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I've got many, many mp3 files that I would like to merge into a single file. I've used the command line method
copy /b 1.mp3+2.mp3 3.mp3
but it's a pain when there's a lot of them and their namings are inconsistent. The time never seems to come out right either.
David's answer is correct that just concatenating the files will leave ID3 tags scattered inside (although this doesn't normally affect playback, so you can do "copy /b" or on UNIX "cat a.mp3 b.mp3 > combined.mp3" in a pinch).
However, mp3wrap isn't exactly the right tool to just combine multiple MP3s into one "clean" file. Rather than using ID3, it actually inserts its own custom data format in amongst the MP3 frames (the "wrap" part), which causes issues with playback, particularly on iTunes and iPods. Although the file will play back fine if you just let them run from start to finish (because players will skip these is arbitrary non-MPEG bytes) the file duration and bitrate will be reported incorrectly, which breaks seeking. Also, mp3wrap will wipe out all your ID3 metadata, including cover art, and fail to update the VBR header with the correct file length.
mp3cat on its own will produce a good concatenated data file (so, better than mp3wrap), but it also strips ID3 tags and fails to update the VBR header with the correct length of the joined file.
Here's a good explanation of these issues and method (two actually) to combine MP3 files and produce a "clean" final result with original metadata intact -- it's command-line so works on Mac/Linux/BSD etc. It uses:
mp3cat to combine the MPEG data frames only into a continuous file, then
id3cp to copy all metadata over to the combined file, and finally
VBRFix to update the VBR header.
For a Windows GUI tool, take a look at Merge MP3 -- it takes care of everything. (VBRFix also comes in GUI form, but it doesn't do the joining.)
As Thomas Owens pointed out, simply concatenating the files will leave multiple ID3 headers scattered throughout the resulting concatenated file - so the time/bitrate info will be wildly wrong.
You're going to need to use a tool which can combine the audio data for you.
mp3wrap would be ideal for this - it's designed to join together MP3 files, without needing to decode + re-encode the data (which would result in a loss of audio quality) and will also deal with the ID3 tags intelligently.
The resulting file can also be split back into its component parts using the mp3splt tool - mp3wrap adds information to the IDv3 comment to allow this.
Use ffmpeg or a similar tool to convert all of your MP3s into a consistent format, e.g.
ffmpeg -i originalA.mp3 -f mp3 -ab 128kb -ar 44100 -ac 2 intermediateA.mp3
ffmpeg -i originalB.mp3 -f mp3 -ab 128kb -ar 44100 -ac 2 intermediateB.mp3
Then, at runtime, concat your files together:
cat intermediateA.mp3 intermediateB.mp3 > output.mp3
Finally, run them through the tool MP3Val to fix any stream errors without forcing a full re-encode:
mp3val output.mp3 -f -nb
The time problem has to do with the ID3 headers of the MP3 files, which is something your method isn't taking into account as the entire file is copied.
Do you have a language of choice that you want to use or doesn't it matter? That will affect what libraries are available that support the operations you want.
MP3 files have headers you need to respect.
You could ether use a library like Open Source Audio Library Project and write a tool around it.
Or you can use a tool that understands mp3 files like Audacity.
What I really wanted was a GUI to reorder them and output them as one file
Playlist Producer does exactly that, decoding and reencoding them into a combined MP3. It's designed for creating mix tapes or simple podcasts, but you might find it useful.
(Disclosure: I wrote the software, and I profit if you buy the Pro Edition. The Lite edition is a free version with a few limitations).
As David says, mp3wrap is the way to go. However, I found that it didn't fix the audio length header, so iTunes refused to play the whole file even though all the data was there. (I merged three 7-minute files, but it only saw up to the first 7 minutes.)
I dug up this blog post, which explains how to fix this and also how to copy the ID3 tags over from the original files (on its own, mp3wrap deletes your ID3 tags). Or to just copy the tags (using id3cp from id3lib), do:
id3cp original.mp3 new.mp3
I would use Winamp to do this. Create a playlist of files you want to merge into one, select Disk Writer output plugin, choose filename and you're done. The file you will get will be correct MP3 file and you can set bitrate etc.
