I am trying to transcode an Apple Prores 444 to H.264 using qsv without success.
If I use this command line:
ffmpeg -i 10minute_Pipeline_Test.mov -c:v h264_qsv -c:a aac -pix_fmt qsv chris.mp4
I get:
ffmpeg version 4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 9 (Ubuntu 9.3.0-17ubuntu1~20.04)
configuration: --prefix=/root/ffmpeg_build --extra-cflags=-I/root/ffmpeg_build/include --extra-ldflags=-L/root/ffmpeg_build/lib --bindir=/root/bin --extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-nonfree --enable-libmfx
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
libswscale 5. 5.100 / 5. 5.100
libswresample 3. 5.100 / 3. 5.100
libpostproc 55. 5.100 / 55. 5.100
Guessed Channel Layout for Input Stream #0.2 : mono
Guessed Channel Layout for Input Stream #0.3 : mono
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '10minute_Pipeline_Test.mov':
Metadata:
major_brand : qt
minor_version : 537134592
compatible_brands: qt
creation_time : 2020-12-19T12:43:38.000000Z
com.apple.quicktime.author:
com.apple.quicktime.comment:
com.apple.quicktime.copyright:
com.apple.quicktime.description:
com.apple.quicktime.director:
com.apple.quicktime.genre:
com.apple.quicktime.information:
com.apple.quicktime.keywords:
com.apple.quicktime.producer:
com.apple.quicktime.displayname:
timecode : 12:43:37;28
Duration: 00:10:06.72, start: 0.000000, bitrate: 167429 kb/s
Stream #0:0(eng): Data: none (tmcd / 0x64636D74)
Metadata:
creation_time : 1970-01-04T00:49:14.000000Z
timecode : 12:43:37;28
Stream #0:1(eng): Video: prores (Standard) (apcn / 0x6E637061), yuv422p10le(tv, GBR, progressive), 1280x720, 164985 kb/s, SAR 1:1 DAR 16:9, 59.94 fps, 59.94 tbr, 60k tbn, 60k tbc (default)
Metadata:
creation_time : 1970-01-01T00:00:04.000000Z
Stream #0:2(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, s32 (24 bit), 1152 kb/s (default)
Stream #0:3(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, s32 (24 bit), 1152 kb/s (default)
Metadata:
creation_time : 2003-10-05T11:26:56.000000Z
File 'chris.mp4' already exists. Overwrite ? [y/N] y
Stream mapping:
Stream #0:1 -> #0:0 (prores (native) -> h264 (h264_qsv))
Stream #0:2 -> #0:1 (pcm_s24le (native) -> aac (native))
Press [q] to stop, [?] for help
[h264_qsv # 0x56265b81a800] Selected ratecontrol mode is unsupported
[h264_qsv # 0x56265b81a800] Low power mode is unsupported
[h264_qsv # 0x56265b81a800] Current frame rate is unsupported
[h264_qsv # 0x56265b81a800] Current picture structure is unsupported
[h264_qsv # 0x56265b81a800] Current resolution is unsupported
[h264_qsv # 0x56265b81a800] Current pixel format is unsupported
[h264_qsv # 0x56265b81a800] some encoding parameters are not supported by the QSV runtime. Please double check the input parameters.
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!
