[Ffmpeg-devel] Buffering audio data in case of overflow
Sat May 6 21:46:06 CEST 2006
On 6-May-06, at 3:19 PM, Alex Beregszaszi wrote:
>>> No, it was an intermediate solution. 192k is one second 48khz,
>>> 16bit integer audio data. That should be enough for most codecs we
>>> support. The real solution would be a different method of passing
>>> chunks, the push method. A codec knows how much data it has,
>>> requests a
>>> buffer large enough for it and pushes to the player.
>> Can you elaborate on this point? Are there any examples in the
>> codebase of this "method"?
> like avcodec_get_buffer or the mechanism in mplayer
> Note: the interface is not designed.
But how does this solve the problem of the given audio output buffer
being too small? Can I override that size and send a larger audio
output buffer to ffmpeg out of my decoder (presumably using the
function you mentioned earlier?)
> Alex Beregszaszi email: alex at fsn.hu
> Free Software Network cell: +36 70 3144424
> ffmpeg-devel mailing list
> ffmpeg-devel at mplayerhq.hu
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