void AUDSimpleAudioChangePlayback(const u32 *playMaskArray,u32 numMasks)
{
u32 i,b;
BOOL old = OSDisableInterrupts();
audioPlayMaskArray = playMaskArray;
audioNumPlayMasks = numMasks;
// Any playback mask specified?
if (audioNumPlayMasks != 0)
{
// Yes. Count the active voices...
audioNumActiveVoices = 0;
for(i=0; i<numMasks; i++)
{
for(b=0; b<32; b++)
{
if (audioPlayMaskArray[i] & (1<<b))
audioNumActiveVoices++;
}
}
// So, did we have too much?
ASSERT(audioNumActiveVoices <= 2);
}
else
audioNumActiveVoices = audioReadBufferNumChannels; // Make all active...
OSRestoreInterrupts(old);
}
/**
* Release an Uninterruptible Spin Lock.
*
* Interrupts will be restored to their previous state before
* the lock was acquired.
*
* \return Returns TRUE if the lock was released.
*/
BOOL
OSUninterruptibleSpinLock_Release(OSSpinLock *spinlock)
{
if (spinReleaseLock(spinlock)) {
OSRestoreInterrupts(spinlock->restoreInterruptState);
}
return TRUE;
}
/*!
******************************************************************************
* \brief
* Stop audio playback without shutting down AI etc.
*
******************************************************************************
*/
void AUDSimpleAudioReset(void)
{
BOOL old;
old = OSDisableInterrupts();
audioReadBufferWritePos = 0;
audioReadBufferReadPos = 0;
audioPlayBufferEnabled = FALSE;
OSRestoreInterrupts(old);
}
/*!
******************************************************************************
* \brief
* Shutdown audio decoder and free resources
*
******************************************************************************
*/
void AUDSimpleExitAudioDecoder(void)
{
// Yes. Unregister callback & stop AI DMA
BOOL old = OSDisableInterrupts();
// Register our default AX callback.
AXRegisterCallback( Sfx::AXUserCBack );
AXSetVoiceState(axVoice[0],AX_PB_STATE_STOP);
AXSetVoiceState(axVoice[1],AX_PB_STATE_STOP);
AXFreeVoice(axVoice[0]);
AXFreeVoice(axVoice[1]);
axVoice[0] = axVoice[1] = NULL;
OSRestoreInterrupts(old);
// Free allocated resources...
audio_player.cbFree(audioPlayBuffer[0]);
audio_player.cbFree(audioReadBuffer);
}
/*!
******************************************************************************
* \brief
* Main audio data decode function.
*
* This function will be used as a callback from the VIDAudioDecode
* function or is called directely in case of PCM or ADPCM.
*
* \param numChannels
* Number of channels present in the sample array
*
* \param samples
* Array of s16 pointers to the sample data
*
* \param sampleNum
* Number of samples in the array. All arrays have the same amount
* of sample data
*
* \param userData
* Some user data
*
* \return
* FALSE if data could not be interpreted properly
*
******************************************************************************
*/
static BOOL audioDecode(u32 numChannels, const s16 **samples, u32 sampleNum, void* _UNUSED(userData))
{
u32 freeSamples;
u32 sampleSize;
u32 len1;
BOOL old;
// we can only play mono or stereo!
ASSERT(numChannels <= 2);
// Disable IRQs. We must make sure we don't get interrupted by the AI callback.
old = OSDisableInterrupts();
// Did the video decoder just jump back to the beginning of the stream?
if (audio_player.readBuffer[audio_player.decodeIndex].frameNumber < audio_player.lastDecodedFrame)
{
// Yes! We have to reset the internal read buffer and disable any audio output
// until we got new video data to display...
//
// Note: we have to reset our buffers because the stream contains more audio data than
// neccessary to cover a single video frame within the first few frames to accumulate
// some safety buffer. If the stream would just be allowed to loop we would get an
// overflow (unless we used up the extra buffer due to read / decode delays) after a few
// loops...
//
AUDSimpleAudioReset();
}
// Calculate the read buffer's sample size
sampleSize = sizeof(s16) * audioReadBufferNumChannels;
// How many samples could we put into the buffer?
if (audioReadBufferWritePos >= audioReadBufferReadPos)
{
freeSamples = audioReadBufferNumSamples - (audioReadBufferWritePos - audioReadBufferReadPos);
if( freeSamples < sampleNum )
{
OSRestoreInterrupts(old);
#ifndef FINAL
OSReport("*** audioDecode: overflow case 1\n");
#endif
return FALSE; // overflow!
