Index: webrtc/audio/audio_send_stream_unittest.cc |
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc |
index 3b1b44af34661726bb50323c7d77171e6cd35fad..08ff9a6de9c9551c5b7f36b7e08a187a83acaa24 100644 |
--- a/webrtc/audio/audio_send_stream_unittest.cc |
+++ b/webrtc/audio/audio_send_stream_unittest.cc |
@@ -43,7 +43,7 @@ const int kEchoReturnLossEnhancement = 101; |
const unsigned int kSpeechInputLevel = 96; |
const CallStatistics kCallStats = { |
1345, 1678, 1901, 1234, 112, 13456, 17890, 1567, -1890, -1123}; |
-const CodecInst kCodecInst = {-121, "codec_name_send", 48000, -231, -451, -671}; |
+const CodecInst kCodecInst = {-121, "codec_name_send", 48000, -231, 0, -671}; |
const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354}; |
const int kTelephoneEventPayloadType = 123; |
const uint8_t kTelephoneEventCode = 45; |