Fixing config for Audio BWE.

webrtc/audio/audio_send_stream.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.
  */
 
@@ skipping 83 matching lines @@
 AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
   channel_proxy_->DeRegisterExternalTransport();
   channel_proxy_->ResetCongestionControlObjects();
   channel_proxy_->SetRtcEventLog(nullptr);
 }
 
 void AudioSendStream::Start() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  if (config_.min_bitrate_kbps != -1 && config_.max_bitrate_kbps != -1) {
-    RTC_DCHECK_GE(config_.max_bitrate_kbps, config_.min_bitrate_kbps);
+  if (config_.min_bitrate_bps != -1 && config_.max_bitrate_bps != -1) {
+    RTC_DCHECK_GE(config_.max_bitrate_bps, config_.min_bitrate_bps);
     rtc::Event thread_sync_event(false /* manual_reset */, false);
     worker_queue_->PostTask([this, &thread_sync_event] {
-      bitrate_allocator_->AddObserver(this, config_.min_bitrate_kbps * 1000,
-                                      config_.max_bitrate_kbps * 1000, 0, true);
+      bitrate_allocator_->AddObserver(this, config_.min_bitrate_bps,
+                                      config_.max_bitrate_bps, 0, true);
       thread_sync_event.Set();
     });
     thread_sync_event.Wait(rtc::Event::kForever);
   }
 
   ScopedVoEInterface<VoEBase> base(voice_engine());
   int error = base->StartSend(config_.voe_channel_id);
   if (error != 0) {
     LOG(LS_ERROR) << "AudioSendStream::Start failed with error: " << error;
   }
@@ skipping 122 matching lines @@
   // calls on the worker thread. We should move towards always using a network
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!thread_checker_.CalledOnValidThread());
   return channel_proxy_->ReceivedRTCPPacket(packet, length);
 }
 
 uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
                                            uint8_t fraction_loss,
                                            int64_t rtt) {
   RTC_DCHECK_GE(bitrate_bps,
-                static_cast<uint32_t>(config_.min_bitrate_kbps * 1000));
+                static_cast<uint32_t>(config_.min_bitrate_bps));
   // The bitrate allocator might allocate an higher than max configured bitrate
   // if there is room, to allow for, as example, extra FEC. Ignore that for now.
-  const uint32_t max_bitrate_bps = config_.max_bitrate_kbps * 1000;
+  const uint32_t max_bitrate_bps = config_.max_bitrate_bps;
   if (bitrate_bps > max_bitrate_bps)
     bitrate_bps = max_bitrate_bps;
 
   channel_proxy_->SetBitrate(bitrate_bps);
 
   // The amount of audio protection is not exposed by the encoder, hence
   // always returning 0.
   return 0;
 }
 
@@ skipping 120 matching lines @@
         LOG(LS_WARNING) << "SetVADStatus() failed: " << base->LastError();
         return false;
       }
     }
   }
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc