/*
 * Copyright (C) 2021 Apple 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"
#include "AudioMediaStreamTrackRendererInternalUnit.h"

#if ENABLE(MEDIA_STREAM)

#include "AudioSampleDataSource.h"
#include "AudioSession.h"
#include "CAAudioStreamDescription.h"
#include "Logging.h"

#include <Accelerate/Accelerate.h>
#include <pal/spi/cocoa/AudioToolboxSPI.h>
#include <wtf/FastMalloc.h>
#include <wtf/Lock.h>

#if PLATFORM(COCOA)
#include "CoreAudioCaptureDevice.h"
#include "CoreAudioCaptureDeviceManager.h"
#endif

#include <pal/cf/AudioToolboxSoftLink.h>
#include <pal/cf/CoreMediaSoftLink.h>

namespace WebCore {

class LocalAudioMediaStreamTrackRendererInternalUnit final : public AudioMediaStreamTrackRendererInternalUnit {
    WTF_MAKE_FAST_ALLOCATED;
public:
    LocalAudioMediaStreamTrackRendererInternalUnit(RenderCallback&&, ResetCallback&&);

private:
    void createAudioUnitIfNeeded();

    // AudioMediaStreamTrackRendererInternalUnit API.
    void start() final;
    void stop() final;
    void retrieveFormatDescription(CompletionHandler<void(const CAAudioStreamDescription*)>&&) final;
    void setAudioOutputDevice(const String&) final;

    OSStatus render(AudioUnitRenderActionFlags*, const AudioTimeStamp*, UInt32 sampleCount, AudioBufferList*);
    static OSStatus renderingCallback(void*, AudioUnitRenderActionFlags*, const AudioTimeStamp*, UInt32 inBusNumber, UInt32 sampleCount, AudioBufferList*);

    RenderCallback m_renderCallback;
    ResetCallback m_resetCallback;
    std::unique_ptr<CAAudioStreamDescription> m_outputDescription;
    AudioComponentInstance m_remoteIOUnit { nullptr };
    bool m_isStarted { false };
    uint64_t m_sampleTime { 0 };
#if PLATFORM(MAC)
    uint32_t m_deviceID { 0 };
#endif
};

UniqueRef<AudioMediaStreamTrackRendererInternalUnit> AudioMediaStreamTrackRendererInternalUnit::createLocalInternalUnit(RenderCallback&& renderCallback, ResetCallback&& resetCallback)
{
    return makeUniqueRef<LocalAudioMediaStreamTrackRendererInternalUnit>(WTFMove(renderCallback), WTFMove(resetCallback));
}

LocalAudioMediaStreamTrackRendererInternalUnit::LocalAudioMediaStreamTrackRendererInternalUnit(RenderCallback&& renderCallback, ResetCallback&& resetCallback)
    : m_renderCallback(WTFMove(renderCallback))
    , m_resetCallback(WTFMove(resetCallback))
{
}

void LocalAudioMediaStreamTrackRendererInternalUnit::retrieveFormatDescription(CompletionHandler<void(const CAAudioStreamDescription*)>&& callback)
{
    createAudioUnitIfNeeded();
    callback(m_outputDescription.get());
}

void LocalAudioMediaStreamTrackRendererInternalUnit::setAudioOutputDevice(const String& deviceID)
{
#if PLATFORM(MAC)
    auto device = CoreAudioCaptureDeviceManager::singleton().coreAudioDeviceWithUID(deviceID);

    if (!device && !deviceID.isEmpty()) {
        RELEASE_LOG(WebRTC, "AudioMediaStreamTrackRendererInternalUnit::setAudioOutputDeviceId - did not find device");
        return;
    }

    auto audioUnitDeviceID = device ? device->deviceID() : 0;
    if (m_deviceID == audioUnitDeviceID)
        return;

    bool shouldRestart = m_isStarted;
    if (m_isStarted)
        stop();

