Summary: | Regression: No right channel sound on Creative Xtreme Audio (CA0106) | ||
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Product: | PulseAudio | Reporter: | Luke <lukebenes> |
Component: | alsa | Assignee: | pulseaudio-bugs |
Status: | RESOLVED WORKSFORME | QA Contact: | pulseaudio-bugs |
Severity: | normal | ||
Priority: | medium | CC: | lennart, superquad.vortex2 |
Version: | unspecified | ||
Hardware: | Other | ||
OS: | All | ||
See Also: |
https://launchpad.net/bugs/1497666 https://bugzilla.kernel.org/show_bug.cgi?id=118161 |
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Whiteboard: | |||
i915 platform: | i915 features: | ||
Attachments: |
PulseAudio verbose log
New PulseAudio Verbose Log with updated analog-output.conf + default.conf |
Description
Luke
2015-10-27 11:06:29 UTC
how about aad the following elements to http://cgit.freedesktop.org/pulseaudio/pulseaudio/plain/src/modules/alsa/mixer/paths/analog-output.conf [Element Analog Front] switch = mute volume = merge override-map.1 = all-front override-map.2 = front-left,front-right [Element Analog Surround] switch = mute volume = merge override-map.1 = all-rear override-map.2 = rear-left,rear-right [Element Analog Side] switch = mute volume = merge override-map.1 = all-side override-map.2 = side-left,side-right [Element Analog Center/LFE] switch = mute volume = merge override-map.1 = all-center override-map.2 = all-center,lfe Hi Luke, Can you try the suggestion by Raymond in the previous comment? I e, edit /usr/share/pulseaudio/alsa-mixer/paths/analog-output.conf and add these lines: ;;; cut here ;;; [Element Analog Front] switch = mute volume = merge override-map.1 = all-front override-map.2 = front-left,front-right [Element Analog Surround] switch = mute volume = merge override-map.1 = all-rear override-map.2 = rear-left,rear-right [Element Analog Side] switch = mute volume = merge override-map.1 = all-side override-map.2 = side-left,side-right [Element Analog Center/LFE] switch = mute volume = merge override-map.1 = all-center override-map.2 = all-center,lfe ;;; cut here ;;; Then reboot your computer for changes to take effect. Do you now have the possibility to change balance on the different outputs correctly, so that PulseAudio controls these alsamixer sliders when you use the normal GUI (e g unity-control-center)? If not, please also attach the output of "pactl list". https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/ca0106/ca0106_mixer.c?id=49c88b85b53767f97eb8c9171cb0b976c62a0114 the virtual master control both analog and digital volume 000) I: [pulseaudio] (alsa-lib)setup.c: Cannot lock ctl elem ( 0.355| 0.000) I: [pulseaudio] (alsa-lib)setup.c: Cannot lock ctl elem ( 0.355| 0.000) I: [pulseaudio] (alsa-lib)setup.c: Cannot lock ctl elem ( 0.355| 0.000) I: [pulseaudio] (alsa-lib)setup.c: Cannot lock ctl elem ( 0.355| 0.000) D: [pulseaudio] alsa-util.c: Set neither period nor buffer size. ( 0.356| 0.000) I: [pulseaudio] (alsa-lib)setup.c: Cannot lock ctl elem ( 0.356| 0.000) I: [pulseaudio] alsa-util.c: snd_pcm_hw_params failed: Device or resource busy ( 0.356| 0.000) I: [pulseaudio] alsa-util.c: Failed to set hardware parameters on plug:iec958:0: Device or resource busy there are locks associated with iec958 playback you need to use type asym for ca0106 iec958 to prevent lock when pulseaudio open iec958 capture http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/paths why pulseaudio only has iec958-stereo-output.conf but no iec958-stereo-input.conf when http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/profile-sets/default.conf [Mapping iec958-stereo] device-strings = iec958:%f channel-map = left,right paths-input = iec958-stereo-input paths-output = iec958-stereo-output priority = 5 you need ice958-stereo-input.conf for selecting digital source Simple mixer control 'Digital Source',0 Capabilities: cenum Items: 'IEC958 out' 'i2s mixer out' 'IEC958 in' 'i2s in' 'AC97 in' 'SRC out' Item0: 'i2s in' After editing edit /usr/share/pulseaudio/alsa-mixer/paths/analog-output.conf, The Unity sound control panel showed the balance as "Left" (it was at the "Center" position before the edit). After moving the slider from "Left" to "Center", sound played out of the Right speaker. Would you like any more information? @Raymond This is still marked NEEDINFO. What else do you need? [Mapping iec958-stereo] device-strings = iec958:%f channel-map = left,right paths-input = iec958-stereo-input paths-output = iec958-stereo-output priority = 5 You can remove the iec958-stereo-input from pulseaudio default.conf http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/profile-sets/default.conf You need to provide pulseaudio verbose log after you added those analog elements and force pulseaudio not probe iec958 input 0.304| 0.000) D: [pulseaudio] alsa-mixer.c: Available mixer paths (after tidying): ( 0.304| 0.000) D: [pulseaudio] alsa-mixer.c: Path Set 0x8628a98, direction=1 ( 0.304| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=255, min_dB=-100000, max_dB=0 ( 0.304| 0.000) D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes ( 0.304| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no It seem ignore those ca0106 analog playback volume control and just found the mono virtual master playback volume Created attachment 123531 [details] New PulseAudio Verbose Log with updated analog-output.conf + default.conf Sorry for the delay, I was away from my desktop for a few months. I have done the following: * Followed the instructions here: http://ubuntuforums.org/showthread.php?t=2210602 to build from source * Removed the 1 line with iec958-stereo-input from src/modules/alsa/mixer/profile-sets/default.conf * Added the elements to src/modules/alsa/mixer/paths/analog-output.conf * sudo make install * Rebooted * Created PulseAudio verbose log As I said earlier, with the new analog-output.conf elements, this issue is fixed. you still have locking problem when pulseaudio try to open ice958 playback and ice958 capture 0.374| 0.000) D: [pulseaudio] alsa-mixer.c: Looking at profile output:iec958-stereo+input:iec958-stereo ( 0.374| 0.000) D: [pulseaudio] alsa-mixer.c: Checking for recording on Digital Stereo (IEC958) (iec958-stereo) ( 0.374| 0.000) D: [pulseaudio] alsa-util.c: Trying iec958:0 with SND_PCM_NO_AUTO_FORMAT ... ( 0.374| 0.000) D: [pulseaudio] alsa-util.c: Managed to open iec958:0 ( 0.375| 0.000) D: [pulseaudio] alsa-util.c: Maximum hw buffer size is 340 ms ( 0.375| 0.000) I: [pulseaudio] (alsa-lib)setup.c: Cannot lock ctl elem ( 0.375| 0.000) I: [pulseaudio] (alsa-lib)setup.c: Cannot lock ctl elem ( 0.375| 0.000) I: [pulseaudio] (alsa-lib)setup.c: Cannot lock ctl elem ( 0.375| 0.000) I: [pulseaudio] (alsa-lib)setup.c: Cannot lock ctl elem you need an asym plugin for ice958 playback and capture so that it only lock playback ctl http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/conf/cards/CA0106.conf;hb=HEAD#l223 Raymond, I' Raymond, I'm not a developer. Could you please give me detailed instructions on what you want me to do? Thank you for all your help on this issue. http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html just make sure that when application open iec958 capture won't lock those playback ctl elems Plugin: asym This plugin is a combination of playback and capture PCM streams. Slave PCMs can be defined asymmetrically for both directions. pcm.name { type asym # Asym PCM playback STR # Playback slave name # or playback { # Playback slave definition pcm STR # Slave PCM name # or pcm { } # Slave PCM definition } capture STR # Capture slave name # or capture { # Capture slave definition pcm STR # Slave PCM name # or pcm { } # Slave PCM definition } } I still don't quite follow what you want. I tried adding your Plugin: asym code to a new file, $HOME/.asoundrc . After a reboot, no sound would play. Do you want a PulseAudio verbose log of this? ( 0.327| 0.000) D: [pulseaudio] alsa-mixer.c: Path analog-output (Analog Output), direction=1, priority=99, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=255, min_dB=-200000, max_dB=12 ( 0.327| 0.000) D: [pulseaudio] alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=yes ( 0.327| 0.000) D: [pulseaudio] alsa-mixer.c: Element Analog Front, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x6, n_channels=2, override_map=yes ( 0.327| 0.000) D: [pulseaudio] alsa-mixer.c: Element Analog Side, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0xc00, n_channels=2, override_map=yes ( 0.327| 0.000) D: [pulseaudio] alsa-mixer.c: Element Analog Center/LFE, direction=1, switch=1, volume=1, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x4900000000098, n_channels=2, override_map=yes ( 0.327| 0.000) D: [pulseaudio] alsa-mixer.c: Element