I'd not heard of mp3wrap before. Looks great. I'm guessing someone's made it into a gui as well somewhere. But, just to respond to the original post, I've written a gui that does the COPY /b method. So, under the covers, nothing new under the sun, but the program is all about making the process less painful if you have a lot of files to merge...AND you don't want to re-encode AND each set of files to merge are the same bitrate. If you have that (and you're on Windows), check out Mp3Merge at: http://www.leighweb.com/david/mp3merge and see if that's what you're looking for.
If you want something free with a simple user interface that makes a completely clean mp3 I recommend MP3 Joiner.
Features:
Strips ID3 data (both ID3v1 and ID3v2.x) and doesn't add it's own (unlike mp3wrap)
Lossless joining (doesn't decode and re-encode the .mp3s). No codecs required.
Simple UI (see below)
Low memory usage (uses streams)
Very fast (compared to mp3wrap)
I wrote it :) - so you can request features and I'll add them.
Links:
MP3 Joiner website: Here
Latest installer: Here
Personally I would use something like mplayer with the audio pass though option eg -oac copy
Instead of using the command line to do
copy /b 1.mp3+2.mp3 3.mp3
you could instead use "The Rename" to rename all the MP3 fragments into a series of names that are in order based on some kind of counter. Then you could just use the same command line format but change it a little to:
copy /b *.mp3 output_name.mp3
That is assuming you ripped all of these fragment MP3's at the same time and they have the same audio settings. Worked great for me when I was converting an Audio book I had in .aa to a single .mp3. I had to burn all the .aa files to 9 CD's then rip all 9 CD's and then I was left with about 90 mp3's. Really a pain in the a55.
Related
I have a flow where iOS app users will record a large video file and upload it to our server. After the fact, the user might want to extract certain portions of that larger video based on specific time stamps and generate a highlight reel that can be viewed and shared locally back on the iOS device.
As a FE developer I don't really have much experience with where to even start here. Our BE will be built in NodeJS. It seems to me that this should be a relatively straightforward problem to solve, but I don't know.
Are there APIs that make movie manipulation easy? Can I easily extract a clip based on a start and stop time and save that as a separate file? Are those costly tasks? Or not too bad?
I'm guessing that the response to this call would be a list of a series of file names that have been generated as a result of these clips being generated, that the iOS app could then pull down and load.
It's not quite as straightforward as it might seem as video files are quite structured with header information and indexing into the individual video and audio tracks and frames. Any splitting up or cropping needs to allow for this and also create new files with the correct headers and indexing etc.
Fortunately, there are indeed libraries that you can use to do this type of thing, one of the most powerful being ffmpeg.
There are projects which allow the ffmpeg command line tool be used programatically - the advantage of this approach is that you get to leverage the vast community knowledge base for ffmpeg command line.
One of the popular ones for nodejs is:
https://github.com/damianociarla/node-ffmpeg
You can then look at the ffmpeg documentation or community answers to find the particularly functionality you need - for example to crop video at a start and end time as you asked:
https://stackoverflow.com/a/42827058/334402
https://superuser.com/a/704118
The general idea is quite simple and will be of the format:
ffmpeg -i yourInputVideo.mp4 -ss 01:30:00 -to 02:30:00 -c copy copy yourNewOutputVideo.mp4
It's worth taking a look at the seeking info in the ffmpeg online documentation (https://ffmpeg.org/ffmpeg.html) to help understand the examples, especially the second one above:
-ss position (input/output)
When used as an input option (before -i), seeks in this input file to position. Note that in most formats it is not possible to seek exactly, so ffmpeg will seek to the closest seek point before position. When transcoding and -accurate_seek is enabled (the default), this extra segment between the seek point and position will be decoded and discarded. When doing stream copy or when -noaccurate_seek is used, it will be preserved.
