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1 Commits
| Author | SHA1 | Date | |
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79c3bc86bd |
@ -1,599 +0,0 @@
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//========= Copyright Valve Corporation, All rights reserved. ============//
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//
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// Purpose:
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//
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//===========================================================================//
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#include "audio_pch.h"
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#include <AudioToolbox/AudioQueue.h>
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#include <AudioToolbox/AudioFile.h>
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#include <AudioToolbox/AudioFormat.h>
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// memdbgon must be the last include file in a .cpp file!!!
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#include "tier0/memdbgon.h"
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extern bool snd_firsttime;
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extern bool MIX_ScaleChannelVolume( paintbuffer_t *ppaint, channel_t *pChannel, int volume[CCHANVOLUMES], int mixchans );
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extern void S_SpatializeChannel( int volume[6], int master_vol, const Vector *psourceDir, float gain, float mono );
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#define NUM_BUFFERS_SOURCES 128
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#define BUFF_MASK (NUM_BUFFERS_SOURCES - 1 )
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#define BUFFER_SIZE 0x0400
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//-----------------------------------------------------------------------------
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//
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// NOTE: This only allows 16-bit, stereo wave out
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//
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//-----------------------------------------------------------------------------
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class CAudioDeviceAudioQueue : public CAudioDeviceBase
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{
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public:
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bool IsActive( void );
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bool Init( void );
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void Shutdown( void );
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void PaintEnd( void );
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int GetOutputPosition( void );
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void ChannelReset( int entnum, int channelIndex, float distanceMod );
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void Pause( void );
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void UnPause( void );
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float MixDryVolume( void );
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bool Should3DMix( void );
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void StopAllSounds( void );
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int PaintBegin( float mixAheadTime, int soundtime, int paintedtime );
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void ClearBuffer( void );
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void UpdateListener( const Vector& position, const Vector& forward, const Vector& right, const Vector& up );
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void MixBegin( int sampleCount );
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void MixUpsample( int sampleCount, int filtertype );
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void Mix8Mono( channel_t *pChannel, char *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress );
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void Mix8Stereo( channel_t *pChannel, char *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress );
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void Mix16Mono( channel_t *pChannel, short *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress );
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void Mix16Stereo( channel_t *pChannel, short *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress );
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void TransferSamples( int end );
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void SpatializeChannel( int volume[CCHANVOLUMES/2], int master_vol, const Vector& sourceDir, float gain, float mono);
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void ApplyDSPEffects( int idsp, portable_samplepair_t *pbuffront, portable_samplepair_t *pbufrear, portable_samplepair_t *pbufcenter, int samplecount );
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const char *DeviceName( void ) { return "AudioQueue"; }
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int DeviceChannels( void ) { return 2; }
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int DeviceSampleBits( void ) { return 16; }
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int DeviceSampleBytes( void ) { return 2; }
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int DeviceDmaSpeed( void ) { return SOUND_DMA_SPEED; }
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int DeviceSampleCount( void ) { return m_deviceSampleCount; }
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void BufferCompleted() { m_buffersCompleted++; }
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void SetRunning( bool bState ) { m_bRunning = bState; }
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private:
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void OpenWaveOut( void );
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void CloseWaveOut( void );
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bool ValidWaveOut( void ) const;
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bool BIsPlaying();
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AudioStreamBasicDescription m_DataFormat;
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AudioQueueRef m_Queue;
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AudioQueueBufferRef m_Buffers[NUM_BUFFERS_SOURCES];
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int m_SndBufSize;
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void *m_sndBuffers;
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CInterlockedInt m_deviceSampleCount;
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int m_buffersSent;
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int m_buffersCompleted;
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int m_pauseCount;
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bool m_bSoundsShutdown;
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bool m_bFailed;
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bool m_bRunning;
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};
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CAudioDeviceAudioQueue *wave = NULL;
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static void AudioCallback(void *pContext, AudioQueueRef pQueue, AudioQueueBufferRef pBuffer)
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{
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if ( wave )
