On iOS 7, how do I get the current microphone input volume in a range between 0 and 1?
I've seen several approaches like this one, but the results I get baffle me.
The return values of peakPowerForChannel: are documented to be in the range of -160 to 0 with 0 being the loudest and -160 near absolute silence.
Problem: Given a quite room and a short but loud noise, the power goes all the way up in an instant but takes very long time to drop back to quite level (way longer than the actual noise...)
What I want: Essentially I want an exact copy of the Audio Input patch of Quartz Composer with its Volume Peak output. Any tips?
To get a similar volume peak measurement, you might have to input raw audio via the iOS Audio Queue API (or the RemoteIO Audio Unit), and analyze the raw PCM waveform samples in each audio callback, looking for a magnitude maxima over your desired frame width or analysis time.
Related
I am trying to implement an SLM app for iOS using AudioKit. Therefore I need to determine different loudness values to a) display the current loudness (averaged over a second) and b) do further calculations (e.g. to calculate the "Equivalent Continuous Sound Level" over a longer time span). The app should be able to track frequency-weighted decibel values like dB(A) and dB(C).
I do understand that some of the issues im facing are related to my general lack of understanding in the field of signal and audio processing. My question is how one would approach this task with AudioKit. I will describe my current process and would like to get some input:
Create an instance of AKMicrophone and a AKFrequencyTracker on this microphone
Create a Timer instance with some interval (currently 1/48_000.0)
Inside the timer: retrieve the amplitude and frequency. Calculate a decibel value from the amplitude with 20 * log10(amplitude) + calibrationOffset (calibration offset will be determined per device model with the help of a professional SLM). Calculate offsets for the retrieved frequency according to frequency-weighting (A and C) and apply these to the initial dB value. Store dB, dB(A) and dB(C) values in an array.
Calculate the average for arrays over the give timeframe (1 second).
I read somewhere else that using a Timer this is not the best approach. What else is there that I could use for the "sampling"? What exactly is the frequency of AKFrequencyTracker? Will this frequency be sufficient to determine dB(A) and dB(C) values or will I need an AKFFTTap for this? How are values retrieved from the AKFrequencyTracker averaged, i.e. what time frame is used for the RMS?
Possibly related questions: Get dB(a) level from AudioKit in swift, AudioKit FFT conversion to dB?
I'm trying to demodulate a signal using GNU Radio Companion. The signal is FSK (Frequency-shift keying), with mark and space frequencies at 1200 and 2200 Hz, respectively.
The data in the signal text data generated by a device called GeoStamp Audio. The device generates audio of GPS data fed into it in real time, and it can also decode that audio. I have the decoded text version of the audio for reference.
I have set up a flow graph in GNU Radio (see below), and it runs without error, but with all the variations I've tried, I still can't get the data.
The output of the flow graph should be binary (1s and 0s) that I can later convert to normal text, right?
Is it correct to feed in a wav audio file the way I am?
How can I recover the data from the demodulated signal -- am I missing something in my flow graph?
This is a FFT plot of the wav audio file before demodulation:
This is the result of the scope sink after demodulation (maybe looks promising?):
UPDATE (August 2, 2016): I'm still working on this problem (occasionally), and unfortunately still cannot retrieve the data. The result is a promising-looking string of 1's and 0's, but nothing intelligible.
If anyone has suggestions for figuring out the settings on the Polyphase Clock Sync or Clock Recovery MM blocks, or the gain on the Quad Demod block, I would greatly appreciate it.
Here is one version of an updated flow graph based on Marcus's answer (also trying other versions with polyphase clock recovery):
However, I'm still unable to recover data that makes any sense. The result is a long string of 1's and 0's, but not the right ones. I've tried tweaking nearly all the settings in all the blocks. I thought maybe the clock recovery was off, but I've tried a wide range of values with no improvement.
So, at first sight, my approach here would look something like:
What happens here is that we take the input, shift it in frequency domain so that mark and space are at +-500 Hz, and then use quadrature demod.
"Logically", we can then just make a "sign decision". I'll share the configuration of the Xlating FIR here:
Notice that the signal is first shifted so that the center frequency (middle between 2200 and 1200 Hz) ends up at 0Hz, and then filtered by a low pass (gain = 1.0, Stopband starts at 1 kHz, Passband ends at 1 kHz - 400 Hz = 600 Hz). At this point, the actual bandwidth that's still present in the signal is much lower than the sample rate, so you might also just downsample without losses (set decimation to something higher, e.g. 16), but for the sake of analysis, we won't do that.
The time sink should now show better values. Have a look at the edges; they are probably not extremely steep. For clock sync I'd hence recommend to just go and try the polyphase clock recovery instead of Müller & Mueller; chosing about any "somewhat round" pulse shape could work.
