* qatar/master:
avcodec_default_reget_buffer(): fix compilation in DEBUG mode
fate: Overhaul WavPack coverage
h264: fix mmxext chroma deblock to use correct TC values.
flvdec: Remove the now redundant check for known broken metadata creator
flvdec: Validate index entries added from metadata while reading
rtsp: Handle requests from server to client
movenc: use timestamps instead of frame_size for samples-per-packet
movenc: use the first cluster duration as the tfhd default duration
movenc: factorize calculation of cluster duration into a separate function
doc/APIchanges: fill in missing dates and hashes.
lavc: reorder AVCodecContext fields.
lavc: reorder AVFrame fields.
Conflicts:
doc/APIchanges
libavcodec/avcodec.h
libavformat/flvdec.c
libavformat/movenc.c
tests/fate/lossless-audio.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This returns 200 OK for OPTIONS requests and 501 Not Implemented
for all other requests.
Even though this doesn't do much actual handling of the requests,
it makes the code properly identify server requests as such, instead
of interpreting it as a reply to the client's request as it did
before.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (40 commits)
swf: check return values for av_get/new_packet().
wavpack: Don't shift minclip/maxclip
rtpenc: Expose the max packet size via an avoption
rtpenc: Move max_packet_size to a context variable
rtpenc: Add an option for not sending RTCP packets
lavc: drop encode() support for video.
snowenc: switch to encode2().
snowenc: don't abuse input picture for storing information.
a64multienc: switch to encode2().
a64multienc: don't write into output buffer when there's no output.
libxvid: switch to encode2().
tiffenc: switch to encode2().
tiffenc: properly forward error codes in encode_frame().
lavc: drop libdirac encoder.
gifenc: switch to encode2().
libvpxenc: switch to encode2().
flashsvenc: switch to encode2().
Remove libpostproc.
lcl: don't overwrite input memory.
swscale: take first/lastline over/underflows into account for MMX.
...
Conflicts:
.gitignore
Makefile
cmdutils.c
configure
doc/APIchanges
libavcodec/Makefile
libavcodec/allcodecs.c
libavcodec/libdiracenc.c
libavcodec/libxvidff.c
libavcodec/qtrleenc.c
libavcodec/tiffenc.c
libavcodec/utils.c
libavformat/mov.c
libavformat/movenc.c
libpostproc/Makefile
libpostproc/postprocess.c
libpostproc/postprocess.h
libpostproc/postprocess_altivec_template.c
libpostproc/postprocess_internal.h
libpostproc/postprocess_template.c
libswscale/swscale.c
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
shorten: Use separate pointers for the allocated memory for decoded samples.
atrac3: Fix crash in tonal component decoding.
ws_snd1: Fix wrong samples counts.
movenc: Don't set a default sample duration when creating ismv
rtp: Factorize the check for distinguishing RTCP packets from RTP
golomb: avoid infinite loop on all-zero input (or end of buffer).
bethsoftvid: synchronize video timestamps with audio sample rate
bethsoftvid: add audio stream only after getting the first audio packet
bethsoftvid: Set video packet duration instead of accumulating pts.
bethsoftvid: set packet key frame flag for audio and I-frame video packets.
bethsoftvid: fix read_packet() return codes.
bethsoftvid: pass palette in side data instead of in a separate packet.
sdp: Ignore RTCP packets when autodetecting RTP streams
proresenc: initialise 'sign' variable
mpegaudio: replace memcpy by SIMD code
vc1: prevent using last_frame as a reference for I/P first frame.
Conflicts:
libavcodec/atrac3.c
libavcodec/golomb.h
libavcodec/shorten.c
libavcodec/ws-snd1.c
tests/ref/fate/bethsoft-vid
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The rtp demuxer which listens for RTP packets and detects the
RTP payload type will currently get confused if the first packet
received is an RTCP packet. Thus ignore such packets.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (26 commits)
avconv: deprecate the -deinterlace option
doc: Fix the name of the new function
aacenc: make sure to encode enough frames to cover all input samples.
aacenc: only use the number of input samples provided by the user.
wmadec: Verify bitstream size makes sense before calling init_get_bits.
kmvc: Log into a context at a log level constant.
