I am looking at a project where we send information from a peripheral device to an iPhone through the microphone input.
In a nutshell, the iPhone would act as a voltmeter. (In reality, the controller we developed will send data encoded as voltages to the iPhone for further processing).
As there are several voltmeter apps on the AppStore that receive their input through the microphone port, this seems to be technically possible.
However, scanning the AudioQueue and AudioFile APIs, there doesn't seem to be a method for directly accessing the voltage.
Does anyone know of APIs, sample code or literature that would allow me to access the voltage information?
The A/D converter on the line-in is a voltmeter, for all practical purposes. The sample values map to the voltage applied at the time the sample was taken. You'll need to do your own testing to figure out what voltages correspond to the various sample values.
As far as I know, it won't be possible to get the voltages directly; you'll have to figure out how to convert them to equivalent 'sounds' such that the iOS APIs will pick them up as sounds, which you can interpret as voltages in your app.
If I were attempting this sort of app, I would hook up some test voltages to the input (very small ones!), capture the sound and then see what it looks like. A bit of reverse engineering should get you to the point where you can interpret the inputs correctly.
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Iam using Unity SDK provided for IBM Watson services. I try to use 'ExampleStreaming.cs' sample provided for speech to text recognition. I test the app in unity editor.
This sample uses Microphone as audio input and gets results for voice input from the user. However, when I use microphone as input, the transcribed results are far from being correct. When I say "Create a black box", the results are inappropriate, with the word results being completely irrelevant to input.
When I use pre-recorded voice clips, the output is perfect.
Does the service perform incorrectly for Indian accent?.
What is the reason for poor microphone input recognition?
The docs say:
"In general, the service is sensitive to background noise. For instance, engine noise, working devices, street noise, and talking can significantly reduce accuracy. In addition, the microphones that are typically installed on mobile devices and tablets are often inadequate. The service performs best when professional microphones are used to capture audio with better quality."
I use Logitech headset mic as input source.
Satish,
Try to "clean up" the audio as best you can - by limiting background noise. Also be aware that you can use one of two different processing models - one for broadband and one for narrowband. Try them both, and see which is most appropriate for your input device.
In addition, you can find that the underlying speech model does not handle all of the domain specific terms that you might be looking for. In these cases you can customize and expand the speech model, as explained in the documentation on Using Custom Language Models (https://console.bluemix.net/docs/services/speech-to-text/custom.html#custom). While this is a bit more involved, it can often make a huge difference in accuracy and overall usability.
I have a series of sounds that a user will play, rearrange, and edit etc. while using my app. When the user is finished, I want them to be able to save their work and record it to an mp3.
I don't want to play it through speakers and record it with the mic since that will result in low sound quality and interference. I cannot think of any ways of doing this that doesn't require extra hardware and/or a computer.
How can I do this using just their device?
Well, I would say it cant be done with AVFoundation.
My suggestion is to use Audio Units, and transform all your interactions to an audio graph. at some point you set a render notify on the RemoteIO so every time it renders sounds to the speakers you get a callback where you can write it down those frames/packets/data into a file.
I will probably suggest to use AAC(m4a) over MP3. I am not very fond of MP3, and to be honest as far as I know the sdk does not provide encoding to MP3, probably due to licensing issues. I could be wrong though. Check this sample code below, probably the best sample code you will ever find on Audio units on the web.
AudioGraph by Tom Zic
Some days ago I saw a interesting device for iphone, square, here: https://squareup.com/
you can plug it into iphone's earphone socket, and it can transfer data to iphone. A running App on iphone can receive it.
does any one know how it implemented? I guess it can encode data to audio stream and "sing" it, and App on phone can record the sound and decode it. but how to? is there a protocol or SDK?
The implemention is likely to be no different to that of a simple acoustic modem. The relevant APIs include Audio Units (low-level) or Audio Queue Services (higher level).
Matt Gallagher has written an excellent (as always!) post on creating an iOS tone generator, which is one way of enabling what you are after.
I am new to Core Audio and really lost, I am trying to record an audio and then apply voice modulation to that recording and play it back. I have looked at the example Speak Here which uses Audio Queue for audio recording. I am stuck at the part of how to change the audio samples. I understand that it can be done using Audio Unit in the call back function to change the audio samples, but I have no idea what to apply to those samples to change them (will changing pitch help ?).
If you could direct me to some source code or tutorial or any site that explains voice modulation for objective C will really really help me. Thank you all in advance.
What you are trying to do here is not that simple. Basically, you would have to implement a vocoder ("voice-coder") to change a voice. The Wikipedia links should help you there.
Then, you still have manipulate those samples in CoreAudio. You can do this using Audio Queue Services but that not exactly an easy-to-use API. It might actually be less trouble to use one of the simpler CoreAudio APIs and wrap your vocoder in an Audio Unit.
Do you have some experience with audio processing? Implementing a vocoder without some prior knowledge about audio processing in general is a tough task.
First, to actually answer your question: When you called the AudioQueueNewInput() function, you pass it the name of a routine that will be called every time data is available to you. You probably called it MyInputBufferHandler() or something. It's third argument is an AudioQueueBufferRef which hold the incoming data.
Be aware that this is not as simple as looking at each sample (amplitude) and lowering or raising it. You receive samples in the temporal (time) domain as amplitudes. There is no pitch or frequency information available. What you need to do is move the incoming samples (waveform) into the frequency domain, wherein each "point" in that space is a frequency and it's accompanying power and phase. You can do that with an FFT (fast Fourier transform) but the mathematics are somewhat sophisticated. Apple does provide FFT routines in the Acceleration framework, but be aware that you are wading into very deep water here.
I'm wondering if there are any examples atomic examples out there for streaming audio FROM the iPhone to a server. I'm not interested in telephony or SIP style solutions, just a simple socket stream to send an audio clip, in .wav format, as it is being recorded. I haven't had much luck with the google or other obvious avenues, although there seem to be many examples of doing this the other way around.
i cant figure out how to register the unregistered account i initially posted with.
anyway, I'm not really interested in the audio format at present, just the streaming aspect. i want to take the microphone input, and stream it from the iphone to a server. i dont presently care about the transfer rate as ill initially just test from a wifi connection, not the 3g setup. the reason i cant cache it is because im interested in trying out some open source speech recognition stuffs for my undergraduate thesis. caching and then sending the recording is possible but then it takes considerably longer to get the voice data to the server. if i can start sending the data as soon as i start recording, then the response time is considerably improved because most of the data will have already reached the server by the time i let go of the record button. furthermore, if i can get this streaming functionality to work from the iphone then on the server side of things i can also start the speech recognizer as soon as the first bit of audio comes through. again this should considerably speech up the final amount of time that the transaction takes from the user perspective.
colin barrett mentions the phones and phone networks, but these are actually a pretty suboptimal solution for asr, mainly because they provide no good way to recover from errors - doing so over a voip dialogue is a horrible experience. however, the iphone and in particular the touch screen provide a great way to do that, through use of an ime or nbest lists for the other recognition candidates.
if i can figure out the basic architecture for streaming the audio, then i can start thinking about doing flac encoding or something to reduce the required transfer rate. maybe even feature extraction, although that limits the later ability to retrain the system with the recordings.