commit c856dc2b6ba89ac6760524dd10f69a27199b8dc1
author Karl Wiberg <kwiberg@webrtc.org> Thu Sep 28 20:13:59 2017 +0200
committer Commit Bot <commit-bot@chromium.org> Thu Sep 28 23:23:07 2017 +0000
tree 4cb38d991d4f8e0375237e59105a9cb6cb15ee6d
parent a6368d17c5ec52355f332d869996a1946535826f

Convert PayloadUnion from a union to a class, step 2

Stop using PayloadUnion's public member variables, since a future CL
will make them private.

BUG=webrtc:8159

Change-Id: Ia3dada56be7ef00ed80f3733209b18c178a36561
Reviewed-on: https://webrtc-review.googlesource.com/4380
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20027}

modules/rtp_rtcp/source/rtp_receiver_impl.cc

diff --git a/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/modules/rtp_rtcp/source/rtp_receiver_impl.cc
index c200c1f..8a8669e 100644
--- a/modules/rtp_rtcp/source/rtp_receiver_impl.cc
+++ b/modules/rtp_rtcp/source/rtp_receiver_impl.cc
@@ -279,9 +279,9 @@
           }
           payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
           strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
-          if (payload->audio) {
-            channels = payload->typeSpecific.Audio.channels;
-            rate = payload->typeSpecific.Audio.rate;
+          if (payload->typeSpecific.is_audio()) {
+            channels = payload->typeSpecific.audio_payload().channels;
+            rate = payload->typeSpecific.audio_payload().rate;
           }
         }
       }
@@ -376,7 +376,7 @@
       rtp_media_receiver_->SetLastMediaSpecificPayload(payload->typeSpecific);
       rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
 
-      if (!payload->audio) {
+      if (!payload->typeSpecific.is_audio()) {
         bool media_type_unchanged =
             rtp_payload_registry_->ReportMediaPayloadType(payload_type);
         if (media_type_unchanged) {