I'm writing a program where I need to be able to rip files from a CD into WAV format (or flac but wav works fine). It must run on Windows. I saw other answers where Express Rip and Audio Commander were recommended as command line tools. But Audio Commander's page doesn't seem to exist anymore. And I'm not so sure about express rip, it seems a bit sketchy.
Then they mentioned mutagen for retrieving the metadata.
Anyone have any experience with these utilities or this goal? I would like to be able to rip a CD in WAV, keep the metadata that is there, and if possible check the CD Archive for metadata as well.
Anyone ever do anything like this or have an suggestions on modules, utilities, methods, etc. to get me going? Even some small examples would help. That is examples of ripping a cd with python, or modules to accomplish the task.
You might want to have a look at PyMedia
PyMedia is a Python module for wav, mp3, ogg,
avi, divx, dvd, cdda etc files manipulations. It allows you to parse,
demutiplex, multiplex, decode and encode all supported formats. It can
be compiled for Windows, Linux and cygwin.
PyMedia was built to be really simple and flexible at the same time.
See tutorial for example. It allows you to create your own multimedia
applications in a matter of minutes and adjust it to your needs using
other components. Python language is chosen because of simple
semantics, complete and reach set of features.
You can also use this as a library:
From their Audio CD Grabber:
import pymedia.removable.cd as cd
def readTrack(track, offset, bytes):
cd.init()
if cd.getCount() == 0:
print 'There is no cdrom found. Bailing out...'
return 0
c = cd.CD(0)
props = c.getProperties()
if props['type'] != 'AudioCD':
print 'Media in %s has type %s, not AudioCD. Cannot read audio data.' % (c.getName(), props['type'])
return 0
tr0 = c.open(props['titles'][track - 1]['name'])
tr0.seek(offset, cd.SEEK_SET)
return tr0.read(bytes)
Update: For accessing metadata about the Audio CD you can use the PyCDDB lirbary.
Related
I'm working on a python3 project (in PyCharm, using Windows 10) and I'm trying to play a series of tones and save them all to a mp3 file.
I can get it to play a series of notes with winsound:
import winsound
while True:
winsound.Beep(1000, 1000)
But I can't save it.
All help appreciated!
If you want to save output to a file, rather than just play it, probably the most ideal way of doing this is to generate midi files. There are a few packages that can help you create midi files in a similar way.
This example uses the package called mido
from mido import Message, MidiFile, MidiTrack
mid = MidiFile()
track = MidiTrack()
mid.tracks.append(track)
track.append(Message('note_on', note=64, velocity=64, time=32))
mid.save('new_song.mid')
You can convert midi files to MP3s easily if needed (many programs or even online tools can do this) although, midi files are generally well-supported in most applications.
See also
Although perhaps dated, you can also check the PythonInMusic wiki for references to more MIDI and other audio libraries.
Alternatively, you can use a separate program to record your audio output when running your python script using winsound.
I am currently attempting to undergo a project to automate my music library cataloging using Python scripts, and I desperately need help before I go insane. For starters, I am using Linux (Arch Linux to be specific). To explain, I recently began using Morituri to rip my CD collection. I especially like the Mortituri script because it checks the files three times over, and generates FLAC files immediately.
This above system works fine, but I ran into another problem. I like FLAC files for my PC, but for mobile usage, I need MP3 files. I came across this famous little bash script:
for f in *.flac; do flac -cd "$f" | lame -b 320 - "${f%.*}".mp3; done
I saw this great little script, I knew there must be some way to replicate this in python. So I began working on a script that converts FLAC to MP3, and uses the basic arguments used by their respecitve CLI interfaces. I decided it be best to start with a single file at a time, and work my way up.After learning a bit about the subprocess module and pipes in python, I wrote the code below:
#!/usr/bin/python3
from subprocess import *
MyTestFile = ('/path/to/file.flac')
def Convert_It(File):
with open(File, 'r') as infile:
FlacDecode = Popen(["flac", "-cd","-","-"],stdin=infile,stdout=PIPE)
LameEncode = Popen(["lame", "-b", "192","-", "/a/differnet/path/test.mp3"],stdin=PIPE)
Convert_It(MyTestFile)
As it turns out, the output of this script is as follows:
flac 1.3.1, Copyright (C) 2000-2009 Josh Coalson, 2011-2014 Xiph.Org Foundation
flac comes with ABSOLUTELY NO WARRANTY. This is free software, and you are
welcome to redistribute it under certain conditions. Type `flac' for details.
-: ERROR while decoding metadata
state = FLAC__STREAM_DECODER_END_OF_STREAM
Warning: unsupported audio format
So I have a few questions:
Why is FLAC acting this way? In testing the script, I set to path to a real FLAC file. Why it say this?
How can I make this work?!
Really, any method using simple commands. Programs likes ffmpeg are not very easy to understand (how to set LAME q value to 0? No idea.) And other python scripts are way to long (like flac2all) for me to reverse engineer.
In review, any help is accepted. Just as long as it follows my specifications of: 1. only using the bare FLAC and LAME tools for linux (no sox or ffmpeg), 2. A script not too long. 3. Doesn't create a temporary WAV file. 4. Can encode any Flac file to MP3. I'm not even worried about metadata. Just the file itself.
Please HELP!
