* qatar/master:
rtsp: Don't use av_malloc(0) if there are no streams
rtsp: Don't use uninitialized data if there are no streams
vaapi: mpeg2: fix slice_vertical_position calculation.
hwaccel: mpeg2: decode first field, if requested.
cosmetics: Fix indentation
rtsp: Don't expose the MS-RTSP RTX data stream to the caller
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This avoids exposing a dummy AVStream which won't get any data
and which will make avformat_find_stream_info wait for info about
this stream.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
asf: only set index_read if the index contained entries.
cabac: add overread protection to BRANCHLESS_GET_CABAC().
cabac: increment jump locations by one in callers of BRANCHLESS_GET_CABAC().
cabac: remove unused argument from BRANCHLESS_GET_CABAC_UPDATE().
cabac: use struct+offset instead of memory operand in BRANCHLESS_GET_CABAC().
h264: add overread protection to get_cabac_bypass_sign_x86().
h264: reindent get_cabac_bypass_sign_x86().
h264: use struct offsets in get_cabac_bypass_sign_x86().
h264: fix overreads in cabac reader.
wmall: fix seeking.
lagarith: fix buffer overreads.
dvdec: drop unnecessary dv_tablegen.h #include
build: fix doc generation errors in parallel builds
Replace memset(0) by zero initializations.
faandct: Remove FAAN_POSTSCALE define and related code.
dvenc: print allowed profiles if the video doesn't conform to any of them.
avcodec_encode_{audio,video}: only reallocate output packet when it has non-zero size.
FATE: add a test for vp8 with changing frame size.
fate: add kgv1 fate test.
oggdec: calculate correct timestamps in Ogg/FLAC
Conflicts:
libavcodec/4xm.c
libavcodec/cook.c
libavcodec/dvdata.c
libavcodec/dvdsubdec.c
libavcodec/lagarith.c
libavcodec/lagarithrac.c
libavcodec/utils.c
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (27 commits)
avconv: free packet in write_frame() when discarding due to frame number limit
FATE: use +/- flag option syntax for vp8 emu-edge tests
lavf: make av_interleave_packet_per_dts() private.
lavf: deprecate av_read_packet().
oggdec: output correct timestamps for Vorbis
avconv: pass input stream timestamps to audio encoders
lavc: shrink encoded audio packet size after encoding.
xa: set correct bit rate
xa: do not set bit_rate, block_align, or bits_per_coded_sample
xa: fix end-of-file handling
xa: fix timestamp calculation
bink: fix typo in FFALIGN() argument
bink: align plane width to 8 when calculating bundle sizes
doc: pass -Idoc texi2html and texi2pod
doc: texi2pod: add -I flag
movenc: Add a min_frag_duration option
rtsp: Set the default delay to 0.1 s for the RTSP/SDP/RTP demuxers
libavformat: Set the default for the max_delay option to -1
Generate manpages for AV{Format,Codec}Context AVOptions.
doc/avconv: remove entries for AVOptions.
...
Conflicts:
doc/Makefile
doc/ffmpeg.texi
doc/muxers.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/options.c
libavcodec/vp8.c
libavformat/options.c
tests/fate/demux.mak
tests/ref/fate/truemotion1-15
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Make the muxers/demuxers that use the field handle the default
-1 in the same way as 0.
This allows distinguishing an intentionally set 0 from the default
value where the user hasn't set it.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (35 commits)
fix space type in Changelog
ZeroCodec Decoder
RealAudio Lossless decoder
rtpenc: Use AVFormatContext.packet_size instead of a private option
url: Document the expected behaviour of url_read
libavformat: Use AVFormatContext.probesize in init_input
docs: Fix a stray reference to tags in the generic doxy on dicts
cosmetics: Align some AVInput/OutputFormat declarations
zmbv: check decompress result
zmbv: correct indentation
adpcm: convert adpcm_thp to bytestream2.
adpcm: convert adpcm_yamaha to bytestream2.
adpcm: convert adpcm_swf to bytestream2.
adpcm: convert adpcm_sbpro to bytestream2.
adpcm: convert adpcm_ct to bytestream2.
adpcm: convert adpcm_ima_amv/smjpeg to bytestream2.
adpcm: convert adpcm_ea_xas to bytestream2.
adpcm: convert adpcm_ea_r1/2/3 to bytestream2.
adpcm: convert ea_maxis_xa to bytestream2.
adpcm: convert adpcm_ea to bytestream2.
...
Conflicts:
Changelog
libavcodec/Makefile
libavcodec/adpcm.c
libavcodec/allcodecs.c
libavcodec/avcodec.h
libavcodec/version.h
libavcodec/zerocodec.c
libavcodec/zmbv.c
libavformat/riff.c
libavformat/url.h
tests/ref/fate/truemotion1-15
tests/ref/fate/truemotion1-24
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
doc/general: update supported devices table.
doc/general: add missing @tab to codecs table.
h264: Fix invalid interlaced/progressive MB combinations for direct mode prediction.
avconv: reindent
avconv: link '-passlogfile' option to libx264 'stats' AVOption.
libx264: add 'stats' private option for setting 2pass stats filename.
libx264: fix help text for slice-max-size option.
http: Clear the auth state on redirects
http: Retry auth if it failed due to being stale
rtsp: Resend new keepalive commands if they used stale auth
rtsp: Retry authentication if failed due to being stale
httpauth: Parse the stale field in digest auth
dxva2_vc1: pass the overlap flag to the decoder
dxva2_vc1: fix decoding of BI frames
FATE: add shorthand to wavpack test
dfa: convert to bytestream2 API
anm decoder: move buffer allocation from decode_init() to decode_frame()
h264: improve parsing of broken AVC SPS
Conflicts:
ffmpeg.c
libavcodec/anm.c
libavcodec/dfa.c
libavcodec/h264.c
libavcodec/h264_direct.c
libavcodec/h264_ps.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
pcm-mpeg: convert to bytestream2 API
Revert "h264: clear trailing bits in partially parsed NAL units"
remove iwmmxt optimizations
mimic: do not continue if swap_buf_size is 0
mimic: convert to bytestream2 API
frwu: use MKTAG to check marker instead of AV_RL32
txd: port to bytestream2 API
c93: convert to bytestream2 API
iff: make .long_name more descriptive
FATE: add test for cdxl demuxer
rtsp: Fix a typo
Conflicts:
libavcodec/arm/dsputil_iwmmxt.c
libavcodec/arm/dsputil_iwmmxt_rnd_template.c
libavcodec/arm/mpegvideo_iwmmxt.c
libavcodec/c93.c
libavcodec/txd.c
libavutil/arm/cpu.c
tests/fate/demux.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
Fix a bunch of common typos.
build: Skip compiling xvmc.h under the correct condition.
configure: darwin: Change dylib install names to include major version.
mpegts: Always honor a registration descriptor if present and there is no other codec information.
aacdec: Fix SCE parity check.
aacdec: Fix out of array writes (stack).
rtsp: Only set the ttl parameter if the server actually gave a value
udp: Set ttl for read-write streams, too, not only for write-only ones
udp: Only bind to the multicast address if in read-only mode
udp: Clarify the comment about binding the multicast address
udp: Reorder comments
Conflicts:
libavcodec/aacdec.c
tools/patcheck
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
