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Replaced all our internal use of the old AudioSampleBuffer name with AudioBuffer<float> (for which AudioSampleBuffer is just a typedef)

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This commit is contained in:
jules 2017-10-30 14:52:47 +00:00
parent 6b45923426
commit aecb819985
67 changed files with 338 additions and 358 deletions
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2
examples/BLOCKS/BlocksSynth/Source/Audio.h
View file

@ -64,7 +64,7 @@ public:
void audioDeviceIOCallback (const float **/*inputChannelData*/, int /*numInputChannels*/,
float **outputChannelData, int numOutputChannels, int numSamples) override
{
AudioSampleBuffer sampleBuffer = AudioSampleBuffer (outputChannelData, numOutputChannels, numSamples);
AudioBuffer<float> sampleBuffer (outputChannelData, numOutputChannels, numSamples);
sampleBuffer.clear();
synthesiser.renderNextBlock (sampleBuffer, MidiBuffer(), 0, numSamples);

2
examples/BLOCKS/BlocksSynth/Source/Oscillators.h
View file

@ -75,7 +75,7 @@ public:
amplitude.setValue (newChannelPressureValue / 127.0);
}
void renderNextBlock (AudioSampleBuffer& outputBuffer, int startSample, int numSamples) override
void renderNextBlock (AudioBuffer<float>& outputBuffer, int startSample, int numSamples) override
{
while (--numSamples >= 0)
{

2
examples/DSP module plugin demo/Source/PluginProcessor.cpp
View file

@ -164,7 +164,7 @@ void DspModulePluginDemoAudioProcessor::process (dsp::ProcessContextReplacing<fl
outputVolume.process (context);
}
void DspModulePluginDemoAudioProcessor::processBlock (AudioSampleBuffer& inoutBuffer, MidiBuffer&)
void DspModulePluginDemoAudioProcessor::processBlock (AudioBuffer<float>& inoutBuffer, MidiBuffer&)
{
auto totalNumInputChannels = getTotalNumInputChannels();
auto totalNumOutputChannels = getTotalNumOutputChannels();

2
examples/DSP module plugin demo/Source/PluginProcessor.h
View file

@ -46,7 +46,7 @@ public:
#endif
void prepareToPlay (double sampleRate, int samplesPerBlock) override;
void releaseResources() override;
void processBlock (AudioSampleBuffer&, MidiBuffer&) override;
void processBlock (AudioBuffer<float>&, MidiBuffer&) override;
void reset() override;
//==============================================================================

4
examples/Demo/Source/Demos/AudioLatencyDemo.cpp
View file

@ -172,7 +172,7 @@ public:
}
private:
AudioSampleBuffer testSound, recordedSound;
AudioBuffer<float> testSound, recordedSound;
Array<int> spikePositions;
int playingSampleNum, recordedSampleNum;
CriticalSection lock;
@ -211,7 +211,7 @@ private:
}
// Searches a buffer for a set of spikes that matches those in the test sound
int findOffsetOfSpikes (const AudioSampleBuffer& buffer) const
int findOffsetOfSpikes (const AudioBuffer<float>& buffer) const
{
const float minSpikeLevel = 5.0f;
const double smooth = 0.975;

4
examples/Demo/Source/Demos/AudioRecordingDemo.cpp
View file

@ -124,8 +124,8 @@ public:
{
activeWriter->write (inputChannelData, numSamples);
// Create an AudioSampleBuffer to wrap our incoming data, note that this does no allocations or copies, it simply references our input data
const AudioSampleBuffer buffer (const_cast<float**> (inputChannelData), thumbnail.getNumChannels(), 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;
}

12
examples/Demo/Source/Demos/AudioSynthesiserDemo.cpp
View file

@ -42,9 +42,7 @@ struct SineWaveSound : public SynthesiserSound
/** Our demo synth voice just plays a sine wave.. */
struct SineWaveVoice : public SynthesiserVoice
{
SineWaveVoice() : currentAngle (0), angleDelta (0), level (0), tailOff (0)
{
}
SineWaveVoice() {}
bool canPlaySound (SynthesiserSound* sound) override
{
@ -94,7 +92,7 @@ struct SineWaveVoice : public SynthesiserVoice
// not interested in controllers in this case.
}
void renderNextBlock (AudioSampleBuffer& outputBuffer, int startSample, int numSamples) override
void renderNextBlock (AudioBuffer<float>& outputBuffer, int startSample, int numSamples) override
{
if (angleDelta != 0.0)
{
@ -102,7 +100,7 @@ struct SineWaveVoice : public SynthesiserVoice
{
while (--numSamples >= 0)
{
const float currentSample = (float) (std::sin (currentAngle) * level * tailOff);
auto currentSample = (float) (std::sin (currentAngle) * level * tailOff);
for (int i = outputBuffer.getNumChannels(); --i >= 0;)
outputBuffer.addSample (i, startSample, currentSample);
@ -125,7 +123,7 @@ struct SineWaveVoice : public SynthesiserVoice
{
while (--numSamples >= 0)
{
const float currentSample = (float) (std::sin (currentAngle) * level);
auto currentSample = (float) (std::sin (currentAngle) * level);
for (int i = outputBuffer.getNumChannels(); --i >= 0;)
outputBuffer.addSample (i, startSample, currentSample);
@ -138,7 +136,7 @@ struct SineWaveVoice : public SynthesiserVoice
}
private:
double currentAngle, angleDelta, level, tailOff;
double currentAngle = 0, angleDelta = 0, level = 0, tailOff = 0;
};
//==============================================================================

11
examples/PlugInSamples/Arpeggiator/Source/Arpeggiator.cpp
View file

@ -57,16 +57,16 @@ public:
void releaseResources() override {}
void processBlock (AudioSampleBuffer& buffer, MidiBuffer& midi) override
void processBlock (AudioBuffer<float>& buffer, MidiBuffer& midi) override
{
// the audio buffer in a midi effect will have zero channels!
jassert (buffer.getNumChannels() == 0);
// however we use the buffer to get timing information
const int numSamples = buffer.getNumSamples();
auto numSamples = buffer.getNumSamples();
// get note duration
const int noteDuration = static_cast<int> (std::ceil (rate * 0.25f * (0.1f + (1.0f - (*speed)))));
auto noteDuration = static_cast<int> (std::ceil (rate * 0.25f * (0.1f + (1.0f - (*speed)))));
MidiMessage msg;
int ignore;
@ -81,7 +81,7 @@ public:
if ((time + numSamples) >= noteDuration)
{
const int offset = jmax (0, jmin ((int) (noteDuration - time), numSamples - 1));
auto offset = jmax (0, jmin ((int) (noteDuration - time), numSamples - 1));
if (lastNoteValue > 0)
{
@ -133,9 +133,6 @@ public:
speed->setValueNotifyingHost (MemoryInputStream (data, static_cast<size_t> (sizeInBytes), false).readFloat());
}
//==============================================================================
private:
//==============================================================================
AudioParameterFloat* speed;

2
examples/PlugInSamples/GainPlugIn/Source/GainProcessor.cpp
View file

@ -48,7 +48,7 @@ public:
void prepareToPlay (double, int) override {}
void releaseResources() override {}
void processBlock (AudioSampleBuffer& buffer, MidiBuffer&) override
void processBlock (AudioBuffer<float>& buffer, MidiBuffer&) override
{
buffer.applyGain (*gain);
}

9
examples/PlugInSamples/InterAppAudioEffect/Source/IAAEffectProcessor.cpp
View file

@ -114,14 +114,14 @@ bool IAAEffectProcessor::isBusesLayoutSupported (const BusesLayout& layouts) con
return true;
}
void IAAEffectProcessor::processBlock (AudioSampleBuffer& buffer, MidiBuffer&)
void IAAEffectProcessor::processBlock (AudioBuffer<float>& buffer, MidiBuffer&)
{
const float gain = *parameters.getRawParameterValue ("gain");
const int totalNumInputChannels = getTotalNumInputChannels();
const int totalNumOutputChannels = getTotalNumOutputChannels();
auto totalNumInputChannels = getTotalNumInputChannels();
auto totalNumOutputChannels = getTotalNumOutputChannels();
const int numSamples = buffer.getNumSamples();
auto numSamples = buffer.getNumSamples();
for (int i = totalNumInputChannels; i < totalNumOutputChannels; ++i)
buffer.clear (i, 0, buffer.getNumSamples());
@ -164,6 +164,7 @@ void IAAEffectProcessor::getStateInformation (MemoryBlock& destData)
void IAAEffectProcessor::setStateInformation (const void* data, int sizeInBytes)
{
auto xmlState = std::unique_ptr<XmlElement> (getXmlFromBinary (data, sizeInBytes));
if (xmlState.get() != nullptr)
if (xmlState->hasTagName (parameters.state.getType()))
parameters.state = ValueTree::fromXml (*xmlState);

2
examples/PlugInSamples/InterAppAudioEffect/Source/IAAEffectProcessor.h
View file

@ -44,7 +44,7 @@ public:
bool isBusesLayoutSupported (const BusesLayout& layouts) const override;
void processBlock (AudioSampleBuffer&, MidiBuffer&) override;
void processBlock (AudioBuffer<float>&, MidiBuffer&) override;
//==============================================================================
AudioProcessorEditor* createEditor() override;

7
examples/PlugInSamples/MultiOutSynth/Source/MultiOutSynth.cpp
View file

@ -90,13 +90,14 @@ public:
void releaseResources() override {}
void processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiBuffer) override
void processBlock (AudioBuffer<float>& buffer, MidiBuffer& midiBuffer) override
{
const int busCount = getBusCount (false);
auto busCount = getBusCount (false);
for (int busNr = 0; busNr < busCount; ++busNr)
{
MidiBuffer midiChannelBuffer = filterMidiMessagesForChannel (midiBuffer, busNr + 1);
AudioSampleBuffer audioBusBuffer = getBusBuffer (buffer, false, busNr);
auto audioBusBuffer = getBusBuffer (buffer, false, busNr);
synth [busNr]->renderNextBlock (audioBusBuffer, midiChannelBuffer, 0, audioBusBuffer.getNumSamples());
}

