Before, header information for ogg format files was sent with the
first encoded packet.
This patch makes it so that it is possible for API users to
differentiate between headers and encoded audio. This is useful, for
example, when creating an audio stream where you want to send one set
of headers for every client that connects and then the encoded stream
of audio.
Signed-off-by: Martin Storsjö <martin@martin.st>
Use it instead of checking CODEC_FLAG_BITEXACT in the first stream's
codec context.
Using codec options inside lavf is fragile and can easily break when the
muxing codec context is not the encoding context.
Support the URL scheme where the playpath is in an RTMP URL is
passed as the slist argument and the app is given infront of the
query part of the URL:
rtmp://host[:port]/[app]?slist=[playpath]
(other arguments in the query part are stripped as they are not used)
Signed-off-by: Martin Storsjö <martin@martin.st>
Also, stop using AVCodecContext.codec_name as fallback, since it will be
deprecated.
Changes the result of the lavf-asf test (and its associated seektest),
since 'msmpeg4v3' gets written instead of just 'msmpeg4'.
This adds a function to export raw replaygain values (i.e. in the (u)int32_t
form). It first checks whether AV_PKT_DATA_REPLAYGAIN side data is present, in
which case it does nothing.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
In all other cases where ff_rtmp_packet_read is used, the packet returned
is passed to rtmp_parse_result more or less immediately. In this single
case, the content of the packet was required to be a connect packet.
Some clients, e.g. Open Broadcaster Software, send a chunk size packet
before the connect packet. If the first packet is a chunk size packet,
handle it and read another one, requiring this to be a connect packet
instead.
Signed-off-by: Martin Storsjö <martin@martin.st>
Previously, if read_connect failed, the ret variable was unmodified
and had the value 0, indicating success, which then was returned from
the rtmp_open function, even though it actually failed.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Instead of using a fixed bitrate_idx, calculate a matching bitrate for
the XING header.
Using a fixed bitrate_idx causes tools such as file(1) and mediainfo(1)
to report wrong bitrate and bitrate mode when using CBR.
Bug-Id: https://bugs.debian.org/736088
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
According to the ReplayGain spec, the peak amplitude may overflow and may result
in peak amplitude values greater than 1.0 with psychoacoustically coded audio,
such as MP3. Fully compliant decoders must allow peak overflows.
Additionally, having peak values in the 0<->UINT32_MAX scale makes it more
difficult for applications to actually use the peak values (e.g. when
implementing clipping prevention) since values have to be rescaled down.
This patch corrects the peak parsing by removing the rescaling of the decoded
values between 0 and UINT32_MAX and the 1.0 upper limit.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The gain sign was incorrectly decoded: since the FFSIGN() macro treats 0 as
negative, gain values starting with "0." were always decoded as negative.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Further performance improvements and security fixes by
Vittorio Giovara, Luca Barbato and Diego Biurrun.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Additional fixes by Nigel Touati-Evans <nigel.touatievans@gmail.com>.
Check the index for streams with a time drift of 2s or a buffer drift
of 64MB.
Bug-Id: 666
CC: libav-stable@libav.org
Sample-Id: yet-another-broken-interleaved-avi.avi
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Additional fixes and enhancements by Vittorio Giovara, Gonzalo Garramuno,
Nicolas George, Paul B Mahol and Michael Niedermayer.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Some files set the PreviousPartition field to point to its own offset.
If we are parsing forward the Previous partition is immediately known
and its value could be used, otherwise we can safely point to the
header.
Reported-By: Jean Baptiste Kempf <jb@videolan.org>
For live audio streams, requiring 500 frames for a stream to
be detected is a bit overkill.
This allows live ADTS streams that don't start nicely at
a frame boundary to start up more quickly, e.g.
http://mp3.streampower.be/radio1.aac.
Signed-off-by: Martin Storsjö <martin@martin.st>
If a portion of the probe buffer seem to resemble ADTS frames,
but some data at the end is a mismatch, disregard the whole
probing attempt. If it actually is ADTS data, there shouldn't be
any mismatches within the sequential frame data.
Signed-off-by: Martin Storsjö <martin@martin.st>
mp4 files embedding DVD subtitles do not use the same extradata format
as the rest of Libav expects. The subtitle decoder in libavcodec in
particular does not understand this format.
