iOS: How to read an audio file into a float buffer - ios

Question

I have a really short audio file, say a 10th of a second in (say) .PCM format
I want to use RemoteIO to loop through the file repeatedly to produce a continuous musical tone. So how do I read this into an array of floats?
EDIT: while I could probably dig out the file format, extract the file into an NSData and process it manually, I'm guessing there is a more sensible generic approach... ( that eg copes with different formats )

Answer 1

You can use ExtAudioFile to read data from any supported data format in numerous client formats. Here is an example to read a file as 16-bit integers:
CFURLRef url = /* ... */;
ExtAudioFileRef eaf;
OSStatus err = ExtAudioFileOpenURL((CFURLRef)url, &eaf);
if(noErr != err)
/* handle error */
AudioStreamBasicDescription format;
format.mSampleRate = 44100;
format.mFormatID = kAudioFormatLinearPCM;
format.mFormatFlags = kAudioFormatFormatFlagIsPacked;
format.mBitsPerChannel = 16;
format.mChannelsPerFrame = 2;
format.mBytesPerFrame = format.mChannelsPerFrame * 2;
format.mFramesPerPacket = 1;
format.mBytesPerPacket = format.mFramesPerPacket * format.mBytesPerFrame;
err = ExtAudioFileSetProperty(eaf, kExtAudioFileProperty_ClientDataFormat, sizeof(format), &format);
/* Read the file contents using ExtAudioFileRead */
If you wanted Float32 data, you would set up format like this:
format.mFormatID = kAudioFormatLinearPCM;
format.mFormatFlags = kAudioFormatFlagsNativeFloatPacked;
format.mBitsPerChannel = 32;

Answer 2

This is the code I have used to convert my audio data (audio file ) into floating point representation and saved into an array.
-(void) PrintFloatDataFromAudioFile {
NSString * name = #"Filename"; //YOUR FILE NAME
NSString * source = [[NSBundle mainBundle] pathForResource:name ofType:#"m4a"]; // SPECIFY YOUR FILE FORMAT
const char *cString = [source cStringUsingEncoding:NSASCIIStringEncoding];
CFStringRef str = CFStringCreateWithCString(
NULL,
cString,
kCFStringEncodingMacRoman
);
CFURLRef inputFileURL = CFURLCreateWithFileSystemPath(
kCFAllocatorDefault,
str,
kCFURLPOSIXPathStyle,
false
);
ExtAudioFileRef fileRef;
ExtAudioFileOpenURL(inputFileURL, &fileRef);
AudioStreamBasicDescription audioFormat;
audioFormat.mSampleRate = 44100; // GIVE YOUR SAMPLING RATE
audioFormat.mFormatID = kAudioFormatLinearPCM;
audioFormat.mFormatFlags = kLinearPCMFormatFlagIsFloat;
audioFormat.mBitsPerChannel = sizeof(Float32) * 8;
audioFormat.mChannelsPerFrame = 1; // Mono
audioFormat.mBytesPerFrame = audioFormat.mChannelsPerFrame * sizeof(Float32); // == sizeof(Float32)
audioFormat.mFramesPerPacket = 1;
audioFormat.mBytesPerPacket = audioFormat.mFramesPerPacket * audioFormat.mBytesPerFrame; // = sizeof(Float32)
// 3) Apply audio format to the Extended Audio File
ExtAudioFileSetProperty(
fileRef,
kExtAudioFileProperty_ClientDataFormat,
sizeof (AudioStreamBasicDescription), //= audioFormat
&audioFormat);
int numSamples = 1024; //How many samples to read in at a time
UInt32 sizePerPacket = audioFormat.mBytesPerPacket; // = sizeof(Float32) = 32bytes
UInt32 packetsPerBuffer = numSamples;
UInt32 outputBufferSize = packetsPerBuffer * sizePerPacket;
// So the lvalue of outputBuffer is the memory location where we have reserved space
UInt8 *outputBuffer = (UInt8 *)malloc(sizeof(UInt8 *) * outputBufferSize);
AudioBufferList convertedData ;//= malloc(sizeof(convertedData));
convertedData.mNumberBuffers = 1; // Set this to 1 for mono
convertedData.mBuffers[0].mNumberChannels = audioFormat.mChannelsPerFrame; //also = 1
convertedData.mBuffers[0].mDataByteSize = outputBufferSize;
convertedData.mBuffers[0].mData = outputBuffer; //
UInt32 frameCount = numSamples;
float *samplesAsCArray;
int j =0;
double floatDataArray[882000] ; // SPECIFY YOUR DATA LIMIT MINE WAS 882000 , SHOULD BE EQUAL TO OR MORE THAN DATA LIMIT
while (frameCount > 0) {
ExtAudioFileRead(
fileRef,
&frameCount,
&convertedData
);
if (frameCount > 0) {
AudioBuffer audioBuffer = convertedData.mBuffers[0];
samplesAsCArray = (float *)audioBuffer.mData; // CAST YOUR mData INTO FLOAT
for (int i =0; i<1024 /*numSamples */; i++) { //YOU CAN PUT numSamples INTEAD OF 1024
floatDataArray[j] = (double)samplesAsCArray[i] ; //PUT YOUR DATA INTO FLOAT ARRAY
printf("\n%f",floatDataArray[j]); //PRINT YOUR ARRAY'S DATA IN FLOAT FORM RANGING -1 TO +1
j++;
}
}
}}

