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. |