Fixed a potential deadlock problem in the AGC.

webrtc/modules/audio_processing/gain_control_impl.cc

 /*
  *  Copyright (c) 2012 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 318 matching lines @@
   return AudioProcessing::kNoError;
 }
 
 GainControl::Mode GainControlImpl::mode() const {
   rtc::CritScope cs(crit_capture_);
   return mode_;
 }
 
 int GainControlImpl::set_analog_level_limits(int minimum,
                                              int maximum) {
-  rtc::CritScope cs(crit_capture_);
   if (minimum < 0) {
     return AudioProcessing::kBadParameterError;
   }
 
   if (maximum > 65535) {
     return AudioProcessing::kBadParameterError;
   }
 
   if (maximum < minimum) {
     return AudioProcessing::kBadParameterError;
   }
 
-  minimum_capture_level_ = minimum;
-  maximum_capture_level_ = maximum;
+  size_t num_proc_channels_local = 0u;
+  int sample_rate_hz_local = 0;
+  {
+    rtc::CritScope cs(crit_capture_);
 
-  RTC_DCHECK(num_proc_channels_);
-  RTC_DCHECK(sample_rate_hz_);
-  Initialize(*num_proc_channels_, *sample_rate_hz_);
+    minimum_capture_level_ = minimum;
+    maximum_capture_level_ = maximum;
+
+    RTC_DCHECK(num_proc_channels_);
+    RTC_DCHECK(sample_rate_hz_);
+    num_proc_channels_local = *num_proc_channels_;
+    sample_rate_hz_local = *sample_rate_hz_;
+  }
+  Initialize(num_proc_channels_local, sample_rate_hz_local);
   return AudioProcessing::kNoError;
 }
 
 int GainControlImpl::analog_level_minimum() const {
   rtc::CritScope cs(crit_capture_);
   return minimum_capture_level_;
 }
 
 int GainControlImpl::analog_level_maximum() const {
   rtc::CritScope cs(crit_capture_);
@@ skipping 117 matching lines @@
   for (auto& gain_controller : gain_controllers_) {
     const int handle_error =
         WebRtcAgc_set_config(gain_controller->state(), config);
     if (handle_error != AudioProcessing::kNoError) {
       error = handle_error;
     }
   }
   return error;
 }
 }  // namespace webrtc