* commit 'd4aef997809167832ecc64e89dda8cb445e5fe10':
rtmp: Follow Flash player numbering for channels.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'fe0337e89bbbe84b7274fbb0d9d56ed992937931':
rtmp: Do not send the first field twice within the handshake
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This more closely corresponds to the usage of the field.
Its usage here is unrelated to the channel ID.
Signed-off-by: Martin Storsjö <martin@martin.st>
Channel 4 is typically used by the Flash player to transmit
audio, channel 6 for video, and various stream-specific invokes
get sent over channel 8, which is designated the source channel.
This more closely matches the behavior of the Flash player,
including the transmission of play requests over channel 8.
Signed-off-by: Martin Storsjö <martin@martin.st>
Sending non-monotonic packets (e.g. when the audio and video
streams are monotonic within themselves but not muxed
monotonically) will lead to negative values the RTMP timestamp
field (where timestamps are transmitted only as deltas for each
channel), and this delta can end up being incorrectly written as
a large unsigned number.
Signed-off-by: Martin Storsjö <martin@martin.st>
When streaming to limelight, the app name is either a full
"appname/subaccount" or "appname/_definst_". In the latter case,
the app name can be simplified into simply "appname", but the
authentication hashing assumes the /_definst_ still to be present.
Signed-off-by: Martin Storsjö <martin@martin.st>
* commit '8e1fe345577a42f99591caf8a06c447613449694':
rtmp: Detect and warn if the user tries to pass librtmp style parameters
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '28306e6d620c109ddd672f7243adfbc2bbb3b18f':
network: factor out bind-listening code
use my full first name instead of short one in copyrights
Conflicts:
libavformat/tcp.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'b3ea76624ad1baab0b6bcc13f3f856be2f958110':
vf_aspect: use the name 's' for the pointer to the private context
Remove commented-out debug #define cruft
Conflicts:
libavcodec/4xm.c
libavcodec/dvdsubdec.c
libavcodec/ituh263dec.c
libavcodec/mpeg12.c
libavfilter/avfilter.c
libavfilter/vf_aspect.c
libavfilter/vf_fieldorder.c
libavformat/rtmpproto.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
do_adobe_auth takes the parameters in the order "opaque, challenge".
Due to the way they are treated, this didn't matter in the tested
setups though - if both are set, we only use one. In the tested
setups (Wowza and Akamai) either one of them were null or they
were both set to the same value, which is why this worked before.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
rtmp: Add support for limelight authentication
rtmp: Add support for adobe authentication
Conflicts:
Changelog
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '33f28a3be3092f642778253d9529dd66fe2a014a':
rtmp: Add a function for writing AMF strings based on two substrings
rtmp: Return a proper error code in handle_invoke_error
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Limelight is a not too uncommon CDN. The authentication scheme is
pretty similar to the adobe authentication, but is even closer to
normal http digest authentication (but not close enough to warrant
sharing code) than the adobe version.
Signed-off-by: Martin Storsjö <martin@martin.st>
This is mostly used to authenticate the client when publishing.
Tested with wowza and akamai.
Some but not all servers support resending a new connect invoke
within the same connection, so always reconnect for sending a new
connection attempt. This matches what other applications do as well.
The authentication scheme is structurally pretty similar to http
digest authentication, but uses base64 instead of hex strings.
Signed-off-by: Martin Storsjö <martin@martin.st>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Fixes: [FFmpeg-devel] rtmpproto compile error
Similar patch: [FFmpeg-devel] [PATCH] call to strncat replaced with av_strlcat to avoid compile issue with systems implementing strncat via strcat.
