Define positive return values as non errors and leave further meaning undefined
This allows future extensions to use these values
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '71549a857b13edf4c4f95037de6ed5bb4c4bd4af':
http: Support auth method detection for POST
Conflicts:
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Inspired by a patch by Jakob van Bethlehem. But instead of doing
an empty POST first to trigger the WWW-Authenticate header (which
would succeed if no auth actually was required), add an Expect:
100-continue header, which is meant to be used exactly for
cases like this.
The header is added if doing a post, and the user has specified
authentication but we don't know the auth method yet.
Not all common HTTP servers support the Expect: 100-continue header,
though, so we only try to use it when it really is needed. The user
can request it to be added for other POST requests as well via
an option - which would allow the caller to know immediately that
the POST has failed (e.g. if no auth was provided but the server
required it, or if the target URL simply doesn't exist).
This is only done for write mode posts (e.g. posts without pre-set
post_data) - for posts with pre-set data, we can just redo the post
if it failed due to 401.
Signed-off-by: Martin Storsjö <martin@martin.st>
The default is to autodetect the auth method. This does require one
extra request (and also closing and reopening the http connection).
For some cases such as HTTP POST, the autodetection is not handled
properly (yet).
No option is added for digest, since this method requires getting
nonce parameters from the server first and can't be used straight
away like Basic.
Signed-off-by: Martin Storsjö <martin@martin.st>
previously only codec_ids could be forced, which did not allow
forcing a specific implementation like libopenjpeg vs jpeg2000.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Also add options for specifying a certificate and key, which can
be used both when operating as client and as server.
Partially based on a patch by Peter Ross.
Signed-off-by: Martin Storsjö <martin@martin.st>
A file containing the trusted CA certificates needs to be
supplied via the ca_file AVOption, unless the TLS library
has got a system default file/database set up.
This doesn't check the hostname of the peer certificate with
openssl, which requires a non-trivial piece of code for
manually matching the desired hostname to the string provided
by the certificate, not provided as a library function.
That is, with openssl, this only validates that the received
certificate is signed with the right CA, but not that it is
the actual server we think we're talking to.
Verification is still disabled by default since we can't count
on a proper CA database existing at all times.
Signed-off-by: Martin Storsjö <martin@martin.st>
ASF markers only have a start time, so we lose the chapter end times,
but that is ASF for you
Signed-off-by: Vladimir Pantelic <vladoman@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
movenc: Add an option for omitting the tfhd base offset
Conflicts:
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This makes the output fragments independent of their position in
the output stream, making the output work better when streamed.
QuickTime Player doesn't support fragmented mp4 without the base
data offset, though.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '596e5d4783ca951258a7c580951fd161f1785ec1':
lavf: Add a flag to enable/disable per-packet flushing
Conflicts:
libavformat/avformat.h
libavformat/mux.c
libavformat/version.h
This adds a 2nd API to set per packet flushing
If the user application indicates through either a non default then this non default takes
precedence over the other still default value
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is enabled by default and can be disabled with
"-fflags -flush_packets".
Inspired by a patch from Nicolas George <nicolas.george@normalesup.org>.
Signed-off-by: Martin Storsjö <martin@martin.st>
In order to represent the codec delay accurately in Matroska, a
new element CodecDelay has been introduced. It contains the
overall delay added by the codec in nanoseconds. This patch adds
support for muxing CodecDelay value in the container.
Matroska spec for CodecDelay element can be found here:
http://matroska.org/technical/specs/index.html#CodecDelay
Signed-off-by: Vignesh Venkatasubramanian <vigneshv@google.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b886f5c2f1e71b3e60e4265c500158d392b4b9a4':
mkv: Allow flushing the current cluster in progress
Conflicts:
libavformat/matroskaenc.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Allow emitting the current cluster that is being written before
starting a new one, simplifying how to figure out where clusters
are positioned in the output stream (for live streaming).
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
Changes since v1 of the patch:
- enable option by default
- add documentation
- move up PTS override code after PES header parsing, to ensure we use the
last PCR before the first packet of the teletext PES packet.
The option overrides teletext packet PTS and DTS values with the timestamps
calculated from the PCR of the first program which the teletext stream is part
of and is not discarded.
