Currently i am trying to create a game and having a radio which you must tune. I was experimenting using Csound and Frequency modulation and while it is possible using oscoil opcode however It does not appear to allow you to modify a Sound File from the DiskIn opcode.
Is there another opcode that may allow modification to an audio diskin File?
Could you explain what you mean about "modifying a sound file from the diskin opcode"? I know you can modulate the playback rate at k-rate but I'm not sure that's what you're looking tod o.
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My program launchs an arbitrary number of sounds streamed from FreeSound.org and plays them using javafx.scene.media AudioClip instances.
I was trying to figure out whether it could be possible to capture the generated output to disk, from within the same program? Any pointers?
Instead of using AudioClip you could use SourceDataLine for playback. This class allows you to progressively read the audio data, exposing it for handling. You would have to decode from bytes to PCM for each incoming line, then add the PCM from all the lines to be merged, and recode that back to bytes for your output.
I suspect with a little minor tweaking you could get the library I wrote AudioCue to work for you. It has an optional mixer that will handle multiple cue inputs. The inputs make use of SourceDataLine and the mixer uses the logic I described. You would have to tweak the code to output to disk. I could probably help with that if this project is still live for you.
I am going to read the PC's music output and getting basic information (beat/tone/..) of the song played (then flash the lights accordingly etc). Can NAudio be used for the purpose and any samples? Sorry for this too general question at the moment.
TIA
-d
You can access your PC's output using WasapiLoopbackCapture. However, NAudio does not include a beat detection algorithm, so you'd need to find one yourself. There is a FFT class though which could be used to determine frequencies present.
I've been looking at the NAudio demo application "Audio file playback". What I'm missing from this demo is a way to get hold of the samples while the audio file is being played.
I figured that it would somehow be possible to fill a BufferedWaveProvider with samples using a callback whenever new samples are needed, but I can't figure out how.
My other (non-preferred) idea is to make a special version of e.g. DirectSoundOut where I can get hold of the samples before they are written to the sound card.
Any ideas?
With audio file playback in NAudio you construct an audio pipeline, starting with your audio file and going through various transformations (e.g. changing volume) along the way before ending up at your output device. The NAudioDemo does in fact show how the samples can be accessed along the way by drawing a waveform (pre-volume adjustment) and by showing a volume meter (post-volume adjustment).
You could, for example, create an implementer of IWaveProvider or ISampleProvider and insert it into the pipeline. Then, in the Read method, you read from your source, and then you can process or examine or write to disk the samples before passing them on to the next stage in the pipeline. Look at the AudioPlaybackPanel.CreateInputStream to see how this is done in the demo.
I am trying to record audio from a microphone/iSight camera from Mac to a NSData object.
I have tried to do it using QTKit, but I found out that you could only save it as a .mov file.
But the fact is that I want to recode the audio into a FLAC file. Is that posible, or I'll need to use another framework?.
Thanks.
Grab the source for VLC (if you can deal w/GPL -- it has limitations on use that many find onerous) and have a read. It does transcoding, amongst other things.
Beyond that, one dead simple approach is to save as AIFF and then use a command line tool (via NSTask) to do the conversion.
Or you could just go with Apple Lossless -- it is open source these days.
Of course, this also begs the question; why do you need lossless compression when recording voice [low bandwidth in the first place] via a relatively sub-par microphone?
I know how to play mp3 files and whatnot in Xcode iOS. But how do I play a certain frequency, like if I just wanted to emit a C# note for 25 seconds; how might I do that? (The synth isn't as important to me as just the pitch of the note.)
You need to generate the PCM audio waveform that corresponds to the note you want to play and store that into a sample buffer in memory. Then you send that buffer to the audio hardware.
Here is a tutorial on generating waveforms of several types. The article goes into some details on the many aspects to a note you need to consider, including the frequency, volume, waveform shape, sampling rate, etc. The article comes with Flash source code, I think you should have no problem taking the concepts and adapting them to iOS.
If you also need a library that you can use to play the generated buffers on iOS, then I recommend the open source Finch.
I hope this helps!
You can synthesize waveforms of your desired frequency and feed them to the callbacks of either the Audio Queue or the RemoteIO Audio Unit API.
Here is a short tutorial on some of the code needed to create sine wave tones for iOS in C.