mpeg12: Pad framerate tab to 16 entries.
kgv1dec: Increase offsets array size so it is large enough.
kmvc: Check palsize.
nsvdec: Propagate errors
nsvdec: Be more careful with av_malloc().
nsvdec: Fix use of uninitialized streams.
movenc: cosmetics: Get rid of camelCase identifiers
swscale: more generic check for planar destination formats with alpha
doc: Document mov/mp4 fragmentation options
build: Use order-only prerequisites for creating FATE reference file dirs.
x86 dsputil: provide SSE2/SSSE3 versions of bswap_buf
rtsp: Remove some unused variables from ff_rtsp_connect().
avutil: make intfloat api public
avformat_write_header(): detail error message
...
Conflicts:
doc/APIchanges
doc/ffmpeg.texi
doc/muxers.texi
ffmpeg.c
libavcodec/kmvc.c
libavcodec/x86/Makefile
libavcodec/x86/dsputil_yasm.asm
libavcodec/x86/pngdsp-init.c
libavformat/movenc.c
libavformat/movenc.h
libavformat/mpegtsenc.c
libavformat/nsvdec.c
libavformat/utils.c
libavutil/avutil.h
libswscale/swscale.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
aacenc: Fix LONG_START windowing.
aacenc: Fix a bug where deinterleaved samples were stored in the wrong place.
avplay: use the correct array size for stride.
lavc: extend doxy for avcodec_alloc_context3().
APIchanges: mention avcodec_alloc_context()/2/3
avcodec_align_dimensions2: set only 4 linesizes, not AV_NUM_DATA_POINTERS.
aacsbr: ARM NEON optimised sbrdsp functions
aacsbr: align some arrays
aacsbr: move some simdable loops to function pointers
cosmetics: Remove extra newlines at EOF
Conflicts:
libavcodec/utils.c
libavfilter/formats.c
libavutil/mem.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (25 commits)
riff: fix invalid av_freep() calls on EOF in ff_read_riff_info
pam: Fix a typo that broke writing and reading PAM files.
mxfdec: fix memleak on av_realloc failures
mxfdec: Do not parse slices or DeltaEntryArrays.
mxfdec: hybrid demuxing/seeking solution
mxfdec: Add Avid's essence element key.
mfxdec: Separate mxf_essence_container_uls for audio and video.
mxfdec: Compute packet offsets properly.
mxfdec: Use MaterialPackage - Track - TrackID instead of the system_item hack.
mxfdec: use av_dlog() for 'no corresponding source package found'
mxfdec: Make mxf->partitions sorted by offset.
mxfdec: parse ThisPartition
mxfdec: Speed up metadata and index parsing.
mxfdec: Make sure DataDefinition is consistent between material track and source track.
mxfdec: add EssenceContainer UL found in 0001GL00.MXF.A1.mxf_opatom.mxf
mxfdec: Add hack that adjusts the n_delta calculation when system items are present.
mxfdec: Parse IndexTableSegments and convert them into AVIndexEntry arrays.
mxfdec: Move FooterPartition to MXFContext and make sure it is never zero.
mxfdec: check return value of avio_seek
mxfdec: skip to end of structural sets
...
Conflicts:
configure
libavcodec/pnm.c
libavformat/mxfdec.c
libavformat/riff.c
libavformat/rtsp.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This avoids (for all practical cases) the issue of reusing
the same UDP port as for an earlier connection. If the remote
doesn't know the previous session was closed, he might keep
on sending packets to that port. If we always start off trying
to open the same UDP port, we might get those packets intermixed
with the new ones.
This is occasionally an issue when testing RTSP stuff with
DSS, perhaps also with other servers.
Signed-off-by: Martin Storsjö <martin@martin.st>
This check isn't relevant in the way the code currently works.
Also change a case of if (x == 0) into if (!x).
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
aac_latm: reconfigure decoder on audio specific config changes
latmdec: fix audio specific config parsing
Add avcodec_decode_audio4().
avcodec: change number of plane pointers from 4 to 8 at next major bump.
