diff --git a/engine/audio/snd_dev_mac_audioqueue.cpp b/engine/audio/snd_dev_mac_audioqueue.cpp deleted file mode 100644 index e58a703f..00000000 --- a/engine/audio/snd_dev_mac_audioqueue.cpp +++ /dev/null @@ -1,599 +0,0 @@ -//========= Copyright Valve Corporation, All rights reserved. ============// -// -// Purpose: -// -//===========================================================================// - -#include "audio_pch.h" -#include -#include -#include - -// memdbgon must be the last include file in a .cpp file!!! -#include "tier0/memdbgon.h" - -extern bool snd_firsttime; -extern bool MIX_ScaleChannelVolume( paintbuffer_t *ppaint, channel_t *pChannel, int volume[CCHANVOLUMES], int mixchans ); -extern void S_SpatializeChannel( int volume[6], int master_vol, const Vector *psourceDir, float gain, float mono ); - -#define NUM_BUFFERS_SOURCES 128 -#define BUFF_MASK (NUM_BUFFERS_SOURCES - 1 ) -#define BUFFER_SIZE 0x0400 - - -//----------------------------------------------------------------------------- -// -// NOTE: This only allows 16-bit, stereo wave out -// -//----------------------------------------------------------------------------- -class CAudioDeviceAudioQueue : public CAudioDeviceBase -{ -public: - bool IsActive( void ); - bool Init( void ); - void Shutdown( void ); - void PaintEnd( void ); - int GetOutputPosition( void ); - void ChannelReset( int entnum, int channelIndex, float distanceMod ); - void Pause( void ); - void UnPause( void ); - float MixDryVolume( void ); - bool Should3DMix( void ); - void StopAllSounds( void ); - - int PaintBegin( float mixAheadTime, int soundtime, int paintedtime ); - void ClearBuffer( void ); - void UpdateListener( const Vector& position, const Vector& forward, const Vector& right, const Vector& up ); - void MixBegin( int sampleCount ); - void MixUpsample( int sampleCount, int filtertype ); - void Mix8Mono( channel_t *pChannel, char *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress ); - void Mix8Stereo( channel_t *pChannel, char *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress ); - void Mix16Mono( channel_t *pChannel, short *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress ); - void Mix16Stereo( channel_t *pChannel, short *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress ); - - void TransferSamples( int end ); - void SpatializeChannel( int volume[CCHANVOLUMES/2], int master_vol, const Vector& sourceDir, float gain, float mono); - void ApplyDSPEffects( int idsp, portable_samplepair_t *pbuffront, portable_samplepair_t *pbufrear, portable_samplepair_t *pbufcenter, int samplecount ); - - const char *DeviceName( void ) { return "AudioQueue"; } - int DeviceChannels( void ) { return 2; } - int DeviceSampleBits( void ) { return 16; } - int DeviceSampleBytes( void ) { return 2; } - int DeviceDmaSpeed( void ) { return SOUND_DMA_SPEED; } - int DeviceSampleCount( void ) { return m_deviceSampleCount; } - - void BufferCompleted() { m_buffersCompleted++; } - void SetRunning( bool bState ) { m_bRunning = bState; } - -private: - void OpenWaveOut( void ); - void CloseWaveOut( void ); - bool ValidWaveOut( void ) const; - bool BIsPlaying(); - - AudioStreamBasicDescription m_DataFormat; - AudioQueueRef m_Queue; - AudioQueueBufferRef m_Buffers[NUM_BUFFERS_SOURCES]; - - int