* commit '194be1f43ea391eb986732707435176e579265aa':
lavf: switch to AVStream.time_base as the hint for the muxer timebase
Conflicts:
doc/APIchanges
libavformat/filmstripenc.c
libavformat/movenc.c
libavformat/mxfenc.c
libavformat/oggenc.c
libavformat/swf.h
libavformat/version.h
tests/ref/lavf/mkv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This avoids the following libass warning when using the subtitles
filter: "Neither PlayResX nor PlayResY defined. Assuming 384x288"
Subtitles tests change because the output is ASS and the PlayRes[XY]
ends up in the output.
Previously, AVStream.codec.time_base was used for that purpose, which
was quite confusing for the callers. This change also opens the path for
removing AVStream.codec.
The change in the lavf-mkv test is due to the native timebase (1/1000)
being used instead of the default one (1/90000), so the packets are now
sent to the crc muxer in the same order in which they are demuxed
(previously some of them got reordered because of inexact timestamp
conversion).
It has not been properly maintained for years and there is little hope
of that changing in the future.
It appears simpler to write a new replacement from scratch than
unbreaking it.
This very slightly improves compression
Found-by: Christophe Gisquet <christophe.gisquet@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The actual predictor value, set by the trellis code, never
was written back into the variable that was written into
the block header. This was accidentally removed in b304244b.
This significantly improves the audio quality of the trellis
case, which was plain broken since b304244b.
Encoding IMA QT with trellis still actually gives a slightly
worse quality than without trellis, since the trellis encoder
doesn't use the exact same way of rounding as in
adpcm_ima_qt_compress_sample and adpcm_ima_qt_expand_nibble.
Fixes part of Ticket3701
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
adpcm_ima_qt does not produce reproducible results, so it is temporarily
disabled (see #3701).
Signed-off-by: Timothy Gu <timothygu99@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This results in DefaultDuration not being written when the framerate is
not known, but as this field is purely informative, this should not
break any sane demuxers.
This corrects the bug that caused the checksums to change in
9767d7c092.
It caused the EOS flag to be set incorrectly; the ogg spec does not
allow it to be set in the middle of a logical bitstream.
Signed-off-by: Andrew Kelley <superjoe30@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
Before, header information for ogg format files was sent with the
first encoded packet.
This patch makes it so that it is possible for API users to
differentiate between headers and encoded audio. This is useful, for
example, when creating an audio stream where you want to send one set
of headers for every client that connects and then the encoded stream
of audio.
Signed-off-by: Martin Storsjö <martin@martin.st>
Based off the srt encoder. The following features are unimplemented:
- fonts, colors, sizes
- alignment and positioning
The rest works well. For example, use ffmpeg to convert subtitles into the .vtt format:
ffmpeg -i input.srt output.vtt
Signed-off-by: Aman Gupta <ffmpeg@tmm1.net>
Signed-off-by: Clément Bœsch <u@pkh.me>
* commit '6656370b858329ca07a60a2de954d5e90daa0206':
avconv: set the "encoder" tag when transcoding
Conflicts:
ffmpeg.c
tests/ref/lavf/mkv
tests/ref/seek/lavf-mkv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '93afb6c98df876b15e3d911a9450ad55f92080ce':
avconv: set output avg_frame_rate when known
Conflicts:
ffmpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '6072184e702b4b631ac72f1b66b75e5f21e0ce2d':
asfenc: use codec descriptors instead of AVCodecs to write codec info
Conflicts:
tests/ref/lavf/asf
tests/ref/seek/lavf-asf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Also, stop using AVCodecContext.codec_name as fallback, since it will be
deprecated.
Changes the result of the lavf-asf test (and its associated seektest),
since 'msmpeg4v3' gets written instead of just 'msmpeg4'.
Partially undoes commit 2c4e08d893:
riff: always generate a proper WAVEFORMATEX structure in
ff_put_wav_header
A new flag, FF_PUT_WAV_HEADER_FORCE_WAVEFORMATEX, is added to force the
use of WAVEFORMATEX rather than PCMWAVEFORMAT even for PCM codecs.
This flag is used in the Matroska muxer (the cause of the original
change) and in the ASF muxer, because the specifications for
these formats indicate explicitly that WAVEFORMATEX should be used.
Muxers for other formats will return to the original behavior of writing
PCMWAVEFORMAT when writing a header for raw PCM.
In particular, this causes raw PCM in WAV to generate the canonical
44-byte header expected by some tools.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The rational for this is another issue that plex has exposed. When it is
conducting a transcode of video to HLS for streaming, my father noticed
artifacts when played on his GoogleTV (NSZ-GT1). He sent me a test file
and I reproduced it on my device of the same model. It is important to
note that the artifacts were not present when streaming to VLC or QuickTime
Player. I copied the command-line that plex used, and conducted all of the
following tests using FFmpeg git.
Transcode to HLS: artifacts on playback
Transcode to TS: playback is fine
Cat HLS segments into a single TS: playback is fine
Segment single TS file to segments: artifacts on playback
Segment single TS file to segments using Apple's HLS segmenter: playback is
fine
At this point I carefully examined the differences between Apple's HLS
segmenter output and FFmpeg's. Among the considerable differences, I
noticed that the video PES packets always had a 0 length. So I continued:
Transcode to HLS using FFmpeg with 0 length PES packets: playback is fine.
