Given an audio file of duration say 10s, how do I find out the no of samples betweeen 2s and 8s?
if it's LPCM (e.g. not compressed), then use the sample rate.
in pseudocode:
double sampleRate = audioFile.getSampleRate();
// you may also need to account for channel count here
size_t sampleCount = sampleRate * (8-2);
-- where (8-2) represents "betweeen 2s and 8s"
You can use the duration property of the audio player for that purpose..
The duration property gives the duration of the current object of audio player(i.e the file that is currently being played)
So then you can supply all the audio files to the player one by one and check their durations and perform the required operations...
Cheers
Related
İ need to record pcmaudio per 10 milisecond without playback in swift.
I have tried this code but i can't find how can i stop playback while recording.
RecordAudio Github Repo
and second question: How can i get PCM data from circular buffer for encode-decode process properly. When I convert recorded audio data to signed byte or unsigned byte or anything else the converted data sometimes will corrupt. What is the best practice for this kind of process
In the RecordAudio sample code, the audio format is specified as Float (32-bit floats). When doing a float to integer conversion, you have to make sure your scale and offset results in a value in legal range for the destination type. e.g. check that -1.0 to 1.0 results in 0 to 256 (unsigned byte), and out-of-range values are clipped to legal values. Also pay attention to the number of samples you convert, as an Audio Unit callback can vary the frameCount sent (number of samples returned). You most likely won't get exactly 10 mS in any single RemoteIO callback, but may have to observe a circular buffer filled by multiple callbacks, or a larger buffer that you will have to split.
When RemoteIO is running in play-and-record mode, you can usually silence playback by zeroing the bufferList buffers (after copying, analyzing, or otherwise using the data in the buffers) before returning from the Audio Unit callback.
I'm trying to make an accurate timer to analyze an input. I'd like to be able to measure 1% deviation in signals of ~200ms.
My understanding is that using an AudioUnit will be able to get <1ms.
I tried implementing the code from Stefan Popp's example
After updating a few things to get it to work on xcode 6.3, I have the example working, however:
While I do eventually want to capture audio, I thought there should be some way to get a notification, like NSTimer, so I tried an AudioUnitAddRenderNotify, but it does exactly what it says it should - i.e it's tied to the render, not just an arbitrary timer. Is there some way to get a callback triggered without having to record or play?
When I examine mSampleTime, I find that the interval between slices does match the inNumberFrames - 512 - which works out to 11.6ms. I see the same interval for both record and play. I need more resolution than that.
I tried playing with kAudioSessionProperty_PreferredHardwareIOBufferDuration but all the examples I could find use the deprecated AudioSessions, so I tried to convert to AudioUnits:
Float32 preferredBufferSize = .001; // in seconds
status = AudioUnitSetProperty(audioUnit, kAudioSessionProperty_PreferredHardwareIOBufferDuration, kAudioUnitScope_Output, kOutputBus, &preferredBufferSize, sizeof(preferredBufferSize));
But I get OSStatus -10879, kAudioUnitErr_InvalidProperty.
Then I tried kAudioUnitProperty_MaximumFramesPerSlice with values of 128 and 256, but inNumberFrames is always 512.
UInt32 maxFrames = 128;
status = AudioUnitSetProperty(audioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global, 0, &maxFrames, sizeof(maxFrames));
[EDIT]
I am trying to compare the timing of an input (user's choice of MIDI or microphone) to when it should be. Specifically, is the instrument being played before or after the beat/metronome and by how much? This is for musicians, not a game, so precision is expected.
[EDIT]
The answers seem re-active to events. i.e. They let me precisely see when something happened, however I don't see how I do something accurately. My fault for not being clear. My app needs to be the metronome as well - synchronize playing a click on the beat and flash a dot on the beat - then I can analyze the user's action to compare timing. But if I can't play the beat accurately, the rest falls apart. Maybe I'm supposed to record audio - even if I don't want it - just to get inTimeStamp from the callback?
