1 /* 2 * Copyright (c) 2013 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 #ifndef WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_ 12 #define WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_ 13 14 #include "webrtc/base/constructormagic.h" 15 #include "webrtc/common_audio/resampler/sinc_resampler.h" 16 #include "webrtc/system_wrappers/interface/scoped_ptr.h" 17 #include "webrtc/typedefs.h" 18 19 namespace webrtc { 20 21 // A thin wrapper over SincResampler to provide a push-based interface as 22 // required by WebRTC. 23 class PushSincResampler : public SincResamplerCallback { 24 public: 25 // Provide the size of the source and destination blocks in samples. These 26 // must correspond to the same time duration (typically 10 ms) as the sample 27 // ratio is inferred from them. 28 PushSincResampler(int source_frames, int destination_frames); 29 virtual ~PushSincResampler(); 30 31 // Perform the resampling. |source_frames| must always equal the 32 // |source_frames| provided at construction. |destination_capacity| must be 33 // at least as large as |destination_frames|. Returns the number of samples 34 // provided in destination (for convenience, since this will always be equal 35 // to |destination_frames|). 36 int Resample(const int16_t* source, int source_frames, 37 int16_t* destination, int destination_capacity); 38 int Resample(const float* source, 39 int source_frames, 40 float* destination, 41 int destination_capacity); 42 43 // Implements SincResamplerCallback. 44 virtual void Run(int frames, float* destination) OVERRIDE; 45 get_resampler_for_testing()46 SincResampler* get_resampler_for_testing() { return resampler_.get(); } AlgorithmicDelaySeconds(int source_rate_hz)47 static float AlgorithmicDelaySeconds(int source_rate_hz) { 48 return 1.f / source_rate_hz * SincResampler::kKernelSize / 2; 49 } 50 51 private: 52 scoped_ptr<SincResampler> resampler_; 53 scoped_ptr<float[]> float_buffer_; 54 const float* source_ptr_; 55 const int16_t* source_ptr_int_; 56 const int destination_frames_; 57 58 // True on the first call to Resample(), to prime the SincResampler buffer. 59 bool first_pass_; 60 61 // Used to assert we are only requested for as much data as is available. 62 int source_available_; 63 64 DISALLOW_COPY_AND_ASSIGN(PushSincResampler); 65 }; 66 67 } // namespace webrtc 68 69 #endif // WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_ 70