Implement ANA statistics.
This CL also makes it possible to enable/prevent ANA controllers from making adaptations using field trials.
BUG=webrtc:8127
Review-Url: https://codereview.webrtc.org/3007983002
Cr-Original-Commit-Position: refs/heads/master@{#19761}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: 17289097f08280b377c4ca550e8d75e46ffcede3
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
index 5eff4b3..e2786ca 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
+++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
@@ -14,6 +14,7 @@
#include "webrtc/rtc_base/logging.h"
#include "webrtc/rtc_base/timeutils.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
namespace webrtc {
@@ -40,7 +41,17 @@
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_);
}
@@ -118,6 +129,55 @@
controller_manager_->GetSortedControllers(last_metrics_))
controller->MakeDecision(&config);
+ // Update ANA stats.
+ auto increment_opt = [](rtc::Optional<uint32_t>& a) {
+ a = rtc::Optional<uint32_t>(a.value_or(0) + 1);
+ };
+ 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<float>(*config.uplink_packet_loss_fraction);
+ }
+ }
+ 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.reset();
+ }
+ if (!enable_dtx_adaptation_ && config.enable_dtx) {
+ config.enable_dtx.reset();
+ }
+ if (!enable_fec_adaptation_ && config.enable_fec) {
+ config.enable_fec.reset();
+ config.uplink_packet_loss_fraction.reset();
+ }
+ if (!enable_frame_length_adaptation_ && config.frame_length_ms) {
+ config.frame_length_ms.reset();
+ }
+ if (!enable_channel_adaptation_ && config.num_channels) {
+ config.num_channels.reset();
+ }
+
if (debug_dump_writer_)
debug_dump_writer_->DumpEncoderRuntimeConfig(config, rtc::TimeMillis());
@@ -136,9 +196,7 @@
}
ANAStats AudioNetworkAdaptorImpl::GetStats() const {
- // TODO(ivoc): Actually implement the stat.
- // Tracking bug: https://crbug.com/webrtc/8127
- return ANAStats();
+ return stats_;
}
void AudioNetworkAdaptorImpl::DumpNetworkMetrics() {
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
index bd2a250..8e76db2 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
+++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
@@ -75,6 +75,16 @@
Controller::NetworkMetrics last_metrics_;
+ rtc::Optional<AudioEncoderRuntimeConfig> prev_config_;
+
+ ANAStats stats_;
+
+ const bool enable_bitrate_adaptation_;
+ const bool enable_dtx_adaptation_;
+ const bool enable_fec_adaptation_;
+ const bool enable_channel_adaptation_;
+ const bool enable_frame_length_adaptation_;
+
RTC_DISALLOW_COPY_AND_ASSIGN(AudioNetworkAdaptorImpl);
};
diff --git a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
index 5695a38..a0dc12c 100644
--- a/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
+++ b/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
@@ -17,6 +17,7 @@
#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h"
#include "webrtc/rtc_base/fakeclock.h"
+#include "webrtc/test/field_trial.h"
#include "webrtc/test/gtest.h"
namespace webrtc {
@@ -178,6 +179,9 @@
TEST(AudioNetworkAdaptorImplTest,
DumpEncoderRuntimeConfigIsCalledOnGetEncoderRuntimeConfig) {
+ test::ScopedFieldTrials override_field_trials(
+ "WebRTC-Audio-BitrateAdaptation/Enabled/WebRTC-Audio-FecAdaptation/"
+ "Enabled/");
rtc::ScopedFakeClock fake_clock;
fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(kClockInitialTimeMs));
auto states = CreateAudioNetworkAdaptor();
@@ -255,6 +259,9 @@
}
TEST(AudioNetworkAdaptorImplTest, LogRuntimeConfigOnGetEncoderRuntimeConfig) {
+ test::ScopedFieldTrials override_field_trials(
+ "WebRTC-Audio-BitrateAdaptation/Enabled/WebRTC-Audio-FecAdaptation/"
+ "Enabled/");
auto states = CreateAudioNetworkAdaptor();
AudioEncoderRuntimeConfig config;
@@ -276,9 +283,17 @@
// Simulate some adaptation, otherwise the stats will not show anything.
AudioEncoderRuntimeConfig config1, config2;
config1.bitrate_bps = rtc::Optional<int>(32000);
+ config1.num_channels = rtc::Optional<size_t>(2);
config1.enable_fec = rtc::Optional<bool>(true);
+ config1.enable_dtx = rtc::Optional<bool>(true);
+ config1.frame_length_ms = rtc::Optional<int>(120);
+ config1.uplink_packet_loss_fraction = rtc::Optional<float>(0.1f);
config2.bitrate_bps = rtc::Optional<int>(16000);
+ config2.num_channels = rtc::Optional<size_t>(1);
config2.enable_fec = rtc::Optional<bool>(false);
+ config2.enable_dtx = rtc::Optional<bool>(false);
+ config2.frame_length_ms = rtc::Optional<int>(60);
+ config1.uplink_packet_loss_fraction = rtc::Optional<float>(0.1f);
EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
.WillOnce(SetArgPointee<0>(config1));
@@ -286,24 +301,19 @@
EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
.WillOnce(SetArgPointee<0>(config2));
states.audio_network_adaptor->GetEncoderRuntimeConfig();
+ EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
+ .WillOnce(SetArgPointee<0>(config1));
+ states.audio_network_adaptor->GetEncoderRuntimeConfig();
auto ana_stats = states.audio_network_adaptor->GetStats();
- // Check that the default stats are returned, as these have not been
- // implemented yet). Tracking bug: https://crbug.com/8127
- auto default_stats = ANAStats();
- EXPECT_EQ(ana_stats.bitrate_action_counter,
- default_stats.bitrate_action_counter);
- EXPECT_EQ(ana_stats.channel_action_counter,
- default_stats.channel_action_counter);
- EXPECT_EQ(ana_stats.dtx_action_counter, default_stats.dtx_action_counter);
- EXPECT_EQ(ana_stats.fec_action_counter, default_stats.fec_action_counter);
- EXPECT_EQ(ana_stats.frame_length_increase_counter,
- default_stats.frame_length_increase_counter);
- EXPECT_EQ(ana_stats.frame_length_decrease_counter,
- default_stats.frame_length_decrease_counter);
- EXPECT_EQ(ana_stats.uplink_packet_loss_fraction,
- default_stats.uplink_packet_loss_fraction);
+ EXPECT_EQ(ana_stats.bitrate_action_counter, 2);
+ EXPECT_EQ(ana_stats.channel_action_counter, 2);
+ EXPECT_EQ(ana_stats.dtx_action_counter, 2);
+ EXPECT_EQ(ana_stats.fec_action_counter, 2);
+ EXPECT_EQ(ana_stats.frame_length_increase_counter, 1);
+ EXPECT_EQ(ana_stats.frame_length_decrease_counter, 1);
+ EXPECT_EQ(ana_stats.uplink_packet_loss_fraction, 0.1f);
}
} // namespace webrtc
