Moved RtcEventLog files from call/ to logging/ (new top level dir)

webrtc/call/rtc_event_log.h

Index: webrtc/call/rtc_event_log.h
diff --git a/webrtc/call/rtc_event_log.h b/webrtc/call/rtc_event_log.h
deleted file mode 100644
index a3359692eb8488dd2796929aaa278476156ee563..0000000000000000000000000000000000000000
--- a/webrtc/call/rtc_event_log.h
+++ /dev/null
@@ -1,142 +0,0 @@
-/*
- *  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.
- */
-
-#ifndef WEBRTC_CALL_RTC_EVENT_LOG_H_
-#define WEBRTC_CALL_RTC_EVENT_LOG_H_
-
-#include <memory>
-#include <string>
-
-#include "webrtc/base/platform_file.h"
-#include "webrtc/video_receive_stream.h"
-#include "webrtc/video_send_stream.h"
-
-namespace webrtc {
-
-// Forward declaration of storage class that is automatically generated from
-// the protobuf file.
-namespace rtclog {
-class EventStream;
-}  // namespace rtclog
-
-class Clock;
-class RtcEventLogImpl;
-
-enum class MediaType;
-
-enum PacketDirection { kIncomingPacket = 0, kOutgoingPacket };
-
-class RtcEventLog {
- public:
-  virtual ~RtcEventLog() {}
-
-  // Factory method to create an RtcEventLog object.
-  static std::unique_ptr<RtcEventLog> Create(const Clock* clock);
-
-  // Create an RtcEventLog object that does nothing.
-  static std::unique_ptr<RtcEventLog> CreateNull();
-
-  // Starts logging a maximum of max_size_bytes bytes to the specified file.
-  // If the file already exists it will be overwritten.
-  // If max_size_bytes <= 0, logging will be active until StopLogging is called.
-  // The function has no effect and returns false if we can't start a new log
-  // e.g. because we are already logging or the file cannot be opened.
-  virtual bool StartLogging(const std::string& file_name,
-                            int64_t max_size_bytes) = 0;
-
-  // Same as above. The RtcEventLog takes ownership of the file if the call
-  // is successful, i.e. if it returns true.
-  virtual bool StartLogging(rtc::PlatformFile platform_file,
-                            int64_t max_size_bytes) = 0;
-
-  // Deprecated. Pass an explicit file size limit.
-  bool StartLogging(const std::string& file_name) {
-    return StartLogging(file_name, 10000000);
-  }
-
-  // Deprecated. Pass an explicit file size limit.
-  bool StartLogging(rtc::PlatformFile platform_file) {
-    return StartLogging(platform_file, 10000000);
-  }
-
-  // Stops logging to file and waits until the thread has finished.
-  virtual void StopLogging() = 0;
-
-  // Logs configuration information for webrtc::VideoReceiveStream.
-  virtual void LogVideoReceiveStreamConfig(
-      const webrtc::VideoReceiveStream::Config& config) = 0;
-
-  // Logs configuration information for webrtc::VideoSendStream.
-  virtual void LogVideoSendStreamConfig(
-      const webrtc::VideoSendStream::Config& config) = 0;
-
-  // Logs the header of an incoming or outgoing RTP packet. packet_length
-  // is the total length of the packet, including both header and payload.
-  virtual void LogRtpHeader(PacketDirection direction,
-                            MediaType media_type,
-                            const uint8_t* header,
-                            size_t packet_length) = 0;
-
-  // Logs an incoming or outgoing RTCP packet.
-  virtual void LogRtcpPacket(PacketDirection direction,
-                             MediaType media_type,
-                             const uint8_t* packet,
-                             size_t length) = 0;
-
-  // Logs an audio playout event.
-  virtual void LogAudioPlayout(uint32_t ssrc) = 0;
-
-  // Logs a bitrate update from the bandwidth estimator based on packet loss.
-  virtual void LogBwePacketLossEvent(int32_t bitrate,
-                                     uint8_t fraction_loss,
-                                     int32_t total_packets) = 0;
-
-  // Reads an RtcEventLog file and returns true when reading was successful.
-  // The result is stored in the given EventStream object.
-  // The order of the events in the EventStream is implementation defined.
-  // The current implementation writes a LOG_START event, then the old
-  // configurations, then the remaining events in timestamp order and finally
-  // a LOG_END event. However, this might change without further notice.
-  // TODO(terelius): Change result type to a vector?
-  static bool ParseRtcEventLog(const std::string& file_name,
-                               rtclog::EventStream* result);
-};
-
-// No-op implementation is used if flag is not set, or in tests.
-class RtcEventLogNullImpl final : public RtcEventLog {
- public:
-  bool StartLogging(const std::string& file_name,
-                    int64_t max_size_bytes) override {
-    return false;
-  }
-  bool StartLogging(rtc::PlatformFile platform_file,
-                    int64_t max_size_bytes) override;
-  void StopLogging() override {}
-  void LogVideoReceiveStreamConfig(
-      const VideoReceiveStream::Config& config) override {}
-  void LogVideoSendStreamConfig(
-      const VideoSendStream::Config& config) override {}
-  void LogRtpHeader(PacketDirection direction,
-                    MediaType media_type,
-                    const uint8_t* header,
-                    size_t packet_length) override {}
-  void LogRtcpPacket(PacketDirection direction,
-                     MediaType media_type,
-                     const uint8_t* packet,
-                     size_t length) override {}
-  void LogAudioPlayout(uint32_t ssrc) override {}
-  void LogBwePacketLossEvent(int32_t bitrate,
-                             uint8_t fraction_loss,
-                             int32_t total_packets) override {}
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_CALL_RTC_EVENT_LOG_H_