* qatar/master:
h264: allow cropping to AVCodecContext.width/height
mov: set AVCodecContext.width/height for h264
iac: generate codec tables as they are supposed to be
indeo4: handle frame type 1 properly
lavu: change versioning script to include all av* prefixed symbols
Conflicts:
libavcodec/h264.c
libavutil/libavutil.v
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
vorbis: Validate that the floor 1 X values contain no duplicates.
avprobe: Identify codec probe failures rather than calling them unsupported codecs.
avformat: Probe codecs at score 0 on buffer exhaustion conditions.
avformat: Factorize codec probing.
Indeo Audio decoder
imc: make IMDCT support stereo output
imc: move channel-specific data into separate context
lavfi: remove request/poll and drawing functions from public API on next bump
lavfi: make avfilter_insert_pad and pals private on next bump.
lavfi: make formats API private on next bump.
avplay: use buffersrc instead of custom input filter.
avtools: move buffer management code from avconv to cmdutils.
avconv: don't use InputStream in the buffer management code.
avconv: fix exiting when max frames is reached.
mpc8: fix maximum bands handling
aacdec: Turn PS off when switching to stereo and turn it to implicit when switching to mono.
Conflicts:
Changelog
cmdutils.h
ffmpeg.c
ffplay.c
ffprobe.c
libavcodec/avcodec.h
libavcodec/mpc8.c
libavcodec/v210dec.h
libavcodec/version.h
libavcodec/vorbisdec.c
libavfilter/avfilter.c
libavfilter/avfilter.h
libavfilter/buffersrc.c
libavfilter/formats.c
libavfilter/src_movie.c
libavfilter/vf_aspect.c
libavfilter/vf_blackframe.c
libavfilter/vf_boxblur.c
libavfilter/vf_crop.c
libavfilter/vf_cropdetect.c
libavfilter/vf_delogo.c
libavfilter/vf_drawbox.c
libavfilter/vf_drawtext.c
libavfilter/vf_fade.c
libavfilter/vf_fifo.c
libavfilter/vf_format.c
libavfilter/vf_frei0r.c
libavfilter/vf_gradfun.c
libavfilter/vf_hflip.c
libavfilter/vf_hqdn3d.c
libavfilter/vf_libopencv.c
libavfilter/vf_lut.c
libavfilter/vf_overlay.c
libavfilter/vf_pad.c
libavfilter/vf_scale.c
libavfilter/vf_select.c
libavfilter/vf_showinfo.c
libavfilter/vf_transpose.c
libavfilter/vf_unsharp.c
libavfilter/vf_yadif.c
libavfilter/vsrc_color.c
libavfilter/vsrc_testsrc.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
movenc: Write chan atom for all audio tracks in mov mode movies.
mpegtsenc: use avio_open_dyn_buf(), zero pointers after freeing
doc/avconv: add some details about the transcoding process.
avidec: make scale and rate unsigned.
avconv: check output stream recording time before each frame returned from filters
avconv: split selecting input file out of transcode().
avconv: split checking for active outputs out of transcode().
avfiltergraph: make some functions static.
Conflicts:
ffmpeg.c
libavfilter/avfiltergraph.c
libavfilter/internal.h
libavformat/mpegtsenc.c
tests/ref/fate/acodec-alac
tests/ref/fate/acodec-pcm-s16be
tests/ref/fate/acodec-pcm-s24be
tests/ref/fate/acodec-pcm-s32be
tests/ref/fate/acodec-pcm-s8
tests/ref/lavf/mov
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Some demuxers set a timecode in the format or streams metadata. The
muxers now make use of this metadata instead of a duplicated private
option.
This makes possible transparent copy of the timecode when transmuxing
and transcoding.
-timecode option for MPEG1/2 codec is also renamed to -gop_timecode. The
global ffmpeg -timecode option will set it anyway so no option change
visible for the user.
* qatar/master:
librtmp: return AVERROR_UNKNOWN instead of -1.
librtmp: don't abuse a variable for two unrelated things.
librtmp: add rtmp_app and rtmp_playpath private options.
bmv: add stricter checks for invalid decoded length
avpacket: fix duplicating side data.
flv: support stream text data as onTextData
Conflicts:
libavcodec/bmv.c
libavformat/flvdec.c
libavformat/flvenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mpegtsenc: Support LATM packetization for AAC
adtsenc: Don't expose the muxer internals to the rest of lavf
mpegtsenc: use AVFormatContext for AAC packetization
mpegtsenc: use AVERROR() for return codes
Conflicts:
libavformat/adts.h
libavformat/mpegtsenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This adds the avoption mpegts_flags and converts the existing
resend_headers option into a flag, keeping the old option as
fallback for now.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This isn't required any longer, when the mpegts muxer uses it
as a proper chained muxer.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
This removes the dependency on adts.c internals, and simplifies
adding other packetization formats.
Signed-off-by: Jindrich Makovicka <makovick@gmail.com>
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
http: Add the url_shutdown function for https, too
http: Simplify code by removing a local variable
http: Clear the old URLContext pointer when closed
tcp: Try enabling SO_REUSEADDR when listening
tcp: Check the return values from bind and accept
avisynth: Make sure the filename passed to avisynth is in the right code page
Conflicts:
libavformat/http.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes issues with opening http urls that have authentication
or redirects, introduced in commit e999b641.
Signed-off-by: Martin Storsjö <martin@martin.st>
avisynth is a non-unicode application and cannot accept UTF-8
characters. Therefore, the input filename should be converted to
the correct code page that it expects.
Signed-off-by: Martin Storsjö <martin@martin.st>
* 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>
This fixes passing junk in stream.
It should not have any user vissible effect.
We are discarding the new data in the decoder as no case is known
where it is needed but it causes problems if used.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
indeo: Make ivi_calc_band_checksum() static, it is only used in one file.
indeo: Drop unused debug function ivi_check_band().
avcodec/utils: cast a function argument to shut up a compiler warning
truemotion1: remove disabled code
fix typo in comment
fate: fix dependencies for non-SAMPLES avconv tests
indeo: check for invalid motion vectors
indeo: check that band output buffer exists
indeo: clear allocated band buffers
indeo: track tile macroblock size
indeo: check custom Huffman tables for errors
factor out common decoding code for Indeo 4 and Indeo 5
mp3: fix start band index for block type 2 in 8kHz audio
lavf: change some (de)muxer names to lowercase
lavf: make output format matching case insensitive
Conflicts:
libavcodec/indeo4.c
libavcodec/indeo5.c
libavcodec/ivi_common.c
libavcodec/utils.c
tests/fate/video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Use codec aspect ratio for frame aspect ratio if AVFrame is NULL.
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* commit '755cd4197d53946208e042f095b930dca18d9430':
mov: enable parsing for VC-1.
lavfi: Add fps filter.
lavfi: initialize pts to AV_NOPTS_VALUE when creating new buffer refs.
avconv: add support for audio in complex filtergraphs.
Conflicts:
ffmpeg.c
libavfilter/version.h
libavformat/mov.c
tests/ref/fate/vc1-ism
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:
fate: Work around non-standard wc implementations at more places
fate: work around non-standard wc implementations
x86: rv40: Mark rv40_weight functions as MMX2; they use MMX2 instructions.
ac3dsp: simplify x86 versions of ac3_max_msb_abs_int16
fate: use standard diff options
tta: Fix comment about channel number; TTA supports >2 channels.
avfilter: Move ff_get_ref_perms_string() to where it is used.
build: Add 'check' target to run all compile and test targets.
indeo3: validate new frame size before resetting decoder
indeo3: when freeing buffers, set pointers referencing them to NULL as well
indeo3: initialise pixel planes on allocation
indeo3: ensure that decoded cell data is in 7-bit range as presumed by decoder
fate: rename psx-str-v3-mdec to mdec-v3
fate: convert psx-str to a demuxer test
lavf: add mdec to is_intra_only() list
Conflicts:
doc/developer.texi
libavcodec/indeo3.c
libavfilter/video.c
libavformat/utils.c
tests/fate/demux.mak
tests/fate/video.mak
tests/lavf-regression.sh
tests/ref/vsynth1/cljr
tests/ref/vsynth1/ffvhuff
tests/ref/vsynth2/cljr
tests/ref/vsynth2/ffvhuff
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Guesses the sample aspect ratio of a frame, based on both the stream and the
frame aspect ratio.
Since the frame aspect ratio is set by the codec but the stream aspect ratio
is set by the demuxer, these two may not be equal. This function tries to
return the value that you should use if you would like to display the frame.
Basic logic is to use the stream aspect ratio if it is set to something sane
otherwise use the frame aspect ratio. This way a container setting, which is
usually easy to modify can override the coded value in the frames.
