| Index: webrtc/test/call_test.cc
|
| diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
|
| index 68d6857e29d7208d85a8af114227b5fb951b25f9..833e53c4652288b3a268103432c07214876d08d5 100644
|
| --- a/webrtc/test/call_test.cc
|
| +++ b/webrtc/test/call_test.cc
|
| @@ -215,7 +215,7 @@
|
| audio_send_config_.voe_channel_id = voe_send_.channel_id;
|
| audio_send_config_.rtp.ssrc = kAudioSendSsrc;
|
| audio_send_config_.send_codec_spec.codec_inst =
|
| - CodecInst{kAudioSendPayloadType, "OPUS", 48000, 960, 2, 64000};
|
| + CodecInst{kAudioSendPayloadType, "ISAC", 16000, 480, 1, 32000};
|
| }
|
|
|
| // TODO(brandtr): Update this when we support multistream protection.
|
|
|