/*
 *  Copyright (c) 2016 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.
 */

#include "webrtc/modules/audio_processing/test/audio_processing_simulator.h"

#include <algorithm>
#include <iostream>
#include <sstream>
#include <string>
#include <vector>

#include "webrtc/base/stringutils.h"
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"

namespace webrtc {
namespace test {
namespace {

void CopyFromAudioFrame(const AudioFrame& src, ChannelBuffer<float>* dest) {
  RTC_CHECK_EQ(src.num_channels_, dest->num_channels());
  RTC_CHECK_EQ(src.samples_per_channel_, dest->num_frames());
  // Copy the data from the input buffer.
  std::vector<float> tmp(src.samples_per_channel_ * src.num_channels_);
  S16ToFloat(src.data_, tmp.size(), tmp.data());
  Deinterleave(tmp.data(), src.samples_per_channel_, src.num_channels_,
               dest->channels());
}

std::string GetIndexedOutputWavFilename(const std::string& wav_name,
                                        int counter) {
  std::stringstream ss;
  ss << wav_name.substr(0, wav_name.size() - 4) << "_" << counter
     << wav_name.substr(wav_name.size() - 4);
  return ss.str();
}

}  // namespace

void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) {
  RTC_CHECK_EQ(src.num_channels(), dest->num_channels_);
  RTC_CHECK_EQ(src.num_frames(), dest->samples_per_channel_);
  for (size_t ch = 0; ch < dest->num_channels_; ++ch) {
    for (size_t sample = 0; sample < dest->samples_per_channel_; ++sample) {
      dest->data_[sample * dest->num_channels_ + ch] =
          src.channels()[ch][sample] * 32767;
    }
  }
}

AudioProcessingSimulator::ScopedTimer::~ScopedTimer() {
  int64_t interval = rtc::TimeNanos() - start_time_;
  proc_time_->sum += interval;
  proc_time_->max = std::max(proc_time_->max, interval);
  proc_time_->min = std::min(proc_time_->min, interval);
}

void AudioProcessingSimulator::ProcessStream(bool fixed_interface) {
  if (fixed_interface) {
    {
      const auto st = ScopedTimer(mutable_proc_time());
      RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->ProcessStream(&fwd_frame_));
    }
    CopyFromAudioFrame(fwd_frame_, out_buf_.get());
  } else {
    const auto st = ScopedTimer(mutable_proc_time());
    RTC_CHECK_EQ(AudioProcessing::kNoError,
                 ap_->ProcessStream(in_buf_->channels(), in_config_,
                                    out_config_, out_buf_->channels()));
  }

  if (buffer_writer_) {
    buffer_writer_->Write(*out_buf_);
  }

  ++num_process_stream_calls_;
}

void AudioProcessingSimulator::ProcessReverseStream(bool fixed_interface) {
  if (fixed_interface) {
    const auto st = ScopedTimer(mutable_proc_time());
    RTC_CHECK_EQ(AudioProcessing::kNoError,
                 ap_->ProcessReverseStream(&rev_frame_));
    CopyFromAudioFrame(rev_frame_, reverse_out_buf_.get());

  } else {
    const auto st = ScopedTimer(mutable_proc_time());
    RTC_CHECK_EQ(AudioProcessing::kNoError,
                 ap_->ProcessReverseStream(
                     reverse_in_buf_->channels(), reverse_in_config_,
                     reverse_out_config_, reverse_out_buf_->channels()));
  }

  if (reverse_buffer_writer_) {
    reverse_buffer_writer_->Write(*reverse_out_buf_);
  }

  ++num_reverse_process_stream_calls_;
}

void AudioProcessingSimulator::SetupBuffersConfigsOutputs(
    int input_sample_rate_hz,
    int output_sample_rate_hz,
    int reverse_input_sample_rate_hz,
    int reverse_output_sample_rate_hz,
    int input_num_channels,
    int output_num_channels,
    int reverse_input_num_channels,
    int reverse_output_num_channels) {
  in_config_ = StreamConfig(input_sample_rate_hz, input_num_channels);
  in_buf_.reset(new ChannelBuffer<float>(
      rtc::CheckedDivExact(input_sample_rate_hz, kChunksPerSecond),
      input_num_channels));

  reverse_in_config_ =
      StreamConfig(reverse_input_sample_rate_hz, reverse_input_num_channels);
  reverse_in_buf_.reset(new ChannelBuffer<float>(
      rtc::CheckedDivExact(reverse_input_sample_rate_hz, kChunksPerSecond),
      reverse_input_num_channels));

