paulxstretch/Source/ps3_BufferingAudioSource.cpp
2017-11-13 17:06:08 +02:00

323 lines
10 KiB
C++

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