The new function accepts a slightly more intuitive order of paramters,
and returns an error code, thus allowing applications to report a
meaningful error message.
* qatar/master:
ffmpeg: get rid of the -vglobal option.
dct32: Add AVX implementation of 32-point DCT
dct32: Change pass 6 permutation to allow for AVX implementation
dct32: port SSE 32-point DCT to YASM
multiple inclusion guard cleanup
avio: document buffer must created with av_malloc() and friends
avio: check AVIOContext malloc failure
swscale: point out an alternative to sws_getContext
svq3: Do initialization after parsing the extradata
add changelog entries for 0.7_beta2
mp3lame: add #include required for AV_RB32 macro.
Conflicts:
Changelog
libavcodec/svq3.c
libavcodec/x86/dct32_sse.c
libavfilter/vsrc_buffer.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
configure: make executable again
LATM/AAC: Free previously initialized context on reinit.
configure: Do not unconditionally add -Wall to host CFLAGS.
configure: Set OS/2 objformat to a.out.
Add support for a.out object format to assembler macros.
fate: disable threading for encoding
fate: add comment field
fate: allow overriding default build and install dirs
mpegtsenc: Add an AVClass pointer to the private data
mpegaudio: clean up #includes
mpegaudio: move all header parsing to mpegaudiodecheader.[ch]
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Since a private class is set for this muxer, the callers will
assume that the private data starts with an AVClass pointer.
If no such member exists, the first few bytes of the struct
will be overwritten, and the class pointer may be broken at
any later time.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
qdm2: Use floating point synthesis filter.
h264: correct border check.
h264: fix loopfilter with threading at slice boundaries.
Fix ff_mpa_synth_filter_fixed() prototype
Rename costablegen.c ---> cos_tablegen.c.
Collapse tableprint.c into tableprint.h.
Simplify trig table rules
Remove potentially unstable filenames from comments in generated files.
Ignore generated tables and generated table generator programs.
Simplify CLEANFILES make variable by using wildcards.
Remove silly insults from avformat_version() Doxygen documentation.
mpegaudiodsp: fix x86 and ppc makefiles
configure: Adjust AVX assembler check.
mpegaudio: remove unused version of SAME_HEADER_MASK
mpegaudio: remove useless #undef at end of file
asfdec: add missing #include for av_bswap32()
mpegaudio: merge two #if CONFIG_FLOAT blocks
mpegaudio: move some struct definitions from mpegaudio.h
Move some mpegaudio functions to new mpegaudiodsp subsystem
Conflicts:
libavcodec/h264.c
libavcodec/x86/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In ff_id3v2_parse(), prevent unsigned integer overflow if data length
indicator is skipped and tlen is < 4.
Fix crash decoding file Allaby_cut.mp3, fix trac issue #182.
* qatar/master:
Fix compilation of iirfilter-test.
libx264: handle closed GOP codec flag
lavf: remove duplicate assignment in avformat_alloc_context.
lavf: use designated initializers for AVClasses.
flvdec: clenup debug code
asfdec: fix possible overread on broken files.
asfdec: do not fall back to binary/generic search
asfdec: reindent after previous commit c7bd5ed
asfdec: fallback to binary search internally
mpegaudio: add _fixed suffix to some names
Modify x86util.asm to ease transitioning to 10-bit H.264 assembly.
dct: build dct32 as separate object files
qdm2: include correct header for rdft
Conflicts:
ffpresets/libx264-fast.ffpreset
ffpresets/libx264-fast_firstpass.ffpreset
ffpresets/libx264-faster.ffpreset
ffpresets/libx264-faster_firstpass.ffpreset
ffpresets/libx264-medium.ffpreset
ffpresets/libx264-medium_firstpass.ffpreset
ffpresets/libx264-placebo.ffpreset
ffpresets/libx264-placebo_firstpass.ffpreset
ffpresets/libx264-slow.ffpreset
ffpresets/libx264-slow_firstpass.ffpreset
ffpresets/libx264-slower.ffpreset
ffpresets/libx264-slower_firstpass.ffpreset
ffpresets/libx264-superfast.ffpreset
ffpresets/libx264-superfast_firstpass.ffpreset
ffpresets/libx264-ultrafast.ffpreset
ffpresets/libx264-ultrafast_firstpass.ffpreset
ffpresets/libx264-veryfast.ffpreset
ffpresets/libx264-veryfast_firstpass.ffpreset
ffpresets/libx264-veryslow.ffpreset
ffpresets/libx264-veryslow_firstpass.ffpreset
libavformat/flvdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Moving the search and parsing of the 'fmt ' info the main loop of wav_read_header() allows tags that precede it to be parsed.
Creating wav_parse_fmt_tag() makes wav_read_header() easier to read.
asf_read_seek() inside the asf demuxer already does the
right thing, it tries the index and if that fails it uses
binary search. If binary search is called from outside of asfdec.c
it will fail because the asf code cannot clean up after itself.
