* commit 'e57daa876bf0cf50782550e366e589441cd8c2bd':
adpcm: decode directly to the user-provided AVFrame
ac3: decode directly to the user-provided AVFrame
aac: decode directly to the user-provided AVFrame
8svx: decode directly to the user-provided AVFrame
Conflicts:
libavcodec/8svx.c
libavcodec/ac3dec.c
libavcodec/adpcm.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Removes limitation of max 2 channels for pcm_s8_planar decoder by moving
it to more natural place.
AV_CODEC_ID_8SVX_RAW is not used by anything anymore and is going to be
removed.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
The fate checksum change is due to the header size having been wrong.
Credit&Authorship for the code belongs to Justin Ruggles
Blame for bugs in this merging of the code belong to the Commiter
Commit message by Commiter
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Otherwise, the last byte of each stream is left uninitialized.
Signed-off-by: Nicolas George <nicolas.george@normalesup.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'bfcd4b6a1691d20aebc6d2308424c2a88334a9f0':
adpcmdec: set AVCodec.sample_fmts
twinvq: use planar sample format
ralf: use planar sample format
mpc7/8: use planar sample format
iac/imc: use planar sample format
dcadec: use float planar sample format
cook: use planar sample format
atrac3: use float planar sample format
apedec: output in planar sample format
8svx: use planar sample format
Conflicts:
libavcodec/8svx.c
libavcodec/dcadec.c
libavcodec/mpc7.c
libavcodec/mpc8.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Fix even more missing includes after the common.h removal
build: Factor out rangecoder dependencies to CONFIG_RANGECODER
build: Factor out error resilience dependencies to CONFIG_ERROR_RESILIENCE
x86: avcodec: Consistently name all init files
Add more missing includes after removing the implicit common.h
Add some more missing includes after removing the implicit common.h
Don't include common.h from avutil.h
rtmp: Automatically compute the hash for SWFVerification
Conflicts:
configure
doc/APIchanges
doc/examples/decoding_encoding.c
libavcodec/Makefile
libavcodec/assdec.c
libavcodec/audio_frame_queue.c
libavcodec/avpacket.c
libavcodec/dv_profile.c
libavcodec/dwt.c
libavcodec/libtheoraenc.c
libavcodec/rawdec.c
libavcodec/rv40dsp.c
libavcodec/tiff.c
libavcodec/tiffenc.c
libavcodec/v210dec.h
libavcodec/vc1dsp.c
libavcodec/x86/Makefile
libavfilter/asrc_anullsrc.c
libavfilter/avfilter.c
libavfilter/buffer.c
libavfilter/formats.c
libavfilter/vf_ass.c
libavfilter/vf_drawtext.c
libavfilter/vf_fade.c
libavfilter/vf_select.c
libavfilter/video.c
libavfilter/vsrc_testsrc.c
libavformat/version.h
libavutil/audioconvert.c
libavutil/error.h
libavutil/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
samples_size and samples_idx are supposed to be multiple of
channels at all time. If they are, the division is exact;
if they are not, something is very wrong in the code.
* qatar/master:
aac_latm: reconfigure decoder on audio specific config changes
latmdec: fix audio specific config parsing
Add avcodec_decode_audio4().
avcodec: change number of plane pointers from 4 to 8 at next major bump.
Update developers documentation with coding conventions.
svq1dec: avoid undefined get_bits(0) call
ARM: h264dsp_neon cosmetics
ARM: make some NEON macros reusable
Do not memcpy raw video frames when using null muxer
fate: update asf seektest
vp8: flush buffers on size changes.
doc: improve general documentation for MacOSX
asf: use packet dts as approximation of pts
asf: do not call av_read_frame
rtsp: Initialize the media_type_mask in the rtp guessing demuxer
Cleaned up alacenc.c
Conflicts:
doc/APIchanges
doc/developer.texi
libavcodec/8svx.c
libavcodec/aacdec.c
libavcodec/ac3dec.c
libavcodec/avcodec.h
libavcodec/nellymoserdec.c
libavcodec/tta.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/wmadec.c
libavformat/asfdec.c
tests/ref/seek/lavf_asf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
avcodec: add support for planar signed 8-bit PCM.
