/*
 * Copyright (C) 2010, Google Inc. All rights reserved.
 *
 * Redistribution and use in source and binary forms, with or without
 * modification, are permitted provided that the following conditions
 * are met:
 * 1.  Redistributions of source code must retain the above copyright
 *    notice, this list of conditions and the following disclaimer.
 * 2.  Redistributions in binary form must reproduce the above copyright
 *    notice, this list of conditions and the following disclaimer in the
 *    documentation and/or other materials provided with the distribution.
 *
 * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
 * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
 * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
 * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
 * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
 * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
 * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
 * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
 * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
 * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
 */

#include "config.h"

#if ENABLE(WEB_AUDIO)

#include "modules/webaudio/ScriptProcessorNode.h"

#include "bindings/core/v8/ExceptionState.h"
#include "core/dom/CrossThreadTask.h"
#include "core/dom/ExecutionContext.h"
#include "modules/webaudio/AudioBuffer.h"
#include "modules/webaudio/AudioContext.h"
#include "modules/webaudio/AudioNodeInput.h"
#include "modules/webaudio/AudioNodeOutput.h"
#include "modules/webaudio/AudioProcessingEvent.h"
#include "public/platform/Platform.h"
#include "wtf/Float32Array.h"

namespace blink {

static size_t chooseBufferSize()
{
    // Choose a buffer size based on the audio hardware buffer size. Arbitarily make it a power of
    // two that is 4 times greater than the hardware buffer size.
    // FIXME: What is the best way to choose this?
    size_t hardwareBufferSize = Platform::current()->audioHardwareBufferSize();
    size_t bufferSize = 1 << static_cast<unsigned>(log2(4 * hardwareBufferSize) + 0.5);

    if (bufferSize < 256)
        return 256;
    if (bufferSize > 16384)
        return 16384;

    return bufferSize;
}

ScriptProcessorNode* ScriptProcessorNode::create(AudioContext* context, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels)
{
    // Check for valid buffer size.
    switch (bufferSize) {
    case 0:
        bufferSize = chooseBufferSize();
        break;
    case 256:
    case 512:
    case 1024:
    case 2048:
    case 4096:
    case 8192:
    case 16384:
        break;
    default:
        return 0;
    }

    if (!numberOfInputChannels && !numberOfOutputChannels)
        return 0;

    if (numberOfInputChannels > AudioContext::maxNumberOfChannels())
        return 0;

    if (numberOfOutputChannels > AudioContext::maxNumberOfChannels())
        return 0;

    return new ScriptProcessorNode(context, sampleRate, bufferSize, numberOfInputChannels, numberOfOutputChannels);
}

ScriptProcessorNode::ScriptProcessorNode(AudioContext* context, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels)
    : AudioNode(context, sampleRate)
    , m_doubleBufferIndex(0)
    , m_doubleBufferIndexForEvent(0)
    , m_bufferSize(bufferSize)
    , m_bufferReadWriteIndex(0)
    , m_numberOfInputChannels(numberOfInputChannels)
    , m_numberOfOutputChannels(numberOfOutputChannels)
    , m_internalInputBus(AudioBus::create(numberOfInputChannels, AudioNode::ProcessingSizeInFrames, false))
{
    // Regardless of the allowed buffer sizes, we still need to process at the granularity of the AudioNode.
    if (m_bufferSize < AudioNode::ProcessingSizeInFrames)
        m_bufferSize = AudioNode::ProcessingSizeInFrames;

    ASSERT(numberOfInputChannels <= AudioContext::maxNumberOfChannels());

    addInput();
    addOutput(AudioNodeOutput::create(this, numberOfOutputChannels));

    setNodeType(NodeTypeJavaScript);
    m_channelCount = numberOfInputChannels;
    m_channelCountMode = Explicit;

    initialize();
}

ScriptProcessorNode::~ScriptProcessorNode()
{
    ASSERT(!isInitialized());
}

void ScriptProcessorNode::dispose()
{
    uninitialize();
    AudioNode::dispose();
}

void ScriptProcessorNode::initialize()
{
    if (isInitialized())
        return;

    float sampleRate = context()->sampleRate();

