Before this patch, the implementation of pipe: inputs/outputs would
silently fall back to stdin/stdout for any argument not successfully
parsed by strtol().
This patch introduces an explicit error for any non-numerical arguments,
which should avoid user confusion as in #10977.
New behavior:
$ cat /tmp/video.mkv | ./ffmpeg -i pipe:aa -acodec copy -vcodec copy -f matroska pipe:1 | cat >/tmp/out.mkv
[pipe @ 0x5618c7bcf740] Non-numerical argument "aa" to pipe:
[in#0 @ 0x5618c7bced00] Error opening input: Invalid argument
Error opening input file pipe:aa.
Error opening input files: Invalid argument
Based on the patch of Nils Goroll <nils.goroll@uplex.de>.
Signed-off-by: Marton Balint <cus@passwd.hu>
Previously we always assumed that the channels are in native order, even if
they were not. The new channel layout API allows us to signal the proper
channel order, so let's do so.
Signed-off-by: Marton Balint <cus@passwd.hu>
We can do this by using an X-macro for channel map declaration and doing a
static_assert() in one pass for the check. Thanks for Andreas Rheinhardt for
the suggestion.
Signed-off-by: Marton Balint <cus@passwd.hu>
We don't know if the protocol used is referring to a local file or a remote
resource, so it is better to simply use slash as separator which works all the
time. (well, except in very special cases when the user specified a \\?\ path)
Fixes ticket #9780.
Signed-off-by: Marton Balint <cus@passwd.hu>
Overriding unknown layouts with the negotiated layout is OK, but the number of
channels should match with what was negotiated.
Signed-off-by: Marton Balint <cus@passwd.hu>
Fixes ffplay playback of unknown layouts, when SDL directly supports the audio
format, such as:
ffplay -f lavfi anullsrc=cl=2C,aformat=s16
Without the patch, "Channel layout change is not supported" errors are
generated because buffersrc (unknown 2 channel) and buffersink (stereo)
negotiated a stereo layout, but the stereo layout was never stored in the
BufferSourceContext.
This fixes a regression of 7251f90972, but this
is more of a regression of the avfilter channel layout conversion
(1f96db959c).
Signed-off-by: Marton Balint <cus@passwd.hu>
av_ts_make_time_string() used "%.6g" format, but this format was losing
precision even when the timestamp to be printed was not that large. For example
for 3 hours (10800) seconds, only 1 decimal digit was printed, which made this
format inaccurate when it was used in e.g. the silencedetect filter. Other
detection filters printing timestamps had similar issues. Also time base
parameter of the function was *AVRational instead of AVRational.
Resolve these problems by introducing a new function, av_ts_make_time_string2().
We change the used format to "%.*f", use a precision of 6, except when printing
values near 0, in which case we calculate the precision dynamically to aim for
a similar precision in normal form as with %.6g. No longer using scientific
representation can make parsing the timestamp easier for the users, we can
safely do this because the theoretical maximum of INT64_MAX*INT32_MAX still
fits into the string buffer in normal form.
We somewhat imitate %g by trimming ending zeroes and the potential decimal
point characters. In order not to trim "inf" as well, we assume that the
decimal point string does not contain the letter "f". Note that depending on
printf %f implementation, we might trim "infinity" to "inf".
Thanks for Allan Cady for bringing up this issue.
Signed-off-by: Marton Balint <cus@passwd.hu>
- Properly initialize all the mappings to -1 by default.
- Make sure every output channel is assigned exactly once
- Autodetect a native layout when only native channels are present
- Always honor the user specified layout, but make sure the mapping is
compatible with it
The last item is a regression from 4af412be71.
Signed-off-by: Marton Balint <cus@passwd.hu>
In this case in_channel_idx was never set and the default 0 was used.
Suprisingly no one noticed that the respective fate test output was wrong.
