osb/source/core/StarAudio.cpp
2024-03-25 01:57:55 +11:00

583 lines
18 KiB
C++

// Fixes unused variable warning
#define OV_EXCLUDE_STATIC_CALLBACKS
#include "vorbis/codec.h"
#include "vorbis/vorbisfile.h"
#include "StarAudio.hpp"
#include "StarBuffer.hpp"
#include "StarFile.hpp"
#include "StarFormat.hpp"
#include "StarLogging.hpp"
#include "StarDataStreamDevices.hpp"
namespace Star {
float const DefaultPerceptualRangeDb = 40.f;
float const DefaultPerceptualBoostRangeDb = 6.f;
// https://github.com/discord/perceptual
float perceptualToAmplitude(float perceptual, float normalizedMax, float range, float boostRange) {
if (perceptual == 0.f) return 0.f;
float dB = perceptual > normalizedMax
? ((perceptual - normalizedMax) / normalizedMax) * boostRange
: (perceptual / normalizedMax) * range - range;
return normalizedMax * pow(10.f, dB / 20.f);
}
float amplitudeToPerceptual(float amp, float normalizedMax, float range, float boostRange) {
if (amp == 0.f) return 0.f;
float const dB = 20.f * log10(amp / normalizedMax);
float perceptual = dB > 0.f
? dB / boostRange + 1
: (range + dB) / range;
return normalizedMax * perceptual;
}
namespace {
struct WaveData {
ByteArrayPtr byteArray;
unsigned channels;
unsigned sampleRate;
};
template <typename T>
T readLEType(IODevicePtr const& device) {
T t;
device->readFull((char*)&t, sizeof(t));
fromByteOrder(ByteOrder::LittleEndian, (char*)&t, sizeof(t));
return t;
}
bool isUncompressed(IODevicePtr device) {
const size_t sigLength = 4;
unique_ptr<char[]> riffSig(new char[sigLength + 1]()); // RIFF\0
unique_ptr<char[]> waveSig(new char[sigLength + 1]()); // WAVE\0
StreamOffset previousOffset = device->pos();
device->seek(0);
device->readFull(riffSig.get(), sigLength);
device->seek(4, IOSeek::Relative);
device->readFull(waveSig.get(), sigLength);
device->seek(previousOffset);
if (strcmp(riffSig.get(), "RIFF") == 0 && strcmp(waveSig.get(), "WAVE") == 0) { // bytes are magic
return true;
}
return false;
}
WaveData parseWav(IODevicePtr device) {
const size_t sigLength = 4;
unique_ptr<char[]> riffSig(new char[sigLength + 1]()); // RIFF\0
unique_ptr<char[]> waveSig(new char[sigLength + 1]()); // WAVE\0
unique_ptr<char[]> fmtSig(new char[sigLength + 1]()); // fmt \0
unique_ptr<char[]> dataSig(new char[sigLength + 1]()); // data\0
// RIFF Chunk Descriptor
device->seek(0);
device->readFull(riffSig.get(), sigLength);
uint32_t fileSize = readLEType<uint32_t>(device);
fileSize += sigLength + sizeof(fileSize);
if (fileSize != device->size())
throw AudioException(strf("Wav file is wrong size, reports {} is actually {}", fileSize, device->size()));
device->readFull(waveSig.get(), sigLength);
if ((strcmp(riffSig.get(), "RIFF") != 0) || (strcmp(waveSig.get(), "WAVE") != 0)) { // bytes are not magic
auto p = [](char a) { return isprint(a) ? a : '?'; };
throw AudioException(strf("Wav file has wrong magic bytes, got `{:c}{:c}{:c}{:c}' and `{:c}{:c}{:c}{:c}' but expected `RIFF' and `WAVE'",
p(riffSig[0]), p(riffSig[1]), p(riffSig[2]), p(riffSig[3]), p(waveSig[0]), p(waveSig[1]), p(waveSig[2]), p(waveSig[3])));
}
// fmt subchunk
device->readFull(fmtSig.get(), sigLength);
if (strcmp(fmtSig.get(), "fmt ") != 0) { // friendship is magic
auto p = [](char a) { return isprint(a) ? a : '?'; };
throw AudioException(strf("Wav file fmt subchunk has wrong magic bytes, got `{:c}{:c}{:c}{:c}' but expected `fmt '",
p(fmtSig[0]),
