Modified source engine (2017) developed by valve and leaked in 2020. Not for commercial purporses
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//========= Copyright 1996-2009, Valve Corporation, All rights reserved. ============//
//
// Purpose:
//
// $NoKeywords: $
//
//=============================================================================//
// This module implements the voice record and compression functions
#include <Carbon/Carbon.h>
#include <AudioUnit/AudioUnit.h>
#include <AudioToolbox/AudioToolbox.h>
#include "tier0/platform.h"
#include "tier0/threadtools.h"
//#include "tier0/vcrmode.h"
#include "ivoicerecord.h"
#define kNumSecAudioBuffer 1.0f
// ------------------------------------------------------------------------------
// VoiceRecord_AudioQueue
// ------------------------------------------------------------------------------
class VoiceRecord_AudioQueue : public IVoiceRecord
{
public:
VoiceRecord_AudioQueue();
virtual ~VoiceRecord_AudioQueue();
// IVoiceRecord.
virtual void Release();
virtual bool RecordStart();
virtual void RecordStop();
// Initialize. The format of the data we expect from the provider is
// 8-bit signed mono at the specified sample rate.
virtual bool Init( int nSampleRate );
virtual void Idle();
// Get the most recent N samples.
virtual int GetRecordedData(short *pOut, int nSamplesWanted );
AudioUnit GetAudioUnit() { return m_AudioUnit; }
AudioConverterRef GetConverter() { return m_Converter; }
void RenderBuffer( const short *pszBuf, int nSamples );
bool BRecording() { return m_bRecordingAudio; }
void ClearThreadHandle() { m_hThread = NULL; m_bFirstInit = false; }
AudioBufferList m_MicInputBuffer;
AudioBufferList m_ConverterBuffer;
void *m_pMicInputBuffer;
int m_nMicInputSamplesAvaialble;
float m_flSampleRateConversion;
int m_nBufferFrameSize;
int m_ConverterBufferSize;
int m_MicInputBufferSize;
int m_InputBytesPerPacket;
private:
bool InitalizeInterfaces(); // Initialize the openal capture buffers and other interfaces
void ReleaseInterfaces(); // Release openal buffers and other interfaces
void ClearInterfaces(); // Clear members.
private:
AudioUnit m_AudioUnit;
char *m_SampleBuffer;
int m_SampleBufferSize;
int m_nSampleRate;
bool m_bRecordingAudio;
bool m_bFirstInit;
ThreadHandle_t m_hThread;
AudioConverterRef m_Converter;
CInterlockedUInt m_SampleBufferReadPos;
CInterlockedUInt m_SampleBufferWritePos;
//UInt32 nPackets = 0;
//bool bHaveListData = false;
};
VoiceRecord_AudioQueue::VoiceRecord_AudioQueue() :
m_nSampleRate( 0 ), m_AudioUnit( NULL ), m_SampleBufferSize(0), m_SampleBuffer(NULL),
m_SampleBufferReadPos(0), m_SampleBufferWritePos(0), m_bRecordingAudio(false), m_hThread( NULL ), m_bFirstInit( true )
{
ClearInterfaces();
}
VoiceRecord_AudioQueue::~VoiceRecord_AudioQueue()
{
ReleaseInterfaces();
if ( m_hThread )
ReleaseThreadHandle( m_hThread );
m_hThread = NULL;
}
void VoiceRecord_AudioQueue::Release()
{
ReleaseInterfaces();
}
uintp StartAudio( void *pRecorder )
{
VoiceRecord_AudioQueue *vr = (VoiceRecord_AudioQueue *)pRecorder;
if ( vr )
{
//printf( "AudioOutputUnitStart\n" );
AudioOutputUnitStart( vr->GetAudioUnit() );
vr->ClearThreadHandle();
}
//printf( "StartAudio thread done\n" );
return 0;
}
bool VoiceRecord_AudioQueue::RecordStart()
{
if ( !m_AudioUnit )
return false;
if ( m_bFirstInit )
m_hThread = CreateSimpleThread( StartAudio, this );
else
AudioOutputUnitStart( m_AudioUnit );
m_SampleBufferReadPos = m_SampleBufferWritePos = 0;
m_bRecordingAudio = true;
//printf( "VoiceRecord_AudioQueue::RecordStart\n" );
return ( !m_bFirstInit || m_hThread != NULL );
}
void VoiceRecord_AudioQueue::RecordStop()
{
// Stop capturing.