  out_config_ = StreamConfig(output_sample_rate_hz, output_num_channels);
  out_buf_.reset(new ChannelBuffer<float>(
      rtc::CheckedDivExact(output_sample_rate_hz, kChunksPerSecond),
      output_num_channels));

  reverse_out_config_ =
      StreamConfig(reverse_output_sample_rate_hz, reverse_output_num_channels);
  reverse_out_buf_.reset(new ChannelBuffer<float>(
      rtc::CheckedDivExact(reverse_output_sample_rate_hz, kChunksPerSecond),
      reverse_output_num_channels));

  fwd_frame_.sample_rate_hz_ = input_sample_rate_hz;
  fwd_frame_.samples_per_channel_ =
      rtc::CheckedDivExact(fwd_frame_.sample_rate_hz_, kChunksPerSecond);
  fwd_frame_.num_channels_ = input_num_channels;

  rev_frame_.sample_rate_hz_ = reverse_input_sample_rate_hz;
  rev_frame_.samples_per_channel_ =
      rtc::CheckedDivExact(rev_frame_.sample_rate_hz_, kChunksPerSecond);
  rev_frame_.num_channels_ = reverse_input_num_channels;

  if (settings_.use_verbose_logging) {
    std::cout << "Sample rates:" << std::endl;
    std::cout << " Forward input: " << input_sample_rate_hz << std::endl;
    std::cout << " Forward output: " << output_sample_rate_hz << std::endl;
    std::cout << " Reverse input: " << reverse_input_sample_rate_hz
              << std::endl;
    std::cout << " Reverse output: " << reverse_output_sample_rate_hz
              << std::endl;
    std::cout << "Number of channels: " << std::endl;
    std::cout << " Forward input: " << input_num_channels << std::endl;
    std::cout << " Forward output: " << output_num_channels << std::endl;
    std::cout << " Reverse input: " << reverse_input_num_channels << std::endl;
    std::cout << " Reverse output: " << reverse_output_num_channels
              << std::endl;
  }

  SetupOutput();
}

void AudioProcessingSimulator::SetupOutput() {
  if (settings_.output_filename) {
    std::string filename;
    if (settings_.store_intermediate_output) {
      filename = GetIndexedOutputWavFilename(*settings_.output_filename,
                                             output_reset_counter_);
    } else {
      filename = *settings_.output_filename;
    }

    std::unique_ptr<WavWriter> out_file(
        new WavWriter(filename, out_config_.sample_rate_hz(),
                      static_cast<size_t>(out_config_.num_channels())));
    buffer_writer_.reset(new ChannelBufferWavWriter(std::move(out_file)));
  }

  if (settings_.reverse_output_filename) {
    std::string filename;
    if (settings_.store_intermediate_output) {
      filename = GetIndexedOutputWavFilename(*settings_.reverse_output_filename,
                                             output_reset_counter_);
    } else {
      filename = *settings_.reverse_output_filename;
    }

    std::unique_ptr<WavWriter> reverse_out_file(
        new WavWriter(filename, reverse_out_config_.sample_rate_hz(),
                      static_cast<size_t>(reverse_out_config_.num_channels())));
    reverse_buffer_writer_.reset(
        new ChannelBufferWavWriter(std::move(reverse_out_file)));
  }

  ++output_reset_counter_;
}

void AudioProcessingSimulator::DestroyAudioProcessor() {
  if (settings_.aec_dump_output_filename) {
    RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->StopDebugRecording());
  }
}

void AudioProcessingSimulator::CreateAudioProcessor() {
  Config config;
  if (settings_.use_bf && *settings_.use_bf) {
    config.Set<Beamforming>(new Beamforming(
        true, ParseArrayGeometry(*settings_.microphone_positions),
        SphericalPointf(DegreesToRadians(settings_.target_angle_degrees), 0.f,
                        1.f)));
  }
  if (settings_.use_ts) {
    config.Set<ExperimentalNs>(new ExperimentalNs(*settings_.use_ts));
  }
  if (settings_.use_ie) {
    config.Set<Intelligibility>(new Intelligibility(*settings_.use_ie));
  }
  if (settings_.use_aec3) {
    config.Set<EchoCanceller3>(new EchoCanceller3(*settings_.use_aec3));
  }
  if (settings_.use_lc) {
    config.Set<LevelControl>(new LevelControl(true));
  }
  if (settings_.use_refined_adaptive_filter) {
    config.Set<RefinedAdaptiveFilter>(
        new RefinedAdaptiveFilter(*settings_.use_refined_adaptive_filter));
  }
  config.Set<ExtendedFilter>(new ExtendedFilter(
      !settings_.use_extended_filter || *settings_.use_extended_filter));
  config.Set<DelayAgnostic>(new DelayAgnostic(!settings_.use_delay_agnostic ||
                                              *settings_.use_delay_agnostic));

