| 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;
|
|
|