Signed-off-by: Marton Balint <cus@passwd.hu>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (25 commits)
vcr1: Add vcr1_ prefixes to all static functions with generic names.
vcr1: Fix return type of common_init to match the function pointer signature.
vcr1enc: Replace obsolete get_bit_count by put_bits_count/flush_put_bits.
motion-test: remove disabled code
gxfenc: remove disabled half-implemented MJPEG tag
x86: use more standard construct for setting ASM functions in FFT code
fate: westwood-aud: disable decoding
fate: caf: disable decoding
fate: film-cvid: drop pcm audio and rename test
fate: d-cinema-demux: drop unnecessary flags
fate: split off dpcm-interplay from interplay-mve tests
fate: rename funcom-iss to adpcm-ima-iss
fate: rename cryo-apc to adpcm-ima-apc
fate: rename adpcm-psx-str-v3 to adpcm-xa
fate: split off adpcm-ms-mono test from dxa-feeble
fate: split off adpcm-ima-ws test from vqa-cc
fate: add adpcm-ima-smjpeg test
fate: split off adpcm-ima-amv from amv test
fate: separate bmv audio and video tests
fate: separate delphine-cin audio and video tests
...
Conflicts:
doc/platform.texi
libavcodec/vcr1.c
tests/fate/audio.mak
tests/fate/demux.mak
tests/fate/video.mak
tests/ref/fate/ea-mad-pcm-planar
tests/ref/fate/interplay-mve-16bit
tests/ref/fate/interplay-mve-8bit
tests/ref/fate/mtv
tests/ref/fate/qtrle-1bit
tests/ref/fate/qtrle-2bit
tests/ref/fate/truemotion1-15
tests/ref/fate/truemotion1-24
tests/ref/fate/vqa-cc
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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: (27 commits)
libxvid: Give more suitable names to libxvid-related files.
libxvid: Separate libxvid encoder from libxvid rate control code.
jpeglsdec: Remove write-only variable in ff_jpegls_decode_lse().
fate: cosmetics: lowercase some comments
fate: Give more consistent names to some RealVideo/RealAudio tests.
lavfi: add avfilter_get_audio_buffer_ref_from_arrays().
lavfi: add extended_data to AVFilterBuffer.
lavc: check that extended_data is properly set in avcodec_encode_audio2().
lavc: pad last audio frame with silence when needed.
samplefmt: add a function for filling a buffer with silence.
samplefmt: add a function for copying audio samples.
lavr: do not try to copy to uninitialized output audio data.
lavr: make avresample_read() with NULL output discard samples.
fate: split idroq audio and video into separate tests
fate: improve dependencies
fate: add convenient shorthands for ea-vp6, libavcodec, libavutil tests
fate: split some combined tests into separate audio and video tests
fate: fix dependencies for probe tests
mips: intreadwrite: fix inline asm for gcc 4.8
mips: intreadwrite: remove unnecessary inline asm
...
Conflicts:
cmdutils.h
configure
doc/APIchanges
doc/filters.texi
ffmpeg.c
ffplay.c
libavcodec/internal.h
libavcodec/jpeglsdec.c
libavcodec/libschroedingerdec.c
libavcodec/libxvid.c
libavcodec/libxvid_rc.c
libavcodec/utils.c
libavcodec/version.h
libavfilter/avfilter.c
libavfilter/avfilter.h
libavfilter/buffersink.h
tests/Makefile
tests/fate/aac.mak
tests/fate/audio.mak
tests/fate/demux.mak
tests/fate/ea.mak
tests/fate/image.mak
tests/fate/libavutil.mak
tests/fate/lossless-audio.mak
tests/fate/lossless-video.mak
tests/fate/microsoft.mak
tests/fate/qt.mak
tests/fate/real.mak
tests/fate/screen.mak
tests/fate/video.mak
tests/fate/voice.mak
tests/fate/vqf.mak
tests/ref/fate/ea-mad
tests/ref/fate/ea-tqi
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* 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:
libschroedinger: Switch to function names more in line with Libav style.
Move code shared between libdirac and libschroedinger to libschroedinger.
lavfi: uninline avfilter_copy_buffer_ref_props().
lavf: add missing '*' in a doxy.
h264: Remove a commented-out function pointer typedef.
txd: Remove write-only variable in txd_decode_frame().
mmvideo.c: Remove unused variable in mm_decode_pal().
build: cosmetics: Add missing end-of-line backslashes to item lists.
build: cosmetics: Split HEADERS/OBJS/PROGS lists into one entry per line.
libschroedinger: Move a function to avoid a forward declaration.
pthread: warn on high thread counts
vf_yadif: fix missing error handling for avfilter_poll_frame()
avprobe: allow showing only one container/stream property.
lavfi: support audio in avfilter_copy_frame_props().
lavfi: avfilter_merge_formats: handle case where inputs are same
lavc: add sample rate and channel layout to AVFrame.
zerocodec: check if the previous frame is missing
doc: clarify check for NULL pointer style
Conflicts:
doc/APIchanges
doc/developer.texi
ffprobe.c
libavcodec/Makefile
libavcodec/avcodec.h
libavcodec/libdirac_libschro.c
libavcodec/libdirac_libschro.h
libavcodec/mmvideo.c
libavcodec/txd.c
libavcodec/version.h
libavcodec/zerocodec.c
libavfilter/Makefile
libavfilter/avfilter.c
libavfilter/version.h
libavformat/Makefile
libavutil/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fixes the Xing tag identification string to be "Info" for MP3 files with
constant bitrate. The previous "Xing" caused some decoders to recognize the
file as VBR.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (28 commits)
dfa: use more meaningful return codes
eatgv: check vector_bits
eatgv: check motion vectors
Mark a number of variables only used in av_dlog() calls as av_unused.
dvdec: drop const qualifier from variable to eliminate a warning
avcodec: Improve comment for thread_safe_callbacks to avoid misinterpretation.
tests/utils: don't ignore the return value of fwrite()
lavfi/formats: use sizeof(var) instead of sizeof(type).
lavfi: remove avfilter_default_config_input_link() declaration
lavfi: always enable the scale filter and depend on sws.
vf_split: support user-specifiable number of outputs.
avconv: remove stray useless comment.
mpegmux: add stuffing to avoid incomplete PCM frames
rtsp: avoid const warnings from strtol() call
avserver: check return value of ftruncate()
lagarith: make offset array type unsigned
dfa: add some checks to ensure that decoder won't write past frame end
aacps: NEON optimisations
aacps: align some arrays
aacps: move some loops to function pointers
...
Conflicts:
configure
doc/filters.texi
libavcodec/dfa.c
libavcodec/eatgv.c
libavfilter/Makefile
libavfilter/allfilters.c
libavfilter/avfilter.h
libavfilter/formats.c
libavfilter/vf_split.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The strtol() interface makes it difficult to use with
const-qualified pointers. With this change, although
the const is still lost, the compiler does not warn
about it.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
tests: Refactor rotozoom/videogen common code into a separate file.
tests: Mark some file-internal symbols as static.
build: Drop leftover .exp pattern from LIBSUFFIXES list.
vsrc_buffer: return EAGAIN if no frame is available.
WMAL: Shift output samples by the specified number of padding zeroes.
WMAL: Restore removed code in mclms_predict()
rtpdec_h264: Remove a useless ifdef
rtpdec_h264: Remove outdated/useless/incorrect comments
rtpdec_h264: Remove useless memory corruption checks
rtpdec_h264: Return proper error codes
rtpdec_h264: Check the available data length before reading
rtpdec_h264: Add input size checks
png: check bit depth for PAL8/Y400A pixel formats.
ea: check chunk_size for validity.
celp filters: Do not read earlier than the start of the 'out' vector.
Conflicts:
libavcodec/pngdec.c
libavfilter/src_buffer.c
tests/rotozoom.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Add/fix spacing, split long lines, align assignments where suitable.
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Signed-off-by: Martin Storsjö <martin@martin.st>
Split long comments, move long comments at the end of lines to
separate lines above, fix vertical alignment, fix up comment style
(unify trailing dots - comments had a mix of 2, 3 or 4 dots, where
it would be just as good without them at all).
Signed-off-by: Martin Storsjö <martin@martin.st>
It is worth keeping instead of removing, in case reading this
bit becomes necessary at some later point.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
configure: add POWER[5-7] support
arm: intreadwrite: revert 16-bit load asm to old version for gcc < 4.6
vqavideo: return error if image size is not a multiple of block size
cosmetics: indentation
avformat: only fill-in interpolated timestamps if duration is non-zero
avformat: remove a workaround for broken timestamps
Conflicts:
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Manually remove that flag again for formats that read an arbitrary
amount of data and thus truncation is not an error.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
mkv: mark corrupted packets and return them
mkv: forward EMBL block data error
avcodec: introduce YCoCg colorspace
avcodec: cosmetic cleanup on header
aac sbr: align struct member by 32 byte.
