/* * 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 COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_ #define COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_ #include #include #include "api/audio/audio_view.h" namespace webrtc { class PushSincResampler; // Wraps PushSincResampler to provide stereo support. // Note: This implementation assumes 10ms buffer sizes throughout. template class PushResampler final { public: PushResampler(); PushResampler(size_t src_samples_per_channel, size_t dst_samples_per_channel, size_t num_channels); ~PushResampler(); // Returns the total number of samples provided in destination (e.g. 32 kHz, // 2 channel audio gives 640 samples). int Resample(InterleavedView src, InterleavedView dst); // For when a deinterleaved/mono channel already exists and we can skip the // deinterleaved operation. int Resample(MonoView src, MonoView dst); private: // Ensures that source and destination buffers for deinterleaving are // correctly configured prior to resampling that requires deinterleaving. void EnsureInitialized(size_t src_samples_per_channel, size_t dst_samples_per_channel, size_t num_channels); // Buffers used for when a deinterleaving step is necessary. std::unique_ptr source_; std::unique_ptr destination_; DeinterleavedView source_view_; DeinterleavedView destination_view_; std::vector> resamplers_; }; } // namespace webrtc #endif // COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_