if ( m_AudioUnit && m_bRecordingAudio )
{
AudioOutputUnitStop( m_AudioUnit );
//printf( "AudioOutputUnitStop\n" );
}
m_SampleBufferReadPos = m_SampleBufferWritePos = 0;
m_bRecordingAudio = false;
if ( m_hThread )
ReleaseThreadHandle( m_hThread );
m_hThread = NULL;
}
OSStatus ComplexBufferFillPlayback( AudioConverterRef inAudioConverter,
UInt32 *ioNumberDataPackets,
AudioBufferList *ioData,
AudioStreamPacketDescription **outDataPacketDesc,
void *inUserData)
{
VoiceRecord_AudioQueue *vr = (VoiceRecord_AudioQueue *)inUserData;
if ( !vr->BRecording() )
return noErr;
if ( vr->m_nMicInputSamplesAvaialble )
{
int nBytesRequired = *ioNumberDataPackets * vr->m_InputBytesPerPacket;
int nBytesAvailable = vr->m_nMicInputSamplesAvaialble*vr->m_InputBytesPerPacket;
if ( nBytesRequired < nBytesAvailable )
{
ioData->mBuffers[0].mData = vr->m_MicInputBuffer.mBuffers[0].mData;
ioData->mBuffers[0].mDataByteSize = nBytesRequired;
vr->m_MicInputBuffer.mBuffers[0].mData = (char *)vr->m_MicInputBuffer.mBuffers[0].mData+nBytesRequired;
vr->m_MicInputBuffer.mBuffers[0].mDataByteSize = nBytesAvailable - nBytesRequired;
}
else
{
ioData->mBuffers[0].mData = vr->m_MicInputBuffer.mBuffers[0].mData;
ioData->mBuffers[0].mDataByteSize = nBytesAvailable;
vr->m_MicInputBuffer.mBuffers[0].mData = vr->m_pMicInputBuffer;
vr->m_MicInputBuffer.mBuffers[0].mDataByteSize = vr->m_MicInputBufferSize;
}
*ioNumberDataPackets = ioData->mBuffers[0].mDataByteSize / vr->m_InputBytesPerPacket;
vr->m_nMicInputSamplesAvaialble = nBytesAvailable / vr->m_InputBytesPerPacket - *ioNumberDataPackets;
}
else
{
*ioNumberDataPackets = 0;
return -1;
}
return noErr;
}
static OSStatus recordingCallback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData)
{
VoiceRecord_AudioQueue *vr = (VoiceRecord_AudioQueue *)inRefCon;
if ( !vr->BRecording() )
return noErr;
OSStatus err = noErr;
if ( vr->m_nMicInputSamplesAvaialble == 0 )
{
err = AudioUnitRender( vr->GetAudioUnit(), ioActionFlags, inTimeStamp, 1, inNumberFrames, &vr->m_MicInputBuffer );
if ( err == noErr )
vr->m_nMicInputSamplesAvaialble = vr->m_MicInputBuffer.mBuffers[0].mDataByteSize / vr->m_InputBytesPerPacket;
}
if ( vr->m_nMicInputSamplesAvaialble > 0 )
{
UInt32 nConverterSamples = ceil(vr->m_nMicInputSamplesAvaialble/vr->m_flSampleRateConversion);
vr->m_ConverterBuffer.mBuffers[0].mDataByteSize = vr->m_ConverterBufferSize;
OSStatus err = AudioConverterFillComplexBuffer( vr->GetConverter(),
ComplexBufferFillPlayback,
vr,
&nConverterSamples,
&vr->m_ConverterBuffer,
NULL );
if ( err == noErr || err == -1 )
vr->RenderBuffer( (short *)vr->m_ConverterBuffer.mBuffers[0].mData, vr->m_ConverterBuffer.mBuffers[0].mDataByteSize/sizeof(short) );
}
return err;
}
void VoiceRecord_AudioQueue::RenderBuffer( const short *pszBuf, int nSamples )
{
int samplePos = m_SampleBufferWritePos;
int samplePosBefore = samplePos;
