[gst-devel] change audio buffer duration

[Julien Isorce]

Hi,

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,
codec_data=(buffer)1000000000000100000000001000800000aa00389b71
and
(type: 118, taglist, audio-codec=(string)\"Uncompressed\\ 16-bit\\ PCM\\
audio\";)

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 :

Implementation:
      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 ?

Sincerely

Julien
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