int AnalyzeSamplesInterleaved(char * samples, long num_samples, int
num_channels){
double leftbuf[MAXSAMPLES], rightbuf[MAXSAMPLES];
long i;
long totSamples = num_samples;
long nSamples = num_samples;
signed short int * samp = (signed short int *)samples;
int result = GAIN_ANALYSIS_ERROR;
/* NOTES:
* leftbuf and rightbuf are arrays of doubles
* samp is a short (16-bit) integer
* inf is the input file
* totSamples is the total number of samples remaining in the input file
* MAXSAMPLES is the maximum number of samples to send to AnalyzeSamples at
* once
*/
while (totSamples > 0) {
if (totSamples > MAXSAMPLES)
nSamples = MAXSAMPLES;
else
nSamples = totSamples;
if (num_channels == 2) {
for (i = 0; i < nSamples; i++) {
leftbuf[i] = *samp++; /* default conversion from short to double */
rightbuf[i] = *samp++;
}
result = AnalyzeSamples(leftbuf, rightbuf, nSamples, 2);
if (result != GAIN_ANALYSIS_OK) return result;
} // end stereo
else { /* Just one channel (mono) */
for (i = 0; i < nSamples; i++) {
leftbuf[i] = *samp++;
}
result = AnalyzeSamples(leftbuf, NULL, nSamples, 1);
if (result != GAIN_ANALYSIS_OK) return result;
} // end just mono
totSamples -= nSamples;
} // end while
return result;
} /* AnalyzeSamplesInterleaved */
FLAC__bool grabbag__replaygain_analyze(const FLAC__int32 * const input[], FLAC__bool is_stereo, unsigned bps, unsigned samples)
{
/* using a small buffer improves data locality; we'd like it to fit easily in the dcache */
static Float_t lbuffer[2048], rbuffer[2048];
static const unsigned nbuffer = sizeof(lbuffer) / sizeof(lbuffer[0]);
FLAC__int32 block_peak = 0, s;
unsigned i, j;
FLAC__ASSERT(bps >= 4 && bps <= FLAC__REFERENCE_CODEC_MAX_BITS_PER_SAMPLE);
FLAC__ASSERT(FLAC__MIN_BITS_PER_SAMPLE == 4);
/*
* We use abs() on a FLAC__int32 which is undefined for the most negative value.
* If the reference codec ever handles 32bps we will have to write a special
* case here.
*/
FLAC__ASSERT(FLAC__REFERENCE_CODEC_MAX_BITS_PER_SAMPLE < 32);
if(bps == 16) {
if(is_stereo) {
j = 0;
while(samples > 0) {
const unsigned n = local_min(samples, nbuffer);
for(i = 0; i < n; i++, j++) {
s = input[0][j];
lbuffer[i] = (Float_t)s;
s = abs(s);
block_peak = local_max(block_peak, s);
s = input[1][j];
rbuffer[i] = (Float_t)s;
s = abs(s);
block_peak = local_max(block_peak, s);
}
samples -= n;
if(AnalyzeSamples(lbuffer, rbuffer, n, 2) != GAIN_ANALYSIS_OK)
return false;
}
}
else {
j = 0;
while(samples > 0) {
const unsigned n = local_min(samples, nbuffer);
for(i = 0; i < n; i++, j++) {
s = input[0][j];
lbuffer[i] = (Float_t)s;
s = abs(s);
block_peak = local_max(block_peak, s);
}
samples -= n;
if(AnalyzeSamples(lbuffer, 0, n, 1) != GAIN_ANALYSIS_OK)
return false;
}
}
}
else { /* bps must be < 32 according to above assertion */
const double scale = (
(bps > 16)?
