void StarSeemTest::star_external_ffem()
{
QFETCH(QString, data);
AudioBuffer c;
QVERIFY2(c.read(data), ("Failed reading test wave; " + data).toUtf8().data());
/*=== WORLD による分析 ===*/
int timeLength = GetSamplesForDIO(c.format().sampleRate(), c.length(), msFramePeriod);
int fftLength = GetFFTSizeForStar(c.format().sampleRate());
double *f0 = new double[timeLength];
double *t = new double[timeLength];
double **specgram = new double*[timeLength];
specgram[0] = new double[timeLength * (fftLength / 2 + 1)];
for(int i = 1; i < timeLength; i++)
{
specgram[i] = specgram[0] + i * (fftLength / 2 + 1);
}
Dio(c.data()[0], c.length(), c.format().sampleRate(), msFramePeriod, t, f0);
Star(c.data()[0], c.length(), c.format().sampleRate(), msFramePeriod, f0, timeLength, specgram);
/*=== WORLD による分析ここまで ===*/
/*=== StarSeem による分析 === */
Envelope *e = new Envelope;
QVERIFY2(DioFfem().estimate(e, c.data()[0], c.length(), c.format().sampleRate(), msFramePeriod), "Failed dio FFEM");
// Envelope が正しいか確認する
for(int i = 0; i < e->size(); i++)
{
QVERIFY2(e->value(i) == f0[i], "Error; invalid f0 envelope");
}
Specgram *testset = new Specgram;
QVERIFY2(StarSeem(e).estimate(testset, c.data()[0], c.length(), fftLength, c.format().sampleRate(), msFramePeriod), "Failed STAR SEEM");
/*=== StarSeem による分析ここまで === */
// 比較開始
for(int t = 0; t < timeLength; t++)
{
for(int f = 0; f <= fftLength / 2; f++)
{
if(testset->value(t, f) != specgram[t][f])
{
QString s;
s = "t = " + QString::number(t) + ", f = " + QString::number(f);
s += " actual :" + QString::number(testset->value(t, f)) + " , expceted :" + QString::number(specgram[t][f]);
// WORLD 自体は同じだから値が全く同じでないとおかしい。
QFAIL(("Error ;" + s).toUtf8().data());
}
}
}
delete[] specgram[0];
delete[] specgram;
delete[] t;
delete[] f0;
}
void JavaScriptAudioNode::process(size_t framesToProcess)
{
// Discussion about inputs and outputs:
// As in other AudioNodes, JavaScriptAudioNode uses an AudioBus for its input and output (see inputBus and outputBus below).
// Additionally, there is a double-buffering for input and output which is exposed directly to JavaScript (see inputBuffer and outputBuffer below).
// This node is the producer for inputBuffer and the consumer for outputBuffer.
// The JavaScript code is the consumer of inputBuffer and the producer for outputBuffer.
// Get input and output busses.
AudioBus* inputBus = this->input(0)->bus();
AudioBus* outputBus = this->output(0)->bus();
// Get input and output buffers. We double-buffer both the input and output sides.
unsigned doubleBufferIndex = this->doubleBufferIndex();
bool isDoubleBufferIndexGood = doubleBufferIndex < 2 && doubleBufferIndex < m_inputBuffers.size() && doubleBufferIndex < m_outputBuffers.size();
ASSERT(isDoubleBufferIndexGood);
if (!isDoubleBufferIndexGood)
return;
AudioBuffer* inputBuffer = m_inputBuffers[doubleBufferIndex].get();
AudioBuffer* outputBuffer = m_outputBuffers[doubleBufferIndex].get();
// Check the consistency of input and output buffers.
unsigned numberOfInputChannels = m_internalInputBus.numberOfChannels();
bool buffersAreGood = outputBuffer && bufferSize() == outputBuffer->length() && m_bufferReadWriteIndex + framesToProcess <= bufferSize();
// If the number of input channels is zero, it's ok to have inputBuffer = 0.
