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..95d1e29ff7af33205880d0c7bd920950073cae9d 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 +168,19 @@ 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);

[minyue-webrtc, 2016/12/28 08:46:07]

try not to deprecate it now. since it is not a necessary change in this CL. but
of course, remove the override in AudioEncoderOpus as you did.

[michaelt, 2017/01/09 09:24:05]

Done.

-
   // 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);
+  RTC_DEPRECATED virtual void OnReceivedTargetAudioBitrate(int target_bps);
+
+  // Provides target audio bitrate to this encoder to allow it to adapt and the

[minyue-webrtc, 2016/12/27 14:48:58]

// Provides target audio bitrate and corresponding probing interval of the
bandwidth estimator to this encoder to allow it to adapt.

[michaelt, 2017/01/09 09:24:05]

Done.

+  // corresponding probing interval of the bandwidth estimator.
+  virtual void OnReceivedTargetAudioBitrate(
+      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);