11
examples/PlugInSamples/NoiseGate/Source/NoiseGate.cpp
View file

@ -47,18 +47,18 @@ public:
bool isBusesLayoutSupported (const BusesLayout& layouts) const override
{
// the sidechain can take any layout, the main bus needs to be the same on the input and output
return (layouts.getMainInputChannelSet() == layouts.getMainOutputChannelSet() &&
(! layouts.getMainInputChannelSet().isDisabled()));
return layouts.getMainInputChannelSet() == layouts.getMainOutputChannelSet()
&& ! layouts.getMainInputChannelSet().isDisabled();
}
//==============================================================================
void prepareToPlay (double /*sampleRate*/, int /*maxBlockSize*/) override { lowPassCoeff = 0.0f; sampleCountDown = 0; }
void releaseResources() override {}
void processBlock (AudioSampleBuffer& buffer, MidiBuffer&) override
void processBlock (AudioBuffer<float>& buffer, MidiBuffer&) override
{
AudioSampleBuffer mainInputOutput = getBusBuffer(buffer, true, 0);
AudioSampleBuffer sideChainInput = getBusBuffer(buffer, true, 1);
auto mainInputOutput = getBusBuffer (buffer, true, 0);
auto sideChainInput = getBusBuffer (buffer, true, 1);
float alphaCopy = *alpha;
float thresholdCopy = *threshold;
@ -66,6 +66,7 @@ public:
for (int j = 0; j < buffer.getNumSamples(); ++j)
{
float mixedSamples = 0.0f;
for (int i = 0; i < sideChainInput.getNumChannels(); ++i)
mixedSamples += sideChainInput.getReadPointer (i) [j];

9
examples/PlugInSamples/Surround/Source/SurroundProcessor.cpp
View file

@ -60,7 +60,7 @@ public:
void releaseResources() override { reset(); }
void processBlock (AudioSampleBuffer& buffer, MidiBuffer&) override
void processBlock (AudioBuffer<float>& buffer, MidiBuffer&) override
{
for (int ch = 0; ch < buffer.getNumChannels(); ++ch)
{
@ -79,7 +79,7 @@ public:
channelTime = jmax (0, channelTime - buffer.getNumSamples());
}
const int fillSamples = jmin (static_cast<int> (std::ceil (getSampleRate())) - sampleOffset,
auto fillSamples = jmin (static_cast<int> (std::ceil (getSampleRate())) - sampleOffset,
buffer.getNumSamples());
if (isPositiveAndBelow (channelClicked, buffer.getNumChannels()))
@ -141,8 +141,8 @@ public:
void handleAsyncUpdate() override
{
if (AudioProcessorEditor* editor = getActiveEditor())
if (SurroundEditor* surroundEditor = dynamic_cast<SurroundEditor*> (editor))
if (auto* editor = getActiveEditor())
if (auto* surroundEditor = dynamic_cast<SurroundEditor*> (editor))
surroundEditor->updateGUI();
}
@ -151,6 +151,7 @@ private:
Array<float> alphaCoeffs;
int channelClicked;
int sampleOffset;
//==============================================================================
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (SurroundProcessor)
};

2
examples/audio plugin demo/Source/SinewaveSynth.h
View file

@ -94,7 +94,7 @@ public:
// not implemented for the purposes of this demo!
}
void renderNextBlock (AudioSampleBuffer& outputBuffer, int startSample, int numSamples) override
void renderNextBlock (AudioBuffer<float>& outputBuffer, int startSample, int numSamples) override
{
if (angleDelta != 0.0)
{

12
extras/Projucer/JuceLibraryCode/BinaryData.cpp
View file

@ -5709,11 +5709,11 @@ static const unsigned char temp_binary_data_28[] =
"}\r\n"
"#endif\r\n"
"\r\n"
"void FILTERCLASSNAME::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)\r\n"
"void FILTERCLASSNAME::processBlock (AudioBuffer<float>& buffer, MidiBuffer& midiMessages)\r\n"
"{\r\n"
" ScopedNoDenormals noDenormals;\r\n"
" const int totalNumInputChannels = getTotalNumInputChannels();\r\n"
" const int totalNumOutputChannels = getTotalNumOutputChannels();\r\n"
" auto totalNumInputChannels = getTotalNumInputChannels();\r\n"
" auto totalNumOutputChannels = getTotalNumOutputChannels();\r\n"
"\r\n"
" // In case we have more outputs than inputs, this code clears any output\r\n"
" // channels that didn't contain input data, (because these aren't\r\n"
@ -5803,7 +5803,7 @@ static const unsigned char temp_binary_data_29[] =
" bool isBusesLayoutSupported (const BusesLayout& layouts) const override;\r\n"
" #endif\r\n"
"\r\n"
" void processBlock (AudioSampleBuffer&, MidiBuffer&) override;\r\n"
" void processBlock (AudioBuffer<float>&, MidiBuffer&) override;\r\n"
"\r\n"
" //==============================================================================\r\n"
" AudioProcessorEditor* createEditor() override;\r\n"
@ -7014,8 +7014,8 @@ const char* getNamedResource (const char* resourceNameUTF8, int& numBytes) throw
case 0xafccbd3f: numBytes = 3141; return jucer_AudioComponentTemplate_cpp;
case 0x27c5a93a: numBytes = 1310; return jucer_AudioPluginEditorTemplate_cpp;
case 0x4d0721bf: numBytes = 938; return jucer_AudioPluginEditorTemplate_h;
case 0x51b49ac5: numBytes = 5647; return jucer_AudioPluginFilterTemplate_cpp;
case 0x488afa0a: numBytes = 2245; return jucer_AudioPluginFilterTemplate_h;
case 0x51b49ac5: numBytes = 5638; return jucer_AudioPluginFilterTemplate_cpp;
case 0x488afa0a: numBytes = 2246; return jucer_AudioPluginFilterTemplate_h;
case 0xabad7041: numBytes = 2151; return jucer_ComponentTemplate_cpp;
case 0xfc72fe86: numBytes = 2064; return jucer_ComponentTemplate_h;
case 0x0b66646c: numBytes = 1029; return jucer_ContentCompTemplate_cpp;

4
extras/Projucer/JuceLibraryCode/BinaryData.h
View file

@ -93,10 +93,10 @@ namespace BinaryData
const int jucer_AudioPluginEditorTemplate_hSize = 938;
extern const char* jucer_AudioPluginFilterTemplate_cpp;
const int jucer_AudioPluginFilterTemplate_cppSize = 5647;
const int jucer_AudioPluginFilterTemplate_cppSize = 5638;
extern const char* jucer_AudioPluginFilterTemplate_h;
const int jucer_AudioPluginFilterTemplate_hSize = 2245;
const int jucer_AudioPluginFilterTemplate_hSize = 2246;
extern const char* jucer_ComponentTemplate_cpp;
const int jucer_ComponentTemplate_cppSize = 2151;

6
extras/Projucer/Source/BinaryData/Templates/jucer_AudioPluginFilterTemplate.cpp
View file

@ -129,11 +129,11 @@ bool FILTERCLASSNAME::isBusesLayoutSupported (const BusesLayout& layouts) const
}
#endif
void FILTERCLASSNAME::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
void FILTERCLASSNAME::processBlock (AudioBuffer<float>& buffer, MidiBuffer& midiMessages)
{
ScopedNoDenormals noDenormals;
const int totalNumInputChannels = getTotalNumInputChannels();
const int totalNumOutputChannels = getTotalNumOutputChannels();
auto totalNumInputChannels = getTotalNumInputChannels();
auto totalNumOutputChannels = getTotalNumOutputChannels();
// In case we have more outputs than inputs, this code clears any output
// channels that didn't contain input data, (because these aren't

2
extras/Projucer/Source/BinaryData/Templates/jucer_AudioPluginFilterTemplate.h
View file

@ -31,7 +31,7 @@ public:
bool isBusesLayoutSupported (const BusesLayout& layouts) const override;
#endif
void processBlock (AudioSampleBuffer&, MidiBuffer&) override;
void processBlock (AudioBuffer<float>&, MidiBuffer&) override;
//==============================================================================
AudioProcessorEditor* createEditor() override;

4
modules/juce_audio_basics/buffers/juce_AudioSampleBuffer.h
View file

@ -25,9 +25,7 @@ namespace juce
//==============================================================================
/**
A multi-channel buffer of floating point audio samples.
@see AudioSampleBuffer
A multi-channel buffer containing floating point audio samples.
*/
template <typename Type>
class AudioBuffer

2
modules/juce_audio_basics/midi/juce_MidiBuffer.h
View file

@ -27,7 +27,7 @@ namespace juce
/**
Holds a sequence of time-stamped midi events.
Analogous to the AudioSampleBuffer, this holds a set of midi events with
Analogous to the AudioBuffer, this holds a set of midi events with
integer time-stamps. The buffer is kept sorted in order of the time-stamps.
If you're working with a sequence of midi events that may need to be manipulated

6
modules/juce_audio_basics/sources/juce_AudioSource.h
View file

@ -35,7 +35,7 @@ struct JUCE_API AudioSourceChannelInfo
}
/** Creates an AudioSourceChannelInfo. */
AudioSourceChannelInfo (AudioSampleBuffer* bufferToUse,
AudioSourceChannelInfo (AudioBuffer<float>* bufferToUse,
int startSampleOffset, int numSamplesToUse) noexcept
: buffer (bufferToUse),
startSample (startSampleOffset),
@ -47,7 +47,7 @@ struct JUCE_API AudioSourceChannelInfo
Note that the buffer provided must not be deleted while the
AudioSourceChannelInfo is still using it.
*/
explicit AudioSourceChannelInfo (AudioSampleBuffer& bufferToUse) noexcept
explicit AudioSourceChannelInfo (AudioBuffer<float>& bufferToUse) noexcept
: buffer (&bufferToUse),
startSample (0),
numSamples (bufferToUse.getNumSamples())
@ -70,7 +70,7 @@ struct JUCE_API AudioSourceChannelInfo
The number of channels in the buffer could be anything, so the AudioSource
must cope with this in whatever way is appropriate for its function.
*/
AudioSampleBuffer* buffer;
AudioBuffer<float>* buffer;
/** The first sample in the buffer from which the callback is expected
to write data. */

2
modules/juce_audio_basics/sources/juce_BufferingAudioSource.h
View file

@ -100,7 +100,7 @@ private:
OptionalScopedPointer<PositionableAudioSource> source;
TimeSliceThread& backgroundThread;
int numberOfSamplesToBuffer, numberOfChannels;
AudioSampleBuffer buffer;
AudioBuffer<float> buffer;
CriticalSection bufferStartPosLock;
WaitableEvent bufferReadyEvent;
int64 volatile bufferValidStart, bufferValidEnd, nextPlayPos;