Convert the extradata to the vobsub .idx format. mp4 stores the palette
as binary 32 bit ints in YUV. The subtitle resolution is stored
separately in the track header, which we access through AVStream.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The icy_metadata_headers string never gets initialized, so,
during the first call to av_strlcatf() in parse_icy(),
strlen() will be called on a pointer to uninitialized memory.
At best this causes some garbage data to be left at the
start of the string.
By initializing icy_metadata_headers to the empty string, the
first call to strlen() will always return 0, so that data is
appended from the start of the string.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Export the metadata as a icy_metadata_packet avoption.
Based on the work of wm4 and Alessandro Ghedini.
Bug-Id: https://bugs.debian.org/739936
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The MSVCRT version of strftime calls the invalid parameter handler
if the struct values in struct tm are invalid. In case no invalid
parameter handler is set for the process, the process is aborted.
This fixes fate failures on MSVC builds since 570af382.
Based on a patch by Hendrik Leppkes.
Signed-off-by: Martin Storsjö <martin@martin.st>
'hvc1' requires that parameter set NAL units be
present only in the samples entry, but not in the
samples themselves, requiring that additional
parameter sets, if present, be filtered out of the
samples and placed in new, additional sample entries
if they override or otherwise conflict with the
parameter sets present in the first sample entry.
We do not have any way of doing this at present, so
the files we produce can only comply with the
restrictions set for the 'hev1' sample entry name in
ISO/IEC 14496-15.
The correct point that seperates ISO and MAC language codes is 0x400
according to the current QT spec. Old QT specs did not list where this
seperation is but apparently only defined the meaning of the first 137.
It is my understanding that "Unless otherwise stated, all data in a
QuickTime movie is stored in big-endian byte ordering" [1] in MOV files.
I have a couple of thousand files, which technically are invalid because
their sound sample description element 4CC is 'lpcm' but its version is
0 - and "Version 0 supports only uncompressed audio in raw ('raw ') or
twos-complement ('twos') format" [2]
Because isom.c only contains a mapping for 4CC 'lpcm' to
AV_CODEC_ID_PCM_S16LE, these files have their audio decoded as LE when
it is actually BE.
This commit adds AV_CODEC_ID_PCM_S16BE as the first match for 4CC 'lpcm'.
[1]
https://developer.apple.com/library/mac/documentation/quicktime/QTFF/qtff.pdf
page 21
[2]
https://developer.apple.com/library/mac/documentation/quicktime/QTFF/qtff.pdf
page 178
Reviewed-by: Yusuke Nakamura <muken.the.vfrmaniac@gmail.com>
Based on a suggestion by Martin Panter. This is more descriptive,
since it's the actual timestamp field from the RTMP packet,
which might or might not be a delta depending on context (in
some packets it's a delta, in some packets it's an absolute
timestamp, and in some packets it's 0xffffff to indicate that
the actual delta or absolute timestamp is transmitted separately).
Signed-off-by: Martin Storsjö <martin@martin.st>
Related fix in "rtmpdump":
https://repo.or.cz/w/rtmpdump.git/commitdiff/79459a2
Adobe's RTMP specification (21 Dec 2012), section 5.3.1.3 ("Extended
Timestamp"), says "this field is present in Type 3 chunks". Type 3 chunks are
those with the one-byte header size.
This resolves intermittent hangs and segfaults caused by the read function,
and also includes an untested fix for the write function.
The read function was tested with ABC (Australia) News 24 streams, however
they are probably restricted to only Australian internet addresses. Some of
the packets at the start of these streams seem to contain junk timestamp
fields, often requiring the extended field. Test command:
avplay rtmp://cp81899.live.edgefcs.net/live/news24-med@28772
Signed-off-by: Martin Storsjö <martin@martin.st>
Get the last partition offset and use it when footer partition
offset is missing.
Footer partition may not be present and even if present footer
partition offset may not be set in any partition except last one.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Extrapolate audio timestamps based on the number of samples demuxed.
Deal with some MXF nastiness involving fractional number of
samples per EditUnit when seeking (the specs handwave this away).
Further fixes from Tomas Härdin.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
We cannot easily determine if an mpeg TS's packet size is DVHS, FEC
or so on, for that we need to expose the internal raw_packet_size
field.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
