/*
 *  Copyright (c) 2017 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 "modules/audio_processing/aec3/block_delay_buffer.h"

#include <string>

#include "modules/audio_processing/aec3/aec3_common.h"
#include "modules/audio_processing/audio_buffer.h"
#include "rtc_base/strings/string_builder.h"
#include "test/gtest.h"

namespace webrtc {

namespace {

float SampleValue(size_t sample_index) {
  return sample_index % 32768;
}

// Populates the frame with linearly increasing sample values for each band.
void PopulateInputFrame(size_t frame_length,
                        size_t num_bands,
                        size_t first_sample_index,
                        float* const* frame) {
  for (size_t k = 0; k < num_bands; ++k) {
    for (size_t i = 0; i < frame_length; ++i) {
      frame[k][i] = SampleValue(first_sample_index + i);
    }
  }
}

std::string ProduceDebugText(int sample_rate_hz, size_t delay) {
  char log_stream_buffer[8 * 1024];
  rtc::SimpleStringBuilder ss(log_stream_buffer);
  ss << "Sample rate: " << sample_rate_hz;
  ss << ", Delay: " << delay;
  return ss.str();
}

}  // namespace

// Verifies that the correct signal delay is achived.
TEST(BlockDelayBuffer, CorrectDelayApplied) {
  for (size_t delay : {0, 1, 27, 160, 4321, 7021}) {
    for (auto rate : {8000, 16000, 32000, 48000}) {
      SCOPED_TRACE(ProduceDebugText(rate, delay));
      size_t num_bands = NumBandsForRate(rate);
      size_t subband_frame_length = rate == 8000 ? 80 : 160;

      BlockDelayBuffer delay_buffer(num_bands, subband_frame_length, delay);

      static constexpr size_t kNumFramesToProcess = 20;
      for (size_t frame_index = 0; frame_index < kNumFramesToProcess;
           ++frame_index) {
        AudioBuffer audio_buffer(rate, 1, rate, 1, rate, 1);
        if (rate > 16000) {
          audio_buffer.SplitIntoFrequencyBands();
        }
        size_t first_sample_index = frame_index * subband_frame_length;
        PopulateInputFrame(subband_frame_length, num_bands, first_sample_index,
                           &audio_buffer.split_bands(0)[0]);
        delay_buffer.DelaySignal(&audio_buffer);

        for (size_t k = 0; k < num_bands; ++k) {
          size_t sample_index = first_sample_index;
          for (size_t i = 0; i < subband_frame_length; ++i, ++sample_index) {
            if (sample_index < delay) {
              EXPECT_EQ(0.f, audio_buffer.split_bands(0)[k][i]);
            } else {
              EXPECT_EQ(SampleValue(sample_index - delay),
                        audio_buffer.split_bands(0)[k][i]);
            }
          }
        }
      }
    }
  }
}

}  // namespace webrtc