2009/12/9 Takashi Iwai <tiwai@xxxxxxx> > At Wed, 9 Dec 2009 19:55:15 +0800, > Raymond Yau wrote: > > > > 2009/12/8 Takashi Iwai <tiwai@xxxxxxx> > > > > > At Sat, 5 Dec 2009 10:17:07 +0800, > > > Raymond Yau wrote: > > > > > > > > 2009/12/4 pl bossart <bossart.nospam@xxxxxxxxx> > > > > > > > > > >> 17. hwptr 24351 buffer 16384 appl 24332, avail=16403 > > > > > > > > > > > > appl pointer is behind hwprtr ( underrun ) , need timestamp to > find > > > out > > > > > > whether PA is unable to write audio data in time since > > > > > > > > > > I commented out the portion of the code that increases the > watermark > > > and > > > > > clamped the number of available samples to the buffer size, and it > > > works > > > > > just fine, no audio > > > > > underflows. > > > > > > > > > > > > > Do you mean that the underrun is related the the portion of code that > > > > increase the watermark (i.e It is a pulseaudio bug instead of driver > bug > > > ) > > > > ? > > > > > > Or, we can try simply to change to periods_min = 2 whether this fixes > > > the PA issues. > > > > > > periods_min = 1 is anyway fairly fragile. With that setup, apps are > > > supposed (implicitly) to wake up frequently enough to get pointers in > > > sync. This is because you can't trust whether the IRQ wake up is > > > really fast enough. There are machines that aren't. > > > > > > > > > thanks, > > > > > > Takashi > > > > > > > The intel8x0 driver has code to handle one period per buffer , > > Yes, but it doesn't work so reliably. > That's why I asked for testing periods_min = 2. > > > Is the feature "one period per buffer" designed for dmix ? > > No. > > > The system timer may be running faster or slower than the sound card > > on differnt machine > > Yes. > > > PA only adjust the watermark/sleeping time in one direction only > > Yes. > > > thanks, > > Takashi > Unlike the convention sound card which has only one set of hardware registers, aureal au88x0 chips has 4 set of hardware registers ( sub-buffers as descibed in patent 6167465 ) The maximum period size is 4k bytes and the driver support 2 to 32 periods. This is exactly as same as the sound card in the figure *Image 1:* *Schematic overview of the playback buffer in the traditional playback model, in the best way the author can visualize this with his limited drawing abilities.* http://0pointer.de/blog/projects/pulse-glitch-free.html 1) what is hwbuf_unused_frames ? 2) why PA use snd_pcm_hw_params_get_buffer_size_max() instead of snd_pcm_hw_params_get_buffer_size() after snd_pcm_hw_params() ? D: alsa-util.c: Maximum hw buffer size is 371 ms I: module-alsa-sink.c: Using 2 fragments of size 4096 bytes, buffer time is 46.44ms D: alsa-util.c: buffer_size : 2048 D: alsa-util.c: period_size : 1024 3) why PA alway open CTL device "front:0" instead of "hw:0" ? I: (alsa-lib)control.c: Invalid CTL front:0 I: alsa-util.c: Unable to attach to mixer front:0: No such file or directory I: alsa-util.c: Successfully attached to mixer 'hw:0' 4) did PA has any data to play when the server probe the devices ? D: module-alsa-sink.c: Thread starting up D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+29 I: module-alsa-sink.c: Starting playback. 