Update smoothed bitrate.

webrtc/modules/audio_coding/codecs/audio_encoder.h

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_
 
 #include <algorithm>
 #include <vector>
 
 #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 {
 
 class Clock;
 
 // This is the interface class for encoders in AudioCoding module. Each codec
 // type must have an implementation of this class.
 class AudioEncoder {
  public:
@@ skipping 130 matching lines @@
   virtual rtc::ArrayView<std::unique_ptr<AudioEncoder>>
   ReclaimContainedEncoders();
 
   // Enables audio network adaptor. Returns true if successful.
   virtual bool EnableAudioNetworkAdaptor(const std::string& config_string,
                                          const Clock* clock);
 
   // 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);
 
   // Provides overhead to this encoder to adapt. The overhead is the number of
   // bytes that will be added to each packet the encoder generates.
   virtual void OnReceivedOverhead(size_t overhead_bytes_per_packet);
 
   // To allow encoder to adapt its frame length, it must be provided the frame
   // length range that receivers can accept.
   virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
                                            int max_frame_length_ms);
 
  protected:
   // Subclasses implement this to perform the actual encoding. Called by
   // Encode().
   virtual EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
                                  rtc::ArrayView<const int16_t> audio,
                                  rtc::Buffer* encoded) = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_AUDIO_ENCODER_H_