p(fmtSig[1]),
p(fmtSig[2]),
p(fmtSig[3])));
}
uint32_t fmtSubchunkSize = readLEType<uint32_t>(device);
fmtSubchunkSize += sigLength;
if (fmtSubchunkSize < 20)
throw AudioException(strf("fmt subchunk is sized wrong, expected 20 got {}. Is this wav file not PCM?", fmtSubchunkSize));
uint16_t audioFormat = readLEType<uint16_t>(device);
if (audioFormat != 1)
throw AudioException("audioFormat data indicates that wav file is something other than PCM format. Unsupported.");
uint16_t wavChannels = readLEType<uint16_t>(device);
uint32_t wavSampleRate = readLEType<uint32_t>(device);
uint32_t wavByteRate = readLEType<uint32_t>(device);
uint16_t wavBlockAlign = readLEType<uint16_t>(device);
uint16_t wavBitsPerSample = readLEType<uint16_t>(device);
if (wavBitsPerSample != 16)
throw AudioException("Only 16-bit PCM wavs are supported.");
if (wavByteRate * 8 != wavSampleRate * wavChannels * wavBitsPerSample)
throw AudioException("Sanity check failed, ByteRate is wrong");
if (wavBlockAlign * 8 != wavChannels * wavBitsPerSample)
throw AudioException("Sanity check failed, BlockAlign is wrong");
device->seek(fmtSubchunkSize - 20, IOSeek::Relative);
// data subchunk
device->readFull(dataSig.get(), sigLength);
if (strcmp(dataSig.get(), "data") != 0) { // magic or more magic?
auto p = [](char a) { return isprint(a) ? a : '?'; };
throw AudioException(strf("Wav file data subchunk has wrong magic bytes, got `{:c}{:c}{:c}{:c}' but expected `data'",
p(dataSig[0]), p(dataSig[1]), p(dataSig[2]), p(dataSig[3])));
}
uint32_t wavDataSize = readLEType<uint32_t>(device);
size_t wavDataOffset = (size_t)device->pos();
if (wavDataSize + wavDataOffset > (size_t)device->size()) {
throw AudioException(strf("Wav file data size reported is inconsistent with file size, got {} but expected {}",
device->size(), wavDataSize + wavDataOffset));
}
ByteArrayPtr pcmData = make_shared<ByteArray>();
pcmData->resize(wavDataSize);
// Copy across data and perform and endianess conversion if needed
device->readFull(pcmData->ptr(), pcmData->size());
for (size_t i = 0; i < pcmData->size() / 2; ++i)
fromByteOrder(ByteOrder::LittleEndian, pcmData->ptr() + i * 2, 2);
return WaveData{std::move(pcmData), wavChannels, wavSampleRate};
}
}
class CompressedAudioImpl {
public:
static size_t readFunc(void* ptr, size_t size, size_t nmemb, void* datasource) {
return static_cast<ExternalBuffer*>(datasource)->read((char*)ptr, size * nmemb) / size;
}
static int seekFunc(void* datasource, ogg_int64_t offset, int whence) {
static_cast<ExternalBuffer*>(datasource)->seek(offset, (IOSeek)whence);
return 0;
};
static long int tellFunc(void* datasource) {
return (long int)static_cast<ExternalBuffer*>(datasource)->pos();
};
CompressedAudioImpl(CompressedAudioImpl const& impl) {
m_audioData = impl.m_audioData;
m_memoryFile.reset(m_audioData->ptr(), m_audioData->size());
m_vorbisInfo = nullptr;
}
CompressedAudioImpl(IODevicePtr audioData) {
audioData->open(IOMode::Read);
audioData->seek(0);
m_audioData = make_shared<ByteArray>(audioData->readBytes((size_t)audioData->size()));
m_memoryFile.reset(m_audioData->ptr(), m_audioData->size());
m_vorbisInfo = nullptr;
}
~CompressedAudioImpl() {
ov_clear(&m_vorbisFile);
}
bool open() {
m_callbacks.read_func = readFunc;
m_callbacks.seek_func = seekFunc;
m_callbacks.tell_func = tellFunc;
m_callbacks.close_func = NULL;
if (ov_open_callbacks(&m_memoryFile, &m_vorbisFile, NULL, 0, m_callbacks) < 0)
