blob: 1e4650c054fd9e7c11a170ce90c21a11af588e5d [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.
*/
#ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
#define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
#include <memory>
#include <string>
#include <vector>
#include "webrtc/api/rtpparameters.h"
#include "webrtc/common_types.h"
#include "webrtc/rtc_base/platform_file.h"
namespace webrtc {
// Forward declaration of storage class that is automatically generated from
// the protobuf file.
namespace rtclog {
class EventStream;
struct StreamConfig {
uint32_t local_ssrc = 0;
uint32_t remote_ssrc = 0;
uint32_t rtx_ssrc = 0;
std::string rsid;
bool remb = false;
std::vector<RtpExtension> rtp_extensions;
RtcpMode rtcp_mode = RtcpMode::kReducedSize;
struct Codec {
Codec(const std::string& payload_name,
int payload_type,
int rtx_payload_type)
: payload_name(payload_name),
payload_type(payload_type),
rtx_payload_type(rtx_payload_type) {}
std::string payload_name;
int payload_type;
int rtx_payload_type;
};
std::vector<Codec> codecs;
};
} // namespace rtclog
class Clock;
struct AudioEncoderRuntimeConfig;
enum class MediaType;
enum class BandwidthUsage;
enum PacketDirection { kIncomingPacket = 0, kOutgoingPacket };
enum ProbeFailureReason {
kInvalidSendReceiveInterval,
kInvalidSendReceiveRatio,
kTimeout
};
class RtcEventLog {
public:
virtual ~RtcEventLog() {}
// Factory method to create an RtcEventLog object.
static std::unique_ptr<RtcEventLog> Create();
// TODO(nisse): webrtc::Clock is deprecated. Delete this method and
// above forward declaration of Clock when
// webrtc/system_wrappers/include/clock.h is deleted.
static std::unique_ptr<RtcEventLog> Create(const Clock* clock) {
return Create();
}
// 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.
RTC_DEPRECATED bool StartLogging(const std::string& file_name) {
return StartLogging(file_name, 10000000);
}
// Deprecated. Pass an explicit file size limit.
RTC_DEPRECATED bool StartLogging(rtc::PlatformFile platform_file) {
return StartLogging(platform_file, 10000000);
}
// Stops logging to file and waits until the file has been closed, after
// which it would be permissible to read and/or modify it.
virtual void StopLogging() = 0;
// Logs configuration information for a video receive stream.
virtual void LogVideoReceiveStreamConfig(
const rtclog::StreamConfig& config) = 0;
// Logs configuration information for a video send stream.
virtual void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) = 0;
// Logs configuration information for an audio receive stream.
virtual void LogAudioReceiveStreamConfig(
const rtclog::StreamConfig& config) = 0;
// Logs configuration information for an audio send stream.
virtual void LogAudioSendStreamConfig(const rtclog::StreamConfig& 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,
const uint8_t* header,
size_t packet_length) = 0;
// Same as above but used on the sender side to log packets that are part of
// a probe cluster.
virtual void LogRtpHeader(PacketDirection direction,
const uint8_t* header,
size_t packet_length,
int probe_cluster_id) = 0;
// Logs an incoming or outgoing RTCP packet.
virtual void LogRtcpPacket(PacketDirection direction,
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 LogLossBasedBweUpdate(int32_t bitrate_bps,
uint8_t fraction_loss,
int32_t total_packets) = 0;
// Logs a bitrate update from the bandwidth estimator based on delay changes.
virtual void LogDelayBasedBweUpdate(int32_t bitrate_bps,
BandwidthUsage detector_state) = 0;
// Logs audio encoder re-configuration driven by audio network adaptor.
virtual void LogAudioNetworkAdaptation(
const AudioEncoderRuntimeConfig& config) = 0;
// Logs when a probe cluster is created.
virtual void LogProbeClusterCreated(int id,
int bitrate_bps,
int min_probes,
int min_bytes) = 0;
// Logs the result of a successful probing attempt.
virtual void LogProbeResultSuccess(int id, int bitrate_bps) = 0;
// Logs the result of an unsuccessful probing attempt.
virtual void LogProbeResultFailure(int id,
ProbeFailureReason failure_reason) = 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 : 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 {
return false;
}
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,
const uint8_t* header,
size_t packet_length) override {}
void LogRtpHeader(PacketDirection direction,
const uint8_t* header,
size_t packet_length,
int probe_cluster_id) override {}
void LogRtcpPacket(PacketDirection direction,
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 {}
void LogProbeClusterCreated(int id,
int bitrate_bps,
int min_probes,
int min_bytes) override{};
void LogProbeResultSuccess(int id, int bitrate_bps) override{};
void LogProbeResultFailure(int id,
ProbeFailureReason failure_reason) override{};
};
} // namespace webrtc
#endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_