[gst-devel] change audio buffer duration
julien.isorce at gmail.com
Wed Jul 8 12:44:59 CEST 2009
I have an avi file which contains audio/x-raw-int (and video, but my
question is just about the audio).
There is the caps:
caps = audio/x-raw-int, endianness=(int)1234, channels=(int)2,
width=(int)16, depth=(int)16, rate=(int)48000, signed=(boolean)true,
(type: 118, taglist, audio-codec=(string)\"Uncompressed\\ 16-bit\\ PCM\\
Using identity and -v, I can see that buffer duration is around 10 sec and
the total is 20 sec.
So there is only 2 audio buffers.
Is there a gstreamer element that can change or split this buffer duration ?
(Usually audio buffer duration is about 20 or 50 ms)
There is also the "gst_query_set_latency" but what would be the inpact on
the video (video buffer duration) ?
Usually I configure the audio latency (=audio buffer duration) when using
alsasrc, but how to do that with an avi file?
Finally I can see :
Has getrangefunc(): gst_base_transform_getrange
Has custom eventfunc(): gst_base_transform_src_event
Has custom queryfunc(): 0xb7916800
Provides query types:
(3): latency (Latency)
in gst-inspect-0.10 audioresample
So audioresample is able to only change the latency ? any example ?
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