Playing Audio on iOS from Socket connection - ios

Question

Hope you can help me with this issue, I have seen a lot of questions related to this, but none of them really helps me to figure out what I am doing wrong here.
So on Android I have an AudioRecord which is recording audio and sending the audio as byte array over a socket connection to clients. This part was super easy on Android and is working perfectly.
When I started working with iOS I found out there is no easy way to go about this, so after 2 days of research and plugging and playing this is what I have got. Which still does not play any audio. It makes a noise when it starts but none of the audio being transferred over the socket is being played. I confirmed that the socket is receiving data by logging each element in the buffer array.
Here is all the code I am using, a lot is reused from a bunch of sites, can't remember all the links. (BTW using AudioUnits)
First up, audio processor:
Play Callback
static OSStatus playbackCallback(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData) {
/**
This is the reference to the object who owns the callback.
*/
AudioProcessor *audioProcessor = (__bridge AudioProcessor*) inRefCon;
// iterate over incoming stream an copy to output stream
for (int i=0; i < ioData->mNumberBuffers; i++) {
AudioBuffer buffer = ioData->mBuffers[i];
// find minimum size
UInt32 size = min(buffer.mDataByteSize, [audioProcessor audioBuffer].mDataByteSize);
// copy buffer to audio buffer which gets played after function return
memcpy(buffer.mData, [audioProcessor audioBuffer].mData, size);
// set data size
buffer.mDataByteSize = size;
}
return noErr;
}
Audio processor initialize
-(void)initializeAudio
{
OSStatus status;
// We define the audio component
AudioComponentDescription desc;
desc.componentType = kAudioUnitType_Output; // we want to ouput
desc.componentSubType = kAudioUnitSubType_RemoteIO; // we want in and ouput
desc.componentFlags = 0; // must be zero
desc.componentFlagsMask = 0; // must be zero
desc.componentManufacturer = kAudioUnitManufacturer_Apple; // select provider
// find the AU component by description
AudioComponent inputComponent = AudioComponentFindNext(NULL, &desc);
// create audio unit by component
status = AudioComponentInstanceNew(inputComponent, &audioUnit);
[self hasError:status:__FILE__:__LINE__];
// define that we want record io on the input bus
UInt32 flag = 1;
// define that we want play on io on the output bus
status = AudioUnitSetProperty(audioUnit,
kAudioOutputUnitProperty_EnableIO, // use io
kAudioUnitScope_Output, // scope to output
kOutputBus, // select output bus (0)
&flag, // set flag
sizeof(flag));
[self hasError:status:__FILE__:__LINE__];
/*
We need to specifie our format on which we want to work.
We use Linear PCM cause its uncompressed and we work on raw data.
for more informations check.
We want 16 bits, 2 bytes per packet/frames at 44khz
*/
AudioStreamBasicDescription audioFormat;
audioFormat.mSampleRate = SAMPLE_RATE;
audioFormat.mFormatID = kAudioFormatLinearPCM;
audioFormat.mFormatFlags = kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger;
audioFormat.mFramesPerPacket = 1;
audioFormat.mChannelsPerFrame = 1;
audioFormat.mBitsPerChannel = 16;
audioFormat.mBytesPerPacket = 2;
audioFormat.mBytesPerFrame = 2;
// set the format on the output stream
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Output,
kInputBus,
&audioFormat,
sizeof(audioFormat));
[self hasError:status:__FILE__:__LINE__];
/**
We need to define a callback structure which holds
a pointer to the recordingCallback and a reference to
the audio processor object
*/
AURenderCallbackStruct callbackStruct;
/*
We do the same on the output stream to hear what is coming
from the input stream
*/
callbackStruct.inputProc = playbackCallback;
callbackStruct.inputProcRefCon = (__bridge void *)(self);
// set playbackCallback as callback on our renderer for the output bus
