Implement ANA statistics.

webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc

 /*
  *  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 "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
 
 #include <utility>
 
 #include "webrtc/rtc_base/logging.h"
 #include "webrtc/rtc_base/timeutils.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
 namespace {
 constexpr int kEventLogMinBitrateChangeBps = 5000;
 constexpr float kEventLogMinBitrateChangeFraction = 0.25;
 constexpr float kEventLogMinPacketLossChangeFraction = 0.5;
 }  // namespace
 
 AudioNetworkAdaptorImpl::Config::Config() : event_log(nullptr){};
 
 AudioNetworkAdaptorImpl::Config::~Config() = default;
 
 AudioNetworkAdaptorImpl::AudioNetworkAdaptorImpl(
     const Config& config,
     std::unique_ptr<ControllerManager> controller_manager,
     std::unique_ptr<DebugDumpWriter> debug_dump_writer)
     : config_(config),
       controller_manager_(std::move(controller_manager)),
       debug_dump_writer_(std::move(debug_dump_writer)),
       event_log_writer_(
           config.event_log
               ? new EventLogWriter(config.event_log,
                                    kEventLogMinBitrateChangeBps,
                                    kEventLogMinBitrateChangeFraction,
                                    kEventLogMinPacketLossChangeFraction)
-              : nullptr) {
+              : nullptr),
+      enable_bitrate_adaptation_(
+          webrtc::field_trial::IsEnabled("WebRTC-Audio-BitrateAdaptation")),
+      enable_dtx_adaptation_(
+          webrtc::field_trial::IsEnabled("WebRTC-Audio-DtxAdaptation")),
+      enable_fec_adaptation_(
+          webrtc::field_trial::IsEnabled("WebRTC-Audio-FecAdaptation")),
+      enable_channel_adaptation_(
+          webrtc::field_trial::IsEnabled("WebRTC-Audio-ChannelAdaptation")),
+      enable_frame_length_adaptation_(webrtc::field_trial::IsEnabled(
+          "WebRTC-Audio-FrameLengthAdaptation")) {
   RTC_DCHECK(controller_manager_);
 }
 
 AudioNetworkAdaptorImpl::~AudioNetworkAdaptorImpl() = default;
 
 void AudioNetworkAdaptorImpl::SetUplinkBandwidth(int uplink_bandwidth_bps) {
   last_metrics_.uplink_bandwidth_bps = rtc::Optional<int>(uplink_bandwidth_bps);
   DumpNetworkMetrics();
 
   Controller::NetworkMetrics network_metrics;
@@ skipping 57 matching lines @@
       rtc::Optional<size_t>(overhead_bytes_per_packet);
   UpdateNetworkMetrics(network_metrics);
 }
 
 AudioEncoderRuntimeConfig AudioNetworkAdaptorImpl::GetEncoderRuntimeConfig() {
   AudioEncoderRuntimeConfig config;
   for (auto& controller :
        controller_manager_->GetSortedControllers(last_metrics_))
     controller->MakeDecision(&config);
 
+  // Update ANA stats.
+  auto increment_opt = [](rtc::Optional<uint32_t>& a) {
+    if (a)
+      (*a)++;
+    else
+      a = rtc::Optional<uint32_t>(1);

[ossu, 2017/09/08 10:56:01]

I think you could just do a = a.value_or(0) + 1 as the implementation of
increment_opt.

[ivoc, 2017/09/08 11:43:47]

Nice! Done, although I needed to wrap the result in an optional to make it work.

[ossu, 2017/09/08 12:05:21]

Ah, that's right. I'll try to look at getting a newer version of Optional from
Chromium, so that we can avoid having to explicitly construct Optionals like
this in future.

+  };
+  if (prev_config_) {
+    if (config.bitrate_bps != prev_config_->bitrate_bps) {
+      increment_opt(stats_.bitrate_action_counter);
+    }
+    if (config.enable_dtx != prev_config_->enable_dtx) {
+      increment_opt(stats_.dtx_action_counter);
+    }
+    if (config.enable_fec != prev_config_->enable_fec) {
+      increment_opt(stats_.fec_action_counter);
+    }
+    if (config.frame_length_ms && prev_config_->frame_length_ms) {
+      if (*config.frame_length_ms > *prev_config_->frame_length_ms) {
+        increment_opt(stats_.frame_length_increase_counter);
+      } else if (*config.frame_length_ms < *prev_config_->frame_length_ms) {
+        increment_opt(stats_.frame_length_decrease_counter);
+      }
+    }
+    if (config.num_channels != prev_config_->num_channels) {
+      increment_opt(stats_.channel_action_counter);
+    }
+    if (config.uplink_packet_loss_fraction) {
+      stats_.uplink_packet_loss_fraction = rtc::Optional<double>(
+          static_cast<double>(*config.uplink_packet_loss_fraction));

[ossu, 2017/09/08 10:56:01]

Why does this need a static_cast? Why isn't uplink_packet_loss_fraction the same
type in both?

[ivoc, 2017/09/08 11:43:47]

That's a good point, I changed the stat into a float so this is no longer
needed.

+    }
+  }
+  prev_config_ = rtc::Optional<AudioEncoderRuntimeConfig>(config);
+
+  // Prevent certain controllers from taking action (determined by field trials)
+  if (!enable_bitrate_adaptation_ && config.bitrate_bps) {
+    config.bitrate_bps = rtc::Optional<int>();

[ossu, 2017/09/08 10:56:01]

You can use .reset() to empty Optionals and save on typing. (Especially having
to fully qualify the template type).

[ivoc, 2017/09/08 11:43:47]

Done!

+  }
+  if (!enable_dtx_adaptation_ && config.enable_dtx) {
+    config.enable_dtx = rtc::Optional<bool>();
+  }
+  if (!enable_fec_adaptation_ && config.enable_fec) {
+    config.enable_fec = rtc::Optional<bool>();
+    config.uplink_packet_loss_fraction = rtc::Optional<float>();
+  }
+  if (!enable_frame_length_adaptation_ && config.frame_length_ms) {
+    config.frame_length_ms = rtc::Optional<int>();
+  }
+  if (!enable_channel_adaptation_ && config.num_channels) {
+    config.num_channels = rtc::Optional<size_t>();
+  }
+
   if (debug_dump_writer_)
     debug_dump_writer_->DumpEncoderRuntimeConfig(config, rtc::TimeMillis());
 
   if (event_log_writer_)
     event_log_writer_->MaybeLogEncoderConfig(config);
 
   return config;
 }
 
 void AudioNetworkAdaptorImpl::StartDebugDump(FILE* file_handle) {
   debug_dump_writer_ = DebugDumpWriter::Create(file_handle);
 }
 
 void AudioNetworkAdaptorImpl::StopDebugDump() {
   debug_dump_writer_.reset(nullptr);
 }
 
 ANAStats AudioNetworkAdaptorImpl::GetStats() const {
-  // TODO(ivoc): Actually implement the stat.
-  // Tracking bug: https://bugs.chromium.org/p/webrtc/issues/detail?id=8127
-  return ANAStats();
+  return stats_;
 }
 
 void AudioNetworkAdaptorImpl::DumpNetworkMetrics() {
   if (debug_dump_writer_)
     debug_dump_writer_->DumpNetworkMetrics(last_metrics_, rtc::TimeMillis());
 }
 
 void AudioNetworkAdaptorImpl::UpdateNetworkMetrics(
     const Controller::NetworkMetrics& network_metrics) {
   for (auto& controller : controller_manager_->GetControllers())
     controller->UpdateNetworkMetrics(network_metrics);
 }
 
 }  // namespace webrtc