }, /* Not used */ + { 0x23, 0x40a701f0 }, /* Not used */ + {} +}; + static const struct hda_fixup cxt_fixups[] = { [CXT_PINCFG_LENOVO_X200] = { .type = HDA_FIXUP_PINS, @@ -4444,10 +4472,24 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_PINS, .v.pins = cxt_pincfg_lenovo_tp410, }, + [CXT_PINCFG_LEMOTE_A1004] = { + .type = HDA_FIXUP_PINS, + .chained = true, + .chain_id = CXT_FIXUP_INC_MIC_BOOST, + .v.pins = cxt_pincfg_lemote, + }, + [CXT_PINCFG_LEMOTE_A1205] = { + .type = HDA_FIXUP_PINS, + .v.pins = cxt_pincfg_lemote, + }, [CXT_FIXUP_STEREO_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_stereo_dmic, }, + [CXT_FIXUP_INC_MIC_BOOST] = { + .type = HDA_FIXUP_FUNC, + .v.func = cxt5066_increase_mic_boost, + }, }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -4463,6 +4505,8 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004), + SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205), {} }; -- 1.7.7.3

2 2

Re: [alsa-devel] [Bug 46011] Capture of sound from USB connected audio card (UGM96) doesn't work in 3.5.1 - kernel bug generated: scheduling while atomic
by Daniel Mack 17 Aug '12

17 Aug '12

On 15.08.2012 12:40, Takashi Iwai wrote: >>> Could you try the patch below? Clemens, the patch below fixes a "sleeping while atomic" bug in the USB audio driver, but it raises a new "not enough bandwidth" issue, which might even be unreleated. I currently don't see the root cause here, but I might be blind. Do you have any idea? Thanks, Daniel > --- > From: Takashi Iwai <tiwai(a)suse.de> > Subject: [PATCH] ALSA: usb-audio: Fix scheduling-while-atomic bug in PCM > capture stream > > A PCM capture stream on usb-audio causes a scheduling-while-atomic > BUG, as reported in the bugzilla entry below. It's because > snd_usb_endpoint_start() is called at first at trigger START for a > capture stream, and this function contains the left-over EP > deactivation codes. The problem doesn't happen for a playback stream > because the function is called at PCM prepare time, which can sleep. > > This patch fixes the BUG by moving the EP deactivation code into the > PCM prepare callback. > > Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=46011 > Cc: <stable(a)vger.kernel.org> [v3.5+] > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > --- > sound/usb/endpoint.c | 4 ---- > sound/usb/pcm.c | 3 +++ > 2 files changed, 3 insertions(+), 4 deletions(-) > > diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c > index 0f647d2..c411812 100644 > --- a/sound/usb/endpoint.c > +++ b/sound/usb/endpoint.c > @@ -821,10 +821,6 @@ int snd_usb_endpoint_start(struct snd_usb_endpoint *ep) > if (++ep->use_count != 1) > return 0; > > - /* just to be sure */ > - deactivate_urbs(ep, 0, 1); > - wait_clear_urbs(ep); > - > ep->active_mask = 0; > ep->unlink_mask = 0; > ep->phase = 0; > diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c > index a1298f3..62ec808 100644 > --- a/sound/usb/pcm.c > +++ b/sound/usb/pcm.c > @@ -544,6 +544,9 @@ static int snd_usb_pcm_prepare(struct snd_pcm_substream *substream) > subs->last_frame_number = 0; > runtime->delay = 0; > > + /* clear the pending deactivation on the target EPs */ > + deactivate_endpoints(subs); > + > /* for playback, submit the URBs now; otherwise, the first hwptr_done > * updates for all URBs would happen at the same time when starting */ > if (subs->direction == SNDRV_PCM_STREAM_PLAYBACK) >

3 2

[alsa-devel] [PATCH] ASoC: wm9712: Make microphone source selectable
by Mark Brown 16 Aug '12

16 Aug '12

Currently the microphone input source is not selectable as while there is a DAPM widget it's not connected to anything so it won't be properly instantiated. Add something more correct for the input structure to get things going, even though it's not hooked into the rest of the routing map and so won't actually achieve anything except allowing the relevant register bits to be written. Reported-by: Christop Fritz <chf.fritz(a)googlemail.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9712.c | 19 +++++++++++++++++-- 1 files changed, 17 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 00f4436..f35e3bd 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -270,7 +270,7 @@ SOC_DAPM_ENUM("Route", wm9712_enum[9]); /* Mic select */ static const struct snd_kcontrol_new wm9712_mic_src_controls = -SOC_DAPM_ENUM("Route", wm9712_enum[7]); +SOC_DAPM_ENUM("Mic Source Select", wm9712_enum[7]); /* diff select */ static const struct snd_kcontrol_new wm9712_diff_sel_controls = @@ -289,7 +289,9 @@ SND_SOC_DAPM_MUX("Left Capture Select", SND_SOC_NOPM, 0, 0, &wm9712_capture_selectl_controls), SND_SOC_DAPM_MUX("Right Capture Select", SND_SOC_NOPM, 0, 0, &wm9712_capture_selectr_controls), -SND_SOC_DAPM_MUX("Mic Select Source", SND_SOC_NOPM, 0, 0, +SND_SOC_DAPM_MUX("Left Mic Select Source", SND_SOC_NOPM, 0, 0, + &wm9712_mic_src_controls), +SND_SOC_DAPM_MUX("Right Mic Select Source", SND_SOC_NOPM, 0, 0, &wm9712_mic_src_controls), SND_SOC_DAPM_MUX("Differential Source", SND_SOC_NOPM, 0, 0, &wm9712_diff_sel_controls), @@ -317,6 +319,7 @@ SND_SOC_DAPM_PGA("Out 3 PGA", AC97_INT_PAGING, 5, 1, NULL, 0), SND_SOC_DAPM_PGA("Line PGA", AC97_INT_PAGING, 2, 1, NULL, 0), SND_SOC_DAPM_PGA("Phone PGA", AC97_INT_PAGING, 1, 1, NULL, 0), SND_SOC_DAPM_PGA("Mic PGA", AC97_INT_PAGING, 0, 1, NULL, 0), +SND_SOC_DAPM_PGA("Differential Mic", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MICBIAS("Mic Bias", AC97_INT_PAGING, 10, 1), SND_SOC_DAPM_OUTPUT("MONOOUT"), SND_SOC_DAPM_OUTPUT("HPOUTL"), @@ -377,6 +380,18 @@ static const struct snd_soc_dapm_route wm9712_audio_map[] = { {"Mic PGA", NULL, "MIC1"}, {"Mic PGA", NULL, "MIC2"}, + /* microphones */ + {"Differential Mic", NULL, "MIC1"}, + {"Differential Mic", NULL, "MIC2"}, + {"Left Mic Select Source", "Mic 1", "MIC1"}, + {"Left Mic Select Source", "Mic 2", "MIC2"}, + {"Left Mic Select Source", "Stereo", "MIC1"}, + {"Left Mic Select Source", "Differential", "Differential Mic"}, + {"Right Mic Select Source", "Mic 1", "MIC1"}, + {"Right Mic Select Source", "Mic 2", "MIC2"}, + {"Right Mic Select Source", "Stereo", "MIC2"}, + {"Right Mic Select Source", "Differential", "Differential Mic"}, + /* left capture selector */ {"Left Capture Select", "Mic", "MIC1"}, {"Left Capture Select", "Speaker Mixer", "Speaker Mixer"}, -- 1.7.9.1

1 0

[alsa-devel] [PATCH] ASoC: wm5102: Add AEC routing control
by Mark Brown 16 Aug '12

16 Aug '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 34 ++++++++++++++++++++++++++++++++++ 1 file changed, 34 insertions(+) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a5aea03..e2fb07e 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -274,6 +274,27 @@ ARIZONA_MIXER_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE); + +static const char *wm5102_aec_loopback_texts[] = { + "HPOUT1L", "HPOUT1R", "HPOUT2L", "HPOUT2R", "EPOUT", + "SPKOUTL", "SPKOUTR", "SPKDAT1L", "SPKDAT1R", +}; + +static const unsigned int wm5102_aec_loopback_values[] = { + 0, 1, 2, 3, 4, 6, 7, 8, 9, +}; + +static const struct soc_enum wm5102_aec_loopback = + SOC_VALUE_ENUM_SINGLE(ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_SRC_SHIFT, + ARIZONA_AEC_LOOPBACK_SRC_MASK, + ARRAY_SIZE(wm5102_aec_loopback_texts), + wm5102_aec_loopback_texts, + wm5102_aec_loopback_values); + +static const struct snd_kcontrol_new wm5102_aec_loopback_mux = + SOC_DAPM_VALUE_ENUM("AEC Loopback", wm5102_aec_loopback); + static const struct snd_soc_dapm_widget wm5102_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, 0, NULL, 0), @@ -425,6 +446,9 @@ SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0, SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, + ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux), + SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), @@ -520,6 +544,7 @@ SND_SOC_DAPM_OUTPUT("SPKDAT1R"), { name, "Noise Generator", "Noise Generator" }, \ { name, "Tone Generator 1", "Tone Generator 1" }, \ { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "AEC", "AEC Loopback" }, \ { name, "IN1L", "IN1L PGA" }, \ { name, "IN1R", "IN1R PGA" }, \ { name, "IN2L", "IN2L PGA" }, \ @@ -685,21 +710,30 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("ASRC2L", "ASRC2L"), ARIZONA_MIXER_ROUTES("ASRC2R", "ASRC2R"), + { "AEC Loopback", "HPOUT1L", "OUT1L" }, + { "AEC Loopback", "HPOUT1R", "OUT1R" }, { "HPOUT1L", NULL, "OUT1L" }, { "HPOUT1R", NULL, "OUT1R" }, + { "AEC Loopback", "HPOUT2L", "OUT2L" }, + { "AEC Loopback", "HPOUT2R", "OUT2R" }, { "HPOUT2L", NULL, "OUT2L" }, { "HPOUT2R", NULL, "OUT2R" }, + { "AEC Loopback", "EPOUT", "OUT3L" }, { "EPOUTN", NULL, "OUT3L" }, { "EPOUTP", NULL, "OUT3L" }, + { "AEC Loopback", "SPKOUTL", "OUT4L" }, { "SPKOUTLN", NULL, "OUT4L" }, { "SPKOUTLP", NULL, "OUT4L" }, + { "AEC Loopback", "SPKOUTR", "OUT4R" }, { "SPKOUTRN", NULL, "OUT4R" }, { "SPKOUTRP", NULL, "OUT4R" }, + { "AEC Loopback", "SPKDAT1L", "OUT5L" }, + { "AEC Loopback", "SPKDAT1R", "OUT5R" }, { "SPKDAT1L", NULL, "OUT5L" }, { "SPKDAT1R", NULL, "OUT5R" }, }; -- 1.7.10.4

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Re: [alsa-devel] [PATCH 1/5] ARM: tegra: config: enable dmaengine based APB DMA driver
by Stephen Warren 16 Aug '12

16 Aug '12

On 08/16/2012 08:13 AM, Laxman Dewangan wrote: > Enable config for dmaengine based Tegra APB DMA driver and > disable the legacy APB DMA driver (SYSTEM_DMA). Laxman, if I apply this series to next-20120816 (plus a few patches in my local work branch plus the CPU hotplug patches from Joseph, although I suspect none of that matters), then audio playback on Tegra20 is broken; the pitch is far too high. Audio playback on Tegra30 works as expected.

1 0

[alsa-devel] [PATCH] ASoC: wm5102: Remove DRC2
by Mark Brown 16 Aug '12

16 Aug '12

It will be removed from future device revisions. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 16 ---------------- 1 file changed, 16 deletions(-) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a00901c..a5aea03 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -128,13 +128,9 @@ SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE), -ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE), SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), -SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5, - ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA), ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), @@ -236,8 +232,6 @@ ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); -ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE); -ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); @@ -353,10 +347,6 @@ SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, NULL, 0), -SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0, - NULL, 0), -SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0, - NULL, 0), SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, NULL, 0), @@ -470,8 +460,6 @@ ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"), ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), -ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"), -ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"), ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), @@ -557,8 +545,6 @@ SND_SOC_DAPM_OUTPUT("SPKDAT1R"), { name, "EQ4", "EQ4" }, \ { name, "DRC1L", "DRC1L" }, \ { name, "DRC1R", "DRC1R" }, \ - { name, "DRC2L", "DRC2L" }, \ - { name, "DRC2R", "DRC2R" }, \ { name, "LHPF1", "LHPF1" }, \ { name, "LHPF2", "LHPF2" }, \ { name, "LHPF3", "LHPF3" }, \ @@ -688,8 +674,6 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), - ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"), - ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"), ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), -- 1.7.10.4

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[alsa-devel] [PATCH] hda-emu: Add idt_set_gpio verb for stac92xx
by David Henningsson 16 Aug '12

16 Aug '12

According to Takashi, this verb is valid and given by Sigmatel. Also merge two identical arrays and sort the table a little. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hda-spec.c | 66 ++++++++++++++++++++++++++++-------------------------------- 1 file changed, 31 insertions(+), 35 deletions(-) diff --git a/hda-spec.c b/hda-spec.c index dd43ee9..eebaa39 100644 --- a/hda-spec.c +++ b/hda-spec.c @@ -155,15 +155,10 @@ static struct xhda_verb_table idt_92hd8xx_verbs[] = { /* * STAC codecs */ -static struct xhda_verb_table stac927x_verbs[] = { +static struct xhda_verb_table stac92xx_verbs[] = { { 0xfeb, get_cached_verb, "get_analog_loop" }, /* undocumented */ { 0x7eb, set_cached_verb, "set_analog_loop" }, - { } /* terminator */ -}; - -static struct xhda_verb_table stac9205_verbs[] = { - { 0xfe0, get_cached_verb, "get_analog_loop" }, /* undocumented */ - { 0x7e0, set_cached_verb, "set_analog_loop" }, + { 0x7e7, set_cached_verb, "idt_set_gpio" }, { } /* terminator */ }; @@ -271,34 +266,35 @@ static struct verb_ext_list extensions[] = { { .id = 0x111d76e5, .verbs = idt_92hd8xx_verbs }, { .id = 0x111d76e7, .verbs = idt_92hd8xx_verbs }, - { .id = 0x83847618, .verbs = stac927x_verbs }, - { .id = 0x83847619, .verbs = stac927x_verbs }, - { .id = 0x83847616, .verbs = stac927x_verbs }, - { .id = 0x83847617, .verbs = stac927x_verbs }, - { .id = 0x83847614, .verbs = stac927x_verbs }, - { .id = 0x83847615, .verbs = stac927x_verbs }, - { .id = 0x83847620, .verbs = stac927x_verbs }, - { .id = 0x83847621, .verbs = stac927x_verbs }, - { .id = 0x83847622, .verbs = stac927x_verbs }, - { .id = 0x83847623, .verbs = stac927x_verbs }, - { .id = 0x83847624, .verbs = stac927x_verbs }, - { .id = 0x83847625, .verbs = stac927x_verbs }, - { .id = 0x83847626, .verbs = stac927x_verbs }, - { .id = 0x83847627, .verbs = stac927x_verbs }, - { .id = 0x83847628, .verbs = stac927x_verbs }, - { .id = 0x83847629, .verbs = stac927x_verbs }, - { .id = 0x83847645, .verbs = stac927x_verbs }, - { .id = 0x83847646, .verbs = stac927x_verbs }, - - { .id = 0x83847698, .verbs = stac9205_verbs }, - { .id = 0x838476a0, .verbs = stac9205_verbs }, - { .id = 0x838476a1, .verbs = stac9205_verbs }, - { .id = 0x838476a2, .verbs = stac9205_verbs }, - { .id = 0x838476a3, .verbs = stac9205_verbs }, - { .id = 0x838476a4, .verbs = stac9205_verbs }, - { .id = 0x838476a5, .verbs = stac9205_verbs }, - { .id = 0x838476a6, .verbs = stac9205_verbs }, - { .id = 0x838476a7, .verbs = stac9205_verbs }, + { .id = 0x83847614, .verbs = stac92xx_verbs }, + { .id = 0x83847615, .verbs = stac92xx_verbs }, + { .id = 0x83847616, .verbs = stac92xx_verbs }, + { .id = 0x83847617, .verbs = stac92xx_verbs }, + { .id = 0x83847618, .verbs = stac92xx_verbs }, + { .id = 0x83847619, .verbs = stac92xx_verbs }, + { .id = 0x83847620, .verbs = stac92xx_verbs }, + { .id = 0x83847621, .verbs = stac92xx_verbs }, + { .id = 0x83847622, .verbs = stac92xx_verbs }, + { .id = 0x83847623, .verbs = stac92xx_verbs }, + { .id = 0x83847624, .verbs = stac92xx_verbs }, + { .id = 0x83847625, .verbs = stac92xx_verbs }, + { .id = 0x83847626, .verbs = stac92xx_verbs }, + { .id = 0x83847627, .verbs = stac92xx_verbs }, + { .id = 0x83847628, .verbs = stac92xx_verbs }, + { .id = 0x83847629, .verbs = stac92xx_verbs }, + { .id = 0x83847645, .verbs = stac92xx_verbs }, + { .id = 0x83847646, .verbs = stac92xx_verbs }, + + { .id = 0x83847690, .verbs = stac92xx_verbs }, + { .id = 0x83847698, .verbs = stac92xx_verbs }, + { .id = 0x838476a0, .verbs = stac92xx_verbs }, + { .id = 0x838476a1, .verbs = stac92xx_verbs }, + { .id = 0x838476a2, .verbs = stac92xx_verbs }, + { .id = 0x838476a3, .verbs = stac92xx_verbs }, + { .id = 0x838476a4, .verbs = stac92xx_verbs }, + { .id = 0x838476a5, .verbs = stac92xx_verbs }, + { .id = 0x838476a6, .verbs = stac92xx_verbs }, + { .id = 0x838476a7, .verbs = stac92xx_verbs }, { .id = 0x11061708, .verbs = vt1708b_verbs }, { .id = 0x11061709, .verbs = vt1708b_verbs }, -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ASoC: wm8994: Add bytes controls for DRC
by Mark Brown 16 Aug '12

16 Aug '12

If DRC coefficients are not configured via platform data then add bytes controls for them instead so they can be configured by applications. This is the normal means of controlling things like this for newer systems, we maintain compatibility with platform data to avoid disruption to existing systems. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 25 +++++++++++++++++++++---- 1 file changed, 21 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 7755ac9..2d60ed3 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -759,6 +759,18 @@ SOC_SINGLE_TLV("AIF2 EQ5 Volume", WM8994_AIF2_EQ_GAINS_2, 6, 31, 0, eq_tlv), }; +static const struct snd_kcontrol_new wm8994_drc_controls[] = { +SND_SOC_BYTES_MASK("AIF1.1 DRC", WM8994_AIF1_DRC1_1, 5, + WM8994_AIF1DAC1_DRC_ENA | WM8994_AIF1ADC1L_DRC_ENA | + WM8994_AIF1ADC1R_DRC_ENA), +SND_SOC_BYTES_MASK("AIF1.2 DRC", WM8994_AIF1_DRC2_1, 5, + WM8994_AIF1DAC2_DRC_ENA | WM8994_AIF1ADC2L_DRC_ENA | + WM8994_AIF1ADC2R_DRC_ENA), +SND_SOC_BYTES_MASK("AIF2 DRC", WM8994_AIF2_DRC_1, 5, + WM8994_AIF2DAC_DRC_ENA | WM8994_AIF2ADCL_DRC_ENA | + WM8994_AIF2ADCR_DRC_ENA), +}; + static const char *wm8958_ng_text[] = { "30ms", "125ms", "250ms", "500ms", }; @@ -3254,14 +3266,19 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) ret = snd_soc_add_codec_controls(wm8994->hubs.codec, controls, ARRAY_SIZE(controls)); - if (ret != 0) - dev_err(wm8994->hubs.codec->dev, - "Failed to add DRC mode controls: %d\n", ret); - for (i = 0; i < WM8994_NUM_DRC; i++) wm8994_set_drc(codec, i); + } else { + ret = snd_soc_add_codec_controls(wm8994->hubs.codec, + wm8994_drc_controls, + ARRAY_SIZE(wm8994_drc_controls)); } + if (ret != 0) + dev_err(wm8994->hubs.codec->dev, + "Failed to add DRC mode controls: %d\n", ret); + + dev_dbg(codec->dev, "%d ReTune Mobile configurations\n", pdata->num_retune_mobile_cfgs); -- 1.7.10.4

1 0

[alsa-devel] [PATCH v2] ALSA: hda - Don't send invalid volume knob command on IDT 92hd75bxx
by David Henningsson 16 Aug '12

16 Aug '12

Instead of blindly initializing a volume knob widget, first check that there actually is a volume knob widget. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 94040cc..ea5775a 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4272,7 +4272,8 @@ static int stac92xx_init(struct hda_codec *codec) unsigned int gpio; int i; - snd_hda_sequence_write(codec, spec->init); + if (spec->init) + snd_hda_sequence_write(codec, spec->init); /* power down adcs initially */ if (spec->powerdown_adcs) @@ -5748,7 +5749,6 @@ again: /* fallthru */ case 0x111d76b4: /* 6 Port without Analog Mixer */ case 0x111d76b5: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5773,7 +5773,6 @@ again: spec->stream_delay = 40; /* 40 milliseconds */ /* disable VSW */ - spec->init = stac92hd71bxx_core_init; unmute_init++; snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); @@ -5788,7 +5787,6 @@ again: /* fallthru */ default: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5796,6 +5794,9 @@ again: break; } + if (get_wcaps_type(get_wcaps(codec, 0x28)) == AC_WID_VOL_KNB) + spec->init = stac92hd71bxx_core_init; + if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) snd_hda_sequence_write_cache(codec, unmute_init); -- 1.7.9.5

2 1

[alsa-devel] [PATCH] hda-emu: Fixup ALC268 input amp on Beep node
by David Henningsson 16 Aug '12

16 Aug '12

According to the ALC268 datasheet, there is actually an input amplifier on node 0x1d, even though this is not exposed in the codec proc file. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- hda-spec.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) Relevant alsa info: alc268-toshiba-tecra-s11-m11--nvidia--ccert-201002-5346 diff --git a/hda-spec.c b/hda-spec.c index f1b03b4..dd43ee9 100644 --- a/hda-spec.c +++ b/hda-spec.c @@ -386,6 +386,20 @@ static void fixup_via_mixer(struct xhda_codec *codec) node->node[node->num_nodes++] = 0x08; } +static void fixup_alc268_beep(struct xhda_codec *codec) +{ + struct xhda_node *node = find_node(codec, 0x1d); + if (!node) + return; + node->wcaps |= (1 << 1); /* AMP-in present */ + + node->amp_in_caps.ofs = 0x0c; + node->amp_in_caps.nsteps = 0x0c; + node->amp_in_caps.stepsize = 0x07; + node->amp_in_caps.mute = 0; + node->amp_in_caps.override = 1; +} + struct fixup_list { unsigned int vendor_id; void (*func)(struct xhda_codec *); @@ -397,6 +411,7 @@ static struct fixup_list fixups[] = { { 0x11060438, fixup_via_mixer }, { 0x11064438, fixup_via_mixer }, { 0x11060448, fixup_via_mixer }, + { 0x10ec0268, fixup_alc268_beep }, { } }; -- 1.7.9.5

2 1

[alsa-devel] command 0x7e7 sent on STAC9200 / STAC9228
by David Henningsson 16 Aug '12

16 Aug '12

Hi Takashi, While working with fixing the stuff that my automated test tool has discovered, here's one I'm not sure what to do with. If you run hda-emu on any of these three: codecs/canonical/stac9200-dell-inspiron-6400-ccert-200705-83 codecs/canonical/stac9200-dell-inspiron-6400-ccert-200712-207 codecs/canonical/stac9228-dell-inspiron-1420-ccert-200708-95 Hda-emu prints out: invalid command: NID=0x1, verb=0x7e7, parm=0x0 In this case, I'm not sure if the command is invalid (and thus should be removed), or if it's a secret IDT verb that we need to set. From patch_sigmatel.c: /* Configure GPIOx as CMOS */ snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0); But in hda-spec.c this verb is only in the verb table of IDT codecs, not STAC codecs. Can you offer some enlightement? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

2 1

[alsa-devel] [PATCH] ALSA: hda - Don't set invalid volume knob command on IDT 92hd75bxx
by David Henningsson 16 Aug '12

16 Aug '12

Instead of blindly initializing a volume knob widget, first check that there actually is a volume knob widget. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) Relevant alsa info: 92hd75b3x5-hp-probook-6550b-ccert-201101-6954 diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 94040cc..1018324 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4272,7 +4272,8 @@ static int stac92xx_init(struct hda_codec *codec) unsigned int gpio; int i; - snd_hda_sequence_write(codec, spec->init); + if (spec->init) + snd_hda_sequence_write(codec, spec->init); /* power down adcs initially */ if (spec->powerdown_adcs) @@ -5748,7 +5749,6 @@ again: /* fallthru */ case 0x111d76b4: /* 6 Port without Analog Mixer */ case 0x111d76b5: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5773,7 +5773,6 @@ again: spec->stream_delay = 40; /* 40 milliseconds */ /* disable VSW */ - spec->init = stac92hd71bxx_core_init; unmute_init++; snd_hda_codec_set_pincfg(codec, 0x0f, 0x40f000f0); snd_hda_codec_set_pincfg(codec, 0x19, 0x40f000f3); @@ -5788,7 +5787,6 @@ again: /* fallthru */ default: - spec->init = stac92hd71bxx_core_init; codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs; spec->num_dmics = stac92xx_connected_ports(codec, stac92hd71bxx_dmic_nids, @@ -5796,6 +5794,10 @@ again: break; } + if (codec->start_nid + codec->num_nodes >= 0x28 && + get_wcaps_type(get_wcaps(codec, 0x28)) == AC_WID_VOL_KNB) + spec->init = stac92hd71bxx_core_init; + if (get_wcaps(codec, 0xa) & AC_WCAP_IN_AMP) snd_hda_sequence_write_cache(codec, unmute_init); -- 1.7.9.5

2 1

Re: [alsa-devel] Question about Conversion to S24_LE from S24_3LE
by Takashi Iwai 16 Aug '12

16 Aug '12

At Thu, 16 Aug 2012 16:55:55 +0900 (JST), etsuko suzuki wrote: > > Thank you for your kindness reply. > > > Did you subscribe to the ML? > > Yes I did. > I subscribed ML and I accepted the mail 'Welcome to the "Alsa-devel" mailing l > ist'. > And I can receive mails of this ML. > But I can't post it to ML. > (The mails were NOT rejected.) > > I had sent e-mail to 'alsa-devel-owner(a)alsa-project.org' several times. > But there was no answer. > (The mails were NOT rejected.) > > Is something done else? > I am sorry about like this question... Hm, I guess your message didn't reach properly, or was filtered by some spam filter. Better to contact with Jaroslav (Cc'ed), who maintains alsa-devel ML. Takashi > > Regards. > > Etsuko >

2 1

[alsa-devel] [PATCH] hda-emu: add python compiled files to gitignore
by David Henningsson 16 Aug '12

16 Aug '12

Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- .gitignore | 1 + 1 file changed, 1 insertion(+) diff --git a/.gitignore b/.gitignore index 2281a23..5ab6918 100644 --- a/.gitignore +++ b/.gitignore @@ -1,6 +1,7 @@ *~ *.o *.a +*.pyc hda-emu hda-decode-verb hda-encode-verb -- 1.7.9.5

2 1

Re: [alsa-devel] Question about Conversion to S24_LE from S24_3LE
by Takashi Iwai 16 Aug '12

16 Aug '12

At Thu, 16 Aug 2012 09:37:18 +0900 (JST), etsuko suzuki wrote: > > Hi. > > I am Etsuko. > Kaz is my coworker. > > This was my question, so I had to post this mail myself. > However my mail was not posted from my E-mail address. Did you subscribe to the ML? Otherwise a post would need a manual approval and it misses sometimes timeframe that the ML server keeps a pending post. > Therefore he had posted this instead of me. > > I am challenging once again using the different mail address. > > ************************************************************* > > >It depends on what format is used in the original PCM stream. > > I recorded this file from 16bits wav file. OK, it's clear now. > >It'd make sense to have 00 in LSB if it was converted from 16 to 24bit. > > Does this mean the following? > > ex.) > > > 0000030 ffffc300 ffffc600 ffffca00 ffffd300 > ++====** > > ++ : padding > == : PCM data > ** : This is added when it was converted from 16 to 24bit. > Yes. When the original source is 16bit, ALSA-lib format conversion just shifts the bits. It's suboptimal, but fast. That's the reason of the left 00. > I'd like to ask one more. > I read the following, in order to know about wav file. > > http://msdn.microsoft.com/en-us/windows/hardware/gg463006.aspx > > In this, I found the statement 'left-alinged'. > (In 'Details about wValidBitsPerSample') > > I understood 'left-alinged' was 'LSB padding'. > So I thought S24_LE was LSB padding. No. S24_LE has the MSB padding of 8 bits in 32 bit container. > Does the format in the ALSA driver differ from the format in wav file? You need to use S32_LE instead. HTH, Takashi

1 0

[alsa-devel] [PATCH v2 0/4] ASoC machine driver for simple SoC with twl4030
by Peter Ujfalusi 15 Aug '12

15 Aug '12

Hello, Changes since v1: - Card name for cm-t35 is "cm-t3x" - Added Ack from Tony for the OMAP patch - Codec sysclock no longer needed by the machine driver The codec itself must know the mclk since it needs it for initial configuration and also in case only the vibra functionality is used. The codec driver ignores the machine driver provided clock, it always uses the mclk freq it got from the MFD core driver. Intro mail from v1: Create new unified ASoC machine driver (with DT support) for boards using simple configuration for audio and using the twl4030 codec. This driver will replace the following machine drivers (since they are basically the same): igep0020, omap3beagle, omap3evm and overo. This series will switch the boards I can identify to use the new unified ASoC machine driver. Changes under arch/arm/mach-omap2/ are trivial, I don't expect merge issues later. If it is OK for the maintainers I would like this to go via audio tree (with the McBSP and the twl4030-audio stack DT support). The series has been tested on BeagleBoard (with the McBSP and twl4030-audio DT support). Regards, Peter --- Peter Ujfalusi (4): ASoC: omap-twl4030: Simple machine driver for TI SoC with twl4030 codec ARM: OMAP: twl-common: Add helper function to register the omap-twl4030 audio driver ARM: OMAP3: Switch to use the unified audio driver (omap-twl4030) for selected boards ASoC: Remove obsolete OMAP3 machine drivers .../devicetree/bindings/sound/omap-twl4030.txt | 17 ++ arch/arm/mach-omap2/board-cm-t35.c | 1 + arch/arm/mach-omap2/board-devkit8000.c | 1 + arch/arm/mach-omap2/board-igep0020.c | 1 + arch/arm/mach-omap2/board-omap3beagle.c | 1 + arch/arm/mach-omap2/board-omap3evm.c | 1 + arch/arm/mach-omap2/board-overo.c | 1 + arch/arm/mach-omap2/twl-common.c | 27 +++ arch/arm/mach-omap2/twl-common.h | 2 + include/linux/platform_data/omap-twl4030.h | 32 ++++ sound/soc/omap/Kconfig | 48 ++---- sound/soc/omap/Makefile | 10 +- sound/soc/omap/igep0020.c | 120 ------------- sound/soc/omap/omap-twl4030.c | 188 ++++++++++++++++++++ sound/soc/omap/omap3beagle.c | 150 ---------------- sound/soc/omap/omap3evm.c | 118 ------------ sound/soc/omap/overo.c | 122 ------------- 17 files changed, 287 insertions(+), 553 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/omap-twl4030.txt create mode 100644 include/linux/platform_data/omap-twl4030.h delete mode 100644 sound/soc/omap/igep0020.c create mode 100644 sound/soc/omap/omap-twl4030.c delete mode 100644 sound/soc/omap/omap3beagle.c delete mode 100644 sound/soc/omap/omap3evm.c delete mode 100644 sound/soc/omap/overo.c -- 1.7.8.6

2 5

[alsa-devel] [PATCH 03/22] ASoC: ab8500: Inform SoC Core that we have our own I/O arrangements
by Lee Jones 15 Aug '12

15 Aug '12

If codec->control_data is not populated SoC Core assumes we want to use regmap, which fails catastrophically, as we don't have one: Unable to handle kernel NULL pointer dereference at virtual address 00000080 pgd = c0004000 [00000080] *pgd=00000000 Internal error: Oops: 17 [#1] PREEMPT SMP ARM Modules linked in: CPU: 1 Not tainted (3.5.0-rc6-00884-g0b2419e-dirty #130) PC is at regmap_read+0x10/0x5c LR is at hw_read+0x80/0x90 pc : [<c01a91b8>] lr : [<c0216804>] psr: 60000013 CC: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/codecs/ab8500-codec.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 3c79592..23b4018 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2406,6 +2406,10 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) /* Setup AB8500 according to board-settings */ pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent); + + /* Inform SoC Core that we have our own I/O arrangements. */ + codec->control_data = (void *)true; + status = ab8500_audio_setup_mics(codec, &pdata->codec->amics); if (status < 0) { pr_err("%s: Failed to setup mics (%d)!\n", __func__, status); -- 1.7.9.5

4 3

[alsa-devel] [RFC PATCH 1/1] ALSA: hda - add a timer to find and suspend idle codecs
by mengdong.lin＠intel.com 15 Aug '12

