* qatar/master:
mingw/cygwin: Stop adding -fno-common to gcc CFLAGS
Restructure av_log_missing_feature message
rtp: Support packetization/depacketization of opus
file: Set the return value type for lseek to int64_t.
ppc: fix Altivec build with old compilers
build: add LTO support for PGI compiler
build: add -Mdse to PGI optimisation flags
rtpenc_vp8: Update the packetizer to the latest spec version
rtpdec_vp8: Make the depacketizer implement the latest spec draft
doc: allow building with old texi2html versions
avutil: skip old_pix_fmts.h since it is just a list
Conflicts:
libavcodec/aacdec.c
libavcodec/h264.c
libavcodec/ppc/fmtconvert_altivec.c
libavcodec/utils.c
libavformat/file.c
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>
RTPDynamicProtocolHandler for speex is added. Initial support for
speex depacketization from RTP stream comes with it.
Currently, only codec audio rate can be applied based on sdp:
* Narrowband ( 8K)
* Wideband (16K)
* Ultrawideband (32K)
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
unistd.h used to be required for gethostname. On windows, gethostname
is provided by winsock2.h. Now network.h includes both unistd.h and
winsock2.h if they exist.
Signed-off-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (35 commits)
h264_idct_10bit: port x86 assembly to cpuflags.
x86inc: clip num_args to 7 on x86-32.
x86inc: sync to latest version from x264.
fft: rename "z" to "zc" to prevent name collision.
wv: return meaningful error codes.
wv: return AVERROR_EOF on EOF, not EIO.
mp3dec: forward errors for av_get_packet().
mp3dec: remove a pointless local variable.
mp3dec: remove commented out cruft.
lavfi: bump minor to mark stabilizing the ABI.
FATE: add tests for yadif.
FATE: add a test for delogo video filter.
FATE: add a test for amix audio filter.
audiogen: allow specifying random seed as a commandline parameter.
vc1dec: Override invalid macroblock quantizer
vc1: avoid reading beyond the last line in vc1_draw_sprites()
vc1dec: check that coded slice positions and interlacing match.
vc1dec: Do not ignore ff_vc1_parse_frame_header_adv return value
configure: Move parts that should not be user-selectable to CONFIG_EXTRA
lavf: remove commented out cruft in avformat_find_stream_info()
...
Conflicts:
Makefile
configure
libavcodec/vc1dec.c
libavcodec/x86/h264_deblock.asm
libavcodec/x86/h264_deblock_10bit.asm
libavcodec/x86/h264dsp_mmx.c
libavfilter/version.h
libavformat/mp3dec.c
libavformat/utils.c
libavformat/wv.c
libavutil/x86/x86inc.asm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
unistd.h used to be required for gethostname. On windows, gethostname
is provided by winsock2.h. Now network.h includes both unistd.h and
winsock2.h if they exist.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (24 commits)
flvdec: remove incomplete, disabled seeking code
mem: add support for _aligned_malloc() as found on Windows
lavc: Extend the documentation for avcodec_init_packet
flvdec: remove incomplete, disabled seeking code
http: replace atoll() with strtoll()
mpegts: remove unused/incomplete/broken seeking code
af_amix: allow float planar sample format as input
af_amix: use AVFloatDSPContext.vector_fmac_scalar()
float_dsp: add x86-optimized functions for vector_fmac_scalar()
float_dsp: Move vector_fmac_scalar() from libavcodec to libavutil
lavr: Add x86-optimized function for flt to s32 conversion
lavr: Add x86-optimized function for flt to s16 conversion
lavr: Add x86-optimized functions for s32 to flt conversion
lavr: Add x86-optimized functions for s32 to s16 conversion
lavr: Add x86-optimized functions for s16 to flt conversion
lavr: Add x86-optimized function for s16 to s32 conversion
rtpenc: Support packetizing iLBC
rtpdec: Add a depacketizer for iLBC
Implement the iLBC storage file format
mov: Support muxing/demuxing iLBC
...
Conflicts:
Changelog
configure
libavcodec/avcodec.h
libavcodec/dsputil.c
libavcodec/version.h
libavformat/movenc.c
libavformat/mpegts.c
libavformat/version.h
libavutil/mem.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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:
tiertexseq: set correct block_align for audio
tiertexseq: set audio stream start time to 0
voc/avs: Do not change the sample rate mid-stream.
segafilm: use the sample rate as the time base for audio streams
ea: fix audio pts
psx-str: fix audio pts
vqf: set packet duration
tta demuxer: set packet duration
mpegaudio_parser: do not ignore information from the first parsed frame
mpegaudio_parser: be less picky about the start position
thp: set audio packet durations
avcodec: add a Vorbis parser to get packet duration
vorbisdec: read the previous window flag for long windows
lavc: free the output packet when encoding failed or produced no output.
lavc: preserve avpkt->destruct in ff_alloc_packet().
lavc: clarify the meaning of AVCodecContext.frame_number.
mpegts: Pad the packet buffer in handle_packet().
mpegts: Do not call read_sl_header() when no bytes remain in the buffer.
