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C++ AudioSampleBuffer::getSample方法代码示例

本文整理汇总了C++中AudioSampleBuffer::getSample方法的典型用法代码示例。如果您正苦于以下问题：C++ AudioSampleBuffer::getSample方法的具体用法？C++ AudioSampleBuffer::getSample怎么用？C++ AudioSampleBuffer::getSample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AudioSampleBuffer的用法示例。


在下文中一共展示了AudioSampleBuffer::getSample方法的6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: processBlock

void BitcrushAudioProcessor::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
{
	this->initializing(buffer);

	for (int i = getNumInputChannels(); i < getNumOutputChannels(); ++i) {
		buffer.clear(i, 0, buffer.getNumSamples());
	}

	float crush = bitcrush->getValue();
	float wet_ = wet->getValue();
	int groupedSamples = std::max(1.f, downsample->getValue() * 100);
	float bitdepth = 12. * (1. - crush) + 1. * crush;
	int steps = exp2(bitdepth);

	// This is the place where you'd normally do the guts of your plugin's
	// audio processing...
	for (int channel = 0; channel < getNumInputChannels(); channel++) 
	{
		for (int sample = 0; sample < buffer.getNumSamples() - groupedSamples; sample += groupedSamples) {
			float averagedSample = 0.;
			for (int i = 0; i < groupedSamples; i++) {
				averagedSample += buffer.getSample(channel, i + sample) / groupedSamples;
			}

			int discretizedSample = averagedSample * steps;
			float crushed = float(discretizedSample) / steps;

			for (int i = 0; i < groupedSamples; i++) {
				float sampleValue = buffer.getSample(channel, i + sample);
				buffer.setSample(channel, i + sample, sampleValue * (1. - wet_) + crushed * wet_);
			}
		}

		float averagedSample = 0.;
		for (int i = (buffer.getNumSamples() / groupedSamples) * groupedSamples; i < buffer.getNumSamples(); i++) {
			averagedSample += buffer.getSample(channel, i) / (buffer.getNumSamples() % groupedSamples);
		}

		float bitdepth = 12. * (1. - crush) + 1. * crush;
		int steps = exp2(bitdepth);
		int discretizedSample = averagedSample * steps;
		float crushed = float(discretizedSample) / steps;

		for (int i = (buffer.getNumSamples() / groupedSamples) * groupedSamples; i < buffer.getNumSamples(); i++) {
			float sampleValue = buffer.getSample(channel, i);
			buffer.setSample(channel, i, sampleValue * (1. - wet_) + crushed * wet_);
		}
	}

	this->meteringBuffer(buffer);
	this->finalizing(buffer);
}
开发者ID:eriser，项目名称:blankenhain，代码行数:52，代码来源:BitcrushAudioProcessor.cpp

示例2: processBlock

void EQNode::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
{
    float in_samp_m1, in_samp_m2;
    float out_samp_m1, out_samp_m2;
    for(int channel = 0; channel < buffer.getNumChannels(); ++channel)
    {
        for(int sample = 0; sample < buffer.getNumSamples(); ++sample)
        {
            float in_samp = buffer.getSample(channel, sample);
            float out_samp = (_b0 * in_samp + _b1 * in_samp_m1 + _b2 * in_samp_m2 - _a1 * out_samp_m1 - _a2 * out_samp_m2) / _a0;
            
            in_samp_m2 = in_samp_m1;
            in_samp_m1 = in_samp;
            out_samp_m2 = out_samp_m1;
            out_samp_m1 = out_samp;
            buffer.setSample(channel, sample, out_samp);
        }
    }
}
开发者ID:dmtaudio，项目名称:mordaw，代码行数:19，代码来源:EQNode.cpp

