* commit '1cd432e167b1a80853760c89a33606e2b5f229c2':
configure: fix libcdio check
rtsp: Allow setting the reordering buffer size via an AVOption
rtsp: Vertically align a constant definition
rtp: Update the check for distinguishing between RTP and RTCP
aac: fix build with hardcoded tables
fate: dependencies for screen codec tests
riff: Move functions around to be covered by appropriate #ifdefs
Conflicts:
configure
tests/fate/screen.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
aac_tablegen.h includes aac.h for the POW_SF2_ZERO definition, but
this also pulls in a raft of other headers, some of which are not
safe to use in code built with the host compiler.
Moving POW_SF2_ZERO to aac_tablegen_decl.h, where the declaration
of the array it relates to already resides, fixes the problems.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
yuv4mpeg: return proper error codes.
Give all anonymously typedeffed structs in headers a name
fate: Add parseutils test
parseutils-test: Drop random colors from parsing test
vf_pad/scale: use double precision for aspect ratios.
build: error on variable-length arrays
ppc: swscale: rework yuv2planeX_altivec()
ppc: fmtconvert: kill VLA in float_to_int16_interleave_altivec()
x86: dsputil: kill VLA in gmc_mmx()
libspeexenc: Updated commentary to reflect recent changes
libspeexenc: Add an option for enabling DTX
doc/APIchanges: fill in missing dates and hashes.
lavr: bump major to 1 and declare it stable.
lavr: change the type of the data buffers to uint8_t**.
lavc: deprecate the audio resampling API.
Conflicts:
cmdutils.h
configure
doc/APIchanges
ffplay.c
libavcodec/dwt.h
libavcodec/libspeexenc.c
libavfilter/vf_pad.c
libavfilter/vf_scale.c
libavformat/asf.h
tests/fate/libavutil.mak
tests/ref/fate/parseutils
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Japanese DTV uses some non standard extensions in AAC audio.
One example is 'dual mono', which combines two independent
audio into one stereo stream, storing them in left and right channels
respectively. Historically, dual mono audio has been used for
multi-lingual audio, one for local/native language, and another for english,
and usually the "main" (local language) channel should be output without
any user interactions.
The frames of those dual mono audio are allowed to set
ADTS channel_config field to 0, and just contain two SCE's *WITHOUT* PCE,
which is a non standard extension by Japanese DTV standard.
(ref. ARIB STD-B32 PartII 5.2.3)
This patch adds an AVPacket side data, AV_PKT_DATA_JP_DUALMONO,
which indicates that the AVPacket is likely to contain an audio frame
with the above dual mono extension, and has the parameter to specify
the desired channel selection in that case.
It also makes aacdec to detect dual mono and output just the desired
channel when this side data is attached.
Signed-off-by: Akihiro Tsukada <atsukada@users.sourceforge.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
float_dsp: ppc: add a separate header for Altivec function prototypes
ARM: fix float_dsp breakage from d5a7229
Add a float DSP framework to libavutil
PPC: Move types_altivec.h and util_altivec.h from libavcodec to libavutil
ARM: Move asm.S from libavcodec to libavutil
vc1dsp: mark put/avg_vc1_mspel_mc() always_inline
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
aacenc: Fix issues with huge values of bit_rate.
dv_tablegen: Drop unnecessary av_unused attribute from dv_vlc_map_tableinit().
proresenc: multithreaded quantiser search
riff: use bps instead of bits_per_coded_sample in the WAVEFORMATEXTENSIBLE header
avconv: only set the "channels" option when it exists for the specified input format
avplay: update get_buffer to be inline with avconv
aacdec: More robust output configuration.
faac: Fix multi-channel ordering
faac: Add .channel_layouts
rtmp: Support 'rtmp_playpath', an option which overrides the stream identifier
rtmp: Support 'rtmp_app', an option which overrides the name of application
avutil: add better documentation for AVSampleFormat
Conflicts:
libavcodec/aac.h
libavcodec/aacdec.c
libavcodec/aacenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Save the old output configuration (if it has been used
successfully) when trying a new configuration. If the new configuration
fails to decode, restore the last successful configuration.