Signed-off-by: Marton Balint <cus@passwd.hu>
Previously we always assumed that the channels are in native order, even if
they were not. The new channel layout API allows us to signal the proper
channel order, so let's do so.
Fixes ticket #98.
Also make initialization/uninitialization behaviour more explicit in the docs,
and make sure we do not leak a channel map on error.
Signed-off-by: Marton Balint <cus@passwd.hu>
Also make use of the av_channel_from_string() function to determine the channel
id. This fixes some parse issues in av_channel_layout_from_string().
Signed-off-by: Marton Balint <cus@passwd.hu>
We lacked tests which supposed to fail, and there are some which should fail
but right now it does not. This will be fixed in a later commit.
Signed-off-by: Marton Balint <cus@passwd.hu>
Deduplicates a lot of code.
Some minor differences (mostly white space and inconsistent use of quotes) are
expected in the fate tests, there was no point aiming for exactly the same
formatting.
Signed-off-by: Marton Balint <cus@passwd.hu>
This makes the wav and pcm demuxer demux bigger packets, which is more
efficient.
As a side effect of the bigger packets, audio durations can become less exact
for command lines such as "ffmpeg -i $INPUT -c:a copy -t 1.0 $OUTPUT".
Signed-off-by: Marton Balint <cus@passwd.hu>
- Remove the 1024 cap on the number of samples, for high sample rate audio it
was suboptimal, calculate the low neighbour power of two for the number of
samples (audio blocks) instead.
- Make the function work correctly also for non-pcm codecs by using the stream
bitrate to estimate the target packet size. A previous version of this patch
used av_get_audio_frame_duration2() the estimate the desired packet size, but
for some codecs that returns the duration of a single audio frame regardless
of frame_bytes.
- Fallback to 4096/block_align*block_align if bitrate is not available.
Signed-off-by: Marton Balint <cus@passwd.hu>
Muxing multiple streams to raw files is allowed but the packets are
interleaved, so the output is dependant of packet size.
Signed-off-by: Marton Balint <cus@passwd.hu>
The samples I found all have 2000 sample packets, and by forcing the packet
size with a bsf we could automagically make muxing work for packets containing
more than 3640 samples.
Signed-off-by: Marton Balint <cus@passwd.hu>
The mpegts code historically tries to strip (the first) metadata access unit
header from synchronous KLV metadata, but the detection for such streams was
unreliable causing strips of asynchronous metadata or ID3 as well.
MISB ST 1402 specifies required stream type, stream id and registration
descriptor (which eventually maps to the codec ID) so let's use all of these
for reliable detection.
Fixes a regression caused by 468615f204.
Fixes ticket #10828, #10883.
Signed-off-by: Marton Balint <cus@passwd.hu>
The wav demuxer by default tried to demux 4096-byte packets which caused
packets with very few number of samples for files with high channel count.
This caused a significant overhead especially since the latest ffmpeg.c
threading changes.
So let's use a similar approach for selecting audio frame size which is already
used in the PCM demuxer, which is to read 25 times per second but at most 1024
samples.
Signed-off-by: Marton Balint <cus@passwd.hu>
Since PTS is changed randomly for every audio frame, it matters. Also add some
forgotten filter dependencies.
Signed-off-by: Marton Balint <cus@passwd.hu>
Depending on input chunk size noticable corrpution was hearable, here is an
example command line:
ffplay -f lavfi -i "sine=440:r=8000:samples_per_frame=32,aresample=24000:filter_size=1:phase_shift=0"
Fix this by rounding the fixed point fractions up instead of down.
Signed-off-by: Marton Balint <cus@passwd.hu>
GEN=1 is used to generate reference files in the source tree, not
auto-generated reference samples.
Without this patch GEN=1 could overwrite the auto generated reference files
in each test where they are used causing failures.
Signed-off-by: Marton Balint <cus@passwd.hu>
We typically are only interesed in a single type of metadata set, so it is
better to keep them separated instead of always filtering for them.
