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xenakios
322
Source/ps3_BufferingAudioSource.cpp
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322
Source/ps3_BufferingAudioSource.cpp
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/*
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==============================================================================
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This file is part of the JUCE library.
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Copyright (c) 2017 - ROLI Ltd.
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JUCE is an open source library subject to commercial or open-source
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licensing.
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The code included in this file is provided under the terms of the ISC license
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http://www.isc.org/downloads/software-support-policy/isc-license. Permission
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To use, copy, modify, and/or distribute this software for any purpose with or
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without fee is hereby granted provided that the above copyright notice and
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this permission notice appear in all copies.
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JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
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EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
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DISCLAIMED.
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==============================================================================
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*/
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#include "ps3_BufferingAudioSource.h"
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MyBufferingAudioSource::MyBufferingAudioSource(PositionableAudioSource* s,
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TimeSliceThread& thread,
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const bool deleteSourceWhenDeleted,
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const int bufferSizeSamples,
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const int numChannels,
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bool prefillBufferOnPrepareToPlay)
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: source (s, deleteSourceWhenDeleted),
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backgroundThread (thread),
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numberOfSamplesToBuffer (jmax (1024, bufferSizeSamples)),
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numberOfChannels (numChannels),
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bufferValidStart (0),
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bufferValidEnd (0),
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nextPlayPos (0),
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sampleRate (0),
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wasSourceLooping (false),
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isPrepared (false),
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prefillBuffer (prefillBufferOnPrepareToPlay)
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{
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jassert (source != nullptr);
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jassert (numberOfSamplesToBuffer > 1024); // not much point using this class if you're
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// not using a larger buffer..
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}
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MyBufferingAudioSource::~MyBufferingAudioSource()
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{
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releaseResources();
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}
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//==============================================================================
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void MyBufferingAudioSource::prepareToPlay (int samplesPerBlockExpected, double newSampleRate)
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{
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const int bufferSizeNeeded = jmax (samplesPerBlockExpected * 2, numberOfSamplesToBuffer);
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if (newSampleRate != sampleRate
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|| bufferSizeNeeded != buffer.getNumSamples()
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|| ! isPrepared)
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{
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backgroundThread.removeTimeSliceClient (this);
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isPrepared = true;
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sampleRate = newSampleRate;
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source->prepareToPlay (samplesPerBlockExpected, newSampleRate);
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buffer.setSize (numberOfChannels, bufferSizeNeeded);
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buffer.clear();
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bufferValidStart = 0;
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bufferValidEnd = 0;
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backgroundThread.addTimeSliceClient (this);
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do
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{
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backgroundThread.moveToFrontOfQueue (this);
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Thread::sleep (5);
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}
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while (prefillBuffer
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&& (bufferValidEnd - bufferValidStart < jmin (((int) newSampleRate) / 4, buffer.getNumSamples() / 2)));
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}
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}
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void MyBufferingAudioSource::releaseResources()
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{
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isPrepared = false;
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backgroundThread.removeTimeSliceClient (this);
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buffer.setSize (numberOfChannels, 0);
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// MSVC2015 seems to need this if statement to not generate a warning during linking.
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// As source is set in the constructor, there is no way that source could
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// ever equal this, but it seems to make MSVC2015 happy.
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if (source != this)
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source->releaseResources();
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}
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void MyBufferingAudioSource::getNextAudioBlock (const AudioSourceChannelInfo& info)
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{
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const ScopedLock sl (bufferStartPosLock);
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const int validStart = (int) (jlimit (bufferValidStart, bufferValidEnd, nextPlayPos) - nextPlayPos);
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const int validEnd = (int) (jlimit (bufferValidStart, bufferValidEnd, nextPlayPos + info.numSamples) - nextPlayPos);
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if (validStart == validEnd)
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{
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// total cache miss
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info.clearActiveBufferRegion();
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}
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else
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{
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if (validStart > 0)
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info.buffer->clear (info.startSample, validStart); // partial cache miss at start
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if (validEnd < info.numSamples)
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info.buffer->clear (info.startSample + validEnd,
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info.numSamples - validEnd); // partial cache miss at end
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if (validStart < validEnd)
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{
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for (int chan = jmin (numberOfChannels, info.buffer->getNumChannels()); --chan >= 0;)
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{
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jassert (buffer.getNumSamples() > 0);
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const int startBufferIndex = (int) ((validStart + nextPlayPos) % buffer.getNumSamples());
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const int endBufferIndex = (int) ((validEnd + nextPlayPos) % buffer.getNumSamples());
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if (startBufferIndex < endBufferIndex)
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{
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info.buffer->copyFrom (chan, info.startSample + validStart,
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buffer,
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chan, startBufferIndex,
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validEnd - validStart);
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}
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else
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{
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const int initialSize = buffer.getNumSamples() - startBufferIndex;
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info.buffer->copyFrom (chan, info.startSample + validStart,
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buffer,
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chan, startBufferIndex,
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initialSize);
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info.buffer->copyFrom (chan, info.startSample + validStart + initialSize,
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buffer,
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chan, 0,