[EDIT]
Currently my metronome is:
- (void) setupAudio
{
AVAudioPlayer *audioPlayer;
NSString *path = [NSString stringWithFormat:#"%#/click.mp3", [[NSBundle mainBundle] resourcePath]];
NSURL *soundUrl = [NSURL fileURLWithPath:path];
audioPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL:soundUrl error:nil];
[audioPlayer prepareToPlay];
CADisplayLink *syncTimer;
syncTimer = [CADisplayLink displayLinkWithTarget:self selector:#selector(syncFired:)];
syncTimer.frameInterval = 30;
[syncTimer addToRunLoop:[NSRunLoop mainRunLoop] forMode:NSDefaultRunLoopMode];
}
-(void)syncFired:(CADisplayLink *)displayLink
{
[audioPlayer play];
}
You should be using a circular buffer, and performing your analysis on the signal in chunks that match your desired frame count on your own timer. To do this you set up a render callback, then feed your circular buffer the input audio in the callback. Then you set up your own timer which will pull from the tail of the buffer and do your analysis. This way you could be feeding the buffer 1024 frames every 0.23 seconds, and your analysis timer could fire maybe every 0.000725 seconds and analyze 32 samples. Here is a related question about circular buffers.
EDIT
To get precision timing using a ring buffer, you could also store the timestamp corresponding to the audio buffer. I use TPCircularBuffer for doing just that. TPCircularBufferPrepareEmptyAudioBufferList, TPCircularBufferProduceAudioBufferList, and TPCircularBufferNextBufferList will copy and retrieve the audio buffer and timestamp to and from a ring buffer. Then when you are doing your analysis, there will be a timestamp corresponding to each buffer, eliminating the need to do all of your work in the render thread, and allowing you to pick and choose your analysis window.
If you are using something like cross-correlation and/or a peak detector to find a matched sample vector within an audio sample buffer (or a ring buffer containing samples), then you should be able to count samples between sharp events to within one sample (1/44100 or 0.0226757 milliseconds at a 44.1k Hz sample rate), plus or minus some time estimation error. For events more than one Audio Unit buffer apart, you can sum and add the number of samples within the intervening buffers to get a more precise time interval than just using (much coarser) buffer timing.
However, note that there is a latency or delay between every sample buffer and speaker audio going out, as well as between microphone sound reception and buffer callbacks. That has to be measured, as in you can measure the round trip time between sending a sample buffer out, and when the input buffer autocorrelation estimation function gets it back. This is how long it takes the hardware to buffer, convert (analog to digital and vice versa) and pass the data. That latency might be around the area of 2 to 6 times 5.8 milliseconds, using appropriate Audio Session settings, but might be different for different iOS devices.
Yes, the most accurate way to measure audio is to capture the audio and look at the data in the actual sampled audio stream.
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.
I was trying to set up an audio unit to render the music (instead of Audio Queue.. which was too opaque for my purposes).. iOS doesn't have this property kAudioDevicePropertyBufferFrameSize.. any idea how I can derive this value to set up the buffer size of my IO unit?
I found this post interesting.. it asks about the possibility of using a combination of kAudioSessionProperty_CurrentHardwareIOBufferDuration and kAudioSessionProperty_CurrentHardwareOutputLatency audio session properties to determine that value.. but there is no answer.. any ideas?
You can use the kAudioSessionProperty_CurrentHardwareIOBufferDuration property, which represents the buffer size in seconds. Multiply this by the sample rate you get from kAudioSessionProperty_CurrentHardwareSampleRate to get the number of samples you should buffer.
The resulting buffer size should be a multiple of 2. I believe either 512 or 4096 are what you're likely to get, but you should always base it off of the values returned from AudioSessionGetProperty.
Example:
Float64 sampleRate;
UInt32 propSize = sizeof(Float64);
AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareSampleRate,
&propSize,
&sampleRate);
Float32 bufferDuration;
propSize = sizeof(Float32);
AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareIOBufferDuration,
&propSize,
&bufferDuration);
UInt32 bufferLengthInFrames = sampleRate * bufferDuration;
The next step is to find out the input stream format of the unit you're sending audio to. Based on your description, I'm assuming that you're programmatically generating audio to send to the speakers. This code assumes unit is an AudioUnit you're sending audio to, whether that's the RemoteIO or something like an effect Audio Unit.
AudioStreamBasicDescription inputASBD;
UInt32 propSize = sizeof(AudioStreamBasicDescription);
AudioUnitGetProperty(unit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
0,
&inputASBD,
&propSize);
After this, inputASBD.mFormatFlags will be a bit field corresponding to the audio stream format that unit is expecting. The two most likely sets of flags are named kAudioFormatFlagsCanonical and kAudioFormatFlagsAudioUnitCanonical. These two have corresponding sample types AudioSampleType and AudioUnitSampleType that you can base your size calculation off of.
