* 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>
Add a new option 'rtmp_flush_interval' that allows specifying the
number of packets to write before sending it off as a HTTP request.
This is mostly relevant for RTMPT - for plain RTMP, it only controls
how often we check the socket for incoming packets, which shouldn't
affect the performance in any noticeable way.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
lavr: add x86-optimized functions for mixing 1-to-2 s16p with flt coeffs
lavr: add x86-optimized functions for mixing 1-to-2 fltp with flt coeffs
Add Dolby/DPLII downmix support to libavresample
vorbisdec: replace div/mod in loop with a counter
fate: vorbis: add 5.1 surround test
rtpenc: Allow requesting H264 RTP packetization mode 0
configure: Sort the library listings in the help text alphabetically
dwt: remove variable-length arrays
RTMPT protocol support
http: Properly handle chunked transfer-encoding for replies to post data
http: Fail reading if the connection has gone away
amr: Mark an array const
amr: More space cleanup
rtpenc: Fix memory leaks in the muxer open function
Conflicts:
Changelog
configure
doc/APIchanges
libavformat/version.h
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
RTMPT protocol implementation uses rtmphttp:// 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:
Avoid C99 variable declarations within for statements.
rtmp: Read and handle incoming packets while writing data
doc: document THREAD_TYPE fate variable
rtpdec: Don't require frames to start with a Mode A packet
avconv: don't try to free threads that were not initialized.
Conflicts:
doc/fate.texi
ffplay.c
libavdevice/dv1394.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This makes sure all incoming packets are read and handled (and reacted
to) while sending an FLV stream over RTMP to a server. If there were
enough incoming data to fill the TCP buffers, this could potentially
make things block at unexpected places. For the upcoming RTMPT support,
we need to consume all incoming data before we can send the next
request.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
rtmp: Add a new option 'rtmp_buffer', for setting the client buffer time
rtmp: Set the client buffer time to 3s instead of 0.26s
rtmp: Handle server bandwidth packets
rtmp: Display a verbose message when an unknown packet type is received
lavfi/audio: use av_samples_copy() instead of custom code.
configure: add all filters hardcoded into avconv to avconv_deps
avfiltergraph: remove a redundant call to avfilter_get_by_name().
lavfi: allow building without swscale.
build: Do not delete tests/vsynth2 directory, which is no longer created.
lavfi: replace AVFilterContext.input/output_count with nb_inputs/outputs
lavfi: make AVFilterPad opaque after two major bumps.
lavfi: add avfilter_pad_get_type() and avfilter_pad_get_name().
lavfi: make avfilter_get_video_buffer() private on next bump.
jack: update to new latency range API as the old one has been deprecated
rtmp: Tokenize the AMF connection parameters manually instead of using strtok_r
ppc: Rename H.264 optimization template file for consistency.
lavfi: add channelsplit audio filter.
golomb: check remaining bits during unary decoding in get_ur_golomb_jpegls()
sws: fix planar RGB input conversions for 9/10/16 bpp.
Conflicts:
Changelog
configure
doc/APIchanges
ffmpeg.c
libavcodec/golomb.h
libavcodec/v210dec.h
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/asrc_anullsrc.c
libavfilter/audio.c
libavfilter/avfilter.c
libavfilter/avfilter.h
libavfilter/avfiltergraph.c
libavfilter/buffersrc.c
libavfilter/formats.c
libavfilter/version.h
libavfilter/vf_frei0r.c
libavfilter/vf_pad.c
libavfilter/vf_scale.c
libavfilter/video.h
libavfilter/vsrc_color.c
libavformat/rtmpproto.c
libswscale/input.c
tests/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This factorizes existing code into a new function gen_buffer_time(),
which generates the client buffer time message and sends it to the
server.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
avfilter: Log an error if avfilter fails to configure a link.
avconv: support only native pthreads.
rtmp: Fix a possible access to invalid memory location when the playpath is too short.
