It is already set to 0 by av_free_packet()

We no longer have any audio encoders using AVCodec.encode().

Rationale is the same as for video width/height etc.

This feature is complex, of questionable utility, and slows down
normal decoding.

Signed-off-by: Mans Rullgard <mans@mansr.com>

This improves output for formats with more than one AVCodec.

Use default alignment in audio_get_buffer()

Set avctx->channel_layout to 0 if the channel count does not match
avctx->channels.

Otherwise realloc would free it, which would result in double free
later.

Signed-off-by: Anton Khirnov <anton@khirnov.net>

This is a utility function for the user to get the frame duration based on
the codec id, frame size in bytes, and various AVCodecContext parameters.

This only returns bits per sample when it is exactly correct. That is, the
codec contains only raw samples with no frame headers or padding. This applies
to basically all PCM codecs and a small subset of ADPCM codecs.

Also, don't bother with saving/restoring data, av_init_packet doesn't
touch it.

Based on a patch by Nicolas George <nicolas.george <at> normalesup.org>

All video encoders have been converted to encode2(), all new encoders
should also use only encode2().

This allows encoders to set pts and/or duration even if they do not use
CODEC_CAP_DELAY.

Signed-off-by: Martin Storsjö <martin@martin.st>

Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>

There are no audio encoders which do frame reordering.

Deprecate avcodec_encode_video().

This fixes crashes in e.g. PNG decoding with SSE2 enabled. In fact, many
x86 optimizations for codecs assume that our buffer strides are 16-byte
aligned.

This will simplify encoders which use this function to request the exact
packet size rather than the maximum size.

Wrapper around av_fast_malloc() that keeps FF_INPUT_BUFFER_PADDING_SIZE
zero-padded bytes at the end of the used buffer.

Based on a patch by Reimar Döffinger <Reimar.Doeffinger@gmx.de>.

This way, if the AVCodecContext is allocated for a specific codec, the
caller doesn't need to store this codec separately and then pass it
again to avcodec_open2().

It also allows to set codec private options using av_opt_set_* before
opening the codec.

I.e. free the priv_data and other stuff allocated in
avcodec_alloc_context3() and not segfault.

It allows to check whether an AVCodecContext is open in a documented
way. Right now the undocumented way this check is done in lavf/lavc is
by checking whether AVCodecContext.codec is NULL. However it's desirable
to be able to set AVCodecContext.codec before avcodec_open2().

This function is video-only, so there's no point in setting more
linesizes.

Fixes stack corruption in avplay.

Earlier, calling avcodec_encode_audio worked fine even if time_base
wasn't set. Now it crashes due to trying to scale the output pts to
the codec context time base. This affects e.g. VLC.

If no time_base is set for audio codecs, set it to the sample
rate.

CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>

The stereo layout and extradata is significantly different from that in
Westwood IMA ADPCM, so a separate codec_id is warranted.

This allows audio encoders to optionally take an AVFrame as input and write
encoded output to an AVPacket.

This also adds AVCodec.encode2() which will also be usable by video and
subtitle encoders once support is implemented in the public functions.

This is a convenience function for the user to fill audio AVFrame information.

Do not fail audio decoding with avcodec_decode_audio3 if user has set a
custom get_buffer. Strictly speaking, this was never allowed by the API,
but it seems that some software packages did so anyways. In order to
unbreak applications (cf. http://bugs.debian.org/655890), this change
clarifies the API and overrides the custom get_buffer() with the defaults.

This change is inspired by a similar
commit (c3846e3eba) in FFmpeg.

Signed-off-by: Reinhard Tartler <siretart@tauware.de>