(double)1. / (double)(1u << (bps - 16)) :
(double)(1u << (16 - bps))
);
if(is_stereo) {
j = 0;
while(samples > 0) {
const unsigned n = local_min(samples, nbuffer);
for(i = 0; i < n; i++, j++) {
s = input[0][j];
lbuffer[i] = (Float_t)(scale * (double)s);
s = abs(s);
block_peak = local_max(block_peak, s);
s = input[1][j];
rbuffer[i] = (Float_t)(scale * (double)s);
s = abs(s);
block_peak = local_max(block_peak, s);
}
samples -= n;
if(AnalyzeSamples(lbuffer, rbuffer, n, 2) != GAIN_ANALYSIS_OK)
return false;
}
}
else {
j = 0;
while(samples > 0) {
const unsigned n = local_min(samples, nbuffer);
for(i = 0; i < n; i++, j++) {
s = input[0][j];
lbuffer[i] = (Float_t)(scale * (double)s);
s = abs(s);
block_peak = local_max(block_peak, s);
}
samples -= n;
if(AnalyzeSamples(lbuffer, 0, n, 1) != GAIN_ANALYSIS_OK)
return false;
}
}
}
{
const double peak_scale = (double)(1u << (bps - 1));
double peak = (double)block_peak / peak_scale;
if(peak > title_peak_)
title_peak_ = peak;
if(peak > album_peak_)
album_peak_ = peak;
}
return true;
}
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;
}
static void
CalcReplayGain ( const char* filename, gain_info_t* G )
{
FILE* fp;
float buffl [NO];
float buffr [NO];
Int16_t buff [NO] [2];
size_t i;
size_t len;
size_t lastlen = 0;
unsigned int max = 0;
float level;
float mult;
if ((fp = pipeopen ( "mppdec --silent --scale 0.5 --gain 0 --raw - - < #", filename)) == NULL) {
stderr_printf ( "Can't decode '%s'\n", filename );
exit (9);
}
memset ( buff, 0, sizeof(buff) );
G->Silence = 0;
lastlen = len = fread (buff, 4, NO, fp);
for ( i = 0; i < len; i++ ) {
buffl [i] = 2. * buff [i] [0];
buffr [i] = 2. * buff [i] [1];
if ( abs(buff[i][0]) > max ) max = abs(buff[i][0]);
if ( abs(buff[i][1]) > max ) max = abs(buff[i][1]);
}
AnalyzeSamples ( buffl, buffr, len, 2 );
level = 0.;
mult = 1.;
for ( i = 0; i < len; i++ ) {
level += mult * (buff [i] [0] * buff [i] [0] + buff [i] [1] * buff [i] [1]);
mult *= 0.95;
}
level = 2*sqrt(level * 0.05);
if ( level > LEVEL_THR )
G->Silence |= 2;
sh ( NULL, level );
while (( len = fread (buff, 4, NO, fp) ) > 0 ) {
lastlen = len;
for ( i = 0; i < len; i++ ) {
buffl [i] = 2. * buff [i] [0];
buffr [i] = 2. * buff [i] [1];
if ( abs(buff[i][0]) > max ) max = abs(buff[i][0]);
if ( abs(buff[i][1]) > max ) max = abs(buff[i][1]);
}
AnalyzeSamples ( buffl, buffr, len, 2 );
}
level = 0.;
mult = 1.;
for ( i = 1; i <= NO; i++ ) {
int idx = (lastlen + NO - i) % NO;
level += mult * (buff [idx] [0] * buff [idx] [0] + buff [idx] [1] * buff [idx] [1]);
mult *= 0.95;
}
level = 2*sqrt(level * 0.05);
if ( level > LEVEL_THR )
G->Silence |= 1;
sh(filename,level);
PCLOSE (fp);
#if 0
GetTitleDynamics ();
#endif
G -> FileName = filename;
G -> TitleGain = GetTitleGain ();
G -> TitlePeak = 2 * max + 1;
G -> AlbumGain = GetAlbumGain ();
G -> AlbumPeak = G->AlbumPeak < G->TitlePeak ? G->TitlePeak : G->AlbumPeak;
return;
}