int readPos = m_SampleBufferReadPos;
bool bBeforeRead = false;
if ( samplePos < readPos )
bBeforeRead = true;
char *pOut = (char *)(m_SampleBuffer + samplePos);
int nFirstCopy = MIN( nSamples*sizeof(short), m_SampleBufferSize - samplePos );
memcpy( pOut, pszBuf, nFirstCopy );
samplePos += nFirstCopy;
if ( nSamples*sizeof(short) > nFirstCopy )
{
nSamples -= ( nFirstCopy / sizeof(short) );
samplePos = 0;
memcpy( m_SampleBuffer, pszBuf + nFirstCopy, nSamples * sizeof(short) );
samplePos += nSamples * sizeof(short);
}
m_SampleBufferWritePos = samplePos%m_SampleBufferSize;
if ( (bBeforeRead && samplePos > readPos) )
{
m_SampleBufferReadPos = (readPos+m_SampleBufferSize/2)%m_SampleBufferSize; // if we crossed the read pointer then bump it forward
//printf( "Crossed %d %d (%d)\n", (int)samplePosBefore, (int)samplePos, readPos );
}
}
bool VoiceRecord_AudioQueue::InitalizeInterfaces()
{
//printf( "Initializing audio queue recorder\n" );
// Describe audio component
ComponentDescription desc;
desc.componentType = kAudioUnitType_Output;
desc.componentSubType = kAudioUnitSubType_HALOutput;
desc.componentManufacturer = kAudioUnitManufacturer_Apple;
desc.componentFlags = 0;
desc.componentFlagsMask = 0;
Component comp = FindNextComponent(NULL, &desc);
if (comp == NULL)
return false;
OSStatus status = OpenAComponent(comp, &m_AudioUnit);
if ( status != noErr )
return false;
// Enable IO for recording
UInt32 flag = 1;
status = AudioUnitSetProperty( m_AudioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input,
1, &flag, sizeof(flag));
if ( status != noErr )
return false;
// disable output on the device
flag = 0;
status = AudioUnitSetProperty( m_AudioUnit,kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output,
0, &flag,sizeof(flag));
if ( status != noErr )
return false;
UInt32 size = sizeof(AudioDeviceID);
AudioDeviceID inputDevice;
status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice,&size, &inputDevice);
if ( status != noErr )
return false;
status =AudioUnitSetProperty( m_AudioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global,
0, &inputDevice, sizeof(inputDevice));
if ( status != noErr )
return false;
// Describe format
AudioStreamBasicDescription audioDeviceFormat;
size = sizeof(AudioStreamBasicDescription);
status = AudioUnitGetProperty( m_AudioUnit,
kAudioUnitProperty_StreamFormat,
kAudioUnitScope_Input,
1, // input bus
&audioDeviceFormat,
&size);
if ( status != noErr )
return false;
// we only want mono audio, so if they have a stero input ask for mono
if ( audioDeviceFormat.mChannelsPerFrame == 2 )
{
audioDeviceFormat.mChannelsPerFrame = 1;
audioDeviceFormat.mBytesPerPacket /= 2;
audioDeviceFormat.mBytesPerFrame /= 2;
}
// Apply format
status = AudioUnitSetProperty( m_AudioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output,
1, &audioDeviceFormat, sizeof(audioDeviceFormat) );
if ( status != noErr )
