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Head down (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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Echoplex (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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Discipline (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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Letting you (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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1 000 000 (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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999 999 (wav version)
26 septembre 2011, par
Mis à jour : Avril 2013
Langue : English
Type : Audio
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Sur d’autres sites (6175)
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FFMPEG update from 5.0 to 6.0 out_0_0 buffer queued [closed]
16 mai 2023, par KevittoI've been using ffmpeg 5.0 for some time, encoding an audio stream to an rtp server, but since I updated to ffmpeg 6.0 I get this :


[out_0_0 @ 0x55ac187b60] 100 buffers queued in out_0_0, something may be wrong.



Below is the ffmpeg call :


ffmpeg -re -f alsa -i default:CARD:card1 -ac 2 -af aresample=async=1 -acodec libopus -b:a 48000 -f rtp "rtp://127.0.0.1:5002"



And here is the full startup log :


ffmpeg version 549430e Copyright (c) 2000-2023 the FFmpeg developers
 built with gcc 10 (Debian 10.2.1-6)
 configuration: --extra-cflags=-I/usr/local/include --extra-ldflags=-L/usr/local/lib --extra-libs='-lpthread -lm -latomic' --arch=arm64 --enable-gmp --enable-gpl --enable-libopus --enable-nonfree --enable-version3 --target-os=linux --enable-pthreads --enable-openssl --enable-hardcoded-tables
 libavutil 58. 2.100 / 58. 2.100
 libavcodec 60. 3.100 / 60. 3.100
 libavformat 60. 3.100 / 60. 3.100
 libavdevice 60. 1.100 / 60. 1.100
 libavfilter 9. 3.100 / 9. 3.100
 libswscale 7. 1.100 / 7. 1.100
 libswresample 4. 10.100 / 4. 10.100
 libpostproc 57. 1.100 / 57. 1.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, alsa, from 'default:CARD=pisound':
 Duration: N/A, start: 1684250059.973334, bitrate: 1536 kb/s
 Stream #0:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
Stream mapping:
 Stream #0:0 -> #0:0 (pcm_s16le (native) -> opus (libopus))
Press [q] to stop, [?] for help
Output #0, rtp, to 'rtp://127.0.0.1:5002':
 Metadata:
 encoder : Lavf60.3.100
 Stream #0:0: Audio: opus, 48000 Hz, stereo, s16, 48 kb/s
 Metadata:
 encoder : Lavc60.3.100 libopus
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 60.3.100
m=audio 5002 RTP/AVP 97
b=AS:48
a=rtpmap:97 opus/48000/2
a=fmtp:97 sprop-stereo=1

size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A 
size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A 
[out_0_0 @ 0x559f530c60] 100 buffers queued in out_0_0, something may be wrong.
size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A 
size= 0kB time=-577014:32:22.77 bitrate= -0.0kbits/s speed=N/A 
size= 0kB time=-00:00:00.00 bitrate= -0.0kbits/s speed=N/A 
size= 0kB time=-00:00:00.00 bitrate= -0.0kbits/s speed=N/A 
[alsa @ 0x559f4db460] ALSA buffer xrun.
size= 6kB time=00:00:02.81 bitrate= 16.7kbits/s speed=0.93x 
size= 6kB time=00:00:02.81 bitrate= 16.7kbits/s speed=0.797x 
size= 6kB time=00:00:02.83 bitrate= 16.6kbits/s speed=0.703x 
size= 6kB time=00:00:02.83 bitrate= 16.6kbits/s speed=0.624x 
size= 6kB time=00:00:02.83 bitrate= 16.6kbits/s speed=0.562x 
size= 6kB time=00:00:02.83 bitrate= 16.6kbits/s speed=0.511x 
[alsa @ 0x559f4db460] ALSA buffer xrun.
size= 10kB time=00:00:05.67 bitrate= 14.8kbits/s speed=0.939x 
size= 10kB time=00:00:05.67 bitrate= 14.8kbits/s speed=0.866x 
size= 10kB time=00:00:05.67 bitrate= 14.8kbits/s speed=0.805x 
size= 10kB time=00:00:05.67 bitrate= 14.8kbits/s speed=0.751x 
size= 10kB time=00:00:05.69 bitrate= 14.8kbits/s speed=0.707x 
[alsa @ 0x559f4db460] ALSA buffer xrun.
size= 13kB time=00:00:05.95 bitrate= 17.8kbits/s speed=0.696x 
size= 16kB time=00:00:08.51 bitrate= 15.2kbits/s speed=0.939x 
size= 16kB time=00:00:08.51 bitrate= 15.2kbits/s speed=0.89x 
size= 16kB time=00:00:08.53 bitrate= 15.2kbits/s speed=0.847x 
size= 16kB time=00:00:08.53 bitrate= 15.2kbits/s speed=0.806x 
size= 16kB time=00:00:08.53 bitrate= 15.2kbits/s speed=0.77x 
[alsa @ 0x559f4db460] ALSA buffer xrun.
size= 21kB time=00:00:11.37 bitrate= 14.8kbits/s speed=0.981x 



