This makes -t sample-accurate for audio and will allow further
simplication in the future.
Most of the FATE changes are due to audio now being sample accurate. In
some cases a video frame was incorrectly passed with the old code, while
its was over the limit.
Most formats do not support negative timestamps, shift them to avoid
unexpected behaviour and a number of bad crashes.
CC:libav-stable@libav.org
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
To define accurately the delay between two frames, it is necessary to
have both available. Before this commit, the first frame had a delay of
0; while in practice the problem is not visible in most situation, it is
problematic with low frame rate and large scene change.
This commit notably fixes output generated with commands such as:
ffmpeg -i big_buck_bunny_1080p_h264.mov
-vf "select='gt(scene,0.4)',scale=320:-1,setpts=N/TB"
-frames:v 5 -y out.gif
Also, to avoid odd loop delays, the N-1 delay is duplicated for the last
frame.
This commit removes the badly duplicated code between the encoder and
the muxer. That may sound surprising, but the encoder is now responsible
from the encoding of the picture when muxing to a .gif file. It also
does not require anymore a manual user intervention such as a -pix_fmt
rgb24 to work properly. To summarize, output gif are now easier to
generate, code is saner and simpler, and files are smaller (thanks to
the lzw encoding which was unused so far with the default .gif output).
We can certainly make things even better, but this is the first step.
FATE is updated because of the output being produced by the encoder and
not the muxer (no lzw in the muxer), and in the seek test only the size
mismatches.
Fixes Ticket #2262
Other software does not store it in this case, and the information
is provided by the codec stream
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The QuickTime specification does not contain any hint that the atom
must not be written in some cases and both the QuickTime and the
AVID decoders do not fail if the atom is present.
This change allows to signal (visually) interlaced streams with
a codec different from uncompressed video.
As a side-effect, this fixes ticket #2202
We have to make some symetric changes elsewhere as this increases
the precission with which samples are stored.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
After making some blind tests on a small collection of music
samples for home usage. It turned out that the default cutoff
was too low.
The impact of filter_size was not clearly distinguishable (the
results were on the edge) with the music samples but turned out
to be clearly audible in some synthetic samples.
Thanks to Daniel for helping out with the listening tests.
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
It is broken, and results will be messed up when seeking.
This also fix duration displayed for streams when using -c copy.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Without this exception files with ".gif" extension by default
recognized as input suitable for image2 demuxer rather than gif.
In order to pass image through gif demuxer it was necessary
to use -f gif option.
This change affected 'make fate' test results because previously
image2 demuxer and gif decoder took only first frame of multiframe
test data, which is no longer true with gif demuxer.
Signed-off-by: Vitaliy E Sugrobov <vsugrob@hotmail.com>
Currently FFM files generated with one versions of ffmpeg generally
cannot be read by another.
By spliting data into chunks, more fields can saftely be appended to
chunks as well as new chunks added.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes playback in some circumstances (like webm in firefox).
Regression after 2c34367b.
It is also matching the Matroska specifications:
http://matroska.org/technical/specs/notes.html, "The quick eye will
notice that if a Cluster's Timecode is set to zero, it is possible to
have Blocks with a negative Raw Timecode. Blocks with a negative Raw
Timecode are not valid."
* qatar/master:
wmaenc: use float planar sample format
(e)ac3enc: use planar sample format
aacenc: use planar sample format
adpcmenc: use planar sample format for adpcm_ima_wav and adpcm_ima_qt
adpcmenc: move 'ch' variable to higher scope
adpcmenc: fix 3 instances of variable shadowing
adpcm_ima_wav: simplify encoding
libvorbis: use planar sample format
libmp3lame: use planar sample formats
vorbisenc: use float planar sample format
ffm: do not write or read the audio sample format
parseutils: fix parsing of invalid alpha values
doc/RELEASE_NOTES: update for the 9 release.
smoothstreamingenc: Add a more verbose error message
smoothstreamingenc: Ignore the return value from mkdir
smoothstreamingenc: Try writing a manifest when opening the muxer
smoothstreamingenc: Move the output_chunk_list and write_manifest functions up
smoothstreamingenc: Properly return errors from ism_flush to the caller
smoothstreamingenc: Check the output UrlContext before accessing it
Conflicts:
doc/RELEASE_NOTES
libavcodec/aacenc.c
libavcodec/ac3enc_template.c
libavcodec/wmaenc.c
tests/ref/lavf/ffm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The timebases before where only guranteed to be 1/fps precisse
and could cause AV sync errors on low fps
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
While a 25 fps stream can in general store frame durations in 1/25
units, this is not true for the timestamps. For example a 25fps
and a 25000/1001 fps stream when they are stored together might have
a matching 0 timestamp point but when for example a chapter from
this is cut the new start is no longer aligned. The issue gets
MUCH worse when the streams are lower fps, like 1 or 2 fps.
