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..6d64fd740c9983f7fcffd1077f3c2b7513be28b1
--- /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 <cmath>
+
+#include "webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.h"
+#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
+
+namespace webrtc {
+
+namespace {
+constexpr int kMinBitreateChangeBps = 5000;

[minyue-webrtc, 2016/12/20 10:39:29]

bitrate

[michaelt, 2016/12/20 13:41:41]

Removed

+constexpr float kMinPacketLossChangeFraction = 0.5;
+}
+
+EventLogWriter::EventLogWriter(RtcEventLog* event_log)

[minyue-webrtc, 2016/12/20 10:39:29]

I'd like to remove this constructor, put kMinBitreateChangeBps and
kMinPacketLossChangeFraction in ana_impl.cc

but preferably rename a bit: kEventLogMinBitrateChangeBps and
kEventLogMinPacketLossChangeFraction

[michaelt, 2016/12/20 13:41:41]

Done.

+    : EventLogWriter(event_log,
+                     kMinBitreateChangeBps,
+                     kMinPacketLossChangeFraction) {}
+
+EventLogWriter::EventLogWriter(RtcEventLog* event_log,
+                               int min_bitreate_change_bps,
+                               float min_packet_loss_change_fraction)
+    : event_log_(event_log),
+      min_bitreate_change_bps_(min_bitreate_change_bps),
+      min_packet_loss_change_fraction_(min_packet_loss_change_fraction) {
+  RTC_CHECK(event_log_);
+}
+
+EventLogWriter::~EventLogWriter() = default;
+
+void EventLogWriter::MayLogEncoderRuntimeConfig(
+    const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
+  if (last_runtime_config_) {

[minyue-webrtc, 2016/12/20 10:39:29]

I prefer adding a

private 
void LogEncoderRuntimeConfig(const AudioNetworkAdaptor::EncoderRuntimeConfig&
config) {
  last_runtime_config_ =
rtc::Optional<AudioNetworkAdaptor::EncoderRuntimeConfig>(config);
  event_log_->LogEncoderRuntimeConfig(config);
}

and here

if (!last_runtime_config_)
  return LogEncoderRuntimeConfig(config);
if (xxx)
  return LogEncoderRuntimeConfig(config);
if (yyy)
  return LogEncoderRuntimeConfig(config);

break all "||"s

[michaelt, 2016/12/20 13:41:41]

Done.

+    if (last_runtime_config_->num_channels != config.num_channels ||
+        last_runtime_config_->enable_dtx != config.enable_dtx ||
+        last_runtime_config_->enable_fec != config.enable_fec ||
+        last_runtime_config_->frame_length_ms != config.frame_length_ms) {
+      event_log_->LogEncoderRuntimeConfig(config);
+
+    } else if (last_runtime_config_->bitrate_bps && config.bitrate_bps &&
+               std::abs(*last_runtime_config_->bitrate_bps -
+                        *config.bitrate_bps) >= min_bitreate_change_bps_) {
+      event_log_->LogEncoderRuntimeConfig(config);
+
+    } else if (last_runtime_config_->uplink_packet_loss_fraction &&
+               config.uplink_packet_loss_fraction &&
+               std::abs(*last_runtime_config_->uplink_packet_loss_fraction -
+                        *config.uplink_packet_loss_fraction) >=
+                   min_packet_loss_change_fraction_ *
+                       *last_runtime_config_->uplink_packet_loss_fraction) {
+      event_log_->LogEncoderRuntimeConfig(config);
+    }
+
+  } else {
+    event_log_->LogEncoderRuntimeConfig(config);
+  }
+  last_runtime_config_ =
+      rtc::Optional<AudioNetworkAdaptor::EncoderRuntimeConfig>(config);
+}
+}  // namespace webrtc