示例3: meteringBuffer

void BlankenhainAudioProcessor::meteringBuffer(AudioSampleBuffer& buffer)
{
	/* METERING CODE */
	for (size_t iteration = 0; iteration < buffer.getNumSamples(); iteration++)
	{
		if (abs(buffer.getSample(/*channel*/ 0, iteration)) > meterValues[0])
		{
			meterValues[0] = abs(buffer.getSample(/*channel*/ 0, iteration));
		}
		if (abs(buffer.getSample(/*channel*/ 1, iteration)) > meterValues[1])
		{
			meterValues[1] = abs(buffer.getSample(/*channel*/ 1, iteration));
		}
		meterValues[2] += buffer.getSample(/*channel*/ 0, iteration) * buffer.getSample(/*channel*/ 0, iteration);
		meterValues[3] += buffer.getSample(/*channel*/ 1, iteration) * buffer.getSample(/*channel*/ 1, iteration);
	}
	/* END METERING CODE*/
}
开发者ID:eriser，项目名称:blankenhain，代码行数:18，代码来源:BlankenhainAudioProcessor.cpp

示例4: processBlock

void VolumeAudioProcessor::processBlock(AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
{
	this->initializing(buffer);

	float currentVolumeL, currentVolumeR, bufferValue, oldVolumeL, oldVolumeR;
	unsigned int maxInterpolation;

	if (stereoCoupling->getBoolValue()) 
	{
		currentVolumeL = pow(10.f, (volumeL->getValue()) / 10.f);
		oldVolumeL = pow(10.f, (volumeL->getOldValue()) / 10.f);
		maxInterpolation = int(buffer.getNumSamples() * volumeL->getBufferScalingValue());
		for (size_t interpolationIteration = 0; interpolationIteration < maxInterpolation; interpolationIteration++)
		{
			bufferValue = buffer.getSample(/*channel*/ 0, interpolationIteration);

			buffer.setSample(/*channel*/ 0, interpolationIteration, bufferValue * \
			(oldVolumeL + ((interpolationIteration + 1) * (currentVolumeL - oldVolumeL) \
			/ maxInterpolation)));

			bufferValue = buffer.getSample(/*channel*/ 1, interpolationIteration);

			buffer.setSample(/*channel*/ 1, interpolationIteration, bufferValue * \
			(oldVolumeL + ((interpolationIteration + 1) * (currentVolumeL - oldVolumeL) \
			/ maxInterpolation)));

		}
		for (size_t bufferIteration = maxInterpolation; bufferIteration < buffer.getNumSamples(); bufferIteration++)
		{
			bufferValue = buffer.getSample(/*channel*/ 0, bufferIteration);
			buffer.setSample(/*channel*/ 0, bufferIteration, bufferValue * currentVolumeL);
			bufferValue = buffer.getSample(/*channel*/ 1, bufferIteration);
			buffer.setSample(/*channel*/ 1, bufferIteration, bufferValue * currentVolumeL);

		}
	}
	else
	{
		currentVolumeR = pow(10.f, volumeR->getValue() / 10.f);
		currentVolumeL = pow(10.f, volumeL->getValue() / 10.f);
		oldVolumeL = pow(10.f, volumeL->getOldValue() / 10.f);
		oldVolumeR = pow(10.f, volumeR->getOldValue() / 10.f);
		maxInterpolation = int(buffer.getNumSamples() * volumeL->getBufferScalingValue());
		for (size_t interpolationIteration = 0; interpolationIteration < maxInterpolation; interpolationIteration++)
		{
			bufferValue = buffer.getSample(/*channel*/ 0, interpolationIteration);

			buffer.setSample(/*channel*/ 0, interpolationIteration, bufferValue * \
			(oldVolumeL + ((interpolationIteration + 1) * (currentVolumeL - oldVolumeL) \
			/ maxInterpolation)));

			bufferValue = buffer.getSample(/*channel*/ 1, interpolationIteration);

			buffer.setSample(/*channel*/ 1, interpolationIteration, bufferValue * \
			(oldVolumeR + ((interpolationIteration + 1) * (currentVolumeR - oldVolumeR) \
			/ maxInterpolation )));


