* commit '1fb8f6a44f06e48386450fe0363aefc02583d24a':
x86: lavr: add SSE2 quantize() for dithering
doc/APIchanges: fill in missing dates and hashes.
rtpdec_vp8: Request a keyframe if RTP packets are lost
Conflicts:
doc/APIchanges
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '42805eda554a7fc44341282771531e7837ac72b7':
rtpdec: Store the dynamic payload handler in the rtpdec context
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '9c80ed836a511293f4cc3a858060969d32f2b1ce':
rtpdec_vp8: Avoid a warning about a possibly unused variable
rtpdec_vp8: Make sure the previous packet is returned
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '92e354b655613b88c3c202a7e19e7037daed37eb':
rtpdec_vp8: Set the timestamp when returning a deferred packet
hlsenc: Make the start_number option set the right variable
Conflicts:
libavformat/hlsenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This allows the caller to either include them (and get more packets
decoded, but possibly some nonperfect frames), or discard them (by
setting fflags=discardcorrupt).
Signed-off-by: Martin Storsjö <martin@martin.st>
Sometimes the muxer modifies the packet, like for instance lavf/mp3enc
changing pkt->destruct in order to keep a copy. These changes must be
kept, even though the muxer behaviour is questionable. Regression since
0072116.
Fixes#2124.
This uses page duration instead of byte size to determine when to buffer
the page. Also, it tries to avoid continued pages by buffering the current
page if there are already packets in the page and adding the next packet
would require it to be continued on a new page. This can improve seeking
performance.
The default page duration is 1 second, which is much saner than filling
all page segments by default.
This sends NACK for missed packets and PLI (picture loss indication)
if a depacketizer indicates that it needs a new keyframe, according
to RFC 4585.
This is only enabled if the SDP indicated that feedback is supported
(via the AVPF or SAVPF profile names).
The feedback packets are throttled to a certain maximum interval
(currently 250 ms) to make sure the feedback packets don't eat up
too much bandwidth (which might be counterproductive). The RFC
specifies a more elaborate feedback packet scheduling.
The feedback packets are currently sent independently from normal
RTCP RR packets, which is not totally spec compliant, but works
fine in the environments I've tested it in. (RFC 5506 allows this,
but requires a SDP attribute for enabling it.)
Signed-off-by: Martin Storsjö <martin@martin.st>
The warning is a false positive, but I prefer actually initializing
it over masking it with av_uninit, since the code is not performance
critical.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is a bug from c7d4de3d73 - if the previous frame wasn't
returned yet (due to missing the final packets), but we have
enough data of it to return the first partition, we write that into
pkt and set returned_old_frame. That commit forgot returning 0 for
the case where this current packet didn't have the end_packet flag
set.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '8729698d50739524665090e083d1bfdf28235724':
rtsp: Recheck the reordering queue if getting a new packet
lavr: log channel conversion description for any-to-any functions
lavr: mix: reduce the mixing matrix when possible
lavr: cosmetics: reindent
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '9a00374cb4512a58a1fee366b850dfa87c76e1f3':
doc: Fix a few typos in the developer documentation
xwma: Remove unused variable
asfdec: Fix printf format string length modifier
Conflicts:
doc/developer.texi
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If we timed out and consumed a packet from the reordering queue,
but didn't return a packet to the caller, recheck the queue status.
Otherwise, we could end up in an infinite loop, trying to consume
a queued packet that has already been consumed.
CC: libav-stable@libav.org
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '89b51b570daa80e6e3790fcd449fe61fc5574e07':
oggdec: free the ogg streams on read_header failure
Conflicts:
libavformat/oggdec.c
Original commit this was based on: (this merge just moves the function up)
commit 07a866282f
Author: Michael Niedermayer <michaelni@gmx.at>
Date: Tue Nov 20 15:12:37 2012 +0100
oggdec: fix memleak on header parsing failure
Fixes Ticket1931
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The following out-of-memory check is broken.
*sorted_segments = av_mallocz(...);
if (!sorted_segments) { ... }
The correct NULL check should use *sorted_segments.
Signed-off-by: Xi Wang <xi.wang@gmail.com>
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
* qatar/master:
lavr: fix missing " in header documentation
aviobuf: Discard old buffered, previously read data in ffio_read_partial
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'e96406eda4f143f101bd44372f7b2d542183000a':
rtsp: Add support for depacketizing RTP data via custom IO
Conflicts:
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '3f95f0dda55fca74b646937095a02a8fa9776622':
rtpdec: Move the URLContext used for RTCP RR out from the context, to a parameter
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Followup to http://thread.gmane.org/gmane.comp.video.ffmpeg.devel/151321
patch by Reimar and Thomas Mundt fixes some AVC-Intra files from
different tickets.
It does not fix http://samples.ffmpeg.org/ffmpeg-
bugs/trac/ticket524/AVCI50.mov
Authors of this commit are: Reimar and Thomas Mundt
Patch and commit message mostly taken from ffmpeg-devel, mail by Carl
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
To use this, set sdpflags=custom_io to the sdp demuxer. During
the avformat_open_input call, the SDP is read from the AVFormatContext
AVIOContext (ctx->pb) - after the avformat_open_input call,
during the av_read_frame() calls, the same ctx->pb is used for reading
packets (and sending back RTCP RR packets).
