Adding OnReceivedOverhead to AudioEncoder.
BUG=webrtc:6762
Review-Url: https://codereview.webrtc.org/2528933002
Cr-Commit-Position: refs/heads/master@{#15457}
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.cc b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
index 16b470b..7b7325d 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.cc
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.cc
@@ -81,6 +81,8 @@
void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
+void AudioEncoder::OnReceivedOverhead(size_t overhead_bytes_per_packet) {}
+
void AudioEncoder::SetReceiverFrameLengthRange(int min_frame_length_ms,
int max_frame_length_ms) {}
diff --git a/webrtc/modules/audio_coding/codecs/audio_encoder.h b/webrtc/modules/audio_coding/codecs/audio_encoder.h
index c913765..5132c2e 100644
--- a/webrtc/modules/audio_coding/codecs/audio_encoder.h
+++ b/webrtc/modules/audio_coding/codecs/audio_encoder.h
@@ -181,6 +181,10 @@
// Provides RTT to this encoder to allow it to adapt.
virtual void OnReceivedRtt(int rtt_ms);
+ // Provides overhead to this encoder to adapt. The overhead is the number of
+ // bytes that will be added to each packet the encoder generates.
+ virtual void OnReceivedOverhead(size_t overhead_bytes_per_packet);
+
// To allow encoder to adapt its frame length, it must be provided the frame
// length range that receivers can accept.
virtual void SetReceiverFrameLengthRange(int min_frame_length_ms,
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 40ad595..af66cd3 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -15,12 +15,14 @@
#include "webrtc/base/analytics/exp_filter.h"
#include "webrtc/base/checks.h"
+#include "webrtc/base/logging.h"
#include "webrtc/base/safe_conversions.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h"
#include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
#include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
namespace webrtc {
@@ -291,10 +293,25 @@
void AudioEncoderOpus::OnReceivedTargetAudioBitrate(
int target_audio_bitrate_bps) {
- if (!audio_network_adaptor_)
- return SetTargetBitrate(target_audio_bitrate_bps);
- audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps);
- ApplyAudioNetworkAdaptor();
+ if (audio_network_adaptor_) {
+ audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps);
+ ApplyAudioNetworkAdaptor();
+ } else if (webrtc::field_trial::FindFullName(
+ "WebRTC-SendSideBwe-WithOverhead") == "Enabled") {
+ if (!overhead_bytes_per_packet_) {
+ LOG(LS_INFO)
+ << "AudioEncoderOpus: Overhead unknown, target audio bitrate "
+ << target_audio_bitrate_bps << " bps is ignored.";
+ return;
+ }
+ const int overhead_bps = static_cast<int>(
+ *overhead_bytes_per_packet_ * 8 * 100 / Num10MsFramesInNextPacket());
+ SetTargetBitrate(std::min(
+ kMaxBitrateBps,
+ std::max(kMinBitrateBps, target_audio_bitrate_bps - overhead_bps)));
+ } else {
+ SetTargetBitrate(target_audio_bitrate_bps);
+ }
}
void AudioEncoderOpus::OnReceivedRtt(int rtt_ms) {
@@ -304,6 +321,16 @@
ApplyAudioNetworkAdaptor();
}
+void AudioEncoderOpus::OnReceivedOverhead(size_t overhead_bytes_per_packet) {
+ if (audio_network_adaptor_) {
+ audio_network_adaptor_->SetOverhead(overhead_bytes_per_packet);
+ ApplyAudioNetworkAdaptor();
+ } else {
+ overhead_bytes_per_packet_ =
+ rtc::Optional<size_t>(overhead_bytes_per_packet);
+ }
+}
+
void AudioEncoderOpus::SetReceiverFrameLengthRange(int min_frame_length_ms,
int max_frame_length_ms) {
// Ensure that |SetReceiverFrameLengthRange| is called before
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 92f2126..0864c7b 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -110,6 +110,7 @@
float uplink_packet_loss_fraction) override;
void OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) override;
void OnReceivedRtt(int rtt_ms) override;
+ void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
void SetReceiverFrameLengthRange(int min_frame_length_ms,
int max_frame_length_ms) override;
rtc::ArrayView<const int> supported_frame_lengths_ms() const {
@@ -159,6 +160,7 @@
std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_;
AudioNetworkAdaptorCreator audio_network_adaptor_creator_;
std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_;
+ rtc::Optional<size_t> overhead_bytes_per_packet_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
};
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index f07279f..9f2c318 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -14,6 +14,7 @@
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_audio_network_adaptor.h"
#include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
+#include "webrtc/test/field_trial.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"
#include "webrtc/system_wrappers/include/clock.h"
@@ -249,7 +250,7 @@
EXPECT_THAT(states.encoder->supported_frame_lengths_ms(), ElementsAre(20));
}
-TEST(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnSetUplinkBandwidth) {
+TEST(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedUplinkBandwidth) {
auto states = CreateCodec(2);
states.encoder->EnableAudioNetworkAdaptor("", nullptr);
@@ -267,7 +268,7 @@
}
TEST(AudioEncoderOpusTest,
- InvokeAudioNetworkAdaptorOnSetUplinkPacketLossFraction) {
+ InvokeAudioNetworkAdaptorOnReceivedUplinkPacketLossFraction) {
auto states = CreateCodec(2);
states.encoder->EnableAudioNetworkAdaptor("", nullptr);
@@ -284,7 +285,8 @@
CheckEncoderRuntimeConfig(states.encoder.get(), config);
}
-TEST(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnSetTargetAudioBitrate) {
+TEST(AudioEncoderOpusTest,
+ InvokeAudioNetworkAdaptorOnReceivedTargetAudioBitrate) {
auto states = CreateCodec(2);
states.encoder->EnableAudioNetworkAdaptor("", nullptr);
@@ -301,7 +303,7 @@
CheckEncoderRuntimeConfig(states.encoder.get(), config);
}
-TEST(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnSetRtt) {
+TEST(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedRtt) {
auto states = CreateCodec(2);
states.encoder->EnableAudioNetworkAdaptor("", nullptr);
@@ -317,6 +319,22 @@
CheckEncoderRuntimeConfig(states.encoder.get(), config);
}
+TEST(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedOverhead) {
+ auto states = CreateCodec(2);
+ states.encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+ auto config = CreateEncoderRuntimeConfig();
+ EXPECT_CALL(**states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+ .WillOnce(Return(config));
+
+ // Since using mock audio network adaptor, any overhead is fine.
