Fixed the render queue item size preallocation and verification, moved constant declarations, remov…

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 15 matching lines @@
     case GainControl::kAdaptiveAnalog:
       return kAgcModeAdaptiveAnalog;
     case GainControl::kAdaptiveDigital:
       return kAgcModeAdaptiveDigital;
     case GainControl::kFixedDigital:
       return kAgcModeFixedDigital;
   }
   assert(false);
   return -1;
 }
+
+// Maximum length that a frame of samples can have.
+static const size_t kMaxAllowedValuesOfSamplesPerFrame = 160;
+// Maximum number of frames to buffer in the render queue.
+// TODO(peah): Decrease this once we properly handle hugely unbalanced
+// reverse and forward call numbers.
+static const size_t kMaxNumFramesToBuffer = 100;
+
 }  // namespace
 
-const size_t GainControlImpl::kAllowedValuesOfSamplesPerFrame1;
-const size_t GainControlImpl::kAllowedValuesOfSamplesPerFrame2;
-
 GainControlImpl::GainControlImpl(const AudioProcessing* apm,
                                  CriticalSectionWrapper* crit)
     : ProcessingComponent(),
       apm_(apm),
       crit_(crit),
       mode_(kAdaptiveAnalog),
       minimum_capture_level_(0),
       maximum_capture_level_(255),
       limiter_enabled_(true),
       target_level_dbfs_(3),
       compression_gain_db_(9),
       analog_capture_level_(0),
       was_analog_level_set_(false),
       stream_is_saturated_(false),
-      render_queue_element_max_size_(0) {
-  AllocateRenderQueue();
-}
+      render_queue_element_max_size_(0) {}
 
 GainControlImpl::~GainControlImpl() {}
 
 int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) {
   if (!is_component_enabled()) {
     return apm_->kNoError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
 
@@ skipping 142 matching lines @@
 
     analog_capture_level_ /= num_handles();
   }
 
   was_analog_level_set_ = false;
   return apm_->kNoError;
 }
 
 // TODO(ajm): ensure this is called under kAdaptiveAnalog.
 int GainControlImpl::set_stream_analog_level(int level) {
-  // TODO(peah): Verify that this is really needed to do the reading
-  // here as well as in ProcessStream. It works since these functions
-  // are called from the same thread, but it is not nice to do it in two
-  // places if not needed.
-  ReadQueuedRenderData();
-
   CriticalSectionScoped crit_scoped(crit_);
   was_analog_level_set_ = true;
   if (level < minimum_capture_level_ || level > maximum_capture_level_) {
     return apm_->kBadParameterError;
   }
   analog_capture_level_ = level;
 
   return apm_->kNoError;
 }
 
@@ skipping 106 matching lines @@
 
   AllocateRenderQueue();
 
   const int n = num_handles();
   RTC_CHECK_GE(n, 0) << "Bad number of handles: " << n;
   capture_levels_.assign(n, analog_capture_level_);
   return apm_->kNoError;
 }
 
 void GainControlImpl::AllocateRenderQueue() {
-  const size_t max_frame_size = std::max<size_t>(
-      kAllowedValuesOfSamplesPerFrame1, kAllowedValuesOfSamplesPerFrame2);
+  const size_t new_render_queue_element_max_size =
+      std::max<size_t>(static_cast<size_t>(1),
+                       kMaxAllowedValuesOfSamplesPerFrame * num_handles());
 
-  const size_t new_render_queue_element_max_size = std::max<size_t>(
-      static_cast<size_t>(1), (max_frame_size * num_handles()));
-
-  if (new_render_queue_element_max_size > render_queue_element_max_size_) {
+  if (render_queue_element_max_size_ < new_render_queue_element_max_size) {
+    render_queue_element_max_size_ = new_render_queue_element_max_size;
     std::vector<int16_t> template_queue_element(render_queue_element_max_size_);
 
     render_signal_queue_.reset(
         new SwapQueue<std::vector<int16_t>, RenderQueueItemVerifier<int16_t>>(
             kMaxNumFramesToBuffer, template_queue_element,
             RenderQueueItemVerifier<int16_t>(render_queue_element_max_size_)));
+
+    render_queue_buffer_.resize(render_queue_element_max_size_);
+    capture_queue_buffer_.resize(render_queue_element_max_size_);
   } else {
     render_signal_queue_->Clear();
   }
-
-  render_queue_buffer_.resize(new_render_queue_element_max_size);
-  capture_queue_buffer_.resize(new_render_queue_element_max_size);
 }
 
 void* GainControlImpl::CreateHandle() const {
   return WebRtcAgc_Create();
 }
 
 void GainControlImpl::DestroyHandle(void* handle) const {
   WebRtcAgc_Free(static_cast<Handle*>(handle));
 }
 
@@ skipping 23 matching lines @@
   return apm_->num_output_channels();
 }
 
 int GainControlImpl::GetHandleError(void* handle) const {
   // The AGC has no get_error() function.
   // (Despite listing errors in its interface...)
   assert(handle != NULL);
   return apm_->kUnspecifiedError;
 }
 }  // namespace webrtc