m_SndBufSize; - - void *m_sndBuffers; - - CInterlockedInt m_deviceSampleCount; - - int m_buffersSent; - int m_buffersCompleted; - int m_pauseCount; - bool m_bSoundsShutdown; - - bool m_bFailed; - bool m_bRunning; - - -}; - -CAudioDeviceAudioQueue *wave = NULL; - - -static void AudioCallback(void *pContext, AudioQueueRef pQueue, AudioQueueBufferRef pBuffer) -{ - if ( wave ) - wave->BufferCompleted(); -} - - -IAudioDevice *Audio_CreateMacAudioQueueDevice( void ) -{ - wave = new CAudioDeviceAudioQueue; - if ( wave->Init() ) - return wave; - - delete wave; - wave = NULL; - - return NULL; -} - - -void OnSndSurroundCvarChanged2( IConVar *pVar, const char *pOldString, float flOldValue ); -void OnSndSurroundLegacyChanged2( IConVar *pVar, const char *pOldString, float flOldValue ); - -//----------------------------------------------------------------------------- -// Init, shutdown -//----------------------------------------------------------------------------- -bool CAudioDeviceAudioQueue::Init( void ) -{ - m_SndBufSize = 0; - m_sndBuffers = NULL; - m_pauseCount = 0; - - m_bSurround = false; - m_bSurroundCenter = false; - m_bHeadphone = false; - m_buffersSent = 0; - m_buffersCompleted = 0; - m_pauseCount = 0; - m_bSoundsShutdown = false; - m_bFailed = false; - m_bRunning = false; - - m_Queue = NULL; - - static bool first = true; - if ( first ) - { - snd_surround.SetValue( 2 ); - snd_surround.InstallChangeCallback( &OnSndSurroundCvarChanged2 ); - snd_legacy_surround.InstallChangeCallback( &OnSndSurroundLegacyChanged2 ); - first = false; - } - - OpenWaveOut(); - - if ( snd_firsttime ) - { - DevMsg( "Wave sound initialized\n" ); - } - return ValidWaveOut() && !m_bFailed; -} - -void CAudioDeviceAudioQueue::Shutdown( void ) -{ - CloseWaveOut(); -} - - -//----------------------------------------------------------------------------- -// WAV out device -//----------------------------------------------------------------------------- -inline bool CAudioDeviceAudioQueue::ValidWaveOut( void ) const -{ - return m_sndBuffers != 0 && m_Queue; -} - - -//----------------------------------------------------------------------------- -// called by the mac audioqueue code when we run out of playback buffers -//----------------------------------------------------------------------------- -void AudioQueueIsRunningCallback( void* inClientData, AudioQueueRef inAQ, AudioQueuePropertyID inID) -{ - CAudioDeviceAudioQueue* audioqueue = (CAudioDeviceAudioQueue*)inClientData; - - UInt32 running = 0; - UInt32 size; - OSStatus err = AudioQueueGetProperty(inAQ, kAudioQueueProperty_IsRunning, &running, &size); - audioqueue->SetRunning( running != 0 ); - //DevWarning( "AudioQueueStart %d\n", running ); -} - - - - -//----------------------------------------------------------------------------- -// Opens the windows wave out device -//----------------------------------------------------------------------------- -void CAudioDeviceAudioQueue::OpenWaveOut( void ) -{ - if ( m_Queue ) - return; - - m_buffersSent = 0; - m_buffersCompleted = 0; - - m_DataFormat.mSampleRate = 44100; - m_DataFormat.mFormatID = kAudioFormatLinearPCM; - m_DataFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked; - m_DataFormat.mBytesPerPacket = 4; // 16-bit samples * 2 channels - m_DataFormat.mFramesPerPacket = 