The rtp_set_prft() function incorrectly calculates the timestamp delta
when RTP timestamps wrap around the 32-bit boundary. The current code:
delta_timestamp = (int64_t)timestamp - (int64_t)s->last_rtcp_timestamp;
treats both timestamps as large positive values, causing wraparound to
produce a large negative delta instead of the correct small positive delta.
For example, with a 90kHz video clock:
- last_rtcp_timestamp = 0xFFFFFF00 (near wraparound)
- timestamp = 0x00000100 (after wraparound)
- Current result: delta ≈ -4.3 billion ticks ≈ -47,721 seconds
- Expected result: delta ≈ +512 ticks ≈ +0.006 seconds
This causes prft->wallclock to jump backward by approximately:
- 90kHz video: ~47,721 seconds (~13.25 hours)
- 48kHz audio: ~89,478 seconds (~24.9 hours)
- 8kHz audio: ~536,871 seconds (~6.2 days)
Fix by casting the subtraction result to int32_t, which correctly
handles wraparound through modular arithmetic:
delta_timestamp = (int32_t)(timestamp - s->last_rtcp_timestamp);
This ensures the delta is always in the range [-2^31, 2^31-1], making
wraparound produce the correct small positive values.
Fixes timing jumps in applications that rely on Producer Reference Time
for media synchronization.
Signed-off-by: Clément Péron <peron.clem@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This allows detecting changes and regressions in side data related code, same as what
framecrc does for before already for packet data itself.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Code like FFMIN(MAX_DURATION_BUFFER_SIZE, avio_size(s->pb)) is not safe
as FFMIN() is a macro and avio_size() is thus evaluated multiple
times
Reviewed-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
add the missing data structure pkey in the tls_context
properly set this pkey and free it
Signed-off-by: Jack Lau <jacklau1222@qq.com>
Reviewed-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
As described in section F.6.1 from ETSI TS 102 366.
Found-by: nyanmisaka
Reviewed-by: Baptiste Coudurier <baptiste.coudurier@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
0. WHIP Version 3.
1. The WHIP muxer has been renamed and refined,
with improved logging context and error messages for SSL, DTLS, and RTC.
2. Magic numbers have been replaced with macros and extracted to functions,
and log levels have been altered for better clarity.
3. DTLS curve list has been updated,
and SRTP profile names have been refined for FFmpeg and OpenSSL.
4. ICE STUN magic number has been refined,
and RTP payload types have been updated based on Chrome's definition.
5. Fixed frame size has been refined to rtc->audio_par->frame_size,
and h264_mp4toannexb is now used to convert MP4/ISOM to annexb.
6. OPUS timestamp issue has been addressed,
and marker setting has been corrected after utilizing BSF.
7. DTLS handshake and ICE handling have been optimized for improved performance,
with a single handshake timeout and server role to prevent ARQ.
8. Consolidated ICE request/response handling and DTLS handshake into a single function,
and fixed OpenSSL build errors to work with Pion.
9. Merge TLS & DTLS implementation, shared BIO callbacks, read, write,
print_ssl_error, openssl_init_ca_key_cert,
init_bio_method function and shared same data structure
10. Modify configure that whip is enabled only dtls is
enabled(just support openssl for now) to fix build error
Co-authored-by: winlin <winlinvip@gmail.com>
Co-authored-by: yangrtc <yangrtc@aliyun.com>
Co-authored-by: cloudwebrtc <duanweiwei1982@gmail.com>
Co-authored-by: Haibo Chen <495810242@qq.com>
Co-authored-by: Steven Liu <lq@chinaffmpeg.org>
Co-authored-by: Jun Zhao <barryjzhao@tencent.com>
Signed-off-by: Jack Lau <jacklau1222@qq.com>
Signed-off-by: Steven Liu <lq@chinaffmpeg.org>
The follow cmd output corrupted file before the patch:
ffmpeg -f lavfi -i color=blue,trim=duration=0.04 \
-f lavfi -i anullsrc,atrim=duration=2 \
-movflags +empty_moov+hybrid_fragmented \
-frag_duration 1000000 \
-frag_interleave 1 \
output.mp4
1. first_track is the first track with track->entry != 0. As in the
command above, video track (track index 0) has a single frame. When
flush the second fragment, first_track is 1, the audio track.
2. write_moof = i == first_track, so write_moof is false for i = 0.
3. When mov->frag_interleave != 0, mov->mdat_buf != NULL, because
it contains audio data. So avio_write is called before write_moof,
that is, the data write before moof, and mov_finish_fragment
executed with wrong mdat_start.
4. With normal fmp4 output, the error isn't obvious. With
hybrid_fragmented, ffplay output.mp4 shows a lot of error messages.
Signed-off-by: Zhao Zhili <zhilizhao@tencent.com>
The default behavior for VVenC (since v1.10.0) is to create an IDR with
leading pictures for the first picture in decoding order (POC 32). This
leads to FFmpeg generating an edit list with an empty entry, skipping
the leading pictures.
This patch fixes the calculation for the start_pts, while the DTS is
negative (as produced by vvenc).
Signed-off-by: Gabriel Hege <g+ffmpeg@hege.cc>
It's not valid to call avio_close() on context that has not been open
with avio_open().
This fixes use of custom IO. (io_open / io_close2 callbacks)
Note that by default io_close2 is set to io_close2_default() which calls
avio_close(), so default case will work the same as before.
