10- and 12-bit DNxHR use the same DC coefficient decoding process and
VLC table, just with a different shift value. From SMPTE 2019-1:2016,
8.2.4 DC Coefficient Decoding:
"For 8-bit video sampling, the maximum value of η=11 and for
10-/12-bit video sampling, the maximum value of η=13."
A sample file will be uploaded to show that with this patch, things
decode correctly:
dnxhr_hqx_12bit_1080p_smpte_colorbars_davinci_resolve.mov
Signed-off-by: Steven Robertson <steven@strobe.cc> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
build: Make API documentation depend on config.mak
The Doxygen command is generated from the list of installed headers,
which may change per configuration (e.g. `--enable-gpl` results in
libpostproc to be built and installed).
Burt P [Fri, 29 Jul 2016 17:36:22 +0000 (12:36 -0500)]
af_hdcd: Warn if there is any resampling or format conversion in the link chain
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>
Burt P [Mon, 1 Aug 2016 19:28:39 +0000 (14:28 -0500)]
af_hdcd: Process stereo channels together, fix #5727
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>
Burt P [Fri, 29 Jul 2016 17:36:18 +0000 (12:36 -0500)]
af_hdcd: Improve error detection logging
* 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>
avfilter/af_stereowiden: fix read/write past the end of buffer
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.
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.
Lucas Cooper [Mon, 25 Jul 2016 18:54:37 +0000 (11:54 -0700)]
avfilter: Add new format for PSNR stats log
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.
lavf/mpegtsenc: add special case for handling timed ID3 packets
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).
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.
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>
avformat/hls: Fix missing streams in some cases with MPEG TS
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.
avformat/hls: Fix regression with ranged media segments
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.
avformat/utils: Fix find_stream_info not considering the extradata it found
Commit 9200514ad8717c6 ("lavf: replace AVStream.codec with
AVStream.codecpar") merged in commit 6f69f7a8bf6a0d01 changed
avformat_find_stream_info() to put the extradata it got from
st->parser->parser->split() to st->internal->avctx instead of st->codec
(extradata in st->internal->avctx will be later copied to st->codecpar).
However, in the same function, the "is stream ready?" check was changed
to check for extradata in st->codecpar instead of st->codec, even
though st->codecpar is not yet updated at that point.
Extradata retrieved from split() is therefore not considered anymore,
and avformat_find_stream_info() will therefore needlessly continue
probing in some cases.
Fix that by checking for the extradata at st->internal->avctx where it
is actually put.
* commit '54dd9b1cdd9e54f1ee39ae25af0324f8aba2831b':
h264: set mb_aff_frame in frame_start()
h264: move the block starting a new field out of slice_header_parse()
Both commits are merged at the same time in order to prevent a
regression with Ticket #4440 (see 38660128).
lavc/ffjni: replace ff_jni_{attach,detach} with ff_jni_get_env
If a JNI environment is not already attached to the thread where the
MediaCodec calls are made the current implementation will attach /
detach an environment for each MediaCodec call wasting some CPU time.
ff_jni_get_env replaces ff_jni_{attach,detach} by permanently attaching
an environment (if it is not already the case) to the current thread.
The environment will be automatically detached at the thread destruction
using a pthread_key callback.
Saves around 5% of CPU time (out of 20%) while decoding a stream with
MediaCodec.
projects / ffmpeg / log