In parse_cinespace(), memory allocated in in_prelut[] and out_prelut[]
would leak if allocate_3dlut() failed. Replace return ret with goto end
to free memory before return error code.
Signed-off-by: Lidong Yan <502024330056@smail.nju.edu.cn>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
In put_main_header(), av_mallocz() allocates memory to local variable
buffer, buffer leaks if av_mallocz() to *out failed. Add av_free(buffer)
before return error code.
Signed-off-by: Lidong Yan <502024330056@smail.nju.edu.cn>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The ret value is checked later on again, so this check
is redundant and would cause the frame to not be unrefd on
failure as well.
So remove this check and add one before av_frame_remove_side_data
to ensure it is not called with an invalid frame.
Fix CID 1648350
Reviewed-by: Frank Plowman <post@frankplowman.com>
This would make the goto dead code and also would not properly
call avtext_context_close.
Fix CID 1646939
Reviewed-by: softworkz <softworkz@hotmail.com>
Add handling here for
sps_scaling_matrix_for_alternative_colour_space_disabled_flag.
Also add parentheses to make behaviour a little more explicit,
where &&'s precedence over || was relied on previously.
Reported-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Frank Plowman <post@frankplowman.com>
Make sure the WHIP protocol performs the SDP offer/answer
exchange with the WebRTC peer over HTTP.
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Reviewed-by: Jack Lau <jacklau1222@qq.com>
h264_annexb_insert_sps_pps (called after write_packet)
reorganizes PPS, SPS, and IDR packets in H.264 streams.
Since write_packet already validates pkt,
redundant null checks in h264_annexb_insert_sps_pps can be removed.
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
Reviewed-by: Marvin Scholz <epirat07@gmail.com>
Since 155508c6e9 specifying multiple
bsfs for different streams was broken:
"[bsfs/a=h264_metadata:bsfs/v=h264_metadata]out.mp4|..."
This incorrectly only parsed the first bsfs specification. The reason
for this is that the dictionary is modified in the iterator, hence
invalidating the iterator. The simplest fix for this is to simply
iterate from the beginning in each loop given that the previous entry
is removed.
The way streams are coded in an IAMF struct follows a scalable model where the
channel layouts for each layer may not match the channel order our API can
represent in a Native order layout.
For example, an audio element may have six coded streams in the form of two
stereo streams, followed by two mono streams, and then by another two stereo
streams, for a total of 10 channels, and define for them four scalable layers
with loudspeaker_layout values "Stereo", "5.1ch", "5.1.2ch", and "5.1.4ch".
The first layer references the first stream, and each following layer will
reference all previous streams plus extra ones.
In this case, the "5.1ch" layer will reference four streams (the first two
stereo and the two mono) to encompass six channels, which does not match out
native layout 5.1(side) given that FC and LFE come after FL+FR but before
SL+SR, and here, they are at the end.
For this reason, we need to build Custom order layouts that properly represent
what we're exporting.
----
Before:
Stream group #0:0[0x12c]: IAMF Audio Element:
Layer 0: stereo
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Layer 1: 5.1(side)
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Stream #0:1[0x1]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:2[0x2]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:3[0x3]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Layer 2: 5.1.2
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Stream #0:1[0x1]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:2[0x2]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:3[0x3]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:4[0x4]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Layer 3: 5.1.4
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Stream #0:1[0x1]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:2[0x2]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:3[0x3]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:4[0x4]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:5[0x5]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
----
AFter:
Stream group #0:0[0x12c]: IAMF Audio Element:
Layer 0: stereo
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Layer 1: 6 channels (FL+FR+SL+SR+FC+LFE)
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Stream #0:1[0x1]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:2[0x2]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:3[0x3]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Layer 2: 8 channels (FL+FR+SL+SR+FC+LFE+TFL+TFR)
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Stream #0:1[0x1]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:2[0x2]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:3[0x3]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:4[0x4]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Layer 3: 10 channels (FL+FR+SL+SR+FC+LFE+TFL+TFR+TBL+TBR)
Stream #0:0[0x0]: Audio: opus, 48000 Hz, stereo, fltp (default)
Stream #0:1[0x1]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:2[0x2]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:3[0x3]: Audio: opus, 48000 Hz, mono, fltp (dependent)
Stream #0:4[0x4]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Stream #0:5[0x5]: Audio: opus, 48000 Hz, stereo, fltp (dependent)
Signed-off-by: James Almer <jamrial@gmail.com>
Follows the proper order defined by the spec, even if mostly cosmetic, and is
also preparation for a following change.
Signed-off-by: James Almer <jamrial@gmail.com>
In most cases, the channel ids will match the standard Ambisonic Order, saving us the
need to use a custom order layout.
Signed-off-by: James Almer <jamrial@gmail.com>
After 3b26b782ee, `ebur128->true_peak` was only set to the maximum of the
current "true peak per frame" values, when it should report the true peak for
the entire stream.
Fixes: 3b26b782ee
It is unnecessary for decoders (they just set
AVCodecContext.sample_fmt). This also fixes a warning
with Clang (because sample_fmts is set directly,
not via CODEC_SAMPLEFMTS).
Reviewed-by: Peter Ross <pross@xvid.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The issue is that vulkan_device_create_internal() is only called for
devices that lavu creates by itself.
For external devices, this was never done.
This also solves some mid-function declaration warnings.
Fixes: division by 0
Fixes: 418396712/clusterfuzz-testcase-minimized-ffmpeg_dem_IFF_fuzzer-6104388018176000
Fixes: 418478219/clusterfuzz-testcase-minimized-ffmpeg_dem_IFF_fuzzer-4569544410857472
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Reviewed-by: Peter Ross <pross@xvid.org>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