user#NUC:~$ ffmpeg -i 10minute_Pipeline_Test.mov -c:v h264_qsv -c:a aac -pix_fmt qsv chris.mp4
ffmpeg version 4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 9 (Ubuntu 9.3.0-17ubuntu1~20.04)
configuration: --prefix=/root/ffmpeg_build --extra-cflags=-I/root/ffmpeg_build/include --extra-ldflags=-L/root/ffmpeg_build/lib --bindir=/root/bin --extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-nonfree --enable-libmfx
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
libswscale 5. 5.100 / 5. 5.100
libswresample 3. 5.100 / 3. 5.100
libpostproc 55. 5.100 / 55. 5.100
Guessed Channel Layout for Input Stream #0.2 : mono
Guessed Channel Layout for Input Stream #0.3 : mono
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '10minute_Pipeline_Test.mov':
Metadata:
major_brand : qt
minor_version : 537134592
compatible_brands: qt
creation_time : 2020-12-19T12:43:38.000000Z
com.apple.quicktime.author:
com.apple.quicktime.comment:
com.apple.quicktime.copyright:
com.apple.quicktime.description:
com.apple.quicktime.director:
com.apple.quicktime.genre:
com.apple.quicktime.information:
com.apple.quicktime.keywords:
com.apple.quicktime.producer:
com.apple.quicktime.displayname:
timecode : 12:43:37;28
Duration: 00:10:06.72, start: 0.000000, bitrate: 167429 kb/s
Stream #0:0(eng): Data: none (tmcd / 0x64636D74)
Metadata:
creation_time : 1970-01-04T00:49:14.000000Z
timecode : 12:43:37;28
Stream #0:1(eng): Video: prores (Standard) (apcn / 0x6E637061), yuv422p10le(tv, GBR, progressive), 1280x720, 164985 kb/s, SAR 1:1 DAR 16:9, 59.94 fps, 59.94 tbr, 60k tbn, 60k tbc (default)
Metadata:
creation_time : 1970-01-01T00:00:04.000000Z
Stream #0:2(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, s32 (24 bit), 1152 kb/s (default)
Stream #0:3(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, s32 (24 bit), 1152 kb/s (default)
Metadata:
creation_time : 2003-10-05T11:26:56.000000Z
File 'chris.mp4' already exists. Overwrite ? [y/N] y
Stream mapping:
Stream #0:1 -> #0:0 (prores (native) -> h264 (h264_qsv))
Stream #0:2 -> #0:1 (pcm_s24le (native) -> aac (native))
Press [q] to stop, [?] for help
Impossible to convert between the formats supported by the filter 'Parsed_null_0' and the filter 'auto_scaler_0'
Error reinitializing filters!
Failed to inject frame into filter network: Function not implemented
Error while processing the decoded data for stream #0:1
Conversion failed!
If I use:
ffmpeg -i 10minute_Pipeline_Test.mov -c:v h264_qsv -c:a aac chris.mp4
I get:
ffmpeg version 4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 9 (Ubuntu 9.3.0-17ubuntu1~20.04)
configuration: --prefix=/root/ffmpeg_build --extra-cflags=-I/root/ffmpeg_build/include --extra-ldflags=-L/root/ffmpeg_build/lib --bindir=/root/bin --extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-nonfree --enable-libmfx
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
libswscale 5. 5.100 / 5. 5.100
libswresample 3. 5.100 / 3. 5.100
libpostproc 55. 5.100 / 55. 5.100
Guessed Channel Layout for Input Stream #0.2 : mono
Guessed Channel Layout for Input Stream #0.3 : mono
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '10minute_Pipeline_Test.mov':
Metadata:
major_brand : qt
minor_version : 537134592
compatible_brands: qt
creation_time : 2020-12-19T12:43:38.000000Z
com.apple.quicktime.author:
com.apple.quicktime.comment:
com.apple.quicktime.copyright:
com.apple.quicktime.description:
com.apple.quicktime.director:
com.apple.quicktime.genre:
com.apple.quicktime.information:
com.apple.quicktime.keywords:
com.apple.quicktime.producer:
com.apple.quicktime.displayname:
timecode : 12:43:37;28
Duration: 00:10:06.72, start: 0.000000, bitrate: 167429 kb/s
Stream #0:0(eng): Data: none (tmcd / 0x64636D74)
Metadata:
creation_time : 1970-01-04T00:49:14.000000Z
timecode : 12:43:37;28
Stream #0:1(eng): Video: prores (Standard) (apcn / 0x6E637061), yuv422p10le(tv, GBR, progressive), 1280x720, 164985 kb/s, SAR 1:1 DAR 16:9, 59.94 fps, 59.94 tbr, 60k tbn, 60k tbc (default)
Metadata:
creation_time : 1970-01-01T00:00:04.000000Z
Stream #0:2(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, s32 (24 bit), 1152 kb/s (default)
Stream #0:3(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, s32 (24 bit), 1152 kb/s (default)
Metadata:
creation_time : 2003-10-05T11:26:56.000000Z
File 'chris.mp4' already exists. Overwrite ? [y/N] y
Stream mapping:
Stream #0:1 -> #0:0 (prores (native) -> h264 (h264_qsv))
Stream #0:2 -> #0:1 (pcm_s24le (native) -> aac (native))
Press [q] to stop, [?] for help
Impossible to convert between the formats supported by the filter 'Parsed_null_0' and the filter 'auto_scaler_0'
Error reinitializing filters!
