logging/rtc_event_log/rtc_event_log_parser.h - src/webrtc - Git at Google

 /*
  *  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 WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
 #define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_

 #include <map>
 #include <string>
 #include <utility>  // pair
 #include <vector>

 #include "webrtc/call/video_receive_stream.h"
 #include "webrtc/call/video_send_stream.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/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 "webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
 #endif
 RTC_POP_IGNORING_WUNDEF()

 namespace webrtc {

 enum class MediaType;

 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;
   };

   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
   };

   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) 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;

  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  // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
	/*
	* 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 WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
	#define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_

	#include <map>
	#include <string>
	#include <utility> // pair
	#include <vector>

	#include "webrtc/call/video_receive_stream.h"
	#include "webrtc/call/video_send_stream.h"
	#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
	#include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
	#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
	#include "webrtc/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 "webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
	#endif
	RTC_POP_IGNORING_WUNDEF()

	namespace webrtc {

	enum class MediaType;

	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;
	};

	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
	};

	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) 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;

	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 // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_