1; - m_DataFormat.mBytesPerFrame = 4; // 16-bit samples * 2 channels - m_DataFormat.mChannelsPerFrame = 2; - m_DataFormat.mBitsPerChannel = 16; - m_DataFormat.mReserved = 0; - - // Create the audio queue that will be used to manage the array of audio - // buffers used to queue samples. - OSStatus err = AudioQueueNewOutput(&m_DataFormat, AudioCallback, this, NULL, NULL, 0, &m_Queue); - if ( err != noErr) - { - DevMsg( "Failed to create AudioQueue output %d\n", (int)err ); - m_bFailed = true; - return; - } - - for ( int i = 0; i < NUM_BUFFERS_SOURCES; ++i) - { - err = AudioQueueAllocateBuffer( m_Queue, BUFFER_SIZE,&(m_Buffers[i])); - if ( err != noErr) - { - DevMsg( "Failed to AudioQueueAllocateBuffer output %d (%i)\n",(int)err,i ); - m_bFailed = true; - } - - m_Buffers[i]->mAudioDataByteSize = BUFFER_SIZE; - Q_memset( m_Buffers[i]->mAudioData, 0, BUFFER_SIZE ); - } - - err = AudioQueuePrime( m_Queue, 0, NULL); - if ( err != noErr) - { - DevMsg( "Failed to create AudioQueue output %d\n", (int)err ); - m_bFailed = true; - return; - } - - AudioQueueSetParameter( m_Queue, kAudioQueueParam_Volume, 1.0); - - err = AudioQueueAddPropertyListener( m_Queue, kAudioQueueProperty_IsRunning, AudioQueueIsRunningCallback, this ); - if ( err != noErr) - { - DevMsg( "Failed to create AudioQueue output %d\n", (int)err ); - m_bFailed = true; - return; - } - - m_SndBufSize = NUM_BUFFERS_SOURCES*BUFFER_SIZE; - m_deviceSampleCount = m_SndBufSize / DeviceSampleBytes(); - - if ( !m_sndBuffers ) - { - m_sndBuffers = malloc( m_SndBufSize ); - memset( m_sndBuffers, 0x0, m_SndBufSize ); - } -} - - -//----------------------------------------------------------------------------- -// Closes the windows wave out device -//----------------------------------------------------------------------------- -void CAudioDeviceAudioQueue::CloseWaveOut( void ) -{ - if ( ValidWaveOut() ) - { - AudioQueueStop(m_Queue, true); - m_bRunning = false; - - AudioQueueRemovePropertyListener( m_Queue, kAudioQueueProperty_IsRunning, AudioQueueIsRunningCallback, this ); - - for ( int i = 0; i < NUM_BUFFERS_SOURCES; i++ ) - AudioQueueFreeBuffer( m_Queue, m_Buffers[i]); - - AudioQueueDispose( m_Queue, true); - - m_Queue = NULL; - } - - if ( m_sndBuffers ) - { - free( m_sndBuffers ); - m_sndBuffers = NULL; - } -} - - - -//----------------------------------------------------------------------------- -// Mixing setup -//----------------------------------------------------------------------------- -int CAudioDeviceAudioQueue::PaintBegin( float mixAheadTime, int soundtime, int paintedtime ) -{ - // soundtime - total samples that have been played out to hardware at dmaspeed - // paintedtime - total samples that have been mixed at speed - // endtime - target for samples in mixahead buffer at speed - - unsigned int endtime = soundtime + mixAheadTime * DeviceDmaSpeed(); - - int samps = DeviceSampleCount() >> (DeviceChannels()-1); - - if ((int)(endtime - soundtime) > samps) - endtime = soundtime + samps; - - if ((endtime - paintedtime) & 0x3) - { - // The difference between endtime and painted time should align on - // boundaries of 4 samples. This is important when upsampling from 11khz -> 44khz. - endtime -= (endtime - paintedtime) & 0x3; - } - - return