ChinnaSuhas / ossbuild

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No playing sound in FLV file #44

Closed GoogleCodeExporter closed 9 years ago

GoogleCodeExporter commented 9 years ago
In previous version was no problem with this file.

Original issue reported on code.google.com by pak...@gmail.com on 13 May 2010 at 6:33

Attachments:

GoogleCodeExporter commented 9 years ago
Works for me with the following pipeline:
gst-launch.exe playbin uri=file:///c:/video_test.flv

There is a bug in the GStreamer registry on Windows which prevents the registry 
to be 
updated properly when new plugins are installed, retry deleting it. It's 
located in 
'$HOME/.gstreamer-0.10'
You can also check with gst-inspect if all the plugins are loaded properly

Original comment by ylatuya on 13 May 2010 at 7:18

GoogleCodeExporter commented 9 years ago
I delete this file, but still have no sound with this file :(
I use Windows 7 64bit, gst-inspect show what is in attachment
gst-launch show:
Setting pipeline to PAUSED ...
Pipeline is PREROLLING ...
Pipeline is PREROLLED ...
Setting pipeline to PLAYING ...
New clock: GstAudioSinkClock

Original comment by pak...@gmail.com on 14 May 2010 at 6:47

Attachments:

GoogleCodeExporter commented 9 years ago
I didn't test previously with sound. 
In fact there is no sound when using directsoundsink, but it works using 
waveformsink.
The bug is therefore in directsoundsink which doesn't seems to like the rate:

Using this pipelin:
gst-launch.exe -v audiotestsrc ! audio/x-raw-int, rate=22050, channels=2,
signed=true, endianness=1234, width=16, depth=16 ! directsoundsink

The followings rates doesn't work:
22050
22060
44070

The following ones work:
22000
22100
44000

It seems that it doesn't like non multiples of 100, althoug 22010 works for 
instance
gst-launch.exe -v audiotestsrc ! audio/x-raw-int, rate=22050, channels=2,
signed=true, endianness=1234, width=16, depth=16 ! directsoundsink
gst-launch.exe -v audiotestsrc ! audio/x-raw-int, rate=44050, channels=2,
signed=true, endianness=1234, width=16, depth=16 ! directsoundsink

There must be some kind of rounding going on that is making buffers to get to 
late in
the sink

Original comment by ylatuya on 14 May 2010 at 8:22

GoogleCodeExporter commented 9 years ago
Would audioresample help here?

Original comment by david.g.hoyt on 14 May 2010 at 10:37

GoogleCodeExporter commented 9 years ago
It won't help either unless you force the caps before the sink to something 
that know
would work. Also decodebin won't add it since it can negociate the caps with 
the sink
without it.
waveforsink has its drawbacks, like completely failing with some audio cards, 
and
that's why I lowered down its rank, but it seems to be accepting any input rate 
:/
I'll take a look at directsoundsink and I'll fix it this weekend.

Original comment by ylatuya on 15 May 2010 at 1:48

GoogleCodeExporter commented 9 years ago
>I'll fix it this weekend
I meant "I'll try to fix it" :P

Original comment by ylatuya on 15 May 2010 at 1:49

GoogleCodeExporter commented 9 years ago
That's now fixed upstream:
http://cgit.freedesktop.org/gstreamer/gst-plugins-good/commit/?id=055586f2d9b75e
9a6912ed36c8718cd94d20c2e8

Original comment by ylatuya on 9 Jun 2010 at 7:37