| /* |
| * Copyright (c) 2013 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. |
| */ |
| |
| #ifndef WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_ |
| #define WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_ |
| |
| #include <memory> |
| |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| |
| class PushSincResampler; |
| |
| // Wraps PushSincResampler to provide stereo support. |
| // TODO(ajm): add support for an arbitrary number of channels. |
| template <typename T> |
| class PushResampler { |
| public: |
| PushResampler(); |
| virtual ~PushResampler(); |
| |
| // Must be called whenever the parameters change. Free to be called at any |
| // time as it is a no-op if parameters have not changed since the last call. |
| int InitializeIfNeeded(int src_sample_rate_hz, int dst_sample_rate_hz, |
| size_t num_channels); |
| |
| // Returns the total number of samples provided in destination (e.g. 32 kHz, |
| // 2 channel audio gives 640 samples). |
| int Resample(const T* src, size_t src_length, T* dst, size_t dst_capacity); |
| |
| private: |
| std::unique_ptr<PushSincResampler> sinc_resampler_; |
| std::unique_ptr<PushSincResampler> sinc_resampler_right_; |
| int src_sample_rate_hz_; |
| int dst_sample_rate_hz_; |
| size_t num_channels_; |
| std::unique_ptr<T[]> src_left_; |
| std::unique_ptr<T[]> src_right_; |
| std::unique_ptr<T[]> dst_left_; |
| std::unique_ptr<T[]> dst_right_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_ |