Split off packet parsing into a separate function. Parse full packets at
once and store them in a queue, eliminating the need for tracking
parsing state in AVStream.
The horrible unreadable loop in read_frame_internal() now isn't weirdly
ordered and doesn't contain evil gotos, so it should be much easier to
understand.
compute_pkt_fields() now invents slightly different timestamps for two
raw vc1 tests, due to has_b_frames being set a bit later. They shouldn't
be more wrong (or right) than previous ones.
* qatar/master: (27 commits)
cmdutils: use new avcodec_is_decoder/encoder() functions.
lavc: make codec_is_decoder/encoder() public.
lavc: deprecate AVCodecContext.sub_id.
libcdio: add a forgotten AVClass to the private context.
swscale: remove "cpu flags" from -sws_flags description.
proresenc: give user a possibility to alter some encoding parameters
vorbisenc: add output buffer overwrite protection
libopencore-amrnbenc: fix end-of-stream handling
ra144enc: fix end-of-stream handling
nellymoserenc: zero any leftover packet bytes
nellymoserenc: use proper MDCT overlap delay
qpeg: Use bytestream2 functions to prevent buffer overreads.
swscale: make %rep unconditional.
vp8: convert simple loopfilter x86 assembly to use named arguments.
vp8: convert idct x86 assembly to use named arguments.
vp8: convert mc x86 assembly to use named arguments.
vp8: convert loopfilter x86 assembly to use cpuflags().
vp8: convert idct/mc x86 assembly to use cpuflags().
swscale: remove now unnecessary hack.
x86inc: don't "bake" stack_offset in named arguments.
...
Conflicts:
cmdutils.c
doc/APIchanges
libavcodec/mpeg12.c
libavcodec/options.c
libavcodec/qpeg.c
libavcodec/utils.c
libavcodec/version.h
libavdevice/libcdio.c
tests/lavf-regression.sh
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes issues when the bitrate is variable or inaccurate but the
frame size has not been determined yet.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
tiertexseq: set correct block_align for audio
tiertexseq: set audio stream start time to 0
voc/avs: Do not change the sample rate mid-stream.
segafilm: use the sample rate as the time base for audio streams
ea: fix audio pts
psx-str: fix audio pts
vqf: set packet duration
tta demuxer: set packet duration
mpegaudio_parser: do not ignore information from the first parsed frame
mpegaudio_parser: be less picky about the start position
thp: set audio packet durations
avcodec: add a Vorbis parser to get packet duration
vorbisdec: read the previous window flag for long windows
lavc: free the output packet when encoding failed or produced no output.
lavc: preserve avpkt->destruct in ff_alloc_packet().
lavc: clarify the meaning of AVCodecContext.frame_number.
mpegts: Pad the packet buffer in handle_packet().
mpegts: Do not call read_sl_header() when no bytes remain in the buffer.
Conflicts:
libavcodec/mpegaudio_parser.c
libavcodec/version.h
libavformat/mpegts.c
tests/ref/fate/pva-demux
Merged-by: Michael Niedermayer <michaelni@gmx.at>
We need to set ms_stereo in encode_init() in order to avoid incorrectly
encoding the first frame as non-m/s while flagging it as m/s. Fixes an
uncomfortable pop in the left channel at the start of playback.
CC:libav-stable@libav.org
Fixes timestamp calculation.
The FATE reference is updated because timestamp calculations are now more
accurate. Previous timestamps were based on average bit rate.
* qatar/master: (29 commits)
amrwb: remove duplicate arguments from extrapolate_isf().
amrwb: error out early if mode is invalid.
h264: change underread for 10bit QPEL to overread.
matroska: check buffer size for RM-style byte reordering.
vp8: disable mmx functions with sse/sse2 counterparts on x86-64.
vp8: change int stride to ptrdiff_t stride.
wma: fix invalid buffer size assumptions causing random overreads.
