push_sinc_resampler.h (3361B)
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 COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_ 12 #define COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_ 13 14 #include <stddef.h> 15 #include <stdint.h> 16 17 #include <memory> 18 19 #include "api/audio/audio_view.h" 20 #include "common_audio/resampler/sinc_resampler.h" 21 22 namespace webrtc { 23 24 // A thin wrapper over SincResampler to provide a push-based interface as 25 // required by WebRTC. SincResampler uses a pull-based interface, and will 26 // use SincResamplerCallback::Run() to request data upon a call to Resample(). 27 // These Run() calls will happen on the same thread Resample() is called on. 28 class PushSincResampler : public SincResamplerCallback { 29 public: 30 // Provide the size of the source and destination blocks in samples. These 31 // must correspond to the same time duration (typically 10 ms) as the sample 32 // ratio is inferred from them. 33 PushSincResampler(size_t source_frames, size_t destination_frames); 34 ~PushSincResampler() override; 35 36 PushSincResampler(const PushSincResampler&) = delete; 37 PushSincResampler& operator=(const PushSincResampler&) = delete; 38 39 // Perform the resampling. `source_frames` must always equal the 40 // `source_frames` provided at construction. `destination_capacity` must be 41 // at least as large as `destination_frames`. Returns the number of samples 42 // provided in destination (for convenience, since this will always be equal 43 // to `destination_frames`). 44 template <typename S, typename D> 45 size_t Resample(const MonoView<S>& source, const MonoView<D>& destination) { 46 return Resample(&source[0], SamplesPerChannel(source), &destination[0], 47 SamplesPerChannel(destination)); 48 } 49 50 size_t Resample(const int16_t* source, 51 size_t source_frames, 52 int16_t* destination, 53 size_t destination_capacity); 54 size_t Resample(const float* source, 55 size_t source_frames, 56 float* destination, 57 size_t destination_capacity); 58 59 // Delay due to the filter kernel. Essentially, the time after which an input 60 // sample will appear in the resampled output. 61 static float AlgorithmicDelaySeconds(int source_rate_hz) { 62 return 1.f / source_rate_hz * SincResampler::kKernelSize / 2; 63 } 64 65 protected: 66 // Implements SincResamplerCallback. 67 void Run(size_t frames, float* destination) override; 68 69 private: 70 friend class PushSincResamplerTest; 71 SincResampler* get_resampler_for_testing() { return resampler_.get(); } 72 73 std::unique_ptr<SincResampler> resampler_; 74 std::unique_ptr<float[]> float_buffer_; 75 const float* source_ptr_; 76 const int16_t* source_ptr_int_; 77 const size_t destination_frames_; 78 79 // True on the first call to Resample(), to prime the SincResampler buffer. 80 bool first_pass_; 81 82 // Used to assert we are only requested for as much data as is available. 83 size_t source_available_; 84 }; 85 86 } // namespace webrtc 87 88 #endif // COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_