		}
		for (size_t bufferIteration = maxInterpolation; bufferIteration < buffer.getNumSamples(); bufferIteration++)
		{
			bufferValue = buffer.getSample(/*channel*/ 0, bufferIteration);
			buffer.setSample(/*channel*/ 0, bufferIteration, bufferValue * currentVolumeL);
			bufferValue = buffer.getSample(/*channel*/ 1, bufferIteration);
			buffer.setSample(/*channel*/ 1, bufferIteration, bufferValue * currentVolumeR);
		}
	}
	//Set current values as old values for interpolation in next buffer iteration
	volumeL->setOldValue(volumeL->getValue());
	volumeR->setOldValue(volumeR->getValue());

	this->meteringBuffer(buffer);
	this->finalizing(buffer);
}
开发者ID:eriser，项目名称:blankenhain，代码行数:74，代码来源:VolumeAudioProcessor.cpp

示例5: processBlock

void PanAudioProcessor::processBlock(AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
{
	this->initializing(buffer);

	for (int i = getNumInputChannels(); i < getNumOutputChannels(); ++i) {
		buffer.clear(i, 0, buffer.getNumSamples());
	}
	// currentPanning: Set by Editor before this buffer iteration
	// oldPanning: Was Set in Editor after last buffer Iteration
	// Interpolation from oldPanning to currentPanning
	// momentaryPanning: Helper Variable, keeps results of current Interpolation iteration during Interpolation
	float currentPanning, oldPanning, bufferValue, momentaryPanning;

	unsigned int maxInterpolation;
	currentPanning = panning->getNormalizedValue();
	oldPanning = panning->getNormalizedOldValue();
	maxInterpolation = int(buffer.getNumSamples() * panning->getBufferScalingValue());

	if (getNumInputChannels() == 1)
	{
		for (size_t interpolationIteration = 0; interpolationIteration < maxInterpolation; interpolationIteration++)
		{
			bufferValue = buffer.getSample(/*channel*/ 0, interpolationIteration);
			momentaryPanning = (oldPanning + ((interpolationIteration + 1) * (currentPanning - oldPanning) / maxInterpolation));
			buffer.setSample(/*channel*/ 0, interpolationIteration, bufferValue * \
				(1.f - 2 * (std::max(0.5f, momentaryPanning) - 0.5f)));

			buffer.setSample(/*channel*/ 1, interpolationIteration, bufferValue * \
				2 * (std::min(0.5f, momentaryPanning)));
		}
		for (size_t bufferIteration = maxInterpolation; bufferIteration < buffer.getNumSamples(); bufferIteration++)
		{
			bufferValue = buffer.getSample(/*channel*/ 0, bufferIteration);
			buffer.setSample(/*channel*/ 0, bufferIteration, bufferValue * (1.f - 2 * (std::max(0.5f, currentPanning) - 0.5f)));
			buffer.setSample(/*channel*/ 1, bufferIteration, bufferValue * 2 * (std::min(0.5f, currentPanning)));
		}
	}
	else
	{
		for (size_t interpolationIteration = 0; interpolationIteration < maxInterpolation; interpolationIteration++)
		{
			momentaryPanning = (oldPanning + ((interpolationIteration + 1) * (currentPanning - oldPanning) / maxInterpolation));

			bufferValue = buffer.getSample(/*channel*/ 0, interpolationIteration);
			buffer.setSample(/*channel*/ 0, interpolationIteration, bufferValue * \
				(1.f - 2 * (std::max(0.5f, momentaryPanning) - 0.5f)));
			bufferValue = buffer.getSample(/*channel*/ 1, interpolationIteration);
			buffer.setSample(/*channel*/ 1, interpolationIteration, bufferValue * \
				2 * (std::min(0.5f, momentaryPanning)));
		}
		for (size_t bufferIteration = maxInterpolation; bufferIteration < buffer.getNumSamples(); bufferIteration++)
		{
			bufferValue = buffer.getSample(/*channel*/ 0, bufferIteration);
			buffer.setSample(/*channel*/ 0, bufferIteration, bufferValue * (1.f - 2 * (std::max(0.5f, currentPanning) - 0.5f)));
			bufferValue = buffer.getSample(/*channel*/ 1, bufferIteration);
			buffer.setSample(/*channel*/ 1, bufferIteration, bufferValue * 2 * (std::min(0.5f, currentPanning)));
		}
	}
	panning->setOldValue();
	this->meteringBuffer(buffer);
	this->finalizing(buffer);
}
开发者ID:eriser，项目名称:blankenhain，代码行数:62，代码来源:PanAudioProcessor.cpp

