commit 94f107454e3e79981aab2f4e15b205344e3bd58d
author Ivo Creusen <ivoc@webrtc.org> Fri Dec 14 13:43:11 2018
committer Commit Bot <commit-bot@chromium.org> Fri Dec 14 15:05:15 2018
tree ff2510e3a71a0f8781222f722310a97427e76fbd
parent 24779d82292e57fdec11bbb540ce4137b07416ee

Only use GetAudio events that correspond to an ssrc matching at least one incoming packet.

Using GetAudio events from SSRCs without incoming packets doesn't make sense, and should be prevented.

Bug: b/116685514
Change-Id: I48e38bb780549c71cb5f68d370a6819634ad487d
Reviewed-on: https://webrtc-review.googlesource.com/c/114321
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26017}

modules/audio_coding/neteq/tools/rtc_event_log_source.cc

diff --git a/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
index 582c2f2..0110b8c 100644
--- a/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
+++ b/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
@@ -13,6 +13,7 @@
 #include <string.h>
 #include <iostream>
 #include <limits>
+#include <set>
 #include <utility>
 
 #include "logging/rtc_event_log/rtc_event_processor.h"
@@ -73,23 +74,30 @@
           ? std::numeric_limits<int64_t>::max()
           : parsed_log.stop_log_events().front().log_time_us();
 
+  std::set<uint32_t> packet_ssrcs;
   auto handle_rtp_packet =
-      [this,
-       first_log_end_time_us](const webrtc::LoggedRtpPacketIncoming& incoming) {
+      [this, first_log_end_time_us,
+       &packet_ssrcs](const webrtc::LoggedRtpPacketIncoming& incoming) {
         if (!filter_.test(incoming.rtp.header.payloadType) &&
             incoming.log_time_us() < first_log_end_time_us) {
           rtp_packets_.emplace_back(absl::make_unique<Packet>(
               incoming.rtp.header, incoming.rtp.total_length,
               incoming.rtp.total_length - incoming.rtp.header_length,
               static_cast<double>(incoming.log_time_ms())));
+          packet_ssrcs.insert(rtp_packets_.back()->header().ssrc);
         }
       };
 
+  std::set<uint32_t> ignored_ssrcs;
   auto handle_audio_playout =
-      [this, first_log_end_time_us](
-          const webrtc::LoggedAudioPlayoutEvent& audio_playout) {
+      [this, first_log_end_time_us, &packet_ssrcs,
+       &ignored_ssrcs](const webrtc::LoggedAudioPlayoutEvent& audio_playout) {
         if (audio_playout.log_time_us() < first_log_end_time_us) {
-          audio_outputs_.emplace_back(audio_playout.log_time_ms());
+          if (packet_ssrcs.count(audio_playout.ssrc) > 0) {
+            audio_outputs_.emplace_back(audio_playout.log_time_ms());
+          } else {
+            ignored_ssrcs.insert(audio_playout.ssrc);
+          }
         }
       };
 
@@ -121,6 +129,12 @@
   // Fills in rtp_packets_ and audio_outputs_.
   event_processor.ProcessEventsInOrder();
 
+  for (const auto& ssrc : ignored_ssrcs) {
+    std::cout << "Ignoring GetAudio events from SSRC 0x" << std::hex << ssrc
+              << " because no packets were found with a matching SSRC."
+              << std::endl;
+  }
+
   return true;
 }