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wave->BufferCompleted();
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}
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IAudioDevice *Audio_CreateMacAudioQueueDevice( void )
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{
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wave = new CAudioDeviceAudioQueue;
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if ( wave->Init() )
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return wave;
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delete wave;
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wave = NULL;
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return NULL;
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}
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void OnSndSurroundCvarChanged2( IConVar *pVar, const char *pOldString, float flOldValue );
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void OnSndSurroundLegacyChanged2( IConVar *pVar, const char *pOldString, float flOldValue );
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//-----------------------------------------------------------------------------
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// Init, shutdown
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//-----------------------------------------------------------------------------
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bool CAudioDeviceAudioQueue::Init( void )
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{
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m_SndBufSize = 0;
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m_sndBuffers = NULL;
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m_pauseCount = 0;
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m_bSurround = false;
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m_bSurroundCenter = false;
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m_bHeadphone = false;
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m_buffersSent = 0;
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m_buffersCompleted = 0;
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m_pauseCount = 0;
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m_bSoundsShutdown = false;
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m_bFailed = false;
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m_bRunning = false;
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m_Queue = NULL;
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static bool first = true;
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if ( first )
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{
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snd_surround.SetValue( 2 );
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snd_surround.InstallChangeCallback( &OnSndSurroundCvarChanged2 );
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snd_legacy_surround.InstallChangeCallback( &OnSndSurroundLegacyChanged2 );
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first = false;
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}
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OpenWaveOut();
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if ( snd_firsttime )
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{
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DevMsg( "Wave sound initialized\n" );
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}
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return ValidWaveOut() && !m_bFailed;
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}
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void CAudioDeviceAudioQueue::Shutdown( void )
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{
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CloseWaveOut();
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}
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//-----------------------------------------------------------------------------
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// WAV out device
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//-----------------------------------------------------------------------------
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inline bool CAudioDeviceAudioQueue::ValidWaveOut( void ) const
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{
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return m_sndBuffers != 0 && m_Queue;
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}
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//-----------------------------------------------------------------------------
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// called by the mac audioqueue code when we run out of playback buffers
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//-----------------------------------------------------------------------------
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void AudioQueueIsRunningCallback( void* inClientData, AudioQueueRef inAQ, AudioQueuePropertyID inID)
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{
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CAudioDeviceAudioQueue* audioqueue = (CAudioDeviceAudioQueue*)inClientData;
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UInt32 running = 0;
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UInt32 size;
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OSStatus err = AudioQueueGetProperty(inAQ, kAudioQueueProperty_IsRunning, &running, &size);
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audioqueue->SetRunning( running != 0 );
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//DevWarning( "AudioQueueStart %d\n", running );
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}
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//-----------------------------------------------------------------------------
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// Opens the windows wave out device
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//-----------------------------------------------------------------------------
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void CAudioDeviceAudioQueue::OpenWaveOut( void )
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{
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if ( m_Queue )
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return;
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m_buffersSent = 0;
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m_buffersCompleted = 0;
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m_DataFormat.mSampleRate = 44100;
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m_DataFormat.mFormatID = kAudioFormatLinearPCM;
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m_DataFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked;
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m_DataFormat.mBytesPerPacket = 4; // 16-bit samples * 2 channels
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m_DataFormat.mFramesPerPacket = 1;
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m_DataFormat.mBytesPerFrame = 4; // 16-bit samples * 2 channels
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m_DataFormat.mChannelsPerFrame = 2;
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m_DataFormat.mBitsPerChannel = 16;
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m_DataFormat.mReserved = 0;
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// Create the audio queue that will be used to manage the array of audio
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// buffers used to queue samples.
