Previously, we required the minimum number of bytes required for
the full box. Don't strictly require the astronomical body and additional
notes fields, but do require an altitude field (which currently isn't
parsed). This matches the initial length check at the start of the function
(which doesn't know about the variable length place field).
Signed-off-by: Martin Storsjö <martin@martin.st>
This was missed in e1eb0fc960, when ff_interleaved_peek was
changed to include const during the evolution of the patch.
Signed-off-by: Martin Storsjö <martin@martin.st>
As long as caller only writes packets using av_interleaved_write_frame
with no manual flushing, this should allow us to always have accurate
durations at the end of fragments, since there should be at least
one queued packet in each stream (except for the stream where the
current packet is being written, but if the muxer itself does the
cutting of fragments, it also has info about the next packet for that
stream).
Signed-off-by: Martin Storsjö <martin@martin.st>
This allows callers with avio write callbacks to get the bytestream
positions that correspond to keyframes, suitable for live streaming.
In the simplest form, a caller could expect that a header is written
to the bytestream during the avformat_write_header, and the data
output to the avio context during e.g. av_write_frame corresponds
exactly to the current packet passed in.
When combined with av_interleaved_write_frame, and with muxers that
do buffering (most muxers that do some sort of fragmenting or
clustering), the mapping from input data to bytestream positions
is nontrivial.
This allows callers to get directly information about what part
of the bytestream is what, without having to resort to assumptions
about the muxer behaviour.
One keyframe/fragment/block can still be split into multiple (if
they are larger than the aviocontext buffer), which would call
the callback with e.g. AVIO_DATA_MARKER_SYNC_POINT, followed by
AVIO_DATA_MARKER_UNKNOWN for the second time it is called with
the following data.
Signed-off-by: Martin Storsjö <martin@martin.st>
Use it to get stream duration, sample rate, channel count and initial padding
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
Previously a partial log message without newline was printed in case of
loglevel=warning.
Signed-off-by: Tobias Rapp <t.rapp@noa-archive.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '0abb07bad7026a945a31ba4047e6583c8b3fa3da':
movenc: Update a comment to reflect how the code actually behaves
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit '06edef3d5e072ef3c4face9ce946d2d9c36cc477':
Generate the lists of enabled protocols/bsfs from configure.
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit 'd12b5b2f135aade4099f4b26b0fe678656158c13':
build: Split test programs off into separate files
Some conversions done by: James Almer <jamrial@gmail.com>
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
When feeding input RTP packets to the depacketizer via custom IO,
it needs to pick the right stream using the payload type for
RTP packets, and using the SSRC for RTCP packets. If the first
packet is an RTCP packet, we don't (currently) know the SSRC
yet and thus can't pick the right RTP depacketizer to handle it.
By parsing the SSRC attribute in the SDP, we can map initial
RTCP packets to the right stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
It doesn't matter what the actual reason for not returning
an AVPacket was - if we didn't return any packet and we have
the next one queued, parse it immediately. (rtp_parse_queued_packet
always consumes a queued packet if one exists, so there's no risk
for infinite loops.)
Signed-off-by: Martin Storsjö <martin@martin.st>
Leaking this private structure opens up the possibility that it may
be re-used when parsing later packets in the stream. This is
problematic if the later packets are not the same codec type (e.g.
private allocated during Vorbis parsing, but later packets are Opus
and the private is assumed to be the oggopus_private type in
opus_header()).
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Larger values would imply file durations of astronomic proportions and cause
overflows
Fixes integer overflow
Fixes: usan_int64_overflow
Found-by: Thomas Guilbert <tguilbert@google.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit 'e3453fd44480d903338c663238bf280215dd9a07':
matroska: Write the field order information
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Replace av_copy_packet and deprecated av_dup_packet by
creating reference using av_packet_ref.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Jan Sebechlebsky <sebechlebskyjan@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
TeeSlave.bsfs is array of pointers to AVBitStreamFilterContext,
so element size should be really size of a pointer, not size
of TeeSlave structure.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Jan Sebechlebsky <sebechlebskyjan@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
The declarations that this comment referred to were removed
in 2439f2ca8 - there is no unbuffered IO in this header now.
Signed-off-by: Martin Storsjö <martin@martin.st>
We still only support one single layer though, but this allows
receiving streams that have this structure present even for
single layer streams.
Signed-off-by: Martin Storsjö <martin@martin.st>
Adds per slave option 'onfail' to the tee muxer allowing an output to
fail, so other slave outputs can continue.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Jan Sebechlebsky <sebechlebskyjan@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
In open_slave failure can happen before bsfs array is initialized,
close_slave must check that bsfs is not NULL before accessing
tee_slave->bsfs[i] element.
Slave muxer expects write_trailer to be called if it's
write_header suceeded (so resources allocated in write_header
are freed). Therefore if failure happens after successfull
write_header call, we must ensure that write_trailer of
that particular slave is called.
Some cleanups are made by Marton Balint.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Jan Sebechlebsky <sebechlebskyjan@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
* commit '3ee2ec5ec1e39a438f89302d949c93a1b5d365a2':
unix: Use rw_timeout for setting the connect timeout
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Until now, the decoding API was restricted to outputting 0 or 1 frames
per input packet. It also enforces a somewhat rigid dataflow in general.
This new API seeks to relax these restrictions by decoupling input and
output. Instead of doing a single call on each decode step, which may
consume the packet and may produce output, the new API requires the user
to send input first, and then ask for output.
For now, there are no codecs supporting this API. The API can work with
codecs using the old API, and most code added here is to make them
interoperate. The reverse is not possible, although for audio it might.
