* qatar/master: (44 commits)
replacement Indeo 3 decoder
gsm demuxer: do not allocate packet twice.
flvenc: use first packet delay as global delay.
ac3enc: doxygen update.
imc: return error codes instead of 0 for error conditions.
imc: return meaningful error codes instead of -1
imc: do not set channel layout for stereo
imc: validate channel count
imc: check for ff_fft_init() failure
imc: check output buffer size before decoding
imc: use DSPContext.bswap16_buf() to byte-swap packet data
rtsp: add allowed_media_types option
libgsm: add flush function to reset the decoder state when seeking
libgsm: simplify decoding by using a loop
gsm: log error message when packet is too small
libgsmdec: do not needlessly set *data_size to 0
gsmdec: do not needlessly set *data_size to 0
gsmdec: add flush function to reset the decoder state when seeking
libgsmdec: check output buffer size before decoding
gsmdec: log error message when output buffer is too small.
...
Conflicts:
Changelog
ffplay.c
libavcodec/indeo3.c
libavcodec/mjpeg_parser.c
libavcodec/vp3.c
libavformat/cutils.c
libavformat/id3v2.c
libavutil/parseutils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Streams from RTSP or SDP that do not match an allowed type will
be skipped entirely, which allows video-only or audio-only
streaming from servers that provide both.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
avconv: add presets
rtsp: Expose the flag options via private AVOptions for sdp and rtp, too
rtsp: Make the rtsp flags avoptions set via a define
rtpenc: Set a default video codec
avoptions: Fix av_opt_flag_is_set
rtp: Fix ff_rtp_get_payload_type
doc: Update the documentation on setting options for RTSP
rtsp: Remove the separate filter_source variable
rtsp: Accept options via private avoptions instead of URL options
rtsp: Simplify AVOption definitions
rtsp: Merge the AVOption lists
lavfi: port libmpcodecs delogo filter
lavfi: port boxblur filter from libmpcodecs
lavfi: add negate filter
lavfi: add LUT (LookUp Table) generic filters
AVOptions: don't segfault on NULL parameter in av_set_options_string()
avio: Check for invalid buffer length.
mpegenc/mpegtsenc: add muxrate private options.
lavf: deprecate AVFormatContext.file_size
mov: add support for TV metadata atoms tves, tvsn and stik
Conflicts:
Changelog
doc/filters.texi
doc/protocols.texi
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/avfilter.h
libavfilter/formats.c
libavfilter/internal.h
libavfilter/vf_boxblur.c
libavfilter/vf_delogo.c
libavfilter/vf_lut.c
libavformat/mpegtsenc.c
libavformat/utils.c
libavformat/version.h
libavutil/opt.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Eventually, the old way of passing options by adding
stuff to the URL can be dropped.
This avoids having to tamper with the user-specified URL to
pass options on the transport mode. This also works better
with redirects, since the options don't need to be parsed out
from the URL.
Signed-off-by: Martin Storsjö <martin@martin.st>
This eases adding options that are common for both. The
AV_OPT_FLAG_EN/DECODING_PARAM still indicates whether they belong
to the muxer or demuxer.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
simple_idct: simplify some ifdeffery
simple_idct: remove code for DCTELEM != int16
Remove VLAs in ff_amrwb_lsp2lpc()
fate: make vsynth tests depend on only the relevant vref
rtsp: remove disabled code
dsputil: restore mistakenly removed hunk of disabled code
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
APIchanges: fill in missing hashes and dates.
Add an APIChanges entry and bump minor versions for recent changes.
ffmpeg: print the low bitrate warning after the codec is openend.
doxygen: Move function documentation into the macro generating the function.
doxygen: Make sure parameter names match between .c and .h files.
h264: move fill_decode_neighbors()/fill_decode_caches() to h264_mvpred.h
H.264: Add more x86 assembly for 10-bit H.264 predict functions
lavf: fix invalid reads in avformat_find_stream_info()
cmdutils: replace opt_default with opt_default2() and remove set_context_opts
ffmpeg: use new avcodec_open2 and avformat_find_stream_info API.
ffplay: use new avcodec_open2 and avformat_find_stream_info API.
cmdutils: store all codec options in one dict instead of video/audio/sub
ffmpeg: check experimental flag after codec is opened.
