Log audio network adapter decisions in event log.

webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc

Index: webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc b/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
new file mode 100644
index 0000000000000000000000000000000000000000..619a2473d93708cf221846151f9b2ebf61d5a815
--- /dev/null
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
@@ -0,0 +1,68 @@
+/*
+ *  Copyright (c) 2016 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 <math.h>
+#include <algorithm>
+
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h"
+
+namespace webrtc {
+
+EventLogWriter::EventLogWriter(RtcEventLog* event_log,
+                               int min_bitrate_change_bps,
+                               float min_bitrate_change_fraction,
+                               float min_packet_loss_change_fraction)
+    : event_log_(event_log),
+      min_bitrate_change_bps_(min_bitrate_change_bps),
+      min_bitrate_change_fraction_(min_bitrate_change_fraction),
+      min_packet_loss_change_fraction_(min_packet_loss_change_fraction) {
+  RTC_DCHECK(event_log_);
+}
+
+EventLogWriter::~EventLogWriter() = default;
+
+void EventLogWriter::MaybeLogEncoderConfig(
+    const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
+  if (last_logged_config_.num_channels != config.num_channels)
+    return LogEncoderConfig(config);
+  if (last_logged_config_.enable_dtx != config.enable_dtx)
+    return LogEncoderConfig(config);
+  if (last_logged_config_.enable_fec != config.enable_fec)
+    return LogEncoderConfig(config);
+  if (last_logged_config_.frame_length_ms != config.frame_length_ms)
+    return LogEncoderConfig(config);
+  if ((!last_logged_config_.bitrate_bps && config.bitrate_bps) ||
+      (last_logged_config_.bitrate_bps && config.bitrate_bps &&
+       std::abs(*last_logged_config_.bitrate_bps - *config.bitrate_bps) >=
+           std::min(static_cast<int>(*last_logged_config_.bitrate_bps *
+                                     min_bitrate_change_fraction_),
+                    min_bitrate_change_bps_))) {
+    return LogEncoderConfig(config);
+  }
+  if ((!last_logged_config_.uplink_packet_loss_fraction &&
+       config.uplink_packet_loss_fraction) ||
+      (last_logged_config_.uplink_packet_loss_fraction &&
+       config.uplink_packet_loss_fraction &&
+       fabs(*last_logged_config_.uplink_packet_loss_fraction -
+            *config.uplink_packet_loss_fraction) >=
+           min_packet_loss_change_fraction_ *
+               *last_logged_config_.uplink_packet_loss_fraction)) {
+    return LogEncoderConfig(config);
+  }
+}
+
+void EventLogWriter::LogEncoderConfig(
+    const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
+  event_log_->LogAudioNetworkAdaptation(config);
+  last_logged_config_ = config;
+}
+
+}  // namespace webrtc