Using the same teletext PID for multiple programs is possible, therefore we
need some kind of heuristics to know which program PCR we should synchronize
to. Using the first non-discarded PCR pid among the programs of the teletext
stream seemed like a good choice.
The patch does not do PCR interpolation to estimate the PCR of the teltetext
packet, it just uses the last PCR of the program, which may cause a slight
error (0.1 sec) in the teletext packet pts-es.
Based on a patch by Reimar Döffinger.
http://lists.ffmpeg.org/pipermail/ffmpeg-devel/2012-September/131610.html
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
avconv uses private and internal fields from libavformat, we thus must
match the layout even of the fields marked non public.
Otherwise ffmpegs libavformat could not be used as a dropin replacement
on debian/ubuntu
The current soname of libavformat was not part of any release nor are any
fields marked public moved thus in theory
no installed shared lib ABI breakage should occur. Still the need for this
change is unfortunate and chilling.
If you installed shared libs from a recent development version of libavformat
that is more recent than the last release. You probably want to check or rebuild
applications that linked to it.
minor versions of avformat & avdevice are bumped to allow detecting this
as both use the updated struct
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The toc is inexact and not using it can thus make sense.
Using it is faster though, thus the opposite can similarly makes sense
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Allow applications to request reading streamcast metadata. This uses
AVOptions as API, and requires the application to explicitly request
and read metadata. Metadata can be updated mid-stream; if an
application is interested in that, it has to poll for the data by
reading the "icy_metadata_packet" option in regular intervals.
There doesn't seem to be a nice way to transfer the metadata in a nicer
way. Converting the metadata to ID3v2 tags might be a nice idea, but
the libavformat mp3 demuxer doesn't seem to read these tags mid-stream,
and even then we couldn't guarantee that tags are not inserted in the
middle of mp3 packet data.
This commit provides the minimum to enable applications to retrieve
this information at all.
Signed-off-by: Stefano Sabatini <stefasab@gmail.com>
There are 4 separate WebVTT text track kinds: subtitles (the default
if not otherwise specified), captions, descriptions, and metadata.
The WebM muxer needs to know which WebVTT text track kind this is, in
order to synthesize the correct track type and codec id.
To allow a demuxer to indicate the text track kind of the input, a new
set of AV_DISPOSITION flag values has been added, corresponding to
each of the non-default text track kind values.
Adding an arbitrary amount of padding bytes at the end of the
ID3 metadata fixes cover art display for some software (iTunes,
Traktor, Serato, Torq).
For reference (ID3 metadata):
[ Apic frames ] -> cover doesn't show up
[ Apic frames, Padding ] -> ok
[ Apic frames, ID3 frames ] -> ok
[ ID3 frames, Apic frames ] -> cover doesn't show up
[ ID3 frames, Apic frames, Padding ] -> ok
* commit '2d2d6a4883479403798f4ed46941d5b365823570':
lavf: add a raw WavPack muxer.
apetag: add support for writing APE tags
matroskaenc: support muxing WavPack
Conflicts:
libavformat/Makefile
libavformat/allformats.c
libavformat/apetag.h
libavformat/version.h
libavformat/wvenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'e3b225a4fe0ff1e64a220b757c6f0a5cf9258521':
matroskaenc: add an option to put the index at the start of the file
Conflicts:
doc/muxers.texi
libavformat/matroskaenc.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Currently, we have a AV_CODEC_ID_SSA, which matches the way the ASS/SSA
markup is muxed in a standalone .ass/.ssa file. This means the AVPacket
data starts with a "Dialogue:" string, followed by a timing information
(start and end of the event as string) and a trailing CRLF after each
line. One packet can contain several lines. We'll refer to this layout
as "SSA" or "SSA lines".
In matroska, this markup is not stored as such: it has no "Dialogue:"
prefix, it contains a ReadOrder field, the timing information is not in
the payload, and it doesn't contain the trailing CRLF. See [1] for more
info. We'll refer to this layout as "ASS".
Since we have only one common codec for both formats, the matroska
demuxer is constructing an AVPacket following the "SSA lines" format.
This causes several problems, so it was decided to change this into
clean ASS packets.
