As part of a project I am working on, there is a requirement to concatenate multiple pieces of audio data into one large audio file. The audio files are generated from four sources, and the individual files are stored in a Google Cloud storage bucket. Each file is an mp3 file, and it is easy to verify that each individual file is generating correctly (individually, I can play them, edit them in my favourite software, etc.).
To merge the audio files together, a nodejs server loads the files from the Google Cloud storage as an array buffer using an axios POST request. From there, it puts each array buffer into a node Buffer using Buffer.from(), so now we have an array of Buffer objects. Then it uses Buffer.concat() to concatenate the Buffer objects into one big Buffer, which we then convert to Base64 data and send to the client server.
This is cool, but the issue arises when concatenating audio generated from different sources. The 4 sources I mentioned above are Text to Speech software platforms, such as Google Cloud Voice and Amazon Polly. Specifically, we have files from Google Cloud Voice, Amazon Polly, IBM Watson, and Microsoft Azure Text to Speech. Essentially just five text to speech solutions. Again, all individual files work, but when concatenating them together via this method there are some interesting effects.
When the sound files are concatenated, seemingly depending on which platform they originate from, the sound data either will or will not be included in the final sound file. Below is a 'compatibility' table based on my testing:
|------------|--------|--------|-----------|-----|
| Platform / | Google | Amazon | Microsoft | IBM |
|------------|--------|--------|-----------|-----|
| Google | Yes | No | No | No |
|------------|--------|--------|-----------|-----|
| Amazon | | No | No | Yes |
|------------|--------|--------|-----------|-----|
| Microsoft | | | Yes | No |
|------------|--------|--------|-----------|-----|
| IBM | | | | Yes |
|------------|--------|--------|-----------|-----|
The effect is as follows: When I play the large output file, it will always start playing the first sound file included. From there, if the next sound file is compatible, it is heard, otherwise it is skipped entirely (no empty sound or anything). If it was skipped, the 'length' of that file (for example 10s long audio file) is included at the end of the generated output sound file. However, the moment that my audio player hits the point where the last 'compatible' audio has played, it immediately skips to the end.
As a scenario:
Input:
sound1.mp3 (3s) -> Google
sound2.mp3 (5s) -> Amazon
sound3.mp3 (7s)-> Google
sound4.mp3 (11s) -> IBM
Output:
output.mp3 (26s) -> first 10s is sound1 and sound3, last 16s is skipped.
In this case, the output sound file would be 26s seconds long. For the first 10 seconds, you would hear the sound1.mp3 and sound3.mp3 played back to back. Then at 10s (at least playing this mp3 file in firefox) the player immediately skips to the end at 26s.
My question is: Does anyone have any ideas why sometimes I can concatenate audio data in this way, and other times I cannot? And how come there is this 'missing' data included at the end of the output file? Shouldn't concatenating the binary data work in all cases if it works for some cases, as all the files have mp3 encoding? If I am wrong please let me know what I can do to successfully concatenate any mp3 files :)
I can provide my nodeJS backend code, but the process and methods used are described above.
Thanks for reading?
Potential Sources of Problems
Sample Rate
44.1 kHz is often used for music, as it's what is used on CD audio. 48 kHz is usually used for video, as it's what was used on DVDs. Both of those sample rates are much higher than is required for speech, so it's likely that your various text-to-speech providers are outputting something different. 22.05 kHz (half of 44.1 kHz) is common, and 11.025 kHz is out there too.
While each frame specifies its own sample rate, making it possible to generate a stream with varying sample rates, I've never seen a decoder attempt to switch sample rates mid-stream. I suspect that the decoder is skipping these frames, or maybe even skipping over an arbitrary block until it gets consistent data again.
Use something like FFmpeg (or FFprobe) to figure out what the sample rates of your files are:
ffmpeg -i sound2.mp3
You'll get an output like this:
Duration: 00:13:50.22, start: 0.011995, bitrate: 192 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, fltp, 192 kb/s
In this example, 44.1 kHz is the sample rate.
Channel Count
I'd expect your voice MP3s to be in mono, but it wouldn't hurt to check to be sure. As with above, check the output of FFmpeg. In my example above, it says stereo.
As with sample rate, technically each frame could specify its own channel count but I don't know of any player that will pull off switching channel count mid-stream. Therefore, if you're concatenating, you need to make sure all the channel counts are the same.
ID3 Tags
It's common for there to be ID3 metadata at the beginning (ID3v2) and/or end (ID3v1) of the file. It's less expected to have this data mid-stream. You would want to make sure this metadata is all stripped out before concatenating.
