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

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
index d004cd8d4fbe6239d6515a2d328cb777807dc33b..abaa4b87151c3dbb69973358b552d890c7bfe744 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
@@ -119,7 +119,8 @@ RTPSender::RTPSender(
     TransportFeedbackObserver* transport_feedback_observer,
     BitrateStatisticsObserver* bitrate_callback,
     FrameCountObserver* frame_count_observer,
-    SendSideDelayObserver* send_side_delay_observer)
+    SendSideDelayObserver* send_side_delay_observer,
+    SendPacketObserver* send_packet_observer)
     : clock_(clock),
       // TODO(holmer): Remove this conversion when we remove the use of
       // TickTime.
@@ -157,6 +158,7 @@ RTPSender::RTPSender(
       rtp_stats_callback_(NULL),
       frame_count_observer_(frame_count_observer),
       send_side_delay_observer_(send_side_delay_observer),
+      send_packet_observer_(send_packet_observer),
       // RTP variables
       start_timestamp_forced_(false),
       start_timestamp_(0),
@@ -885,9 +887,7 @@ bool RTPSender::TimeToSendPacket(uint16_t sequence_number,
     // Packet cannot be found. Allow sending to continue.
     return true;
   }
-  if (!retransmission && capture_time_ms > 0) {
-    UpdateDelayStatistics(capture_time_ms, clock_->TimeInMilliseconds());
-  }
+
   int rtx;
   {
     CriticalSectionScoped lock(send_critsect_.get());
@@ -946,6 +946,11 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
     transport_feedback_observer_->AddPacket(options.packet_id, length, true);
   }
 
+  if (!is_retransmit && !send_over_rtx) {
+    UpdateDelayStatistics(capture_time_ms, now_ms);
+    UpdateOnSendPacket(capture_time_ms, &options.packet_id);
+  }
+
   bool ret = SendPacketToNetwork(buffer_to_send_ptr, length, options);
   if (ret) {
     CriticalSectionScoped lock(send_critsect_.get());
@@ -1065,12 +1070,12 @@ int32_t RTPSender::SendToNetwork(uint8_t* buffer,
     }
     return 0;
   }
-  if (capture_time_ms > 0) {
-    UpdateDelayStatistics(capture_time_ms, now_ms);
-  }
+  PacketOptions options;
+  UpdateDelayStatistics(capture_time_ms, now_ms);
+  UpdateOnSendPacket(capture_time_ms, &options.packet_id);

[stefan-webrtc, 2015/12/10 08:37:49]

I don't think we have to add it here since this path is deprecated and shouldn't
be used. All packets are going through the pacer.

[åsapersson, 2015/12/15 14:28:26]

OK, removed from this path.

 
   size_t length = payload_length + rtp_header_length;
-  bool sent = SendPacketToNetwork(buffer, length, PacketOptions());
+  bool sent = SendPacketToNetwork(buffer, length, options);
 
   // Mark the packet as sent in the history even if send failed. Dropping a
   // packet here should be treated as any other packet drop so we should be
@@ -1089,7 +1094,7 @@ int32_t RTPSender::SendToNetwork(uint8_t* buffer,
 }
 
 void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
-  if (!send_side_delay_observer_)
+  if (!send_side_delay_observer_ || capture_time_ms <= 0)
     return;
 
   uint32_t ssrc;
@@ -1121,6 +1126,22 @@ void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
                                                   ssrc);
 }
 
+void RTPSender::UpdateOnSendPacket(int64_t capture_time_ms, int* packet_id) {
+  if (!send_packet_observer_ || !transport_sequence_number_allocator_ ||
+      capture_time_ms <= 0)
+    return;
+
+  uint32_t ssrc;
+  {
+    CriticalSectionScoped lock(send_critsect_.get());
+    ssrc = ssrc_;

[stefan-webrtc, 2015/12/10 08:37:49]

The ssrc is available without locking from the rtp_header after line 912.

[åsapersson, 2015/12/15 14:28:26]

Done.

+  }
+  if (*packet_id == -1)
+    *packet_id = transport_sequence_number_allocator_->AllocateSequenceNumber();

[stefan-webrtc, 2015/12/10 08:37:49]

Can we move this out of this method and call it once before this method is
called? I think we should call it for all packets we send, but only set the
transport sequence number if the extension is used, see line 942.

[åsapersson, 2015/12/15 14:28:26]

Done.

+
+  send_packet_observer_->OnSendPacket(*packet_id, capture_time_ms, ssrc);

[stefan-webrtc, 2015/12/10 08:37:49]

Is it possible to unify this callback with the callback on line 946, so that we
only have to register a single "OnSendPacket"/"AddPacket" callback to the
module?

[åsapersson, 2015/12/15 14:28:26]

I guess it can be done if we do not want to do it per stream. Wdyt?
Use SendTimeHistory (and add capture_time_ms info for cap -> sent)?
Would audio also be included then?

[stefan-webrtc, 2016/01/18 19:48:25]

Why doesn't it work per stream? I don't think I understand the issue. I was
mostly thinking that we could register a single "observer" object to this class
instead of two separate. That observer could still have two methods, "AddPacket"
and "OnSendPacket".

+}
+
 void RTPSender::ProcessBitrate() {
   CriticalSectionScoped cs(send_critsect_.get());
   total_bitrate_sent_.Process();