| /* |
| * 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_PUSH_SINC_RESAMPLER_H_ |
| #define WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_ |
| |
| #include "webrtc/common_audio/resampler/sinc_resampler.h" |
| #include "webrtc/system_wrappers/interface/constructor_magic.h" |
| #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| |
| // A thin wrapper over SincResampler to provide a push-based interface as |
| // required by WebRTC. |
| class PushSincResampler : public SincResamplerCallback { |
| public: |
| // Provide the size of the source and destination blocks in samples. These |
| // must correspond to the same time duration (typically 10 ms) as the sample |
| // ratio is inferred from them. |
| PushSincResampler(int src_block_size, int dst_block_size); |
| virtual ~PushSincResampler(); |
| |
| // Perform the resampling. |source_length| must always equal the |
| // |src_block_size| provided at construction. |destination_capacity| must be |
| // at least as large as |dst_block_size|. Returns the number of samples |
| // provided in destination (for convenience, since this will always be equal |
| // to |dst_block_size|). |
| int Resample(const int16_t* source, int source_length, |
| int16_t* destination, int destination_capacity); |
| |
| // Implements SincResamplerCallback. |
| virtual void Run(float* destination, int frames); |
| |
| private: |
| scoped_ptr<SincResampler> resampler_; |
| scoped_array<float> float_buffer_; |
| const int16_t* source_ptr_; |
| const int dst_size_; |
| |
| DISALLOW_COPY_AND_ASSIGN(PushSincResampler); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_ |