Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
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logging/rtc_event_log/rtc_event_log_parser.h
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212
logging/rtc_event_log/rtc_event_log_parser.h
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/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
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#define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
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#include <map>
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#include <string>
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#include <utility> // pair
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#include <vector>
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#include "webrtc/call/video_receive_stream.h"
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#include "webrtc/call/video_send_stream.h"
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#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
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#include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
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#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
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#include "webrtc/rtc_base/ignore_wundef.h"
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// Files generated at build-time by the protobuf compiler.
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RTC_PUSH_IGNORING_WUNDEF()
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#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
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#include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
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#else
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#include "webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
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#endif
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RTC_POP_IGNORING_WUNDEF()
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namespace webrtc {
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enum class MediaType;
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class ParsedRtcEventLog {
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friend class RtcEventLogTestHelper;
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public:
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struct BweProbeClusterCreatedEvent {
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uint64_t timestamp;
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uint32_t id;
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uint64_t bitrate_bps;
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uint32_t min_packets;
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uint32_t min_bytes;
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};
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struct BweProbeResultEvent {
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uint64_t timestamp;
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uint32_t id;
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rtc::Optional<uint64_t> bitrate_bps;
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rtc::Optional<ProbeFailureReason> failure_reason;
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};
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struct BweDelayBasedUpdate {
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uint64_t timestamp;
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int32_t bitrate_bps;
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BandwidthUsage detector_state;
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};
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enum EventType {
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UNKNOWN_EVENT = 0,
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LOG_START = 1,
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LOG_END = 2,
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RTP_EVENT = 3,
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RTCP_EVENT = 4,
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AUDIO_PLAYOUT_EVENT = 5,
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LOSS_BASED_BWE_UPDATE = 6,
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DELAY_BASED_BWE_UPDATE = 7,
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VIDEO_RECEIVER_CONFIG_EVENT = 8,
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VIDEO_SENDER_CONFIG_EVENT = 9,
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AUDIO_RECEIVER_CONFIG_EVENT = 10,
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AUDIO_SENDER_CONFIG_EVENT = 11,
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AUDIO_NETWORK_ADAPTATION_EVENT = 16,
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BWE_PROBE_CLUSTER_CREATED_EVENT = 17,
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BWE_PROBE_RESULT_EVENT = 18
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};
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enum class MediaType { ANY, AUDIO, VIDEO, DATA };
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// Reads an RtcEventLog file and returns true if parsing was successful.
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bool ParseFile(const std::string& file_name);
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// Reads an RtcEventLog from a string and returns true if successful.
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bool ParseString(const std::string& s);
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// Reads an RtcEventLog from an istream and returns true if successful.
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bool ParseStream(std::istream& stream);
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// Returns the number of events in an EventStream.
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size_t GetNumberOfEvents() const;
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// Reads the arrival timestamp (in microseconds) from a rtclog::Event.
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int64_t GetTimestamp(size_t index) const;
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// Reads the event type of the rtclog::Event at |index|.
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EventType GetEventType(size_t index) const;
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// Reads the header, direction, header length and packet length from the RTP
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// event at |index|, and stores the values in the corresponding output
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// parameters. Each output parameter can be set to nullptr if that value
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// isn't needed.
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// NB: The header must have space for at least IP_PACKET_SIZE bytes.
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// Returns: a pointer to a header extensions map acquired from parsing
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// corresponding Audio/Video Sender/Receiver config events.
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// Warning: if the same SSRC is reused by both video and audio streams during
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// call, extensions maps may be incorrect (the last one would be returned).
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webrtc::RtpHeaderExtensionMap* GetRtpHeader(size_t index,
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PacketDirection* incoming,
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uint8_t* header,
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size_t* header_length,
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size_t* total_length) const;
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// Reads packet, direction and packet length from the RTCP event at |index|,
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// and stores the values in the corresponding output parameters.
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// Each output parameter can be set to nullptr if that value isn't needed.
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// NB: The packet must have space for at least IP_PACKET_SIZE bytes.
