void AudioSource::AddAudioSegment(AudioSegment *newSegment, float curVolume)
{
for (UINT i=0; i<audioFilters.Num(); i++)
{
if (newSegment)
newSegment = audioFilters[i]->Process(newSegment);
}
if (newSegment)
{
MultiplyAudioBuffer(newSegment->audioData.Array(), newSegment->audioData.Num(), curVolume*sourceVolume);
audioSegments << newSegment;
}
}
/**************************************************
获取音频数据函数
参数：
lpData ：输入数据的内存
size：输入数据的长度
pts：输入数据的时间戳
***************************************************/
void CDemandMediaAudio::PushAudio(const void *lpData, unsigned int size, int64_t pts, IBaseVideo *Video, bool bCanPlay)
{
VideoSource *Source = dynamic_cast<VideoSource*>(Video);
if (!m_uBlockSize || !Source)
return;
if (m_sAudioParam.iChannel <= 2)
{
if (fVolume != 1.0f)
{
short *Tem = (short*)lpData;
for (int i = 0; i < size; i += 2)
{
long sVolume = Tem[i / 2];
sVolume *= fVolume;
if (sVolume > 0x7fff)
{
sVolume = 0x7fff;
}
else if (sVolume < -0x8000)
{
sVolume = -0x8000;
}
Tem[i / 2] = (short)sVolume;
}
}
Source->PlayCallBackAudio((LPBYTE)lpData, size);
}
else
{
UINT totalSamples = size * 8 / m_sAudioParam.iBitPerSample;
if (TemconvertBuffer.Num() < totalSamples)
TemconvertBuffer.SetSize(totalSamples);
OutputconvertBuffer.SetSize(totalSamples / m_sAudioParam.iChannel * 2);
if (m_sAudioParam.iBitPerSample == 8)
{
float *tempConvert = TemconvertBuffer.Array();
char *tempSByte = (char*)lpData;
while (totalSamples--)
{
*(tempConvert++) = float(*(tempSByte++)) / 127.0f;
}
}
else if (m_sAudioParam.iBitPerSample == 16)
{
float *tempConvert = TemconvertBuffer.Array();
short *tempShort = (short*)lpData;
while (totalSamples--)
{
*(tempConvert++) = float(*(tempShort++)) / 32767.0f;
}
}
else if (m_sAudioParam.iBitPerSample == 24)
{
float *tempConvert = TemconvertBuffer.Array();
BYTE *tempTriple = (BYTE*)lpData;
TripleToLong valOut;
while (totalSamples--)
{
TripleToLong &valIn = (TripleToLong&)tempTriple;
valOut.wVal = valIn.wVal;
valOut.tripleVal = valIn.tripleVal;
if (valOut.tripleVal > 0x7F)
valOut.lastByte = 0xFF;
*(tempConvert++) = float(double(valOut.val) / 8388607.0);
tempTriple += 3;
}
}
else if (m_sAudioParam.iBitPerSample == 32)
{
float *tempConvert = TemconvertBuffer.Array();
long *tempShort = (long*)lpData;
while (totalSamples--)
{
*(tempConvert++) = float(double(*(tempShort++)) / 2147483647.0);
}
}
float *inputTemp = TemconvertBuffer.Array();
float *outputTemp = OutputconvertBuffer.Array();
UINT numFloats = size * 8 / m_sAudioParam.iBitPerSample;
if (m_sAudioParam.iChannel == 3)
{
float *endTemp = inputTemp + numFloats;
while (inputTemp < endTemp)
{
float left = inputTemp[0];
float right = inputTemp[1];
// Drop LFE since we don't need it
//float lfe = inputTemp[2]*lowFreqMix;
*(outputTemp++) = left;
*(outputTemp++) = right;
inputTemp += 3;
}
}
else if (m_sAudioParam.iChannel == 4)
{
float *endTemp = inputTemp + numFloats;
while (inputTemp < endTemp)
{
float left = inputTemp[0];
float right = inputTemp[1];
float frontCenter = inputTemp[2];
float lowFreq = inputTemp[3];
*(outputTemp++) = left;
*(outputTemp++) = right;
inputTemp += 4;
}
}
else if (m_sAudioParam.iChannel == 5)
{
float *endTemp = inputTemp + numFloats;
while (inputTemp < endTemp)
{
float left = inputTemp[0];
float right = inputTemp[1];
// Skip LFE , we don't really need it.