15 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> (sorry, this patch is resent with title updated) This patch sets up a timer to monitor changes of the parameter 'power_save' and suspend any idle codecs if power save is re-enabled by the user. A codec won't be suspended if it becomes idle when power save is disabled. And it will remain unsuspended if power save is enabled later but no further user operations comes. So a timer is used here to find and suspend such idle codecs when power save is re-enabled. Now the timer has a fixed interval of 1 second. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_codec.c | 109 +++++++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_codec.h | 9 ++++ sound/pci/hda/hda_intel.c | 2 +- 3 files changed, 119 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 629131a..6bd8873 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -98,6 +98,8 @@ EXPORT_SYMBOL_HDA(snd_hda_delete_codec_preset); static void hda_power_work(struct work_struct *work); static void hda_keep_power_on(struct hda_codec *codec); #define hda_codec_is_power_on(codec) ((codec)->power_on) +#define POWER_SAVE_MONITOR_INTERVAL 1000 +static void power_save_monitor_timeout(unsigned long data); #else static inline void hda_keep_power_on(struct hda_codec *codec) {} #define hda_codec_is_power_on(codec) 1 @@ -685,6 +687,9 @@ static int snd_hda_bus_free(struct hda_bus *bus) if (!bus) return 0; + + snd_hda_power_save_monitor_stop(bus); + if (bus->workq) flush_workqueue(bus->workq); if (bus->unsol) @@ -4558,6 +4563,106 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, return 0; } EXPORT_SYMBOL_HDA(snd_hda_check_amp_list_power); + +#define BUS_POWER_SAVE(bus) \ + ((bus)->power_save && *(bus)->power_save ? 1 : 0) + +static void power_save_monitor_timeout(unsigned long data) +{ + int ret; + struct hda_bus *bus = (struct hda_bus *)data; + struct list_head *codec_list = &bus->codec_list; + struct hda_codec *codec, *next; + + if (BUS_POWER_SAVE(bus) == bus->power_save_enabled) + goto out; + + bus->power_save_enabled = BUS_POWER_SAVE(bus); + if (!bus->power_save_enabled) { + snd_printd("hda-bus: power save disabled\n"); + goto out; + } + + snd_printd("hda-bus: power save enabled\n"); + list_for_each_entry_safe(codec, next, codec_list, list) { + spin_lock(&codec->power_lock); + if (codec->power_on + && !codec->power_count + && !codec->power_transition) { + snd_printd("hda codec %d idle\n", codec->addr); + codec->power_transition = -1; /* avoid reentrance */ + queue_delayed_work(codec->bus->workq, + &codec->power_work, + msecs_to_jiffies(power_save(codec) * 1000)); + } + spin_unlock(&codec->power_lock); + } + +out: + ret = mod_timer(&bus->power_save_timer, + jiffies + \ + msecs_to_jiffies(POWER_SAVE_MONITOR_INTERVAL)); + if (ret) + snd_printd("hda-bus: error in mod_timer\n"); + + return; +} + +/** + * snd_hda_power_save_monitor_start - Setup a timer to monitor the changes of + * power_save parameter. + * @bus: HD-audio bus + * + * Setup a timer to monitor whether parameter power_save is changed by the user. + * A codec won't be suspended if it becomes idle when power save is disabled, + * and can remain unsuspended after power save is enabled later. + * This monitor will find and suspened such idle codecs when power save is + * re-enabled. + */ +void snd_hda_power_save_monitor_start(struct hda_bus *bus) +{ + int ret; + + if (bus->power_save_timer_inited) { + snd_printd("hda-bus: power save timer already inited\n"); + return; + } + + setup_timer(&bus->power_save_timer, + power_save_monitor_timeout, (unsigned long)bus); + bus->power_save_timer_inited = 1; + + snd_printd("hda-bus: start power save monitor\n"); + ret = mod_timer(&bus->power_save_timer, + jiffies + \ + msecs_to_jiffies(POWER_SAVE_MONITOR_INTERVAL)); + if (ret) + snd_printd("hda-bus: error in mod_timer\n"); + + return; +} +EXPORT_SYMBOL_HDA(snd_hda_power_save_monitor_start); + +/** + * snd_hda_power_save_monitor_stop - Delete the timer to monitor the changes of + * power_save parameter. + * @bus: HD-audio bus + */ +void snd_hda_power_save_monitor_stop(struct hda_bus *bus) +{ + int ret; + + if (bus->power_save_timer_inited) { + snd_printd("hda-bus: stop power save monitor\n"); + ret = del_timer_sync(&bus->power_save_timer); + if (ret < 0) + snd_printd("hda-bus: error in del_timer_sync\n"); + bus->power_save_timer_inited = 0; + } + + return; +} +EXPORT_SYMBOL_HDA(snd_hda_power_save_monitor_stop); #endif /* @@ -5045,6 +5150,8 @@ int snd_hda_suspend(struct hda_bus *bus) { struct hda_codec *codec; + snd_hda_power_save_monitor_stop(bus); + list_for_each_entry(codec, &bus->codec_list, list) { if (hda_codec_is_power_on(codec)) hda_call_codec_suspend(codec); @@ -5069,6 +5176,8 @@ int snd_hda_resume(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { hda_call_codec_resume(codec); } + + snd_hda_power_save_monitor_start(bus); return 0; } EXPORT_SYMBOL_HDA(snd_hda_resume); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c422d33..8b88ec5 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -653,6 +653,8 @@ struct hda_bus { struct hda_bus_unsolicited *unsol; char workq_name[16]; struct workqueue_struct *workq; /* common workqueue for codecs */ + /* timer to monitor changes of power_save parameter */ + struct timer_list power_save_timer; /* assigned PCMs */ DECLARE_BITMAP(pcm_dev_bits, SNDRV_PCM_DEVICES); @@ -667,6 +669,9 @@ struct hda_bus { unsigned int response_reset:1; /* controller was reset */ unsigned int in_reset:1; /* during reset operation */ unsigned int power_keep_link_on:1; /* don't power off HDA link */ + /* power save flags */ + unsigned int power_save_enabled:1; /* enabled at last check */ + unsigned int power_save_timer_inited:1; /* timer is initialized */ }; /* @@ -1062,10 +1067,14 @@ void snd_hda_power_up(struct hda_codec *codec); void snd_hda_power_up_d3wait(struct hda_codec *codec); void snd_hda_power_down(struct hda_codec *codec); void snd_hda_update_power_acct(struct hda_codec *codec); +extern void snd_hda_power_save_monitor_start(struct hda_bus *bus); +extern void snd_hda_power_save_monitor_stop(struct hda_bus *bus); #else static inline void snd_hda_power_up(struct hda_codec *codec) {} static inline void snd_hda_power_up_d3wait(struct hda_codec *codec) {} static inline void snd_hda_power_down(struct hda_codec *codec) {} +#define snd_hda_power_save_monitor_start(bus) {} +#define snd_hda_power_save_monitor_stop(bus) {} #endif #ifdef CONFIG_SND_HDA_PATCH_LOADER diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c8aced1..fe015d0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3236,7 +3236,7 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) chip->running = 1; power_down_all_codecs(chip); azx_notifier_register(chip); - + snd_hda_power_save_monitor_start(chip->bus); return 0; out_free: -- 1.7.9.5

5 12

[alsa-devel] [RFC PATCH 1/1] ALSA: hda - add a timer to find and suspend idle codecs
by mengdong.lin＠intel.com 14 Aug '12

14 Aug '12

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch sets up a timer to monitor changes of the parameter 'power_save' and suspend any idle codecs if power save is re-enabled by the user. A codec won't be suspended if it becomes idle when power save is disabled. And it will remain unsuspended if power save is enabled later but no further user operations comes. So a timer is used here to find and suspend such idle codecs when power save is re-enabled. Now the timer has a fixed interval of 1 second. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_codec.c | 109 +++++++++++++++++++++++++++++++++++++++++++++ sound/pci/hda/hda_codec.h | 9 ++++ sound/pci/hda/hda_intel.c | 2 +- 3 files changed, 119 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 629131a..6bd8873 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -98,6 +98,8 @@ EXPORT_SYMBOL_HDA(snd_hda_delete_codec_preset); static void hda_power_work(struct work_struct *work); static void hda_keep_power_on(struct hda_codec *codec); #define hda_codec_is_power_on(codec) ((codec)->power_on) +#define POWER_SAVE_MONITOR_INTERVAL 1000 +static void power_save_monitor_timeout(unsigned long data); #else static inline void hda_keep_power_on(struct hda_codec *codec) {} #define hda_codec_is_power_on(codec) 1 @@ -685,6 +687,9 @@ static int snd_hda_bus_free(struct hda_bus *bus) if (!bus) return 0; + + snd_hda_power_save_monitor_stop(bus); + if (bus->workq) flush_workqueue(bus->workq); if (bus->unsol) @@ -4558,6 +4563,106 @@ int snd_hda_check_amp_list_power(struct hda_codec *codec, return 0; } EXPORT_SYMBOL_HDA(snd_hda_check_amp_list_power); + +#define BUS_POWER_SAVE(bus) \ + ((bus)->power_save && *(bus)->power_save ? 1 : 0) + +static void power_save_monitor_timeout(unsigned long data) +{ + int ret; + struct hda_bus *bus = (struct hda_bus *)data; + struct list_head *codec_list = &bus->codec_list; + struct hda_codec *codec, *next; + + if (BUS_POWER_SAVE(bus) == bus->power_save_enabled) + goto out; + + bus->power_save_enabled = BUS_POWER_SAVE(bus); + if (!bus->power_save_enabled) { + snd_printd("hda-bus: power save disabled\n"); + goto out; + } + + snd_printd("hda-bus: power save enabled\n"); + list_for_each_entry_safe(codec, next, codec_list, list) { + spin_lock(&codec->power_lock); + if (codec->power_on + && !codec->power_count + && !codec->power_transition) { + snd_printd("hda codec %d idle\n", codec->addr); + codec->power_transition = -1; /* avoid reentrance */ + queue_delayed_work(codec->bus->workq, + &codec->power_work, + msecs_to_jiffies(power_save(codec) * 1000)); + } + spin_unlock(&codec->power_lock); + } + +out: + ret = mod_timer(&bus->power_save_timer, + jiffies + \ + msecs_to_jiffies(POWER_SAVE_MONITOR_INTERVAL)); + if (ret) + snd_printd("hda-bus: error in mod_timer\n"); + + return; +} + +/** + * snd_hda_power_save_monitor_start - Setup a timer to monitor the changes of + * power_save parameter. + * @bus: HD-audio bus + * + * Setup a timer to monitor whether parameter power_save is changed by the user. + * A codec won't be suspended if it becomes idle when power save is disabled, + * and can remain unsuspended after power save is enabled later. + * This monitor will find and suspened such idle codecs when power save is + * re-enabled. + */ +void snd_hda_power_save_monitor_start(struct hda_bus *bus) +{ + int ret; + + if (bus->power_save_timer_inited) { + snd_printd("hda-bus: power save timer already inited\n"); + return; + } + + setup_timer(&bus->power_save_timer, + power_save_monitor_timeout, (unsigned long)bus); + bus->power_save_timer_inited = 1; + + snd_printd("hda-bus: start power save monitor\n"); + ret = mod_timer(&bus->power_save_timer, + jiffies + \ + msecs_to_jiffies(POWER_SAVE_MONITOR_INTERVAL)); + if (ret) + snd_printd("hda-bus: error in mod_timer\n"); + + return; +} +EXPORT_SYMBOL_HDA(snd_hda_power_save_monitor_start); + +/** + * snd_hda_power_save_monitor_stop - Delete the timer to monitor the changes of + * power_save parameter. + * @bus: HD-audio bus + */ +void snd_hda_power_save_monitor_stop(struct hda_bus *bus) +{ + int ret; + + if (bus->power_save_timer_inited) { + snd_printd("hda-bus: stop power save monitor\n"); + ret = del_timer_sync(&bus->power_save_timer); + if (ret < 0) + snd_printd("hda-bus: error in del_timer_sync\n"); + bus->power_save_timer_inited = 0; + } + + return; +} +EXPORT_SYMBOL_HDA(snd_hda_power_save_monitor_stop); #endif /* @@ -5045,6 +5150,8 @@ int snd_hda_suspend(struct hda_bus *bus) { struct hda_codec *codec; + snd_hda_power_save_monitor_stop(bus); + list_for_each_entry(codec, &bus->codec_list, list) { if (hda_codec_is_power_on(codec)) hda_call_codec_suspend(codec); @@ -5069,6 +5176,8 @@ int snd_hda_resume(struct hda_bus *bus) list_for_each_entry(codec, &bus->codec_list, list) { hda_call_codec_resume(codec); } + + snd_hda_power_save_monitor_start(bus); return 0; } EXPORT_SYMBOL_HDA(snd_hda_resume); diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c422d33..8b88ec5 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -653,6 +653,8 @@ struct hda_bus { struct hda_bus_unsolicited *unsol; char workq_name[16]; struct workqueue_struct *workq; /* common workqueue for codecs */ + /* timer to monitor changes of power_save parameter */ + struct timer_list power_save_timer; /* assigned PCMs */ DECLARE_BITMAP(pcm_dev_bits, SNDRV_PCM_DEVICES); @@ -667,6 +669,9 @@ struct hda_bus { unsigned int response_reset:1; /* controller was reset */ unsigned int in_reset:1; /* during reset operation */ unsigned int power_keep_link_on:1; /* don't power off HDA link */ + /* power save flags */ + unsigned int power_save_enabled:1; /* enabled at last check */ + unsigned int power_save_timer_inited:1; /* timer is initialized */ }; /* @@ -1062,10 +1067,14 @@ void snd_hda_power_up(struct hda_codec *codec); void snd_hda_power_up_d3wait(struct hda_codec *codec); void snd_hda_power_down(struct hda_codec *codec); void snd_hda_update_power_acct(struct hda_codec *codec); +extern void snd_hda_power_save_monitor_start(struct hda_bus *bus); +extern void snd_hda_power_save_monitor_stop(struct hda_bus *bus); #else static inline void snd_hda_power_up(struct hda_codec *codec) {} static inline void snd_hda_power_up_d3wait(struct hda_codec *codec) {} static inline void snd_hda_power_down(struct hda_codec *codec) {} +#define snd_hda_power_save_monitor_start(bus) {} +#define snd_hda_power_save_monitor_stop(bus) {} #endif #ifdef CONFIG_SND_HDA_PATCH_LOADER diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c8aced1..fe015d0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3236,7 +3236,7 @@ static int DELAYED_INIT_MARK azx_probe_continue(struct azx *chip) chip->running = 1; power_down_all_codecs(chip); azx_notifier_register(chip); - + snd_hda_power_save_monitor_start(chip->bus); return 0; out_free: -- 1.7.9.5

1 0

[alsa-devel] [PATCH v2 00/14] MFD/ASoC/Input: twl4030-audio submodule DT support
by Peter Ujfalusi 14 Aug '12

14 Aug '12

Hello, Changes since v1: - Get the MCLK frequencey from twl-core driver (via new API) - hs_extmute_disable_level parameter has been removed - empty of_find_node_by_name() in of.h for !CONFIG_OF builds Mark: the extmute GPIO handling (when it is used) remained in the codec driver for now. I can think more on how to add support for this type of mute control. If you can point me to other codec needing this it would help on designing something which is common enough for other users. As for now I have a separate series to add GPIO controlled output amps (hp/spk). I will send that to review soon. Introl mail from v1: The following series adds DT support for the twl4030 audio submodule which provides audio codec and vibra functionality. The MFD core driver is probed via DT, it will create the needed child devices based on the provided information in the DT blob. Child drivers (vibra, ASoC codec) will parse the core's node if needed to get the needed parameters for their configuration. In the ASoC codec driver the hs_extmute callback (which was used to toggle a GPIO line) has been removed. The codec driver will receive the GPIO number (if it is needed on the platform). If the series is OK (and no objections from the maintainers), it would be good if this can go via audio. Changed files are well contained within the twl4030-audio stack so I do not expect merge issues later. The series has been tested on BeagleBoard (with the McBSP DT series, and with the upcoming DT audio support for BeagleBoard). Regards, Peter --- Peter Ujfalusi (14): MFD: twl4030-audio: Clean up MODULE_* and platform_driver part MFD: twl4030-audio: Convert to use devm_kzalloc MFD: twl4030-audio: Rearange and clean-up the probe function MFD: twl-core: Add API to query the HFCLK rate MFD: twl4030-audio: Get audio MCLK via twl-core API instead of pdata dt: Add empty of_find_node_by_name() function MFD: twl4030-audio: Add DT support Input: twl4030-vibra: Support for DT booted kernel ASoC: twl4030: Move hs_extmute GPIO handling to driver ARM: OMAP/ASoC: Zoom2: Let the codec to handle the hs_extmute GPIO ASoC/MFD: twl4030: Remove set_hs_extmute callback from platform data ASoC: twl4030: Convert to use devm_kzalloc ASoC: twl4030: Add pointer to pdata within the private data ASoC: twl4030: Support for DT booted kernel .../devicetree/bindings/mfd/twl4030-audio.txt | 46 +++++++++ arch/arm/mach-omap2/board-zoom-peripherals.c | 9 +-- arch/arm/mach-omap2/include/mach/board-zoom.h | 2 - drivers/input/misc/twl4030-vibra.c | 20 ++++- drivers/mfd/twl-core.c | 32 ++++++ drivers/mfd/twl4030-audio.c | 107 +++++++++++++------- include/linux/i2c/twl.h | 3 +- include/linux/of.h | 6 + sound/soc/codecs/twl4030.c | 103 +++++++++++++++---- sound/soc/omap/zoom2.c | 4 - 10 files changed, 261 insertions(+), 71 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/twl4030-audio.txt -- 1.7.8.6

1 14

[alsa-devel] [PATCH 22/22] Documentation: Add the AB8500 CODEC device to the MFD AB8500 doc
by Lee Jones 14 Aug '12

14 Aug '12

Here we document the AB8500 Audio CODEC in the existing MFD AB8500 document. CC: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- Documentation/devicetree/bindings/mfd/ab8500.txt | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/Documentation/devicetree/bindings/mfd/ab8500.txt b/Documentation/devicetree/bindings/mfd/ab8500.txt index 69e757a..ce83c8d 100644 --- a/Documentation/devicetree/bindings/mfd/ab8500.txt +++ b/Documentation/devicetree/bindings/mfd/ab8500.txt @@ -23,6 +23,7 @@ Device IRQ Names Supply Names Description ab8500-bm : : : Battery Manager ab8500-btemp : : : Battery Temperature ab8500-charger : : : Battery Charger +ab8500-codec : : : Audio Codec ab8500-fg : : : Fuel Gauge ab8500-gpadc : HW_CONV_END : vddadc : Analogue to Digital Converter SW_CONV_END : : @@ -52,6 +53,14 @@ Optional child device properties: supplied in the interrupts property - <supply_name>-supply : contains a phandle to the regulator supply node in Device Tree +Non-standard child device properties: + - Audio CODEC: + - stericsson,amic[1|2]-type-single-ended : Single-ended Analoge Mic (default: differential) + - stericsson,amic1a-bias-vamic2 : Analoge Mic wishes to use a non-standard Vamic + - stericsson,amic1b-bias-vamic2 : Analoge Mic wishes to use a non-standard Vamic + - stericsson,amic2-bias-vamic1 : Analoge Mic wishes to use a non-standard Vamic + - stericsson,earpeice-cmv : Earpeice voltage (only: 950 | 1100 | 1270 | 1580) + ab8500@5 { compatible = "stericsson,ab8500"; reg = <5>; /* mailbox 5 is i2c */ @@ -110,6 +119,12 @@ ab8500@5 { compatible = "stericsson,ab8500-pwm"; }; + codec: ab8500-codec { + compatible = "stericsson,ab8500-codec"; + + stericsson,earpeice-cmv = <950>; /* Units in mV. */ + }; + ab8500-regulators { compatible = "stericsson,ab8500-regulator"; -- 1.7.9.5

2 1

[alsa-devel] Automated test suite for HDA codecs, now up and running for daily builds
by David Henningsson 14 Aug '12

14 Aug '12

As some of you know, I've been working on extending hda-emu to become an automated test suite. A little status report follows: * Daniel Manrique provided me with alsa-info from ~110 machines that Canonical have certified. These alsa-infos are now upstreamed (except for a few which were marked confidential or never reached the market), and distributed with hda-emu [1]. * The test suite (which also is a part of hda-emu) loops over all these codecs, and sends a few commands to hda-emu for each one of them. E g, it tries setting volume controls and verifies that the value read back is the same it tried to set. It also looks for errors from hda-emu after each command sent. * I already run daily imports of Takashi's sound tarballs to provide automatic DKMS-based builds. What's new is that this import script [3] now also runs the test suite on that code. It uses the summary of this test run as commit message, which is then imported into Launchpad [2], and sent out by email (to anyone subscribed to that branch). I've just set this up, so maybe there will be some bumps before the import script runs correctly, but hopefully everything will run smoothly soon, and that we'll catch some regressions before they affect anyone. :-) -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic [1] http://git.kernel.org/?p=linux/kernel/git/tiwai/hda-emu.git;a=commit;h=fb3b… [2] https://code.launchpad.net/~ubuntu-audio-dev/alsa-driver/tiwai-trunk2 [3] https://code.launchpad.net/~ubuntu-audio-dev/alsa-driver/filter-sound-script

1 0

[alsa-devel] [PATCH 1/2] hda-emu: configure.ac: Allow relative paths in hdadir
by David Henningsson 14 Aug '12

14 Aug '12

Previously, adding a path that was relative could fail because the symlinks created were broken. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- configure.ac | 2 ++ 1 file changed, 2 insertions(+) diff --git a/configure.ac b/configure.ac index 315e759..18118f3 100644 --- a/configure.ac +++ b/configure.ac @@ -51,6 +51,8 @@ AC_ARG_WITH(hdadir, [path where HD-audio kernel files are stored]), hdadir="$withval", hdadir="") +hdadir="$( cd "$hdadir" && pwd )" + if test -n "$hdadir"; then test -d "$hdadir" || \ -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ALSA: hda - Fix 'Beep Playback Switch' with no underlying mute switch
by David Henningsson 14 Aug '12

14 Aug '12

Some conexant devices has no mute capability on their Beep widgets. This patch makes sure we don't try setting mutes on those widgets. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Another Beep fix. It's not that I'm suddenly obsessed with beeping; it's my automated test tool that finds them :-) I admit "knew->info == snd_hda_mixer_amp_switch_info" and "kctl->private_value = 0x10000" looks a bit hacky, feel free to suggest something more elegant if you wish. Example alsa info: codecs/canonical/cx20590-lenovo-thinkpad-t420-ccert-201102-7230 sound/pci/hda/hda_beep.c | 6 ++++-- sound/pci/hda/patch_conexant.c | 8 +++++++- 2 files changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index d26ae65..3a72543 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -237,9 +237,9 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; - if (beep && !beep->enabled) { + if (beep && (!get_amp_nid(kcontrol) || !beep->enabled)) { ucontrol->value.integer.value[0] = - ucontrol->value.integer.value[1] = 0; + ucontrol->value.integer.value[1] = beep->enabled; return 0; } return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); @@ -263,6 +263,8 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, enable |= *valp; snd_hda_enable_beep_device(codec, enable); } + if (!get_amp_nid(kcontrol)) + return 0; return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep); diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 5e22a8f..fa36c5e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -537,12 +537,18 @@ static int conexant_build_controls(struct hda_codec *codec) /* create beep controls if needed */ if (spec->beep_amp) { const struct snd_kcontrol_new *knew; + unsigned int caps; + caps = snd_hda_param_read(codec, spec->beep_amp, AC_PAR_AMP_OUT_CAP); + for (knew = cxt_beep_mixer; knew->name; knew++) { struct snd_kcontrol *kctl; kctl = snd_ctl_new1(knew, codec); if (!kctl) return -ENOMEM; - kctl->private_value = spec->beep_amp; + if (knew->info == snd_hda_mixer_amp_switch_info && !(caps & AC_AMPCAP_MUTE)) + kctl->private_value = 0x10000; /* Mono channel, no nid */ + else + kctl->private_value = spec->beep_amp; err = snd_hda_ctl_add(codec, 0, kctl); if (err < 0) return err; -- 1.7.9.5

2 5

[alsa-devel] [PATCH v2 0/9] ARM/ASoC: OMAP McBSP device tree support
by Peter Ujfalusi 14 Aug '12

14 Aug '12

Hello, Changes since v1: - ICLK workaround for the sidetone module has been kept for legacy mode. Intro mail ffrom v1 (updated): in order to be able to add DT support for the McBSP driver which is used on all OMAP platforms (OMAP1/2/3/4/5) I needed to make some cleanups to the stack: - Removing callbacks used by the driver for - clock re-parenting (code moved to sound driver) - FSR/CLKR mux configuration (from now on it need to be done at board level) This mux is board specific and depends on the wiring. - Replaced the cpu_is_omap* checks with future proof check. - The FIFO was not enabled on OMAP2430 SoC It has been tested on BeagleBoard (with legacy and DT boot). The series changes McBSP related files mostly. It would be great if the whole series could go via audio tree (if the patches are OK and it is fine by Tony). For DT booted kernel on OMAP3 proper ICLK woraround need to be implemented. At the moment I don't see this as a stopper since we can not properly boot OMAP3 devices (especially rx51 which is using the ST). Other OMAP revisions does not have ST block so the missing workaround is not going to hurt them. I'm working on a solution which can replace the current workaround with a generic implementation to support DT and non DT booted kernels. Regards, Peter --- Peter Ujfalusi (9): ARM/ASoC: omap-mcbsp: Move OMAP2+ clock parenting code to ASoC driver ARM: OMAP: mcbsp: Enable FIFO use for OMAP2430 ARM: OMAP: board-am3517evm: Configure McBSP1 CLKR/FSR signal source ASoC: am3517evm: Do not configure McBSP1 CLKR/FSR signal muxing ARM/ASoC: omap-mcbsp: Remove CLKR/FSR mux configuration code ASoC: omap-mcbsp: Remove unused defines ASoC: omap-mcbsp: Remove cpu_is_omap* checks from the code ARM: OMAP2+: McBSP: Do not create legacy devices when booting with DT data ASoC: omap-mcbsp: Add device tree bindings .../devicetree/bindings/sound/omap-mcbsp.txt | 45 +++++++ arch/arm/mach-omap2/board-am3517evm.c | 13 ++ arch/arm/mach-omap2/mcbsp.c | 126 +------------------ arch/arm/plat-omap/include/plat/mcbsp.h | 2 - sound/soc/omap/am3517evm.c | 20 +--- sound/soc/omap/mcbsp.c | 54 ++++----- sound/soc/omap/mcbsp.h | 3 - sound/soc/omap/omap-mcbsp.c | 106 +++++++++++----- sound/soc/omap/omap-mcbsp.h | 20 --- 9 files changed, 162 insertions(+), 227 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/omap-mcbsp.txt -- 1.7.8.6

2 11

[alsa-devel] [PATCH] ASoC: jack: Always notify full jack status
by Mark Brown 13 Aug '12

13 Aug '12

Don't just notify for the bits we've updated, notify the full state of the jack otherwise users might get confused by misleading reports. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-jack.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index e19fd6d..9ece60a 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -98,7 +98,7 @@ void snd_soc_jack_report(struct snd_soc_jack *jack, int status, int mask) } /* Report before the DAPM sync to help users updating micbias status */ - blocking_notifier_call_chain(&jack->notifier, status, jack); + blocking_notifier_call_chain(&jack->notifier, jack->status, jack); snd_soc_dapm_sync(dapm); -- 1.7.10.4

1 0

[alsa-devel] Moving hda-analyzer to alsa-tools?
by Takashi Iwai 13 Aug '12

13 Aug '12

Hi Jaroslav, do you have any plan to move hda-analyzer to alsa-tools package? Looking at the recent demand on it, I think it's worth to merge into the common alsa-tools repo, instead of keeping it separately. FWIW, HD-audio specific tools like hda-verb and hdajackretask have been already merged there. thanks, Takashi

1 0

[alsa-devel] [PATCH V5 14/18] ALSA: HDA: Make hda sound card usable for Loongson.
by Huacai Chen 13 Aug '12

13 Aug '12

Lemote A1004(Laptop) and A1205(All-In-One) use Conexant's hda codec, this patch modify patch_conexant.c to add Lemote specific code. Signed-off-by: Jie Chen <chenj(a)lemote.com> Signed-off-by: Huacai Chen <chenhc(a)lemote.com> Signed-off-by: Hongliang Tao <taohl(a)lemote.com> Signed-off-by: Hua Yan <yanh(a)lemote.com> Cc: alsa-devel(a)alsa-project.org --- include/linux/pci_ids.h | 2 ++ sound/pci/hda/patch_conexant.c | 24 ++++++++++++++++++++++++ 2 files changed, 26 insertions(+), 0 deletions(-) diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index fc35260..b28270e 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -2918,3 +2918,5 @@ #define PCI_DEVICE_ID_XEN_PLATFORM 0x0001 #define PCI_VENDOR_ID_OCZ 0x1b85 + +#define PCI_VENDOR_ID_LEMOTE 0x1c06 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1436118..b7de368 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -4414,6 +4414,8 @@ static const struct hda_codec_ops cx_auto_patch_ops = { enum { CXT_PINCFG_LENOVO_X200, CXT_PINCFG_LENOVO_TP410, + CXT_PINCFG_LEMOTE_A1004, + CXT_PINCFG_LEMOTE_A1205, CXT_FIXUP_STEREO_DMIC, }; @@ -4441,6 +4443,18 @@ static const struct hda_pintbl cxt_pincfg_lenovo_tp410[] = { {} }; +/* Lemote A1004/A1205 with cxt5066 */ +static const struct hda_pintbl cxt_pincfg_lemote[] = { + { 0x1a, 0x90a10020 }, /* Internal mic */ + { 0x1b, 0x03a11020 }, /* External mic */ + { 0x1d, 0x400101f0 }, /* Not used */ + { 0x1e, 0x40a701f0 }, /* Not used */ + { 0x20, 0x404501f0 }, /* Not used */ + { 0x22, 0x404401f0 }, /* Not used */ + { 0x23, 0x40a701f0 }, /* Not used */ + {} +}; + static const struct hda_fixup cxt_fixups[] = { [CXT_PINCFG_LENOVO_X200] = { .type = HDA_FIXUP_PINS, @@ -4450,6 +4464,14 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_PINS, .v.pins = cxt_pincfg_lenovo_tp410, }, + [CXT_PINCFG_LEMOTE_A1004] = { + .type = HDA_FIXUP_PINS, + .v.pins = cxt_pincfg_lemote, + }, + [CXT_PINCFG_LEMOTE_A1205] = { + .type = HDA_FIXUP_PINS, + .v.pins = cxt_pincfg_lemote, + }, [CXT_FIXUP_STEREO_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_stereo_dmic, @@ -4467,6 +4489,8 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x215f, "Lenovo T510", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21ce, "Lenovo T420", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x21cf, "Lenovo T520", CXT_PINCFG_LENOVO_TP410), + SND_PCI_QUIRK(PCI_VENDOR_ID_LEMOTE, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004), + SND_PCI_QUIRK(PCI_VENDOR_ID_LEMOTE, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205), SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), {} -- 1.7.7.3

2 2

[alsa-devel] ICT Bits for HDMI HBR feature
by Wang, Xingchao 13 Aug '12

13 Aug '12

Hi Takashi, Should we set ICT bits when enable HDMI HBR features? Here's quote from HDA Specification Rev1.0a(page 150). " ICT[3:0] (IEC Coding Type): Programmed according to IEC standards to enable stream types for High Bit Rate Encoding. This is valid only for HDMI and Display port digital Pin Widgets. " Since HDMI HBR already works on some chips, I wonder if they set the bits correctly. Thanks --xingchao

3 3

[alsa-devel] [PATCH 1/3] ALSA: hda - show ICT/KAE control bits
by Wang Xingchao 13 Aug '12

13 Aug '12

Enable two debug options for S/PDIF Converter Control. KAE: Keep Alive Enable; ICT: IEC Coding Type. Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_codec.h | 4 ++++ sound/pci/hda/hda_proc.c | 4 ++++ 2 files changed, 8 insertions(+) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c422d33..141b3b8 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -386,6 +386,10 @@ enum { /* DIGITAL2 bits */ #define AC_DIG2_CC (0x7f<<0) +/* DIGITAL3 bits */ +#define AC_DIG3_ICT (0xf<<0) +#define AC_DIG3_KAE (1<<23) + /* Pin widget control - 8bit */ #define AC_PINCTL_EPT (0x3<<0) #define AC_PINCTL_EPT_NATIVE 0 diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 7e46258..a222aae 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -419,9 +419,13 @@ static void print_digital_conv(struct snd_info_buffer *buffer, snd_iprintf(buffer, " Pro"); if (digi1 & AC_DIG1_LEVEL) snd_iprintf(buffer, " GenLevel"); + if (digi1 & AC_DIG3_KAE) + snd_iprintf(buffer, " KAE"); snd_iprintf(buffer, "\n"); snd_iprintf(buffer, " Digital category: 0x%x\n", (digi1 >> 8) & AC_DIG2_CC); + snd_iprintf(buffer, " IEC Coding Type: 0x%x\n", + (digi1 >> 16) & AC_DIG3_ICT); } static const char *get_pwr_state(u32 state) -- 1.7.9.5

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[alsa-devel] [PATCH v2] ALSA: hda - show ICT/KAE control bits
by Wang Xingchao 13 Aug '12

13 Aug '12

Enable two debug options for S/PDIF Converter Control. KAE: Keep Alive Enable; ICT: IEC Coding Type. Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_codec.h | 4 ++++ sound/pci/hda/hda_proc.c | 9 ++++++++- 2 files changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c422d33..afd2af0 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -386,6 +386,10 @@ enum { /* DIGITAL2 bits */ #define AC_DIG2_CC (0x7f<<0) +/* DIGITAL3 bits */ +#define AC_DIG3_ICT (0xf<<0) +#define AC_DIG3_KAE (1<<7) + /* Pin widget control - 8bit */ #define AC_PINCTL_EPT (0x3<<0) #define AC_PINCTL_EPT_NATIVE 0 diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 7e46258..32bb3d9 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -402,6 +402,9 @@ static void print_digital_conv(struct snd_info_buffer *buffer, { unsigned int digi1 = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); + unsigned char digi2 = digi1 >> 8; + unsigned char digi3 = digi1 >> 16; + snd_iprintf(buffer, " Digital:"); if (digi1 & AC_DIG1_ENABLE) snd_iprintf(buffer, " Enabled"); @@ -419,9 +422,13 @@ static void print_digital_conv(struct snd_info_buffer *buffer, snd_iprintf(buffer, " Pro"); if (digi1 & AC_DIG1_LEVEL) snd_iprintf(buffer, " GenLevel"); + if (digi3 & AC_DIG3_KAE) + snd_iprintf(buffer, " KAE"); snd_iprintf(buffer, "\n"); snd_iprintf(buffer, " Digital category: 0x%x\n", - (digi1 >> 8) & AC_DIG2_CC); + digi2 & AC_DIG2_CC); + snd_iprintf(buffer, " IEC Coding Type: 0x%x\n", + digi3 & AC_DIG3_ICT); } static const char *get_pwr_state(u32 state) -- 1.7.9.5

2 1

[alsa-devel] HDA Mono out
by Raymond Yau 13 Aug '12

13 Aug '12

How can the driver differentiate those mono internal chassis speaker of desktop from those subwoofer speaker of laptop? https://launchpadlibrarian.net/34863455/Card0.Codecs.codec.2.txt Node 0x13 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x00] Pincap 0x00000010: OUT Pin Default 0x911711f0: [Fixed] Speaker at Int Rear Conn = Analog, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x2d refer to the patch hda: Added mono_out_pin to autoconfig Added a mono_out_pin field to autocfg struct, and code to parse for the mono_out_line. http://git.kernel.org/?p=linux/kernel/git/tiwai/sound.git;a=blobdiff;f=soun… does any computer really has the mono jack using the mono pin complex? do pulseaudio classify those mono internal chassis speaker as desktop speaker path or speaker path? http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mix… in patch_analog.c, "mono playback switch" is created in 6stack. 3stack and laptop model. the auto model need to handle the speaker of the laptop implementation but lenovo 3000 workstation contain the internal chassis speaker which hda auto parser put it in speaker_pins instead of mono_out if ((err = ad1988_auto_create_multi_out_ctls(spec, &spec->autocfg)) < 0 || (err = ad1988_auto_create_extra_out(codec, spec->autocfg.speaker_pins[0], "Speaker")) < 0 || (err = ad1988_auto_create_extra_out(codec, spec->autocfg.hp_pins[0], "Headphone")) < 0 || (err = ad1988_auto_create_analog_input_ctls(codec, &spec->autocfg)) < 0) return err; one possible way to fix this bug for lenovo 3000 by checking wcaps of the pin support mono/stereo in ad1988_auto_create_extra_out and create the mono playback switch for the internal chassis speaker instead of the speaker playback volume/switch since mono out pin have to share audio output 0x04 with green jack at rear panel.