Some insight about what is changed or unchanged in this commit:
CODECS
------
- the decoding process still writes "SSA lines" markup inside the ass
fields of the subtitles rectangles (sub->rects[n]->ass), which is
still the current common way of representing decoded subtitles
markup. It is meant to change later.
- new ASS codec id: AV_CODEC_ID_ASS (which is different from the
legacy AV_CODEC_ID_SSA)
- lavc/assdec: the "ass" decoder is renamed into "ssa" (instead of
"ass") for consistency with the codec id and allows to add a real
ass decoder. This ass decoder receives clean ASS lines (so it starts
with a ReadOrder, is followed by the Layer, etc). We make sure this
is decoded properly in a new ass-line rectangle of the decoded
subtitles (the ssa decoder OTOH is doing a simple straightforward
copy). Using the packet timing instead of data string makes sure the
ass-line now contains the appropriate timing.
- lavc/assenc: just like the ass decoder, the "ssa" encoder is renamed
into "ssa" (instead of "ass") for consistency with the codec id, and
allows to add a real "ass" encoder.
One important thing about this encoder is that it only supports one
ass rectangle: we could have put several dialogue events in the
AVPacket (separated by a \0 for instance) but this would have cause
trouble for the muxer which needs not only the start time, but also
the duration: typically, you have merged events with the same start
time (stored in the AVPacket->pts) but a different duration. At the
moment, only the matroska do the merge with the SSA-line codec.
We will need to make sure all the decoders in the future can't add
more than one rectangle (and only one Dialogue line in it
obviously).
FORMATS
-------
- lavf/assenc: the .ass/.ssa muxer can take both SSA and ASS packets.
In the case of ASS packets as input, it adds the timing based on the
AVPacket pts and duration, and mux it with "Dialogue:", trailing
CRLF, etc.
- lavf/assdec: unchanged; it currently still only outputs SSA-lines
packets.
- lavf/mkv: the demuxer can now output ASS packets without the need of
any "SSA-lines" reconstruction hack. It will become the default at
next libavformat bump, and the SSA support will be dropped from the
demuxer. The muxer can take ASS packets since it's muxed normally,
and still supports the old SSA packets. All the SSA support and
hacks in Matroska code will be dropped at next lavf bump.
[1]: http://www.matroska.org/technical/specs/subtitles/ssa.html
* commit '85a5bc054c1290699ccbf5799ba6c4e2fbcc3530':
lavf: remove disabled FF_API_R_FRAME_RATE cruft
Conflicts:
libavformat/avformat.h
libavformat/electronicarts.c
libavformat/rawdec.c
libavformat/utils.c
libavformat/version.h
The field is in use and no semantically equivalent field is available,
thus not removed
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This reverts commit 6cc12353a8.
Conflicts:
libavformat/version.h
Allowing to automatically select the concat demuxer raises
security concerns, as it allows a possibly hostile file to
access any file on the system. Guessing the format based on
the file name extension does not allow to enable the safe
mode designed to avoid it.
* qatar/master:
lavf: Add a fate test for the noproxy pattern matching
lavf: Handle the environment variable no_proxy more properly
Conflicts:
libavformat/Makefile
libavformat/internal.h
libavformat/tls.c
libavformat/utils.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The handling of the environment variable no_proxy, present since
one of the initial commits (de6d9b6404), is inconsistent with
how many other applications and libraries interpret this
variable. Its bare presence does not indicate that the use of
proxies should be skipped, but it is some sort of pattern for
hosts that does not need using a proxy (e.g. for a local network).
As investigated by Rudolf Polzer, different libraries handle this
in different ways, some supporting IP address masks, some supporting
arbitrary globbing using *, some just checking that the pattern matches
the end of the hostname without regard for whether it actually is
the right domain or a domain that ends in the same string.
