Modularize RtcEventLog

webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc

Index: webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
diff --git a/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc b/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
new file mode 100644
index 0000000000000000000000000000000000000000..95f24623a47276c0c5c152e746f7dc3ac70ff9bb
--- /dev/null
+++ b/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
@@ -0,0 +1,80 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h"
+
+namespace webrtc {
+
+void RtcEventLogEncoderLegacy::LoggingStart() {}
+
+void RtcEventLogEncoderLegacy::LoggingStopped() {}
+
+void RtcEventLogEncoderLegacy::VideoReceiveStreamConfig(
+    const rtclog::StreamConfig& config) {}
+
+void RtcEventLogEncoderLegacy::VideoSendStreamConfig(
+    const rtclog::StreamConfig& config) {}
+
+void RtcEventLogEncoderLegacy::AudioReceiveStreamConfig(
+    const rtclog::StreamConfig& config) {}
+
+void RtcEventLogEncoderLegacy::AudioSendStreamConfig(
+    const rtclog::StreamConfig& config) {}
+
+void RtcEventLogEncoderLegacy::RtpHeader(PacketDirection direction,
+                                         const uint8_t* header,
+                                         size_t packet_length) {}
+
+void RtcEventLogEncoderLegacy::RtpHeader(PacketDirection direction,
+                                         const uint8_t* header,
+                                         size_t packet_length,
+                                         int probe_cluster_id) {}
+
+void RtcEventLogEncoderLegacy::IncomingRtpHeader(
+    const RtpPacketReceived& packet) {}
+
+void RtcEventLogEncoderLegacy::OutgoingRtpHeader(const RtpPacketToSend& packet,
+                                                 int probe_cluster_id) {}
+
+void RtcEventLogEncoderLegacy::RtcpPacket(PacketDirection direction,
+                                          const uint8_t* packet,
+                                          size_t length) {}
+
+void RtcEventLogEncoderLegacy::IncomingRtcpPacket(
+    rtc::ArrayView<const uint8_t> packet) {}
+
+void RtcEventLogEncoderLegacy::OutgoingRtcpPacket(
+    rtc::ArrayView<const uint8_t> packet) {}
+
+void RtcEventLogEncoderLegacy::AudioPlayout(uint32_t ssrc) {}
+
+void RtcEventLogEncoderLegacy::LossBasedBweUpdate(int32_t bitrate_bps,
+                                                  uint8_t fraction_loss,
+                                                  int32_t total_packets) {}
+
+void RtcEventLogEncoderLegacy::DelayBasedBweUpdate(
+    int32_t bitrate_bps,
+    BandwidthUsage detector_state) {}
+
+void RtcEventLogEncoderLegacy::AudioNetworkAdaptation(
+    const AudioEncoderRuntimeConfig& config) {}
+
+void RtcEventLogEncoderLegacy::ProbeClusterCreated(int id,
+                                                   int bitrate_bps,
+                                                   int min_probes,
+                                                   int min_bytes) {}
+
+void RtcEventLogEncoderLegacy::ProbeResultSuccess(int id, int bitrate_bps) {}
+
+void RtcEventLogEncoderLegacy::ProbeResultFailure(
+    int id,
+    ProbeFailureReason failure_reason) {}
+
+}  // namespace webrtc