I would like to convert a AC3 audio file (ATSC A/52 aka AC-3 aka Dolby Digital stream 6 channels) to Wave audio file (16khz mono/1 channel).
While searching on the internet, a lot of people just used ffmpeg -i file.ac3 file.wav however, i'm not sure if that even works.
I keep getting
[ac3 # 0x55ac1a0b0660] exponent -1 is out-of-rangets/s speed= 125x
[ac3 # 0x55ac1a0b0660] error decoding the audio block
[ac3 # 0x55ac1a0b0660] frame sync error
Error while decoding stream #0:0: Invalid data found when processing input
etc
while I do the same command.
How do I convert ac3 to wav (16khz mono)?
*Note:
I also tried ffmpeg -i file.ac3 -codec:a:1 ac3 -codec copy -b:a 384 file.wav -ac 1 -ar 16000. But this doesn't output an actual wav file.
ffmpeg -i file.ac3 -vcodec copy -acodec pcm_s16le -ar 16000 -ab 128k -ac 1 file.wav should do it!
Also you can convert Eac3 to wav in high quality mode.
The wav file will be 48KHz 24Bit 6 Channels.
ffmpeg -i "input.eac3" -acodec pcm_s24le -ar 48000 -ac 6 "output.wav"
If you want to export in 8 channels, just write 8 instead of 6.
Related
I need to convert a 44KHz stero m4a audio file to 22KHz mono mp3 VBR file, how can I do that with ffmpeg on linux terminal? Thanks.
Gyan's comment is what I want, here is the full command line:
ffmpeg -i in.m4a -ac 1 -ar 22050 -c:a libmp3lame -q:a 9 out.mp3
with the option for VBR encoding. The number after -q:a specifies encoding quality (bitrate), with 0 being the best quality (largest file) and 9 being the worst quality (smallest file).
Here is the document on ffmpeg wiki.
I attempt to create a video slideshow from a number of image files and an audio file in 2 steps:
Create a temporary video file from a sequence of image files
Add an audio file to the temporary video file with a delay of 5 seconds
The audio file is an uncompressed stereo wav file, encoded with a sample rate of 44100 Hz and a bit depth of 32 bits, with a size of 40.1 MB. To preserve the lossless quality of the input audio file I use the option -c:a aac -b:a 192k as per Slideshow Wiki. However, the final output video file has a size of only 4.49 MB.
How can the output video file be about 10 times smaller than the input audio file and still preserve the original lossless quality?
My code:
ffmpeg -f concat -i slide-sequence.txt -c:v libx264 -r 30 -filter_complex format=yuv420p temp.mp4
ffmpeg -i temp.mp4 -i audio.wav -af "adelay=5000|5000" -c:v copy -c:a aac -b:a 192k out.mp4
How can the output video file be about 10 times smaller than the input audio file and still preserve the original lossless quality?
It does not. AAC is a lossy format. It uses encoding methods to make it sound good although it is lossy.
There are formats that are both compressed and lossless, such as FLAC. YouTube supports this, so use:
ffmpeg -i temp.mp4 -i audio.wav -af "adelay=5000|5000" -c:v copy -c:a flac out.mkv
Note the change of the output container format from MP4 to Matroska (.mkv). YouTube supports Matroska.
I have a MP4a file which I am looking to convert to WAV file, containing signed 16-bit PCM samples. I have ffmpeg at my disposal, and looking at previous SOF posts, I have tried:
ffmpeg -y -i input.mp4 -acodec pcm_s16le -f s16le -ac 1 -ar 16000 output.pcm
but, the program I use complains that this converted file has data in unknown format. I was wondering if anyone had any pointers on how to go from m4a to wav with pcm samples.
ffmpeg -i input.mp4 output.wav
This command will output WAV file containing signed 16-bit PCM samples. Your command is outputting raw PCM, not WAV.
You can add -c:a pcm_s16le output option if you prefer, but that's the default encoder for WAV so it can be omitted.
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