/*
 *  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.
 */

#include "modules/audio_coding/acm2/acm_resampler.h"

#include <string.h>

#include "rtc_base/logging.h"

namespace webrtc {
namespace acm2 {

ACMResampler::ACMResampler() {}

ACMResampler::~ACMResampler() {}

int ACMResampler::Resample10Msec(const int16_t* in_audio,
                                 int in_freq_hz,
                                 int out_freq_hz,
                                 size_t num_audio_channels,
                                 size_t out_capacity_samples,
                                 int16_t* out_audio) {
  size_t in_length = in_freq_hz * num_audio_channels / 100;
  if (in_freq_hz == out_freq_hz) {
    if (out_capacity_samples < in_length) {
      RTC_NOTREACHED();
      return -1;
    }
    memcpy(out_audio, in_audio, in_length * sizeof(int16_t));
    return static_cast<int>(in_length / num_audio_channels);
  }

  if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
                                    num_audio_channels) != 0) {
    RTC_LOG(LS_ERROR) << "InitializeIfNeeded(" << in_freq_hz << ", "
                      << out_freq_hz << ", " << num_audio_channels
                      << ") failed.";
    return -1;
  }

  int out_length =
      resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
  if (out_length == -1) {
    RTC_LOG(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", "
                      << out_audio << ", " << out_capacity_samples
                      << ") failed.";
    return -1;
  }

  return static_cast<int>(out_length / num_audio_channels);
}

}  // namespace acm2
}  // namespace webrtc