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) {
   RTC_DCHECK(controller_manager_);
+  // Set the field trial flags to determine if adaptation is enabled.
+  enable_bitrate_adaptation_ =
+      webrtc::field_trial::IsEnabled("WebRTC-Audio-BitrateAdaptation");
+  enable_channel_adaptation_ =
+      webrtc::field_trial::IsEnabled("WebRTC-Audio-ChannelAdaptation");
+  enable_dtx_adaptation_ =
+      webrtc::field_trial::IsEnabled("WebRTC-Audio-DtxAdaptation");
+  enable_fec_adaptation_ =
+      webrtc::field_trial::IsEnabled("WebRTC-Audio-FecAdaptation");
+  enable_frame_length_adaptation_ =
+      webrtc::field_trial::IsEnabled("WebRTC-Audio-FrameLengthAdaptation");
 }
 
 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;
   network_metrics.uplink_bandwidth_bps =
@@ skipping 56 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<int>& a) {
+    if (a)
+      (*a)++;
+    else
+      a = rtc::Optional<int>(1);
+  };
+  if (prev_config_) {
+    if (config.bitrate_bps != prev_config_->bitrate_bps) {
+      increment_opt(stats_.ana_bitrate_action_counter);
+    }
+    if (config.enable_dtx != prev_config_->enable_dtx) {
+      increment_opt(stats_.ana_dtx_action_counter);
+    }
+    if (config.enable_fec != prev_config_->enable_fec) {

[alexnarest, 2017/09/04 11:24:02]

In addition I think we want to record the fact FEC controller smoothed the PL or
in case of RPLR-FEC sent RPL instead of RTCP PL in the control group.
I suggest we increment here on FEC enable or PL is greater than 1%.

[ivoc, 2017/09/05 09:01:40]

Another option is to add another stat for uplink packet loss fraction, what do
you think? It could also be implemented as a cumulative stat since the start of
the call, which would make it possible to calculate the average value between
the GetStats() calls. Would that be useful to have?

[alexnarest, 2017/09/05 10:43:27]

Yes, I think it is a good idea

+      increment_opt(stats_.ana_fec_action_counter);
+    }
+    if (config.frame_length_ms != prev_config_->frame_length_ms) {

[alexnarest, 2017/09/05 10:43:27]

It would be helpful to report inc/dec counters rather than the general change
counter.

+      increment_opt(stats_.ana_frame_length_action_counter);
+    }
+    if (config.num_channels != prev_config_->num_channels) {
+      increment_opt(stats_.ana_channel_action_counter);
+    }
+  }
+  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>();
+  }
+  if (!enable_dtx_adaptation_ && config.enable_dtx) {
+    config.enable_dtx = rtc::Optional<bool>();
+  }
+  if (!enable_fec_adaptation_ && config.enable_fec) {

[alexnarest, 2017/09/04 11:24:01]

In addition to enable_fec we want to prevent PL smoothing and RPL reporting in
case of RPLR-FEC

[ivoc, 2017/09/05 09:01:40]

Good point, done.

+    config.enable_fec = rtc::Optional<bool>();
+  }
+  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);
 }
 
 AudioNetworkAdaptorImpl::AudioNetworkAdaptorStats
 AudioNetworkAdaptorImpl::GetStats() const {
-  // TODO(ivoc): Actually implement the stat.
-  return AudioNetworkAdaptorStats();
+  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