* commit 'e002e3291e6dc7953f843abf56fc14f08f238b21':
Use the new aes/md5/sha/tree allocation functions
avutil: Add functions for allocating opaque contexts for algorithms
svq3: fix pointer type warning
svq3: replace unsafe pointer casting with intreadwrite macros
parseutils-test: various cleanups
Conflicts:
doc/APIchanges
libavcodec/svq3.c
libavutil/parseutils.c
libavutil/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit '124134e42455763b28cc346fed1d07017a76e84e':
avopt: Store defaults for AV_OPT_TYPE_CONST in the i64 union member
Conflicts:
libavcodec/aacenc.c
libavcodec/libopenjpegenc.c
libavcodec/options_table.h
libavdevice/bktr.c
libavdevice/v4l2.c
libavdevice/x11grab.c
libavfilter/af_amix.c
libavfilter/vf_drawtext.c
libavformat/movenc.c
libavformat/options_table.h
libavutil/opt.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
doc/APIchanges: add an entry for codec descriptors.
vorbisenc: set AVCodecContext.bit_rate to 0
vorbisenc: fix quality parameter
FATE: add ALAC encoding tests
lpc: fix alignment of windowed samples for odd maximum LPC order
alacenc: use s16p sample format as input
alacenc: remove unneeded sample_fmt check
alacenc: fix max_frame_size calculation for the final frame
adpcm_swf: Use correct sample offsets when using trellis.
rtmp: support strict rtmp servers
mjpegdec: support AVRn interlaced
x86: remove FASTDIV inline asm
Conflicts:
doc/APIchanges
libavcodec/mjpegdec.c
libavcodec/vorbisenc.c
libavutil/x86/intmath.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
In order to send or receive a stream FCPublish, FCSubscribe and _checkbw
are completely optional and often not implemented. releaseStream over a
non-existen stream might report an error instead of being silent.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* qatar/master:
libvpxenc: use the default bitrate if not set
utvideo: Rename utvideo.c to utvideodec.c
doc: Fix syntax errors in sample Emacs config
mjpegdec: more meaningful return values
configure: clean up Altivec detection
getopt: Remove an unnecessary define
rtmp: Use int instead of ssize_t
getopt: Add missing includes
rtmp: Add support for receiving incoming streams
Add missing includes for code relying on external libraries
Conflicts:
libavcodec/libopenjpegenc.c
libavcodec/libvpxenc.c
libavcodec/mjpegdec.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Not all compilers support ssize_t (MSVC doesn't), and none of these
variables need to be larger than 32 bit.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
Fix even more missing includes after the common.h removal
build: Factor out rangecoder dependencies to CONFIG_RANGECODER
build: Factor out error resilience dependencies to CONFIG_ERROR_RESILIENCE
x86: avcodec: Consistently name all init files
Add more missing includes after removing the implicit common.h
Add some more missing includes after removing the implicit common.h
Don't include common.h from avutil.h
rtmp: Automatically compute the hash for SWFVerification
Conflicts:
configure
doc/APIchanges
doc/examples/decoding_encoding.c
libavcodec/Makefile
libavcodec/assdec.c
libavcodec/audio_frame_queue.c
libavcodec/avpacket.c
libavcodec/dv_profile.c
libavcodec/dwt.c
libavcodec/libtheoraenc.c
libavcodec/rawdec.c
libavcodec/rv40dsp.c
libavcodec/tiff.c
libavcodec/tiffenc.c
libavcodec/v210dec.h
libavcodec/vc1dsp.c
libavcodec/x86/Makefile
libavfilter/asrc_anullsrc.c
libavfilter/avfilter.c
libavfilter/buffer.c
libavfilter/formats.c
libavfilter/vf_ass.c
libavfilter/vf_drawtext.c
libavfilter/vf_fade.c
libavfilter/vf_select.c
libavfilter/video.c
libavfilter/vsrc_testsrc.c
libavformat/version.h
libavutil/audioconvert.c
libavutil/error.h
libavutil/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtmp: Add support for SWFVerification
api-example: use new video encoding API.