Windows Media Audio Lossless decoder
rv10/20: Fix slice overflow with checked bitstream reader.
h263dec: Disallow width/height changing with frame threads.
rv10/20: Fix a buffer overread caused by losing track of the remaining buffer size.
rmdec: Honor .RMF tag size rather than assuming 18.
g722: Fix the QMF scaling
r3d: don't set codec timebase.
electronicarts: set timebase for tgv video.
electronicarts: parse the framerate for cmv video.
ogg: don't set codec timebase
electronicarts: don't set codec timebase
avs: don't set codec timebase
wavpack: Fix an integer overflow
...
Conflicts:
libavcodec/arm/vp8dsp_init_arm.c
libavcodec/fraps.c
libavcodec/h264.c
libavcodec/mpeg4videodec.c
libavcodec/mpegvideo.c
libavcodec/msmpeg4.c
libavcodec/pnmdec.c
libavcodec/qpeg.c
libavcodec/rawenc.c
libavcodec/ulti.c
libavcodec/vcr1.c
libavcodec/version.h
libavcodec/wmalosslessdec.c
libavformat/electronicarts.c
libswscale/ppc/yuv2rgb_altivec.c
tests/ref/acodec/g722
tests/ref/fate/ea-cmv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes clipping if the encoder input used the full 16 bit
input range (samples with a magnitude below 16383 worked fine).
The filtered subband samples should be 15 bit maximum, while
the code earlier produced them scaled to 16 bit.
This makes the decoder output have double the magnitude
compared to before.
The spec reference samples doesn't test the QMF at all, which
was why this part slipped past initially.
Signed-off-by: Martin Storsjö <martin@martin.st>
The container has no timestamps and the framerate isn't stored in the
data either.
The decoder sets codec timebase to experimentally found value 1/15. Do
the same for the demuxer too, it should at least be better than the
default 1/90000.
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:
mpegvideo_enc: only allocate output packet when we know there will be output
Add names for more channel layouts to the channel layout map.
sunrast: Add a sample request for RMP_RAW colormap.
avcodec: do not override pts or duration from the audio encoder
Add prores regression test.
Enable already existing rso regression test.
Add regression test for "sox" format muxer/demuxer.
Add dpx encoding regression test.
swscale: K&R formatting cosmetics for PowerPC code (part I/II)
img2: Use ff_guess_image2_codec(filename) shorthand where appropriate.
Clarify licensing information about files borrowed from libjpeg.
Mark mutable static data const where appropriate.
avplay: fix -threads option
dvbsubdec: avoid undefined signed left shift in RGBA macro
mlpdec: use av_log_ask_for_sample()
gif: K&R formatting cosmetics
png: make .long_name more descriptive
movdec: Adjust keyframe flagging in fragmented files
rv34: change most "int stride" into "ptrdiff_t stride".
Conflicts:
avprobe.c
ffplay.c
libavcodec/mlpdec.c
libavcodec/mpegvideo_enc.c
libavcodec/pngenc.c
libavcodec/x86/v210-init.c
libavfilter/vf_boxblur.c
libavfilter/vf_crop.c
libavfilter/vf_drawtext.c
libavfilter/vf_lut.c
libavfilter/vf_overlay.c
libavfilter/vf_pad.c
libavfilter/vf_scale.c
libavfilter/vf_select.c
libavfilter/vf_setpts.c
libavfilter/vf_settb.c
libavformat/img2.c
libavutil/audioconvert.c
tests/codec-regression.sh
tests/lavf-regression.sh
tests/ref/lavf/dpx
tests/ref/vsynth1/prores
tests/ref/vsynth2/prores
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (36 commits)
adpcmenc: Use correct frame_size for Yamaha ADPCM.