Conflicts:
libavcodec/avcodec.h
libavformat/matroskadec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
It is possible that just extending the RMMuxContext.streams
array would avoid it.
It is also possible that two audio streams will fail to mux
correctly as well, though at least it should not crash for
this reason.
I do not feel like checking either of these.
This patch fixes trac issue #1022 (at least it makes it
exit with a proper error message instead of crashing).
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Also fixes an (incorrect) "control reaches end of non-void function"
warning with some compilers.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
h264: new assembly version of get_cabac for x86_64 with PIC
h264: use one table instead of several for cabac functions
h264: (trivial) remove unneeded macro argument in x86/cabac.h
libschroedingerdec: check malloc
segment: reorder seg_write_header allocation
avio: make avio_close(NULL) a no-op
mov: Parse EC3SpecificBox (dec3 atom).
Conflicts:
libavcodec/cabac.c
libavcodec/x86/cabac.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
this hunk was merged in 8b97ae64 and cbf767a8 although the check was there a
few lines above since cdced09e. I removed the first check to reduce the differences
to libav.
Signed-off-by: Jean First <jeanfirst@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mpegts: Make sure we don't return uninitialized packets
gitignore: replace library catch-all pattern by more specific patterns
Conflicts:
.gitignore
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Skip to parse fields for additional independent substreams and its
associated dependent substreams since libavcodec's E-AC-3 decoder does not
support them yet.
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
This fixes crashes, where the demuxer could return 0 even
if the returned AVPacket isn't initialized at all. This
could happen if running into EOF or running out of probesize
with non-seekable sources.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
avconv: fix a segfault on -c copy with -filter_complex.
isom: Support more DTS codec identifiers.
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
matroska: Clear prev_pkt between seeks.
avutil: change default buffer size alignment for sample buffer functions
audemux: Add a sanity check for the number of channels
Remove libdirac decoder.
matroska: Add incremental parsing of clusters.
avconv: fix off by one check in complex_filter
mpegts: Try seeking back even for nonseekable protocols
swscale: K&R formatting cosmetics (part III)
Conflicts:
configure
doc/general.texi
doc/platform.texi
ffmpeg.c
libavcodec/Makefile
libavcodec/allcodecs.c
libavcodec/libdirac.h
libavcodec/libdiracdec.c
libavformat/au.c
libavformat/mpegts.c
libswscale/input.c
tests/ref/seek/lavf_mkv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The new incremental parser doesn't always clear prev_pkt,
however the packet queue is cleared when seeking. Which leads
to a use-after-free.
Verified using Valgrind.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
Reduces the amount of upfront data required for cluster parsing
thus decreasing latency on seek and startup.
The change in the seek-lavf_mkv FATE test is due to incremental
parsing no longer reading as much data as the old parser and
thus not having that additional data to generate index entries
based on keyframes. Index entries are added correctly as the
file is parsed.
All FATE tests pass and Chrome has been using this patch for ~6
months without issue.
Currently incremental parsing is not supported for files with
SSA tracks since they require merging packets between clusters.
In this case the code falls back to non-incremental parsing.
Signed-off-by: Aaron Colwell <acolwell@chromium.org>
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
The mpegts demuxer reads 5 KB at startup just for discovering
the packet size. Since the default avio buffer size is 32 KB,
the seek back to the start will in most cases be within the
avio buffer, and will in most cases succeed even if the actual
protocol isn't seekable.
This makes the demuxer startup faster/with less data when
reading data from a non-seekable input, by not skipping
the first few KB.
If it fails, don't warn if the protocol isn't seekable, making
it behave as before in the failure case.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
ARM: allow runtime masking of CPU features
dsputil: remove unused functions
mov: Treat keyframe indexes as 1-origin if starting at non-zero.
mov: Take stps entries into consideration also about key_off.
Remove lowres video decoding
Conflicts:
ffmpeg.c
ffplay.c
libavcodec/arm/vp8dsp_init_arm.c
libavcodec/libopenjpegdec.c
libavcodec/mjpegdec.c
libavcodec/mpegvideo.c
libavcodec/utils.c
libavformat/mov.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes a null ptr dereference with attachments
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
avcodec: remove AVCodecContext.dsp_mask
avconv: fix a segfault when default encoder for a format doesn't exist.
utvideo: general cosmetics
aac: Handle HE-AACv2 when sniffing a channel order.
movenc: Support high sample rates in isomedia formats by setting the sample rate field in stsd to 0.
xxan: Remove write-only variable in xan_decode_frame_type0().
ivi_common: Initialize a variable at declaration in ff_ivi_decode_blocks().
Conflicts:
ffmpeg.c
libavcodec/utvideo.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
avcodec: add a cook parser to get subpacket duration
FATE: allow lavf tests to alter input parameters
FATE: replace the acodec-pcm_s24daud test with an enc_dec_pcm checksum test
FATE: replace the acodec-g726 test with 4 new encode/decode tests
FATE: replace current g722 encoding tests with an encode/decode test
FATE: add a pattern rule for generating asynth wav files
FATE: optionally write a WAVE header in audiogen
avutil: add audio fifo buffer
Conflicts:
doc/APIchanges
libavcodec/version.h
libavutil/avutil.h
tests/Makefile
tests/codec-regression.sh
tests/fate/voice.mak
tests/lavf-regression.sh
tests/ref/acodec/g722
tests/ref/acodec/g726
tests/ref/acodec/pcm_s24daud
tests/ref/lavf/dv_fmt
tests/ref/lavf/gxf
tests/ref/lavf/mxf
tests/ref/lavf/mxf_d10
tests/ref/seek/lavf_dv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
dv: Initialize encoder tables during encoder init.
dv: Replace some magic numbers by the appropriate #define.
FATE: pass the decoded output format and audio source file to enc_dec_pcm
FATE: specify the input format when decoding in enc_dec_pcm()
x86inc: support AVX abstraction for 2-operand instructions
configure: detect PGI compiler and set suitable flags
avconv: check for an incompatible changing channel layout
avio: make AVIOContext.av_class pointer to const
nutdec: add malloc check and fix const to non-const conversion warnings
Conflicts:
ffmpeg.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Fix this warning:
libavformat/aviobuf.c:663:20: warning: assignment discards qualifiers from pointer target type
Although this is a public header, it should remain source and
binary compatible.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* qatar/master:
ppc: drop unused function dct_quantize_altivec()
mpegaudiodec: Do not discard mp_decode_frame() return value.
matroska: do not set invalid default duration if frame rate is zero
mkv: use av_reduce instead of av_d2q for framerate estimation
mkv: report average framerate as minimal as well
avcodec_string: Favor AVCodecContext.codec over the default codec.
cook: Make constants passed to AV_BE2NE32C() unsigned to avoid signed overflow.
Conflicts:
libavcodec/cook.c
libavcodec/ppc/mpegvideo_altivec.c
libavcodec/utils.c
libavformat/matroskadec.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
If a video track specifies a zero frame rate (invalid but occurs),
this results in a division by zero and subsequent undefined conversion
to integer. Setting the default duration from the frame rate only
if the latter is greater than zero avoids such problems.
Signed-off-by: Mans Rullgard <mans@mansr.com>
* 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>
Other cases are not supported and lead to inconsistencies which
can lead to out of array writes.
Reported-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This replaces the matroskadec one with the same name.
The advantage is not only easier reuse in other demuxers
but also that we can make the decisions after the parser.
This fixes seeking in files that mark the keyframes incorrectly,
for example the file in track ticket #1003.
The matroska variable is still kept to be able to complain
about such broken files.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
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 new values lead to error messages when used
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Instead of allocating over the original, free first. MOVStreamContext
is zero initialized so no double free will occur. Same style as other
fixes for the same problem in this file.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
* qatar/master:
vsrc_buffer: fix check from 7ae7c41.
libxvid: Reorder functions to avoid forward declarations; make functions static.
libxvid: drop some pointless dead code
wmal: vertical alignment cosmetics
wmal: Warn about missing bitstream splicing feature and ask for sample.
wmal: Skip seekable_frame_in_packet.
wmal: Drop unused variable num_possible_block_size.
avfiltergraph: make the AVFilterInOut alloc/free API public
graphparser: allow specifying sws flags in the graph description.
graphparser: fix the order of connecting unlabeled links.
graphparser: add avfilter_graph_parse2().
vsrc_buffer: allow using a NULL buffer to signal EOF.
swscale: handle last pixel if lines have an odd width.
qdm2: fix a dubious pointer cast
WMAL: Do not try to read rawpcm coefficients if bits is invalid
mov: Fix detecting there is no sync sample.
tiffdec: K&R cosmetics
avf: has_duration does not check the global one
dsputil: fix optimized emu_edge function on Win64.
Conflicts:
doc/APIchanges
libavcodec/libxvid_rc.c
libavcodec/libxvidff.c
libavcodec/tiff.c
libavcodec/wmalosslessdec.c
libavfilter/avfiltergraph.h
libavfilter/graphparser.c
libavfilter/version.h
libavfilter/vsrc_buffer.c
libswscale/output.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
0-sized packets are used to implement variable fps.
However there seems to be a variation where these are not
even stored in the main file but as 0-size index entries
only.
This fixes the sample in trac issue #957, it now plays both
the same ways as in MPlayer and in a way that looks correct.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
They will only cause us to skip writing the Xing header,
not cause any serious breakage.
Related to trac issue #1027.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
swscale: K&R formatting cosmetics (part II)
tiffdec: Add a malloc check and refactor another.
faxcompr: Check malloc results and unify return path
configure: escape colons in values written to config.fate
ac3dsp: call femms/emms at the end of float_to_fixed24() for 3DNow and SSE
matroska: Fix leaking memory allocated for laces.
pthread: Fix crash due to fctx->delaying not being cleared.
vp3: Assert on invalid filter_limit values.
h264: fix 10bit biweight functions after recent x86inc.asm fixes.
ffv1: Fix size mismatch in encode_line.
movenc: Remove a dead initialization
git-howto: Explain how to avoid Windows line endings in git checkouts.
build: Move all arch OBJS declarations into arch subdirectory Makefiles.
Conflicts:
configure
libavcodec/vp3.c
libavformat/matroskadec.c
libavutil/Makefile
libswscale/Makefile
libswscale/swscale.c
libswscale/swscale_internal.h
libswscale/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Its bad to free things without zeroing them.
This fixes a potential issue when mov_read_close() would be called twice.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
If mov_read_header exits under error, the memory allocated is
not freed.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
During error conditions matroska_parse_block may exit without
freeing the memory allocated for laces.
Found via valgrind: http://pastebin.com/E54k8QFU
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Convert key_off initialize to use the same sc->keyframe_count as
used elsewhere in the function.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
During error conditions matroska_parse_block may exit without
freeing the memory allocated for laces.
Found via valgrind: http://pastebin.com/E54k8QFU
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
Fixes potential out-of-bounds writes.
This is mostly possible when muxing ALS files where from
an extradata size of about 1050 put_bits would write data
outside the buffer.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
For the FATE test sample used, this only avoids a warning
message.
However for other samples like al05_44.mp4 the converted
file can be played only after this fix.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This will only work for DSEs that are first in a packet, but
that is enough to fix handling of the reference files in
fate-suite/aac (though most of them still have other issues).
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Otherwise for muxers like e.g. latmenc that never call
avio_flush (and do not have a write_trailer function)
a part of the data will always be missing.
Also update references for the voc muxer, which was also
buggy before and did not write out all data.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
avplay: Don't free video filters string until the end of decoding.
movenc: small refactor mov_write_packet
movenc: remove redundant check
interplayvideo: fix av_dlog parameter type mismatch
Drop some pointless #ifdefs.
Conflicts:
libavcodec/interplayvideo.c
libavcodec/libxvidff.c
libavcodec/snowenc.c
libavformat/movenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Previously it would just silently write out incorrect data.
This also fixes a potential integer overflow in the allocation.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
"Fix" in so far as at least it will no longer overread and possibly
crash and makes somewhat sense, but no idea whether there is anything
that can play the resulting files (FFmpeg can't).
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Share the formerly internal write_packet with the hinter and move the
fragment flush logic to the user facing one since it is not concerned
about movtrack-only streams.
Fixes bug #263
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (22 commits)
rv40dsp x86: use only one register, for both increment and loop counter
rv40dsp: implement prescaled versions for biweight.
avconv: use default channel layouts when they are unknown
avconv: parse channel layout string
nutdec: K&R formatting cosmetics
vda: Signal 4 byte NAL headers to the decoder regardless of what's in the extradata
mem: Consistently return NULL for av_malloc(0)
vf_overlay: implement poll_frame()
vf_scale: support named constants for sws flags.
lavc doxy: add all installed headers to doxy groups.
lavc doxy: add avfft to the main lavc group.
lavc doxy: add remaining avcodec.h functions to a misc doxygen group.
lavc doxy: add AVPicture functions to a doxy group.
lavc doxy: add resampling functions to a doxy group.
lavc doxy: replace \ with /
lavc doxy: add encoding functions to a doxy group.
lavc doxy: add decoding functions to a doxy group.
lavc doxy: fix formatting of AV_PKT_DATA_{PARAM_CHANGE,H263_MB_INFO}
lavc doxy: add AVPacket-related stuff to a separate doxy group.
lavc doxy: add core functions/definitions to a doxy group.
...
Conflicts:
ffmpeg.c
libavcodec/avcodec.h
libavcodec/vda.c
libavcodec/x86/rv40dsp.asm
libavfilter/vf_scale.c
libavformat/nutdec.c
libavutil/mem.c
tests/ref/acodec/pcm_s24daud
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:
rtpdec_asf: Set the no_resync_search option for the chained asf demuxer
asfdec: Add an option for not searching for the packet markers
cosmetics: Clean up the tiffenc pix_fmts declaration to match the style of others
cosmetics: Align codec declarations
cosmetics: Convert mimic.c to utf-8
avconv: remove an unused function parameter.
avconv: remove now pointless variables.
avconv: drop support for building without libavfilter.
nellymoserenc: fix crash due to memsetting the wrong area.
libavformat: Only require first packet to be known for audio/video streams
avplay: Don't try to scale timestamps if the tb isn't set
Conflicts:
Changelog
configure
ffmpeg.c
libavcodec/aacenc.c
libavcodec/bmpenc.c
libavcodec/dnxhddec.c
libavcodec/dnxhdenc.c
libavcodec/ffv1.c
libavcodec/flacenc.c
libavcodec/fraps.c
libavcodec/huffyuv.c
libavcodec/libopenjpegdec.c
libavcodec/mpeg12enc.c
libavcodec/mpeg4videodec.c
libavcodec/pamenc.c
libavcodec/pgssubdec.c
libavcodec/pngenc.c
libavcodec/qtrleenc.c
libavcodec/rawdec.c
libavcodec/sgienc.c
libavcodec/tiffenc.c
libavcodec/v210dec.c
libavcodec/wmv2dec.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The format is slightly proprietary.
DVDs use a format of
code byte (0x00, 0x01, 0xfe or 0xff), two data bytes
MOV uses instead
cdat/cdt2 atom, two data bytes
Auto-detecting and supporting both in one decoder is trivial,
so a single codec ID is used.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Searching for packet markers doesn't make sense for this use case,
where packets are fed one at a time to the demuxer.
This fixes playing back streams that have packets not starting
with the 0x82, 0x00, 0x00 marker.
Signed-off-by: Martin Storsjö <martin@martin.st>
It can take a long time before subtitles or data streams show up,
so we shouldn't wait for those before assuming we have all info
for streams.
Signed-off-by: Martin Storsjö <martin@martin.st>
Also add missing trailing commas, break long codec_tag lines and
add spaces in codec_tag declarations.
Signed-off-by: Martin Storsjö <martin@martin.st>
The correct point that seperates ISO and MAC language codes is 0x400
according to the current QT spec. Old QT specs did not list where this
seperation is but apparently only defined the meaning of the first 137.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The handler name is stored as a pascal string in the QT specs (first
byte is the length of the string), thus leading to an invalid metadata
string export.
Also add a second length check based on the first character to avoid
overwriting an already specified handler_name (it happens with Youtube
videos for instance, the handler_name get masked), or specifying an
empty string metadata.
To reproduce the problem, using ffprobe:
./ffprobe -show_packets -print_format compact -fflags +genpts -i
fate_samples/mxf/C0023S01.mxf
You will notice that the last video frame does not have it's PTS being
set, even with using genpts.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Allows avoiding the buffer when using avio read, write and seek functions.
When using the ffmpeg executable -avioflags direct can be used to enable
this mode for input files, but has no effect on output files.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The reason for this is that such files have IndexTableSegments which when parsed
cover EditUnit ranges like this:
[0,1)
[249,250)
[249,377)
[0,249)
where each interval is [IndexStartPosition,IndexStartPosition+IndexDuration).