x86: avcodec: Appropriately name files containing only init functions
mpegvideo_mmx_template: drop some commented-out cruft
libavresample: add mix level normalization option
w32pthreads: Add missing #includes to make header compile standalone
rtmp: Gracefully ignore _checkbw errors by tracking them
rtmp: Do not send _checkbw calls as notifications
prores: interlaced ProRes encoding
Conflicts:
doc/examples/decoding_encoding.c
libavcodec/proresenc_kostya.c
libavcodec/w32pthreads.h
libavcodec/x86/Makefile
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Specifies how the server verifies client SWF files before allowing the
files to connect to an application. Verifying SWF files is a security
measure that prevents someone from creating their own SWF files that can
attempt to stream your resources.
Signed-off-by: Martin Storsjö <martin@martin.st>
The _checkbw calls were changed to use transactionId 0 in commit
82613564 so that servers would not return _result/_error about it.
While this is the strict interpretation of the spec, there are
servers that return _error about it, even if transactionId was 0.
The latest version of EvoStream Media Server (the commercial version
of crtmpserver) behaves properly as described, i.e. returning an
_error normally but not returning anything when using transactionId
0. The latest version of crtmpserver (right now at least) doesn't
behave like this though, it returns an error even if transactionId
was 0.
There are also other servers that return errors even if transactionId
is set to 0. Therefore set a proper transaction id so that the invoke
can be tracked and the error properly ignored instead.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
lavf: Detect discontinuities in timestamps for framerate/analyzeduration calculation
lavf: Initialize the stream info timestamps in avformat_new_stream
id3v2: Match PIC mimetype/format case-insensitively
configure: Rename check_asm() to more fitting check_inline_asm()
fate: Only test enabled filters
avresample: De-doxygenize some comments where Doxygen is not appropriate
rtmp: split chunk_size var into in_chunk_size and out_chunk_size
rtmp: Factorize the code by adding find_tracked_method
Conflicts:
configure
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
g723.1: fix addition overflow
g723.1: simplify and fix multiplication overflow
g723.1: deobfuscate an expression
g723.1: remove unused #includes
ARM: add missing "cc" clobber in av_clipl_int32_arm()
rtmp: Factorize the code by adding handle_invoke_error
rtmp: Factorize the code by adding handle_invoke_status
rtmp: Factorize the code by adding handle_invoke_result
libavutil: remove unused av_abort() macro
ffmenc: replace if/abort with assert()
libavutil: drop offsetof() fallback definition
libavutil: drop fallback definitions of INTxx_MIN/MAX
configure: Check for a sctp struct instead of just the header
configure: suncc: Add -xc99 to dependency flags, required on Solaris
doxygen: Fix function parameter names to match the code
doc: Drop obsolete shared libs cflags hint to workaround Cygwin gcc bugs
swf: Move shared table out of the header file
swf: Move swf_audio_codec_tags table to the only place it is used
fate: add G.723.1 decoder tests
Conflicts:
configure
doc/platform.texi
libavformat/Makefile
libavutil/arm/intmath.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mpegvideo: reduce excessive inlining of mpeg_motion()
mpegvideo: convert mpegvideo_common.h to a .c file
build: factor out mpegvideo.o dependencies to CONFIG_MPEGVIDEO
Move MASK_ABS macro to libavcodec/mathops.h
x86: move MANGLE() and related macros to libavutil/x86/asm.h
x86: rename libavutil/x86_cpu.h to libavutil/x86/asm.h
aacdec: Don't fall back to the old output configuration when no old configuration is present.
rtmp: Add message tracking
rtsp: Support mpegts in raw udp packets
rtsp: Support receiving plain data over UDP without any RTP encapsulation
rtpdec: Remove an unused include
rtpenc: Remove an av_abort() that depends on user-supplied data
vsrc_movie: discourage its use with avconv.
avconv: allow no input files.
avconv: prevent invalid reads in transcode_init()
avconv: rename OutputStream.is_past_recording_time to finished.
