| /* |
| * 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 |