I have a FLAC file that contains a single song. I want to create a HLS stream with the best quality stream being FLAC (matching the input so just pass-through but chunked) followed by AAC-LC 320, 160 and 96Kbps.
How might I do this?
The following command is a good point to start:
./ffmpeg -i my.flac -map 0:a -map 0:a -map 0:a -map 0:a -c:a:0 flac -c:a:1 aac -c:a:2 aac -c:a:3 aac -b:a:1 320k -b:a:2 160k -b:a:3 96k -f hls -hls_playlist_type vod -master_pl_name master.m3u8 -var_stream_map "a:0 a:1 a:2 a:3" stream_%v.m3u8
Some explanation:
-map 0:a adds the audio input 4 times (0:a means from the first input file (0) the audio stream)
-c:a:0 sets the codec for each stream (0 = flac; 1+2+3 for aac)
-b:a:0 sets the bitrate for each steam (1-3 for aac 320/160/96kbits)
-f hls because you want HLS
-hls_playlist_type vod to keep all segments of the HLS stream
-master_pl_name master.m3u8 name of the master playlist (contains all 4 streams)
-var_stream_map "a:0 a:1 a:2 a:3" add all 4 audio streams to the output
stream_%v.m3u8 name of the HLS playlist for each stream (with the placeholder %v for the stream variant
Related
I have a mkv file with an audio track (raw AAC file with ADTS headers) and I would like to merge it with another audio file that has my comments on it (AAC m4a file recorded with my phone.
I know I can merge both audio files in one track with te following command:
ffmpeg -i input.mkv -i audio.m4a -i audio.aac -filter_complex
"[1][2]amix=inputs=2[a]" -map 0:v -map "[a]" -c:a:1 aac copy test.mkv
However I would like my voice (audio.m4a) to be around 40% louder than the original audio (audio.acc). How can i do that?
I have the following problem.
In the folder there is video.mp4 file (contain 1 audio stream). There are also 3 different files audio1.wav, audio2.wav, audio3.wav. These files I need to 'attach' as multi stream to the video file - so the user can choose the audio language in VLC player or similar. Result must be one audio at the time - no mixing it all together.
Now, I've done it via Premiere Pro with multitrack (quicktime export to mov), and then I run a script to change audio stream names to correspond with the audio language (iso 639-2 ) and output the mp4 file. All works well, but I wonder if there is simple way to do everything via ffmpeg ( .bat script ). I have a working script for replacing audio in the video, but I need to add few additional .wav to the video file as separate audio tracks.
Any help will be appreciated!
To add a new audio track into an existing video with audio, use
the -i parameter to specify all the input files (original video and additional audios)
the -map option to manually select the tracks of each input (https://trac.ffmpeg.org/wiki/Map)
in your case,
-map 0 to copy all streams from the input #0 (video)
-map 1:a to include all audio streams from input#1 file (audio1)
-map 2:a to include all audio streams from input#2 file (audio2)
and so on
and
-shortest to crop the output to the shortest input
and additionally you may want to use
-c:v copy to copy the video stream without reencoding.
so, try this (line split for readability)
ffmpeg -i video.mp4 -i input1.mp3 -i input2.mp3
-map 0 -map 1:a -map 2:a
-c:v copy -shortest
output.mp4
and (addording to your comment) adding metadata for the audio tracks
ffmpeg -i video.mp4 -i input1.mp3 -i input2.mp3
map 0 -map 1:a -map 2:a
-metadata:s:a:0 language=eng
-metadata:s:a:1 language=ger
-metadata:s:a:2 language=fra
-disposition:a:0 default
-c:v copy -shortest
output.mp4
I want to perform HLS segmentation and get audio and video in separated files for each segment from .mp4 video with audio. No need to adapt bitrate, because that will cause an extra delay.
I tried to use
ffmpeg -i videoInput.mp4 -c:v copy –c:a copy -hls_segment_type mpegts –map a:0 –map v:0 -hls_time 1 out.m3u8 audio.ts
Which generates the video files perfectly but just one audio file.
Thank you.
From the FFmpeg documentation I understand:
When there are two or more variant streams, the output filename
pattern must contain the string "%v", this string specifies the
position of variant stream index in the output media playlist
filenames. The string "%v" may be present in the filename or in the
last directory name containing the file. If the string is present
in the directory name, then sub-directories are created after
expanding the directory name pattern. This enables creation of
variant streams in subdirectories.
'''
For me:
ffmpeg -hide_banner -t 5:00 -i "$IN" -c copy -map a:0 -map v:0 -hls_segment_type mpegts -hls_time 1 -f hls -var_stream_map "a:0 v:0" v_%v/file.m3u8
creates both streams in v_0 & v_1.
I don't know the exact solution, but as a workaround, you can extract the audio first with:
ffmpeg -i video.mp4 -c:a copy -vn audio.aac
Then apply segmentation for video and audio files separately
Audio:
ffmpeg -i audio.aac -c:a copy -hls_segment_type mpegts -map a:0 -hls_time 1 out_aud.m3u8
Video:
ffmpeg -i video.mp4 -c:v copy -hls_segment_type mpegts -map v:0 -hls_time 1 out_vid.m3u8
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.
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