return false;
m_vorbisInfo = ov_info(&m_vorbisFile, -1);
return true;
}
unsigned channels() {
return m_vorbisInfo->channels;
}
unsigned sampleRate() {
return m_vorbisInfo->rate;
}
double totalTime() {
return ov_time_total(&m_vorbisFile, -1);
}
uint64_t totalSamples() {
return ov_pcm_total(&m_vorbisFile, -1);
}
void seekTime(double time) {
int ret = ov_time_seek(&m_vorbisFile, time);
if (ret != 0)
throw StarException("Cannot seek ogg stream Audio::seekTime");
}
void seekSample(uint64_t pos) {
int ret = ov_pcm_seek(&m_vorbisFile, pos);
if (ret != 0)
throw StarException("Cannot seek ogg stream in Audio::seekSample");
}
double currentTime() {
return ov_time_tell(&m_vorbisFile);
}
uint64_t currentSample() {
return ov_pcm_tell(&m_vorbisFile);
}
size_t readPartial(int16_t* buffer, size_t bufferSize) {
int bitstream;
int read;
// ov_read takes int parameter, so do some magic here to make sure we don't
// overflow
bufferSize *= 2;
#if STAR_LITTLE_ENDIAN
read = ov_read(&m_vorbisFile, (char*)buffer, bufferSize, 0, 2, 1, &bitstream);
#else
read = ov_read(&m_vorbisFile, (char*)buffer, bufferSize, 1, 2, 1, &bitstream);
#endif
if (read < 0)
throw AudioException("Error in Audio::read");
// read in bytes, returning number of int16_t samples.
return read / 2;
}
private:
ByteArrayConstPtr m_audioData;
ExternalBuffer m_memoryFile;
ov_callbacks m_callbacks;
OggVorbis_File m_vorbisFile;
vorbis_info* m_vorbisInfo;
};
class UncompressedAudioImpl {
public:
UncompressedAudioImpl(UncompressedAudioImpl const& impl) {
m_channels = impl.m_channels;
m_sampleRate = impl.m_sampleRate;
m_audioData = impl.m_audioData;
m_memoryFile.reset(m_audioData->ptr(), m_audioData->size());
}
UncompressedAudioImpl(CompressedAudioImpl& impl) {
m_channels = impl.channels();
m_sampleRate = impl.sampleRate();
int16_t buffer[1024];
Buffer uncompressBuffer;
while (true) {
size_t ramt = impl.readPartial(buffer, 1024);
if (ramt == 0) {
// End of stream reached
break;
} else {
uncompressBuffer.writeFull((char*)buffer, ramt * 2);
}
}
m_audioData = make_shared<ByteArray>(uncompressBuffer.takeData());
m_memoryFile.reset(m_audioData->ptr(), m_audioData->size());
}
UncompressedAudioImpl(ByteArrayConstPtr data, unsigned channels, unsigned sampleRate) {
m_channels = channels;
m_sampleRate = sampleRate;
m_audioData = std::move(data);
m_memoryFile.reset(m_audioData->ptr(), m_audioData->size());
}
bool open() {
return true;
}
unsigned channels() {
return m_channels;
}
unsigned sampleRate() {
return m_sampleRate;
}
double totalTime() {
return (double)totalSamples() / m_sampleRate;
}
uint64_t totalSamples() {
return m_memoryFile.dataSize() / 2 / m_channels;
}
void seekTime(double time) {
seekSample((uint64_t)(time * m_sampleRate));
}
void seekSample(uint64_t pos) {
m_memoryFile.seek(pos * 2 * m_channels);
}
double currentTime() {
return (double)currentSample() / m_sampleRate;
}
uint64_t currentSample() {
return m_memoryFile.pos() / 2 / m_channels;
}
size_t readPartial(int16_t* buffer, size_t bufferSize) {
if (bufferSize != NPos)
bufferSize = bufferSize * 2;
return m_memoryFile.read((char*)buffer, bufferSize) / 2;
}
private:
unsigned m_channels;
unsigned m_sampleRate;
ByteArrayConstPtr m_audioData;
ExternalBuffer m_memoryFile;
};
Audio::Audio(IODevicePtr device) {
if (!device->isOpen())
device->open(IOMode::Read);
if (isUncompressed(device)) {
