BWE allocation strategy

webrtc/api/peerconnectioninterface.h

 /*
  *  Copyright 2012 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.
  */
 
@@ skipping 66 matching lines @@
 #include "webrtc/api/datachannelinterface.h"
 #include "webrtc/api/dtmfsenderinterface.h"
 #include "webrtc/api/jsep.h"
 #include "webrtc/api/mediastreaminterface.h"
 #include "webrtc/api/rtcerror.h"
 #include "webrtc/api/rtpreceiverinterface.h"
 #include "webrtc/api/rtpsenderinterface.h"
 #include "webrtc/api/stats/rtcstatscollectorcallback.h"
 #include "webrtc/api/statstypes.h"
 #include "webrtc/api/umametrics.h"
+#include "webrtc/base/bitrateallocationstrategy.h"
 #include "webrtc/base/fileutils.h"
 #include "webrtc/base/network.h"
 #include "webrtc/base/rtccertificate.h"
 #include "webrtc/base/rtccertificategenerator.h"
 #include "webrtc/base/socketaddress.h"
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/call/callfactoryinterface.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log_factory_interface.h"
 #include "webrtc/media/base/mediachannel.h"
 #include "webrtc/media/base/videocapturer.h"
@@ skipping 655 matching lines @@
   // SetBitrate limits the bandwidth allocated for all RTP streams sent by
   // this PeerConnection. Other limitations might affect these limits and
   // are respected (for example "b=AS" in SDP).
   //
   // Setting |current_bitrate_bps| will reset the current bitrate estimate
   // to the provided value.
   virtual RTCError SetBitrate(const BitrateParameters& bitrate) {
     return RTCError::OK();
   }
 
+  // SetBitrateAllocationStrategy sets current strategy. If not set default
+  // WEBRTC allocator will be used. May be changed during an active session.
+  virtual RTCError SetBitrateAllocationStrategy(

[stefan-webrtc, 2017/08/14 07:46:03]

I would prefer if we could pass the allocation strategy to PeerConnection and
Call upon creation via the constructor instead of using a setter. It's easier to
see life time, and doesn't require locks.

It should be doable by adding an argument to this method:
https://cs.chromium.org/chromium/src/third_party/webrtc/pc/peerconnectionfact...

WDYT?

[alexnarest, 2017/08/15 06:02:36]

Helper would let users to change strategies during a call. For example audio
only hangout session may not need a custom strategy but as soon as video/screen
share tracks are added it may want to change allocations. Having strategy
allocated in constructor will require users to implement allocations even for
the cases the default implementation is ok.

+      rtc::BitrateAllocationStrategy* bitrate_allocation_strategy) {
+    return RTCError(RTCErrorType::UNSUPPORTED_OPERATION);
+  }
+
   // Returns the current SignalingState.
   virtual SignalingState signaling_state() = 0;
   virtual IceConnectionState ice_connection_state() = 0;
   virtual IceGatheringState ice_gathering_state() = 0;
 
   // Starts RtcEventLog using existing file. Takes ownership of |file| and
   // passes it on to Call, which will take the ownership. If the
   // operation fails the file will be closed. The logging will stop
   // automatically after 10 minutes have passed, or when the StopRtcEventLog
   // function is called.
@@ skipping 408 matching lines @@
     cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
     cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
     rtc::scoped_refptr<AudioMixer> audio_mixer,
     std::unique_ptr<cricket::MediaEngineInterface> media_engine,
     std::unique_ptr<CallFactoryInterface> call_factory,
     std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory);
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_PEERCONNECTIONINTERFACE_H_