//float lfe = inputTemp[2];
float rearLeft = inputTemp[3] * surroundMix4;
float rearRight = inputTemp[4] * surroundMix4;
// Same idea as with 5.1 downmix
*(outputTemp++) = (left + rearLeft) * attn4dotX;
*(outputTemp++) = (right + rearRight) * attn4dotX;
inputTemp += 5;
}
}
else if (m_sAudioParam.iChannel == 6)
{
float *endTemp = inputTemp + numFloats;
while (inputTemp < endTemp)
{
float left = inputTemp[0];
float right = inputTemp[1];
float center = inputTemp[2] * centerMix;
float rearLeft = inputTemp[4] * surroundMix;
float rearRight = inputTemp[5] * surroundMix;
*(outputTemp++) = (left + center + rearLeft) * attn5dot1;
*(outputTemp++) = (right + center + rearRight) * attn5dot1;
inputTemp += 6;
}
}
else if (m_sAudioParam.iChannel == 8)
{
float *endTemp = inputTemp + numFloats;
while (inputTemp < endTemp)
{
float left = inputTemp[0];
float right = inputTemp[1];
float center = inputTemp[2] * centerMix;
// Drop LFE since we don't need it
//float lowFreq = inputTemp[3]*lowFreqMix;
float rearLeft = inputTemp[4] * surroundMix;
float rearRight = inputTemp[5] * surroundMix;
// Drop SPEAKER_FRONT_LEFT_OF_CENTER , SPEAKER_FRONT_RIGHT_OF_CENTER
//float centerLeft = inputTemp[6];
//float centerRight = inputTemp[7];
// Downmix from 5.1 to stereo
*(outputTemp++) = (left + center + rearLeft) * attn5dot1;
*(outputTemp++) = (right + center + rearRight) * attn5dot1;
inputTemp += 8;
}
}
if (fVolume != 1.0f)
MultiplyAudioBuffer(OutputconvertBuffer.Array(), OutputconvertBuffer.Num(),fVolume);
Source->PlayCallBackAudio((LPBYTE)OutputconvertBuffer.Array(), OutputconvertBuffer.Num() * 4);
if (bCanPlay)
{
bool bPlayLive = false;
if (bLiveInstance)
{
AudioTimestamp audioTimestamp;
EnterCriticalSection(&sampleBufferLock);
sampleBuffer.AppendArray((BYTE *)(OutputconvertBuffer.Array()), OutputconvertBuffer.Num() * 4);
audioTimestamp.count = size / m_uBlockSize;
audioTimestamp.pts = pts;
//sampleBufferPts.push_back(audioTimestamp);
LeaveCriticalSection(&sampleBufferLock);
bPlayLive = m_bPlayPcmLive;
}
else
{
EnterCriticalSection(&sampleBufferLock);
sampleBuffer.RemoveRange(0, sampleBuffer.Num());
LeaveCriticalSection(&sampleBufferLock);
}
int Len = OutputconvertBuffer.Num();
char *OutBuffer;
CaculateVolume((LPVOID)OutputconvertBuffer.Array(), Len, (void**)&OutBuffer);
EnterCriticalSection(&sampleBufferLock);
if (m_pAudioWaveOut && (m_bPlayPcmLocal || bPlayLive))
{
m_pAudioWaveOut->push_pcm_data((char*)OutBuffer, Len * 4);
if (!bSameDevice && bProjector && m_pSecWaveOut)
m_pSecWaveOut->push_pcm_data((char*)OutBuffer, Len * 4);
}
else if (bProjector)
{
if (bSameDevice && m_pAudioWaveOut)
{
m_pAudioWaveOut->push_pcm_data((char*)OutBuffer, Len * 4);
}
else if (m_pSecWaveOut)
{
m_pSecWaveOut->push_pcm_data((char*)OutBuffer, Len * 4);
}
}
LeaveCriticalSection(&sampleBufferLock);