Signed-off-by: Kacper Michajłow <kasper93@gmail.com>
Reviewed-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Up until now, our muxer wrote FFV1 in video-for-windows
compatibility mode out of concern for old demuxers that
only support that (whereas the demuxer accepts V_FFV1).
This commit switches to using native mode, because
a) V_FFV1 is around long enough so that old demuxers
should not be an issue (support in FFmpeg has been added
in commit 9ae762da7e
in March 2017/FFmpeg 3.3),
b) using native mode uses fewer bytes for the CodecPrivate,
c) the VfW extradata is zero-padded to an even length
if necessary, but our demuxer forgot to undo the padding
until very recently (92e310eb82),
so that there are many versions of our demuxer around that
are buggy wrt VFW, but not V_FFV1.
This affects the FFV1 extradata checksums, specifically
the (experimental) version 4 files with error check version 2*
as created by
ffmpeg -i ../fate-suite/mpeg2/sony-ct3.bs -c:v ffv1 \
-slices 16 -frames 1 -level 4 -strict experimental ffv1.mkv
VFW files like the above created by this muxer before this patch
would not work with an old demuxer.
*: Without error check version 2, the CRC for the whole extradata
is zero, which is not changed by appending a zero byte.
Reviewed-by: compn <ff@hawaiiantel.net>
Reviewed-by: Dave Rice <dave@dericed.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
Legal since commit 1cd0a9be4b2d1e7c60184ec68404e00e46e3123e
(Jan 4) in the Cellar Matroska specification git repo.
We still hold out on muxing it due to compatibility with
old demuxers.
Reviewed-by: compn <ff@hawaiiantel.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
The structure is padded to an even length with an internal
size field to indicate the real size.
The matroska-matroska-display-metadata test (writing FFV1
in VFW mode) was affected by this.
It should also fix ticket #11613.
Reviewed-by: compn <ff@hawaiiantel.net>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
io_open and io_close2 callbacks may use opaque pointer stored in the
context. They are already inherited, so opaque should also be passed
through.
Fixes IMF playback in mpv.
Signed-off-by: Kacper Michajłow <kasper93@gmail.com>
Reviewed-by: Pierre-Anthony Lemieux <pal@sandflow.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This commit will properly set the duration field of Opus AVPackets.
Currently, duration is set to 0 on Opus packets from the RTP demuxer.
The Ogg muxer depends on the duration field to properly compute the page granule
value. Without a proper duration, the granule will be wrong, and result in
negative pts values in ogg files.
See oggenc.c:657 (ogg_write_packet_internal)
This commit calculates using the opus_duration function, which was copied
from oggparseopus.c
I moved this functionality and the existing opus extradata functionality
(added by me in 6c24f2b) into a new rtpdec_opus.c file.
Reviewed-by: Tristan Matthews <tmatth@videolan.org>
Signed-off-by: Marvin Scholz <epirat07@gmail.com>
The backwards scanning done for incomplete final packets should not
assume a specific alignment at the end of the file. Truncated files
result in hundreds of thousands of seeks if the final packet does not
fall on a specific byte boundary, which can be extremely slow.
For example, with HTTP, each backwards seek results in a separate
HTTP request.
This changes the scanning to check for the end tag 1 byte at a time
and buffers the last 1 MiB to avoid additional seek operations.
Co-authored-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Justin Ruggles <justinr@vimeo.com>
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
I have several .ts captures where video and audio codec changes even
though the PMT version does not change and the PIDs stay the same.
This happens during transition to/from slate (mpeg2 video and audio)
to network broadcast (hevc video and eac3 audio in private PES).
I've updated fate ts-demux expected results.
Fixes: signed integer overflow: -1384566925600903168 * 16 cannot be represented in type 'long'
Fixes: 407069502/clusterfuzz-testcase-minimized-ffmpeg_dem_WEBM_DASH_MANIFEST_fuzzer-5159255372267520
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
AV1DemuxContext.temporal_unit_size is zero after reading the header,
so the position set when reading the header will not be used at all.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
We may write up to 43 bits, so 5 bytes is not enough.
Fixes: Assertion n>=0 && n<=32 failed at ./libavcodec/get_bits.h:406
Fixes: 398527871/clusterfuzz-testcase-minimized-ffmpeg_dem_IAMF_fuzzer-6602025714647040
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: James Almer <jamrial@gmail.com>
The current vvc_probe function checks for SPS, PPS, and IRAP NAL units but does not enforce their detection order. This can lead to false positives when PPS or IRAP appear before SPS, causing malformed or non-compliant bitstreams to be incorrectly identified as valid VVC streams.
This commit modifies the logic to use boolean flags and enforces the correct detection sequence: SPS must be detected before PPS, and PPS before IRAP. This ensures more accurate probing and prevents invalid bitstreams from being misidentified.
For streams with wrong ordering a lower score is returned
This change addresses issues reported in:
- https://trac.ffmpeg.org/ticket/11496
- https://trac.ffmpeg.org/ticket/11087
Signed-off-by: xiongweixiao <xiongweixiaoxxw@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Codec probing was primarily added to the wav demuxer to support DTS-in-wav
files, but DTS probing functions return AVPROBE_SCORE_EXTENSION+1, so we can be
a bit more strict with the required score.
This fixes MP3 misdetections for some wav files.
Fixes ticket #11581.
Signed-off-by: Marton Balint <cus@passwd.hu>
Fixes: division by 0
Fixes: 395163171/clusterfuzz-testcase-minimized-ffmpeg_dem_IFF_fuzzer-542604339373670
Reviewed-by: Peter Ross <pross@xvid.org>
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