}
// We might have a two buffer update to do. Check for it...
if ((len1 = (audioReadBufferNumSamples - audioReadBufferWritePos)) >= sampleNum)
{
// No. We got ourselfs a nice, simple single buffer update.
writeChannelData((s16 *)((u32)audioReadBuffer + audioReadBufferWritePos * sampleSize),numChannels,samples,0,sampleNum);
}
else
{
// Dual buffer case
writeChannelData((s16 *)((u32)audioReadBuffer + audioReadBufferWritePos * sampleSize),numChannels,samples,0,len1);
writeChannelData((s16 *)audioReadBuffer,numChannels,samples,len1,sampleNum-len1);
}
}
else
{
freeSamples = audioReadBufferReadPos - audioReadBufferWritePos;
if (freeSamples < sampleNum)
{
OSRestoreInterrupts(old);
#ifndef FINAL
OSReport("*** audioDecode: overflow case 2\n");
#endif
return FALSE; // overflow!
}
// We're save to assume to have a single buffer update in any case...
writeChannelData((s16 *)((u32)audioReadBuffer + audioReadBufferWritePos * sampleSize),numChannels,samples,0,sampleNum);
}
// Advance write position...
audioReadBufferWritePos += sampleNum;
if (audioReadBufferWritePos >= audioReadBufferNumSamples)
audioReadBufferWritePos -= audioReadBufferNumSamples;
// We're done with all critical stuff. IRQs may be enabled again...
OSRestoreInterrupts(old);
return TRUE;
}
/*!
******************************************************************************
* \brief
* Initialize audio decoder
*
* This function allocates all neccessary memory for the audio processing
* and sets the audio decoder into an idle state, waiting for first data.
* A file must be opened with the VIDSimplePlayer before calling this
* function.
*
* \return
* FALSE if any problem was detected
*
******************************************************************************
*/
BOOL AUDSimpleInitAudioDecoder(void)
{
u32 i, ratio;
BOOL old;
AXPBMIX axMix[2];
AXPBVE axVE;
AXPBSRC axSRC;
AXPBADDR axAddr;
AXPBADPCM axADPCM;
// Calculate buffer size to allocate proper memry to keep a bit of "extra" audio data around...
audioReadBufferNumSamples = (u32)((f32)AUD_AUDIO_READAHEADFRAMES * audio_player.audioInfo.vaud.frq);
audioReadBufferNumChannels = audio_player.audioInfo.vaud.numChannels <= 2 ? audio_player.audioInfo.vaud.numChannels : 2;
// Allocate read buffer
audioReadBuffer = audio_player.cbAlloc(audioReadBufferNumSamples * sizeof(s16) * audio_player.audioInfo.vaud.numChannels);
if (audioReadBuffer == NULL)
return FALSE; // error
// Reset ring buffer
audioReadBufferReadPos = audioReadBufferWritePos = 0;
// What frquency is best?
audioPlayBufferFrq = audio_player.audioInfo.vaud.frq;
// Allocate AI playback buffer
audioPlayBuffer[0] = audio_player.cbAlloc(2 * sizeof(s16) * AUD_AUDIO_AIBUFFERSAMPLES * AUD_AUDIO_NUMAIBUFFERS);
if (audioPlayBuffer[0] == NULL)
return FALSE; // error
for(i=1; i<AUD_AUDIO_NUMAIBUFFERS; i++)
audioPlayBuffer[i] = (void *)((u32)audioPlayBuffer[i - 1] + (2 * sizeof(s16) * AUD_AUDIO_AIBUFFERSAMPLES));
// Reset buffer index
audioPlayBufferWriteIndex = 0;
// We disable AI output for now (logically)
audioPlayBufferEnabled = FALSE;
// We assume to playback all we get by default
audioPlayMaskArray = NULL;
audioNumPlayMasks = 0;
audioNumActiveVoices = 2;
// Clear out AI buffers to avoid any noise what so ever
memset(audioPlayBuffer[0],0,2 * sizeof(s16) * AUD_AUDIO_AIBUFFERSAMPLES * AUD_AUDIO_NUMAIBUFFERS);
DCFlushRange(audioPlayBuffer[0],2 * sizeof(s16) * AUD_AUDIO_AIBUFFERSAMPLES * AUD_AUDIO_NUMAIBUFFERS);
// Init GCN audio system
old = OSDisableInterrupts();
axVoice[0] = AXAcquireVoice(AX_PRIORITY_NODROP,NULL,0);
ASSERT(axVoice[0] != NULL);
axVoice[1] = AXAcquireVoice(AX_PRIORITY_NODROP,NULL,0);
ASSERT(axVoice[1] != NULL);
memset(&axMix[0],0,sizeof(axMix[0]));
axMix[0].vL = 0x7FFF;
