Chromium Code Reviews| 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..d3b7af0abc19b50c61df4d204de47aa43b4d97db 100644 |
| --- a/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h |
| +++ b/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h |
| @@ -20,11 +20,9 @@ |
| #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/intelligibility/intelligibility_utils.h" |
| -struct WebRtcVadInst; |
| -typedef struct WebRtcVadInst VadInst; |
| - |
| namespace webrtc { |
| // Speech intelligibility enhancement module. Reads render and capture |
| @@ -33,32 +31,45 @@ 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(); |
| + 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; |
| + 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; |
| + }; |
| + |
| + explicit IntelligibilityEnhancer(const Config& config); |
| + IntelligibilityEnhancer(); // Initialize with default config. |
| // Reads and processes chunk of noise stream in time domain. |
| - void ProcessCaptureAudio(float* const* audio); |
| + 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); |
| + void ProcessRenderAudio(float* const* audio, |
| + int sample_rate_hz, |
| + int num_channels); |
| + bool active() const; |
| private: |
| enum AudioSource { |
| @@ -127,15 +138,18 @@ class IntelligibilityEnhancer { |
| // 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 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 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. |
| + |
| + bool active_; // Whether render gains are being updated. |
|
Andrew MacDonald
2015/07/29 03:52:27
Make this const for now.
ekm
2015/07/29 23:35:06
Done.
|
| + // TODO(ekm): Add logic for updating |active_|. |
| intelligibility::VarianceArray clear_variance_; |
| intelligibility::VarianceArray noise_variance_; |
| @@ -149,12 +163,11 @@ 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_; |
| @@ -162,14 +175,6 @@ class IntelligibilityEnhancer { |
| rtc::scoped_ptr<LappedTransform> 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<int16_t[]> vad_tmp_buffer_; |
| - bool has_voice_low_; // Whether voice detected in speech stream. |
| }; |
| } // namespace webrtc |