    m_deviceID = audioUnitDeviceID;

    if (shouldRestart)
        start();
#else
    UNUSED_PARAM(deviceID);
#endif
}

void LocalAudioMediaStreamTrackRendererInternalUnit::start()
{
    RELEASE_LOG_INFO(WebRTC, "LocalAudioMediaStreamTrackRendererInternalUnit::start");
    if (m_isStarted)
        return;

    createAudioUnitIfNeeded();
    if (!m_remoteIOUnit)
        return;

    m_sampleTime = 0;
    if (auto error = PAL::AudioOutputUnitStart(m_remoteIOUnit)) {
        RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererInternalUnit::start AudioOutputUnitStart failed, error = %d", error);
        PAL::AudioComponentInstanceDispose(m_remoteIOUnit);
        m_remoteIOUnit = nullptr;
        return;
    }
    m_isStarted = true;
    RELEASE_LOG(WebRTC, "AudioMediaStreamTrackRendererInternalUnit is started");
}

void LocalAudioMediaStreamTrackRendererInternalUnit::stop()
{
    RELEASE_LOG_INFO(WebRTC, "LocalAudioMediaStreamTrackRendererInternalUnit::stop");
    if (!m_remoteIOUnit)
        return;

    if (m_isStarted) {
        PAL::AudioOutputUnitStop(m_remoteIOUnit);
        m_isStarted = false;
    }

    PAL::AudioComponentInstanceDispose(m_remoteIOUnit);
    m_remoteIOUnit = nullptr;
}

void LocalAudioMediaStreamTrackRendererInternalUnit::createAudioUnitIfNeeded()
{
    ASSERT(!m_remoteIOUnit || m_outputDescription);
    if (m_remoteIOUnit)
        return;

    AudioComponentInstance remoteIOUnit { nullptr };

    AudioComponentDescription ioUnitDescription { kAudioUnitType_Output, 0, kAudioUnitManufacturer_Apple, 0, 0 };
#if PLATFORM(IOS_FAMILY)
    ioUnitDescription.componentSubType = kAudioUnitSubType_RemoteIO;
#else
    ioUnitDescription.componentSubType = kAudioUnitSubType_DefaultOutput;
#endif

    AudioComponent ioComponent = PAL::AudioComponentFindNext(nullptr, &ioUnitDescription);
    ASSERT(ioComponent);
    if (!ioComponent) {
        RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererInternalUnit::createAudioUnit unable to find remote IO unit component");
        return;
    }

    auto error = PAL::AudioComponentInstanceNew(ioComponent, &remoteIOUnit);
    if (error) {
        RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererInternalUnit::createAudioUnit unable to open vpio unit, error = %d", error);
        return;
    }

#if PLATFORM(IOS_FAMILY)
    UInt32 param = 1;
    error = PAL::AudioUnitSetProperty(remoteIOUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &param, sizeof(param));
    if (error) {
        RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererInternalUnit::createAudioUnit unable to enable vpio unit output, error = %d", error);
        return;
    }
#endif

#if PLATFORM(MAC)
    if (m_deviceID) {
        error = PAL::AudioUnitSetProperty(remoteIOUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &m_deviceID, sizeof(m_deviceID));
        if (error) {
            RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererInternalUnit::createAudioUnit unable to set unit device ID %d, error %d (%.4s)", (int)m_deviceID, (int)error, (char*)&error);
            return;
        }
    }
#endif