* qatar/master: (38 commits)
v210enc: remove redundant check for pix_fmt
wavpack: allow user to disable CRC checking
v210enc: Use Bytestream2 functions
cafdec: Check return value of avio_seek and avoid modifying state if it fails
yop: Check return value of avio_seek and avoid modifying state if it fails
tta: Check return value of avio_seek and avoid modifying state if it fails
tmv: Check return value of avio_seek and avoid modifying state if it fails
r3d: Check return value of avio_seek and avoid modifying state if it fails
nsvdec: Check return value of avio_seek and avoid modifying state if it fails
mpc8: Check return value of avio_seek and avoid modifying state if it fails
jvdec: Check return value of avio_seek and avoid modifying state if it fails
filmstripdec: Check return value of avio_seek and avoid modifying state if it fails
ffmdec: Check return value of avio_seek and avoid modifying state if it fails
dv: Check return value of avio_seek and avoid modifying state if it fails
bink: Check return value of avio_seek and avoid modifying state if it fails
Check AVCodec.pix_fmts in avcodec_open2()
svq3: Prevent illegal reads while parsing extradata.
remove ParseContext1
vc1: use ff_parse_close
mpegvideo parser: move specific fields into private context
...
Conflicts:
libavcodec/4xm.c
libavcodec/aacdec.c
libavcodec/h264.c
libavcodec/h264.h
libavcodec/h264_cabac.c
libavcodec/h264_cavlc.c
libavcodec/mpeg4video_parser.c
libavcodec/svq3.c
libavcodec/v210enc.c
libavformat/cafdec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fate: Add tests for more AAC features.
aacps: Add missing newline in error message.
fate: Add tests for vc1/wmapro in ism.
aacdec: Add a fate test for 5.1 channel SBR.
aacdec: Turn off PS for multichannel files that use PCE based configs.
cabac: remove put_cabac_u/ueg from cabac-test.
swscale: RGB4444 and BGR444 input
FATE: add test for xWMA demuxer.
FATE: add test for SMJPEG demuxer and associated IMA ADPCM audio decoder.
mpegaudiodec: optimized iMDCT transform
mpegaudiodec: change imdct window arrangment for better pointer alignment
mpegaudiodec: move imdct and windowing function to mpegaudiodsp
mpegaudiodec: interleave iMDCT buffer to simplify future SIMD implementations
swscale: convert yuy2/uyvy/nv12/nv21ToY/UV from inline asm to yasm.
FATE: test to exercise WTV demuxer.
mjpegdec: K&R formatting cosmetics
swscale: K&R formatting cosmetics for code examples
swscale: K&R reformatting cosmetics for header files
FATE test: cvid-grayscale; ensures that the grayscale Cinepak variant is exercised.
Conflicts:
libavcodec/cabac.c
libavcodec/mjpegdec.c
libavcodec/mpegaudiodec.c
libavcodec/mpegaudiodsp.c
libavcodec/mpegaudiodsp.h
libavcodec/mpegaudiodsp_template.c
libavcodec/x86/Makefile
libavcodec/x86/imdct36_sse.asm
libavcodec/x86/mpegaudiodec_mmx.c
libswscale/swscale-test.c
libswscale/swscale.c
libswscale/swscale_internal.h
libswscale/x86/swscale_template.c
tests/fate/demux.mak
tests/fate/microsoft.mak
tests/fate/video.mak
tests/fate/wma.mak
tests/ref/lavfi/pixfmts_scale
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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:
fifo: add FIFO API test program, and fate test
fifo: add av_fifo_peek2(), and deprecate av_fifo_peek()
postprocess.c: filter name needs to be double 0 terminated
doxygen: fix wrong comment syntax, //< vs. ///<
doxygen: drop pointless star from pointer variable names
Replace deprecated av_find_stream_info() by avformat_find_stream_info().
xmv: eliminate superfluous zeroing of zero data
configure: fix typo in avconv dependency list
Conflicts:
configure
doc/APIchanges
libavutil/Makefile
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
ARM: ac3: update ff_ac3_extract_exponents_neon per 8b7b2d6
ARM: NEON optimised vector_clip_int32()
swscale: disable full_chroma_int when converting to non-24/32bpp RGB.