Update developers documentation with coding conventions.
svq1dec: avoid undefined get_bits(0) call
ARM: h264dsp_neon cosmetics
ARM: make some NEON macros reusable
Do not memcpy raw video frames when using null muxer
fate: update asf seektest
vp8: flush buffers on size changes.
doc: improve general documentation for MacOSX
asf: use packet dts as approximation of pts
asf: do not call av_read_frame
rtsp: Initialize the media_type_mask in the rtp guessing demuxer
Cleaned up alacenc.c
Conflicts:
doc/APIchanges
doc/developer.texi
libavcodec/8svx.c
libavcodec/aacdec.c
libavcodec/ac3dec.c
libavcodec/avcodec.h
libavcodec/nellymoserdec.c
libavcodec/tta.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/wmadec.c
libavformat/asfdec.c
tests/ref/seek/lavf_asf
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The media_type_mask is initialized via AVOptions for the
rtsp and sdp demuxers, but it isn't available as an option
for the rtp guessing demuxer (since it doesn't really make
sense there). Therefore, it must be manually initialized
instead, since a zero value means no media types at all
are accepted.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
rtpdec: Templatize the code for different g726 bitrate variants
rv40: move loop filter to rv34dsp context
lavf: make av_set_pts_info private.
rtpdec: Add support for G726 audio
rtpdec: Add an init function that can do custom codec context initialization
avconv: make copy_tb on by default.
matroskadec: don't set codec timebase.
rmdec: don't set codec timebase.
avconv: compute next_pts from input packet duration when possible.
lavf: estimate frame duration from r_frame_rate.
avconv: update InputStream.pts in the streamcopy case.
Conflicts:
avconv.c
libavdevice/alsa-audio-dec.c
libavdevice/bktr.c
libavdevice/fbdev.c
libavdevice/libdc1394.c
libavdevice/oss_audio.c
libavdevice/v4l.c
libavdevice/v4l2.c
libavdevice/vfwcap.c
libavdevice/x11grab.c
libavformat/au.c
libavformat/eacdata.c
libavformat/flvdec.c
libavformat/mpegts.c
libavformat/mxfenc.c
libavformat/rtpdec_g726.c
libavformat/wtv.c
libavformat/xmv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
lavf: pass options from AVFormatContext to avio.
avformat: Use avio_open2, pass the AVFormatContext interrupt_callback onwards
avio: add avio_open2, taking an interrupt callback and options
avio: add support for passing options to protocols.
avio: add and use ffurl_protocol_next().
avformat: Pass the interrupt callback on to chained muxers/demuxers
avio: Add an AVIOInterruptCB parameter to ffurl_open/ffurl_alloc
avformat: Use ff_check_interrupt
avio: Add an internal utility function for checking the new interrupt callback
avio: Add AVIOInterruptCB
texi2html: remove stray \n
doc: prettyfy the texi2html documentation
swscale: handle unaligned buffers in yuv2plane1
Conflicts:
libavformat/avformat.h
libavformat/avio.c
libavformat/mov.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (29 commits)
doc: update libavfilter documentation
tls: Use the URLContext as logging context
aes: Avoid illegal read and don't generate more key than we use.
mpc7: Fix memset call in mpc7_decode_frame function
atrac1: use correct context for av_log()
apedec: consume the whole packet when copying to the decoder buffer.
apedec: do not needlessly copy s->samples to nblocks.
apedec: check output buffer size after calculating actual output size
apedec: remove unneeded entropy decoder normalization.
truespeech: use memmove() in truespeech_update_filters()
vorbisdec: remove AVCODEC_MAX_AUDIO_FRAME_SIZE check
vorbisdec: remove unneeded buf_size==0 check
vorbisdec: return proper error codes instead of made-up ones
http: Don't add a Range: bytes=0- header for POST
sunrast: Check for invalid/corrupted bitstream
http: Change the chunksize AVOption into chunked_post
http: Add encoding/decoding flags to the AVOptions
avconv: remove some codec-specific hacks
crypto: add decoding flag to options.
tls: use AVIO_FLAG_NONBLOCK instead of deprecated URL_FLAG_NONBLOCK
...
Conflicts:
doc/libavfilter.texi
libavcodec/atrac1.c
libavcodec/sunrast.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The chunksize internal variable has two different uses - for
reading, it's the amount of data left of the current chunk
(or -1 if the server doesn't send data in chunked mode), where
it's only an internal state variable. For writing, it's used
to decide whether to enable chunked encoding (by default), by
using the value 0, or disable chunked encoding (value -1).
