void
OscillatorNode::Start(double aWhen, ErrorResult& aRv)
{
if (!WebAudioUtils::IsTimeValid(aWhen)) {
aRv.Throw(NS_ERROR_DOM_NOT_SUPPORTED_ERR);
return;
}
if (mStartCalled) {
aRv.Throw(NS_ERROR_DOM_INVALID_STATE_ERR);
return;
}
mStartCalled = true;
if (!mStream) {
// Nothing to play, or we're already dead for some reason
return;
}
// TODO: Perhaps we need to do more here.
mStream->SetStreamTimeParameter(OscillatorNodeEngine::START,
Context(), aWhen);
MarkActive();
}
void
AudioBufferSourceNode::SendOffsetAndDurationParametersToStream(AudioNodeStream* aStream)
{
NS_ASSERTION(mBuffer && mStartCalled,
"Only call this when we have a buffer and start() has been called");
float rate = mBuffer->SampleRate();
aStream->SetInt32Parameter(SAMPLE_RATE, rate);
int32_t bufferEnd = mBuffer->Length();
int32_t offsetSamples = std::max(0, NS_lround(mOffset * rate));
// Don't set parameter unnecessarily
if (offsetSamples > 0) {
aStream->SetInt32Parameter(BUFFERSTART, offsetSamples);
}
if (mDuration != std::numeric_limits<double>::min()) {
MOZ_ASSERT(mDuration >= 0.0); // provided by Start()
MOZ_ASSERT(rate >= 0.0f); // provided by AudioBuffer::Create()
static_assert(std::numeric_limits<double>::digits >=
std::numeric_limits<decltype(bufferEnd)>::digits,
"bufferEnd should be represented exactly by double");
// + 0.5 rounds mDuration to nearest sample when assigned to bufferEnd.
bufferEnd = std::min<double>(bufferEnd,
offsetSamples + mDuration * rate + 0.5);
}
aStream->SetInt32Parameter(BUFFEREND, bufferEnd);
MarkActive();
}
void
ScriptProcessorNode::UpdateConnectedStatus()
{
bool isConnected = mHasPhantomInput ||
!(OutputNodes().IsEmpty() && OutputParams().IsEmpty()
&& InputNodes().IsEmpty());
// Events are queued even when there is no listener because a listener
// may be added while events are in the queue.
SendInt32ParameterToStream(ScriptProcessorNodeEngine::IS_CONNECTED,
isConnected);
if (isConnected && HasListenersFor(nsGkAtoms::onaudioprocess)) {
MarkActive();
} else {
MarkInactive();
}
}
void
AudioBufferSourceNode::SendOffsetAndDurationParametersToStream(AudioNodeStream* aStream)
{
NS_ASSERTION(mBuffer && mStartCalled,
"Only call this when we have a buffer and start() has been called");
float rate = mBuffer->SampleRate();
int32_t bufferEnd = mBuffer->Length();
int32_t offsetSamples = std::max(0, NS_lround(mOffset * rate));
// Don't set parameter unnecessarily
if (offsetSamples > 0) {
aStream->SetInt32Parameter(BUFFERSTART, offsetSamples);
}
if (mDuration != std::numeric_limits<double>::min()) {
bufferEnd = std::min(bufferEnd,
offsetSamples + NS_lround(mDuration * rate));
}
aStream->SetInt32Parameter(BUFFEREND, bufferEnd);
MarkActive();
}
