CBS doesn't change its contents in any way whatsoever internally, and most
users already set it to a const array.
Signed-off-by: James Almer <jamrial@gmail.com>
These fields were added to support -merge_pmt_versions, but the mpegts demuxer
is also keeping track its programs internally, so that should be a better place
to handle it.
Also it is not a very good idea to keep fields like program_num or
pmt_stream_idx in an AVStream, because a single stream can be part of multiple
programs, multiple PMTs, so the stream attributes can refer to any program the
stream is part of.
Since they are not part of public API, lets simply remove them, or rather
replace them with placeholders for ABI compatibility with libavdevice.
Signed-off-by: Marton Balint <cus@passwd.hu>
Also make sure we are checking the old state of the streams because otherwise
some streams might already have the newly parsed stream identifiers which
corrupts matching.
Fixes streams having the same identifier mixed up on pmt version change.
Fixes ticket #9006.
Signed-off-by: Marton Balint <cus@passwd.hu>
Otherwise there can be a small period when the programs only contain the PMT
pid.
Also make sure skip_clear only affects AVProgram clear, and that pmt_pid is
always kept as the first entry of the PID list of the programs. Also reject
PMTs for programs on the wrong PID.
Signed-off-by: Marton Balint <cus@passwd.hu>
PID 0 was removed from the pid list when then PMT was parsed, it is better
to explictly avoid it from being discarded instead of keeing it in the list of
every program.
Signed-off-by: Marton Balint <cus@passwd.hu>
av_new_program returns the existing program if that already exists, in that
case it makes no sense to overwrite existing attributes.
Signed-off-by: Marton Balint <cus@passwd.hu>
INT32_MAX (2147483647) isn't exactly representable by a floating point
value, with the closest being 2147483648.0. So when rescaling a value
of 1.0, this could overflow when casting the 64-bit value returned from
lrintf() into 32 bits.
Unfortunately the properties of integer overflows don't match up well
with how a Fourier Transform operates. So clip the value before
casting to a 32-bit int.
Should be noted we don't have overflows with the table values we're
currently using. However, converting a Kaiser-Bessel window function
with a length of 256 and a parameter of 5.0 to fixed point did create
overflows. So this is more of insurance to save debugging time
in case something changes in the future.
The macro is only used during init, so it being a little slower is
not a problem.
Commit bdd31feec9 changed the SBC decoder to only set the output
sample format on init, instead of setting it explicitly on each frame,
which is correct. But the SBC parser overrides the sample format to S16,
which triggers a crash when combining the parser and the decoder.
Fix the issue by not setting the sample format anymore in the parser,
which is wrong.
Signed-off-by: James Almer <jamrial@gmail.com>
The function is not used anywhere else and is causing mingw-w64 clang
builds to fail with
ffmpeg-git/libavdevice/decklink_dec.cpp:792:5: error: no previous prototype for function 'get_bmd_timecode' [-Werror,-Wmissing-prototypes]
int get_bmd_timecode(AVFormatContext *avctx, AVTimecode *tc, AVRational frame_rate, BMDTimecodeFormat tc_format, IDeckLinkVideoInputFrame *videoFrame)
Signed-off-by: Christopher Degawa <ccom@randomderp.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
Currently skip_samples is set to start_pad if sample_time is lesser or
equal to 0. This can cause issues if the stream starts with packets that
have negative pts. Calling avformat_seek_file() with ts set to 0 on such
streams makes the mov demuxer return the right corresponding packets
(near the 0 timestamp) but set skip_samples to start_pad which is
incorrect as the audio decoder will discard the returned samples
according to skip_samples from the first packet it receives (which has
its timestamp near 0).
For example, considering the following audio stream with start_pad=1344:
[PKT pts=-1344] [PKT pts=-320] [PKT pts=704] [PKT pts=1728] [...]
Calling avformat_seek_file() with ts=0 makes the next call to
av_read_frame() return the packet with pts=-320 and a skip samples
side data set to 1344 (start_pad). This makes the audio decoder
incorrectly discard (1344 - 320) samples.
This commit makes the move demuxer adjust skip_samples according to the
stream start_pad, seek timestamp and first sample timestamp.
The above example will now result in av_read_frame() still returning the
packet with pts=-320 but with a skip samples side data set to 320
(src_pad - (seek_timestamp - first_timestamp)). This makes the audio
decoder only discard 320 samples (from pts=-320 to pts=0).
Signed-off-by: Marton Balint <cus@passwd.hu>
Runtime checks for whether the encoder is fixed-point or not are
unnecessary here as this is a template; furthermore, there is no
fixed-point EAC-3 encoder, so some checks for whether one is in EAC-3
mode can be omitted when doing fixed-point encoding.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
ff_eac3_exponent_init() set values twice when initializing a static
table; ergo the initialization code must not run concurrently with
a running EAC-3 encoder. Yet this code is executed every time an EAC-3
encoder is initialized. So use ff_thread_once() for this and also for a
similar initialization performed for all AC-3 encoders to make them all
init-threadsafe.
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Fixes: left shift of negative value -25824
Fixes: 27754/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_XMA2_fuzzer-5760255962906624
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: 9223372036854775807 + 32768 cannot be represented in type 'long'
Fixes: 27744/clusterfuzz-testcase-minimized-ffmpeg_dem_DHAV_fuzzer-5179319491756032
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Also do it for FFT_FLOAT only, as this is the only combination for which
it can be set.
Reviewed-by: Lynne <dev@lynne.ee>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
The Opus header initial padding preskip amount is always to be expressed
relative to 48kHz. However, the encoder delay returned from querying
libopus is relative to the encoding samplerate. Multiply by the
samplerate conversion factor to correct.
Signed-off-by: Arthur Taylor <art@ified.ca>
Fixes: division by 0
Fixes: 28597/clusterfuzz-testcase-minimized-ffmpeg_dem_VIVIDAS_fuzzer-5752201490333696
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes: signed integer overflow: -210824 * 16384 cannot be represented in type 'int'
Fixes: 28670/clusterfuzz-testcase-minimized-ffmpeg_AV_CODEC_ID_ALS_fuzzer-5682310846480384
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Nuo Mi
James Almer
Paul B Mahol
Gyan Doshi
Marton Balint
Lynne
Arnaud Vrac
Christopher Degawa
Matthieu Bouron
Lingjiang Fang
Timo Rothenpieler
Andreas Rheinhardt
Michael Niedermayer
Arthur Taylor