MP3 Bit Reservoir
MP3 frames don't necessarily stand alone. If you have a constant bitrate stream, the encoder may still use less data to encode one frame, and more data to encode another. When this happens, some frames contain data for other frames. That way, frames that could benefit from the extra bandwidth can get it while still fitting the whole stream within a constant bitrate. This is the "bit reservoir".
If you cut a stream and splice in another stream, you may split up a frame and its dependent frames. This typically causes an audio glitch, but may also cause the decoder to skip ahead. Some badly behaving decoders will just stop playing altogether. In your example, you're not cutting anything so this probably isn't the source of your trouble... but I mention it here because it's definitely relevant to the way you're working these streams.
See also: http://wiki.hydrogenaud.io/index.php?title=Bit_reservoir
Solutions
Pick a "normal" format, resample and rencode non-conforming files
If most of your sources are all the exact same format and only one or two outstanding, you could convert the non-conforming file. From there, strip ID3 tags from everything and concatenate away.
To do the conversion, I'd recommend kicking it over to FFmpeg as a child process.
child_process.spawn('ffmpeg' [
// Input
'-i', inputFile, // Use '-' to write to STDIN instead
// Set sample rate
'-ar', '44100',
// Set audio channel count
'-ac', '1',
// Audio bitrate... try to match others, but not as critical
'-b:a', '64k',
// Ensure we output an MP3
'-f', 'mp3',
// Output
outputFile // As with input, use '-' to write to STDOUT
]);
Best Solution: Let FFmpeg (or similar) do the work for you
The simplest, most robust solution to all of this is to let FFmpeg build a brand new stream for you. This will cause your audio files to be decoded to PCM, and a new stream made. You can add parameters to resample those inputs, and modify channel counts if needed. Then output one stream. Use the concat filter.
This way, you can accept audio files of any type, you don't have to write the code to hack those streams together, and once setup you won't have to worry about it.
The only downside is that it will require a re-encoding of everything, meaning another generation of quality lost. This would be required for any non-conforming files anyway, and it's just speech, so I wouldn't give it a second thought.
#Brad's answer was the solution! The first solution he suggested worked. It took some messing around getting FFMpeg to work correctly, but in the end using the fluent-ffmpeg library worked.
Each file in my case was stored on Google Cloud Storage, and not on the server's hard drive. This posed some problems for FFmpeg, as it requires file paths to have multiple files, or an input stream (but only one is supported, as there is only one STDIN).
One solution is to put the files on the hard drive temporarily, but this would not work for our use case as we may have a lot of use in this function and the hard drive adds latency.
So, instead we did as suggested and loaded each file into ffmpeg to convert it into a standardized format. This was a bit tricky, but in the end requesting each file as a stream, using that stream as an input for ffmpeg, then using fluent-ffmpeg's pipe() method (which returns a stream) as output worked.
We then bound an event listener to the 'data' event for this pipe, and pushed the data to an array (bufs.push(data)), and on stream 'end' we concatenated this array using Buffer.concat(bufs), followed by a promise resolve.
Then once all requests promises were resolved, we could be sure ffmpeg had processed each file, and then those buffers were concatenated in the required groups as before using Buffer.concat(), converted to base64 data, and sent to the client.
This works great, and now it seems to be able to handle every combination of files/sources I can throw at it!
In conclusion:
The answer to the question was that the mp3 data must have been encoded differently (different channels, sample rates, etc.), and loading it through ffmpeg and outputing it in a 'unified' way made the mp3 data compatible.
The solution was to process each file in ffmpeg separately, pipe the ffmpeg output into a buffer, then concatenate the buffers.
Thanks #Brad for your suggestions and detailed answer!
Is there a way to add lossless data to an AAC audio stream?
Essentially I am looking to be able to inject "this frame of audio should be played at XXX time" every n frames in.
If I use a lossless codec I suppose I could just inject my own header mid stream and that data would be intact as it needs to be the same on the way out just like gzip does not loose data.
Any ideas? I suppose I could encode the data into chunks of AAC on the server and on the network layer add a timestamp saying play the following chunk of AAC at time x but I'd prefer to figure a way to add it to the audio itself.
This is not really possible (short of writing your own specialized encoder), as AAC (and MP3) frames are not truly standalone.
There is a concept of the bit reservoir, where unused bandwidth from one frame can be utilized for a later frame that may need more bandwidth to store a more complicated sound. That is, data from frame 1 might be needed in frame 2 and/or 3. If you cut the stream between frames 1 and 2 and insert your alternative frames, the reference to the bit reservoir data is broken and you have damaged frame 2's ability to be decoded.
There are encoders that can work in a mode where the bit reservoir isn't used (at the cost of quality). If operating in this mode, you should be able to cut the stream more freely along frame boundaries.