Failed to inject frame into filter network: Function not implemented
Error while processing the decoded data for stream #0:1
Conversion failed!
user#NUC:~$ ffmpeg -i 10minute_Pipeline_Test.mov -c:v h264_qsv -c:a aac chris.mp4
ffmpeg version 4.2.1 Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 9 (Ubuntu 9.3.0-17ubuntu1~20.04)
configuration: --prefix=/root/ffmpeg_build --extra-cflags=-I/root/ffmpeg_build/include --extra-ldflags=-L/root/ffmpeg_build/lib --bindir=/root/bin --extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-nonfree --enable-libmfx
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
libswscale 5. 5.100 / 5. 5.100
libswresample 3. 5.100 / 3. 5.100
libpostproc 55. 5.100 / 55. 5.100
Guessed Channel Layout for Input Stream #0.2 : mono
Guessed Channel Layout for Input Stream #0.3 : mono
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '10minute_Pipeline_Test.mov':
Metadata:
major_brand : qt
minor_version : 537134592
compatible_brands: qt
creation_time : 2020-12-19T12:43:38.000000Z
com.apple.quicktime.author:
com.apple.quicktime.comment:
com.apple.quicktime.copyright:
com.apple.quicktime.description:
com.apple.quicktime.director:
com.apple.quicktime.genre:
com.apple.quicktime.information:
com.apple.quicktime.keywords:
com.apple.quicktime.producer:
com.apple.quicktime.displayname:
timecode : 12:43:37;28
Duration: 00:10:06.72, start: 0.000000, bitrate: 167429 kb/s
Stream #0:0(eng): Data: none (tmcd / 0x64636D74)
Metadata:
creation_time : 1970-01-04T00:49:14.000000Z
timecode : 12:43:37;28
Stream #0:1(eng): Video: prores (Standard) (apcn / 0x6E637061), yuv422p10le(tv, GBR, progressive), 1280x720, 164985 kb/s, SAR 1:1 DAR 16:9, 59.94 fps, 59.94 tbr, 60k tbn, 60k tbc (default)
Metadata:
creation_time : 1970-01-01T00:00:04.000000Z
Stream #0:2(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, s32 (24 bit), 1152 kb/s (default)
Stream #0:3(eng): Audio: pcm_s24le (in24 / 0x34326E69), 48000 Hz, mono, s32 (24 bit), 1152 kb/s (default)
Metadata:
creation_time : 2003-10-05T11:26:56.000000Z
File 'chris.mp4' already exists. Overwrite ? [y/N] y
Stream mapping:
Stream #0:1 -> #0:0 (prores (native) -> h264 (h264_qsv))
Stream #0:2 -> #0:1 (pcm_s24le (native) -> aac (native))
Press [q] to stop, [?] for help
[h264_qsv # 0x55b3bb6e8800] Selected ratecontrol mode is unsupported
[h264_qsv # 0x55b3bb6e8800] Low power mode is unsupported
[h264_qsv # 0x55b3bb6e8800] Current frame rate is unsupported
[h264_qsv # 0x55b3bb6e8800] Current picture structure is unsupported
[h264_qsv # 0x55b3bb6e8800] Current resolution is unsupported
[h264_qsv # 0x55b3bb6e8800] Current pixel format is unsupported
[h264_qsv # 0x55b3bb6e8800] some encoding parameters are not supported by the QSV runtime. Please double check the input parameters.
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!
I cannot get ANYTHING to work. I can transcode other h264 files without issue. I cannot seem to transcode this prores file.