This simple logic should be pretty similar to the logic used by
lynx and curl.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '4f56e773fe8a554b8c2662650aaf799c2ece2721':
x86: ac3: Fix HAVE_MMXEXT condition to only refer to external assembly
rtpenc: Start the sequence numbers from a random offset
Conflicts:
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Expose the current sequence number via an AVOption - this can
be used both for setting the initial sequence number, or for
querying the current number.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit 'a2a991b2ddf951454ffceb7bcedc9db93e26c610':
srtp: Improve the minimum encryption buffer size check
srtp: Add support for a few DTLS-SRTP related crypto suites
Conflicts:
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The main difference to the existing suites from RFC 4568 is
that the version with a 32 bit HMAC still uses 80 bit HMAC
for RTCP packets.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '2f3bada63e57345329c4f9b48e9b81b5cfc03d05':
lavf: Add a protocol for SRTP encryption/decryption
rtsp: Support decryption of SRTP signalled via RFC 4568 (SDES)
Conflicts:
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is mostly useful for encryption together with the RTP muxer,
but could also be set up as IO towards the peer with the SDP
demuxer with custom IO.
Signed-off-by: Martin Storsjö <martin@martin.st>
This only takes care of decrypting incoming packets; the outgoing
RTCP packets are not encrypted. This is enough for some use cases,
and signalling crypto keys for use with outgoing RTCP packets
doesn't fit as simply into the API. If the SDP demuxer is hooked
up with custom IO, the return packets can be encrypted e.g. via the
SRTP protocol.
If the SRTP keys aren't available within the SDP, the decryption
can be handled externally as well (when using custom IO).
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
* commit 'e96406eda4f143f101bd44372f7b2d542183000a':
rtsp: Add support for depacketizing RTP data via custom IO
Conflicts:
libavformat/version.h
Merged-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>
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).
* 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>
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.
The new options reset the timestamps at each new segment, so that the
generated segments will have timestamps starting close to 0.
It is meant to address trac ticket #1425.
Gif demuxer is capable of extracting multiple frames from gif file.
In conjunction with gif decoder it implements support for reading
animated gifs.
Demuxer has two options available to user: default_delay and min_delay.
These options are for protection from too rapid gif animations. In practice
it is standard approach to slow down rendering of this kind of gifs. If you try to
play gif with delay between frames of one hundredth of second (100fps) using
one of major web browsers, you get significantly slower playback,
around 10 fps. This is because browser detects that delay value is less than some
threshold (usually 2 hundredths of second) and reset it to default value (usually 10
hundredths of second, which corresponds to 10fps). Manipulating these options user
can achieve the same effect during conversion to some video format. Otherwise user
can set them to not protect from rapid animations at all.
The other case when these options necessary is for gif images encoded according to
gif87a standard since prior to gif89a there was no delay information included in file.
Bump lavf minor version.
Signed-off-by: Vitaliy E Sugrobov <vsugrob@hotmail.com>
* commit 'b522000e9b2ca36fe5b2751096b9a5f5ed8f87e6':
avio: introduce avio_closep
mpegtsenc: set muxing type notification to verbose
vc1dec: Use correct spelling of "opposite"
a64multienc: change mc_frame_counter to unsigned
arm: call arm-specific rv34dsp init functions under if (ARCH_ARM)
svq1: Drop a bunch of useless parentheses
parseutils-test: do not print numerical error codes
svq1: K&R formatting cosmetics
Conflicts:
doc/APIchanges
libavcodec/svq1dec.c
libavcodec/svq1enc.c
libavformat/version.h
libavutil/parseutils.c
tests/ref/fate/parseutils
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
dwt: Drop unused functions spatial_compose{53|97}i()
nutdec: Remove unused and broken debug function stub
avcodec: Drop long-deprecated imgconvert.h header
Add Opus support to the Ogg muxer.
Add Opus codec id and codec description.
avformat: Identify anonymous AVIO typedef structs.
Conflicts:
libavcodec/avcodec.h
libavcodec/codec_desc.c
libavcodec/imgconvert.h
libavcodec/version.h
libavformat/oggenc.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
nutdec: const correctness for get_v_trace/get_s_trace function arguments
truemotion2: Request samples for old TM2 headers
rtpdec: Remove a useless ff_ prefix from a static symbol
rtpdec: Support depacketizing speex
rtpenc: Add support for packetizing speex
Conflicts:
libavformat/rtpdec.c
libavformat/sdp.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This packetization scheme simply places the full packets into the
RTP packet without any extra header bytes.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
rtp: Packetization of JPEG (RFC 2435)
smoothstreamingenc: Copy the SAR on the AVStreams as well
Conflicts:
Changelog
libavformat/rtpenc.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '3f7fd59d151a2773f0e2e93e56b6b13ec6e5334b':
avformat: fix typo in avformat_close_input
mp3enc: write Xing TOC
mp3enc: support MPEG-2 and MPEG-2.5 in Xing header.
