commit d95138b684b82527b8ccd1ccc631b5ff550de320
author Jakob Ivarsson <jakobi@webrtc.org> Thu Jun 18 08:19:11 2020
committer Commit Bot <commit-bot@chromium.org> Wed Jun 24 11:22:21 2020
tree 9605d47aa82dc8ad9d7d9c3a8957a49e8747446a
parent ee5b6de5aa2bf48c0dd27d6a7a2cb3a03b359f12

Make stable target adaptation enabled by default.

This will result in slightly higher encode bitrates and longer frame
lengths compared to using the smoothing filter.

Bug: webrtc:10981
Change-Id: I64704196c56b0ad910895c908baad38c994a971b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177425
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31556}

modules/audio_coding/codecs/opus/audio_encoder_opus.cc

diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
index ff5d71b..220e96f 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
@@ -357,7 +357,7 @@
     : payload_type_(payload_type),
       send_side_bwe_with_overhead_(
           webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
-      use_stable_target_for_adaptation_(webrtc::field_trial::IsEnabled(
+      use_stable_target_for_adaptation_(!webrtc::field_trial::IsDisabled(
           "WebRTC-Audio-StableTargetAdaptation")),
       adjust_bandwidth_(
           webrtc::field_trial::IsEnabled("WebRTC-AdjustOpusBandwidth")),

modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc

diff --git a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index 1c73eaf..1cbc4a3 100644
--- a/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -254,6 +254,8 @@
 
 TEST_P(AudioEncoderOpusTest,
        InvokeAudioNetworkAdaptorOnReceivedUplinkBandwidth) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-StableTargetAdaptation/Disabled/");
   auto states = CreateCodec(sample_rate_hz_, 2);
   states->encoder->EnableAudioNetworkAdaptor("", nullptr);
 
@@ -275,6 +277,28 @@
   CheckEncoderRuntimeConfig(states->encoder.get(), config);
 }
 
+TEST_P(AudioEncoderOpusTest,
+       InvokeAudioNetworkAdaptorOnReceivedUplinkAllocation) {
+  auto states = CreateCodec(sample_rate_hz_, 2);
+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
+
+  auto config = CreateEncoderRuntimeConfig();
+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
+      .WillOnce(Return(config));
+
+  BitrateAllocationUpdate update;
+  update.target_bitrate = DataRate::BitsPerSec(30000);
+  update.stable_target_bitrate = DataRate::BitsPerSec(20000);
+  update.bwe_period = TimeDelta::Millis(200);
+  EXPECT_CALL(*states->mock_audio_network_adaptor,
+              SetTargetAudioBitrate(update.target_bitrate.bps()));
+  EXPECT_CALL(*states->mock_audio_network_adaptor,
+              SetUplinkBandwidth(update.stable_target_bitrate.bps()));
+  states->encoder->OnReceivedUplinkAllocation(update);
+
+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
+}
+
 TEST_P(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedRtt) {
   auto states = CreateCodec(sample_rate_hz_, 2);
   states->encoder->EnableAudioNetworkAdaptor("", nullptr);
@@ -501,6 +525,8 @@
 }
 
 TEST_P(AudioEncoderOpusTest, UpdateUplinkBandwidthInAudioNetworkAdaptor) {
+  test::ScopedFieldTrials override_field_trials(
+      "WebRTC-Audio-StableTargetAdaptation/Disabled/");
   auto states = CreateCodec(sample_rate_hz_, 2);
   states->encoder->EnableAudioNetworkAdaptor("", nullptr);
   const size_t opus_rate_khz = rtc::CheckedDivExact(sample_rate_hz_, 1000);