5) it seem bug in snd_pcm_sw_params_set_avail_min() since 4661 is larger than the buffer size 2048 since PA still think that 371ms is the maximum buffer size D: alsa-util.c: buffer_size : 2048 D: alsa-util.c: period_size : 1024 D: alsa-util.c: period_time : 23219 D: alsa-util.c: tstamp_mode : NONE D: alsa-util.c: period_step : 1 D: alsa-util.c: avail_min : 4661 6) when using the minimum period size of alsa-pulse plugin and aplay on pulseaudio 0.9.19 it seem that underrun occur between aplay and PA server I: alsa-sink.c: Increasing wakeup watermark to 7,98 ms D: protocol-native.c: Underrun on 'ALSA Playback', 0 bytes in queue. D: sink-input.c: Requesting rewind due to corking For pulseaudio-0.9.14 on Fedora 10 D: module-hal-detect.c: Not loaded device /org/freedesktop/Hal/devices/pci_12eb_2_sound_card_0_alsa_playback_1 D: module-hal-detect.c: Loading module-alsa-sink with arguments 'device_id=0 sink_name=alsa_output.pci_12eb_2_sound_card_0_alsa_playback_0 tsched=1' D: alsa-util.c: Trying front:0 with SND_PCM_NO_AUTO_FORMAT ... D: alsa-util.c: Maximum hw buffer size is 371 ms ... I: module-alsa-sink.c: Successfully opened device front:0. I: module-alsa-sink.c: Successfully enabled mmap() mode. I: module-alsa-sink.c: Successfully enabled timer-based scheduling mode. I: (alsa-lib)control.c: Invalid CTL front:0 I: alsa-util.c: Unable to attach to mixer front:0: No such file or directory I: alsa-util.c: Successfully attached to mixer 'hw:0' I: alsa-util.c: Using mixer control "Master". I: sink.c: Created sink 0 "alsa_output.pci_12eb_2_sound_card_0_alsa_playback_0" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: source.c: Created source 0 "alsa_output.pci_12eb_2_sound_card_0_alsa_playback_0.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: module-alsa-sink.c: Using 2 fragments of size 4096 bytes, buffer time is 46.44ms I: module-alsa-sink.c: Time scheduling watermark is 20.00ms D: module-alsa-sink.c: hwbuf_unused_frames=0 D: module-alsa-sink.c: setting avail_min=4661 I: module-alsa-sink.c: Volume ranges from 0 to 31. I: module-alsa-sink.c: Volume ranges from -46.50 dB to 0.00 dB. I: alsa-util.c: All 2 channels can be mapped to mixer channels. I: module-alsa-sink.c: Using hardware volume control. Hardware dB scale supported. D: alsa-util.c: snd_pcm_dump(): D: alsa-util.c: Hardware PCM card 0 'Aureal Vortex au8830' device 0 subdevice 0 D: alsa-util.c: Its setup is: D: alsa-util.c: stream : PLAYBACK D: alsa-util.c: access : MMAP_INTERLEAVED D: alsa-util.c: format : S16_LE D: alsa-util.c: subformat : STD D: alsa-util.c: channels : 2 D: alsa-util.c: rate : 44100 D: alsa-util.c: exact rate : 44100 (44100/1) D: alsa-util.c: msbits : 16 D: alsa-util.c: buffer_size : 2048 D: alsa-util.c: period_size : 1024 D: alsa-util.c: period_time : 23219 D: alsa-util.c: tstamp_mode : NONE D: alsa-util.c: period_step : 1 D: alsa-util.c: avail_min : 4661 D: alsa-util.c: period_event : 0 D: alsa-util.c: start_threshold : -1 D: alsa-util.c: stop_threshold : -1 D: alsa-util.c: silence_threshold: 0 D: alsa-util.c: silence_size : 0 D: alsa-util.c: boundary : 1073741824 D: alsa-util.c: appl_ptr : 0 D: alsa-util.c: hw_ptr : 0 D: module-alsa-sink.c: Read hardware volume: 0: 64% 1: 64% D: module-alsa-sink.c: Thread starting up D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+29 I: module-alsa-sink.c: Starting playback. ... I: module-alsa-sink.c: Trying resume... D: alsa-util.c: Maximum hw buffer size is 371 ms D: module-alsa-sink.c: hwbuf_unused_frames=0 D: module-alsa-sink.c: setting avail_min=4661 I: module-alsa-sink.c: Resumed successfully... I: module-alsa-sink.c: Starting playback. D: module-suspend-on-idle.c: Sink alsa_output.pci_12eb_2_sound_card_0_alsa_playback_0 becomes idle. D: module-suspend-on-idle.c: Sink alsa_output.pci_12eb_2_sound_card_0_alsa_playback_0 becomes busy. D: memblockq.c: memblockq requested: maxlength=33554432, tlength=0, base=4, prebuf=0, minreq=1 