Gstreamer: error de análisis de mayúsculas opus, ¿alguien sabe cómo solucionarlo?

$ gst-launch-0.10 -v gstrtpbin name=rtpbin latency=100 udpsrc caps="application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)X-GST-OPUS-DRAFT-SPITTKA-00, caps=(string)\"audio/x-opus\,\ multistream\=\(boolean\)false\,\ streamheader\=\(buffer\)\<\ 4f707573486561640101000080bb0000000000\,\ 4f707573546167731e000000456e636f6465642077697468204753747265616d6572204f707573656e63010000001a0000004445534352495054494f4e3d617564696f74657374207761766501\ \>\", ssrc=(uint)3090172512, payload=(int)96, clock-base=(uint)4268257583, seqnum-base=(uint)10001" port=5002 ! rtpbin.recv_rtp_sink_1 rtpbin. ! rtpopusdepay ! opusdec ! audioconvert ! audioresample ! alsasink device=2 name=uudpsink0 udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=%s sync=false async=false

(gst-plugin-scanner:25672): GStreamer-WARNING **: Failed to load plugin '/usr/lib/gstreamer-0.10/libgstsimsyn.so': /usr/lib/gstreamer-0.10/libgstsimsyn.so: undefined symbol: gst_controller_sync_values

(gst-plugin-scanner:25672): GStreamer-WARNING **: Failed to load plugin '/usr/lib/gstreamer-0.10/libgstaudiodelay.so': /usr/lib/gstreamer-0.10/libgstaudiodelay.so: undefined symbol: gst_base_transform_set_gap_aware

(gst-plugin-scanner:25672): GStreamer-WARNING **: Failed to load plugin '/usr/lib/gstreamer-0.10/libgstbml.so': /usr/lib/gstreamer-0.10/libgstbml.so: undefined symbol: gst_base_src_set_format
WARNING: erroneous pipeline: could not set property "caps" in element "udpsrc0" to "application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)X-GST-OPUS-DRAFT-SPITTKA-00, caps=(string)"audio/x-opus\,\ multistream\=\(boolean\)false\,\ streamheader\=\(buffer\)\<\ 4f707573486561640101000080bb0000000000\,\ 4f707573546167731e000000456e636f6465642077697468204753747265616d6572204f707573656e63010000001a0000004445534352495054494f4e3d617564696f74657374207761766501\ \>", ssrc=(uint)3090172512, payload=(int)96, clock-base=(uint)4268257583, seqnum-base=(uint)10001"

- user285594

Responder

• No creo que se necesite un escape especial. Si su canalización es correcta, esto debería funcionar:

  gst-launch-0.10 -v gstrtpbin name=rtpbin latency=100 udpsrc caps="application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)X-GST-OPUS-DRAFT-SPITTKA-00, caps=(string)audio/x-opus, multistream=(boolean)false, streamheader=(buffer)<4f707573486561640101000080bb0000000000,4f707573546167731e000000456e636f6465642077697468204753747265616d6572204f707573656e63010000001a0000004445534352495054494f4e3d617564696f74657374207761766501>, ssrc=(uint)3090172512, payload=(int)96, clock-base=(uint)4268257583, seqnum-base=(uint)10001" port=5002 ! rtpbin.recv_rtp_sink_1 rtpbin. ! rtpopusdepay ! opusdec ! audioconvert ! audioresample ! alsasink device=2 name=uudpsink0 udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 rtpbin.send_rtcp_src_1 ! udpsink port=5007 host=%s sync=false async=false
  

  Si necesita cuidar los caracteres especiales que bash podría estar interpretando, cambie caps="..." a caps='...'.

  Aquí hay una versión torpe de Python:

  import subprocess                                                                                                                                                                                  
  
  args=[  'gst-launch-0.10',
          '-v',
          'gstrtpbin',
          'name=rtpbin',
          'latency=100',
          'udpsrc',
          'caps="application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)X-GST-OPUS-DRAFT-SPITTKA-00, caps=(string)audio/x-opus, multistream=(boolean)false, streamheader=(buffer)<4f707573486561640101000080bb0000000000,4f707573546167731e000000456e636f6465642077697468204753747265616d6572204f707573656e63010000001a0000004445534352495054494f4e3d617564696f74657374207761766501>, ssrc=(uint)3090172512, payload=(int)96, clock-base=(uint)4268257583, seqnum-base=(uint)10001"',
          'port=5002',
          '!',
          'rtpbin.recv_rtp_sink_1',
          'rtpbin.',
          '!',
          'rtpopusdepay',
          '!',
          'opusdec',
          '!',
          'audioconvert',
          '!',
          'audioresample',
          '!',
          'alsasink',
          'device=2',
          'name=uudpsink0',
          'udpsrc',
          'port=5003',
          '!',
          'rtpbin.recv_rtcp_sink_1',
          'rtpbin.send_rtcp_src_1',
          '!',
          'udpsink',
          'port=5007',
          'host=%s',
          'sync=false',
          'async=false',
  
          ]
  
  child = subprocess.Popen(args, stdout=subprocess.PIPE)
  streamdata = child.communicate()[0] # streamdata will contain output of gst-launch-0.10
  rc = child.returncode # rc will contain the returncode of gst-launch-0.10
  
  print streamdata
  print "\nprocess returned %d" %(rc)
  

  Creo que es mejor encontrar un buen módulo de python para gstreamer que usar un subproceso o similar.

  ¡Espero que esto ayude!