2
modules/juce_audio_basics/sources/juce_ChannelRemappingAudioSource.h
View file

@ -129,7 +129,7 @@ private:
Array<int> remappedInputs, remappedOutputs;
int requiredNumberOfChannels;
AudioSampleBuffer buffer;
AudioBuffer<float> buffer;
AudioSourceChannelInfo remappedInfo;
CriticalSection lock;

2
modules/juce_audio_basics/sources/juce_MixerAudioSource.h
View file

@ -87,7 +87,7 @@ private:
Array<AudioSource*> inputs;
BigInteger inputsToDelete;
CriticalSection lock;
AudioSampleBuffer tempBuffer;
AudioBuffer<float> tempBuffer;
double currentSampleRate;
int bufferSizeExpected;

2
modules/juce_audio_basics/sources/juce_ResamplingAudioSource.h
View file

@ -75,7 +75,7 @@ private:
//==============================================================================
OptionalScopedPointer<AudioSource> input;
double ratio, lastRatio;
AudioSampleBuffer buffer;
AudioBuffer<float> buffer;
int bufferPos, sampsInBuffer;
double subSampleOffset;
double coefficients[6];

4
modules/juce_audio_devices/audio_io/juce_AudioDeviceManager.cpp
View file

@ -970,7 +970,7 @@ double AudioDeviceManager::LevelMeter::getCurrentLevel() const noexcept
void AudioDeviceManager::playTestSound()
{
{ // cunningly nested to swap, unlock and delete in that order.
ScopedPointer<AudioSampleBuffer> oldSound;
ScopedPointer<AudioBuffer<float>> oldSound;
{
const ScopedLock sl (audioCallbackLock);
@ -990,7 +990,7 @@ void AudioDeviceManager::playTestSound()
const double phasePerSample = double_Pi * 2.0 / (sampleRate / frequency);
AudioSampleBuffer* const newSound = new AudioSampleBuffer (1, soundLength);
auto* newSound = new AudioBuffer<float> (1, soundLength);
for (int i = 0; i < soundLength; ++i)
newSound->setSample (0, i, amplitude * (float) std::sin (i * phasePerSample));

4
modules/juce_audio_devices/audio_io/juce_AudioDeviceManager.h
View file

@ -458,7 +458,7 @@ private:
BigInteger inputChannels, outputChannels;
ScopedPointer<XmlElement> lastExplicitSettings;
mutable bool listNeedsScanning;
AudioSampleBuffer tempBuffer;
AudioBuffer<float> tempBuffer;
struct MidiCallbackInfo
{
@ -474,7 +474,7 @@ private:
ScopedPointer<MidiOutput> defaultMidiOutput;
CriticalSection audioCallbackLock, midiCallbackLock;
ScopedPointer<AudioSampleBuffer> testSound;
ScopedPointer<AudioBuffer<float>> testSound;
int testSoundPosition;
double cpuUsageMs, timeToCpuScale, msPerBlock;

2
modules/juce_audio_devices/native/juce_android_Audio.cpp
View file

@ -423,7 +423,7 @@ private:
String lastError;
BigInteger activeOutputChans, activeInputChans;
GlobalRef outputDevice, inputDevice;
AudioSampleBuffer inputChannelBuffer, outputChannelBuffer;
AudioBuffer<float> inputChannelBuffer, outputChannelBuffer;
jmethodID getUnderrunCount = 0;
void closeDevices()

14
modules/juce_audio_devices/native/juce_android_OpenSL.cpp
View file

@ -166,9 +166,9 @@ struct BufferHelpers<int16>
dataFormat.representation = 0;
}
static void prepareCallbackBuffer (AudioSampleBuffer&, int16*) {}
static void prepareCallbackBuffer (AudioBuffer<float>&, int16*) {}
static void convertFromOpenSL (const int16* srcInterleaved, AudioSampleBuffer& audioBuffer)
static void convertFromOpenSL (const int16* srcInterleaved, AudioBuffer<float>& audioBuffer)
{
for (int i = 0; i < audioBuffer.getNumChannels(); ++i)
{
@ -181,7 +181,7 @@ struct BufferHelpers<int16>
}
}
static void convertToOpenSL (const AudioSampleBuffer& audioBuffer, int16* dstInterleaved)
static void convertToOpenSL (const AudioBuffer<float>& audioBuffer, int16* dstInterleaved)
{
for (int i = 0; i < audioBuffer.getNumChannels(); ++i)
{
@ -215,13 +215,13 @@ struct BufferHelpers<float>
dataFormat.representation = SL_ANDROID_PCM_REPRESENTATION_FLOAT;
}
static void prepareCallbackBuffer (AudioSampleBuffer& audioBuffer, float* native)
static void prepareCallbackBuffer (AudioBuffer<float>& audioBuffer, float* native)
{
if (audioBuffer.getNumChannels() == 1)
audioBuffer.setDataToReferTo (&native, 1, audioBuffer.getNumSamples());
}
static void convertFromOpenSL (const float* srcInterleaved, AudioSampleBuffer& audioBuffer)
static void convertFromOpenSL (const float* srcInterleaved, AudioBuffer<float>& audioBuffer)
{
if (audioBuffer.getNumChannels() == 1)
{
@ -240,7 +240,7 @@ struct BufferHelpers<float>
}
}
static void convertToOpenSL (const AudioSampleBuffer& audioBuffer, float* dstInterleaved)
static void convertToOpenSL (const AudioBuffer<float>& audioBuffer, float* dstInterleaved)
{
if (audioBuffer.getNumChannels() == 1)
{
@ -374,7 +374,7 @@ public:
int numChannels;
HeapBlock<T> nativeBuffer;
AudioSampleBuffer scratchBuffer, sampleBuffer;
AudioBuffer<float> scratchBuffer, sampleBuffer;
Atomic<int> nextBlock, numBlocksOut;
};

6
modules/juce_audio_devices/native/juce_linux_ALSA.cpp
View file

@ -310,7 +310,7 @@ public:
}
//==============================================================================
bool writeToOutputDevice (AudioSampleBuffer& outputChannelBuffer, const int numSamples)
bool writeToOutputDevice (AudioBuffer<float>& outputChannelBuffer, const int numSamples)
{
jassert (numChannelsRunning <= outputChannelBuffer.getNumChannels());
float* const* const data = outputChannelBuffer.getArrayOfWritePointers();
@ -348,7 +348,7 @@ public:
return true;
}
bool readFromInputDevice (AudioSampleBuffer& inputChannelBuffer, const int numSamples)
bool readFromInputDevice (AudioBuffer<float>& inputChannelBuffer, const int numSamples)
{
jassert (numChannelsRunning <= inputChannelBuffer.getNumChannels());
float* const* const data = inputChannelBuffer.getArrayOfWritePointers();
@ -801,7 +801,7 @@ private:
CriticalSection callbackLock;
AudioSampleBuffer inputChannelBuffer, outputChannelBuffer;
AudioBuffer<float> inputChannelBuffer, outputChannelBuffer;
Array<const float*> inputChannelDataForCallback;
Array<float*> outputChannelDataForCallback;

