blob: 643891877486c5fadb673a58bd79a1a4d23c91fb [file] [log] [blame]
/*
* Copyright (c) 2016 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 LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
#define LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
#include <map>
#include <string>
#include <utility> // pair
#include <vector>
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/rtc_stream_config.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "rtc_base/ignore_wundef.h"
// Files generated at build-time by the protobuf compiler.
RTC_PUSH_IGNORING_WUNDEF()
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
#else
#include "logging/rtc_event_log/rtc_event_log.pb.h"
#endif
RTC_POP_IGNORING_WUNDEF()
namespace webrtc {
enum class BandwidthUsage;
enum class MediaType;
struct AudioEncoderRuntimeConfig;
class ParsedRtcEventLog {
friend class RtcEventLogTestHelper;
public:
struct BweProbeClusterCreatedEvent {
uint64_t timestamp;
uint32_t id;
uint64_t bitrate_bps;
uint32_t min_packets;
uint32_t min_bytes;
};
struct BweProbeResultEvent {
uint64_t timestamp;
uint32_t id;
rtc::Optional<uint64_t> bitrate_bps;
rtc::Optional<ProbeFailureReason> failure_reason;
};
struct BweDelayBasedUpdate {
uint64_t timestamp;
int32_t bitrate_bps;
BandwidthUsage detector_state;
};
struct AlrStateEvent {
uint64_t timestamp;
bool in_alr;
};
enum EventType {
UNKNOWN_EVENT = 0,
LOG_START = 1,
LOG_END = 2,
RTP_EVENT = 3,
RTCP_EVENT = 4,
AUDIO_PLAYOUT_EVENT = 5,
LOSS_BASED_BWE_UPDATE = 6,
DELAY_BASED_BWE_UPDATE = 7,
VIDEO_RECEIVER_CONFIG_EVENT = 8,
VIDEO_SENDER_CONFIG_EVENT = 9,
AUDIO_RECEIVER_CONFIG_EVENT = 10,
AUDIO_SENDER_CONFIG_EVENT = 11,
AUDIO_NETWORK_ADAPTATION_EVENT = 16,
BWE_PROBE_CLUSTER_CREATED_EVENT = 17,
BWE_PROBE_RESULT_EVENT = 18,
ALR_STATE_EVENT = 19
};
enum class MediaType { ANY, AUDIO, VIDEO, DATA };
// Reads an RtcEventLog file and returns true if parsing was successful.
bool ParseFile(const std::string& file_name);
// Reads an RtcEventLog from a string and returns true if successful.
bool ParseString(const std::string& s);
// Reads an RtcEventLog from an istream and returns true if successful.
bool ParseStream(std::istream& stream);
// Returns the number of events in an EventStream.
size_t GetNumberOfEvents() const;
// Reads the arrival timestamp (in microseconds) from a rtclog::Event.
int64_t GetTimestamp(size_t index) const;
// Reads the event type of the rtclog::Event at |index|.
EventType GetEventType(size_t index) const;
// Reads the header, direction, header length and packet length from the RTP
// event at |index|, and stores the values in the corresponding output
// parameters. Each output parameter can be set to nullptr if that value
// isn't needed.
// NB: The header must have space for at least IP_PACKET_SIZE bytes.
// Returns: a pointer to a header extensions map acquired from parsing
// corresponding Audio/Video Sender/Receiver config events.
// Warning: if the same SSRC is reused by both video and audio streams during
// call, extensions maps may be incorrect (the last one would be returned).
webrtc::RtpHeaderExtensionMap* GetRtpHeader(size_t index,
PacketDirection* incoming,
uint8_t* header,
size_t* header_length,
size_t* total_length,
int* probe_cluster_id) const;
// Reads packet, direction and packet length from the RTCP event at |index|,
// and stores the values in the corresponding output parameters.
// Each output parameter can be set to nullptr if that value isn't needed.
