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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[Sebastian Dröge-7]

Am Mittwoch, den 08.07.2009, 12:44 +0200 schrieb 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 ?

There's no "audiosplit" element that does what you want, it should be
quite easy to implement though (do you want to do it? :) ).

The latency query is something different though and the latency is not
influenced by the buffer sizes. Simple said it's the amount of time that
is buffered inside the element.

audioresample adds some latency to the pipeline but has no effect on
buffer sizes, instead it changes the sampling rate of the audio.

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[Julien Isorce]

2009/7/8 Sebastian Dröge <[hidden email]>

Am Mittwoch, den 08.07.2009, 12:44 +0200 schrieb 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 ?

There's no "audiosplit" element that does what you want, it should be
quite easy to implement though (do you want to do it? :) ).

 

In some cases where the source is encoded in a bad way,  it happens that the audio buffers are very big.
i do not know how much is the buffer size of an audio device renderer. But it would be better to split the buffers before to give them to the device.
Moreover, I do not understand why an element is needed. It should be configurable through gstreamer API.

For video buffer, the smaller thing is 32 bits (for example), all the 32 bits packets of an image have a meaning together (a whole frame). But for audio, the smaller thing is 16 bits (for example) but we can group them where we want, we just need to keep order.
I not sure to be clear.

The latency query is something different though and the latency is not
influenced by the buffer sizes. Simple said it's the amount of time that
is buffered inside the element.

 

you right, I missed to say "with a given and fixed rate", then latency gives the buffer sizes.

audioresample adds some latency to the pipeline but has no effect on
buffer sizes, instead it changes the sampling rate of the audio.

ok

 

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