318
modules/juce_audio_devices/native/juce_mac_CoreAudio.cpp
View file

@ -209,7 +209,7 @@ public:
};
// returns the number of actual available channels
StringArray getChannelInfo (const bool input, Array<CallbackDetailsForChannel>& newChannelInfo) const
StringArray getChannelInfo (bool input, Array<CallbackDetailsForChannel>& newChannelInfo) const
{
StringArray newNames;
int chanNum = 0;
@ -300,7 +300,7 @@ public:
}
}
if (newSampleRates.size() == 0 && sampleRate > 0)
if (newSampleRates.isEmpty() && sampleRate > 0)
newSampleRates.add (sampleRate);
return newSampleRates;
@ -342,7 +342,7 @@ public:
}
}
if (newBufferSizes.size() == 0 && bufferSize > 0)
if (newBufferSizes.isEmpty() && bufferSize > 0)
newBufferSizes.add (bufferSize);
return newBufferSizes;
@ -419,20 +419,20 @@ public:
// this collects all the new details from the device without any locking, then
// locks + swaps them afterwards.
const double newSampleRate = getNominalSampleRate();
const int newBufferSize = getFrameSizeFromDevice();
auto newSampleRate = getNominalSampleRate();
auto newBufferSize = getFrameSizeFromDevice();
Array<int> newBufferSizes = getBufferSizesFromDevice();
Array<double> newSampleRates = getSampleRatesFromDevice();
auto newBufferSizes = getBufferSizesFromDevice();
auto newSampleRates = getSampleRatesFromDevice();
const int newInputLatency = getLatencyFromDevice (kAudioDevicePropertyScopeInput);
const int newOutputLatency = getLatencyFromDevice (kAudioDevicePropertyScopeOutput);
auto newInputLatency = getLatencyFromDevice (kAudioDevicePropertyScopeInput);
auto newOutputLatency = getLatencyFromDevice (kAudioDevicePropertyScopeOutput);
Array<CallbackDetailsForChannel> newInChans, newOutChans;
auto newInNames = isInputDevice ? getChannelInfo (true, newInChans) : StringArray();
auto newOutNames = isOutputDevice ? getChannelInfo (false, newOutChans) : StringArray();
const int newBitDepth = jmax (getBitDepthFromDevice (kAudioDevicePropertyScopeInput),
auto newBitDepth = jmax (getBitDepthFromDevice (kAudioDevicePropertyScopeInput),
getBitDepthFromDevice (kAudioDevicePropertyScopeOutput));
{
@ -467,7 +467,7 @@ public:
{
StringArray s;
HeapBlock<OSType> types;
const int num = getAllDataSourcesForDevice (deviceID, types);
auto num = getAllDataSourcesForDevice (deviceID, types);
for (int i = 0; i < num; ++i)
{
@ -722,11 +722,10 @@ public:
{
for (int i = numInputChans; --i >= 0;)
{
const CallbackDetailsForChannel& info = inputChannelInfo.getReference(i);
float* dest = tempInputBuffers [i];
const float* src = ((const float*) inInputData->mBuffers[info.streamNum].mData)
+ info.dataOffsetSamples;
const int stride = info.dataStrideSamples;
auto& info = inputChannelInfo.getReference(i);
auto dest = tempInputBuffers[i];
auto src = ((const float*) inInputData->mBuffers[info.streamNum].mData) + info.dataOffsetSamples;
auto stride = info.dataStrideSamples;
if (stride != 0) // if this is zero, info is invalid
{
@ -746,11 +745,10 @@ public:
for (int i = numOutputChans; --i >= 0;)
{
const CallbackDetailsForChannel& info = outputChannelInfo.getReference(i);
const float* src = tempOutputBuffers [i];
float* dest = ((float*) outOutputData->mBuffers[info.streamNum].mData)
+ info.dataOffsetSamples;
const int stride = info.dataStrideSamples;
auto& info = outputChannelInfo.getReference(i);
auto src = tempOutputBuffers[i];
auto dest = ((float*) outOutputData->mBuffers[info.streamNum].mData) + info.dataOffsetSamples;
auto stride = info.dataStrideSamples;
if (stride != 0) // if this is zero, info is invalid
{
@ -1145,8 +1143,8 @@ public:
{
Array<AudioIODevice*> devs;
for (int i = 0; i < devices.size(); ++i)
devs.add (devices.getUnchecked(i)->device);
for (auto* d : devices)
devs.add (d->device);
return devs;
}
@ -1155,8 +1153,8 @@ public:
{
StringArray names;
for (int i = 0; i < devices.size(); ++i)
names.addArray (devices.getUnchecked(i)->getOutputChannelNames());
for (auto* d : devices)
names.addArray (d->getOutputChannelNames());
names.appendNumbersToDuplicates (false, true);
return names;
@ -1166,8 +1164,8 @@ public:
{
StringArray names;
for (int i = 0; i < devices.size(); ++i)
names.addArray (devices.getUnchecked(i)->getInputChannelNames());
for (auto* d : devices)
names.addArray (d->getInputChannelNames());
names.appendNumbersToDuplicates (false, true);
return names;
@ -1176,16 +1174,22 @@ public:
Array<double> getAvailableSampleRates() override
{
Array<double> commonRates;
bool first = true;
for (int i = 0; i < devices.size(); ++i)
for (auto* d : devices)
{
Array<double> rates (devices.getUnchecked(i)->device->getAvailableSampleRates());
auto rates = d->device->getAvailableSampleRates();
if (i == 0)
if (first)
{
first = false;
commonRates = rates;
}
else
{
commonRates.removeValuesNotIn (rates);
}
}
return commonRates;
}
@ -1193,16 +1197,22 @@ public:
Array<int> getAvailableBufferSizes() override
{
Array<int> commonSizes;
bool first = true;
for (int i = 0; i < devices.size(); ++i)
for (auto* d : devices)
{
Array<int> sizes (devices.getUnchecked(i)->device->getAvailableBufferSizes());
auto sizes = d->device->getAvailableBufferSizes();
if (i == 0)
if (first)
{
first = false;
commonSizes = sizes;
}
else
{
commonSizes.removeValuesNotIn (sizes);
}
}
return commonSizes;
}
@ -1216,8 +1226,8 @@ public:
{
int depth = 32;
for (int i = 0; i < devices.size(); ++i)
depth = jmin (depth, devices.getUnchecked(i)->device->getCurrentBitDepth());
for (auto* d : devices)
depth = jmin (depth, d->device->getCurrentBitDepth());
return depth;
}
@ -1226,8 +1236,8 @@ public:
{
int size = 0;
for (int i = 0; i < devices.size(); ++i)
size = jmax (size, devices.getUnchecked(i)->device->getDefaultBufferSize());
for (auto* d : devices)
size = jmax (size, d->device->getDefaultBufferSize());
return size;
}
@ -1244,7 +1254,7 @@ public:
if (sampleRate <= 0)
{
Array<double> rates (getAvailableSampleRates());
auto rates = getAvailableSampleRates();
for (int i = 0; i < rates.size() && sampleRate < 44100.0; ++i)
sampleRate = rates.getUnchecked(i);
@ -1257,20 +1267,18 @@ public:
int totalInputChanIndex = 0, totalOutputChanIndex = 0;
int chanIndex = 0;
for (int i = 0; i < devices.size(); ++i)
for (auto* d : devices)
{
DeviceWrapper& d = *devices.getUnchecked(i);
BigInteger ins (inputChannels >> totalInputChanIndex);
BigInteger outs (outputChannels >> totalOutputChanIndex);
int numIns = d.getInputChannelNames().size();
int numOuts = d.getOutputChannelNames().size();
int numIns = d->getInputChannelNames().size();
int numOuts = d->getOutputChannelNames().size();
totalInputChanIndex += numIns;
totalOutputChanIndex += numOuts;
String err = d.open (ins, outs, sampleRate, bufferSize,
String err = d->open (ins, outs, sampleRate, bufferSize,
chanIndex, fifoSize);
if (err.isNotEmpty())
@ -1280,7 +1288,7 @@ public:
return err;
}
chanIndex += d.numInputChans + d.numOutputChans;
chanIndex += d->numInputChans + d->numOutputChans;
}
fifos.setSize (chanIndex, fifoSize);
@ -1297,8 +1305,8 @@ public:
fifos.clear();
active = false;
for (int i = 0; i < devices.size(); ++i)
devices.getUnchecked(i)->close();
for (auto* d : devices)
d->close();
}
BigInteger getActiveOutputChannels() const override
@ -1306,13 +1314,13 @@ public:
BigInteger chans;
int start = 0;
for (int i = 0; i < devices.size(); ++i)
for (auto* d : devices)
{
const int numChans = devices.getUnchecked(i)->getOutputChannelNames().size();
auto numChans = d->getOutputChannelNames().size();
if (numChans > 0)
{
chans |= (devices.getUnchecked(i)->device->getActiveOutputChannels() << start);
chans |= (d->device->getActiveOutputChannels() << start);
start += numChans;
}
}
@ -1325,13 +1333,13 @@ public:
BigInteger chans;
int start = 0;
for (int i = 0; i < devices.size(); ++i)
for (auto* d : devices)
{
const int numChans = devices.getUnchecked(i)->getInputChannelNames().size();
auto numChans = d->getInputChannelNames().size();
if (numChans > 0)
{
chans |= (devices.getUnchecked(i)->device->getActiveInputChannels() << start);
chans |= (d->device->getActiveInputChannels() << start);
start += numChans;
}
}
@ -1343,8 +1351,8 @@ public:
{
int lat = 0;
for (int i = 0; i < devices.size(); ++i)
lat = jmax (lat, devices.getUnchecked(i)->device->getOutputLatencyInSamples());
for (auto* d : devices)
lat = jmax (lat, d->device->getOutputLatencyInSamples());
return lat + currentBufferSize * 2;
}
@ -1353,8 +1361,8 @@ public:
{
int lat = 0;
for (int i = 0; i < devices.size(); ++i)
lat = jmax (lat, devices.getUnchecked(i)->device->getInputLatencyInSamples());
for (auto* d : devices)
lat = jmax (lat, d->device->getInputLatencyInSamples());
return lat + currentBufferSize * 2;
}
@ -1366,8 +1374,8 @@ public:
stop();
fifos.clear();
for (int i = 0; i < devices.size(); ++i)
devices.getUnchecked(i)->start();
for (auto* d : devices)
d->start();
if (newCallback != nullptr)
newCallback->audioDeviceAboutToStart (this);
@ -1392,34 +1400,31 @@ private:
int currentBufferSize = 0;
bool active = false;
String lastError;
AudioSampleBuffer fifos;
AudioBuffer<float> fifos;
void run() override
{
const int numSamples = currentBufferSize;
auto numSamples = currentBufferSize;
AudioSampleBuffer buffer (fifos.getNumChannels(), numSamples);
AudioBuffer<float> buffer (fifos.getNumChannels(), numSamples);
buffer.clear();
Array<const float*> inputChans;
Array<float*> outputChans;
for (int i = 0; i < devices.size(); ++i)
for (auto* d : devices)
{
DeviceWrapper& d = *devices.getUnchecked(i);
for (int j = 0; j < d.numInputChans; ++j) inputChans.add (buffer.getReadPointer (d.inputIndex + j));
for (int j = 0; j < d.numOutputChans; ++j) outputChans.add (buffer.getWritePointer (d.outputIndex + j));
for (int j = 0; j < d->numInputChans; ++j) inputChans.add (buffer.getReadPointer (d->inputIndex + j));
for (int j = 0; j < d->numOutputChans; ++j) outputChans.add (buffer.getWritePointer (d->outputIndex + j));
}
const int numInputChans = inputChans.size();
const int numOutputChans = outputChans.size();
auto numInputChans = inputChans.size();
auto numOutputChans = outputChans.size();
inputChans.add (nullptr);
outputChans.add (nullptr);
const int blockSizeMs = jmax (1, (int) (1000 * numSamples / currentSampleRate));
auto blockSizeMs = jmax (1, (int) (1000 * numSamples / currentSampleRate));
jassert (numInputChans + numOutputChans == buffer.getNumChannels());
@ -1462,8 +1467,8 @@ private:
std::swap (callback, lastCallback);
}
for (int i = 0; i < devices.size(); ++i)
devices.getUnchecked(i)->device->stop();
for (auto* d : devices)
d->device->stop();
if (lastCallback != nullptr)
{
@ -1476,36 +1481,31 @@ private:
void reset()
{
for (int i = 0; i < devices.size(); ++i)
devices.getUnchecked(i)->reset();
for (auto* d : devices)
