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 |
deleted file mode 100644 |
index a50ec20391de1b53bf76187b6f993f69c8cde9e7..0000000000000000000000000000000000000000 |
--- a/webrtc/call/rtc_event_log_parser.h |
+++ /dev/null |
@@ -1,123 +0,0 @@ |
-/* |
- * 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/base/ignore_wundef.h" |
-#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. |
-RTC_PUSH_IGNORING_WUNDEF() |
-#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 |
-RTC_POP_IGNORING_WUNDEF() |
- |
-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); |
- |
- // 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, 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> events_; |
-}; |
- |
-} // namespace webrtc |
- |
-#endif // WEBRTC_CALL_RTC_EVENT_LOG_PARSER_H_ |