Index: webrtc/call/rtc_event_log_parser.h |
diff --git a/webrtc/call/rtc_event_log_parser.h b/webrtc/call/rtc_event_log_parser.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..acdfa77da177e605ddfa94e80dccea6abef37847 |
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+++ b/webrtc/call/rtc_event_log_parser.h |
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+/* |
+ * 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_ |