| /* |
| * 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_TIME_STRETCH_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_ |
| |
| #include <assert.h> |
| #include <string.h> // memset, size_t |
| |
| #include "webrtc/base/constructormagic.h" |
| #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| |
| // Forward declarations. |
| class BackgroundNoise; |
| |
| // This is the base class for Accelerate and PreemptiveExpand. This class |
| // cannot be instantiated, but must be used through either of the derived |
| // classes. |
| class TimeStretch { |
| public: |
| enum ReturnCodes { |
| kSuccess = 0, |
| kSuccessLowEnergy = 1, |
| kNoStretch = 2, |
| kError = -1 |
| }; |
| |
| TimeStretch(int sample_rate_hz, size_t num_channels, |
| const BackgroundNoise& background_noise) |
| : sample_rate_hz_(sample_rate_hz), |
| fs_mult_(sample_rate_hz / 8000), |
| num_channels_(num_channels), |
| master_channel_(0), // First channel is master. |
| background_noise_(background_noise), |
| max_input_value_(0) { |
| assert(sample_rate_hz_ == 8000 || |
| sample_rate_hz_ == 16000 || |
| sample_rate_hz_ == 32000 || |
| sample_rate_hz_ == 48000); |
| assert(num_channels_ > 0); |
| assert(master_channel_ < num_channels_); |
| memset(auto_correlation_, 0, sizeof(auto_correlation_)); |
| } |
| |
| virtual ~TimeStretch() {} |
| |
| // This method performs the processing common to both Accelerate and |
| // PreemptiveExpand. |
| ReturnCodes Process(const int16_t* input, |
| size_t input_len, |
| bool fast_mode, |
| 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. This method must be |
| // implemented by the sub-classes. |
| virtual void SetParametersForPassiveSpeech(size_t input_length, |
| int16_t* best_correlation, |
| size_t* peak_index) const = 0; |
| |
| // Checks the criteria for performing the time-stretching operation and, |
| // if possible, performs the time-stretching. This method must be implemented |
| // by the sub-classes. |
| virtual 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 = 0; |
| |
| static const size_t kCorrelationLen = 50; |
| static const size_t kLogCorrelationLen = 6; // >= log2(kCorrelationLen). |
| static const size_t kMinLag = 10; |
| static const size_t kMaxLag = 60; |
| static const size_t kDownsampledLen = kCorrelationLen + kMaxLag; |
| static const int kCorrelationThreshold = 14746; // 0.9 in Q14. |
| |
| const int sample_rate_hz_; |
| const int fs_mult_; // Sample rate multiplier = sample_rate_hz_ / 8000. |
| const size_t num_channels_; |
| const size_t master_channel_; |
| const BackgroundNoise& background_noise_; |
| int16_t max_input_value_; |
| int16_t downsampled_input_[kDownsampledLen]; |
| // Adding 1 to the size of |auto_correlation_| because of how it is used |
| // by the peak-detection algorithm. |
| int16_t auto_correlation_[kCorrelationLen + 1]; |
| |
| private: |
| // Calculates the auto-correlation of |downsampled_input_| and writes the |
| // result to |auto_correlation_|. |
| void AutoCorrelation(); |
| |
| // Performs a simple voice-activity detection based on the input parameters. |
| bool SpeechDetection(int32_t vec1_energy, int32_t vec2_energy, |
| size_t peak_index, int scaling) const; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(TimeStretch); |
| }; |
| |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TIME_STRETCH_H_ |