Answer 3

I'm not familiar with RemoteIO, but I am familiar with WAV's and thought I'd post some format information on them. If you need, you should be able to easily parse out information such as duration, bit rate, etc...
First, here is an excellent website detailing the WAVE PCM soundfile format. This site also does an excellent job illustrating what the different byte addresses inside the "fmt" sub-chunk refer to.
WAVE File format
A WAVE is composed of a "RIFF" chunk and subsequent sub-chunks
Every chunk is at least 8 bytes
First 4 bytes is the Chunk ID
Next 4 bytes is the Chunk Size (The Chunk Size gives the size of the remainder of the chunk excluding the 8 bytes used for the Chunk ID and Chunk Size)
Every WAVE has the following chunks / sub chunks
"RIFF" (first and only chunk. All the rest are technically sub-chunks.)
"fmt " (usually the first sub-chunk after "RIFF" but can be anywhere between "RIFF" and "data". This chunk has information about the WAV such as number of channels, sample rate, and byte rate)
"data" (must be the last sub-chunk and contains all the sound data)
Common WAVE Audio Formats:
PCM
IEEE_Float
PCM_EXTENSIBLE (with a sub format of PCM or IEEE_FLOAT)
WAVE Duration and Size
A WAVE File's duration can be calculated as follows:
seconds = DataChunkSize / ByteRate
Where
ByteRate = SampleRate * NumChannels * BitsPerSample/8
and DataChunkSize does not include the 8 bytes reserved for the ID and Size of the "data" sub-chunk.
Knowing this, the DataChunkSize can be calculated if you know the duration of the WAV and the ByteRate.
DataChunkSize = seconds * ByteRate
This can be useful for calculating the size of the wav data when converting from formats like mp3 or wma. Note that a typical wav header is 44 bytes followed by DataChunkSize (this is always the case if the wav was converted using the Normalizer tool - at least as of this writing).

Answer 4

Update for Swift 5
This is a simple function that helps get your audio file into an array of floats. This is for both mono and stereo audio, To get the second channel of stereo audio, just uncomment sample 2
import AVFoundation
//..
do {
guard let url = Bundle.main.url(forResource: "audio_example", withExtension: "wav") else { return }
let file = try AVAudioFile(forReading: url)
if let format = AVAudioFormat(commonFormat: .pcmFormatFloat32, sampleRate: file.fileFormat.sampleRate, channels: file.fileFormat.channelCount, interleaved: false), let buf = AVAudioPCMBuffer(pcmFormat: format, frameCapacity: AVAudioFrameCount(file.length)) {
try file.read(into: buf)
guard let floatChannelData = buf.floatChannelData else { return }
let frameLength = Int(buf.frameLength)
let samples = Array(UnsafeBufferPointer(start:floatChannelData[0], count:frameLength))
// let samples2 = Array(UnsafeBufferPointer(start:floatChannelData[1], count:frameLength))
print("samples")
print(samples.count)
print(samples.prefix(10))
// print(samples2.prefix(10))
}
} catch {
print("Audio Error: \(error)")
}