}
else
{
int Len = OutputconvertBuffer.Num();
char *OutBuffer;
CaculateVolume((LPVOID)OutputconvertBuffer.Array(), Len, (void**)&OutBuffer, true);
}
return;
}
if (bCanPlay)
{
bool bPlayLive = false;
size = size / m_uBlockSize;
if (bLiveInstance)
{
AudioTimestamp audioTimestamp;
EnterCriticalSection(&sampleBufferLock);
sampleBuffer.AppendArray(static_cast<const BYTE *>(lpData), size * m_uBlockSize);
audioTimestamp.count = size;
audioTimestamp.pts = pts;
//sampleBufferPts.push_back(audioTimestamp);
LeaveCriticalSection(&sampleBufferLock);
bPlayLive = m_bPlayPcmLive;
}
else
{
EnterCriticalSection(&sampleBufferLock);
sampleBuffer.RemoveRange(0, sampleBuffer.Num());
LeaveCriticalSection(&sampleBufferLock);
}
EnterCriticalSection(&sampleBufferLock);
int Len = size * m_uBlockSize;
char *OutBuffer;
CaculateVolume((LPVOID)lpData, Len, (void**)&OutBuffer);
if (m_pAudioWaveOut && (m_bPlayPcmLocal || bPlayLive))
{
m_pAudioWaveOut->push_pcm_data((char*)OutBuffer, Len);
if (!bSameDevice && bProjector && m_pSecWaveOut)
m_pSecWaveOut->push_pcm_data((char*)OutBuffer, Len);
}
else if (bProjector)
{
if (bSameDevice && m_pAudioWaveOut)
{
m_pAudioWaveOut->push_pcm_data((char*)OutBuffer, Len);
}
else if (m_pSecWaveOut)
{
m_pSecWaveOut->push_pcm_data((char*)OutBuffer, Len);
}
}
LeaveCriticalSection(&sampleBufferLock);
}
else
{
int Len = size;
char *OutBuffer;
CaculateVolume((LPVOID)lpData, Len, (void**)&OutBuffer,true);
}
}
UINT MMDeviceAudioSource::GetNextBuffer(float curVolume)
{
UINT captureSize = 0;
HRESULT err = mmCapture->GetNextPacketSize(&captureSize);
if(FAILED(err))
{
RUNONCE AppWarning(TEXT("MMDeviceAudioSource::GetBuffer: GetNextPacketSize failed"));
return NoAudioAvailable;
}
float *outputBuffer = NULL;
if(captureSize)
{
LPBYTE captureBuffer;
DWORD dwFlags = 0;
UINT numAudioFrames = 0;
UINT64 devPosition;
UINT64 qpcTimestamp;
err = mmCapture->GetBuffer(&captureBuffer, &numAudioFrames, &dwFlags, &devPosition, &qpcTimestamp);
if(FAILED(err))
{
RUNONCE AppWarning(TEXT("MMDeviceAudioSource::GetBuffer: GetBuffer failed"));
return NoAudioAvailable;
}
QWORD newTimestamp;
if(dwFlags & AUDCLNT_BUFFERFLAGS_TIMESTAMP_ERROR)
{
RUNONCE AppWarning(TEXT("MMDeviceAudioSource::GetBuffer: woa woa woa, getting timestamp errors from the audio subsystem. device = %s"), GetDeviceName().Array());
if(!bBrokenTimestamp)
newTimestamp = lastUsedTimestamp + numAudioFrames*1000/inputSamplesPerSec;
}
else
{
if(!bBrokenTimestamp)
newTimestamp = qpcTimestamp/10000;
/*UINT64 freq;
mmClock->GetFrequency(&freq);
Log(TEXT("position: %llu, numAudioFrames: %u, freq: %llu, newTimestamp: %llu, test: %llu"), devPosition, numAudioFrames, freq, newTimestamp, devPosition*8000/freq);*/
}
//have to do this crap to account for broken devices or device drivers. absolutely unbelievable.
