1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
12
13 #include "webrtc/base/checks.h"
14 #include "webrtc/base/scoped_ptr.h"
15
16 namespace webrtc {
17 namespace test {
18
Read(size_t samples,int output_rate_hz,int16_t * destination)19 bool ResampleInputAudioFile::Read(size_t samples,
20 int output_rate_hz,
21 int16_t* destination) {
22 const size_t samples_to_read = samples * file_rate_hz_ / output_rate_hz;
23 RTC_CHECK_EQ(samples_to_read * output_rate_hz, samples * file_rate_hz_)
24 << "Frame size and sample rates don't add up to an integer.";
25 rtc::scoped_ptr<int16_t[]> temp_destination(new int16_t[samples_to_read]);
26 if (!InputAudioFile::Read(samples_to_read, temp_destination.get()))
27 return false;
28 resampler_.ResetIfNeeded(file_rate_hz_, output_rate_hz, 1);
29 size_t output_length = 0;
30 RTC_CHECK_EQ(resampler_.Push(temp_destination.get(), samples_to_read,
31 destination, samples, output_length),
32 0);
33 RTC_CHECK_EQ(samples, output_length);
34 return true;
35 }
36
Read(size_t samples,int16_t * destination)37 bool ResampleInputAudioFile::Read(size_t samples, int16_t* destination) {
38 RTC_CHECK_GT(output_rate_hz_, 0) << "Output rate not set.";
39 return Read(samples, output_rate_hz_, destination);
40 }
41
set_output_rate_hz(int rate_hz)42 void ResampleInputAudioFile::set_output_rate_hz(int rate_hz) {
43 output_rate_hz_ = rate_hz;
44 }
45
46 } // namespace test
47 } // namespace webrtc
48