Add histogram stats for send delay for a sent video stream.

webrtc/common_types.h

 /*
  *  Copyright (c) 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 278 matching lines @@
 
 // Callback, used to notify an observer whenever the send-side delay is updated.
 class SendSideDelayObserver {
  public:
   virtual ~SendSideDelayObserver() {}
   virtual void SendSideDelayUpdated(int avg_delay_ms,
                                     int max_delay_ms,
                                     uint32_t ssrc) = 0;
 };
 
+// Callback, used to notify an observer whenever a packet is sent to the
+// transport.
+// TODO(asapersson): This class will remove the need for SendSideDelayObserver.
+// Remove SendSideDelayObserver once possible.
+class SendPacketObserver {
+ public:
+  virtual ~SendPacketObserver() {}
+  virtual void OnSendPacket(uint16_t packet_id,
+                            int64_t capture_time_ms,
+                            uint32_t ssrc) = 0;
+};
+
 // ==================================================================
 // Voice specific types
 // ==================================================================
 
 // Each codec supported can be described by this structure.
 struct CodecInst {
   int pltype;
   char plname[RTP_PAYLOAD_NAME_SIZE];
   int plfreq;
   int pacsize;
@@ skipping 572 matching lines @@
 enum class RtcpMode { kOff, kCompound, kReducedSize };
 
 enum NetworkState {
   kNetworkUp,
   kNetworkDown,
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_COMMON_TYPES_H_