Split LogRtpHeader and LogRtcpPacket into separate versions for incoming and outgoing packets.

webrtc/logging/rtc_event_log/rtc_event_log.h

 /*
  *  Copyright (c) 2015 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_RTC_EVENT_LOG_H_
 #define WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_
 
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/rtpparameters.h"
 #include "webrtc/common_types.h"
+#include "webrtc/rtc_base/array_view.h"

[danilchap, 2017/09/05 08:47:16]

just moved to webrtc/api/

[terelius, 2017/09/07 12:53:55]

Done.

 #include "webrtc/rtc_base/platform_file.h"
 
 namespace webrtc {
 
 // Forward declaration of storage class that is automatically generated from
 // the protobuf file.
 namespace rtclog {
 class EventStream;
 
 struct StreamConfig {
@@ skipping 20 matching lines @@
     int rtx_payload_type;
   };
   std::vector<Codec> codecs;
 };
 
 }  // namespace rtclog
 
 class Clock;
 class RtcEventLogImpl;
 struct AudioEncoderRuntimeConfig;
+class RtpPacketReceived;
+class RtpPacketToSend;
 
 enum class MediaType;
 enum class BandwidthUsage;
 
 enum PacketDirection { kIncomingPacket = 0, kOutgoingPacket };
 enum ProbeFailureReason {
   kInvalidSendReceiveInterval,
   kInvalidSendReceiveRatio,
   kTimeout
 };
@@ skipping 47 matching lines @@
   // Logs configuration information for a video send stream.
   virtual void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) = 0;
 
   // Logs configuration information for an audio receive stream.
   virtual void LogAudioReceiveStreamConfig(
       const rtclog::StreamConfig& config) = 0;
 
   // Logs configuration information for an audio send stream.
   virtual void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) = 0;
 
-  // Logs the header of an incoming or outgoing RTP packet. packet_length
+  RTC_DEPRECATED virtual void LogRtpHeader(PacketDirection direction,
+                                           const uint8_t* header,
+                                           size_t packet_length) {}
+
+  RTC_DEPRECATED virtual void LogRtpHeader(PacketDirection direction,
+                                           const uint8_t* header,
+                                           size_t packet_length,
+                                           int probe_cluster_id) {}
+
+  // Logs the header of an incoming RTP packet. |packet_length|
   // is the total length of the packet, including both header and payload.
-  virtual void LogRtpHeader(PacketDirection direction,
-                            const uint8_t* header,
-                            size_t packet_length) = 0;
+  virtual void LogIncomingRtpHeader(const RtpPacketReceived& packet) = 0;
 
-  // Same as above but used on the sender side to log packets that are part of
-  // a probe cluster.
-  virtual void LogRtpHeader(PacketDirection direction,
-                            const uint8_t* header,
-                            size_t packet_length,
-                            int probe_cluster_id) = 0;
+  // Logs the header of an incoming RTP packet. |packet_length|
+  // is the total length of the packet, including both header and payload.
+  virtual void LogOutgoingRtpHeader(const RtpPacketToSend& packet,
+                                    int probe_cluster_id) = 0;
 
-  // Logs an incoming or outgoing RTCP packet.
-  virtual void LogRtcpPacket(PacketDirection direction,
-                             const uint8_t* packet,
-                             size_t length) = 0;
+  RTC_DEPRECATED virtual void LogRtcpPacket(PacketDirection direction,
+                                            const uint8_t* header,
+                                            size_t packet_length) {}
+
+  // Logs an incoming RTCP packet.
+  virtual void LogIncomingRtcpPacket(rtc::ArrayView<const uint8_t> packet) = 0;
+
+  // Logs an outgoing RTCP packet.
+  virtual void LogOutgoingRtcpPacket(rtc::ArrayView<const uint8_t> packet) = 0;
 
   // Logs an audio playout event.
   virtual void LogAudioPlayout(uint32_t ssrc) = 0;
 
   // Logs a bitrate update from the bandwidth estimator based on packet loss.
   virtual void LogLossBasedBweUpdate(int32_t bitrate_bps,
                                      uint8_t fraction_loss,
                                      int32_t total_packets) = 0;
 
   // Logs a bitrate update from the bandwidth estimator based on delay changes.
@@ skipping 39 matching lines @@
                     int64_t max_size_bytes) override {
     return false;
   }
   void StopLogging() override {}
   void LogVideoReceiveStreamConfig(
       const rtclog::StreamConfig& config) override {}
   void LogVideoSendStreamConfig(const rtclog::StreamConfig& config) override {}
   void LogAudioReceiveStreamConfig(
       const rtclog::StreamConfig& config) override {}
   void LogAudioSendStreamConfig(const rtclog::StreamConfig& config) override {}
-  void LogRtpHeader(PacketDirection direction,
-                    const uint8_t* header,
-                    size_t packet_length) override {}
-  void LogRtpHeader(PacketDirection direction,
-                    const uint8_t* header,
-                    size_t packet_length,
-                    int probe_cluster_id) override {}
-  void LogRtcpPacket(PacketDirection direction,
-                     const uint8_t* packet,
-                     size_t length) override {}
+  void LogIncomingRtpHeader(const RtpPacketReceived& packet) override {}
+  void LogOutgoingRtpHeader(const RtpPacketToSend& packet,
+                            int probe_cluster_id) override {}
+  void LogIncomingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override {}
+  void LogOutgoingRtcpPacket(rtc::ArrayView<const uint8_t> packet) override {}
   void LogAudioPlayout(uint32_t ssrc) override {}
   void LogLossBasedBweUpdate(int32_t bitrate_bps,
                              uint8_t fraction_loss,
                              int32_t total_packets) override {}
   void LogDelayBasedBweUpdate(int32_t bitrate_bps,
                               BandwidthUsage detector_state) override {}
   void LogAudioNetworkAdaptation(
       const AudioEncoderRuntimeConfig& config) override {}
   void LogProbeClusterCreated(int id,
                               int bitrate_bps,
                               int min_probes,
                               int min_bytes) override{};
   void LogProbeResultSuccess(int id, int bitrate_bps) override{};
   void LogProbeResultFailure(int id,
                              ProbeFailureReason failure_reason) override{};
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_H_