I'm looking for a way to re-encode the video stream of a movie only and keep all other streams as they are using ffmpeg or more specific streamio/streamio-ffmpeg (Github - StreamIO-FFMPEG).
I already tried various combinations of -map 0 or -map a:0 -map s:0, but in all combinations I tried, either nothing is encoded at all, or not all other streams are copied to the new file. In most cases there is only one audio stream after encoding, when there were two before, and sometimes the subtitle streams are lost, too. Also most times the info what language the streams are in gets lost.
So when I have a movie file (mkv) with the following streams:
0: video [H.264, 1080p]
1: audio [english, mp3]
2: audio [french, mp3]
3: subtitle [english (forced)]
4: subtitle [english]
What should the ffmpeg parameters be, if I want to encode the video file to H.265 and 720p and keep all other streams as they are?
What should the parameters be, if I additionally want to encode the audio streams as AAC?
Thanks in advance!
Use
ffmpeg -i in.mkv -vf scale=hd720 -map 0 -c copy -c:v libx265 out.mkv
To encode audio as well,
ffmpeg -i in.mkv -vf scale=hd720 -map 0 -c copy -c:v libx265 -c:a aac out.mkv
(The order of the arguments above matter)
Related
I'm trying to encode 6 arbitrary mono audio streams into a single AAC 5.1 track in an mp4 container (here with test streams):
ffmpeg -f lavfi -i testsrc=duration=10:size=100x100:rate=30 -f lavfi -i aevalsrc="-2+random(0)" -filter_complex "[1:a][1:a][1:a][1:a][1:a][1:a]join=inputs=6:channel_layout=5.1:map=0.0-FL|1.0-FR|2.0-FC|3.0-LFE|4.0-BL|5.0-BR[a]" -map '0:v' -map "[a]" -c:a aac -channel_layout 5.1 -t 10 testlfe.mp4
5 of the channels replicate the input audio just fine (modulo encoding). However, the LFE channel is lowpassed. Extracting with:
ffmpeg -i testlfe.mp4 -filter_complex "channelsplit=channel_layout=5.1:channels=LFE[LFE]" -map '[LFE]' testlfe.wav
I get a lowpassed rumble, instead of the original full white noise
(from ffmpeg -i testlfe.wav -lavfi showspectrumpic=s=640x320 testlfe.png)
Is there a way to prevent the lowpass from happening?
I couldn't find any references whether that's inherent to the AAC 5.1 encoding, something that ffmpeg does, or inherent to the decoding process. (I did decode my same test files using something that uses Microsoft MediaFoundation and the LFE channel was still lowpassed).
Turns out, the AAC codec inherently limits the LFE bandwidth, so there's no way around it.
(thanks to kesh in the comments) Wikipedia's Advanced Audio Encoding article claims the upper limit is 120Hz which matches my spectrogram, but doesn't cite a source. The actual ISO/IEC 13818-7:2006(en) Standard costs a bunch of money to read as usual, but in the free glossary there is an entry:
low frequency enhancement ( LFE ) channel:
limited bandwidth channel for low frequency audio effects in a multichannel system
Encode with
ffmpeg -i 6channels.wav -filter "channelmap=0|1|2|3|4|5:6.0(front)" -c:a libfdk_aac -ac 6 -profile:a aac_he -vbr 1 -cutoff 18000 -movflags +faststart 6channels-vbr1-fdk.m4a
It can also be done with regular aac codec.
I am using the following command to stream a video and it's audio to localhost:
ffmpeg -re -i out.mp4 -map 0:0 -vcodec libx264 -f h264 udp://127.0.0.1:1234 -map 0:1 -acodec libfaac -f mp4a udp://127.0.0.1:2020
FFmpeg is not recognising my audio codec and my audio format so I get the following error message:
Error
What audio format and codec do I need to use? The codec information of the video I wish to send is as follows:
Codecs used
When I convert the audio track to mp3 I can run the above command and stream the video and audio properly. However I dont want to convert all my video audio-tracks to mp3.
(I am confused by all the encoders, decoders, codec names in the ffmpeg documentation) Is there a way of finding the right encoder to use with the mp4a audio codec other than reading the whole list of codecs and options?
Thanks.
I'm using latest static build of ffmpeg windows.
My input file (.mkv) is:
[video] - 1080, V_MPEG4/ISO/AVC, 14.6 Mbps, ID#0
[audio] - DTS 5.1, 1510 Kbps, ID#1
[subtitles] - S_TEXT/ASS Lossless English, ID#14
My problem is this: I convert the audio, so that my target player, a XB1 console (media support faq), is able to play audio/video. However sometimes its rather difficult to hear or parts may be in foreign language, so I want to force the english subtitles into the mix at the same time I convert the audio.
Currently for the audio, I use the following command
ffmpeg -i input.mkv -codec copy -acodec ac3 output.mkv
Can I somehow tie in the forced subtitles (onto the video) in order to save an extra process of taking the output.mkv and trying to force subtitles on?