mp3enc: downgrade some errors in writing Xing frame to warnings
lavf: flush the output AVIOContext in av_write_trailer().
lavf: cosmetics, reformat av_write_trailer().
avio: flush the internal buffer in avio_close()
Enhance doc on asyncts audiofilter
cmdutils: avoid setting data pointers to invalid values in alloc_buffer()
libavcodec: remove av_destruct_packet_nofree()
Conflicts:
libavcodec/avpacket.c
libavformat/mp3enc.c
libavformat/nutenc.c
libavformat/utils.c
libavformat/version.h
tests/ref/lavf/voc
tests/ref/lavf/voc_s16
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '07584eaf4a95db3f11d3bc411f9786932829e82b':
mpegts: check substreams before discarding
Add a smooth streaming segmenter muxer
file: Add an avoption for disabling truncating existing files on open
img2dec: always close AVIOContexts
rtpdec_jpeg: Error out on other unsupported type values as well
rtpdec_jpeg: Disallow using the reserved q values
rtpdec_jpeg: Fold the default qtables case into an existing if statement
rtpdec_jpeg: Store and reuse old qtables for q values 128-254
rtpdec_jpeg: Simplify the calculation of the number of qtables
rtpdec_jpeg: Add more comments about the fields in the SOF0 section
rtpdec_jpeg: Clarify where the subsampling magic numbers come from
rtpdec_jpeg: Don't use a bitstream writer for the EOI marker
rtpdec_jpeg: Don't needlessly use a bitstream writer for the header
rtpdec_jpeg: Simplify writing of the jpeg header
rtpdec_jpeg: Merge two if statements
rtpdec_jpeg: Write the DHT section properly
Conflicts:
libavformat/Makefile
libavformat/allformats.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This muxer splits the output from the ismv muxer into individual
files, in realtime.
The same can also be done by the standalone tool ismindex, but this
muxer is needed for doing it in realtime (especially for live
streams that need extra handling for updating the lookahead fields
in the fragment headers).
Using this muxer, one can deliver live smooth streaming from a
normal static file web server. (Using ismindex, one can deliver
premade smooth streaming files from a static file web server,
or prepare files for serving with IIS.)
Signed-off-by: Martin Storsjö <martin@martin.st>
Allow to specify options affecting the segment list generation.
In particular: add +live and +cache flags.
For a full discussion read trac ticket #1642:
http://ffmpeg.org/trac/ffmpeg/ticket/1642
Also add live M3U8 generation example.
* qatar/master:
x86: dsputil: Only compile motion_est code when encoders are enabled
mem: fix typo in check for __ICC
fate: mp3: drop redundant CMP setting
rtp: Depacketization of JPEG (RFC 2435)
Rename ff_put_string to avpriv_put_string
mjpeg: Rename some symbols to avpriv_* instead of ff_*
yadif: cosmetics
Conflicts:
Changelog
libavcodec/mjpegenc.c
libavcodec/x86/Makefile
libavfilter/vf_yadif.c
libavformat/version.h
libavutil/mem.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Make internal small_strptime() function public, and use it in place of
strptime().
This allows to avoid a dependency on strptime() on systems which do not
support it.
In particular, fix trac ticket #992.