2 1

[alsa-devel] Gitcompile for Kernel 3.2?
by Brand, Mark (Mr) (Summerstrand Campus South) 13 Aug '12

13 Aug '12

Hi all, Having cloned alsa-driver and alsa-kmirror, my gitcompile and sudo make install-modules goes smoothly. However, the modules won't load. First errors in dmesg are: [ 24.415466] snd_hwdep: disagrees about version of symbol snd_info_register [ 24.415475] snd_hwdep: Unknown symbol snd_info_register (err -22) [ 24.415489] snd_hwdep: disagrees about version of symbol snd_info_create_module_entry [ 24.415495] snd_hwdep: Unknown symbol snd_info_create_module_entry (err -22) ... and so on. My initial assumption is that kmirror is specific to a 2.6 kernel, while I'm running 3.2, though some of what I've read seems to contradict this. I've tried foregoing kmirror for a full 3.2 kernel source, but that failed on gitcompile. I also tried with kmirror on a 2.6 kernel, but those modules also failed to load. I initially thought my problems stemmed from trying to do this against a lowlatency kernel, but I've subsequently had the same against vanilla Ubuntu kernels. Would someone kindly point me in the right direction? * Is it possible to compile and load a subset of alsa modules without having full kernel sources on hand? * If so, is this restricted to particular kernels (docs seem to suggest 2.6, 3.0 and 3.2 are supported)? * If not, is alsa-driver expected to compile against 3.2 sources as is (mine fails on some of the patching - presumably the make dep step)? * Does the alsa-driver.new.git merge address this, or is development expected to take place in a 2.6 environment? Sorry if any or all of this is incoherent (please help me to clarify), but any pointers would be greatly appreciated. What might be pertinent: Ubuntu 12.04 (kernels 3.2.0 and 2.6.38 tested), gcc 4.6.3, anything else? I am attempting to add device support for a multi-port MIDI interface. I managed to get a working prototype running a year ago, but had to shelve the project due to work pressures. I'd really love to get it back up and submitted. Kind regards, Mark Brand. ________________________________ NOTICE: Please note that this eMail, and the contents thereof, is subject to the standard NMMU eMail disclaimer which may be found at: http://www.nmmu.ac.za/disclaimer/email.htm

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[alsa-devel] [PATCH] ALSA: hda - show ICT/KAE control bits
by Wang Xingchao 13 Aug '12

13 Aug '12

Enable two debug options for S/PDIF Converter Control. KAE: Keep Alive Enable; ICT: IEC Coding Type. Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/hda_codec.h | 4 ++++ sound/pci/hda/hda_proc.c | 4 ++++ 2 files changed, 8 insertions(+) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index c422d33..7167f37 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -386,6 +386,10 @@ enum { /* DIGITAL2 bits */ #define AC_DIG2_CC (0x7f<<0) +/* DIGITAL3 bits */ +#define AC_DIG3_ICT (0xf<<16) +#define AC_DIG3_KAE (1<<23) + /* Pin widget control - 8bit */ #define AC_PINCTL_EPT (0x3<<0) #define AC_PINCTL_EPT_NATIVE 0 diff --git a/sound/pci/hda/hda_proc.c b/sound/pci/hda/hda_proc.c index 7e46258..a222aae 100644 --- a/sound/pci/hda/hda_proc.c +++ b/sound/pci/hda/hda_proc.c @@ -419,9 +419,13 @@ static void print_digital_conv(struct snd_info_buffer *buffer, snd_iprintf(buffer, " Pro"); if (digi1 & AC_DIG1_LEVEL) snd_iprintf(buffer, " GenLevel"); + if (digi1 & AC_DIG3_KAE) + snd_iprintf(buffer, " KAE"); snd_iprintf(buffer, "\n"); snd_iprintf(buffer, " Digital category: 0x%x\n", (digi1 >> 8) & AC_DIG2_CC); + snd_iprintf(buffer, " IEC Coding Type: 0x%x\n", + (digi1 >> 16) & AC_DIG3_ICT); } static const char *get_pwr_state(u32 state) -- 1.7.9.5

1 0

[alsa-devel] [PATCH 00/11] MFD/ASoC/Input: twl4030-audio submodule DT support
by Peter Ujfalusi 12 Aug '12

12 Aug '12

Hello, The following series adds DT support for the twl4030 audio submodule which provides audio codec and vibra functionality. The MFD core driver is probed via DT, it will create the needed child devices based on the provided information in the DT blob. Child drivers (vibra, ASoC codec) will parse the core's node if needed to get the needed parameters for their configuration. In the ASoC codec driver the hs_extmute callback (which was used to toggle a GPIO line) has been removed. The codec driver will receive the GPIO number (if it is needed on the platform) and also the level to disable the mute. If the series is OK (and no objections from the maintainers), it would be good if this can go via audio. Changed files are well contained within the twl4030-audio stack so I do not expect merge issues later. The series has been tested on BeagleBoard (with the McBSP DT series, and with the upcoming DT audio support for BeagleBoard). Regards, Peter --- Peter Ujfalusi (11): MFD: twl4030-audio: Clean up MODULE_* and platform_driver part MFD: twl4030-audio: Convert to use devm_kzalloc MFD: twl4030-audio: Rearange and clean-up the probe function MFD: twl4030-audio: Add DT support Input: twl4030-vibra: Support for DT booted kernel ASoC: twl4030: Move hs_extmute GPIO handling to driver ARM: OMAP/ASoC: Zoom2: Let the codec to handle the hs_extmute GPIO ASoC/MFD: twl4030: Remove set_hs_extmute callback from platform data ASoC: twl4030: Convert to use devm_kzalloc ASoC: twl4030: Add pointer to pdata within the private data ASoC: twl4030: Support for DT booted kernel .../devicetree/bindings/mfd/twl4030-audio.txt | 51 +++++++++ arch/arm/mach-omap2/board-zoom-peripherals.c | 9 +- arch/arm/mach-omap2/include/mach/board-zoom.h | 2 - drivers/input/misc/twl4030-vibra.c | 20 +++- drivers/mfd/twl4030-audio.c | 114 +++++++++++++------ include/linux/i2c/twl.h | 4 +- sound/soc/codecs/twl4030.c | 117 ++++++++++++++++---- sound/soc/omap/zoom2.c | 4 - 8 files changed, 250 insertions(+), 71 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/twl4030-audio.txt -- 1.7.8.6

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Re: [alsa-devel] [PATCHv2.1-CA0132 HDA Codec 1/2] Add DSP loader code to CA0132 codec
by Takashi Iwai 11 Aug '12

11 Aug '12

At Fri, 10 Aug 2012 23:17:51 -0700, Ian Minett wrote: > > From: Ian Minett <ian_minett(a)creativelabs.com> > > Hi Takashi, > Great, it's good to hear that you are able to assist with adding the DSP > loader modifications to hda_intel.c. > > I've attached a draft patch as you requested. Please let us know if you have > any questions concerning it. The first patch is the previous update to add DSP > features to patch_ca0132, then the second is the draft patch, to add the > new DMA loader calls. Well, for cleanness, I think we should start rather from scratch: - The first patch to add a new hda_bus_ops for DSP transfer; it won't touch patch_ca0132.c but only the core part (hda_intel.c, hda_codec.[ch]) - Add the DSP transfer to patch_ca0132.c; at this point, just do DMA transfer but no new mixers etc won't be added - Add new features to patch_ca0132.c At the early development stage, the first two items can be a single patch. We can split to two when applied to the upstream. But, mixing all changes in a single patch is pretty bad. So the point is to concentrate only on DMA transfer stuff at first. Once after confirmed it's working, continue to the new DSP features. > We're also currently ascertaining the license information to go along with the > firmware binary release. Good. This is really mandatory for upstreaming. > The project team would like to know, beyond the outstanding items mentioned > above, do you see any other potential issues with the updates to the codec that > could possibly slow the submission process? We're starting to get rather close > to our target date. > Also, once the codec has made it into the ALSA tree, how long do you estimate > it would take before it is upstreamed to the Linux kernel? If you could > provide any info on the timeframe for this process, we'd greatly appreciate it. Heh, this depends on how hard we work :) If it can be quickly done, it'll be merged even in 3.7 kernel. Takashi

1 0

[alsa-devel] [PATCH][trivial] ASoC: isabelle: Remove unneeded include of version.h
by Jesper Juhl 10 Aug '12

10 Aug '12

There is no need to include version.h in sound/soc/codecs/isabelle.c - this patch removes the pointless include. Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> --- sound/soc/codecs/isabelle.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 5d8f39e..1bf5560 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -13,7 +13,6 @@ */ #include <linux/module.h> #include <linux/moduleparam.h> -#include <linux/version.h> #include <linux/kernel.h> #include <linux/init.h> #include <linux/delay.h> -- 1.7.11.4 -- Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ Don't top-post http://www.catb.org/jargon/html/T/top-post.html Plain text mails only, please.

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Re: [alsa-devel] [PATCH] ASoC: Samsung: Fix build error
by Mark Brown 10 Aug '12

10 Aug '12

On Wed, Aug 08, 2012 at 11:34:43AM +0530, Sachin Kamat wrote: > Fixes the following build error: > In file included from arch/arm/mach-exynos/include/mach/dma.h:24:0, > from arch/arm/plat-samsung/include/plat/dma-ops.h:17, Applied, thanks.

1 0

Re: [alsa-devel] [PATCH] ASoC: core: remove unused variable in soc_probe() in linux-next
by Mark Brown 10 Aug '12

10 Aug '12

On Thu, Aug 09, 2012 at 11:16:26PM -0700, Jerry Snitselaar wrote: > With commit 28d528c8 "ASoC: core: Remove pointless error on card > registration failure", the variable ret is no longer used in > soc_probe() and generates an unused variable warning during a build. Applied, thanks.

1 0

[alsa-devel] [PATCH] ALSA : hda - bug fix on return value check when getting supported power states of a codec
by Lin, Mengdong 10 Aug '12

10 Aug '12

The return value of snd_hda_param_read() is -1 for an error, otherwise it's the supported power states of a codec. The supported power states is a 32-bit value. Bit 31 will be set to 1 if the codec supports EPSS, thus making "sup" negative. And the bit 28:5 is reserved as "0". So a negative value other than -1 shall be further checked. Please refer to High-Definition spec 7.3.4.12 "Supported Power States", thanks! Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 88a9c20..629131a 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3497,7 +3497,7 @@ static bool snd_hda_codec_get_supported_ps(struct hda_codec *codec, hda_nid_t fg { int sup = snd_hda_param_read(codec, fg, AC_PAR_POWER_STATE); - if (sup < 0) + if (sup == -1) return false; if (sup & power_state) return true; -- 1.7.9.5

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[alsa-devel] [PATCH 00/11] ARM/ASoC: OMAP McBSP device tree support
by Peter Ujfalusi 10 Aug '12

10 Aug '12

Hello, in order to be able to add DT support for the McBSP driver which is used on all OMAP platforms (OMAP1/2/3/4/5) I needed to make some cleanups to the stack: - Removing callbacks used by the driver for - clock re-parenting (code moved to sound driver) - FSR/CLKR mux configuration (from now on it need to be done at board level) This mux is board specific and depends on the wiring. - OMAP3 sidetone ICLK workaround has been reimplemented using the SYSCONFIG register's SIDLEMODE filed to prevent ICLK gating while ST is running. - Replaced the cpu_is_omap* checks with future proof check. - The FIFO was not enabled on OMAP2430 SoC It has been tested on BeagleBoard (with legacy and DT boot). The series changes McBSP related files mostly. It would be great if the whole series could go via audio tree (if the patches are OK and it is fine by Tony). Regards, Peter --- Peter Ujfalusi (11): ARM/ASoC: omap-mcbsp: Move OMAP2+ clock parenting code to ASoC driver ARM: OMAP2+: McBSP: Do not create legacy devices when booting with DT data ARM: OMAP: mcbsp: Enable FIFO use for OMAP2430 ARM: OMAP: board-am3517evm: Configure McBSP1 CLKR/FSR signal source ASoC: am3517evm: Do not configure McBSP1 CLKR/FSR signal muxing ARM/ASoC: omap-mcbsp: Remove CLKR/FSR mux configuration code ASoC: omap-mcbsp: Sidetone: Use SIDLE bits in SYSCONFIG register to select noidle mode ARM: OMAP3: Remove callback for McBSP sidetone ICLK workaround ASoC: omap-mcbsp: Remove unused defines ASoC: omap-mcbsp: Remove cpu_is_omap* checks from the code ASoC: omap-mcbsp: Add device tree bindings .../devicetree/bindings/sound/omap-mcbsp.txt | 45 ++++++ arch/arm/mach-omap2/board-am3517evm.c | 13 ++ arch/arm/mach-omap2/mcbsp.c | 152 +------------------- arch/arm/plat-omap/include/plat/mcbsp.h | 3 - sound/soc/omap/am3517evm.c | 20 +--- sound/soc/omap/mcbsp.c | 72 +++++----- sound/soc/omap/mcbsp.h | 4 +- sound/soc/omap/omap-mcbsp.c | 106 ++++++++++----- sound/soc/omap/omap-mcbsp.h | 20 --- 9 files changed, 177 insertions(+), 258 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/omap-mcbsp.txt -- 1.7.8.6

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[alsa-devel] [PATCH 1/2] ASoC: wm5102: Add missing input PGA routes
by Mark Brown 10 Aug '12

10 Aug '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5102.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 4c2fc36..a00901c 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -643,6 +643,15 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "AIF2 Capture", NULL, "SYSCLK" }, { "AIF3 Capture", NULL, "SYSCLK" }, + { "IN1L PGA", NULL, "IN1L" }, + { "IN1R PGA", NULL, "IN1R" }, + + { "IN2L PGA", NULL, "IN2L" }, + { "IN2R PGA", NULL, "IN2R" }, + + { "IN3L PGA", NULL, "IN3L" }, + { "IN3R PGA", NULL, "IN3R" }, + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), -- 1.7.10.4

1 1

[alsa-devel] Question about Conversion to S24_LE from S24_3LE
by Fujikawa, Kazutoshi 10 Aug '12

10 Aug '12

Hello. I'd like to play a wav file converted from S24_3LE format to S24_LE by alsa-lib. (I use aplay command.) I recognize the following: <PCM format in wav file> ex. Valid PCM data is 0x123456. S24_LE : 0x12345600(in 4bytes, LSB 0 padding) S24_3LE : 0x123456 (in 3bytes, No padding) When the S24_LE format file is played, the ALSA driver receives data format is as follows: 0x12345600 (LSB 0 padding) However, when the S24_3LE format file which is converted to S24_LE by alsa-lib is played, the ALSA driver receives data formats is as follows: 0x00123456 (MSB 0(or 0xFF) padding) Why is the how to pad 0 different(LSB/MSB)? cf. When "snd_pcm_linear_getput" is called 'get_idx' is set 16. 'put_idx' is set 8. [alsa-lib:pcm_linear.c] 211 void snd_pcm_linear_getput(const snd_pcm_channel_area_t *dst_areas, snd_ pcm_uframes_t dst_offset, 212 const snd_pcm_channel_area_t *src_areas, snd_ pcm_uframes_t src_offset, 213 unsigned int channels, snd_pcm_uframes_t fram es, 214 unsigned int get_idx, unsigned int put_idx) 215 { 216 #define CONV24_LABELS 217 #include "plugin_ops.h" 218 #undef CONV24_LABELS 219 void *get = get32_labels[get_idx]; 220 void *put = put32_labels[put_idx]; 221 unsigned int channel; 222 u_int32_t sample = 0; 223 for (channel = 0; channel < channels; ++channel) { 224 const char *src; When 'get_idx' is 16 and 'put_idx' is 8, it is converted as follows: 0x123456(S24_3LE) -> 0x00123456(MSB 0 padding) I expect the following: 0x123456(S24_3LE) -> 0x12345600(LSB 0 padding) (If 'get_idx' is 16 and 'put_idx' is 12, it is converted like this.) I'm doing something wrong? -------------------- alsa-lib-1.0.24.1 alsa-utils-1.0.24.2 -------------------- Regards. Kaz

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[alsa-devel] Error with "Unsigned 8 bit, Rate 8000 Hz, Mono" track
by Fabio Estevam 10 Aug '12

10 Aug '12

Hi, Running 3.6-rc1 on a mx28evk I get the following error with this wav format: root@freescale /home$ aplay k3b_error1.wav Playing WAVE 'k3b_error1.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono [ 303.640000] asoc: machine hw_params failed: -22 aplay: set_params:959: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: U8 SUBFORMAT: STD SAMPLE_BITS: 8 FRAME_BITS: 8 CHANNELS: 1 RATE: 8000 PERIOD_TIME: 125000 PERIOD_SIZE: 1000 PERIOD_BYTES: 1000 PERIODS: 4 BUFFER_TIME: 500000 BUFFER_SIZE: 4000 BUFFER_BYTES: 4000 TICK_TIME: 0 Haven't started debugging yet, but if anyone sees something obvious, please let me know. This same file plays correctly on mx31/mx27/mx51/mx6q boards. Regards, Fabio Estevam

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[alsa-devel] [PATCH] Add *.swp to .gitignore.
by Tanu Kaskinen 10 Aug '12

10 Aug '12

These are vim swap files. Signed-off-by: Tanu Kaskinen <tanu.kaskinen(a)digia.com> --- .gitignore | 1 + 1 file changed, 1 insertion(+) diff --git a/.gitignore b/.gitignore index 64d0bc6..7f1827e 100644 --- a/.gitignore +++ b/.gitignore @@ -27,6 +27,7 @@ ltconfig *.pc *.o *~ +*.swp include/config.h include/stamp-h1 include/stamp-vh -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix panned "Beep Playback Switch"
by David Henningsson 10 Aug '12

10 Aug '12

When "Beep Playback Switch" had a different value on left and right channels (such as muting left but not right, or vice versa), this could result in the right channel being ignored. This patch enables beep to be sounding from right channel only, and also give correct result back to userspace (e g amixer). Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/hda_beep.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/hda_beep.c b/sound/pci/hda/hda_beep.c index 0bc2315..d26ae65 100644 --- a/sound/pci/hda/hda_beep.c +++ b/sound/pci/hda/hda_beep.c @@ -237,10 +237,9 @@ int snd_hda_mixer_amp_switch_get_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; - if (beep) { + if (beep && !beep->enabled) { ucontrol->value.integer.value[0] = - ucontrol->value.integer.value[1] = - beep->enabled; + ucontrol->value.integer.value[1] = 0; return 0; } return snd_hda_mixer_amp_switch_get(kcontrol, ucontrol); @@ -252,9 +251,18 @@ int snd_hda_mixer_amp_switch_put_beep(struct snd_kcontrol *kcontrol, { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct hda_beep *beep = codec->beep; - if (beep) - snd_hda_enable_beep_device(codec, - *ucontrol->value.integer.value); + if (beep) { + u8 chs = get_amp_channels(kcontrol); + int enable = 0; + long *valp = ucontrol->value.integer.value; + if (chs & 1) { + enable |= *valp; + valp++; + } + if (chs & 2) + enable |= *valp; + snd_hda_enable_beep_device(codec, enable); + } return snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); } EXPORT_SYMBOL_HDA(snd_hda_mixer_amp_switch_put_beep); -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/6] Bugfixes and clean-ups bound for the v3.6 RCs
by Lee Jones 10 Aug '12

10 Aug '12

This patch-set includes some important changes which should make their way to the Mainline Release Candidates for the v3.6 release. Without them Audio doesn't even probe (at all) when booting with Device Tree selected. The kernel can't compile with ux500 audio enabled. Even if we can get that far SoC Core will assume we're using regmaps and attempt to use them; however, none exist for this driver, so the kernel will oops. arch/arm/mach-ux500/Kconfig | 1 + arch/arm/mach-ux500/board-mop500-msp.c | 2 +- arch/arm/mach-ux500/board-mop500.c | 9 ++++----- sound/soc/codecs/ab8500-codec.c | 4 ++++ sound/soc/soc-dapm.c | 2 -- 5 files changed, 10 insertions(+), 8 deletions(-)

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[alsa-devel] [patch] ALSA: cs46xx - signedness bug in snd_cs46xx_codec_read()
by Dan Carpenter 10 Aug '12

10 Aug '12

This function returns its own error codes instead of normal negative error codes. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index f75f5ff..a71d1c1 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -94,7 +94,7 @@ static unsigned short snd_cs46xx_codec_read(struct snd_cs46xx *chip, if (snd_BUG_ON(codec_index != CS46XX_PRIMARY_CODEC_INDEX && codec_index != CS46XX_SECONDARY_CODEC_INDEX)) - return -EINVAL; + return 0xffff; chip->active_ctrl(chip, 1);

2 1

[alsa-devel] PCM substream
by arun kumar 10 Aug '12

10 Aug '12

Hi All, >From the last couple of days, I have been trying to understand the Linux Alsa Subsystem.I have read the Alsa Driver API and Application API's. But I could not understand the concept of PCM substreams, Could you please give me some pointers to get a good understanding on the PCM substream. regards, Arun.

3 4

[alsa-devel] [PATCH 0/4] ASoC/OMAP: ASoC machine driver for simple SoC with twl4030
by Peter Ujfalusi 10 Aug '12

10 Aug '12

Hello, Create new unified ASoC machine driver (with DT support) for boards using simple configuration for audio and using the twl4030 codec. This driver will replace the following machine drivers (since they are basically the same): igep0020, omap3beagle, omap3evm and overo. This series will switch the boards I can identify to use the new unified ASoC machine driver. Changes under arch/arm/mach-omap2/ are trivial, I don't expect merge issues later. If it is OK for the maintainers I would like this to go via audio tree (with the McBSP and the twl4030-audio stack DT support). The series has been tested on BeagleBoard (with the McBSP and twl4030-audio DT support). Regards, Peter --- Peter Ujfalusi (4): ASoC: omap-twl4030: Simple machine driver for TI SoC with twl4030 codec ARM: OMAP: twl-common: Add helper function to register the omap-twl4030 audio driver ARM: OMAP3: Switch to use the unified audio driver (omap-twl4030) for selected boards ASoC: Remove obsolete OMAP3 machine drivers .../devicetree/bindings/sound/omap-twl4030.txt | 19 ++ arch/arm/mach-omap2/board-cm-t35.c | 1 + arch/arm/mach-omap2/board-devkit8000.c | 1 + arch/arm/mach-omap2/board-igep0020.c | 1 + arch/arm/mach-omap2/board-omap3beagle.c | 1 + arch/arm/mach-omap2/board-omap3evm.c | 1 + arch/arm/mach-omap2/board-overo.c | 1 + arch/arm/mach-omap2/twl-common.c | 28 +++ arch/arm/mach-omap2/twl-common.h | 2 + include/linux/platform_data/omap-twl4030.h | 33 +++ sound/soc/omap/Kconfig | 48 ++--- sound/soc/omap/Makefile | 10 +- sound/soc/omap/igep0020.c | 120 ----------- sound/soc/omap/omap-twl4030.c | 219 ++++++++++++++++++++ sound/soc/omap/omap3beagle.c | 150 ------------- sound/soc/omap/omap3evm.c | 118 ----------- sound/soc/omap/overo.c | 122 ----------- 17 files changed, 322 insertions(+), 553 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/omap-twl4030.txt create mode 100644 include/linux/platform_data/omap-twl4030.h delete mode 100644 sound/soc/omap/igep0020.c create mode 100644 sound/soc/omap/omap-twl4030.c delete mode 100644 sound/soc/omap/omap3beagle.c delete mode 100644 sound/soc/omap/omap3evm.c delete mode 100644 sound/soc/omap/overo.c -- 1.7.8.6

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Re: [alsa-devel] [PATCH] ASoC: Samsung: Fix build error
by Kukjin Kim 10 Aug '12

10 Aug '12

Sachin Kamat wrote: > > Fixes the following build error: > In file included from arch/arm/mach-exynos/include/mach/dma.h:24:0, > from arch/arm/plat-samsung/include/plat/dma-ops.h:17, > from arch/arm/plat-samsung/include/plat/dma.h:128, > from sound/soc/samsung/pcm.c:23: > arch/arm/plat-samsung/include/plat/dma-pl330.h:106:8: > error: redefinition of ‘struct s3c2410_dma_client’ > arch/arm/plat-samsung/include/plat/dma.h:40:8: note: originally defined > here > make[3]: *** [sound/soc/samsung/pcm.o] Error 1 > > Cc: Kukjin Kim <kgene.kim(a)samsung.com> > Signed-off-by: Sachin Kamat <sachin.kamat(a)linaro.org> > Signed-off-by: Sachin Kamat <sachin.kamat(a)samsung.com> BTW, Sachin, do you _really_ need to keep above double signoff with different e-mail? I think, last one should be removed? Because in the field of author, 'linaro.org' e-mail is used? Thanks. Best regards, Kgene. -- Kukjin Kim <kgene.kim(a)samsung.com>, Senior Engineer, SW Solution Development Team, Samsung Electronics Co., Ltd.

1 0

Re: [alsa-devel] [PATCH] ASoC: Samsung: Fix build error
by Kukjin Kim 10 Aug '12

10 Aug '12

Kukjin Kim wrote: > > Sachin Kamat wrote: > > > > On 8 August 2012 17:08, Kukjin Kim <kgene.kim(a)samsung.com> wrote: > > > Sachin Kamat wrote: > > >> > > >> Hi Mark, > > >> > > >> On 8 August 2012 16:39, Mark Brown > > <broonie(a)opensource.wolfsonmicro.com> > > >> wrote: > > >> > On Wed, Aug 08, 2012 at 11:34:43AM +0530, Sachin Kamat wrote: > > >> >> Fixes the following build error: > > >> >> In file included from arch/arm/mach-exynos/include/mach/dma.h:24:0, > > >> >> from arch/arm/plat-samsung/include/plat/dma-ops.h:17, > > >> >> from arch/arm/plat-samsung/include/plat/dma.h:128, > > >> >> from sound/soc/samsung/pcm.c:23: > > >> >> arch/arm/plat-samsung/include/plat/dma-pl330.h:106:8: > > >> >> error: redefinition of 'struct > > > s3c2410_dma_client' > > >> >> arch/arm/plat-samsung/include/plat/dma.h:40:8: note: originally > > defined > > >> here > > >> >> make[3]: *** [sound/soc/samsung/pcm.o] Error 1 > > >> > > > >> > Shouldn't this be being fixed in the headers? It looks like they > > >> > reference and disagree with each other on Exynos (my primary > > development > > >> > platform is S3C6410 based so I'd notice if it was a generic Samsung > > >> > issue. > > >> > > > >> > Pasting the error message isn't really that useful unless the fix > is > > >> > obvious, it's better to explain in words what's wrong. > > >> > > >> I had sent the initial patch trying to fix this in the headers and it > > >> worked for exynos4 config. However Kukjin pointed out that the patch > > >> created similar errors with 2410 and 6400 configs. Fixing it further > > >> in the headers did not look quite straight forward as there are many > > >> headers trying to include one another. Hence I felt it is better > > >> handled in the current way. I understand i should have added this > > >> information in the commit message. Below is the link for your > > >> reference that provides the complete discussion. > > >> > > >> http://comments.gmane.org/gmane.linux.kernel.samsung-soc/11274 > > >> > > >> Please let me know your opinion about this. > > >> > > >> > > > I also agree with Mark's opinion and current Sachin's approach cannot > > avoid > > > whole similar build error. > > > > With the current change I did not see any build error with the configs > > (2410, 6400, 64x0 and exyno4) i checked with (after enabling audio). > > Please let me know if you observed any such build error after adding > > this patch. > > > > > > Happens following error with s5pv210_defcofnig (after enabling s5p-sss). > > In file included from arch/arm/mach-s5pv210/include/mach/dma.h:24, > from arch/arm/plat-samsung/include/plat/dma-ops.h:17, > from arch/arm/plat-samsung/include/plat/dma.h:128, > from drivers/crypto/s5p-sss.c:33: > arch/arm/plat-samsung/include/plat/dma-pl330.h:106: error: redefinition of > 'struct s3c2410_dma_client' > make[2]: *** [drivers/crypto/s5p-sss.o] Error 1 > make[1]: *** [drivers/crypto] Error 2 > make[1]: *** Waiting for unfinished jobs.... > make: *** [drivers] Error 2 > make: *** Waiting for unfinished jobs.... > > Actually, <mach/dma.h> is including <plat/dma-pl330.h> for s5p64x0, > s5pc100, s5pv210 and exynos or <plat/dma.h> for s3c24xx and s3c64xx. And > the cause of build error you reported is duplicated declaration of > s3c2410_dma_client. So it can happen when <plat/dma-pl330.h> and > <plat/dma.h> are included together at the same time... > Of course, following can fix above build error similarly... --- diff --git a/drivers/crypto/s5p-sss.c b/drivers/crypto/s5p-sss.c index bc986f8..6a3e98e 100644 --- a/drivers/crypto/s5p-sss.c +++ b/drivers/crypto/s5p-sss.c @@ -30,7 +30,7 @@ #include <crypto/ctr.h> #include <plat/cpu.h> -#include <plat/dma.h> +#include <mach/dma.h> #define _SBF(s, v) ((v) << (s)) #define _BIT(b) _SBF(b, 1) --- Hmm...I need to sort this out... Thanks. Best regards, Kgene. -- Kukjin Kim <kgene.kim(a)samsung.com>, Senior Engineer, SW Solution Development Team, Samsung Electronics Co., Ltd.