Segment single TS to segments with 0 length PES packets: playback is fine.
All failures mentioned are only on the GTV since it is the only player on
which I could reproduce artifacts. I only tested the GTV, VLC, and
QuickTime Player though, so my test case is limited. I do not know if
other players exhibit this issue.
Since it was useful last time, I have uploaded the test file as
hls_pes_packet_length.m4v along with its associated txt file which contains
the transcode command-line that was used.
Reviewed-by: Kieran Kunhya <kierank@obe.tv>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes conversion of pal8 to rgb formats with alpha.
Updated references for 2 FATE tests which previously encoded fully
transparent images.
Based on a patch by Baptiste Coudurier <baptiste.coudurier@gmail.com>
If 384k is too high for the samplerate, choose the closest
possible
Idea to increase the bitrate from: 46439e1562
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This causes us to favor RGB8 over PAL8 when FF_LOSS_COLORQUANT is used
It probably makes sense to reinvestigate the exact scoring of pal8 when
our pal8 support improves to be supperior to rgb8
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '92b099daf4b8ef93513e38b43899cb8458a2fde3':
swscale: support converting YVYU422 pixel format
Conflicts:
libswscale/input.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Improves compatibility with XDCAM HD formats. It has been set for a long time
in ffmbc.
Reviewed-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Marton Balint <cus@passwd.hu>
* commit '3e4e2142d246699a1a3a0045ba7124b18bc34d7a':
fate: Convert the paletted output in the brenderpix tests to rgb24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Improves rgb -> gray16 conversion
Fixes Ticket3422
The pam and png output files look visually similar, in both cases the
dynamics increase to 0x0 -> 0xfffb.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Extrapolate audio timestamps based on the number of samples demuxed.
Deal with some MXF nastiness involving fractional number of
samples per EditUnit when seeking (the specs handwave this away).
Further fixes from Tomas Härdin.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The old one didn't use segmentation. One uses segmentation in all frame
types (--aq-mode=1), and the other uses all segmentation features, but
only in inter frames (mbgraph).
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This disables backward probability updates, which makes the codec more
friendly for frame-level multi-threading.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
fate: force the simple idct for xvid custom matrix test
Conflicts:
tests/fate/xvid.mak
tests/ref/fate/xvid-custom-matrix
See: ef034cbf18
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The original test without a forced idct is still useful since it tests
the switching of the idct algorithm/permutation on x86 with MMX. MMXext
or SSE2. Make sure the test runs only if MMX inline asm is available and
force -cpuflags to all.
Add the required bitexact flag for both tests.
no changes in either standard deviation or PSNR is seen in any of the changed fate
cases
MSE changes from 0.05012422 to 0.04890000
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'cfb4ee30977732674d30c20e93a761c33c743972':
fate: add a pngparser test
Conflicts:
tests/fate/image.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is useful for debugging.
Reference and ffprobe.xsd changes done and tested by Stefano Sabatini.
Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
Since we don't write lavf's string when bitexact is requested, this will
prevent the tag from being copied from the source stream.
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '58a868968df445068a143f327ced03b6a02baf0d':
FATE: drop the last partial frame in the wmv8-drm test
Conflicts:
tests/ref/fate/wmv8-drm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The encoder uses almost none of the mpegvideo infrastructure, only some
fields from MpegEncContext.
The FATE results change because now an all-zero quant matrix is written
into the file. Since it is not used for anything for ljpeg, this should
not be a problem.
This makes the USE_FLOATS == 0 available to the end user
More float optimizations can easily be added as well now
common code should be factored out into a common file once all
fixed point & floating point optimizations are done, this is to
avoid having to move code back and forth between files.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The old one didn't use segmentation. One uses segmentation in all frame
types (--aq-mode=1), and the other uses all segmentation features, but
only in inter frames (mbgraph).
* commit '874838dc6589d978611c89a40694a5074f892a76':
fate: add one select filter test
Conflicts:
tests/fate/filter-video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Originally written by Ronald S. Bultje <rsbultje@gmail.com> and
Clément Bœsch <u@pkh.me>
Further contributions by:
Anton Khirnov <anton@khirnov.net>
Diego Biurrun <diego@biurrun.de>
Luca Barbato <lu_zero@gentoo.org>
Martin Storsjö <martin@martin.st>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This changes the tests that used the internal hevc checksum to use framecrc
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Conflicts:
tests/fate/hevc.mak
tests/ref/fate/hevc-conformance-DBLK_A_SONY_3
tests/ref/fate/hevc-conformance-DBLK_B_SONY_3
tests/ref/fate/hevc-conformance-DBLK_C_SONY_3
tests/ref/fate/hevc-conformance-DELTAQP_B_SONY_3
tests/ref/fate/hevc-conformance-DELTAQP_C_SONY_3
tests/ref/fate/hevc-conformance-POC_A_Bossen_3
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The tests are disabled as 2 do not pass yet
(fate-hevc-conformance-PPS_A_qualcomm_7 and fate-hevc-conformance-RAP_A_docomo_4)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Here is an extract of fate-samples/sub/vobsub.idx, with an additional
text at the end of each line to better identify each bitmap:
timestamp: 00:04:55:445, filepos: 00001b000 Ace!
