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 > > Nah, it's not an audio underflow. In the default PulseAudio code, the > timer value is increased by 20ms each time this errror condition is > reached. PA ends-up waking-up 150ms or so before the deadline and yet > it still encounters this error. I don't think it's a problem with PA > not refilling the buffer on time, it's more that the information > provided by snd_pcm_avail() is misleading. 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. > - P > Since PA use one period per buffer for intel8x0 ( the hwptr is already at the outside of the buffer ) It is hard to debug without timestamp since it is not easy to know the value of sleeping time Can you explain what is "hwbuf_unused" ? 1. I: alsa-sink.c: Using 1.0 fragments of size 65536 bytes (371.52ms), buffer size is 65536 bytes (371.52ms) 2. I: alsa-sink.c: Time scheduling watermark is 20.00ms 3. D: alsa-sink.c: hwbuf_unused=0 4. D: alsa-sink.c: Buffer time: 371 ms; Sleep time: 351 ms; Process time: 20 ms 5. D: alsa-sink.c: setting avail_min=15502 6. D: alsa-mixer.c: Activating path analog-output 7. D: 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=31, min_dB=-81, max_dB=12 8. 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 9. D: alsa-mixer.c: Element Master Mono, direction=1, switch=2, volume=2, enumeration=0, required=0, required_absent=0, mask=0x7ffffffffffff, n_channels=1, override_map=no 10. D: alsa-mixer.c: Element Headphone, direction=1, switch=1, volume=3, enumeration=0, required=0, required_absent=0, mask=0x6, n_channels=2, override_map=no 11. 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 12. 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 13. D: alsa-mixer.c: Option on (output-amplifier-on/Amplifier) index=1, priority=10 14. D: alsa-mixer.c: Option off (output-amplifier-off/No Amplifier) index=0, priority=0 15. D: alsa-mixer.c: Setting output-amplifier-on (Amplifier) priority=10 16. D: alsa-mixer.c: Setting output-amplifier-off (No Amplifier) priority=0 17. I: alsa-sink.c: Hardware volume ranges from -81.00 dB to 12.00 dB. 18. I: alsa-sink.c: Fixing base volume to -12.00 dB 19. I: alsa-sink.c: Using hardware volume control. Hardware dB scale supported. 20. I: alsa-sink.c: Using hardware mute control. 21. D: alsa-util.c: snd_pcm_dump(): 22. D: alsa-util.c: Hardware PCM card 0 'Intel ICH6' device 0 subdevice 0 23. D: alsa-util.c: Its setup is: 24. D: alsa-util.c: stream : PLAYBACK 25. D: alsa-util.c: access : MMAP_INTERLEAVED 26. D: alsa-util.c: format : S16_LE 27. D: alsa-util.c: subformat : STD 28. D: alsa-util.c: channels : 2 29. D: alsa-util.c: rate : 44100 30. D: alsa-util.c: exact rate : 44100 (44100/1) 31. D: alsa-util.c: msbits : 16 32. D: alsa-util.c: buffer_size : 16384 33. D: alsa-util.c: period_size : 16384 34. D: alsa-util.c: period_time : 371519 35. D: alsa-util.c: tstamp_mode : ENABLE 36. D: alsa-util.c: period_step : 1 37. D: alsa-util.c: avail_min : 16384 38. D: alsa-util.c: period_event : 0 39. D: alsa-util.c: start_threshold : -1 40. D: alsa-util.c: stop_threshold : 1073741824 41. D: alsa-util.c: silence_threshold: 0 42. D: alsa-util.c: silence_size : 0 43. D: alsa-util.c: boundary : 1073741824 44. D: alsa-util.c: appl_ptr : 0 45. D: alsa-util.c: hw_ptr : 0 46. D: alsa-sink.c: Requested volume: 0: 3% 1: 3% 47. D: alsa-sink.c: Got hardware volume: 0: 3% 1: 3% 48. D: alsa-sink.c: Calculated software volume: 0: 100% 1: 100% (accurate-enough=yes) 49. D: alsa-sink.c: Thread starting up 50. I: core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 4, which is lower than the requested 5. 