[Libav-user] Audio sample rate conversion and fixed frame size

[Anton Shekhovtsov]

2015-01-23 14:34 GMT+02:00 Max Vlasov <max.vlasov at gmail.com>:

> On Fri, Jan 23, 2015 at 2:20 PM, Anton Shekhovtsov <shekh.anton at gmail.com>
> wrote:
>
>> I used swresample only to convert format but it looks simple as brick to
>> me.
>>
>>
> Is there somewhere a hidden question "What is the problem in the first
> place?" :)
>
> Probably I missed the point somewhere, but some codecs report particular
> frame_size so one should feed data only with blocks having this particular
> size. A quote from the sources about
> AVCodecContext.frame_size
> ...
>      * - encoding: ... Each submitted frame
>      *   except the last must contain exactly frame_size samples per
> channel.
>      *   May be 0 when the codec has CODEC_CAP_VARIABLE_FRAME_SIZE set,
> then the
>      *   frame size is not restricted.
>      *)
>
>
> If incoming data has the same sample_rate as outgoing, no problem,
> swr_convert will output the same amount of frames as it accepted. But if
> the sample rate are different (44.1k vs 48 k), you can't avoid tricky
> arithmetic/logic or caching extra data somewhere unless you have plans to
> violate the rule.
>
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>
Yes, in the code fragment above "buf_data" is assumed to hold data between
invocations. Exactly that - caching somewhere
But it is plain memory buffer, not tricky :)
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