[英]How to access Audio data from JUCE Demo Audio Plugin Host?
我正在做一个项目,要求我从JUCE演示音频插件主机中加载的MIDI Synth插件将音频数据记录为.wav文件(每个1秒)。 基本上,我需要从MIDI Synth自动创建一个数据集(对应于不同的参数配置)。
我是否需要发送MIDI Note On / Off信息来生成音频数据? 还是有更好的方式获取音频数据?
AudioBuffer<FloatType> getBusBuffer (AudioBuffer<FloatType>& processBlockBuffer) const
这是可以满足我需求的功能吗? 如果是,我将如何存储数据? 如果没有,请有人指导我正确的功能/解决方案。 谢谢。
我不确定您要问的是什么,所以我要猜测:
您需要以编程方式触发合成器中的一些MIDI音符,然后将所有音频写入.wav文件,对吗?
假设您已经了解JUCE,那么制作一个用于打开插件,发送MIDI和录制音频的应用程序将是微不足道的,但是调整AudioPluginHost
项目可能更容易。
让我们将其分为几个简单步骤(首先打开AudioPluginHost
项目):
查看GraphEditorPanel.h
,特别是GraphDocumentComponent
类。 它具有一个私有成员变量: MidiKeyboardState keyState;
。 这将收集传入的MIDI消息,然后将它们插入到发送到插件的传入的音频和MIDI缓冲区中。
您可以简单地调用keyState.noteOn (midiChannel, midiNoteNumber, velocity)
和keyState.noteOff (midiChannel, midiNoteNumber, velocity)
来触发音符。
在JUCE中这是一件非常简单的事情-您应该先查看JUCE演示。 下面的示例在后台记录输出音频,但是还有许多其他方法可以执行此操作:
class AudioRecorder : public AudioIODeviceCallback
{
public:
AudioRecorder (AudioThumbnail& thumbnailToUpdate)
: thumbnail (thumbnailToUpdate)
{
backgroundThread.startThread();
}
~AudioRecorder()
{
stop();
}
//==============================================================================
void startRecording (const File& file)
{
stop();
if (sampleRate > 0)
{
// Create an OutputStream to write to our destination file...
file.deleteFile();
ScopedPointer<FileOutputStream> fileStream (file.createOutputStream());
if (fileStream.get() != nullptr)
{
// Now create a WAV writer object that writes to our output stream...
WavAudioFormat wavFormat;
auto* writer = wavFormat.createWriterFor (fileStream.get(), sampleRate, 1, 16, {}, 0);
if (writer != nullptr)
{
fileStream.release(); // (passes responsibility for deleting the stream to the writer object that is now using it)
// Now we'll create one of these helper objects which will act as a FIFO buffer, and will
// write the data to disk on our background thread.
threadedWriter.reset (new AudioFormatWriter::ThreadedWriter (writer, backgroundThread, 32768));
// Reset our recording thumbnail
thumbnail.reset (writer->getNumChannels(), writer->getSampleRate());
nextSampleNum = 0;
// And now, swap over our active writer pointer so that the audio callback will start using it..
const ScopedLock sl (writerLock);
activeWriter = threadedWriter.get();
}
}
}
}
void stop()
{
// First, clear this pointer to stop the audio callback from using our writer object..
{
const ScopedLock sl (writerLock);
activeWriter = nullptr;
}
// Now we can delete the writer object. It's done in this order because the deletion could
// take a little time while remaining data gets flushed to disk, so it's best to avoid blocking
// the audio callback while this happens.
threadedWriter.reset();
}
bool isRecording() const
{
return activeWriter != nullptr;
}
//==============================================================================
void audioDeviceAboutToStart (AudioIODevice* device) override
{
sampleRate = device->getCurrentSampleRate();
}
void audioDeviceStopped() override
{
sampleRate = 0;
}
void audioDeviceIOCallback (const float** inputChannelData, int numInputChannels,
float** outputChannelData, int numOutputChannels,
int numSamples) override
{
const ScopedLock sl (writerLock);
if (activeWriter != nullptr && numInputChannels >= thumbnail.getNumChannels())
{
activeWriter->write (inputChannelData, numSamples);
// Create an AudioBuffer to wrap our incoming data, note that this does no allocations or copies, it simply references our input data
AudioBuffer<float> buffer (const_cast<float**> (inputChannelData), thumbnail.getNumChannels(), numSamples);
thumbnail.addBlock (nextSampleNum, buffer, 0, numSamples);
nextSampleNum += numSamples;
}
// We need to clear the output buffers, in case they're full of junk..
for (int i = 0; i < numOutputChannels; ++i)
if (outputChannelData[i] != nullptr)
FloatVectorOperations::clear (outputChannelData[i], numSamples);
}
private:
AudioThumbnail& thumbnail;
TimeSliceThread backgroundThread { "Audio Recorder Thread" }; // the thread that will write our audio data to disk
ScopedPointer<AudioFormatWriter::ThreadedWriter> threadedWriter; // the FIFO used to buffer the incoming data
double sampleRate = 0.0;
int64 nextSampleNum = 0;
CriticalSection writerLock;
AudioFormatWriter::ThreadedWriter* volatile activeWriter = nullptr;
};
请注意,包含来自插件的音频数据的实际音频回调发生在FilterGraph
的AudioProcessorGraph
内部。 每秒有多次音频回调发生,原始音频数据被传入。在AudioPluginHost
内更改此设置可能很麻烦,除非您知道自己在做什么–使用上面的示例可能会更简单或创建自己的具有自己音频流的应用。
您询问的功能:
AudioBuffer<FloatType> getBusBuffer (AudioBuffer<FloatType>& processBlockBuffer) const
是无关紧要的。 一旦您已经进入音频回调,这将使您将音频发送到插件的总线(又名,如果您的合成器具有侧链)。 相反,您要做的是从回调中获取音频,并将其传递给AudioFormatWriter
,或者最好传递给AudioFormatWriter::ThreadedWriter
以便实际写入发生在不同的线程上。
如果您完全不熟悉C ++或JUCE,则Max / MSP或Pure Data可能会更容易使您快速掌握一些东西。
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