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GStreamer audio streaming on Windows

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Callewaert Sven

GStreamer audio streaming on Windows

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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sumit kumar-11

Re: GStreamer audio streaming on Windows

Hi,

try using directsoundsink...or if you are using autoaudiosink than give sync=false and see

On Tue, Nov 30, 2010 at 8:14 PM, Callewaert Sven <[hidden email]> 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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Callewaert Sven

GStreamer audio streaming on Windows

In reply to this post by Callewaert Sven

Hi,

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:

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?

sumit kumar-11

Re: 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 <[hidden email]> 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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Callewaert Sven

Re: GStreamer audio streaming on Windows

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:[hidden email]]
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 <[hidden email]> wrote:

Hi,

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:

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?

sumit kumar-11

Re: GStreamer audio streaming on Windows

try setting sync=false for directsoundsink properties.

On Tue, Dec 7, 2010 at 2:23 PM, Callewaert Sven <[hidden email]> 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:[hidden email]]
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 <[hidden email]> 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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Callewaert Sven

Re: GStreamer audio streaming on Windows

In reply to this post by sumit kumar-11

The problem of the distortion on the audio stream appears to be linked to the rtpL16pay payloader. When I use the rtpmp4apay payloader the problem disappears.

From: Callewaert Sven
Sent: dinsdag 7 december 2010 9:53
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 <[hidden email]> wrote:

Hi,

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:

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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Callewaert Sven

Distortion when using directsoundsrc

In reply to this post by sumit kumar-11

On my Vista computer I hear some distortions on the audio that is captured from my microphone. The distortion is a periodical noisy crack. Playing around with the buffer-time parameter changes the number of cracks, but it doesn’t seem possible to make them go away. I use the ossbuild of gstreamer version 0.10.7. On my Windows 7 computer there is no problem. This is the pipeline:

gst-launch directsoundsrc ! directsoundsink

Is there any way to get rid of the distortion?

Marco Ballesio

Re: Distortion when using directsoundsrc

Hi,

On Thu, Dec 9, 2010 at 10:41 AM, Callewaert Sven
<[hidden email]> wrote:

> On my Vista computer I hear some distortions on the audio that is captured
> from my microphone. The distortion is a periodical noisy crack. Playing
> around with the buffer-time parameter changes the number of cracks, but it
> doesn’t seem possible to make them go away. I use the ossbuild of gstreamer
> version 0.10.7. On my Windows 7 computer there is no problem. This is the
> pipeline:
>
>
>
> gst-launch directsoundsrc ! directsoundsink
>
>
>
> Is there any way to get rid of the distortion?
>

I've never used GStreamer under Microsoft stuff, btw maybe I could get
more help by attaching here the log you get with, let's say,
GST_DEBUG=3 (no copy-paste in the email please).

Does it get better if you add a queue element bw the source and the sink?

Regards

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