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 df47de597885ed61d9dd9a824d2c6505c1be99a4..12c7e732b99d9fd8568c7d108717a9d260acf5e6 100644 |
--- a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h |
+++ b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h |
@@ -20,11 +20,10 @@ |
#include "webrtc/base/scoped_ptr.h" |
#include "webrtc/common_audio/lapped_transform.h" |
+#include "webrtc/common_audio/channel_buffer.h" |
+#include "webrtc/modules/audio_processing/vad/voice_activity_detector.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 |
@@ -33,32 +32,64 @@ namespace webrtc { |
// 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); |
+ 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), |
+ capture_vad_thresh(1.f), |
+ render_vad_thresh(0.f), |
+ activate_snr_thresh(0.f), |
+ deactivate_snr_thresh(100000.f) {} |
+ int sample_rate_hz; |
+ int num_capture_channels; |
+ int num_render_channels; |
+ intelligibility::VarianceArray::StepType var_type; |
+ float var_decay_rate; |
+ int var_window_size; |
+ int analysis_rate; |
+ float gain_change_limit; |
+ float rho; |
+ float capture_vad_thresh; |
+ float render_vad_thresh; |
+ float activate_snr_thresh; |
+ float deactivate_snr_thresh; |
+ }; |
+ |
+ explicit IntelligibilityEnhancer(const Config& config); |
+ IntelligibilityEnhancer(); // Initialize with default config. |
+ |
+ // Reads and processes chunk of noise stream in time domain. Only updates |
+ // noise estimate when |voice_probability| below a threshold. Uses internal |
+ // VAD when |voice_probability| not provided. |
+ void AnalyzeCaptureAudio(float* const* audio, |
+ int sample_rate_hz, |
+ int num_channels, |
+ float voice_probability); |
+ void AnalyzeCaptureAudio(float* const* audio, |
+ int sample_rate_hz, |
+ int num_channels); |
// Reads chunk of speech in time domain and updates with modified signal. |
- void ProcessRenderAudio(float* const* audio); |
+ // Only updates speech estimate when |voice_probability| above a threshold. |
+ // Uses internal VAD when |voice_probability| not provided. |
+ void ProcessRenderAudio(float* const* audio, |
+ int sample_rate_hz, |
+ int num_channels, |
+ float voice_probability); |
+ void ProcessRenderAudio(float* const* audio, |
+ int sample_rate_hz, |
+ int num_channels); |
private: |
enum AudioSource { |
@@ -124,6 +155,12 @@ class IntelligibilityEnhancer { |
// Stores in |result|. |
void FilterVariance(const float* var, float* result); |
+ // Returns ratio of total variance of clear to noise. |
+ float SNR(); |
Andrew MacDonald
2015/07/24 23:50:40
Sorry to do this, but could you please move these
turaj
2015/07/27 20:01:05
This method is const.
ekm
2015/07/29 00:37:19
Done.
|
+ |
+ // Updates |active_| based on SNR. |
+ void UpdateActivity(); |
+ |
// Returns dot product of vectors specified by size |length| arrays |a|,|b|. |
static float DotProduct(const float* a, const float* b, int length); |
@@ -133,9 +170,16 @@ class IntelligibilityEnhancer { |
const int bank_size_; // Num ERB filters. |
const int sample_rate_hz_; |
const int erb_resolution_; |
- const int channels_; // Num channels. |
+ const int num_capture_channels_; |
+ const int num_render_channels_; |
const int analysis_rate_; // Num blocks before gains recalculated. |
- const int variance_rate_; // Num recalculations before history is cleared. |
+ const float capture_vad_thresh_; // Threshold for updating noise estimate. |
+ const float render_vad_thresh_; // Threshold for updating speech estimate. |
+ const float activate_snr_thresh_; // Threshold for activating gain updates. |
+ const float deactivate_snr_thresh_; // Threshold for deactivating. |
+ |
+ bool active_; // Whether render gains are being updated. |
+ bool deactivating_; // True when we are smoothing enhancer off. |
intelligibility::VarianceArray clear_variance_; |
intelligibility::VarianceArray noise_variance_; |
@@ -149,12 +193,11 @@ class IntelligibilityEnhancer { |
rtc::scoped_ptr<float[]> gains_eq_; // Pre-filter modified gains. |
intelligibility::GainApplier gain_applier_; |
- // Destination buffer used to reassemble blocked chunks before overwriting |
+ // Destination buffers used to reassemble blocked chunks before overwriting |
// the original input array with modifications. |
- // TODO(ekmeyerson): Switch to using ChannelBuffer. |
- float** temp_out_buffer_; |
+ ChannelBuffer<float> temp_render_out_buffer_; |
+ ChannelBuffer<float> temp_capture_out_buffer_; |
- rtc::scoped_ptr<float* []> input_audio_; |
rtc::scoped_ptr<float[]> kbd_window_; |
TransformCallback render_callback_; |
TransformCallback capture_callback_; |
@@ -163,13 +206,13 @@ class IntelligibilityEnhancer { |
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_; |
+ VoiceActivityDetector capture_vad_; |
+ VoiceActivityDetector render_vad_; |
+ float capture_voice_probability_; |
+ float render_voice_probability_; |
+ bool using_capture_vad_; |
+ bool using_render_vad_; |
rtc::scoped_ptr<int16_t[]> vad_tmp_buffer_; |
- bool has_voice_low_; // Whether voice detected in speech stream. |
}; |
} // namespace webrtc |