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(validEnd - validStart) - initialSize);
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}
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}
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}
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nextPlayPos += info.numSamples;
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}
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}
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bool MyBufferingAudioSource::waitForNextAudioBlockReady (const AudioSourceChannelInfo& info, const uint32 timeout)
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{
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if (!source || source->getTotalLength() <= 0)
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return false;
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if (nextPlayPos + info.numSamples < 0)
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return true;
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if (! isLooping() && nextPlayPos > getTotalLength())
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return true;
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uint32 now = Time::getMillisecondCounter();
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const uint32 startTime = now;
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uint32 elapsed = (now >= startTime ? now - startTime
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: (std::numeric_limits<uint32>::max() - startTime) + now);
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while (elapsed <= timeout)
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{
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{
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const ScopedLock sl (bufferStartPosLock);
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const int validStart = static_cast<int> (jlimit (bufferValidStart, bufferValidEnd, nextPlayPos) - nextPlayPos);
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const int validEnd = static_cast<int> (jlimit (bufferValidStart, bufferValidEnd, nextPlayPos + info.numSamples) - nextPlayPos);
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if (validStart <= 0 && validStart < validEnd && validEnd >= info.numSamples)
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return true;
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}
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if (elapsed < timeout && (! bufferReadyEvent.wait (static_cast<int> (timeout - elapsed))))
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return false;
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now = Time::getMillisecondCounter();
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elapsed = (now >= startTime ? now - startTime
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: (std::numeric_limits<uint32>::max() - startTime) + now);
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}
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return false;
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}
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double MyBufferingAudioSource::getPercentReady()
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{
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if (bufferValidEnd == bufferValidStart)
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return 0.0;
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if (numberOfSamplesToBuffer == 0)
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return 0.0;
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return 1.0 / numberOfSamplesToBuffer * (bufferValidEnd - bufferValidStart);
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}
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int64 MyBufferingAudioSource::getNextReadPosition() const
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{
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jassert (source->getTotalLength() > 0);
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return (source->isLooping() && nextPlayPos > 0)
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? nextPlayPos % source->getTotalLength()
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: nextPlayPos;
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}
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void MyBufferingAudioSource::setNextReadPosition (int64 newPosition)
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{
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const ScopedLock sl (bufferStartPosLock);
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nextPlayPos = newPosition;
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backgroundThread.moveToFrontOfQueue (this);
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}
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bool MyBufferingAudioSource::readNextBufferChunk()
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{
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int64 newBVS, newBVE, sectionToReadStart, sectionToReadEnd;
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{
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const ScopedLock sl (bufferStartPosLock);
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if (wasSourceLooping != isLooping())
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{
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wasSourceLooping = isLooping();
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bufferValidStart = 0;
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bufferValidEnd = 0;
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}
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newBVS = jmax ((int64) 0, nextPlayPos);
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newBVE = newBVS + buffer.getNumSamples() - 4;
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sectionToReadStart = 0;
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sectionToReadEnd = 0;
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const int maxChunkSize = 2048;
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if (newBVS < bufferValidStart || newBVS >= bufferValidEnd)
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{
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newBVE = jmin (newBVE, newBVS + maxChunkSize);
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sectionToReadStart = newBVS;
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sectionToReadEnd = newBVE;
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bufferValidStart = 0;
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bufferValidEnd = 0;
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}
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else if (std::abs ((int) (newBVS - bufferValidStart)) > 512
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|| std::abs ((int) (newBVE - bufferValidEnd)) > 512)
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{
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newBVE = jmin (newBVE, bufferValidEnd + maxChunkSize);
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sectionToReadStart = bufferValidEnd;
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sectionToReadEnd = newBVE;
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bufferValidStart = newBVS;
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bufferValidEnd = jmin (bufferValidEnd, newBVE);
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}
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}
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if (sectionToReadStart == sectionToReadEnd)
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return false;
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jassert (buffer.getNumSamples() > 0);
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const int bufferIndexStart = (int) (sectionToReadStart % buffer.getNumSamples());
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const int bufferIndexEnd = (int) (sectionToReadEnd % buffer.getNumSamples());
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if (bufferIndexStart < bufferIndexEnd)
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{
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readBufferSection (sectionToReadStart,
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(int) (sectionToReadEnd - sectionToReadStart),
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bufferIndexStart);
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}
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else
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{
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const int initialSize = buffer.getNumSamples() - bufferIndexStart;
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readBufferSection (sectionToReadStart,
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initialSize,
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bufferIndexStart);
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readBufferSection (sectionToReadStart + initialSize,
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(int) (sectionToReadEnd - sectionToReadStart) - initialSize,
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0);
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}
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{
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const ScopedLock sl2 (bufferStartPosLock);
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bufferValidStart = newBVS;
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bufferValidEnd = newBVE;
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}
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bufferReadyEvent.signal();
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return true;
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}
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void MyBufferingAudioSource::readBufferSection (const int64 start, const int length, const int bufferOffset)
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{
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if (source->getNextReadPosition() != start)
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source->setNextReadPosition (start);
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AudioSourceChannelInfo info (&buffer, bufferOffset, length);
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source->getNextAudioBlock (info);
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}
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int MyBufferingAudioSource::useTimeSlice()
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{
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return readNextBufferChunk() ? 1 : 100;
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}
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