[gst-devel] GStreamer audio streaming on Windows

[sumit kumar]

try setting sync=false for directsoundsink properties.

On Tue, Dec 7, 2010 at 2:23 PM, Callewaert Sven <S.Callewaert at televic.com>wrote:

> Replacing the autoaudiosink element with directsoundsink did the trick. Now
> I hear a continuous beep but there is a lot of distortion on it. When I
> stream the microphone by replacing audiotestsrc with directsoundsrc on the
> sender side, I also hear a lot of distortion. The distortion however only
> occurs when the microphone captures moderate to loud noises. There is no
> distortion with quiet noises. Could it be a signed/unsigned issue? Can I
> force the sender the use signed=false in some way?
>
>
>
> *From:* sumit kumar [mailto:sumit.git at gmail.com]
> *Sent:* dinsdag 7 december 2010 8:18
> *To:* Discussion of the development of GStreamer
> *Subject:* Re: [gst-devel] GStreamer audio streaming on Windows
>
>
>
> Hi,
>
>
>
> Try with directsoundsink.. else try setting sync=false for autoaudiosink
>
>
>
> -Sumit
>
> On Tue, Dec 7, 2010 at 3:55 AM, Callewaert Sven <S.Callewaert at televic.com>
> wrote:
>
> Hi,
>
>
>
> I'm experimenting a bit with GStreamer (ossbuild 0.10.7) on Windows, but I
> can't seem to make audio streaming between two computers work. All I hear at
> the receiver side is a short beep followed by silence.
>
>
>
> This is the sender pipeline:
>
> gst-launch -v audiotestsrc  ! audioconvert ! rtpL16pay ! udpsink
> host=224.0.0.7 auto-multicast=true port=4444
>
>
>
> This is the receiver pipeline:
>
> gst-launch -v udpsrc multicast-group=224.0.0.7 port=4444
> caps="application/x-rtp, media=(string)audio, channels=(int)1,
> clock-rate=(int)44100, encoding-name=(string)L16" ! gstrtpbin ! rtpL16depay
> ! audioconvert ! queue ! autoaudiosink
>
>
>
> I've already tried different queue settings and codecs. Same thing when I
> try to stream an audio file, all I hear is about 1 second of it. What
> could be the problem?
>
>
>
>
>
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