<div dir="ltr">Hello,<div><br></div><div>I'm using an iPhone 5 to send and receive audio using RTP over UDP. The problem is that I can't hear anything in any direction. I get:</div><div><br></div>
0:00:27.758603000 [333m 1490[00m 0x16a9b460 [33;01mWARN [00m [00m audiobasesrc gstaudiobasesrc.c:858:GstFlowReturn gst_audio_base_src_create(GstBaseSrc *, guint64, guint, GstBuffer **):<autoaudiosrc0-actual-src-osxaudi>[00m create DISCONT of 1600 samples at sample 3680<br>0:00:27.758815000 [333m 1490[00m 0x16a9b460 [33;01mWARN [00m [00m audiobasesrc gstaudiobasesrc.c:863:GstFlowReturn gst_audio_base_src_create(GstBaseSrc *, guint64, guint, GstBuffer **):<autoaudiosrc0-actual-src-osxaudi>[00m warning: Can't record audio fast enough<br>0:00:27.758919000 [333m 1490[00m 0x16a9b460 [33;01mWARN [00m [00m audiobasesrc gstaudiobasesrc.c:863:GstFlowReturn gst_audio_base_src_create(GstBaseSrc *, guint64, guint, GstBuffer **):<autoaudiosrc0-actual-src-osxaudi>[00m warning: Dropped 1600 samples. This is most likely because downstream can't keep up and is consuming samples too slowly.<div><br></div><div>IMPORTANT NOTE: If I use only the receiving part of the pipeline it works fine (please see P.S.1). Same happens if I use only the sending part of the pipeline (please see P.S.2).</div><div><br></div><div>Code:</div><div><br></div><font face="monospace, monospace" size="1"><div><font face="monospace, monospace" size="1"> ...</font></div> gst_debug_set_threshold_from_string("2,*audio*:3", TRUE);<br> // Bidirectional<br> self->sm_pipeline = gst_parse_launch("udpsrc name=udp-src caps=\"application/x-rtp,media=audio,clock-rate=8000,encoding-name=PCMU\" ! rtppcmudepay ! mulawdec ! audioconvert ! audioresample ! autoaudiosink autoaudiosrc ! audioconvert ! audioresample ! mulawenc ! rtppcmupay ! udpsink name=udp-sink async=false", &error);<br><br> if (error) {</font><div><font face="monospace, monospace" size="1"> ...<br> }<br><br> gst_element_set_state (self->sm_pipeline, GST_STATE_PLAYING);</font></div><div><font face="monospace, monospace" size="1"> ...<br></font><br></div><div>Any ideas?</div><div><br></div><div>Thank you,</div><div>Antonis</div><div><br></div><div>P.S.1. Pipeline the works (only sending):</div><br> self->sm_pipeline = gst_parse_launch("autoaudiosrc ! audioconvert ! audioresample ! mulawenc ! rtppcmupay ! udpsink name=udp-sink async=false", &error);<div><br></div><div>P.S.2. Pipeline that works (only receiving):</div><div>
<p class=""><span class=""> self->sm_pipeline = gst_parse_launch("udpsrc name=udp-src caps=\"application/x-rtp,clock-rate=8000,encoding-name=PCMU\" ! rtppcmudepay ! mulawdec ! audioconvert ! audioresample ! autoaudiosink", &error);</span></p></div><div><br></div></div>