Here is a link to the source file if anyone can help I would REALLY appreciate it...
https://www.dropbox.com/s/ejrfzad20yzaifm/10minute_Pipeline_Test.mov?dl=1
I use H264_QSV daily, and I find you have to declare the QSV device as being available.
Try this:
ffmpeg -err_detect ignore_err -hide_banner -loglevel verbose -init_hw_device qsv:qsv,child_device_type=qsv ^ -hwaccel qsv -hwaccel_output_format qsv -i "input.mov" -q:v 30 -preset slow -c:a aac output.mp4
There are many more options that can be added to improve efficiency, change the quality (the -q:v setting), etc.
I've found that QSV speeds things up so much that you can use a -preset of slow or very slow to get more compression for a given quality setting without significantly increasing the time it takes to convert the file.
I may not have done the copy as well as I should have.
This is a more complete copy of how I use ffmpeg.
ffmpeg -err_detect ignore_err -hide_banner -loglevel verbose -stats -benchmark -init_hw_device qsv:qsv,child_device_type=qsv ^
-hwaccel qsv -hwaccel_output_format qsv ^
-i "input file" ^
-c:a aac -q:a 1.9 -strict normal -sws_flags lanczos ^
-vf "vpp_qsv=cw=704:ch=480:cx=11:cy=0:w=640:h=480" ^
-async_depth 128 -q:v 28 -c:v h264_qsv -preset veryslow (a bunch of optimization options on how I want the compression to be done go in here, which can be discussed separately) -movflags +faststart "output file.mp4"
This is on Windows so the carat "^" is the command line continuation character.
-err_detect suppresses some of the more useless messages. -hide_banner suppresses things that I normally don't need to see at all.
-loglevel is usually set to "info" or "quiet", but if you want to know exactly which codecs are being used, set it to "verbose" as it is here.
This is the simple answer to the original question, "Am I using the QSV codec?".
-strict normal is optional, but I found some applications didn't do well with some of the newer optimizations. It does not appear to increase file size to any significant extent, and I don't run into problems running videos on old equipment.
I put the audio processing first as it seems to work better that way.
I let the codec choose the bit rate by setting the quality, as with the video (see below).
I have also included an example of the vpp_qsv video processing filter, as I find it speeds up many operations. It can, of course, be left out if you don't need it. I put it before the compression codec: ffmpeg will process them in the proper order, but I find it's easier to keep track of what's going on if I put the commands in about the same order as they will eventually be processed. When I put the commands in this order and "verbose" is on, ffmpeg reports that the output of the vpp_qsv filter remains in video memory as the input to the h264_qsv codec. This speeds things up in my tests: or, at least, it reduces the CPU load so other programs can run at the same time.
-async_depth is optional, increases the number of frames that are read before compression is done; I find this also usually makes things go a bit faster. -q:v is the compression quality setting: I've found 28 to 30 gives me good results for watching videos on a reasonably large TV, but you will have to make tests for yourself to see what setting is right for you. Doing this is much, much better than guessing what bitrate you need, the codec can do better optimizations, and so on. You will, in most cases, get variable bit rate compression, and sometimes variable frame rates. This improves compression for parts of the video that don't have much going on, while still providing higher bit rates when needed. You may be surprised at how low a bit rate can be produced this way and still have a good quality video.
I put -movflags +faststart in ALL of my MP4 videos. This moves a copy of the MOOV atom from the end of the video to the beginning. This does at least two things. First, for many players, the video will start playing faster as the information the player needs about the video is read immediately. Second, if an MP4 file ever gets truncated and the MOOV atom is missing, you will not be able to play the file at all. There are programs that pretend to be able to recover the missing information, but I have yet to see one actually work. But if the MOOV atom is also included at the beginning of the video, you will at least be able to start processing the video, and should at least get to the point where the file is damaged. It's cheap insurance, and only takes a moment or two. (This won't work if your output is a live stream, the video has to be "finished" before the atom is created.)
-stats and -benchmark are optional, I like to see how fast processing is going and be able to compare it to other times I process videos to see if any changes I make to the options are helping or not.
If there is an interest in the various vpp_qsv filter options, or in what other compression settings I use, or what settings will allow videos to work with Roku Media Player, let me know which topic I should post that in.
