本文整理汇总了C++中AudioContext::sampleRate方法的典型用法代码示例。如果您正苦于以下问题：C++ AudioContext::sampleRate方法的具体用法？C++ AudioContext::sampleRate怎么用？C++ AudioContext::sampleRate使用的例子？那么, 这里精选的方法代码示例或许可以为您提供帮助。您也可以进一步了解该方法所在类AudioContext的用法示例。
在下文中一共展示了AudioContext::sampleRate方法的5个代码示例，这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞，您的评价将有助于系统推荐出更棒的C++代码示例。
示例1: AudioBasicProcessorNode
WaveShaperNode::WaveShaperNode(AudioContext& context)
: AudioBasicProcessorNode(context, context.sampleRate())
{
m_processor = std::make_unique<WaveShaperProcessor>(context.sampleRate(), 1);
setNodeType(NodeTypeWaveShaper);
initialize();
}示例2: AudioNode
MediaElementAudioSourceNode::MediaElementAudioSourceNode(AudioContext& context, HTMLMediaElement& mediaElement)
: AudioNode(context, context.sampleRate())
, m_mediaElement(mediaElement)
, m_sourceNumberOfChannels(0)
, m_sourceSampleRate(0)
{
// Default to stereo. This could change depending on what the media element .src is set to.
addOutput(std::make_unique<AudioNodeOutput>(this, 2));
setNodeType(NodeTypeMediaElementAudioSource);
initialize();
}示例3: valueForContextTime
float AudioParamTimeline::valueForContextTime(AudioContext& context, float defaultValue, bool& hasValue)
{
{
std::unique_lock<Lock> lock(m_eventsMutex, std::try_to_lock);
if (!lock.owns_lock() || !m_events.size() || context.currentTime() < m_events[0].time()) {
hasValue = false;
return defaultValue;
}
}
// Ask for just a single value.
float value;
double sampleRate = context.sampleRate();
double startTime = context.currentTime();
double endTime = startTime + 1.1 / sampleRate; // time just beyond one sample-frame
double controlRate = sampleRate / AudioNode::ProcessingSizeInFrames; // one parameter change per render quantum
value = valuesForTimeRange(startTime, endTime, defaultValue, &value, 1, sampleRate, controlRate);
hasValue = true;
return value;
}示例4: AudioNode
PannerNode::PannerNode(AudioContext& context, float sampleRate)
: AudioNode(context, sampleRate)
, m_panningModel(PanningModelType::HRTF)
, m_lastGain(-1.0)
, m_connectionCount(0)
{
// Load the HRTF database asynchronously so we don't block the Javascript thread while creating the HRTF database.
m_hrtfDatabaseLoader = HRTFDatabaseLoader::createAndLoadAsynchronouslyIfNecessary(context.sampleRate());
addInput(std::make_unique<AudioNodeInput>(this));
addOutput(std::make_unique<AudioNodeOutput>(this, 2));
// Node-specific default mixing rules.
m_channelCount = 2;
m_channelCountMode = ClampedMax;
m_channelInterpretation = AudioBus::Speakers;
m_distanceGain = AudioParam::create(context, "distanceGain", 1.0, 0.0, 1.0);
m_coneGain = AudioParam::create(context, "coneGain", 1.0, 0.0, 1.0);
m_position = FloatPoint3D(0, 0, 0);
m_orientation = FloatPoint3D(1, 0, 0);
m_velocity = FloatPoint3D(0, 0, 0);
setNodeType(NodeTypePanner);
initialize();
}示例5: AudioNode
WaveShaperNode::WaveShaperNode(AudioContext& context)
: AudioNode(context)
{
setHandler(AudioBasicProcessorHandler::create(AudioHandler::NodeTypeWaveShaper, *this, context.sampleRate(), adoptPtr(new WaveShaperProcessor(context.sampleRate(), 1))));
handler().initialize();
}