if (m_internalInputBus.numberOfChannels())
buffersAreGood = buffersAreGood && inputBuffer && bufferSize() == inputBuffer->length();
ASSERT(buffersAreGood);
if (!buffersAreGood)
return;
// We assume that bufferSize() is evenly divisible by framesToProcess - should always be true, but we should still check.
bool isFramesToProcessGood = framesToProcess && bufferSize() >= framesToProcess && !(bufferSize() % framesToProcess);
ASSERT(isFramesToProcessGood);
if (!isFramesToProcessGood)
return;
unsigned numberOfOutputChannels = outputBus->numberOfChannels();
bool channelsAreGood = (numberOfInputChannels == m_numberOfInputChannels) && (numberOfOutputChannels == m_numberOfOutputChannels);
ASSERT(channelsAreGood);
if (!channelsAreGood)
return;
for (unsigned i = 0; i < numberOfInputChannels; i++)
m_internalInputBus.setChannelMemory(i, inputBuffer->getChannelData(i)->data() + m_bufferReadWriteIndex, framesToProcess);
if (numberOfInputChannels)
m_internalInputBus.copyFrom(*inputBus);
// Copy from the output buffer to the output.
for (unsigned i = 0; i < numberOfOutputChannels; ++i)
memcpy(outputBus->channel(i)->mutableData(), outputBuffer->getChannelData(i)->data() + m_bufferReadWriteIndex, sizeof(float) * framesToProcess);
// Update the buffering index.
m_bufferReadWriteIndex = (m_bufferReadWriteIndex + framesToProcess) % bufferSize();
// m_bufferReadWriteIndex will wrap back around to 0 when the current input and output buffers are full.
// When this happens, fire an event and swap buffers.
if (!m_bufferReadWriteIndex) {
// Avoid building up requests on the main thread to fire process events when they're not being handled.
// This could be a problem if the main thread is very busy doing other things and is being held up handling previous requests.
if (m_isRequestOutstanding) {
// We're late in handling the previous request. The main thread must be very busy.
// The best we can do is clear out the buffer ourself here.
outputBuffer->zero();
} else {
// Reference ourself so we don't accidentally get deleted before fireProcessEvent() gets called.
ref();
// Fire the event on the main thread, not this one (which is the realtime audio thread).
m_doubleBufferIndexForEvent = m_doubleBufferIndex;
m_isRequestOutstanding = true;
callOnMainThread(fireProcessEventDispatch, this);
}
swapBuffers();
}
}
void ScriptProcessorHandler::process(size_t framesToProcess)
{
// Discussion about inputs and outputs:
// As in other AudioNodes, ScriptProcessorNode uses an AudioBus for its input and output (see inputBus and outputBus below).
// Additionally, there is a double-buffering for input and output which is exposed directly to JavaScript (see inputBuffer and outputBuffer below).
// This node is the producer for inputBuffer and the consumer for outputBuffer.
// The JavaScript code is the consumer of inputBuffer and the producer for outputBuffer.
// Get input and output busses.
AudioBus* inputBus = input(0).bus();
AudioBus* outputBus = output(0).bus();
// Get input and output buffers. We double-buffer both the input and output sides.
unsigned doubleBufferIndex = this->doubleBufferIndex();
bool isDoubleBufferIndexGood = doubleBufferIndex < 2 && doubleBufferIndex < m_inputBuffers.size() && doubleBufferIndex < m_outputBuffers.size();
ASSERT(isDoubleBufferIndexGood);
if (!isDoubleBufferIndexGood)
return;
AudioBuffer* inputBuffer = m_inputBuffers[doubleBufferIndex].get();
AudioBuffer* outputBuffer = m_outputBuffers[doubleBufferIndex].get();
// Check the consistency of input and output buffers.
unsigned numberOfInputChannels = m_internalInputBus->numberOfChannels();
bool buffersAreGood = outputBuffer && bufferSize() == outputBuffer->length() && m_bufferReadWriteIndex + framesToProcess <= bufferSize();
// If the number of input channels is zero, it's ok to have inputBuffer = 0.