I tried changing the output to
-f null /dev/null
to see if the rtp was the issue, but I get the same thing. I made sure the user running it was a member to the "audio" group andarecord -l
andaplay -l
both show the card with the right name and information. I even tried to use its hw code instead of the default name, and same issue.

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'unsupported input sample rate set' error while converting mkv to mp3 with ffmpeg on python
15 décembre 2020, par Agent MerlotI'm getting this error on trying to convert some mkv files to mp3 via python. Nearly all files got converted, but some are facing this issue.

https://cdn.discordapp.com/attachments/663255565451001866/788424224661569596/Error.txt

Please help me fix this issue.


ffmpeg output extracted from the discord link above :


ffmpeg version git-2020-06-04-7f81785 Copyright (c) 2000-2020 the FFmpeg developers
 built with gcc 9.3.1 (GCC) 20200523
 configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
 libavutil 56. 49.100 / 56. 49.100
 libavcodec 58. 90.100 / 58. 90.100
 libavformat 58. 44.100 / 58. 44.100
 libavdevice 58. 9.103 / 58. 9.103
 libavfilter 7. 84.100 / 7. 84.100
 libswscale 5. 6.101 / 5. 6.101
 libswresample 3. 6.100 / 3. 6.100
 libpostproc 55. 6.100 / 55. 6.100
Input #0, matroska,webm, from 'J:\DC ED\Original\045 'Kimi no Egao ga Nani Yori mo Suki Datta' by 'Chicago Poodle'.mkv':
 Metadata:
 title : 045 'Kimi no Egao ga Nani Yori mo Suki Datta' by 'Chicago Poodle'.mkv
 COPYRIGHT : © 2013 APTX4869 Fansub
 creation_time : 2020-11-18T05:03:06.000000Z
 COMPOSER : Chicago Poodle
 ENCODER : Lavf58.44.100
 Duration: 00:01:20.04, start: 0.000000, bitrate: 2023 kb/s
 Stream #0:0(jpn): Video: hevc (Main), yuv420p(tv), 1440x1080 [SAR 4:3 DAR 16:9], 23.98 fps, 23.98 tbr, 1k tbn, 23.98 tbc (default)
 Metadata:
 title : VIDEO[AVC]
 ENCODER : Lavc58.90.100 libx265
 BPS-eng : 1831510
 DURATION-eng : 00:01:20.039000000
 NUMBER_OF_FRAMES-eng: 1919
 NUMBER_OF_BYTES-eng: 18324032
 _STATISTICS_WRITING_APP-eng: mkvmerge v49.0.0 ('Sick Of Losing Soulmates') 64-bit
 _STATISTICS_WRITING_DATE_UTC-eng: 2020-11-18 05:03:06
 _STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES
 Stream #0:1(jpn): Audio: aac (LC), 96000 Hz, stereo, fltp (default)
 Metadata:
 title : AUDIO[AAC]
 BPS-eng : 188626
 DURATION-eng : 00:01:19.999000000
 NUMBER_OF_FRAMES-eng: 3750
 NUMBER_OF_BYTES-eng: 1886246
 _STATISTICS_WRITING_APP-eng: mkvmerge v49.0.0 ('Sick Of Losing Soulmates') 64-bit
 _STATISTICS_WRITING_DATE_UTC-eng: 2020-11-18 05:03:06
 _STATISTICS_TAGS-eng: BPS DURATION NUMBER_OF_FRAMES NUMBER_OF_BYTES
Stream mapping:
 Stream #0:1 -> #0:0 (aac (native) -> mp3 (mp3_mf))
Press [q] to stop, [?] for help
[mp3_mf @ 000002142f4a5fc0] MFT name: 'MP3 Encoder ACM Wrapper MFT'
[mp3_mf @ 000002142f4a5fc0] unsupported input sample rate set
Error initializing output stream 0:0 -- Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height
Conversion failed!



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A Comprehensive Guide to Robust Digital Marketing Analytics
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