Mostly just re-arranges some code to make it easier to update this
filter and libhdcd together. filter_frame() is much simpler as a
result.
* use the HDCD detection data structure and functions from libhdcd,
moved detection code out of filter_frame()
* moved analyze_mode preparation out of filter_frame() into
hdcd_analyze_prepare(), from libhdcd
* moved some macro definitions to the top so they are all together
Signed-off-by: Burt P <pburt0@gmail.com>
The names are hard to understand and also seem to break build on at least some solaris
versions
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Anton Khirnov:
"[av_frame_copy_props()] potentially contains memory allocation,
so the return value needs to be checked."
Signed-off-by: Burt P <pburt0@gmail.com>
Should fix compilation with non C99 compilers like msvc 2012, where
acosh() is not available.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Restore a check added in 440af105f2
but lost sometime after. avfilter_graph_set_auto_convert() will
have an effect once again.
Signed-off-by: Burt P <pburt0@gmail.com>
Also checking AVFilterLink->type is AVMEDIA_TYPE_AUDIO before
calling av_get_sample_fmt_name() on AVFilterLink->format.
Signed-off-by: Burt P <pburt0@gmail.com>
A new mode, selected by filter option, to aid in analysis of HDCD
encoded audio. In this mode the audio is replaced by a solid tone and
the amplitude is adjusted to signal some specified aspect of the process.
The output file can be loaded in an audio editor alongside the original,
where the user can see where different features or states are present.
Signed-off-by: Burt P <pburt0@gmail.com>
In order to avoid possible clipping at the very
beginning. Especially apparent when "alternative"
boundary mode is used.
Adopted from original commit by committer.
HDCD is only encoded in s16@44100Hz. Scan the chain of AVFilterLinks
for any resampling or format conversion/truncation that might cause
problems for the filter and issue warnings.
Signed-off-by: Burt P <pburt0@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Issue #5727: gain adjustment should only be applied if matching
gain value from a valid packet in both channels. The existing functions process
each channel separately, so it was not possible.
* New versions of hdcd_process(), hdcd_scan(), hdcd_integrate() named
hdcd_*_stereo() that process both channels together.
* target_gain applied will be the last matching target_gain.
* The old single channel functions remain as an option. They can be
used by: -af hdcd=process_stereo=0.
Signed-off-by: Burt P <pburt0@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Used to attempt replication of some results from
http://www.audiomisc.co.uk/HFN/HDCD/Examined.html
May not be generally useful, defaults to off.
Signed-off-by: Burt P <pburt0@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* Moves the filter context member out of state and into HDCDContext
* More useful information when an error is detected
* Gives a location near where the error was detected
Signed-off-by: Burt P <pburt0@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The counter is now -1 if the code detect timer was never set,
and 0 if it was set but never expired.
Signed-off-by: Burt P <pburt0@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The stereowiden filter uses a buffer, s->buffer[], and a pointer
within the buffer, s->write, to implement inter-channel delays.
The loop which applies the delayed samples turns out to be faulty.
109 for (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) {
110 const float left = src[0], right = src[1];
111 float *read = s->write + 2;
112
113 if (read > s->buffer + s->length)
114 read = s->buffer;
115
116 dst[0] = drymix * left - crossfeed * right - feedback * read[1];
117 dst[1] = drymix * right - crossfeed * left - feedback * read[0];
118
119 s->write[0] = left;
120 s->write[1] = right;
121
122 if (s->write == s->buffer + s->length)
123 s->write = s->buffer;
124 else
125 s->write += 2;
126 }
For one, the buffer gets written past its end in lines 119-120, before
the bound check is done in lines 122-123. This can be easily confirmed
by valgrind.
==3544== Invalid read of size 4
==3544== at 0x593B41: filter_frame (af_stereowiden.c:116)
==3544== Address 0xb1b03c4 is 4 bytes after a block of size 7,680 alloc'd
==3544==
==3544== Invalid read of size 4
==3544== at 0x593B66: filter_frame (af_stereowiden.c:117)
==3544== Address 0xb1b03c0 is 0 bytes after a block of size 7,680 alloc'd
==3544==
==3544== Invalid write of size 4
==3544== at 0x593B79: filter_frame (af_stereowiden.c:119)
==3544== Address 0xb1b03c0 is 0 bytes after a block of size 7,680 alloc'd
==3544==
==3544== Invalid write of size 4
==3544== at 0x593B7D: filter_frame (af_stereowiden.c:120)
==3544== Address 0xb1b03c4 is 4 bytes after a block of size 7,680 alloc'd
Also, using two separate pointers, s->write and read = s->write + 2,
does not seem to be well thought out. To apply the delay of s->buffer[],
it is enough to read the delayed samples at the current position within
the buffer, and then to store new samples at the same current position.
Thus the application of delayed samples can probably be best described
with a single pointer s->cur.
I also introduce a minor change to ensure that the size of s->buffer[]
is always a multiple of 2. Since the delay parameter is a float, it is
otherwise possible to trick the code into allocating off-by-one buffer.
Add an AVOption stats_version with a new header for V2 stats, which
specifies the stats log version and lists the fields that will be
present in the log (to ease parsing).
The primary motivation is to facilitate the addition of optional fields
to the log without breaking backwards compatibility, while making the
logs easier to parse.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The Peak Extend feature could be enabled permanently or only
when needed. This is now reported.