1 1

[alsa-devel] ASoC: how to calculate dma size in tdm mode
by Scott Jiang 10 Aug '12

10 Aug '12

Hi Mark, If user app use 2 channels but codec need 8 channels in tdm mode, I need to copy this data from user space to dma buffer. I wonder how to calculate dma size, snd_pcm_hardware.buffer_bytes_max or buffer_bytes_max * tdm_channels / channels_min? Thanks, Scott

2 2

[alsa-devel] [asoc:for-next 82/87] sound/soc/soc-core.c:1817:6: warning: unused variable 'ret'
by Fengguang Wu 10 Aug '12

10 Aug '12

Hi Mark, FYI, there are new compile warnings show up in tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: cd1d5e1e60bb455bf28c0d460f6f3782e7c8d9a0 commit: 28d528c8dbab5c6b3cf2fc4f0e91470a9a63dbc0 [82/87] ASoC: core: Remove pointless error on card registration failure config: i386-randconfig-m007 (attached as .config) All error/warnings: sound/soc/soc-core.c: In function 'soc_probe': sound/soc/soc-core.c:1817:6: warning: unused variable 'ret' [-Wunused-variable] vim +1817 sound/soc/soc-core.c 1814 static int soc_probe(struct platform_device *pdev) 1815 { 1816 struct snd_soc_card *card = platform_get_drvdata(pdev); > 1817 int ret = 0; 1818 1819 /* 1820 * no card, so machine driver should be registering card --- 0-DAY kernel build testing backend Open Source Technology Centre Fengguang Wu <wfg(a)linux.intel.com> Intel Corporation

1 0

[alsa-devel] [PATCH 1/2] ASoC: ad1836: convert to use snd_soc_register_card
by Scott Jiang 10 Aug '12

10 Aug '12

Cpu dai and codec name are passed in through platform data. Signed-off-by: Scott Jiang <scott.jiang.linux(a)gmail.com> --- sound/soc/blackfin/bf5xx-ad1836.c | 73 +++++++++++++++++++------------------ 1 files changed, 37 insertions(+), 36 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index d542d40..16b9c9e 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -59,62 +59,63 @@ static struct snd_soc_ops bf5xx_ad1836_ops = { #define BF5XX_AD1836_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ SND_SOC_DAIFMT_CBM_CFM) -static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { - { - .name = "ad1836", - .stream_name = "AD1836", - .cpu_dai_name = "bfin-tdm.0", - .codec_dai_name = "ad1836-hifi", - .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "spi0.4", - .ops = &bf5xx_ad1836_ops, - .dai_fmt = BF5XX_AD1836_DAIFMT, - }, - { - .name = "ad1836", - .stream_name = "AD1836", - .cpu_dai_name = "bfin-tdm.1", - .codec_dai_name = "ad1836-hifi", - .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "spi0.4", - .ops = &bf5xx_ad1836_ops, - .dai_fmt = BF5XX_AD1836_DAIFMT, - }, +static struct snd_soc_dai_link bf5xx_ad1836_dai = { + .name = "ad1836", + .stream_name = "AD1836", + .codec_dai_name = "ad1836-hifi", + .platform_name = "bfin-tdm-pcm-audio", + .ops = &bf5xx_ad1836_ops, + .dai_fmt = BF5XX_AD1836_DAIFMT, }; static struct snd_soc_card bf5xx_ad1836 = { .name = "bfin-ad1836", .owner = THIS_MODULE, - .dai_link = &bf5xx_ad1836_dai[CONFIG_SND_BF5XX_SPORT_NUM], + .dai_link = &bf5xx_ad1836_dai, .num_links = 1, }; -static struct platform_device *bfxx_ad1836_snd_device; - -static int __init bf5xx_ad1836_init(void) +static __devinit int bf5xx_ad1836_driver_probe(struct platform_device *pdev) { + struct snd_soc_card *card = &bf5xx_ad1836; + const char **link_name; int ret; - bfxx_ad1836_snd_device = platform_device_alloc("soc-audio", -1); - if (!bfxx_ad1836_snd_device) - return -ENOMEM; + link_name = pdev->dev.platform_data; + if (!link_name) { + dev_err(&pdev->dev, "No platform data supplied\n"); + return -EINVAL; + } + bf5xx_ad1836_dai.cpu_dai_name = link_name[0]; + bf5xx_ad1836_dai.codec_name = link_name[1]; - platform_set_drvdata(bfxx_ad1836_snd_device, &bf5xx_ad1836); - ret = platform_device_add(bfxx_ad1836_snd_device); + card->dev = &pdev->dev; + platform_set_drvdata(pdev, card); + ret = snd_soc_register_card(card); if (ret) - platform_device_put(bfxx_ad1836_snd_device); - + dev_err(&pdev->dev, "Failed to register card\n"); return ret; } -static void __exit bf5xx_ad1836_exit(void) +static int __devexit bf5xx_ad1836_driver_remove(struct platform_device *pdev) { - platform_device_unregister(bfxx_ad1836_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + return 0; } -module_init(bf5xx_ad1836_init); -module_exit(bf5xx_ad1836_exit); +static struct platform_driver bf5xx_ad1836_driver = { + .driver = { + .name = "bfin-snd-ad1836", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = bf5xx_ad1836_driver_probe, + .remove = __devexit_p(bf5xx_ad1836_driver_remove), +}; +module_platform_driver(bf5xx_ad1836_driver); /* Module information */ MODULE_AUTHOR("Barry Song"); -- 1.7.0.4

2 3

Re: [alsa-devel] XMOS UAC2
by Daniel Mack 09 Aug '12

09 Aug '12

(Cc linux-usb) On 09.08.2012 21:59, adelias(a)freemail.gr wrote: > I've attached the dmesg of each. > > On 9/8/2012 10:45 μμ, Daniel Mack wrote: >> > On 09.08.2012 21:33, adelias wrote: >>> >> I have an XMOS Reference board with the latest firmware (v3.3). With >>> >> ALSA 1.0.23 it enumerates and functions properly. With ALSA 1.0.25 it >>> >> appears to enumerate properly after boot up but will not output sound. >>> >> Removing and reinserting snd-usb-audio with modprobe doesn't help. Only >>> >> after I reset it by unpluging and repluging the usb cable will it >>> >> function as it should. >> > >> > Could you provide the full boot logs of both scenarios please? > > dmesg.1.0.24 Thanks. To summarize, you're not only using two different versions of ALSA but two versions of the kernel (2.6.37 vs 3.0.36+), and the problem is not caused by ALSA but by the USB host controller driver (see below). Is there any reason why you can't update to a more recent kernel (3.5)? Daniel > [ 0.000000] Initializing cgroup subsys cpuset > [ 0.000000] Initializing cgroup subsys cpu > [ 0.000000] Linux version 3.0.36+ (gnexus(a)a10linux.org) (gcc version 4.6.1 (Sourcery CodeBench Lite 2011.09-70) ) #34 PREEMPT Sat Jul 21 11:34:33 IST 2012 > [ 0.000000] CPU: ARMv7 Processor [413fc082] revision 2 (ARMv7), cr=10c5387d > [ 0.000000] CPU: VIPT nonaliasing data cache, VIPT aliasing instruction cache > [ 0.000000] Machine: sun4i > [ 0.000000] Total Detected Memory: 512MB with 1 banks > [ 0.000000] Ignoring unrecognised tag 0x00000000 > [ 0.000000] Memory Reserved(in bytes): > [ 0.000000] SYS: 0x43000000, 0x00010000 > [ 0.000000] LCD: 0x5a000000, 0x02000000 > [ 0.000000] G2D: 0x58000000, 0x01000000 > [ 0.000000] VE : 0x44000000, 0x05000000 > [ 0.000000] Memory policy: ECC disabled, Data cache writeback > [ 0.000000] On node 0 totalpages: 114688 > [ 0.000000] free_area_init_node: node 0, pgdat c054f6bc, node_mem_map c0604000 > [ 0.000000] Normal zone: 896 pages used for memmap > [ 0.000000] Normal zone: 0 pages reserved > [ 0.000000] Normal zone: 113792 pages, LIFO batch:31 > [ 0.000000] pcpu-alloc: s0 r0 d32768 u32768 alloc=1*32768 > [ 0.000000] pcpu-alloc: [0] 0 > [ 0.000000] Built 1 zonelists in Zone order, mobility grouping on. Total pages: 113792 > [ 0.000000] Kernel command line: console=ttyS0,115200 root=/dev/mmcblk0p2 init=/init loglevel=8 rootfstype=ext4 rootwait > [ 0.000000] PID hash table entries: 2048 (order: 1, 8192 bytes) > [ 0.000000] Dentry cache hash table entries: 65536 (order: 6, 262144 bytes) > [ 0.000000] Inode-cache hash table entries: 32768 (order: 5, 131072 bytes) > [ 0.000000] allocated 1835008 bytes of page_cgroup > [ 0.000000] please try 'cgroup_disable=memory' option if you don't want memory cgroups > [ 0.000000] Memory: 448MB = 448MB total > [ 0.000000] Memory: 315636k/315636k available, 143116k reserved, 0K highmem > [ 0.000000] Virtual kernel memory layout: > [ 0.000000] vector : 0xffff0000 - 0xffff1000 ( 4 kB) > [ 0.000000] fixmap : 0xfff00000 - 0xfffe0000 ( 896 kB) > [ 0.000000] DMA : 0xffc00000 - 0xffe00000 ( 2 MB) > [ 0.000000] vmalloc : 0xdc800000 - 0xf0000000 ( 312 MB) > [ 0.000000] lowmem : 0xc0000000 - 0xdc000000 ( 448 MB) > [ 0.000000] pkmap : 0xbfe00000 - 0xc0000000 ( 2 MB) > [ 0.000000] modules : 0xbf000000 - 0xbfe00000 ( 14 MB) > [ 0.000000] .init : 0xc0008000 - 0xc002c000 ( 144 kB) > [ 0.000000] .text : 0xc002c000 - 0xc0517850 (5039 kB) > [ 0.000000] .data : 0xc0518000 - 0xc0550040 ( 225 kB) > [ 0.000000] .bss : 0xc0550064 - 0xc0603630 ( 718 kB) > [ 0.000000] SLUB: Genslabs=11, HWalign=64, Order=0-3, MinObjects=0, CPUs=1, Nodes=1 > [ 0.000000] NR_IRQS:96 nr_irqs:96 96 > [ 0.000000] timer0: Periodic Mode > [ 0.000000] Console: colour dummy device 80x30 > [ 0.000000] Calibrating delay loop... 1001.88 BogoMIPS (lpj=5009408) > [ 0.040000] pid_max: default: 32768 minimum: 301 > [ 0.040000] Security Framework initialized > [ 0.040000] Mount-cache hash table entries: 512 > [ 0.050000] Initializing cgroup subsys cpuacct > [ 0.050000] Initializing cgroup subsys memory > [ 0.060000] Initializing cgroup subsys devices > [ 0.060000] Initializing cgroup subsys freezer > [ 0.060000] Initializing cgroup subsys blkio > [ 0.070000] Initializing cgroup subsys perf_event > [ 0.070000] CPU: Testing write buffer coherency: ok > [ 0.070000] hw perfevents: enabled with ARMv7 Cortex-A8 PMU driver, 5 counters available > [ 0.080000] devtmpfs: initialized > [ 0.080000] print_constraints: dummy: > [ 0.090000] NET: Registered protocol family 16 > [ 0.090000] hw-breakpoint: debug architecture 0x4 unsupported. > [ 0.100000] SOFTWINNER DMA Driver, (c) 2003-2004,2006 Simtec Electronics > [ 0.100000] Initialize DMAC OK > [ 0.100000] bio: create slab <bio-0> at 0 > [ 0.110000] SCSI subsystem initialized > [ 0.110000] usbcore: registered new interface driver usbfs > [ 0.120000] usbcore: registered new interface driver hub > [ 0.120000] usbcore: registered new device driver usb > [ 0.120000] Advanced Linux Sound Architecture Driver Version 1.0.24. > [ 0.130000] Init eGon pin module V2.0 > [ 0.130000] Switching to clocksource aw 64bits couter > [ 0.140000] FS-Cache: Loaded > [ 0.140000] CacheFiles: Loaded > [ 0.150000] Switched to NOHz mode on CPU #0 > [ 0.150000] [usb_manager]: CONFIG_USB_SW_SUN4I_USB0_OTG > [ 0.150000] [sw_hcd0]: usb host driver initialize........ > [ 0.160000] [sw_udc]: udc_init: version 20080411 > [ 0.160000] [sw_udc]: usb_vbase = 0xf1c13000 > [ 0.160000] [sw_udc]: sram_vbase = 0xf1c00000 > [ 0.170000] [sw_udc]: open_usb_clock > [ 0.190000] udc->driver is null, udc is need not stop > [ 0.190000] NET: Registered protocol family 2 > [ 0.200000] IP route cache hash table entries: 4096 (order: 2, 16384 bytes) > [ 0.200000] TCP established hash table entries: 16384 (order: 5, 131072 bytes) > [ 0.210000] TCP bind hash table entries: 16384 (order: 4, 65536 bytes) > [ 0.210000] TCP: Hash tables configured (established 16384 bind 16384) > [ 0.220000] TCP reno registered > [ 0.220000] UDP hash table entries: 256 (order: 0, 4096 bytes) > [ 0.230000] UDP-Lite hash table entries: 256 (order: 0, 4096 bytes) > [ 0.230000] NET: Registered protocol family 1 > [ 0.240000] [pm]aw_pm_init! > [ 0.240000] audit: initializing netlink socket (disabled) > [ 0.240000] type=2000 audit(0.239:1): initialized > [ 0.260000] VFS: Disk quotas dquot_6.5.2 > [ 0.260000] Dquot-cache hash table entries: 1024 (order 0, 4096 bytes) > [ 0.270000] msgmni has been set to 616 > [ 0.270000] alg: No test for stdrng (krng) > [ 0.270000] Block layer SCSI generic (bsg) driver version 0.4 loaded (major 253) > [ 0.280000] io scheduler noop registered > [ 0.280000] io scheduler deadline registered > [ 0.290000] io scheduler cfq registered (default) > [ 0.290000] Serial: 8250/16550 driver, 8 ports, IRQ sharing disabled > [ 0.320000] serial8250.0: ttyS0 at MMIO 0x1c28000 (irq = 1) is a U6_16550A > [ 0.860000] console [ttyS0] enabled > [ 0.870000] [uart]: used uart info.: 0x05 > [ 0.900000] sunxi-uart.0: ttyS0 at MMIO 0x1c28000 (irq = 1) is a U6_16550A > [ 0.910000] [uart]: serial probe 0, membase (null) irq 1 mapbase 0x01c28000 > [ 0.940000] sunxi-uart.2: ttyS1 at MMIO 0x1c28800 (irq = 3) is a U6_16550A > [ 0.960000] [uart]: serial probe 2, membase (null) irq 3 mapbase 0x01c28800 > [ 0.970000] G2D: drv_g2d_init > [ 0.970000] G2D: g2dmem: g2d_start=58000000, g2d_size=1000000 > [ 0.980000] G2D: head:d8000000,tail:d9000000 > [ 0.990000] G2D: Module initialized.major:251 > [ 1.000000] [drm] Initialized drm 1.1.0 20060810 > [ 1.010000] brd: module loaded > [ 1.010000] [NAND]nand driver, init. > [ 1.020000] cmu_clk is 360 > [ 1.020000] nand clk init end > [ 1.030000] offset 0xc: 0x6141 > [ 1.030000] offset 0x14: 0x82000008 > [ 1.040000] [NAND] nand gpio_request > [ 1.050000] [NAND] nand driver version: 0x2 0x9 > [ 1.050000] nand interrupte register ok > [ 1.060000] ret of NFC_ChangMode is 0 > [ 1.070000] dma_hdle is 0 > [ 1.070000] dma_hdle is 10000008 > [ 1.080000] cmu_clk is 360 > [ 1.080000] nand clk init end > [ 1.080000] offset 0xc: 0x6141 > [ 1.090000] offset 0x14: 0x82000005 > [ 1.110000] The 0 disk name = bootloader, class name = DISK, disk size = 32768 > [ 1.130000] The 1 disk name = env, class name = DISK, disk size = 4096 > [ 1.140000] The 2 disk name = boot, class name = DISK, disk size = 65536 > [ 1.150000] The 3 disk name = system, class name = DISK, disk size = 524288 > [ 1.160000] The 4 disk name = data, class name = DISK, disk size = 524288 > [ 1.180000] The 5 disk name = misc, class name = DISK, disk size = 2048 > [ 1.190000] The 6 disk name = recovery, class name = DISK, disk size = 65536 > [ 1.200000] The 7 disk name = cache, class name = DISK, disk size = 262144 > [ 1.210000] The 8 disk name = private, class name = DISK, disk size = 2048 > [ 1.230000] The 9 disk name = sysrecovery, class name = DISK, disk size = 655360 > [ 1.240000] The 10 disk name = UDISK, class name = DISK, disk size = 0 > [ 1.250000] The 10 disk size = 5642240 > [ 1.260000] part total count = 11 > [ 1.260000] nanda: > [ 1.270000] nandb: unknown partition table > [ 1.280000] nandc: unknown partition table > [ 1.290000] nandd: unknown partition table > [ 1.300000] nande: unknown partition table > [ 1.300000] nandf: unknown partition table > [ 1.310000] nandg: unknown partition table > [ 1.320000] nandh: unknown partition table > [ 1.330000] nandi: > [ 1.340000] nandj: unknown partition table > [ 1.340000] nandk: > [ 1.350000] benn: nand probe enter > [ 1.350000] [NAND]nand driver, ok. > [ 1.360000] ehci_hcd: USB 2.0 'Enhanced' Host Controller (EHCI) Driver > [ 1.370000] ohci_hcd: USB 1.1 'Open' Host Controller (OHCI) Driver > [ 1.380000] [sw-ehci1]: probe, pdev->name: sw-ehci, pdev->id: 1, sw_ehci: 0xc05fe5dc > [ 1.400000] [sw-ehci1]: open clock > [ 1.420000] [sw-ehci1]: Set USB Power ON > [ 1.430000] sw-ehci sw-ehci.1: SW USB2.0 'Enhanced' Host Controller (EHCI) Driver > [ 1.440000] sw-ehci sw-ehci.1: new USB bus registered, assigned bus number 1 > [ 1.460000] sw-ehci sw-ehci.1: irq 39, io mem 0xf1c14000 > [ 1.490000] sw-ehci sw-ehci.1: USB 0.0 started, EHCI 1.00 > [ 1.490000] ehci_irq: port change detect > [ 1.500000] usb usb1: New USB device found, idVendor=1d6b, idProduct=0002 > [ 1.510000] usb usb1: New USB device strings: Mfr=3, Product=2, SerialNumber=1 > [ 1.530000] usb usb1: Product: SW USB2.0 'Enhanced' Host Controller (EHCI) Driver > [ 1.540000] usb usb1: Manufacturer: Linux 3.0.36+ ehci_hcd > [ 1.550000] usb usb1: SerialNumber: sw-ehci > [ 1.560000] hub 1-0:1.0: USB hub found > [ 1.570000] hub 1-0:1.0: 1 port detected > [ 1.570000] [sw-ohci1]: probe, pdev->name: sw-ohci, pdev->id: 1, sw_ohci: 0xc05fe6ec > [ 1.590000] [sw-ohci1]: open clock > [ 1.610000] sw-ohci sw-ohci.1: SW USB2.0 'Open' Host Controller (OHCI) Driver > [ 1.630000] sw-ohci sw-ohci.1: new USB bus registered, assigned bus number 2 > [ 1.640000] sw-ohci sw-ohci.1: irq 64, io mem 0xf1c14400 > [ 1.710000] usb usb2: New USB device found, idVendor=1d6b, idProduct=0001 > [ 1.720000] usb usb2: New USB device strings: Mfr=3, Product=2, SerialNumber=1 > [ 1.730000] usb usb2: Product: SW USB2.0 'Open' Host Controller (OHCI) Driver > [ 1.750000] usb usb2: Manufacturer: Linux 3.0.36+ ohci_hcd > [ 1.760000] usb usb2: SerialNumber: sw-ohci > [ 1.770000] hub 2-0:1.0: USB hub found > [ 1.770000] hub 2-0:1.0: 1 port detected > [ 1.780000] [sw-ehci2]: probe, pdev->name: sw-ehci, pdev->id: 2, sw_ehci: 0xc05fe800 > [ 1.790000] [sw-ehci2]: open clock > [ 1.820000] [sw-ehci2]: Set USB Power ON > [ 1.830000] sw-ehci sw-ehci.2: SW USB2.0 'Enhanced' Host Controller (EHCI) Driver > [ 1.840000] sw-ehci sw-ehci.2: new USB bus registered, assigned bus number 3 > [ 1.850000] sw-ehci sw-ehci.2: irq 40, io mem 0xf1c1c000 > [ 1.890000] usb 1-1: new high speed USB device number 2 using sw-ehci > [ 1.920000] sw-ehci sw-ehci.2: USB 0.0 started, EHCI 1.00 > [ 1.920000] ehci_irq: port change detect > [ 1.930000] usb usb3: New USB device found, idVendor=1d6b, idProduct=0002 > [ 1.940000] usb usb3: New USB device strings: Mfr=3, Product=2, SerialNumber=1 > [ 1.960000] usb usb3: Product: SW USB2.0 'Enhanced' Host Controller (EHCI) Driver > [ 47.380000] usb usb3: Manufacturer: Linux 3.0.36+ ehci_hcd > [ 47.390000] usb usb3: SerialNumber: sw-ehci > [ 47.400000] hub 3-0:1.0: USB hub found > [ 47.410000] hub 3-0:1.0: 1 port detected > [ 47.410000] [sw-ehci2]: sw_usb_disable_ehci > [ 47.420000] [sw-ehci2]: remove, pdev->name: sw-ehci, pdev->id: 2, sw_ehci: 0xc05fe800 > [ 47.440000] sw-ehci sw-ehci.2: remove, state 1 > [ 47.440000] usb usb3: USB disconnect, device number 1 > [ 47.450000] BUG: scheduling while atomic: swapper/1/0x40000002 > [ 47.460000] Modules linked in: > [ 47.470000] [<c0036e80>] (unwind_backtrace+0x0/0xe0) from [<c03c263c>] (__schedule+0x5c/0x4ec) > [ 47.480000] [<c03c263c>] (__schedule+0x5c/0x4ec) from [<c00517fc>] (__cond_resched+0x14/0x20) > [ 47.500000] [<c00517fc>] (__cond_resched+0x14/0x20) from [<c03c2b64>] (_cond_resched+0x34/0x44) > [ 47.520000] [<c03c2b64>] (_cond_resched+0x34/0x44) from [<c0245794>] (wakeup_source_unregister+0xc/0x18) > [ 47.530000] [<c0245794>] (wakeup_source_unregister+0xc/0x18) from [<c0245964>] (device_wakeup_disable+0x64/0x78) > [ 47.550000] [<c0245964>] (device_wakeup_disable+0x64/0x78) from [<c02447ac>] (device_pm_remove+0x48/0x58) > [ 47.570000] [<c02447ac>] (device_pm_remove+0x48/0x58) from [<c023d144>] (device_del+0x34/0x168) > [ 47.590000] [<c023d144>] (device_del+0x34/0x168) from [<c02847f4>] (usb_disconnect+0x9c/0xf8) > [ 47.600000] [<c02847f4>] (usb_disconnect+0x9c/0xf8) from [<c0287244>] (usb_remove_hcd+0xb8/0x174) > [ 47.620000] [<c0287244>] (usb_remove_hcd+0xb8/0x174) from [<c0295bbc>] (sw_ehci_hcd_remove+0x9c/0x104) > [ 47.640000] [<c0295bbc>] (sw_ehci_hcd_remove+0x9c/0x104) from [<c0295cec>] (sw_usb_disable_ehci+0xc8/0xf0) > [ 47.650000] [<c0295cec>] (sw_usb_disable_ehci+0xc8/0xf0) from [<c0295ed0>] (sw_ehci_hcd_probe+0x1bc/0x228) > [ 47.670000] [<c0295ed0>] (sw_ehci_hcd_probe+0x1bc/0x228) from [<c02409c4>] (platform_drv_probe+0x14/0x18) > [ 47.690000] [<c02409c4>] (platform_drv_probe+0x14/0x18) from [<c023f8a8>] (driver_probe_device+0xc8/0x184) > [ 47.710000] [<c023f8a8>] (driver_probe_device+0xc8/0x184) from [<c023e86c>] (bus_for_each_drv+0x48/0x80) > [ 47.730000] [<c023e86c>] (bus_for_each_drv+0x48/0x80) from [<c023f700>] (device_attach+0x74/0xa0) > [ 47.740000] [<c023f700>] (device_attach+0x74/0xa0) from [<c023f078>] (bus_probe_device+0x24/0x40) > [ 47.760000] [<c023f078>] (bus_probe_device+0x24/0x40) from [<c023d884>] (device_add+0x39c/0x528) > [ 47.780000] [<c023d884>] (device_add+0x39c/0x528) from [<c0240fb0>] (platform_device_add+0x108/0x164) > [ 47.790000] [<c0240fb0>] (platform_device_add+0x108/0x164) from [<c001e6c0>] (sw_hci_sun4i_init+0x154/0x1f0) > [ 47.810000] [<c001e6c0>] (sw_hci_sun4i_init+0x154/0x1f0) from [<c002c408>] (do_one_initcall+0x90/0x160) > [ 47.830000] [<c002c408>] (do_one_initcall+0x90/0x160) from [<c0008a90>] (kernel_init+0xb0/0x15c) > [ 47.840000] [<c0008a90>] (kernel_init+0xb0/0x15c) from [<c0032598>] (kernel_thread_exit+0x0/0x8) > [ 47.860000] sw-ehci sw-ehci.2: USB bus 3 deregistered Well, this is your problem appearantly, and it has nothing to do with ALSA. Not sure what causes this and whether this was a known problem in that 3.0.36+ kernel that you're using. > [ 47.870000] [sw-ehci2]: Set USB Power OFF > [ 47.880000] [sw-ehci2]: close clock > [ 47.880000] [sw-ohci2]: probe, pdev->name: sw-ohci, pdev->id: 2, sw_ohci: 0xc05fe910 > [ 47.900000] [sw-ohci2]: open clock > [ 47.920000] [sw-ohci2]: Set USB Power ON > [ 47.930000] sw-ohci sw-ohci.2: SW USB2.0 'Open' Host Controller (OHCI) Driver > [ 47.940000] sw-ohci sw-ohci.2: new USB bus registered, assigned bus number 3 > [ 47.960000] sw-ohci sw-ohci.2: irq 65, io mem 0xf1c1c400 > [ 47.970000] BUG: scheduling while atomic: swapper/1/0x00000002 > [ 47.980000] Modules linked in: > [ 47.980000] [<c0036e80>] (unwind_backtrace+0x0/0xe0) from [<c03c263c>] (__schedule+0x5c/0x4ec) > [ 48.000000] [<c03c263c>] (__schedule+0x5c/0x4ec) from [<c03c3560>] (schedule_timeout+0x1d0/0x20c) > [ 48.010000] [<c03c3560>] (schedule_timeout+0x1d0/0x20c) from [<c00659e4>] (msleep+0x14/0x20) > [ 48.030000] [<c00659e4>] (msleep+0x14/0x20) from [<c029c91c>] (ohci_run+0xc0/0x434) > [ 48.040000] [<c029c91c>] (ohci_run+0xc0/0x434) from [<c03bc290>] (sw_ohci_start+0x24/0x68) > [ 48.060000] [<c03bc290>] (sw_ohci_start+0x24/0x68) from [<c0287fe4>] (usb_add_hcd+0x2cc/0x60c) > [ 48.070000] [<c0287fe4>] (usb_add_hcd+0x2cc/0x60c) from [<c029cfcc>] (sw_ohci_hcd_probe+0x148/0x220) > [ 48.090000] [<c029cfcc>] (sw_ohci_hcd_probe+0x148/0x220) from [<c02409c4>] (platform_drv_probe+0x14/0x18) > [ 48.110000] [<c02409c4>] (platform_drv_probe+0x14/0x18) from [<c023f8a8>] (driver_probe_device+0xc8/0x184) > [ 48.130000] [<c023f8a8>] (driver_probe_device+0xc8/0x184) from [<c023e86c>] (bus_for_each_drv+0x48/0x80) > [ 48.140000] [<c023e86c>] (bus_for_each_drv+0x48/0x80) from [<c023f700>] (device_attach+0x74/0xa0) > [ 48.160000] [<c023f700>] (device_attach+0x74/0xa0) from [<c023f078>] (bus_probe_device+0x24/0x40) > [ 48.180000] [<c023f078>] (bus_probe_device+0x24/0x40) from [<c023d884>] (device_add+0x39c/0x528) > [ 48.190000] [<c023d884>] (device_add+0x39c/0x528) from [<c0240fb0>] (platform_device_add+0x108/0x164) > [ 48.210000] [<c0240fb0>] (platform_device_add+0x108/0x164) from [<c001e6fc>] (sw_hci_sun4i_init+0x190/0x1f0) > [ 48.230000] [<c001e6fc>] (sw_hci_sun4i_init+0x190/0x1f0) from [<c002c408>] (do_one_initcall+0x90/0x160) > [ 48.250000] [<c002c408>] (do_one_initcall+0x90/0x160) from [<c0008a90>] (kernel_init+0xb0/0x15c) > [ 48.260000] [<c0008a90>] (kernel_init+0xb0/0x15c) from [<c0032598>] (kernel_thread_exit+0x0/0x8) > [ 48.280000] usb usb3: New USB device found, idVendor=1d6b, idProduct=0001 > [ 48.300000] usb usb3: New USB device strings: Mfr=3, Product=2, SerialNumber=1 > [ 48.310000] usb usb3: Product: SW USB2.0 'Open' Host Controller (OHCI) Driver > [ 48.320000] usb usb3: Manufacturer: Linux 3.0.36+ ohci_hcd > [ 48.330000] usb usb3: SerialNumber: sw-ohci > [ 48.340000] hub 3-0:1.0: USB hub found > [ 48.350000] hub 3-0:1.0: 1 port detected > [ 48.350000] [sw-ohci2]: sw_usb_disable_ohci > [ 48.360000] [sw-ohci2]: remove, pdev->name: sw-ohci, pdev->id: 2, sw_ohci: 0xc05fe910 > [ 48.380000] sw-ohci sw-ohci.2: remove, state 1 > [ 48.380000] usb usb3: USB disconnect, device number 1 > [ 48.390000] BUG: scheduling while atomic: swapper/1/0x40000002 > [ 48.400000] Modules linked in: > [ 48.410000] [<c0036e80>] (unwind_backtrace+0x0/0xe0) from [<c03c263c>] (__schedule+0x5c/0x4ec) > [ 48.420000] [<c03c263c>] (__schedule+0x5c/0x4ec) from [<c00517fc>] (__cond_resched+0x14/0x20) > [ 48.440000] [<c00517fc>] (__cond_resched+0x14/0x20) from [<c03c2b64>] (_cond_resched+0x34/0x44) > [ 48.460000] [<c03c2b64>] (_cond_resched+0x34/0x44) from [<c0245794>] (wakeup_source_unregister+0xc/0x18) > [ 48.470000] [<c0245794>] (wakeup_source_unregister+0xc/0x18) from [<c0245964>] (device_wakeup_disable+0x64/0x78) > [ 48.490000] [<c0245964>] (device_wakeup_disable+0x64/0x78) from [<c02447ac>] (device_pm_remove+0x48/0x58) > [ 48.510000] [<c02447ac>] (device_pm_remove+0x48/0x58) from [<c023d144>] (device_del+0x34/0x168) > [ 48.530000] [<c023d144>] (device_del+0x34/0x168) from [<c02847f4>] (usb_disconnect+0x9c/0xf8) > [ 48.540000] [<c02847f4>] (usb_disconnect+0x9c/0xf8) from [<c0287244>] (usb_remove_hcd+0xb8/0x174) > [ 48.560000] [<c0287244>] (usb_remove_hcd+0xb8/0x174) from [<c029cd30>] (sw_ohci_hcd_remove+0xa0/0x104) > [ 48.580000] [<c029cd30>] (sw_ohci_hcd_remove+0xa0/0x104) from [<c029ce5c>] (sw_usb_disable_ohci+0xc8/0xf0) > [ 48.590000] [<c029ce5c>] (sw_usb_disable_ohci+0xc8/0xf0) from [<c029d034>] (sw_ohci_hcd_probe+0x1b0/0x220) > [ 48.610000] [<c029d034>] (sw_ohci_hcd_probe+0x1b0/0x220) from [<c02409c4>] (platform_drv_probe+0x14/0x18) > [ 48.630000] [<c02409c4>] (platform_drv_probe+0x14/0x18) from [<c023f8a8>] (driver_probe_device+0xc8/0x184) > [ 48.650000] [<c023f8a8>] (driver_probe_device+0xc8/0x184) from [<c023e86c>] (bus_for_each_drv+0x48/0x80) > [ 48.670000] [<c023e86c>] (bus_for_each_drv+0x48/0x80) from [<c023f700>] (device_attach+0x74/0xa0) > [ 48.680000] [<c023f700>] (device_attach+0x74/0xa0) from [<c023f078>] (bus_probe_device+0x24/0x40) > [ 48.700000] [<c023f078>] (bus_probe_device+0x24/0x40) from [<c023d884>] (device_add+0x39c/0x528) > [ 48.710000] [<c023d884>] (device_add+0x39c/0x528) from [<c0240fb0>] (platform_device_add+0x108/0x164) > [ 48.730000] [<c0240fb0>] (platform_device_add+0x108/0x164) from [<c001e6fc>] (sw_hci_sun4i_init+0x190/0x1f0) > [ 48.750000] [<c001e6fc>] (sw_hci_sun4i_init+0x190/0x1f0) from [<c002c408>] (do_one_initcall+0x90/0x160) > [ 48.770000] [<c002c408>] (do_one_initcall+0x90/0x160) from [<c0008a90>] (kernel_init+0xb0/0x15c) > [ 48.780000] [<c0008a90>] (kernel_init+0xb0/0x15c) from [<c0032598>] (kernel_thread_exit+0x0/0x8) > [ 48.800000] sw-ohci sw-ohci.2: USB bus 3 deregistered > [ 48.810000] [sw-ohci2]: Set USB Power OFF > [ 48.820000] [sw-ohci2]: close clock > [ 48.820000] initcall sw_hci_sun4i_init+0x0/0x1f0 returned with preemption imbalance > [ 48.840000] mousedev: PS/2 mouse device common for all mice > [ 48.850000] sunxi RTC version 0.1 > [ 48.850000] sunxi-rtc sunxi-rtc: sunxi_rtc_probe tmp_data = 380239881 > [ 48.860000] sunxi-rtc sunxi-rtc: sunxi_rtc_gettime > [ 48.870000] usb 1-1: New USB device found, idVendor=05e3, idProduct=0608 > [ 48.890000] sunxi-rtc sunxi-rtc: read time 2010-1-1 2:0:12 > [ 48.900000] using rtc device, rtc, for alarms > [ 48.900000] sunxi-rtc sunxi-rtc: rtc core: registered rtc as rtc0 > [ 48.910000] usb 1-1: New USB device strings: Mfr=0, Product=1, SerialNumber=0 > [ 48.930000] i2c /dev entries driver > [ 48.930000] !!! base_Addr = 0xdc84ac00 > [ 48.940000] config i2c gpio with gpio_config api > [ 48.950000] usb 1-1: Product: USB2.0 Hub > [ 48.960000] twi0, apb clock = 24000000 > [ 48.960000] hub 1-1:1.0: USB hub found > [ 48.970000] hub 1-1:1.0: 4 ports detected > [ 48.980000] axp_mfd 0-0034: AXP (CHIP ID: 0x21) detected > [ 48.990000] I2C: i2c-0: AW16XX I2C adapter > [ 48.990000] **********start************ > [ 49.000000] 0x40 > [ 49.000000] 0xf8 > [ 49.010000] 0x28 > [ 49.010000] 0x0 > [ 49.010000] 0x0 > [ 49.010000] **********end************ > [ 49.020000] !!! base_Addr = 0xdc84c000 > [ 49.030000] twi1, apb clock = 24000000 > [ 49.030000] I2C: i2c-1: AW16XX I2C adapter > [ 49.040000] **********start************ > [ 49.050000] 0x40 > [ 49.050000] 0xf8 > [ 49.050000] 0x58 > [ 49.050000] 0x0 > [ 49.060000] 0x0 > [ 49.060000] **********end************ > [ 49.060000] !!! base_Addr = 0xdc84e400 > [ 49.070000] twi2, apb clock = 24000000 > [ 49.080000] I2C: i2c-2: AW16XX I2C adapter > [ 49.080000] **********start************ > [ 49.090000] 0x40 > [ 49.090000] 0xf8 > [ 49.100000] 0x58 > [ 49.100000] 0x0 > [ 49.100000] 0x0 > [ 49.100000] **********end************ > [ 49.110000] Linux media interface: v0.10 > [ 49.120000] [cedar dev]: install start!!! > [ 49.120000] [cedar dev]: install end!!! > [ 49.130000] print_constraints: axp20_ldo1: 1300 mV > [ 49.140000] print_constraints: axp20_ldo2: 1800 <--> 3300 mV at 3000 mV > [ 49.150000] print_constraints: axp20_ldo3: 700 <--> 3500 mV at 2800 mV > [ 49.170000] print_constraints: axp20_ldo4: 1250 <--> 3300 mV at 2800 mV > [ 49.180000] print_constraints: axp20_buck2: 700 <--> 2275 mV at 1400 mV > [ 49.190000] print_constraints: axp20_buck3: 700 <--> 3500 mV at 1250 mV > [ 49.200000] print_constraints: axp20_ldoio0: 1800 <--> 3300 mV at 2800 mV > [ 49.220000] input: axp20-supplyer as /devices/platform/sun4i-i2c.0/i2c-0/0-0034/axp20-supplyer.28/input/input0 > [ 49.240000] [AXP]axp driver uning configuration failed(1579) > [ 49.250000] [AXP]pmu_suspendpwroff_vol = 3500 > [ 49.270000] cpuidle: using governor ladder > [ 49.280000] cpuidle: using governor menu > [ 49.280000] [mmc_pm]: no sdio card used in configuration > [ 49.290000] [mmc]: sunximmc_init > [ 49.300000] [mmc]: sunxi mmc controller using config : 0xb > [ 49.310000] [mmc]: sunxi-mmc.0: pdev->name: sunxi-mmc, pdev->id: 0 > [ 49.320000] [mmc]: mmc 0 power off !! > [ 49.320000] usb 1-1.2: new high speed USB device number 3 using sw-ehci > [ 49.340000] [mmc]: mmc0 Probe: base:0xdc854000 irq:32 dma:0 pdes:0xd9a04000, ret 0. > [ 49.350000] [mmc]: sunxi-mmc.1: pdev->name: sunxi-mmc, pdev->id: 1 > [ 49.360000] [mmc]: mmc 1 power off !! > [ 49.370000] [mmc]: mmc1 Probe: base:0xdc856000 irq:33 dma:0 pdes:0xd9a10000, ret 0. > [ 49.380000] [mmc]: sunxi-mmc.3: pdev->name: sunxi-mmc, pdev->id: 3 > [ 49.390000] [mmc]: mmc 3 power off !! > [ 49.400000] [mmc]: mmc3 Probe: base:0xdc858000 irq:35 dma:0 pdes:0xd9a14000, ret 0. > [ 49.410000] enter sun4i Audio codec!!! > [ 49.420000] sun4i audio support initialized > [ 49.430000] audiocodec_adap_awxx_init: script_parser_fetch err. > [ 49.440000] sun4i Audio codec successfully loaded.. > [ 49.450000] asoc: sndhdmi <-> sun4i-hdmiaudio.0 mapping ok > [ 49.460000] asoc: sndspdif <-> sun4i-spdif.0 mapping ok > [ 49.470000] [I2S]sun4i-i2s cannot find any using configuration for controllers, return directly! > [ 49.490000] [I2S]sndi2s cannot find any using configuration for controllers, return directly! > [ 49.500000] [I2S]sun4i_sndi2s cannot find any using configuration for controllers, return directly! > [ 49.520000] ALSA device list: > [ 49.530000] #0: sun4i-CODEC Audio Codec > [ 49.530000] #1: sun4i-sndhdmi > [ 49.540000] #2: sun4i-sndspdif > [ 49.540000] usb 1-1.2: New USB device found, idVendor=20b1, idProduct=0002 > [ 49.560000] TCP cubic registered > [ 49.560000] NET: Registered protocol family 10 > [ 49.570000] NET: Registered protocol family 17 > [ 49.580000] usb 1-1.2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 > [ 49.590000] Registering the dns_resolver key type > [ 49.600000] usb 1-1.2: Product: XMOS USB Audio 2.0 > [ 49.610000] VFP support v0.3: > [ 49.620000] usb 1-1.2: Manufacturer: XMOS > [ 49.620000] usb 1-1.2: SerialNumber: 0000 > [ 49.630000] implementor 41 architecture 3 part 30 variant c rev 3 > [ 49.640000] ThumbEE CPU extension supported. > [ 49.650000] Registering SWP/SWPB emulation handler > [ 49.660000] registered taskstats version 1 > [ 49.670000] [LCD] lcd_module_init > [ 49.730000] [DISP WRN] file:drivers/video/sun4i/disp/de_bsp/de/disp_lcd.c,line:592: lcd1_para.lcd_used not exit > [ 49.750000] regulator_init_complete: axp20_buck3: incomplete constraints, leaving on > [ 49.770000] regulator_init_complete: axp20_buck2: incomplete constraints, leaving on > [ 49.780000] regulator_init_complete: axp20_ldo4: incomplete constraints, leaving on > [ 49.800000] regulator_init_complete: axp20_ldo3: incomplete constraints, leaving on > [ 49.810000] regulator_init_complete: axp20_ldo2: incomplete constraints, leaving on > [ 49.830000] regulator_init_complete: axp20_ldo1: incomplete constraints, leaving on > [ 49.840000] sunxi-rtc sunxi-rtc: sunxi_rtc_gettime > [ 49.850000] sunxi-rtc sunxi-rtc: read time 2010-1-1 2:0:13 > [ 49.860000] sunxi-rtc sunxi-rtc: setting system clock to 2010-01-01 02:00:13 UTC (1262311213) > [ 49.880000] Waiting for root device /dev/mmcblk0p2... > [ 50.340000] [mmc]: mmc 0 detect change, present 1 > [ 50.640000] [mmc]: mmc 0 power on !! > [ 50.680000] [mmc]: sdxc_request_done(L1113): smc 0 err, cmd 52, RTO !! > [ 50.690000] [mmc]: sdxc_request_done(L1113): smc 0 err, cmd 52, RTO !! > [ 50.700000] [mmc]: sdxc_request_done(L1113): smc 0 err, cmd 5, RTO !! > [ 50.720000] [mmc]: sdxc_request_done(L1113): smc 0 err, cmd 5, RTO !! > [ 50.730000] [mmc]: sdxc_request_done(L1113): smc 0 err, cmd 5, RTO !! > [ 50.740000] [mmc]: sdxc_request_done(L1113): smc 0 err, cmd 5, RTO !! > [ 50.790000] mmc0: new high speed SDHC card at address b368 > [ 50.800000] mmcblk0: mmc0:b368 USD 7.46 GiB > [ 50.810000] mmcblk0: p1 p2 p3 > [ 50.880000] EXT4-fs (mmcblk0p2): mounted filesystem with ordered data mode. Opts: (null) > [ 50.890000] VFS: Mounted root (ext4 filesystem) on device 179:2. > [ 50.910000] devtmpfs: mounted > [ 50.910000] Freeing init memory: 144K > [ 50.920000] Failed to execute /init. Attempting defaults... > [ 52.020000] udevd[136]: starting version 175 > [ 54.290000] ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0xa00, type = 4 > [ 54.310000] ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0xa00, type = 4 > [ 54.330000] usbcore: registered new interface driver snd-usb-audio > [ 54.920000] Adding 262136k swap on /dev/mmcblk0p3. Priority:-1 extents:1 across:262136k SS > [ 56.840000] EXT4-fs (mmcblk0p2): re-mounted. Opts: (null) > [ 57.280000] UMP<2>: Inserting UMP device driver. Compiled: Jul 21 2012, time: 11:43:11 > [ 57.300000] UMP<2>: Using OS memory backend, allocation limit: 134217728 > [ 57.310000] UMP: UMP device driver loaded > [ 57.360000] mali: use config clk_div 3 > [ 57.360000] mali: clk_div 3 > [ 57.380000] Mali: mali clock set completed, clock is 320000000 Mhz > [ 57.390000] Mali: Mali device driver loaded > [ 57.460000] wemac Ethernet Driver, V1.00 in file:drivers/net/sun4i/sun4i_wemac.c > [ 57.490000] [EMAC] ahb clk enable > [ 57.490000] [EMAC] ahb gate clk: 0x26147 > [ 57.520000] wemac wemac.0: PHY SETUP, reg 0 value: 0 > [ 57.530000] wemac wemac.0: resetting device > [ 57.540000] release temp resource > [ 62.140000] wemac wemac.0: enabling eth0 > [ 62.140000] wemac wemac.0: WARNING: no IRQ resource flags set. > [ 62.150000] wemac wemac.0: resetting device > [ 62.160000] wemac wemac.0: eth0: link up, 100Mbps, full-duplex, lpa 0xFFFF > [ 67.460000] RPC: Registered named UNIX socket transport module. > [ 67.470000] RPC: Registered udp transport module. > [ 67.480000] RPC: Registered tcp transport module. > [ 67.490000] RPC: Registered tcp NFSv4.1 backchannel transport module. > [ 67.750000] FS-Cache: Netfs 'nfs' registered for caching > [ 67.870000] Installing knfsd (copyright (C) 1996 okir(a)monad.swb.de). > [ 72.600000] eth0: no IPv6 routers present > [ 81.520000] usbcore: deregistering interface driver snd-usb-audio > [ 82.050000] ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0xa00, type = 4 > [ 82.070000] ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0xa00, type = 4 > [ 82.100000] usbcore: registered new interface driver snd-usb-audio > [ 288.510000] usbcore: deregistering interface driver snd-usb-audio > [ 296.390000] ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0xa00, type = 4 > [ 296.420000] ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0xa00, type = 4 > [ 296.460000] usbcore: registered new interface driver snd-usb-audio > [ 384.700000] usb 1-1.2: USB disconnect, device number 3 > [ 392.090000] ehci_irq: port change detect > [ 392.350000] usb 1-1.2: new high speed USB device number 4 using sw-ehci > [ 392.470000] usb 1-1.2: New USB device found, idVendor=20b1, idProduct=0002 > [ 392.490000] usb 1-1.2: New USB device strings: Mfr=1, Product=2, SerialNumber=3 > [ 392.510000] usb 1-1.2: Product: XMOS USB Audio 2.0 > [ 392.510000] usb 1-1.2: Manufacturer: XMOS > [ 392.530000] usb 1-1.2: SerialNumber: 0000 > [ 393.010000] ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0xa00, type = 4 > [ 393.040000] ALSA sound/usb/mixer.c:342 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0xa00, type = 4 > > > dmesg.1.0.23 > > > Linux version 2.6.37-dockstar (root@debian) (gcc version 4.4.5 (Debian 4.4.5-4) ) #6 Mon Jan 17 01:55:44 CET 2011 > CPU: Feroceon 88FR131 [56251311] revision 1 (ARMv5TE), cr=00053977 > CPU: VIVT data cache, VIVT instruction cache > Machine: Seagate Dockstar Board > Memory policy: ECC disabled, Data cache writeback > On node 0 totalpages: 32768 > free_area_init_node: node 0, pgdat c0559fa4, node_mem_map c0588000 > Normal zone: 256 pages used for memmap > Normal zone: 0 pages reserved > Normal zone: 32512 pages, LIFO batch:7 > pcpu-alloc: s0 r0 d32768 u32768 alloc=1*32768 > pcpu-alloc: [0] 0 > Built 1 zonelists in Zone order, mobility grouping on. Total pages: 32512 > Kernel command line: console=ttyS0,115200 root=/dev/sda1 rootdelay=10 rootfstype=ext2 mtdparts=orion_nand:1M(u-boot),4M(uImage),32M(rootfs),-(data) > PID hash table entries: 512 (order: -1, 2048 bytes) > Dentry cache hash table entries: 16384 (order: 4, 65536 bytes) > Inode-cache hash table entries: 8192 (order: 3, 32768 bytes) > Memory: 128MB = 128MB total > Memory: 124248k/124248k available, 6824k reserved, 0K highmem > Virtual kernel memory layout: > vector : 0xffff0000 - 0xffff1000 ( 4 kB) > fixmap : 0xfff00000 - 0xfffe0000 ( 896 kB) > DMA : 0xffc00000 - 0xffe00000 ( 2 MB) > vmalloc : 0xc8800000 - 0xfe800000 ( 864 MB) > lowmem : 0xc0000000 - 0xc8000000 ( 128 MB) > modules : 0xbf000000 - 0xc0000000 ( 16 MB) > .init : 0xc0008000 - 0xc0030000 ( 160 kB) > .text : 0xc0030000 - 0xc0525000 (5076 kB) > .data : 0xc0526000 - 0xc055b680 ( 214 kB) > SLUB: Genslabs=13, HWalign=32, Order=0-3, MinObjects=0, CPUs=1, Nodes=1 > NR_IRQS:114 > Console: colour dummy device 80x30 > Calibrating delay loop... 1192.75 BogoMIPS (lpj=5963776) > pid_max: default: 32768 minimum: 301 > Mount-cache hash table entries: 512 > CPU: Testing write buffer coherency: ok > regulator: core version 0.5 > regulator: dummy: > NET: Registered protocol family 16 > i2c-core: driver [dummy] registered > Kirkwood: MV88F6281-A0, TCLK=200000000. > Feroceon L2: Cache support initialised. > initial MPP regs: 01111111 11113322 00001111 00100000 00000000 00000000 00000000 > final MPP regs: 01111111 11113322 00001111 00000000 00000000 00000000 00000000 > bio: create slab <bio-0> at 0 > vgaarb: loaded > i2c-core: driver [88PM860x] registered > SCSI subsystem initialized > usbcore: registered new interface driver usbfs > usbcore: registered new interface driver hub > usbcore: registered new device driver usb > Switching to clocksource orion_clocksource > FS-Cache: Loaded > NET: Registered protocol family 2 > IP route cache hash table entries: 1024 (order: 0, 4096 bytes) > TCP established hash table entries: 4096 (order: 3, 32768 bytes) > TCP bind hash table entries: 4096 (order: 2, 16384 bytes) > TCP: Hash tables configured (established 4096 bind 4096) > TCP reno registered > UDP hash table entries: 256 (order: 0, 4096 bytes) > UDP-Lite hash table entries: 256 (order: 0, 4096 bytes) > NET: Registered protocol family 1 > RPC: Registered udp transport module. > RPC: Registered tcp transport module. > RPC: Registered tcp NFSv4.1 backchannel transport module. > PCI: CLS 0 bytes, default 32 > FS-Cache: Netfs 'nfs' registered for caching > nfs4filelayout_init: NFSv4 File Layout Driver Registering... > Installing knfsd (copyright (C) 1996 okir(a)monad.swb.de). > fuse init (API version 7.15) > msgmni has been set to 242 > io scheduler noop registered > io scheduler deadline registered > io scheduler cfq registered (default) > Serial: 8250/16550 driver, 2 ports, IRQ sharing disabled > serial8250.0: ttyS0 at MMIO 0xf1012000 (irq = 33) is a 16550A > console [ttyS0] enabled > brd: module loaded > loop: module loaded > physmap-flash.0: failed to claim resource 0 > ONFI flash detected > ONFI param page 0 valid > NAND device: Manufacturer ID: 0x2c, Chip ID: 0xda (Micron NAND 256MiB 3,3V 8-bit) > Scanning device for bad blocks > Bad eraseblock 877 at 0x000006da0000 > Bad eraseblock 952 at 0x000007700000 > Bad eraseblock 1020 at 0x000007f80000 > 4 cmdlinepart partitions found on MTD device orion_nand > Creating 4 MTD partitions on "orion_nand": > 0x000000000000-0x000000100000 : "u-boot" > 0x000000100000-0x000000500000 : "uImage" > 0x000000500000-0x000002500000 : "rootfs" > 0x000002500000-0x000010000000 : "data" > MV-643xx 10/100/1000 ethernet driver version 1.4 > mv643xx_eth smi: probed > net eth0: port 0 with MAC address 00:10:75:1a:42:2a > ehci_hcd: USB 2.0 'Enhanced' Host Controller (EHCI) Driver > orion-ehci orion-ehci.0: Marvell Orion EHCI > orion-ehci orion-ehci.0: new USB bus registered, assigned bus number 1 > orion-ehci orion-ehci.0: irq 19, io mem 0xf1050000 > orion-ehci orion-ehci.0: USB 2.0 started, EHCI 1.00 > hub 1-0:1.0: USB hub found > hub 1-0:1.0: 1 port detected > ohci_hcd: USB 1.1 'Open' Host Controller (OHCI) Driver > uhci_hcd: USB Universal Host Controller Interface driver > Initializing USB Mass Storage driver... > usbcore: registered new interface driver usb-storage > USB Mass Storage support registered. > usbcore: registered new interface driver usbserial > USB Serial support registered for generic > usbcore: registered new interface driver usbserial_generic > usbserial: USB Serial Driver core > mice: PS/2 mouse device common for all mice > i2c /dev entries driver > i2c-core: driver [dev_driver] registered > lirc_dev: IR Remote Control driver registered, major 253 > IR NEC protocol handler initialized > IR RC5(x) protocol handler initialized > IR RC6 protocol handler initialized > IR JVC protocol handler initialized > IR Sony protocol handler initialized > IR RC5 (streamzap) protocol handler initialized > IR LIRC bridge handler initialized > Orion Watchdog Timer: Initial timeout 21 sec > cpuidle: using governor ladder > cpuidle: using governor menu > mv_xor_shared mv_xor_shared.0: Marvell shared XOR driver > mv_xor_shared mv_xor_shared.1: Marvell shared XOR driver > mv_xor mv_xor.0: Marvell XOR: ( xor cpy ) > mv_xor mv_xor.1: Marvell XOR: ( xor fill cpy ) > mv_xor mv_xor.2: Marvell XOR: ( xor cpy ) > mv_xor mv_xor.3: Marvell XOR: ( xor fill cpy ) > sdhci: Secure Digital Host Controller Interface driver > sdhci: Copyright(c) Pierre Ossman > mmc0: mvsdio driver initialized, lacking card detect (fall back to polling) > Registered led device: dockstar:green:health > Registered led device: dockstar:orange:health > Driver for HIFN 795x crypto accelerator chip has been successfully registered. > usbcore: registered new interface driver usbhid > usbhid: USB HID core driver > zram: num_devices not specified. Using default: 1 > zram: Creating 1 devices ... > oprofile: hardware counters not available > oprofile: using timer interrupt. > TCP cubic registered > NET: Registered protocol family 17 > Gating clock of unused units > before: 0x00dfc3fd > after: 0x00c701d9 > drivers/rtc/hctosys.c: unable to open rtc device (rtc0) > Waiting 10sec before mounting root device... > usb 1-1: new high speed USB device using orion-ehci and address 2 > hub 1-1:1.0: USB hub found > hub 1-1:1.0: 4 ports detected > usb 1-1.2: new high speed USB device using orion-ehci and address 3 > scsi0 : usb-storage 1-1.2:1.0 > scsi 0:0:0:0: Direct-Access Generic- SD/MMC 1.00 PQ: 0 ANSI: 0 CCS > sd 0:0:0:0: Attached scsi generic sg0 type 0 > sd 0:0:0:0: [sda] 7880704 512-byte logical blocks: (4.03 GB/3.75 GiB) > sd 0:0:0:0: [sda] Write Protect is off > sd 0:0:0:0: [sda] Mode Sense: 03 00 00 00 > sd 0:0:0:0: [sda] Assuming drive cache: write through > sd 0:0:0:0: [sda] Assuming drive cache: write through > sda: sda1 sda2 > sd 0:0:0:0: [sda] Assuming drive cache: write through > sd 0:0:0:0: [sda] Attached SCSI removable disk > VFS: Mounted root (ext2 filesystem) on device 8:1. > Freeing init memory: 160K > rtc-mv rtc-mv: internal RTC not ticking > Adding 260092k swap on /dev/sda2. Priority:-1 extents:1 across:260092k > eth0: link up, 100 Mb/s, full duplex, flow control disabled > sshd (1097): /proc/1097/oom_adj is deprecated, please use /proc/1097/oom_score_adj instead. > usb 1-1.3: new high speed USB device using orion-ehci and address 4 > usbcore: registered new interface driver snd-usb-audio >

