Allow ANA to receive RPLR (recoverable packet loss rate) indications
This is part of a series of CLs. Next CLs:
1. CL for RPLR-based FecController
2. CL for allowing experiment-driven configuration of the above (through both field-trials and protobuf)
BUG=webrtc:7058
Review-Url: https://codereview.webrtc.org/2661043003
Cr-Commit-Position: refs/heads/master@{#17368}
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
index 612d2d3..e1952f4 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
@@ -68,6 +68,18 @@
UpdateNetworkMetrics(network_metrics);
}
+void AudioNetworkAdaptorImpl::SetUplinkRecoverablePacketLossFraction(
+ float uplink_recoverable_packet_loss_fraction) {
+ last_metrics_.uplink_recoverable_packet_loss_fraction =
+ rtc::Optional<float>(uplink_recoverable_packet_loss_fraction);
+ DumpNetworkMetrics();
+
+ Controller::NetworkMetrics network_metrics;
+ network_metrics.uplink_recoverable_packet_loss_fraction =
+ rtc::Optional<float>(uplink_recoverable_packet_loss_fraction);
+ UpdateNetworkMetrics(network_metrics);
+}
+
void AudioNetworkAdaptorImpl::SetRtt(int rtt_ms) {
last_metrics_.rtt_ms = rtc::Optional<int>(rtt_ms);
DumpNetworkMetrics();
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
index 82062ab..3713bda 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
@@ -45,6 +45,9 @@
void SetUplinkPacketLossFraction(float uplink_packet_loss_fraction) override;
+ void SetUplinkRecoverablePacketLossFraction(
+ float uplink_recoverable_packet_loss_fraction) override;
+
void SetRtt(int rtt_ms) override;
void SetTargetAudioBitrate(int target_audio_bitrate_bps) override;
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
index db2a466..c434be3 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
@@ -36,7 +36,10 @@
arg.target_audio_bitrate_bps == metric.target_audio_bitrate_bps &&
arg.rtt_ms == metric.rtt_ms &&
arg.overhead_bytes_per_packet == metric.overhead_bytes_per_packet &&
- arg.uplink_packet_loss_fraction == metric.uplink_packet_loss_fraction;
+ arg.uplink_packet_loss_fraction ==
+ metric.uplink_packet_loss_fraction &&
+ arg.uplink_recoverable_packet_loss_fraction ==
+ metric.uplink_recoverable_packet_loss_fraction;
}
MATCHER_P(EncoderRuntimeConfigIs, config, "") {
@@ -127,6 +130,18 @@
states.audio_network_adaptor->SetUplinkPacketLossFraction(kPacketLoss);
}
+TEST(AudioNetworkAdaptorImplTest,
+ UpdateNetworkMetricsIsCalledOnSetUplinkRecoverablePacketLossFraction) {
+ auto states = CreateAudioNetworkAdaptor();
+ constexpr float kRecoverablePacketLoss = 0.1f;
+ Controller::NetworkMetrics check;
+ check.uplink_recoverable_packet_loss_fraction =
+ rtc::Optional<float>(kRecoverablePacketLoss);
+ SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
+ states.audio_network_adaptor->SetUplinkRecoverablePacketLossFraction(
+ kRecoverablePacketLoss);
+}
+
TEST(AudioNetworkAdaptorImplTest, UpdateNetworkMetricsIsCalledOnSetRtt) {
auto states = CreateAudioNetworkAdaptor();
constexpr int kRtt = 100;
@@ -186,6 +201,7 @@
constexpr int kBandwidth = 16000;
constexpr float kPacketLoss = 0.7f;
+ const auto kRecoverablePacketLoss = 0.2f;
constexpr int kRtt = 100;
constexpr int kTargetAudioBitrate = 15000;
constexpr size_t kOverhead = 64;
@@ -205,6 +221,15 @@
DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
states.audio_network_adaptor->SetUplinkPacketLossFraction(kPacketLoss);
+ states.simulated_clock->AdvanceTimeMilliseconds(50);
+ timestamp_check += 50;
+ check.uplink_recoverable_packet_loss_fraction =
+ rtc::Optional<float>(kRecoverablePacketLoss);
+ EXPECT_CALL(*states.mock_debug_dump_writer,
+ DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
+ states.audio_network_adaptor->SetUplinkRecoverablePacketLossFraction(
+ kRecoverablePacketLoss);
+
states.simulated_clock->AdvanceTimeMilliseconds(200);
timestamp_check += 200;
check.rtt_ms = rtc::Optional<int>(kRtt);
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/controller.h b/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
index 7679a58..0ed23c8 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
@@ -23,6 +23,7 @@
~NetworkMetrics();
rtc::Optional<int> uplink_bandwidth_bps;
rtc::Optional<float> uplink_packet_loss_fraction;
+ rtc::Optional<float> uplink_recoverable_packet_loss_fraction;
rtc::Optional<int> target_audio_bitrate_bps;
rtc::Optional<int> rtt_ms;
rtc::Optional<size_t> overhead_bytes_per_packet;
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h
index 338a365..155b749 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h
@@ -81,6 +81,7 @@
// Scoring point is a subset of NetworkMetrics that is used for comparing the
// significance of controllers.
