commit 17e14fdf3497506c818f61f6f6415b616fb97178
author Alessio Bazzica <alessiob@webrtc.org> Wed Dec 07 17:08:45 2022 +0100
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Wed Dec 07 21:47:45 2022 +0000
tree d557d5547fc01eadd642c348802f61234ad928cc
parent 3c529893e008e8fa1bebfe0565215a4232933efe

APM AGC2: consolidate `GainController2`

Now that `InputVolumeController` is finalized, it's time to
consolidate AGC2.

Main changes:
- Remove `AdaptiveDigitalGainController`: it's too simple to justify
  a dedicated class and some components of it are also used by
  `InputVolumeController`
- Remove unwanted temporal dependency: make `InputVolumeController`
  adapt the volume based on the current speech level estimation and
  not on the estimation from the previous frame

Tested: AGC2 adaptive digital bit-exactness verified

Bug: webrtc:7494
Change-Id: I175c2741cafc52be81794219c996a3824c3bbf5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280560
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38841}

modules/audio_processing/audio_processing_impl_unittest.cc

diff --git a/modules/audio_processing/audio_processing_impl_unittest.cc b/modules/audio_processing/audio_processing_impl_unittest.cc
index 346b5f5..7a45c45 100644
--- a/modules/audio_processing/audio_processing_impl_unittest.cc
+++ b/modules/audio_processing/audio_processing_impl_unittest.cc
@@ -648,11 +648,10 @@
   rtc::scoped_refptr<AudioProcessing> apm = AudioProcessingBuilder().Create();
   ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError);
   webrtc::AudioProcessing::Config apm_config;
-  // Disable AGC1 analog.
   apm_config.gain_controller1.enabled = false;
-  // Enable AGC2 digital.
   apm_config.gain_controller2.enabled = true;
   apm_config.gain_controller2.adaptive_digital.enabled = true;
+  apm_config.transient_suppression.enabled = true;
   apm->ApplyConfig(apm_config);
   constexpr int kSampleRateHz = 48000;
   constexpr int kNumChannels = 1;
@@ -680,11 +679,10 @@
   rtc::scoped_refptr<AudioProcessing> apm = AudioProcessingBuilder().Create();
   ASSERT_EQ(apm->Initialize(), AudioProcessing::kNoError);
   webrtc::AudioProcessing::Config apm_config;
-  // Disable AGC1 analog.
   apm_config.gain_controller1.enabled = false;
-  // Enable AGC2 digital.
   apm_config.gain_controller2.enabled = true;
   apm_config.gain_controller2.adaptive_digital.enabled = true;
+  apm_config.transient_suppression.enabled = true;
   apm->ApplyConfig(apm_config);
   constexpr int kSampleRateHz = 48000;
   constexpr int kNumChannels = 1;