2 2

Re: [alsa-devel] [PATCH v8] media: Add stk1160 new driver
by Mauro Carvalho Chehab 09 Aug '12

09 Aug '12

Patch looks ok. Just a few comments: Em 06-08-2012 10:38, Ezequiel Garcia escreveu: > This driver adds support for stk1160 usb bridge as used in some > video/audio usb capture devices. > It is a complete rewrite of staging/media/easycap driver and > it's expected as a replacement. > --- Please don't add a "---" here. Everything after a --- are discarded by my scripts (and by most other kernel developer scripts). > Cc: Mauro Carvalho Chehab <mchehab(a)redhat.com> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: Hans Verkuil <hverkuil(a)xs4all.nl> > Cc: Sylwester Nawrocki <sylvester.nawrocki(a)gmail.com> Hmm... weren't it reviewed already be them? > Signed-off-by: Ezequiel Garcia <elezegarcia(a)gmail.com> > diff --git a/drivers/media/video/stk1160/Makefile b/drivers/media/video/stk1160/Makefile > new file mode 100644 > index 0000000..8f66a78 > --- /dev/null > +++ b/drivers/media/video/stk1160/Makefile > @@ -0,0 +1,12 @@ > +obj-stk1160-ac97-$(CONFIG_VIDEO_STK1160_AC97) := stk1160-ac97.o > + > +stk1160-y := stk1160-core.o \ > + stk1160-v4l.o \ > + stk1160-video.o \ > + stk1160-i2c.o \ > + $(obj-stk1160-ac97-y) > + > +obj-$(CONFIG_VIDEO_STK1160) += stk1160.o > + > +ccflags-y += -Wall You shouldn't be adding the above here. > +ccflags-y += -Idrivers/media/video Ah, please split this patch into two patches: one with the new driver addition, and another one with the removal of the driver at staging. That will help to make the patch smaller, and avoids mixing two different things at the same place. Thanks, Mauro

2 1

[alsa-devel] XMOS UAC2
by adelias 09 Aug '12

09 Aug '12

I have an XMOS Reference board with the latest firmware (v3.3). With ALSA 1.0.23 it enumerates and functions properly. With ALSA 1.0.25 it appears to enumerate properly after boot up but will not output sound. Removing and reinserting snd-usb-audio with modprobe doesn't help. Only after I reset it by unpluging and repluging the usb cable will it function as it should.

2 1

[alsa-devel] [PATCH] ASoC: bells: Add machine driver for Wolfson Bells boards
by Mark Brown 09 Aug '12

09 Aug '12

The Wolfson Bells board takes submodules for various audio functions but since the system integrations are virtually identical for most of them we can support the overwhemling majority from the same machine driver. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/Kconfig | 8 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/bells.c | 346 ++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 356 insertions(+) create mode 100644 sound/soc/samsung/bells.c diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index fe3995c..fb56000 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -199,6 +199,14 @@ config SND_SOC_TOBERMORY select SND_SAMSUNG_I2S select SND_SOC_WM8962 +config SND_SOC_BELLS + tristate "Audio support for Wolfson Bells" + depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 + select SND_SAMSUNG_I2S + select SND_SOC_WM5102 + select SND_SOC_WM5110 + select SND_SOC_WM9081 + config SND_SOC_LOWLAND tristate "Audio support for Wolfson Lowland" depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 9d03beb..709f605 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -42,6 +42,7 @@ snd-soc-speyside-objs := speyside.o snd-soc-tobermory-objs := tobermory.o snd-soc-lowland-objs := lowland.o snd-soc-littlemill-objs := littlemill.o +snd-soc-bells-objs := bells.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o obj-$(CONFIG_SND_SOC_SAMSUNG_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o @@ -65,3 +66,4 @@ obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o obj-$(CONFIG_SND_SOC_TOBERMORY) += snd-soc-tobermory.o obj-$(CONFIG_SND_SOC_LOWLAND) += snd-soc-lowland.o obj-$(CONFIG_SND_SOC_LITTLEMILL) += snd-soc-littlemill.o +obj-$(CONFIG_SND_SOC_BELLS) += snd-soc-bells.o diff --git a/sound/soc/samsung/bells.c b/sound/soc/samsung/bells.c new file mode 100644 index 0000000..2aff223 --- /dev/null +++ b/sound/soc/samsung/bells.c @@ -0,0 +1,346 @@ +/* + * Bells audio support + * + * Copyright 2012 Wolfson Microelectronics + * + * 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 <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/jack.h> +#include <linux/gpio.h> +#include <linux/module.h> + +#include "../codecs/wm5102.h" +#include "../codecs/wm9081.h" + +/* + * 44.1kHz based clocks for the SYSCLK domain, use a very high clock + * to allow all the DSP functionality to be enabled if desired. + */ +#define SYSCLK_RATE (44100 * 1024) + +/* 48kHz based clocks for the ASYNC domain */ +#define ASYNCCLK_RATE (48000 * 512) + +/* BCLK2 is fixed at this currently */ +#define BCLK2_RATE (64 * 8000) + +/* + * Expect a 24.576MHz crystal if one is fitted (the driver will function + * if this is not fitted). + */ +#define MCLK_RATE 24576000 + +#define WM9081_AUDIO_RATE 44100 +#define WM9081_MCLK_RATE (WM9081_AUDIO_RATE * 256) + +static int bells_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (dapm->bias_level == SND_SOC_BIAS_STANDBY) { + ret = snd_soc_codec_set_pll(codec, WM5102_FLL1, + ARIZONA_FLL_SRC_MCLK1, + MCLK_RATE, + SYSCLK_RATE); + if (ret < 0) + pr_err("Failed to start FLL: %d\n", ret); + + ret = snd_soc_codec_set_pll(codec, WM5102_FLL2, + ARIZONA_FLL_SRC_AIF2BCLK, + BCLK2_RATE, + ASYNCCLK_RATE); + if (ret < 0) + pr_err("Failed to start FLL: %d\n", ret); + } + break; + + default: + break; + } + + return 0; +} + +static int bells_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + int ret; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + ret = snd_soc_codec_set_pll(codec, WM5102_FLL1, 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_pll(codec, WM5102_FLL2, 0, 0, 0); + if (ret < 0) { + pr_err("Failed to stop FLL: %d\n", ret); + return ret; + } + break; + + default: + break; + } + + dapm->bias_level = level; + + return 0; +} + +static int bells_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_codec *codec = card->rtd[0].codec; + struct snd_soc_dai *aif1_dai = card->rtd[0].codec_dai; + struct snd_soc_dai *aif2_dai = card->rtd[1].cpu_dai; + struct snd_soc_dai *aif3_dai = card->rtd[2].cpu_dai; + struct snd_soc_dai *wm9081_dai = card->rtd[2].codec_dai; + int ret; + + ret = snd_soc_dai_set_sysclk(aif1_dai, ARIZONA_CLK_SYSCLK, 0, 0); + if (ret != 0) { + dev_err(aif1_dai->dev, "Failed to set AIF1 clock: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(aif2_dai, ARIZONA_CLK_ASYNCCLK, 0, 0); + if (ret != 0) { + dev_err(aif2_dai->dev, "Failed to set AIF2 clock: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(aif3_dai, ARIZONA_CLK_SYSCLK, 0, 0); + if (ret != 0) { + dev_err(aif1_dai->dev, "Failed to set AIF1 clock: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_SYSCLK, + ARIZONA_CLK_SRC_FLL1, SYSCLK_RATE, + SND_SOC_CLOCK_IN); + if (ret != 0) { + dev_err(codec->dev, "Failed to set SYSCLK: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_OPCLK, 0, + WM9081_MCLK_RATE, SND_SOC_CLOCK_OUT); + if (ret != 0) { + dev_err(codec->dev, "Failed to set OPCLK: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_sysclk(codec, ARIZONA_CLK_ASYNCCLK, + ARIZONA_CLK_SRC_FLL2, ASYNCCLK_RATE, + SND_SOC_CLOCK_IN); + if (ret != 0) { + dev_err(codec->dev, "Failed to set SYSCLK: %d\n", ret); + return ret; + } + + ret = snd_soc_codec_set_sysclk(wm9081_dai->codec, WM9081_SYSCLK_MCLK, + 0, WM9081_MCLK_RATE, 0); + if (ret != 0) { + dev_err(wm9081_dai->dev, "Failed to set MCLK: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct snd_soc_pcm_stream baseband_params = { + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rate_min = 8000, + .rate_max = 8000, + .channels_min = 2, + .channels_max = 2, +}; + +static const struct snd_soc_pcm_stream sub_params = { + .formats = SNDRV_PCM_FMTBIT_S24_LE, + .rate_min = WM9081_AUDIO_RATE, + .rate_max = WM9081_AUDIO_RATE, + .channels_min = 2, + .channels_max = 2, +}; + +static struct snd_soc_dai_link bells_dai_wm5102[] = { + { + .name = "CPU", + .stream_name = "CPU", + .cpu_dai_name = "samsung-i2s.0", + .codec_dai_name = "wm5102-aif1", + .platform_name = "samsung-audio", + .codec_name = "wm5102-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + }, + { + .name = "Baseband", + .stream_name = "Baseband", + .cpu_dai_name = "wm5102-aif2", + .codec_dai_name = "wm1250-ev1", + .codec_name = "wm1250-ev1.1-0027", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + .params = &baseband_params, + }, + { + .name = "Sub", + .stream_name = "Sub", + .cpu_dai_name = "wm5102-aif3", + .codec_dai_name = "wm9081-hifi", + .codec_name = "wm9081.1-006c", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .params = &sub_params, + }, +}; + +static struct snd_soc_dai_link bells_dai_wm5110[] = { + { + .name = "CPU", + .stream_name = "CPU", + .cpu_dai_name = "samsung-i2s.0", + .codec_dai_name = "wm5110-aif1", + .platform_name = "samsung-audio", + .codec_name = "wm5110-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + }, + { + .name = "Baseband", + .stream_name = "Baseband", + .cpu_dai_name = "wm5110-aif2", + .codec_dai_name = "wm1250-ev1", + .codec_name = "wm1250-ev1.1-0027", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ignore_suspend = 1, + .params = &baseband_params, + }, + { + .name = "Sub", + .stream_name = "Sub", + .cpu_dai_name = "wm5102-aif3", + .codec_dai_name = "wm9081-hifi", + .codec_name = "wm9081.1-006c", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .params = &sub_params, + }, +}; + +static struct snd_soc_codec_conf bells_codec_conf[] = { + { + .dev_name = "wm9081.1-006c", + .name_prefix = "Sub", + }, +}; + +static struct snd_soc_dapm_route bells_routes[] = { + { "Sub CLK_SYS", NULL, "OPCLK" }, +}; + +static struct snd_soc_card bells_cards[] = { + { + .name = "Bells WM5102", + .owner = THIS_MODULE, + .dai_link = bells_dai_wm5102, + .num_links = ARRAY_SIZE(bells_dai_wm5102), + .codec_conf = bells_codec_conf, + .num_configs = ARRAY_SIZE(bells_codec_conf), + + .late_probe = bells_late_probe, + + .dapm_routes = bells_routes, + .num_dapm_routes = ARRAY_SIZE(bells_routes), + + .set_bias_level = bells_set_bias_level, + .set_bias_level_post = bells_set_bias_level_post, + }, + { + .name = "Bells WM5110", + .owner = THIS_MODULE, + .dai_link = bells_dai_wm5110, + .num_links = ARRAY_SIZE(bells_dai_wm5110), + .codec_conf = bells_codec_conf, + .num_configs = ARRAY_SIZE(bells_codec_conf), + + .late_probe = bells_late_probe, + + .dapm_routes = bells_routes, + .num_dapm_routes = ARRAY_SIZE(bells_routes), + + .set_bias_level = bells_set_bias_level, + .set_bias_level_post = bells_set_bias_level_post, + }, +}; + + +static __devinit int bells_probe(struct platform_device *pdev) +{ + int ret; + + bells_cards[pdev->id].dev = &pdev->dev; + + ret = snd_soc_register_card(&bells_cards[pdev->id]); + if (ret) { + dev_err(&pdev->dev, + "snd_soc_register_card(%s) failed: %d\n", + bells_cards[pdev->id].name, ret); + return ret; + } + + return 0; +} + +static int __devexit bells_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&bells_cards[pdev->id]); + + return 0; +} + +static struct platform_driver bells_driver = { + .driver = { + .name = "bells", + .owner = THIS_MODULE, + .pm = &snd_soc_pm_ops, + }, + .probe = bells_probe, + .remove = __devexit_p(bells_remove), +}; + +module_platform_driver(bells_driver); + +MODULE_DESCRIPTION("Bells audio support"); +MODULE_AUTHOR("Mark Brown <broonie(a)opensource.wolfsonmicro.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:bells"); -- 1.7.10.4

