Trying to use '-acodec libopus' in my npm project as I use in the command line like in the following format;
ffmpeg -acodec libopus -i 1.webm 1.wav
This works perfectly! But I would like to make it possible in my NPM project.
How can I set the parameters?
This is what I have , but not working. The output file is broken in a way that some of the frames of the audio file are missing. It is like there is sound and then there is not. And vice versa.
var proc = new ffmpeg({
source: file,
nolog: false
});
format = "opus"; // or could be wav as well!
proc.addOptions([
'-f ' + format,
'-acodec libopus',
'-vn'
]);
The purpose is to take audio file from the video file seamlessly.
Without the codec libopus, I get the following errors in the command prompt, so I assume I should handle the same issue in my NPM project as well.
[opus # 00000000006d4520] LBRR frames is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[opus # 00000000006d4520] Error decoding a SILK frame.
[opus # 00000000006d4520] Error decoding an Opus frame.
My library is up to date, I just need to use the codec libopus properly.
Any suggestions?
\node-js>ffmpeg -version
ffmpeg version N-86175-g64ea4d1 Copyright (c) 2000-2017 the FFmpeg
developers
built with gcc 6.3.0 (GCC)
Output in command line;
xtranscribe transcodeWatson: file : ./data/that/2.webm
progress 62.625273103421605%
progress 100.01224534515762%
SAVED - transcodeWatson : .mp3
out of transcode!
fileSizeInBytes : 16284033
According to the README, you can add input options to the process:
proc.addInputOption('-acodec libopus');
It matters where you place an option in ffmpeg. If you put it before -i, it applies to that particular input. If you put it before an output file name, it applies to that output.
Related
I wanted to edit my .m4a voice recording from Samsung Voice Recorder using ffmpeg 2.2.2, however, I got the error Invalid data found when processing input. I tried to open it through Audacity, but it returned an error claiming that the ffmpeg library is missing, which is definitely not the case. Eventually I tried to use online .m4a to .mp3 converters, but they all returned error, so I assume there may be an issue with the encoding of the original file and ffmpeg should be configured accordingly. What settings shall I use? (The original file can be played on the phone without any problem.)
ffmpeg -ss 00:00:19 -i "C:\Your\Folder\original.m4a" edited.m4a
I'm new to FFMPEG and I'm having a really hard time understanding the documentation: https://ffmpeg.org/ffmpeg-all.html#aspectralstats
I want to get the centroid value for an audio file using command line.
ffmpeg -i file.wav -af aspectralstats=measure=centroid -f null -
I get the following errors
[Parsed_aspectralstats_0 # 000002a19b1b9380] Option 'measure' not found
[AVFilterGraph # 000002a19b1c99c0] Error initializing filter 'aspectralstats' with args 'measure=centroid'
Error reinitializing filters!
Failed to inject frame into filter network: Option not found
Error while processing the decoded data for stream #0:0
Conversion failed!
What am I doing wrong?
The measure option was added mere 4 weeks ago. So, yeah, you probably missed it by a couple days. Grab the latest snapshot if you want to only retrieve the centroids. The snapshot you have should get you the centroids along with other parameters if you just call aspectralstats (no options).
Also, the aspectralstats outputs only goes to the frame metadata and not printed on stdout by default. So you need to append ametadata=print:file=- to your -af.
ffmpeg -i file.wav -af aspectralstats=measure=centroid,ametadata=print:file=- -f null -
<Shameless plug> FYI, if you're calling it from Python, I have implemented an interface for this in ffmpegio if interested.</sp>
I am trying to add chapters to a ogg file containing vorbis audio.
From this link I copied the following ffmpeg command.
ffmpeg -threads auto -y -i in.ogg -i metadata_OGG.txt -map_metadata 1 -codec copy out_METADATA.ogg
My metadata_OGG.txt file is as given below.
CHAPTER00=00:00:00.000
CHAPTER00NAME=Chapter 01
CHAPTER01=00:00:05.000
CHAPTER01NAME=Chapter 02
CHAPTER02=00:00:10.000
CHAPTER02NAME=Chapter 03
I am getting the following error.
[ogg # 00000000006d6900] Unsupported codec id in stream 0
Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument
But if i change -codec copy to -acodec copy there is no error in ffmpeg but the text file is converted to video. i.e. the output file will have a static video frame with the text of metadata_OGG.txt in it. Also, I observe the following log message during conversion.
Stream #1:0 -> #0:0 (ansi (native) -> theora (libtheora))
Stream #0:0 -> #0:1 (copy)
Anybody please tell me what is going wrong here?
Also, I would like to know what is the right way to add chapters to ogg. I searched for some tools also. I did not get any.
Here is what worked for me using ffmpeg 4.3.1.
I have a metadata file which almost respects ffmpeg's metadata file format:
;FFMETADATA1
title=Evolution theory
[CHAPTER]
TIMEBASE=1/1000
START=0
END=
title=Darwin's point of view
[CHAPTER]
TIMEBASE=1/1000
START=78880
END=
title=Genghis Khan's children
Notice the file format requires an END time, but leaving it empty didn't bother in my case.