Conflicts:
ffmpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtmp: Do not send extension for flv files
rtmp: support connection parameters
doc: Add documentation for the newly added rtmp_* options
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Allow using connection parameters in order to append arbitrary
AMF data like "B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0" to the
Connect message. You can pass these parameters through the -rtmp_conn
option.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
opt: Add av_opt_set_bin()
avconv: Display the error returned by avformat_write_header
rtpenc_chain: Return an error code instead of just a plain pointer
rtpenc_chain: Free the URLContext on failure
rtpenc: Expose the ssrc as an avoption
avprobe: display the codec profile in show_stream()
avprobe: fix function prototype
cosmetics: Fix indentation
avprobe: changelog entry
avprobe: update documentation
avprobe: provide JSON output
avprobe: output proper INI format
avprobe: improve formatting
rtmp: fix url parsing
fate: document TARGET_EXEC and its usage
Conflicts:
doc/APIchanges
doc/fate.texi
doc/ffprobe.texi
ffprobe.c
libavformat/version.h
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
movenc: Don't write the 'wave' atom or its child 'enda' for lpcm audio.
imc: some cosmetics
rtmp: Pass the proper return code in rtmp_handshake
rtmp: Check return codes of net IO operations
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtmp: Return a proper error code instead of -1
rtmp: Check malloc calls
rtmp: Check ff_rtmp_packet_create calls
lavfi: add audio mix filter
flvdec: Make sure sample_rate is set to the updated value
tqi: Pass errors from the MB decoder
Conflicts:
Changelog
doc/filters.texi
libavcodec/eatqi.c
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
doc: Replace some @file tags by more suitable markup.
fate: Set FUZZ factor of vorbis-13 test to 2.
fate: Set FUZZ factor of (e)ac3-encode test to 3.
fate: remove unused code from regressions-funcs.sh
rtmp: Don't assume path points to a string of nonzero length
avconv: fix behavior with -ss as an output option.
Conflicts:
doc/platform.texi
doc/protocols.texi
ffmpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If using the new -rtmp_app and -rtmp_playpath parameters,
one can in many cases set the main url to just rtmp://server/.
If the trailing slash is omitted, path is a string of zero length,
and using path+1 will end up reading uninitialized data.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (25 commits)
rv40dsp x86: MMX/MMX2/3DNow/SSE2/SSSE3 implementations of MC
ape: Use unsigned integer maths
arm: dsputil: fix overreads in put/avg_pixels functions
h264: K&R formatting cosmetics for header files (part II/II)
h264: K&R formatting cosmetics for header files (part I/II)
rtmp: Implement check bandwidth notification.
rtmp: Support 'rtmp_swfurl', an option which specifies the URL of the SWF player.
rtmp: Support 'rtmp_flashver', an option which overrides the version of the Flash plugin.
rtmp: Support 'rtmp_tcurl', an option which overrides the URL of the target stream.
cmdutils: Add fallback case to switch in check_stream_specifier().
sctp: be consistent with socket option level
configure: Add _XOPEN_SOURCE=600 to Solaris preprocessor flags.
vcr1enc: drop pointless empty encode_init() wrapper function
vcr1: drop pointless write-only AVCodecContext member from VCR1Context
vcr1: group encoder code together to save #ifdefs
vcr1: cosmetics: K&R prettyprinting, typos, parentheses, dead code, comments
mov: make one comment slightly more specific
lavr: replace the SSE version of ff_conv_fltp_to_flt_6ch() with SSE4 and AVX
lavfi: move audio-related functions to a separate file.
lavfi: remove some audio-related function from public API.
...
Conflicts:
cmdutils.c
libavcodec/h264.h
libavcodec/h264_mvpred.h
libavcodec/vcr1.c
libavfilter/avfilter.c
libavfilter/avfilter.h
libavfilter/defaults.c
libavfilter/internal.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
According to the behaviour of librtmp, it is recommended to send this
message to the server after receiving the 'onBWDone' callback in order
to do bandwidth checking and improve compatibility with some servers.