1 0

[alsa-devel] [PATCH 1/3] ASoC: arizona: Implement OPCLK support
by Mark Brown 09 Aug '12

09 Aug '12

Arizona devices support two output system clocks. Provide support for configuring these via set_sysclk(). Once the clock API is more useful we should migrate over to that. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 66 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 6 ++-- 2 files changed, 70 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 5c9caca..5e96a0a 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -229,6 +229,69 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(arizona_out_ev); +static unsigned int arizona_sysclk_48k_rates[] = { + 6144000, + 12288000, + 22579200, + 49152000, +}; + +static unsigned int arizona_sysclk_44k1_rates[] = { + 5644800, + 11289600, + 24576000, + 45158400, +}; + +static int arizona_set_opclk(struct snd_soc_codec *codec, unsigned int clk, + unsigned int freq) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + unsigned int *rates; + int ref, div, refclk; + + switch (clk) { + case ARIZONA_CLK_OPCLK: + reg = ARIZONA_OUTPUT_SYSTEM_CLOCK; + refclk = priv->sysclk; + break; + case ARIZONA_CLK_ASYNC_OPCLK: + reg = ARIZONA_OUTPUT_ASYNC_CLOCK; + refclk = priv->asyncclk; + break; + default: + return -EINVAL; + } + + if (refclk % 8000) + rates = arizona_sysclk_44k1_rates; + else + rates = arizona_sysclk_48k_rates; + + for (ref = 0; ref < ARRAY_SIZE(arizona_sysclk_48k_rates) && + rates[ref] <= refclk; ref++) { + div = 1; + while (rates[ref] / div >= freq && div < 32) { + if (rates[ref] / div == freq) { + dev_dbg(codec->dev, "Configured %dHz OPCLK\n", + freq); + snd_soc_update_bits(codec, reg, + ARIZONA_OPCLK_DIV_MASK | + ARIZONA_OPCLK_SEL_MASK, + (div << + ARIZONA_OPCLK_DIV_SHIFT) | + ref); + return 0; + } + div++; + } + } + + dev_err(codec->dev, "Unable to generate %dHz OPCLK\n", freq); + return -EINVAL; +} + int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir) { @@ -252,6 +315,9 @@ int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, reg = ARIZONA_ASYNC_CLOCK_1; clk = &priv->asyncclk; break; + case ARIZONA_CLK_OPCLK: + case ARIZONA_CLK_ASYNC_OPCLK: + return arizona_set_opclk(codec, clk_id, freq); default: return -EINVAL; } diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 59caca8..eb66b52 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -17,8 +17,10 @@ #include <sound/soc.h> -#define ARIZONA_CLK_SYSCLK 1 -#define ARIZONA_CLK_ASYNCCLK 2 +#define ARIZONA_CLK_SYSCLK 1 +#define ARIZONA_CLK_ASYNCCLK 2 +#define ARIZONA_CLK_OPCLK 3 +#define ARIZONA_CLK_ASYNC_OPCLK 4 #define ARIZONA_CLK_SRC_MCLK1 0x0 #define ARIZONA_CLK_SRC_MCLK2 0x1 -- 1.7.10.4

1 2

[alsa-devel] [PATCH 1/2] ASoC: core: Upgrade the severity of probe deferral errors to dev_err()
by Mark Brown 09 Aug '12

09 Aug '12

In the past when ASoC had a custom probe deferral mechanism people complained about the logspam it generated and didn't want to know about the fact that we were doing probe deferral so all the error messages for it were at dev_dbg(), making diagnostics hard. Now that we have probe deferral as an accepted thing and it's generating log messages anyway there's no need to worry about this so upgrade the severity of all the probe deferral sources to dev_err() so that they are displayed by default. Also add one for missing aux_devs since there wasn't one. Reported-by: Russell King <linux(a)arm.linux.org.uk> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 91b096a..97fb881 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -826,7 +826,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } if (!rtd->cpu_dai) { - dev_dbg(card->dev, "CPU DAI %s not registered\n", + dev_err(card->dev, "CPU DAI %s not registered\n", dai_link->cpu_dai_name); return -EPROBE_DEFER; } @@ -857,14 +857,14 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) } if (!rtd->codec_dai) { - dev_dbg(card->dev, "CODEC DAI %s not registered\n", + dev_err(card->dev, "CODEC DAI %s not registered\n", dai_link->codec_dai_name); return -EPROBE_DEFER; } } if (!rtd->codec) { - dev_dbg(card->dev, "CODEC %s not registered\n", + dev_err(card->dev, "CODEC %s not registered\n", dai_link->codec_name); return -EPROBE_DEFER; } @@ -888,7 +888,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) rtd->platform = platform; } if (!rtd->platform) { - dev_dbg(card->dev, "platform %s not registered\n", + dev_err(card->dev, "platform %s not registered\n", dai_link->platform_name); return -EPROBE_DEFER; } @@ -1481,6 +1481,8 @@ static int soc_check_aux_dev(struct snd_soc_card *card, int num) return 0; } + dev_err(card->dev, "%s not registered\n", aux_dev->codec_name); + return -EPROBE_DEFER; } -- 1.7.10.4

1 1

Re: [alsa-devel] [PATCH 3/5] hda_intel: Add Device IDs for Intel Lynx Point-LP PCH
by Takashi Iwai 09 Aug '12

09 Aug '12

At Thu, 9 Aug 2012 09:38:59 -0700, james.d.ralston(a)intel.com wrote: > > From: James Ralston <james.d.ralston(a)intel.com> > > This patch adds the Intel HD Audio Device IDs for the Intel Lynx Point-LP PCH > > Signed-off-by: James Ralston <james.d.ralston(a)intel.com> Applied now. Thanks. Takashi > --- > sound/pci/hda/hda_intel.c | 9 +++++++++ > 1 files changed, 9 insertions(+), 0 deletions(-) > > diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c > index c8aced1..60882c6 100644 > --- a/sound/pci/hda/hda_intel.c > +++ b/sound/pci/hda/hda_intel.c > @@ -151,6 +151,7 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}," > "{Intel, CPT}," > "{Intel, PPT}," > "{Intel, LPT}," > + "{Intel, LPT_LP}," > "{Intel, HPT}," > "{Intel, PBG}," > "{Intel, SCH}," > @@ -3270,6 +3271,14 @@ static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { > { PCI_DEVICE(0x8086, 0x8c20), > .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > + /* Lynx Point-LP */ > + { PCI_DEVICE(0x8086, 0x9c20), > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > + /* Lynx Point-LP */ > + { PCI_DEVICE(0x8086, 0x9c21), > + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_SCH_SNOOP | > + AZX_DCAPS_BUFSIZE | AZX_DCAPS_POSFIX_COMBO }, > /* Haswell */ > { PCI_DEVICE(0x8086, 0x0c0c), > .driver_data = AZX_DRIVER_SCH | AZX_DCAPS_SCH_SNOOP | > -- > 1.7.7.6 >

1 0

[alsa-devel] uda1380 and pxa255 plays nothing but noise
by nerwusek 09 Aug '12

09 Aug '12

Hello, I'm trying to enable sound on my iPAQ h2200 device using ALSA ASoC. I've created driver basing on hx4700.c and magician.c files, but I can't hear is anything. When I am playing music on device file uda1380-codec.0-0018/dapm/Playback shows that stream Playback is active, but in DAC that stream is inactive (but DAC is on). I also noticed that on I2S bus is nothing going on. I've lost many hours on looking for problem. Is there anyone who could help me? Łukasz Dałek

3 10

Re: [alsa-devel] [sound-unstable:master 94/97] sound/pci/hda/hda_intel.c:3174:6: error: 'struct azx' has no member named 'fw'
by Takashi Iwai 09 Aug '12

09 Aug '12

At Thu, 9 Aug 2012 23:34:06 +0800, Fengguang Wu wrote: > > Hi Takashi, > > FYI, kernel build failed on > > tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable.git master > head: d2dfd82739080fefb704c0d8e4c9b2b2093a601c > commit: 5cb543dba9867588786f87af2e64fca371b69283 [94/97] ALSA: hda - Deferred probing with request_firmware_nowait() > config: i386-defconfig (attached as .config) > > All related error/warning messages: > > sound/pci/hda/hda_intel.c: In function 'azx_firmware_cb': > sound/pci/hda/hda_intel.c:3174:6: error: 'struct azx' has no member named 'fw' > sound/pci/hda/hda_intel.c: At top level: > sound/pci/hda/hda_intel.c:3163:13: warning: 'azx_firmware_cb' defined but not used [-Wunused-function] Oh, a typical mistake, a forgotten ifdef. Fixed now. Thanks! Takashi

1 0

[alsa-devel] [PATCH 10/22] Documentation: Define the MSP Driver Device Tree bindings
by Lee Jones 09 Aug '12

09 Aug '12

Here we add the required documentation for the new Device Tree bindings pertaining to the MSP CPU-side DAI Driver. CC: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- .../devicetree/bindings/sound/ux500-msp.txt | 43 ++++++++++++++++++++ 1 file changed, 43 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ux500-msp.txt diff --git a/Documentation/devicetree/bindings/sound/ux500-msp.txt b/Documentation/devicetree/bindings/sound/ux500-msp.txt new file mode 100644 index 0000000..99acd9c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ux500-msp.txt @@ -0,0 +1,43 @@ +* ux500 MSP (CPU-side Digital Audio Interface) + +Required properties: + - compatible :"stericsson,ux500-msp-i2s" + - reg : Physical base address and length of the device's registers. + +Optional properties: + - interrupts : The interrupt output from the device. + - interrupt-parent : The parent interrupt controller. + - <name>-supply : Phandle to the regulator <name> supply + +Example: + + sound { + compatible = "stericsson,snd-soc-mop500"; + + stericsson,platform-pcm-dma = <&pcm>; + stericsson,cpu-dai = <&msp1 &msp3>; + stericsson,audio-codec = <&codec>; + }; + + pcm: ux500-pcm { + compatible = "stericsson,ux500-pcm"; + }; + + msp1: msp@80124000 { + compatible = "stericsson,ux500-msp-i2s"; + reg = <0x80124000 0x1000>; + interrupts = <0 62 0x4>; + v-ape-supply = <&db8500_vape_reg>; + }; + + msp3: msp@80125000 { + compatible = "stericsson,ux500-msp-i2s"; + reg = <0x80125000 0x1000>; + interrupts = <0 62 0x4>; + v-ape-supply = <&db8500_vape_reg>; + }; + + codec: ab8500-codec { + compatible = "stericsson,ab8500-codec"; + stericsson,earpeice-cmv = <950>; /* Units in mV. */ + }; -- 1.7.9.5

1 0

[alsa-devel] [PATCH 09/22] Documentation: Define the MOP500 Audio Machine Driver Device Tree bindings
by Lee Jones 09 Aug '12

09 Aug '12

Here we add the required documentation for the new Device Tree bindings pertaining to the MOP500 Audio Machine driver. CC: alsa-devel(a)alsa-project.org Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- .../devicetree/bindings/sound/ux500-mop500.txt | 39 ++++++++++++++++++++ 1 file changed, 39 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ux500-mop500.txt diff --git a/Documentation/devicetree/bindings/sound/ux500-mop500.txt b/Documentation/devicetree/bindings/sound/ux500-mop500.txt new file mode 100644 index 0000000..48e071c --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ux500-mop500.txt @@ -0,0 +1,39 @@ +* MOP500 Audio Machine Driver + +This node is responsible for linking together all ux500 Audio Driver components. + +Required properties: + - compatible : "stericsson,snd-soc-mop500" + +Non-standard properties: + - stericsson,cpu-dai : Phandle to the CPU-side DAI + - stericsson,audio-codec : Phandle to the Audio CODEC + - stericsson,card-name : Over-ride default card name + +Example: + + sound { + compatible = "stericsson,snd-soc-mop500"; + + stericsson,cpu-dai = <&msp1 &msp3>; + stericsson,audio-codec = <&codec>; + }; + + msp1: msp@80124000 { + compatible = "stericsson,ux500-msp-i2s"; + reg = <0x80124000 0x1000>; + interrupts = <0 62 0x4>; + v-ape-supply = <&db8500_vape_reg>; + }; + + msp3: msp@80125000 { + compatible = "stericsson,ux500-msp-i2s"; + reg = <0x80125000 0x1000>; + interrupts = <0 62 0x4>; + v-ape-supply = <&db8500_vape_reg>; + }; + + codec: ab8500-codec { + compatible = "stericsson,ab8500-codec"; + stericsson,earpeice-cmv = <950>; /* Units in mV. */ + }; -- 1.7.9.5

1 0

[alsa-devel] [PATCH] ucm: Allow cset commands to have values with spaces.
by Tanu Kaskinen 09 Aug '12

09 Aug '12

An example: cset "name='Input Select' Digital Mic" The old parsing code interpreted "name='Input Select' Digital" as the element id, which of course didn't work. Signed-off-by: Tanu Kaskinen <tanu.kaskinen(a)digia.com> --- src/ucm/main.c | 19 +++++++++++++++++-- 1 file changed, 17 insertions(+), 2 deletions(-) diff --git a/src/ucm/main.c b/src/ucm/main.c index 4b37776..05a7b0a 100644 --- a/src/ucm/main.c +++ b/src/ucm/main.c @@ -170,8 +170,23 @@ static int execute_cset(snd_ctl_t *ctl, char *cset) snd_ctl_elem_value_malloc(&value); snd_ctl_elem_info_malloc(&info); - pos = strrchr(cset, ' '); - if (pos == NULL) { + /* Find the space after the element id, taking quoting with + single-quotes into account. */ + for (pos = cset; *pos != '\0'; pos += strcspn(pos, "' ")) { + if (*pos == ' ') + break; + if (*pos == '\'') { + pos++; + pos += strcspn(pos, "'"); + if (*pos == '\0') { + uc_error("invalid element id (closing single-quote not found): %s", cset); + err = -EINVAL; + goto __fail; + } + pos++; + } + } + if (*pos == '\0') { uc_error("undefined value for cset >%s<", cset); err = -EINVAL; goto __fail; -- 1.7.9.5

3 5

[alsa-devel] [PATCH] ALSA: hda - Defer probe when loading patch firmware
by Thierry Reding 09 Aug '12

09 Aug '12

Recent changes to the firmware loading helpers cause drivers to stall when firmware is loaded during the module_init() call. The snd-hda-intel module requests firmware if the patch= parameter is used to load a patch file. This patch works around the problem by deferring the probe in such cases, which will cause the module to load successfully and the driver binding to the device outside the module_init() call. Signed-off-by: Thierry Reding <thierry.reding(a)avionic-design.de> --- sound/pci/hda/hda_intel.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index c8aced1..296616e 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -69,6 +69,7 @@ static int probe_only[SNDRV_CARDS]; static bool single_cmd; static int enable_msi = -1; #ifdef CONFIG_SND_HDA_PATCH_LOADER +static bool deferred[SNDRV_CARDS]; static char *patch[SNDRV_CARDS]; #endif #ifdef CONFIG_SND_HDA_INPUT_BEEP @@ -3156,6 +3157,16 @@ static int __devinit azx_probe(struct pci_dev *pci, if (dev >= SNDRV_CARDS) return -ENODEV; + +#ifdef CONFIG_SND_HDA_PATCH_LOADER + if (patch[dev] && *patch[dev] && !deferred[dev]) { + snd_printk(KERN_ERR SFX "deferring probe for patch %s\n", + patch[dev]); + deferred[dev] = true; + return -EPROBE_DEFER; + } +#endif + if (!enable[dev]) { dev++; return -ENOENT; -- 1.7.11.4

3 29

Re: [alsa-devel] [PATCH v8] media: Add stk1160 new driver
by Ezequiel Garcia 09 Aug '12

09 Aug '12

Hi Mauro, On Mon, Aug 6, 2012 at 10:38 AM, Ezequiel Garcia <elezegarcia(a)gmail.com> wrote: > This driver adds support for stk1160 usb bridge as used in some > video/audio usb capture devices. > It is a complete rewrite of staging/media/easycap driver and > it's expected as a replacement. > --- > I just sent v8, but it looks it wasn't received by patchwork either. What's going on? Regards, Ezequiel.

3 8

[alsa-devel] [PATCH] ASoC: wm8994: Add missing dapm routes for WM8958 rev A
by Mark Brown 09 Aug '12

09 Aug '12

From: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> Signed-off-by: Chris Rattray <crattray(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 04ef031..6c9eeca 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -4038,6 +4038,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; case WM8958: if (wm8994->revision < 1) { + snd_soc_dapm_add_routes(dapm, wm8994_intercon, + ARRAY_SIZE(wm8994_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, ARRAY_SIZE(wm8994_revd_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon, -- 1.7.10.4

1 0

[alsa-devel] [PATCH] ASoC: Davinci: McASP: remove unused header include
by Hebbar, Gururaja 09 Aug '12

09 Aug '12

Defines or parameters from <mach/mux.h> isn't used anywhere. Hence remove the header include. Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> --- This patch was tested on AM18x Board and is based on repo at http://git.kernel.org/?p=linux/kernel/git/broonie/sound.git; a=shortlog;h=refs/heads/for-3.6 :100644 100644 10a2d8c... c80c20a... M sound/soc/davinci/davinci-evm.c sound/soc/davinci/davinci-evm.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/davinci/davinci-evm.c b/sound/soc/davinci/davinci-evm.c index 10a2d8c..c80c20a 100644 --- a/sound/soc/davinci/davinci-evm.c +++ b/sound/soc/davinci/davinci-evm.c @@ -24,7 +24,6 @@ #include <mach/asp.h> #include <mach/edma.h> -#include <mach/mux.h> #include "davinci-pcm.h" #include "davinci-i2s.h" -- 1.7.1

2 3

[alsa-devel] Question about surround default pin config
by David Henningsson 09 Aug '12

09 Aug '12

Hi, Sometimes it's better to just ask than getting confused trying to figure it out yourself. When you have 2 stereo pairs of line-out or speaker, they are meant as either 4.0 (front left, front right, rear left, rear right), and sometimes they are meant as 2.1 (front left, front right, unused, lfe). We should support both variations, I assume. What is really the way to write my pin fixups to specify either 4.0 or 2.1? Are there any de-facto standards? If I look at e g check_output_sfx (in hda_auto_parser.c), it looks like it would be impossible to have 2.1, i e, impossible to have LFE without also having rear left and rear right. But I might be missing something? -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

2 1

[alsa-devel] plughw vs default / per-application volume changer
by antony k 09 Aug '12

09 Aug '12

Hello, I am new to ALSA and I cant figure out 2 things. a) if and why should I prefer the "plughw:0,0" over the "default" device name. I am developing a simple audio player which is based on the latter. I made this choice because when I was using plughw:0,0 with other audio programs(flash player,vlc) open in the background I would get the "Device is busy" error code. With default, it's fine. b) can I have an application specific volume changer? So far I have managed to tweak the master volume of the "default" device which has a global effect. Im guessing a) and b) are connected. Thanks for your time

2 1

Re: [alsa-devel] adding an external dsp under soc-dsp framework
by Mark Brown 09 Aug '12

09 Aug '12

On Wed, Aug 08, 2012 at 10:49:19PM +0300, Ziv Haziz wrote: > Looking at the littlemill driver on kernel 3.4.7 the device name > "Baseband" and his dai link doesn't contain platform device can be > triggered without side effects? Yes, this is the most common case for DAI links that don't involve a CPU in modern smartphones.

1 0

Re: [alsa-devel] [PATCH 3/4] ARM: OMAP3: Switch to use the unified audio driver (omap-twl4030) for selected boards
by Peter Ujfalusi 09 Aug '12

09 Aug '12

On 08/08/2012 04:06 PM, Igor Grinberg wrote: > Well, yes I know we will need to adjust the user space. > I wanted to change that for a long time and did not get to it... > Now, with your patches, it seems like the best time for doing this. Done for the v2. You will have card named "cm-t3x". -- Péter

1 0

[alsa-devel] [PATCH] ALSA: hda - Fix pop noise in headphones on S3 for Asus X55A, X55V
by David Henningsson 09 Aug '12

09 Aug '12

To turn off pin control for the pin was tested, and helped against this issue. BugLink: https://bugs.launchpad.net/bugs/1034779 Tested-by: Chih-Hsyuan Ho <chih.ho(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_via.c | 8 ++++++++ 1 file changed, 8 insertions(+) Alsa-info at: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1034779/+attachment/32… diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 80d90cb..c9484fe 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1752,6 +1752,14 @@ static int via_suspend(struct hda_codec *codec) { struct via_spec *spec = codec->spec; vt1708_stop_hp_work(spec); + + if (spec->codec_type == VT1802) { + /* Fix pop noise on headphones */ + int i; + for (i = 0; i < spec->autocfg.hp_outs; i++) + snd_hda_set_pin_ctl(codec, spec->autocfg.hp_pins[i], 0); + } + return 0; } #endif -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix pop noise in headphones on S3 for Asus X55A, X55V
by David Henningsson 09 Aug '12

09 Aug '12

To turn off pin control for the pin was tested, and helped against this issue. BugLink: https://bugs.launchpad.net/bugs/1034779 Tested-by: Chih-Hsyuan Ho <chih.ho(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_via.c | 8 ++++++++ 1 file changed, 8 insertions(+) Alsa-info at: https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1034779/+attachment/32… diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 80d90cb..c9484fe 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -1752,6 +1752,14 @@ static int via_suspend(struct hda_codec *codec) { struct via_spec *spec = codec->spec; vt1708_stop_hp_work(spec); + + if (spec->codec_type == VT1802) { + /* Fix pop noise on headphones */ + int i; + for (i = 0; i < spec->autocfg.hp_outs; i++) + snd_hda_set_pin_ctl(codec, spec->autocfg.hp_pins[0], 0); + } + return 0; } #endif -- 1.7.9.5

2 2

[alsa-devel] [PATCH] ALSA: hda/realtek - Keep analog bypass muted on ALC892
by Thierry Reding 09 Aug '12

09 Aug '12

Unmuting the analog bypass causes any input to be directly bypassed to the outputs, which is usually not expected by the user. Signed-off-by: Thierry Reding <thierry.reding(a)avionic-design.de> --- Note that this patch could be generalized to include all codec variants that have this second channel. As I don't have access to all the data sheets I'll leave that decision up to somebody with more hardware available. sound/pci/hda/patch_realtek.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4f81dd4..54d0917 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3752,8 +3752,15 @@ static void alc_auto_set_output_and_unmute(struct hda_codec *codec, if (nid_has_mute(codec, mix, HDA_INPUT)) { snd_hda_codec_write(codec, mix, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)); - snd_hda_codec_write(codec, mix, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AMP_IN_UNMUTE(1)); + /* + * On ALC892, the second entry in the connection list is the + * analog bypass. Unmuting it will cause any inputs to be + * directly forwarded to the corresponding outputs. + */ + if (codec->subsystem_id != 0x10ec0892) + snd_hda_codec_write(codec, mix, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(1)); } /* initialize volume */ nid = alc_look_for_out_vol_nid(codec, pin, dac); -- 1.7.11.4

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Re: [alsa-devel] [PATCH V4 12/16] ALSA: HDA: Make hda sound card usable for Loongson.
by Takashi Iwai 09 Aug '12

09 Aug '12

At Thu, 9 Aug 2012 11:35:15 +0800, Chen Jie wrote: > > Hi, > > 2012/8/3 Takashi Iwai <tiwai(a)suse.de>: > > At Fri, 3 Aug 2012 18:36:40 +0800, > > Huacai Chen wrote: > >> > >> We write these quirks on 2.6.36 some time ago, and then we port them > >> to 3.x (3.2, 3.3, 3.4 and 3.5). As you say, PMON (BIOS for Loongson) > >> doesn't set the pins correctly. Anyway, I'll try your suggestions. > > > > Thanks. I guess it should work by just adding a new entry for your > > device in cxt_fixups[] containing the right default pin-configuration > > table, then point it in cxt5066_fixups[] with the corresponding PCI > > (or codec) SSID. > > > > > > Takashi > > I've found it is a little difficult to get proper pincfg values. The > original patch builds 'input/output path' manually, the new way does > it automatically as long as providing proper pincfgs for 'end points'. > > I tried to copy related pincfgs from a workable lemote a1004 > laptop(kernel with the original patch, read from > /proc/asound/card0/codec#0), didn't help much. It doesn't help because your old patch doesn't change the default pin-configuration values. It changes the pin-control values or amp values, but not the former one. > I guess the pincfgs are not correct, and on the platform, the pincfgs > are not touched by BIOS, so I have to calculate proper pincfgs, how? Just follow the HD-audio specification :) > HDA spec explains a pincfg as four bytes. For each byte, it has bits > indicate amp/in/out/vref/ept. No, you are referring to pin control bits. This has nothing to do with the pin default configuration. You must misunderstand the spec completely. Read section 7.3.3.31 (page 179) for details. Takashi > In the kernel side, it reads a pincfg and explains it as a 32bits > flag, indicating > connect/location/device/jack_connect_type/jack_color/misc/association/sequence. > > Why they differ? How to get proper pincfg values? > > > > Regards, > -- Chen Jie >

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[alsa-devel] hw_params ignores ioplug's buffer size requirements
by Andrew Eikum 08 Aug '12

08 Aug '12

Hi folks, I've got either a bug or a misunderstanding in how ALSA handles ioplugs that provide hw_params restrictions. My understanding is that the snd_pcm_ioplug_set_param_* functions are supposed to restrict which hw_params settings a client may set. This seems to work for at least access and format. But I noticed that it's broken for buffer_bytes and periods at least. Attached is two source files demonstrating the problem. alsa-example.c is a dead-simple ioplug module. It sets some hw_params restrictions, and does basically nothing else. The important thing to notice is that it sets /only one/ allowable period size, period count, and buffer size, all of which are compatible with each other (256 byte period * 8 periods == 2048 byte buffer size). There is also test client, alsa-test.c, which attempts to violate those hw_params restrictions, and succeeds! The eventual call to the ioplug's example_hw_params() callback gets an invalid buffer_size parameter (it happens to be 512). Here is the output when run on my Arch Linux with alsa-lib 1.0.25: [aeikum@aeikum ioplug_buffer_bytes]$ ./alsa-test ALSA-example: Opening bytes: 512 ALSA-example: hw_params: 512 TEST FAILURE: buffer_size != 8812: 512 ALSA-example: stop ALSA-example: close [aeikum@aeikum ioplug_buffer_bytes]$ Am I misunderstanding the snd_pcm_ioplug_set_param_* methods, or is this a bug? I am seeing this behavior in the wild with a pulseaudio client that ignores my hw_params restrictions. Thanks for any insight, Andrew

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[alsa-devel] [PATCH] test/latency.c: Fix string argument to getopt_long, so now it handles -E and -B
by Mario Domenech Goulart 08 Aug '12

08 Aug '12

Hi, Attached is a small patch for alsa-lib's test/latency.c to fix command line arguments handling. Best wishes. Mario -- http://parenteses.org/mario

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Re: [alsa-devel] DT support for Davinci ASoC platform
by Daniel Mack 08 Aug '12

08 Aug '12

Hi Serkar, On 03.08.2012 11:09, Sekhar Nori wrote: > On 8/3/2012 2:02 AM, Daniel Mack wrote: >> On 05.07.2012 12:58, Daniel Mack wrote: >>> Hi Gururaja, >>> >>> On 04.07.2012 14:48, Hebbar, Gururaja wrote: >>>> All, >>>> >>>> Are there any plans or work going on for providing Device Tree Support for Davinci >>>> ASoC? >>>> >>>> I am planning to add DT support for TI AM335x (McASP) and I don't want to do duplicate >>>> task. >>> >>> That's good to know. But isn't there a dependency on the EDMA engine? Is >>> that part ported yet? >> >> May I ask whether you made any progress on that yet? > > Matt Porter from TI is working on DMA engine migration. This would still > be some time away since there is a bunch of work involved here. > > To make progress on Audio DT conversion Gururaja is doing an up port of > AM335x EDMA support from TI release to mainline kernel. If you are > interested in it, he can probably share that with you once he posts his > DT patches. As the DMA engine is also a requirement for the MMC controller, I would be interested in helping, in case help is required. Is there any repository with code that I can have a look at? Thanks, Daniel

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[alsa-devel] [PATCH v2] ASoC: imx-ssi: Remove mono support
by Fabio Estevam 08 Aug '12

08 Aug '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Playing a mono track results in incorrect playback rate, ie, the audio is played at a faster rate. Remove mono support in the driver by setting 'channes_min' to dual-channel and this allows mono tracks to be played correctly. Reported-by: Gaëtan Carlier <gcembed(a)gmail.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- Changes since v1: - Also change 'channels_min' for the capture - Improve the commit log sound/soc/fsl/imx-ssi.c | 5 +++-- 1 files changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 3c520c4..7074ae6 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -380,13 +380,14 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver imx_ssi_dai = { .probe = imx_ssi_dai_probe, .playback = { - .channels_min = 1, + /* The SSI does not support monaural audio. */ + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, -- 1.7.1

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Re: [alsa-devel] [PATCH 3/4] ARM: OMAP3: Switch to use the unified audio driver (omap-twl4030) for selected boards
by Peter Ujfalusi 08 Aug '12

08 Aug '12

Hi Igor, On 08/08/2012 03:24 PM, Igor Grinberg wrote: >> diff --git a/arch/arm/mach-omap2/board-cm-t35.c b/arch/arm/mach-omap2/board-cm-t35.c >> index 97d7190..b73fa5c 100644 >> --- a/arch/arm/mach-omap2/board-cm-t35.c >> +++ b/arch/arm/mach-omap2/board-cm-t35.c >> @@ -731,6 +731,7 @@ static void __init cm_t3x_common_init(void) >> cm_t35_init_ethernet(); >> cm_t35_init_led(); >> cm_t35_init_display(); >> + omap_twl4030_audio_init("overo", 26000000); > > Can the card_name here be "cm-t3x" ? I have used the "overo" as card name here since what I gathered is that cm-t35 used the sound/soc/omap/overo.c machine driver for audio support. If the card name is changing your existing user space might get confused by the card name change. But sure, I can change the card name to "cm-t3x" if it is better for you. -- Péter

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Re: [alsa-devel] [PATCH] ASoC: Samsung: Fix build error
by Kukjin Kim 08 Aug '12

08 Aug '12

Sachin Kamat wrote: > > On 8 August 2012 17:08, Kukjin Kim <kgene.kim(a)samsung.com> wrote: > > Sachin Kamat wrote: > >> > >> Hi Mark, > >> > >> On 8 August 2012 16:39, Mark Brown > <broonie(a)opensource.wolfsonmicro.com> > >> wrote: > >> > On Wed, Aug 08, 2012 at 11:34:43AM +0530, Sachin Kamat wrote: > >> >> Fixes the following build error: > >> >> In file included from arch/arm/mach-exynos/include/mach/dma.h:24:0, > >> >> from arch/arm/plat-samsung/include/plat/dma-ops.h:17, > >> >> from arch/arm/plat-samsung/include/plat/dma.h:128, > >> >> from sound/soc/samsung/pcm.c:23: > >> >> arch/arm/plat-samsung/include/plat/dma-pl330.h:106:8: > >> >> error: redefinition of 'struct > > s3c2410_dma_client' > >> >> arch/arm/plat-samsung/include/plat/dma.h:40:8: note: originally > defined > >> here > >> >> make[3]: *** [sound/soc/samsung/pcm.o] Error 1 > >> > > >> > Shouldn't this be being fixed in the headers? It looks like they > >> > reference and disagree with each other on Exynos (my primary > development > >> > platform is S3C6410 based so I'd notice if it was a generic Samsung > >> > issue. > >> > > >> > Pasting the error message isn't really that useful unless the fix is > >> > obvious, it's better to explain in words what's wrong. > >> > >> I had sent the initial patch trying to fix this in the headers and it > >> worked for exynos4 config. However Kukjin pointed out that the patch > >> created similar errors with 2410 and 6400 configs. Fixing it further > >> in the headers did not look quite straight forward as there are many > >> headers trying to include one another. Hence I felt it is better > >> handled in the current way. I understand i should have added this > >> information in the commit message. Below is the link for your > >> reference that provides the complete discussion. > >> > >> http://comments.gmane.org/gmane.linux.kernel.samsung-soc/11274 > >> > >> Please let me know your opinion about this. > >> > >> > > I also agree with Mark's opinion and current Sachin's approach cannot > avoid > > whole similar build error. > > With the current change I did not see any build error with the configs > (2410, 6400, 64x0 and exyno4) i checked with (after enabling audio). > Please let me know if you observed any such build error after adding > this patch. > > Happens following error with s5pv210_defcofnig (after enabling s5p-sss). In file included from arch/arm/mach-s5pv210/include/mach/dma.h:24, from arch/arm/plat-samsung/include/plat/dma-ops.h:17, from arch/arm/plat-samsung/include/plat/dma.h:128, from drivers/crypto/s5p-sss.c:33: arch/arm/plat-samsung/include/plat/dma-pl330.h:106: error: redefinition of 'struct s3c2410_dma_client' make[2]: *** [drivers/crypto/s5p-sss.o] Error 1 make[1]: *** [drivers/crypto] Error 2 make[1]: *** Waiting for unfinished jobs.... make: *** [drivers] Error 2 make: *** Waiting for unfinished jobs.... Actually, <mach/dma.h> is including <plat/dma-pl330.h> for s5p64x0, s5pc100, s5pv210 and exynos or <plat/dma.h> for s3c24xx and s3c64xx. And the cause of build error you reported is duplicated declaration of s3c2410_dma_client. So it can happen when <plat/dma-pl330.h> and <plat/dma.h> are included together at the same time... Thanks. Best regards, Kgene. -- Kukjin Kim <kgene.kim(a)samsung.com>, Senior Engineer, SW Solution Development Team, Samsung Electronics Co., Ltd.