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_SetRenderCallback,
kAudioUnitScope_Global,
kOutputBus,
&callbackStruct,
sizeof(callbackStruct));
[self hasError:status:__FILE__:__LINE__];
// reset flag to 0
flag = 0;
/*
we need to tell the audio unit to allocate the render buffer,
that we can directly write into it.
*/
status = AudioUnitSetProperty(audioUnit,
kAudioUnitProperty_ShouldAllocateBuffer,
kAudioUnitScope_Output,
kInputBus,
&flag,
sizeof(flag));
/*
we set the number of channels to mono and allocate our block size to
1024 bytes.
*/
audioBuffer.mNumberChannels = 1;
audioBuffer.mDataByteSize = 512 * 2;
audioBuffer.mData = malloc( 512 * 2 );
// Initialize the Audio Unit and cross fingers =)
status = AudioUnitInitialize(audioUnit);
[self hasError:status:__FILE__:__LINE__];
NSLog(#"Started");
}
Start Playing
-(void)start;
{
// start the audio unit. You should hear something, hopefully <img src="http://www.stefanpopp.de/wp-includes/images/smilies/icon_smile.gif" alt=":)" class="wp-smiley">
OSStatus status = AudioOutputUnitStart(audioUnit);
[self hasError:status:__FILE__:__LINE__];
}
Adding data to the buffer
-(void)processBuffer: (AudioBufferList*) audioBufferList
{
AudioBuffer sourceBuffer = audioBufferList->mBuffers[0];
// we check here if the input data byte size has changed
if (audioBuffer.mDataByteSize != sourceBuffer.mDataByteSize) {
// clear old buffer
free(audioBuffer.mData);
// assing new byte size and allocate them on mData
audioBuffer.mDataByteSize = sourceBuffer.mDataByteSize;
audioBuffer.mData = malloc(sourceBuffer.mDataByteSize);
}
// loop over every packet
// copy incoming audio data to the audio buffer
memcpy(audioBuffer.mData, audioBufferList->mBuffers[0].mData, audioBufferList->mBuffers[0].mDataByteSize);
}
Stream connection callback (Socket)
-(void)stream:(NSStream *)aStream handleEvent:(NSStreamEvent)eventCode
{
if(eventCode == NSStreamEventHasBytesAvailable)
{
if(aStream == inputStream) {
uint8_t buffer[1024];
UInt32 len;
while ([inputStream hasBytesAvailable]) {
len = (UInt32)[inputStream read:buffer maxLength:sizeof(buffer)];
if(len > 0)
{
AudioBuffer abuffer;
abuffer.mDataByteSize = len; // sample size
abuffer.mNumberChannels = 1; // one channel
abuffer.mData = buffer;
int16_t audioBuffer[len];
for(int i = 0; i <= len; i++)
{
audioBuffer[i] = MuLaw_Decode(buffer[i]);
}
AudioBufferList bufferList;
bufferList.mNumberBuffers = 1;
bufferList.mBuffers[0] = abuffer;
NSLog(#"%", bufferList.mBuffers[0]);
[audioProcessor processBuffer:&bufferList];
}
}
}
}
}
The MuLaw_Decode
#define MULAW_BIAS 33
int16_t MuLaw_Decode(uint8_t number)
{
uint8_t sign = 0, position = 0;
int16_t decoded = 0;
number =~ number;
if(number&0x80)
{
number&=~(1<<7);
sign = -1;
}
position= ((number & 0xF0) >> 4) + 5;
decoded = ((1<<position) | ((number&0x0F) << (position - 4)) |(1<<(position-5))) - MULAW_BIAS;
return (sign == 0) ? decoded : (-(decoded));
}
And the code that opens the connection and initialises the audio processor
CFReadStreamRef readStream;
CFWriteStreamRef writeStream;
CFStreamCreatePairWithSocketToHost(NULL, (CFStringRef)#"10.0.0.14", 6000, &readStream, &writeStream);
inputStream = (__bridge_transfer NSInputStream *)readStream;
outputStream = (__bridge_transfer NSOutputStream *)writeStream;
[inputStream setDelegate:self];
[outputStream setDelegate:self];
[inputStream scheduleInRunLoop:[NSRunLoop currentRunLoop] forMode:NSDefaultRunLoopMode];
[outputStream scheduleInRunLoop:[NSRunLoop currentRunLoop] forMode:NSDefaultRunLoopMode];
[inputStream open];
[outputStream open];
audioProcessor = [[AudioProcessor alloc] init];
[audioProcessor start];
[audioProcessor setGain:1];
I believe the issue in my code is with the socket connection callback, that I am not doing the right thing with the data.

Answer 1

I solved this in the end, see my answer here
I intended putting the code here, but it would be a lot of copy pasting