push_resampler.h (1944B)
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_INCLUDE_PUSH_RESAMPLER_H_ 12 #define COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_ 13 14 #include <cstddef> 15 #include <memory> 16 #include <vector> 17 18 #include "api/audio/audio_view.h" 19 20 namespace webrtc { 21 22 class PushSincResampler; 23 24 // Wraps PushSincResampler to provide stereo support. 25 // Note: This implementation assumes 10ms buffer sizes throughout. 26 template <typename T> 27 class PushResampler final { 28 public: 29 PushResampler(); 30 PushResampler(size_t src_samples_per_channel, 31 size_t dst_samples_per_channel, 32 size_t num_channels); 33 ~PushResampler(); 34 35 void Resample(InterleavedView<const T> src, InterleavedView<T> dst); 36 // For when a deinterleaved (or mono) channel already exists and we can skip 37 // the deinterleaved operation. 38 void Resample(MonoView<const T> src, MonoView<T> dst); 39 40 private: 41 // Ensures that source and destination buffers for deinterleaving are 42 // correctly configured prior to resampling that requires deinterleaving. 43 void EnsureInitialized(size_t src_samples_per_channel, 44 size_t dst_samples_per_channel, 45 size_t num_channels); 46 47 // Buffers used for when a deinterleaving step is necessary. 48 std::unique_ptr<T[]> source_; 49 std::unique_ptr<T[]> destination_; 50 DeinterleavedView<T> source_view_; 51 DeinterleavedView<T> destination_view_; 52 53 std::vector<std::unique_ptr<PushSincResampler>> resamplers_; 54 }; 55 } // namespace webrtc 56 57 #endif // COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_