For fun and giggles, I clicked together a quick demo demod (GRC here):
which shows:
I am looking for a low-latency way of finding out how many seconds have elapsed in an audio file to guaranteed millisecond precision in real-time. According to the AVAudioPlayer class reference, a call to -currentTime will return "the offset of the current playback position, measured in seconds from the start of the sound", however an NSTimeInterval is a double and this implies fractions of a second are possible.
As a testing scenario, I have an audio file playing and the user taps a button. Playback DOES NOT pause/stop, but at the moment the button was tapped I would like to obtain information about the elapsed time. In the real application, the "button may be pressed" many times in one second, hence the need for millisecond precision.
My files are stored as AIFF files and are around 1-10 minutes in length. Ideally I would like to find out exactly which sample frame is 'up-next' when playback resumes - however, this level of precision is a little excessive and millisecond precision is perfectly acceptable.
Is AVAudioPlayer's -currentTime method sufficient to achieve guaranteed millisecond precision for a currently-playing audio file? Or, would it be preferable to use a lower-level API such as iOS's Audio Units?
If you want sub-millisecond relative time resolution, convert to raw PCM and count buffers * length + samples using a low latency RemoteIO Audio Unit configuration. Most iOS devices will support as small as 6 mS RemoteIO buffers of 256 samples, with a callback for each buffer.
This one keeps me awake:
I have an OS X audio application which has to react if the user changes the current sample rate of the device.
To do this I register a callback for both in- and output devices on ‘kAudioDevicePropertyNominalSampleRate’.
So if one of the devices sample rates get changed I get the callback and set the new sample rate on the devices with 'AudioObjectSetPropertyData' and 'kAudioDevicePropertyNominalSampleRate' as the selector.
The next steps were mentioned on the apple mailing list and i followed them:
stop the input AudioUnit and the AUGraph which consists of a mixer and the output AudioUnit
uninitalize them both.
check for the node count, step over them and use AUGraphDisconnectNodeInput to disconnect the mixer from the output
now set the new sample rate on the output scope of the input unit
and on the in- and output scope on the mixer unit
reconnect the mixer node to the output unit
update the graph
init input and graph
start input and graph
Render and Output callbacks start again but now the audio is distorted. I believe it's the input render callback which is responsible for the signal but I'm not sure.
What did I forget?
The sample rate doesn't affect the buffer size as far as i know.
If I start my application with the other sample rate everything is OK, it's the change that leads to the distorted signal.
I look at the stream format (kAudioUnitProperty_StreamFormat) before and after. Everything stays the same except the sample rate which of course changes to the new value.
As I said I think it's the input render callback which needs to be changed. Do I have to notify the callback that more samples are needed? I checked the callbacks and buffer sizes with 44k and 48k and nothing was different.
I wrote a small test application so if you want me to provide code, I can show you.
Edit: I recorded the distorted audio(a sine) and looked at it in Audacity.
What I found was that after every 495 samples the audio drops for another 17 samples.
I think you see where this is going: 495 samples + 17 samples = 512 samples. Which is the buffer size of my devices.
But I still don't know what I can do with this finding.
I checked my Input and Output render procs and their access of the RingBuffer(I'm using the fixed Version of CARingBuffer)
Both store and fetch 512 frames so nothing is missing here...
Got it!
After disconnecting the Graph it seems to be necessary to tell both devices the new sample rate.
I already did this before the callback but it seems this has to be done at a later time.
Is there an API in one of the iOS layers that I can use to generate a tone by just specifying its Hertz. What I´m looking to do is generate a DTMF tone. This link explains how DTMF tones consists of 2 tones:
http://en.wikipedia.org/wiki/Telephone_keypad
Which basically means that I should need playback of 2 tones at the same time...
So, does something like this exist:
SomeCleverPlayerAPI(697, 1336);
If spent the whole morning searching for this, and have found a number of ways to playback a sound file, but nothing on how to generate a specific tone. Does anyone know, please...
Check out the AU (AudioUnit) API. It's pretty low-level, but it can do what you want. A good intro (that probably already gives you what you need) can be found here:
http://cocoawithlove.com/2010/10/ios-tone-generator-introduction-to.html
There is no iOS API to do this audio synthesis for you.
But you can use the Audio Queue or Audio Unit RemoteIO APIs to play raw audio samples, generate an array of samples of 2 sine waves summed (say 44100 samples for 1 seconds worth), and then copy the results in the audio callback (1024 samples, or whatever the callback requests, at a time).
See Apple's aurioTouch and SpeakHere sample apps for how to use these audio APIs.
The samples can be generated by something as simple as:
sample[i] = (short int)(v1*sinf(2.0*pi*i*f1/sr) + v2*sinf(2.0*pi*i*f2/sr));
where sr is the sample rate, f1 and f1 are the 2 frequencies, and v1 + v2 sum to less than 32767.0. You can add rounding or noise dithering to this for cleaner results.
Beware of clicking if your generated waveforms don't taper to zero at the ends.