avcodec: add ff_samples_to_time_base() convenience function to internal.h
adx parser: set duration
mlp parser: set duration instead of frame_size
gsm parser: set duration
mpegaudio parser: set duration instead of frame_size
(e)ac3 parser: set duration instead of frame_size
flac parser: set duration instead of frame_size
avcodec: add duration field to AVCodecParserContext
avutil: add av_rescale_q_rnd() to allow different rounding
pnmdec: remove useless .pix_fmts
libmp3lame: support float and s32 sample formats
libmp3lame: renaming, rearrangement, alignment, and comments
libmp3lame: use the LAME default bit rate
libmp3lame: use avpriv_mpegaudio_decode_header() for output frame parsing
libmp3lame: cosmetics: remove some pointless comments
libmp3lame: convert some debugging code to av_dlog()
libmp3lame: remove outdated comment.
libmp3lame: do not set coded_frame->key_frame.
libmp3lame: improve error handling in MP3lame_encode_init()
...
Conflicts:
doc/APIchanges
libavcodec/libmp3lame.c
libavcodec/pcxenc.c
libavcodec/pnmdec.c
libavcodec/pnmenc.c
libavcodec/sgienc.c
libavcodec/utils.c
libavformat/hls.c
libavutil/avutil.h
libswscale/x86/swscale_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mov: Use defines for sample flags in fragments
mov: Use defines for trun flags
mov: Use defines for tfhd flags
proresenc: force bitrate not to exceed given limit
vc1parse: call vc1_init_common().
wma: don't return 0 on invalid packets.
asf: prevent packet_size_left from going negative if hdrlen > pktlen.
mjpegb: don't return 0 at the end of frame decoding.
rtpdec: Identify incorrectly signalled H263
vp8dsp: split long line.
aiff: don't skip block_align==0 check on COMM-after-SSND files.
dpcm: ignore extra unpaired bytes in stereo streams.
mp3on4: require a minimum framesize.
mpc7: assign an error level + context to av_log() msg.
huffyuv: error out on bit overrun.
dct-test: Add the missing ff_ prefix to the altivec functions
dct-test: Remove a stray declaration of a nonexistent function
movenc: Write the unknown duration as 64 bit fields in ismv
movenc: Write track durations with all bits set if duration is unknown
Conflicts:
libavcodec/dct-test.c
libavcodec/wmadec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The previous condition of 0 page size was wrong,
that would disable the mechanism for all frames at
a start of a page, thus some keyframes still would not
get their own granule.
The real problem is that header packets must not be flushed,
but they have (and must have) 0 granule and thus would
be detected as keyframes.
Add a separate parameter to mark header packets.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* 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.
Seek beyond the end will now directly return an error instead
of claiming to succeed and then return EOF immediately on next read.
This change is because before 47e015e6f1
mkv seek incorrectly never failed.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
FATE: update reference for seek-alac_mp4
sunrast: Return AVERROR values instead of -1.
sunrast: Add support for gray8 decoding.
swscale: enforce a minimum filtersize.
alacenc: use AVCodec.encode2()
alacenc: cosmetics: indentation
alacenc: consolidate bitstream writing into a single function.
alacenc: only encode frame size in header for a final smaller frame
alacenc: store current frame size in AlacEncodeContext.
alacenc: return AVERROR codes in alac_encode_frame()
alacenc: calculate a new max frame size for the final small frame
alacenc: pretty-printing and other cosmetics
alacenc: fix error handling and potential memleaks in alac_encode_init()
alacenc: do not set coded_frame->key_frame
alacenc: do not set bits_per_coded_sample
alacenc: remove unneeded frame_size check in alac_encode_frame()
tta: error out if samplerate is zero.
ttadec: fix invalid free when an error occurs while decoding 24-bit tta
wavpack: add needed braces for 2 statements inside an if block
Conflicts:
tests/ref/acodec/alac
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is so that TS fragments produced by
http://code.google.com/p/httpsegmenter/
would be compatible with JW Player.
A new member variable prev_payload_key was added to MpegTSWriteStream
to help detect transition from non-key to key frame, so that
PAT/PMT would not be produced for every keyframe in intra-only videos.
Signed-off-by: Pavel Koshevoy <pkoshevoy@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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>