Summary
Context
The software I use
The problem
Results
4.1. Actual Results
4.2. Expected Results
What did I try to fix the bug?
How to reproduce this bug: minimal and testable example with the provided required data
The question
Sources
Context
I would want to set an audio WAV as the background sound of a video WEBM. The video can be shorter or longer than the audio. At the moment I add the audio over the video, I don't know the length of both streams. The audio must repeat until the video ends (the audio can be truncated if the video ends before the end of the last repetition of the audio).
The software I use
I use ffmpeg version 4.2.2-1ubuntu1~18.04.sav0.
The problem
ffmpeg seems to enter in an infinite loop when it proccesses in order to mix the audio and the video. Also, the length of the currently-generating-output-file (which contains both video and audio) is equal to the length of the audio, instead of the length of the video.
The problem seems to be triggered by this command line:
ffmpeg -i directory_1/video.webm -stream_loop -1 -fflags +shortest -max_interleave_delta 50000 -i directory_2/audio.wav directory_3/video_and_audio.webm
Results
Actual Results
Three things:
The infinite loop of the ffmpeg process: I must manually stop the ffmpeg process
The output video file with music (which is currently generating but output anyway): it contains both audio and video. But the length of the output file is equal to the length of the audio, instead of the length of the video.
The following output logs:
ffmpeg version 4.2.2-1ubuntu1~18.04.sav0 Copyright (c) 2000-2019 the
FFmpeg developers built with gcc 7 (Ubuntu 7.5.0-3ubuntu1~18.04)
configuration: --prefix=/usr --extra-version='1ubuntu1~18.04.sav0'
--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 Input #0, matroska,webm, from 'youtubed/my_youtube_video.webm': Metadata:
encoder : Chrome Duration: N/A, start: 0.000000, bitrate: N/A
Stream #0:0(eng): Video: vp8, yuv420p(progressive), 3200x1608, SAR 1:1 DAR 400:201, 1k tbr, 1k tbn, 1k tbc (default)
Metadata:
alpha_mode : 1 Guessed Channel Layout for Input Stream #1.0 : stereo Input #1, wav, from 'tmp_music/original_music.wav':
Duration: 00:00:11.78, bitrate: 1411 kb/s
Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, s16, 1411 kb/s Stream mapping: Stream #0:0 -> #0:0 (vp8
(native) -> vp9 (libvpx-vp9)) Stream #1:0 -> #0:1 (pcm_s16le
(native) -> opus (libopus)) Press [q] to stop, [?] for help
[libvpx-vp9 # 0x5645268aed80] v1.8.2 [libopus # 0x5645268b09c0] No bit
rate set. Defaulting to 96000 bps. Output #0, webm, to
'youtubed/my_youtube_video_with_music.webm': Metadata:
encoder : Lavf58.29.100
Stream #0:0(eng): Video: vp9 (libvpx-vp9), yuv420p(progressive), 3200x1608 [SAR 1:1 DAR 400:201], q=-1--1, 200 kb/s, 1k fps, 1k tbn, 1k
tbc (default)
Metadata:
alpha_mode : 1
encoder : Lavc58.54.100 libvpx-vp9
Side data:
cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1
Stream #0:1: Audio: opus (libopus), 48000 Hz, stereo, s16, 96 kb/s
Metadata:
encoder : Lavc58.54.100 libopus
Expected Results
No infinite loop during the ffmpeg process
Concerning the output logs, I don't know what it should look.
The output file with the audio and the video should:
3.1. If the video is longer than the audio, then the audio is repeated until it exactly fits the video. The audio can be truncated.
3.2. If the video is shorter than the audio, then the audio is truncated and exactly fits the video.
3.3. If both video and audio are of the same length, then the audio exactly fits the video.
How to reproduce this bug? (+ required data)
Download the following files (resp. audio and video) (I must refresh these download links every 24 hours):
1.1. https://a.uguu.se/dmgsmItjJMDq_audio.wav
1.2. https://a.uguu.se/w3qHDlGq6mOW_video.webm
Move them into the directory/directories of your choice.