d->reset();
}
void underrun()
{
}
void readInput (AudioSampleBuffer& buffer, const int numSamples, const int blockSizeMs)
void readInput (AudioBuffer<float>& buffer, const int numSamples, const int blockSizeMs)
{
for (int i = 0; i < devices.size(); ++i)
{
DeviceWrapper& d = *devices.getUnchecked(i);
d.done = (d.numInputChans == 0);
}
for (auto* d : devices)
d->done = (d->numInputChans == 0);
for (int tries = 5;;)
{
bool anyRemaining = false;
for (int i = 0; i < devices.size(); ++i)
for (auto* d : devices)
{
DeviceWrapper& d = *devices.getUnchecked(i);
if (! d.done)
if (! d->done)
{
if (d.isInputReady (numSamples))
if (d->isInputReady (numSamples))
{
d.readInput (buffer, numSamples);
d.done = true;
d->readInput (buffer, numSamples);
d->done = true;
}
else
anyRemaining = true;
@ -1521,38 +1521,29 @@ private:
wait (blockSizeMs);
}
for (int j = 0; j < devices.size(); ++j)
{
DeviceWrapper& d = *devices.getUnchecked(j);
if (! d.done)
for (int i = 0; i < d.numInputChans; ++i)
buffer.clear (d.inputIndex + i, 0, numSamples);
}
for (auto* d : devices)
if (! d->done)
for (int i = 0; i < d->numInputChans; ++i)
buffer.clear (d->inputIndex + i, 0, numSamples);
}
void pushOutputData (AudioSampleBuffer& buffer, const int numSamples, const int blockSizeMs)
void pushOutputData (AudioBuffer<float>& buffer, const int numSamples, const int blockSizeMs)
{
for (int i = 0; i < devices.size(); ++i)
{
DeviceWrapper& d = *devices.getUnchecked(i);
d.done = (d.numOutputChans == 0);
}
for (auto* d : devices)
d->done = (d->numOutputChans == 0);
for (int tries = 5;;)
{
bool anyRemaining = false;
for (int i = 0; i < devices.size(); ++i)
for (auto* d : devices)
{
DeviceWrapper& d = *devices.getUnchecked(i);
if (! d.done)
if (! d->done)
{
if (d.isOutputReady (numSamples))
if (d->isOutputReady (numSamples))
{
d.pushOutputData (buffer, numSamples);
d.done = true;
d->pushOutputData (buffer, numSamples);
d->done = true;
}
else
anyRemaining = true;
@ -1572,28 +1563,39 @@ private:
auto newSampleRate = device->getCurrentSampleRate();
auto commonRates = getAvailableSampleRates();
if (! commonRates.contains (newSampleRate))
{
commonRates.sort();
if (newSampleRate < commonRates.getFirst() || newSampleRate > commonRates.getLast())
{
newSampleRate = jlimit (commonRates.getFirst(), commonRates.getLast(), newSampleRate);
}
else
{
for (auto it = commonRates.begin(); it < commonRates.end() - 1; ++it)
{
if (it[0] < newSampleRate && it[1] > newSampleRate)
{
newSampleRate = newSampleRate - it[0] < it[1] - newSampleRate ? it[0] : it[1];
break;
}
}
currentSampleRate = newSampleRate;
}
}
currentSampleRate = newSampleRate;
bool anySampleRateChanges = false;
for (int i = 0; i < devices.size(); ++i)
if (devices.getUnchecked(i)->getCurrentSampleRate() != currentSampleRate)
for (auto* d : devices)
{
devices.getUnchecked(i)->setCurrentSampleRate (currentSampleRate);
if (d->getCurrentSampleRate() != currentSampleRate)
{
d->setCurrentSampleRate (currentSampleRate);
anySampleRateChanges = true;
}
}
if (anySampleRateChanges)
owner.audioDeviceListChanged();
@ -1609,9 +1611,8 @@ private:
struct DeviceWrapper : private AudioIODeviceCallback
{
DeviceWrapper (AudioIODeviceCombiner& cd, CoreAudioIODevice* d, bool useIns, bool useOuts)
: owner (cd), device (d), inputIndex (0), outputIndex (0),
useInputs (useIns), useOutputs (useOuts),
inputFifo (32), outputFifo (32), done (false)
: owner (cd), device (d),
useInputs (useIns), useOutputs (useOuts)
{
}
@ -1621,18 +1622,16 @@ private:
}
String open (const BigInteger& inputChannels, const BigInteger& outputChannels,
double sampleRate, int bufferSize,
int channelIndex,
int fifoSize)
double sampleRate, int bufferSize, int channelIndex, int fifoSize)
{
inputFifo.setTotalSize (fifoSize);
outputFifo.setTotalSize (fifoSize);
inputFifo.reset();
outputFifo.reset();
String err (device->open (useInputs ? inputChannels : BigInteger(),
auto err = device->open (useInputs ? inputChannels : BigInteger(),
useOutputs ? outputChannels : BigInteger(),
sampleRate, bufferSize));
sampleRate, bufferSize);
numInputChans = useInputs ? device->getActiveInputChannels().countNumberOfSetBits() : 0;
numOutputChans = useOutputs ? device->getActiveOutputChannels().countNumberOfSetBits() : 0;
@ -1668,7 +1667,7 @@ private:
return numInputChans == 0 || inputFifo.getNumReady() >= numSamples;
}
void readInput (AudioSampleBuffer& destBuffer, int numSamples)
void readInput (AudioBuffer<float>& destBuffer, int numSamples)
{
if (numInputChans == 0)
return;
@ -1678,9 +1677,9 @@ private:
for (int i = 0; i < numInputChans; ++i)
{
const int index = inputIndex + i;
float* const dest = destBuffer.getWritePointer (index);
const float* const src = owner.fifos.getReadPointer (index);
auto index = inputIndex + i;
auto dest = destBuffer.getWritePointer (index);
auto src = owner.fifos.getReadPointer (index);
if (size1 > 0) FloatVectorOperations::copy (dest, src + start1, size1);
if (size2 > 0) FloatVectorOperations::copy (dest + size1, src + start2, size2);
@ -1694,7 +1693,7 @@ private:
return numOutputChans == 0 || outputFifo.getFreeSpace() >= numSamples;
}
void pushOutputData (AudioSampleBuffer& srcBuffer, int numSamples)
void pushOutputData (AudioBuffer<float>& srcBuffer, int numSamples)
{
if (numOutputChans == 0)
return;
@ -1704,9 +1703,9 @@ private:
for (int i = 0; i < numOutputChans; ++i)
{
const int index = outputIndex + i;
float* const dest = owner.fifos.getWritePointer (index);
const float* const src = srcBuffer.getReadPointer (index);
auto index = outputIndex + i;
auto dest = owner.fifos.getWritePointer (index);
auto src = srcBuffer.getReadPointer (index);
if (size1 > 0) FloatVectorOperations::copy (dest + start1, src, size1);
if (size2 > 0) FloatVectorOperations::copy (dest + start2, src + size1, size2);
@ -1719,7 +1718,7 @@ private:
float** outputChannelData, int numOutputChannels,
int numSamples) override
{
AudioSampleBuffer& buf = owner.fifos;
auto& buf = owner.fifos;
if (numInputChannels > 0)
{
@ -1734,8 +1733,8 @@ private:
for (int i = 0; i < numInputChannels; ++i)
{
float* const dest = buf.getWritePointer (inputIndex + i);
const float* const src = inputChannelData[i];
auto dest = buf.getWritePointer (inputIndex + i);
auto src = inputChannelData[i];
if (size1 > 0) FloatVectorOperations::copy (dest + start1, src, size1);
if (size2 > 0) FloatVectorOperations::copy (dest + start2, src + size1, size2);
@ -1765,8 +1764,8 @@ private:
for (int i = 0; i < numOutputChannels; ++i)
{
float* const dest = outputChannelData[i];
const float* const src = buf.getReadPointer (outputIndex + i);
auto dest = outputChannelData[i];
auto src = buf.getReadPointer (outputIndex + i);
if (size1 > 0) FloatVectorOperations::copy (dest, src + start1, size1);
if (size2 > 0) FloatVectorOperations::copy (dest + size1, src + start2, size2);
@ -1795,10 +1794,10 @@ private:
AudioIODeviceCombiner& owner;
ScopedPointer<CoreAudioIODevice> device;
int inputIndex, numInputChans, outputIndex, numOutputChans;
bool useInputs, useOutputs;
AbstractFifo inputFifo, outputFifo;
bool done;
int inputIndex = 0, numInputChans = 0, outputIndex = 0, numOutputChans = 0;
bool useInputs = false, useOutputs = false;
AbstractFifo inputFifo { 32 }, outputFifo { 32 };
bool done = false;
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (DeviceWrapper)
};
@ -1814,9 +1813,7 @@ class CoreAudioIODeviceType : public AudioIODeviceType,
private AsyncUpdater
{
public:
CoreAudioIODeviceType()
: AudioIODeviceType ("CoreAudio"),
hasScanned (false)
CoreAudioIODeviceType() : AudioIODeviceType ("CoreAudio")
{
AudioObjectPropertyAddress pa;
pa.mSelector = kAudioHardwarePropertyDevices;
@ -1860,7 +1857,8 @@ public:
if (AudioObjectGetPropertyData (kAudioObjectSystemObject, &pa, 0, nullptr, &size, devs) == noErr)
{
const int num = size / (int) sizeof (AudioDeviceID);
auto num = (int) size / (int) sizeof (AudioDeviceID);
for (int i = 0; i < num; ++i)
{
char name[1024];
@ -1869,9 +1867,9 @@ public:
if (AudioObjectGetPropertyData (devs[i], &pa, 0, nullptr, &size, name) == noErr)
{
const String nameString (String::fromUTF8 (name, (int) strlen (name)));
const int numIns = getNumChannels (devs[i], true);
const int numOuts = getNumChannels (devs[i], false);
auto nameString = String::fromUTF8 (name, (int) strlen (name));
auto numIns = getNumChannels (devs[i], true);
auto numOuts = getNumChannels (devs[i], false);
if (numIns > 0)
{
@ -1940,17 +1938,15 @@ public:
{
jassert (hasScanned); // need to call scanForDevices() before doing this
if (CoreAudioIODevice* const d = dynamic_cast<CoreAudioIODevice*> (device))
if (auto* d = dynamic_cast<CoreAudioIODevice*> (device))
return asInput ? d->inputIndex
: d->outputIndex;
if (AudioIODeviceCombiner* const d = dynamic_cast<AudioIODeviceCombiner*> (device))
if (auto* d = dynamic_cast<AudioIODeviceCombiner*> (device))
{
const Array<AudioIODevice*> devs (d->getDevices());
for (int i = 0; i < devs.size(); ++i)
for (auto* dev : d->getDevices())
{
const int index = getIndexOfDevice (devs.getUnchecked(i), asInput);
auto index = getIndexOfDevice (dev, asInput);
if (index >= 0)
return index;
@ -1967,16 +1963,17 @@ public:
{
jassert (hasScanned); // need to call scanForDevices() before doing this
const int inputIndex = inputDeviceNames.indexOf (inputDeviceName);
const int outputIndex = outputDeviceNames.indexOf (outputDeviceName);
auto inputIndex = inputDeviceNames.indexOf (inputDeviceName);
auto outputIndex = outputDeviceNames.indexOf (outputDeviceName);
AudioDeviceID inputDeviceID = inputIds [inputIndex];
AudioDeviceID outputDeviceID = outputIds [outputIndex];
auto inputDeviceID = inputIds[inputIndex];
auto outputDeviceID = outputIds[outputIndex];
if (inputDeviceID == 0 && outputDeviceID == 0)
return nullptr;
String combinedName (outputDeviceName.isEmpty() ? inputDeviceName : outputDeviceName);
auto combinedName = outputDeviceName.isEmpty() ? inputDeviceName
: outputDeviceName;
if (inputDeviceID == outputDeviceID)
return new CoreAudioIODevice (*this, combinedName, inputDeviceID, inputIndex, outputDeviceID, outputIndex);
@ -2014,7 +2011,7 @@ private:
StringArray inputDeviceNames, outputDeviceNames;
Array<AudioDeviceID> inputIds, outputIds;
bool hasScanned;
bool hasScanned = false;
static int getNumChannels (AudioDeviceID deviceID, bool input)
{
@ -2033,13 +2030,10 @@ private:
if (AudioObjectGetPropertyData (deviceID, &pa, 0, nullptr, &size, bufList) == noErr)
{
const int numStreams = (int) bufList->mNumberBuffers;
auto numStreams = (int) bufList->mNumberBuffers;
for (int i = 0; i < numStreams; ++i)
{
const ::AudioBuffer& b = bufList->mBuffers[i];
total += b.mNumberChannels;
}
total += bufList->mBuffers[i].mNumberChannels;
}
}

