AVFormatContext->priv_data is not always a MpegTSContext, it can be
RTSPState when decoding a RTP stream. So it is necessary to pass
MpegTSContext pointer explicitly.
Within libav, the write_section_data function doesn't actually use
the MpegTSContext at all, so this doesn't change anything at the
moment (no memory was corrupted before), but it reduces the risk of
anybody trying to touch the MpegTSContext via AVFormatContext->priv_data
in the future.
Signed-off-by: Martin Storsjö <martin@martin.st>
Its contents are meaningful only if the stream codec context is the one
actually used for encoding, which is often not the case (and is
discouraged).
Use AVCodecContext.field_order instead.
librtmp can keep pointers to this string internally, and may
use them at shutdown as well.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
This typo has existed since this code was added in c16582579.
Newer versions of clang pointed out that this comparison always
was true (since the result of the negation is either 0 or 1, while
AVDISCARD_ALL has the value 48).
Signed-off-by: Martin Storsjö <martin@martin.st>
default-base-is-moof shall be set to track fragments compatible with DASH
Media Segments. So, this is a fundamental support for ISOBMFF ver. DASH.
This is meaningful only when base-data-offset-present is absent and two or
more track fragments are present in a movie fragment.
Signed-off-by: Martin Storsjö <martin@martin.st>
It makes more sense to print the timebase exactly as it is set. Also,
this avoids a divide by zero when av_dump_format() is called on a format
context before writing the header.
As indicated in the function documentation, the header MUST be
checked prior to calling it because no consistency check is done
there.
CC:libav-stable@libav.org
Previously, AVStream.codec.time_base was used for that purpose, which
was quite confusing for the callers. This change also opens the path for
removing AVStream.codec.
The change in the lavf-mkv test is due to the native timebase (1/1000)
being used instead of the default one (1/90000), so the packets are now
sent to the crc muxer in the same order in which they are demuxed
(previously some of them got reordered because of inexact timestamp
conversion).
It has not been properly maintained for years and there is little hope
of that changing in the future.
It appears simpler to write a new replacement from scratch than
unbreaking it.
On big endian machines, the default value set via the faulty
AVOption ended up as 2^32 times too big.
This fixes the fate-lavf-ogg test which currently is broken on
big endian machines, broken since 3831362. Since that commit,
a final zero-sized packet is written to the ogg muxer in that test,
which caused different flushing behaviour on little and big endian
depending on whether the pref_duration option was handled as it
should or not.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
The '?xyz' form is used by android devices (and according to apple
mailing list archives, also by older iOS devices). The 'loci' field
(defined in 3GPP 26.244) is used by recent iOS devices.
Even though the loci field can contain an altitude, it was plain
0 in my sample. Just export longitude and latitude, in a string
format matching the one used by the '?xyz' metadata field.
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows the caller to write all buffered data to disk, allowing
the caller to know at what byte position in the file a certain
packet starts (any packet written after the flush will be located
after that byte position).
Signed-off-by: Martin Storsjö <martin@martin.st>
In the presence of no metadata, do not set any stream flag in the FLV
header but let the demuxer handle the detection and creation of streams
as data arrives.
Signed-off-by: Martin Storsjö <martin@martin.st>
If no streams were indicated in the FLV header, do not automatically
allocate by default a video and an audio stream. Instead, in the case
that the header did not indicate the presence of any data, allocate no
stream until data actually arrives for one type.
Signed-off-by: Martin Storsjö <martin@martin.st>
The other format (full flac header blocks) should not be exported by any
demuxers anymore.
This allows to drop an avpriv_ function and also simplify the following
commits.
Only copy it manually in the muxers where it makes sense (rtspenc,
sapenc). Don't touch the original AVStream in movenchint, where
the original AVStream should be kept untouched.
This fixes the normal tracks in RTP hinted files after
abb810db - the hint tracks were ok while the normal media tracks
were broken, noticed by Michael Niedermayer.
This reverts abb810db but achieves the same effect for the other
muxers.
Signed-off-by: Martin Storsjö <martin@martin.st>
While it strictly isn't necessary to copy the time base (since
any use of it is scaled in ff_write_chained), it still is better
to signal the actual time base to the caller, avoiding one
unnecessary rescaling. This also lets the caller know what the
actual internal time base is, in case that is useful info
for some caller.
This reverts commit 397ffde115.
Signed-off-by: Martin Storsjö <martin@martin.st>
This results in DefaultDuration not being written when the framerate is
not known, but as this field is purely informative, this should not
break any sane demuxers.
Add the low overhead pipe mode and the extended broadcast mode.
Export the options as 'syncponts' since it impacts only that.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This corrects the bug that caused the checksums to change in
9767d7c092.
It caused the EOS flag to be set incorrectly; the ogg spec does not
allow it to be set in the middle of a logical bitstream.
Signed-off-by: Andrew Kelley <superjoe30@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This avoids all the ABI troubles associated with avpriv_.
Since this function is very small and does not depend on any tables,
making it inline should have no adverse effects.
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>