WaveData data = parseWav(device);
m_uncompressed = make_shared<UncompressedAudioImpl>(std::move(data.byteArray), data.channels, data.sampleRate);
} else {
m_compressed = make_shared<CompressedAudioImpl>(device);
if (!m_compressed->open())
throw AudioException("File does not appear to be a valid ogg bitstream");
}
}
Audio::Audio(Audio const& audio) {
*this = audio;
}
Audio::Audio(Audio&& audio) {
operator=(std::move(audio));
}
Audio& Audio::operator=(Audio const& audio) {
if (audio.m_uncompressed) {
m_uncompressed = make_shared<UncompressedAudioImpl>(*audio.m_uncompressed);
m_uncompressed->open();
} else {
m_compressed = make_shared<CompressedAudioImpl>(*audio.m_compressed);
m_compressed->open();
}
seekSample(audio.currentSample());
return *this;
}
Audio& Audio::operator=(Audio&& audio) {
m_compressed = std::move(audio.m_compressed);
m_uncompressed = std::move(audio.m_uncompressed);
return *this;
}
unsigned Audio::channels() const {
if (m_uncompressed)
return m_uncompressed->channels();
else
return m_compressed->channels();
}
unsigned Audio::sampleRate() const {
if (m_uncompressed)
return m_uncompressed->sampleRate();
else
return m_compressed->sampleRate();
}
double Audio::totalTime() const {
if (m_uncompressed)
return m_uncompressed->totalTime();
else
return m_compressed->totalTime();
}
uint64_t Audio::totalSamples() const {
if (m_uncompressed)
return m_uncompressed->totalSamples();
else
return m_compressed->totalSamples();
}
bool Audio::compressed() const {
return (bool)m_compressed;
}
void Audio::uncompress() {
if (m_compressed) {
m_uncompressed = make_shared<UncompressedAudioImpl>(*m_compressed);
m_compressed.reset();
}
}
void Audio::seekTime(double time) {
if (m_uncompressed)
m_uncompressed->seekTime(time);
else
m_compressed->seekTime(time);
}
void Audio::seekSample(uint64_t pos) {
if (m_uncompressed)
m_uncompressed->seekSample(pos);
else
m_compressed->seekSample(pos);
}
double Audio::currentTime() const {
if (m_uncompressed)
return m_uncompressed->currentTime();
else
return m_compressed->currentTime();
}
uint64_t Audio::currentSample() const {
if (m_uncompressed)
return m_uncompressed->currentSample();
else
return m_compressed->currentSample();
}
size_t Audio::readPartial(int16_t* buffer, size_t bufferSize) {
if (bufferSize == 0)
return 0;
if (m_uncompressed)
return m_uncompressed->readPartial(buffer, bufferSize);
else
return m_compressed->readPartial(buffer, bufferSize);
}
size_t Audio::read(int16_t* buffer, size_t bufferSize) {
if (bufferSize == 0)
return 0;
size_t readTotal = 0;
while (readTotal < bufferSize) {
size_t toGo = bufferSize - readTotal;
size_t ramt = readPartial(buffer + readTotal, toGo);
readTotal += ramt;
// End of stream reached
if (ramt == 0)
break;
}
return readTotal;
}
size_t Audio::resample(unsigned destinationChannels, unsigned destinationSampleRate, int16_t* destinationBuffer, size_t destinationBufferSize, double velocity) {
unsigned destinationSamples = destinationBufferSize / destinationChannels;
if (destinationSamples == 0)
return 0;
unsigned sourceChannels = channels();
unsigned sourceSampleRate = sampleRate();
if (velocity != 1.0)
sourceSampleRate = (unsigned)(sourceSampleRate * velocity);
if (destinationChannels == sourceChannels && destinationSampleRate == sourceSampleRate) {
// If the destination and source channel count and sample rate are the
// same, this is the same as a read.