2
modules/juce_audio_devices/native/juce_win32_DirectSound.cpp
View file

@ -892,7 +892,7 @@ private:
int bufferSizeSamples = 0;
double sampleRate = 0;
BigInteger enabledInputs, enabledOutputs;
AudioSampleBuffer inputBuffers, outputBuffers;
AudioBuffer<float> inputBuffers, outputBuffers;
AudioIODeviceCallback* callback = nullptr;
CriticalSection startStopLock;

4
modules/juce_audio_devices/native/juce_win32_WASAPI.cpp
View file

@ -1246,8 +1246,8 @@ public:
const int numOutputBuffers = getActiveOutputChannels().countNumberOfSetBits();
bool sampleRateHasChanged = false;
AudioSampleBuffer ins (jmax (1, numInputBuffers), bufferSize + 32);
AudioSampleBuffer outs (jmax (1, numOutputBuffers), bufferSize + 32);
AudioBuffer<float> ins (jmax (1, numInputBuffers), bufferSize + 32);
AudioBuffer<float> outs (jmax (1, numOutputBuffers), bufferSize + 32);
float** const inputBuffers = ins.getArrayOfWritePointers();
float** const outputBuffers = outs.getArrayOfWritePointers();
ins.clear();

9
modules/juce_audio_devices/sources/juce_AudioSourcePlayer.cpp
View file

@ -24,11 +24,6 @@ namespace juce
{
AudioSourcePlayer::AudioSourcePlayer()
: source (nullptr),
sampleRate (0),
bufferSize (0),
lastGain (1.0f),
gain (1.0f)
{
}
@ -41,7 +36,7 @@ void AudioSourcePlayer::setSource (AudioSource* newSource)
{
if (source != newSource)
{
AudioSource* const oldSource = source;
auto* oldSource = source;
if (newSource != nullptr && bufferSize > 0 && sampleRate > 0)
newSource->prepareToPlay (bufferSize, sampleRate);
@ -137,7 +132,7 @@ void AudioSourcePlayer::audioDeviceIOCallback (const float** inputChannelData,
}
}
AudioSampleBuffer buffer (channels, numActiveChans, numSamples);
AudioBuffer<float> buffer (channels, numActiveChans, numSamples);
AudioSourceChannelInfo info (&buffer, 0, numSamples);
source->getNextAudioBlock (info);

10
modules/juce_audio_devices/sources/juce_AudioSourcePlayer.h
View file

@ -96,14 +96,14 @@ public:
private:
//==============================================================================
CriticalSection readLock;
AudioSource* source;
double sampleRate;
int bufferSize;
AudioSource* source = nullptr;
double sampleRate = 0;
int bufferSize = 0;
float* channels[128];
float* outputChans[128];
const float* inputChans[128];
AudioSampleBuffer tempBuffer;
float lastGain, gain;
AudioBuffer<float> tempBuffer;
float lastGain = 1.0f, gain = 1.0f;
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (AudioSourcePlayer)
};

2
modules/juce_audio_formats/codecs/juce_FlacAudioFormat.cpp
View file

@ -366,7 +366,7 @@ public:
private:
FlacNamespace::FLAC__StreamDecoder* decoder;
AudioSampleBuffer reservoir;
AudioBuffer<float> reservoir;
int reservoirStart = 0, samplesInReservoir = 0;
bool ok = false, scanningForLength = false;

2
modules/juce_audio_formats/codecs/juce_OggVorbisAudioFormat.cpp
View file

@ -263,7 +263,7 @@ public:
private:
OggVorbisNamespace::OggVorbis_File ovFile;
OggVorbisNamespace::ov_callbacks callbacks;
AudioSampleBuffer reservoir;
AudioBuffer<float> reservoir;
int reservoirStart = 0, samplesInReservoir = 0;
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (OggReader)

2
modules/juce_audio_formats/codecs/juce_WavAudioFormat.cpp
View file

@ -1838,7 +1838,7 @@ struct WaveAudioFormatTests : public UnitTest
32, metadataValues, 0));
expect (writer != nullptr);
AudioSampleBuffer buffer (numTestAudioBufferChannels, numTestAudioBufferSamples);
AudioBuffer<float> buffer (numTestAudioBufferChannels, numTestAudioBufferSamples);
buffer.clear();
beginTest ("Writing audio data to the basic wave writer");

27
modules/juce_audio_formats/format/juce_AudioFormatReader.cpp
View file

@ -100,10 +100,8 @@ bool AudioFormatReader::read (int* const* destSamples,
return true;
}
static void readChannels (AudioFormatReader& reader,
int** const chans, AudioSampleBuffer* const buffer,
const int startSample, const int numSamples,
const int64 readerStartSample, const int numTargetChannels)
static void readChannels (AudioFormatReader& reader, int** chans, AudioBuffer<float>* buffer,
int startSample, int numSamples, int64 readerStartSample, int numTargetChannels)
{
for (int j = 0; j < numTargetChannels; ++j)
chans[j] = reinterpret_cast<int*> (buffer->getWritePointer (j, startSample));
@ -112,7 +110,7 @@ static void readChannels (AudioFormatReader& reader,
reader.read (chans, numTargetChannels, readerStartSample, numSamples, true);
}
void AudioFormatReader::read (AudioSampleBuffer* buffer,
void AudioFormatReader::read (AudioBuffer<float>* buffer,
int startSample,
int numSamples,
int64 readerStartSample,
@ -186,16 +184,17 @@ void AudioFormatReader::readMaxLevels (int64 startSampleInFile, int64 numSamples
return;
}
const int bufferSize = (int) jmin (numSamples, (int64) 4096);
AudioSampleBuffer tempSampleBuffer ((int) channelsToRead, bufferSize);
auto bufferSize = (int) jmin (numSamples, (int64) 4096);
AudioBuffer<float> tempSampleBuffer ((int) channelsToRead, bufferSize);
float* const* const floatBuffer = tempSampleBuffer.getArrayOfWritePointers();
int* const* intBuffer = reinterpret_cast<int* const*> (floatBuffer);
auto floatBuffer = tempSampleBuffer.getArrayOfWritePointers();
auto intBuffer = reinterpret_cast<int* const*> (floatBuffer);
bool isFirstBlock = true;
while (numSamples > 0)
{
const int numToDo = (int) jmin (numSamples, (int64) bufferSize);
auto numToDo = (int) jmin (numSamples, (int64) bufferSize);
if (! read (intBuffer, channelsToRead, startSampleInFile, numToDo, false))
break;
@ -209,7 +208,7 @@ void AudioFormatReader::readMaxLevels (int64 startSampleInFile, int64 numSamples
}
else
{
Range<int> intRange (Range<int>::findMinAndMax (intBuffer[i], numToDo));
auto intRange = Range<int>::findMinAndMax (intBuffer[i], numToDo);
r = Range<float> (intRange.getStart() / (float) std::numeric_limits<int>::max(),
intRange.getEnd() / (float) std::numeric_limits<int>::max());
@ -248,9 +247,9 @@ void AudioFormatReader::readMaxLevels (int64 startSampleInFile, int64 numSamples
int64 AudioFormatReader::searchForLevel (int64 startSample,
int64 numSamplesToSearch,
const double magnitudeRangeMinimum,
const double magnitudeRangeMaximum,
const int minimumConsecutiveSamples)
double magnitudeRangeMinimum,
double magnitudeRangeMaximum,
int minimumConsecutiveSamples)
{
if (numSamplesToSearch == 0)
return -1;

4
modules/juce_audio_formats/format/juce_AudioFormatReader.h
View file

@ -111,14 +111,14 @@ public:
int numSamplesToRead,
bool fillLeftoverChannelsWithCopies);
/** Fills a section of an AudioSampleBuffer from this reader.
/** Fills a section of an AudioBuffer from this reader.
This will convert the reader's fixed- or floating-point data to
the buffer's floating-point format, and will try to intelligently
cope with mismatches between the number of channels in the reader
and the buffer.
*/
void read (AudioSampleBuffer* buffer,
void read (AudioBuffer<float>* buffer,
int startSampleInDestBuffer,
int numSamples,
int64 readerStartSample,