When used as an output option (before an output url), decodes but discards input until the timestamps reach position.
position must be a time duration specification, see (ffmpeg-utils)the Time duration section in the ffmpeg-utils(1) manual.
I want to be able to set the "Title" and "Comments" (listed in properties->details) of some mp3 files in Windows using python. Is this possible, perhaps with a library like PyWin32? Also, would these details be visible in other operating systems or are they Windows-specific? Thanks.
Simple Answer:
Yes, you can set 'Title' and 'Comments' (and many other fields) of an mp3 file in Windows using Python.
Also, the details are visible on all operating systems and are not windows specific.
First you have to understand what is mp3 file and how data is organized within an mp3 file.
Detailed Answer:
Raw audio consumes a lot of size. For example, an audio signal of 10 sec sampled 48 kHz and having a bit depth of 16 bits per sample will be of size 10*48000*16 bits, which is close to 1 MB. So, for a 5 minute song, it will almost take 30 MB. But, if you observe, most 5 min mp3 songs are of size around 5 MB (of course it depends on sampling frequency, bit depth and amount of compression used). How is it possible? It is possible because we compress the data using signal processing techniques which in itself is a big topic altogether which we will not discuss here. So, to create an mp3 file we need something called encoder which converts the raw audio data to compressed data and every time you play an mp3 song, decoder is used which converts the data from compressed format to raw audio, which is what you can only listen. So, compression is done for saving storage and also transmission bandwidth (basically saving amount of data to be transmitted over internet).
Now, coming to how data is organized inside an mp3 file. mp3 file will obviously contain the compressed data. In addition many mp3 files contain some meta data (like Title and Comments you mentioned in your question). There are several formats for storing this meta data. So, a decoder which is decoding mp3 file should also support decoding of meta-data, then only you can see the information, other wise you can't see. The meta data is operating system independent, and can be seen on any operating system provided you have a proper decoder.
Finally, yes you can edit the meta data on windows (for that matter on any OS) using python. If you want to do this, using only python without any library, you need to understand how data is organized inside an mp3 file, find the meta-data inside it, edit it and store it back. But, there are libraries and packages in python which support editing meta-data of mp3 file. You can use them directly. Also, the meta data is independent of OS, and once you edit your properties, you should be able to see the properties in any OS provided the decoder you use has the support.
Some links which will help you:
mp3 tag tool
Another stack overflow question which gives details about libraries that support viewing and editing of meta data using Python
Another post here answered the question of creating 30 second preview clips from WAV audio files (Create mp3 previews from wav and aiff files). My needs slightly overlap, but differing details are beyond my knowledge.
Requirements/Options: clip length; beginning & ending fade length; input filetypes: m4a/AAC/AIFF; output filetype: mp3; kbps (e.g. 192); original files unaltered; suffix new mp3 names with " (Preview)"
Limitations: no uploading of original files to a server (desktop processing); no compiling (unix/Terminal/Bash script only); recursive processing of files in sub-directories
Any/All assistance and advice is welcome.
You'll most likely get the best results with a DAW (digital audio workstation) and an audio file converter.
For a DAW, Reaper comes with a 60 day trial, and it has everything you need to cut the songs where you need and to do fade ins/fade outs, and other effects if you'd like.
www.reaper.fm
Simply use a converter to convert the m4a file to .wav, .mp3 or whatever you prefer, and then if you need it back in m4a, convert it back. I say this because some DAWs can't work with m4a files, but if which ever one you choose to work with can then no conversion is necessary,
There are many options for what DAW and what converter you use, I recommend Reaper for a DAW, and most converters essentially do the same thing, so it doesn't make much of a difference which one you choose.
Hope this helps!
Apple gives an example of support for byte-range segments in m3u8 files for HLS
#EXTM3U
#EXT-X-TARGETDURATION:11
#EXT-X-MEDIA-SEQUENCE:0
#EXT-X-VERSION:4
#EXTINF:10.0,
#EXT-X-BYTERANGE:75232#0
media.ts
#EXTINF:10.0,
#EXT-X-BYTERANGE:82112#752321
media.ts
#EXTINF:10.0,
#EXT-X-BYTERANGE:69864
media.ts
But I cannot figure out how to create such playlist for given .ts file.