Edit: I've tried using the following command to extract subtitles to be able to edit them
ffmpeg -i Movie.mkv -map 0:s:14 subs.srt
However i get the error: Stream map '0:s:14' matches no streams
Edit2: attempted to extract subtitles and succeeded with
ffmpeg -i input.mkv -map 0:14 -c copy subtitles.ass
but still looking to force the subtitles, nonetheless!
Also - a little bonus to this question - can I somehow extract the .ass file and edit it to only produce subtitles for foreign parts - so english audio doesn't have subtitles during the movie but foreign audio does have subtitles?
Cheers
Edit3:
When I try to use both of the commands at once (my earlier mentioned audio converter & one from the ffmpeg wiki)
ffmpeg -i input.mkv -codec copy -acodec ac3 -vf "ass=subs.ass" output.mkv
I get the following error from ffmpeg,
Filtergraph 'ass=subs.ass' was defined for video output stream 0:0 but codec copy was selected.
Filtering and streamcopy cannot be used together.
Since your media player does not support subtitles, the text has to be burnt onto the video image. For that, use
ffmpeg -i input.mkv -vf "ass=subs.ass" -c:v libx264 -crf 20 -c:a ac3 output.mkv
This will re-encode the video, since text is being added. The CRF value controls the video quality. Lower values produce better quality but larger files. 18 to 28 is a decent range to try.
I have converted my .mkv file into an .mp4 by using the command:
sudo avconv -i input.mkv -codec copy output.mp4
I am trying to play the mp4 file in the browser, but the audio is not playing. The video player shows that it is on mute, but the button is disabled so you cannot turn it off of mute.
Other mp4s are working, but they were not converted from .mkv. Any help would be much appreciated.
In an MP4 container, browsers usually support only H.264 video and either AAC or MP3 audio. The output from your avconv command should show the format of your video and audio; look under "Input #0" for the lines that start with "Stream #". If you audio is not already AAC or MP3 you will want to convert it instead of just copying it to the MP4 container. You can copy the video and convert only the audio with a command like this:
avconv -i input.mkv -c:v copy -c:a libmp3lame -q:a 2 output.mp4
The -c:v copy will copy the video, and -c:a libmp3lame will convert the audio to MP3 using the libmp3lame encoder. -q:a 2 sets the audio quality; use a lower number for better quality (and a larger file). You could instead convert to AAC audio if your avconv was configured with non-free codecs enabled and a good quality AAC encoder.
I have an avi file that has different levels of audio. Is there a way to decrease and increase appropriately where needed the audio of my file using ffmpeg?
In ffmpeg you can use the volume filter to change the volume of a track. Make sure you download a recent version of the program.
Find out the gain to apply
First you need to analyze the audio stream for the maximum volume to see if normalizing would even pay off:
ffmpeg -i video.avi -af "volumedetect" -f null /dev/null
Replace /dev/null with NUL on Windows. This will output something like the following:
[Parsed_volumedetect_0 # 0x7f8ba1c121a0] mean_volume: -16.0 dB
[Parsed_volumedetect_0 # 0x7f8ba1c121a0] max_volume: -5.0 dB
[Parsed_volumedetect_0 # 0x7f8ba1c121a0] histogram_0db: 87861
As you can see, our maximum volume is -5.0 dB, so we can apply 5 dB gain. If you get a value of 0 dB, then you don't need to normalize the audio.
Apply the volume filter:
Now we apply the volume filter to an audio file. Note that applying the filter means we will have to re-encode the audio stream. What codec you want for audio depends on the original format, of course. Here are some examples:
Plain audio file: Just encode the file with whatever encoder you need:
ffmpeg -i input.wav -af "volume=5dB" output.mp3
Your options are very broad, of course.
AVI format: Usually there's MP3 audio with video that comes in an AVI container:
ffmpeg -i video.avi -af "volume=5dB" -c:v copy -c:a libmp3lame -q:a 2 output.avi
Here we chose quality level 2. Values range from 0–9 and lower means better. Check the MP3 VBR guide for more info on setting the quality. You can also set a fixed bitrate with -b:a 192k, for example.
MP4 format: With an MP4 container, you will typically find AAC audio. We can use ffmpeg's build-in AAC encoder.
ffmpeg -i video.mp4 -af "volume=5dB" -c:v copy -c:a aac -strict experimental -b:a 192k output.mp4
Here you can also use other AAC encoders. Some of them support VBR, too. See this answer and the AAC encoding guide for some tips.
In the above examples, the video stream will be copied over using -c:v copy. If there are subtitles in your input file, or multiple video streams, use the option -map 0 before the output filename.
The author's info is: Jon Skarpeteig in SuperUser
You can use my ffmpeg-normalize script for that.
First, install a recent version of ffmpeg. Then, install via pip install ffmpeg_normalize, then run it on an AVI file:
ffmpeg-normalize input.avi -o output.mkv -c:a aac -b:a 192k
Here, we're choosing to re-encode the audio with AAC at 192 kBit/s, and copy the video stream over to the output. This will perform EBU R128 normalization, but simple peak/RMS normalization is also possible.