| 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) {
|
|
|