struct ScoringPoint {
+ // TODO(elad.alon): Do we want to experiment with RPLR-based scoring?
ScoringPoint(int uplink_bandwidth_bps, float uplink_packet_loss_fraction);
// Calculate the normalized [0,1] distance between two scoring points.
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto
index f425244..3f9275a 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.proto
@@ -7,14 +7,20 @@
optional float uplink_packet_loss_fraction = 2;
optional int32 target_audio_bitrate_bps = 3;
optional int32 rtt_ms = 4;
+ optional int32 uplink_recoverable_packet_loss_fraction = 5;
}
message EncoderRuntimeConfig {
optional int32 bitrate_bps = 1;
optional int32 frame_length_ms = 2;
+ // Note: This is what we tell the encoder. It doesn't have to reflect
+ // the actual NetworkMetrics; it's subject to our decision.
optional float uplink_packet_loss_fraction = 3;
optional bool enable_fec = 4;
optional bool enable_dtx = 5;
+ // Some encoders can encode fewer channels than the actual input to make
+ // better use of the bandwidth. |num_channels| sets the number of channels
+ // to encode.
optional uint32 num_channels = 6;
}
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
index 7770e65..e0af336 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
@@ -94,6 +94,11 @@
if (metrics.rtt_ms)
dump_metrics->set_rtt_ms(*metrics.rtt_ms);
+ if (metrics.uplink_recoverable_packet_loss_fraction) {
+ dump_metrics->set_uplink_recoverable_packet_loss_fraction(
+ *metrics.uplink_recoverable_packet_loss_fraction);
+ }
+
DumpEventToFile(event, dump_file_.get());
#endif // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
}
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
index 14ddbca..0ad4a1e 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
@@ -26,6 +26,8 @@
~EncoderRuntimeConfig();
rtc::Optional<int> bitrate_bps;
rtc::Optional<int> frame_length_ms;
+ // Note: This is what we tell the encoder. It doesn't have to reflect
+ // the actual NetworkMetrics; it's subject to our decision.
rtc::Optional<float> uplink_packet_loss_fraction;
rtc::Optional<bool> enable_fec;
rtc::Optional<bool> enable_dtx;
@@ -43,6 +45,9 @@
virtual void SetUplinkPacketLossFraction(
float uplink_packet_loss_fraction) = 0;
+ virtual void SetUplinkRecoverablePacketLossFraction(
+ float uplink_recoverable_packet_loss_fraction) = 0;
+
virtual void SetRtt(int rtt_ms) = 0;
virtual void SetTargetAudioBitrate(int target_audio_bitrate_bps) = 0;
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h b/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h
index a826911..104dde6 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h
@@ -26,6 +26,9 @@
MOCK_METHOD1(SetUplinkPacketLossFraction,
void(float uplink_packet_loss_fraction));
+ MOCK_METHOD1(SetUplinkRecoverablePacketLossFraction,
+ void(float uplink_recoverable_packet_loss_fraction));
+
MOCK_METHOD1(SetRtt, void(int rtt_ms));
MOCK_METHOD1(SetTargetAudioBitrate, void(int target_audio_bitrate_bps));
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
index 9e0cf4a..c9d85f8 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
@@ -76,6 +76,9 @@
void AudioEncoder::OnReceivedUplinkPacketLossFraction(
float uplink_packet_loss_fraction) {}
+void AudioEncoder::OnReceivedUplinkRecoverablePacketLossFraction(
+ float uplink_recoverable_packet_loss_fraction) {}
+
void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
OnReceivedUplinkBandwidth(target_audio_bitrate_bps, rtc::Optional<int64_t>());
}
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index 47152f9..b58f964 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -175,6 +175,12 @@
virtual void OnReceivedUplinkPacketLossFraction(
float uplink_packet_loss_fraction);
+ // Provides 1st-order-FEC-recoverable uplink packet loss rate to this encoder
+ // to allow it to adapt.