12
modules/juce_audio_formats/format/juce_AudioFormatWriter.cpp
View file

@ -84,7 +84,7 @@ bool AudioFormatWriter::writeFromAudioReader (AudioFormatReader& reader,
int64 numSamplesToRead)
{
const int bufferSize = 16384;
AudioSampleBuffer tempBuffer ((int) numChannels, bufferSize);
AudioBuffer<float> tempBuffer ((int) numChannels, bufferSize);
int* buffers[128] = { 0 };
@ -128,11 +128,11 @@ bool AudioFormatWriter::writeFromAudioReader (AudioFormatReader& reader,
bool AudioFormatWriter::writeFromAudioSource (AudioSource& source, int numSamplesToRead, const int samplesPerBlock)
{
AudioSampleBuffer tempBuffer (getNumChannels(), samplesPerBlock);
AudioBuffer<float> tempBuffer (getNumChannels(), samplesPerBlock);
while (numSamplesToRead > 0)
{
const int numToDo = jmin (numSamplesToRead, samplesPerBlock);
auto numToDo = jmin (numSamplesToRead, samplesPerBlock);
AudioSourceChannelInfo info (&tempBuffer, 0, numToDo);
info.clearActiveBufferRegion();
@ -185,9 +185,9 @@ bool AudioFormatWriter::writeFromFloatArrays (const float* const* channels, int
return true;
}
bool AudioFormatWriter::writeFromAudioSampleBuffer (const AudioSampleBuffer& source, int startSample, int numSamples)
bool AudioFormatWriter::writeFromAudioSampleBuffer (const AudioBuffer<float>& source, int startSample, int numSamples)
{
const int numSourceChannels = source.getNumChannels();
auto numSourceChannels = source.getNumChannels();
jassert (startSample >= 0 && startSample + numSamples <= source.getNumSamples() && numSourceChannels > 0);
if (startSample == 0)
@ -326,7 +326,7 @@ public:
private:
AbstractFifo fifo;
AudioSampleBuffer buffer;
AudioBuffer<float> buffer;
TimeSliceThread& timeSliceThread;
ScopedPointer<AudioFormatWriter> writer;
CriticalSection thumbnailLock;

8
modules/juce_audio_formats/format/juce_AudioFormatWriter.h
View file

@ -96,7 +96,7 @@ public:
//==============================================================================
/** Writes a set of samples to the audio stream.
Note that if you're trying to write the contents of an AudioSampleBuffer, you
Note that if you're trying to write the contents of an AudioBuffer, you
can use writeFromAudioSampleBuffer().
@param samplesToWrite an array of arrays containing the sample data for
@ -154,8 +154,8 @@ public:
int samplesPerBlock = 2048);
/** Writes some samples from an AudioSampleBuffer. */
bool writeFromAudioSampleBuffer (const AudioSampleBuffer& source,
/** Writes some samples from an AudioBuffer. */
bool writeFromAudioSampleBuffer (const AudioBuffer<float>& source,
int startSample, int numSamples);
/** Writes some samples from a set of float data channels. */
@ -217,7 +217,7 @@ public:
virtual ~IncomingDataReceiver() {}
virtual void reset (int numChannels, double sampleRate, int64 totalSamplesInSource) = 0;
virtual void addBlock (int64 sampleNumberInSource, const AudioSampleBuffer& newData,
virtual void addBlock (int64 sampleNumberInSource, const AudioBuffer<float>& newData,
int startOffsetInBuffer, int numSamples) = 0;
};

2
modules/juce_audio_formats/format/juce_BufferingAudioFormatReader.h
View file

@ -78,7 +78,7 @@ private:
BufferedBlock (AudioFormatReader& reader, int64 pos, int numSamples);
Range<int64> range;
AudioSampleBuffer buffer;
AudioBuffer<float> buffer;
};
CriticalSection lock;

4
modules/juce_audio_formats/sampler/juce_Sampler.cpp
View file

@ -44,7 +44,7 @@ SamplerSound::SamplerSound (const String& soundName,
length = jmin ((int) source.lengthInSamples,
(int) (maxSampleLengthSeconds * sourceSampleRate));
data = new AudioSampleBuffer (jmin (2, (int) source.numChannels), length + 4);
data = new AudioBuffer<float> (jmin (2, (int) source.numChannels), length + 4);
source.read (data, 0, length + 4, 0, true, true);
@ -129,7 +129,7 @@ void SamplerVoice::pitchWheelMoved (int /*newValue*/) {}
void SamplerVoice::controllerMoved (int /*controllerNumber*/, int /*newValue*/) {}
//==============================================================================
void SamplerVoice::renderNextBlock (AudioSampleBuffer& outputBuffer, int startSample, int numSamples)
void SamplerVoice::renderNextBlock (AudioBuffer<float>& outputBuffer, int startSample, int numSamples)
{
if (auto* playingSound = static_cast<SamplerSound*> (getCurrentlyPlayingSound().get()))
{

6
modules/juce_audio_formats/sampler/juce_Sampler.h
View file

@ -79,7 +79,7 @@ public:
/** Returns the audio sample data.
This could return nullptr if there was a problem loading the data.
*/
AudioSampleBuffer* getAudioData() const noexcept { return data; }
AudioBuffer<float>* getAudioData() const noexcept { return data; }
//==============================================================================
@ -92,7 +92,7 @@ private:
friend class SamplerVoice;
String name;
ScopedPointer<AudioSampleBuffer> data;
ScopedPointer<AudioBuffer<float>> data;
double sourceSampleRate;
BigInteger midiNotes;
int length = 0, attackSamples = 0, releaseSamples = 0;
@ -130,7 +130,7 @@ public:
void pitchWheelMoved (int newValue) override;
void controllerMoved (int controllerNumber, int newValue) override;
void renderNextBlock (AudioSampleBuffer&, int startSample, int numSamples) override;
void renderNextBlock (AudioBuffer<float>&, int startSample, int numSamples) override;
private:

2
modules/juce_audio_plugin_client/AAX/juce_AAX_Wrapper.cpp
View file

@ -1237,7 +1237,7 @@ namespace AAXClasses
void process (float* const* channels, const int numChans, const int bufferSize,
const bool bypass, AAX_IMIDINode* midiNodeIn, AAX_IMIDINode* midiNodesOut)
{
AudioSampleBuffer buffer (channels, numChans, bufferSize);
AudioBuffer<float> buffer (channels, numChans, bufferSize);
midiBuffer.clear();

2
modules/juce_audio_plugin_client/AU/juce_AU_Wrapper.mm
View file

@ -1690,7 +1690,7 @@ private:
}
}
void processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiBuffer) noexcept
void processBlock (AudioBuffer<float>& buffer, MidiBuffer& midiBuffer) noexcept
{
const ScopedLock sl (juceFilter->getCallbackLock());

4
modules/juce_audio_plugin_client/AU/juce_AUv3_Wrapper.mm
View file

@ -1056,7 +1056,7 @@ private:
AudioBufferList* bufferList = nullptr;
int maxFrames, numberOfChannels;
bool isInterleaved;
AudioSampleBuffer scratchBuffer;
AudioBuffer<float> scratchBuffer;
};
//==============================================================================
@ -1357,7 +1357,7 @@ private:
return noErr;
}
void processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiBuffer) noexcept
void processBlock (AudioBuffer<float>& buffer, MidiBuffer& midiBuffer) noexcept
{
auto& processor = getAudioProcessor();
const ScopedLock sl (processor.getCallbackLock());

2
modules/juce_audio_plugin_client/RTAS/juce_RTAS_Wrapper.cpp
View file

@ -599,7 +599,7 @@ public:
channels [i] = inputs [i];
}
AudioSampleBuffer chans (channels, totalChans, numSamples);
AudioBuffer<float> chans (channels, totalChans, numSamples);
if (mBypassed)
juceFilter->processBlockBypassed (chans, midiEvents);

2
modules/juce_audio_plugin_client/Standalone/juce_StandaloneFilterWindow.h
View file

@ -384,7 +384,7 @@ public:
// avoid feedback loop by default
bool processorHasPotentialFeedbackLoop = true;
Value shouldMuteInput;
AudioSampleBuffer emptyBuffer;
AudioBuffer<float> emptyBuffer;
bool autoOpenMidiDevices;
ScopedPointer<AudioDeviceManager::AudioDeviceSetup> options;

6
modules/juce_audio_processors/format_types/juce_AU_Shared.h
View file

@ -213,7 +213,7 @@ struct AudioUnitHelpers
}
//==============================================================================
AudioSampleBuffer& getBuffer (UInt32 frames) noexcept
AudioBuffer<float>& getBuffer (UInt32 frames) noexcept
{
jassert (pushIdx == scratch.getNumChannels());
@ -240,9 +240,7 @@ struct AudioUnitHelpers
}
//==============================================================================
AudioSampleBuffer scratch;
AudioSampleBuffer mutableBuffer;
AudioBuffer<float> scratch, mutableBuffer;
HeapBlock<float*> channels;
int pushIdx, popIdx;
};

12
modules/juce_audio_processors/format_types/juce_AudioUnitPluginFormat.mm
View file

@ -787,9 +787,9 @@ public:
for (int i = 0; i < getBusCount (false); ++i) AudioUnitReset (audioUnit, kAudioUnitScope_Output, static_cast<UInt32> (i));
}
void processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages) override
void processBlock (AudioBuffer<float>& buffer, MidiBuffer& midiMessages) override
{
const int numSamples = buffer.getNumSamples();
auto numSamples = buffer.getNumSamples();
if (prepared)
{
@ -816,6 +816,7 @@ public:
{
int chIdx = 0;
numOutputBuses = getBusCount (false);
for (int i = 0; i < numOutputBuses; ++i)
{
if (AUBuffer* buf = outputBufferList[i])
@ -1298,7 +1299,7 @@ private:
OwnedArray<AUBuffer> outputBufferList;
AudioTimeStamp timeStamp;
AudioSampleBuffer* currentBuffer;
AudioBuffer<float>* currentBuffer;
Array<Array<AudioChannelSet>> supportedInLayouts, supportedOutLayouts;
int numChannelInfos;
@ -1464,10 +1465,9 @@ private:
{
// if this ever happens, might need to add extra handling
jassert (inNumberFrames == (UInt32) currentBuffer->getNumSamples());
AudioSampleBuffer buffer =
(static_cast<int> (inBusNumber) < getBusCount (true)
auto buffer = static_cast<int> (inBusNumber) < getBusCount (true)
? getBusBuffer (*currentBuffer, true, static_cast<int> (inBusNumber))
: AudioSampleBuffer());
: AudioBuffer<float>();
for (int i = 0; i < static_cast<int> (ioData->mNumberBuffers); ++i)
{

6
modules/juce_audio_processors/format_types/juce_LADSPAPluginFormat.cpp
View file

@ -286,9 +286,9 @@ public:
tempBuffer.setSize (1, 1);
}
void processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
void processBlock (AudioBuffer<float>& buffer, MidiBuffer& midiMessages)
{
const int numSamples = buffer.getNumSamples();
auto numSamples = buffer.getNumSamples();
if (initialised && plugin != nullptr && handle != nullptr)
{
@ -484,7 +484,7 @@ private:
String name;
CriticalSection lock;
bool initialised;
AudioSampleBuffer tempBuffer;
AudioBuffer<float> tempBuffer;
Array<int> inputs, outputs, parameters;
struct ParameterValue

