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Sintel MP4 Surround 5.1 Full
13 mai 2011, par
Mis à jour : Février 2012
Langue : English
Type : Video
Autres articles (29)
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Sur d’autres sites (2430)
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When I append a silent audio (mp3) to an existing list of audio it garbles the final audio ?
6 février 2020, par MarieAfter several hours I have narrowed down the issue with the garbled audio to be the 2-seconds silence audio mp3 I am appending (I think I had produced it once with Wavelab)
However, I tried using ffmpeg according to a post to produce a similar 2 seconds audio but it too will corrupt/garble/chop voice in the final concatenation of audio files.
ffmpeg -f lavfi -i anullsrc=r=44100:cl=mono -t 2 -q:a 9 -acodec libmp3lame SILENCE_2sec.MP3
I typically will have several audio files to concatenate together but for simplicity I have able to narrow it to a couple of files simplifying to the following script. A simple Windows batch file you should be able to use and reproduce the issue at your end.
rem
rem
SET EXE="S:\_BINS\FFmpeg 4.2.1 20200112\bin\ffmpeg.exe"
SET ROOTPATH=.\
SET IN_FILE="%ROOTPATH%MyList.txt"
ECHO file '%ROOTPATH%HELLO.mp3' > MyList.txt
ECHO file 'SILENCE_2sec.MP3' >> MyList.txt
SET OPTIONS= -f concat -safe 0 -i %IN_FILE% -c copy -y
SET OUT_FILE="%ROOTPATH%CONCATENATED_AUDIO_2.MP3"
SET INFO_FILE="INFO.TXT"
%EXE% %OPTIONS% %OUT_FILE% 1> %INFO_FILE% 2>&1
ECHO ======================== >> %INFO_FILE%
ECHO IN_FILE=%IN_FILE% >> %INFO_FILE%
ECHO EXE=%EXE% >> %INFO_FILE%
ECHO OPTIONS=%OPTIONS% >> %INFO_FILE%
ECHO ======================== >> %INFO_FILE%Here is the console info output from the ffmpeg, let me know if you need other output include ones from ffprobe
ffmpeg version git-2020-01-10-3d894db Copyright (c) 2000-2020 the FFmpeg developers
built with gcc 9.2.1 (GCC) 20191125
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-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-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
libavutil 56. 38.100 / 56. 38.100
libavcodec 58. 65.103 / 58. 65.103
libavformat 58. 35.101 / 58. 35.101
libavdevice 58. 9.103 / 58. 9.103
libavfilter 7. 70.101 / 7. 70.101
libswscale 5. 6.100 / 5. 6.100
libswresample 3. 6.100 / 3. 6.100
libpostproc 55. 6.100 / 55. 6.100
[mp3 @ 000000000036af80] Estimating duration from bitrate, this may be inaccurate
Input #0, concat, from '.\MyList.txt':
Duration: N/A, start: 0.000000, bitrate: 32 kb/s
Stream #0:0: Audio: mp3, 24000 Hz, mono, fltp, 32 kb/s
Output #0, mp3, to '.\CONCATENATED_AUDIO_2.MP3':
Metadata:
TSSE : Lavf58.35.101
Stream #0:0: Audio: mp3, 24000 Hz, mono, fltp, 32 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
[mp3 @ 0000000000372d00] Application provided invalid, non monotonically increasing dts to muxer in stream 0: 17280 >= 17255
size= 11kB time=00:00:02.73 bitrate= 33.2kbits/s speed=2.73e+03x
video:0kB audio:11kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 2.137446%
========================
IN_FILE=".\MyList.txt"
EXE="S:\_BINS\FFmpeg 4.2.1 20200112\bin\ffmpeg.exe"
OPTIONS= -f concat -safe 0 -i ".\MyList.txt" -c copy -y
========================I believe I am running FFmpeg 4.2.1, recently installed (20200112)
You may produce the HELLO.mp3 by saving the following link
https://translate.google.com.vn/translate_tts?en=UTF-8&q=Hello+&tl=en&client=tw-ob
FYI, I am still a novice of ffmpeg and using it more like a black box with the help I received in this very super forum.
Please be as explicit as you can with command line options on how I can fix this issue.
Thank you.Additional Hints Debugging :
If I append more files after the silence audio it seems that the silence audio impacts (garbles, chops) the previous audio.
You may try the following for the list of audio files input.ECHO file '%ROOTPATH%HELLO.mp3' > MyList.txt
ECHO file 'SILENCE_2sec.MP3' >> MyList.txt
ECHO file '%ROOTPATH%HELLO.mp3' >> MyList.txt
ECHO file '%ROOTPATH%HELLO.mp3' >> MyList.txtI typically add one or more silence file to derive a post silence effect after the actual audio. That’s my current logic. However if you have an alternative to appending a silence in the process of concatenating several audio files or appending x-seconds silence to an existing audio file. I can use that method as well from my coding.
Thank you.
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Picture size 0x0 is invalid : trying to downsample 4k raw video into 1080 using ffmpeg
25 février 2021, par neetu siggerffmpeg -i CityAlley_3840x2160_50fps_8bit.yuv -vf scale=1920:1080 CityAlly.yuv



