<html>
<head>
<meta content="text/html; charset=utf-8" http-equiv="Content-Type">
</head>
<body bgcolor="#FFFFFF" text="#000000">
Well, it could be that your iPhone does not have the guts to do
everything that you want. However, I don't see any 'queue' elements
in your pipeline. A queue causes gstreamer to create another thread
and it is possible that a few extra threads would allow both streams
to proceed.<br>
<br>
Ian<br>
<br>
<div class="moz-cite-prefix">On 24/01/2015 20:55, Antonis Tsakiridis
wrote:<br>
</div>
<blockquote
cite="mid:CAO97BBSMdZ9sJ_Q+52Gi8QrY8Un4y4kNMJ6E-JHRAs-L00PrNA@mail.gmail.com"
type="cite">
<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>
<br>
<fieldset class="mimeAttachmentHeader"></fieldset>
<br>
<pre wrap="">_______________________________________________
gstreamer-devel mailing list
<a class="moz-txt-link-abbreviated" href="mailto:gstreamer-devel@lists.freedesktop.org">gstreamer-devel@lists.freedesktop.org</a>
<a class="moz-txt-link-freetext" href="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel">http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel</a>
</pre>
</blockquote>
<br>
</body>
</html>