timestamp: 00:05:00:049, filepos: 00001b800 Wake up, honey!
timestamp: 00:05:02:018, filepos: 00001c800 I gotta go to work.
timestamp: 00:05:02:035, filepos: 00001d000 <???>
timestamp: 00:05:04:203, filepos: 00001d800 Look after Clayton, okay?
timestamp: 00:05:05:947, filepos: 00001e800 I'll be back tonight.
timestamp: 00:05:07:957, filepos: 00001f800 Bye! Love you.
timestamp: 00:05:21:295, filepos: 000020800 Hey, Ace! What's up?
timestamp: 00:05:23:356, filepos: 000021800 Hey, how's it going?
timestamp: 00:05:24:640, filepos: 000022800 Remember what today is? The 3rd!
timestamp: 00:05:27:193, filepos: 000023800 Look over there!
timestamp: 00:05:28:369, filepos: 000024800 Where are they going?
timestamp: 00:05:28:361, filepos: 000025000 <???>
timestamp: 00:05:29:946, filepos: 000025800 Let's go see.
timestamp: 00:05:31:230, filepos: 000026000 I can't, man. I got Clayton.
Note the two "<???>": they are basically split subtitles (with the
previous one), which the dvdsub decoder is now supposed to reconstruct
with a previous commit. But also note that while the first chunk has
increasing timestamps,
timestamp: 00:05:02:018, filepos: 00001c800
timestamp: 00:05:02:035, filepos: 00001d000
...it's not the case of the second one (and this is not an exception in the
original file):
timestamp: 00:05:28:369, filepos: 000024800
timestamp: 00:05:28:361, filepos: 000025000
For the dvdsub decoder, they need to be "filepos'ed" ordered, but the
FFDemuxSubtitlesQueue is timestamps ordered, which is the reason of the
introduction of a sub sort method in the context, to allow giving
priority to the position, and then the timestamps. With that change, the
dvdsub decoder get fed with ordered packets.
Now the packet size estimation was also broken: the filepos differences
in the vobsub index defines the full data read between two subtitles
chunks, and it is necessary to take into account what is read by the
mpegps_read_pes_header() function since the length returned by that
function doesn't count the size of the data it reads. This is fixed with
the introduction of total_read, and {old,new}_pos. By doing this change,
we can drop the unreliable len16 heuristic and simplify the whole loop.
Note that mpegps_read_pes_header() often read more than one PES packet
(typically in one call it can read 0x1ba and 0x1be chunk along with the
relevant 0x1bd packet), which triggers the "total_read + pkt_size >
psize" check. This is an expected behaviour, which could be avoided by
having a more chunked version of mpegps_read_pes_header().
The latest change is the extraction of each stream into its own
subtitles queue. If we don't do this, the maximum size for a subtitle
chunk is broken, and the previous changes can not work. Having each
stream in a different queue requires some little adjustments in the
seek code of the demuxer.
This commit is only meaningful as a whole change and can not be easily
split. The FATE test changes because it uses the vobsub demuxer.
Fixes sync in some samples (e.g. bugs 7581 and 8374 in VLC).
Based on a commit by Matthieu Bouron <matthieu.bouron@gmail.com>
Reported-by: Jean-Baptiste Kempf <jb@videolan.org>
CC: libav-stable@libav.org
This may improve compatibility of lgpegs generated by libavcodec
also encoded ljpegs become slightly smaller
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'f1eac2b8a0370b908cd691086d11f51342054730':
movenc: Use keyframes as default fragmentation point in ismv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
For codecs where decoding of a whole plane can simply
be skipped, we should offer applications to not decode
alpha for better performance (ca. 30% less CPU usage
and 40% reduced memory bandwidth).
It also means applications do not need to implement support
(even if it is rather simple) for YUVA formats in order to be
able to play these files.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Use it only on subtitle CuePoints.
With proper demuxer/splitter support this should improve the display
of subtitles right after seeking to a given point in the stream.
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Files won't validate with mkvalidtor if these two elements are missing.
Use a const "Lavf" string that wont change with library version bumps.
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The muxer has been creating files with v4 elements for some time now,
and especially now that we can mux non-experimental Opus files, reporting
the DocTypeVersion as 2 is not correct.
Signed-off-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The element was only being written when the value == 1. But the default
value of this element is 1, so this has no useful effect. This element
needs to be written when the value == 0.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The fate tests change as they used 1.2 previously
The increased size is due to:
32bit CRCs per slice by default (can be disabled),
it adds slice headers to allow decoding one slice without the others
an additional slice size field is added to make it possible to find
slices within corrupted surroundings.
these add up to about 57bit per slice more
at 50 frames and 4 slices thats 1425 byte
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
QuickTime will play multiple audio tracks concurrently if this flag is
set for multiple audio tracks. And if no subtitle track has this flag
set, QuickTime will show no subtitles in the subtitle menu.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