endtime; -} - - -//----------------------------------------------------------------------------- -// Actually performs the mixing -//----------------------------------------------------------------------------- -void CAudioDeviceAudioQueue::PaintEnd( void ) -{ - int cblocks = 4 << 1; - - if ( m_bRunning && m_buffersSent == m_buffersCompleted ) - { - // We are running the audio queue but have become starved of buffers. - // Stop the audio queue so we force a restart of it. - AudioQueueStop( m_Queue, true ); - } - - // - // submit a few new sound blocks - // - // 44K sound support - while (((m_buffersSent - m_buffersCompleted) >> SAMPLE_16BIT_SHIFT) < cblocks) - { - int iBuf = m_buffersSent&BUFF_MASK; - - m_Buffers[iBuf]->mAudioDataByteSize = BUFFER_SIZE; - Q_memcpy( m_Buffers[iBuf]->mAudioData, (char *)m_sndBuffers + iBuf*BUFFER_SIZE, BUFFER_SIZE); - - // Queue the buffer for playback. - OSStatus err = AudioQueueEnqueueBuffer( m_Queue, m_Buffers[iBuf], 0, NULL); - if ( err != noErr) - { - DevMsg( "Failed to AudioQueueEnqueueBuffer output %d\n", (int)err ); - } - - m_buffersSent++; - } - - - if ( !m_bRunning ) - { - DevMsg( "Restarting sound playback\n" ); - m_bRunning = true; - AudioQueueStart( m_Queue, NULL); - } - -} - -int CAudioDeviceAudioQueue::GetOutputPosition( void ) -{ - int s = m_buffersSent * BUFFER_SIZE; - - s >>= SAMPLE_16BIT_SHIFT; - - s &= (DeviceSampleCount()-1); - - return s / DeviceChannels(); -} - - -//----------------------------------------------------------------------------- -// Pausing -//----------------------------------------------------------------------------- -void CAudioDeviceAudioQueue::Pause( void ) -{ - m_pauseCount++; - if (m_pauseCount == 1) - { - m_bRunning = false; - AudioQueueStop(m_Queue, true); - } -} - - -void CAudioDeviceAudioQueue::UnPause( void ) -{ - if ( m_pauseCount > 0 ) - { - m_pauseCount--; - } - - if ( m_pauseCount == 0 ) - { - m_bRunning = true; - AudioQueueStart( m_Queue, NULL); - } -} - -bool CAudioDeviceAudioQueue::IsActive( void ) -{ - return ( m_pauseCount == 0 ); -} - -float CAudioDeviceAudioQueue::MixDryVolume( void ) -{ - return 0; -} - - -bool CAudioDeviceAudioQueue::Should3DMix( void ) -{ - return false; -} - - -void CAudioDeviceAudioQueue::ClearBuffer( void ) -{ - if ( !m_sndBuffers ) - return; - - Q_memset( m_sndBuffers, 0x0, DeviceSampleCount() * DeviceSampleBytes() ); -} - -void CAudioDeviceAudioQueue::UpdateListener( const Vector& position, const Vector& forward, const Vector& right, const Vector& up ) -{ -} - - -bool CAudioDeviceAudioQueue::BIsPlaying() -{ - UInt32 isRunning; - UInt32 propSize = sizeof(isRunning); - - OSStatus result = AudioQueueGetProperty( m_Queue, kAudioQueueProperty_IsRunning, &isRunning, &propSize); - return isRunning != 0; -} - - -void CAudioDeviceAudioQueue::MixBegin( int sampleCount ) -{ - MIX_ClearAllPaintBuffers( sampleCount, false ); -} - - -void CAudioDeviceAudioQueue::MixUpsample( int sampleCount, int filtertype ) -{ - paintbuffer_t *ppaint = MIX_GetCurrentPaintbufferPtr(); - int ifilter = ppaint->ifilter; - - Assert (ifilter < CPAINTFILTERS); - - S_MixBufferUpsample2x( sampleCount, ppaint->pbuf, &(ppaint->fltmem[ifilter][0]), CPAINTFILTERMEM, filtertype ); - - ppaint->ifilter++; -} - -void CAudioDeviceAudioQueue::Mix8Mono( channel_t *pChannel, char *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress ) -{ - int volume[CCHANVOLUMES]; - paintbuffer_t *ppaint = MIX_GetCurrentPaintbufferPtr(); - - if (!MIX_ScaleChannelVolume( ppaint, pChannel, volume, 1)) - return; - - Mix8MonoWavtype( pChannel, ppaint->pbuf + outputOffset, volume, (byte *)pData, inputOffset, rateScaleFix, outCount ); -} - - -void CAudioDeviceAudioQueue::Mix8Stereo( channel_t *pChannel, char *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress ) -{ - int volume[CCHANVOLUMES]; - paintbuffer_t *ppaint = MIX_GetCurrentPaintbufferPtr(); - - if (!MIX_ScaleChannelVolume( ppaint, pChannel, volume, 2 )) - return; - - Mix8StereoWavtype( pChannel, ppaint->pbuf + outputOffset, volume, (byte *)pData, inputOffset, rateScaleFix, outCount ); -} - - -void CAudioDeviceAudioQueue::Mix16Mono( channel_t *pChannel, short *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress ) -{ - int volume[CCHANVOLUMES]; - paintbuffer_t *ppaint = MIX_GetCurrentPaintbufferPtr(); - - if (!MIX_ScaleChannelVolume( ppaint, pChannel, volume, 1 )) - return; - - Mix16MonoWavtype( pChannel, ppaint->pbuf + outputOffset, volume, pData, inputOffset, rateScaleFix, outCount ); -} - - -void CAudioDeviceAudioQueue::Mix16Stereo( channel_t *pChannel, short *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress ) -{ - int volume[CCHANVOLUMES]; - paintbuffer_t *ppaint = MIX_GetCurrentPaintbufferPtr(); - - if (!MIX_ScaleChannelVolume( ppaint, pChannel, volume, 2 )) - return; - - Mix16StereoWavtype( pChannel, ppaint->pbuf + outputOffset, volume, pData, inputOffset, rateScaleFix, outCount ); -} - - -void CAudioDeviceAudioQueue::ChannelReset( int entnum, int channelIndex, float distanceMod ) -{ -} - - -void CAudioDeviceAudioQueue::TransferSamples( int end ) -{ - int lpaintedtime = g_paintedtime; - int endtime = end; - - // resumes playback... - - if ( m_sndBuffers ) - { - S_TransferStereo16( m_sndBuffers, PAINTBUFFER, lpaintedtime, endtime ); - } -} - -void CAudioDeviceAudioQueue::SpatializeChannel( int volume[CCHANVOLUMES/2], int master_vol, const Vector& sourceDir, float gain, float mono ) -{ - VPROF("CAudioDeviceAudioQueue::SpatializeChannel"); - S_SpatializeChannel( volume, master_vol, &sourceDir, gain, mono ); -} - -void CAudioDeviceAudioQueue::StopAllSounds( void ) -{ - m_bSoundsShutdown = true; - m_bRunning = false; - AudioQueueStop(m_Queue, true); -} - - - -void CAudioDeviceAudioQueue::ApplyDSPEffects( int idsp, portable_samplepair_t *pbuffront, portable_samplepair_t *pbufrear, portable_samplepair_t *pbufcenter, int samplecount ) -{ - //SX_RoomFX( endtime, filter, timefx ); - DSP_Process( idsp, pbuffront, pbufrear, pbufcenter, samplecount ); -} - - -static uint32 GetOSXSpeakerConfig() -{ - return 2; -} - -static uint32 GetSpeakerConfigForSurroundMode( int surroundMode, const char **pConfigDesc ) -{ - uint32 newSpeakerConfig = 2; - *pConfigDesc = "stereo speaker"; - return newSpeakerConfig; -} - - - -void OnSndSurroundCvarChanged2( IConVar *pVar, const char *pOldString, float flOldValue ) -{ - // if the old value is -1, we're setting this from the detect routine for the first time - // no need to reset the device - if ( flOldValue == -1 ) - return; - - // get the user's previous speaker config - uint32 speaker_config = GetOSXSpeakerConfig(); - - // get the new config - uint32 newSpeakerConfig = 0; - const char *speakerConfigDesc = ""; - - ConVarRef var( pVar ); - newSpeakerConfig = GetSpeakerConfigForSurroundMode( var.GetInt(), &speakerConfigDesc ); - // make sure the config has changed - if (newSpeakerConfig == speaker_config) - return; - - // set new configuration - //SetWindowsSpeakerConfig(newSpeakerConfig); - - Msg("Speaker configuration has been changed to %s.\n", speakerConfigDesc); - - // restart sound system so it takes effect - //g_pSoundServices->RestartSoundSystem(); -} - -void OnSndSurroundLegacyChanged2( IConVar *pVar, const char *pOldString, float flOldValue ) -{ -} - - diff --git a/engine/audio/snd_win.cpp b/engine/audio/snd_win.cpp index 818312e3..33538009 100644 --- a/engine/audio/snd_win.cpp +++ b/engine/audio/snd_win.cpp @@ -11,10 +11,6 @@ #endif #ifdef OSX #include "snd_dev_openal.h" -#include "snd_dev_mac_audioqueue.h" - -ConVar snd_audioqueue( "snd_audioqueue", "1" ); - #endif // memdbgon must be the last include file in a .cpp file!!! @@ -94,11 +90,6 @@ IAudioDevice *IAudioDevice::AutoDetectInit( bool waveOnly ) pDevice = Audio_CreateWaveDevice(); } #elif defined(OSX) - if ( !CommandLine()->CheckParm( "-snd_openal" ) ) - { - DevMsg( "Using AudioQueue Interface\n" ); - pDevice = Audio_CreateMacAudioQueueDevice(); - } if ( !pDevice ) { DevMsg( "Using OpenAL Interface\n" ); diff --git a/engine/audio/voice.cpp b/engine/audio/voice.cpp index de5e0232..da7102e1 100644 --- a/engine/audio/voice.cpp +++ b/engine/audio/voice.cpp @@ -189,8 +189,6 @@ bool g_bUsingSteamVoice = false; #ifdef WIN32 extern IVoiceRecord* CreateVoiceRecord_DSound(int nSamplesPerSec); -#elif defined( OSX ) -extern IVoiceRecord* CreateVoiceRecord_AudioQueue(int sampleRate); #endif #ifdef POSIX @@ -643,13 +641,8 @@ bool Voice_Init( const char *pCodecName, int nSampleRate ) return false; // Get the voice input device. -#ifdef OSX - g_pVoiceRecord = CreateVoiceRecord_AudioQueue( Voice_SamplesPerSec() ); - if ( !g_pVoiceRecord ) - { - // Fall back to OpenAL - g_pVoiceRecord = CreateVoiceRecord_OpenAL( Voice_SamplesPerSec() ); - } +#if defined( OSX ) + g_pVoiceRecord = CreateVoiceRecord_OpenAL( Voice_SamplesPerSec() ); #elif defined( WIN32 ) g_pVoiceRecord = CreateVoiceRecord_DSound( Voice_SamplesPerSec() ); #elif defined( USE_SDL ) diff --git a/engine/audio/voice_record_mac_audioqueue.cpp b/engine/audio/voice_record_mac_audioqueue.cpp deleted file mode 100644 index e26ba4ee..00000000 --- a/engine/audio/voice_record_mac_audioqueue.cpp +++ /dev/null @@ -1,528 +0,0 @@ -//========= Copyright 1996-2009, Valve Corporation, All rights reserved. ============// -// -// Purpose: -// -// $NoKeywords: $ -// -//=============================================================================// -// This module implements the voice record and compression functions - -#include -#include -#include - -#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; - } -} diff --git a/engine/wscript b/engine/wscript index 54ffdc4e..fbf93f8e 100755 --- a/engine/wscript +++ b/engine/wscript @@ -341,8 +341,7 @@ def build(bld): if bld.env.DEST_OS == 'darwin': source += [ 'audio/snd_dev_openal.cpp', # [$OSXALL] - 'audio/snd_dev_mac_audioqueue.cpp',# [$OSXALL] - 'audio/voice_record_mac_audioqueue.cpp', #[$OSXALL] + 'audio/snd_dev_mac_audioqueue.cpp', # [$OSXALL] ] includes = [