if (m_internalInputBus->numberOfChannels())
buffersAreGood = buffersAreGood && inputBuffer && bufferSize() == inputBuffer->length();
ASSERT(buffersAreGood);
if (!buffersAreGood)
return;
// We assume that bufferSize() is evenly divisible by framesToProcess - should always be true, but we should still check.
bool isFramesToProcessGood = framesToProcess && bufferSize() >= framesToProcess && !(bufferSize() % framesToProcess);
ASSERT(isFramesToProcessGood);
if (!isFramesToProcessGood)
return;
unsigned numberOfOutputChannels = outputBus->numberOfChannels();
bool channelsAreGood = (numberOfInputChannels == m_numberOfInputChannels) && (numberOfOutputChannels == m_numberOfOutputChannels);
ASSERT(channelsAreGood);
if (!channelsAreGood)
return;
for (unsigned i = 0; i < numberOfInputChannels; ++i)
m_internalInputBus->setChannelMemory(i, inputBuffer->getChannelData(i)->data() + m_bufferReadWriteIndex, framesToProcess);
if (numberOfInputChannels)
m_internalInputBus->copyFrom(*inputBus);
// Copy from the output buffer to the output.
for (unsigned i = 0; i < numberOfOutputChannels; ++i)
memcpy(outputBus->channel(i)->mutableData(), outputBuffer->getChannelData(i)->data() + m_bufferReadWriteIndex, sizeof(float) * framesToProcess);
// Update the buffering index.
m_bufferReadWriteIndex = (m_bufferReadWriteIndex + framesToProcess) % bufferSize();
// m_bufferReadWriteIndex will wrap back around to 0 when the current input and output buffers are full.
// When this happens, fire an event and swap buffers.
if (!m_bufferReadWriteIndex) {
// Avoid building up requests on the main thread to fire process events when they're not being handled.
// This could be a problem if the main thread is very busy doing other things and is being held up handling previous requests.
// The audio thread can't block on this lock, so we call tryLock() instead.
MutexTryLocker tryLocker(m_processEventLock);
if (!tryLocker.locked()) {
// We're late in handling the previous request. The main thread must be very busy.
// The best we can do is clear out the buffer ourself here.
outputBuffer->zero();
} else if (context()->executionContext()) {
// Fire the event on the main thread, not this one (which is the realtime audio thread).
m_doubleBufferIndexForEvent = m_doubleBufferIndex;
context()->executionContext()->postTask(BLINK_FROM_HERE, createCrossThreadTask(&ScriptProcessorHandler::fireProcessEvent, PassRefPtr<ScriptProcessorHandler>(this)));
}
swapBuffers();
}
}
void JavaScriptAudioNode::process(size_t framesToProcess)
{
// Discussion about inputs and outputs:
// As in other AudioNodes, JavaScriptAudioNode uses an AudioBus for its input and output (see inputBus and outputBus below).
// Additionally, there is a double-buffering for input and output which is exposed directly to JavaScript (see inputBuffer and outputBuffer below).
// This node is the producer for inputBuffer and the consumer for outputBuffer.
// The JavaScript code is the consumer of inputBuffer and the producer for outputBuffer.
// Get input and output busses.
AudioBus* inputBus = this->input(0)->bus();
AudioBus* outputBus = this->output(0)->bus();
// Get input and output buffers. We double-buffer both the input and output sides.
unsigned doubleBufferIndex = this->doubleBufferIndex();
bool isDoubleBufferIndexGood = doubleBufferIndex < 2 && doubleBufferIndex < m_inputBuffers.size() && doubleBufferIndex < m_outputBuffers.size();
ASSERT(isDoubleBufferIndexGood);
if (!isDoubleBufferIndexGood)
return;
AudioBuffer* inputBuffer = m_inputBuffers[doubleBufferIndex].get();
AudioBuffer* outputBuffer = m_outputBuffers[doubleBufferIndex].get();
// Check the consistency of input and output buffers.
bool buffersAreGood = inputBuffer && outputBuffer && bufferSize() == inputBuffer->length() && bufferSize() == outputBuffer->length()
&& m_bufferReadWriteIndex + framesToProcess <= bufferSize();
ASSERT(buffersAreGood);
if (!buffersAreGood)
return;
// We assume that bufferSize() is evenly divisible by framesToProcess - should always be true, but we should still check.