+ constexpr size_t kOverhead = 64;
+ EXPECT_CALL(**states.mock_audio_network_adaptor, SetOverhead(kOverhead));
+ states.encoder->OnReceivedOverhead(kOverhead);
+
+ CheckEncoderRuntimeConfig(states.encoder.get(), config);
+}
+
TEST(AudioEncoderOpusTest,
PacketLossFractionSmoothedOnSetUplinkPacketLossFraction) {
auto states = CreateCodec(2);
@@ -343,4 +361,64 @@
EXPECT_FLOAT_EQ(0.05f, states.encoder->packet_loss_rate());
}
+TEST(AudioEncoderOpusTest, DoNotInvokeSetTargetBitrateIfOverheadUnknown) {
+ test::ScopedFieldTrials override_field_trials(
+ "WebRTC-SendSideBwe-WithOverhead/Enabled/");
+
+ auto states = CreateCodec(2);
+
+ states.encoder->OnReceivedTargetAudioBitrate(kDefaultOpusSettings.rate * 2);
+
+ // Since |OnReceivedOverhead| has not been called, the codec bitrate should
+ // not change.
+ EXPECT_EQ(kDefaultOpusSettings.rate, states.encoder->GetTargetBitrate());
+}
+
+TEST(AudioEncoderOpusTest, OverheadRemovedFromTargetAudioBitrate) {
+ test::ScopedFieldTrials override_field_trials(
+ "WebRTC-SendSideBwe-WithOverhead/Enabled/");
+
+ auto states = CreateCodec(2);
+
+ constexpr size_t kOverheadBytesPerPacket = 64;
+ states.encoder->OnReceivedOverhead(kOverheadBytesPerPacket);
+
+ constexpr int kTargetBitrateBps = 40000;
+ states.encoder->OnReceivedTargetAudioBitrate(kTargetBitrateBps);
+
+ int packet_rate = rtc::CheckedDivExact(48000, kDefaultOpusSettings.pacsize);
+ EXPECT_EQ(kTargetBitrateBps -
+ 8 * static_cast<int>(kOverheadBytesPerPacket) * packet_rate,
+ states.encoder->GetTargetBitrate());
+}
+
+TEST(AudioEncoderOpusTest, BitrateBounded) {
+ test::ScopedFieldTrials override_field_trials(
+ "WebRTC-SendSideBwe-WithOverhead/Enabled/");
+
+ constexpr int kMinBitrateBps = 500;
+ constexpr int kMaxBitrateBps = 512000;
+
+ auto states = CreateCodec(2);
+
+ constexpr size_t kOverheadBytesPerPacket = 64;
+ states.encoder->OnReceivedOverhead(kOverheadBytesPerPacket);
+
+ int packet_rate = rtc::CheckedDivExact(48000, kDefaultOpusSettings.pacsize);
+
+ // Set a target rate that is smaller than |kMinBitrateBps| when overhead is
+ // subtracted. The eventual codec rate should be bounded by |kMinBitrateBps|.
+ int target_bitrate =
+ kOverheadBytesPerPacket * 8 * packet_rate + kMinBitrateBps - 1;
+ states.encoder->OnReceivedTargetAudioBitrate(target_bitrate);
+ EXPECT_EQ(kMinBitrateBps, states.encoder->GetTargetBitrate());
+
+ // Set a target rate that is greater than |kMaxBitrateBps| when overhead is
+ // subtracted. The eventual codec rate should be bounded by |kMaxBitrateBps|.
+ target_bitrate =
+ kOverheadBytesPerPacket * 8 * packet_rate + kMaxBitrateBps + 1;
+ states.encoder->OnReceivedTargetAudioBitrate(target_bitrate);
+ EXPECT_EQ(kMaxBitrateBps, states.encoder->GetTargetBitrate());
+}
+
} // namespace webrtc