Fix for NetEq simulations containing large gaps and multiple SSRCs.

This CL fixes 2 issues that affect NetEq simulations.
- When using event logs with multiple SSRCs, it does not make sense to
  use more than a single SSRC. If the user does not provide an SSRC
  filter, we should use the first SSRC we find and no others.
- It is possible for event logs to have a gap in the middle, and
  sometimes we don't store/mark the gap properly. If is possible to
  detect gaps by looking at the wallclock time delta between getAudio
  events. These should be 10 ms nominally, so values greater than 1000
  should never happen and indicate an error.

Bug: webrtc:11855
Change-Id: Idc3b8a7902be4159da48b063ef5c5c82fd484071
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181940
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31960}

modules/audio_coding/neteq/tools/neteq_test.cc

@@ -267,8 +267,12 @@ NetEqTest::SimulationStepResult NetEqTest::RunToNextGetAudio() {
       prev_lifetime_stats_ = lifetime_stats;
       const bool no_more_packets_to_decode =
           !input_->NextPacketTime() && !operations_state.next_packet_available;
-      result.is_simulation_finished =
-          no_more_packets_to_decode || input_->ended();
+      // End the simulation if the gap is too large. This indicates an issue
+      // with the event log file.
+      const bool simulation_step_too_large = result.simulation_step_ms > 1000;
+      result.is_simulation_finished = simulation_step_too_large ||
+                                      no_more_packets_to_decode ||
+                                      input_->ended();
       prev_ops_state_ = operations_state;
       return result;
     }

modules/audio_coding/neteq/tools/rtc_event_log_source.cc

@@ -138,6 +138,11 @@ bool RtcEventLogSource::Initialize(const ParsedRtcEventLog& parsed_log,
       continue;
     }
     event_processor.AddEvents(rtp_packets.incoming_packets, handle_rtp_packet);
+    // If no SSRC filter has been set, use the first SSRC only. The simulator
+    // does not work properly with interleaved packets from multiple SSRCs.
+    if (!ssrc_filter.has_value()) {
+      ssrc_filter = rtp_packets.ssrc;
+    }
   }
 
   for (const auto& audio_playouts : parsed_log.audio_playout_events()) {