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* framehash: convert to new channel layout APIAnton Khirnov2022-03-151-3/+3
| | | | Signed-off-by: James Almer <jamrial@gmail.com>
* avformat/framecrcenc: print basic side data information againJames Almer2021-05-181-3/+3
| | | | | | This partially reverts c6ae560a18d67b9ddaa25a0338b7fb55e3312e57. Signed-off-by: James Almer <jamrial@gmail.com>
* lavf/framecrcenc: do not hash side dataAnton Khirnov2021-05-091-3/+3
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | There are no guarantees that all side data types have the same representation on all platforms. Tests that change output due to this: id3v2-priv-remux, cover-art-mp3-id3v2-remux, gapless-mp3: SKIP_SAMPLES, which is tested by fate-gapless-mp3-side-data matroska-vp8-alpha-remux: MATROSKA_BLOCKADDITIONAL, which is tested by remux itself (side data is written into output) matroska-mastering-display-metadata: MASTERING_DISPLAY_METADATA and CONTENT_LIGHT_LEVEL, which are tested by ffprobe invocation in the same test matroska-spherical-mono-remux: STEREO3D and SPHERICAL, which are tested by ffprobe invocation in the same test segment-mp4-to-ts: MPEGTS_STREAM_ID, which is tested by ts remuxing tests webm-webvtt-remux: WEBVTT_IDENTIFIER/SETTINGS, which is tested by the ffprobe invocation in the same test mxf-d10-user-comments: CPB_PROPERTIES, which is tested by mxf-probe-d10 mov-cover-image: SKIP_SAMPLES, which is tested for mov by mov-aac-2048-priming copy-trac3074: AUDIO_SERVICE_TYPE, which is tested by fate-hls-fmp4_ac3
* Revert "lavf/mp3dec: don't adjust start time; packets are not adjusted."Michael Niedermayer2020-06-081-1/+1
| | | | | | | | | This causes regressions in end to end timestamps with mp3s and ffmpeg. The revert is to avoid this regression in the 4.3 release See: [FFmpeg-devel] [PATCH] Don't adjust start time for MP3 files; packets are not adjusted. This reverts commit 460132c9980f8a1f501a1f69477bca49e1641233.
* lavf/mp3dec: don't adjust start time; packets are not adjusted.Dale Curtis2020-05-271-1/+1
| | | | | | | | | | | | | | | | | | | | 7546ac2fee4 made it so that the start_time for mp3 files is adjusted for skip_samples. However, this appears incorrect because subsequent packet timestamps are not adjusted and skip_samples are applied by deleting data from a packet without changing the timestamp. E.g., we are told the start_time is ~25ms and we get a packet with a timestamp of 0 that has had the skip_samples discarded from it. As such rendering engines may incorrectly discard everything prior to the 25ms thinking that is where playback should officially start. Since the samples were deleted without adjusting timestamps though, the true start_time is still 0. Other formats like MP4 with edit lists will adjust both the start time and the timestamps of subsequent packets to avoid this issue. Signed-off-by: Dale Curtis <dalecurtis@chromium.org> Signed-off-by: Anton Khirnov <anton@khirnov.net>
* avformat/framehash: also print channel layout as a stringJames Almer2016-11-051-3/+3
| | | | | | | | This should be more useful for users since numerical values for channel layout can be confusing and unintuitive. Reviewed-by: Michael Niedermayer <michael@niedermayer.cc> Signed-off-by: James Almer <jamrial@gmail.com>
* avformat/framecrc: enable new outputJames Almer2016-04-131-3/+3
| | | | | Reviewed-by: Michael Niedermayer <michael@niedermayer.cc> Signed-off-by: James Almer <jamrial@gmail.com>
* avcodec/mpegaudio_parser: Discard ID3v1 tag at the endMichael Niedermayer2015-06-241-1/+1
| | | | | | | | | | | | | | Ideally this should be discarded by the demuxer but this is not possible without fully parsing which would be then very similar to this. The current ID3v1 discard code in the demuxer does not work and will be removed in a subsequent commit The discard code could be adjusted if needed to also discard tags at other locations than the end or to limit this possibly to input from the mp3 demuxer or even to move the discarding to the decoder. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* avformat/mp3dec: allow enabling generic seek modewm42015-04-221-1/+1
| | | | | | | | | | | | | | "-usetoc 2" now invokes the generic seek and indexing mode. This mode skips data until the seek target is reached, and this is exact. It also makes gapless audio actually work if a seek past the start of the file is involved. Change the fate-gapless-mp3 test to use the new mode, and move the old one to fate-gapless-mp3-toc (since the test forces use of the Xing TOC). The new mode has a different result for the seek - this result is actually correct. Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* fate: gapless: fix mp3 testswm42015-04-201-2/+2
| | | | | | | | | | | | | | | | | | Seeking to a negative time did not have the desired effect of seeking to the next valid position (the file start). On the other hand, just "-ss 0" will normally seek to a position higher than 0, because it adds the start time of the file. (The start time is not 0 because the gapless code skips a few samples from the start.) Fix this by using the "-seek_timestamp 1" option, which makes "-ss 0" do what you'd expect it would do. Also put the -ss option at the right place, before -i. This actually makes it seek, instead of something completely else. The ".out-3" test is no different in the -usetoc 0/1 cases, because the seeking is inaccurate (in both cases). Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* fate: gapless: test seeking to a specific positionwm42015-04-171-0/+1
| | | | Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* avcodec/mpegaudiodec_template: use double to build csa tablesMichael Niedermayer2015-04-171-2/+2
| | | | | | | Fixes rounding difference between 32bit x86 and 64bit Fixes fate failure with gapless mp3 Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
* fate: add mp3 gapless testwm42015-04-161-0/+4
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
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