From 63c38e21dae66e8c708d5a35d3d6a9f0049d34c8 Mon Sep 17 00:00:00 2001
From: Sebastian Jansson <srte@webrtc.org>
Date: Tue, 6 Aug 2019 17:17:43 +0200
Subject: [PATCH] Fix for incorrect transport sequence number config for audio
 in scenario tests.

Bug: webrtc:9883
Change-Id: Iafe1db4b4dbfa81c7901640114057806821de760
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148280
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28778}
---
 test/scenario/audio_stream.cc | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/test/scenario/audio_stream.cc b/test/scenario/audio_stream.cc
index 4e5396e34c..f5d21167ff 100644
--- a/test/scenario/audio_stream.cc
+++ b/test/scenario/audio_stream.cc
@@ -182,8 +182,8 @@ ReceiveAudioStream::ReceiveAudioStream(
   receiver->ssrc_media_types_[recv_config.rtp.remote_ssrc] = MediaType::AUDIO;
   if (config.stream.in_bandwidth_estimation) {
     recv_config.rtp.transport_cc = true;
-    recv_config.rtp.extensions = {
-        {RtpExtension::kTransportSequenceNumberUri, 8}};
+    recv_config.rtp.extensions = {{RtpExtension::kTransportSequenceNumberUri,
+                                   kTransportSequenceNumberExtensionId}};
   }
   receiver_->AddExtensions(recv_config.rtp.extensions);
   recv_config.decoder_factory = decoder_factory;