/*
 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_MAC_H
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_MAC_H

#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/audio_device/audio_device_generic.h"
#include "webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"

#include <AudioToolbox/AudioConverter.h>
#include <CoreAudio/CoreAudio.h>
#include <mach/semaphore.h>

struct PaUtilRingBuffer;

namespace rtc {
class PlatformThread;
}  // namespace rtc

namespace webrtc
{
class EventWrapper;

const uint32_t N_REC_SAMPLES_PER_SEC = 48000;
const uint32_t N_PLAY_SAMPLES_PER_SEC = 48000;

const uint32_t N_REC_CHANNELS = 1; // default is mono recording
const uint32_t N_PLAY_CHANNELS = 2; // default is stereo playout
const uint32_t N_DEVICE_CHANNELS = 64;

const uint32_t ENGINE_REC_BUF_SIZE_IN_SAMPLES = (N_REC_SAMPLES_PER_SEC / 100);
const uint32_t ENGINE_PLAY_BUF_SIZE_IN_SAMPLES = (N_PLAY_SAMPLES_PER_SEC / 100);

const int N_BLOCKS_IO = 2;
const int N_BUFFERS_IN = 2;  // Must be at least N_BLOCKS_IO.
const int N_BUFFERS_OUT = 3;  // Must be at least N_BLOCKS_IO.

const uint32_t TIMER_PERIOD_MS = (2 * 10 * N_BLOCKS_IO * 1000000);

const uint32_t REC_BUF_SIZE_IN_SAMPLES =
    ENGINE_REC_BUF_SIZE_IN_SAMPLES * N_DEVICE_CHANNELS * N_BUFFERS_IN;
const uint32_t PLAY_BUF_SIZE_IN_SAMPLES =
    ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * N_PLAY_CHANNELS * N_BUFFERS_OUT;

class AudioDeviceMac: public AudioDeviceGeneric
{
public:
    AudioDeviceMac(const int32_t id);
    ~AudioDeviceMac();

    // Retrieve the currently utilized audio layer
    virtual int32_t
        ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;

    // Main initializaton and termination
    virtual int32_t Init();
    virtual int32_t Terminate();
    virtual bool Initialized() const;

    // Device enumeration
    virtual int16_t PlayoutDevices();
    virtual int16_t RecordingDevices();
    virtual int32_t PlayoutDeviceName(
        uint16_t index,
        char name[kAdmMaxDeviceNameSize],
        char guid[kAdmMaxGuidSize]);
    virtual int32_t RecordingDeviceName(
        uint16_t index,
        char name[kAdmMaxDeviceNameSize],
        char guid[kAdmMaxGuidSize]);

    // Device selection
    virtual int32_t SetPlayoutDevice(uint16_t index);
    virtual int32_t SetPlayoutDevice(
        AudioDeviceModule::WindowsDeviceType device);
    virtual int32_t SetRecordingDevice(uint16_t index);
    virtual int32_t SetRecordingDevice(
        AudioDeviceModule::WindowsDeviceType device);

    // Audio transport initialization
    virtual int32_t PlayoutIsAvailable(bool& available);
    virtual int32_t InitPlayout();
    virtual bool PlayoutIsInitialized() const;
    virtual int32_t RecordingIsAvailable(bool& available);
    virtual int32_t InitRecording();
    virtual bool RecordingIsInitialized() const;

    // Audio transport control
    virtual int32_t StartPlayout();
    virtual int32_t StopPlayout();
    virtual bool Playing() const;
    virtual int32_t StartRecording();
    virtual int32_t StopRecording();
    virtual bool Recording() const;

    // Microphone Automatic Gain Control (AGC)
    virtual int32_t SetAGC(bool enable);
    virtual bool AGC() const;

    // Volume control based on the Windows Wave API (Windows only)
    virtual int32_t SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight);
    virtual int32_t WaveOutVolume(uint16_t& volumeLeft,
                                  uint16_t& volumeRight) const;

    // Audio mixer initialization
    virtual int32_t InitSpeaker();
    virtual bool SpeakerIsInitialized() const;
    virtual int32_t InitMicrophone();
    virtual bool MicrophoneIsInitialized() const;

    // Speaker volume controls
    virtual int32_t SpeakerVolumeIsAvailable(bool& available);
    virtual int32_t SetSpeakerVolume(uint32_t volume);
    virtual int32_t SpeakerVolume(uint32_t& volume) const;
    virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
    virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const;
    virtual int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const;

