A two byte sync word is not enough to ensure we got a real syncframe, nor are
all the range checks we do in the first seven bytes. Do therefore an integrity
check for the sync frame in order to prevent the parser from filling avctx with
bogus information.
Signed-off-by: James Almer <jamrial@gmail.com>
Have it only find frame boundaries. The stream props will then be filled once
we have an assembled frame.
Signed-off-by: James Almer <jamrial@gmail.com>
This avoids unnecessary rebuilds of most source files if only the
list of enabled components has changed, but not the other properties
of the build, set in config.h.
Signed-off-by: Martin Storsjö <martin@martin.st>
The new code is analog to how it's done in our mpegaudio parser.
Acked-by: Jun Zhao <barryjzhao@tencent.com>
Signed-off-by: Alexander Strasser <eclipse7@gmx.net>
Now we just use one ADTS raw frame to calculate the bit rate, it's
lead to a larger error when get the duration from bit rate, the
improvement cumulate Nth ADTS frames to get the average bit rate.
e,g used the command get the duration like:
ffprobe -show_entries format=duration -i fate-suite/aac/foo.aac
before this improvement dump the duration=2.173935
after this improvement dump the duration=1.979267
in fact, the real duration can be get by command like:
ffmpeg -i fate-suite/aac/foo.aac -f null /dev/null with time=00:00:01.97
Also update the fate-adtstoasc_ticket3715.
Signed-off-by: Jun Zhao <barryjzhao@tencent.com>
If a frame starts very close to a packet boundary, the start code may
already have been added to the parsing buffer, indicated by a small
negative value of "i", while the header is still being tracked in the
"state" variable.
Reduce the remaining size accordingly, otherwise trying to find the next
frame could skip over the frame header and lump two frames together as
one.
request_channel_layout is a decoder option and it makes no sense
to have it in a parser.
This feature was needed in the past when the decoder was allowed
to reuse the avctx from the demuxer. Nowadays the decoder receives
only the parameters from it, already containing the real channel
layout (and the correct request_channel_layout option).
After initialization the decoder overwrites the channel layout
with the downmixed one that is actually output, so there is no need
to preserve this functionality in the parser.
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com>
* commit 'dc70c19476e76f1118df73b5d97cc76f0e5f6f6c':
lavc: Drop deprecated request_channels related functions
Merged-by: Hendrik Leppkes <h.leppkes@gmail.com>
* qatar/master:
Fix even more missing includes after the common.h removal
build: Factor out rangecoder dependencies to CONFIG_RANGECODER
build: Factor out error resilience dependencies to CONFIG_ERROR_RESILIENCE
x86: avcodec: Consistently name all init files
Add more missing includes after removing the implicit common.h
Add some more missing includes after removing the implicit common.h
Don't include common.h from avutil.h
rtmp: Automatically compute the hash for SWFVerification
Conflicts:
configure
doc/APIchanges
doc/examples/decoding_encoding.c
libavcodec/Makefile
libavcodec/assdec.c
libavcodec/audio_frame_queue.c
libavcodec/avpacket.c
libavcodec/dv_profile.c
libavcodec/dwt.c
libavcodec/libtheoraenc.c
libavcodec/rawdec.c
libavcodec/rv40dsp.c
libavcodec/tiff.c
libavcodec/tiffenc.c
libavcodec/v210dec.h
libavcodec/vc1dsp.c
libavcodec/x86/Makefile
libavfilter/asrc_anullsrc.c
libavfilter/avfilter.c
libavfilter/buffer.c
libavfilter/formats.c
libavfilter/vf_ass.c
libavfilter/vf_drawtext.c
libavfilter/vf_fade.c
libavfilter/vf_select.c
libavfilter/video.c
libavfilter/vsrc_testsrc.c
libavformat/version.h
libavutil/audioconvert.c
libavutil/error.h
libavutil/version.h
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* qatar/master: (36 commits)
adpcmenc: Use correct frame_size for Yamaha ADPCM.
avcodec: add ff_samples_to_time_base() convenience function to internal.h
adx parser: set duration
mlp parser: set duration instead of frame_size
gsm parser: set duration
mpegaudio parser: set duration instead of frame_size
(e)ac3 parser: set duration instead of frame_size
flac parser: set duration instead of frame_size
avcodec: add duration field to AVCodecParserContext
avutil: add av_rescale_q_rnd() to allow different rounding
pnmdec: remove useless .pix_fmts
libmp3lame: support float and s32 sample formats
libmp3lame: renaming, rearrangement, alignment, and comments
libmp3lame: use the LAME default bit rate
libmp3lame: use avpriv_mpegaudio_decode_header() for output frame parsing
libmp3lame: cosmetics: remove some pointless comments
libmp3lame: convert some debugging code to av_dlog()
libmp3lame: remove outdated comment.
libmp3lame: do not set coded_frame->key_frame.
libmp3lame: improve error handling in MP3lame_encode_init()
...
Conflicts:
doc/APIchanges
libavcodec/libmp3lame.c
libavcodec/pcxenc.c
libavcodec/pnmdec.c
libavcodec/pnmenc.c
libavcodec/sgienc.c
libavcodec/utils.c
libavformat/hls.c
libavutil/avutil.h
libswscale/x86/swscale_mmx.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
* newdev/master:
mov: set audio service type for AC-3 from bitstream mode in the 'dac3' atom.
Get audio_service_type for AC-3 based on bitstream mode in the AC-3 parser and decoder, and vice-versa for the AC-3 encoder.
Use audio_service_type to set stream disposition.
Add APIchanges entry for audio_service_type.
Add audio_service_type field to AVCodecContext for encoding and reporting of the service type in the audio bitstream.
configure: in check_ld, place new -l flags before existing ones
support @heading, @subheading, @subsubheading, and @subsubsection in texi2pod.pl
doc: update build system documentation
aacenc: indentation
aacenc: fix the side calculation in search_for_ms
vp8.c: rename EDGE_* to VP8_EDGE_*.
Conflicts:
doc/APIchanges
libavcodec/avcodec.h
libavcodec/version.h
libavcodec/vp8.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Thanks to backwards compatible HE-AAC signalling these values are unreliable.
Originally committed as revision 22194 to svn://svn.ffmpeg.org/ffmpeg/trunk
