[gst-devel] Choppy Audio over UDP

[Marco Ballesio]

Hi,

sorry for double posting, I've forgot to add a few comments in my previous
email.

On Mon, Oct 18, 2010 at 9:03 PM, Wes Miller <wmiller at sdr.com> wrote:

>
> Marco et al,
>
>
> Ok, new pipelines with rtpbin.  These work though I am not sure they are
> completely correct.
>
> SENDER <><><><><><><><><><><><><><><><>
>
> gst-launch-0.10 -v gstrtpbin name=rtpbin \
>         alsasrc do-timestamp=true \
>            ! queue \
>            ! audioresample \
>            ! audioconvert \
>

Are the audioresample/audioconvert necessary? What if you use pulsesrc /
pulsesink and remove them?


>             ! dmaienc_aac outputBufferSize=10000000 outputformat=2 \
>             ! rtpmp4apay         \
>            ! rtpbin.send_rtp_sink_1 \
>         rtpbin.send_rtp_src_1 \
>            ! udpsink port=5002 host=$1  ts-offset=0 name=artpsink \
>         rtpbin.send_rtcp_src_1 \
>            ! udpsink port=5003 host=$1 sync=false async=false
> name=artcpsink \
>         udpsrc port=5007 name=artpsrc \
>            ! rtpbin.recv_rtcp_sink_1
>
>
> RECEIVER <><><><><><><><><><><><><><><>
>
>  gst-launch-0.10  -v gstrtpbin name=rtpbin latency=2000 \
>         udpsrc
>
> caps="application/x-rtp,media=audio,clock-rate=44100,encoding-name=MP4A-LATM,


here clock-rate is the RTP clock, you should set it to a value according
with your sample rate, check the caps negotiated from the source element on
the sender side to have an hint. You can do this using gst-launch -v


> payload=96"
> port=5002 \
>            ! rtpbin.recv_rtp_sink_1 \
>         rtpbin. \
>            ! rtpmp4adepay  \
>            ! decodebin     \
>

you don't need decodebin here. Just put the decoder element.


>            ! audioconvert  \
>            ! audioresample \
>            ! alsasink sync=true \
>

Again, try replacing audioconvert/audioresample/alsasink with a plain
pulsesink (if you can ;) ).

Regards,
Marco


>         udpsrc port=5002   \
>            ! rtpbin.recv_rtcp_sink_1 \
>         rtpbin.send_rtcp_src_1 \
>            ! udpsink port=5003 host=$1 sync=false async=false
>
>
> I am still getting playback that is too fast and choppy.  Sounds like the
> receiver pipe is just playing each packet too fast.  I don't think I'm
> loosing any of the music, it just plays each packet too fast and has a dead
> spot in between them.  Imagine the Moody Blues sung by the Chipmunks with
> hiccups.
>
> So, first, do I haave the ports mapped correctly?  Am I missing any
> connections between the rtpbin in's and out's?
>
> Secondly, is there a retimesynch trick I can use?  Replacing alsasink with
> fakesink -v tells me that the packets are timestamped and that the
> durations
> match the time between timestamps.
>
> I really intend to replace the mp3 player I am using with a microphone.
>  The
> Leopardboard only has linelevel input so I can't use a mic right now.  If
> passing speech, would I be more likely to hear the speed=up and gaps?  Will
> speex through an rtpxxxpayloader still cause me to need 4k of Caps?
>
> Thanks,
>
> Wes
>
>
>
>
>
>
>
> --
> View this message in context:
> http://gstreamer-devel.966125.n4.nabble.com/Choppy-Audio-over-UDP-tp2997741p3000742.html
> Sent from the GStreamer-devel mailing list archive at Nabble.com.
>
>
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