bool isFramesToProcessGood = framesToProcess && bufferSize() >= framesToProcess && !(bufferSize() % framesToProcess);
ASSERT(isFramesToProcessGood);
if (!isFramesToProcessGood)
return;
unsigned numberOfInputChannels = inputBus->numberOfChannels();
bool channelsAreGood = (numberOfInputChannels == 1 || numberOfInputChannels == 2) && outputBus->numberOfChannels() == 2;
ASSERT(channelsAreGood);
if (!channelsAreGood)
return;
const float* sourceL = inputBus->channel(0)->data();
const float* sourceR = numberOfInputChannels > 1 ? inputBus->channel(1)->data() : 0;
float* destinationL = outputBus->channel(0)->mutableData();
float* destinationR = outputBus->channel(1)->mutableData();
// Copy from the input to the input buffer. See "buffersAreGood" check above for safety.
size_t bytesToCopy = sizeof(float) * framesToProcess;
memcpy(inputBuffer->getChannelData(0)->data() + m_bufferReadWriteIndex, sourceL, bytesToCopy);
if (numberOfInputChannels == 2)
memcpy(inputBuffer->getChannelData(1)->data() + m_bufferReadWriteIndex, sourceR, bytesToCopy);
else if (numberOfInputChannels == 1) {
// If the input is mono, then also copy the mono input to the right channel of the AudioBuffer which the AudioProcessingEvent uses.
// FIXME: it is likely the audio API will evolve to present an AudioBuffer with the same number of channels as our input.
memcpy(inputBuffer->getChannelData(1)->data() + m_bufferReadWriteIndex, sourceL, bytesToCopy);
}
// Copy from the output buffer to the output. See "buffersAreGood" check above for safety.
memcpy(destinationL, outputBuffer->getChannelData(0)->data() + m_bufferReadWriteIndex, bytesToCopy);
memcpy(destinationR, outputBuffer->getChannelData(1)->data() + m_bufferReadWriteIndex, bytesToCopy);
// Update the buffering index.
m_bufferReadWriteIndex = (m_bufferReadWriteIndex + framesToProcess) % bufferSize();
// m_bufferReadWriteIndex will wrap back around to 0 when the current input and output buffers are full.
// When this happens, fire an event and swap buffers.
if (!m_bufferReadWriteIndex) {
// Avoid building up requests on the main thread to fire process events when they're not being handled.
// This could be a problem if the main thread is very busy doing other things and is being held up handling previous requests.
if (m_isRequestOutstanding) {
// We're late in handling the previous request. The main thread must be very busy.
// The best we can do is clear out the buffer ourself here.
outputBuffer->zero();
} else {
// Reference ourself so we don't accidentally get deleted before fireProcessEvent() gets called.
ref();
// Fire the event on the main thread, not this one (which is the realtime audio thread).
m_doubleBufferIndexForEvent = m_doubleBufferIndex;
m_isRequestOutstanding = true;
callOnMainThread(fireProcessEventDispatch, this);
}
swapBuffers();
}
}
void ScriptProcessorHandler::process(size_t framesToProcess) {
// Discussion about inputs and outputs:
// As in other AudioNodes, ScriptProcessorNode uses an AudioBus for its input
// and output (see inputBus and outputBus below). Additionally, there is a
// double-buffering for input and output which is exposed directly to
// JavaScript (see inputBuffer and outputBuffer below). This node is the
// producer for inputBuffer and the consumer for outputBuffer. The JavaScript
// code is the consumer of inputBuffer and the producer for outputBuffer.
// Get input and output busses.