return read(destinationBuffer, destinationBufferSize);
} else if (destinationSampleRate == sourceSampleRate) {
// If the destination and source sample rate are the same, then we can skip
// the super-sampling math.
unsigned sourceBufferSize = destinationSamples * sourceChannels;
m_workingBuffer.resize(sourceBufferSize * sizeof(int16_t));
int16_t* sourceBuffer = (int16_t*)m_workingBuffer.ptr();
unsigned readSamples = read(sourceBuffer, sourceBufferSize) / sourceChannels;
for (unsigned sample = 0; sample < readSamples; ++sample) {
unsigned sourceBufferIndex = sample * sourceChannels;
unsigned destinationBufferIndex = sample * destinationChannels;
for (unsigned destinationChannel = 0; destinationChannel < destinationChannels; ++destinationChannel) {
// If the destination channel count is greater than the source
// channels, simply copy the last channel
unsigned sourceChannel = min(destinationChannel, sourceChannels - 1);
destinationBuffer[destinationBufferIndex + destinationChannel] =
sourceBuffer[sourceBufferIndex + sourceChannel];
}
}
return readSamples * destinationChannels;
} else {
// Otherwise, we have to do a full resample.
unsigned sourceSamples = ((uint64_t)sourceSampleRate * destinationSamples + destinationSampleRate - 1) / destinationSampleRate;
unsigned sourceBufferSize = sourceSamples * sourceChannels;
m_workingBuffer.resize(sourceBufferSize * sizeof(int16_t));
int16_t* sourceBuffer = (int16_t*)m_workingBuffer.ptr();
unsigned readSamples = read(sourceBuffer, sourceBufferSize) / sourceChannels;
if (readSamples == 0)
return 0;
unsigned writtenSamples = 0;
for (unsigned destinationSample = 0; destinationSample < destinationSamples; ++destinationSample) {
unsigned destinationBufferIndex = destinationSample * destinationChannels;
for (unsigned destinationChannel = 0; destinationChannel < destinationChannels; ++destinationChannel) {
static int const SuperSampleFactor = 8;
// If the destination channel count is greater than the source
// channels, simply copy the last channel
unsigned sourceChannel = min(destinationChannel, sourceChannels - 1);
int sample = 0;
int sampleCount = 0;
for (int superSample = 0; superSample < SuperSampleFactor; ++superSample) {
unsigned sourceSample = (unsigned)((destinationSample * SuperSampleFactor + superSample) * sourceSamples / destinationSamples) / SuperSampleFactor;
if (sourceSample < readSamples) {
unsigned sourceBufferIndex = sourceSample * sourceChannels;
starAssert(sourceBufferIndex + sourceChannel < sourceBufferSize);
sample += sourceBuffer[sourceBufferIndex + sourceChannel];
++sampleCount;
}
}
// If sampleCount is zero, then we are past the end of our read data
// completely, and can stop
if (sampleCount == 0)
return writtenSamples * destinationChannels;
sample /= sampleCount;
destinationBuffer[destinationBufferIndex + destinationChannel] = (int16_t)sample;
writtenSamples = destinationSample + 1;
}
}
return writtenSamples * destinationChannels;
}
}
}