This makes sure that audio preroll for e.g. AAC is signaled correctly.
Previously we only wrote the edit list correctly if we had negative
dts but started with pts == 0 (e.g. for video with B-frames).
Signed-off-by: Martin Storsjö <martin@martin.st>
In these cases, only drop dts. Because if we drop both we have no
timestamps at all for some files.
This improves playback of HLS streams from GoPro cameras.
Signed-off-by: Martin Storsjö <martin@martin.st>
Trying to write to a stream id larger the the maximum requested is
a programming error, still there is no reason to leave a
reachable abort() in the codebase.
CC: libav-stable@libav.org
This makes the field consistent with AVInputFormat.mime_type and the
argument type of av_match_name.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
By using ff_avc_write_annexb_extradata instead of the h264_mp4toannexb
BSF, the code for doing the conversion itself is kept much shorter,
there's less state to restore at the end, we don't risk leaving the
AVCodecContext in an inconsistent state if returning early due to
errors, etc.
Also add a missing free if the base64 encoding fails.
Signed-off-by: Martin Storsjö <martin@martin.st>
The -hls_allow_cache parameter enables explicitly setting the
EXT-X-ALLOW-CACHE tag in the manifest file. That tag indicates
whether the client MAY or MUST NOT cache downloaded media
segments for later replay.
Valid values are 1 (=YES) or 0 (=NO) and the EXT-X-ALLOW-CACHE
will not show in the manifest for other values (or if
-hls_allow_cache is not used.
Signed-off-by: Martin Storsjö <martin@martin.st>
When AVFMT_FLAG_NOBUFFER is set, the packets are not added to the
AVFormatContext packet list, so they need to be freed when they are
no longer needed.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The RFC spec draft only specifies the "H265" name - there is no
specification saying how to interpret "HEVC" (if such a packet
format is specified it could be an entirely different format).
Since this is a very new standard (still a draft), there is little
need for compatibility with existing, broken implementations. Therefore
remove the extra alias, to avoid the risk of encouraging incorrect
usage.
Intentionally keeping the ff_hevc_dynamic_handler name for the
handler, to use "hevc" consistently as name for the codec instead
of "h265" within the library internals as long as there only is one
single variant in actual use.
Signed-off-by: Martin Storsjö <martin@martin.st>
In practice this hint is ignored - the rtp muxer always overwrites
the stream time base without taking the hint into account. But as
a general practice this is the correct way to pass a time base hint
on to a chained muxer.
This avoids warnings about using the codec time base as hint
being deprecated.
Signed-off-by: Martin Storsjö <martin@martin.st>
The size variable is (correctly) unsigned, but is passed to several functions
which take signed parameters, such as avio_read, sometimes after having
numbers added to it. So ensure that size remains within the bounds that
these functions can handle.
CC: libav-stable@libav.org
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Previously, the returned error codes were intentionally ignored
(see fadd3a6821), to avoid aborting if the directory already
existed. If the mkdir actually failed, this was caught when
opening files within the directory fails anyway.
By handling the error code here (but explicitly ignoring EEXIST),
the error messages and return codes in these cases are more
appropriate and less confusing.
Signed-off-by: Martin Storsjö <martin@martin.st>
Convert the Matroska stereo format to the Stereo3D format, and add a
Stereo3D side data to the stream.
Bump the doctype version supported.
Bug-Id: 728 / https://bugs.debian.org/757185
If the remote end of a connection oriented socket hangs up, generating
an EPIPE error is preferable over an unhandled SIGPIPE signal.
Signed-off-by: Martin Storsjö <martin@martin.st>
At least one FATE sample contains such chunks and happens to work simply
by accident (due to find_stream_info() swallowing the error).
CC: libav-stable@libav.org
Update mxf_set_audio_pts to use the container-provided information.
The UL is marked as "to be changed in the future", but the current
samples in the wild do use it.
Prevent out of array writes.
Similar to what Michael Niedermayer did to address the same issue.
Bug-Id: CVE-2014-2263
CC: libav-stable@libav.org
Signed-off-by: Diego Biurrun <diego@biurrun.de>
It is basically a wrapper around av_get_audio_frame_duration(), with a
fallback to AVCodecContext.frame_size. However, that field is set only
when the stream codec context is actually used for encoding or decoding,
which is discouraged.
For muxing, it is generally the responsibility of the caller to set the
packet duration.
For demuxing, if the duration is not stored at the container level, it
should be set by the parser.
Therefore, removing the frame_size fallback should not break any
important case.
The cur_*auth_type variables were set before the http_connect call
prior to 6a463e7fb - their sole purpose is to record the
authentication type used to do the latest request, since parsing
the http response sets the new type in the auth state.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
Originally, AVFormatContext and a metadata dict were provided to ff_vorbis_comment(),
but this presented issues if an AVStream was being updated or the metadata on
AVFormatContext wasn't actually being updated. To remedy this, ff_vorbis_stream_comment()
explicitly updates a stream's metadata and sets any necessary flags.
ff_vorbis_comment() does not modify any flags, and any calls to it that update
AVFormatContext's metadata (just a single call) must also update
AVFormatContext.event_flags after detecting any metadata changes to the provided
dictionary, as signaled by a positive return value.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Currently, only onMetaData is used, but some providers (wrongly)
put metadata into onCuePoint events, and it's still nice to be
able to use that data.
onCuePoint events also present metadata slightly differently than
onMetaData events: all metadata is found inside an object called
"parameters". In order to extract this metadata, it's easiest to
recurse through the object tree and pull out anything found in
child objects and put it in the top-level metadata.
Reference: http://help.adobe.com/en_US/FlashPlatform/reference/actionscript/2/help.html?content=00001404.html
Signed-off-by: Anton Khirnov <anton@khirnov.net>
If any option named "metadata" is set inside the context, it is pulled up to
the context and then the option is cleared.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
The only flags, for now, indicate if metadata was updated and are set after each call to
av_read_frame(). This comes with the caveat that, on stream start, it might not be set properly
as packets might be buffered in AVFormatContext.packet_buffer before being given to the user
in av_read_frame().
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Previously this logic was only used if the server didn't
respond with Connection: close, but use it even for that case,
if the server response is non-chunked.
Originally the http code has relied on Connection: close to close
the socket when the file/stream is received - the http protocol
code just kept reading from the socket until the socket was closed.
In f240ed18 we added a check for the file size, because some
http servers didn't respond with Connection: close (and wouldn't
close the socket) even though we requested it, which meant that the
http protocol blocked for a long time at the end of files, waiting
for a socket level timeout.
When reading over tls, trying to read at the end of the connection,
when the peer has closed the connection, can produce spurious (but
harmless) warnings. Therefore always voluntarily stop reading when
the specified file size has been received, if not using a chunked
transfer encoding. (For chunked transfers, we already return 0
as soon as we get the chunk header indicating end of stream.)
Signed-off-by: Martin Storsjö <martin@martin.st>
Split return value handling from the actual opening.
Incidentally fixes the https -> http redirect issue reported by
Compn on behalf of rcombs.
CC: libav-stable@libav.org
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>
