Split webrtc/video into webrtc/{audio,call,video}.

webrtc/video/rtc_event_log.h

Index: webrtc/video/rtc_event_log.h
diff --git a/webrtc/video/rtc_event_log.h b/webrtc/video/rtc_event_log.h
deleted file mode 100644
index daab3c116ff0944d0f99723b946ee2fd2542a694..0000000000000000000000000000000000000000
--- a/webrtc/video/rtc_event_log.h
+++ /dev/null
@@ -1,78 +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_VIDEO_RTC_EVENT_LOG_H_
-#define WEBRTC_VIDEO_RTC_EVENT_LOG_H_
-
-#include <string>
-
-#include "webrtc/base/scoped_ptr.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 RtcEventLogImpl;
-
-enum class MediaType;
-
-class RtcEventLog {
- public:
-  virtual ~RtcEventLog() {}
-
-  static rtc::scoped_ptr<RtcEventLog> Create();
-
-  // Starts logging for the specified duration to the specified file.
-  // The logging will stop automatically after the specified duration.
-  // If the file already exists it will be overwritten.
-  // If the file cannot be opened, the RtcEventLog will not start logging.
-  virtual void StartLogging(const std::string& file_name, int duration_ms) = 0;
-
-  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(bool incoming,
-                            MediaType media_type,
-                            const uint8_t* header,
-                            size_t packet_length) = 0;
-
-  // Logs an incoming or outgoing RTCP packet.
-  virtual void LogRtcpPacket(bool incoming,
-                             MediaType media_type,
-                             const uint8_t* packet,
-                             size_t length) = 0;
-
-  // Logs an audio playout event
-  virtual void LogAudioPlayout(uint32_t ssrc) = 0;
-
-  // Reads an RtcEventLog file and returns true when reading was successful.
-  // The result is stored in the given EventStream object.
-  static bool ParseRtcEventLog(const std::string& file_name,
-                               rtclog::EventStream* result);
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_VIDEO_RTC_EVENT_LOG_H_