This would be reduced to a sparse index like:
[0,1), [249,250)
instead of the full range:
[0,249), [249,377)
See TimeCode_HD.mxf, UMID =
060a2b340101010101010410130000000004001aa0e59175025b2a5600da4101.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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:
vsrc_buffer: allow buffering arbitrary number of frames.
vf_scale: avoid a pointless memcpy in no-op conversion.
avfiltergraph: try to reduce format conversions in filters.
avfiltergraph: add an AVClass to AVFilterGraph on next major bump.
id3v2: fix skipping extended header in id3v2.4
Conflicts:
libavfilter/vf_scale.c
libavfilter/vsrc_buffer.c
libavformat/id3v2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
apedec: check bits <= 32.
cavs: Remove unused code.
oggenc: fix condition when not to flush due to keyframe granule.
oggenc: add pagesize option to set preferred page size
libspeexdec: set frame size in libspeex_decode_init()
smacker audio: sign-extend the initial 16-bit predicted value
Conflicts:
libavcodec/apedec.c
libavcodec/libspeexdec.c
libavformat/oggenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes crashes when copying a data track as in trac
issue #236.
No proper timecode tracks will be written though.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This patch fixes the sample from trac issue #522.
The issue is that the mov demuxer insists on using its
calculated sample_size (which is nonsense for old-style tracks)
instead of the one encoded in the track.
The old raw audio code should be using the value in stsz, because
the size of a single sample never makes sense for the size of
a full audio packet, whereas the new code will multiply the
sample size by the chunk size, so it should use the calculated value.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
This patch fixes the sample from trac issue #733.
The issue is that the size of the trak elements is coded
too large, so that the next trak element would be parsed
as part of the first and truncated incorrectly.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
The previous condition of 0 page size was wrong,
that would disable the mechanism for all frames at
a start of a page, thus some keyframes still would not
get their own granule.
The real problem is that header packets must not be flushed,
but they have (and must have) 0 granule and thus would
be detected as keyframes.
Add a separate parameter to mark header packets.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
When set, if an Ogg stream buffer has enough data, a page is made
instead of filling maximum-size pages. Using smaller pages results
smaller seek intervals at the expense of higher container overhead.
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* qatar/master:
h264: drop ff_h264_ prefix from static function ff_h264_decode_rbsp_trailing()
h264: Make ff_h264_decode_end() static, it is not used externally.
output-example: K&R formatting cosmetics, comment spelling fixes
avf: make the example output the proper message
avf: fix audio writing in the output-example
mov: don't overwrite existing indexes.
lzw: fix potential integer overflow.
truemotion: forbid invalid VLC bitsizes and token values.
truemotion2: handle out-of-frame motion vectors through edge extension.
configure: Check for a different SDL function
Conflicts:
configure
doc/examples/muxing.c
libavcodec/truemotion2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
mp3dec: perform I/S and M/S only when frame mode is joint stereo.
id3v2: add another mimetype for JPEG image
lzw: prevent buffer overreads.
WMAL: Remove inaccurate and unnecessary doxy
h264: fix cabac-on-stack after safe cabac reader.
truemotion2: convert packet header reading to bytestream2.
Conflicts:
libavcodec/lzw.c
libavcodec/truemotion2.c
libavformat/id3v2.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
tilde expansion should/can be done by the shell
Reviewed-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This changes globbing support to only be used if the character
contains at least one glob meta character that is preceded by
an unescaped %. To escape a literal % one would use %% which is
identical to the way to match a % with image2 sequence generation
feature.
* Makes it possible to have patterns like %04d-[720p].jpg work
again with sequence number generation. Previously this would
always be detected as a glob pattern and was interpreted by
the image2 glob code instead.
* Makes it possible to use %*-[720p].jpg to match above pattern
without having to double escape it to be not interpreted by most
shells and not by the image2 glob code (previously one would
need to use \*-\\\[720p\\\].jpg to achieve the same)
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
Version from vqa header does not dictate which sound chunks may
appear in file.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Justin Ruggles <justin.ruggles@gmail.com>
* qatar/master:
make av_interleaved_write_frame() flush packets when pkt is NULL
mpegts: Fix dead error checks
vc1: Do not read from array if index is invalid.
targa: convert to bytestream2.
rv34: set mb_num_left to 0 after finishing a frame
Conflicts:
libavcodec/targa.c
libavcodec/vc1data.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This patch allows the user to force flushing of all queued packets
by calling av_interleaved_write_frame() with pkt set to NULL.
Signed-off-by: Jindrich Makovicka <jindrich.makovicka@nangu.tv>
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
build: ppc: drop stray leftover backslash
build: Only clean the architecture subdirectory we build for.
build: drop some unnecessary dependencies from the H.264 parser
build: prettyprinting cosmetics
libavutil: Remove pointless rational test program.
libavutil: Remove broken and pointless lzo test program.
lavf doxy: expand AVStream.codec doxy.
lavf doxy: improve AVStream.time_base doxy.
lavf doxy: add some basic documentation about reading from the demuxer.
lavf doxy: document passing options to demuxers.
lavf doxy: clarify that an AVPacket contains encoded data.
mpegtsenc: allow user triggered PES packet flushing
APIchanges: mark the place where 0.7 was cut.
APIchanges: mark the place where 0.8 was cut.
APIchanges: fill in missing dates and hashes.
smacker: convert palette and header reading to bytestream2.
alac: convert extradata reading to bytestream2.
Conflicts:
doc/APIchanges
libavcodec/smacker.c
libavcodec/x86/Makefile
libavfilter/Makefile
libavutil/Makefile
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This depends on the proposed parser change for 0-size packets
in previous mail, otherwise video now plays far too fast.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
Currently, the duration of those packets is just discarded
when enabling parsing, thus the output of the Metal Gear Solid
demuxer breaks completely when just setting AVSTREAM_PARSE_HEADERS.
The result will not be correct if a parser creates a delay even
with PARSER_FLAG_COMPLETE_FRAMES and there might be other cases
where it does not work correct, but just discarding them as it
is done currently seems worse.
Signed-off-by: Reimar Döffinger <Reimar.Doeffinger@gmx.de>
* qatar/master:
avc: Add a function for converting mp4 style extradata to annex b
pthread: free progress if buffer allocation failed.
lavc/avconv: support changing frame sizes in codecs with frame mt.
libavformat: Document who sets the AVStream.id field
utvideo: mark output picture as keyframe.
sunrast: Add support for negative linesize.
vp8: fix update_lf_deltas in libavcodec/vp8.c
ralf: read Huffman code lengths without GetBitContext
Conflicts:
ffmpeg.c
libavcodec/sunrastenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This isn't exactly equivalent with the earlier code for codecs
other than H264 and VC1, but those are two only codecs supported
by this codepath anyway, and it simplifies it a bit.
Signed-off-by: Martin Storsjö <martin@martin.st>
This prevents a null ptr dereference.
It could be checked differently but this way it should
be possible to return some data.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
xwma: Validate channels and bits_per_coded_sample.
mov: Do not read past the end of the ctts_data table.
mov: Add missing terminator to mov_ch_layout_map_1ch.
asf: reset side data elements on packet copy.
wmavoice: fix stack overread.
wmalossless: error out if a subframe is not used by any channel.
vqa: check palette chunk size before reading data.
wmalossless: reset sample pointer for each subframe.
wmalossless: error out on invalid values for order.
Conflicts:
libavcodec/vqavideo.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
FATE: Add ZeroCodec test
oggparseogm: fix order of arguments of avpriv_set_pts_info().
pngenc: better upper bound for encoded frame size.
aiffdec: set block_duration to 1 for PCM codecs that are supported in AIFF-C
aiffdec: factor out handling of integer PCM for AIFF-C and plain AIFF
aiffdec: use av_get_audio_frame_duration() to set block_duration for AIFF-C
aiffdec: do not set bit rate if block duration is unknown
wmall: output packet only if we have decoded some samples
Conflicts:
libavcodec/pngenc.c
tests/fate/lossless-video.mak
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Version from vqa header does not dictate which sound chunks may
appear in file.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Signed-off-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>
Takes encoder delay into account by comparing first the coded page
duration with the calculated page duration. Handles last packet duration
if needed, also by comparing coded duration with calculated duration.
Also does better handling of timestamp generation for packets in the
first page for streamed ogg files where the start time is not
necessarily zero.
The other fragmentation options (frag_duration, frag_size and
frag_keyframe) are combined with OR, cutting fragments at the
first of the conditions being fulfilled.
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
An obscure Japanese lossless video codec, originally intended
for use with a remote desktop application.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Kostya Shishkov <kostya.shishkov@gmail.com>
The private option has not been part of any release yet (and
it is only of use in quite rare cases), so just remove it instead
of keeping it with deprecation warnings.
Signed-off-by: Martin Storsjö <martin@martin.st>
This was forgotten in the transition from av_open_input_file to
avformat_open_input, see 603b8bc2a1.
This doesn't change anything for the default case where the
option isn't set, since PROBE_BUF_MAX is 1048576 (which was
used as max probe size earlier) while the default value for
the probesize option is 5000000, which for the probe function
is clipped to PROBE_BUF_MAX anyway.
