Update common_audio/smoothing_filter.

webrtc/common_audio/smoothing_filter.cc

Index: webrtc/common_audio/smoothing_filter.cc
diff --git a/webrtc/common_audio/smoothing_filter.cc b/webrtc/common_audio/smoothing_filter.cc
index ff79ab8799e9de332ff324152e97c43947837e6a..162fe2a1721f139008f8cbd3594c30860d138aa2 100644
--- a/webrtc/common_audio/smoothing_filter.cc
+++ b/webrtc/common_audio/smoothing_filter.cc
@@ -8,60 +8,95 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "webrtc/common_audio/smoothing_filter.h"
+
 #include <cmath>
 
-#include "webrtc/common_audio/smoothing_filter.h"
+#include "webrtc/base/logging.h"
 
 namespace webrtc {
 
-SmoothingFilterImpl::SmoothingFilterImpl(int time_constant_ms,
-                                         const Clock* clock)
-    : time_constant_ms_(time_constant_ms),
-      clock_(clock),
-      first_sample_received_(false),
-      initialized_(false),
-      first_sample_time_ms_(0),
-      last_sample_time_ms_(0),
-      filter_(0.0) {}
+SmoothingFilterImpl::SmoothingFilterImpl(int init_time_ms_, const Clock* clock)
+    : init_time_ms_(init_time_ms_),
+      // Duing the initalization time, we use an increasing alpha. Specifically,
+      //   alpha(n) = exp(pow(init_factor_, n)),
+      // where |init_factor_| is chosen such that
+      //   alpha(init_time_ms_) = exp(-1.0f / init_time_ms_),
+      init_factor_(pow(init_time_ms_, 1.0f / init_time_ms_)),
+      init_const_(1.0f / (init_time_ms_ -
+                          pow(init_time_ms_, 1.0f - 1.0f / init_time_ms_))),
+      clock_(clock) {
+  UpdateAlpha(init_time_ms_);
+}
+
+SmoothingFilterImpl::~SmoothingFilterImpl() = default;
 
 void SmoothingFilterImpl::AddSample(float sample) {
-  if (!first_sample_received_) {
-    last_sample_time_ms_ = first_sample_time_ms_ = clock_->TimeInMilliseconds();
-    first_sample_received_ = true;
-    RTC_DCHECK_EQ(rtc::ExpFilter::kValueUndefined, filter_.filtered());
+  int64_t now_ms = clock_->TimeInMilliseconds();
 
-    // Since this is first sample, any value for argument 1 should work.
-    filter_.Apply(0.0f, sample);
+  if (!first_sample_time_ms_) {
+    // This is equivalent to assuming the filter has been receiving the same
+    // value as the first sample since time -infinity.
+    state_ = last_sample_ = sample;
+    first_sample_time_ms_ = rtc::Optional<int64_t>(now_ms);
+    last_state_time_ms_ = now_ms;
     return;
   }
 
-  int64_t now_ms = clock_->TimeInMilliseconds();
-  if (!initialized_) {
-    float duration = now_ms - first_sample_time_ms_;
-    if (duration < static_cast<int64_t>(time_constant_ms_)) {
-      filter_.UpdateBase(exp(1.0f / duration));
-    } else {
-      initialized_ = true;
-      filter_.UpdateBase(exp(1.0f / time_constant_ms_));
-    }
-  }
-
-  // The filter will do the following:
-  //   float alpha = pow(base, last_update_time_ms_ - now_ms);
-  //   filtered_ = alpha * filtered_ + (1 - alpha) * sample;
-  filter_.Apply(static_cast<float>(last_sample_time_ms_ - now_ms), sample);
-  last_sample_time_ms_ = now_ms;
+  ExtrapolateLastSample(now_ms);
+  last_sample_ = sample;
 }
 
-rtc::Optional<float> SmoothingFilterImpl::GetAverage() const {
-  float value = filter_.filtered();
-  return value == rtc::ExpFilter::kValueUndefined ? rtc::Optional<float>()
-                                                  : rtc::Optional<float>(value);
+rtc::Optional<float> SmoothingFilterImpl::GetAverage() {
+  if (!first_sample_time_ms_)
+    return rtc::Optional<float>();
+  ExtrapolateLastSample(clock_->TimeInMilliseconds());
+  return rtc::Optional<float>(state_);
 }
 
 void SmoothingFilterImpl::SetTimeConstantMs(int time_constant_ms) {
-  time_constant_ms_ = time_constant_ms;
-  filter_.UpdateBase(exp(1.0f / time_constant_ms_));
+  if (!first_sample_time_ms_ ||
+      last_state_time_ms_ < *first_sample_time_ms_ + init_time_ms_) {
+    LOG(LS_INFO) << "SmoothingFilterImpl: Cannot set time constant "
+                 << time_constant_ms << " ms during the initialization time.";
+    return;
+  }
+  UpdateAlpha(time_constant_ms);
+}
+
+void SmoothingFilterImpl::UpdateAlpha(int time_constant_ms) {
+  alpha_ = exp(-1.0f / time_constant_ms);
+}
+
+void SmoothingFilterImpl::ExtrapolateLastSample(int64_t time_ms) {
+  RTC_DCHECK_GE(time_ms, last_state_time_ms_);
+  RTC_DCHECK(first_sample_time_ms_);
+
+  float multiplier = 0.0f;
+  if (time_ms <= *first_sample_time_ms_ + init_time_ms_) {
+    // Current update is to be made during initialization phase.
+    // We update the state as if the |alpha| has been increased according
+    //   alpha(n) = exp(pow(init_factor_, n)),
+    // where n is the time (in millisecond) since the first sample received.
+    // With algebraic derivation, we can find that the state can be updated

[michaelt, 2016/12/12 09:05:30]

Could you add the algebraic derivation ?

[minyue-webrtc, 2016/12/12 09:12:14]

I will try. But pure text will be a bit cumbersome to write derivation. I think
it would good to add LaTeX syntax in an appendix?

+    // in a similar manner as if alpha is a constant, except for a different
+    // multiplier.
+    multiplier = exp(-init_const_ *
+        (pow(init_factor_,
+             *first_sample_time_ms_ + init_time_ms_ - last_state_time_ms_) -
+         pow(init_factor_, *first_sample_time_ms_ + init_time_ms_ - time_ms)));
+  } else {
+    if (last_state_time_ms_ < *first_sample_time_ms_ + init_time_ms_) {
+      // The latest state update was made during initialization phase.
+      // We first extrapolate to the initialization time.
+      ExtrapolateLastSample(*first_sample_time_ms_ + init_time_ms_);
+      // Then extrapolate the rest by the following.
+    }
+    multiplier = pow(alpha_, time_ms - last_state_time_ms_);
+  }
+
+  state_ = multiplier * state_ + (1.0f - multiplier) * last_sample_;
+  last_state_time_ms_ = time_ms;
 }
 
 }  // namespace webrtc