This commit thus makes the muxer choose a multiple of the
framerate as timebase that is at least about 20 micro seconds precise
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
With this, when we use a finer timebase than neccessary to store
durations the demuxer still knows what the original timebase was.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Also factorize the common options for the different mov-based tests.
Since the header is now on top in the last generated file, the data
offset in the seek test needed some updates as well.
This is consistent with stdio and is what we want to do in all cases.
Fixes a bug in the voc muxer which didn't flush in write_trailer()
previously. This is the cause of the change in the test results.
Rewrite 10 bit dpx decoder to decode into GBRP10 color space
instead of converting to RGB48.
Add 12 bit decoder to decode into GBRP12 color space.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes Ticket1627
The fate change is due to ffmpeg no longer pushing audio timestamps
aggressively up (which is what caused the AV sync issues in the ticket)
but leaving them as they are.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtmp: Add credit/copyright to librtmp authors for parts of the RTMPE code
rtmp: Move the CONFIG_ condition into the if conditions
aac: Mention abbreviation as well in long_name
build: Skip compiling rtmpdh.h if ffrtmpcrypt protocol is not enabled
doc: Add Git configuration section
configure: Add a dependency on https for rtmpts
rtp: Only choose static payload types if the sample rate and channels are right
Conflicts:
doc/git-howto.texi
libavformat/rtmpproto.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
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.
* qatar/master:
movenc: Write chan atom for all audio tracks in mov mode movies.
mpegtsenc: use avio_open_dyn_buf(), zero pointers after freeing
doc/avconv: add some details about the transcoding process.
avidec: make scale and rate unsigned.
avconv: check output stream recording time before each frame returned from filters
avconv: split selecting input file out of transcode().
avconv: split checking for active outputs out of transcode().
avfiltergraph: make some functions static.
Conflicts:
ffmpeg.c
libavfilter/avfiltergraph.c
libavfilter/internal.h
libavformat/mpegtsenc.c
tests/ref/fate/acodec-alac
tests/ref/fate/acodec-pcm-s16be
tests/ref/fate/acodec-pcm-s24be
tests/ref/fate/acodec-pcm-s32be
tests/ref/fate/acodec-pcm-s8
tests/ref/lavf/mov
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
pcmenc: set correct bitrate value
avprobe: don't print format entry name when only one was requested
Conflicts:
ffprobe.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Some of the FATE changes are due to off-by-one different rounding being used
(lrintf vs av_rescale_q).
Some fate changes are due to 1 audio frame less being encoded (the new variant seems
matching what qatar does and according to ffprobe its closer to the requested duration)
the mapchan feature sadly is lost in this commit because it depends on resampling
being done in ffmpeg.c which is now moved completely into the av filter layer
-async is broken after this commit, this will be fixed in subsequent commits
the new filter reconfiguration system is flawed and will drop a frame on each
parameter change which is why the nelly moser checksums need updating.
Conflicts:
ffmpeg.c
tests/ref/fate/smjpeg
* qatar/master:
fate: Work around non-standard wc implementations at more places
fate: work around non-standard wc implementations
x86: rv40: Mark rv40_weight functions as MMX2; they use MMX2 instructions.
ac3dsp: simplify x86 versions of ac3_max_msb_abs_int16
fate: use standard diff options
tta: Fix comment about channel number; TTA supports >2 channels.
avfilter: Move ff_get_ref_perms_string() to where it is used.
build: Add 'check' target to run all compile and test targets.
indeo3: validate new frame size before resetting decoder
indeo3: when freeing buffers, set pointers referencing them to NULL as well
indeo3: initialise pixel planes on allocation
indeo3: ensure that decoded cell data is in 7-bit range as presumed by decoder
fate: rename psx-str-v3-mdec to mdec-v3
fate: convert psx-str to a demuxer test
lavf: add mdec to is_intra_only() list
Conflicts:
doc/developer.texi
libavcodec/indeo3.c
libavfilter/video.c
libavformat/utils.c
tests/fate/demux.mak
tests/fate/video.mak
tests/lavf-regression.sh
tests/ref/vsynth1/cljr
tests/ref/vsynth1/ffvhuff
tests/ref/vsynth2/cljr
tests/ref/vsynth2/ffvhuff
Merged-by: Michael Niedermayer <michaelni@gmx.at>
diff -w is not a standard option. This fixes the reference files
to match what the tests actually output and switches to using the
standard diff -b which is sufficient to handle different line ending
styles.
Signed-off-by: Mans Rullgard <mans@mansr.com>
This partially reverts acb1730218
which would only have needed to change the checksums if channel mixing had
been properly avoided. This changes the output file size reference and the
seek test reference back to the previous values.