return false;
AudioStreamBasicDescription audioOutputFormat;
audioOutputFormat = audioDeviceFormat;
audioOutputFormat.mFormatID = kAudioFormatLinearPCM;
audioOutputFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
audioOutputFormat.mBytesPerPacket = 2; // 16-bit samples * 1 channels
audioOutputFormat.mFramesPerPacket = 1;
audioOutputFormat.mBytesPerFrame = 2; // 16-bit samples * 1 channels
audioOutputFormat.mChannelsPerFrame = 1;
audioOutputFormat.mBitsPerChannel = 16;
audioOutputFormat.mReserved = 0;
audioOutputFormat.mSampleRate = m_nSampleRate;
m_flSampleRateConversion = audioDeviceFormat.mSampleRate / audioOutputFormat.mSampleRate;
// setup sample rate conversion
status = AudioConverterNew( &audioDeviceFormat, &audioOutputFormat, &m_Converter );
if ( status != noErr )
return false;
UInt32 primeMethod = kConverterPrimeMethod_None;
status = AudioConverterSetProperty( m_Converter, kAudioConverterPrimeMethod, sizeof(UInt32), &primeMethod);
if ( status != noErr )
return false;
UInt32 quality = kAudioConverterQuality_Medium;
status = AudioConverterSetProperty( m_Converter, kAudioConverterSampleRateConverterQuality, sizeof(UInt32), &quality);
if ( status != noErr )
return false;
// Set input callback
AURenderCallbackStruct callbackStruct;
callbackStruct.inputProc = recordingCallback;
callbackStruct.inputProcRefCon = this;
status = AudioUnitSetProperty( m_AudioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global,
0, &callbackStruct, sizeof(callbackStruct) );
if ( status != noErr )
return false;
UInt32 bufferFrameSize;
size = sizeof(bufferFrameSize);
status = AudioDeviceGetProperty( inputDevice, 1, 1, kAudioDevicePropertyBufferFrameSize, &size, &bufferFrameSize );
if ( status != noErr )
return false;
m_nBufferFrameSize = bufferFrameSize;
// allocate the input and conversion sound storage buffers
m_MicInputBuffer.mNumberBuffers = 1;
m_MicInputBuffer.mBuffers[0].mDataByteSize = m_nBufferFrameSize*audioDeviceFormat.mBitsPerChannel/8*audioDeviceFormat.mChannelsPerFrame;
m_MicInputBuffer.mBuffers[0].mData = malloc( m_MicInputBuffer.mBuffers[0].mDataByteSize );
m_MicInputBuffer.mBuffers[0].mNumberChannels = audioDeviceFormat.mChannelsPerFrame;
m_pMicInputBuffer = m_MicInputBuffer.mBuffers[0].mData;
m_MicInputBufferSize = m_MicInputBuffer.mBuffers[0].mDataByteSize;
m_InputBytesPerPacket = audioDeviceFormat.mBytesPerPacket;
m_ConverterBuffer.mNumberBuffers = 1;
m_ConverterBuffer.mBuffers[0].mDataByteSize = m_nBufferFrameSize*audioOutputFormat.mBitsPerChannel/8*audioOutputFormat.mChannelsPerFrame;
m_ConverterBuffer.mBuffers[0].mData = malloc( m_MicInputBuffer.mBuffers[0].mDataByteSize );
m_ConverterBuffer.mBuffers[0].mNumberChannels = 1;
m_ConverterBufferSize = m_ConverterBuffer.mBuffers[0].mDataByteSize;
m_nMicInputSamplesAvaialble = 0;
m_SampleBufferReadPos = m_SampleBufferWritePos = 0;