* qatar/master:
rtmp: Support 'rtmp_live', an option which specifies if the media is a live stream.
av_samples_fill_array: Mark unmodified function argument as const.
lagarith: add YUY2 decoding support
Support decoding unaligned rgb24 lagarith.
dv: Split profile handling code into a separate file.
flvenc: use AVFormatContext, not AVCodecContext for logging.
mov: Remove write-only variable in mov_read_chan().
fate: Change the probe-format refs to match the final text format committed.
fate: Add avprobe as a make dependency
Add probe fate tests to test for regressions in detecting media types.
fate: Add oneline comparison method
qdm2: clip array indices returned by qdm2_get_vlc().
avplay: properly close/reopen AVAudioResampleContext on channel layout change
avcodec: do not needlessly set packet size to 0 in avcodec_encode_audio2()
avcodec: for audio encoding, reset output packet when it is not valid
avcodec: refactor avcodec_encode_audio2() to merge common branches
avcodec: remove fallbacks for AVCodec.encode() in avcodec_encode_audio2()
Conflicts:
ffplay.c
libavcodec/Makefile
libavcodec/dvdata.c
libavcodec/dvdata.h
libavcodec/qdm2.c
libavcodec/utils.c
libavformat/flvenc.c
libavformat/mov.c
tests/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
aacenc: Fix issues with huge values of bit_rate.
dv_tablegen: Drop unnecessary av_unused attribute from dv_vlc_map_tableinit().
proresenc: multithreaded quantiser search
riff: use bps instead of bits_per_coded_sample in the WAVEFORMATEXTENSIBLE header
avconv: only set the "channels" option when it exists for the specified input format
avplay: update get_buffer to be inline with avconv
aacdec: More robust output configuration.
faac: Fix multi-channel ordering
faac: Add .channel_layouts
rtmp: Support 'rtmp_playpath', an option which overrides the stream identifier
rtmp: Support 'rtmp_app', an option which overrides the name of application
avutil: add better documentation for AVSampleFormat
Conflicts:
libavcodec/aac.h
libavcodec/aacdec.c
libavcodec/aacenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This option is the stream identifier to play or to publish.
Sometimes the URL parser cannot determine the correct
playpath automatically, so it must be given explicitly
using this option (ie. -rtmp_playpath).
Signed-off-by: Martin Storsjö <martin@martin.st>
This option is the name of application to connect on the RTMP server.
Sometimes the URL parser cannot determine the app name automatically,
so it must be given explicitly using this option (ie. -rtmp_app).
Signed-off-by: Martin Storsjö <martin@martin.st>
The audio codecs property is composed by all values except
SUPPORT_SND_INTEL (0x0008) and SUPPORT_SND_UNUSED (0x0010) which are
unused.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* qatar/master: (21 commits)
ipmovie: do not read audio packets before the codec is known
truemotion2: check size before GetBitContext initialisation
avio: Only do implicit network initialization for network protocols
avio: Add an URLProtocol flag for indicating that a protocol uses network
adpcm: ADPCM Electronic Arts has always two channels
matroskadec: Fix a bug where a pointer was cached to an array that might later move due to a realloc()
fate: Add missing reference file from 9b4767e4.
mov: Support MOV_CH_LAYOUT_USE_DESCRIPTIONS for labeled descriptions.
4xm: Prevent buffer overreads.
mjpegdec: parse RSTn to prevent skipping other data in mjpeg_decode_scan
vp3: add fate test for non-zero last coefficient
vp3: fix streams with non-zero last coefficient
swscale: remove unused U/V arguments from yuv2rgb_write().
timer: K&R formatting cosmetics
lavf: cosmetics, reformat av_read_frame().
lavf: refactor av_read_frame() to make it easier to understand.
Report an error if pitch_lag is zero in AMR-NB decoder.
Revert "4xm: Prevent buffer overreads."
4xm: Prevent buffer overreads.
4xm: pass the correct remaining buffer size to decode_i2_frame().
...