Therefore introduce AVFMT_NOBINSEARCH that prevents the seek
code to fallback to binary search and AVFMT_NOGENSEARCH that
prevents the seek code to fallback to generic search.
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.
Create separate fields 8svx_compression (for audio compression), and
bitmap_compression (for video compression), and perform minor related
logging tweaks.
Improve clarity, also simplify the case when both types of compression
are employed in the same file.
* qatar/master: (33 commits)
rtpdec_qdm2: Don't try to parse data packet if no configuration is received
ac3enc: put the counting of stereo rematrixing bits in the same place to make the code easier to understand.
ac3enc: clean up count_frame_bits() and count_frame_bits_fixed()
mpegvideo: make FF_DEBUG_DCT_COEFF output coeffs via av_log() instead of just via AVFrame.
srtdec: make sure we don't write past the end of buffer
wmaenc: improve channel count and bitrate error handling in encode_init()
matroskaenc: make sure we don't produce invalid file with no codec ID
matroskadec: check that pointers were initialized before accessing them
lavf: fix function name in compute_pkt_fields2 av_dlog message
lavf: fix av_find_best_stream when providing a wanted stream.
lavf: fix av_find_best_stream when decoder_ret is given and using a related stream.
ffmpeg: factorize quality calculation
tiff: add support for SamplesPerPixel tag in tiff_decode_tag()
tiff: Prefer enum TiffCompr over int for TiffContext.compr.
mov: Support edit list atom version 1.
configure: Enable libpostproc automatically if GPL code is enabled.
Cosmetics: fix prototypes in oggdec
oggdec: fix memleak with continuous streams.
matroskaenc: add missing new line in av_log() call
dnxhdenc: add AVClass in private context.
...
swscale changes largely rewritten by me or replaced by baptsites due to lots of bugs in ronalds code.
Above code is also just in case its not obvios to a large extended duplicates that where cherry picked
from ffmpeg.
Conflicts:
configure
ffmpeg.c
libavformat/matroskaenc.c
libavutil/pixfmt.h
libswscale/ppc/swscale_template.c
libswscale/swscale.c
libswscale/swscale_template.c
libswscale/utils.c
libswscale/x86/swscale_template.c
tests/fate/h264.mak
tests/ref/lavfi/pixdesc_le
tests/ref/lavfi/pixfmts_copy_le
tests/ref/lavfi/pixfmts_null_le
tests/ref/lavfi/pixfmts_scale_le
tests/ref/lavfi/pixfmts_vflip_le
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The later parsing of payload data depends on the configuration
being present. If it hasn't been configured properly yet,
parsing a data packet may lead to a crash.
Signed-off-by: Martin Storsjö <martin@martin.st>
In the main loop, stream_number is incremented after checking the stream type,
so the search usually will not find the wanted stream.
This patch eliminates the useless stream_number variable and introduces a new
one, called real_stream_index to store the real stream index of the current
stream, no matter if we are looping through all the streams or only the streams
of a program.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Yet another fix for the code originally designed for use without related_stream.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Add an extra size validity check in asf_read_frame_header(). Without
this asf->packet_size_left may become negative, which triggers an
assertion failure later.
This avoids the creation of a new AVStream instead of replacing it when
a stream reset occurs (track change with some webradios for example).
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Adding _POSIX_C_SOURCE to CPPFLAGS globally produces all sorts of problems
since it causes certain system functions to be hidden on some (BSD) systems.
The solution is to only add the flag on systems that really require it, i.e.
glibc-based ones.
This change makes BSD systems compile out-of-the-box without the need for
adding specific flags manually. It also allows dropping a number of flags
set manually on a file-per-file basis, but were only present to work around
breakage introduced by the presence of _POSIX_C_SOURCE.
Also add _XOPEN_SOURCE to CPPFLAGS for glibc systems. We use XSI extensions
in several places already, so it is preferable to define it globally instead
of littering source files with individual #defines only needed for glibc.
If an MP3 file contains the string NSVs, the NSV probe will confuse it for an
NSV file. Check for 0xBEEF after a Video/Audio chunk to achieve more accuracy.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
* qatar/master: (32 commits)
10-bit H.264 x86 chroma v loopfilter asm
Port SMPTE S302M audio decoder from FFmbc 0.3. [Copyright headers corrected]
Fix crash of interlaced MPEG2 decoding
h264pred: fix one more aliasing violation.
doc/APIchanges: fill in missing hashes and dates.
flacenc: use proper initializers for AVOption default values.
lavc: deprecate named constants for deprecated antialias_algo.
aac: workaround for compilation on cygwin
swscale: extend YUV422p support to 10bits depth
tiff: add support for inverted FillOrder for uncompressed data
Remove unused softfloat implementation.
h264pred: fix aliasing violations.
rotozoom: Eliminate French variable name.
rotozoom: Check return value of fread().
rotozoom: Return an error value instead of calling exit().
rotozoom: Make init_demo() return int and check for errors on invocation.
rotozoom: Drop silly UINT8 typedef.
rotozoom: Drop some unnecessary parentheses.
rotozoom: K&R coding style cosmetics
rtsp: Only do keepalive using GET_PARAMETER if the server supports it
...
