Modularize RtcEventLog

webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder.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_ENCODER_RTC_EVENT_LOG_ENCODER_H_
+#define WEBRTC_LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_H_
+
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "webrtc/api/array_view.h"
+#include "webrtc/api/rtpparameters.h"
+#include "webrtc/common_types.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;
+};
+
+}  // namespace rtclog
+
+struct AudioEncoderRuntimeConfig;
+class RtpPacketReceived;
+class RtpPacketToSend;
+enum class BandwidthUsage;
+
+enum PacketDirection { kIncomingPacket = 0, kOutgoingPacket };
+
+enum ProbeFailureReason {
+  kInvalidSendReceiveInterval,
+  kInvalidSendReceiveRatio,
+  kTimeout
+};
+
+class RtcEventLogEncoder {
+ public:
+  virtual ~RtcEventLogEncoder() = default;
+
+  virtual void LoggingStart() = 0;
+  virtual void LoggingStopped() = 0;
+  virtual void VideoReceiveStreamConfig(const rtclog::StreamConfig& config) = 0;
+  virtual void VideoSendStreamConfig(const rtclog::StreamConfig& config) = 0;
+  virtual void AudioReceiveStreamConfig(const rtclog::StreamConfig& config) = 0;
+  virtual void AudioSendStreamConfig(const rtclog::StreamConfig& config) = 0;
+  virtual void RtpHeader(PacketDirection direction,
+                         const uint8_t* header,
+                         size_t packet_length) = 0;
+  virtual void RtpHeader(PacketDirection direction,
+                         const uint8_t* header,
+                         size_t packet_length,
+                         int probe_cluster_id) = 0;
+  virtual void IncomingRtpHeader(const RtpPacketReceived& packet) = 0;
+  virtual void OutgoingRtpHeader(const RtpPacketToSend& packet,
+                                 int probe_cluster_id) = 0;
+  virtual void RtcpPacket(PacketDirection direction,
+                          const uint8_t* packet,
+                          size_t length) = 0;
+  virtual void IncomingRtcpPacket(rtc::ArrayView<const uint8_t> packet) = 0;
+  virtual void OutgoingRtcpPacket(rtc::ArrayView<const uint8_t> packet) = 0;
+  virtual void AudioPlayout(uint32_t ssrc) = 0;
+  virtual void LossBasedBweUpdate(int32_t bitrate_bps,
+                                  uint8_t fraction_loss,
+                                  int32_t total_packets) = 0;
+  virtual void DelayBasedBweUpdate(int32_t bitrate_bps,
+                                   BandwidthUsage detector_state) = 0;
+  virtual void AudioNetworkAdaptation(
+      const AudioEncoderRuntimeConfig& config) = 0;
+  virtual void ProbeClusterCreated(int id,
+                                   int bitrate_bps,
+                                   int min_probes,
+                                   int min_bytes) = 0;
+  virtual void ProbeResultSuccess(int id, int bitrate_bps) = 0;
+  virtual void ProbeResultFailure(int id,
+                                  ProbeFailureReason failure_reason) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_LOGGING_RTC_EVENT_LOG_ENCODER_RTC_EVENT_LOG_ENCODER_H_