    AURenderCallbackStruct callback = { LocalAudioMediaStreamTrackRendererInternalUnit::renderingCallback, this };
    error = PAL::AudioUnitSetProperty(remoteIOUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Global, 0, &callback, sizeof(callback));
    if (error) {
        RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererInternalUnit::createAudioUnit unable to set vpio unit speaker proc, error = %d", error);
        return;
    }

    if (!m_outputDescription) {
        CAAudioStreamDescription outputDescription;
        UInt32 size = sizeof(outputDescription.streamDescription());
        error  = PAL::AudioUnitGetProperty(remoteIOUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &outputDescription.streamDescription(), &size);
        if (error) {
            RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererInternalUnit::createAudioUnit unable to get input stream format, error = %d", error);
            return;
        }

        outputDescription.streamDescription().mSampleRate = AudioSession::sharedSession().sampleRate();
        m_outputDescription = makeUnique<CAAudioStreamDescription>(outputDescription);
    }
    error = PAL::AudioUnitSetProperty(remoteIOUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &m_outputDescription->streamDescription(), sizeof(m_outputDescription->streamDescription()));
    if (error) {
        RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererInternalUnit::createAudioUnit unable to set input stream format, error = %d", error);
        return;
    }

    error = PAL::AudioUnitInitialize(remoteIOUnit);
    if (error) {
        RELEASE_LOG_ERROR(WebRTC, "AudioMediaStreamTrackRendererInternalUnit::createAudioUnit AudioUnitInitialize() failed, error = %d", error);
        return;
    }

    m_remoteIOUnit = remoteIOUnit;
}

static void clipAudioBuffer(float* vector, size_t size)
{
    float minimum = -1;
    float maximum = 1;
    vDSP_vclip(vector, 1, &minimum, &maximum, vector, 1, size);
}

static void clipAudioBuffer(double* vector, size_t size)
{
    double minimum = -1;
    double maximum = 1;
    vDSP_vclipD(vector, 1, &minimum, &maximum, vector, 1, size);
}

static void clipAudioBufferList(AudioBufferList& list, AudioStreamDescription::PCMFormat format)
{
    switch (format) {
    case AudioStreamDescription::Int16:
        break;
    case AudioStreamDescription::Int32:
        break;
    case AudioStreamDescription::Float32:
        for (size_t index = 0; index < list.mNumberBuffers ; ++index)
            clipAudioBuffer(static_cast<float*>(list.mBuffers[index].mData), list.mBuffers[index].mDataByteSize / sizeof(float));
        break;
    case AudioStreamDescription::Float64:
        for (size_t index = 0; index < list.mNumberBuffers ; ++index)
            clipAudioBuffer(static_cast<double*>(list.mBuffers[index].mData), list.mBuffers[index].mDataByteSize / sizeof(double));
        break;
    case AudioStreamDescription::None:
        ASSERT_NOT_REACHED();
        break;
    }
}

OSStatus LocalAudioMediaStreamTrackRendererInternalUnit::render(AudioUnitRenderActionFlags* actionFlags, const AudioTimeStamp* timeStamp, UInt32 sampleCount, AudioBufferList* ioData)
{
    auto sampleTime = timeStamp->mSampleTime;
    // If we observe an irregularity in the timeline, we trigger a reset.
    if (m_sampleTime && (m_sampleTime + 2 * sampleCount < sampleTime || sampleTime <= m_sampleTime))
        m_resetCallback();
    m_sampleTime = sampleTime < std::numeric_limits<Float64>::max() - sampleCount ? sampleTime : 0;

    auto result = m_renderCallback(sampleCount, *ioData, sampleTime, timeStamp->mHostTime, *actionFlags);
    // FIXME: We should probably introduce a limiter to limit the amount of clipping.
    clipAudioBufferList(*ioData, m_outputDescription->format());
    return result;
}

OSStatus LocalAudioMediaStreamTrackRendererInternalUnit::renderingCallback(void* processor, AudioUnitRenderActionFlags* actionFlags, const AudioTimeStamp* timeStamp, UInt32, UInt32 sampleCount, AudioBufferList* ioData)
{
    return static_cast<LocalAudioMediaStreamTrackRendererInternalUnit*>(processor)->render(actionFlags, timeStamp, sampleCount, ioData);
}

} // namespace WebCore

#endif // ENABLE(MEDIA_STREAM)