GitLab

CC: libav-stable@libav.org
Bug-Id: CID 1005311

It limits the duration of the data read from a given input.

Insert (a)trim filters on the corresponding inputs, so the extra frames
are decoded and discarded.

Using -ss 0 to drop frames with negative timestamps is a perfectly valid
use case.

This will allow using the same code when one filter can be inserted both
on input and output.

'filter' is too generic, since the functions in question deal with
several filters.

Also add const qualifiers to those variables.

This makes output -ss sample-accurate for audio and will allow further
simplication in the future.

This makes -t sample-accurate for audio and will allow further
simplication in the future.

Most of the FATE changes are due to audio now being sample accurate. In
some cases a video frame was incorrectly passed with the old code, while
its was over the limit.

The separators were changed from ',' to '|' in all the filters.

We do not support using filters without AVFilterGraph in practice
anyway, so there is no point in pretending we do.

This fixes crashes when there is no encoder for the default codec of
selected format.
Signed-off-by: Martin Storsjö <martin@martin.st>

Remove the code for volume scaling in avconv.c and instead auto-insert a
volume filter into the beginning of the filter chain.

Also reorder some other #include when applicable.

The buffer sink does not take any parameters.

Fixes an uninitialized variable warning.