Move RTP module activation into PayloadRouter.

webrtc/video/payload_router.h

 /*
  *  Copyright (c) 2015 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 18 matching lines @@
 // PayloadRouter routes outgoing data to the correct sending RTP module, based
 // on the simulcast layer in RTPVideoHeader.
 class PayloadRouter {
  public:
   PayloadRouter();
   ~PayloadRouter();
 
   static size_t DefaultMaxPayloadLength();
 
   // Rtp modules are assumed to be sorted in simulcast index order.
-  void SetSendingRtpModules(const std::vector<RtpRtcp*>& rtp_modules);
+  void Init(const std::vector<RtpRtcp*>& rtp_modules);
+
+  void SetSendingRtpModules(size_t num_sending_modules);
 
   // PayloadRouter will only route packets if being active, all packets will be
   // dropped otherwise.
   void set_active(bool active);
   bool active();
 
   // Input parameters according to the signature of RtpRtcp::SendOutgoingData.
   // Returns true if the packet was routed / sent, false otherwise.
   bool RoutePayload(FrameType frame_type,
                     int8_t payload_type,
                     uint32_t time_stamp,
                     int64_t capture_time_ms,
                     const uint8_t* payload_data,
                     size_t payload_size,
                     const RTPFragmentationHeader* fragmentation,
                     const RTPVideoHeader* rtp_video_hdr);
 
   // Configures current target bitrate per module. 'stream_bitrates' is assumed
   // to be in the same order as 'SetSendingRtpModules'.
   void SetTargetSendBitrates(const std::vector<uint32_t>& stream_bitrates);
 
   // Returns the maximum allowed data payload length, given the configured MTU
   // and RTP headers.
   size_t MaxPayloadLength() const;
 
-  void AddRef() { ++ref_count_; }
-  void Release() { if (--ref_count_ == 0) { delete this; } }
+ private:
+  void UpdateModuleSendingState() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
- private:
-  // TODO(mflodman): When the new video API has launched, remove crit_ and
-  // assume rtp_modules_ will never change during a call.
+  // TODO(pbos): Set once and for all on construction and make const.
+  std::vector<RtpRtcp*> rtp_modules_;
+
   rtc::CriticalSection crit_;
-
-  // Active sending RTP modules, in layer order.
-  std::vector<RtpRtcp*> rtp_modules_ GUARDED_BY(crit_);
   bool active_ GUARDED_BY(crit_);
-
-  Atomic32 ref_count_;
+  size_t num_sending_modules_ GUARDED_BY(crit_);
 
   RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_PAYLOAD_ROUTER_H_