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=33554432, tlength=33554432, base=4, prebuf=0, minreq=4 maxrewind=0 I: sink-input.c: Created input 1 "ALSA Playback" on alsa_output.pci_12eb_2_sound_card_0_alsa_playback_0 with sample spec s16le 2ch 44100Hz and channel map front-left,front-right I: protocol-native.c: Requested tlength=500.00 ms, minreq=124.99 ms D: protocol-native.c: Early requests mode enabled, configuring sink latency to minreq. D: memblockq.c: memblockq requested: maxlength=4194304, tlength=88200, base=4, prebuf=66152, minreq=22048 maxrewind=0 D: memblockq.c: memblockq sanitized: maxlength=4194304, tlength=88200, base=4, prebuf=66152, minreq=22048 maxrewind=0 I: protocol-native.c: Final latency 624.99 ms = 250.02 ms + 2*124.99 ms + 124.99 ms D: module-alsa-sink.c: latency set to 46.44ms D: module-alsa-sink.c: hwbuf_unused_frames=1 D: module-alsa-sink.c: setting avail_min=4662 D: module-alsa-sink.c: Requesting rewind due to latency change. D: module-alsa-sink.c: Requested to rewind 8192 bytes. D: module-alsa-sink.c: Limited to 3712 bytes. D: module-alsa-sink.c: before: 928 D: module-alsa-sink.c: after: 928 D: module-alsa-sink.c: Rewound 3712 bytes. D: sink.c: Processing rewind... D: sink-input.c: Have to rewind 3712 bytes on render memblockq. https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4776 hyper_sonic 's au8830 a bit lagged .But if switch to integrated i8x0 sound chip music played very well. on pulseaudio 0.9.19 PA seem to use 16 fragments instead of 2 fragments for au8830 I: alsa-sink.c: Using 16,0 fragments of size 4096 bytes (23,22ms), buffer size is 65536 bytes (371,52ms) I: alsa-sink.c: Time scheduling watermark is 20,00ms D: alsa-sink.c: hwbuf_unused=0 D: alsa-sink.c: setting avail_min=15502 D: alsa-mixer.c: Activating path analog-output-lfe-on-mono D: alsa-mixer.c: Path analog-output-lfe-on-mono (Analog Output (LFE)), direction=1, priority=40, probed=yes, supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0, max_volume=31, min_dB=-81, max_dB=12 D: alsa-mixer.c: Element Master, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x3600000000f66, n_channels=2, override_map=yes D: alsa-mixer.c: Element Master Mono, direction=1, switch=1, volume=1, enumeration=0, required=4, required_absent=0, mask=0x80, n_channels=1, override_map=yes D: alsa-mixer.c: Element PCM, direction=1, switch=1, volume=1, enumeration=0, required=0, required_absent=0, mask=0x402b600000000f66, n_channels=2, override_map=yes D: alsa-mixer.c: Element External Amplifier, direction=1, switch=4, volume=0, enumeration=0, required=0, required_absent=0, mask=0x0, n_channels=0, override_map=no D: alsa-mixer.c: Option on (output-amplifier-on/Amplifier) index=1, priority=10 D: alsa-mixer.c: Option off (output-amplifier-off/No Amplifier) index=0, priority=0 D: alsa-mixer.c: Setting output-amplifier-on (Amplifier) priority=10 D: alsa-mixer.c: Setting output-amplifier-off (No Amplifier) priority=0 I: alsa-sink.c: Hardware volume ranges from -81,00 dB to 12,00 dB. I: alsa-sink.c: Fixing base volume to -12,00 dB I: alsa-sink.c: Using hardware volume control. Hardware dB scale supported. I: alsa-sink.c: Using hardware mute control. D: alsa-util.c: snd_pcm_dump(): D: alsa-util.c: Hardware PCM card 1 'Aureal Vortex au8830' device 0 subdevice 0 D: alsa-util.c: Its setup is: D: alsa-util.c: stream : PLAYBACK D: alsa-util.c: access : MMAP_INTERLEAVED D: alsa-util.c: format : S16_LE D: alsa-util.c: subformat : STD _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