* qatar/master:
avcodec: add a cook parser to get subpacket duration
FATE: allow lavf tests to alter input parameters
FATE: replace the acodec-pcm_s24daud test with an enc_dec_pcm checksum test
FATE: replace the acodec-g726 test with 4 new encode/decode tests
FATE: replace current g722 encoding tests with an encode/decode test
FATE: add a pattern rule for generating asynth wav files
FATE: optionally write a WAVE header in audiogen
avutil: add audio fifo buffer
Conflicts:
doc/APIchanges
libavcodec/version.h
libavutil/avutil.h
tests/Makefile
tests/codec-regression.sh
tests/fate/voice.mak
tests/lavf-regression.sh
tests/ref/acodec/g722
tests/ref/acodec/g726
tests/ref/acodec/pcm_s24daud
tests/ref/lavf/dv_fmt
tests/ref/lavf/gxf
tests/ref/lavf/mxf
tests/ref/lavf/mxf_d10
tests/ref/seek/lavf_dv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This way we don't require a clearly defined corresponding input stream.
The result for the xwd test changes because rgb24 is now chosen instead
of bgra.
Otherwise for muxers like e.g. latmenc that never call
avio_flush (and do not have a write_trailer function)
a part of the data will always be missing.
Also update references for the voc muxer, which was also
buggy before and did not write out all data.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master: (22 commits)
rv40dsp x86: use only one register, for both increment and loop counter
rv40dsp: implement prescaled versions for biweight.
avconv: use default channel layouts when they are unknown
avconv: parse channel layout string
nutdec: K&R formatting cosmetics
vda: Signal 4 byte NAL headers to the decoder regardless of what's in the extradata
mem: Consistently return NULL for av_malloc(0)
vf_overlay: implement poll_frame()
vf_scale: support named constants for sws flags.
lavc doxy: add all installed headers to doxy groups.
lavc doxy: add avfft to the main lavc group.
lavc doxy: add remaining avcodec.h functions to a misc doxygen group.
lavc doxy: add AVPicture functions to a doxy group.
lavc doxy: add resampling functions to a doxy group.
lavc doxy: replace \ with /
lavc doxy: add encoding functions to a doxy group.
lavc doxy: add decoding functions to a doxy group.
lavc doxy: fix formatting of AV_PKT_DATA_{PARAM_CHANGE,H263_MB_INFO}
lavc doxy: add AVPacket-related stuff to a separate doxy group.
lavc doxy: add core functions/definitions to a doxy group.
...
Conflicts:
ffmpeg.c
libavcodec/avcodec.h
libavcodec/vda.c
libavcodec/x86/rv40dsp.asm
libavfilter/vf_scale.c
libavformat/nutdec.c
libavutil/mem.c
tests/ref/acodec/pcm_s24daud
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Decode output must be converted to rgb24 to avoid CRC difference
due to palette being stored in machine endianness.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master: (26 commits)
adxenc: use AVCodec.encode2()
adxenc: Use the AVFrame in ADXContext for coded_frame
indeo4: fix out-of-bounds function call.
configure: Restructure help output.
configure: Internal-only components should not be command-line selectable.
vorbisenc: use AVCodec.encode2()
libvorbis: use AVCodec.encode2()
libopencore-amrnbenc: use AVCodec.encode2()
ra144enc: use AVCodec.encode2()
nellymoserenc: use AVCodec.encode2()
roqaudioenc: use AVCodec.encode2()
libspeex: use AVCodec.encode2()
libvo_amrwbenc: use AVCodec.encode2()
libvo_aacenc: use AVCodec.encode2()
wmaenc: use AVCodec.encode2()
mpegaudioenc: use AVCodec.encode2()
libmp3lame: use AVCodec.encode2()
libgsmenc: use AVCodec.encode2()
libfaac: use AVCodec.encode2()
g726enc: use AVCodec.encode2()
...
Conflicts:
configure
libavcodec/Makefile
libavcodec/ac3enc.c
libavcodec/adxenc.c
libavcodec/libgsm.c
libavcodec/libvorbis.c
libavcodec/vorbisenc.c
libavcodec/wmaenc.c
tests/ref/acodec/g722
tests/ref/lavf/asf
tests/ref/lavf/ffm
tests/ref/lavf/mkv
tests/ref/lavf/mpg
tests/ref/lavf/rm
tests/ref/lavf/ts
tests/ref/seek/lavf_asf
tests/ref/seek/lavf_ffm
tests/ref/seek/lavf_mkv
tests/ref/seek/lavf_mpg
tests/ref/seek/lavf_rm
tests/ref/seek/lavf_ts
Merged-by: Michael Niedermayer <michaelni@gmx.at>
With this we can always know if a timestamp is based on added durations
from an unknown origin or if it is based on a correct timestamp (and possibly
added durations)
This should fix some bugs where this distinction was mixed up.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
* 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>