ffmpeg: do not set GLOBAL_HEADER flag in the options context
Conflicts:
cmdutils.c
doc/APIchanges
ffmpeg.c
ffplay.c
libavcodec/version.h
libavformat/version.h
libswscale/swscale_unscaled.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (40 commits)
H.264: template left MB handling
H.264: faster fill_decode_caches
H.264: faster write_back_*
H.264: faster fill_filter_caches
H.264: make filter_mb_fast support the case of unavailable top mb
Do not include log.h in avutil.h
Do not include pixfmt.h in avutil.h
Do not include rational.h in avutil.h
Do not include mathematics.h in avutil.h
Do not include intfloat_readwrite.h in avutil.h
Remove return statements following infinite loops without break
RTSP: Doxygen comment cleanup
doxygen: Escape '\' in Doxygen documentation.
md5: cosmetics
md5: use AV_WL32 to write result
md5: add fate test
md5: include correct headers
md5: fix test program
doxygen: Drop array size declarations from Doxygen parameter names.
doxygen: Fix parameter names to match the function prototypes.
...
Conflicts:
libavcodec/x86/dsputil_mmx.c
libavformat/flvenc.c
libavformat/oggenc.c
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
bitstream: Properly promote av_reverse values before shifting.
libavutil/swscale: YUV444P10/YUV444P9 support.
H.264: Fix high bit depth explicit biweight
h264: Fix 10-bit H.264 x86 chroma v loopfilter asm.
Replace DEBUG_SEEK/DEBUG_SI + av_log combinations by av_dlog.
Update copyright year for ac3enc_opts_template.c.
adts: Adjust frame size mask to follow the specification.
movenc: Add RTP muxer/hinter options
movenc: Pass the RTP AVFormatContext to the SDP generation
rtspenc: Add RTP muxer options
rtspenc: Add an AVClass for setting muxer specific options
rtpenc_chain: Pass the rtpflags options through to the chained muxer
rtpenc: Declare the rtp flags private AVOptions in rtpenc.h
sdp: Reindent after the previous commit
rtpenc: MP4A-LATM payload support
avoptions: Add an av_opt_flag_is_set function for inspecting flag fields
sdp: Allow passing an AVFormatContext to the SDP generation
mov: Fix wrong timestamp generation for fragmented movies that have time offset caused by the first edit list entry.
mpeg12: more advanced ffmpeg mpeg2 aspect guessing code.
swscale: split YUYV output out of yuv2packed[12X]_c().
Conflicts:
doc/APIchanges
libavcodec/Makefile
libavcodec/h264dsp_template.c
libavcodec/mpeg12.c
libavformat/aacdec.c
libavformat/avidec.c
libavformat/internal.h
libavformat/movenc.c
libavformat/rtpenc.c
libavformat/rtpenc_latm.c
libavformat/sdp.c
libavformat/version.h
libavutil/avutil.h
libavutil/pixfmt.h
libswscale/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (32 commits)
10-bit H.264 x86 chroma v loopfilter asm
Port SMPTE S302M audio decoder from FFmbc 0.3. [Copyright headers corrected]
Fix crash of interlaced MPEG2 decoding
h264pred: fix one more aliasing violation.
doc/APIchanges: fill in missing hashes and dates.
flacenc: use proper initializers for AVOption default values.
lavc: deprecate named constants for deprecated antialias_algo.
aac: workaround for compilation on cygwin
swscale: extend YUV422p support to 10bits depth
tiff: add support for inverted FillOrder for uncompressed data
Remove unused softfloat implementation.
h264pred: fix aliasing violations.
rotozoom: Eliminate French variable name.
rotozoom: Check return value of fread().
rotozoom: Return an error value instead of calling exit().
rotozoom: Make init_demo() return int and check for errors on invocation.
rotozoom: Drop silly UINT8 typedef.
rotozoom: Drop some unnecessary parentheses.
rotozoom: K&R coding style cosmetics
rtsp: Only do keepalive using GET_PARAMETER if the server supports it
...