Signed-off-by: Burt P <pburt0@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Adds a counter for when the "code detect timer" expired without
finding a valid packet.
Signed-off-by: Burt P <pburt0@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Add some comments describing the fields in hdcd_state_t.
Signed-off-by: Burt P <pburt0@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
HDCD is now only considered detected if a valid packet
is active in both channels simultaneously.
Signed-off-by: Burt P <pburt0@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
There is actually a need for the origin and end point not to be defined.
We can not automatically insert them with the y value of the first and
last point as it will influence the curves in a wrong way.
Fixes#5397
Count and report when a code is signaled but fails to match a known pattern.
For example try Līve - Secret Samadhi.
Signed-off-by: Burt P <pburt0@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Only call hdcd_update_info() when the control code changes
instead of every frame, so the counters are more meaningful.
Signed-off-by: Burt P <pburt0@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This is an - once again - updated patch, that uses avio_write instead
of avio_puts to stream clean text output without null characters. Works
now for me as intended.
Changes metadata filter to accept general urls as file argument without
breaking former behaviour. As a byproduct, it also allows for writing to
file "-" if specified as "file:-".
Example:
ffmpeg -i test.wav -filter_complex "silencedetect=n=-40dB:d=0.1,ametadata=mode=print:file='pipe\:4'" -f null
Signed-off-by: Sami Hult <sami.hult@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The new HDCD filter really does nothing to show that it is working or
that HDCD control information was even detected in the stream. This
patch collects information about the decode, like which features were
used, and reports it to the user at the end.
Also,
* Fixes low-level gain adjustment
* Updates the documentation
Signed-off-by: Burt P <pburt0@gmail.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Even though this is not part of the public API, some external
applications access fields after it, thus breaking after updating from
ffmpeg 3.0 or earlier.
Since it is not public, it can be freely moved to the end to avoid
that problem in the future.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Fixes crash due to unprocessed input being passed through
This fixes the last segfault caused by mixing 3.0 and 3.1 libs and
applications
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Benjamin Steffes <benjaminst123@gmail.com>
(comment by ronald)
prevent the theoretical case where the container type (int)
would be 64 bit on some platforms, which would waste some space
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
this allow a filter to be written like this:
aformat =
sample_fmts = fltp|flt:
sample_rates = 44100|44800
Signed-off-by: Muhammad Faiz <mfcc64@gmail.com>
Split version files into one line per symbol/directive to allow compatibility
with the Solaris linker without preprocessing and eliminate $ from version file
templates to simplify the postprocessing shell command.
When a filter is no longer part of a graph, its pointers should be cleared
so no stale pointers remain.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
If acmax can be 0 then it could result in a division by 0
Fixes CID1351345
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit 'd12b5b2f135aade4099f4b26b0fe678656158c13':
build: Split test programs off into separate files
Some conversions done by: James Almer <jamrial@gmail.com>
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
We can only remove the logo if it is inside the picture. We need at
least one pixel around the logo area for interpolation.
Fixes ticket #5527 (Delogo crash with x=0 and/or y=0).
Signed-off-by: Jean Delvare <jdelvare@suse.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Also check return value of av_frame_copy() in passthrough mode, so that
if a copy fails (as it did here, because bitdepth didn't match), the filter
doesn't return success, which would mean sending an uninitialized framebuffer
further down the filtergraph.
Update print mode output header line format to be more consistent with
other log output of FFmpeg. The printf-modifiers have been inspired by
the showinfo filter.
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This allows e.g. to correlate signalstats metadata to time position
without having to find out the filter chain timebase first.
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit 'a638e9184d63e57e67901f34afe919fd56fd3ac4':
vf_fade: make sure the slice end is always in the frame
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Add number of input and output frames to possible variables.
Add option eval to reevaluate coordinate expressions during
initialization or for every frame.
Add a filter to scan the top lines of video frames for vertical interval
timecode (VITC) information and attach it as metadata keys.
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
It is practical to de-noise only on luma while keeping chroma unchanged.
However, libavfilter/vf_owdenoise.c always do the Wavelet transform/retransform on all 3 channels without check whether chroma_strength is 0.
Thus I make this patch. De-noise on Y only for yuv420 is now 1.5 times faster.
Signed-off-by: Yuuki Galaxy <galaxy001@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Restore alphabetical order in lists, break overly long lines, do some
prettyprinting, add some explanatory section comments, group parts
together that belong together logically.
Currently, if the movie source filter is used and a seek_point is
specified on a file that has a negative start time, ffmpeg will fail.
An easy way to reproduce this is as follows:
$ ffmpeg -vsync passthrough -filter_complex 'color=d=10,setpts=PTS-1/TB' test.mp4
$ ffmpeg -filter_complex 'movie=filename=test.mp4:seek_point=2' -f null -
The problem is caused by checking for int64_t overflow the wrong way.
In general, to check whether a + b overflows, it is not enough to do:
a > INT64_MAX - b
because b might be negative; the correct way is:
b > 0 && > a > INT64_MAX - b
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The idea is to use ffmath.h for internal implementations of math functions.
Currently, it is used for variants of libm functions, but is by no means
limited to such things.
Note that this is not exported; use lavu/mathematics for such purposes.
Reviewed-by: Ronald S. Bultje <rsbultje@gmail.com>
Signed-off-by: Ganesh Ajjanagadde <gajjanag@gmail.com>
Remove achroma filter, as same output can be done with lowpass filter
and multiple components with overlay display.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