Open your CLI, move to the adequat directory and copy/paste/execute the instruction given in Part. The Problem (don't forget to eventually modify this instruction by indicating the adequat directories, according to step 2.).
You'll face my problem.
What did I try to fix the bug?
Nothing, since I don't even understand why the bug occures.
The question
How to correct my command in order to mix these audio and video streams without any infinite loop during the ffmpeg process, keeping in mind that I don't know their length, and that audio must be repeated in order to fit the video, even if audio must be truncated (in the case of the last repetition of the audio file must be truncated because the video stream has just ended)?
Sources
The source is the command line you can find in Part. The problem.
The placement of some of your options is wrong. All of the shortest related options belong in front of the output.
ffmpeg -i directory_1/video.webm -stream_loop -1 -i directory_2/audio.wav -c:v copy -shortest -fflags +shortest -max_interleave_delta 100M directory_3/video_and_audio.webm
There's no need to transcode the video unless you wish to.
I used ffmpeg to transcode some files into new format and with certain parameters. After transcoding, I found some output file's metadata is not what I expected, the output value is not the same with I set in the cmd line.
Before transcoding I check the media info of the inputfile:
ffmpeg -i dz2015082000010.mpg
ffmpeg version 3.2.4 Copyright (c) 2000-2017 the FFmpeg developers
built with gcc 4.8.3 (GCC) 20140911 (Red Hat 4.8.3-9)
configuration: --enable-static --enable-memalign-hack --enable-libx264
--enable-gpl --enable-pthreads --enable-version3 --enable-avisynth --enable-bzlib --enable-iconv --enable-zlib --enable-nonfree --extra-cflags=-I/usr/local/include/ --extra-ldflags=-L/usr/local/lib --enable-debug=3 --disable-optimizations --enable-nonfree --enable-libmp3lame libavutil 55. 34.101 / 55. 34.101 libavcodec 57. 64.101 / 57. 64.101 libavformat 57. 56.101 /
57. 56.101 libavdevice 57. 1.100 / 57. 1.100 libavfilter 6. 65.100 / 6. 65.100 libswscale 4. 2.100 / 4. 2.100 libswresample 2. 3.100 / 2. 3.100 libpostproc 54. 1.100 /
54. 1.100 Input #0, mpeg, from 'dz2015082000010.mpg': Duration: 00:01:49.30, start: 0.685389, bitrate: 15723 kb/s
Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p(tv, top first), 1920x1080 [SAR 1:1 DAR 16:9], 15000 kb/s, 25 fps, 25 tbr,
90k tbn, 50 tbc
Stream #0:1[0x1c0]: Audio: mp2, 48000 Hz, stereo, s16p, 384 kb/s At least one output file must be specified
Next, transcoding with the cmd line:
ffmpeg -i dz2015082000010.mpg -vcodec libx264 -b:v 4000k -s 1920x1080 -r 25 -g 25 -vprofile main -acodec aac -strict -2 -b:a 128k -ac 2 -ar 44100 -y output.ts
After transcoding, I check the media info of the output file:
ffmpeg -i output.ts
ffmpeg version 3.2.4 Copyright (c) 2000-2017 the FFmpeg developers built with gcc 4.8.3 (GCC) 20140911 (Red Hat
4.8.3-9) configuration: --enable-static --enable-memalign-hack --enable-libx264 --enable-gpl --enable-pthreads --enable-version3 --enable-avisynth --enable-bzlib --enable-iconv --enable-zlib --enable-nonfree --extra-cflags=-I/usr/local/include/ --extra-ldflags=-L/usr/local/lib --enable-debug=3 --disable-optimizations --enable-nonfree --enable-libmp3lame libavutil 55. 34.101 / 55. 34.101 libavcodec 57. 64.101
/ 57. 64.101 libavformat 57. 56.101 / 57. 56.101
libavdevice 57. 1.100 / 57. 1.100 libavfilter 6. 65.100
/ 6. 65.100 libswscale 4. 2.100 / 4. 2.100
libswresample 2. 3.100 / 2. 3.100 libpostproc 54. 1.100
/ 54. 1.100 Input #0, mpegts, from 'full-2.ts': Duration:
00:01:49.30, start: 1.456778, bitrate: 4455 kb/s Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0:0[0x100]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p(progressive), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr,
90k tbn, 50 tbc
Stream #0:1[0x101]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 4 kb/s At least one output file must be
specified
I don't know why the audio bitrate is changed to 4 kb/s after transcoding, I set the value with -b:a 128k before, anybody can help me? BTW, the output file sounds all right.