// NB: The packet must have space for at least IP_PACKET_SIZE bytes.
void GetRtcpPacket(size_t index,
PacketDirection* incoming,
uint8_t* packet,
size_t* length) const;
// Reads a video receive config event to a StreamConfig struct.
// Only the fields that are stored in the protobuf will be written.
rtclog::StreamConfig GetVideoReceiveConfig(size_t index) const;
// Reads a video send config event to a StreamConfig struct. If the proto
// contains multiple SSRCs and RTX SSRCs (this used to be the case for
// simulcast streams) then we return one StreamConfig per SSRC,RTX_SSRC pair.
// Only the fields that are stored in the protobuf will be written.
std::vector<rtclog::StreamConfig> GetVideoSendConfig(size_t index) const;
// Reads a audio receive config event to a StreamConfig struct.
// Only the fields that are stored in the protobuf will be written.
rtclog::StreamConfig GetAudioReceiveConfig(size_t index) const;
// Reads a config event to a StreamConfig struct.
// Only the fields that are stored in the protobuf will be written.
rtclog::StreamConfig GetAudioSendConfig(size_t index) const;
// Reads the SSRC from the audio playout event at |index|. The SSRC is stored
// in the output parameter ssrc. The output parameter can be set to nullptr
// and in that case the function only asserts that the event is well formed.
void GetAudioPlayout(size_t index, uint32_t* ssrc) const;
// Reads bitrate, fraction loss (as defined in RFC 1889) and total number of
// expected packets from the loss based BWE event at |index| and stores the
// values in
// the corresponding output parameters. Each output parameter can be set to
// nullptr if that
// value isn't needed.
void GetLossBasedBweUpdate(size_t index,
int32_t* bitrate_bps,
uint8_t* fraction_loss,
int32_t* total_packets) const;
// Reads bitrate and detector_state from the delay based BWE event at |index|
// and stores the values in the corresponding output parameters. Each output
// parameter can be set to nullptr if that
// value isn't needed.
BweDelayBasedUpdate GetDelayBasedBweUpdate(size_t index) const;
// Reads a audio network adaptation event to a (non-NULL)
// AudioEncoderRuntimeConfig struct. Only the fields that are
// stored in the protobuf will be written.
void GetAudioNetworkAdaptation(size_t index,
AudioEncoderRuntimeConfig* config) const;
BweProbeClusterCreatedEvent GetBweProbeClusterCreated(size_t index) const;
BweProbeResultEvent GetBweProbeResult(size_t index) const;
MediaType GetMediaType(uint32_t ssrc, PacketDirection direction) const;
AlrStateEvent GetAlrState(size_t index) const;
private:
rtclog::StreamConfig GetVideoReceiveConfig(const rtclog::Event& event) const;
std::vector<rtclog::StreamConfig> GetVideoSendConfig(
const rtclog::Event& event) const;
rtclog::StreamConfig GetAudioReceiveConfig(const rtclog::Event& event) const;
rtclog::StreamConfig GetAudioSendConfig(const rtclog::Event& event) const;
std::vector<rtclog::Event> events_;
struct Stream {
Stream(uint32_t ssrc,
MediaType media_type,
webrtc::PacketDirection direction,
webrtc::RtpHeaderExtensionMap map)
: ssrc(ssrc),
media_type(media_type),
direction(direction),
rtp_extensions_map(map) {}
uint32_t ssrc;
MediaType media_type;
webrtc::PacketDirection direction;
webrtc::RtpHeaderExtensionMap rtp_extensions_map;
};
// All configured streams found in the event log.
std::vector<Stream> streams_;
// To find configured extensions map for given stream, what are needed to
// parse a header.
typedef std::pair<uint32_t, webrtc::PacketDirection> StreamId;
std::map<StreamId, webrtc::RtpHeaderExtensionMap*> rtp_extensions_maps_;
};
} // namespace webrtc
#endif // LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_