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void GetRtcpPacket(size_t index,
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PacketDirection* incoming,
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uint8_t* packet,
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size_t* length) const;
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// Reads a video receive config event to a StreamConfig struct.
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// Only the fields that are stored in the protobuf will be written.
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rtclog::StreamConfig GetVideoReceiveConfig(size_t index) const;
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// Reads a video send config event to a StreamConfig struct. If the proto
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// contains multiple SSRCs and RTX SSRCs (this used to be the case for
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// simulcast streams) then we return one StreamConfig per SSRC,RTX_SSRC pair.
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// Only the fields that are stored in the protobuf will be written.
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std::vector<rtclog::StreamConfig> GetVideoSendConfig(size_t index) const;
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// Reads a audio receive config event to a StreamConfig struct.
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// Only the fields that are stored in the protobuf will be written.
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rtclog::StreamConfig GetAudioReceiveConfig(size_t index) const;
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// Reads a config event to a StreamConfig struct.
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// Only the fields that are stored in the protobuf will be written.
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rtclog::StreamConfig GetAudioSendConfig(size_t index) const;
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// Reads the SSRC from the audio playout event at |index|. The SSRC is stored
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// in the output parameter ssrc. The output parameter can be set to nullptr
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// and in that case the function only asserts that the event is well formed.
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void GetAudioPlayout(size_t index, uint32_t* ssrc) const;
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// Reads bitrate, fraction loss (as defined in RFC 1889) and total number of
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// expected packets from the loss based BWE event at |index| and stores the
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// values in
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// the corresponding output parameters. Each output parameter can be set to
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// nullptr if that
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// value isn't needed.
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void GetLossBasedBweUpdate(size_t index,
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int32_t* bitrate_bps,
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uint8_t* fraction_loss,
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int32_t* total_packets) const;
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// Reads bitrate and detector_state from the delay based BWE event at |index|
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// and stores the values in the corresponding output parameters. Each output
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// parameter can be set to nullptr if that
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// value isn't needed.
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BweDelayBasedUpdate GetDelayBasedBweUpdate(size_t index) const;
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// Reads a audio network adaptation event to a (non-NULL)
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// AudioEncoderRuntimeConfig struct. Only the fields that are
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// stored in the protobuf will be written.
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void GetAudioNetworkAdaptation(size_t index,
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AudioEncoderRuntimeConfig* config) const;
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BweProbeClusterCreatedEvent GetBweProbeClusterCreated(size_t index) const;
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BweProbeResultEvent GetBweProbeResult(size_t index) const;
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MediaType GetMediaType(uint32_t ssrc, PacketDirection direction) const;
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private:
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rtclog::StreamConfig GetVideoReceiveConfig(const rtclog::Event& event) const;
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std::vector<rtclog::StreamConfig> GetVideoSendConfig(
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const rtclog::Event& event) const;
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rtclog::StreamConfig GetAudioReceiveConfig(const rtclog::Event& event) const;
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rtclog::StreamConfig GetAudioSendConfig(const rtclog::Event& event) const;
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std::vector<rtclog::Event> events_;
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struct Stream {
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Stream(uint32_t ssrc,
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MediaType media_type,
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webrtc::PacketDirection direction,
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webrtc::RtpHeaderExtensionMap map)
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: ssrc(ssrc),
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media_type(media_type),
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direction(direction),
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rtp_extensions_map(map) {}
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uint32_t ssrc;
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MediaType media_type;
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webrtc::PacketDirection direction;
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webrtc::RtpHeaderExtensionMap rtp_extensions_map;
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};
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// All configured streams found in the event log.
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std::vector<Stream> streams_;
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// To find configured extensions map for given stream, what are needed to
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// parse a header.
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typedef std::pair<uint32_t, webrtc::PacketDirection> StreamId;
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std::map<StreamId, webrtc::RtpHeaderExtensionMap*> rtp_extensions_maps_;
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};
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} // namespace webrtc
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#endif // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_H_
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