Conflicts:
libavcodec/mpegaudio_parser.c
libavcodec/version.h
libavformat/mpegts.c
tests/ref/fate/pva-demux
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>
* qatar/master: (21 commits)
CDXL demuxer and decoder
hls: Re-add legacy applehttp name to preserve interface compatibility.
hlsproto: Rename the functions and context
hlsproto: Encourage users to try the hls demuxer instead of the proto
doc: Move the hls protocol section into the right place
libavformat: Rename the applehttp protocol to hls
hls: Rename the functions and context
libavformat: Rename the applehttp demuxer to hls
rtpdec: Support H263 in RFC 2190 format
rv30: check block type validity
ttadec: CRC checking
movenc: Support muxing VC1
avconv: Don't split out inline sequence headers when stream copying VC1
rv34: handle size changes during frame multithreading
rv40: prevent undefined signed overflow in rv40_loop_filter()
rv34: use AVERROR return values in ff_rv34_decode_frame()
rv34: use uint16_t for RV34DecContext.deblock_coefs
librtmp: Add "lib" prefix to librtmp URLProtocol declarations.
movenc: Use defines instead of hardcoded numbers for RTCP types
smjpegdec: implement seeking
...
Conflicts:
Changelog
doc/general.texi
libavcodec/avcodec.h
libavcodec/rv30.c
libavcodec/tta.c
libavcodec/version.h
libavformat/Makefile
libavformat/allformats.c
libavformat/version.h
libswscale/x86/swscale_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This is different from the "modern" RTP payload formats for H263
as defined by RFC 4629, 2429 and 3555. According to the newer RFCs,
this old one is to be considered deprecated and only be used for
interoperating with legacy systems.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
rtpdec: Use our own SSRC in the SDES field when sending RRs
Finalize changelog for 0.8 Release
Prepare for 0.8 Release
threads: change the default for threads back to 1
threads: update slice_count and slice_offset from user context
aviocat: Remove useless includes
doc/APIChanges: fill in missing dates and hashes
Revert "avserver: fix build after the next bump."
mpegaudiodec: switch error detection check to AV_EF_BUFFER
lavf: rename fer option and document resulting (f_)err_detect options
lavc: rename err_filter option to err_detect and document it
mpegvideo: fix invalid memory access for small video dimensions
movenc: Reorder entries in the MOVIentry struct, for tigheter packing
rtsp: Remove extern declarations for variables that don't exist
aviocat: Flush the output before closing
Conflicts:
Changelog
RELEASE
libavcodec/mpegaudiodec.c
libavcodec/pthread.c
libavformat/options.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The s->ssrc field is the sender's SSRC, we use ssrc + 1 to get
a collision free "unique" SSRC for ourselves in the RR part.
The SDES block in the RTCP packet should describe ourselves,
not the sender.
This was fixed for the RR part in 952139a322, but wasn't
fixed for the SDES part until now.
This could cause some Axis cameras to send RTCP BYE packets
to us due to the SSRC collision.
Signed-off-by: Martin Storsjö <martin@martin.st>
This requires using a separate init function, since there
isn't necessarily any fmtp lines for this codec, so
parse_sdp_a_line won't be called. Incorporating it with the
alloc function wouldn't do either, since it is called before
the full rtpmap line is parsed (where the sample rate is
extracted).
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (22 commits)
configure: add check for w32threads to enable it automatically
rtmp: do not hardcode invoke numbers
cinepack: return non-generic errors
fate-lavf-ts: use -mpegts_transport_stream_id option.
Add an APIchanges entry and a minor bump for avio changes.
avio: Mark the old interrupt callback mechanism as deprecated
avplay: Set the new interrupt callback
avconv: Set new interrupt callbacks for all AVFormatContexts, use avio_open2() everywhere
cinepak: remove redundant coordinate checks
cinepak: check strip_size
cinepak, simplify, use AV_RB24()
cinepak: simplify, use FFMIN()
cinepak: Fix division by zero, ask for sample if encoded_buf_size is 0
applehttp: Fix seeking in streams not starting at DTS=0
http: Don't use the normal http proxy mechanism for https
tls: Handle connection via a http proxy
http: Reorder two code blocks
http: Add a new protocol for opening connections via http proxies
http: Split out the non-chunked buffer reading part from http_read
segafilm: add support for raw videos
...