51. D: alsa-sink.c: Loop 52. D: alsa-sink.c: Loop 53. D: alsa-sink.c: Buffer time: 371 ms; Sleep time: 351 ms; Process time: 20 ms 54. hwptr 0 buffer 16384 appl 0, avail=16384 55. D: alsa-sink.c: avail: 65536 (samples 16384) 56. D: alsa-sink.c: 0.00 ms left to play; inc threshold = 0.00 ms; dec threshold = 100.00 ms 57. D: alsa-sink.c: Filling up 58. hwptr 0 buffer 16384 appl 0, avail=16384 59. hwptr 0 buffer 16384 appl 0, avail=16384 60. D: alsa-sink.c: Wrote 65496 bytes (of possible 65536 bytes) 61. hwptr 0 buffer 16384 appl 16374, avail=10 62. hwptr 0 buffer 16384 appl 16374, avail=10 63. D: alsa-sink.c: Wrote 40 bytes (of possible 40 bytes) 64. hwptr 0 buffer 16384 appl 16384, avail=0 65. D: alsa-sink.c: avail: 0 (samples 0) 66. D: alsa-sink.c: 371.52 ms left to play; inc threshold = 0.00 ms; dec threshold = 100.00 ms 67. D: alsa-sink.c: Not filling up, because too early. 68. I: alsa-sink.c: Starting playback. 69. D: alsa-sink.c: Cutting sleep time for the initial iterations by half. In the default PulseAudio code, the > timer value is increased by 20ms each time this errror condition is > reached. D: alsa-sink.c: Cutting sleep time for the initial iterations by half. Are you sure the time value is increased by 20 ms in pulseaudio-0.9.21 ? My au8830 work quite well on pulseaudio-0.9.14 ( Fedora 10 ) with 2 fragments 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: 72% 1: 72% D: module-alsa-sink.c: Thread starting up D: rtpoll.c: Acquired POSIX realtime signal SIGRTMIN+29 I: module-alsa-sink.c: Starting playback. 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 ( 16 fragments ) 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 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 : 16384 D: alsa-util.c: period_size : 1024 D: alsa-util.c: period_time : 23219 D: alsa-util.c: tstamp_mode : ENABLE D: alsa-util.c: period_step : 1 D: alsa-util.c: avail_min : 15502 D: alsa-util.c: period_event : 0 D: alsa-util.c: start_threshold : -1 D: alsa-util.c: stop_threshold : 1073741824 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: alsa-sink.c: Thread starting up I: core-util.c: Successfully enabled SCHED_RR scheduling for thread, with priority 4, which is lower than the requested 5. D: alsa-sink.c: Requested volume: 0: 23% 1: 23% D: alsa-sink.c: Got hardware volume: 0: 24% 1: 24% D: alsa-sink.c: Calculated software volume: 0: 97% 1: 97% (accurate-enough=no) I: alsa-sink.c: Starting playback. D: alsa-sink.c: Cutting sleep time for the initial iterations by half. D: alsa-sink.c: Cutting sleep time for the initial iterations by half. D: alsa-sink.c: Cutting sleep time for the initial iterations by half. > PA ends-up waking-up 150ms or so before the deadline and yet > it still encounters this error. I don't think it's a problem with PA > not refilling the buffer on time, it's more that the information > provided by snd_pcm_avail() is misleading. 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. > - P > Adding the timestamp is the easy way to prove the pulseaudio is writing the audio in time The problem may be the timer is not running at the same rate as the sound card 's clock or snd_pcm_rewindable() and snd_pcm_rewind() has bug since PA server is the only application use snd_pcm_rewind() http://thread.gmane.org/gmane.linux.alsa.devel/66826/focus=66952 Unfortunately there is no test program to test the snd_pcm_rewind() function http://thread.gmane.org/gmane.linux.alsa.devel/59470/focus=59641 PCI audio controllers fetch audio from memory in brust mode (i.e. fetch data must faster than sampling rate 48000Hz) , it is doubt that the application can really rewind the application pointer to hardware pointer and perform an immedate rewrite _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
