I have audio in opus format and need to make mp4 container
ffmpeg -i input.opus -c:a opus -strict experimental output.mp4
Throws Could not find tag for codec opus in stream
What is wrong here?
Related
I am using ffmpeg to extract the audio from a video. Below code downlaods the audio from a video file. I'm not sure how efficient this program is but I do know that it downloaods it in 48KHZ.
How do I use this program to extract audio from a video in 8Khz because the file is getting too big.
ffmpeg -i video_link -vn output.wav
Use -ar option to change frequency rate
ffmpeg -i video_link -vn -ar 8000 output.wav
If you want to try different formats of audio check the available formats in ffmpeg using ffmpeg -formats and available codecs using ffmpeg -codecs
Here's an example to extract to mp3 file
ffmpeg -i video_link -vn -ar 8000 -f mp3 output.mp3
Edit: as #llogan pointed out, -f option is not needed, ffmpeg automatically mux mp3 file.
ffmpeg -i video_link -vn -ar 8000 output.mp3
I'm using ffmpeg -i video.mp4 -i audio.mp3 -c:a aac -shortest output.mp4 to join a part of an mp3 file to a video while converting to AAC, but the last 3 centiseconds of the audio are silent. Is there a way to fix this?
Log: https://jpst.it/2aIIh - This one had 10 ms of silence at the end.
FFMPEG documentation says this can be achieved by doing something similar to the following:
ffmpeg -f lavfi -i anullsrc=channel_layout=stereo:sample_rate=44100 -i $media -shortest -c:v copy -c:a aac $output
However when i try this with a WebM video I get the following error:
Could not write header for output file #0 (incorrect codec parameters ?): Invalid argument
Error initializing output stream 0:1
I there a workaround for this? Thanks.
AAC is not a compatible codec for WebM. You can simply drop -c:a aac and FFmpeg will select the default audio codec for the output format. Opus (libopus) and Vorbis (libvorbis) are acceptable audio codecs for WebM.
I'm currently writing a small script that coverts an MP4 to Opus audio on the fly and sends it to Discord in golang. Initially my script would pass an MP4 as it was downloading to ffmpeg through stdin and then pass stdout to an Opus encoder, then to Discord (exactly like this). After learning I could build ffmpeg with Opus, I'd like to cut out the opus encoder I previous had and pass ffmpeg's output directly to Discord.
Previous, my ffmpeg command looked like this (with using the second opus encoder)
ffmpeg -i - -f s16le -ar 48000 -ac 2 pipe:1
Now, without the encoder and letting ffmpeg do all the work, this is what I've come up with so far.
ffmpeg -i - -f s16le -ar 48000 -ac 2 -acodec libopus -b:a 192k -vbr on -compression_level 10 pipe:1
With this command however the audio doesn't get accepted by Discord's server, meaning I'm suspecting opus audio isn't coming out the other end. No errors outputted. Have I done something wrong with ffmpeg that could of caused this?
Try
ffmpeg -i - -sample_fmt s16 -ar 48000 -ac 2 -acodec libopus -b:a 192k -vbr on -compression_level 10 -f opus pipe:1
You can't use -f s16le as that specifies an uncompressed output format (of a specific sample type), whereas you need a compressed data stream of a certain codec. Instead, you can use sample_fmt and -f opus
I have some video files that I need to re-encode due to compatibility issues. They are currently mkv files with h.264 video and ac3-a52 audio. I want to keep the h.264 video, convert the container to m4v and create two audio tracks, one with the original ac3-a52 and one copied from that but in aac stereo.
I assume there has to be some sort of audio stream mapping command but I don't know how to map and re-encode at the same time. What command should I enter into ffmpeg to achieve this?
Also, what is the difference between ac3 and ac3-a52? Will an apple TV still be able to pass through ac3-a52 or does that have to be converted to ac3?
this works for me:
ffmpeg -y -i Source.mkv -map 0:v -c:v copy -map 0:a -c:a copy -map 0:a -strict -2 -c:a aac out.mkv
-y – A global option to overwrite the output file if it already exists.
-map 0:v – Designate the video stream(s) from the first input as a source for the output file.
-c:v copy – Stream copy the video. This just muxes the input to the output. No re-encoding occurs.
-map 0:a – Designate the audio stream(s) from the first input as a source for the output file.
-c:a copy – Stream copy the audio. This just muxes the input to the output. No re-encoding occurs.
-strict -2 -c:a aac – Use the native FFmpeg AAC audio encoder. -strict -2 is required as a way that you acknowledge that the encoder is designated as experimental. It is not a great encoder, but it is not too bad at higher bitrates.
According to wikipedia, there is no difference between AC3 and ATSC A/52: the 1st one is the name of the codec, the 2nd is the name of the standard specifying the AC3 codec. Maybe someone have more knowledge about it?
I'm doing the same as the OP, but with an m4v container. I'm using the MacPorts "nonfree" variant of ffmpeg so that I can use libfaac, which gives better audio quality than the built-in AAC encoder and also had the same issue as #dkam. The command line I ended using is like this:
ffmpeg -i input.m4v -map 0:v -c:v copy -map 0:a -c:a:0 copy -map 0:a -c:a:1 libfaac output.m4v
(The videos are for playback on an iPad, which doesn't seem to be able to handle ac3.)
This command will take a video with 1 audio stream, and downmix to stereo and convert the audio stream and add it as a 2nd audio stream. It will be in AAC 384k.
ffmpeg -i INPUT.mkv -strict -2 -map 0 -c copy -map 0:a:0 -c:a:1 aac -b:a 384k -ac 2 OUTPUT.mkv
Explanation of the command
ffmpeg -i INPUT.mkv The application and input file
-strict -2 Enable downmixing support
-map 0 Tell ffmpeg read all Video, Audio, and Subtitle streams for the following arguments
-c copy Copy everything
-map 0:a:0 Tell ffmpeg to read the first audio stream for the following arguments
-c:a:1 aac Output the audio to a 2nd audio channel (0 = first channel) in aac format. Important! You must change the output channel to a higher number if there are multiple audio streams to prevent overwriting them.
-b:a 384k 384k bitrate (I don't know what's good for aac stereo but this is really high since it's for 5.1 aac)
-ac 2 Downmix to stereo
OUTPUT.mkv Output file
More examples
A video with two audio streams. Creating a third audio stream by encoding the first.
ffmpeg -i INPUT.mkv -strict -2 -map 0 -c copy -map 0:a:0 -c:a:2 aac -b:a 384k -ac 2 OUTPUT.mkv
Again a video with two audio streams, but you want to encode the second one
ffmpeg -i INPUT.mkv -strict -2 -map 0 -c copy -map 0:a:1 -c:a:2 aac -b:a 384k -ac 2 OUTPUT.mkv