* qatar/master:
libvpxenc: use the default bitrate if not set
utvideo: Rename utvideo.c to utvideodec.c
doc: Fix syntax errors in sample Emacs config
mjpegdec: more meaningful return values
configure: clean up Altivec detection
getopt: Remove an unnecessary define
rtmp: Use int instead of ssize_t
getopt: Add missing includes
rtmp: Add support for receiving incoming streams
Add missing includes for code relying on external libraries
Conflicts:
libavcodec/libopenjpegenc.c
libavcodec/libvpxenc.c
libavcodec/mjpegdec.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Fix even more missing includes after the common.h removal
build: Factor out rangecoder dependencies to CONFIG_RANGECODER
build: Factor out error resilience dependencies to CONFIG_ERROR_RESILIENCE
x86: avcodec: Consistently name all init files
Add more missing includes after removing the implicit common.h
Add some more missing includes after removing the implicit common.h
Don't include common.h from avutil.h
rtmp: Automatically compute the hash for SWFVerification
Conflicts:
configure
doc/APIchanges
doc/examples/decoding_encoding.c
libavcodec/Makefile
libavcodec/assdec.c
libavcodec/audio_frame_queue.c
libavcodec/avpacket.c
libavcodec/dv_profile.c
libavcodec/dwt.c
libavcodec/libtheoraenc.c
libavcodec/rawdec.c
libavcodec/rv40dsp.c
libavcodec/tiff.c
libavcodec/tiffenc.c
libavcodec/v210dec.h
libavcodec/vc1dsp.c
libavcodec/x86/Makefile
libavfilter/asrc_anullsrc.c
libavfilter/avfilter.c
libavfilter/buffer.c
libavfilter/formats.c
libavfilter/vf_ass.c
libavfilter/vf_drawtext.c
libavfilter/vf_fade.c
libavfilter/vf_select.c
libavfilter/video.c
libavfilter/vsrc_testsrc.c
libavformat/version.h
libavutil/audioconvert.c
libavutil/error.h
libavutil/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtmp: Add support for SWFVerification
api-example: use new video encoding API.
x86: avcodec: Appropriately name files containing only init functions
mpegvideo_mmx_template: drop some commented-out cruft
libavresample: add mix level normalization option
w32pthreads: Add missing #includes to make header compile standalone
rtmp: Gracefully ignore _checkbw errors by tracking them
rtmp: Do not send _checkbw calls as notifications
prores: interlaced ProRes encoding
Conflicts:
doc/examples/decoding_encoding.c
libavcodec/proresenc_kostya.c
libavcodec/w32pthreads.h
libavcodec/x86/Makefile
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Specifies how the server verifies client SWF files before allowing the
files to connect to an application. Verifying SWF files is a security
measure that prevents someone from creating their own SWF files that can
attempt to stream your resources.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
mpegvideo: reduce excessive inlining of mpeg_motion()
mpegvideo: convert mpegvideo_common.h to a .c file
build: factor out mpegvideo.o dependencies to CONFIG_MPEGVIDEO
Move MASK_ABS macro to libavcodec/mathops.h
x86: move MANGLE() and related macros to libavutil/x86/asm.h
x86: rename libavutil/x86_cpu.h to libavutil/x86/asm.h
aacdec: Don't fall back to the old output configuration when no old configuration is present.
rtmp: Add message tracking
rtsp: Support mpegts in raw udp packets
rtsp: Support receiving plain data over UDP without any RTP encapsulation
rtpdec: Remove an unused include
rtpenc: Remove an av_abort() that depends on user-supplied data
vsrc_movie: discourage its use with avconv.
avconv: allow no input files.
avconv: prevent invalid reads in transcode_init()
avconv: rename OutputStream.is_past_recording_time to finished.
Conflicts:
configure
doc/filters.texi
ffmpeg.c
ffmpeg.h
libavcodec/Makefile
libavcodec/aacdec.c
libavcodec/mpegvideo.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (23 commits)
build: cosmetics: Reorder some lists in a more logical fashion
x86: pngdsp: Fix assembly for OS/2
fate: add test for RTjpeg in nuv with frameheader
rtmp: send check_bw as notification
g723_1: clip argument for 15-bit version of normalize_bits()
g723_1: use all LPC vectors in formant postfilter
id3v2: Support v2.2 PIC
avplay: fix build with lavfi disabled.
avconv: split configuring filter configuration to a separate file.
avconv: split option parsing into a separate file.
mpc8: do not leave padding after last frame in buffer for the next decode call
mpegaudioenc: list supported channel layouts.
mpegaudiodec: don't print an error on > 1 frame in a packet.
api-example: update to new audio encoding API.
configure: add --enable/disable-random option
doc: cygwin: Update list of FATE package requirements
build: Remove all installed headers and header directories on uninstall
build: change checkheaders to use regular build rules
rtmp: Add a new option 'rtmp_subscribe'
rtmp: Add support for subscribing live streams
...
