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.
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>
While we correctly "register" the side data when we split it,
the application (in this case FFmpeg) might not update the
AVPacket pool it uses to finally free the packet, thus
causing a leak.
This also makes the av_dup_packet unnecessary which could
cause an even worse leak in this situation.
Also change the code to not modify the user-provide AVPacket at all.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Some of these encoders may produce invalid bitstreams, which should not
be done without the user knowing.
Some of these decoders may be unfinished and may contain security issues.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The format is a per-frame property, having it in AVFrame simplify the
operation of extraction of that information, since avoids the need to
access the codec/stream context.
width and height are per-frame properties, setting these values in
AVFrame simplify the operation of extraction of that information,
since avoids the need to check the codec/stream context.
The sample aspect ratio is a per-frame property, so it makes sense to
define it in AVFrame rather than in the codec/stream context.
Simplify application-level sample aspect ratio information extraction,
and allow further simplifications.
This way ffmpeg can be distinguished from the fork by a user
application or a encoded file by a decoder.
The highest value micro had, in the past, that i could find, was 6
thus 100 should be safe.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Interlaced content for most codec requires it.
This patch is a stop-gap pending a serious rework to support
codecs with non 16 pixel macroblocks.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This was intended as an optimisation for skipped blocks in MPEG2
P-frames and never used elsewhere. Removing this "optimisation"
speeds up MPEG2 decoding by 1-2% (ARM Cortex-A9).
Signed-off-by: Mans Rullgard <mans@mansr.com>
When the buf and last pointers are equal, the FFSWAP() results
in an invalid call to memcpy() with same source and destination
on some targets. Although assigning a struct to itself is valid
C99, gcc does not check for this before calling memcpy().
See http://gcc.gnu.org/bugzilla/show_bug.cgi?id=32667
Signed-off-by: Mans Rullgard <mans@mansr.com>
Earlier, bits per sample was defined as 8, since
bits_per_coded_sample was used to indicate whether to ignore
the lower bits of the codeword, having values 6, 7 or 8.
g722 encodes 2 samples into one byte codeword, therefore the
bits per sample is 4. By changing this, the generated timestamps
for streams encoded with g722 become correct.
This makes timestamp generation for g722 data correct (both when
encoding and when demuxing from raw g722 files).
Signed-off-by: Martin Storsjö <martin@martin.st>
Add AV_NUM_DATA_POINTERS to simplify the bump transition.
This will allow for supporting more planar audio channels without having to
allocate separate pointer arrays.
A new field, AVCodecContext.internal is used to hold a new struct
AVCodecInternal, which has private fields that are not codec-specific and are
used by general libavcodec functions.
Moved internal_buffer, internal_buffer_count, and is_copy.
Changing the lowres value is risky because the user application may have a
local copy and not read back into it, or not undo some lowres dependant things.
A patch implementing this in ffplay is already on ffmpeg-dev, so this feature
should be back soon.
This reverts commit 125ea3ee06.
This extends the lock manager in avcodec to manage two separate
mutexes via the user-specified lock functions.
Signed-off-by: Martin Storsjö <martin@martin.st>
This code was already added by Yuvi in c82cbea68273c6f08c4d0e94fc9fd50bfdea4e2b
It was subsequently lost somehow
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Replace our incomplete w32threads implementation with x264's pthreads
w32threads wrapper.
Relicensed to LGPL with kind permission by Pegasys Inc.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
There is no valid reason the user should ever send such packets in the
first place, but the documentation for CODEC_CAP_DELAY states that the
codec is guaranteed not to get a NULL packet unless that capability is
set. That isn't true without preventing this case.
Justin Ruggles
Reinhard Tartler
Reimar Döffinger
Carl Eugen Hoyos
Michael Niedermayer
Alex Converse
Martin Storsjö
Janne Grunau
Stefano Sabatini
Rafaël Carré
Mans Rullgard
Sjoerd Simons
Ronald S. Bultje
Dustin Brody
Anton Khirnov
Steven Walters