| Index: webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
|
| diff --git a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
|
| index fade1449ccdb3886af3dae5154fef3ee6f176bac..be20e11d50bb36b8d1e547d7c9638944ee0711fb 100644
|
| --- a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
|
| +++ b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
|
| @@ -8,10 +8,6 @@
|
| * 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_
|
|
|
| @@ -22,43 +18,18 @@
|
| #include "webrtc/common_audio/lapped_transform.h"
|
| #include "webrtc/common_audio/channel_buffer.h"
|
| #include "webrtc/modules/audio_processing/intelligibility/intelligibility_utils.h"
|
| +#include "webrtc/modules/audio_processing/vad/voice_activity_detector.h"
|
|
|
| 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.
|
| +// Details of the model and algorithm can be found in the original paper:
|
| +// http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=6882788
|
| class IntelligibilityEnhancer {
|
| public:
|
| - struct Config {
|
| - // |var_*| are parameters for the VarianceArray constructor for the
|
| - // clear speech stream.
|
| - // TODO(bercic): the |var_*|, |*_rate| and |gain_limit| parameters should
|
| - // probably go away once fine tuning is done.
|
| - Config()
|
| - : sample_rate_hz(16000),
|
| - num_capture_channels(1),
|
| - num_render_channels(1),
|
| - var_type(intelligibility::VarianceArray::kStepDecaying),
|
| - var_decay_rate(0.9f),
|
| - var_window_size(10),
|
| - analysis_rate(800),
|
| - gain_change_limit(0.1f),
|
| - rho(0.02f) {}
|
| - int sample_rate_hz;
|
| - size_t num_capture_channels;
|
| - size_t num_render_channels;
|
| - intelligibility::VarianceArray::StepType var_type;
|
| - float var_decay_rate;
|
| - size_t var_window_size;
|
| - int analysis_rate;
|
| - float gain_change_limit;
|
| - float rho;
|
| - };
|
| -
|
| - explicit IntelligibilityEnhancer(const Config& config);
|
| - IntelligibilityEnhancer(); // Initialize with default config.
|
| + IntelligibilityEnhancer(int sample_rate_hz, size_t num_render_channels);
|
|
|
| // Sets the capture noise magnitude spectrum estimate.
|
| void SetCaptureNoiseEstimate(std::vector<float> noise);
|
| @@ -90,14 +61,11 @@ class IntelligibilityEnhancer {
|
| FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, TestErbCreation);
|
| FRIEND_TEST_ALL_PREFIXES(IntelligibilityEnhancerTest, TestSolveForGains);
|
|
|
| - // Updates variance computation and analysis with |in_block_|,
|
| + // Updates power computation and analysis with |in_block_|,
|
| // and writes modified speech to |out_block|.
|
| void ProcessClearBlock(const std::complex<float>* in_block,
|
| std::complex<float>* 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);
|
|
|
| @@ -114,29 +82,24 @@ class IntelligibilityEnhancer {
|
| // Negative gains are set to 0. Stores the results in |sols|.
|
| void SolveForGainsGivenLambda(float lambda, size_t start_freq, float* sols);
|
|
|
| + // Returns true if the audio is speech.
|
| + bool IsSpeech(const float* audio);
|
| +
|
| const size_t freqs_; // Num frequencies in frequency domain.
|
| - const size_t window_size_; // Window size in samples; also the block size.
|
| const size_t chunk_length_; // Chunk size in samples.
|
| const size_t bank_size_; // Num ERB filters.
|
| const int sample_rate_hz_;
|
| - const int erb_resolution_;
|
| - const size_t num_capture_channels_;
|
| const size_t num_render_channels_;
|
| - const int analysis_rate_; // Num blocks before gains recalculated.
|
| -
|
| - const bool active_; // Whether render gains are being updated.
|
| - // TODO(ekm): Add logic for updating |active_|.
|
|
|
| - intelligibility::VarianceArray clear_variance_;
|
| - std::vector<float> noise_power_;
|
| - rtc::scoped_ptr<float[]> filtered_clear_var_;
|
| - rtc::scoped_ptr<float[]> filtered_noise_var_;
|
| + intelligibility::PowerEstimator clear_power_estimator_;
|
| + rtc::scoped_ptr<intelligibility::PowerEstimator> noise_power_estimator_;
|
| + rtc::scoped_ptr<float[]> filtered_clear_pow_;
|
| + rtc::scoped_ptr<float[]> filtered_noise_pow_;
|
| rtc::scoped_ptr<float[]> center_freqs_;
|
| std::vector<std::vector<float>> capture_filter_bank_;
|
| std::vector<std::vector<float>> render_filter_bank_;
|
| size_t start_freq_;
|
| - rtc::scoped_ptr<float[]> rho_; // Production and interpretation SNR.
|
| - // for each ERB band.
|
| +
|
| rtc::scoped_ptr<float[]> gains_eq_; // Pre-filter modified gains.
|
| intelligibility::GainApplier gain_applier_;
|
|
|
| @@ -144,11 +107,13 @@ class IntelligibilityEnhancer {
|
| // the original input array with modifications.
|
| ChannelBuffer<float> temp_render_out_buffer_;
|
|
|
| - rtc::scoped_ptr<float[]> kbd_window_;
|
| TransformCallback render_callback_;
|
| rtc::scoped_ptr<LappedTransform> render_mangler_;
|
| - int block_count_;
|
| - int analysis_step_;
|
| +
|
| + VoiceActivityDetector vad_;
|
| + std::vector<int16_t> audio_s16_;
|
| + size_t chunks_since_voice_;
|
| + bool is_speech_;
|
| };
|
|
|
| } // namespace webrtc
|
|
|