Conflicts:
Makefile
common.mak
configure
doc/examples/decoding_encoding.c
ffmpeg.c
libavcodec/g723_1.c
libavcodec/mpegaudiodec.c
libavcodec/x86/pngdsp.asm
libavformat/version.h
library.mak
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Allow to override the default 'glob_sequence' value, which is deprecated
in favor of the new 'glob' and 'sequence' options.
The new pattern types should be easier on the user since they are more
predictable than 'glob_sequence', and do not require awkward escaping.
* qatar/master:
flvdec: remove spurious use of stream id
lavf: deprecate r_frame_rate.
lavf: round estimated average fps to a "standard" fps.
Conflicts:
ffmpeg.c
ffprobe.c
libavformat/avformat.h
libavformat/electronicarts.c
libavformat/flvdec.c
libavformat/rawdec.c
libavformat/utils.c
tests/ref/fate/iv8-demux
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'fe1c1198e670242f3cf9e3e1eef27cff77f3ee23':
lavf: use dts difference instead of AVPacket.duration in find_stream_info()
avf: introduce nobuffer option
fate: make yadif tests consistent across systems
vf_hqdn3d: support 9 and 10bit colordepth
vf_hqdn3d: reduce intermediate precision
vf_hqdn3d: simplify and optimize
factor identical ff_inplace_start_frame out of two filters
vf_hqdn3d: cosmetics
avprobe/avconv: fix tentative declaration compile errors on MSVS.
Conflicts:
doc/APIchanges
ffmpeg.c
ffprobe.c
libavformat/avformat.h
libavformat/options_table.h
libavformat/utils.c
libavformat/version.h
tests/fate/filter.mak
tests/ref/fate/filter-yadif-mode0
tests/ref/fate/filter-yadif-mode1
Merged-by: Michael Niedermayer <michaelni@gmx.at>
According to its description, it is supposed to be the LCM of all the
frame durations. The usability of such a thing is vanishingly small,
especially since we cannot determine it with any amount of reliability.
Therefore get rid of it after the next bump.
Replace it with the average framerate where it makes sense.
FATE results for the wtv and xmv demux tests change. In the wtv case
this is caused by the file being corrupted (or possibly badly cut) and
containing invalid timestamps. This results in lavf estimating the
framerate wrong and making up wrong frame durations.
In the xmv case the file contains pts jumps, so again the estimated
framerate is far from anything sane and lavf again makes up different
frame durations.
In some other tests lavf starts making up frame durations from different
frame.
Useful in cases where a significant analyzeduration is
still needed, while minimizing buffering before output.
An example is processing low-latency streams where all
media types won't necessarily come in if the
analyzeduration is small.
Additional changes by Josh Allmann <joshua.allmann@gmail.com>
Signed-off-by: Anton Khirnov <anton@khirnov.net>
* qatar/master:
libopenjpeg: support YUV and deep RGB pixel formats
Fix typo in v410 decoder.
vf_yadif: unset cur_buf on the input link.
vf_overlay: ensure the overlay frame does not get leaked.
vf_overlay: prevent premature freeing of cur_buf
Support urlencoded http authentication credentials
rtmp: Return an error when the client bandwidth is incorrect
rtmp: Return proper error code in handle_server_bw
rtmp: Return proper error code in handle_client_bw
rtmp: Return proper error codes in handle_chunk_size
lavr: x86: add missing vzeroupper in ff_mix_1_to_2_fltp_flt()
vp8: Replace x*155/100 by x*101581>>16.
vp3: don't use calls to inline asm in yasm code.
x86/dsputil: put inline asm under HAVE_INLINE_ASM.
dsputil_mmx: fix incorrect assembly code
rtmp: Factorize the code by adding handle_invoke
rtmp: Factorize the code by adding handle_chunk_size
rtmp: Factorize the code by adding handle_ping
rtmp: Factorize the code by adding handle_client_bw
rtmp: Factorize the code by adding handle_server_bw
Conflicts:
libavcodec/libopenjpegdec.c
libavcodec/x86/dsputil_mmx.c
libavfilter/vf_overlay.c
libavformat/Makefile
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It should be possible to specify usernames in http requests containing
urlencoded characters. This patch adds support for decoding the auth
strings.