The native encoder won't waste bits on silent portions. And it doesn't do strict CBR. If you really need an output to be around the target bitrate, you can mix in a very low level of noise.
I want to record audio and video with my raspberry pi b+ 2.
I tried to accomplish this with one ffmpeg command but this is to slow. and i could not get it working correctly
I have a raspberry pi camera module and a Cirrus audio card. On the raspberry i have compiled a new kernel with support for the audio card. I also compiled ffmpeg on the raspberr with alsa support
~$ ffmpeg
ffmpeg version N-71470-g2db24cf Copyright (c) 2000-2015 the FFmpeg developers
built with gcc 4.6 (Debian 4.6.3-14+rpi1)
configuration: --arch=armel --target-os=linux --enable-gpl --extra-libs=-lasound --enable-nonfree
libavutil 54. 22.101 / 54. 22.101
libavcodec 56. 34.100 / 56. 34.100
libavformat 56. 30.100 / 56. 30.100
libavdevice 56. 4.100 / 56. 4.100
libavfilter 5. 14.100 / 5. 14.100
libswscale 3. 1.101 / 3. 1.101
libswresample 1. 1.100 / 1. 1.100
libpostproc 53. 3.100 / 53. 3.100
Hyper fast Audio and Video encoder
usage: ffmpeg [options] [[infile options] -i infile]... {[outfile options] outfile}...
Now i try to record an audio stream and a video stream 'at the same time'
I do this my running a shell script
raspivid -t 60000 -vs -w 1280 -h 720 -b 5000000 -fps 25 -o video.h264 &
arecord -Dhw:sndrpiwsp -r 44100 -c 2 -d 60 -f S32_LE audio.aac
i also tried with -r 22050 and -f S16_LE
when running this it sometimes gives an (i think)
overrun!!! (at least 1038.725 ms long)
at the end of the script i have two files. a video and a audio file.
now i want to merge those two together by using ffmpeg
ffmpeg -i video.h264 -i audio.aac -c:v copy -c:a aac -strict experimental output.mp4
this gives the output:
ffmpeg version N-71470-g2db24cf Copyright (c) 2000-2015 the FFmpeg developers
built with gcc 4.6 (Debian 4.6.3-14+rpi1)
configuration: --arch=armel --target-os=linux --enable-gpl --extra-libs=-lasound --enable-nonfree
libavutil 54. 22.101 / 54. 22.101
libavcodec 56. 34.100 / 56. 34.100
libavformat 56. 30.100 / 56. 30.100
libavdevice 56. 4.100 / 56. 4.100
libavfilter 5. 14.100 / 5. 14.100
libswscale 3. 1.101 / 3. 1.101
libswresample 1. 1.100 / 1. 1.100
libpostproc 53. 3.100 / 53. 3.100
Input #0, h264, from 'video_1min_3.h264':
Duration: N/A, bitrate: N/A
Stream #0:0: Video: h264 (High), yuv420p, 1280x720, 25 fps, 25 tbr, 1200k tbn, 50 tbc
Guessed Channel Layout for Input Stream #1.0 : stereo
Input #1, wav, from 'audio_1min_3.aac':
Duration: 00:01:00.00, bitrate: 705 kb/s
Stream #1:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 22050 Hz, 2 channels, s16, 705 kb/s
[mp4 # 0x3230f20] Codec for stream 0 does not use global headers but container format requires global headers
Output #0, mp4, to 'output_1min_3.mp4':
Metadata:
encoder : Lavf56.30.100
Stream #0:0: Video: h264 ([33][0][0][0] / 0x0021), yuv420p, 1280x720, q=2-31, 25 fps, 25 tbr, 1200k tbn, 1200k tbc
Stream #0:1: Audio: aac ([64][0][0][0] / 0x0040), 22050 Hz, stereo, fltp, 128 kb/s
Metadata:
encoder : Lavc56.34.100 aac
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Stream #1:0 -> #0:1 (pcm_s16le (native) -> aac (native))
Press [q] to stop, [?] for help
frame= 1822 fps=310 q=-1.0 Lsize= 33269kB time=00:01:12.84 bitrate=3741.7kbits/s
video:32300kB audio:941kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.086073%
so finally i have a file output.mp4 that is a movie with audio that is in sync at the beginning but drifts away to a difference of about 4 seconds. where the audio is ahead of the video.
