| OLD | NEW |
| 1 /* | 1 /* |
| 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. |
| 3 * | 3 * |
| 4 * Use of this source code is governed by a BSD-style license | 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 | 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 | 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ | 9 */ |
| 10 | 10 |
| 11 #ifndef WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_ | 11 #ifndef WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_ |
| 12 #define WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_ | 12 #define WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_ |
| 13 | 13 |
| 14 #include "webrtc/base/scoped_ptr.h" | 14 #include <memory> |
| 15 |
| 15 #include "webrtc/typedefs.h" | 16 #include "webrtc/typedefs.h" |
| 16 | 17 |
| 17 namespace webrtc { | 18 namespace webrtc { |
| 18 | 19 |
| 19 class PushSincResampler; | 20 class PushSincResampler; |
| 20 | 21 |
| 21 // Wraps PushSincResampler to provide stereo support. | 22 // Wraps PushSincResampler to provide stereo support. |
| 22 // TODO(ajm): add support for an arbitrary number of channels. | 23 // TODO(ajm): add support for an arbitrary number of channels. |
| 23 template <typename T> | 24 template <typename T> |
| 24 class PushResampler { | 25 class PushResampler { |
| 25 public: | 26 public: |
| 26 PushResampler(); | 27 PushResampler(); |
| 27 virtual ~PushResampler(); | 28 virtual ~PushResampler(); |
| 28 | 29 |
| 29 // Must be called whenever the parameters change. Free to be called at any | 30 // Must be called whenever the parameters change. Free to be called at any |
| 30 // time as it is a no-op if parameters have not changed since the last call. | 31 // time as it is a no-op if parameters have not changed since the last call. |
| 31 int InitializeIfNeeded(int src_sample_rate_hz, int dst_sample_rate_hz, | 32 int InitializeIfNeeded(int src_sample_rate_hz, int dst_sample_rate_hz, |
| 32 size_t num_channels); | 33 size_t num_channels); |
| 33 | 34 |
| 34 // Returns the total number of samples provided in destination (e.g. 32 kHz, | 35 // Returns the total number of samples provided in destination (e.g. 32 kHz, |
| 35 // 2 channel audio gives 640 samples). | 36 // 2 channel audio gives 640 samples). |
| 36 int Resample(const T* src, size_t src_length, T* dst, size_t dst_capacity); | 37 int Resample(const T* src, size_t src_length, T* dst, size_t dst_capacity); |
| 37 | 38 |
| 38 private: | 39 private: |
| 39 rtc::scoped_ptr<PushSincResampler> sinc_resampler_; | 40 std::unique_ptr<PushSincResampler> sinc_resampler_; |
| 40 rtc::scoped_ptr<PushSincResampler> sinc_resampler_right_; | 41 std::unique_ptr<PushSincResampler> sinc_resampler_right_; |
| 41 int src_sample_rate_hz_; | 42 int src_sample_rate_hz_; |
| 42 int dst_sample_rate_hz_; | 43 int dst_sample_rate_hz_; |
| 43 size_t num_channels_; | 44 size_t num_channels_; |
| 44 rtc::scoped_ptr<T[]> src_left_; | 45 std::unique_ptr<T[]> src_left_; |
| 45 rtc::scoped_ptr<T[]> src_right_; | 46 std::unique_ptr<T[]> src_right_; |
| 46 rtc::scoped_ptr<T[]> dst_left_; | 47 std::unique_ptr<T[]> dst_left_; |
| 47 rtc::scoped_ptr<T[]> dst_right_; | 48 std::unique_ptr<T[]> dst_right_; |
| 48 }; | 49 }; |
| 49 | 50 |
| 50 } // namespace webrtc | 51 } // namespace webrtc |
| 51 | 52 |
| 52 #endif // WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_ | 53 #endif // WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_ |
| OLD | NEW |