EDIT 08/05/2015:
I discovered I was using pipes incorrectly. Here is the code I created that solves my problem:
FlacDecode = subprocess.Popen(["flac","-cds",MyFile,"-"],stdout=subprocess.PIPE)
LameEncode = subprocess.Popen(['lame','-b','320','-'(NewFilePath)],stdin=FlacDecode.stdout,stdout=subprocess.PIPE)
EndLame = LameEncode.communicate()[0]
I had a similar problem, converting flac files to mp3, and found a one line bash solution here which reads:
for f in *.flac; do ffmpeg -i "$f" -aq 1 "${f%flac}mp3"; done
I appreciate that this doesn't answer the question but it is an elegant solution to your problem.
I'd like to seperate the channels of a mp3 file in Python and save it in two other files.
Does anybody know a library for this.
Thanks in advance.
I assume you want to split the channels losslessly, without decoding MP3 and re-encoding it - otherwise you would not have mentioned MP3 at all and would have easily found many tools like Audacity to do that.
There are 4 channel modes of MP3 frames - this means 4 types of MP3 files: simple stereo, joint-stereo, dual-channel, mono. joint-stereo files can't be split without loss. mono files doesn't need splitting. The rest: stereo and dual-channel, consists of less than 0.1% of all MP3 files, technically can be split into 2 files, each for a channel, without loss. However there is not any tool on the Internet to do that - not any command line tool nor any GUI tool, because few need the function.
There are not any python library for you neither. Most libraries abstracted MP3 files into a common audio which you can manipulate, after decoding. pymad is the only one specific to MP3 file, and it can tell if a file is using any of the 4 channel modes, but does not offer to extract a channel without decoding it. If you write a new tool, you will have to work on raw MP3 files or produce a library for it.
And it is not easy to write a tool or library for it. It's one stream with 2 channels and not two streams interleaved on a frame level. You cannot simply work on MP3 frames, drop some frames, keep others, and manage to extract a channel out that way. It's a task for a professional, and perhaps best happen in a decoder project (like lame or libmad) and not in a file manipulation project (like mp3info or the python eyeD3). In other words, this feature is likely written in C, not python.
Implementaiton Note:
The task to build such a tool thus suits well for a computer science C-programming language course project:
1. it takes a lot of time to do;
2. it requires every skill learned from C programming course;
3. it can get wrong easily;
4. it is likely built on the work of other projects, a lesson of adaptating existing work;
5. is a damn-hard endeavor that no-one did before and thus very rewarding
6. perhaps can be done in 300 difficult lines of code instead of bloated simple Visual Basic code, thus is a good lession of modesty and quality;
7. and finally: nobody is waiting in an hurry for a working implementation.
All condition fits perfectly for a C-programming course project.
Implementation Note 2:
some bit-rates are only possible in mono mode (80kbps), and some bit-rates are only possible in stereo mode (e.g. 320kpbs). Luckily this does not present a problem in this task, because all dual-mp3 bit-rate can be mapped into a fitting mono-mp3 bit-rate -- but not vice versa!
I'm looking for a cat for aac music files (the stuff iTunes uses).
Use Case: My father in law will not touch computers except for audiobooks he downloads to his iPod. I have taught him some iTunes (Windows) basics, but his library is a mess. It turns out, that iTunes is optimized for listening to podcasts and random songs from your library, not for audiobooks.
I would like to write a script (preferably python, but comfortable with other stuff too) to import his audiobook cds in a sane fashion, combining the tracks of each cd into a bookmarkable aac file (.m4b?) and then adding that to iTunes so it shows up in the audiobooks section.
I have figured out how to talk to iTunes (there is a COM interface in Windows, look for the iTunes SDK). Using that interface, I can use iTunes to rip the CD to aac format. It's the actual concatenation of the aac files I'm having trouble with. Can't find the right stuff on the net...
I created a freeware program called "Chapter and Verse" to concatenate m4a (AAC) files into a single m4b audiobook file with chapter marks and metadata.
If you have already ripped the CD's to AAC using itunes (which you say you have) then the rest is easy with my software. I wrote it for this exact reason and scenario. You can download it from www.lodensoftware.com
After trying to work with SlideShow Assembler, the QT SDK and a bunch of other command line tools, I ended up building my own application based on the publicly available MP4v2 library. The concatenating of files and adding of chapters is done using the MP4v2 library.
There are quite a few nuances in building an audiobook properly formatted for the iPod. The information is hard to find. Working with Apple documentation and open libraries has its own challenges as well.
Best of Luck.
Not programming related (well, kinda.)
iTunes already has functionality to rip as a single track (e.g. an audiobook.) Check this out: http://www.ehow.com/how_2108906_merge-cd-single-track-itunes.html
That fixes your immediate problem, but I guess people can keep discussing how to do it programatically.
The most powerful Python audio manipulation module out there seems to be Python Audio Tools. The download comes with CLI tools that would probably do everything you'd want to do, even ripping, so you can even get by with shell scripting the whole thing. The module itself is also pretty powerful and has a handy set of functions to manipulate audio files. If you want to stick with writing everything in python, you can possibly learn enough to do what you want to do after studying their CLI source code. Specifically they have a tool that just does audio file cat in any codec. (They do depend on FAAC/FAAD2 for AAC support, but that'd be true for every library you'll find)
I haven't seen an aac codec library for python, but you could use wav files as an intermediary format.
You can pull the tracks off the cd as wav files, and then use the wave module to concatenate them into one large file, which could then be converted by itunes to aac. This may increase your processing time considerably because of the size of the data, but it would be fairly easy, and you don't need any external libraries.
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?
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.
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.
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.
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.
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).