suggest to use av_get_bytes_per_sample() in av_get_bits_per_sample_format() doxy
ffmpeg: use av_get_bytes_per_sample() in place of av_get_bits_per_sample_fmt()
put_bits: remove ALT_BITSTREAM_WRITER
put_bits: always use intreadwrite.h macros
libavformat: Add an example how to use the metadata API
doxygen: Prefer member groups over grouping into modules
doxygen: be more permissive when searching for API examples
avformat: doxify the Metadata API
lavf: restore old behavior for custom AVIOContex with an AVFMT_NOFILE format.
lavf: use the correct pointer in av_open_input_stream().
avidec: infer absolute vs relative index from first packet
Conflicts:
libavformat/Makefile
libavformat/avidec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Before this, almost all module groups have been used for grouping functions
and fields in structures semantically. This causes them to not appear
properly in the file documentation and needlessly clutters up the "Modules"
index.
Additionally, this commit streamlines some spelling and appearances.
* qatar/master:
aacdec: Use float instead of int16_t for ltp_state to avoid needless rounding.
acelp: Remove unused gray_decode table.
dfa: Remove unused variable.
configure: Include AVX availability in summary output.
configure: use same CPPFLAGS in kFreeBSD as Linux
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
vorbisdec: Rename silly "class_" variable to plain "class".
simple_idct_alpha: Drop some useless casts.
Simplify av_log_missing_feature().
ac3enc: remove check for mismatching channels and channel_layout
If AVCodecContext.channels is 0 and AVCodecContext.channel_layout is non-zero, set channels based on channel_layout.
If AVCodecContext.channel_layout and AVCodecContext.channels are both non-zero, check to make sure they do not contradict eachother.
cosmetics: indentation
Check AVCodec.supported_samplerates and AVCodec.channel_layouts in avcodec_open().
aacdec: remove sf_scale and sf_offset.
aacdec: use a scale of 2 in the LTP MDCT rather than doubling the coefficient table values from the spec.
Define POW_SF2_ZERO in aac.h and use for ff_aac_pow2sf_tabp[] offsets instead of hardcoding 200 everywhere.
Large intensity stereo and PNS indices are legal. Clip them instead of erroring out. A magnitude of 100 corresponds to 2^25 so the will most likely result in clipped output anyway.
qpeg: use reget_buffer() in decode_frame()
ultimotion: use reget_buffer() in ulti_decode_frame()
smacker: remove unnecessary call to avctx->release_buffer in decode_frame()
avparser: don't av_malloc(0).
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Instead, scalefactors are adjusted by the offset amount, removing the need
for sf_scale, and the MDCT scales are adjusted to compensate for the higher
scalefactors. Floating-point output will be handled by modifying the MDCT
scales.
This will be beneficial for use with the audio conversion API without
requiring it to depend on all of dsputil.
Signed-off-by: Mans Rullgard <mans@mansr.com>
(cherry picked from commit c73d99e672)
This will be beneficial for use with the audio conversion API without
requiring it to depend on all of dsputil.
Signed-off-by: Mans Rullgard <mans@mansr.com>
For a PCE based configuration map the channels solely based on tags.
For an indexed configuration map the channels solely based on position.
This works with all known exotic samples including al17, elem_id0, bad_concat,
and lfe_is_sce.
Originally committed as revision 25098 to svn://svn.ffmpeg.org/ffmpeg/trunk
Passing an explicit filename to this command is only necessary if the
documentation in the @file block refers to a file different from the
one the block resides in.
Originally committed as revision 22921 to svn://svn.ffmpeg.org/ffmpeg/trunk
A large portion of this code was orignally authored by Robert Swain. The rest
was written by me. Full history is available at:
svn://svn.ffmpeg.org/soc/aac-sbr
http://github.com/aconverse/ffmpeg-heaac/tree/sbr_pub
Originally committed as revision 22316 to svn://svn.ffmpeg.org/ffmpeg/trunk
These macros are redundant. All uses are replaced with the generic
DECLARE_ALIGNED macro instead.
Originally committed as revision 22233 to svn://svn.ffmpeg.org/ffmpeg/trunk
non extradata formats. Instead lock it only after the successful decoding of a
frame. This fixes issue 999.
Originally committed as revision 20448 to svn://svn.ffmpeg.org/ffmpeg/trunk
