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>