Signed-off-by: Martin Storsjö <martin@martin.st>
An obscure Japanese lossless video codec, originally intended
for use with a remote desktop application.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master:
fate: make compare() function compatible with POSIX bc
Update Janne's email address.
APIchanges: Replace Subversion revision numbers by Git hashes.
bytestream: Eliminate one level of pointless macro indirection.
xwd: convert to bytestream2.
vqavideo: port to bytestream2 API
Read preset files with suffix .avpreset
prores: allow user to set fixed quantiser
lavf: remove some disabled code.
lavf: only set average frame rate for video.
lavf: remove a pointless check.
avcodec: add XBM encoder
Conflicts:
Changelog
cmdutils.c
cmdutils.h
doc/APIchanges
libavcodec/Makefile
libavcodec/avcodec.h
libavcodec/version.h
libavcodec/vqavideo.c
libavformat/img2enc.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
We can't do this in general since we could be reading a file with B-frames while
lacking an index.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
In non-blocking mode, lowest-level read protocols are
supposed block only for a short amount of time to let
retry_transfer_wrapper() check for interrupts.
Also, checking the interrupt_callback in the receiving thread is
wrong, as interrupt_callback is not guaranteed to be thread-safe
and the job is already done by retry_transfer_wrapper(). The error
code was also incorrect.
Bug reported by Andrey Utkin.
* qatar/master:
dxa: remove useless code
lavf: don't select an attached picture as default stream for seeking.
avconv: remove pointless checks.
avconv: check for get_filtered_frame() failure.
avconv: remove a pointless check.
swscale: convert hscale() to use named arguments.
x86inc: add *mp named argument support to DEFINE_ARGS.
swscale: convert hscale to cpuflags().
Conflicts:
ffmpeg.c
libswscale/x86/scale.asm
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This avoids problems
where avio_tell() returns 0. I've updated all the checks against
cluster_pos
Signed-off-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>
Allow up to 4 retries for normal requests, where both the
proxy and the target server might need to authenticate.
Signed-off-by: Martin Storsjö <martin@martin.st>
These commands are sent asynchronously, not waiting for the reply.
This reply is parsed later by ff_rtsp_tcp_read_packet or
udp_read_packet. If the reply indicates that we used stale
authentication and need to use a new nonce, resend a new keepalive
command immediately.
This is the only request sent asynchronously, so currently there's
no other command that needs to be resent in the same way.
Signed-off-by: Martin Storsjö <martin@martin.st>
* 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>
According to video_file_format_spec_v10_1.pdf flv stores AAC RAW
thanks to Baptiste Coudurier for pointing that out
thanks to Aℓex Converse for explaining:
This can't be at the start of a non-ADTS payload. 111 is the
EndOfFrame syntax element.
Together these proof that the check was correctly rejecting ADTS which
is not supposed to be in flv. Many players are able to play such ADTS
in flv though but its better if we conform to the spec as this should
ensure that not many but all players can play files generated by ffmpeg.
This reverts commit 3c9a86df0e.
mpjpeg video streamings would break and stop on Firefox after 1 - 30
seconds.
In order to fix this, two changes were made:
1. Replaced all occurrences of '\n' character in mjpeg metadata
with occurences of "\r\n".
2. Added "Content-length: <packet-size>" metadata entry for each
sent frame.
The change has been tested on Google Chrome 17.0.963.78 and Firefox 10.0.2
on lubuntu 11.10 and the streaming seems to work fine now.
* 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>
With this we can always know if a timestamp is based on added durations
from an unknown origin or if it is based on a correct timestamp (and possibly
added durations)
This should fix some bugs where this distinction was mixed up.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes sending back RTCP RR packets if receiving RTP over
multicast.
If the multicast stream is sent on demand (set up and signalled
via RTSP), the sender might depend on getting RTCP RR packets
knowing that there are listeners, otherwise the stream can be
closed after a certain timeout.
This fixes receiving RTSP streams over multicast on unix, from
certain Axis cameras.
Signed-off-by: Martin Storsjö <martin@martin.st>
When this code was added in 36b532815c, the new code was added
between the existing comment and the existing line of code, making
the old comment seem to refer to the new code. This makes it read
correctly.
Signed-off-by: Martin Storsjö <martin@martin.st>
Fixes trac #1045.
Thanks to Peter Ross for his help with this patch.
Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The ogg decoder wasn't padding the input buffer with the appropriate
FF_INPUT_BUFFER_PADDING_SIZE bytes. Which led to uninitialized reads in
various pieces of parsing code when they thought they had more data than
they actually did.
Signed-off-by: Dale Curtis <dalecurtis@chromium.org>
Signed-off-by: Ronald S. Bultje <rsbultje@gmail.com>
* qatar/master: (29 commits)
sbrdsp.asm: convert all instructions to float/SSE ones.
dv: cosmetics.
dv: check buffer size before reading profile.
Revert "AAC SBR: group some writes."
udp: Print an error message if bind fails
cook: extend channel uncoupling tables so the full bit range is covered.
roqvideo: cosmetics.
roqvideo: convert to bytestream2 API.
dca: don't use av_clip_uintp2().
wmall: fix build with -DDEBUG enabled.
smc: port to bytestream2 API.
AAC SBR: group some writes.
dsputil: remove shift parameter from scalarproduct_int16
SBR DSP: unroll sum_square
rv34: remove dead code in intra availability check
rv34: clean a bit availability checks.
v4l2: update documentation
tgq: convert to bytestream2 API.
parser: remove forward declaration of MpegEncContext
dca: prevent accessing static arrays with invalid indexes.
...
Conflicts:
doc/indevs.texi
libavcodec/Makefile
libavcodec/dca.c
libavcodec/dvdata.c
libavcodec/eatgq.c
libavcodec/mmvideo.c
libavcodec/roqvideodec.c
libavcodec/smc.c
libswscale/output.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The "ECs != 1 -> OP1a" assumption was wrong. Luckily, the file that triggered
that behavior had two ECs, not zero. Hence distinguishing between them is
simple in this case.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This fixes rare cases where OPAtom may be treated as OP1a, causing all essence
to be read into RAM.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (31 commits)
cdxl demux: do not create packets with uninitialized data at EOF.
Replace computations of remaining bits with calls to get_bits_left().
amrnb/amrwb: Remove get_bits usage.
cosmetics: reindent
avformat: do not require a pixel/sample format if there is no decoder
avformat: do not fill-in audio packet duration in compute_pkt_fields()
lavf: Use av_get_audio_frame_duration() in get_audio_frame_size()
dca_parser: parse the sample rate and frame durations
libspeexdec: do not set AVCodecContext.frame_size
libopencore-amr: do not set AVCodecContext.frame_size
alsdec: do not set AVCodecContext.frame_size
siff: do not set AVCodecContext.frame_size
amr demuxer: do not set AVCodecContext.frame_size.
aiffdec: do not set AVCodecContext.frame_size
mov: do not set AVCodecContext.frame_size
ape: do not set AVCodecContext.frame_size.
rdt: remove workaround for infinite loop with aac
avformat: do not require frame_size in avformat_find_stream_info() for CELT
avformat: do not require frame_size in avformat_find_stream_info() for MP1/2/3
avformat: do not require frame_size in avformat_find_stream_info() for AAC
...
Conflicts:
doc/APIchanges
libavcodec/Makefile
libavcodec/avcodec.h
libavcodec/h264.c
libavcodec/h264_ps.c
libavcodec/utils.c
libavcodec/version.h
libavcodec/x86/dsputil_mmx.c
libavformat/utils.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
All colorspaces are supported.
Renamed libutvideo.cpp to libutvideodec.cpp.
Signed-off-by: Derek Buitenhuis <derek.buitenhuis@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Also, do not keep trying to find and open a decoder in try_decode_frame() if
we already tried and failed once.
Fixes always searching until max_analyze_duration in
avformat_find_stream_info() when demuxing codecs without a decoder.
Also, do not give AVCodecContext.frame_size priority for muxing.
Updated 2 FATE references:
dxa-feeble - adds 1 audio frame that is still within 2 seconds as specified
by -t 2 in the FATE test
wmv8-drm-nodec - durations are not needed. previously they were estimated
using the packet size and average bit rate.
It is unnecessary. Also, for some codecs we're reading more than 1 frame per
packet. Instead we use a private context variable to calculate the bit rate,
stream duration, and packet durations.
Updated FATE seek test, which has slightly different timestamps due to a
more accurate bit rate calculation.
For encoding, frame_size is not a reliable indicator of packet duration.
Also, we don't want to have to force the demuxer to find frame_size for
stream copy to work.
Split off packet parsing into a separate function. Parse full packets at
once and store them in a queue, eliminating the need for tracking
parsing state in AVStream.