Conflicts:
avconv.c
configure
doc/APIchanges
libavcodec/cinepak.c
libavformat/applehttp.c
libavformat/version.h
tests/lavf-regression.sh
tests/ref/lavf/ts
Merged-by: Michael Niedermayer <michaelni@gmx.at>
RTCP timestamps are only necessary to synchronize time between
multiple streams. For a single stream, the RTP packet timestamp
provides more reliable timing. As a result, single-stream RTP
sessions should now have accurate and monotonic PTS.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
vble: remove vble_error_close
VBLE Decoder
tta: use an integer instead of a pointer to iterate output samples
shorten: do not modify samples pointer when interleaving
mpc7: only support stereo input.
dpcm: do not try to decode empty packets
dpcm: remove unneeded buf_size==0 check.
twinvq: add SSE/AVX optimized sum/difference stereo interleaving
vqf/twinvq: pass vqf COMM chunk info in extradata
vqf: do not set bits_per_coded_sample for TwinVQ.
twinvq: check for allocation failure in init_mdct_win()
swscale: add padding to conversion buffer.
rtpdec: Simplify finalize_packet
http: Handle proxy authentication
http: Print an error message for Authorization Required, too
AVOptions: don't return an invalid option when option list is empty
AIFF: add 'twos' FourCC for the mux/demuxer (big endian PCM audio)
Conflicts:
libavcodec/avcodec.h
libavcodec/tta.c
libavcodec/vble.c
libavcodec/version.h
libavutil/opt.c
libswscale/utils.c
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>
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>
Previous code could read 4 bytes past the end of the buffer on a RTCP_SR
packet or offset a pointer by an unchecked external value (payload_len),
though neither will reliably cause a crash or other misbehavior beyond
garbage timestamps.
Additionally, unknown RTCP packet types, even in compounded packets, are
now ignored as per RFC 3550 section 6.1, page 22, though currently this
only has any practical effect if a sender puts an unrecognized type
before RTCP_BYE in a compounded packet, or (incorrectly) does not put
RTCP_SR first.
Signed-off-by: John Brooks <john.brooks@bluecherry.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This allows skipping past unsupported RTCP packet types, as
RFC 3550 section 6.1 mandates.
Currently this only has any practical effect if a sender puts
an unrecognized type before RTCP_BYE in a compounded packet, or
(incorrectly) does not put RTCP_SR first.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
dnxhddec: optimise dnxhd_decode_dct_block()
rtp: remove disabled code
eac3enc: use different numbers of blocks per frame to allow higher bitrates
dnxhd: add regression test for 10-bit
dnxhd: 10-bit support
dsputil: update per-arch init funcs for non-h264 high bit depth
dsputil: template get_pixels() for different bit depths
dsputil: create 16/32-bit dctcoef versions of some functions
jfdctint: add 10-bit version
mov: add clcp type track as Subtitle stream.
mpeg4: add Mpeg4 Profiles names.
mpeg4: decode Level Profile for MPEG4 Part 2.
ffprobe: display bitstream level.
imgconvert: remove unused glue and xglue macros
Conflicts:
libavcodec/dsputil_template.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: (21 commits)
build: simplify commands for clean target
swscale: split swscale.c in unscaled and generic conversion routines.
swscale: cosmetics.
swscale: integrate (literally) swscale_template.c in swscale.c.
swscale: split out x86/swscale_template.c from swscale.c.
swscale: enable hScale_altivec_real.
swscale: split out ppc _template.c files from main swscale.c.
swscale: remove indirections in ppc/swscale_template.c.
swscale: split out unscaled altivec YUV converters in their own file.
mpegvideoenc: fix multislice fate tests with threading disabled.
mpegts: Wrap #ifdef DEBUG and av_hex_dump_log() combination in a macro.
build: Simplify texi2html invocation through the --output option.
Mark some variables with av_unused
Replace avcodec_get_pix_fmt_name() by av_get_pix_fmt_name().
svq3: Check negative mb_type to fix potential crash.
svq3: Move svq3-specific fields to their own context.
rawdec: initialize return value to 0.
Remove unused get_psnr() prototype
rawdec: don't leak option strings.
bktr: get default framerate from video standard.
...
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Most of these variables are only used in av_dlog statements, some
are required but not used by other macros.
Signed-off-by: Mans Rullgard <mans@mansr.com>
