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>
dv_frame_offset() is static and called only from dv_read_seek(), where
c->sys->frame_size is already used.
This simplifies the incoming codecpar merge where
avctx->{coded_width,coded_height,time_base} are not accessible anymore.
Needed for noStreams.wtv unless something else forces continued parsing (like looking for more than 1
frame in attachments)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This reverts commit 9f9ed79d4c.
The hlsopts member was never set anywhere and always NULL, furthermore
the HLS demuxer needs to retrieve the proper options from the underlying
http protocol (cookies, user-agent, etc), so a dummy context won't help.
Instead, use the AVIOContext directly to access the options.
For example you can split a file, keeping a continuous timecode between
each segment:
ffmpeg -i src.mov -timecode 10:00:00:00 -vcodec copy -f segment \
-segment_time 2 -reset_timestamps 1 -increment_tc 1 target_%03d.mov
Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
Check if the size is written the first 4 bytes and read the next 4
as fourcc candidate, fallback checking the initial for 4 bytes.
"The CodecPrivate contains all additional data that is stored in the
'stsd' (sample description) atom in the QuickTime file after the
mandatory video descriptor structure (starting with the size and FourCC
fields)"
CC: libav-stable@libav.org
Fill DTS if all packets have been read in avformat_find_stream_info, and still
has_decode_delay_been_guessed returns false.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Should fix xvid/cache on windows with --enable-shared
May be related to Ticket 4780
Tested-by: Hendrik Leppkes <h.leppkes@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This is safer, as a selected demuxer could still mean that it was auto-detected
by a user application
Reviewed-previously-by: Nicolas George <george@nsup.org>
Reviewed-previously-by: Andreas Cadhalpun <andreas.cadhalpun@googlemail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
"Skipping 0 bytes of junk" is useless to the user, and essentially
indicates a NOP. At 0 bytes, this message is now pushed back to
the verbose log level.
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
Example found in the wild:
0:00:03:25.000
0:01:47:A legend is sung
0:01:50:Of when England was young
0:01:53:And knights|were brave and bold
0:01:59:The good king had died
Reported-by: wm4
We cannot play multiple multicast streams with the same port at the
same time. This is because both rtp and rtcp port are opened in
read-write mode, so they will not bind to the multicast address. Try
to make rtp port as read-only by default to solve this bug.
Signed-off-by: Zhao Zhili <wantlamy@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This fixes a problem where ffmpeg would hang if there is already an open
data connection, and the server sends a 125 response code in reply to a
STOR or RETR command.
Signed-off-by: Raymond Hilseth <rhi@vizrt.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
RTCP synchronization packet was broken since commit in ffmpeg version > 2.8.3
(commit: e04b039b15) Since this commit (2e814d0329)
"rtpenc: Simplify code by introducing a macro for rescaling NTP timestamps", NTP_TO_RTP_FORMAT
uses av_rescale_rnd() function to add the data to the packet.
This causes an overflow in the av_rescale_rnd() function and it will return INT64_MIN.
Causing the NTP stamp in the RTCP packet to have an invalid value.
Github: Closes#182
Reverting commit '2e814d0329aded98c811d0502839618f08642685' solves the problem.
Use the context and level specified to av_pkt_dump_log2(),
instead of panic level (0), for dumping packet payload.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Samples produced by Omneon (Harmonic) store external references with
paths ending with 0s. Such movs cannot be loaded properly since every
0 is converted to '/', to keep the same parsing code for dref type 2
and type 18: this makes the external reference point to a non-existing
direactory, rather than to the actual referenced file.
Add a brief trimming loop that drops all ending 0s before trying to
parse the external reference path.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
Store the file duration in the same timebase it arrives (i.e.
milliseconds) and only convert it to the file duration units (100ns)
when it's actually written, thus simplifying some calculations. Also,
store the duration as unsigned, since it cannot be negative.
CC: libav-stable@libav.org
Bug-ID: CVE-2016-2326
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
I've left out elements for pix_fmt related things as there still
seems to be some discussion around these, and the max_cll/max_fall
are currently not propagated as there is not yet side data for them.
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>
This can be used for formats which write all format metadata as string to
files, therefore non-standard creation times such as 'now' will be parsed.
The standardized creation time is UTC ISO 8601 with microsecond precision.
Reviewed-by: wm4 <nfxjfg@googlemail.com>
Signed-off-by: Marton Balint <cus@passwd.hu>
* commit '832a202c47a246ed15e3edc6b05dfcfa7d82c4b2':
protocols: make the list of protocols static
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This commit also disables the async fate test, because it
used internal APIs in a non-kosher way, which no longer
exists.
* commit '2758cdedfb7ac61f8b5e4861f99218b6fd43491d':
lavf: reorganize URLProtocols
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit 'e192cd9ce2b51c2e6919f2a78b1ce53e0024e728':
smoothstreamingenc: do not open the files as read+write
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit '7fbb3b5b9857276b4cd17b2a530c7e0880d2bc0a':
lavf: use the io_open callbacks for files opened from open_input() as well
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
FATE tests have been updated to patch. They do not differ in
any meaningful way.
* commit 'dc6527ed908e4d330738f139074455ffbe56a2de':
nutenc: do not use AVCodecContext.frame_size
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This also fixes reading gapless metadata when the entries do not start with the
mean atom. Such samples can be found here:
https://hydrogenaud.io/index.php/topic,93310.0.html
Reviewed-by: Michael Niedermayer <michael@niedermayer.cc>
Signed-off-by: Marton Balint <cus@passwd.hu>
https://developer.apple.com/library/mac/technotes/tn2174/_index.html
- Enabled creation of timecode tracks for MP4 in the same way as MOV.
