Audio Pipe with using the Sink Clock (need help)

Maik Scholz Scholz.Maik at t-online.de
Thu Jan 24 15:47:24 UTC 2019


My GST_DEBUG log was removed from the mailing daemon.

*gst.log:*

    audioclock
gstaudioclock.c:70:gst_audio_clock_init:<GstAudioClock at 0x1912100> init
     GST_CLOCK gstsystemclock.c:270:gst_system_clock_set_property:
clock-type set to 2
          alsa gstalsasink.c:257:gst_alsasink_init:<GstAlsaSink at 0x19117d0>
initializing alsasink
      pipeline gstpipeline.c:237:gst_pipeline_init:<GstPipeline at 0x1915070>
set bus <bus1> on pipeline
     GST_CLOCK gstelement.c:489:gst_element_set_base_time:<audiotestsrc0>
set base_time=0:00:00.000000000, old 0:00:00.000000000
     GST_CLOCK gstelement.c:551:gst_element_set_start_time:<audiotestsrc0>
set start_time=0:00:00.000000000, old 0:00:00.000000000
     GST_CLOCK gstelement.c:429:gst_element_set_clock:<audiotestsrc0>
setting clock (nil)
     GST_CLOCK gstelement.c:489:gst_element_set_base_time:<alsasink0> set
base_time=0:00:00.000000000, old 0:00:00.000000000
     GST_CLOCK gstelement.c:551:gst_element_set_start_time:<alsasink0> set
start_time=0:00:00.000000000, old 0:00:00.000000000
     GST_CLOCK gstelement.c:429:gst_element_set_clock:<alsasink0> setting
clock (nil)
          alsa gstalsasink.c:292:gst_alsasink_getcaps:<alsasink0> device not
open, using template caps
          alsa gstalsasink.c:292:gst_alsasink_getcaps:<alsasink0> device not
open, using template caps
    audioclock gstaudioclock.c:144:gst_audio_clock_reset:<GstAudioSinkClock>
reset clock to 0:00:00.000000000, last 0:00:00.000000000, offset
+0:00:00.000000000
    ringbuffer
gstaudioringbuffer.c:453:gst_audio_ring_buffer_open_device:<audiosinkringbuffer0>
opening device
          alsa gstalsasink.c:849:gst_alsasink_open:<alsasink0> Opened device
hw:0,0
    ringbuffer
gstaudioringbuffer.c:471:gst_audio_ring_buffer_open_device:<audiosinkringbuffer0>
opened device
      pipeline gstpipeline.c:306:reset_start_time:<pipeline0> reset
start_time to 0
    ringbuffer
gstaudioringbuffer.c:1991:gst_audio_ring_buffer_may_start:<audiosinkringbuffer0>
may start: 0
          alsa gstalsa.c:191:gst_alsa_detect_formats:<alsasink0> skipping
non-raw format
          alsa gstalsa.c:191:gst_alsa_detect_formats:<alsasink0> skipping
non-raw format
          alsa gstalsa.c:191:gst_alsa_detect_formats:<alsasink0> skipping
non-raw format
          alsa gstalsa.c:191:gst_alsa_detect_formats:<alsasink0> skipping
non-raw format
          alsa gstalsa.c:30:gst_alsa_detect_rates:<alsasink0> probing sample
rates ...
          alsa gstalsa.c:49:gst_alsa_detect_rates:<alsasink0> Min. rate =
44100 (44100)
          alsa gstalsa.c:50:gst_alsa_detect_rates:<alsasink0> Max. rate =
48000 (48000)
          alsa gstalsa.c:348:gst_alsa_detect_channels:<alsasink0> probing
channels ...
          alsa gstalsa.c:392:gst_alsa_detect_channels:<alsasink0> Min.
channels = 2 (2)
          alsa gstalsa.c:393:gst_alsa_detect_channels:<alsasink0> Max.
channels = 2 (2)
      pipeline gstpipeline.c:412:gst_pipeline_change_state:<pipeline0>
selecting clock and base_time
      pipeline gstpipeline.c:433:gst_pipeline_change_state:<pipeline0> Need
to update start_time
          alsa gstalsa.c:472:gst_alsa_open_iec958_pcm:<alsasink0> Generated
device string "hw:0,0:{AES0 0x02 AES1 0x82 AES2 0x00 AES3 0x02}"
      pipeline gstpipeline.c:438:gst_pipeline_change_state:<pipeline0> Need
to update clock.
          alsa conf.c:4913:parse_args: alsalib error: Parameter DEV must be
an integer
          alsa conf.c:5018:snd_config_expand: alsalib error: Parse arguments
error: 0,0:{AES0 0x02 AES1 0x82 AES2 0x00 AES3 0x02} Invalid argument
          alsa pcm.c:2565:snd_pcm_open_noupdate: alsalib error: Unknown PCM
hw:0,0:{AES0 0x02 AES1 0x82 AES2 0x00 AES3 0x02}
          alsa gstalsa.c:478:gst_alsa_open_iec958_pcm:<alsasink0> failed
opening IEC958 device: Invalid argument
 audiobasesink
gstaudiobasesink.c:362:gst_audio_base_sink_provide_clock:<alsasink0>
ringbuffer not acquired

*For my understanding, at this point in time, the audio sink deny's
providing a clock because the ringbuffer is not ready.*

          alsa gstalsasink.c:332:gst_alsasink_getcaps:<alsasink0> returning
caps audio/x-raw, format=(string)S16LE, layout=(string)interleaved,
rate=(int)[ 44100, 48000 ], channels=(int)2,
channel-mask=(bitmask)0x0000000000000003
     GST_CLOCK gstsystemclock.c:353:gst_system_clock_obtain: creating new
static system clock
     GST_CLOCK gstpipeline.c:732:gst_pipeline_provide_clock_func: pipeline
obtained system clock: 0x19270f0 (GstSystemClock)
     GST_CLOCK gstclock.c:1046:gst_clock_get_internal_time:<GstSystemClock>
internal time 0:51:32.592473507
     GST_CLOCK gstclock.c:1091:gst_clock_get_time:<GstSystemClock> adjusted
time 0:51:32.592473507
     GST_CLOCK gstelement.c:429:gst_element_set_clock:<pipeline0> setting
clock 0x19270f0
     GST_CLOCK gstelement.c:429:gst_element_set_clock:<alsasink0> setting
clock 0x19270f0
          alsa gstalsasink.c:303:gst_alsasink_getcaps:<alsasink0> Returning
cached caps audio/x-raw, format=(string)S16LE, layout=(string)interleaved,
rate=(int)[ 44100, 48000 ], channels=(int)2,
channel-mask=(bitmask)0x0000000000000003 with filter audio/x-raw,
format=(string)S16LE, layout=(string)interleaved, rate=(int)44100,
channels=(int)2, channel-mask=(bitmask)0x0000000000000003 applied:
audio/x-raw, format=(string)S16LE, layout=(string)interleaved,
rate=(int)44100, channels=(int)2, channel-mask=(bitmask)0x0000000000000003
     GST_CLOCK gstelement.c:429:gst_element_set_clock:<audiotestsrc0>
setting clock 0x19270f0

*Now the system clock is used!*



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