I'm learning about Speech Processing, and I want to know whether or not we can use Companding in DPCM? Based on what I've learned so far, Companding is used in PCM for compressing and then expanding an analog or digital signal, but I haven't seen anything about it being used in DPCM. Why is that?
Thank you
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I want to write an application which takes a video stream and in the cycle do processing frames and fragments of audio.
I saw such loop here:
I saw on the forums some solution with pipes here
Is it any library wrapper on ffmpeg/avlib which I could just write my callback without tones of spaghetti around like here
I would look into using Opencv for this. It takes care of a lot of the problems with video streams for you and allows processing on the individual frames.
Here simple and easy to follow examples:
https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples
I'm assuming you have experienced C++ programmer so these examples will give you a good start (btw, examples in C).
I created a book app and used Cocos2D and physics engine (Chipmunk) to create it. I would like to add "read aloud" feature to it.
So far I found instructions/books and tutorials how to add read aloud feature when book is created with iBook Author (but I couldn't use iBook Author due to some limitations) using Epub3 and SMIL.
I also found a good tutorial from J. Shapiro how to make narrated book using AVSpeechSynthesizer. This helps, only that I would like to use recorded voice, rather than synthesized sound. I don't know if this approach can be modified to do so?
I also know how it can be done in Sprite Kit framework.
The only info that I couldn't find is how to add "read aloud" feature to the app written using Cocos2D. Could it be done within SimpleAudioEngine, or it can be combined with some other engine (possibly from Sprite Kit framework)?
I would appreciate very much if somebody can give me some references/pointers or tutorial links where to look for some answers how to add this feature.
Thanking you in advance.
I would like to use recorded voice, rather than synthesized sound
Good. Add your voice recording audio files (caf, wav or mp3 format) to the project. Play it back at the appropriate time using:
[[SimpleAudioEngine sharedEngine] playEffect:#"someVoiceRecordingFile.wav"];
Define what read aloud means to you because I find that a lot of terms, especially semi-vague ones like this, are used differently depending on who is using it.
When you say read aloud book do you essentially mean a digital storybook that reads the story to you by simply playing narration audio? I've created dozens of these and what you are asking has multiple steps depending on what features you are going for in your book. If you mean simply playing audio and that is it, then yes you could do that in cocos2d using SimpleAudioEngine (as one option) but I assume you already knew that which is why this question has a tab bit of vagueness to it. Either way you probably wouldn't want to play narration as an effect but rather stream it. To do that along with background music you'd stream background music via the left channel and narration via the right. You can easily add a method to SimpleAudioEngine to make this nice and neat. To get you started something similar to this can be used to access the right channel:
CDLongAudioSource* sound = [[CDAudioManager sharedManager] audioSourceForChannel:kASC_Right];
if ([sound isPlaying])
{
[sound stop];
}
[sound load:fileName];
Also use the proper settings and recommended formats for streaming audio such as aifc (or really all audio in general). Although I believe you can stream mp3 without it being decompressed first, the problem is with timing. If you are using highlighted text or looping audio then aifc is the better option. Personally I've never had a reason to use mp3. Wav with narration is something I'd avoid even if just for the file size increase. If the mp3 is decompressed even for streaming (which I'm not sure if it is off the top of my head) then you'd have a huge spike in memory that will be both highly unwanted and at times down right bad.
There are many other things that can go into it but those are the basic first steps. If you want to do things like highlighted text, per-word animations, etc then that will take more work of course and you'd need to be comfortable with cocos2d, SpriteKit, or whatever you decide to use. I'll be doing a tutorial series on it one day soon so I'll cover all of that stuff.
On the other hand, if you are talking about recording someone's voice and having it playback i.e. a mother recording herself reading the story so her child can hear her voice whenever they are using your app, then you'd simply record the audio like you would any other piece of audio, save it to the device, and play it back when the page is displayed in the proper reading mode (or whatever you personally call it). One place to look is the AVAudioRecorder that is part of the AVFoundation framework. Simply Google "iOS audio recording" for examples if you need them.
I am working on "de-multiplexing a midi file", which means extracting all the the instruments played on channel at any moment in the song. I thought that the solution could be extracting the instrument played on one note at the time and store it in a data structure. I am supposed to use C++.
Does anyone know a good reference to start with? I have no idea about how to manipulate midi files using C++; I just read some good references about midi structure but no more.