memset(&axMix[1],0,sizeof(axMix[1]));
axMix[1].vR = 0x7FFF;
AXSetVoiceMix(axVoice[0],&axMix[0]);
AXSetVoiceMix(axVoice[1],&axMix[1]);
axVE.currentDelta = 0;
axVE.currentVolume = 0x7FFF;
AXSetVoiceVe(axVoice[0],&axVE);
AXSetVoiceVe(axVoice[1],&axVE);
memset(&axSRC,0,sizeof(AXPBSRC));
ratio = (u32)(65536.0f * (f32)audioPlayBufferFrq / (f32)AX_IN_SAMPLES_PER_SEC);
axSRC.ratioHi = (u16)(ratio >> 16);
axSRC.ratioLo = (u16)ratio;
AXSetVoiceSrcType(axVoice[0],AX_SRC_TYPE_4TAP_16K);
AXSetVoiceSrc(axVoice[0],&axSRC);
AXSetVoiceSrcType(axVoice[1],AX_SRC_TYPE_4TAP_16K);
AXSetVoiceSrc(axVoice[1],&axSRC);
*(u32 *)&axAddr.currentAddressHi = AX_ARAM_LEFT_CHANNEL;
*(u32 *)&axAddr.loopAddressHi = AX_ARAM_LEFT_CHANNEL;
*(u32 *)&axAddr.endAddressHi = AX_ARAM_LEFT_CHANNEL + AX_ARAM_BUFFER_SIZE - 1;
axAddr.format = AX_PB_FORMAT_PCM16;
axAddr.loopFlag = AXPBADDR_LOOP_ON;
AXSetVoiceAddr(axVoice[0],&axAddr);
*(u32 *)&axAddr.currentAddressHi = AX_ARAM_RIGHT_CHANNEL;
*(u32 *)&axAddr.loopAddressHi = AX_ARAM_RIGHT_CHANNEL;
*(u32 *)&axAddr.endAddressHi = AX_ARAM_RIGHT_CHANNEL + AX_ARAM_BUFFER_SIZE - 1;
AXSetVoiceAddr(axVoice[1],&axAddr);
memset(&axADPCM,0,sizeof(axADPCM));
axADPCM.gain = 0x0800;
AXSetVoiceAdpcm(axVoice[0],&axADPCM);
AXSetVoiceAdpcm(axVoice[1],&axADPCM);
AXSetVoiceType(axVoice[0],AX_PB_TYPE_STREAM);
AXSetVoiceType(axVoice[1],AX_PB_TYPE_STREAM);
AXRegisterCallback( AXCallback );
axLastAddr = AX_ARAM_LEFT_CHANNEL;
axPlayedSamples = AUD_AUDIO_AIBUFFERSAMPLES * AUD_AUDIO_NUMAIBUFFERS;
axPlayedSamplesTotal = 0;
axPhase = AX_PHASE_STARTUP;
// All is setup for the voices. We'll start them inside the AX callback as soon as we got data in the ARAM buffers
OSRestoreInterrupts(old);
return TRUE;
}
/*!
******************************************************************************
* \brief
* Decode all frame data
*
* This function operates on the full frame input data. It forwards this
* data to the required decoder.
*
******************************************************************************
*/
BOOL AUDSimpleDecode(void)
{
BOOL enabled;
u8* chunkStart;
u32 chunkSize;
u32 frameSize;
if( audio_player.readBuffer[audio_player.decodeIndex].valid )
{
// ptr to our (pre-) loaded data INSIDE (!) 'FRAM' chunk
// (in other words, the 'FRAM' chunk itself is not visible here)
chunkStart = audio_player.readBuffer[audio_player.decodeIndex].ptr;
// usually, we read additional 32 bytes for getting info about the NEXT chunk.
// We only deal with the actual 'FRAM' chunk data here and adjust the size by 32 bytes.
frameSize = audio_player.readBuffer[audio_player.decodeIndex].size - 32;
// loop across ALL chunks inside 'FRAM'
while(frameSize >= 32)
{
chunkSize = VID_CHUNK_LEN(chunkStart);
if( VID_CHUNK_ID(chunkStart) == VID_FCC('A','U','D','D') )
{
// Get the data to the audio system...
if(! AUDSimpleAudioDecode(chunkStart + VID_CHUNK_HEADER_SIZE, chunkSize - VID_CHUNK_HEADER_SIZE))
{
#ifdef _DEBUG
OSReport("*** AUDSimpleAudioDecode failed!\n");
#endif
}
}
#ifdef _DEBUG
else
{
OSReport("*** AUDSimpleDecode: unknown chunk type!\n");
}
#endif
// goto next chunk
chunkStart += chunkSize;
frameSize -= chunkSize;
}
audio_player.lastDecodedFrame = audio_player.readBuffer[audio_player.decodeIndex].frameNumber;
audio_player.readBuffer[audio_player.decodeIndex].valid = FALSE;
audio_player.decodeIndex = (audio_player.decodeIndex + 1) % AUD_NUM_READ_BUFFERS;
// check if loading is still running
enabled = OSDisableInterrupts();
if (!audio_player.readBuffer[audio_player.readIndex].valid && !audio_player.asyncDvdRunning)
ReadFrameAsync();
OSRestoreInterrupts(enabled);
return TRUE;
}
#ifdef _DEBUG
OSReport("*** AUDSimpleDecode: No valid decode buffer found (?).\n");
#endif
return FALSE;
}