Marton Balint [Sun, 27 Dec 2020 22:49:06 +0000 (23:49 +0100)]
avformat/mpegts: use stream index based lookup with merge_pmt_versions if stream identifier matches multiple streams
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.
Marton Balint [Sun, 27 Dec 2020 22:05:16 +0000 (23:05 +0100)]
avformat/mpegts: only clear programs which no longer exist or have a new PMT
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.
Marton Balint [Sun, 27 Dec 2020 16:35:36 +0000 (17:35 +0100)]
avformat/mpegts: never discard PAT pid
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.
Lynne [Sat, 9 Jan 2021 19:41:25 +0000 (20:41 +0100)]
lavu/tx: clip when converting table values to fixed-point
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.
Arnaud Vrac [Tue, 5 Jan 2021 12:47:43 +0000 (13:47 +0100)]
sbc: do not set sample format in parser
Commit bdd31feec934 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.
avdevice/decklink_dec: mark get_frame_timecode and get_bmd_timecode static
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>
Matthieu Bouron [Fri, 30 Oct 2020 14:38:51 +0000 (15:38 +0100)]
avformat/mov: adjust skip_samples according to seek timestamp
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:
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).
avcodec/ac3enc_template: Perform compile-time checks at compile-time
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>
avcodec/[e]ac3enc: Make encoders init-threadsafe, fix race
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>
Arthur Taylor [Thu, 7 Jan 2021 20:55:59 +0000 (12:55 -0800)]
avcodec/libopusenc: Fix for header pre-skip value
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.
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>
avcodec/alsdec: Fix integer overflow with quant_cof
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>
James Almer [Sun, 15 Nov 2020 21:55:40 +0000 (18:55 -0300)]
avcodec/cbs_av1: add an option to select an operating point
This implements the function drop_obu() as defined in Setion 6.2.1 from the
spec.
In a reading only scenario, units that belong to an operating point the
caller doesn't want should not be parsed.
James Almer [Sun, 15 Nov 2020 21:55:39 +0000 (18:55 -0300)]
avcodec/cbs: allow cbs_read_fragment_content() to skip decomposition of units
The caller may not need all units in a fragment in reading only scenarios.
They could in fact alter global state stored in the private CodedBitstreamType
fields in an undesirable way.
With this change, unit decomposition can be skipped based on parsed values
within the unit.
erankor [Thu, 3 Dec 2020 08:42:52 +0000 (10:42 +0200)]
avformat/http: support retry on connection error
Add 2 new options:
- reconnect_on_http_error - a list of http status codes that should be
retried. the list can contain explicit status codes / the strings
4xx/5xx.
- reconnect_on_network_error - reconnects on arbitrary errors during
connect, e.g. ECONNRESET/ETIMEDOUT
the retry employs the same exponential backoff logic as the existing
reconnect/reconnect_at_eof flags.
related tickets:
https://trac.ffmpeg.org/ticket/6066
https://trac.ffmpeg.org/ticket/7768
Anton Khirnov [Tue, 10 Mar 2020 10:45:55 +0000 (11:45 +0100)]
mpegvideo: use the AVVideoEncParams API for exporting QP tables
Do it only when requested with the AV_CODEC_EXPORT_DATA_VIDEO_ENC_PARAMS
flag.
Drop previous code using the long-deprecated AV_FRAME_DATA_QP_TABLE*
API. Temporarily disable fate-filter-pp, fate-filter-pp7,
fate-filter-spp. They will be reenabled once these filters are converted
in following commits.
The sbr_qmf_window_us array is basically symmetric around its middle
element and therefore the latter half is currently initialized from the
first half at runtime. Yet because the first half is initialized, the
array can't be placed in .bss at all, so that one gains nothing from not
already initializing the whole array statically. Therefore this commit
does exactly this.
(There are two exceptions to the symmetry: Elements 384 and 512 are the
negations of their mirror element; for the fixed-point decoder, Q31(-x)
does not equal -Q31(x). In order to keep the array exactly the same, the
latter form has been used for these two elements.)
Reviewed-by: Lynne <dev@lynne.ee> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Finding the best codebook involves comparing different paths, where each
path is a sequence of several decisions (namely which codebook to use).
Up until now, these sequence was encoded in a NUL-terminated string and
the actual decisions were encoded as ’\0'..'\3' (which encoded 0..3).
This commit modifies this to actually encode it via 0..3 by switching
away from a C-string to a simple array with an explicit length field.
Reviewed-by: Lynne <dev@lynne.ee> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
avcodec/motion_est, mpegvideo: Make pointers to static storage const
Modifying static storage must not happen because of multithreading
(except initialization of course), so add const to the pointed-to type
for pointers that point to static storage.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
avcodec/mpegvideo: Merge ff_mpv_decode_defaults into ff_mpv_decode_init
These two are always called directly after each other (with the
exception of the calls in mpeg_decode_init() where some irrelevant
modifications of the avctx (which could just as well be done before
ff_mpv_decode_defaults(), because it doesn't have a pointer to the
AVCodecContext at all and therefore can't see these modifications at
all) are performed in between), so merge ff_mpv_decode_defaults() in
ff_mpv_decode_init().
Reviewed-by: Anton Khirnov <anton@khirnov.net> Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>