IEC958, direction=1, switch=2, volume=0, volume_limit=-1, enumeration=0, required=0, required_any=0, required_absent=0, mask=0x0, n_channels=0, override_map=no why iec958 switch in analog path ? as 99999.99 represent infinite dB , how can mindB equal to -20000dB control.5 { iface MIXER name 'Analog Front Playback Volume' value.0 207 value.1 207 comment { access 'read write' type INTEGER count 2 range '0 - 255' dbmin -9999999 dbmax 1200 dbvalue.0 0 dbvalue.1 0 } } control.38 { iface MIXER name 'Master Playback Volume' value 202 comment { access 'read write' type INTEGER count 1 range '0 - 255' dbmin -9999999 dbmax 0 dbvalue.0 -1325 } } if the master playback volume control is virtual master, it is strange that dB max is different from slaves https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/ca0106?id=601e1cc5df940b59e71c947726640811897d30df CA0106.pcm.iec958.0 { @args [ CARD AES0 AES1 AES2 AES3 ] @args.CARD { type string } @args.AES0 { type integer } @args.AES1 { type integer } @args.AES2 { type integer } @args.AES3 { type integer } type asym playback { type hooks slave.pcm { type hw card $CARD } hooks.0 { type ctl_elems hook_args [ { name "IEC958 Front Playback Volume" index 0 lock true preserve true value [ 207 207 ] # Puts 0x30303030 in the Volume register. 0xff - 0x30 = 0xcf = 207 } { name "IEC958 Playback Switch" lock true preserve true value 1 } { interface PCM name "IEC958 Playback Default" index 1 lock true preserve true optional true value [ $AES0 $AES1 $AES2 $AES3 ] } { # for compatibility with older drivers name "IEC958 Playback Default" index 1 lock true preserve true optional true value [ $AES0 $AES1 $AES2 $AES3 ] } ] } } capture { slave.pcm { type hw card $CARD } } } Raymond, Again, I'm not sure what you need me to do. Please give me clear instructions and I'll do my best to follow them. you have to change CA0106.pcm.iec958.0 in /usr/share/alsa/cards/CA0106.conf to use asym plugin so that open iec958 capture won't lock those iec958 playback controls seem syntax of capture should be capture { type hw card $CARD } instead of capture { slave.pcm { type hw card $CARD } } for the max dB bug. you have to ask the author of this patch +static int slave_tlv_cmd(struct snd_kcontrol *kcontrol, + int op_flag, unsigned int size, + unsigned int __user *tlv) +{ + struct link_slave *slave = snd_kcontrol_chip(kcontrol); + /* FIXME: this assumes that the max volume is 0 dB */ + return slave->slave.tlv.c(&slave->slave, op_flag, size, tlv); +}ound.git/commit/sound/core/vmaster.c?id=e922b0028fad87de0d262f9fa51f98595d2df258 the modified ca0106.conf should allow you to run arecord and aplay with ice958 of your ca0106 at the same time without "Cannot lock ctl elem" error Raymond, Since you understand this issue far better than I do, could you please ask him for me? I opened a new issue over here: https://bugzilla.kernel.org/show_bug.cgi?id=118161 After I modify A0106.pcm.iec958.0 in /usr/share/alsa/cards/CA0106.conf, what next? Another PulseAudio verbose log? how do you test ? speaker-test -c 2 -t wav -D hw:CARD=CA0106 do you get the correct sound if the application use alsa hw device ? speaker-test -c2 -t wav -D pulse From https://bugzilla.kernel.org/show_bug.cgi?id=118161 > I installed Lubuntu 12.04, 3.2.0-102-generic-pae. Both channels played. I > upgraded the kernel to 3.10.101-0310101-generic from here: > http://kernel.ubuntu.com/~kernel-ppa/mainline/ > Unlike the bootable iso image, once installed it had a /var/lib/alsa/asound.state > I removed this and did a $ alsa force-reload and $ alsactl init, but > asound.state wasn't recreated. Also the old alsamixer settings seemed to > remain. How do I reset ALL the alsa settings to reproduce this bug with newer > kernels? Since upgrading the kernel didn't trigger this bug, does it mean that it's an also configuration issue? I can reproduce this issue on Lubuntu (no PA), so should I close this bug? https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/log/sound/pci/ca0106 only a few change in ca0106 driver, so it is unlikely driver bug if speaker-test with hw devuce still work Verified fixed with a clean install of Ubuntu 16.10. $ uname -a Linux luke-PowerEdge-T105 4.4.0-9136-generic #55-Ubuntu SMP Fri Aug 26 05:58:34 UTC 2016 x86_64 x86_64 x86_64 GNU/Linux Raymond, Thank you so much for all your help on this issue! |
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