Removed RtcEventLog deps to call:call_interfaces.

webrtc/logging/rtc_event_log/rtc_event_log.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/logging/rtc_event_log/rtc_event_log.h"
 
 #include <limits>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/protobuf_utils.h"
 #include "webrtc/base/swap_queue.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/timeutils.h"
-#include "webrtc/call/call.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.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"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
@@ skipping 25 matching lines @@
   bool StartLogging(const std::string& file_name,
                     int64_t max_size_bytes) override;
   bool StartLogging(rtc::PlatformFile platform_file,
                     int64_t max_size_bytes) override;
   void StopLogging() override;
   void LogVideoReceiveStreamConfig(const rtclog::StreamConfig& config) override;
   void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) override;
   void LogAudioReceiveStreamConfig(const rtclog::StreamConfig& config) override;
   void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override;
   void LogRtpHeader(PacketDirection direction,
-                    MediaType media_type,
                     const uint8_t* header,
                     size_t packet_length) override;
   void LogRtpHeader(PacketDirection direction,
-                    MediaType media_type,
                     const uint8_t* header,
                     size_t packet_length,
                     int probe_cluster_id) override;
   void LogRtcpPacket(PacketDirection direction,
-                     MediaType media_type,
                      const uint8_t* packet,
                      size_t length) override;
   void LogAudioPlayout(uint32_t ssrc) override;
   void LogLossBasedBweUpdate(int32_t bitrate_bps,
                              uint8_t fraction_loss,
                              int32_t total_packets) override;
   void LogDelayBasedBweUpdate(int32_t bitrate_bps,
                               BandwidthUsage detector_state) override;
   void LogAudioNetworkAdaptation(
       const AudioEncoderRuntimeConfig& config) override;
@@ skipping 35 matching lines @@
     case RtcpMode::kReducedSize:
       return rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE;
     case RtcpMode::kOff:
       RTC_NOTREACHED();
       return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
   }
   RTC_NOTREACHED();
   return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
 }
 
-rtclog::MediaType ConvertMediaType(MediaType media_type) {
-  switch (media_type) {
-    case MediaType::ANY:
-      return rtclog::MediaType::ANY;
-    case MediaType::AUDIO:
-      return rtclog::MediaType::AUDIO;
-    case MediaType::VIDEO:
-      return rtclog::MediaType::VIDEO;
-    case MediaType::DATA:
-      return rtclog::MediaType::DATA;
-  }
-  RTC_NOTREACHED();
-  return rtclog::ANY;
-}
-
 rtclog::DelayBasedBweUpdate::DetectorState ConvertDetectorState(
     BandwidthUsage state) {
   switch (state) {
     case BandwidthUsage::kBwNormal:
       return rtclog::DelayBasedBweUpdate::BWE_NORMAL;
     case BandwidthUsage::kBwUnderusing:
       return rtclog::DelayBasedBweUpdate::BWE_UNDERUSING;
     case BandwidthUsage::kBwOverusing:
       return rtclog::DelayBasedBweUpdate::BWE_OVERUSING;
   }
@@ skipping 223 matching lines @@
     rtclog::RtpHeaderExtension* extension =
         sender_config->add_header_extensions();
     extension->set_name(e.uri);
     extension->set_id(e.id);
   }
 
   StoreEvent(&event);
 }
 
 void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
-                                   MediaType media_type,
                                    const uint8_t* header,
                                    size_t packet_length) {
-  LogRtpHeader(direction, media_type, header, packet_length,
-               PacedPacketInfo::kNotAProbe);
+  LogRtpHeader(direction, header, packet_length, PacedPacketInfo::kNotAProbe);
 }
 
 void RtcEventLogImpl::LogRtpHeader(PacketDirection direction,
-                                   MediaType media_type,
                                    const uint8_t* header,
                                    size_t packet_length,
                                    int probe_cluster_id) {
   // Read header length (in bytes) from packet data.
   if (packet_length < 12u) {
     return;  // Don't read outside the packet.
   }
   const bool x = (header[0] & 0x10) != 0;
   const uint8_t cc = header[0] & 0x0f;
   size_t header_length = 12u + cc * 4u;
 
   if (x) {
     if (packet_length < 12u + cc * 4u + 4u) {
       return;  // Don't read outside the packet.
     }
     size_t x_len = ByteReader<uint16_t>::ReadBigEndian(header + 14 + cc * 4);
     header_length += (x_len + 1) * 4;
   }
 
   std::unique_ptr<rtclog::Event> rtp_event(new rtclog::Event());
   rtp_event->set_timestamp_us(rtc::TimeMicros());
   rtp_event->set_type(rtclog::Event::RTP_EVENT);
   rtp_event->mutable_rtp_packet()->set_incoming(direction == kIncomingPacket);
-  rtp_event->mutable_rtp_packet()->set_type(ConvertMediaType(media_type));
   rtp_event->mutable_rtp_packet()->set_packet_length(packet_length);
   rtp_event->mutable_rtp_packet()->set_header(header, header_length);
   if (probe_cluster_id != PacedPacketInfo::kNotAProbe)
     rtp_event->mutable_rtp_packet()->set_probe_cluster_id(probe_cluster_id);
   StoreEvent(&rtp_event);
 }
 
 void RtcEventLogImpl::LogRtcpPacket(PacketDirection direction,
-                                    MediaType media_type,
                                     const uint8_t* packet,
                                     size_t length) {
   std::unique_ptr<rtclog::Event> rtcp_event(new rtclog::Event());
   rtcp_event->set_timestamp_us(rtc::TimeMicros());
   rtcp_event->set_type(rtclog::Event::RTCP_EVENT);
   rtcp_event->mutable_rtcp_packet()->set_incoming(direction == kIncomingPacket);
-  rtcp_event->mutable_rtcp_packet()->set_type(ConvertMediaType(media_type));
 
   rtcp::CommonHeader header;
   const uint8_t* block_begin = packet;
   const uint8_t* packet_end = packet + length;
   RTC_DCHECK(length <= IP_PACKET_SIZE);
   uint8_t buffer[IP_PACKET_SIZE];
   uint32_t buffer_length = 0;
   while (block_begin < packet_end) {
     if (!header.Parse(block_begin, packet_end - block_begin)) {
       break;  // Incorrect message header.
@@ skipping 169 matching lines @@
 #else
   return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
 #endif  // ENABLE_RTC_EVENT_LOG
 }
 
 std::unique_ptr<RtcEventLog> RtcEventLog::CreateNull() {
   return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
 }
 
 }  // namespace webrtc