Update smoothed bitrate.

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

Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index 0864c7bfb561b6af167c5ba13fd395571b81b8bf..d67cedfc9c4a5773e036b8f15ba8c1b81414c48c 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -18,6 +18,7 @@
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/optional.h"
+#include "webrtc/common_audio/smoothing_filter.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
@@ -62,7 +63,8 @@ class AudioEncoderOpus final : public AudioEncoder {
     int complexity_threshold_window_bps = 1500;
     bool dtx_enabled = false;
     std::vector<int> supported_frame_lengths_ms;
-    const Clock* clock = nullptr;
+    const Clock* clock = Clock::GetRealTimeClock();

[the sun, 2017/01/09 12:56:52]

The Clock:: API is being deprecated - use those in base/timeutils.h instead.

[michaelt, 2017/01/09 15:09:26]

I changed the direct in this CL used time function to the new API.
Will add a followup CL which will change smoothing filter and other dependencies
as well.

+    int uplink_bandwidth_update_interval_ms = 200;
 
    private:
 #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS) || defined(WEBRTC_ARCH_ARM)
@@ -79,7 +81,8 @@ class AudioEncoderOpus final : public AudioEncoder {
                                                          const Clock*)>;
   AudioEncoderOpus(
       const Config& config,
-      AudioNetworkAdaptorCreator&& audio_network_adaptor_creator = nullptr);
+      AudioNetworkAdaptorCreator&& audio_network_adaptor_creator = nullptr,
+      std::unique_ptr<SmoothingFilter> bitrate_smoother = nullptr);
 
   explicit AudioEncoderOpus(const CodecInst& codec_inst);
 
@@ -105,10 +108,11 @@ class AudioEncoderOpus final : public AudioEncoder {
   bool EnableAudioNetworkAdaptor(const std::string& config_string,
                                  const Clock* clock) override;
   void DisableAudioNetworkAdaptor() override;
-  void OnReceivedUplinkBandwidth(int uplink_bandwidth_bps) override;
   void OnReceivedUplinkPacketLossFraction(
       float uplink_packet_loss_fraction) override;
-  void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) override;
+  void OnReceivedTargetAudioBitrate(
+      int target_audio_bitrate_bps,
+      rtc::Optional<int64_t> probing_interval_ms) override;
   void OnReceivedRtt(int rtt_ms) override;
   void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
   void SetReceiverFrameLengthRange(int min_frame_length_ms,
@@ -149,6 +153,8 @@ class AudioEncoderOpus final : public AudioEncoder {
       const std::string& config_string,
       const Clock* clock) const;
 
+  void MaybeUpdateUplinkBandwidth();
+
   Config config_;
   float packet_loss_rate_;
   std::vector<int16_t> input_buffer_;
@@ -161,6 +167,8 @@ class AudioEncoderOpus final : public AudioEncoder {
   AudioNetworkAdaptorCreator audio_network_adaptor_creator_;
   std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_;
   rtc::Optional<size_t> overhead_bytes_per_packet_;
+  const std::unique_ptr<SmoothingFilter> bitrate_smoother_;
+  rtc::Optional<int64_t> bitrate_smoother_last_update_time_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
 };