Resolve dependency between rtc_event_log_api and remote_bitrate_estimator

webrtc/logging/rtc_event_log/rtc_event_log_parser.cc

 /*
  *  Copyright (c) 2016 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/logging/rtc_event_log/rtc_event_log_parser.h"
 
 #include <stdint.h>
 #include <string.h>
 
 #include <algorithm>
 #include <fstream>
 #include <istream>
 #include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/protobuf_utils.h"
 #include "webrtc/call/call.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
+#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
 namespace {
 MediaType GetRuntimeMediaType(rtclog::MediaType media_type) {
   switch (media_type) {
     case rtclog::MediaType::ANY:
       return MediaType::ANY;
     case rtclog::MediaType::AUDIO:
@@ skipping 53 matching lines @@
       return ParsedRtcEventLog::EventType::BWE_PROBE_RESULT_EVENT;
   }
   RTC_NOTREACHED();
   return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
 }
 
 BandwidthUsage GetRuntimeDetectorState(
     rtclog::DelayBasedBweUpdate::DetectorState detector_state) {
   switch (detector_state) {
     case rtclog::DelayBasedBweUpdate::BWE_NORMAL:
-      return kBwNormal;
+      return BandwidthUsage::kBwNormal;
     case rtclog::DelayBasedBweUpdate::BWE_UNDERUSING:
-      return kBwUnderusing;
+      return BandwidthUsage::kBwUnderusing;
     case rtclog::DelayBasedBweUpdate::BWE_OVERUSING:
-      return kBwOverusing;
+      return BandwidthUsage::kBwOverusing;
   }
   RTC_NOTREACHED();
-  return kBwNormal;
+  return BandwidthUsage::kBwNormal;
 }
 
 std::pair<uint64_t, bool> ParseVarInt(std::istream& stream) {
   uint64_t varint = 0;
   for (size_t bytes_read = 0; bytes_read < 10; ++bytes_read) {
     // The most significant bit of each byte is 0 if it is the last byte in
     // the varint and 1 otherwise. Thus, we take the 7 least significant bits
     // of each byte and shift them 7 bits for each byte read previously to get
     // the (unsigned) integer.
     int byte = stream.get();
@@ skipping 466 matching lines @@
         rtc::Optional<ProbeFailureReason>(kInvalidSendReceiveRatio);
   } else if (pr_event.result() == rtclog::BweProbeResult::TIMEOUT) {
     res.failure_reason = rtc::Optional<ProbeFailureReason>(kTimeout);
   } else {
     RTC_NOTREACHED();
   }
 
   return res;
 }
 }  // namespace webrtc