  ap_.reset(AudioProcessing::Create(config));

  if (settings_.use_aec) {
    RTC_CHECK_EQ(AudioProcessing::kNoError,
                 ap_->echo_cancellation()->Enable(*settings_.use_aec));
  }
  if (settings_.use_aecm) {
    RTC_CHECK_EQ(AudioProcessing::kNoError,
                 ap_->echo_control_mobile()->Enable(*settings_.use_aecm));
  }
  if (settings_.use_agc) {
    RTC_CHECK_EQ(AudioProcessing::kNoError,
                 ap_->gain_control()->Enable(*settings_.use_agc));
  }
  if (settings_.use_hpf) {
    RTC_CHECK_EQ(AudioProcessing::kNoError,
                 ap_->high_pass_filter()->Enable(*settings_.use_hpf));
  }
  if (settings_.use_ns) {
    RTC_CHECK_EQ(AudioProcessing::kNoError,
                 ap_->noise_suppression()->Enable(*settings_.use_ns));
  }
  if (settings_.use_le) {
    RTC_CHECK_EQ(AudioProcessing::kNoError,
                 ap_->level_estimator()->Enable(*settings_.use_le));
  }
  if (settings_.use_vad) {
    RTC_CHECK_EQ(AudioProcessing::kNoError,
                 ap_->voice_detection()->Enable(*settings_.use_vad));
  }
  if (settings_.use_agc_limiter) {
    RTC_CHECK_EQ(AudioProcessing::kNoError, ap_->gain_control()->enable_limiter(
                                                *settings_.use_agc_limiter));
  }
  if (settings_.agc_target_level) {
    RTC_CHECK_EQ(AudioProcessing::kNoError,
                 ap_->gain_control()->set_target_level_dbfs(
                     *settings_.agc_target_level));
  }

  if (settings_.agc_mode) {
    RTC_CHECK_EQ(
        AudioProcessing::kNoError,
        ap_->gain_control()->set_mode(
            static_cast<webrtc::GainControl::Mode>(*settings_.agc_mode)));
  }

  if (settings_.use_drift_compensation) {
    RTC_CHECK_EQ(AudioProcessing::kNoError,
                 ap_->echo_cancellation()->enable_drift_compensation(
                     *settings_.use_drift_compensation));
  }

  if (settings_.aec_suppression_level) {
    RTC_CHECK_EQ(AudioProcessing::kNoError,
                 ap_->echo_cancellation()->set_suppression_level(
                     static_cast<webrtc::EchoCancellation::SuppressionLevel>(
                         *settings_.aec_suppression_level)));
  }

  if (settings_.aecm_routing_mode) {
    RTC_CHECK_EQ(AudioProcessing::kNoError,
                 ap_->echo_control_mobile()->set_routing_mode(
                     static_cast<webrtc::EchoControlMobile::RoutingMode>(
                         *settings_.aecm_routing_mode)));
  }

  if (settings_.use_aecm_comfort_noise) {
    RTC_CHECK_EQ(AudioProcessing::kNoError,
                 ap_->echo_control_mobile()->enable_comfort_noise(
                     *settings_.use_aecm_comfort_noise));
  }

  if (settings_.vad_likelihood) {
    RTC_CHECK_EQ(AudioProcessing::kNoError,
                 ap_->voice_detection()->set_likelihood(
                     static_cast<webrtc::VoiceDetection::Likelihood>(
                         *settings_.vad_likelihood)));
  }
  if (settings_.ns_level) {
    RTC_CHECK_EQ(
        AudioProcessing::kNoError,
        ap_->noise_suppression()->set_level(
            static_cast<NoiseSuppression::Level>(*settings_.ns_level)));
  }

  if (settings_.use_ts) {
    ap_->set_stream_key_pressed(*settings_.use_ts);
  }

  if (settings_.aec_dump_output_filename) {
    size_t kMaxFilenameSize = AudioProcessing::kMaxFilenameSize;
    RTC_CHECK_LE(settings_.aec_dump_output_filename->size(), kMaxFilenameSize);
    RTC_CHECK_EQ(AudioProcessing::kNoError,
                 ap_->StartDebugRecording(
                     settings_.aec_dump_output_filename->c_str(), -1));
  }
}

}  // namespace test
}  // namespace webrtc