if(!bFirstFrameReceived)
{
LARGE_INTEGER clockFreq;
QueryPerformanceFrequency(&clockFreq);
QWORD curTime = GetQPCTimeMS(clockFreq.QuadPart);
if(newTimestamp < (curTime-1000) || newTimestamp > (curTime+1000))
{
bBrokenTimestamp = true;
Log(TEXT("MMDeviceAudioSource::GetNextBuffer: Got bad audio timestamp offset %lld from device: '%s', timestamps for this device will be calculated. curTime: %llu, newTimestamp: %llu"), (LONGLONG)(newTimestamp - curTime), GetDeviceName().Array(), curTime, newTimestamp);
lastUsedTimestamp = newTimestamp = curTime;
}
else
lastUsedTimestamp = newTimestamp;
bFirstFrameReceived = true;
}
if(tempBuffer.Num() < numAudioFrames*2)
tempBuffer.SetSize(numAudioFrames*2);
outputBuffer = tempBuffer.Array();
float *tempOut = outputBuffer;
//------------------------------------------------------------
// channel upmix/downmix
if(inputChannels == 1)
{
UINT numFloats = numAudioFrames;
float *inputTemp = (float*)captureBuffer;
float *outputTemp = outputBuffer;
if(App->SSE2Available() && (UPARAM(inputTemp) & 0xF) == 0 && (UPARAM(outputTemp) & 0xF) == 0)
{
UINT alignedFloats = numFloats & 0xFFFFFFFC;
for(UINT i=0; i<alignedFloats; i += 4)
{
__m128 inVal = _mm_load_ps(inputTemp+i);
__m128 outVal1 = _mm_unpacklo_ps(inVal, inVal);
__m128 outVal2 = _mm_unpackhi_ps(inVal, inVal);
_mm_store_ps(outputTemp+(i*2), outVal1);
_mm_store_ps(outputTemp+(i*2)+4, outVal2);
}
numFloats -= alignedFloats;
inputTemp += alignedFloats;
outputTemp += alignedFloats*2;
}
while(numFloats--)
{
float inputVal = *inputTemp;
*(outputTemp++) = inputVal;
*(outputTemp++) = inputVal;
inputTemp++;
}
}
else if(inputChannels == 2) //straight up copy
{
if(App->SSE2Available())
SSECopy(outputBuffer, captureBuffer, numAudioFrames*2*sizeof(float));
else
mcpy(outputBuffer, captureBuffer, numAudioFrames*2*sizeof(float));
}
else
{
//todo: downmix optimization, also support for other speaker configurations than ones I can merely "think" of. ugh.
float *inputTemp = (float*)captureBuffer;
float *outputTemp = outputBuffer;
if(inputChannelMask == KSAUDIO_SPEAKER_QUAD)
{
UINT numFloats = numAudioFrames*4;
float *endTemp = inputTemp+numFloats;
while(inputTemp < endTemp)
{
float left = inputTemp[0];
float right = inputTemp[1];
float rear = (inputTemp[2]+inputTemp[3])*surroundMix;
*(outputTemp++) = left - rear;
*(outputTemp++) = right + rear;
inputTemp += 4;
}
}
else if(inputChannelMask == KSAUDIO_SPEAKER_2POINT1)
{
UINT numFloats = numAudioFrames*3;
float *endTemp = inputTemp+numFloats;
while(inputTemp < endTemp)
{
float left = inputTemp[0];
float right = inputTemp[1];
float lfe = inputTemp[2]*lowFreqMix;
*(outputTemp++) = left + lfe;
*(outputTemp++) = right + lfe;
inputTemp += 3;
}
}
else if(inputChannelMask == KSAUDIO_SPEAKER_4POINT1)
{
UINT numFloats = numAudioFrames*5;
float *endTemp = inputTemp+numFloats;
while(inputTemp < endTemp)
{
float left = inputTemp[0];
float right = inputTemp[1];
float lfe = inputTemp[2]*lowFreqMix;
float rear = (inputTemp[3]+inputTemp[4])*surroundMix;
*(outputTemp++) = left + lfe - rear;
*(outputTemp++) = right + lfe + rear;
inputTemp += 5;
}
}