1 0

Re: [alsa-devel] [PATCH] ASoC: Samsung: Fix build error
by Kukjin Kim 08 Aug '12

08 Aug '12

Sachin Kamat wrote: > > Hi Mark, > > On 8 August 2012 16:39, Mark Brown <broonie(a)opensource.wolfsonmicro.com> > wrote: > > On Wed, Aug 08, 2012 at 11:34:43AM +0530, Sachin Kamat wrote: > >> Fixes the following build error: > >> In file included from arch/arm/mach-exynos/include/mach/dma.h:24:0, > >> from arch/arm/plat-samsung/include/plat/dma-ops.h:17, > >> from arch/arm/plat-samsung/include/plat/dma.h:128, > >> from sound/soc/samsung/pcm.c:23: > >> arch/arm/plat-samsung/include/plat/dma-pl330.h:106:8: > >> error: redefinition of 'struct s3c2410_dma_client' > >> arch/arm/plat-samsung/include/plat/dma.h:40:8: note: originally defined > here > >> make[3]: *** [sound/soc/samsung/pcm.o] Error 1 > > > > Shouldn't this be being fixed in the headers? It looks like they > > reference and disagree with each other on Exynos (my primary development > > platform is S3C6410 based so I'd notice if it was a generic Samsung > > issue. > > > > Pasting the error message isn't really that useful unless the fix is > > obvious, it's better to explain in words what's wrong. > > I had sent the initial patch trying to fix this in the headers and it > worked for exynos4 config. However Kukjin pointed out that the patch > created similar errors with 2410 and 6400 configs. Fixing it further > in the headers did not look quite straight forward as there are many > headers trying to include one another. Hence I felt it is better > handled in the current way. I understand i should have added this > information in the commit message. Below is the link for your > reference that provides the complete discussion. > > http://comments.gmane.org/gmane.linux.kernel.samsung-soc/11274 > > Please let me know your opinion about this. > > I also agree with Mark's opinion and current Sachin's approach cannot avoid whole similar build error. Thanks. Best regards, Kgene. -- Kukjin Kim <kgene.kim(a)samsung.com>, Senior Engineer, SW Solution Development Team, Samsung Electronics Co., Ltd.

1 0

[alsa-devel] [PATCH] ASoC: mxs: Fix the name of the SoC family
by Fabio Estevam 08 Aug '12

08 Aug '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> SND_SOC_MXS_SGTL5000 is used on MXS boards, so fix the SoC family name. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/mxs/Kconfig b/sound/soc/mxs/Kconfig index 99a997f..b6fa776 100644 --- a/sound/soc/mxs/Kconfig +++ b/sound/soc/mxs/Kconfig @@ -10,7 +10,7 @@ menuconfig SND_MXS_SOC if SND_MXS_SOC config SND_SOC_MXS_SGTL5000 - tristate "SoC Audio support for i.MX boards with sgtl5000" + tristate "SoC Audio support for MXS boards with sgtl5000" depends on I2C select SND_SOC_SGTL5000 help -- 1.7.1

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Re: [alsa-devel] [PATCH] ASoC: Samsung: Fix build error
by Mark Brown 08 Aug '12

08 Aug '12

On Wed, Aug 08, 2012 at 11:34:43AM +0530, Sachin Kamat wrote: > Fixes the following build error: > In file included from arch/arm/mach-exynos/include/mach/dma.h:24:0, > from arch/arm/plat-samsung/include/plat/dma-ops.h:17, > from arch/arm/plat-samsung/include/plat/dma.h:128, > from sound/soc/samsung/pcm.c:23: > arch/arm/plat-samsung/include/plat/dma-pl330.h:106:8: > error: redefinition of ‘struct s3c2410_dma_client’ > arch/arm/plat-samsung/include/plat/dma.h:40:8: note: originally defined here > make[3]: *** [sound/soc/samsung/pcm.o] Error 1 Shouldn't this be being fixed in the headers? It looks like they reference and disagree with each other on Exynos (my primary development platform is S3C6410 based so I'd notice if it was a generic Samsung issue. Pasting the error message isn't really that useful unless the fix is obvious, it's better to explain in words what's wrong.

1 0

Re: [alsa-devel] adding an external dsp under soc-dsp framework
by Mark Brown 08 Aug '12

08 Aug '12

On Wed, Aug 08, 2012 at 11:21:20AM +0300, Ziv Haziz wrote: > >I n that case what you're saying above with a DAI link is exactly what > > you're supposed to do, I don't see any structural problem here? > Creating two dai links will create two separated devices which must be > open for setting configurations, but the dsp-codec dai cannot be started > (or I will get an error). This means changing the codec driver to > enable everything not from trigger callback. > Is it possible to have a dai link without a platform for data transfer? > I saw in the plumbers agenda you are going to talk on codec-codec > interface - do you have any preliminary suggestion? This is already supported in mainline, look at littlemill for one example user. Alternatively for older kernels you can easily support this from userspace by just not starting the DMA, this has been deployed for some considerable time on many systems (OpenMoko is one long standing mainline example).

1 0

[alsa-devel] [PATCH 1/2] hda-emu: Flush output before prompt
by David Henningsson 08 Aug '12

08 Aug '12

When stdout is redirected, there is a risk not all stdout is sent before the prompt is sent on stderr. Thanks to Jeremy Kerr for help with this issue. --- hda-ctlsh.c | 1 + hda-log.c | 5 +++++ include/hda-log.h | 1 + 3 files changed, 7 insertions(+) diff --git a/hda-ctlsh.c b/hda-ctlsh.c index 4c21341..d84cc71 100644 --- a/hda-ctlsh.c +++ b/hda-ctlsh.c @@ -851,6 +851,7 @@ int cmd_loop(FILE *fp) init_completion(); for (;;) { + hda_log_flush(); line = readline("> "); if (!line) break; diff --git a/hda-log.c b/hda-log.c index 65f3044..67ee8e2 100644 --- a/hda-log.c +++ b/hda-log.c @@ -179,6 +179,11 @@ int hda_log_level_set(int level) return saved; } +void hda_log_flush() +{ + fflush(logfp); +} + FILE *hda_get_logfp(void) { return logfp; diff --git a/include/hda-log.h b/include/hda-log.h index 2e2137b..8b0d392 100644 --- a/include/hda-log.h +++ b/include/hda-log.h @@ -18,6 +18,7 @@ int hda_log_init(const char *file, unsigned int flags); void hda_log(int level, const char *fmt, ...); void hda_log_printk(const char *fmt, ...); void hda_log_echo(int level, const char *fmt, ...); +void hda_log_flush(void); extern int hda_log_trap_on_error; -- 1.7.9.5

2 3

Re: [alsa-devel] [PATCHv2 - CA0132 HDA Codec 1/2] ALSA: Update Creative CA0132 codec to add DSP features.
by Takashi Iwai 08 Aug '12

08 Aug '12

At Mon, 30 Jul 2012 15:49:00 -0700, Ian Minett wrote: > > >> One thought we have is to handle the spin_lock_irqxxx() as a special > >> case or quirk for the CA0132 codec. Using an approach similar to > >> 'needs_damn_long_delay' in the CA0110 patch, we could define another > >> flag e.g. 'needs_trigger_irqsave' in hda_codec.h struct hda_bus{}. In > >> the CA0132 patch, we would set this flag, which would then be read in > >> hda_intel.c,azx_pcm_trigger(). Either spin_lock_irqxxx() or the current > >> spin_lock() would then be called depending on the flag condition. > >> What is your opinion on an approach like this? Would this be acceptable? > > >No. The fact that you are calling the PCM functions is already > >wrong. The irqsave() is just a side-effect by that, and we can forget > >about it completely now. > > >The only question is where to implement the DSP loader stuff and how. > > Ok, we understand. Given that the codec cannot call into the controller's > PCM interface to load the DSP image, is there an acceptable way you could > recommend for us to add this functionality to the module, without breaking > the architecture? Well, as mentioned earlier, we need to push it up to the controller layer, i.e. hda_intel.c. For example, add a new op for firmware DMA transfer in hda_bus_ops. Could you describe briefly the procedure for the DMA transfer of CA0132? The code looks too complex to follow. thanks, Takashi

2 6

[alsa-devel] [PATCH] ALSA: hda - Fix double quirk for Quanta FL1
by David Henningsson 08 Aug '12

08 Aug '12

The same ID is twice in the quirk table, so the second one is not used. You probably meant to chain them, like this? Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index daa5103..4f81dd4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6099,6 +6099,8 @@ static const struct alc_fixup alc269_fixups[] = { [ALC269_FIXUP_PCM_44K] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_pcm_44k, + .chained = true, + .chain_id = ALC269_FIXUP_QUANTA_MUTE }, [ALC269_FIXUP_STEREO_DMIC] = { .type = ALC_FIXUP_FUNC, @@ -6210,8 +6212,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x21fb, "Thinkpad T430s", ALC269_FIXUP_LENOVO_DOCK), SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), - SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE), - SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K), + SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), #if 0 -- 1.7.9.5

2 1

[alsa-devel] alsa-project.org is down
by Gabriel M. Beddingfield 08 Aug '12

08 Aug '12

Howdy, Several have noticed that alsa-project.org is down. I don't know if this is a planned outage or not. -gabriel

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Re: [alsa-devel] [PATCH] ASoC: imx-ssi: Fix mono playback
by Mark Brown 07 Aug '12

07 Aug '12

On Mon, Aug 06, 2012 at 04:36:25PM -0300, Fabio Estevam wrote: > Currently mono playback is played on a faster rate. > > Adjust channels_min to 2 channels in order to play it back correctly. This doesn't actually fix mono playback so much as remove support for it...

6 19

[alsa-devel] [PATCH] ASoC: omap-mcbsp: Fix 6pin mux configuration
by Peter Ujfalusi 07 Aug '12

07 Aug '12

The check for the mux_signal callback was wrong which prevents us to configure the 6pin port's FSR/CLKR signal mux. Reported-by: CF Adad <cfadad(a)rocketmail.com> Cc: stable(a)kernel.org (3.4+) Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Mark, The omap-mcbsp FSR/CLKR mux configuration code seams to be broken since 3.4. Would it be possible to apply this patch for the 3.6 release? CC-ing stable(a)kernel.org since this part is broken in 3.4 and 3.5 kernels as well. Regards, Peter sound/soc/omap/mcbsp.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/mcbsp.c b/sound/soc/omap/mcbsp.c index 1b08d98..5bda7c9 100644 --- a/sound/soc/omap/mcbsp.c +++ b/sound/soc/omap/mcbsp.c @@ -766,7 +766,7 @@ int omap_mcbsp_6pin_src_mux(struct omap_mcbsp *mcbsp, u8 mux) { const char *signal, *src; - if (mcbsp->pdata->mux_signal) + if (!mcbsp->pdata->mux_signal) return -EINVAL; switch (mux) { -- 1.7.8.6

3 3

[alsa-devel] [PATCH] ALSA: Initialize power-down register on AC97 codec wm9711/12/15.
by m.kiepfer＠fs-net.de 07 Aug '12

07 Aug '12

From: Martin Kiepfer <m.kiepfer(a)fs-net.de> Signed-off-by: Martin Kiepfer <m.kiepfer(a)fs-net.de> --- sound/pci/ac97/ac97_patch.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/ac97/ac97_patch.c b/sound/pci/ac97/ac97_patch.c index a872d0a..b8e493ac 100644 --- a/sound/pci/ac97/ac97_patch.c +++ b/sound/pci/ac97/ac97_patch.c @@ -697,6 +697,8 @@ static int patch_wolfson11(struct snd_ac97 * ac97) ac97->flags |= AC97_HAS_NO_REC_GAIN | AC97_STEREO_MUTES | AC97_HAS_NO_MIC | AC97_HAS_NO_PC_BEEP | AC97_HAS_NO_VIDEO | AC97_HAS_NO_CD; + + snd_ac97_write_cache(ac97, AC97_POWERDOWN, 0); return 0; } -- 1.7.4.4

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[alsa-devel] [PATCH] ASoC: mxs-saif: Use devm_clk_get()
by Fabio Estevam 07 Aug '12

07 Aug '12

From: Fabio Estevam <fabio.estevam(a)freescale.com> Using devm_clk_get can make the code simpler and smaller. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/mxs/mxs-saif.c | 20 +++++++------------- 1 files changed, 7 insertions(+), 13 deletions(-) diff --git a/sound/soc/mxs/mxs-saif.c b/sound/soc/mxs/mxs-saif.c index b303071..aa037b2 100644 --- a/sound/soc/mxs/mxs-saif.c +++ b/sound/soc/mxs/mxs-saif.c @@ -704,7 +704,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) return ret; } - saif->clk = clk_get(&pdev->dev, NULL); + saif->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(saif->clk)) { ret = PTR_ERR(saif->clk); dev_err(&pdev->dev, "Cannot get the clock: %d\n", @@ -717,8 +717,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) saif->base = devm_request_and_ioremap(&pdev->dev, iores); if (!saif->base) { dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENODEV; - goto failed_get_resource; + return -ENODEV; } dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); @@ -731,7 +730,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) &saif->dma_param.chan_num); if (ret) { dev_err(&pdev->dev, "failed to get dma channel\n"); - goto failed_get_resource; + return ret; } } else { saif->dma_param.chan_num = dmares->start; @@ -742,7 +741,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) ret = saif->irq; dev_err(&pdev->dev, "failed to get irq resource: %d\n", ret); - goto failed_get_resource; + return ret; } saif->dev = &pdev->dev; @@ -750,7 +749,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) "mxs-saif", saif); if (ret) { dev_err(&pdev->dev, "failed to request irq\n"); - goto failed_get_resource; + return ret; } saif->dma_param.chan_irq = platform_get_irq(pdev, 1); @@ -758,7 +757,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) ret = saif->dma_param.chan_irq; dev_err(&pdev->dev, "failed to get dma irq resource: %d\n", ret); - goto failed_get_resource; + return ret; } platform_set_drvdata(pdev, saif); @@ -766,7 +765,7 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) ret = snd_soc_register_dai(&pdev->dev, &mxs_saif_dai); if (ret) { dev_err(&pdev->dev, "register DAI failed\n"); - goto failed_get_resource; + return ret; } ret = mxs_pcm_platform_register(&pdev->dev); @@ -779,19 +778,14 @@ static int __devinit mxs_saif_probe(struct platform_device *pdev) failed_pdev_alloc: snd_soc_unregister_dai(&pdev->dev); -failed_get_resource: - clk_put(saif->clk); return ret; } static int __devexit mxs_saif_remove(struct platform_device *pdev) { - struct mxs_saif *saif = platform_get_drvdata(pdev); - mxs_pcm_platform_unregister(&pdev->dev); snd_soc_unregister_dai(&pdev->dev); - clk_put(saif->clk); return 0; } -- 1.7.1

2 1

[alsa-devel] [PATCH] ASoC: core: Set dapm->idle_bias_off for DAIs not mapped with a codec
by Peter Ujfalusi 07 Aug '12

07 Aug '12

The idle_bias_off flag is not configured for DAIs not mapped with a codec. This causes the pm counter to be increased at probe time for the CPU dai which unbalances the pm counter handling. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hello Mark, I'm not sure if this is the correct way to fix the problem but this patch works for me. The issue in short: Since commit: '18d7564 ASoC: when removing a CPU DAI, clean up its DAPM context' After booting up the kernel on BeagleBoard for example I see that the clocks are enabled for the McBSP CPU dai. This alone causing the first stream start to fail due to unbalanced pm_runtime counter. Later it stabilizes, but I see failures with full duplex operation (arecord | aplay). The root cause of this is not commit 18d7564, but the fact that we did not configured the dapm->idle_bias_off for the CPU dai to be 1 (we do this for the platform driver). The pm_runtime_get_sync is called from dapm_pre_sequence_async() - since we add the cpu_dai dapm context to the list. With this patch we set the dapm->idle_bias_off for DAIs not linked with a codec which means cpu_dais. If this is not acceptable we need to set this flag in the cpu_dai driver itself. Regards, Peter sound/soc/soc-core.c | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f81c597..f10f00b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3715,6 +3715,9 @@ int snd_soc_register_dai(struct device *dev, } } + if (!dai->codec) + dai->dapm.idle_bias_off = 1; + list_add(&dai->list, &dai_list); mutex_unlock(&client_mutex); @@ -3803,6 +3806,9 @@ int snd_soc_register_dais(struct device *dev, } } + if (!dai->codec) + dai->dapm.idle_bias_off = 1; + list_add(&dai->list, &dai_list); mutex_unlock(&client_mutex); -- 1.7.8.6

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Re: [alsa-devel] [PATCH] ASoC: cs42l52: Remove duplicate inclusion of slab.h header file
by Mark Brown 07 Aug '12

07 Aug '12

On Tue, Aug 07, 2012 at 12:07:27PM +0530, Sachin Kamat wrote: > slab.h header file was included twice. Applied, thanks.

1 0

Re: [alsa-devel] adding an external dsp under soc-dsp framework
by Mark Brown 07 Aug '12

07 Aug '12

On Mon, Aug 06, 2012 at 06:11:33PM +0300, Ziv Haziz wrote: > ok thanks. This means that I need to introduce two codecs and make a dai > link between them? > It doesn't really fits the dai structure - how do you suggest to do it? No, if they're part of the same physical chip then they should be handled internally to the device. > ok - finger crossed you will publish an agreed interface soon - > currently I saw in codeaurora kernel an implementation of exposing mixer > controls, so currently this is the only option yes? That's not upstream, standard Linux has no support for Slimbus at all.

2 2

[alsa-devel] Plumbers: Please split audio topics into separate sessions
by David Henningsson 07 Aug '12

07 Aug '12

Looking at: http://summit.linuxplumbersconf.org/lpc-2012/track/lpc2012-audio/ 45 minutes is not enough to handle two topics: Audio uConf: HD-audio cooking && QA/Testing Audio uConf: Expose Routing && DSPs in ASoC Audio uConf: Time Alignment && PA on Android Please split these three sessions into six. It is better to have a little more time than necessary (even if I believe we easily fill 45 minutes per topic), than having to skip a topic just because the other topic took more time than expected. -- David Henningsson, Canonical Ltd. https://launchpad.net/~diwic

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[alsa-devel] [PATCH 1/2] ALSA: hda - remove quirk for Dell Vostro 1015
by David Henningsson 07 Aug '12

07 Aug '12

This computer is confirmed working with model=auto on kernel 3.2. Also, parsing fails with hda-emu with the current model. Cc: stable(a)kernel.org (3.2+) Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_conexant.c | 1 - 1 file changed, 1 deletion(-) Alsa info: see hda-emu/codecs/canonical/cx20585-dell-vostro-1015-ccert-201010-6649 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1436118..6f538eb 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2972,7 +2972,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x02d8, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x02f5, "Dell Vostro 320", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO), - SND_PCI_QUIRK(0x1028, 0x0402, "Dell Vostro", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD), -- 1.7.9.5

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Re: [alsa-devel] [PATCH v3 4/4] fbdev: sh_mobile_lcdc: use dma_mmap_coherent if available
by Laurent Pinchart 07 Aug '12

07 Aug '12

Hi Eiraku-san, On Monday 06 August 2012 18:55:24 Hideki EIRAKU wrote: > fb_mmap() implemented in fbmem.c uses smem_start as the physical > address of the frame buffer. In the sh_mobile_lcdc driver, the > smem_start is a dma_addr_t that is not a physical address when IOMMU is > enabled. dma_mmap_coherent() maps the address correctly. It is > available on ARM platforms. > > Signed-off-by: Hideki EIRAKU <hdk(a)igel.co.jp> Acked-by: Hideki EIRAKU <hdk(a)igel.co.jp> As this patch doesn't depend on any other patch in your series (ARCH_HAS_DMA_MMAP_COHERENT will not be defined without 1/4, so this patch will be a no-op until then), I've applied it to my tree and will push it to avoid merge conflicts, unless you would prefer to push it yourself. > --- > drivers/video/sh_mobile_lcdcfb.c | 28 ++++++++++++++++++++++++++++ > 1 files changed, 28 insertions(+), 0 deletions(-) > > diff --git a/drivers/video/sh_mobile_lcdcfb.c > b/drivers/video/sh_mobile_lcdcfb.c index 8cb653b..c8cba7a 100644 > --- a/drivers/video/sh_mobile_lcdcfb.c > +++ b/drivers/video/sh_mobile_lcdcfb.c > @@ -1614,6 +1614,17 @@ static int sh_mobile_lcdc_overlay_blank(int blank, > struct fb_info *info) return 1; > } > > +#ifdef ARCH_HAS_DMA_MMAP_COHERENT > +static int > +sh_mobile_lcdc_overlay_mmap(struct fb_info *info, struct vm_area_struct > *vma) +{ > + struct sh_mobile_lcdc_overlay *ovl = info->par; > + > + return dma_mmap_coherent(ovl->channel->lcdc->dev, vma, ovl->fb_mem, > + ovl->dma_handle, ovl->fb_size); > +} > +#endif > + > static struct fb_ops sh_mobile_lcdc_overlay_ops = { > .owner = THIS_MODULE, > .fb_read = fb_sys_read, > @@ -1626,6 +1637,9 @@ static struct fb_ops sh_mobile_lcdc_overlay_ops = { > .fb_ioctl = sh_mobile_lcdc_overlay_ioctl, > .fb_check_var = sh_mobile_lcdc_overlay_check_var, > .fb_set_par = sh_mobile_lcdc_overlay_set_par, > +#ifdef ARCH_HAS_DMA_MMAP_COHERENT > + .fb_mmap = sh_mobile_lcdc_overlay_mmap, > +#endif > }; > > static void > @@ -2093,6 +2107,17 @@ static int sh_mobile_lcdc_blank(int blank, struct > fb_info *info) return 0; > } > > +#ifdef ARCH_HAS_DMA_MMAP_COHERENT > +static int > +sh_mobile_lcdc_mmap(struct fb_info *info, struct vm_area_struct *vma) > +{ > + struct sh_mobile_lcdc_chan *ch = info->par; > + > + return dma_mmap_coherent(ch->lcdc->dev, vma, ch->fb_mem, > + ch->dma_handle, ch->fb_size); > +} > +#endif > + > static struct fb_ops sh_mobile_lcdc_ops = { > .owner = THIS_MODULE, > .fb_setcolreg = sh_mobile_lcdc_setcolreg, > @@ -2108,6 +2133,9 @@ static struct fb_ops sh_mobile_lcdc_ops = { > .fb_release = sh_mobile_lcdc_release, > .fb_check_var = sh_mobile_lcdc_check_var, > .fb_set_par = sh_mobile_lcdc_set_par, > +#ifdef ARCH_HAS_DMA_MMAP_COHERENT > + .fb_mmap = sh_mobile_lcdc_mmap, > +#endif > }; > > static void -- Regards, Laurent Pinchart

1 1

[alsa-devel] Connecting a codec to McASP in Linux 2.6.37
by Mike Looijmans 07 Aug '12

07 Aug '12

We are using the (Arago) linux kernel 2.6.37 (also labelled v2.6.37_DAVINCIPSP_03.21.00.04). There are several TLV320AIC3256 codec connected to an OMAP-L138 SoM board via I2C and McASP. The board is based on the DA850 EVM. To support the codec, I've backported the tlv320aic32x4 driver from the 2.6.39 linux kernel. The I2C part is configured correctly, and I can get and set mixer components and other settings via I2C to the codec. However, setting up the McASP data path is a complete riddle, and I need help to get this to work. In the board files, the following structs "glue" the codec to the audio data transfer unit: static struct snd_soc_dai_link da8xx_evm_dai[] = { { .name = "TLV320AIC3X-1", .stream_name = "AIC3X-1", .cpu_dai_name= "davinci-mcasp.0", .codec_dai_name = "tlv320aic32x4-hifi", .codec_name = "tlv320aic32x4.11-0018", .platform_name = "davinci-pcm-audio", .init = evm_aic3x_init_analog, .ops = &evm_ops, }, { .name = "TLV320AIC3X-2", .stream_name = "AIC3X-2", .cpu_dai_name= "davinci-mcasp.0", .codec_dai_name = "tlv320aic32x4-hifi", .codec_name = "tlv320aic32x4.12-0018", .platform_name = "davinci-pcm-audio", .init = evm_aic3x_init_analog, .ops = &evm_ops, }, I cannot determine what "stream_name" is supposed to do. Documentation in "dapm.txt" claims that "the stream name must match the corresponding stream name in your codec snd_soc_codec_dai", but that appears to be untrue, because in the existing SoC drivers, they never match. Also, "snd_soc_codec_dai" seems to be a typo in the documentation, because it only appears in the alsa soc documentation and nowhere else in the kernel. The string "davinci-mcasp.0" refers to the struct in sound/soc/davinci/davinci-mcasp.c: static struct snd_soc_dai_driver davinci_mcasp_dai[] = { { .name = "davinci-mcasp.0", .playback = { .channels_min = 2, .channels_max = 2, .rates = DAVINCI_MCASP_RATES, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, }, .capture = { .channels_min = 2, .channels_max = 2, .rates = DAVINCI_MCASP_RATES, .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, }, .ops = &davinci_mcasp_dai_ops, }, And in the board config files, there is also a struct that configures the McASP ports: static u8 da850_iis_serializer_direction[] = { INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, RX_MODE, TX_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_MODE, INACTIVE_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, .asp_chan_q = EVENTQ_1, .version = MCASP_VERSION_2, .txnumevt = 1, .rxnumevt = 1, }; ... da8xx_register_mcasp(0, &da850_evm_snd_data); Now this part is a complete mystery. To have multiple CODECs, do i need to register multiple instances of the McASP driver and fill in the serializer structs for each? And if not, how does it know which input is for which codec? And what are all the fields? There's no documentation whatsoever, so the above was my best guess to get at least one codec functioning. It sort of seems to do something, but all samples are zero and I get lots of overrun errors when using arecord to capture data from it. If there's more you need to know in order to help, don't hesitate to ask, I'll be happy to supply whatever information you need on this. Mike. Met vriendelijke groet / kind regards, Mike Looijmans TOPIC Embedded Systems Eindhovenseweg 32-C, NL-5683 KH Best Postbus 440, NL-5680 AK Best Telefoon: (+31) - (0)499 - 33.69.79 Telefax: (+31) - (0)499 - 33.69.70 E-mail: mike.looijmans(a)topic.nl Website: www.topic.nl Dit e-mail bericht en de eventueel daarbij behorende bijlagen zijn uitsluitend bestemd voor de geadresseerde, zoals die blijkt uit het e-mail bericht en/of de bijlagen. Er kunnen gegevens met betrekking tot een derde instaan. Indien u als niet-geadresseerde dit bericht en de bijlagen ontvangt, terwijl u niet bevoegd of gemachtigd bent om dit bericht namens de geadresseerde te ontvangen, wordt u verzocht de afzender hierover direct te informeren en het e-mail bericht met de bijlagen te vernietigen. 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Any use of the contents of this message and/or the enclosures by any other person than the addressee or person who is empowered to receive this message, is illegal towards the sender and/or the aforementioned third party. TOPIC Embedded Systems is not liable for any damage as a result of the use and/or acceptance of this message and as well as any enclosures.

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[alsa-devel] [PATCH] ALSA: hda - Set channels count for HBR format
by Wang Xingchao 07 Aug '12

07 Aug '12

If it's HBR format, the channels count should be 8. Signed-off-by: Wang Xingchao <xingchao.wang(a)intel.com> --- sound/pci/hda/patch_hdmi.c | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index ad319d4..ae4e67e 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1153,8 +1153,11 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hdmi_spec *spec = codec->spec; int pin_idx = hinfo_to_pin_index(spec, hinfo); hda_nid_t pin_nid = spec->pins[pin_idx].pin_nid; - - hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); + + if (is_hbr_format(format)) + hdmi_set_channel_count(codec, cvt_nid, 8); + else + hdmi_set_channel_count(codec, cvt_nid, substream->runtime->channels); hdmi_setup_audio_infoframe(codec, pin_idx, substream); -- 1.7.9.5

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Re: [alsa-devel] [PATCH] ALSA: hda - add dock support for Thinkpad X230
by Takashi Iwai 07 Aug '12

07 Aug '12

At Mon, 6 Aug 2012 23:02:01 +0200, Felix Kaechele wrote: n> > As with the ThinkPad Models X230 Tablet and T530 the X230 needs a qurik to > correctly set up the pins for the dock port. > > Signed-off-by: Felix Kaechele <felix(a)fetzig.org> Applied now (with Cc to stable like others). Thanks. Takashi > --- > sound/pci/hda/patch_realtek.c | 1 + > 1 file changed, 1 insertion(+) > > diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c > index b9a5c45..6163826 100644 > --- a/sound/pci/hda/patch_realtek.c > +++ b/sound/pci/hda/patch_realtek.c > @@ -6207,6 +6207,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { > SND_PCI_QUIRK(0x17aa, 0x21ca, "Thinkpad L412", ALC269_FIXUP_SKU_IGNORE), > SND_PCI_QUIRK(0x17aa, 0x21e9, "Thinkpad Edge 15", ALC269_FIXUP_SKU_IGNORE), > SND_PCI_QUIRK(0x17aa, 0x21f6, "Thinkpad T530", ALC269_FIXUP_LENOVO_DOCK), > + SND_PCI_QUIRK(0x17aa, 0x21fa, "Thinkpad X230", ALC269_FIXUP_LENOVO_DOCK), > SND_PCI_QUIRK(0x17aa, 0x2203, "Thinkpad X230 Tablet", ALC269_FIXUP_LENOVO_DOCK), > SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_QUANTA_MUTE), > SND_PCI_QUIRK(0x17aa, 0x3bf8, "Lenovo Ideapd", ALC269_FIXUP_PCM_44K), > -- > 1.7.11.2 >

1 0

Re: [alsa-devel] [PATCH 1/1] mx27_3ds: Add sound support
by Fabio Estevam 06 Aug '12

06 Aug '12

Gaëtan, On Mon, Aug 6, 2012 at 11:24 AM, Gaëtan Carlier <gcembed(a)gmail.com> wrote: > Hi Fabio, > Sorry, I have missed your patch in my email archive. I didn't search the > right keywords. > I applied your [patch 1/2] (2/2 is already merged) but I have same effect. I > use aplay with file k3b_error1.wav (Unsigned 8 bit, Rate 8000 Hz, Mono) and > alsa/Front_Left.wav (Signed 16 bit Little Endian, Rate 48000 Hz, Mono). Two > sounds are played too fast. > It is maybe a bad configuration of Alsa (but this is working with kernel > 2.6.22 from Freescale). I have just sent the following patch to the alsa-list which fixes this mono playback issue. Regards, Fabio Estevam >From bc46087110bd48a41f1f7dcb826853349662c766 Mon Sep 17 00:00:00 2001 From: Fabio Estevam <fabio.estevam(a)freescale.com> Date: Mon, 6 Aug 2012 16:30:51 -0300 Subject: [PATCH] ASoC: imx-ssi: Fix mono playback Currently mono playback is played on a faster rate. Adjust channels_min to 2 channels in order to play it back correctly. Reported-by: Gaëtan Carlier <gcembed(a)gmail.com> Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- sound/soc/fsl/imx-ssi.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index 28dd76c..c42df31 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -380,7 +380,7 @@ static int imx_ssi_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver imx_ssi_dai = { .probe = imx_ssi_dai_probe, .playback = { - .channels_min = 1, + .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, -- 1.7.1

1 0

[alsa-devel] [RFC PATCH 0/2] ASoC: add compress stream support
by Vinod Koul 06 Aug '12

06 Aug '12

Here is the much delayed support for compress audio streams in ASoC. This patch series is in RFC form, once we have agreed on this I will send the patches supporting Intel driver as well. Mainly we indicate if the dai is of compressed nature and based on that register a new compressed device. Also the platform driver adds the compress stream operations. Since the codecs handles PCM only, the assumption is that they will not have anything to do here and machine needs to set codec appropriately for compressed streams. In future if codec with such support are available then we can simply add compressed operations into codec as well Comments, criticism welcome... -- Namarta Kohli (1): ASoC: add compress stream support Vinod Koul (1): ASoC: add definations for compressed operations include/sound/compress_driver.h | 1 + include/sound/soc-dai.h | 2 + include/sound/soc.h | 17 ++- sound/soc/Makefile | 2 +- sound/soc/soc-compress.c | 297 +++++++++++++++++++++++++++++++++++++++ sound/soc/soc-core.c | 10 ++- 6 files changed, 326 insertions(+), 3 deletions(-) create mode 100644 sound/soc/soc-compress.c

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[alsa-devel] adding an external dsp under soc-dsp framework
by ziv haziz 06 Aug '12

06 Aug '12

i plan on adding an external dsp which has slimbus interface under soc-dsp framwork. currently i would like to make the following type of connection: CPU->DSP->Codec and as the dsp is an external one (physically seperated from both the CPU and the Codec) i have a difficulty defining how to represent it: do i need to make two entities of the dsp: - one for the BE connection and one for the FE connection and add some internal routing using mixer controls? this will create two devices in the user space which the user application should open? - or is there a way to make one device which is composed of all three blocks? last small question: as slimbus ports id should be passed how do you suggest a device should share his slimbus port id?

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Re: [alsa-devel] [PATCH 1/1] mx27_3ds: Add sound support
by Fabio Estevam 06 Aug '12

06 Aug '12

On Mon, Aug 6, 2012 at 11:24 AM, Gaëtan Carlier <gcembed(a)gmail.com> wrote: > Hi Fabio, > Sorry, I have missed your patch in my email archive. I didn't search the > right keywords. > I applied your [patch 1/2] (2/2 is already merged) but I have same effect. I > use aplay with file k3b_error1.wav (Unsigned 8 bit, Rate 8000 Hz, Mono) and > alsa/Front_Left.wav (Signed 16 bit Little Endian, Rate 48000 Hz, Mono). Two > sounds are played too fast. > It is maybe a bad configuration of Alsa (but this is working with kernel > 2.6.22 from Freescale). Ok, we will need to debug this problem then. Can you please send me offline these two files? Regards, Fabio Estevam

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Re: [alsa-devel] [PATCH 00/27] ASoC: Use module_i2c_driver
by Mark Brown 06 Aug '12

06 Aug '12

On Mon, Aug 06, 2012 at 05:25:33PM +0530, Sachin Kamat wrote: > module_i2c_driver() is helper macro for I2C drivers which do not do anything > special in module init/exit. This eliminates a lot of boilerplate > and calling it replaces module_init() and module_exit(). Applied all, thanks.

