This fixes duplicate timestamps on mp2 in ts with non seekable input.
It also fixed the fate pva demux timestamps.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (58 commits)
amrnbdec: check frame size before decoding.
cscd: use negative error values to indicate decode_init() failures.
h264: prevent overreads in intra PCM decoding.
FATE: do not decode audio in the nuv test.
dxa: set audio stream time base using the sample rate
psx-str: do not allow seeking by bytes
asfdec: Do not set AVCodecContext.frame_size
vqf: set packet parameters after av_new_packet()
mpegaudiodec: use DSPUtil.butterflies_float().
FATE: add mp3 test for sample that exhibited false overreads
fate: add cdxl test for bit line plane arrangement
vmnc: return error on decode_init() failure.
libvorbis: add/update error messages
libvorbis: use AVFifoBuffer for output packet buffer
libvorbis: remove unneeded e_o_s check
libvorbis: check return values for functions that can return errors
libvorbis: use float input instead of s16
libvorbis: do not flush libvorbis analysis if dsp state was not initialized
libvorbis: use VBR by default, with default quality of 3
libvorbis: fix use of minrate/maxrate AVOptions
...
Conflicts:
Changelog
doc/APIchanges
libavcodec/avcodec.h
libavcodec/dpxenc.c
libavcodec/libvorbis.c
libavcodec/vmnc.c
libavformat/asfdec.c
libavformat/id3v2enc.c
libavformat/internal.h
libavformat/mp3enc.c
libavformat/utils.c
libavformat/version.h
libswscale/utils.c
tests/fate/video.mak
tests/ref/fate/nuv
tests/ref/fate/prores-alpha
tests/ref/lavf/ffm
tests/ref/vsynth1/prores
tests/ref/vsynth2/prores
Merged-by: Michael Niedermayer <michaelni@gmx.at>
ProRes codes chroma blocks in 444 mode in different order than luma blocks,
so make both decoder and encoder read/write chroma blocks in right order.
Reported by Phil Barrett
WavPack has a comprehensive test suite, and a bunch
of corner cases.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
r_frame_rate should in theory have something to do with input framerate,
but in practice it is often made up from thin air by lavf. So unless we
are targeting a constant output framerate, it's better to just use input
stream timebase.
Brings back dropped frames in nuv and cscd tests introduced in
cd1ad18a65
* qatar/master: (34 commits)
mlp_parser: fix the channel mask value used for the top surround channel
vorbisenc: check all allocations for failure
roqaudioenc: return AVERROR codes instead of -1
roqaudioenc: set correct bit rate
roqaudioenc: use AVCodecContext.frame_size correctly.
roqaudioenc: remove unneeded sample_fmt check
ra144enc: use int16_t* for input samples rather than void*
ra144enc: set AVCodecContext.coded_frame
ra144enc: remove unneeded sample_fmt check
nellymoserenc: set AVCodecContext.coded_frame
nellymoserenc: improve error checking in encode_init()
nellymoserenc: return AVERROR codes instead of -1
libvorbis: improve error checking in oggvorbis_encode_init()
mpegaudioenc: return AVERROR codes instead of -1
libfaac: improve error checking and handling in Faac_encode_init()
avutil: add AVERROR_UNKNOWN
check for coded_frame allocation failure in several audio encoders
audio encoders: do not set coded_frame->key_frame.
g722enc: check for trellis data allocation error
libspeexenc: export encoder delay through AVCodecContext.delay
...
Conflicts:
doc/APIchanges
libavcodec/avcodec.h
libavcodec/fraps.c
libavcodec/kgv1dec.c
libavcodec/libfaac.c
libavcodec/libgsm.c
libavcodec/libvorbis.c
libavcodec/mlp_parser.c
libavcodec/roqaudioenc.c
libavcodec/vorbisenc.c
libavutil/avutil.h
libavutil/error.c
libavutil/error.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
WavPack has a comprehensive test suite, and a bunch
of corner cases.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
It is not supposed to be done outside lavc.
This is basically a revert of 818062f2f3.
It is unclear what issue this was supposed to fix, if it reappears again
it will have to be fixed in a more proper place.
The wtv-demux test change is because the sample starts with a B-frame.
* qatar/master:
shorten: Use separate pointers for the allocated memory for decoded samples.
atrac3: Fix crash in tonal component decoding.
ws_snd1: Fix wrong samples counts.
movenc: Don't set a default sample duration when creating ismv
rtp: Factorize the check for distinguishing RTCP packets from RTP
golomb: avoid infinite loop on all-zero input (or end of buffer).
bethsoftvid: synchronize video timestamps with audio sample rate
bethsoftvid: add audio stream only after getting the first audio packet
bethsoftvid: Set video packet duration instead of accumulating pts.
bethsoftvid: set packet key frame flag for audio and I-frame video packets.
bethsoftvid: fix read_packet() return codes.
bethsoftvid: pass palette in side data instead of in a separate packet.
sdp: Ignore RTCP packets when autodetecting RTP streams
proresenc: initialise 'sign' variable
mpegaudio: replace memcpy by SIMD code
vc1: prevent using last_frame as a reference for I/P first frame.