Are there any tools for that?
There is -hls_flags as a ffmpeg option. (https://www.ffmpeg.org/ffmpeg-formats.html)
Following command generates single ts file which is segmented by byte range feature(supported from HLS version 4) in m3u8 index file.
$ ffmpeg -i sample.mp3 -hls_time 20 -hls_flags single_file out.m3u8
Looks like
ffprobe -show_frames media.ts -print_format json
gives enough information about frames to build such playlist, although some scripting will be required to construct it.
I'll update this answer with script if I succeed with that approach.
Update:
Here is couple of useful links I've found by now:
Bash scripts for generating iframe playlists - needs a bit of optimization, as it calls ffprobe multiple times
iframe-playlist-generator - project on python that can be used to generate iframe playlists from usual ones
It is not exactly what I've searched initially, but I-Frame playlists are similar to byte-range ones and fit for my task even better, so I'm going to use these two projects as a reference/starting point to create something a bit more suitable for me.
The projects actually use different methods to find size of I-Frame - the bash script just uses what ffprobe shows in pkt_size, and the python project adds a bit of voodoo by calculating size as difference of positions of packets and adding 188 to match example playlists from apple. 188 bytes is the size of mpeg-ts packet, probably that is related somehow, I have not managed to understand how, however. This difference in size calculation causes different playlists to be generated, probably one of them is incorrect in some way, but actually VLC plays both without any problems, so I'm going to stick to simpler method until it will be proven as incorrect.
Update 2:
I've created a ruby module that can extract I-Frame information of given .ts file with ffprobe and build both I-Frame and usual byterange m3u8 playlist (as it was requested in question) based on that information.
I've found the simple method of creating I-Frame playlist I mentioned before to be incorrect, so I used the method from iframe-playlist-generator. The output is almost similar to the I-Frame playlist generated by mediafilesegmenter -output-single-file -file-base output-dir/ input.ts, mentioned by Duvrai, but sometimes there are some 188-byte size misses for random frames, I could not understand the pattern, so it is currently ignored.
You can use a standard segmenter such as Apple's mediafilesegmenter, check the lengths of the files, and then concatenate (with the cat program) them into a single file. From the file sizes you have all the information needed to specify the byte ranges in a playlist file.
Not as nice as just downloading a tool from the net, but it's not a very complicated algorithm.
Unified Streaming also offers a tool that can do this for you:
mp4split --package-hls output-single-file -o prog_index.m3u8 input.mp4
This is part of their commercial streaming package (they offer a free trial upon request). They also provide an Amazon AWS instance with hourly fees.
I'm looking for a solution to this task: I want to open any audio file (MP3,FLAC,WAV), then proceed it to the extracted form and hash this data. The thing is: I don't know how to get this extracted audio data. DirectX could do the job, right? And also, I suppose if I have fo example two MP3 files, both 320kbps and only ID3 tags differ and there's a garbage inside on of the files mixed with audio data (MP3 format allows garbage to be inside) and I extract both files, I should get the exactly same audio data, right? I'd only differ if one file is 128 and the other 320, for example. Okay so, the question is, is there a way to use DirectX to get this extracted audio data? I imagine it'd be some function returning byte array or something. Also, it would be handy to just extract whole file without playback. I want to process hundreds of files so 3-10min/s each (if files have to be played at natural speed for decoding) is way worse that one second for each file (only extracting)
I hope my question is understandable.
Thanks a lot for answers,
Aaron
Use http://sox.sourceforge.net/ (multiplatform). It's faster than realtime as you'd like, and it's designed for batch mode much more than DirectX. For example, sox -r 48k -b 16 -L -c 1 in.mp3 out.raw. Loop that over your hundreds of files using whatever scripting language you like (bash, python, .bat, ...).