commit b190ca9e70ca448e36ec9951c46faedd5cd1dbb9
author Alessio Bazzica <alessiob@webrtc.org> Mon Sep 05 16:04:31 2022 +0200
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Fri Sep 09 10:34:58 2022 +0000
tree dcb77d3ebe403b81471c320100c24afed5d95fcd
parent c1e7080e5142c953434aeb39b91e2125a666e1b9

Disable Analog AGC based on the APM config

Fixing a bug due to which the analog controller could not be disabled.
AudioProcessing::Config::GainController1::AnalogGainController::enabled
was ignored and therefore `recommended_stream_analog_level_locked()` in
APM was returning the level recommended by `AgcManagerDirect`.

When the analog controller is disabled, `stream_analog_level()` now
returns the last value set via `set_stream_analog_level()`.
However, the analog controller code is still running and, in particular,
the existing metrics are reported as if the controller were enabled.
This choice was made to reduce the risks of adding bugs in the digital
compression gain selection part, which is tied to the analog
controller. The metric drawback will be solved in a follow-up CL.

Additional changes:
- log `WebRTC.Audio.GainController.Analog.Enabled` when
AGC1 is created or when its config changes
- first step to replace "analog level" with "input volume"

Bug: webrtc:7909, b/180019868
Change-Id: I28ce9556dd98f3dd9ad546799406c55478730435
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270663
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38044}

modules/audio_processing/audio_processing_impl.cc

diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index 57d51a2..5256f9f 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -1374,7 +1374,7 @@
     if (config_.capture_level_adjustment.analog_mic_gain_emulation.enabled) {
       if (submodules_.agc_manager) {
         submodules_.capture_levels_adjuster->SetAnalogMicGainLevel(
-            submodules_.agc_manager->stream_analog_level());
+            submodules_.agc_manager->recommended_analog_level());
       } else if (submodules_.gain_control) {
         submodules_.capture_levels_adjuster->SetAnalogMicGainLevel(
             submodules_.gain_control->stream_analog_level());
@@ -1641,7 +1641,7 @@
   }
 
   if (submodules_.agc_manager) {
-    return submodules_.agc_manager->stream_analog_level();
+    return submodules_.agc_manager->recommended_analog_level();
   }
 
   if (submodules_.gain_control) {
@@ -1852,6 +1852,10 @@
     return;
   }
 
+  RTC_HISTOGRAM_BOOLEAN(
+      "WebRTC.Audio.GainController.Analog.Enabled",
+      config_.gain_controller1.analog_gain_controller.enabled);
+
   if (!submodules_.gain_control) {
     submodules_.gain_control.reset(new GainControlImpl());
   }
@@ -1887,7 +1891,7 @@
     int stream_analog_level = -1;
     const bool re_creation = !!submodules_.agc_manager;
     if (re_creation) {
-      stream_analog_level = submodules_.agc_manager->stream_analog_level();
+      stream_analog_level = submodules_.agc_manager->recommended_analog_level();
     }
     submodules_.agc_manager.reset(new AgcManagerDirect(
         num_proc_channels(), config_.gain_controller1.analog_gain_controller));