    // Create double buffers on both the input and output sides.
    // These AudioBuffers will be directly accessed in the main thread by JavaScript.
    for (unsigned i = 0; i < 2; ++i) {
        AudioBuffer* inputBuffer = m_numberOfInputChannels ? AudioBuffer::create(m_numberOfInputChannels, bufferSize(), sampleRate) : 0;
        AudioBuffer* outputBuffer = m_numberOfOutputChannels ? AudioBuffer::create(m_numberOfOutputChannels, bufferSize(), sampleRate) : 0;

        m_inputBuffers.append(inputBuffer);
        m_outputBuffers.append(outputBuffer);
    }

    AudioNode::initialize();
}

void ScriptProcessorNode::uninitialize()
{
    if (!isInitialized())
        return;

    m_inputBuffers.clear();
    m_outputBuffers.clear();

    AudioNode::uninitialize();
}

void ScriptProcessorNode::process(size_t framesToProcess)
{
    // Discussion about inputs and outputs:
    // As in other AudioNodes, ScriptProcessorNode uses an AudioBus for its input and output (see inputBus and outputBus below).
    // Additionally, there is a double-buffering for input and output which is exposed directly to JavaScript (see inputBuffer and outputBuffer below).
    // This node is the producer for inputBuffer and the consumer for outputBuffer.
    // The JavaScript code is the consumer of inputBuffer and the producer for outputBuffer.

    // Get input and output busses.
    AudioBus* inputBus = this->input(0)->bus();
    AudioBus* outputBus = this->output(0)->bus();

    // Get input and output buffers. We double-buffer both the input and output sides.
    unsigned doubleBufferIndex = this->doubleBufferIndex();
    bool isDoubleBufferIndexGood = doubleBufferIndex < 2 && doubleBufferIndex < m_inputBuffers.size() && doubleBufferIndex < m_outputBuffers.size();
    ASSERT(isDoubleBufferIndexGood);
    if (!isDoubleBufferIndexGood)
        return;

    AudioBuffer* inputBuffer = m_inputBuffers[doubleBufferIndex].get();
    AudioBuffer* outputBuffer = m_outputBuffers[doubleBufferIndex].get();

    // Check the consistency of input and output buffers.
    unsigned numberOfInputChannels = m_internalInputBus->numberOfChannels();
    bool buffersAreGood = outputBuffer && bufferSize() == outputBuffer->length() && m_bufferReadWriteIndex + framesToProcess <= bufferSize();

    // If the number of input channels is zero, it's ok to have inputBuffer = 0.
    if (m_internalInputBus->numberOfChannels())
        buffersAreGood = buffersAreGood && inputBuffer && bufferSize() == inputBuffer->length();

    ASSERT(buffersAreGood);
    if (!buffersAreGood)
        return;

    // We assume that bufferSize() is evenly divisible by framesToProcess - should always be true, but we should still check.
    bool isFramesToProcessGood = framesToProcess && bufferSize() >= framesToProcess && !(bufferSize() % framesToProcess);
    ASSERT(isFramesToProcessGood);
    if (!isFramesToProcessGood)
        return;

    unsigned numberOfOutputChannels = outputBus->numberOfChannels();

    bool channelsAreGood = (numberOfInputChannels == m_numberOfInputChannels) && (numberOfOutputChannels == m_numberOfOutputChannels);
    ASSERT(channelsAreGood);
    if (!channelsAreGood)
        return;

    for (unsigned i = 0; i < numberOfInputChannels; i++)
        m_internalInputBus->setChannelMemory(i, inputBuffer->getChannelData(i)->data() + m_bufferReadWriteIndex, framesToProcess);

    if (numberOfInputChannels)
        m_internalInputBus->copyFrom(*inputBus);

