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.
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().
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>
This is needed in case the get_buffer() callback doesnt set
width/height.
Ideally all decoders would make calls through some wraper
to the callbacks and that wraper would call ff_init_buffer_info()
But until thats done, the default reget buffer must call this
itself as it needs the values for the changed size check later.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The same as av_fast_malloc but uses av_mallocz and keeps extra
always-0 padding.
This does not mean the memory will be 0-initialized after each call,
but actually only after each growth of the buffer.
However this makes sure that
a) all data anywhere in the buffer is always initialized
b) the padding is always 0
c) the user does not have to bother with adding the padding themselves
Fixes another valgrind warning about use of uninitialized data,
this time with fate-vsynth1-jpegls.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This combination is quite odd and almost certainly a bug if
it happens.
Reviewed-by: Justin Ruggles <justin.ruggles@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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.
Justin Ruggles
Janne Grunau
Anton Khirnov
Martin Storsjö
Michael Niedermayer
Reimar Döffinger
Reinhard Tartler
Carl Eugen Hoyos
Alex Converse
Stefano Sabatini
Rafaël Carré
Mans Rullgard
Sjoerd Simons