Conflicts:
Changelog
cmdutils.c
doc/APIchanges
doc/general.texi
ffmpeg.c
ffplay.c
libavcodec/h264pred_template.c
libavcodec/resample.c
libavutil/pixfmt.h
libavutil/softfloat.c
libavutil/softfloat.h
tests/rotozoom.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is more like what VLC does. If the server doesn't mention
supporting GET_PARAMETER in response to an OPTIONS request,
VLC doesn't send any keepalive requests at all. After this patch,
libavformat will still send OPTIONS keepalives if GET_PARAMETER
isn't explicitly said to be supported.
Some RTSP cameras don't support GET_PARAMETER, and will
close the connection if this is sent as keepalive request
(but support OPTIONS just fine, but probably don't need any
keepalive at all). Some other cameras don't support using
OPTIONS as keepalive, but require GET_PARAMETER instead.
Signed-off-by: Martin Storsjö <martin@martin.st>
If filtered, only packets from the right source address and port
are received.
To test, play back e.g. some mpeg4 video RTSP stream (where the
video stream is the first stream in the presentation) over UDP.
While receiving this stream, send another stream to the same port:
ffmpeg -re -i <whatever> -vcodec mpeg4 -an -f rtp
rtp://127.0.0.1:5000?localport=1234
Normally, the RTSP playback reports lots of errors at this point.
If the RTSP stream has the ?filter_src option enabled, these
interferring packets are ignored.
Originally committed as revision 26246 to svn://svn.ffmpeg.org/ffmpeg/trunk
This avoids having a large temporary buffer in the struct used for
storing the rtsp reply headers.
Originally committed as revision 26192 to svn://svn.ffmpeg.org/ffmpeg/trunk
This allows ff_rtsp_parse_line to do more changes directly in RTSPState
when parsing the reply, instead of having to store large amounts of
temporary data in RTSPMessageHeader.
Originally committed as revision 26190 to svn://svn.ffmpeg.org/ffmpeg/trunk
message, if available (RFC 2326, section 12.39), fixes issue 2212.
Patch by John Wimer <john at god vtic net>.
Originally committed as revision 25032 to svn://svn.ffmpeg.org/ffmpeg/trunk
Done in preparation for RTSP over HTTP.
Patch by Josh Allmann, joshua dot allmann at gmail
Originally committed as revision 23494 to svn://svn.ffmpeg.org/ffmpeg/trunk
from 2 to 1, which is the actual value used in the spec. Fixes issue1978.
Path by John Wimer <john at god dot vtic dot net>.
Originally committed as revision 23414 to svn://svn.ffmpeg.org/ffmpeg/trunk
This makes sure that the streams get correctly synchronized when viewed,
previously the streams were out of sync by as much time as it took
between the initialization of the individual muxers.
Originally committed as revision 22545 to svn://svn.ffmpeg.org/ffmpeg/trunk
future use of the rtsp* codebase for RTSP muxing.
Patch by Martin Storsjö <$firstname $firstname st>.
Originally committed as revision 21896 to svn://svn.ffmpeg.org/ffmpeg/trunk
if present. This fixes playback of a number of MS-RTSP streams, mostly these
for which playback contains a session key in the URI. Fixes issue 1697.
Patch by Alan Steremberg <$firstname dot $lastname () gmail com>.
Originally committed as revision 21381 to svn://svn.ffmpeg.org/ffmpeg/trunk
All the error codes 3xx got managed the same way.
After setup/early play redirection will not be managed
REDIRECT method is yet to be supported (if somebody knows a server implementing
it please contact me)
Originally committed as revision 20369 to svn://svn.ffmpeg.org/ffmpeg/trunk
(philip coombes zoneminder com), see "[PATCH]RTSP Basic Authentication"
thread on mailinglist.
Originally committed as revision 19905 to svn://svn.ffmpeg.org/ffmpeg/trunk
implement RTCP/bye. See "[PATCH] rtsp.c: EOS support" thread from a few
months back.
Originally committed as revision 19517 to svn://svn.ffmpeg.org/ffmpeg/trunk
discussion in "[PATCH] RTSP-MS 14/15: ASF packet parsing" thread on mailinglist.
Originally committed as revision 19516 to svn://svn.ffmpeg.org/ffmpeg/trunk
a PLAY with Range alone while in PLAY status should be interpreted
as an enqueue
a PAUSE followed by a PLAY with Range is the proper way to ask to
seek to a point.
