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| 1 /* |
| 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. |
| 3 * |
| 4 * Use of this source code is governed by a BSD-style license |
| 5 * that can be found in the LICENSE file in the root of the source |
| 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ |
| 10 |
| 11 #ifndef WEBRTC_LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_LEGACY_H_ |
| 12 #define WEBRTC_LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_LEGACY_H_ |
| 13 |
| 14 #include <memory> |
| 15 #include <string> |
| 16 |
| 17 #include "webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder.h" |
| 18 #include "webrtc/rtc_base/buffer.h" |
| 19 |
| 20 namespace webrtc { |
| 21 |
| 22 namespace rtclog { |
| 23 class Event; |
| 24 } // namespace rtclog |
| 25 |
| 26 namespace rtp { |
| 27 class Packet; |
| 28 } // namespace rtp |
| 29 |
| 30 class RtcEventAudioNetworkAdaptation; |
| 31 class RtcEventAudioPlayout; |
| 32 class RtcEventAudioReceiveStreamConfig; |
| 33 class RtcEventAudioSendStreamConfig; |
| 34 class RtcEventBweUpdateDelayBased; |
| 35 class RtcEventBweUpdateLossBased; |
| 36 class RtcEventLoggingStarted; |
| 37 class RtcEventLoggingStopped; |
| 38 class RtcEventProbeClusterCreated; |
| 39 class RtcEventProbeResultFailure; |
| 40 class RtcEventProbeResultSuccess; |
| 41 class RtcEventRtcpPacketIncoming; |
| 42 class RtcEventRtcpPacketOutgoing; |
| 43 class RtcEventRtpPacketIncoming; |
| 44 class RtcEventRtpPacketOutgoing; |
| 45 class RtcEventVideoReceiveStreamConfig; |
| 46 class RtcEventVideoSendStreamConfig; |
| 47 |
| 48 class RtcEventLogEncoderLegacy final : public RtcEventLogEncoder { |
| 49 public: |
| 50 ~RtcEventLogEncoderLegacy() override = default; |
| 51 |
| 52 std::string Encode(const RtcEvent& event) override; |
| 53 |
| 54 private: |
| 55 // Encoding entry-point for the various RtcEvent subclasses. |
| 56 std::string AudioNetworkAdaptation( |
| 57 const RtcEventAudioNetworkAdaptation& event); |
| 58 std::string AudioPlayout(const RtcEventAudioPlayout& event); |
| 59 std::string AudioReceiveStreamConfig( |
| 60 const RtcEventAudioReceiveStreamConfig& event); |
| 61 std::string AudioSendStreamConfig(const RtcEventAudioSendStreamConfig& event); |
| 62 std::string BweUpdateDelayBased(const RtcEventBweUpdateDelayBased& event); |
| 63 std::string BweUpdateLossBased(const RtcEventBweUpdateLossBased& event); |
| 64 std::string LoggingStarted(const RtcEventLoggingStarted& event); |
| 65 std::string LoggingStopped(const RtcEventLoggingStopped& event); |
| 66 std::string ProbeClusterCreated(const RtcEventProbeClusterCreated& event); |
| 67 std::string ProbeResultFailure(const RtcEventProbeResultFailure& event); |
| 68 std::string ProbeResultSuccess(const RtcEventProbeResultSuccess&); |
| 69 std::string RtcpPacketIncoming(const RtcEventRtcpPacketIncoming& event); |
| 70 std::string RtcpPacketOutgoing(const RtcEventRtcpPacketOutgoing& event); |
| 71 std::string RtpPacketIncoming(const RtcEventRtpPacketIncoming& event); |
| 72 std::string RtpPacketOutgoing(const RtcEventRtpPacketOutgoing& event); |
| 73 std::string VideoReceiveStreamConfig( |
| 74 const RtcEventVideoReceiveStreamConfig& event); |
| 75 std::string VideoSendStreamConfig(const RtcEventVideoSendStreamConfig& event); |
| 76 |
| 77 // RTCP/RTP are handled similarly for incoming/outgoing. |
| 78 std::string RtcpPacket(int64_t timestamp_us, |
| 79 const rtc::Buffer& packet, |
| 80 bool is_incoming); |
| 81 std::string RtpPacket(int64_t timestamp_us, |
| 82 const rtp::Packet& header, |
| 83 size_t packet_length, |
| 84 bool is_incoming); |
| 85 |
| 86 std::string Serialize(rtclog::Event* event); |
| 87 }; |
| 88 |
| 89 } // namespace webrtc |
| 90 |
| 91 #endif // WEBRTC_LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_LEGACY_H_ |
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