Surface the noise estimate of the NS to be used by other components

webrtc/modules/audio_processing/noise_suppression_impl.cc

Index: webrtc/modules/audio_processing/noise_suppression_impl.cc
diff --git a/webrtc/modules/audio_processing/noise_suppression_impl.cc b/webrtc/modules/audio_processing/noise_suppression_impl.cc
index de7e856676ef14e86bebe2a53511d16f7a204943..5ee8160cdddc3b9ec4a263592e3b4e762e225373 100644
--- a/webrtc/modules/audio_processing/noise_suppression_impl.cc
+++ b/webrtc/modules/audio_processing/noise_suppression_impl.cc
@@ -65,6 +65,7 @@ void NoiseSuppressionImpl::Initialize(size_t channels, int sample_rate_hz) {
   }
   suppressors_.swap(new_suppressors);
   set_level(level_);
+  noise_estimate_updated_ = false;
 }
 
 void NoiseSuppressionImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
@@ -81,6 +82,7 @@ void NoiseSuppressionImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
     WebRtcNs_Analyze(suppressors_[i]->state(),
                      audio->split_bands_const_f(i)[kBand0To8kHz]);
   }
+  noise_estimate_updated_ = true;
 #endif
 }
 
@@ -104,6 +106,7 @@ void NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
                       audio->split_bands_const(i),
                       audio->num_bands(),
                       audio->split_bands(i));
+    noise_estimate_updated_ = true;
 #endif
   }
 }
@@ -172,4 +175,31 @@ float NoiseSuppressionImpl::speech_probability() const {
   return AudioProcessing::kUnsupportedFunctionError;
 #endif
 }
+
+const std::vector<float>& NoiseSuppressionImpl::noise_estimate() {
+  rtc::CritScope cs(crit_);
+  if (noise_estimate_updated_) {
+#if defined(WEBRTC_NS_FLOAT)
+    noise_estimate_.assign(WebRtcNs_num_freq(), 0.f);
+    for (auto& suppressor : suppressors_) {
+      float* noise = WebRtcNs_noise_estimate(suppressor->state());
+      for (size_t i = 0; i < noise_estimate_.size(); ++i) {
+        noise_estimate_[i] += noise[i] / suppressors_.size();
+      }
+    }
+#elif defined(WEBRTC_NS_FIXED)
+    noise_estimate_.assign(WebRtcNsx_num_freq(), 0.f);
+    for (auto& suppressor : suppressors_) {
+      uint32_t* noise = WebRtcNsx_noise_estimate(suppressor->state());
+      for (size_t i = 0; i < noise_estimate_.size(); ++i) {
+        noise_estimate_[i] +=
+            static_cast<float>(noise[i] >> 17) / suppressors_.size();

[turaj, 2016/02/05 01:47:58]

Is nois[] is Q17? maybe a comment helps. And if it is in Q17 by shifting >> 17
we lose precision, I don't know how important it is, perhaps not much, but just
to be fair we can do the division in float.

[aluebs-webrtc, 2016/02/06 00:12:27]

It is an unsigned 32-bit integer, and we want to get it to the level of a signed
16-bit integer, so it needs 17 shifts. I thought about making the division in
float to not lose precision, but I thought it was not going to affect and this
was easier to reed. But I guess you are right. Added comment and did division in
float.

+      }
+    }
+#endif
+    noise_estimate_updated_ = false;
+  }
+  return noise_estimate_;
+}
+
 }  // namespace webrtc