    // Microphone volume controls
    virtual int32_t MicrophoneVolumeIsAvailable(bool& available);
    virtual int32_t SetMicrophoneVolume(uint32_t volume);
    virtual int32_t MicrophoneVolume(uint32_t& volume) const;
    virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
    virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
    virtual int32_t
        MicrophoneVolumeStepSize(uint16_t& stepSize) const;

    // Microphone mute control
    virtual int32_t MicrophoneMuteIsAvailable(bool& available);
    virtual int32_t SetMicrophoneMute(bool enable);
    virtual int32_t MicrophoneMute(bool& enabled) const;

    // Speaker mute control
    virtual int32_t SpeakerMuteIsAvailable(bool& available);
    virtual int32_t SetSpeakerMute(bool enable);
    virtual int32_t SpeakerMute(bool& enabled) const;

    // Microphone boost control
    virtual int32_t MicrophoneBoostIsAvailable(bool& available);
    virtual int32_t SetMicrophoneBoost(bool enable);
    virtual int32_t MicrophoneBoost(bool& enabled) const;

    // Stereo support
    virtual int32_t StereoPlayoutIsAvailable(bool& available);
    virtual int32_t SetStereoPlayout(bool enable);
    virtual int32_t StereoPlayout(bool& enabled) const;
    virtual int32_t StereoRecordingIsAvailable(bool& available);
    virtual int32_t SetStereoRecording(bool enable);
    virtual int32_t StereoRecording(bool& enabled) const;

    // Delay information and control
    virtual int32_t
        SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
                         uint16_t sizeMS);
    virtual int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,
                                  uint16_t& sizeMS) const;
    virtual int32_t PlayoutDelay(uint16_t& delayMS) const;
    virtual int32_t RecordingDelay(uint16_t& delayMS) const;

    // CPU load
    virtual int32_t CPULoad(uint16_t& load) const;

    virtual bool PlayoutWarning() const;
    virtual bool PlayoutError() const;
    virtual bool RecordingWarning() const;
    virtual bool RecordingError() const;
    virtual void ClearPlayoutWarning();
    virtual void ClearPlayoutError();
    virtual void ClearRecordingWarning();
    virtual void ClearRecordingError();

    virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);

private:
    virtual int32_t MicrophoneIsAvailable(bool& available);
    virtual int32_t SpeakerIsAvailable(bool& available);

    static void AtomicSet32(int32_t* theValue, int32_t newValue);
    static int32_t AtomicGet32(int32_t* theValue);

    static void logCAMsg(const TraceLevel level,
                         const TraceModule module,
                         const int32_t id, const char *msg,
                         const char *err);

    int32_t GetNumberDevices(const AudioObjectPropertyScope scope,
                             AudioDeviceID scopedDeviceIds[],
                             const uint32_t deviceListLength);

    int32_t GetDeviceName(const AudioObjectPropertyScope scope,
                          const uint16_t index, char* name);

    int32_t InitDevice(uint16_t userDeviceIndex,
                       AudioDeviceID& deviceId, bool isInput);

    // Always work with our preferred playout format inside VoE.
    // Then convert the output to the OS setting using an AudioConverter.
    OSStatus SetDesiredPlayoutFormat();

    static OSStatus
        objectListenerProc(AudioObjectID objectId, UInt32 numberAddresses,
                           const AudioObjectPropertyAddress addresses[],
                           void* clientData);

    OSStatus
        implObjectListenerProc(AudioObjectID objectId, UInt32 numberAddresses,
                               const AudioObjectPropertyAddress addresses[]);

    int32_t HandleDeviceChange();

    int32_t
        HandleStreamFormatChange(AudioObjectID objectId,
                                 AudioObjectPropertyAddress propertyAddress);

    int32_t
        HandleDataSourceChange(AudioObjectID objectId,
                               AudioObjectPropertyAddress propertyAddress);

    int32_t
        HandleProcessorOverload(AudioObjectPropertyAddress propertyAddress);

    static OSStatus deviceIOProc(AudioDeviceID device,
                                 const AudioTimeStamp *now,
                                 const AudioBufferList *inputData,
                                 const AudioTimeStamp *inputTime,
                                 AudioBufferList *outputData,
                                 const AudioTimeStamp* outputTime,
                                 void *clientData);

    static OSStatus
        outConverterProc(AudioConverterRef audioConverter,
                         UInt32 *numberDataPackets, AudioBufferList *data,
                         AudioStreamPacketDescription **dataPacketDescription,
                         void *userData);

    static OSStatus inDeviceIOProc(AudioDeviceID device,
                                   const AudioTimeStamp *now,
                                   const AudioBufferList *inputData,
                                   const AudioTimeStamp *inputTime,
                                   AudioBufferList *outputData,
                                   const AudioTimeStamp *outputTime,
                                   void *clientData);

    static OSStatus
        inConverterProc(AudioConverterRef audioConverter,
                        UInt32 *numberDataPackets, AudioBufferList *data,
                        AudioStreamPacketDescription **dataPacketDescription,
                        void *inUserData);

