Hooking up audio network adaptor to VoE.

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index ae9dae2fbea4bc9e66d6a8b6826236258cc2ecab..60a98f6c6023070f764c9f53d66c8f15fae55967 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -13,6 +13,7 @@
 #include <algorithm>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/base/exp_filter.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
@@ -82,6 +83,35 @@ double OptimizePacketLossRate(double new_loss_rate, double old_loss_rate) {
 
 }  // namespace
 
+class AudioEncoderOpus::PacketLossFractionSmoother {
+ public:
+  explicit PacketLossFractionSmoother(const Clock* clock)
+      : clock_(clock),
+        last_sample_time_ms_(clock_->TimeInMilliseconds()),
+        smoother_(0.9999f) {}

[the sun, 2016/10/11 17:52:42]

Would be nice with a comment about what the filter coeff means physically - e.g.
settles to within 1% in N millisecs.

[minyue-webrtc, 2016/10/12 11:27:35]

I do not quite recall what that number means :P, that is why comments are
important. Now added.

+
+  // Gets the smoothed packet loss fraction.
+  float GetAverage() const {
+    float value = smoother_.filtered();
+    return (value == rtc::ExpFilter::kValueUndefined) ? 0.0f : value;
+  }
+
+  // Add new observation to the packet loss fraction smoother.
+  void AddSample(float packet_loss_fraction) {
+    int64_t now_ms = clock_->TimeInMilliseconds();
+    smoother_.Apply(static_cast<float>(now_ms - last_sample_time_ms_),
+                    packet_loss_fraction);
+    last_sample_time_ms_ = now_ms;
+  }
+
+ private:
+  const Clock* const clock_;
+  int64_t last_sample_time_ms_;
+
+  // An exponential filter is used to smooth the packet loss fraction.
+  rtc::ExpFilter smoother_;
+};
+
 AudioEncoderOpus::Config::Config() = default;
 AudioEncoderOpus::Config::Config(const Config&) = default;
 AudioEncoderOpus::Config::~Config() = default;
@@ -113,9 +143,11 @@ AudioEncoderOpus::AudioEncoderOpus(
     AudioNetworkAdaptorCreator&& audio_network_adaptor_creator)
     : packet_loss_rate_(0.0),
       inst_(nullptr),
+      packet_loss_fraction_smoother_(new PacketLossFractionSmoother(
+          config.clock ? config.clock : Clock::GetRealTimeClock())),
       audio_network_adaptor_creator_(
           audio_network_adaptor_creator
-              ? audio_network_adaptor_creator
+              ? std::move(audio_network_adaptor_creator)
               : [this](const std::string& config_string, const Clock* clock) {
                   return DefaultAudioNetworkAdaptorCreator(config_string,
                                                            clock);
@@ -234,8 +266,11 @@ void AudioEncoderOpus::OnReceivedUplinkBandwidth(int uplink_bandwidth_bps) {
 
 void AudioEncoderOpus::OnReceivedUplinkPacketLossFraction(
     float uplink_packet_loss_fraction) {
-  if (!audio_network_adaptor_)
-    return;
+  if (!audio_network_adaptor_) {
+    packet_loss_fraction_smoother_->AddSample(uplink_packet_loss_fraction);
+    float average_fraction_loss = packet_loss_fraction_smoother_->GetAverage();
+    return SetProjectedPacketLossRate(average_fraction_loss);
+  }
   audio_network_adaptor_->SetUplinkPacketLossFraction(
       uplink_packet_loss_fraction);
   ApplyAudioNetworkAdaptor();
@@ -244,7 +279,7 @@ void AudioEncoderOpus::OnReceivedUplinkPacketLossFraction(
 void AudioEncoderOpus::OnReceivedTargetAudioBitrate(
     int target_audio_bitrate_bps) {
   if (!audio_network_adaptor_)
-    return;
+    return SetTargetBitrate(target_audio_bitrate_bps);
   audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps);
   ApplyAudioNetworkAdaptor();
 }