示例6: renderNextBlock

void StreamingSamplerVoice::renderNextBlock(AudioSampleBuffer &outputBuffer, int startSample, int numSamples)
{
	const StreamingSamplerSound *sound = loader.getLoadedSound();

#if USE_SAMPLE_DEBUG_COUNTER
	const int startDebug = startSample;
	const int numDebug = numSamples;
#endif

	if (sound != nullptr)
	{
		const double startAlpha = fmod(voiceUptime, 1.0);

		jassert(pitchCounter != 0);

		auto tempVoiceBuffer = getTemporaryVoiceBuffer();

		jassert(tempVoiceBuffer != nullptr);

		tempVoiceBuffer->clear();

		// Copy the not resampled values into the voice buffer.
		StereoChannelData data = loader.fillVoiceBuffer(*tempVoiceBuffer, pitchCounter + startAlpha);

		float* outL = outputBuffer.getWritePointer(0, startSample);
		float* outR = outputBuffer.getWritePointer(1, startSample);

		const int startFixed = startSample;
		const int numSamplesFixed = numSamples;


#if USE_SAMPLE_DEBUG_COUNTER
		jassert((int)voiceUptime == data.leftChannel[0]);
#endif

		double indexInBuffer = startAlpha;

		if (data.isFloatingPoint)
		{
			const float* const inL = static_cast<const float*>(data.leftChannel);
			const float* const inR = static_cast<const float*>(data.rightChannel);

			interpolateStereoSamples(inL, inR, pitchData, outL, outR, startSample, indexInBuffer, uptimeDelta, numSamples, true);
		}
		else
		{
			const int16* const inL = static_cast<const int16*>(data.leftChannel);
			const int16* const inR = static_cast<const int16*>(data.rightChannel);

			interpolateStereoSamples(inL, inR, pitchData, outL, outR, startSample, indexInBuffer, uptimeDelta, numSamples, false);

		}

#if USE_SAMPLE_DEBUG_COUNTER 

		for (int i = startDebug; i < numDebug; i++)
		{
			const float l = outputBuffer.getSample(0, i);
			const float r = outputBuffer.getSample(1, i);

			jassert(l == r);
			jassert((abs(l - voiceUptime) < 0.000001) || l == 0.0f);

			voiceUptime += uptimeDelta;

		}

		outputBuffer.clear();
#else
		voiceUptime += pitchCounter;
#endif

		if (!loader.advanceReadIndex(voiceUptime))
		{
#if LOG_SAMPLE_RENDERING
			logger->addStreamingFailure(voiceUptime);
#endif

			outputBuffer.clear(startFixed, numSamplesFixed);

			resetVoice();
			return;
		}

		const bool enoughSamples = sound->hasEnoughSamplesForBlock((int)(voiceUptime));// +numSamples * MAX_SAMPLER_PITCH));

#if LOG_SAMPLE_RENDERING
		logger->checkSampleData(nullptr, DebugLogger::Location::SampleVoiceBufferFillPost, true, outputBuffer.getReadPointer(0, startFixed), numSamplesFixed);
		logger->checkSampleData(nullptr, DebugLogger::Location::SampleVoiceBufferFillPost, false, outputBuffer.getReadPointer(1, startFixed), numSamplesFixed);
#endif

		if (!enoughSamples) resetVoice();
	}
	else
	{
		resetVoice();
	}
};
开发者ID:azeteg，项目名称:HISE，代码行数:98，代码来源:StreamingSamplerVoice.cpp


注：本文中的AudioSampleBuffer::getSample方法示例由纯净天空整理自Github/MSDocs等开源代码及文档管理平台，相关代码片段筛选自各路编程大神贡献的开源项目，源码版权归原作者所有，传播和使用请参考对应项目的License；未经允许，请勿转载。