Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(823)

Unified Diff: webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h

Issue 3009333002: Create RtcEventLogEncoderLegacy (Closed)
Patch Set: Rebased Created 3 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h
diff --git a/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h b/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h
new file mode 100644
index 0000000000000000000000000000000000000000..1f616112db660d146e6800750d048404a29229d7
--- /dev/null
+++ b/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h
@@ -0,0 +1,91 @@
+/*
+ * Copyright (c) 2017 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_LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_LEGACY_H_
+#define WEBRTC_LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_LEGACY_H_
+
+#include <memory>
+#include <string>
+
+#include "webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder.h"
+#include "webrtc/rtc_base/buffer.h"
+
+namespace webrtc {
+
+namespace rtclog {
+class Event;
+} // namespace rtclog
+
+namespace rtp {
+class Packet;
+} // namespace rtp
+
+class RtcEventAudioNetworkAdaptation;
+class RtcEventAudioPlayout;
+class RtcEventAudioReceiveStreamConfig;
+class RtcEventAudioSendStreamConfig;
+class RtcEventBweUpdateDelayBased;
+class RtcEventBweUpdateLossBased;
+class RtcEventLoggingStarted;
+class RtcEventLoggingStopped;
+class RtcEventProbeClusterCreated;
+class RtcEventProbeResultFailure;
+class RtcEventProbeResultSuccess;
+class RtcEventRtcpPacketIncoming;
+class RtcEventRtcpPacketOutgoing;
+class RtcEventRtpPacketIncoming;
+class RtcEventRtpPacketOutgoing;
+class RtcEventVideoReceiveStreamConfig;
+class RtcEventVideoSendStreamConfig;
+
+class RtcEventLogEncoderLegacy final : public RtcEventLogEncoder {
+ public:
+ ~RtcEventLogEncoderLegacy() override = default;
+
+ std::string Encode(const RtcEvent& event) override;
+
+ private:
+ // Encoding entry-point for the various RtcEvent subclasses.
+ std::string AudioNetworkAdaptation(
+ const RtcEventAudioNetworkAdaptation& event);
+ std::string AudioPlayout(const RtcEventAudioPlayout& event);
+ std::string AudioReceiveStreamConfig(
+ const RtcEventAudioReceiveStreamConfig& event);
+ std::string AudioSendStreamConfig(const RtcEventAudioSendStreamConfig& event);
+ std::string BweUpdateDelayBased(const RtcEventBweUpdateDelayBased& event);
+ std::string BweUpdateLossBased(const RtcEventBweUpdateLossBased& event);
+ std::string LoggingStarted(const RtcEventLoggingStarted& event);
+ std::string LoggingStopped(const RtcEventLoggingStopped& event);
+ std::string ProbeClusterCreated(const RtcEventProbeClusterCreated& event);
+ std::string ProbeResultFailure(const RtcEventProbeResultFailure& event);
+ std::string ProbeResultSuccess(const RtcEventProbeResultSuccess&);
+ std::string RtcpPacketIncoming(const RtcEventRtcpPacketIncoming& event);
+ std::string RtcpPacketOutgoing(const RtcEventRtcpPacketOutgoing& event);
+ std::string RtpPacketIncoming(const RtcEventRtpPacketIncoming& event);
+ std::string RtpPacketOutgoing(const RtcEventRtpPacketOutgoing& event);
+ std::string VideoReceiveStreamConfig(
+ const RtcEventVideoReceiveStreamConfig& event);
+ std::string VideoSendStreamConfig(const RtcEventVideoSendStreamConfig& event);
+
+ // RTCP/RTP are handled similarly for incoming/outgoing.
+ std::string RtcpPacket(int64_t timestamp_us,
+ const rtc::Buffer& packet,
+ bool is_incoming);
+ std::string RtpPacket(int64_t timestamp_us,
+ const rtp::Packet& header,
+ size_t packet_length,
+ bool is_incoming);
+
+ std::string Serialize(rtclog::Event* event);
+};
+
+} // namespace webrtc
+
+#endif // WEBRTC_LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_LEGACY_H_

Powered by Google App Engine
This is Rietveld 408576698