I hope you can help me trying to solve this issue so the audio does not drift away anymore.
Thanks in advance
( i tried to be as clear as possible )
We can try to use the -async and -vsync options to correct the audio and video time shift.
for example, i have used the below option to reduce the time lag of 2 sec seen in the audio.
./ffmpeg -async 1 -i "weatherinput.mov" -strict -2 -vcodec libx264 -movflags +faststart -vprofile high -preset slow -b:v 500k -maxrate 500k -bufsize 1000k -threads 0 -b:a 128k -pix_fmt yuv420p "weatheroutput.mp4"
Also we can use vsync options if required apart from the ioffset.
The link below can also referred for other combination of using th async, vsync and i offset to avoid the drift.
I'm implementing a http live streaming server to send audio file to iOS devices.
No problem with Apple's tools, mediafilesegmenter, my files are valid and it works fine.
I'm trying now to segment the same file using ffmpeg. I've downloaded the last stable version which is the 0.10.2 for now.
Here is how I try to segment my mp3 file:
./ffmpeg -re -i input.mp3 -f segment -segment_time 10 -segment_list outputList.m3u8 -acodec libmp3lame -map 0 output%03d.mp3
It starts the mapping like expected but finish with only one .mp3 file.
Did I miss something in the process?
Thanks in advance.
edit
Ok here is my latest command line:
ffmpeg -i input.mp3 -c:a libmp3lame -b:a 128k -map 0:0 -f segment -segment_time 10 -segment_list outputlist.m3u8 -segment_format mp3 'output%03d.mp3'
It still gives me only one file but the file is the hole song, not only one part.
Here is the output of ffmpeg:
ffmpeg version 0.10.2 Copyright (c) 2000-2012 the FFmpeg developers
built on Apr 20 2012 07:08:29 with gcc 4.5.2
configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libmp3lame
libavutil 51. 35.100 / 51. 35.100
libavcodec 53. 61.100 / 53. 61.100
libavformat 53. 32.100 /
53. 32.100
libavdevice 53. 4.100 / 53. 4.100
libavfilter 2. 61.100 / 2. 61.100
libswscale 2. 1.100 / 2. 1.100
libswresample 0. 6.100 / 0. 6.100
libpostproc 52. 0.100 / 52. 0.100
[mp3 # 0x8e4f120] max_analyze_duration 5000000 reached at 5015510
Input #0, mp3, from 'BeachHouse-Myth.mp3':
Metadata:
title : Myth
artist : Beach House
track : /
album : Bloom
disc : /
genre : Alternative
TSRC : USSUB1296501
Duration: 00:04:18.69, start: 0.000000, bitrate: 320 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16, 320 kb/s Output #0, segment, to 'stream%03d.mp3': Metadata:
title : Myth
artist : Beach House
track : /
album : Bloom
disc : /
genre : Alternative
TSRC : USSUB1296501
encoder : Lavf53.32.100
Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16, 128 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (mp3 -> libmp3lame)
Press [q] to stop, [?] for help
Truncating packet of size 1024 to 105ate= 0.0kbits/s
Truncating packet of size 1024 to 1
size= 0kB time=00:04:18.71 bitrate= 0.0kbits/s video:0kB audio:4042kB global headers:0kB muxing overhead -100.000000%
Audio only might be a bug. I contacted the FFMPEG player bug list, and a bug is filed: http://ffmpeg.org/trac/ffmpeg/ticket/1290