//
// libtgvoip is free and unencumbered public domain software.
// For more information, see http://unlicense.org or the UNLICENSE file
// you should have received with this source code distribution.
//

#ifndef TGVOIP_NO_DSP
#include "modules/audio_processing/include/audio_processing.h"
#include "api/audio/audio_frame.h"
#endif

#include "EchoCanceller.h"
#include "audio/AudioOutput.h"
#include "audio/AudioInput.h"
#include "logging.h"
#include "VoIPServerConfig.h"
#include <string.h>
#include <stdio.h>
#include <math.h>

using namespace tgvoip;

EchoCanceller::EchoCanceller(bool enableAEC, bool enableNS, bool enableAGC){
#ifndef TGVOIP_NO_DSP
	this->enableAEC=enableAEC;
	this->enableAGC=enableAGC;
	this->enableNS=enableNS;
	isOn=true;

	webrtc::Config extraConfig;

	apm=webrtc::AudioProcessingBuilder().Create(extraConfig);

	webrtc::AudioProcessing::Config config;
	config.echo_canceller.enabled = enableAEC;
#ifndef TGVOIP_USE_DESKTOP_DSP
	config.echo_canceller.mobile_mode = true;
#else
	config.echo_canceller.mobile_mode = false;
#endif
	config.high_pass_filter.enabled = enableAEC;
	config.gain_controller2.enabled = enableAGC;

    using Level = webrtc::AudioProcessing::Config::NoiseSuppression::Level;
    Level nsLevel;
#ifdef __APPLE__
	switch(ServerConfig::GetSharedInstance()->GetInt("webrtc_ns_level_vpio", 0)){
#else
    switch (ServerConfig::GetSharedInstance()->GetInt("webrtc_ns_level", 2)) {
#endif
        case 0:
            nsLevel=Level::kLow;
            break;
        case 1:
            nsLevel=Level::kModerate;
            break;
        case 3:
            nsLevel=Level::kVeryHigh;
            break;
        case 2:
        default:
            nsLevel=Level::kHigh;
            break;
    }
    config.noise_suppression.level = nsLevel;
    config.noise_suppression.enabled = enableNS;
    if (enableAGC) {
        config.gain_controller1.mode = webrtc::AudioProcessing::Config::GainController1::kAdaptiveDigital;
        config.gain_controller1.target_level_dbfs = ServerConfig::GetSharedInstance()->GetInt("webrtc_agc_target_level", 9);
        config.gain_controller1.enable_limiter = ServerConfig::GetSharedInstance()->GetBoolean("webrtc_agc_enable_limiter", true);
        config.gain_controller1.compression_gain_db = ServerConfig::GetSharedInstance()->GetInt("webrtc_agc_compression_gain", 20);
    }
    config.voice_detection.enabled = true;

    apm->ApplyConfig(config);

    audioFrame = new webrtc::AudioFrame();
    audioFrame->samples_per_channel_ = 480;
    audioFrame->sample_rate_hz_ = 48000;
    audioFrame->num_channels_ = 1;

    farendQueue = new BlockingQueue<int16_t *>(11);
    farendBufferPool = new BufferPool(960 * 2, 10);
    running = true;
    bufferFarendThread = new Thread(std::bind(&EchoCanceller::RunBufferFarendThread, this));
    bufferFarendThread->Start();

#else
    this->enableAEC=this->enableAGC=enableAGC=this->enableNS=enableNS=false;
    isOn=true;
#endif
}

EchoCanceller::~EchoCanceller() {
#ifndef TGVOIP_NO_DSP
    delete apm;
    delete audioFrame;
    delete farendBufferPool;
#endif
}

void EchoCanceller::Start() {

}

void EchoCanceller::Stop() {

}

void EchoCanceller::SpeakerOutCallback(unsigned char* data, size_t len) {
    if (len != 960 * 2 || !enableAEC || !isOn)
        return;
#ifndef TGVOIP_NO_DSP
    int16_t *buf = (int16_t *) farendBufferPool->Get();
    if (buf) {
        memcpy(buf, data, 960 * 2);
        farendQueue->Put(buf);
    }
#endif
}

