New parser for event log. Manually parse the outermost EventStream

webrtc/call/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_CALL_RTC_EVENT_LOG_PARSER_H_
+#define WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_
+
+#include <string>
+#include <vector>
+
+#include "webrtc/call/rtc_event_log.h"
+#include "webrtc/video_receive_stream.h"
+#include "webrtc/video_send_stream.h"
+
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/call/rtc_event_log.pb.h"
+#else
+#include "webrtc/call/rtc_event_log.pb.h"
+#endif
+
+namespace webrtc {
+
+enum class MediaType;
+
+class ParsedRtcEventLog {
+  friend class RtcEventLogTestHelper;
+
+ public:
+  enum EventType {
+    UNKNOWN_EVENT = 0,
+    LOG_START = 1,
+    LOG_END = 2,
+    RTP_EVENT = 3,
+    RTCP_EVENT = 4,
+    AUDIO_PLAYOUT_EVENT = 5,
+    BWE_PACKET_LOSS_EVENT = 6,
+    BWE_PACKET_DELAY_EVENT = 7,
+    VIDEO_RECEIVER_CONFIG_EVENT = 8,
+    VIDEO_SENDER_CONFIG_EVENT = 9,
+    AUDIO_RECEIVER_CONFIG_EVENT = 10,
+    AUDIO_SENDER_CONFIG_EVENT = 11
+  };
+
+  // Reads an RtcEventLog file and returns true if parsing was successful.
+  bool ParseFile(const std::string& file_name);
+
+  // 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, media type, header length and packet length
+  // from the RTP event at |index|, and stores the values in the corresponding
+  // output parameters. The output parameters can be set to nullptr if those
+  // values aren't needed.
+  // NB: The header must have space for at least IP_PACKET_SIZE bytes.
+  void GetRtpHeader(size_t index,
+                    PacketDirection* incoming,
+                    MediaType* media_type,
+                    uint8_t* header,
+                    size_t* header_length,
+                    size_t* total_length) const;
+
+  // Reads packet, direction, media type and packet length from the RTCP event
+  // at |index|, and stores the values in the corresponding output parameters.
+  // The output parameters can be set to nullptr if those values aren't needed.
+  // NB: The packet must have space for at least IP_PACKET_SIZE bytes.
+  void GetRtcpPacket(size_t index,
+                     PacketDirection* incoming,
+                     MediaType* media_type,
+                     uint8_t* packet,
+                     size_t* length) const;
+
+  // Reads a config event to a (non-NULL) VideoReceiveStream::Config struct.
+  // Only the fields that are stored in the protobuf will be written.
+  void GetVideoReceiveConfig(size_t index,
+                             VideoReceiveStream::Config* config) const;
+
+  // Reads a config event to a (non-NULL) VideoSendStream::Config struct.
+  // Only the fields that are stored in the protobuf will be written.
+  void GetVideoSendConfig(size_t index, VideoSendStream::Config* config) 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 BWE event at |index| and stores the values in
+  // the corresponding output parameters. The output parameters can be set to
+  // nullptr if those values aren't needed.
+  // NB: The packet must have space for at least IP_PACKET_SIZE bytes.
+  void GetBwePacketLossEvent(size_t index,
+                             int32_t* bitrate,
+                             uint8_t* fraction_loss,
+                             int32_t* total_packets) const;
+
+ private:
+  std::vector<rtclog::Event> stream_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_