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>
* commit '70b1dcef2d859ae6b3e21d61de928c3dd0cf1aa4':
h264: tighten the valid range for ref_frame_count
Conflicts:
libavcodec/h264_ps.c
Merged-by: James Almer <jamrial@gmail.com>
* commit 'f638b67e5790735f34620bf82025c9b9d6fc7216':
h264: move the parameter set definitions to a new header file
Conflicts:
libavcodec/h264_parse.h
libavcodec/h264_ps.c
libavcodec/h264dec.h
Merged-by: James Almer <jamrial@gmail.com>
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>
Most systems have this, so it isn't really a problem to include it
even if it's not used, but some do not have memory.h as it is
non-standard. Since it's unused just remove it anyway.
* commit '17e7c03e12d1e4490921e7bffaeaa6b46a7ada4e':
h264: only allow ending a field/starting a new one before finish_setup()
This commit is a noop. According to Michael, after 8385e171 this commit
should not be necessary anymore.
Merged-by: Clément Bœsch <u@pkh.me>
Set the stream_id to 0xbd (private_stream_id_1). Tools seem to assume
that value, and this is consistent with MPEG TS specification (ITU-T
H.222.0 section 2.12.3).
The original code assumes av_realloc() will free ptr if size is zero.
The assumes is incorrect now.
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>
Rev #2: Fixes doubled header writing, checked FATE running without errors
Rev #3: Fixed coding style
This commit addresses the following scenario:
we are using ffmpeg to transcode or remux mkv (or something else) to mkv. The result is being streamed on-the-fly to an HTML5 client (streaming starts while ffmpeg is still running). The problem here is that the client is unable to detect the duration because the duration is only written to the mkv at the end of the transcoding/remoxing process. In matroskaenc.c, the duration is only written during mkv_write_trailer but not during mkv_write_header.
The approach:
FFMPEG is currently putting quite some effort to estimate the durations of source streams, but in many cases the source stream durations are still left at 0 and these durations are nowhere mapped to or used for output streams. As much as I would have liked to deduct or estimate output durations based on input stream durations - I realized that this is a hard task (as Nicolas already mentioned in a previous conversation). It would involve changes to the duration calculation/estimation/deduction for input streams and propagating these durations to output streams or the output context in a correct way.
So I looked for a simple and small solution with better chances to get accepted. In webmdashenc.c I found that a duration is written during write_header and this duration is taken from the streams' metadata, so I decided for a similar approach.
And here's what it does:
At first it is checking the duration of the AVFormatContext. In typical cases this value is not set, but: It is set in cases where the user has specified a recording_time or an end_time via the -t or -to parameters.
Then it is looking for a DURATION metadata field in the metadata of the output context (AVFormatContext::metadata). This would only exist in case the user has explicitly specified a metadata DURATION value from the command line.
Then it is iterating all streams looking for a "DURATION" metadata (this works unless the option "-map_metadata -1" has been specified) and determines the maximum value.
The precendence is as follows: 1. Use duration of AVFormatContext - 2. Use explicitly specified metadata duration value - 3. Use maximum (mapped) metadata duration over all streams.
To test this:
1. With explicit recording time:
ffmpeg -i file:"src.mkv" -loglevel debug -t 01:38:36.000 -y "dest.mkv"
2. Take duration from metadata specified via command line parameters:
ffmpeg -i file:"src.mkv" -loglevel debug -map_metadata -1 -metadata Duration="01:14:33.00" -y "dest.mkv"
3. Take duration from mapped input metadata:
ffmpeg -i file:"src.mkv" -loglevel debug -y "dest.mkv"
Regression risk:
Very low IMO because it only affects the header while ffmpeg is still running. When ffmpeg completes the process, the duration is rewritten to the header with the usual value (same like without this commit).
Signed-off-by: SoftWorkz <softworkz@hotmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
HLS demuxer calls the subdemuxer avformat_find_stream_info() while
overriding the subdemuxer AVFMTCTX_NOHEADER flag by clearing it.
However, this prevents some streams in some MPEG TS streams from being
detected properly.
Simply removing the clearing of the flag would cause the inner
avformat_find_stream_info() call to take longer in some cases, without
a way to control it.
To fix the issue, do not clear the flag but propagate it to HLS demuxer.
To avoid the above-mentioned mandatory delay, the call to
avformat_find_stream_info() is dropped except in the HLS ID3 timestamped
case. The HLS demuxer user should be calling avformat_find_stream_info()
on the HLS demuxer if it wants to find the stream info.
The main streams are now created dynamically after read_header time if
the subdemuxer uses AVFMTCTX_NOHEADER (mpegts).
Subdemuxer avformat_find_stream_info() is still called for the HLS ID3
timestamped case as the HLS demuxer needs to know the packet durations
to properly interleave ID3 timestamped streams with MPEG TS streams on
sub-segment level.
Fixes ticket #4930.
Creation of main demuxer streams from subdemuxer streams is moved to
update_streams_from_subdemuxer() which can be called repeatedly.
There should be no functional changes.
Commit 81306fd4bdf ("hls: eliminate ffurl_* usage", merged in d0fc5de3a6)
changed the hls demuxer to use AVIOContext instead of URLContext for its
HTTP requests.
HLS demuxer uses the "offset" option of the http demuxer, requesting
the initial file offset for the I/O (http URLProtocol uses the "Range:"
HTTP header to try to accommodate that).
However, the code in libavformat/aviobuf.c seems to be doing its own
accounting for the current file offset (AVIOContext.pos), with the
assumption that the initial offset is always zero.
HLS demuxer does an explicit seek after open_url to account for cases
where the "offset" was not effective (due to the URL being a local file
or the HTTP server not obeying it), which should be a no-op in case the
file offset is already at that position.
However, since aviobuf.c code thinks the starting offset is 0, this
doesn't work properly.
This breaks retrieval of ranged media segments.
To fix the regression, just drop the seek call from the HLS demuxer when
the HTTP(S) protocol is used.
