text: /*
   *  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 <assert.h>
  #include <string.h>
  
  #include "common_audio/resampler/include/resampler.h"
  #include BOSS_WEBRTC_U_rtc_base__logging_h //original-code:"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) {
        assert(false);
        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
  