Now I add chapter information to my opus/ogg file:
ffmpeg -i darwin.opus.ogg -i darwin_chapters.txt -map_metadata 1 -c copy darwin_withchapters.opus.ogg
Note: if you want to overwrite existing chapter information from the file, you may need to add a -map_chapters 1 parameter in the ffmpeg command line above.
That creates the file darwin_withchapters.opus.ogg. I check if chapter info has really been added to the file:
opusinfo darwin_withchapters.opus.ogg
You would use ogginfo for Ogg/Vorbis files.
And here is the result (I removed a few irrelevant lines):
ENCODER=opusenc from opus-tools 0.1.10
ENCODER_OPTIONS=--bitrate 112
title=Evolution theory
CHAPTER000=00:00:00.000
CHAPTER000NAME=Darwin's point of view
CHAPTER001=00:01:19.880
CHAPTER001NAME=Genghis Khan's children
[...]
Here you go. ffmpeg did the conversion between its metadata file format to the vorbis tag/comment chapter format.
You could also directly write metadata in the Vorbis Chapter Extension format, and use the classic vorbiscomment tool, or other tools which allow editing of opus/ogg in-file tags.
Opus has been mentioned here. I was trying to make opusenc from opus-tools add chapters when encoding and couldn’t find a command line example anywhere. Thanks to the hints in this thread I managed to figure it out, perhaps someone may find it helpful.
opusenc --comment "CHAPTER000=00:00:00.000" --comment "CHAPTER000NAME=Hello" --comment "CHAPTER001=01:23:45.678" --comment "CHAPTER001NAME=World" input.wav output.opus
The chapter key/value scheme is the aforementioned Ogg/Matroska one. Of course, more metadata options like --title, --artist etc. can be added.
Using ffmpeg to add the chapters resulted in two problems for me: The artwork image in the ogg/opus input file was missing in the output file, and ffmpeg rejected empty END chapter times.
I did this on Windows 10 using
opusenc opus-tools 0.2-3-gf5f571b (using libopus 1.3)
ffmpeg version 4.4.1-essentials_build-www.gyan.dev
opusinfo, MPC-HC (64-bit) v1.7.11 and VLC Media Player 3.0.14 Vetinari to confirm.
I found the issue.
For ffmpeg to work, the metadata file should have the following header.
;FFMETADATA1
I followed the steps given in ffmpeg documentation for metadata.
But the issue is not resolved completely.
With the above steps I am able to add metadata to mp4, mkv and other container files but not to ogg files. I am not sure whether ffmpeg supports adding chapters to ogg files.
How to transfer metadata using FFMPEG or other tools with CMD ?
I'm trying to encode video/audio and since they already have metadata inside obviously i want to preserve them into my new file
btw since i'm using mediamonkey as main player, there's also some Custom metadata. this is the one who wont transfer
for Video output file using mp4/mkv (using x264)
for Audio output file using m4a (using neroAac)
Thank You!
ps. which container is best for neroAac and x264? since i can't seem to edit mkv metadata (when i remove from mediamonkey playlist, they're all gone), mp4 is fine though and i can't seem to play AAC, although it's fine when muxed into video
Copy all custom and global metadata tag information using the following command:
ffmpeg <inputfile> -movflags use_metadata_tags -c copy <outputfile>
I am trying to use avs2yuv to pipe avs output to ffmpeg for further conversion.
My video file is called "sample.avi" (No sound, just video)
My audio file is called "sample.wav"
My avs file(s) is called sample.avs, and looks like this:
V = AviSource("sample.avi")
A = WavSource("sample.wav")
AudioDub(V ,A)
or
V = DirectShowSource("sample.avi")
A = DirectShowSource("sample.wav")
AudioDub(V ,A)
Here is how I pipe:
avs2yuv sample.avs - | ffmpeg -y -f yuv4mpegpipe -i - output.mp4
Now here is the PROBLEM: No matter what files I try as an input, there is NO SOUND in my output. I do not understand what I am doing wrong, and why my audio does not make it to the output. If anyone has experience with avisynth and avs2yuv, your help would be GREATLY appreciated.
Thank you!
I would try to play your avs file with ffplay in order to check your avs file.
And you can also try to build some GRaph with GraphEdit in order to do something like that
A = DirectShowSource("sample_audio.grf", video=false)
V = DirectShowSource("sample_video.grf", audio=false)
AudioDub(V ,A)
With DirectShow you can add several parameter like fps, frame-count etc... sometime it helps.
Good Luck
As per this link:
Avs2YUV is a command-line program, intended for use under Wine, to
interface between Avisynth and Linux-based video tools.
avs2yuv.exe only handles the video stream which it output in a YUV color-space. It is that simple: the audio stream is ignored.
Here are some ways to process both audio and video streaams in .avs. These methods work in Linux using wine, and do of course work in Windows:
Encode in Avidemux via AvsProxy (AvsProxy ships with Avidemux)
Use VirutalDub as the encoder gui
otherwise encode the audio seperately, then mux in the video in a seperate step.
I believe avs2pipe can handle both video and audio streams fron a .avs, but I haven't tried it yet. Here is a link to some info about avs2pipe
Summary: Using avs2yuv mainly makes sense in a Linux/Unix environment.
Try makeAVIS.exe from the ffdshow package:
wine makeavis.exe -p -i example.avs -a output.wav