Index: webrtc/call/rampup_tests.cc |
diff --git a/webrtc/call/rampup_tests.cc b/webrtc/call/rampup_tests.cc |
index df64f41051b57dfccf91cf2f06aa6536d77b6196..3cdcc991d2517265121b4474e8f24deb7fdcaea6 100644 |
--- a/webrtc/call/rampup_tests.cc |
+++ b/webrtc/call/rampup_tests.cc |
@@ -212,8 +212,8 @@ void RampUpTester::ModifyAudioConfigs( |
send_config->rtp.ssrc = audio_ssrcs_[0]; |
send_config->rtp.extensions.clear(); |
- send_config->min_bitrate_kbps = 6; |
- send_config->max_bitrate_kbps = 60; |
+ send_config->min_bitrate_bps = 6000; |
+ send_config->max_bitrate_bps = 60000; |
bool transport_cc = false; |
if (extension_type_ == RtpExtension::kAbsSendTimeUri) { |