#ifndef TGVOIP_NO_DSP
void EchoCanceller::RunBufferFarendThread() {
    webrtc::AudioFrame frame;
    frame.num_channels_ = 1;
    frame.sample_rate_hz_ = 48000;
    frame.samples_per_channel_ = 480;

    webrtc::StreamConfig input_config(frame.sample_rate_hz_, frame.num_channels_,
            /*has_keyboard=*/false);
    webrtc::StreamConfig output_config(frame.sample_rate_hz_, frame.num_channels_,
            /*has_keyboard=*/false);

    while (running) {
        int16_t *samplesIn = farendQueue->GetBlocking();
        if (samplesIn) {
            memcpy(frame.mutable_data(), samplesIn, 480 * 2);
            apm->ProcessReverseStream(frame.data(), input_config,
                                      output_config, frame.mutable_data());
            memcpy(frame.mutable_data(), samplesIn + 480, 480 * 2);
            apm->ProcessReverseStream(frame.data(), input_config,
                                      output_config, frame.mutable_data());
            didBufferFarend = true;
            farendBufferPool->Reuse(reinterpret_cast<unsigned char *>(samplesIn));
        }
    }
}
#endif

void EchoCanceller::Enable(bool enabled) {
    isOn = enabled;
}

void EchoCanceller::ProcessInput(int16_t* inOut, size_t numSamples, bool& hasVoice) {
#ifndef TGVOIP_NO_DSP
    if (!isOn || (!enableAEC && !enableAGC && !enableNS)) {
        return;
    }
    int delay = audio::AudioInput::GetEstimatedDelay() + audio::AudioOutput::GetEstimatedDelay();
    assert(numSamples == 960);

    webrtc::StreamConfig input_config(audioFrame->sample_rate_hz_, audioFrame->num_channels_,
            /*has_keyboard=*/false);
    webrtc::StreamConfig output_config(audioFrame->sample_rate_hz_, audioFrame->num_channels_,
            /*has_keyboard=*/false);

    memcpy(audioFrame->mutable_data(), inOut, 480 * 2);
    if (enableAEC)
        apm->set_stream_delay_ms(delay);
    apm->ProcessStream(audioFrame->data(), input_config,
                       output_config, audioFrame->mutable_data());
    if (enableVAD)
        hasVoice= apm->GetStatistics().voice_detected.value_or(false);
    memcpy(inOut, audioFrame->data(), 480 * 2);
    memcpy(audioFrame->mutable_data(), inOut + 480, 480 * 2);
    if (enableAEC)
        apm->set_stream_delay_ms(delay);
    apm->ProcessStream(audioFrame->data(), input_config,
                       output_config, audioFrame->mutable_data());
    if (enableVAD) {
        hasVoice=hasVoice || apm->GetStatistics().voice_detected.value_or(false);
    }
    memcpy(inOut + 480, audioFrame->data(), 480 * 2);
#endif
}

void EchoCanceller::SetAECStrength(int strength) {
#ifndef TGVOIP_NO_DSP
    /*if(aec){
#ifndef TGVOIP_USE_DESKTOP_DSP
        AecmConfig cfg;
        cfg.cngMode=AecmFalse;
        cfg.echoMode=(int16_t) strength;
        WebRtcAecm_set_config(aec, cfg);
#endif
    }*/
#endif
}

void EchoCanceller::SetVoiceDetectionEnabled(bool enabled) {
    enableVAD = enabled;
#ifndef TGVOIP_NO_DSP
    auto config = apm->GetConfig();
    config.voice_detection.enabled = enabled;
    apm->ApplyConfig(config);
#endif
}

using namespace tgvoip::effects;

AudioEffect::~AudioEffect() {

}

void AudioEffect::SetPassThrough(bool passThrough) {
    this->passThrough = passThrough;
}

Volume::Volume() {

}

Volume::~Volume() {

}

void Volume::Process(int16_t* inOut, size_t numSamples) {
    if (level == 1.0f || passThrough) {
        return;
    }
    for (size_t i = 0; i < numSamples; i++) {
        float sample = (float) inOut[i] * multiplier;
        if (sample > 32767.0f)
            inOut[i] = INT16_MAX;
        else if (sample < -32768.0f)
            inOut[i] = INT16_MIN;
        else
            inOut[i] = (int16_t) sample;
    }
}

void Volume::SetLevel(float level) {
    this->level = level;
    float db;
    if (level < 1.0f)
        db = -50.0f * (1.0f - level);
    else if (level > 1.0f && level <= 2.0f)
        db = 10.0f * (level - 1.0f);
    else
        db = 0.0f;
    multiplier = expf(db / 20.0f * logf(10.0f));
}

float Volume::GetLevel() {
    return level;
}