Conflicts:
libavcodec/atrac3.c
libavcodec/golomb.h
libavcodec/shorten.c
libavcodec/ws-snd1.c
tests/ref/fate/bethsoft-vid
Merged-by: Michael Niedermayer <michaelni@gmx.at>
According to unofficial documentation, the video rate is locked to the audio
sample rate. This results in proper synchronization of audio and video
timestamps from the demuxer. This only works if the first audio packet occurs
before the first video packet or the audio sample rate is the default rate of
11111 Hz, both of which are true for all samples in our archive.
Update FATE reference to account for now non-existent palette packet.
This also fixes the FATE test if frame data is not initialized in
get_buffer(), so update comment in avconv accordingly.
This changes a number of FATE results, since before this commit, the
timestamps in all tests using rawenc were made up by lavf.
In most cases, the previous timestamps were completely bogus.
In some other cases -- raw formats, mostly h264 -- the new timestamps
are bogus as well. The only difference is that timestamps invented by
the muxer are replaced by timestamps invented by the demuxer.
cscd -- avconv sets output codec timebase from r_frame_rate
and r_frame_rate is in this case some guessed number 31.42 (377/12),
which is not accurate enough to represent all timestamps. This results
in some frames having duplicate pts. Therefore, vsync 0 needs to be
changed to vsync 2 and avconv drops two frames. A proper fix in the
future would be to set output timebase to something saner in avconv.
nuv -- previous timestamps for video were wrong AND the cscd
comment applies, one frame is dropped.
vp8-signbias -- the file contains two frames with identical timestamps,
so -vsync 0 needs to be removed/changed to -vsync 2 and avconv drops one
frame.
vc1-ism -- apparrently either the demuxer lies about timestamps or the
file is broken, since dts == pts on all packets, but reordering clearly
takes place.
Current code compares the desired recording time with InputStream.pts,
which has a very unclear meaning. Change the code to use actual
timestamps of the frames passed to the encoder.
In several tests, one less frame is encoded, which is more correct.
In the idroq test one more frame is encoded, which is again more
correct.
Behavior with stream copy should be unchanged.
The output is obviously not supposed to contain video (since only
-acodec copy is specified), but that only happens because of the way -t
handling is implemented currently.
Right now those muxers use the default timebase in all cases(1/90000).
This patch avoid unnecessary rescaling and makes the printed timestamps
more readable.
Also, extend the printed information to include the timebases and packet
pts/duration and align the columns.
Obviously changes the results of all fate tests which use those two
muxers.
* qatar/master:
libx264: fix indentation.
vorbis: fix overflows in floor1[] vector and inverse db table index.
win64: add a XMM clobber test configure option.
movdec: Parse the dvc1 atom
ARM: ac3: fix ac3_bit_alloc_calc_bap_armv6
swscale: K&R formatting cosmetics for Blackfin code
frwu: lowercase the FRWU codec name
movdec: fix dts generation in fragmented files
fate: make acodec-ac3_fixed test output raw AC3
APIchanges: add missing commit hashes
swscale: implement MMX, SSE2 and AVX functions for RGB32 input.
ra144enc: drop pointless "encoder" from .long_name
bethsoftvideo: fix palette reading.
mpc7: use av_fast_padded_malloc()
mpc7: simplify handling of packet sizes that are not a multiple of 4 bytes
doc: decoding Forward Uncompressed is supported
Fix a typo in the x86 asm version of ff_vector_clip_int32()
pcmenc: Do not set avpkt->size.
ff_alloc_packet: modify the size of the packet to match the requested size
Conflicts:
doc/APIchanges
libavcodec/libx264.c
libavcodec/mpc7.c
libavformat/isom.h
libswscale/Makefile
libswscale/bfin/yuv2rgb_bfin.c
tests/ref/fate/bethsoft-vid
tests/ref/seek/ac3_ac3
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (22 commits)
frwu: Employ more meaningful return values.
fraps: Use av_fast_padded_malloc() instead of av_realloc()
mjpegdec: use av_fast_padded_malloc()
eatqi: use av_fast_padded_malloc()
asv1: use av_fast_padded_malloc()
avcodec: Add av_fast_padded_malloc().
swscale: enable dithering in MMX functions.
swscale: make rgb24 function macros slightly smaller.
avcodec.h: Remove some disabled cruft.
swscale: remove obsolete comment.
swscale-test: Drop unused argc and argv arguments from main().
zmbv: Employ more meaningful return values.
zmbvenc: Employ more meaningful return values.
vc1: prevent null pointer dereference on broken files
zmbv: check av_realloc() return values and avoid memleaks on ENOMEM
truespeech: align buffer
ac3: Do not read past the end of ff_ac3_band_start_tab.
dv: Fix small stack overread related to CVE-2011-3929 and CVE-2011-3936.
dv: Fix null pointer dereference due to ach=0
dv: check stype
...
