Removed RtcEventLog deps to call:call_interfaces.

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 <map>
 #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:
-      return MediaType::AUDIO;
-    case rtclog::MediaType::VIDEO:
-      return MediaType::VIDEO;
-    case rtclog::MediaType::DATA:
-      return MediaType::DATA;
-  }
-  RTC_NOTREACHED();
-  return MediaType::ANY;
-}
-
 RtcpMode GetRuntimeRtcpMode(rtclog::VideoReceiveConfig::RtcpMode rtcp_mode) {
   switch (rtcp_mode) {
     case rtclog::VideoReceiveConfig::RTCP_COMPOUND:
       return RtcpMode::kCompound;
     case rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE:
       return RtcpMode::kReducedSize;
   }
   RTC_NOTREACHED();
   return RtcpMode::kOff;
 }
@@ skipping 113 matching lines @@
 
   while (1) {
     // Check whether we have reached end of file.
     stream.peek();
     if (stream.eof()) {
       return true;
     }
 
     // Read the next message tag. The tag number is defined as
     // (fieldnumber << 3) | wire_type. In our case, the field number is
-    // supposed to be 1 and the wire type for an length-delimited field is 2.
+    // supposed to be 1 and the wire type for an
+    // length-deli"../call:call_interfaces",mited field is 2.
     const uint64_t kExpectedTag = (1 << 3) | 2;
     std::tie(tag, success) = ParseVarInt(stream);
     if (!success) {
       LOG(LS_WARNING) << "Missing field tag from beginning of protobuf event.";
       return false;
     } else if (tag != kExpectedTag) {
       LOG(LS_WARNING) << "Unexpected field tag at beginning of protobuf event.";
       return false;
     }
 
@@ skipping 39 matching lines @@
     size_t index) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
   const rtclog::Event& event = events_[index];
   RTC_CHECK(event.has_type());
   return GetRuntimeEventType(event.type());
 }
 
 // The header must have space for at least IP_PACKET_SIZE bytes.
 void ParsedRtcEventLog::GetRtpHeader(size_t index,
                                      PacketDirection* incoming,
-                                     MediaType* media_type,
                                      uint8_t* header,
                                      size_t* header_length,
                                      size_t* total_length) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
   const rtclog::Event& event = events_[index];
   RTC_CHECK(event.has_type());
   RTC_CHECK_EQ(event.type(), rtclog::Event::RTP_EVENT);
   RTC_CHECK(event.has_rtp_packet());
   const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
   // Get direction of packet.
   RTC_CHECK(rtp_packet.has_incoming());
   if (incoming != nullptr) {
     *incoming = rtp_packet.incoming() ? kIncomingPacket : kOutgoingPacket;
   }
-  // Get media type.
-  RTC_CHECK(rtp_packet.has_type());
-  if (media_type != nullptr) {
-    *media_type = GetRuntimeMediaType(rtp_packet.type());
-  }
   // Get packet length.
   RTC_CHECK(rtp_packet.has_packet_length());
   if (total_length != nullptr) {
     *total_length = rtp_packet.packet_length();
   }
   // Get header length.
   RTC_CHECK(rtp_packet.has_header());
   if (header_length != nullptr) {
     *header_length = rtp_packet.header().size();
   }
   // Get header contents.
   if (header != nullptr) {
     const size_t kMinRtpHeaderSize = 12;
     RTC_CHECK_GE(rtp_packet.header().size(), kMinRtpHeaderSize);
     RTC_CHECK_LE(rtp_packet.header().size(),
                  static_cast<size_t>(IP_PACKET_SIZE));
     memcpy(header, rtp_packet.header().data(), rtp_packet.header().size());
   }
 }
 
 // The packet must have space for at least IP_PACKET_SIZE bytes.
 void ParsedRtcEventLog::GetRtcpPacket(size_t index,
                                       PacketDirection* incoming,
-                                      MediaType* media_type,
                                       uint8_t* packet,
                                       size_t* length) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
   const rtclog::Event& event = events_[index];
   RTC_CHECK(event.has_type());
   RTC_CHECK_EQ(event.type(), rtclog::Event::RTCP_EVENT);
   RTC_CHECK(event.has_rtcp_packet());
   const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
   // Get direction of packet.
   RTC_CHECK(rtcp_packet.has_incoming());
   if (incoming != nullptr) {
     *incoming = rtcp_packet.incoming() ? kIncomingPacket : kOutgoingPacket;
   }
-  // Get media type.
-  RTC_CHECK(rtcp_packet.has_type());
-  if (media_type != nullptr) {
-    *media_type = GetRuntimeMediaType(rtcp_packet.type());
-  }
   // Get packet length.
   RTC_CHECK(rtcp_packet.has_packet_data());
   if (length != nullptr) {
     *length = rtcp_packet.packet_data().size();
   }
   // Get packet contents.
   if (packet != nullptr) {
     RTC_CHECK_LE(rtcp_packet.packet_data().size(),
                  static_cast<unsigned>(IP_PACKET_SIZE));
     memcpy(packet, rtcp_packet.packet_data().data(),
@@ skipping 268 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