blob: bea57b01cd11eb4729b83f1967cb6258862ac7c2 [file] [log] [blame]
* 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 <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 {
virtual ~RtcEventLog() {}
// Factory method to create an RtcEventLog object.
static std::unique_ptr<RtcEventLog> Create(const Clock* clock);
// 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);
} // namespace webrtc