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OSStatus err = AudioQueueNewOutput(&m_DataFormat, AudioCallback, this, NULL, NULL, 0, &m_Queue);
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if ( err != noErr)
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{
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DevMsg( "Failed to create AudioQueue output %d\n", (int)err );
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m_bFailed = true;
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return;
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}
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for ( int i = 0; i < NUM_BUFFERS_SOURCES; ++i)
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{
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err = AudioQueueAllocateBuffer( m_Queue, BUFFER_SIZE,&(m_Buffers[i]));
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if ( err != noErr)
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{
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DevMsg( "Failed to AudioQueueAllocateBuffer output %d (%i)\n",(int)err,i );
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m_bFailed = true;
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}
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m_Buffers[i]->mAudioDataByteSize = BUFFER_SIZE;
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Q_memset( m_Buffers[i]->mAudioData, 0, BUFFER_SIZE );
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}
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err = AudioQueuePrime( m_Queue, 0, NULL);
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if ( err != noErr)
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{
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DevMsg( "Failed to create AudioQueue output %d\n", (int)err );
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m_bFailed = true;
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return;
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}
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AudioQueueSetParameter( m_Queue, kAudioQueueParam_Volume, 1.0);
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err = AudioQueueAddPropertyListener( m_Queue, kAudioQueueProperty_IsRunning, AudioQueueIsRunningCallback, this );
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if ( err != noErr)
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{
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DevMsg( "Failed to create AudioQueue output %d\n", (int)err );
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m_bFailed = true;
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return;
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}
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m_SndBufSize = NUM_BUFFERS_SOURCES*BUFFER_SIZE;
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m_deviceSampleCount = m_SndBufSize / DeviceSampleBytes();
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if ( !m_sndBuffers )
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{
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m_sndBuffers = malloc( m_SndBufSize );
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memset( m_sndBuffers, 0x0, m_SndBufSize );
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}
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}
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//-----------------------------------------------------------------------------
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// Closes the windows wave out device
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//-----------------------------------------------------------------------------
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void CAudioDeviceAudioQueue::CloseWaveOut( void )
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{
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if ( ValidWaveOut() )
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{
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AudioQueueStop(m_Queue, true);
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m_bRunning = false;
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AudioQueueRemovePropertyListener( m_Queue, kAudioQueueProperty_IsRunning, AudioQueueIsRunningCallback, this );
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for ( int i = 0; i < NUM_BUFFERS_SOURCES; i++ )
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AudioQueueFreeBuffer( m_Queue, m_Buffers[i]);
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AudioQueueDispose( m_Queue, true);
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m_Queue = NULL;
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}
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if ( m_sndBuffers )
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{
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free( m_sndBuffers );
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m_sndBuffers = NULL;
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}
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}
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//-----------------------------------------------------------------------------
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// Mixing setup
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//-----------------------------------------------------------------------------
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int CAudioDeviceAudioQueue::PaintBegin( float mixAheadTime, int soundtime, int paintedtime )
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{
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// soundtime - total samples that have been played out to hardware at dmaspeed
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// paintedtime - total samples that have been mixed at speed
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// endtime - target for samples in mixahead buffer at speed
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unsigned int endtime = soundtime + mixAheadTime * DeviceDmaSpeed();
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int samps = DeviceSampleCount() >> (DeviceChannels()-1);
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if ((int)(endtime - soundtime) > samps)
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endtime = soundtime + samps;
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if ((endtime - paintedtime) & 0x3)
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{
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// The difference between endtime and painted time should align on
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// boundaries of 4 samples. This is important when upsampling from 11khz -> 44khz.
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endtime -= (endtime - paintedtime) & 0x3;
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}
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return endtime;
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}
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//-----------------------------------------------------------------------------
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// Actually performs the mixing
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//-----------------------------------------------------------------------------
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void CAudioDeviceAudioQueue::PaintEnd( void )
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{
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int cblocks = 4 << 1;
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if ( m_bRunning && m_buffersSent == m_buffersCompleted )
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{
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// We are running the audio queue but have become starved of buffers.
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// Stop the audio queue so we force a restart of it.
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AudioQueueStop( m_Queue, true );
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}
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//
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// submit a few new sound blocks
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//
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// 44K sound support
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while (((m_buffersSent - m_buffersCompleted) >> SAMPLE_16BIT_SHIFT) < cblocks)
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{
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int iBuf = m_buffersSent&BUFF_MASK;
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m_Buffers[iBuf]->mAudioDataByteSize = BUFFER_SIZE;
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Q_memcpy( m_Buffers[iBuf]->mAudioData, (char *)m_sndBuffers + iBuf*BUFFER_SIZE, BUFFER_SIZE);
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// Queue the buffer for playback.
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OSStatus err = AudioQueueEnqueueBuffer( m_Queue, m_Buffers[iBuf], 0, NULL);
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if ( err != noErr)
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{
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DevMsg( "Failed to AudioQueueEnqueueBuffer output %d\n", (int)err );
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}
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m_buffersSent++;
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}
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if ( !m_bRunning )
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{
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DevMsg( "Restarting sound playback\n" );
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m_bRunning = true;
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AudioQueueStart( m_Queue, NULL);
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}
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}
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int CAudioDeviceAudioQueue::GetOutputPosition( void )
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{
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int s = m_buffersSent * BUFFER_SIZE;
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s >>= SAMPLE_16BIT_SHIFT;
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s &= (DeviceSampleCount()-1);
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return s / DeviceChannels();
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}
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//-----------------------------------------------------------------------------
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// Pausing
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//-----------------------------------------------------------------------------
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void CAudioDeviceAudioQueue::Pause( void )
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{
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m_pauseCount++;
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if (m_pauseCount == 1)
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{
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m_bRunning = false;
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AudioQueueStop(m_Queue, true);
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}
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}
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void CAudioDeviceAudioQueue::UnPause( void )
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{
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if ( m_pauseCount > 0 )
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{
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m_pauseCount--;
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}
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if ( m_pauseCount == 0 )
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{
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m_bRunning = true;
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AudioQueueStart( m_Queue, NULL);
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}
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}
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bool CAudioDeviceAudioQueue::IsActive( void )
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{
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return ( m_pauseCount == 0 );
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}
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float CAudioDeviceAudioQueue::MixDryVolume( void )
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{
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return 0;
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}
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bool CAudioDeviceAudioQueue::Should3DMix( void )
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{
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return false;
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}
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void CAudioDeviceAudioQueue::ClearBuffer( void )
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{
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if ( !m_sndBuffers )
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return;
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Q_memset( m_sndBuffers, 0x0, DeviceSampleCount() * DeviceSampleBytes() );
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}
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void CAudioDeviceAudioQueue::UpdateListener( const Vector& position, const Vector& forward, const Vector& right, const Vector& up )
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{
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}
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bool CAudioDeviceAudioQueue::BIsPlaying()
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{
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UInt32 isRunning;
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UInt32 propSize = sizeof(isRunning);
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OSStatus result = AudioQueueGetProperty( m_Queue, kAudioQueueProperty_IsRunning, &isRunning, &propSize);
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return isRunning != 0;
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}
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void CAudioDeviceAudioQueue::MixBegin( int sampleCount )
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{
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MIX_ClearAllPaintBuffers( sampleCount, false );
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}
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void CAudioDeviceAudioQueue::MixUpsample( int sampleCount, int filtertype )
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{
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paintbuffer_t *ppaint = MIX_GetCurrentPaintbufferPtr();
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int ifilter = ppaint->ifilter;
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Assert (ifilter < CPAINTFILTERS);
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S_MixBufferUpsample2x( sampleCount, ppaint->pbuf, &(ppaint->fltmem[ifilter][0]), CPAINTFILTERMEM, filtertype );
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ppaint->ifilter++;
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}
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void CAudioDeviceAudioQueue::Mix8Mono( channel_t *pChannel, char *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress )
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{
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int volume[CCHANVOLUMES];
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paintbuffer_t *ppaint = MIX_GetCurrentPaintbufferPtr();
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if (!MIX_ScaleChannelVolume( ppaint, pChannel, volume, 1))
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return;
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Mix8MonoWavtype( pChannel, ppaint->pbuf + outputOffset, volume, (byte *)pData, inputOffset, rateScaleFix, outCount );
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}
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void CAudioDeviceAudioQueue::Mix8Stereo( channel_t *pChannel, char *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress )
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{
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int volume[CCHANVOLUMES];
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paintbuffer_t *ppaint = MIX_GetCurrentPaintbufferPtr();
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if (!MIX_ScaleChannelVolume( ppaint, pChannel, volume, 2 ))
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return;
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Mix8StereoWavtype( pChannel, ppaint->pbuf + outputOffset, volume, (byte *)pData, inputOffset, rateScaleFix, outCount );
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}
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void CAudioDeviceAudioQueue::Mix16Mono( channel_t *pChannel, short *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress )
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{
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int volume[CCHANVOLUMES];
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paintbuffer_t *ppaint = MIX_GetCurrentPaintbufferPtr();
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if (!MIX_ScaleChannelVolume( ppaint, pChannel, volume, 1 ))
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return;
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Mix16MonoWavtype( pChannel, ppaint->pbuf + outputOffset, volume, pData, inputOffset, rateScaleFix, outCount );
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}
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void CAudioDeviceAudioQueue::Mix16Stereo( channel_t *pChannel, short *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress )
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{
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int volume[CCHANVOLUMES];
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paintbuffer_t *ppaint = MIX_GetCurrentPaintbufferPtr();
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||||
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if (!MIX_ScaleChannelVolume( ppaint, pChannel, volume, 2 ))
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return;
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|
||||
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 )
|
||||
{
|
||||
}
|
||||
|
||||
|
||||
@ -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" );
|
||||
|
||||
@ -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 )
|
||||
|
||||
@ -1,528 +0,0 @@
|
||||
//========= 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;
|
||||
}
|
||||
}
|
||||
@ -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 = [
|
||||
|
||||
Loading…
Reference in New Issue
Block a user