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 28 matching lines @@
 #include "webrtc/voice_engine/include/voe_base.h"
 
 namespace {
 
 constexpr int kSendStatsPollingIntervalMs = 1000;
 constexpr int kPayloadTypeH264 = 122;
 constexpr int kPayloadTypeVP8 = 123;
 constexpr int kPayloadTypeVP9 = 124;
 constexpr size_t kMaxComparisons = 10;
 constexpr char kSyncGroup[] = "av_sync";
-constexpr int kOpusMinBitrate = 6000;
-constexpr int kOpusBitrateFb = 32000;
+constexpr int kOpusMinBitrateBps = 6000;
+constexpr int kOpusBitrateFbBps = 32000;
 
 struct VoiceEngineState {
   VoiceEngineState()
       : voice_engine(nullptr),
         base(nullptr),
         send_channel_id(-1),
         receive_channel_id(-1) {}
 
   webrtc::VoiceEngine* voice_engine;
   webrtc::VoEBase* base;
@@ skipping 1196 matching lines @@
   audio_send_config_.voe_channel_id = 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 = kOpusMinBitrateBps;
+    audio_send_config_.max_bitrate_bps = kOpusBitrateFbBps;
   }
   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 = receive_channel_id;
@@ skipping 166 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