+ // |uplink_recoverable_packet_loss_fraction| is in the range [0.0, 1.0].
+ virtual void OnReceivedUplinkRecoverablePacketLossFraction(
+ float uplink_recoverable_packet_loss_fraction);
+
// Provides target audio bitrate to this encoder to allow it to adapt.
virtual void OnReceivedTargetAudioBitrate(int target_bps);
diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
index bcb11ee..993fffe 100644
--- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
+++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
@@ -190,6 +190,12 @@
uplink_packet_loss_fraction);
}
+void AudioEncoderCng::OnReceivedUplinkRecoverablePacketLossFraction(
+ float uplink_recoverable_packet_loss_fraction) {
+ speech_encoder_->OnReceivedUplinkRecoverablePacketLossFraction(
+ uplink_recoverable_packet_loss_fraction);
+}
+
void AudioEncoderCng::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
rtc::Optional<int64_t> probing_interval_ms) {
diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
index 0075bd0..827e463 100644
--- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
+++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
@@ -65,6 +65,8 @@
override;
void OnReceivedUplinkPacketLossFraction(
float uplink_packet_loss_fraction) override;
+ void OnReceivedUplinkRecoverablePacketLossFraction(
+ float uplink_recoverable_packet_loss_fraction) override;
void OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
rtc::Optional<int64_t> probing_interval_ms) override;
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index bac963f..103ec9b 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -304,6 +304,15 @@
ApplyAudioNetworkAdaptor();
}
+void AudioEncoderOpus::OnReceivedUplinkRecoverablePacketLossFraction(
+ float uplink_recoverable_packet_loss_fraction) {
+ if (!audio_network_adaptor_)
+ return;
+ audio_network_adaptor_->SetUplinkRecoverablePacketLossFraction(
+ uplink_recoverable_packet_loss_fraction);
+ ApplyAudioNetworkAdaptor();
+}
+
void AudioEncoderOpus::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
rtc::Optional<int64_t> probing_interval_ms) {
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index 3d0483b..15ded47 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -114,6 +114,8 @@
void DisableAudioNetworkAdaptor() override;
void OnReceivedUplinkPacketLossFraction(
float uplink_packet_loss_fraction) override;
+ void OnReceivedUplinkRecoverablePacketLossFraction(
+ float uplink_recoverable_packet_loss_fraction) override;
void OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
rtc::Optional<int64_t> probing_interval_ms) override;
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
index 3faad68..251a104 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
@@ -126,6 +126,12 @@
uplink_packet_loss_fraction);
}
+void AudioEncoderCopyRed::OnReceivedUplinkRecoverablePacketLossFraction(
+ float uplink_recoverable_packet_loss_fraction) {
+ speech_encoder_->OnReceivedUplinkRecoverablePacketLossFraction(
+ uplink_recoverable_packet_loss_fraction);
+}
+
void AudioEncoderCopyRed::OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
rtc::Optional<int64_t> probing_interval_ms) {
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
index b3ec085..b9b46a8 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -53,6 +53,8 @@
override;
void OnReceivedUplinkPacketLossFraction(
float uplink_packet_loss_fraction) override;
+ void OnReceivedUplinkRecoverablePacketLossFraction(
+ float uplink_recoverable_packet_loss_fraction) override;
void OnReceivedUplinkBandwidth(
int target_audio_bitrate_bps,
rtc::Optional<int64_t> probing_interval_ms) override;