Conflicts:
configure
doc/filters.texi
ffmpeg.c
ffmpeg.h
libavcodec/Makefile
libavcodec/aacdec.c
libavcodec/mpegvideo.c
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (23 commits)
build: cosmetics: Reorder some lists in a more logical fashion
x86: pngdsp: Fix assembly for OS/2
fate: add test for RTjpeg in nuv with frameheader
rtmp: send check_bw as notification
g723_1: clip argument for 15-bit version of normalize_bits()
g723_1: use all LPC vectors in formant postfilter
id3v2: Support v2.2 PIC
avplay: fix build with lavfi disabled.
avconv: split configuring filter configuration to a separate file.
avconv: split option parsing into a separate file.
mpc8: do not leave padding after last frame in buffer for the next decode call
mpegaudioenc: list supported channel layouts.
mpegaudiodec: don't print an error on > 1 frame in a packet.
api-example: update to new audio encoding API.
configure: add --enable/disable-random option
doc: cygwin: Update list of FATE package requirements
build: Remove all installed headers and header directories on uninstall
build: change checkheaders to use regular build rules
rtmp: Add a new option 'rtmp_subscribe'
rtmp: Add support for subscribing live streams
...
Conflicts:
Makefile
common.mak
configure
doc/examples/decoding_encoding.c
ffmpeg.c
libavcodec/g723_1.c
libavcodec/mpegaudiodec.c
libavcodec/x86/pngdsp.asm
libavformat/version.h
library.mak
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
When streaming live streams using the Akamai, Edgecast or Limelight CDN,
players cannot simply connect to the live stream. Instead, they have to
subscribe to it, by sending an FC Subscribe call to the server.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
mpc8: return more meaningful error codes.
mpc: return more meaningful error codes.
wv,mpc8: don't return apetag data in packets.
rtmp: do not warn about receiving metadata packets
x86: h264dsp: Adjust YASM #ifdefs
x86: yadif: Mark mmxext optimizations as such
h264: convert loop filter strength dsp function to yasm.
Improve descriptiveness of a number of codec and container long names
Conflicts:
libavcodec/flvdec.c
libavcodec/libopenjpegdec.c
libavformat/apetag.c
libavformat/mp3dec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
proresdsp: port x86 assembly to cpuflags.
lavr: x86: improve non-SSE4 version of S16_TO_S32_SX macro
lavfi: better channel layout negotiation
alac: check for truncated packets
alac: reverse lpc coeff order, simplify filter
lavr: add x86-optimized mixing functions
x86: add support for fmaddps fma4 instruction with abstraction to avx/sse
tscc2: fix typo in array index
build: use COMPILE template for HOSTOBJS
build: do full flag handling for all compiler-type tools
eval: fix printing of NaN in eval fate test.
build: Rename aandct component to more descriptive aandcttables
mpegaudio: bury inline asm under HAVE_INLINE_ASM.
x86inc: automatically insert vzeroupper for YMM functions.
rtmp: Check the buffer length of ping packets
rtmp: Allow having more unknown data at the end of a chunk size packet without failing
rtmp: Prevent reading outside of an allocate buffer when receiving server bandwidth packets
Conflicts:
Makefile
configure
libavcodec/x86/proresdsp.asm
libavutil/eval.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
libopenjpeg: support YUV and deep RGB pixel formats
Fix typo in v410 decoder.
vf_yadif: unset cur_buf on the input link.
vf_overlay: ensure the overlay frame does not get leaked.
vf_overlay: prevent premature freeing of cur_buf
Support urlencoded http authentication credentials
rtmp: Return an error when the client bandwidth is incorrect
rtmp: Return proper error code in handle_server_bw
rtmp: Return proper error code in handle_client_bw
rtmp: Return proper error codes in handle_chunk_size
lavr: x86: add missing vzeroupper in ff_mix_1_to_2_fltp_flt()
vp8: Replace x*155/100 by x*101581>>16.
