Move StreamConfig into its own file

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/array_view.h"
 #include "webrtc/api/rtpparameters.h"

[terelius, 2017/09/08 13:16:29]

I don't think this include is needed anymore.

[eladalon, 2017/09/08 13:19:57]

It is - RtpExtension.

[eladalon, 2017/09/08 13:23:52]

Oh, wrong file. Oops.

 #include "webrtc/common_types.h"
+#include "webrtc/logging/rtc_event_log/rtc_stream_config.h"
 #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 {
-  uint32_t local_ssrc = 0;
-  uint32_t remote_ssrc = 0;
-  uint32_t rtx_ssrc = 0;
-  std::string rsid;
-
-  bool remb = false;
-  std::vector<RtpExtension> rtp_extensions;
-
-  RtcpMode rtcp_mode = RtcpMode::kReducedSize;
-
-  struct Codec {
-    Codec(const std::string& payload_name,
-          int payload_type,
-          int rtx_payload_type)
-        : payload_name(payload_name),
-          payload_type(payload_type),
-          rtx_payload_type(rtx_payload_type) {}
-
-    std::string payload_name;
-    int payload_type;
-    int rtx_payload_type;
-  };
-  std::vector<Codec> codecs;
-};
-
+class EventStream;  // Storage class automatically generated from protobuf.
 }  // namespace rtclog
 
 class Clock;
 struct AudioEncoderRuntimeConfig;
 class RtpPacketReceived;
 class RtpPacketToSend;
 
 enum class MediaType;
 enum class BandwidthUsage;
 
@@ skipping 165 matching lines @@
                               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_