/*
 *  Copyright (c) 2015 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/base/checks.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"

namespace webrtc {

RawFile::RawFile(const std::string& filename)
    : file_handle_(fopen(filename.c_str(), "wb")) {}

RawFile::~RawFile() {
  fclose(file_handle_);
}

void RawFile::WriteSamples(const int16_t* samples, size_t num_samples) {
#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
#error "Need to convert samples to little-endian when writing to PCM file"
#endif
  fwrite(samples, sizeof(*samples), num_samples, file_handle_);
}

void RawFile::WriteSamples(const float* samples, size_t num_samples) {
  fwrite(samples, sizeof(*samples), num_samples, file_handle_);
}

void WriteIntData(const int16_t* data,
                  size_t length,
                  WavWriter* wav_file,
                  RawFile* raw_file) {
  if (wav_file) {
    wav_file->WriteSamples(data, length);
  }
  if (raw_file) {
    raw_file->WriteSamples(data, length);
  }
}

void WriteFloatData(const float* const* data,
                    int samples_per_channel,
                    int num_channels,
                    WavWriter* wav_file,
                    RawFile* raw_file) {
  size_t length = num_channels * samples_per_channel;
  rtc::scoped_ptr<float[]> buffer(new float[length]);
  Interleave(data, samples_per_channel, num_channels, buffer.get());
  if (raw_file) {
    raw_file->WriteSamples(buffer.get(), length);
  }
  // TODO(aluebs): Use ScaleToInt16Range() from audio_util
  for (size_t i = 0; i < length; ++i) {
    buffer[i] = buffer[i] > 0 ?
                buffer[i] * std::numeric_limits<int16_t>::max() :
                -buffer[i] * std::numeric_limits<int16_t>::min();
  }
  if (wav_file) {
    wav_file->WriteSamples(buffer.get(), length);
  }
}

FILE* OpenFile(const std::string& filename, const char* mode) {
  FILE* file = fopen(filename.c_str(), mode);
  if (!file) {
    printf("Unable to open file %s\n", filename.c_str());
    exit(1);
  }
  return file;
}

int SamplesFromRate(int rate) {
  return AudioProcessing::kChunkSizeMs * rate / 1000;
}

void SetFrameSampleRate(AudioFrame* frame,
                        int sample_rate_hz) {
  frame->sample_rate_hz_ = sample_rate_hz;
  frame->samples_per_channel_ = AudioProcessing::kChunkSizeMs *
      sample_rate_hz / 1000;
}

AudioProcessing::ChannelLayout LayoutFromChannels(int num_channels) {
  switch (num_channels) {
    case 1:
      return AudioProcessing::kMono;
    case 2:
      return AudioProcessing::kStereo;
    default:
      assert(false);
      return AudioProcessing::kMono;
  }
}

std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
                                      size_t num_mics) {
  const std::vector<float> values = ParseList<float>(mic_positions);
  RTC_CHECK_EQ(values.size(), 3 * num_mics)
      << "Could not parse mic_positions or incorrect number of points.";

  std::vector<Point> result;
  result.reserve(num_mics);
  for (size_t i = 0; i < values.size(); i += 3) {
    double x = values[i + 0];
    double y = values[i + 1];
    double z = values[i + 2];
    result.push_back(Point(x, y, z));
  }

  return result;
}


}  // namespace webrtc