See rfc2326
Originally committed as revision 19143 to svn://svn.ffmpeg.org/ffmpeg/trunk
improve plain text doxy readability.
See the thread: "[RFC] Should we use doxygen markup?".
Originally committed as revision 19122 to svn://svn.ffmpeg.org/ffmpeg/trunk
Real wants OPTIONS) while the connection is idle, otherwise it will
be aborted after a short period (usually a minute). See the thread
"[PATCH] rtsp.c: keep-alive" on the mailinglist.
Originally committed as revision 18525 to svn://svn.ffmpeg.org/ffmpeg/trunk
the future, requested by Luca in "[PATCH] rtsp.c: read TCP server
notifications/messages" thread.
Originally committed as revision 18120 to svn://svn.ffmpeg.org/ffmpeg/trunk
structure is meant to represent. See "[PATCH] rtsp.[ch]: RTSPHeader ->
RTSPServerResponse" and "[PATCH] document rtsp.h" threads on ML.
Originally committed as revision 17504 to svn://svn.ffmpeg.org/ffmpeg/trunk
which isn't installed anyway (so it doesn't work).
In the process, also remove public/private API comments from rtsp headers
because they are unnecessary.
Originally committed as revision 17379 to svn://svn.ffmpeg.org/ffmpeg/trunk
"cur_transport_priv", as discussed in the "[PATCH] rtsp.h: rename tx
variables" thread.
Originally committed as revision 17012 to svn://svn.ffmpeg.org/ffmpeg/trunk
move move a struct/typedef in rtsp.h that is only used in ffserver.c into
ffserver.c. See "[PATCH] rtsp.h: move/remove unused thingies" thread on ML.
Originally committed as revision 17005 to svn://svn.ffmpeg.org/ffmpeg/trunk
of this type can be properly attributed as such (in this case, transport in
the RTSPTransportField struct). See "[PATCH] RTSP-MS 10/15: ASF header parsing"
thread on mailinglist.
Originally committed as revision 16989 to svn://svn.ffmpeg.org/ffmpeg/trunk
to is a Microsoft Windows Media Server (the field will be "WMServer/version").
See "[PATCH] RTSP-MS 3/15: Add Windows Media Server type" thread on
mailinglist.
Originally committed as revision 16472 to svn://svn.ffmpeg.org/ffmpeg/trunk
access to these structures in functions that will be located in rtp_asf.c.
See "[PATCH] RTSP-MS 2/15: export RTSPState and RTSPStream" mailinglist
thread.
Originally committed as revision 16471 to svn://svn.ffmpeg.org/ffmpeg/trunk
types and their non-standard extensions, and the data they serve. Practically,
this patch allows Real servers to serve normal non-RDT (standard RTP) data.
See discussion on ML in "Realmedia patch" thread.
Originally committed as revision 15484 to svn://svn.ffmpeg.org/ffmpeg/trunk
that it only describes the lower-level transport (TCP vs. UDP) and not the
actual data layout (e.g. RDT vs. RTP). See discussion in "Realmedia patch"
thread on ML.
Originally committed as revision 15481 to svn://svn.ffmpeg.org/ffmpeg/trunk
Consistently apply this rule: the guard name is obtained from the
filename by stripping the leading "lib", converting '/' and '.' to
'_' and uppercasing the resulting name. Guard names in the root
directory have to be prefixed by "FFMPEG_".
Originally committed as revision 15120 to svn://svn.ffmpeg.org/ffmpeg/trunk
thread on mailinglist for discussion. This patch also implements a
RTSPServerType enum, which allows the RTSP to keep track of what kind of a
stream we're handling: standard-compliant RTP or a proprietary derivative.
This will be used in subsequent patches to implement more Realmedia-specific
extensions required to receive and parse data coming from a Realmedia server.
Originally committed as revision 15104 to svn://svn.ffmpeg.org/ffmpeg/trunk
one doesn't work, we can try the next one (i.e. trial-error protocol auto-
probing).
Discussed and approved in "[PATCH] RTSP alternate protocol 2-3/3".
Originally committed as revision 12504 to svn://svn.ffmpeg.org/ffmpeg/trunk
