Renaming probing_interval to bwe_period globally.

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 287 matching lines @@
   // TODO(solenberg): Tests call this function on a network thread, libjingle
   // calls on the worker thread. We should move towards always using a network
   // thread. Then this check can be enabled.
   // RTC_DCHECK(!worker_thread_checker_.CalledOnValidThread());
   return channel_proxy_->ReceivedRTCPPacket(packet, length);
 }
 
 uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
                                            uint8_t fraction_loss,
                                            int64_t rtt,
-                                           int64_t probing_interval_ms) {
+                                           int64_t bwe_period_ms) {
   // A send stream may be allocated a bitrate of zero if the allocator decides
   // to disable it. For now we ignore this decision and keep sending on min
   // bitrate.
   if (bitrate_bps == 0) {
     bitrate_bps = config_.min_bitrate_bps;
   }
   RTC_DCHECK_GE(bitrate_bps,
                 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_bps;
   if (bitrate_bps > max_bitrate_bps)
     bitrate_bps = max_bitrate_bps;
 
-  channel_proxy_->SetBitrate(bitrate_bps, probing_interval_ms);
+  channel_proxy_->SetBitrate(bitrate_bps, bwe_period_ms);
 
   // The amount of audio protection is not exposed by the encoder, hence
   // always returning 0.
   return 0;
 }
 
 void AudioSendStream::OnPacketAdded(uint32_t ssrc, uint16_t seq_num) {
   RTC_DCHECK(pacer_thread_checker_.CalledOnValidThread());
   // Only packets that belong to this stream are of interest.
   if (ssrc == config_.rtp.ssrc) {
@@ skipping 263 matching lines @@
     if (rtpRtcpModule->RegisterSendPayload(codec) != 0) {
       LOG(LS_ERROR) << "RegisterCngPayloadType() failed to register CN to "
                        "RTP/RTCP module";
     }
   }
 }
 
 
 }  // namespace internal
 }  // namespace webrtc