The horrible unreadable loop in read_frame_internal() now isn't weirdly
ordered and doesn't contain evil gotos, so it should be much easier to
understand.
compute_pkt_fields() now invents slightly different timestamps for two
raw vc1 tests, due to has_b_frames being set a bit later. They shouldn't
be more wrong (or right) than previous ones.
Make packet buffer a parameter, don't hardcode it to be
AVFormatContext.packet_buffer.
Also move the function higher in the file, since it will be called from
read_frame_internal().
This fixes issues when the bitrate is variable or inaccurate but the
frame size has not been determined yet.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* 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>
The time base is 1 / sample_rate, not 90000.
Several more codecs encode the sample count in the first 4 bytes of the
chunk, so we set the durations accordingly. Also, we can set start_time and
packet duration instead of keeping track of the sample count in the demuxer.
Fixes timestamp calculation.
The FATE reference is updated because timestamp calculations are now more
accurate. Previous timestamps were based on average bit rate.
This allows it to be used with get_bits without the thread of overreads.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
* qatar/master: (29 commits)
amrwb: remove duplicate arguments from extrapolate_isf().
amrwb: error out early if mode is invalid.
h264: change underread for 10bit QPEL to overread.
matroska: check buffer size for RM-style byte reordering.
vp8: disable mmx functions with sse/sse2 counterparts on x86-64.
vp8: change int stride to ptrdiff_t stride.
wma: fix invalid buffer size assumptions causing random overreads.
Windows Media Audio Lossless decoder
rv10/20: Fix slice overflow with checked bitstream reader.
h263dec: Disallow width/height changing with frame threads.
rv10/20: Fix a buffer overread caused by losing track of the remaining buffer size.
rmdec: Honor .RMF tag size rather than assuming 18.
g722: Fix the QMF scaling
r3d: don't set codec timebase.
electronicarts: set timebase for tgv video.
electronicarts: parse the framerate for cmv video.
ogg: don't set codec timebase
electronicarts: don't set codec timebase
avs: don't set codec timebase
wavpack: Fix an integer overflow
...
Conflicts:
libavcodec/arm/vp8dsp_init_arm.c
libavcodec/fraps.c
libavcodec/h264.c
libavcodec/mpeg4videodec.c
libavcodec/mpegvideo.c
libavcodec/msmpeg4.c
libavcodec/pnmdec.c
libavcodec/qpeg.c
libavcodec/rawenc.c
libavcodec/ulti.c
libavcodec/vcr1.c
libavcodec/version.h
libavcodec/wmalosslessdec.c
libavformat/electronicarts.c
libswscale/ppc/yuv2rgb_altivec.c
tests/ref/acodec/g722
tests/ref/fate/ea-cmv
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The container has no timestamps and the framerate isn't stored in the
data either.
The decoder sets codec timebase to experimentally found value 1/15. Do
the same for the demuxer too, it should at least be better than the
default 1/90000.
This fixes duplicate timestamps on mp2 in ts with non seekable input.
It also fixed the fate pva demux timestamps.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
The fields "Number of Bytes" and "Number of Frames" are mixed up. "Bytes"
come first, "Frames" behind.
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Signed-off-by: Alex Converse <alex.converse@gmail.com>
* qatar/master:
h264: error out on invalid bitdepth.
aacsbr: use a swap index for the Y matrix rather than copy buffers.
huffyuv: do not abort on unknown pix_fmt; instead, return an error.
lcl: return negative error codes on decode_init() errors.
rtpenc: Use MB info side data for splitting H263 packets for RFC 2190
h263enc: Add an option for outputting info about MBs as side data
avpacket: Add a function for shrinking already allocated side data
nellymoserdec: Saner and faster IMDCT windowing
Conflicts:
doc/APIchanges
libavcodec/avpacket.c
libavcodec/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This makes the packetization spec compliant for cases where one single
GOB doesn't fit into an RTP packet.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master: (58 commits)
amrnbdec: check frame size before decoding.
cscd: use negative error values to indicate decode_init() failures.
h264: prevent overreads in intra PCM decoding.
FATE: do not decode audio in the nuv test.
dxa: set audio stream time base using the sample rate
psx-str: do not allow seeking by bytes
asfdec: Do not set AVCodecContext.frame_size
vqf: set packet parameters after av_new_packet()
mpegaudiodec: use DSPUtil.butterflies_float().
FATE: add mp3 test for sample that exhibited false overreads
fate: add cdxl test for bit line plane arrangement
vmnc: return error on decode_init() failure.
libvorbis: add/update error messages
libvorbis: use AVFifoBuffer for output packet buffer
libvorbis: remove unneeded e_o_s check
libvorbis: check return values for functions that can return errors
libvorbis: use float input instead of s16
libvorbis: do not flush libvorbis analysis if dsp state was not initialized
libvorbis: use VBR by default, with default quality of 3
libvorbis: fix use of minrate/maxrate AVOptions
...
Conflicts:
Changelog
doc/APIchanges
libavcodec/avcodec.h
libavcodec/dpxenc.c
libavcodec/libvorbis.c
libavcodec/vmnc.c
libavformat/asfdec.c
libavformat/id3v2enc.c
libavformat/internal.h
libavformat/mp3enc.c
libavformat/utils.c
libavformat/version.h
libswscale/utils.c
tests/fate/video.mak
tests/ref/fate/nuv
tests/ref/fate/prores-alpha
tests/ref/lavf/ffm
tests/ref/vsynth1/prores
tests/ref/vsynth2/prores
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This fixes cases where the user had specified one desired MTU
via an option, and the protocol indicates another one.
Signed-off-by: Martin Storsjö <martin@martin.st>
Frame sizes in ID3v2.3 are not synchsafe, they are simply 32be numbers.
In practice this bug is not noticeable unless the frame size takes more
than 7 bits (which is almost never for text frames).
Seeking back on EOF will reset the EOF flag, causing us to re-enter
the loop to find the next marker in the ASF file, thus potentially
causing an infinite loop.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
* qatar/master:
adpcm: Clip step_index values read from the bitstream at the beginning of each frame.
oma: don't read beyond end of leaf_table.
doxygen: Remove documentation for non-existing parameters; misc small fixes.
Indeo3: fix crashes on corrupt bitstreams.
msmpeg4: Replace forward declaration by proper #include.
segment: implement wrap around
avf: reorder AVStream and AVFormatContext
aacdec: Remove erroneous reference to global gain from the out of bounds scalefactor error message.
Conflicts:
libavcodec/indeo3.c
libavformat/avformat.h
libavutil/avutil.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
The properties of the CDCI Descriptor are insufficient to specify
the pixel format for uncompressed picture data. SMPTE 377-1 and
RP224v10 have defined a set of picture coding labels to indicate what
formatting was used.
This patch uses 2 labels to detect UYVY422 or YUYV422 pixel formats.
It defaults to UYVY422 for 8-bit 4:2:2 pictures to support files
that were created before the coding labels were introduced ~2008
The codec pix_fmt default was changed from 0 (PIX_FMT_YUV420P) to
-1 (PIX_FMT_NONE)
Reviewed-by: Baptiste Coudurier <baptiste.coudurier@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This supports detection of uncompressed picture in files that
didn't include a Picture Coding Label. The lables weren't
available until SMPTE 377-1 and RP224v10
Reviewed-by: Baptiste Coudurier <baptiste.coudurier@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This matches the order used for the index table edit rate.
Reviewed-by: Baptiste Coudurier <baptiste.coudurier@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Must check all 16 bytes because there is a planar 10-bit format
label that has equal first 15 bytes
Review-by: Baptiste Coudurier <baptiste.coudurier@gmail.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Provide a way to wrap around the segment index so pseudostreaming
live through a web server and html5 browser is simpler.
Also ensure that 0 (disable) is a valid value across the options
providing wrap around.
* 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>
By validating the index entries while reading, we don't need to
seek at startup to validate the entries. If the error in the
index entries is not pointing to (our definition of) the start
of packets, and there is an index entry pointing at some of the
first packets after the metadata, the invalid index can be discarded
almost immediately.
Signed-off-by: Martin Storsjö <martin@martin.st>
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>
For encoding, AVCodecContext.frame_size is the number of input samples to
send to the encoder and does not necessarily correspond directly to the
timestamps of the output packets.
* qatar/master: (34 commits)
mlp_parser: fix the channel mask value used for the top surround channel
vorbisenc: check all allocations for failure
roqaudioenc: return AVERROR codes instead of -1
roqaudioenc: set correct bit rate
roqaudioenc: use AVCodecContext.frame_size correctly.
roqaudioenc: remove unneeded sample_fmt check
ra144enc: use int16_t* for input samples rather than void*
ra144enc: set AVCodecContext.coded_frame
ra144enc: remove unneeded sample_fmt check
nellymoserenc: set AVCodecContext.coded_frame
nellymoserenc: improve error checking in encode_init()
nellymoserenc: return AVERROR codes instead of -1
libvorbis: improve error checking in oggvorbis_encode_init()
mpegaudioenc: return AVERROR codes instead of -1
libfaac: improve error checking and handling in Faac_encode_init()
avutil: add AVERROR_UNKNOWN
check for coded_frame allocation failure in several audio encoders
audio encoders: do not set coded_frame->key_frame.
g722enc: check for trellis data allocation error
libspeexenc: export encoder delay through AVCodecContext.delay
...
Conflicts:
doc/APIchanges
libavcodec/avcodec.h
libavcodec/fraps.c
libavcodec/kgv1dec.c
libavcodec/libfaac.c
libavcodec/libgsm.c
libavcodec/libvorbis.c
libavcodec/mlp_parser.c
libavcodec/roqaudioenc.c
libavcodec/vorbisenc.c
libavutil/avutil.h
libavutil/error.c
libavutil/error.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
This prevents certain tags with a default value assigned to them (as per
the EBML syntax elements) from ever being assigned a NULL value. Other
parts of the code rely on these being non-NULL (i.e. they don't check for
NULL before e.g. using the string in strcmp() or similar), and thus in
effect this prevents crashes when reading of such specific tags fails,
either because of low memory or because of targeted file corruption.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
* qatar/master:
docs: use -bsf:[vas] instead of -[vas]bsf.
mpegaudiodec: Prevent premature clipping of mp3 input buffer.
lavf: move the packet keyframe setting code.
oggenc: free comment header for all codecs
lcl: error out if uncompressed input buffer is smaller than framesize.
mjpeg: abort decoding if packet is too large.
golomb: use HAVE_BITS_REMAINING() macro to prevent infloop on EOF.
get_bits: add HAVE_BITS_REMAINING macro.
lavf/output-example: use new audio encoding API correctly.
lavf/output-example: more proper usage of the new API.
tiff: Prevent overreads in the type_sizes array.
tiff: Make the TIFF_LONG and TIFF_SHORT types unsigned.
apetag: do not leak memory if avio_read() fails
apetag: propagate errors.
SBR DSP x86: implement SSE sbr_hf_g_filt
SBR DSP x86: implement SSE sbr_sum_square_sse
SBR DSP: use intptr_t for the ixh parameter.
Conflicts:
doc/bitstream_filters.texi
doc/examples/muxing.c
doc/ffmpeg.texi
libavcodec/golomb.h
libavcodec/x86/Makefile
libavformat/oggenc.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
compute_pkt_fields() is for unreliable estimates or guessing. The
keyframe information from the parser is (at least in theory) reliable,
so it should be used even when the other guessing is disabled with the
AVFMT_FLAG_NOFILLIN flag.
Therefore, move setting the packet keyframe flag based on parser
information from compute_pkt_fields() to read_frame_internal().
fixes a memleak for Vorbis and Theora, where the comment header from
avpriv_split_xiph_headers() is replaced by a buffer that must be freed
separately.
* 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>
This allows opting for a lower MTU than what the AVIOContext
indicated, and allows writing into outputs that don't indicate
an MTU at all (such as plain files, which is useful for testing).
This also allows querying for the MTU via the avoption.
Signed-off-by: Martin Storsjö <martin@martin.st>
According to newer RFCs, this packetization scheme should only
be used for interfacing with legacy systems.
Implementing this packetization mode properly requires parsing
the full H263 bitstream to find macroblock boundaries (and knowing
their macroblock and gob numbers and motion vector predictors).
This implementation tries to look for GOB headers (which
can be inserted by using -ps <small number>), but if the GOBs
aren't small enough to fit into the MTU, the packetizer blindly
splits packets at any offset and claims it to be a GOB boundary
(by using Mode A from the RFC). While not correct, this seems
to work with some receivers.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
dxva2: don't check for DXVA_PictureParameters->wDecodedPictureIndex
img2: split muxer and demuxer into separate files
rm: prevent infinite loops for index parsing.
aac: fix infinite loop on end-of-frame with sequence of 1-bits.
mov: Add more HDV and XDCAM FourCCs.
lavf: don't set AVCodecContext.has_b_frames in compute_pkt_fields().
rmdec: when using INT4 deinterleaving, error out if sub_packet_h <= 1.
cdxl: correctly synchronize video timestamps to audio
mlpdec_parser: fix a few channel layouts.
Add channel names to channel_names[] array for channels added in b2890f5
movenc: Buffer the mdat for the initial moov fragment, too
flvdec: Ignore the index if the ignidx flag is set
flvdec: Fix indentation
movdec: Don't parse all fragments if ignidx is set
movdec: Restart parsing root-level atoms at the right spot
prores: use natural integer type for the codebook index
mov: Add support for MPEG2 HDV 720p24 (hdv4)
swscale: K&R formatting cosmetics (part I)
swscale: variable declaration and placement cosmetics
Conflicts:
configure
libavcodec/aacdec.c
libavcodec/mlp_parser.c
libavformat/flvdec.c
libavformat/img2.c
libavformat/isom.h
libavformat/mov.c
libavformat/movenc.c
libswscale/rgb2rgb.c
libswscale/rgb2rgb_template.c
libswscale/yuv2rgb.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Specifically, prevent jumping back in the file for the next index, since
this can lead to infinite loops where we jump between indexes referring
to each other, and don't read indexes that don't fit in the file.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
It is not supposed to be done outside lavc.
This is basically a revert of 818062f2f3.
It is unclear what issue this was supposed to fix, if it reappears again
it will have to be fixed in a more proper place.
The wtv-demux test change is because the sample starts with a B-frame.
We read sub_packet_h / 2 packets per line of data (during deinterleaving),
which equals zero if sub_packet_h <= 1, thus causing us to not read any
data, leading to an infinite loop.
Found-by: Mateusz "j00ru" Jurczyk and Gynvael Coldwind
CC: libav-stable@libav.org
This allows writing QuickTime-compatible fragmented mp4 (with
a non-empty moov atom) to a non-seekable output.
This buffers the mdat for the initial fragment just as it does
for all normal fragments, too. Previously, the resulting
atom structure was mdat,moov, moof,mdat ..., while it now
is moov,mdat, moof,mdat.
Signed-off-by: Martin Storsjö <martin@martin.st>
In nonseekable files, we already stop parsing the toplevel atoms
after finding moov and one mdat. In large seekable files (or files
that are seekable, but slowly, e.g. http), reading all the fragments
at the start can take a considerable amount of time. This allows
opting out from this behaviour.
Signed-off-by: Martin Storsjö <martin@martin.st>
If parsing moov+mdat in a non-seekable file, we currently
abort parsing directly after parsing the header of the mdat
atom. If we want to continue parsing later (if looking to
parse later fragments), we need to skip past the content of the
mdat atom, otherwise we end up parsing the content of the mdat
atom as root level atoms.
Signed-off-by: Martin Storsjö <martin@martin.st>
* qatar/master:
mpegvideo_enc: only allocate output packet when we know there will be output
Add names for more channel layouts to the channel layout map.
sunrast: Add a sample request for RMP_RAW colormap.
avcodec: do not override pts or duration from the audio encoder
Add prores regression test.
Enable already existing rso regression test.
Add regression test for "sox" format muxer/demuxer.
Add dpx encoding regression test.
swscale: K&R formatting cosmetics for PowerPC code (part I/II)
img2: Use ff_guess_image2_codec(filename) shorthand where appropriate.
Clarify licensing information about files borrowed from libjpeg.
Mark mutable static data const where appropriate.
avplay: fix -threads option
dvbsubdec: avoid undefined signed left shift in RGBA macro
mlpdec: use av_log_ask_for_sample()
gif: K&R formatting cosmetics
png: make .long_name more descriptive
movdec: Adjust keyframe flagging in fragmented files
rv34: change most "int stride" into "ptrdiff_t stride".
Conflicts:
avprobe.c
ffplay.c
libavcodec/mlpdec.c
libavcodec/mpegvideo_enc.c
libavcodec/pngenc.c
libavcodec/x86/v210-init.c
libavfilter/vf_boxblur.c
libavfilter/vf_crop.c
libavfilter/vf_drawtext.c
libavfilter/vf_lut.c
libavfilter/vf_overlay.c
libavfilter/vf_pad.c
libavfilter/vf_scale.c
libavfilter/vf_select.c
libavfilter/vf_setpts.c
libavfilter/vf_settb.c
libavformat/img2.c
libavutil/audioconvert.c
tests/codec-regression.sh
tests/lavf-regression.sh
tests/ref/lavf/dpx
tests/ref/vsynth1/prores
tests/ref/vsynth2/prores
Merged-by: Michael Niedermayer <michaelni@gmx.at>