Unfortunately, the best way to handle this is to do it in the time domain when dealing with your raw PCM samples. This gives you more control over the timing placement anyway, and ensures that your stream can also be used with other codecs.
I use ffmpeg to convert videos from one format to another.
Is bitrate the only parameter which decides the output size of a video/audio file?
Yes, bitrate is essentially what will control the file size (for a given playback duration). It is the number of bits used to represent each second of material.
However, there are some subtleties, e.g. :
a video file encoded at a certain video bitrate probably contains a separate audio stream, with a separately-specified bitrate
most file formats will contain some metadata that won't be counted towards the basic video stream bitrate
sometimes the algorithm will not actually aim to achieve the specified bitrate - for example, using the CRF factor. http://trac.ffmpeg.org/wiki/x264EncodingGuide explains how two-pass would be preferred if targeting a specific file size.
So you may want to do a little experimenting with a particular set of options for a particular file format.
Bitrate describes the quality of an audio or video file.
For example, an MP3 audio file that is compressed at 192 Kbps will have a greater dynamic range and may sound slightly more clear than the same audio file compressed at 128 Kbps. This is because more bits are used to represent the audio data for each second of playback.
Similarly, a video file that is compressed at 3000 Kbps will look better than the same file compressed at 1000 Kbps. Just like the quality of an image is measured in resolution, the quality of an audio or video file is measured by the bitrate.
When I start receiving the live audio (radio) stream (e.g. MP3 or AAC) I think the received data are not kind of raw bitstream (i.e. raw encoder output), but they are always wrapped into some container format. If this assumption is correct, then I guess I cannot start streaming from arbitrary place of the stream, but I have to wait to some sync byte. Is that right? Is it usual to have some sync bytes? Is there any header following the sync byte, from which I can guess the used codec, number of channels, sample rate, etc.?
When I connect to live stream, will I receive data starting by the nearest sync byte or I will get them from the actual position and I have to check for the sync byte first?
Some streams like icecast use headers in the HTTP response, where stream related information are included, but i think i can skip them and deal directly with the steam format.
Is that correct?
Regards,
STeN
When you look at SHOUTcast/Icecast, the data that comes across is pure MPEG Layer III audio data, and nothing more. (Provided you haven't requested metadata.)
It can be cut at an arbitrary place, so you need to sync to the stream. This is usually done by finding a potential header, and using the data in that header to find sequential headers. Once you have found a few frame headers, you can safely assume you have synced up to the stream and start decoding for playback.
Again, there is no "container format" for these. It's just raw data.
Now, if you want metadata, you have to request it from the server. The data is then just injected into the stream every x number of bytes. See http://www.smackfu.com/stuff/programming/shoutcast.html.
Doom9 has great starting info about both mpeg and aac frame formats. Shoutcast will add some 'metadata' now and then, and it's really trivial. The thing I want to share with you is this; I have an application that can capture all kind of streams, and shoutcast, both aac and mp3 is among them. First versions had their files cut at arbitrary point according to the time, for example every 5 minutes, regardless of the mp3/aac frames. It was somehow OK for the mp3 (the files were playable) but was very bad for aacplus.
The thing is - aacplus decoder ISN'T that forgiving about wrong data, and I had everything from access violations to mysterious software shutdowns with no errors of any kind.
Anyway, if you want to capture stream, open the socket to the server, read the response, you'll have some header there, then use that info to strip metadata that will be injected now and then. Use the header information for both aacplus and mp3 to determine frame boundaries, and try to honor them and split the file at the right place.
mp3 frame header:
http://www.mp3-tech.org/programmer/frame_header.html
aacplus frame header:
http://wiki.multimedia.cx/index.php?title=Understanding_AAC
also this:
aacplus frame alignment problems
Unfortunately it's not always that easy, check the format and notes here:
MPEG frame header format
I will continue the discussion byu answering myself (even we are discouraged to do that):
I was also looking into streamed data and I have found that frequently the sequence ff f3 82 70 is repeated - this I suggest is the MPEG frame header, so I try to look what that means:
ff f3 82 70 (hex) = 11111111 11110011 10000010 01110000 (bin)
Analysis
11111111111 | SYNC
10 | MPEG version 2
01 | Layer III
1 | No CRC
1000 | 64 kbps
00 | 22050Hz
1 | Padding
0 | Private
01 | Joint stereo
11 | ...
Any comments to that?
When starting receiving the streaming data, should I discard all data prior this header before giving the buffer to the class which deals with the DSP? I know this can be implementation specific, but I would like to know what are in general the proceedings here...
BR
STeN