Also use av_dynarray_add for increasing their array.
Signed-off-by: Marton Balint <cus@passwd.hu>
UUIDs do not have to be unique if their type sets them apart, so avoid using
AnyType, since we are only interested in specific types.
Signed-off-by: Marton Balint <cus@passwd.hu>
Jpeg2000 decoder is decoding in native endian, so let's use the same workaround
as in fate-mxf-probe-applehdr10.
Fixes ticket #10868.
Signed-off-by: Marton Balint <cus@passwd.hu>
This is the proper poll mode for waiting for an incoming connection according
to the SRT API docs.
Fixes ticket #9142.
Signed-off-by: Marton Balint <cus@passwd.hu>
By using AnyType for resolving a strong reference we searched among all types,
not just the ones which can be the target of the reference, which in some cases
caused to find the wrong type, if the metadata set UUIDs were not unique.
UUIDs do not have to be unique if their type sets them apart, SMPTE 377M says:
> StrongRef: 'One to One’ relationship between sets and implemented in MXF
> with UUIDs. Strong References are typed which means that the definition
> identifies the kind of set which is the target of the reference.
Fixes ticket #10865.
Signed-off-by: Marton Balint <cus@passwd.hu>
The old layout happened to be a native layout and therefore missed some
recently fixed layout parsing bugs.
Signed-off-by: Marton Balint <cus@passwd.hu>
A lot of changes and fixes to channel layout parsing, notably
- get rid of dynamic allocation of channel positions
- signal unimplemented speaker positions as unknown instead of failure, but
warn the user about it
- native order, and that a single channel only appears once was always assumed
for less than 64 channels, obviously this was incorrect
Signed-off-by: Marton Balint <cus@passwd.hu>
The channel designation metadata should not override the number of channels.
Let's warn the user if it is inconsistent, and keep the channel layout
unspecified.
Before the conversion to the channel layout API the code only set the mask, but
never overridden the channel count, so this restores the old behaviour.
Signed-off-by: Marton Balint <cus@passwd.hu>
Existing code could have caused wrong channel order signalling or reduced
channel count if a channel designation appeared multiple times. This is
actually an old bug, but the conversion to the new channel layout API made it
visible, because now the code overrides the proper channel count with the one
calculated from the mask.
Signed-off-by: Marton Balint <cus@passwd.hu>
Yadif filter assumed that the output timebase is always half of the input
timebase. This is not true if halving the input time base is not representable
as an AVRational causing the output timestamps to be invalidly scaled in such a
case.
So let's use av_reduce instead of av_mul_q when calculating the output time
base and if the conversion is inexact then let's fall back to the original
timebase which probably makes more parctical sense than using x/INT_MAX.
Fixes invalidly scaled pts_time values in this command line:
ffmpeg -f lavfi -i testsrc -vf settb=tb=1/2000000000,yadif,showinfo -f null none
Signed-off-by: Marton Balint <cus@passwd.hu>
Fixes parsing small timebases from expressions (where the expression API
converts the result to double), like in this command line:
ffprobe -f lavfi -i testsrc=d=1,settb=1/2000000000 -show_streams -show_entries stream=time_base
Before the patch timebase was parsed as 1/1999999999.
Signed-off-by: Marton Balint <cus@passwd.hu>
Filter graph reconfigurations can cause nontrivial issues, e.g. PTS
discontinuities, so it is better to warn the user about them.
Signed-off-by: Marton Balint <cus@passwd.hu>
- Fixes YA formats, because previous code always assumed alpha as the 4th
component.
- Fixes PAL format (as long as 0 is black, as in a systematic palette), because
previous code assumed it as limited Y.
- Fixes XYZ format because it does not need nonzero chroma components
- Fixes xv30be as the bitstream mode got merged to the non-bitstream mode.
Signed-off-by: Marton Balint <cus@passwd.hu>
Also rename the contexts and the functions so their names will reflect their
intended size.
