Adding activation logic of the new APM level control functionality using MediaConstraints

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 542 matching lines @@
     options.stereo_swapping = rtc::Optional<bool>(false);
     options.audio_jitter_buffer_max_packets = rtc::Optional<int>(50);
     options.audio_jitter_buffer_fast_accelerate = rtc::Optional<bool>(false);
     options.typing_detection = rtc::Optional<bool>(true);
     options.adjust_agc_delta = rtc::Optional<int>(0);
     options.experimental_agc = rtc::Optional<bool>(false);
     options.extended_filter_aec = rtc::Optional<bool>(false);
     options.delay_agnostic_aec = rtc::Optional<bool>(false);
     options.experimental_ns = rtc::Optional<bool>(false);
     options.intelligibility_enhancer = rtc::Optional<bool>(false);
+    options.level_control = rtc::Optional<bool>(false);
     bool error = ApplyOptions(options);
     RTC_DCHECK(error);
   }
 
   SetDefaultDevices();
 }
 
 WebRtcVoiceEngine::~WebRtcVoiceEngine() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "WebRtcVoiceEngine::~WebRtcVoiceEngine";
@@ skipping 103 matching lines @@
 #endif
     if (ec_mode == webrtc::kEcAecm) {
       bool cn = options.aecm_generate_comfort_noise.value_or(false);
       if (voep->SetAecmMode(aecm_mode, cn) != 0) {
         LOG_RTCERR2(SetAecmMode, aecm_mode, cn);
         return false;
       }
     }
   }
 
+  // Use optional to avoid uneccessary calls to BuiltInAGCIsAvailable while
+  // complying with the unittest requirements of only 1 call per test.
+  rtc::Optional<bool> built_in_agc_avaliable;
+  if (options.level_control) {
+    if (!built_in_agc_avaliable) {
+      built_in_agc_avaliable =
+          rtc::Optional<bool>(adm()->BuiltInAGCIsAvailable());
+    }
+    RTC_DCHECK(built_in_agc_avaliable);
+    if (*built_in_agc_avaliable) {
+      // Disable internal software level control if built-in AGC is enabled,
+      // i.e., replace the software AGC with the built-in AGC.
+      options.level_control = rtc::Optional<bool>(false);
+    }
+  }
+
   if (options.auto_gain_control) {
-    const bool built_in_agc = adm()->BuiltInAGCIsAvailable();
-    if (built_in_agc) {
+    if (!built_in_agc_avaliable) {
+      built_in_agc_avaliable =
+          rtc::Optional<bool>(adm()->BuiltInAGCIsAvailable());
+    }
+    RTC_DCHECK(built_in_agc_avaliable);
+    if (*built_in_agc_avaliable) {
       if (adm()->EnableBuiltInAGC(*options.auto_gain_control) == 0 &&
           *options.auto_gain_control) {
         // Disable internal software AGC if built-in AGC is enabled,
         // i.e., replace the software AGC with the built-in AGC.
         options.auto_gain_control = rtc::Optional<bool>(false);
         LOG(LS_INFO) << "Disabling AGC since built-in AGC will be used instead";
       }
     }
     if (voep->SetAgcStatus(*options.auto_gain_control, agc_mode) == -1) {
       LOG_RTCERR2(SetAgcStatus, *options.auto_gain_control, agc_mode);
@@ skipping 136 matching lines @@
         new webrtc::ExperimentalNs(*experimental_ns_));
   }
 
   if (intelligibility_enhancer_) {
     LOG(LS_INFO) << "Intelligibility Enhancer is enabled? "
                  << *intelligibility_enhancer_;
     config.Set<webrtc::Intelligibility>(
         new webrtc::Intelligibility(*intelligibility_enhancer_));
   }
 
+  if (options.level_control) {
+    level_control_ = options.level_control;
+  }
+
+  LOG(LS_INFO) << "Level control: "
+               << (!!level_control_ ? *level_control_ : -1);
+  if (level_control_) {
+    config.Set<webrtc::LevelControl>(new webrtc::LevelControl(*level_control_));
+  }
+
   // We check audioproc for the benefit of tests, since FakeWebRtcVoiceEngine
   // returns NULL on audio_processing().
   webrtc::AudioProcessing* audioproc = voe_wrapper_->base()->audio_processing();
   if (audioproc) {
     audioproc->SetExtraOptions(config);
   }
 
   if (options.recording_sample_rate) {
     LOG(LS_INFO) << "Recording sample rate is "
                  << *options.recording_sample_rate;
@@ skipping 1752 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE