[Bug 712182] New: iOS: multiple audio streams get "AUIOClient_StartIO failed (560034163)" error

GStreamer (bugzilla.gnome.org) bugzilla at gnome.org
Tue Nov 12 13:07:50 PST 2013


https://bugzilla.gnome.org/show_bug.cgi?id=712182
  GStreamer | gst-plugins-good | git

           Summary: iOS: multiple audio streams get "AUIOClient_StartIO
                    failed (560034163)" error
    Classification: Platform
           Product: GStreamer
           Version: git
        OS/Version: Mac OS
            Status: UNCONFIRMED
          Severity: normal
          Priority: Normal
         Component: gst-plugins-good
        AssignedTo: gstreamer-bugs at lists.freedesktop.org
        ReportedBy: elio.francesconi at gmail.com
         QAContact: gstreamer-bugs at lists.freedesktop.org
     GNOME version: ---


I'm using iphone5 version 7.0.3 and I noticed that opening two rtp stream,
needed for a voip communication using the pipelines below, I got this error:

2013-11-09 08:17:33.668 ipDoor[314:8403] 08:17:33.668 ERROR:     [0x7e44000]
1207: AUIOClient_StartIO failed (560034163)
0:00:39.088255000 [333m  314[00m  0x2a737e8 [31;01mERROR  [00m [00;04m         
   default
gstosxcoreaudiocommon.c:128:gst_core_audio_io_proc_start:<osxaudioringbuffer0>[00m
AudioOutputUnitStart failed: sqa!
2013-11-09 08:17:33.670 ipDoor[314:8903] 08:17:33.670 ERROR:     [0x7f70000]
>aurioc> 1455: AURemoteIO at 0x31ab420: IOThread exiting with error 0x10004006
2013-11-09 08:17:33.672 ipDoor[314:8403] 08:17:33.672 ERROR:     [0x7e44000]
1207: AUIOClient_StartIO failed (560034163)
0:00:39.092167000 [333m  314[00m  0x2a737e8 [31;01mERROR  [00m [00;04m         
   default
gstosxcoreaudiocommon.c:128:gst_core_audio_io_proc_start:<osxaudioringbuffer0>[00m
AudioOutputUnitStart failed: sqa!

as result the sinkPipeline doesn't produce any audio.

#define LAUNCH_SERVER_AUDIO "osxaudiosrc ! capsfilter
caps=audio/x-raw,rate=44100 ! audioconvert ! audioresample ! alawenc !
rtppcmapay ! udpsink host=127.0.0.1 port=5011"

#define LAUNCH_CLIENT_AUDIO "udpsrc
caps=application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA
port=5011 ! rtppcmadepay ! alawdec ! audioconvert ! audioresample !
autoaudiosink"


    sourcePipeline = gst_parse_launch(LAUNCH_SERVER_AUDIO, &error);
    sinkPipeline = gst_parse_launch(LAUNCH_CLIENT_AUDIO, &error);

    if(gst_element_set_state(sourcePipeline, GST_STATE_PLAYING) ==
GST_STATE_CHANGE_FAILURE) {
        GST_ERROR("GST_STATE_CHANGE_FAILURE ...");
    }
    if(gst_element_set_state(sinkPipeline, GST_STATE_PLAYING) ==
GST_STATE_CHANGE_FAILURE) {
        GST_ERROR("GST_STATE_CHANGE_FAILURE ...");
    }

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