Opened 5 years ago
Last modified 5 years ago
#8131 new defect
s302m in combination with loudnorm filter
Reported by: | jb_alvarado | Owned by: | |
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Priority: | normal | Component: | undetermined |
Version: | git-master | Keywords: | s302m loudnorm |
Cc: | Blocked By: | ||
Blocking: | Reproduced by developer: | no | |
Analyzed by developer: | no |
Description
Summary of the bug:
The audio codec s302m produce in combination with loudnorm filter an error:
[s302m @ 00000275a32cba00] number of samples in frame too big Audio encoding failed
How to reproduce:
% ffmpeg -f lavfi -i testsrc=size=1280x720:rate=25:duration=5 -f lavfi -i sine=frequency=1000:duration=5 -af "loudnorm=I=-23.0:TP=-1.5:LRA=11.0:dual_mono=true" -ac 2 -c:a s302m -strict -2 -f null -
Uncut output:
ffmpeg -f lavfi -i testsrc=size=1280x720:rate=25:duration=5 -f lavfi -i sine=frequency=1000:duration=5 -af "loudnorm=I=-23.0:TP=-1.5:LRA=11.0:dual_mono=true" -ac 2 -c:a s302m -strict -2 -f null - ffmpeg version N-94713-g9c369b3222 Copyright (c) 2000-2019 the FFmpeg developers built with gcc 9.2.0 (Rev1, Built by MSYS2 project) configuration: --enable-sdl2 --enable-gmp --enable-libmp3lame --enable-libvpx --enable-libx264 --enable-libx265 --enable-fontconfig --enable-libfreetype --enable-libmysof a --enable-libopenjpeg --enable-libsoxr --enable-libtwolame --enable-libwavpack --enable-libwebp --enable-libxml2 --enable-libzimg --enable-gpl --enable-avisynth --enable-ch romaprint --enable-libfdk-aac --enable-libfribidi --enable-librubberband --enable-libzmq --enable-opengl --enable-libsrt --enable-libaom --enable-schannel --extra-cflags=-DL IBTWOLAME_STATIC --extra-cflags=-DCHROMAPRINT_NODLL --extra-libs=-lstdc++ --extra-cflags=-DZMQ_STATIC --extra-cflags=-DLIBXML_STATIC --enable-version3 --enable-nonfree --dis able-stripping libavutil 56. 33.100 / 56. 33.100 libavcodec 58. 56.100 / 58. 56.100 libavformat 58. 31.104 / 58. 31.104 libavdevice 58. 9.100 / 58. 9.100 libavfilter 7. 58.101 / 7. 58.101 libswscale 5. 6.100 / 5. 6.100 libswresample 3. 6.100 / 3. 6.100 libpostproc 55. 6.100 / 55. 6.100 Input #0, lavfi, from 'testsrc=size=1280x720:rate=25:duration=5': Duration: N/A, start: 0.000000, bitrate: N/A Stream #0:0: Video: rawvideo (RGB[24] / 0x18424752), rgb24, 1280x720 [SAR 1:1 DAR 16:9], 25 tbr, 25 tbn, 25 tbc Input #1, lavfi, from 'sine=frequency=1000:duration=5': Duration: N/A, start: 0.000000, bitrate: 705 kb/s Stream #1:0: Audio: pcm_s16le, 44100 Hz, mono, s16, 705 kb/s Stream mapping: Stream #0:0 -> #0:0 (rawvideo (native) -> wrapped_avframe (native)) Stream #1:0 -> #0:1 (pcm_s16le (native) -> s302m (native)) Press [q] to stop, [?] for help Output #0, null, to 'pipe:': Metadata: encoder : Lavf58.31.104 Stream #0:0: Video: wrapped_avframe, rgb24(progressive), 1280x720 [SAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 25 fps, 25 tbn, 25 tbc Metadata: encoder : Lavc58.56.100 wrapped_avframe Stream #0:1: Audio: s302m, 48000 Hz, stereo, s32 (24 bit), 2688 kb/s Metadata: encoder : Lavc58.56.100 s302m [Parsed_sine_0 @ 00000275a32aef00] EOF timestamp not reliable [s302m @ 00000275a32cba00] number of samples in frame too big Audio encoding failed Conversion failed!
,asetnsamples=n=1024
could fix it, but produce randomly, depending from input, other errors and warnings.
Attachments (1)
Change History (9)
comment:1 by , 5 years ago
comment:2 by , 5 years ago
I tried -frame_size 1024, but it had no effect. When I use the filter asetnsamples=n=1024, it works with the command on top (sine source), but if I use a real audio source I get:
[out_0_0 @ 000002660d73d200] 100 buffers queued in out_0_0, something may be wrong.
In debug mode it looks also that not all packets are encoded:
cur_dts is invalid st:0 (0) [init:1 i_done:0 finish:0] (this is harmless if it occurs once at the start per stream) Last message repeated 139 times [out_0_0 @ 0000021e3727e140] 100 buffers queued in out_0_0, something may be wrong. [out_0_0 @ 0000021e3727e140] EOF on sink link out_0_0:default. No more output streams to write to, finishing. size=N/A time=00:00:25.12 bitrate=N/A speed=21.2x video:0kB audio:8263kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown Input file #0 (D:/stream/loudness/Sprache Leise.wav): Input stream #0:0 (audio): 1178 packets read (4823244 bytes); 1178 frames decoded (1205811 samples); Total: 1178 packets (4823244 bytes) demuxed Output file #0 (pipe:): Output stream #0:0 (audio): 4712 frames encoded (1206016 samples); 4712 packets muxed (8460960 bytes); Total: 4712 packets (8460960 bytes) muxed 1178 frames successfully decoded, 0 decoding errors [AVIOContext @ 0000021e3725e200] Statistics: 4888878 bytes read, 1 seeks
comment:3 by , 5 years ago
I can not reproduce in any way issue with asetnsamples, please provide input file.
by , 5 years ago
Attachment: | audio-test.zip added |
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comment:4 by , 5 years ago
Ok, here the file. I use this command:
ffmpeg.exe -i "Sprache Leise.wav" -af "loudnorm=I=-23.0:TP=-1.5:LRA=11.0,asetnsamples=n=1024" -c:a s302m -strict -2 -f null -
comment:5 by , 5 years ago
s302 encoder is marked as experimental for the reason that proper muxing need correct set for frame size in each packet.
100 queued buffers message is caused by bug/misfeature in loudnorm filter.
comment:6 by , 5 years ago
Yes I know is experimental, but s302 is the only supported lossless codec for mpegts. And in general works very nice in my project, better then mp2 and aac.
Is there a workaround to prevent this bug?
comment:7 by , 5 years ago
Keywords: | loudnorm added |
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comment:8 by , 5 years ago
I think I have now the "workaround". The correct numbers of samples per frame needs to be calculated. For example for a 25 FPS Video: 192000 / 25 = 7680
, this can be used in asetnsamples=7680, or in combination with aresample=48000: 48000 / 25 = 1920
-> asetnsamples=1920.
Is a bit confusing, because there is not always a video line and it also looks like that higher numbers like 8192 or also working...
The limit in s302 is by design, max size of packet is 16bit number.
Can't you use -frame_size to set number of samples?