ra144enc: add sample_fmts list to ff_ra_144_encoder
smackaud: use uint8_t* for 8-bit output buffer type
smackaud: clip output samples
smackaud: use sign_extend() for difference value instead of casting
sipr: use a function pointer to select the decode_frame function
sipr: set mode based on block_align instead of bit_rate
sipr: do not needlessly set *data_size to 0 when returning an error
ra288: fix formatting of LOCAL_ALIGNED_16
udp: Allow specifying the local IP address
VC1: Add bottom field offset to block_index[] to avoid rewriting (+10L)
vc1dec: move an if() block.
vc1dec: use correct hybrid prediction threshold.
vc1dec: Partial rewrite of vc1_pred_mv()
vc1dec: take ME precision into account while scaling MV predictors.
lavf: don't leak corrupted packets
Conflicts:
libavcodec/8svx.c
libavcodec/ra288.c
libavcodec/version.h
libavformat/iff.c
libavformat/udp.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It is found in some 8svx files (e.g. ones created by SoX).
Currently the decoder reuses the 8svx functions because we already have
handling of a single large planar packet for the compressed 8svx codecs.
* qatar/master:
Add LATM demuxer
avplay: flush audio decoder with empty packets at EOF if the decoder has CODEC_CAP_DELAY set.
8svx/iff: fix decoding of compressed stereo 8svx files.
8svx: log an error message if output buffer is too small
8svx: check packet size before reading the initial sample value.
8svx: output 8-bit samples instead of 16-bit.
8svx: split delta decoding into a separate function.
mp4: Don't read an empty Decoder Config Descriptor
fate.sh: Ignore errors from rm command during cleanup.
fate.sh: Run git-pull in quiet mode to avoid console spam.
Apple ProRes decoder
rtmp: Make the input FLV parser handle data cut at any point
rv34: Check for invalid slices offsets
eval: test isnan(sqrt(-1)) instead of just sqrt(-1)
Conflicts:
Changelog
libavcodec/8svx.c
libavcodec/proresdec.c
libavcodec/version.h
libavformat/iff.c
libavformat/version.h
tests/ref/fate/eval
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Make the iff demuxer send the whole audio chunk to the decoder as a
single packet and move stereo interleaving from the iff demuxer to the
decoder.
Based on a patch by Stefano Sabatini.
git.videolan.org/ffmpeg.git
commit e280a4da2a
* qatar/master:
configure: Automatically add more flags required on symbian
mem.h: switch doxygen parameter order to match function prototype
doxygen: replace @sa tag by the more readable but equivalent @see
doxygen: use Doxygen markup for authors and web links where appropriate
doxygen: do not include license boilerplate in Doxygen documentation
ac3enc: Mark AVClasses const
ffserver: Replace two loops with one loop.
ffmpeg: Fix the check for experimental codecs
swscale: extend mmx padding.
swscale: clip unscaled colorspace conversion path.
doxygen: misc consistency cosmetics
doc: remove file name from @file directive in Doxygen usage example
doxygen: consistently place brief description
doxygen: place empty line between brief description and detailed description
avformat_open_input(): Add braces to shut up gcc warning.
Conflicts:
libavcodec/8svx.c
libavcodec/tiff.c
libavcodec/tiff.h
libavcodec/vaapi_h264.c
libavcodec/vorbis.c
libavcodec/vorbisdec.c
libavcodec/vp6.c
libswscale/swscale_unscaled.c
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Make the iff demuxer send the whole audio chunk to the decoder as a
single packet, move stereo interleaving from the iff demuxer to the
decoder, and introduce an 8svx_raw decoder which performs
stereo interleaving.
This is required for handling stereo data correctly, indeed samples
are stored like:
LLLLLL....RRRRRR
that is all left samples are at the beginning of the chunk, all right
samples at the end, so it is necessary to store and process the whole
buffer in order to decode each frame. Thus the decoder needs all the
audio chunk before it can return interleaved data.
Fix decoding of files 8svx_exp.iff and 8svx_fib.iff, fix trac issue #169.
