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>
