/* * Copyright (c) 2014 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. */ // // Specifies core class for intelligbility enhancement. // #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_ #define WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_ #include #include #include "webrtc/base/scoped_ptr.h" #include "webrtc/common_audio/lapped_transform.h" #include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h" struct WebRtcVadInst; typedef struct WebRtcVadInst VadInst; namespace webrtc { // Speech intelligibility enhancement module. Reads render and capture // audio streams and modifies the render stream with a set of gains per // frequency bin to enhance speech against the noise background. // Note: assumes speech and noise streams are already separated. class IntelligibilityEnhancer { public: // Construct a new instance with the given filter bank resolution, // sampling rate, number of channels and analysis rates. // |analysis_rate| sets the number of input blocks (containing speech!) // to elapse before a new gain computation is made. |variance_rate| specifies // the number of gain recomputations after which the variances are reset. // |cv_*| are parameters for the VarianceArray constructor for the // clear speech stream. // TODO(bercic): the |cv_*|, |*_rate| and |gain_limit| parameters should // probably go away once fine tuning is done. They override the internal // constants in the class (kGainChangeLimit, kAnalyzeRate, kVarianceRate). IntelligibilityEnhancer(int erb_resolution, int sample_rate_hz, int channels, int cv_type, float cv_alpha, int cv_win, int analysis_rate, int variance_rate, float gain_limit); ~IntelligibilityEnhancer(); // Reads and processes chunk of noise stream in time domain. void ProcessCaptureAudio(float* const* audio); // Reads chunk of speech in time domain and updates with modified signal. void ProcessRenderAudio(float* const* audio); private: enum AudioSource { kRenderStream = 0, // Clear speech stream. kCaptureStream, // Noise stream. }; // Provides access point to the frequency domain. class TransformCallback : public LappedTransform::Callback { public: TransformCallback(IntelligibilityEnhancer* parent, AudioSource source); // All in frequency domain, receives input |in_block|, applies // intelligibility enhancement, and writes result to |out_block|. virtual void ProcessAudioBlock(const std::complex* const* in_block, int in_channels, int frames, int out_channels, std::complex* const* out_block); private: IntelligibilityEnhancer* parent_; AudioSource source_; }; friend class TransformCallback; FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, TestErbCreation); FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, TestSolveForGains); // Sends streams to ProcessClearBlock or ProcessNoiseBlock based on source. void DispatchAudio(AudioSource source, const std::complex* in_block, std::complex* out_block); // Updates variance computation and analysis with |in_block_|, // and writes modified speech to |out_block|. void ProcessClearBlock(const std::complex* in_block, std::complex* out_block); // Computes and sets modified gains. void AnalyzeClearBlock(float power_target); // Bisection search for optimal |lambda|. void SolveForLambda(float power_target, float power_bot, float power_top); // Transforms freq gains to ERB gains. void UpdateErbGains(); // Updates variance calculation for noise input with |in_block|. void ProcessNoiseBlock(const std::complex* in_block, std::complex* out_block); // Returns number of ERB filters. static int GetBankSize(int sample_rate, int erb_resolution); // Initializes ERB filterbank. void CreateErbBank(); // Analytically solves quadratic for optimal gains given |lambda|. // Negative gains are set to 0. Stores the results in |sols|. void SolveForGainsGivenLambda(float lambda, int start_freq, float* sols); // Computes variance across ERB filters from freq variance |var|. // Stores in |result|. void FilterVariance(const float* var, float* result); // Returns dot product of vectors specified by size |length| arrays |a|,|b|. static float DotProduct(const float* a, const float* b, int length); const int freqs_; // Num frequencies in frequency domain. const int window_size_; // Window size in samples; also the block size. const int chunk_length_; // Chunk size in samples. const int bank_size_; // Num ERB filters. const int sample_rate_hz_; const int erb_resolution_; const int channels_; // Num channels. const int analysis_rate_; // Num blocks before gains recalculated. const int variance_rate_; // Num recalculations before history is cleared. intelligibility::VarianceArray clear_variance_; intelligibility::VarianceArray noise_variance_; rtc::scoped_ptr filtered_clear_var_; rtc::scoped_ptr filtered_noise_var_; std::vector> filter_bank_; rtc::scoped_ptr center_freqs_; int start_freq_; rtc::scoped_ptr rho_; // Production and interpretation SNR. // for each ERB band. rtc::scoped_ptr gains_eq_; // Pre-filter modified gains. intelligibility::GainApplier gain_applier_; // Destination buffer used to reassemble blocked chunks before overwriting // the original input array with modifications. // TODO(ekmeyerson): Switch to using ChannelBuffer. float** temp_out_buffer_; rtc::scoped_ptr input_audio_; rtc::scoped_ptr kbd_window_; TransformCallback render_callback_; TransformCallback capture_callback_; rtc::scoped_ptr render_mangler_; rtc::scoped_ptr capture_mangler_; int block_count_; int analysis_step_; // TODO(bercic): Quick stopgap measure for voice detection in the clear // and noise streams. // Note: VAD currently does not affect anything in IntelligibilityEnhancer. VadInst* vad_high_; VadInst* vad_low_; rtc::scoped_ptr vad_tmp_buffer_; bool has_voice_low_; // Whether voice detected in speech stream. }; } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_PROCESSING_INTELLIGIBILITY_INTELLIGIBILITY_ENHANCER_H_