This, while consistent, doesn't make much sense to expose
as an AVOption. This splits the usage of the internal variable
into two variables, chunksize which is used for reading (as
before), and chunked_post which is the user-settable option,
with the values 0 and 1, where 1 is default.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
avformat: Avoid a warning about mixed declarations and code
BMV demuxer and decoder
matroskaenc: Make sure the seekhead struct is freed even on seek failure
mpeg12enc: Remove write-only variables.
mpeg12enc: Don't set up run-level info for level 0.
msmpeg4: Don't set up run-level info for level 0.
avformat: Warn about using network functions without calling avformat_network_init
avformat: Revise wording
rdt: Set AVFMT_NOFILE on ff_rdt_demuxer
rdt: Check the return value of avformat_open
rtsp: Discard the dynamic handler, if it has an alloc function which failed
dsputil: use cpuflags in x86 versions of vector_clip_int32()
Conflicts:
libavcodec/avcodec.h
libavcodec/version.h
libavformat/Makefile
libavformat/allformats.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
All current usages of it are incompatible with localization.
For example strcasecmp("i", "I") != 0 is possible, but would
break many of the places where it is used.
Instead use our own implementations that always treat the data
as ASCII.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
http: Remove the custom function for disabling chunked posts
rtsp: Disable chunked http post through AVOptions
movdec: Set frame_size for AMR
h264_weight: remove duplication functions.
swscale: align vertical filtersize by 2 on x86.
libavfilter: reindent.
matroskadec: empty blocks are in fact valid.
avfilter: don't abort() on zero-size allocations.
h264: improve calculation of codec delay.
movenc: Set a correct packet size for AMR-NB mode 15, "no data"
avformat: Add functions for doing global network initialization
avformat: Add the https protocol
avformat: Add the tls protocol, using OpenSSL or gnutls
avformat: Initialize gnutls in ff_tls_init()
w32threads: Wrap the mutex functions in inline functions returning int
configure: Allow linking to the gnutls library
avformat: Add ff_tls_init()/deinit() that initialize OpenSSL
configure: Allow linking to openssl
avcodec: Allow locking and unlocking an avformat specific mutex
avformat: Split out functions from network.h to a new file, network.c
Conflicts:
Changelog
configure
doc/APIchanges
libavcodec/internal.h
libavcodec/version.h
libavfilter/formats.c
libavformat/matroskadec.c
libavformat/mov.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
All current usages of it are incompatible with localization.
For example strcasecmp("i", "I") != 0 is possible, but would
break many of the places where it is used.
Instead use our own implementations that always treat the data
as ASCII.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* 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:
presets: rename presets directory
lavc: make avcodec_get_context_defaults3 "officially" public
lavf: replace av_new_stream->avformat_new_stream part II.
lavf,lavd: replace av_new_stream->avformat_new_stream part I.
lavf: add avformat_new_stream as a replacement for av_new_stream.
Use correct scaling table for bwd-pred MVs in second B-field
Ut Video decoder
Makefile: change presets extension to .avpreset
lavfi: add rgbtestsrc source, ported from MPlayer libmpcodecs
lavfi: add testsrc source
AVOptions: add documentation.
presets: update libx264 ffpresets
Conflicts:
Changelog
doc/APIchanges
doc/ffmpeg.texi
ffpresets/libx264-ipod320.ffpreset
ffpresets/libx264-ipod640.ffpreset
ffserver.c
libavcodec/avcodec.h
libavcodec/options.c
libavcodec/version.h
libavdevice/libdc1394.c
libavfilter/avfilter.h
libavfilter/vsrc_testsrc.c
libavformat/flvdec.c
libavformat/riff.c
libavformat/version.h
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Manual replacements are done in this commit.
In many cases, the id is some constant made up number (e.g. 0 for video
and 1 for audio), which is then not used in the demuxer for anything.
Those ids are removed.
* 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>
This allows setting the filter_src option for these demuxers, too,
which wasn't possible at all before (where the option only was set
via URL parameters for RTSP).
Signed-off-by: Martin Storsjö <martin@martin.st>
