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/*
* 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 <string>
#include "webrtc/base/platform_file.h"
#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();
// Sets the time that events are stored in the internal event buffer
// before the user calls StartLogging. The default is 10 000 000 us = 10 s
virtual void SetBufferDuration(int64_t buffer_duration_us) = 0;
// 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;
// Starts logging until either the 10 minute timer runs out or the StopLogging
// function is called. The RtcEventLog takes ownership of the supplied
// rtc::PlatformFile.
virtual bool StartLogging(rtc::PlatformFile log_file) = 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_CALL_RTC_EVENT_LOG_H_
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