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This should be more useful for users since numerical values for channel
layout can be confusing and unintuitive.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
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Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
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The bug it was working seems to have been fixed.
This change causes ffmpeg to use the trim filter to implement
the -t option.
FATE tests are updated due to the more accurate handling of
the last packets.
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This commit is based on libav's implementation and
makes sure to compare output timestamps together.
It also reduces the differences with avconv.
The changes to the test reference files are caused
by an additional packet at the end, the timestamp
of the frame encoded by this packet is always
strictly below the limit stated by the -t option.
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* 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>
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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.
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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.
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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.
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* qatar/master:
dsputil: remove debug message in dsputil_init().
movdec: Avoid av_malloc(0) in stss
build: Drop YASM-OBJS-FFT from SUBDIR_VARS.
build: Drop unused X86-OBJS variable.
avconv: remove debugging cruft from do_video_out().
avconv: factorize setting stream_index for the output packet.
frame{crc/md5}: set the stream timebase from codec timebase.
apedec: remove unneeded #include of get_bits.h and associated macro
apedec: av_fast_malloc() instead of av_realloc()
apedec: fix handling of packet sizes that are not a multiple of 4 bytes
Conflicts:
libavcodec/apedec.c
tests/ref/fate/4xm-1
tests/ref/fate/4xm-2
tests/ref/fate/aasc
tests/ref/fate/armovie-escape124
tests/ref/fate/bethsoft-vid
tests/ref/fate/cljr
tests/ref/fate/creatureshock-avs
tests/ref/fate/cscd
tests/ref/fate/cvid-partial
tests/ref/fate/deluxepaint-anm
tests/ref/fate/dfa1
tests/ref/fate/dfa10
tests/ref/fate/dfa11
tests/ref/fate/dfa2
tests/ref/fate/dfa3
tests/ref/fate/dfa4
tests/ref/fate/dfa5
tests/ref/fate/dfa6
tests/ref/fate/dfa7
tests/ref/fate/dfa8
tests/ref/fate/dfa9
tests/ref/fate/film-cvid-pcm-stereo-8bit
tests/ref/fate/flic-af11-palette-change
tests/ref/fate/flic-magiccarpet
tests/ref/fate/fraps-v2
tests/ref/fate/fraps-v3
tests/ref/fate/h264-lossless
tests/ref/fate/interplay-mve-16bit
tests/ref/fate/interplay-mve-8bit
tests/ref/fate/mimic
tests/ref/fate/motionpixels
tests/ref/fate/mpeg2-field-enc
tests/ref/fate/msvideo1-16bit
tests/ref/fate/mtv
tests/ref/fate/nuv
tests/ref/fate/pictor
tests/ref/fate/prores-alpha
tests/ref/fate/ptx
tests/ref/fate/qtrle-16bit
tests/ref/fate/qtrle-1bit
tests/ref/fate/quickdraw
tests/ref/fate/rpza
tests/ref/fate/sierra-vmd
tests/ref/fate/targa-conformance-CCM8
tests/ref/fate/targa-conformance-UCM8
tests/ref/fate/tiertex-seq
tests/ref/fate/truemotion1-15
tests/ref/fate/truemotion1-24
tests/ref/fate/tscc-15bit
tests/ref/fate/tscc-32bit
tests/ref/fate/v210
tests/ref/fate/vc1-ism
tests/ref/fate/vc1_sa00040
tests/ref/fate/vc1_sa00050
tests/ref/fate/vc1_sa10091
tests/ref/fate/vc1_sa20021
tests/ref/fate/vmnc-16bit
tests/ref/fate/vmnc-32bit
tests/ref/fate/vp5
tests/ref/fate/vp8-sign-bias
tests/ref/fate/vqa-cc
tests/ref/fate/wmv8-drm
tests/ref/fate/yop
tests/ref/fate/zmbv-8bit
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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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.
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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.
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Also check for sufficient packet size.
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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>
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This allows for more reproducible results when using multi-threading.
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
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This adds a "fate" make target which runs the full FATE test suite.
Individual tests can be run with "make fate-$testname".
The location of the FATE test samples must be specified with the
--samples=PATH option to configure.
The tests/fate-update.sh script regenerates the references files and
test list from the online FATE database. These are checked in since
generating them requires non-standard tools.
Originally committed as revision 22552 to svn://svn.ffmpeg.org/ffmpeg/trunk
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