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..4c45c4954429c5fbcaffd8a24ed1e39adcde593d 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),
@@ -608,9 +610,6 @@ size_t RTPSender::SendPadData(size_t bytes,
   size_t padding_bytes_in_packet =
       std::min(MaxDataPayloadLength(), kMaxPaddingLength);
   size_t bytes_sent = 0;
-  bool using_transport_seq = rtp_header_extension_map_.IsRegistered(
-                                 kRtpExtensionTransportSequenceNumber) &&
-                             transport_sequence_number_allocator_;
   for (; bytes > 0; bytes -= padding_bytes_in_packet) {
     if (bytes < padding_bytes_in_packet)
       bytes = padding_bytes_in_packet;
@@ -677,13 +676,13 @@ size_t RTPSender::SendPadData(size_t bytes,
     UpdateAbsoluteSendTime(padding_packet, length, rtp_header, now_ms);
 
     PacketOptions options;
-    if (using_transport_seq) {
-      options.packet_id =
-          UpdateTransportSequenceNumber(padding_packet, length, rtp_header);
-    }
-
-    if (using_transport_seq && transport_feedback_observer_) {
-      transport_feedback_observer_->AddPacket(options.packet_id, length, true);
+    if (UseTransportSequenceNumber(&options.packet_id)) {
+      UpdateTransportSequenceNumber(options.packet_id, padding_packet, length,
+                                    rtp_header);
+      if (transport_feedback_observer_) {
+        transport_feedback_observer_->AddPacket(options.packet_id, length,
+                                                true);
+      }
     }
 
     if (!SendPacketToNetwork(padding_packet, length, options))
@@ -885,9 +884,7 @@ 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());
@@ -931,19 +928,17 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
                                diff_ms);
   UpdateAbsoluteSendTime(buffer_to_send_ptr, length, rtp_header, now_ms);
 
-  // TODO(sprang): Potentially too much overhead in IsRegistered()?
-  bool using_transport_seq = rtp_header_extension_map_.IsRegistered(
-                                 kRtpExtensionTransportSequenceNumber) &&
-                             transport_sequence_number_allocator_;
-
   PacketOptions options;
-  if (using_transport_seq) {
-    options.packet_id =
-        UpdateTransportSequenceNumber(buffer_to_send_ptr, length, rtp_header);
+  if (UseTransportSequenceNumber(&options.packet_id)) {
+    UpdateTransportSequenceNumber(options.packet_id, buffer_to_send_ptr, length,
+                                  rtp_header);
+    if (transport_feedback_observer_)
+      transport_feedback_observer_->AddPacket(options.packet_id, length, true);
   }
 
-  if (using_transport_seq && transport_feedback_observer_) {
-    transport_feedback_observer_->AddPacket(options.packet_id, length, true);
+  if (!is_retransmit && !send_over_rtx) {
+    UpdateDelayStatistics(capture_time_ms, now_ms);
+    UpdateOnSendPacket(options.packet_id, capture_time_ms, rtp_header.ssrc);
   }
 
   bool ret = SendPacketToNetwork(buffer_to_send_ptr, length, options);
@@ -956,6 +951,17 @@ bool RTPSender::PrepareAndSendPacket(uint8_t* buffer,
   return ret;
 }
 
+bool RTPSender::UseTransportSequenceNumber(int* packet_id) const {

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

Maybe this should say that it actually allocates a sequence number even if the
extension isn't registered? I think it might be worth the extra code to split it
in two because of this. WDYT?

[åsapersson, 2016/04/06 14:52:37]

Changed to AllocateTransportSequenceNumber. Removed the registration check
(checked in UpdateTransportSequenceNumber).

+  if (!transport_sequence_number_allocator_)
+    return false;
+
+  *packet_id = transport_sequence_number_allocator_->AllocateSequenceNumber();
+
+  // TODO(sprang): Potentially too much overhead in IsRegistered()?
+  return rtp_header_extension_map_.IsRegistered(
+      kRtpExtensionTransportSequenceNumber);
+}
+
 void RTPSender::UpdateRtpStats(const uint8_t* buffer,
                                size_t packet_length,
                                const RTPHeader& header,
@@ -1065,9 +1071,8 @@ int32_t RTPSender::SendToNetwork(uint8_t* buffer,
     }
     return 0;
   }
-  if (capture_time_ms > 0) {
-    UpdateDelayStatistics(capture_time_ms, now_ms);
-  }
+
+  UpdateDelayStatistics(capture_time_ms, now_ms);
 
   size_t length = payload_length + rtp_header_length;
   bool sent = SendPacketToNetwork(buffer, length, PacketOptions());
@@ -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)

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

Do you think it would be possible to instead use a DCHECK:
DCHECK(capture_time_ms <= 0 || audio_configured_)?

[åsapersson, 2016/04/06 14:52:37]

Not sure if set in all codec wrappers, would need to check separately.

     return;
 
   uint32_t ssrc;
@@ -1121,6 +1126,15 @@ void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
                                                   ssrc);
 }
 
+void RTPSender::UpdateOnSendPacket(int packet_id,
+                                   int64_t capture_time_ms,
+                                   uint32_t ssrc) {
+  if (!send_packet_observer_ || capture_time_ms <= 0 || packet_id == -1)

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

Same here about DCHECK. Should be able to DCHECK both on capture_time_ms as
above and also have one on packet_id >= 0 since it always should be set, right?

[åsapersson, 2016/04/06 14:52:37]

packet_id could be -1 as it is now.

+    return;
+
+  send_packet_observer_->OnSendPacket(packet_id, capture_time_ms, ssrc);
+}
+
 void RTPSender::ProcessBitrate() {
   CriticalSectionScoped cs(send_critsect_.get());
   total_bitrate_sent_.Process();
@@ -1611,7 +1625,8 @@ void RTPSender::UpdateAbsoluteSendTime(uint8_t* rtp_packet,
                                           ConvertMsTo24Bits(now_ms));
 }
 
-uint16_t RTPSender::UpdateTransportSequenceNumber(
+bool RTPSender::UpdateTransportSequenceNumber(
+    uint16_t sequence_number,
     uint8_t* rtp_packet,
     size_t rtp_packet_length,
     const RTPHeader& rtp_header) const {
@@ -1622,19 +1637,18 @@ uint16_t RTPSender::UpdateTransportSequenceNumber(
                           rtp_packet_length, rtp_header,
                           kTransportSequenceNumberLength, &offset)) {
     case ExtensionStatus::kNotRegistered:
-      return 0;
+      return false;
     case ExtensionStatus::kError:
       LOG(LS_WARNING) << "Failed to update transport sequence number";
-      return 0;
+      return false;
     case ExtensionStatus::kOk:
       break;
     default:
       RTC_NOTREACHED();
   }
 
-  uint16_t seq = transport_sequence_number_allocator_->AllocateSequenceNumber();
-  BuildTransportSequenceNumberExtension(rtp_packet + offset, seq);
-  return seq;
+  BuildTransportSequenceNumberExtension(rtp_packet + offset, sequence_number);
+  return true;
 }
 
 void RTPSender::SetSendingStatus(bool enabled) {