acm_resampler.h (1393B)
1 /* 2 * Copyright (c) 2012 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 MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_ 12 #define MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_ 13 14 #include <stddef.h> 15 #include <stdint.h> 16 17 #include <array> 18 19 #include "api/audio/audio_frame.h" 20 #include "common_audio/resampler/include/push_resampler.h" 21 22 namespace webrtc { 23 namespace acm2 { 24 25 // Helper class to perform resampling if needed, meant to be used after 26 // receiving the audio_frame from NetEq. Provides reasonably glitch free 27 // transitions between different output sample rates from NetEq. 28 class ResamplerHelper { 29 public: 30 ResamplerHelper(); 31 32 // Resamples audio_frame if it is not already in desired_sample_rate_hz. 33 bool MaybeResample(int desired_sample_rate_hz, AudioFrame* audio_frame); 34 35 private: 36 PushResampler<int16_t> resampler_; 37 bool resampled_last_output_frame_ = true; 38 std::array<int16_t, AudioFrame::kMaxDataSizeSamples> last_audio_buffer_; 39 }; 40 41 } // namespace acm2 42 } // namespace webrtc 43 44 #endif // MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_