| /* |
| * Copyright (c) 2012 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_ |
| |
| #include <assert.h> |
| |
| #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" |
| #include "webrtc/modules/audio_coding/neteq/time_stretch.h" |
| #include "webrtc/rtc_base/constructormagic.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| |
| // Forward declarations. |
| class BackgroundNoise; |
| |
| // This class implements the Accelerate operation. Most of the work is done |
| // in the base class TimeStretch, which is shared with the PreemptiveExpand |
| // operation. In the Accelerate class, the operations that are specific to |
| // Accelerate are implemented. |
| class Accelerate : public TimeStretch { |
| public: |
| Accelerate(int sample_rate_hz, size_t num_channels, |
| const BackgroundNoise& background_noise) |
| : TimeStretch(sample_rate_hz, num_channels, background_noise) { |
| } |
| |
| // This method performs the actual Accelerate operation. The samples are |
| // read from |input|, of length |input_length| elements, and are written to |
| // |output|. The number of samples removed through time-stretching is |
| // is provided in the output |length_change_samples|. The method returns |
| // the outcome of the operation as an enumerator value. If |fast_accelerate| |
| // is true, the algorithm will relax the requirements on finding strong |
| // correlations, and may remove multiple pitch periods if possible. |
| ReturnCodes Process(const int16_t* input, |
| size_t input_length, |
| bool fast_accelerate, |
| AudioMultiVector* output, |
| size_t* length_change_samples); |
| |
| protected: |
| // Sets the parameters |best_correlation| and |peak_index| to suitable |
| // values when the signal contains no active speech. |
| void SetParametersForPassiveSpeech(size_t len, |
| int16_t* best_correlation, |
| size_t* peak_index) const override; |
| |
| // Checks the criteria for performing the time-stretching operation and, |
| // if possible, performs the time-stretching. |
| ReturnCodes CheckCriteriaAndStretch(const int16_t* input, |
| size_t input_length, |
| size_t peak_index, |
| int16_t best_correlation, |
| bool active_speech, |
| bool fast_mode, |
| AudioMultiVector* output) const override; |
| |
| private: |
| RTC_DISALLOW_COPY_AND_ASSIGN(Accelerate); |
| }; |
| |
| struct AccelerateFactory { |
| AccelerateFactory() {} |
| virtual ~AccelerateFactory() {} |
| |
| virtual Accelerate* Create(int sample_rate_hz, |
| size_t num_channels, |
| const BackgroundNoise& background_noise) const; |
| }; |
| |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_ACCELERATE_H_ |