I have been using FFMPEG for some time for audio file conversions. I just downloaded the newest version and the command below:
ffmpeg -i x.flac -ab 128k x.mp3
no longer produces a 128k file.
It produces a 48k file no matter what the -ab command requests. Any suggestions?
According to the documentation
`-ab bitrate'
Set the audio bitrate in bit/s (default = 64k).
So this is the bitrate, which means that per second by default 64k bits are used for storing sound information. So only a one second file would return a 128k file.
If I misunderstood your question and you mean that the bitrate is 48k instead of 128k then double check if ffmpeg is not using variable bit rate. If this is not the case I suggest you submit a bug report.
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.
24-bit sample sizes are not at all uncommon for PCM/WAV data, so I was surprised to see this:
Invalid sample format 's24'
... when I ran this:
ffmpeg -i input.oga -y -f wav -ar 44100 -sample_fmt s24 -ac 2 output.wav
When I look at the ffmpeg FAQ page it says that it doesn't support signed 24-bit sample sizes.
Fair enough, but I'm having a hard time accepting that this very powerful tool which supports an impressively large number of formats is somehow missing support for this really common sample width.
All I can think of is that maybe it's a build configuration issue.
So this question is...
Is there some way to configure ffmpeg to include support for signed 24-bit WAV output?
There is no sample format to compactly store 24-bit samples, but they can be stored in 32-bits with padding. For that, select a 24-bit PCM encoder
ffmpeg -i input.oga -y -f wav -ar 44100 -c:a pcm_s24le -ac 2 output.wav
Run ffmpeg -encoders | grep 24 to get a list of all 24-bit encoders.
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 am writing a small mp3 conversion tool. We upload a mp3 file & would like to convert it to a 96kbps file & a 320 kbps file. I have written the conversion script & it runs. But these files do not play.
Am i missing something?
the code i've written is:
/usr/local/bin/ffmpeg -i test.mp3 -vn -ar 441000 -ac 2 -ab 96k -f mp2 music/96/test.mp3 2>&1
Thanks!
There are two major issues here:
The audio rate you're setting (-ar 441000) is incorrect. You want 44.1 kHz (-ar 44100).
You're forcing the use of MPEG2 audio (-f mp2), which is not what you want, and is probably not supported by the player you're using either. Leave that option out entirely; the .mp3 extension on the output file will be used as a hint anyway.
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.