    OSStatus implDeviceIOProc(const AudioBufferList *inputData,
                              const AudioTimeStamp *inputTime,
                              AudioBufferList *outputData,
                              const AudioTimeStamp *outputTime);

    OSStatus implOutConverterProc(UInt32 *numberDataPackets,
                                  AudioBufferList *data);

    OSStatus implInDeviceIOProc(const AudioBufferList *inputData,
                                const AudioTimeStamp *inputTime);

    OSStatus implInConverterProc(UInt32 *numberDataPackets,
                                 AudioBufferList *data);

    static bool RunCapture(void*);
    static bool RunRender(void*);
    bool CaptureWorkerThread();
    bool RenderWorkerThread();

    bool KeyPressed();

    AudioDeviceBuffer* _ptrAudioBuffer;

    CriticalSectionWrapper& _critSect;

    EventWrapper& _stopEventRec;
    EventWrapper& _stopEvent;

    // TODO(pbos): Replace with direct members, just start/stop, no need to
    // recreate the thread.
    // Only valid/running between calls to StartRecording and StopRecording.
    rtc::scoped_ptr<rtc::PlatformThread> capture_worker_thread_;

    // Only valid/running between calls to StartPlayout and StopPlayout.
    rtc::scoped_ptr<rtc::PlatformThread> render_worker_thread_;

    int32_t _id;

    AudioMixerManagerMac _mixerManager;

    uint16_t _inputDeviceIndex;
    uint16_t _outputDeviceIndex;
    AudioDeviceID _inputDeviceID;
    AudioDeviceID _outputDeviceID;
#if __MAC_OS_X_VERSION_MAX_ALLOWED >= 1050
    AudioDeviceIOProcID _inDeviceIOProcID;
    AudioDeviceIOProcID _deviceIOProcID;
#endif
    bool _inputDeviceIsSpecified;
    bool _outputDeviceIsSpecified;

    uint8_t _recChannels;
    uint8_t _playChannels;

    Float32* _captureBufData;
    SInt16* _renderBufData;

    SInt16 _renderConvertData[PLAY_BUF_SIZE_IN_SAMPLES];

    AudioDeviceModule::BufferType _playBufType;

    bool _initialized;
    bool _isShutDown;
    bool _recording;
    bool _playing;
    bool _recIsInitialized;
    bool _playIsInitialized;
    bool _AGC;

    // Atomically set varaibles
    int32_t _renderDeviceIsAlive;
    int32_t _captureDeviceIsAlive;

    bool _twoDevices;
    bool _doStop; // For play if not shared device or play+rec if shared device
    bool _doStopRec; // For rec if not shared device
    bool _macBookPro;
    bool _macBookProPanRight;

    AudioConverterRef _captureConverter;
    AudioConverterRef _renderConverter;

    AudioStreamBasicDescription _outStreamFormat;
    AudioStreamBasicDescription _outDesiredFormat;
    AudioStreamBasicDescription _inStreamFormat;
    AudioStreamBasicDescription _inDesiredFormat;

    uint32_t _captureLatencyUs;
    uint32_t _renderLatencyUs;

    // Atomically set variables
    mutable int32_t _captureDelayUs;
    mutable int32_t _renderDelayUs;

    int32_t _renderDelayOffsetSamples;

    uint16_t _playBufDelayFixed; // fixed playback delay

    uint16_t _playWarning;
    uint16_t _playError;
    uint16_t _recWarning;
    uint16_t _recError;

    PaUtilRingBuffer* _paCaptureBuffer;
    PaUtilRingBuffer* _paRenderBuffer;

    semaphore_t _renderSemaphore;
    semaphore_t _captureSemaphore;

    int _captureBufSizeSamples;
    int _renderBufSizeSamples;

    // Typing detection
    // 0x5c is key "9", after that comes function keys.
    bool prev_key_state_[0x5d];
};

}  // namespace webrtc

#endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_AUDIO_DEVICE_MAC_H_