You can take a look at JUCE. It has a lot of MIDI functions that may help you figure out what you need to know. You can also look here for a description of the MIDI file format.
I am currently creating a game. My game will use music from an mp3 file that the user sends in in order to make decisions on where to place things, how fast the level moves, etc. I am fairly new at this, I have been reading information about mp3. Currently I have found all the frames in the mp3 file that I am using. I don't really know where to go from here. What I want to do is measure the frequencies of the sound wave of the music at certain times (like every sec) and then based on that frequency, do what I need to for the game. I don't know whether I should decode the mp3, that looks like a lot of work and I don't want to do that if I don't have 2 or if I can just read the bytes in the frame and convert them without decoding anything. I am developing this in c#, using the game engine FlatRedBall. I am not using any libraries. I am also planning on selling this game so I would like to avoid using other people's code if I can avoid it. Please someone help me, I just need a direction to go from here. I know how to parse the header and calculate the framelength, I just don't know the next step in what I want to do...
Convert your music to .ogg format which is free and use free library to play it.
Note: I was going to post this as a comment but it quickly grew too big. :)
Writing your own MP3 enconder/decoder is probably going to take a good ammount of effort; effort which would probably be better spent on your game itself. Therefore, is possible, I would be all means try to use an open source library.
That said, most good MP3 libraries are LGPL/GPL licensed. This means you can use it in a commercial setting, as long as you dynamically link to it. Also the SDL Mixer library, as of version 1.2.12, supports MP3s and is under a more permissive zlib license, but since you mention C# I don't know if stable and up-to-date bindings are available. Also since your project isn't written in SDL to begin with, it might be hard to integrate it.
Also, as #pro_metedor hinted, perhaps using a more open format could help in licensing issues. In general, OGG achieves better compression than MP3, which is a plus for things like download size, bandwidth/resource usage, etc.
Just shop around for a while, and try to be a little flexible. I'm sure you'll find something nice! :)
This question has been in my mind for a few years and I never actually found the answer for this.
What I would like to do is extract the actual waveform/PCM of an MP3 file, so that I can play it using the soundcard (of course).
Ideally I would be experimenting some DSP effects.
My first step was to look into LAME, but I didn't find anything relevant about MP3 decoding in a program or stuff like that.
So I'm asking where I could find something like this.
What language should I use? I was thinking C, but maybe there are programming languages out there that would do the job more efficiently.
Thanks!
Guillaume.
The question boils down to: what are you trying to accomplish?
From the description of your question of decoding an MP3 and playing it on the sound card makes it sounds as if you are trying to make a media player.
However, if your intent is to play around with DSP effects, then it sounds like the question is more about processing the sound rather than decoding MP3s. if that's the case, probably looking into writing plug-ins for existing media players (such as Windows Media Player and Winamp) would be easiest path to what you're trying to accomplish.
Frankly, learning to write your own decoder from scratch is not just a programming problem but a mathematical one, so using existing libraries are the way to go. Talking to the operating system or libraries like DirectSound to output audio seems like unnecessary work if anything. I feel that working on plug-ins for existing players would be the way to go, unless your goal is to make your own media player.
If what you really want to accomplish is playing with audio data, then probably decoding an MP3 to uncompressed PCM using any MP3 decoder, then manipulating it in the language of your choice would accomplish your goal of dealing with effects with sound.
The language choice is going to depend on whether you are going to interact directly with MP3 decoding libraries, or whether you can just use raw audio input, which would allow you to use pretty much any language of your choice.
There was a similar question a while back, Getting started with programmatic audio, where I posted an answer on some basic ways to manipulate audio, such as amplification, changing playback speed, and doing some work with FFT.
libmpg123 should do the trick.
I have been using the Windows Media SDK, not for this purpose, but I am pretty sure there are hooks let that let you intercept the audio stream, or convert MP4 to uncompressed WAV. I used C++.
Lots:
http://www.mp3-tech.org/programmer/decoding.html
Pick your poison...
Also, LAME does decode MP3s (check out --decode option), so you might find something interesting in that source.
-Adam
It really depends what platform you are programming on and what you want to do with the code. If you are on Windows you should look at the windows media format sdk or DirectShow. They should both have the ability to decode mp3 files into the raw waveform. On the Mac, I would expect Quicktime to have this same ability. Others have already suggested source for Linux/open source code.
I would recommend looking at Cubase and Wavelab as both will convert MP3 to WAV etc and allow you to play around with the waveform