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

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  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 101 matching lines @@
 RTPSender::RTPSender(
     bool audio,
     Clock* clock,
     Transport* transport,
     RtpAudioFeedback* audio_feedback,
     RtpPacketSender* paced_sender,
     TransportSequenceNumberAllocator* sequence_number_allocator,
     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.
       clock_delta_ms_(clock_->TimeInMilliseconds() -
                       TickTime::MillisecondTimestamp()),
       bitrates_(new BitrateAggregator(bitrate_callback)),
       total_bitrate_sent_(clock, bitrates_->total_bitrate_observer()),
       audio_configured_(audio),
       audio_(audio ? new RTPSenderAudio(clock, this, audio_feedback) : nullptr),
       video_(audio ? nullptr : new RTPSenderVideo(clock, this)),
@@ skipping 17 matching lines @@
       // NACK.
       nack_byte_count_times_(),
       nack_byte_count_(),
       nack_bitrate_(clock, bitrates_->retransmit_bitrate_observer()),
       packet_history_(clock),
       // Statistics
       statistics_crit_(CriticalSectionWrapper::CreateCriticalSection()),
       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),
       ssrc_db_(*SSRCDatabase::GetSSRCDatabase()),
       remote_ssrc_(0),
       sequence_number_forced_(false),
       ssrc_forced_(false),
       timestamp_(0),
       capture_time_ms_(0),
       last_timestamp_time_ms_(0),
@@ skipping 708 matching lines @@
 
   if (!packet_history_.GetPacketAndSetSendTime(sequence_number,
                                                0,
                                                retransmission,
                                                data_buffer,
                                                &length,
                                                &stored_time_ms)) {
     // 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());
     rtx = rtx_;
   }
   return PrepareAndSendPacket(data_buffer,
                               length,
                               capture_time_ms,
                               retransmission && (rtx & kRtxRetransmitted) > 0,
                               retransmission);
@@ skipping 38 matching lines @@
   PacketOptions options;
   if (using_transport_seq) {
     options.packet_id =
         UpdateTransportSequenceNumber(buffer_to_send_ptr, length, rtp_header);
   }
 
   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(capture_time_ms, &options.packet_id);
+  }
+
   bool ret = SendPacketToNetwork(buffer_to_send_ptr, length, options);
   if (ret) {
     CriticalSectionScoped lock(send_critsect_.get());
     media_has_been_sent_ = true;
   }
   UpdateRtpStats(buffer_to_send_ptr, length, rtp_header, send_over_rtx,
                  is_retransmit);
   return ret;
 }
 
@@ skipping 99 matching lines @@
                                 payload_length, false);
     if (last_capture_time_ms_sent_ == 0 ||
         corrected_time_ms > last_capture_time_ms_sent_) {
       last_capture_time_ms_sent_ = corrected_time_ms;
       TRACE_EVENT_ASYNC_BEGIN1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                                "PacedSend", corrected_time_ms,
                                "capture_time_ms", corrected_time_ms);
     }
     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
   // ready for a retransmission.
   packet_history_.SetSent(rtp_header.sequenceNumber);
 
   if (!sent)
     return -1;
 
   {
     CriticalSectionScoped lock(send_critsect_.get());
     media_has_been_sent_ = true;
   }
   UpdateRtpStats(buffer, length, rtp_header, false, false);
   return 0;
 }
 
 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;
   int avg_delay_ms = 0;
   int max_delay_ms = 0;
   {
     CriticalSectionScoped lock(send_critsect_.get());
     ssrc = ssrc_;
   }
   {
@@ skipping 11 matching lines @@
       ++num_delays;
     }
     if (num_delays == 0)
       return;
     avg_delay_ms = (avg_delay_ms + num_delays / 2) / num_delays;
   }
   send_side_delay_observer_->SendSideDelayUpdated(avg_delay_ms, max_delay_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();
   nack_bitrate_.Process();
   if (audio_configured_) {
     return;
   }
   video_->ProcessBitrate();
 }
 
@@ skipping 771 matching lines @@
   CriticalSectionScoped lock(send_critsect_.get());
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc