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.
* qatar/master:
fate: split ADPCM and DPCM test references into separate files.
mov, mxfdec: Employ more meaningful return values.
lavc: Relax API strictness in avcodec_decode_audio3 with a custom get_buffer()
wavpack: fix clipping for 32-bit lossy mode
vb: Use bytestream2 functions
Conflicts:
libavcodec/utils.c
libavcodec/vb.c
libavformat/mxfdec.c
tests/fate/dpcm.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Extract processing of intra 16x16 blocks from intra macroblock
processing.
Also implement a function performing inverse transform and block
reconstruction for DC-only blocks in 1 pass instead of 2.
Split inter/intra macroblock handling code. This will allow further
optimizations such as performing inverse transform and block reconstruction
in a single pass as well as specialize code.
Signed-off-by: Janne Grunau <janne-libav@jannau.net>
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>
Current code would just return uninitialized data with no way
to detect this condition.
Instead, fill the whole GUID with 0 in that case.
Fixes valgrind uninitialized data errors in fate-seek-lavf_asf.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Reference decoder clips data before shifting it to final range and also
forces 32-bit lossy mode to be actually 24-bit lossy mode in order to be
able to perform proper clipping.
This is not a real error and memsetting always even when the
size did not change is overkill, but it still should be
an acceptable trade-off.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
sgidec: Use bytestream2 functions to prevent buffer overreads.
cosmetics: Move static and inline attributes to more standard places.
configure: provide libavfilter/version.h header to get_version()
swscale: change yuv2yuvX code to use cpuflag().
libx264: Don't leave max_b_frames as -1 if the user didn't set it
FATE: convert output to rgba for the targa tests which currently output pal8
fate: add missing reference files for targa tests in 9c2f9b0e2
FATE: enable the 2 remaining targa conformance suite tests
targa: add support for rgb555 palette
FATE: fix targa tests on big-endian systems
Conflicts:
libavcodec/sgidec.c
libavcodec/targa.c
libswscale/x86/output.asm
tests/fate/image.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
At the very least this should fix warnings about unused static
functions if one or more of these is not defined.
However even compilation might be broken if the compiler does
not optimize the function away completely.
This actually happens in case of the AVX function, since the
function pointer is used in an assignment that is not under
an #if and thus probably only optimized away after the function
was already marked as used.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
We do this for all other codec_tag checks in mpegvideo*/h26*
doing it here too makes the code more consistent.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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>
* cus/stable:
ffplay: silence buffer size must be a multiple of frame size
ffplay: use swr_set_compensation for audio synchronization
Merged-by: Michael Niedermayer <michaelni@gmx.at>