    // Copy from the output buffer to the output.
    for (unsigned i = 0; i < numberOfOutputChannels; ++i)
        memcpy(outputBus->channel(i)->mutableData(), outputBuffer->getChannelData(i)->data() + m_bufferReadWriteIndex, sizeof(float) * framesToProcess);

    // Update the buffering index.
    m_bufferReadWriteIndex = (m_bufferReadWriteIndex + framesToProcess) % bufferSize();

    // m_bufferReadWriteIndex will wrap back around to 0 when the current input and output buffers are full.
    // When this happens, fire an event and swap buffers.
    if (!m_bufferReadWriteIndex) {
        // Avoid building up requests on the main thread to fire process events when they're not being handled.
        // This could be a problem if the main thread is very busy doing other things and is being held up handling previous requests.
        // The audio thread can't block on this lock, so we call tryLock() instead.
        MutexTryLocker tryLocker(m_processEventLock);
        if (!tryLocker.locked()) {
            // We're late in handling the previous request. The main thread must be very busy.
            // The best we can do is clear out the buffer ourself here.
            outputBuffer->zero();
        } else if (context()->executionContext()) {
            // Fire the event on the main thread, not this one (which is the realtime audio thread).
            m_doubleBufferIndexForEvent = m_doubleBufferIndex;
            context()->executionContext()->postTask(createCrossThreadTask(&ScriptProcessorNode::fireProcessEvent, this));
        }

        swapBuffers();
    }
}

void ScriptProcessorNode::fireProcessEvent()
{
    ASSERT(isMainThread());

    bool isIndexGood = m_doubleBufferIndexForEvent < 2;
    ASSERT(isIndexGood);
    if (!isIndexGood)
        return;

    AudioBuffer* inputBuffer = m_inputBuffers[m_doubleBufferIndexForEvent].get();
    AudioBuffer* outputBuffer = m_outputBuffers[m_doubleBufferIndexForEvent].get();
    ASSERT(outputBuffer);
    if (!outputBuffer)
        return;

    // Avoid firing the event if the document has already gone away.
    if (context()->executionContext()) {
        // This synchronizes with process().
        MutexLocker processLocker(m_processEventLock);

        // Calculate a playbackTime with the buffersize which needs to be processed each time onaudioprocess is called.
        // The outputBuffer being passed to JS will be played after exhuasting previous outputBuffer by double-buffering.
        double playbackTime = (context()->currentSampleFrame() + m_bufferSize) / static_cast<double>(context()->sampleRate());

        // Call the JavaScript event handler which will do the audio processing.
        dispatchEvent(AudioProcessingEvent::create(inputBuffer, outputBuffer, playbackTime));
    }
}

double ScriptProcessorNode::tailTime() const
{
    return std::numeric_limits<double>::infinity();
}

double ScriptProcessorNode::latencyTime() const
{
    return std::numeric_limits<double>::infinity();
}

void ScriptProcessorNode::setChannelCount(unsigned long channelCount, ExceptionState& exceptionState)
{
    ASSERT(isMainThread());
    AudioContext::AutoLocker locker(context());

    if (channelCount != m_channelCount) {
        exceptionState.throwDOMException(
            NotSupportedError,
            "channelCount cannot be changed from " + String::number(m_channelCount) + " to " + String::number(channelCount));
    }
}

void ScriptProcessorNode::setChannelCountMode(const String& mode, ExceptionState& exceptionState)
{
    ASSERT(isMainThread());
    AudioContext::AutoLocker locker(context());

    if ((mode == "max") || (mode == "clamped-max")) {
        exceptionState.throwDOMException(
            NotSupportedError,
            "channelCountMode cannot be changed from 'explicit' to '" + mode + "'");
    }
}

void ScriptProcessorNode::trace(Visitor* visitor)
{
    visitor->trace(m_inputBuffers);
    visitor->trace(m_outputBuffers);
    AudioNode::trace(visitor);
}

} // namespace blink

#endif // ENABLE(WEB_AUDIO)