As an aside, AudioSampleType typically represents samples coming from the mic or destined for the speakers, whereas AudioUnitSampleType is usually for samples that are intended to be processed (by an audio unit, for example). At the moment on iOS, AudioSampleType is a SInt16 and AudioUnitSampleType is fixed 8.24 number stored in a SInt32 container. Here's a post on the Core Audio mailing list explaining this design choice
The reason I hold back from saying something like "just use Float32, it'll work" is because the actual bit representation of the stream is subject to change if Apple feels like it.
The audio unit itself decides on the actual buffer size, so the app's audio unit callback has to be able to handle any reasonable size given to it. You can suggest and poll the kAudioSessionProperty_CurrentHardwareIOBufferDuration property, but note that this value can while your app is running (especially during screen lock or call interruptions, etc.) outside of what the app can control.
EDIT: In the end I used exactly as I explained below, AVRecorder for recording the speech and openAL for the pitch shift and playback. It worked out quite well.
I got a question regarding recording, modifying and playing back audio. I asked a similar question before ( Record, modify pitch and play back audio in real time on iOS ) but I now have more information and could do with some further advice please.
So firstly this is what I am trying to do (on a separate thread to the main thread):
monitor the iphone mic
check for sound greater than a certain volume
if above threshold start recording e.g. person starts talking
continue to record until volume drops below threshold e.g. person stops talking
modify pitch of recorded sound.
playback sound
I was thinking of using the AVRecorder to monitor and record the sound, good tutorial here: http://mobileorchard.com/tutorial-detecting-when-a-user-blows-into-the-mic/
and I was thinking of using openAL to modify the pitch of the recorded audio.
So my question is, is my thinking correct in the list of points above, am I missing something or is there a better/easier way to do it. Can I avoid mixing audio libraries and just use AVFoundation to change the pitch too?
You can either use AVRecorder or something lower like the realtime IO audio unit.
The concept of 'volume' is pretty vague. You might want to look at the difference between calculating peak and RMS values, and understanding how to integrate an RMS value over a given time (say 300ms which is what a VU meter uses).
Basically you sum all the squares of the values. You would take the square root and convert to dBFS with 10 * log10f(sqrt(sum/num_samples)), but you can do that without the sqrt in one step with 20 * log10f(sum/num_samples).
You'll need to do a lot of adjusting of integration times and thresholds to get it to behave the way you want.
For pitch shifting, I think OpenAL with do the trick, the technique behind it is called band limited interpolation - https://ccrma.stanford.edu/~jos/resample/Theory_Ideal_Bandlimited_Interpolation.html
This example shows a rms calculation as a running average. The circular buffer maintains a history of squares, and eliminates the need to sum the squares every operation. I haven't run it so treat it as pseudo code ;)
Example:
class VUMeter
{
protected:
// samples per second
float _sampleRate;
// the integration time in seconds (vu meter is 300ms)
float _integrationTime;
// these maintain a circular buffer which contains
// the 'squares' of the audio samples
int _integrationBufferLength;
float *_integrationBuffer;
float *_integrationBufferEnd;
float *_cursor;
// this is a sort of accumulator to make a running
// average more efficient
float _sum;
public:
VUMeter()
: _sampleRate(48000.0f)
, _integrationTime(0.3f)
, _sum(0.)
{
// create a buffer of values to be integrated
// e.g 300ms # 48khz is 14400 samples
_integrationBufferLength = (int) (_integrationTime * _sampleRate);
_integrationBuffer = new float[_integrationBufferLength + 1];
bzero(_integrationBuffer, _integrationBufferLength);
// set the pointers for our ciruclar buffer
_integrationBufferEnd = _integrationBuffer + _integrationBufferLength;
_cursor = _integrationBuffer;
}
~VUMeter()
{
delete _integrationBuffer;
}
float getRms(float *audio, int samples)
{
// process the samples
// this part accumulates the 'squares'
for (int i = 0; i < samples; ++i)
{
// get the input sample
float s = audio[i];
// remove the oldest value from the sum
_sum -= *_cursor;
// calculate the square and write it into the buffer
double square = s * s;
*_cursor = square;
// add it to the sum
_sum += square;
// increment the buffer cursor and wrap
++_cursor;
if (_cursor == _integrationBufferEnd)
_cursor = _integrationBuffer;
}
// now calculate the 'root mean' value in db
return 20 * log10f(_sum / _integrationBufferLength);
}
};
OpenAL resampling will change the pitch and the duration inversely. e.g. a sound resampled to a higher pitch will play for a shorter amount of time and thus faster.