* qatar/master:
fate: Fix fate-ac3-fixed-encode for pre-ssse3 x86 machines
http: Pass the proper return code of net IO operations
http: Add 'post_data', a new option which sets custom HTTP post data
lavfi: amix: check active input count before calling request_samples
vp8: move block coeff arithcoder on stack.
mp3/ac3 probe: search for PES headers to prevent probing MPEG-PS as MP3.
Conflicts:
libavformat/ac3dec.c
libavformat/mp3dec.c
tests/fate/ac3.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* commit 'f919cc7df6ab844bc12f89fe7bef4fb915a47725':
fate: fix acodec/vsynth tests for make 3.81
pcm_mpeg: fix number of consumed bytes to include the header.
avfilter: include required header file avfilter.h in video.h
x86: Avoid movs on BUTTERFLYPS when in AVX mode
x86: use new schema for ASM macros
fate: convert codec-regression.sh to makefile rules
fate: allow tests to specify unit size for psnr comparison
fate: teach videogen/rotozoom to output a single raw video stream
http: Add support for reusing the http socket for subsequent requests
http: Add support for using persistent connections
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Such files are currently not supported as the table is used at several points
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
There is basic support for muxing chapter information into the
Apple Quicktime format already, but there are two errors which
prevent correct detection on the player side.
1) A special apple 'text' atom needs to be included inside the
gmhd atom.
2) The *different* 'text' atom inside the 'stsd' atom needs a
proper header.
With these changes, the chapters are now picked up by Apple
players and reported correctly by tools like mediainfo and mp4chaps.
v3 Update: The stub TextSampleEntry creation is moved to where the
chapter track is created so it's now specific to this track.
Signed-off-by: Philip Langdale <philipl@overt.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Introduce ff_http_do_new_request(), a new function which sends a new
HTTP request, reusing the existing connection to the server.
Signed-off-by: Martin Storsjö <martin@martin.st>
Add a new AVOption 'multiple_requests', which indicates if we want
to use persistent connections (ie. Connection: keep-alive).
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
avprobe: restore pseudo-INI old style format for compatibility.
avprobe: fix formatting.
log: make colored output more colorful.
rtsp: Check for dynamic payload handlers if no static payload mapping was found
Conflicts:
Changelog
doc/ffprobe.texi
ffprobe.c
libavutil/log.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
Some systems abuse the static payload types 35 or 36 (which
according to IANA are unassigned) for H264.
Signed-off-by: Martin Storsjö <martin@martin.st>
If an URLContext is passed in, its ownership is given to this
function, and is either owned by the returned AVFormatContext
on a successful return, or freed on failure.
Signed-off-by: Martin Storsjö <martin@martin.st>
* 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:
os_support: Define SHUT_RD, SHUT_WR and SHUT_RDWR on OS/2
http: Add support for reading http POST reply headers
http: Add http_shutdown() for ending writing of posts
tcp: Allow signalling end of reading/writing
avio: Add a function for signalling end of reading/writing
lavfi: fix comment, audio is supported now.
lavfi: fix incorrect comment.
lavfi: remove avfilter_null_* from public API on next bump.
lavfi: remove avfilter_default_* from public API on next bump.
lavfi: deprecate default config_props() callback and refactor avfilter_config_links()
avfiltergraph: smarter sample format selection.
avconv: rename transcode_audio/video to decode_audio/video.
asyncts: reset delta to 0 when it's not used.
x86: lavc: use %if HAVE_AVX guards around AVX functions in yasm code.
dwt: return errors from ff_slice_buffer_init()
Conflicts:
ffmpeg.c
libavfilter/avfilter.c
libavfilter/avfilter.h
libavfilter/formats.c
libavfilter/version.h
libavfilter/vf_blackframe.c
libavfilter/vf_drawtext.c
libavfilter/vf_fade.c
libavfilter/vf_format.c
libavfilter/vf_showinfo.c
libavfilter/video.c
libavfilter/video.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The sample_rate variable is used for checks for audio format
changes at the end of the function.
This fixes cases where the sample rate was set from the codec
id by flv_set_audio_codec (as for nellymoser 8 kHz/16 kHz),
so the value set to last_sample_rate wasn't equal to sample_rate
at this point. This caused the demuxer otherwise reports a spurious
change to 5512 Hz and back to the correct one.
Updating channels in the same way is only done for consistency.
Currently, flv_set_audio_codec doesn't update that value.
Signed-off-by: Martin Storsjö <martin@martin.st>
tcp_shutdown() isn't needed at the moment, but is added for
consistency to explain how the function is supposed to be used.
Signed-off-by: Martin Storsjö <martin@martin.st>