else if(inputChannelMask == KSAUDIO_SPEAKER_SURROUND)
{
UINT numFloats = numAudioFrames*4;
float *endTemp = inputTemp+numFloats;
while(inputTemp < endTemp)
{
float left = inputTemp[0];
float right = inputTemp[1];
float center = inputTemp[2]*centerMix;
float rear = inputTemp[3]*(surroundMix*dbMinus3);
*(outputTemp++) = left + center - rear;
*(outputTemp++) = right + center + rear;
inputTemp += 4;
}
}
//don't think this will work for both
else if(inputChannelMask == KSAUDIO_SPEAKER_5POINT1)
{
UINT numFloats = numAudioFrames*6;
float *endTemp = inputTemp+numFloats;
while(inputTemp < endTemp)
{
float left = inputTemp[0];
float right = inputTemp[1];
float center = inputTemp[2]*centerMix;
float lowFreq = inputTemp[3]*lowFreqMix;
float rear = (inputTemp[4]+inputTemp[5])*surroundMix;
*(outputTemp++) = left + center + lowFreq - rear;
*(outputTemp++) = right + center + lowFreq + rear;
inputTemp += 6;
}
}
//todo ------------------
//not sure if my 5.1/7.1 downmixes are correct
else if(inputChannelMask == KSAUDIO_SPEAKER_5POINT1_SURROUND)
{
UINT numFloats = numAudioFrames*6;
float *endTemp = inputTemp+numFloats;
while(inputTemp < endTemp)
{
float left = inputTemp[0];
float right = inputTemp[1];
float center = inputTemp[2]*centerMix;
float lowFreq = inputTemp[3]*lowFreqMix;
float sideLeft = inputTemp[4]*dbMinus3;
float sideRight = inputTemp[5]*dbMinus3;
*(outputTemp++) = left + center + sideLeft + lowFreq;
*(outputTemp++) = right + center + sideRight + lowFreq;
inputTemp += 6;
}
}
else if(inputChannelMask == KSAUDIO_SPEAKER_7POINT1)
{
UINT numFloats = numAudioFrames*8;
float *endTemp = inputTemp+numFloats;
while(inputTemp < endTemp)
{
float left = inputTemp[0];
float right = inputTemp[1];
float center = inputTemp[2]*(centerMix*dbMinus3);
float lowFreq = inputTemp[3]*lowFreqMix;
float rear = (inputTemp[4]+inputTemp[5])*surroundMix;
float centerLeft = inputTemp[6]*dbMinus6;
float centerRight = inputTemp[7]*dbMinus6;
*(outputTemp++) = left + centerLeft + center + lowFreq - rear;
*(outputTemp++) = right + centerRight + center + lowFreq + rear;
inputTemp += 8;
}
}
else if(inputChannelMask == KSAUDIO_SPEAKER_7POINT1_SURROUND)
{
UINT numFloats = numAudioFrames*8;
float *endTemp = inputTemp+numFloats;
while(inputTemp < endTemp)
{
float left = inputTemp[0];
float right = inputTemp[1];
float center = inputTemp[2]*centerMix;
float lowFreq = inputTemp[3]*lowFreqMix;
float rear = (inputTemp[4]+inputTemp[5])*(surroundMix*dbMinus3);
float sideLeft = inputTemp[6]*dbMinus6;
float sideRight = inputTemp[7]*dbMinus6;
*(outputTemp++) = left + sideLeft + center + lowFreq - rear;
*(outputTemp++) = right + sideLeft + center + lowFreq + rear;
inputTemp += 8;
}
}
}
mmCapture->ReleaseBuffer(numAudioFrames);
//------------------------------------------------------------
// resample
if(bResample)
{
UINT frameAdjust = UINT((double(numAudioFrames) * resampleRatio) + 1.0);
UINT newFrameSize = frameAdjust*2;
if(tempResampleBuffer.Num() < newFrameSize)
tempResampleBuffer.SetSize(newFrameSize);
SRC_DATA data;
data.src_ratio = resampleRatio;
data.data_in = tempBuffer.Array();
data.input_frames = numAudioFrames;
