Update smoothed bitrate.

webrtc/modules/audio_coding/codecs/audio_encoder.h

Index: webrtc/modules/audio_coding/codecs/audio_encoder.h
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index 5132c2ed8ba332f4589cba3431f5f37a96f66d08..8f07bd81cc7f70bcd5d20791742d7f521706bbf9 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -17,6 +17,7 @@
 #include "webrtc/base/array_view.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/deprecation.h"
+#include "webrtc/base/optional.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -167,16 +168,15 @@ class AudioEncoder {
   // Disables audio network adaptor.
   virtual void DisableAudioNetworkAdaptor();
 
-  // Provides uplink bandwidth to this encoder to allow it to adapt.
-  virtual void OnReceivedUplinkBandwidth(int uplink_bandwidth_bps);
-
   // Provides uplink packet loss fraction to this encoder to allow it to adapt.
   // |uplink_packet_loss_fraction| is in the range [0.0, 1.0].
   virtual void OnReceivedUplinkPacketLossFraction(
       float uplink_packet_loss_fraction);
 
   // Provides target audio bitrate to this encoder to allow it to adapt.
-  virtual void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps);
+  virtual void OnReceivedTargetAudioBitrate(

[ossu, 2016/12/22 14:01:15]

Going forward, I think we should try and find a cleaner interface between
encoders and the BWE, decoupling encoder implementations from BWE specifics.
This is fine for now, though!

[minyue-webrtc, 2016/12/22 14:51:26]

RTC_DEPRECATED virtual void OnReceivedTargetAudioBitrate(int target_bps); and
add a default implementation

OnReceivedTargetAudioBitrate(int target_bps) {
  OnReceivedTargetAudioBitrate(target_bps, rtc::Optional<int64_t>())
}

then add 

virtual void OnReceivedTargetAudioBitrate(int target_bps, rtc::Optional<int64_t>
probing_interval_ms);

May break internal bots otherwise (not likely though, since it is new)

[michaelt, 2016/12/22 15:10:10]

Do you think this is necessary, wouldn't it be enough to run internal server ?

[minyue-webrtc, 2016/12/22 15:12:02]

No hurt to be cautious. then we avoid risk of revert.

[michaelt, 2016/12/22 15:35:25]

Done.

+      int target_audio_bitrate_bps,
+      rtc::Optional<int64_t> probing_interval_ms);
 
   // Provides RTT to this encoder to allow it to adapt.
   virtual void OnReceivedRtt(int rtt_ms);