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Re: [alsa-devel] [PATCH 1/1] mx27_3ds: Add sound support
by Fabio Estevam 06 Aug '12

06 Aug '12

Hi Gaëtan, On Mon, Aug 6, 2012 at 10:07 AM, Gaëtan Carlier <gcembed(a)gmail.com> wrote: > Hi, > I have ported code "Enable sound on mx31_3ds" from Philippe Rétornaz to > MX27 3DS platform on master branch of linux-next (tag next-20120806). > I also applied patch "dma: imx-dma: Fix kernel crash due to missing clock > conversion" of Fabio Estevam which is not yet merged in linux-next. > Thanks to them, everything is working good on MX27 3DS platform except that > sound is played too fast. > Which parameter must be adjusted to have sound playing at normal speed ? > In mc13783 codec driver, imx_add_imx_ssi(), clock configuration in > mach-mx27_3ds file ? > > Thanks for your help. > > Signed-off-by: Gaëtan Carlier <gcembed(a)gmail.com> I have added audio support for mx27pdk: https://patchwork.kernel.org/patch/1152951/ ,and in my tests the audio was played at the correct rate. Can you please have a try? What is the sampling rate of your file? Regards, Fabio Estevam

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Re: [alsa-devel] [PATCH v7] media: Add stk1160 new driver
by Ezequiel Garcia 06 Aug '12

06 Aug '12

Hi Mauro, On Sat, Jul 28, 2012 at 11:18 AM, Ezequiel Garcia <elezegarcia(a)gmail.com> wrote: > This driver adds support for stk1160 usb bridge as used in some > video/audio usb capture devices. > It is a complete rewrite of staging/media/easycap driver and > it's expected as a future replacement. > > Cc: Mauro Carvalho Chehab <mchehab(a)redhat.com> > Cc: Takashi Iwai <tiwai(a)suse.de> > Cc: Hans Verkuil <hverkuil(a)xs4all.nl> > Cc: Sylwester Nawrocki <sylvester.nawrocki(a)gmail.com> > Signed-off-by: Ezequiel Garcia <elezegarcia(a)gmail.com> > --- > Did you take a look at this? Perhaps we can discuss now you're previous comments: 1. Place for ac97 code: media or alsa? (see Takashis' comments) 2. current_norm usage (see Hans' comments) 3. vb2_dqbuf and O_NONBLOCK flag (also see Hans' comments) I know it's a big patch*, so there is no need to rush. I just wanted to discuss a bit about this before my brain-cache flushes completely :-) Thanks, Ezequiel. * Actually it's huge (12k lines) since I'm removing staging/easycap.

2 3

[alsa-devel] Enable docking station speaker jack for new Lenovo Thinkpads
by Mario 06 Aug '12

06 Aug '12

Hi all, for new Lenovo Thinkpads (mine T530, will probably affect all new [TXW][45]30 models), the model=thinkpad module option does not enable the speaker jack of the docking station. Sound works perfectly with the laptop, only the docking station needs extra work. I managed to get it work using hda-analyzer. It produced the following diff: Diff for codec 0/0 (0x10ec0269): --- +++ @@ -210,28 +210,28 @@ Unsolicited: tag=0x00, enabled=0 Power: setting=D0, actual=D0 Connection: 2 0x0c* 0x0d Node 0x1b [Pin Complex] wcaps 0x40058f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 - Amp-Out vals: [0x80 0x80] + Amp-Out vals: [0x00 0x00] Pincap 0x00003734: IN OUT Detect Vref caps: HIZ 50 GRD 80 100 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 VREF_HIZ - Unsolicited: tag=0x00, enabled=0 - Power: setting=D0, actual=D0 - Connection: 2 - 0x0c* 0x0d + Pin-ctls: 0x40: OUT VREF_HIZ + Unsolicited: tag=0x00, enabled=0 + Power: setting=D0, actual=D0 + Connection: 2 + 0x0c 0x0d* Node 0x1c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1d [Pin Complex] wcaps 0x400400: Mono Pincap 0x00000020: IN Pin Default 0x40138205: [N/A] Speaker at Ext N/A Conn = ATAPI, Color = Purple DefAssociation = 0x0, Sequence = 0x5 Pin-ctls: 0x20: IN Power: setting=D0, actual=D0 Relevent lspci entry: 00:1b.0 Audio device [0403]: Intel Corporation 7 Series/C210 Series Chipset Family High Definition Audio Controller [8086:1e20] (rev 04) Chip: Realtek ALC269VC Further alsa-info: http://www.alsa-project.org/db/?f=e297bdda4fd099425516c7c831e28a2c7d8ae6b6 Unfortunately, I failed to create a working hda-verb script. I can change the out-pin with: hda-verb /dev/snd/hwC0D0 0x1b SET_PIN_WIDGET_CONTROL 0x40 but I did not find the command to unmute the channel. Can anyone tell me which options are necessary and can this easily be fixed in the kernel? Also, hda-analyzer showed me two mixer devices, they produce a slightly different sound. I don't know if this is important. Thank you very much! Mario

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Re: [alsa-devel] [PATCH] ASoC: sgtl5000: remove device tree probe match table
by Mark Brown 06 Aug '12

06 Aug '12

On Mon, Aug 06, 2012 at 06:00:31PM +0800, Hui Wang wrote: > The sgtl5000 codec driver is an i2c driver instead of a platform > driver, current of_i2c subsystem will convert an i2c device node > into a traditional i2c device, the of_match_table is not used. No, this is a regression - even though we can take a good guess for I2C devices it is always good to provide a compatible property to avoid ambiguity.

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Re: [alsa-devel] [PATCH] sound: print small buffers via %*ph[C]
by Takashi Iwai 06 Aug '12

06 Aug '12

At Thu, 2 Aug 2012 16:52:41 +0300, Andy Shevchenko wrote: > > Signed-off-by: Andy Shevchenko <andriy.shevchenko(a)linux.intel.com> Applied now. Thanks. Takashi > --- > sound/isa/gus/interwave.c | 5 ++--- > sound/usb/6fire/firmware.c | 5 ++--- > 2 files changed, 4 insertions(+), 6 deletions(-) > > diff --git a/sound/isa/gus/interwave.c b/sound/isa/gus/interwave.c > index a76bc8d..3fc8b66 100644 > --- a/sound/isa/gus/interwave.c > +++ b/sound/isa/gus/interwave.c > @@ -443,9 +443,8 @@ static void __devinit snd_interwave_detect_memory(struct snd_gus_card * gus) > for (i = 0; i < 8; ++i) > iwave[i] = snd_gf1_peek(gus, bank_pos + i); > #ifdef CONFIG_SND_DEBUG_ROM > - printk(KERN_DEBUG "ROM at 0x%06x = %02x:%02x:%02x:%02x:%02x:%02x:%02x:%02x\n", bank_pos, > - iwave[0], iwave[1], iwave[2], iwave[3], > - iwave[4], iwave[5], iwave[6], iwave[7]); > + printk(KERN_DEBUG "ROM at 0x%06x = %*phC\n", bank_pos, > + 8, iwave); > #endif > if (strncmp(iwave, "INTRWAVE", 8)) > continue; /* first check */ > diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c > index 56ad923..a1d9b07 100644 > --- a/sound/usb/6fire/firmware.c > +++ b/sound/usb/6fire/firmware.c > @@ -346,11 +346,10 @@ static int usb6fire_fw_check(u8 *version) > if (!memcmp(version, known_fw_versions + i, 4)) > return 0; > > - snd_printk(KERN_ERR PREFIX "invalid fimware version in device: " > - "%02x %02x %02x %02x. " > + snd_printk(KERN_ERR PREFIX "invalid fimware version in device: %*ph. " > "please reconnect to power. if this failure " > "still happens, check your firmware installation.", > - version[0], version[1], version[2], version[3]); > + 4, version); > return -EINVAL; > } > > -- > 1.7.10.4 >

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[alsa-devel] multistreaming realtek hda codec
by Raymond Yau 06 Aug '12

06 Aug '12

>I have been trying to get a second codec working on a computer that has two codecs - one for the front speakers, >one for the rear panel line outs. So far I haven't been able to get this working. I received this message from the manufacturer of the board: >"R&D has determined that the ability to have selectable or dual line-out in the current hardware design is >limited by Linux OS kernel. The HDA sound driver did not anticipate and did not include support for two >identical audio codecs (ALC888) in a system - therefore, the 2nd codec cannot be successfully >initialized. To fix this issue, the Linux kernel level sound driver (controls function) would need to be >modified to cover full functional control of the 2nd codec. This modification is very challenging and >involves time, risk, and potential unknown side effects - because we are talking about modifying the >kernel of the Linux OS and different version of kernel may have different structures as well. Do you mean multiple streaming ? The ALC888 series are high-performance 7.1+2 Channel High Definition Audio Codecs providing ten DAC channels that simultaneously support 7.1 sound playback, plus 2 channels of independent stereo sound output (multiple streaming) through the front panel stereo outputs Both via and adi hda codecs support multiple streaming by independent headphone, you just need to assign spec->alt_dac_nid and spec->analog_stream_alt_playback in patch_realtek.c when the hda controller support enough stream and hda codec has a dac for the headphone.

1 0

[alsa-devel] [PATCH 3/3] sound/soc/soc-core.c: drop kfree of devm_kzalloc's data
by Julia Lawall 04 Aug '12

04 Aug '12

From: Julia Lawall <Julia.Lawall(a)lip6.fr> Using kfree to free data allocated with devm_kzalloc causes double frees. The semantic patch that fixes this problem is as follows: (http://coccinelle.lip6.fr/) // <smpl> @@ expression x; @@ x = devm_kzalloc(...) ... ?-kfree(x); // </smpl> Signed-off-by: Julia Lawall <Julia.Lawall(a)lip6.fr> --- The kfrees were introduced in b761c0ca. sound/soc/soc-core.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index f81c597..86d7a36 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4136,7 +4136,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "Property '%s' index %d could not be read: %d\n", propname, 2 * i, ret); - kfree(routes); return -EINVAL; } ret = of_property_read_string_index(np, propname, @@ -4145,7 +4144,6 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "Property '%s' index %d could not be read: %d\n", propname, (2 * i) + 1, ret); - kfree(routes); return -EINVAL; } }

2 1

Re: [alsa-devel] [PATCH 1/6] ASoC: dapm: If one widget fails, do not force all subsequent widgets to fail too
by Mark Brown 04 Aug '12

04 Aug '12

On Thu, Aug 02, 2012 at 08:45:18AM +0100, Lee Jones wrote: > Over time, the requests for Maintainers have Snowballed (pun intended). My > task now seems to be enabling drivers for Device Tree _and_ fix all > associated driver bugs, including any requested restructuring and API One thing to bear in mind with device tree is that it's all about defining an ABI - it's supposed to be stable and OS agnostic so it puts a lot of pressure on to really address quality problems that become visible in the bindings. This stuff is much less critical with platform data, it's relatively easy to change. > adoption. What you fail to notice is that I am only one person, and hopping > all over the code-base trying to do everyone's bidding is no mean feat. In It's fairly obvious that it's only you, or at least only you posting stuff (I know sometimes there are bigger teams behind people) - the pressure you're under to get something in is very clear. A big part of what I'm saying here is that it would be really helpful if could you slow down a bit, discuss problems more and avoid cutting corners so much. This is likely to save you time overall, you'll have a much higher success rate and you'll get much better feedback if it's clear what's going on and that there's an interest in understanding the issues. If you need to focus on device tree enablement and there are underlying problems in the subsystems you're looking at perhaps you need to push back on whoever's asking you to do the work and say you need the domain experts to pitch in and help you out. > reality I am no more obliged to fix driver bugs than you are. In fact as > the Maintainer of some of these subsystems, perhaps you could even help out > a bit? You're not telling us about the problems you see so it's very difficult for anyone to help you. For example with this patch the only information you've sent is the patch and the fact that you're seeing the error you're ignoring going off on the system you're working with (which I had to ask to find out about...). You've not shown the error message or provided any other hint which would help anyone understand why the error might be occuring and what a good solution to the problem is. Ola's guess seems very likely but nobody's got enough information to confirm that unless there's been some off-list ST/Linaro communication. Obviously with the stuff that's device specific you also have to contend with the fact that you're working on hardware which just isn't all that widely available and quite possibly has closed datasheets. > We are all trying to do good things here. Please try not to be too over- > critical. We all know Mainlining is a good thing. Perhaps less people > would be so frightened of it, thus more people would do it if the reviews > weren't such a ball-ache ( for want of a better expression :) ). This is a two way thing - one of the tools that maintainers have to try to drive up the quality of the code is to push back on poor quality submissions and devote less bandwith to sources of those submissions. It is true that there's a bunch of people who seem to view review as being just an annoying obstacle to be dealt with with the minimum possible effort but in practice all that does is make things more painful for everyone, they do tend to be more noticable because "applied, thanks" doesn't make for a big thread.

2 2

[alsa-devel] Recording sound from microphone isn't good working that playback sound is distortion.
by Xiaoyong Wen 04 Aug '12

04 Aug '12

Hi, alsa-devel Platform: S3C6410, WM9713G codec with AC'97, Linux kernel 2.6.36. ALSL 1.0.23 lib and utils. Test process: 1. aplay WAV file is good working #aplay -D 'hw:0,0' -c 2 /opt/music.wav 2. arecord WAV file from Line in is good working. #arecord -D 'hw:0,0' -r 48000 -f S16_LE -c 2 | aplay -D 'hw:0,0' -c 2 & 3. Recording sound from microphone isn't good working that playback sound is distortion. #arecord -D 'hw:0,0' -r 48000 -f S16_LE -c 2 | aplay -D 'hw:0,0' -c 2 & or #arecord -d 10 -fdat test2.wav I hear the sound is very distortion and low. ~~~~~~~~~~~~~~~~~~~~~~~~~~ If I excute "amixer cset numid=9 3; amixer cset numid=7 31; amixer cset numid=87 on; amixer cset numid=6 31,31" command, and then volume is high. Please help me to fix up this bug, thanks. Best regards. Xiaoyong Wen, <wen_kernel(a)163.com>, 2012-07-31, 14:46:17.

2 1

[alsa-devel] ASoC updates for 3.6
by Mark Brown 04 Aug '12

04 Aug '12

The following changes since commit 15d47763b354256053c390a9b7bc9f2b4f197711: Merge branch 'for-3.5' into for-3.6 (2012-07-23 10:45:07 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-3.6 for you to fetch changes up to 98d3088e534a2a61f6690b5426909b0c3b57a785: ASoC: core: Fix check before defaulting to regmap (2012-08-02 11:43:45 +0100) (this is a new subkey on my key which I just realised I didn't upload to the server - will do that on Monday.) ---------------------------------------------------------------- ASoC: Additional updates for 3.6 A few updates for issues discovered during the merge window, the main one being the fix for the issues with defaulting to use of regmap without properly checking if there was I/O in place already. ---------------------------------------------------------------- Dong Aisheng (4): ASoC: mxs-saif: fix clock prepare and enable unbalance issue ASoC: mxs-saif: set a base clock rate for EXTMASTER mode work ASoC: sgtl5000: remove unneeded snd_soc_dapm_new_widgets in probe ASoC: sgtl5000: enable VAG_POWER for LINE_IN Fabio Estevam (1): ASoC: mc13783: Provide codec->control_data Guillaume Gardet (1): ASoC: omap: Add missing modules aliases to get sound working on omap devices Lee Jones (2): ASoC: ux500: Include the correct header files ASoC: ab8500: Inform SoC Core that we have our own I/O arrangements Manuel Lauss (1): ASoC: AC97 doesn't use regmap by default Mark Brown (4): ASoC: wm8994: Ensure there are enough BCLKs for four channels ASoC: wm8994: Hold runtime PM reference while handling mic and jack IRQs ASoC: wm8962: Allow VMID time to fully ramp ASoC: core: Fix check before defaulting to regmap Roland Stigge (2): sound: tegra_wm8903: Adjust to of_get_named_gpio() change sound: tegra_alc5632: Adjust to of_get_named_gpio() change sound/soc/codecs/ab8500-codec.c | 4 ++++ sound/soc/codecs/ad1980.c | 1 + sound/soc/codecs/mc13783.c | 2 ++ sound/soc/codecs/sgtl5000.c | 3 +-- sound/soc/codecs/stac9766.c | 1 + sound/soc/codecs/wm8962.c | 3 +++ sound/soc/codecs/wm8994.c | 15 ++++++++++++++- sound/soc/codecs/wm9712.c | 1 + sound/soc/codecs/wm9713.c | 1 + sound/soc/mxs/mxs-saif.c | 24 ++++++++++++++++++++++++ sound/soc/omap/omap-mcbsp.c | 1 + sound/soc/omap/omap-pcm.c | 1 + sound/soc/soc-core.c | 2 +- sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 10 +++++----- sound/soc/ux500/ux500_msp_dai.c | 2 +- sound/soc/ux500/ux500_msp_i2s.c | 2 +- sound/soc/ux500/ux500_msp_i2s.h | 2 +- 18 files changed, 64 insertions(+), 13 deletions(-)

2 5

[alsa-devel] Clock generators in ASoC setups
by Daniel Mack 03 Aug '12

03 Aug '12

Hi Mark and Liam, some board I'm working on feature chips (connected via I2C) that are in charge of generating clocks for both the CPU DAI and the codecs. Previously, I configured them manually in platform specific code, but I wonder if there's a better, more generic way to get the job done, especially with regards to DT driven boards? Thanks, Daniel

2 3

[alsa-devel] [PATCH V4 12/16] ALSA: HDA: Make hda sound card usable for Loongson.
by Huacai Chen 03 Aug '12

03 Aug '12

Lemote A1004(Laptop) and A1205(All-In-One) use Conexant's hda codec, this patch modify patch_conexant.c to add Lemote specific code. Signed-off-by: Huacai Chen <chenhc(a)lemote.com> Signed-off-by: Hongliang Tao <taohl(a)lemote.com> Signed-off-by: Hua Yan <yanh(a)lemote.com> Cc: alsa-devel(a)alsa-project.org --- include/linux/pci_ids.h | 2 + sound/pci/hda/patch_conexant.c | 52 ++++++++++++++++++++++++++++++++++++++- 2 files changed, 52 insertions(+), 2 deletions(-) diff --git a/include/linux/pci_ids.h b/include/linux/pci_ids.h index fc35260..b28270e 100644 --- a/include/linux/pci_ids.h +++ b/include/linux/pci_ids.h @@ -2918,3 +2918,5 @@ #define PCI_DEVICE_ID_XEN_PLATFORM 0x0001 #define PCI_VENDOR_ID_OCZ 0x1b85 + +#define PCI_VENDOR_ID_LEMOTE 0x1c06 diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 1436118..6bbac39 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -142,6 +142,7 @@ struct conexant_spec { unsigned int thinkpad:1; unsigned int hp_laptop:1; unsigned int asus:1; + unsigned int lemote:1; unsigned int pin_eapd_ctrls:1; unsigned int fixup_stereo_dmic:1; @@ -2280,7 +2281,7 @@ static void cxt5066_automic(struct hda_codec *codec) cxt5066_thinkpad_automic(codec); else if (spec->hp_laptop) cxt5066_hp_laptop_automic(codec); - else if (spec->asus) + else if (spec->asus || spec->lemote) cxt5066_asus_automic(codec); } @@ -2913,6 +2914,32 @@ static const struct hda_verb cxt5066_init_verbs_hp_laptop[] = { { } /* end */ }; +static struct hda_verb cxt5066_init_verbs_lemote[] = { + {0x14, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1: Connection index: 0 */ + {0x19, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_HP_EVENT}, + {0x1b, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | CONEXANT_MIC_EVENT}, + + /* DAC2: unused */ + {0x11, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + + /* ADC2, ADC3: unused */ + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(2)}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(3)}, + + /* Disable digital microphone port */ + {0x23, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + + /* Disable SPDIF */ + {0x20, AC_VERB_SET_PIN_WIDGET_CONTROL, 0}, + { } /* end */ +}; + /* initialize jack-sensing, too */ static int cxt5066_init(struct hda_codec *codec) { @@ -2950,6 +2977,8 @@ enum { CXT5066_THINKPAD, /* Lenovo ThinkPad T410s, others? */ CXT5066_ASUS, /* Asus K52JU, Lenovo G560 - Int mic at 0x1a and Ext mic at 0x1b */ CXT5066_HP_LAPTOP, /* HP Laptop */ + CXT5066_LEMOTE_A1004, /* Lemote Laptop A1004 */ + CXT5066_LEMOTE_A1205, /* Lemote All-In-One A1205 */ CXT5066_AUTO, /* BIOS auto-parser */ CXT5066_MODELS }; @@ -2963,6 +2992,8 @@ static const char * const cxt5066_models[CXT5066_MODELS] = { [CXT5066_THINKPAD] = "thinkpad", [CXT5066_ASUS] = "asus", [CXT5066_HP_LAPTOP] = "hp-laptop", + [CXT5066_LEMOTE_A1004] = "lemote-laptop-a1004", + [CXT5066_LEMOTE_A1205] = "lemote-aio-a1205", [CXT5066_AUTO] = "auto", }; @@ -2995,6 +3026,8 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo U350", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo G560", CXT5066_ASUS), SND_PCI_QUIRK(0x17aa, 0x3938, "Lenovo G565", CXT5066_AUTO), + SND_PCI_QUIRK(PCI_VENDOR_ID_LEMOTE, 0x2011, "Lemote A1004", CXT5066_LEMOTE_A1004), + SND_PCI_QUIRK(PCI_VENDOR_ID_LEMOTE, 0x2012, "Lemote A1205", CXT5066_LEMOTE_A1205), SND_PCI_QUIRK(0x1b0a, 0x2092, "CyberpowerPC Gamer Xplorer N57001", CXT5066_AUTO), {} }; @@ -3075,7 +3108,22 @@ static int patch_cxt5066(struct hda_codec *codec) spec->port_d_mode = 0; spec->mic_boost = 3; /* default 30dB gain */ break; - + case CXT5066_LEMOTE_A1004: + case CXT5066_LEMOTE_A1205: + codec->patch_ops.init = cxt5066_init; + codec->patch_ops.unsol_event = cxt5066_unsol_event; + spec->init_verbs[spec->num_init_verbs] = + cxt5066_init_verbs_lemote; + spec->num_init_verbs++; + spec->lemote = 1; + spec->mixers[spec->num_mixers++] = cxt5066_mixer_master; + spec->mixers[spec->num_mixers++] = cxt5066_mixers; + /* no S/PDIF out */ + /* input source automatically selected */ + spec->input_mux = NULL; + spec->port_d_mode = 0; + spec->mic_boost = 3; /* default 30dB gain */ + break; case CXT5066_OLPC_XO_1_5: codec->patch_ops.init = cxt5066_olpc_init; codec->patch_ops.unsol_event = cxt5066_olpc_unsol_event; -- 1.7.7.3

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[alsa-devel] Move snd_legacy_find_free_ioport to initval.h
by Ondrej Zary 03 Aug '12

03 Aug '12

Move snd_legacy_find_free_ioport() function back to initval.h as it is used by two drivers. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> diff --git a/include/sound/initval.h b/include/sound/initval.h index f99a0d2..ac62c67 100644 --- a/include/sound/initval.h +++ b/include/sound/initval.h @@ -50,6 +50,20 @@ #define SNDRV_DEFAULT_DMA_SIZE { [0 ... (SNDRV_CARDS-1)] = SNDRV_AUTO_DMA_SIZE } #define SNDRV_DEFAULT_PTR SNDRV_DEFAULT_STR +#ifdef SNDRV_LEGACY_FIND_FREE_IOPORT +static long snd_legacy_find_free_ioport(long *port_table, long size) +{ + while (*port_table != -1) { + if (request_region(*port_table, size, "ALSA test")) { + release_region(*port_table, size); + return *port_table; + } + port_table++; + } + return -1; +} +#endif + #ifdef SNDRV_LEGACY_FIND_FREE_IRQ #include <linux/interrupt.h> diff --git a/sound/isa/opti9xx/miro.c b/sound/isa/opti9xx/miro.c index c24594c..3d1afb6 100644 --- a/sound/isa/opti9xx/miro.c +++ b/sound/isa/opti9xx/miro.c @@ -37,6 +37,7 @@ #include <sound/opl4.h> #include <sound/control.h> #include <sound/info.h> +#define SNDRV_LEGACY_FIND_FREE_IOPORT #define SNDRV_LEGACY_FIND_FREE_IRQ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> @@ -770,20 +771,6 @@ static int __devinit snd_miro_mixer(struct snd_card *card, return 0; } -static long snd_legacy_find_free_ioport(long *port_table, long size) -{ - while (*port_table != -1) { - struct resource *res; - if ((res = request_region(*port_table, size, - "ALSA test")) != NULL) { - release_and_free_resource(res); - return *port_table; - } - port_table++; - } - return -1; -} - static int __devinit snd_miro_init(struct snd_miro *chip, unsigned short hardware) { diff --git a/sound/isa/opti9xx/opti92x-ad1848.c b/sound/isa/opti9xx/opti92x-ad1848.c index f8fbe22..2899c9f 100644 --- a/sound/isa/opti9xx/opti92x-ad1848.c +++ b/sound/isa/opti9xx/opti92x-ad1848.c @@ -39,6 +39,7 @@ #ifndef OPTi93X #include <sound/opl4.h> #endif +#define SNDRV_LEGACY_FIND_FREE_IOPORT #define SNDRV_LEGACY_FIND_FREE_IRQ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> @@ -185,19 +186,6 @@ static char * snd_opti9xx_names[] = { "82C930", "82C931", "82C933" }; - -static long __devinit snd_legacy_find_free_ioport(long *port_table, long size) -{ - while (*port_table != -1) { - if (request_region(*port_table, size, "ALSA test")) { - release_region(*port_table, size); - return *port_table; - } - port_table++; - } - return -1; -} - static int __devinit snd_opti9xx_init(struct snd_opti9xx *chip, unsigned short hardware) { -- Ondrej Zary

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[alsa-devel] DT support for Davinci ASoC platform
by Hebbar, Gururaja 02 Aug '12

02 Aug '12

All, Are there any plans or work going on for providing Device Tree Support for Davinci ASoC? I am planning to add DT support for TI AM335x (McASP) and I don't want to do duplicate task. Regards Gururaja

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[alsa-devel] Alsa: USB Class 2 Audio soundcard device disappears suddenly
by Joao Bonina 02 Aug '12

02 Aug '12

It doesn't show anything when that happens. Looking at dmesg (I'm attaching it), it shows this: [ 350.170478] usb 1-4: config 1 has an invalid interface number: 3 but max is 2 [ 350.170500] usb 1-4: config 1 has no interface number 2 [ 350.170843] usb 1-4: config 1 has an invalid interface number: 3 but max is 2 [ 350.170864] usb 1-4: config 1 has no interface number 2 [ 350.171344] usb 1-4: New USB device found, idVendor=21b4, idProduct=0230 [ 350.171366] usb 1-4: New USB device strings: Mfr=1, Product=2, SerialNumber=3 [ 350.171385] usb 1-4: Product: Ayre USB Interface [ 350.171401] usb 1-4: Manufacturer: Ayre Acoustics [ 350.171418] usb 1-4: SerialNumber: (C) 2010 XMOS/Wavelength Audio Don't know if those 'invalid interface number' messages are relevant for the issue. Joao ----- Original Message ----- From: Daniel Mack <zonque(a)gmail.com> To: Joao Bonina <bonina_2001(a)yahoo.com> Cc: "alsa-devel(a)alsa-project.org" <alsa-devel(a)alsa-project.org> Sent: Tuesday, 31 July 2012, 6:24 Subject: Re: [alsa-devel] Alsa: USB Class 2 Audio soundcard device disappears suddenly On 31.07.2012 00:32, Joao Bonina wrote: > Hi, > > I'm using an USB Class 2 Audio DAC (basically a soundcard) with my Linux box and MPD software. > > It works great when playing files stored in a remote share, but when > playing files from an USB pen, it stops playing in a few minutes. > > Mpd.log shows this: > > Code: > Jul 30 13:03 : output: "USB" [alsa] failed to play: No such device > Jul 30 13:03 : output: closed plugin=alsa name="USB" > ALSA lib pcm_hw.c:1401:(_snd_pcm_hw_open) Invalid value for card > Jul 30 13:03 : output: Failed to open "USB" [alsa]: Failed to open ALSA device "hw:0,0": No such file or directoryIt seems to be somewhat related to the USB bus not being able to > cope with both PCM streaming to the soundcard and reading from the flash pen simultaneously, I guess. > > So, anybody faced this issue previously and can provide some hints to solve this? > > MPD is 0.17, kernel is 3.2.17, alsa-utils is 1.0.25. Does the kernel log or 'dmesg' show anything weird what that happens? Daniel _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] [PATCH 01/19] Configure ux500 Audio Drivers for Device Tree
by Lee Jones 02 Aug '12

02 Aug '12

This patch-set sees many of the ux500 related Audio drivers gain Device Tree enablement. There are also some fixes to issues which where encountered along the way. Documentation/devicetree/bindings/mfd/ab8500.txt | 15 + .../devicetree/bindings/sound/ux500-mop500.txt | 39 ++ .../devicetree/bindings/sound/ux500-msp.txt | 43 ++ arch/arm/boot/dts/db8500.dtsi | 597 ------------------ arch/arm/boot/dts/dbx5x0.dtsi | 636 ++++++++++++++++++++ arch/arm/boot/dts/snowball.dts | 18 +- arch/arm/mach-ux500/Makefile | 2 +- arch/arm/mach-ux500/board-mop500-audio.c | 189 ++++++ arch/arm/mach-ux500/board-mop500-msp.c | 267 -------- arch/arm/mach-ux500/board-mop500-msp.h | 14 - arch/arm/mach-ux500/board-mop500.c | 22 +- arch/arm/mach-ux500/board-mop500.h | 6 + arch/arm/mach-ux500/include/mach/msp.h | 2 - include/linux/mfd/abx500/ab8500-codec.h | 6 +- sound/soc/codecs/ab8500-codec.c | 81 +++ sound/soc/ux500/mop500.c | 42 +- sound/soc/ux500/ux500_msp_dai.c | 20 +- sound/soc/ux500/ux500_msp_i2s.c | 97 ++- sound/soc/ux500/ux500_msp_i2s.h | 2 - sound/soc/ux500/ux500_pcm.c | 19 +- sound/soc/ux500/ux500_pcm.h | 3 + 21 files changed, 1189 insertions(+), 931 deletions(-)

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[alsa-devel] [PATCH] ASoC: core: Fix check before defaulting to regmap
by Mark Brown 02 Aug '12

02 Aug '12

Check if the chip has provided a write operation (which is mandatory for I/O) rather than looking for control data as some of the MFDs use a global for this. Also skip the attempt if there's no regmap available by device in case things get confused by the attempt to default. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b1fd043..5359623 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1096,7 +1096,7 @@ static int soc_probe_codec(struct snd_soc_card *card, } /* If the driver didn't set I/O up try regmap */ - if (!codec->control_data) + if (!codec->write && dev_get_regmap(codec->dev, NULL)) snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); if (driver->controls) -- 1.7.10.4

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[alsa-devel] [ALC882/883/885/888/889] Problems with snd-hda-intel after 3.1.6 and the FIX
by Misha Ketslah 02 Aug '12

02 Aug '12

Hello, As i can't reach bugtracker (is it down, or problem on my side?), i'll post here. Though there is already few open bugs there i saw. I just found temp solution. Bug itslef: After updating to greater version of kernel than 3.1.6 speakers stoped working anymore on most vaio z and similar models, though there is still sound through headphones. (ithere are a lot of people reporting it on different distro bug trackers and forums as well) My specific cards are 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 05) 01:00.1 Audio device: NVIDIA Corporation High Definition Audio Controller (rev ff) The only solution was to use model=generic, but than most of features wouldn't work.(plugin jack automute, missing most settings in alsamixer). Solution: And as bug was reported for a long time, i waited through kernel updates to fix it. But yesterday as i updated to 3.5 it was still there. And as 3.5 has fewer models for [ALC882/883/885/888/889], as my card is ALC889 i just tried all 3 of them, and it seems everyting working when setting model=acer-aspire-8930g (cant and didn't try HDMI out). But the problem is that it still not that obvious for people with other than acer laptops to set this as model, as well as before (<3.1.6) it worked out-of-box with not model setting at all. thanks, Michael Ketslah. -- 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

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[alsa-devel] [RFC PATCH 1/1] ASoC: soc-dapm: warn if no avaiable path found for widget kcontrol
by Dong Aisheng 02 Aug '12

02 Aug '12

From: Dong Aisheng <dong.aisheng(a)linaro.org> This could help people know what's wrong going on if the required path is not registered. Signed-off-by: Dong Aisheng <dong.aisheng(a)linaro.org> --- sound/soc/soc-dapm.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4d181df..b9b94c5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1879,6 +1879,9 @@ static int soc_dapm_mux_update_power(struct snd_soc_dapm_widget *widget, } } + WARN(!found, "did not find dapm widget path associated with kcontrol %s\n", + kcontrol->id.name); + if (found) { dapm_mark_dirty(widget, "mux change"); dapm_power_widgets(widget->dapm, SND_SOC_DAPM_STREAM_NOP); -- 1.7.0.4

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