- Used nmhd as media information header of timecode track of MP4 instead
of gmhd used in MOV, thus avoiding tcmi also, as recommended above.
- Bypassed adding source reference field for MP4, as suggested above.
Issue: https://trac.ffmpeg.org/ticket/4704
Signed-off-by: Syed Andaleeb Roomy <andaleebcse@gmail.com>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This broke packed_maindata.mp3.mp4
Its unknown to me what this commit would have fixed
Reviewed-by: James Almer <jamrial@gmail.com>
This reverts commit 79127dbbef, reversing
changes made to 9fad1ce7c9.
* commit '0d1229f1d2b8f26dd50c6be7917bb8ed8cb95364':
voc: Split ff_voc_get_packet into a separate file
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* commit 'b92962436bdcfae478c8598dca397a397762eef8':
mov: Fix the format specifier type for size
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This allows to copy information related to the stream ID from the demuxer
to the muxer, thus allowing for example to retain information related to
synchronous and asynchronous KLV data packets. This information is used
in the muxer when remuxing to distinguish the two kind of packets (if the
information is lacking, data packets are considered synchronous).
The fate reference changes are due to the use of
av_packet_merge_side_data(), which increases the size of the output
packet size, since side data is merged into the packet data.
Currently, AVStream contains an embedded AVCodecContext instance, which
is used by demuxers to export stream parameters to the caller and by
muxers to receive stream parameters from the caller. It is also used
internally as the codec context that is passed to parsers.
In addition, it is also widely used by the callers as the decoding (when
demuxer) or encoding (when muxing) context, though this has been
officially discouraged since Libav 11.
There are multiple important problems with this approach:
- the fields in AVCodecContext are in general one of
* stream parameters
* codec options
* codec state
However, it's not clear which ones are which. It is consequently
unclear which fields are a demuxer allowed to set or a muxer allowed to
read. This leads to erratic behaviour depending on whether decoding or
encoding is being performed or not (and whether it uses the AVStream
embedded codec context).
- various synchronization issues arising from the fact that the same
context is used by several different APIs (muxers/demuxers,
parsers, bitstream filters and encoders/decoders) simultaneously, with
there being no clear rules for who can modify what and the different
processes being typically delayed with respect to each other.
- avformat_find_stream_info() making it necessary to support opening
and closing a single codec context multiple times, thus
complicating the semantics of freeing various allocated objects in the
codec context.
Those problems are resolved by replacing the AVStream embedded codec
context with a newly added AVCodecParameters instance, which stores only
the stream parameters exported by the demuxers or read by the muxers.
Instead of a linked list constructed at av_register_all(), store them
in a constant array of pointers.
Since no registration is necessary now, this removes some global state
from lavf. This will also allow the urlprotocol layer caller to limit
the available protocols in a simple and flexible way in the following
commits.
Improves streaming compatibility with Windows Media Services. Also tested for
compatilbility in Windows Media Player, Windows Media ASF Viewer and VLC.
This version of the patch only writes exclusion among audio streams, therefore
choosing a subtitle language should be possible independently of audio language.
Signed-off-by: Marton Balint <cus@passwd.hu>
I discovered that ffserver streaming was broken (it seems like it has been since 20th November) and I opened a ticket for this (https://trac.ffmpeg.org/ticket/5250 <https://trac.ffmpeg.org/ticket/5250>).
I spent yesterday learning git bisect (with the kind help of cehoyos) to painstakingly track down the cause. This was made more difficult due to the presence of a segfault in ffserver during the period where the bug was introduced so I first had to identify when and how that was fixed and then retrospectively apply that fix again for each step of the second git bisect to find the actual bug.
Anyway, the fruits of my labour are the innocent looking patch below to correct a couple of typos and define a valid range for two variables.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Fix cases where unknown data (data beyond p->buf_size) could produce a
higher ico probe score than if the unknown data was known and valid.
For example:
Header: OK, 2 frames
Frame 0: Unknown (offset points beyond end of probe buffer)
Frame 1: Invalid
Previously this example had a score of 25, even though the score would
be 1 if the unknown frame was known to be valid or 0 if it was known
to be invalid. For this example the score is now 1.
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Some muxer might or might not fit incomplete mp3 frames in
their packets.
Bug-Id: 899
CC: libav-stable@libav.org
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
This can be made more efficient, but first and the main goal of this change is to
store it at all
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
It is only used in a boolean context. Also clarify its documentation.
Signed-off-by: Moritz Barsnick <barsnick@gmx.net>
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
Use with -use_localtime, and set -hls_segment_filename to a path which
contains a subdirectory i.e. /some/path/%Y%m%d/%Y%m%dT%H%M%S-%s.ts
This will mkdir the %Y%m%d-part of the path if it does not already
exist.
In addition, each filename in the playlist output will be prefixed with
this subdirectory (if playlist and segment shares the same base path).
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
This commit is a no-op.
* commit '5eb562831b3a9bea8026c413ef1338e06450d005':
mov: Use the correct type for size
Merged-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
This avoid "libavformat/genh.c:43:14: warning: variable coef_splitted set but not used"
Fewer warnings makes it easier to see new and important warnings
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