Conflicts:
libavcodec/4xm.c
libavcodec/mjpegdec.c
libavcodec/truemotion2.c
libavformat/ipmovie.c
libavformat/mov_chan.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This definition is in two files, since the definitions will move
to the private header at the next bump.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
x86: cabac: replace explicit memory references with "m" operands
avplay: don't request a stereo downmix
wmapro: use av_float2int()
lavc: avoid invalid memcpy() in avcodec_default_release_buffer()
lavu: replace int/float punning functions
lavfi: install libavfilter/vsrc_buffer.h
Remove extraneous semicolons
sdp: Restore the original mp4 format h264 extradata if converted
rtpenc: Add support for mp4 format h264
rtpenc: Simplify code by introducing a separate end pointer
movenc: Use the actual converted sample for RTP hinting
Fix a bunch of common typos.
Conflicts:
doc/developer.texi
doc/eval.texi
doc/filters.texi
doc/protocols.texi
ffmpeg.c
ffplay.c
libavcodec/mpegvideo.h
libavcodec/x86/cabac.h
libavfilter/Makefile
libavformat/avformat.h
libavformat/cafdec.c
libavformat/flvdec.c
libavformat/flvenc.c
libavformat/gxfenc.c
libavformat/img2.c
libavformat/movenc.c
libavformat/mpegts.c
libavformat/rtpenc_h264.c
libavformat/utils.c
libavformat/wtv.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The existing functions defined in intfloat_readwrite.[ch] are
both slow and incorrect (infinities are not handled).
This introduces a new header with fast, inline conversion
functions using direct union punning assuming an IEEE-754
system, an assumption already made throughout the code.
The one use of Intel/Motorola extended 80-bit format is
replaced by simpler code sufficient under the present
constraints (positive normal values).
The old functions are marked deprecated and retained for
compatibility.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
drawtext: remove typo
pcm-mpeg: implement new audio decoding api
w32thread: port fixes to pthread_cond_broadcast() from x264.
doc: add editor configuration section with Vim and Emacs settings
dxva2.h: include d3d9.h to define LPDIRECT3DSURFACE9
avformat/utils: Drop unused goto label.
doxygen: Replace '\' by '@' in Doxygen markup tags.
cosmetics: drop some completely pointless parentheses
cljr: simplify CLJRContext
drawtext: introduce rand(min, max)
drawtext: introduce explicit draw/hide variable
rtmp: Use nb_invokes for all invoke commands
Conflicts:
libavcodec/mpegvideo.c
libavfilter/vf_drawtext.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
704af3e29c broke publishing
of rtmp streams, at least publishing to Wowza servers.
This changes all invoke commands to use nb_invokes.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (25 commits)
rtpenc: Add support for G726 audio
rtpdec: Interpret the different G726 names as bits_per_coded_sample
rtpenc: Change rtp_send_samples to handle sample sizes other than even bytes
rtpenc: Cast a rescaling parameter to int64_t
h264: cap max has_b_frames at MAX_DELAYED_PIC_COUNT - 1.
ARM: fix indentation in ff_dsputil_init_neon()
ARM: NEON put/avg_pixels8/16 cosmetics
ARM: add remaining NEON avg_pixels8/16 functions
ARM: clean up NEON put/avg_pixels macros
fate: split acodec-pcm into individual tests
swscale: #include "libavutil/mathematics.h"
pmpdec: don't use deprecated av_set_pts_info.
rv34: align temporary block of "dct" coefs
Add PlayStation Portable PMP format demuxer
proto: Realign struct initializers
proto: Use .priv_data_size to allocate the private context
mmsh: Properly clean up if the second ffurl_alloc failed
rtmp: Clean up properly if the handshake failed
md5proto: Remove the get_file_handle function
applehttpproto: Use the close function if the open function fails
...
Conflicts:
libavcodec/vble.c
libavformat/mmsh.c
libavformat/pmpdec.c
libavformat/udp.c
tests/ref/acodec/pcm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