Conflicts:
Changelog
cmdutils.c
doc/APIchanges
doc/general.texi
ffmpeg.c
ffplay.c
libavcodec/h264pred_template.c
libavcodec/resample.c
libavutil/pixfmt.h
libavutil/softfloat.c
libavutil/softfloat.h
tests/rotozoom.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This FourCC is used by "mpegable AVC" codec and the file encoded with this codec decodes correctly with FFmpeg's H264 decoder.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This is more like what VLC does. If the server doesn't mention
supporting GET_PARAMETER in response to an OPTIONS request,
VLC doesn't send any keepalive requests at all. After this patch,
libavformat will still send OPTIONS keepalives if GET_PARAMETER
isn't explicitly said to be supported.
Some RTSP cameras don't support GET_PARAMETER, and will
close the connection if this is sent as keepalive request
(but support OPTIONS just fine, but probably don't need any
keepalive at all). Some other cameras don't support using
OPTIONS as keepalive, but require GET_PARAMETER instead.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (30 commits)
AVOptions: make default_val a union, as proposed in AVOption2.
arm/h264pred: add missing argument type.
h264dsp_mmx: place bracket outside #if/#endif block.
lavf/utils: fix ff_interleave_compare_dts corner case.
fate: add 10-bit H264 tests.
h264: do not print "too many references" warning for intra-only.
Enable decoding of high bit depth h264.
Adds 8-, 9- and 10-bit versions of some of the functions used by the h264 decoder.
Add support for higher QP values in h264.
Add the notion of pixel size in h264 related functions.
Make the h264 loop filter bit depth aware.
Template dsputil_template.c with respect to pixel size, etc.
Template h264idct_template.c with respect to pixel size, etc.
Preparatory patch for high bit depth h264 decoding support.
Move some functions in dsputil.c into a new file dsputil_template.c.
Move the functions in h264idct into a new file h264idct_template.c.
Move the functions in h264pred.c into a new file h264pred_template.c.
Preparatory patch for high bit depth h264 decoding support.
Add pixel formats for 9- and 10-bit yuv420p.
Choose h264 chroma dc dequant function dynamically.
...
Conflicts:
doc/APIchanges
ffmpeg.c
ffplay.c
libavcodec/alpha/dsputil_alpha.c
libavcodec/arm/dsputil_init_arm.c
libavcodec/arm/dsputil_init_armv6.c
libavcodec/arm/dsputil_init_neon.c
libavcodec/arm/dsputil_iwmmxt.c
libavcodec/arm/h264pred_init_arm.c
libavcodec/bfin/dsputil_bfin.c
libavcodec/dsputil.c
libavcodec/h264.c
libavcodec/h264.h
libavcodec/h264_cabac.c
libavcodec/h264_cavlc.c
libavcodec/h264_loopfilter.c
libavcodec/h264_ps.c
libavcodec/h264_refs.c
libavcodec/h264dsp.c
libavcodec/h264idct.c
libavcodec/h264pred.c
libavcodec/mlib/dsputil_mlib.c
libavcodec/options.c
libavcodec/ppc/dsputil_altivec.c
libavcodec/ppc/dsputil_ppc.c
libavcodec/ppc/h264_altivec.c
libavcodec/ps2/dsputil_mmi.c
libavcodec/sh4/dsputil_align.c
libavcodec/sh4/dsputil_sh4.c
libavcodec/sparc/dsputil_vis.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/x86/dsputil_mmx.c
libavformat/options.c
libavformat/utils.c
libavutil/pixfmt.h
libswscale/swscale.c
libswscale/swscale_internal.h
libswscale/swscale_template.c
tests/ref/seek/lavf_avi
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This should fix behavior introduced by commit
96573c0d76. Av_rescale_rnd() is not
lossless so if two timestamps are equal after being rescaled they are
not always actually identical. This patch use av_compare_ts() to get
always a correct result.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master:
mpegaudiodec: group #includes more sanely
mpegaudio: remove #if 0 blocks
ffmpeg.c: reset avoptions after each input/output file.
ffmpeg.c: store per-output stream sws flags.
mpegaudio: remove CONFIG_MPEGAUDIO_HP option
mpegtsenc: Clear st->priv_data when freeing it
udp: Fix receiving RTP data over multicast
rtpproto: Remove an unused variable
regtest: fix wma tests
NOT pulled: mpegaudio: remove CONFIG_AUDIO_NONSHORT
regtest: separate flags for encoding and decoding
Conflicts:
ffmpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