6
modules/juce_audio_processors/format_types/juce_VST3Common.h
View file

@ -537,10 +537,8 @@ struct VST3BufferExchange
static inline void assignRawPointer (Steinberg::Vst::AudioBusBuffers& vstBuffers, float** raw) { vstBuffers.channelBuffers32 = raw; }
static inline void assignRawPointer (Steinberg::Vst::AudioBusBuffers& vstBuffers, double** raw) { vstBuffers.channelBuffers64 = raw; }
/** Assigns a series of AudioSampleBuffer's channels to an AudioBusBuffers'
@warning For speed, does not check the channel count and offsets
according to the AudioSampleBuffer
/** Assigns a series of AudioBuffer's channels to an AudioBusBuffers'
@warning For speed, does not check the channel count and offsets according to the AudioBuffer
*/
static void associateBufferTo (Steinberg::Vst::AudioBusBuffers& vstBuffers,
Bus& bus,

4
modules/juce_audio_processors/format_types/juce_VST3PluginFormat.cpp
View file

@ -169,8 +169,8 @@ static void setStateForAllBusesOfType (Vst::IComponent* component,
}
//==============================================================================
/** Assigns a complete AudioSampleBuffer's channels to an AudioBusBuffers' */
static void associateWholeBufferTo (Vst::AudioBusBuffers& vstBuffers, AudioSampleBuffer& buffer) noexcept
/** Assigns a complete AudioBuffer's channels to an AudioBusBuffers' */
static void associateWholeBufferTo (Vst::AudioBusBuffers& vstBuffers, AudioBuffer<float>& buffer) noexcept
{
vstBuffers.channelBuffers32 = buffer.getArrayOfWritePointers();
vstBuffers.numChannels = buffer.getNumChannels();

8
modules/juce_audio_processors/processors/juce_AudioProcessor.h
View file

@ -441,7 +441,7 @@ public:
//==============================================================================
/** Returns the position of a bus's channels within the processBlock buffer.
This can be called in processBlock to figure out which channel of the master AudioSampleBuffer
This can be called in processBlock to figure out which channel of the master AudioBuffer
maps onto a specific bus's channel.
*/
int getChannelIndexInProcessBlockBuffer (int channelIndex) const noexcept;
@ -449,7 +449,7 @@ public:
/** Returns an AudioBuffer containing a set of channel pointers for a specific bus.
This can be called in processBlock to get a buffer containing a sub-group of the master
AudioSampleBuffer which contains all the plugin channels.
AudioBuffer which contains all the plugin channels.
*/
template <typename FloatType>
AudioBuffer<FloatType> getBusBuffer (AudioBuffer<FloatType>& processBlockBuffer) const
@ -605,7 +605,7 @@ public:
//==============================================================================
/** Returns the position of a bus's channels within the processBlock buffer.
This can be called in processBlock to figure out which channel of the master AudioSampleBuffer
This can be called in processBlock to figure out which channel of the master AudioBuffer
maps onto a specific bus's channel.
*/
int getChannelIndexInProcessBlockBuffer (bool isInput, int busIndex, int channelIndex) const noexcept;
@ -620,7 +620,7 @@ public:
/** Returns an AudioBuffer containing a set of channel pointers for a specific bus.
This can be called in processBlock to get a buffer containing a sub-group of the master
AudioSampleBuffer which contains all the plugin channels.
AudioBuffer which contains all the plugin channels.
*/
template <typename FloatType>
AudioBuffer<FloatType> getBusBuffer (AudioBuffer<FloatType>& processBlockBuffer, bool isInput, int busIndex) const

2
modules/juce_audio_utils/gui/juce_AudioThumbnail.cpp
View file

@ -693,7 +693,7 @@ int64 AudioThumbnail::getHashCode() const
return source == nullptr ? 0 : source->hashCode;
}
void AudioThumbnail::addBlock (const int64 startSample, const AudioSampleBuffer& incoming,
void AudioThumbnail::addBlock (int64 startSample, const AudioBuffer<float>& incoming,
int startOffsetInBuffer, int numSamples)
{
jassert (startSample >= 0

2
modules/juce_audio_utils/gui/juce_AudioThumbnail.h
View file

@ -107,7 +107,7 @@ public:
/** Adds a block of level data to the thumbnail.
Call reset() before using this, to tell the thumbnail about the data format.
*/
void addBlock (int64 sampleNumberInSource, const AudioSampleBuffer& newData,
void addBlock (int64 sampleNumberInSource, const AudioBuffer<float>& newData,
int startOffsetInBuffer, int numSamples) override;
//==============================================================================

2
modules/juce_audio_utils/gui/juce_AudioVisualiserComponent.cpp
View file

@ -129,7 +129,7 @@ void AudioVisualiserComponent::pushBuffer (const float** d, int numChannels, int
channels.getUnchecked(i)->pushSamples (d[i], num);
}
void AudioVisualiserComponent::pushBuffer (const AudioSampleBuffer& buffer)
void AudioVisualiserComponent::pushBuffer (const AudioBuffer<float>& buffer)
{
pushBuffer (buffer.getArrayOfReadPointers(),
buffer.getNumChannels(),

2
modules/juce_audio_utils/gui/juce_AudioVisualiserComponent.h
View file

@ -74,7 +74,7 @@ public:
The number of channels provided here is expected to match the number of channels
that this AudioVisualiserComponent has been told to use.
*/
void pushBuffer (const AudioSampleBuffer& bufferToPush);
void pushBuffer (const AudioBuffer<float>& bufferToPush);
/** Pushes a buffer of channels data.
The number of channels provided here is expected to match the number of channels

2
modules/juce_audio_utils/native/juce_mac_AudioCDBurner.mm
View file

@ -129,7 +129,7 @@ private:
if (numSamples > 0)
{
AudioSampleBuffer tempBuffer (2, numSamples);
AudioBuffer<float> tempBuffer (2, numSamples);
AudioSourceChannelInfo info (tempBuffer);
source->source->getNextAudioBlock (info);

2
modules/juce_audio_utils/native/juce_win32_AudioCDBurner.cpp
View file

@ -373,7 +373,7 @@ bool AudioCDBurner::addAudioTrack (AudioSource* audioSource, int numSamples)
hr = pimpl->redbook->CreateAudioTrack ((long) numSamples / (bytesPerBlock * 4));
HeapBlock<byte> buffer (bytesPerBlock);
AudioSampleBuffer sourceBuffer (2, samplesPerBlock);
AudioBuffer<float> sourceBuffer (2, samplesPerBlock);
int samplesDone = 0;
source->prepareToPlay (samplesPerBlock, 44100.0);

2
modules/juce_audio_utils/players/juce_AudioProcessorPlayer.cpp
View file

@ -141,7 +141,7 @@ void AudioProcessorPlayer::audioDeviceIOCallback (const float** const inputChann
}
}
AudioSampleBuffer buffer (channels, totalNumChans, numSamples);
AudioBuffer<float> buffer (channels, totalNumChans, numSamples);
{
const ScopedLock sl (lock);

19
modules/juce_audio_utils/players/juce_SoundPlayer.cpp
View file

@ -85,12 +85,11 @@ private:
};
// An AudioSource which simply outputs a buffer
class AudioSampleBufferSource : public PositionableAudioSource
class AudioBufferSource : public PositionableAudioSource
{
public:
AudioSampleBufferSource (AudioSampleBuffer* audioBuffer, bool ownBuffer, bool playOnAllChannels)
AudioBufferSource (AudioBuffer<float>* audioBuffer, bool ownBuffer, bool playOnAllChannels)
: buffer (audioBuffer, ownBuffer),
position (0), looping (false),
playAcrossAllChannels (playOnAllChannels)
{}
@ -144,11 +143,11 @@ public:
private:
//==============================================================================
OptionalScopedPointer<AudioSampleBuffer> buffer;
int position;
bool looping, playAcrossAllChannels;
OptionalScopedPointer<AudioBuffer<float>> buffer;
int position = 0;
bool looping = false, playAcrossAllChannels;
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (AudioSampleBufferSource)
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (AudioBufferSource)
};
SoundPlayer::SoundPlayer()
@ -185,10 +184,10 @@ void SoundPlayer::play (AudioFormatReader* reader, bool deleteWhenFinished)
play (new AudioFormatReaderSource (reader, deleteWhenFinished), true, reader->sampleRate);
}
void SoundPlayer::play (AudioSampleBuffer* buffer, bool deleteWhenFinished, bool playOnAllOutputChannels)
void SoundPlayer::play (AudioBuffer<float>* buffer, bool deleteWhenFinished, bool playOnAllOutputChannels)
{
if (buffer != nullptr)
play (new AudioSampleBufferSource (buffer, deleteWhenFinished, playOnAllOutputChannels), true);
play (new AudioBufferSource (buffer, deleteWhenFinished, playOnAllOutputChannels), true);
}
void SoundPlayer::play (PositionableAudioSource* audioSource, bool deleteWhenFinished, double fileSampleRate)
@ -232,7 +231,7 @@ void SoundPlayer::playTestSound()
const double phasePerSample = double_Pi * 2.0 / (sampleRate / frequency);
AudioSampleBuffer* newSound = new AudioSampleBuffer (1, soundLength);
auto* newSound = new AudioBuffer<float> (1, soundLength);
for (int i = 0; i < soundLength; ++i)
newSound->setSample (0, i, amplitude * (float) std::sin (i * phasePerSample));

2
modules/juce_audio_utils/players/juce_SoundPlayer.h
View file

@ -96,7 +96,7 @@ public:
multiple outputs so that something is sent to all output channels. If it
is false, then the buffer will just be played on the first output channels.
*/
void play (AudioSampleBuffer* buffer,
void play (AudioBuffer<float>* buffer,
bool deleteWhenFinished = false,
bool playOnAllOutputChannels = false);

2
modules/juce_dsp/containers/juce_AudioBlock.h
View file

@ -47,7 +47,7 @@ namespace SampleTypeHelpers // Internal classes needed for handling sample type
This class doesn't own any of the data which it points to, it's simply a view
into data that is owned elsewhere. You can construct one from some raw data
that you've allocated yourself, or give it a HeapBlock to use, or give it
an AudioSampleBuffer which it can refer to, but in all cases the user is
an AudioBuffer which it can refer to, but in all cases the user is
responsible for making sure that the data doesn't get deleted while there's
still an AudioBlock using it.
*/