commit da2bf4e150a1bded53b74340b9dcc879da91b93f
author kwiberg <kwiberg@webrtc.org> Mon Oct 24 13:47:09 2016 -0700
committer Commit bot <commit-bot@chromium.org> Mon Oct 24 20:47:16 2016 +0000
tree a7a3dcaea9126012593fa9179b4588bf5af2d8ca
parent 88b70747451a41a1d0f8dc607c959ce8548d83cb

Stop using old AudioCodingModule::RegisterReceiveCodec overloads

BUG=webrtc:5801

Review-Url: https://codereview.webrtc.org/2388153004
Cr-Commit-Position: refs/heads/master@{#14753}

webrtc/modules/audio_coding/test/delay_test.cc

diff --git a/webrtc/modules/audio_coding/test/delay_test.cc b/webrtc/modules/audio_coding/test/delay_test.cc
index 99ec0b8..9adeca1 100644
--- a/webrtc/modules/audio_coding/test/delay_test.cc
+++ b/webrtc/modules/audio_coding/test/delay_test.cc
@@ -17,6 +17,7 @@
 #include "gflags/gflags.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/acm2/acm_common_defs.h"
+#include "webrtc/modules/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
 #include "webrtc/modules/audio_coding/test/Channel.h"
@@ -107,8 +108,9 @@
         continue;
       if (STR_CASE_CMP(my_codec_param.plname, "telephone-event") == 0)
         continue;
-      ASSERT_EQ(0, acm_b_->RegisterReceiveCodec(my_codec_param)) <<
-          "Couldn't register receive codec.\n";
+      ASSERT_EQ(true,
+                acm_b_->RegisterReceiveCodec(my_codec_param.pltype,
+                                             CodecInstToSdp(my_codec_param)));
     }
 
     // Create and connect the channel