Signed-off-by: Antti Seppälä <a.seppala@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
v410dec: Implement explode mode support
zerocodec: fix direct rendering.
wav: init st to NULL to avoid a false-positive warning.
wavpack: set bits_per_raw_sample for S32 samples to properly identify 24-bit
h264: refactor NAL decode loop
RTMPTE protocol support
RTMPE protocol support
rtmp: Add ff_rtmp_calc_digest_pos()
rtmp: Rename rtmp_calc_digest to ff_rtmp_calc_digest and make it global
swscale: add missing HAVE_INLINE_ASM check.
lavfi: place x86 inline assembly under HAVE_INLINE_ASM.
vc1: Add a test for interlaced field pictures
swscale: Mark all init functions as av_cold
swscale: x86: Drop pointless _mmx suffix from filenames
lavf: use conditional notation for default codec in muxer declarations.
swscale: place inline assembly bilinear scaler under HAVE_INLINE_ASM.
dsputil: ppc: cosmetics: pretty-print
dsputil: x86: add SHUFFLE_MASK_W macro
configure: respect CC_O setting in check_cc
Conflicts:
Changelog
configure
libavcodec/v410dec.c
libavcodec/zerocodec.c
libavformat/asfenc.c
libavformat/version.h
libswscale/utils.c
libswscale/x86/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This adds two protocols, but one of them is an internal implementation
detail just used as an abstraction layer/generalization in the code. The
RTMPE protocol implementation uses ffrtmpcrypt:// as an alternative to the
tcp:// protocol. This allows moving most of the lower level logic out
from the higher level generic rtmp code.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
libopenjpeg: introduce encoding support
libopenjpeg: rename decoder source file.
RTMPTS protocol support
RTMPS protocol support
avconv: print an error message when demuxing fails.
tscc2: DCT output should not be clipped
rtmp: Rename rtmphttp to ffrtmphttp
Conflicts:
Changelog
configure
doc/general.texi
libavcodec/libopenjpegenc.c
libavcodec/version.h
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
configure: Check for the math function rint
TechSmith Screen Codec 2 decoder
rtsp: Add listen mode
rtsp: Make rtsp_open_transport_ctx() non-static
rtsp: Move rtsp_read_close
rtsp: Parse the mode=receive/record parameter in transport lines
Conflicts:
Changelog
libavcodec/avcodec.h
libavcodec/version.h
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Add list extended format which specifies in the list file the start and
ending time for each segment. This is required to make it available this
information to external tools, avoiding the need to perform file analysis
in the output segments.
* qatar/master: (29 commits)
lavfi: reclassify showfiltfmts as a TESTPROG
graph2dot: fix printf format specifier
swscale: yuv2planeX 8bit >=sse2 functions need aligned stack on x86-32.
vp8: loopfilter >=sse2 functions need aligned stack on x86-32.
amr: remove shift out of the AMR_BIT() macro.
dsputilenc: group yasm and inline asm function pointer assignment.
mov: use forward declaration of a function instead of a table.
Clarify Doxygen comment for FF_API_* #defines.
configure: simplify get_version()
Create version.h headers for libraries that lack them
gitignore: Use full path instead of relative path to specify patterns
mpegvideo: remove VLAs
Add XTEA encryption support in libavutil
Add Blowfish encryption support in libavutil
eval: Add the isinf() function and tests for it
flacdec: move lpc filter to flacdsp
flacdec: split off channel decorrelation as flacdsp
avplay: Add an option for not limiting the input buffer size
FATE: add a test for WMA cover art.
FATE: add a test for apetag cover art
...
Conflicts:
.gitignore
configure
ffplay.c
libavcodec/Makefile
libavcodec/error_resilience.c
libavcodec/mpegvideo.c
libavcodec/ratecontrol.c
libavdevice/avdevice.h
libavfilter/Makefile
libavfilter/filtfmts.c
libavfilter/version.h
libavformat/mov.c
libavformat/version.h
libavutil/Makefile
libavutil/avutil.h
libavutil/version.h
libswscale/swscale.h
libswscale/x86/swscale_mmx.c
tests/fate/libavutil.mak
tests/lavfi-regression.sh
tools/graph2dot.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