m_SampleBufferSize = ceil( kNumSecAudioBuffer * m_nSampleRate * audioOutputFormat.mBytesPerPacket );
m_SampleBuffer = (char *)malloc( m_SampleBufferSize );
memset( m_SampleBuffer, 0x0, m_SampleBufferSize );
DevMsg( "Initialized AudioQueue record interface\n" );
return true;
}
bool VoiceRecord_AudioQueue::Init( int nSampleRate )
{
if ( m_AudioUnit && m_nSampleRate != nSampleRate )
{
// Need to recreate interfaces with different sample rate
ReleaseInterfaces();
ClearInterfaces();
}
m_nSampleRate = nSampleRate;
// Re-initialize the capture buffer if neccesary
if ( !m_AudioUnit )
{
InitalizeInterfaces();
}
m_SampleBufferReadPos = m_SampleBufferWritePos = 0;
//printf( "VoiceRecord_AudioQueue::Init()\n" );
// Initialise
OSStatus status = AudioUnitInitialize( m_AudioUnit );
if ( status != noErr )
return false;
return true;
}
void VoiceRecord_AudioQueue::ReleaseInterfaces()
{
AudioOutputUnitStop( m_AudioUnit );
AudioConverterDispose( m_Converter );
AudioUnitUninitialize( m_AudioUnit );
m_AudioUnit = NULL;
m_Converter = NULL;
}
void VoiceRecord_AudioQueue::ClearInterfaces()
{
m_AudioUnit = NULL;
m_Converter = NULL;
m_SampleBufferReadPos = m_SampleBufferWritePos = 0;
if ( m_SampleBuffer )
free( m_SampleBuffer );
m_SampleBuffer = NULL;
if ( m_MicInputBuffer.mBuffers[0].mData )
free( m_MicInputBuffer.mBuffers[0].mData );
if ( m_ConverterBuffer.mBuffers[0].mData )
free( m_ConverterBuffer.mBuffers[0].mData );
m_MicInputBuffer.mBuffers[0].mData = NULL;
m_ConverterBuffer.mBuffers[0].mData = NULL;
}
void VoiceRecord_AudioQueue::Idle()
{
}
int VoiceRecord_AudioQueue::GetRecordedData(short *pOut, int nSamples )
{
if ( !m_SampleBuffer )
return 0;
int cbSamples = nSamples*2; // convert to bytes
int writePos = m_SampleBufferWritePos;
int readPos = m_SampleBufferReadPos;
int nOutstandingSamples = ( writePos - readPos );
if ( readPos > writePos ) // writing has wrapped around
{
nOutstandingSamples = writePos + ( m_SampleBufferSize - readPos );
}
if ( !nOutstandingSamples )
return 0;
if ( nOutstandingSamples < cbSamples )
cbSamples = nOutstandingSamples; // clamp to the number of samples we have available
memcpy( (char *)pOut, m_SampleBuffer + readPos, MIN( cbSamples, m_SampleBufferSize - readPos ) );
if ( cbSamples > ( m_SampleBufferSize - readPos ) )
{
int offset = m_SampleBufferSize - readPos;
cbSamples -= offset;
readPos = 0;
memcpy( (char *)pOut + offset, m_SampleBuffer, cbSamples );
}
readPos+=cbSamples;
m_SampleBufferReadPos = readPos%m_SampleBufferSize;
//printf( "Returning %d samples, %d %d (%d)\n", cbSamples/2, (int)m_SampleBufferReadPos, (int)m_SampleBufferWritePos, m_SampleBufferSize );
return cbSamples/2;
}
VoiceRecord_AudioQueue g_AudioQueueVoiceRecord;
IVoiceRecord* CreateVoiceRecord_AudioQueue( int sampleRate )
{
if ( g_AudioQueueVoiceRecord.Init( sampleRate ) )
{
return &g_AudioQueueVoiceRecord;
}
else
{
g_AudioQueueVoiceRecord.Release();
return NULL;
}
}