The test is not supposed to cover audio.
Also, using -vframes along with an audio stream depends on
the exact order the frames are processed by filters, it is
too much constraint to guarantee.
Add keyframe index metadata
Used to facilitate seeking; particularly for HTTP pseudo streaming.
1. read live streaming or file by sequence
2. if use add_keyframe_index option, add a mark flag at the position,
use to insert new context at the last step.
3. add the keyframes *offset* and *timestamp* into a list
4. if use add_keyframe_index option, shift the metadata data from
mark flag offset
5. insert the keyframes *offset* and *timestamp* from the list by
sequence
6. free the list
7. end.
Add FATE test case;
Reviewed-by: Lou Logan <lou@lrcd.com>
Signed-off-by: Steven Liu <liuqi@gosun.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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>
The dynamic buffer does not contain the CRC32 element so calls to avio_tell()
don't take it into account. This resulted in CueRelativePosition values being
six bytes short.
This is a regression since 6724525a15
Instead of adding yet another custom check for CRC32 to fix a size or an offset,
remove the existing ones and reserve the six bytes in the dynamic buffer.
Signed-off-by: James Almer <jamrial@gmail.com>
This also fixes a minor bug introduced in the codecpar conversion, where
the termination condition for extracting the extradata does not match
the actual extradata setting code. As a result, the packet durations
made up by lavf go back to their values before the codecpar conversion.
That is of little consequence since that code should eventually be
dropped completely.
We don't currently support values 1 (centimeters), 2 (inches) or 3 (DAR),
only the default value 0 (pixels) which doesn't need to be written.
The fate refs are updated as unknown SAR is now signaled in the output
files with the addition of the new element.
Reviewed-by: Carl Eugen Hoyos <ceffmpeg@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Using the stream timebase simply overflows
Fix integer overflow in psp framerate computation
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Implements part of ticket #4347
Tested-by: Dave Rice <dave@dericed.com>
Tested-by: Jerome Martinez <jerome@mediaarea.net>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Implements part of ticket #4347
Tested-by: Dave Rice <dave@dericed.com>
Tested-by: Jerome Martinez <jerome@mediaarea.net>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Implements part of ticket #4347
Tested-by: Dave Rice <dave@dericed.com>
Tested-by: Jerome Martinez <jerome@mediaarea.net>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Implements part of ticket #4347
Tested-by: Dave Rice <dave@dericed.com>
Tested-by: Jerome Martinez <jerome@mediaarea.net>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
Implements part of ticket #4347
Tested-by: Dave Rice <dave@dericed.com>
Tested-by: Jerome Martinez <jerome@mediaarea.net>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
The durations are never written in that situation.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: James Almer <jamrial@gmail.com>
This is a bit messy, mainly due to timestamp handling.
decode_video() relied on the fact that it could set dts on a flush/drain
packet. This is not possible with the old API, and won't be. (I think
doing this was very questionable with the old API. Flush packets should
not contain any information; they just cause a FIFO to be emptied.) This
is replaced with checking the best_effort_timestamp for AV_NOPTS_VALUE,
and using the suggested DTS in the drain case.
The modified tests (fate-cavs and others) still fails due to dropping
the last frame. This happens because the timestamp of the last frame
goes backwards (ffprobe -show_frames shows the same thing). I suspect
that this "worked" due to the best effort timestamp logic picking the
DTS over the decreasing PTS. Since this logic is in libavcodec (where
it probably shouldn't be), this can't be easily fixed. The timestamps
of the cavs samples are weird anyway, so I chose not to fix it.
Another strange thing is the timestamp handling in the video path of
process_input_packet (after the decode_video() call). It looks like
the code to increase next_dts and next_pts should be run every time
a frame is decoded - but it's needed even if output is skipped.
Fixes gapless decoding. Adjust skip_samples field correctly in case of DISCARDed audio frames.
Signed-off-by: Sasi Inguva <isasi@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This commit is initially largely based on commit 4426540 from Anton
Khirnov <anton@khirnov.net> and two following fixes (80fb19b and
fe7b21c) which were previously skipped respectively in 98e3153, c9ee36e,
and 7fe7cdc.
mpeg4-bsf-unpack-bframes FATE reference is updated because the bsf
filter now actually fixes the extradata (mpeg4_unpack_bframes_init()
changing one byte is now honored on the output extradata).
The FATE references for remove_extra change because the packet flags
were wrong and the keyframes weren't marked, causing the bsf relying on
these proprieties to not actually work as intended.
The following was fixed by James Almer:
The filter option arguments are now also parsed correctly.
A hack to propagate extradata changed by bitstream filters after the
first av_bsf_receive_packet() call is added to maintain the current
behavior. This was previously done by av_bitstream_filter_filter() and
is needed for the aac_adtstoasc bsf.
The exit_on_error was not being checked anymore, and led to an exit
error in the last frame of h264_mp4toannexb test. Restoring this
behaviour prevents erroring out. The test is still changed as a result
due to the badly filtered frame now not being written after the failure.
Signed-off-by: Clément Bœsch <u@pkh.me>
Signed-off-by: James Almer <jamrial@gmail.com>
This commit is largely based on commit 15e84ed3 from Anton Khirnov
<anton@khirnov.net> which was previously skipped in bbf5ef9d.
There are still a bunch of things raising codecpar related warnings that
need fixing, such as:
- the use of codec->debug in the interactive debug mode
- read_ffserver_streams(): it's probably broken now but there is no test
- lowres stuff
- codec copy apparently required by bitstream filters
The matroska references are updated because they now properly forward
the field_order (previously unknown, now progressive).
Thanks to James Almer for fixing a bunch of FATE issues in this commit.
Signed-off-by: Clément Bœsch <clement@stupeflix.com>
Signed-off-by: James Almer <jamrial@gmail.com>
add tests/ref/fate/filter-hls-append for FATE
add hls-list-append fate use filter make audio data and test hls_flags
append options
Signed-off-by: Steven Liu <lingjiujianke@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes regressions with stream copy and output timebase/fps being twice as fine as needed
Makes the timebase and ticks per frame handled identical which should make the
code easier to understand and work with. It does not solve the problem without
st->codec access
Suggested-by: Hendrik Leppkes
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Allows testing simple_idct12 correctness/bitexactness, as the sample
was generated using faani as idct.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
As Nvidia has put the most recent Video Codec SDK behind a double
registration wall, of which one needs manual approval of a lenghty
application, bundling this header saves everyone trying to use NVENC
from that headache.
The header is still MIT licensed and thus fine to bundle with ffmpeg.
Not bundling this header would get ffmpeg stuck at SDK v6, which is
still freely available, holding back future development of the NVENC
encoder.
If this still doesnt give the same results on all platforms then this should be
disabled
Found-by: James Almer <jamrial@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