With the earlier patch this fixes the audio corruption regression caused by
6ba0aa1770.
Fixes ticket #10029.
Signed-off-by: Marton Balint <cus@passwd.hu>
Improves the audio corruption regression caused by
6ba0aa1770 reported in ticket #10029.
There is still however a noticable audio glitch, so the FFT conversion to AVTX
probably also needs some modifications.
Signed-off-by: Marton Balint <cus@passwd.hu>
av_random_bytes() can use OS provided strong random functions and does not
depend soley on openssl/gcrypt external libraries.
Fixes ticket #10441.
Signed-off-by: Marton Balint <cus@passwd.hu>
Also fix a couple of possible overflows while at it.
Fixes the negative initial timestamps in ticket #10358.
Signed-off-by: Marton Balint <cus@passwd.hu>
BMDTimeValue is defined as LONGLONG on Windows, but int64_t on Linux/Mac.
Fixes format string warnings:
libavdevice/decklink_enc.cpp: In function ‘void construct_cc(AVFormatContext*, decklink_ctx*, AVPacket*, klvanc_line_set_s*)’:
libavdevice/decklink_enc.cpp:424:48: warning: format ‘%lld’ expects argument of type ‘long long int’, but argument 4 has type ‘BMDTimeValue {aka long int}’ [-Wformat=]
ctx->bmd_tb_num, ctx->bmd_tb_den);
~~~~~~~~~~~~~~~ ^
libavdevice/decklink_enc.cpp:424:48: warning: format ‘%lld’ expects argument of type ‘long long int’, but argument 5 has type ‘BMDTimeValue {aka long int}’ [-Wformat=]
Signed-off-by: Marton Balint <cus@passwd.hu>
Also make duration check for mvhd more consistent with the others, write
version 1 of mvhd if duration is at least INT32_MAX instead of UINT32_MAX.
Signed-off-by: Marton Balint <cus@passwd.hu>
Commit 23eeffcd48 added a hack to support invalid
files where the creation date was encoded as a classic unix timestamp. Let's
reduce the scope of the hack by only applying it to version 0 mdhd/mvhd atoms.
Also warn the user of such possibly broken files.
Signed-off-by: Marton Balint <cus@passwd.hu>
RIP, if exists is the last KLV item in the MXF files therefore we can stop
parsing the file if it is encountered. This allows us to support files created by
broken muxers such as OpenCube MXFTk Advanced 2.8.0.0.1. which dumps some extra
garbage after the RIP.
Signed-off-by: Marton Balint <cus@passwd.hu>
When writing a subtitle SSA/ASS subtitle file, the AVCodecParameters::extradata
buffer is written directly to the output. In the case where the buffer is
filled from a matroska source file produced by some older versions of
Handbrake, this buffer ends with a null terminating character, which is then
erroneously copied into the middle of the output file. The change here avoids
this problem by treating it as a string rather than a raw buffer. This way it
is agnostic as to whether the source buffer was null terminated or not.
Fixes ticket #10203.
Reported-by: Tim Angus <tim at ngus.net>
Signed-off-by: Marton Balint <cus@passwd.hu>
CC libavcodec/libx265.o
libavcodec/libx265.c: In function ‘libx265_encode_frame’:
libavcodec/libx265.c:781:5: warning: this ‘else’ clause does not guard... [-Wmisleading-indentation]
else
^~~~
libavcodec/libx265.c:782:1: note: ...this statement, but the latter is misleadingly indented as if it were guarded by the ‘else’
FF_DISABLE_DEPRECATION_WARNINGS
^~~
Signed-off-by: Marton Balint <cus@passwd.hu>
Their usefulness is questionable, very few decoders set them, and their type
should have been int64_t. A replacement field can be added later if a valid use
case is found.
Signed-off-by: Marton Balint <cus@passwd.hu>
Marton Balint