Fixing config for Audio BWE.

webrtc/call/rampup_tests.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 194 matching lines @@
     std::vector<AudioReceiveStream::Config>* receive_configs) {
   if (num_audio_streams_ == 0)
     return;
 
   EXPECT_NE(RtpExtension::kTimestampOffsetUri, extension_type_)
       << "Audio BWE not supported with toffset.";
 
   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) {
     transport_cc = false;
     send_config->rtp.extensions.push_back(
         RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId));
   } else if (extension_type_ == RtpExtension::kTransportSequenceNumberUri) {
     transport_cc = true;
     send_config->rtp.extensions.push_back(RtpExtension(
         extension_type_.c_str(), kTransportSequenceNumberExtensionId));
@@ skipping 393 matching lines @@
                     true);
   RunBaseTest(&test);
 }
 
 TEST_F(RampUpTest, TransportSequenceNumberSingleStreamWithHighStartBitrate) {
   RampUpTester test(1, 0, 0.9 * kSingleStreamTargetBps,
                     RtpExtension::kTransportSequenceNumberUri, false, false);
   RunBaseTest(&test);
 }
 }  // namespace webrtc