'Can't record audio fast enough' issue - On iOS device

[Antonis Tsakiridis]

Thanks for the reply Ian.

Well I'm separating the two directions of the pipeline with a space
character (i.e. not a link). Doesn't this separate them and allow them to
run in parallel?

Any ideas how I could update my pipeline so that it uses a queue the way
you describe?

Best regards,
Antonis


On Sun, Jan 25, 2015 at 6:52 PM, Ian Davidson <id012c3076 at blueyonder.co.uk>
wrote:

>  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.
>
> Ian
>
>
> On 24/01/2015 20:55, Antonis Tsakiridis wrote:
>
> Hello,
>
>  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:
>
>  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
> 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
> 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.
>
>  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).
>
>  Code:
>
>     ...
>    gst_debug_set_threshold_from_string("2,*audio*:3", TRUE);
>    // Bidirectional
>    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);
>
>    if (error) {
>        ...
>    }
>
>    gst_element_set_state (self->sm_pipeline, GST_STATE_PLAYING);
>    ...
>
>  Any ideas?
>
>  Thank you,
> Antonis
>
>  P.S.1. Pipeline the works (only sending):
>
>     self->sm_pipeline = gst_parse_launch("autoaudiosrc ! audioconvert !
> audioresample ! mulawenc ! rtppcmupay ! udpsink name=udp-sink async=false",
> &error);
>
>  P.S.2. Pipeline that works (only receiving):
>
>     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);
>
>
>
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