AudioBus* inputBus = input(0).bus();
AudioBus* outputBus = output(0).bus();
// Get input and output buffers. We double-buffer both the input and output
// sides.
unsigned doubleBufferIndex = this->doubleBufferIndex();
bool isDoubleBufferIndexGood = doubleBufferIndex < 2 &&
doubleBufferIndex < m_inputBuffers.size() &&
doubleBufferIndex < m_outputBuffers.size();
DCHECK(isDoubleBufferIndexGood);
if (!isDoubleBufferIndexGood)
return;
AudioBuffer* inputBuffer = m_inputBuffers[doubleBufferIndex].get();
AudioBuffer* outputBuffer = m_outputBuffers[doubleBufferIndex].get();
// Check the consistency of input and output buffers.
unsigned numberOfInputChannels = m_internalInputBus->numberOfChannels();
bool buffersAreGood =
outputBuffer && bufferSize() == outputBuffer->length() &&
m_bufferReadWriteIndex + framesToProcess <= bufferSize();
// If the number of input channels is zero, it's ok to have inputBuffer = 0.
if (m_internalInputBus->numberOfChannels())
buffersAreGood =
buffersAreGood && inputBuffer && bufferSize() == inputBuffer->length();
DCHECK(buffersAreGood);
if (!buffersAreGood)
return;
// We assume that bufferSize() is evenly divisible by framesToProcess - should
// always be true, but we should still check.
bool isFramesToProcessGood = framesToProcess &&
bufferSize() >= framesToProcess &&
!(bufferSize() % framesToProcess);
DCHECK(isFramesToProcessGood);
if (!isFramesToProcessGood)
return;
unsigned numberOfOutputChannels = outputBus->numberOfChannels();
bool channelsAreGood = (numberOfInputChannels == m_numberOfInputChannels) &&
(numberOfOutputChannels == m_numberOfOutputChannels);
DCHECK(channelsAreGood);
if (!channelsAreGood)
return;
for (unsigned i = 0; i < numberOfInputChannels; ++i)
m_internalInputBus->setChannelMemory(
i, inputBuffer->getChannelData(i)->data() + m_bufferReadWriteIndex,
framesToProcess);
if (numberOfInputChannels)
m_internalInputBus->copyFrom(*inputBus);
// Copy from the output buffer to the output.
for (unsigned i = 0; i < numberOfOutputChannels; ++i)
memcpy(outputBus->channel(i)->mutableData(),
outputBuffer->getChannelData(i)->data() + m_bufferReadWriteIndex,
sizeof(float) * framesToProcess);
// Update the buffering index.
m_bufferReadWriteIndex =
(m_bufferReadWriteIndex + framesToProcess) % bufferSize();
// m_bufferReadWriteIndex will wrap back around to 0 when the current input
// and output buffers are full.
// When this happens, fire an event and swap buffers.
if (!m_bufferReadWriteIndex) {
// Avoid building up requests on the main thread to fire process events when
// they're not being handled. This could be a problem if the main thread is
// very busy doing other things and is being held up handling previous
// requests. The audio thread can't block on this lock, so we call
// tryLock() instead.
MutexTryLocker tryLocker(m_processEventLock);
if (!tryLocker.locked()) {
// We're late in handling the previous request. The main thread must be
// very busy. The best we can do is clear out the buffer ourself here.
outputBuffer->zero();
} else if (context()->getExecutionContext()) {
// With the realtime context, execute the script code asynchronously
// and do not wait.
if (context()->hasRealtimeConstraint()) {
// Fire the event on the main thread with the appropriate buffer
// index.
context()->getExecutionContext()->postTask(
BLINK_FROM_HERE,
createCrossThreadTask(&ScriptProcessorHandler::fireProcessEvent,
crossThreadUnretained(this),
m_doubleBufferIndex));
} else {
// If this node is in the offline audio context, use the
// waitable event to synchronize to the offline rendering thread.
std::unique_ptr<WaitableEvent> waitableEvent =
wrapUnique(new WaitableEvent());
context()->getExecutionContext()->postTask(
BLINK_FROM_HERE,
createCrossThreadTask(
&ScriptProcessorHandler::fireProcessEventForOfflineAudioContext,
crossThreadUnretained(this), m_doubleBufferIndex,
crossThreadUnretained(waitableEvent.get())));
// Okay to block the offline audio rendering thread since it is
// not the actual audio device thread.
waitableEvent->wait();
}
}
swapBuffers();
}
}