void WaveformViewDemo::GetWaveformOverview(WaveformOverview* data)
{
AudioBufferList *bufferList = &mFetchingBufferList->GetModifiableBufferList();
UInt32 num = data->mNumDataPoints;
SampleTime t = (SampleTime) data->mFetchStamp.mSampleTime;
mAudioBuffer->Fetch(bufferList, num, t, false);
Float32* b = (Float32*) bufferList->mBuffers[data->mChannel].mData;
memcpy(data->mOverview, b, num*sizeof(Float32));
vDSP_maxmgv(b, 1, &(data->mMax), num);
vDSP_minmgv (b, 1,&(data->mMin), num);
#pragma warning we are pulling all the data but only need a certain channel
data->mFetchStamp.mSampleTime += (Float64) num;
}
void vmaxmgv(const float* sourceP,
int sourceStride,
float* maxP,
size_t framesToProcess) {
vDSP_maxmgv(sourceP, sourceStride, maxP, framesToProcess);
}
bool SFB::Audio::ReplayGainAnalyzer::AnalyzeURL(CFURLRef url, CFErrorRef *error)
{
if(nullptr == url)
return false;
auto decoder = Decoder::CreateDecoderForURL(url, error);
if(!decoder || !decoder->Open(error))
return false;
AudioStreamBasicDescription inputFormat = decoder->GetFormat();
// Higher sampling rates aren't natively supported but are handled via resampling
int32_t decoderSampleRate = (int32_t)inputFormat.mSampleRate;
bool validSampleRate = EvenMultipleSampleRateIsSupported(decoderSampleRate);
if(!validSampleRate) {
if(error) {
SFB::CFString description = CFCopyLocalizedString(CFSTR("The file “%@” does not contain audio at a supported sample rate."), "");
SFB::CFString failureReason = CFCopyLocalizedString(CFSTR("Only sample rates of 8.0 KHz, 11.025 KHz, 12.0 KHz, 16.0 KHz, 22.05 KHz, 24.0 KHz, 32.0 KHz, 44.1 KHz, 48 KHz and multiples are supported."), "");
SFB::CFString recoverySuggestion = CFCopyLocalizedString(CFSTR("The file's extension may not match the file's type."), "");
*error = CreateErrorForURL(ReplayGainAnalyzer::ErrorDomain, ReplayGainAnalyzer::FileFormatNotSupportedError, description, url, failureReason, recoverySuggestion);
}
return false;
}
Float64 replayGainSampleRate = GetBestReplayGainSampleRateForSampleRate(decoderSampleRate);
if(!(1 == inputFormat.mChannelsPerFrame || 2 == inputFormat.mChannelsPerFrame)) {
if(error) {
SFB::CFString description = CFCopyLocalizedString(CFSTR("The file “%@” does not contain mono or stereo audio."), "");
SFB::CFString failureReason = CFCopyLocalizedString(CFSTR("Only mono or stereo files supported"), "");
SFB::CFString recoverySuggestion = CFCopyLocalizedString(CFSTR("The file's extension may not match the file's type."), "");
*error = CreateErrorForURL(ReplayGainAnalyzer::ErrorDomain, ReplayGainAnalyzer::FileFormatNotSupportedError, description, url, failureReason, recoverySuggestion);
}
return false;
}
AudioStreamBasicDescription outputFormat = {
.mFormatID = kAudioFormatLinearPCM,
.mFormatFlags = kAudioFormatFlagsNativeFloatPacked | kAudioFormatFlagIsNonInterleaved,
.mReserved = 0,
.mSampleRate = replayGainSampleRate,
.mChannelsPerFrame = inputFormat.mChannelsPerFrame,
.mBitsPerChannel = 32,
.mBytesPerPacket = 4,
.mBytesPerFrame = 4,
.mFramesPerPacket = 1
};
if(!SetSampleRate((int32_t)outputFormat.mSampleRate)) {
if(error) {
SFB::CFString description = CFCopyLocalizedString(CFSTR("The file “%@” does not contain audio at a supported sample rate."), "");
SFB::CFString failureReason = CFCopyLocalizedString(CFSTR("Only sample rates of 8.0 KHz, 11.025 KHz, 12.0 KHz, 16.0 KHz, 22.05 KHz, 24.0 KHz, 32.0 KHz, 44.1 KHz, 48 KHz and multiples are supported."), "");
SFB::CFString recoverySuggestion = CFCopyLocalizedString(CFSTR("The file's extension may not match the file's type."), "");
*error = CreateErrorForURL(ReplayGainAnalyzer::ErrorDomain, ReplayGainAnalyzer::FileFormatNotSupportedError, description, url, failureReason, recoverySuggestion);
}
return false;
}
// Converter takes ownership of decoder
Converter converter(std::move(decoder), outputFormat);
if(!converter.Open(error))
return false;
const UInt32 bufferSizeFrames = 512;
BufferList outputBuffer(outputFormat, bufferSizeFrames);
bool isStereo = (2 == outputFormat.mChannelsPerFrame);
for(;;) {
UInt32 frameCount = converter.ConvertAudio(outputBuffer, bufferSizeFrames);
if(0 == frameCount)
break;
// Find the peak sample magnitude
float lpeak, rpeak;
vDSP_maxmgv((const float *)outputBuffer->mBuffers[0].mData, 1, &lpeak, frameCount);
if(isStereo) {
vDSP_maxmgv((const float *)outputBuffer->mBuffers[1].mData, 1, &rpeak, frameCount);
priv->trackPeak = std::max(priv->trackPeak, std::max(lpeak, rpeak));
}
else
priv->trackPeak = std::max(priv->trackPeak, lpeak);
// The replay gain analyzer expects 16-bit sample size passed as floats
const float scale = 1u << 15;
vDSP_vsmul((const float *)outputBuffer->mBuffers[0].mData, 1, &scale, (float *)outputBuffer->mBuffers[0].mData, 1, frameCount);
if(isStereo) {
vDSP_vsmul((const float *)outputBuffer->mBuffers[1].mData, 1, &scale, (float *)outputBuffer->mBuffers[1].mData, 1, frameCount);
AnalyzeSamples((const float *)outputBuffer->mBuffers[0].mData, (const float *)outputBuffer->mBuffers[1].mData, frameCount, true);
}
else
AnalyzeSamples((const float *)outputBuffer->mBuffers[0].mData, nullptr, frameCount, false);
}
priv->albumPeak = std::max(priv->albumPeak, priv->trackPeak);
return true;
}
bool SFB::Audio::ReplayGainAnalyzer::GetTrackGain(float& trackGain)
{
if(!analyzeResult(priv->A, sizeof(priv->A) / sizeof(*(priv->A)), trackGain))
return false;
for(uint32_t i = 0; i < sizeof(priv->A) / sizeof(*(priv->A)); ++i) {
priv->B[i] += priv->A[i];
priv->A[i] = 0;
}
priv->Zero();
priv->totsamp = 0;
priv->lsum = priv->rsum = 0.;
return true;
}
bool SFB::Audio::ReplayGainAnalyzer::GetTrackPeak(float& trackPeak)
{
trackPeak = priv->trackPeak;
priv->trackPeak = 0.;
return true;
}
