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>
Since 2007, the Xiph.org Foundation recommends that .ogg only be used
for Ogg Vorbis audio files.
Source: http://wiki.xiph.org/index.php/MIME_Types_and_File_Extensions
However we only do it if we have libvorbis available because the
built in vorbis encoder is not as good.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Currently ff_interleave_packet_per_dts() waits until it gets a frame for
each stream before outputting packets in interleaved order.
Sparse streams (i.e. streams with much fewer packets than the other
streams, like subtitles or audio with DTX) tend to add up latency and in
specific cases end up allocating a large amount of memory.
Emit the top packet from the packet_buffer if it has a time delta
larger than a specified threshold.
Original report of the issue and initial proposed solution by
mus.svz@gmail.com.
Bug-id: 31
Signed-off-by: Anton Khirnov <anton@khirnov.net>
