| /* |
| * Copyright (c) 2014 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_coding/neteq/tools/resample_input_audio_file.h" |
| |
| #include "webrtc/base/checks.h" |
| #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| bool ResampleInputAudioFile::Read(size_t samples, |
| int output_rate_hz, |
| int16_t* destination) { |
| const size_t samples_to_read = samples * file_rate_hz_ / output_rate_hz; |
| CHECK_EQ(samples_to_read * output_rate_hz, samples * file_rate_hz_) |
| << "Frame size and sample rates don't add up to an integer."; |
| scoped_ptr<int16_t[]> temp_destination(new int16_t[samples_to_read]); |
| if (!InputAudioFile::Read(samples_to_read, temp_destination.get())) |
| return false; |
| resampler_.ResetIfNeeded( |
| file_rate_hz_, output_rate_hz, kResamplerSynchronous); |
| int output_length = 0; |
| CHECK_EQ(resampler_.Push(temp_destination.get(), |
| static_cast<int>(samples_to_read), |
| destination, |
| static_cast<int>(samples), |
| output_length), |
| 0); |
| CHECK_EQ(static_cast<int>(samples), output_length); |
| return true; |
| } |
| |
| } // namespace test |
| } // namespace webrtc |