Normally, one would use this with a read-only AVIOContext for the
SDP during the avformat_open_input call, then close that one and
replace it with a read-write one for the packets after the
avformat_open_input call has returned.
This allows using the RTP depacketizers as "pure" demuxers, without
having them tied to the libavformat network IO.
Signed-off-by: Martin Storsjö <martin@martin.st>
So far, aviocontexts are used either in pure-read or pure-write
mode - full read/write mode doesn't work well (and implementing it
is a much larger, not totally trivial change).
This patch allows using avio_read and ffio_read_partial on
read/write aviocontexts, where the read operations are passed
through directly unbuffered, while writes are buffered as usual.
This is enough to support the operations needed by packet based
data transfer like in udp/rtp, where aviocontext is the only
public API for hooking up custom IO.
Signed-off-by: Martin Storsjö <martin@martin.st>
The function find_things() in configure is confused by component
registration calls as part of multiline macros defining combined
component registration. Coalesce those macros into one line to
work around the issue.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Signed-off-by: Martin Storsjö <martin@martin.st>
The data does not contain timing or trailing line breaks anymore. In
addition to being less idiotic, it is consistent with other codecs and
thus allows more switches between formats and codecs. It also fixes the
issue of the trailing line returns being simple \n instead of CRLF in
the ASS rectangle dialogue (this is the reason of the FATE update).
"que" sounds like a slang word to me. This commit renames a few
variables, fix the comments and the logging messages (sometimes along
with small other typo fixes).
* qatar/master:
rtmp: Add support for limelight authentication
rtmp: Add support for adobe authentication
Conflicts:
Changelog
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '33f28a3be3092f642778253d9529dd66fe2a014a':
rtmp: Add a function for writing AMF strings based on two substrings
rtmp: Return a proper error code in handle_invoke_error
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Limelight is a not too uncommon CDN. The authentication scheme is
pretty similar to the adobe authentication, but is even closer to
normal http digest authentication (but not close enough to warrant
sharing code) than the adobe version.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is mostly used to authenticate the client when publishing.
Tested with wowza and akamai.
Some but not all servers support resending a new connect invoke
within the same connection, so always reconnect for sending a new
connection attempt. This matches what other applications do as well.
The authentication scheme is structurally pretty similar to http
digest authentication, but uses base64 instead of hex strings.
Signed-off-by: Martin Storsjö <martin@martin.st>
Current MicroDVD AVPackets contain timing information and trailing line
breaks. The data is now only composed of the markup data. Doing this
consistently between text subtitles decoders allows to use different
codec for various formats. For instance, MicroDVD markup is sometimes
found in some VPlayer files. Also, generally speaking, the subtitles
text decoders have no use of these timings (and they must not use them
since it would break any user timing adjustment).
Technically, this is a major ABI break. In practice, a mismatching
lavf/lavc will now error out for MicroDVD decoding. Supporting both
formats requires unnecessary complex and fragile code.
FATE needs update because line breaks in the ASS file were "\n" (because
that's what is used in the original file). ASS format expect "\r\n" line
breaks; this commit fixes this issue. Also note that this "\r\n"
trailing need to be moved at some point from the decoders to the ASS
muxer.
Note that the linebreaks text codec option (but not the feature) has
been removed; its main goal was to allow demuxers to configure the text
decoder (and not meant to be used by users), but the AVOption are not a
viable solution. This is solved differently in this commit.
This commit also makes sure the extradata and subtitle_header are NUL
terminated, without taking into account the trailing '\0' in account in
the size.
At the same time, it should fix 'warning: dereferencing type-punned
pointer will break strict-aliasing rules' warning for compilers who
don't consider uint8_t** and char** compatibles.
* commit '30a76487304e7250294c9c0e9fa179bf07fd822a':
hlsenc: make segment number unsigned
hlsenc: make EXT-X-MEDIA-SEQUENCE always increase
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '9b1370aced385698bc783747917544ab69ecb373':
hlsenc: do not add timestamps in different timebases
hlsenc: use the correct AV_TIME_BASE macro
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '0448f26c97c5ab4858d31e456a4f1738ae783242':
hlsenc: keep the playlist to the correct number of items
hlsenc: use the segment filename in the playlist entry
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '6dd93ee6f1b050ad7c4b247899e83efa293ee405':
hlsenc: check append_entry return value
hlsenc: use the basename to generate the list entries
avstring: add av_basename and av_dirname
Conflicts:
Changelog
doc/APIchanges
libavutil/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Some wav files report a data size that is bigger than the actual file size.
Fall back to estimation from bitrate in such cases.
Fixes ticket #2065.
Signed-off-by: James Almer <jamrial@gmail.com>
Also fixes linking in various configs with only individual parts enabled
because the RTP muxer chaining code depends on the general RTP code,
which is now accounted for.
Since 83cab07 audio stream time bases are based on SampleRate, not EditRate.
This fixes trac ticket #2029 and a few seeking issues.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If s->filename doesn't contain any period/filename extension to strip
away, the buffer will be too small to fit both strings. This isn't
any buffer overflow since the concatenation uses av_strlcat with
the right buffer size.
Signed-off-by: Martin Storsjö <martin@martin.st>
