Fixing config for Audio BWE.

webrtc/video/video_quality_test.cc

 /*
  *  Copyright (c) 2015 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.
  */
 #include "webrtc/video/video_quality_test.h"
@@ skipping 1314 matching lines @@
     audio_send_config_.voe_channel_id = voe.send_channel_id;
     audio_send_config_.rtp.ssrc = kAudioSendSsrc;
 
     // Add extension to enable audio send side BWE, and allow audio bit rate
     // adaptation.
     audio_send_config_.rtp.extensions.clear();
     if (params_.call.send_side_bwe) {
       audio_send_config_.rtp.extensions.push_back(webrtc::RtpExtension(
           webrtc::RtpExtension::kTransportSequenceNumberUri,
           test::kTransportSequenceNumberExtensionId));
-      audio_send_config_.min_bitrate_kbps = kOpusMinBitrate / 1000;
-      audio_send_config_.max_bitrate_kbps = kOpusBitrateFb / 1000;
+      audio_send_config_.min_bitrate_bps = kOpusMinBitrate;
+      audio_send_config_.max_bitrate_bps = kOpusBitrateFb;
     }
     audio_send_config_.send_codec_spec.codec_inst =
         CodecInst{120, "OPUS", 48000, 960, 2, 64000};
 
     audio_send_stream_ = call->CreateAudioSendStream(audio_send_config_);
 
     AudioReceiveStream::Config audio_config;
     audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
     audio_config.rtcp_send_transport = &transport;
     audio_config.voe_channel_id = voe.receive_channel_id;
@@ skipping 76 matching lines @@
     std::ostringstream str;
     str << receive_logs_++;
     std::string path =
         params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
     stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
                                         10000000);
   }
 }
 
 }  // namespace webrtc