Python library to modify MP3 audio without transcoding
I am looking for some general advice about the mp3 format before I start a small project to make sure I am not on a wild-goose chase.
My understanding of the internals of the mp3 format is minimal. Ideally, I am looking for a library that would abstract those details away. I would prefer to use Python (but could be convinced otherwise).
I would like to modify a set of mp3 files in a fairly simple way. I am not so much interested in the ID3 tags but in the audio itself. I want to be able to delete sections (e.g. drop 10 seconds from the 3rd minute), and insert sections (e.g. add credits to the end.)
My understanding is that the mp3 format is lossy, and so decoding it to (for example) PCM format, making the modifications, and then encoding it again to MP3 will lower the audio quality. (I would love to hear that I am wrong.)
I conjecture that if I stay in mp3 format, there will be some sort of minimum frame or packet-size to deal with, so the granularity of the operations may be coarser. I can live with that, as long as I get an accuracy of within a couple of seconds.
I have looked at PyMedia, but it requires me to migrate to PCM to process the data. Similarly, LAME wants to help me encode, but not access the data in place. I have seen several other libraries that only deal with the ID3 tags.
Can anyone recommend a Python MP3 library? Alternatively, can you disabuse me of my assumption that going to PCM and back is bad and avoidable?
Asked by: Catherine593 | Posted: 05-10-2021
If you want to do things low-level, use pymad. It turns MP3s into a buffer of sample data.
If you want something a little higher-level, use the Echo Nest Remix API (disclosure: I wrote part of it for my dayjob). It includes a few examples. If you look at the cowbell example (i.e., MoreCowbell.dj), you'll see a fork of pymad that gives you a NumPy array instead of a buffer. That datatype makes it easier to slice out sections and do math on them.Answered by: Fenton665 | Posted: 06-11-2021
I got three quality answers, and I thank you all for them. I haven't chosen any as the accepted answer, because each addressed one aspect, so I wanted to write a summary.
Do you need to work in MP3?
Transcoding to PCM and back to MP3 is unlikely to result in a drop in quality.
Don't optimise audio-quality prematurely; test it with a simple prototype and listen to it.
Working in MP3
Wikipedia has a summary of the MP3 File Format.
MP3 frames are short (1152 samples, or just a few milliseconds) allowing for moderate precision at that level.
However, Wikipedia warns that "Frames are not independent items ("byte reservoir") and therefore cannot be extracted on arbitrary frame boundaries."
Existing libraries are unlikely to be of assistance, if I really want to avoid decoding.
Working in PCM
There are several libraries at this level:
- LAME (latest release: October 2017)
- PyMedia (latest release: February 2006)
- PyMad (Linux only? Decoder only? Latest release: January 2007)
Working at a higher level
Echo Nest Remix API (Mac or Linux only, at the moment) is an API to a web-service that supports quite sophisticated operations (e.g. finding the locations of music beats and tempo, etc.)
mp3DirectCut (Windows only) is a GUI that apparently performs the operations I want, but as an app. It is not open-source. (I tried to run it, got an Access Denied installer error, and didn't follow up. A GUI isn't suitably for me, as I want to repeatedly run these operations on a changing library of files.)
My plan is now to start out in PyMedia, using PCM.Answered by: Anna222 | Posted: 06-11-2021
Mp3 is lossy, but it is lossy in a very specific way. The algorithms used as designed to discard certain parts of the audio which your ears are unable to hear (or are very difficult to hear). Re-doing the compression process at the same level of compression over and over is likely to yield nearly identical results for a given piece of audio. However, some additional losses may slowly accumulate. If you're going to be modifying files a lot, this might be a bad idea. It would also be a bad idea if you were concerned about quality, but then using MP3 if you are concerned about quality is a bad idea over all.
You could construct a test using an encoder and a decoder to re-encode a few different mp3 files a few times and watch how they change, this could help you determine the rate of deterioration and figure out if it is acceptable to you. Sounds like you have libraries you could use to run this simple test already.
MP3 files are composed of "frames" of audio and so it should be possible, with some effort, to remove entire frames with minimal processing (remove the frame, update some minor details in the file header). I believe frames are pretty short (a few milliseconds each) which would give the precision you're looking for. So doing some reading on the MP3 File Format should give you enough information to code your own python library to do this. This is a fair bit different than traditional "audio processing" (since you don't care about precision) and so you're unlikely to find an existing library that does this. Most, as you've found, will decompress the audio first so you can have complete fine-grained control.Answered by: Paul368 | Posted: 06-11-2021
Not a direct answer to your needs, but check the mp3DirectCut software that does what you want (as a GUI app). I think that the source code is available, so even if you don't find a library, you could build one of your own, or build a python extension using code from mp3DirectCut.Answered by: Fiona782 | Posted: 06-11-2021
As for removing or extracting mp3 segments from an mp3 file while staying in the MP3 domain (that is, without conversion to PCM format and back), there is also the open source package PyMp3Cut.
As for splicing MP3 files together (adding e.g. 'Credits' to the end or beginning of an mp3 file) I've found you can simply concatenate the MP3 files providing that the files have the same sampling rate (e.g. 44.1khz) and the same number of channels (e.g. both are stereo or both are mono).Answered by: Chelsea240 | Posted: 06-11-2021
python - fuse & gstreamer transcoding
I'm trying to create a FUSE fs which transcodes all sound files to mp3. My first idea is to use gstreamer as the backend for transcoding. I thought about using this pipeline: gst-launch -v filesrc location=01\ New\ Born.flac ! decodebin ! audioconvert ! lame vbr=4 vbr-quality=9 ! id3v2mux ! appsink The python bindings of fuse expect calls this function when a file is being read: ...
ffmpeg transcoding in python with aws-lambda limits
I want to transcode audio file into HLS with segments in aws-lambda environment: ffmpeg -i 2540.mp3 -f segment -segment_time 20 -segment_list battle.m3u8 -segment_format mpegts battle%05d.ts But there are cases when I need to transcode files larger than 500 MB (disc space limit for aws-lambda). How can I trancode audio file parts with segments? Is it possible to redirect segments t...
python - ffmpeg Inserting Extra Frame when Transcoding a Stack of Images Using H.264
So I have a stack of images named 1.jpeg, 2.jpeg, and so on. I've been using the following command in Python successfully to create a .mp4 from these images: os.system("ffmpeg -f image2 -i %d.jpeg -vcodec libx264 -crf 17 -r " + str(FrameRate) + " " + OutputDirectory + "\\" + OutputName + ".mp4") The problem I'm encountering right now is that...
python - Replacing pixels and saving RTSP streams as files without transcoding
Essentially, the issue is this; I have 8 security cameras all streaming via RTSP with very high unmodifiable resolutions (specifically, 2560x1440). I want to place a small tag in the corner of the each video stream to display a string of characters with a white background, then save the file locally. Currently, I am using Python and OpenCV to process the frame, receive the string to apply, modify the frame, then sav...