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Implement ANA statistics.

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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-Commit-Position: refs/heads/master@{#19761}
This commit is contained in:
ivoc
2017-09-09 08:45:40 -07:00
committed by Commit Bot
parent a37de39216
commit 17289097f0
3 changed files with 97 additions and 19 deletions
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66
webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
View File

@ -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 @@ AudioNetworkAdaptorImpl::AudioNetworkAdaptorImpl(
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 @@ AudioEncoderRuntimeConfig AudioNetworkAdaptorImpl::GetEncoderRuntimeConfig() {
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 @@ void AudioNetworkAdaptorImpl::StopDebugDump() {
}
ANAStats AudioNetworkAdaptorImpl::GetStats() const {
// TODO(ivoc): Actually implement the stat.
// Tracking bug: https://crbug.com/webrtc/8127
return ANAStats();
return stats_;
}
void AudioNetworkAdaptorImpl::DumpNetworkMetrics() {

10
webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
View File

@ -75,6 +75,16 @@ class AudioNetworkAdaptorImpl final : public AudioNetworkAdaptor {
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);
};

40
webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
View File

@ -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,
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,
}
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 @@ TEST(AudioNetworkAdaptorImplTest, TestANAStats) {
// 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 @@ TEST(AudioNetworkAdaptorImplTest, TestANAStats) {
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