vp3: don't use calls to inline asm in yasm code.
x86/dsputil: put inline asm under HAVE_INLINE_ASM.
dsputil_mmx: fix incorrect assembly code
rtmp: Factorize the code by adding handle_invoke
rtmp: Factorize the code by adding handle_chunk_size
rtmp: Factorize the code by adding handle_ping
rtmp: Factorize the code by adding handle_client_bw
rtmp: Factorize the code by adding handle_server_bw
Conflicts:
libavcodec/libopenjpegdec.c
libavcodec/x86/dsputil_mmx.c
libavfilter/vf_overlay.c
libavformat/Makefile
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtmp: Add a new option 'rtmp_pageurl'
doc: Update the description of the rtmp_tcurl option
rtmp: Make the description of the rtmp_tcurl option more generic
libfdk-aacenc: add LATM/LOAS encapsulation support
sctp: add port missing error message
tcp: add port missing error message
avfilter: Fix printf format string conversion specifier
Conflicts:
libavcodec/version.h
libavfilter/avfilter.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtmp: Add credit/copyright to librtmp authors for parts of the RTMPE code
rtmp: Move the CONFIG_ condition into the if conditions
aac: Mention abbreviation as well in long_name
build: Skip compiling rtmpdh.h if ffrtmpcrypt protocol is not enabled
doc: Add Git configuration section
configure: Add a dependency on https for rtmpts
rtp: Only choose static payload types if the sample rate and channels are right
Conflicts:
doc/git-howto.texi
libavformat/rtmpproto.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This makes sure these calls are removed by dead code elimination
even if optimization is disabled. This fixes building without
crypto libraries without optimization.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
v410dec: Implement explode mode support
zerocodec: fix direct rendering.
wav: init st to NULL to avoid a false-positive warning.
wavpack: set bits_per_raw_sample for S32 samples to properly identify 24-bit
h264: refactor NAL decode loop
RTMPTE protocol support
RTMPE protocol support
rtmp: Add ff_rtmp_calc_digest_pos()
rtmp: Rename rtmp_calc_digest to ff_rtmp_calc_digest and make it global
swscale: add missing HAVE_INLINE_ASM check.
lavfi: place x86 inline assembly under HAVE_INLINE_ASM.
vc1: Add a test for interlaced field pictures
swscale: Mark all init functions as av_cold
swscale: x86: Drop pointless _mmx suffix from filenames
lavf: use conditional notation for default codec in muxer declarations.
swscale: place inline assembly bilinear scaler under HAVE_INLINE_ASM.
dsputil: ppc: cosmetics: pretty-print
dsputil: x86: add SHUFFLE_MASK_W macro
configure: respect CC_O setting in check_cc
Conflicts:
Changelog
configure
libavcodec/v410dec.c
libavcodec/zerocodec.c
libavformat/asfenc.c
libavformat/version.h
libswscale/utils.c
libswscale/x86/swscale.c
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
RTMPE protocol implementation uses ffrtmpcrypt:// 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:
libopenjpeg: introduce encoding support
libopenjpeg: rename decoder source file.
RTMPTS protocol support
RTMPS protocol support
avconv: print an error message when demuxing fails.
tscc2: DCT output should not be clipped
rtmp: Rename rtmphttp to ffrtmphttp
Conflicts:
Changelog
configure
doc/general.texi
libavcodec/libopenjpegenc.c
libavcodec/version.h
libavformat/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
rtmp: rtmp_parse_result() add case for video and audio packets to avoid undesired debug output.
configure: Move the getaddrinfo function check into the network block
configure: Remove an unused 'have' item
mpeg: remove disabled code
libfdk-aac: Check if cutoff value is valid
network: Always use our version of gai_strerror on windows
network: Undefine existing gai_strerror definitions
network: Extend the fallback gai_strerror implementation to handle more error codes
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