data.data_out = tempResampleBuffer.Array();
data.output_frames = frameAdjust;
data.end_of_input = 0;
int err = src_process(resampler, &data);
if(err)
{
RUNONCE AppWarning(TEXT("Was unable to resample audio"));
return NoAudioAvailable;
}
if(data.input_frames_used != numAudioFrames)
{
RUNONCE AppWarning(TEXT("Failed to downsample buffer completely, which shouldn't actually happen because it should be using 10ms of samples"));
return NoAudioAvailable;
}
numAudioFrames = data.output_frames_gen;
}
//-----------------------------------------------------------------------------
// sort all audio frames into 10 millisecond increments (done because not all devices output in 10ms increments)
// NOTE: 0.457+ - instead of using the timestamps from windows, just compare and make sure it stays within a 100ms of their timestamps
float *newBuffer = (bResample) ? tempResampleBuffer.Array() : tempBuffer.Array();
if(storageBuffer.Num() == 0 && numAudioFrames == 441)
{
lastUsedTimestamp += 10;
if(!bBrokenTimestamp)
{
QWORD difVal = GetQWDif(newTimestamp, lastUsedTimestamp);
if(difVal > 70)
lastUsedTimestamp = newTimestamp;
}
if(lastUsedTimestamp > lastSentTimestamp)
{
QWORD adjustVal = (lastUsedTimestamp-lastSentTimestamp);
if(adjustVal < 10)
lastUsedTimestamp += 10-adjustVal;
AudioSegment &newSegment = *audioSegments.CreateNew();
newSegment.audioData.CopyArray(newBuffer, numAudioFrames*2);
newSegment.timestamp = lastUsedTimestamp;
MultiplyAudioBuffer(newSegment.audioData.Array(), numAudioFrames*2, curVolume);
lastSentTimestamp = lastUsedTimestamp;
}
}
else
{
UINT storedFrames = storageBuffer.Num();
storageBuffer.AppendArray(newBuffer, numAudioFrames*2);
if(storageBuffer.Num() >= (441*2))
{
lastUsedTimestamp += 10;
if(!bBrokenTimestamp)
{
QWORD difVal = GetQWDif(newTimestamp, lastUsedTimestamp);
if(difVal > 70)
lastUsedTimestamp = newTimestamp - (QWORD(storedFrames)/2*1000/44100);
}
//------------------------
// add new data
if(lastUsedTimestamp > lastSentTimestamp)
{
QWORD adjustVal = (lastUsedTimestamp-lastSentTimestamp);
if(adjustVal < 10)
lastUsedTimestamp += 10-adjustVal;
AudioSegment &newSegment = *audioSegments.CreateNew();
newSegment.audioData.CopyArray(storageBuffer.Array(), (441*2));
newSegment.timestamp = lastUsedTimestamp;
MultiplyAudioBuffer(newSegment.audioData.Array(), 441*2, curVolume);
storageBuffer.RemoveRange(0, (441*2));
}
//------------------------
// if still data pending (can happen)
while(storageBuffer.Num() >= (441*2))
{
lastUsedTimestamp += 10;
if(lastUsedTimestamp > lastSentTimestamp)
{
QWORD adjustVal = (lastUsedTimestamp-lastSentTimestamp);
if(adjustVal < 10)
lastUsedTimestamp += 10-adjustVal;
AudioSegment &newSegment = *audioSegments.CreateNew();
newSegment.audioData.CopyArray(storageBuffer.Array(), (441*2));
storageBuffer.RemoveRange(0, (441*2));
MultiplyAudioBuffer(newSegment.audioData.Array(), 441*2, curVolume);
newSegment.timestamp = lastUsedTimestamp;
lastSentTimestamp = lastUsedTimestamp;
}
}
}
}
//-----------------------------------------------------------------------------
return ContinueAudioRequest;
}
return NoAudioAvailable;
}