From Libav commit 05f66706d1.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
This codepath isn't quite as bad as it used to sound, if fragments
are cut automatically at video packets.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '933dec0e29ec4d2cb83474279a6c52d62fdb7310':
file: Add an option for following a file that is being written
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit 'd44f3e4059506a182f59218b1e967d42b01e097c':
avio: Apply avoptions on the URLContext itself as well
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit '65a802401c6cc136576bb2e613c0577cbf622aa8':
build: Add component for the SRTP common code
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit '709c0f79d8032fcf733bfe58e79ca7ff0858c8bc':
nuv: Use the correct context for av_image_check_size
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Also, make every addition except for sidedata part of version 1 instead of the
new version 2.
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Closing single slave operation is pulled out into separate
function close_slave(TeeSlave*).
Both close_slave and close_slaves function are moved before
open_slave function.
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Jan Sebechlebsky <sebechlebskyjan@gmail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
Previously, the bug was that if -1 < start_time < 0, the reported
"start" time would lose the negative-sign.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
* commit '7e01d48cfd168c3dfc663f03a3b6a98e0ecba328':
mov: Check the entries value when parsing dref boxes
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Broke a lot of stuff and didn't fix anything.
This reverts commit 3c461eecd4, reversing
changes made to 884dd175f0.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
Signed-off-by: James Almer <jamrial@gmail.com>
* commit '3e8fd93b6ab219221e17fa2b6243cc72cf2d69dc':
lavf: add a missing bump and APIchanges for the codecpar switch
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit 'fa55addd23c2f168163175aee17adb125c2c0710':
img2: Drop av_ prefix for a static function
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Restore alphabetical order in lists, break overly long lines, do some
prettyprinting, add some explanatory section comments, group parts
together that belong together logically.
It will be used by text subtitle demuxers to construct format instructions
straight into extradata. They all currently a similar function that accepts
an AVCodecContext instead.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
WAV is not a NOHEADER format, and thus should not be changing
stream codec IDs and probing in read_packet.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
The problem is that the argument 'q' is of the type uint8_t.
According to the JPEG standard, if 1 <= q <= 50, the scale factor
'S' should be 5000 / Q. Because the create_default_qtables() reuses
the variable 'q' to store the result of this calculation, for small
values of q < 19, q wil subsequently overflow and give wrong results
in the calculated quantization tables.
Instead, use a new variable 'S' (same name as in RFC2435) with the
proper range to store the result of the division.
Signed-off-by: Martin Storsjö <martin@martin.st>
Original mail and my own followup on ffmpeg-user earlier today:
I have a device sending out a MJPEG/RTP stream on a low quality setting.
Decoding and displaying the video with libavformat results in a washed
out, low contrast, greyish image. Playing the same stream with VLC results
in proper color representation.
Screenshots for comparison:
http://zevv.nl/div/libav/shot-ffplay.jpghttp://zevv.nl/div/libav/shot-vlc.jpg
A pcap capture of a few seconds of video and SDP file for playing the
stream are available at
http://zevv.nl/div/libav/mjpeg.pcaphttp://zevv.nl/div/libav/mjpeg.sdp
I believe the problem might be in the calculation of the quantization
tables in the function create_default_qtables(), the attached patch
solves the issue for me.
The problem is that the argument 'q' is of the type uint8_t. According to the
JPEG standard, if 1 <= q <= 50, the scale factor 'S' should be 5000 / Q.
Because the create_default_qtables() reuses the variable 'q' to store the
result of this calculation, for small values of q < 19, q wil subsequently
overflow and give wrong results in the calculated quantization tables. The
patch below uses a new variable 'S' (same name as in RFC2435) with the proper
range to store the result of the division.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Apply the default value for timeout in code instead of via the
avoption, to allow distinguishing the default value from the user
not setting anything at all.
Signed-off-by: Martin Storsjö <martin@martin.st>
Since all URLContexts have the same AVOptions, such AVOptions
will be applied on the outermost context only and removed from the
dict, while they probably make sense on all contexts.
This makes sure that rw_timeout gets propagated to the innermost
URLContext (to make sure it gets passed to the tcp protocol, when
opening a http connection for instance).
Alternatively, such matching options would be kept in the dict
and only removed after the ffurl_connect call.
Signed-off-by: Martin Storsjö <martin@martin.st>
Using this requires setting the rw_timeout option to make it
terminate, alternatively using the interrupt callback (if used via
the API).
Signed-off-by: Martin Storsjö <martin@martin.st>
If set non-zero, this limits duration of the retry_transfer_wrapper()
loop, thus affecting ffurl_read*(), ffurl_write(). As soon as
one single byte is successfully received/transmitted, the timer
restarts.
This has further changes by Michael Niedermayer and Martin Storsjö.
Signed-off-by: Martin Storsjö <martin@martin.st>
Until now, the decoding API was restricted to outputting 0 or 1 frames
per input packet. It also enforces a somewhat rigid dataflow in general.
This new API seeks to relax these restrictions by decoupling input and
output. Instead of doing a single call on each decode step, which may
consume the packet and may produce output, the new API requires the user
to send input first, and then ask for output.
For now, there are no codecs supporting this API. The API can work with
codecs using the old API, and most code added here is to make them
interoperate. The reverse is not possible, although for audio it might.
Signed-off-by: Anton Khirnov <anton@khirnov.net>
Adding early support for a subset of the proposed colour elements
according to the latest version of spec:
https://mailarchive.ietf.org/arch/search/?email_list=cellar&gbt=1&index=hIKLhMdgTMTEwUTeA4ct38h0tmE
Like matroskadec, I've left out elements for pix_fmt related things
as there still seems to be some discussion around these.
The new elements are exposed under strict experimental mode.
Signed-off-by: Neil Birkbeck <neil.birkbeck@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