But I got the error :


ffmpeg version 4.2.4-1ubuntu0.1 Copyright (c) 2000-2020 the FFmpeg developers
 built with gcc 9 (Ubuntu 9.3.0-10ubuntu2)
 configuration: --prefix=/usr --extra-version=1ubuntu0.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-nvenc --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
 libavutil 56. 31.100 / 56. 31.100
 libavcodec 58. 54.100 / 58. 54.100
 libavformat 58. 29.100 / 58. 29.100
 libavdevice 58. 8.100 / 58. 8.100
 libavfilter 7. 57.100 / 7. 57.100
 libavresample 4. 0. 0 / 4. 0. 0
 libswscale 5. 5.100 / 5. 5.100
 libswresample 3. 5.100 / 3. 5.100
 libpostproc 55. 5.100 / 55. 5.100
[IMGUTILS @ 0x7fffc0509b50] Picture size 0x0 is invalid
CityAlley_3840x2160_50fps_8bit.yuv: Invalid argument



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Need help configuring FFMPEG to work with a webcams h264 stream
9 août 2020, par The Welsh DragonI have been trying to get a H264 stream from a H264 usb webcam working but I am not making much progress so I'm hoping someone knows FFMPEG better than me !


There are dozens of questions/answers on SO but none solve my problem.


In short, I get a very pixelated (or sometimes mostly green) screen. I am using VLC to test the stream which is coming via an RTSP server. I am using FFMPEG to copy the webcam stream to the local RTSP server.


The webcam also supports YUYV which I can get working - it is just the h264 stream causing me problems.


So this is how the device is presented :


H264 USB Camera: USB Camera (usb-20980000.usb-1):
 /dev/video0
 /dev/video1
 /dev/video2
 /dev/video3



/dev/video0 is the YUYV and MPEG stream
/dev/video2 is the h264 stream that has the following capabilities :


ioctl: VIDIOC_ENUM_FMT
 Type: Video Capture

 [0]: 'H264' (H.264, compressed)
 Size: Discrete 1920x1080
 Interval: Discrete 0.033s (30.000 fps)
 Interval: Discrete 0.040s (25.000 fps)
 Interval: Discrete 0.067s (15.000 fps)
 Interval: Discrete 0.033s (30.000 fps)
 Interval: Discrete 0.040s (25.000 fps)
 Interval: Discrete 0.067s (15.000 fps)
 Size: Discrete 1280x720
 Interval: Discrete 0.033s (30.000 fps)
 Interval: Discrete 0.040s (25.000 fps)
 Interval: Discrete 0.067s (15.000 fps)
 Size: Discrete 800x600
 Interval: Discrete 0.033s (30.000 fps)
 Interval: Discrete 0.040s (25.000 fps)
 Interval: Discrete 0.067s (15.000 fps)
 Size: Discrete 640x480
 Interval: Discrete 0.033s (30.000 fps)
 Interval: Discrete 0.040s (25.000 fps)
 Interval: Discrete 0.067s (15.000 fps)
 Size: Discrete 640x360
 Interval: Discrete 0.033s (30.000 fps)
 Interval: Discrete 0.040s (25.000 fps)
 Interval: Discrete 0.067s (15.000 fps)
 Size: Discrete 352x288
 Interval: Discrete 0.033s (30.000 fps)
 Interval: Discrete 0.040s (25.000 fps)
 Interval: Discrete 0.067s (15.000 fps)
 Size: Discrete 320x240
 Interval: Discrete 0.033s (30.000 fps)
 Interval: Discrete 0.040s (25.000 fps)
 Interval: Discrete 0.067s (15.000 fps)
 Size: Discrete 1920x1080
 Interval: Discrete 0.033s (30.000 fps)
 Interval: Discrete 0.040s (25.000 fps)
 Interval: Discrete 0.067s (15.000 fps)
 Interval: Discrete 0.033s (30.000 fps)
 Interval: Discrete 0.040s (25.000 fps)
 Interval: Discrete 0.067s (15.000 fps)