Conflicts:
doc/APIchanges
libavcodec/asv1.c
libavcodec/avcodec.h
libavcodec/eatqi.c
libavcodec/fraps.c
libavcodec/frwu.c
libavcodec/zmbv.c
libavformat/dv.c
libswscale/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Return the correct number of consumed bytes and set *data_size = 0.
Returned size is 1 too small, leading to that 1 byte being read as the next
frame, which results in an extra blank frame at the beginning of the stream.
* qatar/master: (29 commits)
fate: add golomb-test
golomb-test: K&R formatting cosmetics
h264: Split h264-test off into a separate file - golomb-test.c.
h264-test: cleanup: drop timer invocations, commented out code and other cruft
h264-test: Remove unused DSP and AVCodec contexts and related init calls.
adpcm: Add missing stdint.h #include to fix standalone header compilation.
lavf: add functions for accessing the fourcc<->CodecID mapping tables.
lavc: set AVCodecContext.codec in avcodec_get_context_defaults3().
lavc: make avcodec_close() work properly on unopened codecs.
lavc: add avcodec_is_open().
lavf: rename AVInputFormat.value to raw_codec_id.
lavf: remove the pointless value field from flv and iv8
lavc/lavf: remove unnecessary symbols from the symbol version script.
lavc: reorder AVCodec fields.
lavf: reorder AVInput/OutputFormat fields.
mp3dec: Fix a heap-buffer-overflow
adpcmenc: remove some unneeded casts
adpcmenc: use int16_t and uint8_t instead of short and unsigned char.
adpcmenc: fix adpcm_ms extradata allocation
adpcmenc: return proper AVERROR codes instead of -1
...
Conflicts:
doc/APIchanges
libavcodec/Makefile
libavcodec/adpcmenc.c
libavcodec/avcodec.h
libavcodec/h264.c
libavcodec/libavcodec.v
libavcodec/mpc7.c
libavcodec/mpegaudiodec.c
libavcodec/options.c
libavformat/Makefile
libavformat/avformat.h
libavformat/flvdec.c
libavformat/libavformat.v
Merged-by: Michael Niedermayer <michaelni@gmx.at>
get_ue_golomb_long() is only tested for values up to 2^15 - 2 since
we can not write larger values.
Silence the test on success and return a non-zero value on error.
Use an heap scratch buffer instead of large stack buffer.
Remove unneeded includes.
Codec has only I- and skip-frames, so there is no
need for reget_buffer, change it so it works with
get_buffer.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Causes FFmpeg to pass through the correct pts values,
instead of clobbering all to AV_NOPTS_VALUE (the av_init_packet
default) to then make up new ones based on only fps when muxing.
Included are also the related FATE ref changes, which all
some reasonable on quick investigation.
Also set all H.264 references to us -vsync drop to reduce the
diff for the ref files.
Otherwise almost all H.264 references need to change, mostly due
to now starting with negative pts values.
About 20 additional H.264 conformance tests needed -vsync
drop anyway because they create pts values that are out of
order and thus not possible to mux otherwise.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Previously the decoder would raise an error.
The end result is the same, the time stamps only change
because regression tests create time stamps incorrectly.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
mpeg12: check for available bits to avoid an infinite loop
fate: add some shorthands to run groups of tests
fate: Give some tests more sensible names.
cosmetics: Rename ffsink to avsink.
Conflicts:
avconv.c
cmdutils.c
cmdutils.h
ffmpeg.c
ffplay.c
tests/fate/audio.mak
tests/fate/demux.mak
tests/fate/dpcm.mak
tests/fate/image.mak
tests/fate/lossless-audio.mak
tests/fate/lossless-video.mak
tests/fate/microsoft.mak
tests/fate/pcm.mak
tests/fate/real.mak
tests/fate/screen.mak
tests/fate/video.mak
tests/fate/voice.mak
tests/fate/wma.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
They allow to implement the if/then/else logic, which cannot be
implemented otherwise.
For example the expression:
A*B + not(A)*C
always evaluates to NaN if B is NaN, even in the case where A is 0.
* 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>
* qatar/master:
FATE: add tests for targa
ARM: fix Thumb-mode simple_idct_arm
ARM: 4-byte align start of all asm functions
rgb2rgb: rgb12to15()
swscale-test: fix stack overread.
swscale: fix invalid conversions and memory problems.
cabac: split cabac.h into declarations and function definitions
cabac: Mark ff_h264_mps_state array as static, it is only used within cabac.c.
cabac: Remove ff_h264_lps_state array.
Conflicts:
libswscale/rgb2rgb.h
libswscale/swscale_unscaled.c
tests/fate/image.mak
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>
(Does not attempt to decode percetual audio data inside.)
Code coverage: libavformat/xwma.c: 3% -> 75%
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
(Don't attempt to decode JPEG data.)
Code coverage: libavformat/smjpeg.c: 0% -> 69%
libavcodec/adpcm.c: 0% -> 10% (fresh run); 92.4% -> 93% following a FATE run
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