I have tried various resolutions, the smaller giving slightly less pixelated images but none are usable and definitely dont compare to the YUYV high resolution results.


This (YUYV) command works :


ffmpeg -input_format yuyv422 -f video4linux2 -s 1280x720 -r 10 -i /dev/video0 -c:v h264_omx -r 10 -b:v 2M -an -f rtsp rtsp://localhost:80/live/stream



These two h264 options dont work :


ffmpeg -input_format h264 -f video4linux2 -video_size 1920x1080 -framerate 30 -i /dev/video0 -c:v copy -an -f rtsp rtsp://localhost:80/live/stream



ffmpeg -re -i /dev/video2 -video_size 800x600 -framerate 15 -pix_fmt yuv420p -tune zerolatency -c:v copy -an -f rtsp rtsp://localhost:80/live/stream



For that last command the FFMPEG output looks like this :


ffmpeg version git-2020-08-07-6fdf3cc Copyright (c) 2000-2020 the FFmpeg developers
 built with gcc 8 (Raspbian 8.3.0-6+rpi1)
 configuration: --extra-ldflags=-latomic --arch=armel --target-os=linux --enable-gpl --enable-omx --enable-omx-rpi --enable-nonfree --enable-libfreetype --enable-libx264 --enable-libmp3lame --enable-mmal --enable-indev=alsa --enable-outdev=alsa
 libavutil 56. 58.100 / 56. 58.100
 libavcodec 58.100.100 / 58.100.100
 libavformat 58. 50.100 / 58. 50.100
 libavdevice 58. 11.101 / 58. 11.101
 libavfilter 7. 87.100 / 7. 87.100
 libswscale 5. 8.100 / 5. 8.100
 libswresample 3. 8.100 / 3. 8.100
 libpostproc 55. 8.100 / 55. 8.100
Input #0, video4linux2,v4l2, from '/dev/video2':
 Duration: N/A, start: 1353.265049, bitrate: N/A
 Stream #0:0: Video: h264 (Main), yuv420p(progressive), 1920x1080, 30 fps, 30 tbr, 1000k tbn, 2000k tbc
[udp @ 0x38c29f0] attempted to set receive buffer to size 393216 but it only ended up set as 360448
[udp @ 0x38d7b50] attempted to set receive buffer to size 393216 but it only ended up set as 360448
Output #0, rtsp, to 'rtsp://localhost:80/live/stream':
 Metadata:
 encoder : Lavf58.50.100
 Stream #0:0: Video: h264 (Main), yuv420p(progressive), 1920x1080, q=2-31, 30 fps, 30 tbr, 90k tbn, 1000k tbc
Stream mapping:
 Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
[rtsp @ 0x38fd890] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly
[rtsp @ 0x38fd890] Non-monotonous DTS in output stream 0:0; previous: 0, current: 0; changing to 1. This may result in incorrect timestamps in the output file.
frame= 348 fps= 18 q=-1.0 size=N/A time=00:00:21.03 bitrate=N/A speed=1.09x



The issue looks like it is bandwidth related or the lack of processing power in the device being used BUT the YUYV works at a high resolution and (taking a completely different approach i.e. not using FFMPEG) I can get a very decent MPEG stream working on the same device.


So any FFMPEG experts out there who can help me with getting the correct parameters for a h264 stream ?