Moved the AGC render sample queue into the audio processing module

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 @@
       return kAgcModeAdaptiveAnalog;
     case GainControl::kAdaptiveDigital:
       return kAgcModeAdaptiveDigital;
     case GainControl::kFixedDigital:
       return kAgcModeFixedDigital;
   }
   RTC_DCHECK(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
 
 class GainControlImpl::GainController {
  public:
   explicit GainController() {
     state_ = WebRtcAgc_Create();
     RTC_CHECK(state_);
   }
 
   ~GainController() {
@@ skipping 43 matching lines @@
     : crit_render_(crit_render),
       crit_capture_(crit_capture),
       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) {
+      stream_is_saturated_(false) {
   RTC_DCHECK(crit_render);
   RTC_DCHECK(crit_capture);
 }
 
 GainControlImpl::~GainControlImpl() {}
 
-int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) {
-  rtc::CritScope cs(crit_render_);
-  if (!enabled_) {
-    return AudioProcessing::kNoError;
-  }
-
-  RTC_DCHECK_GE(160u, audio->num_frames_per_band());
-
-  render_queue_buffer_.resize(0);
-  for (auto& gain_controller : gain_controllers_) {
-    int err = WebRtcAgc_GetAddFarendError(gain_controller->state(),
-                                          audio->num_frames_per_band());
-
-    if (err != AudioProcessing::kNoError) {
-      return AudioProcessing::kUnspecifiedError;
-    }
-
-    // Buffer the samples in the render queue.
-    render_queue_buffer_.insert(
-        render_queue_buffer_.end(), audio->mixed_low_pass_data(),
-        (audio->mixed_low_pass_data() + audio->num_frames_per_band()));
-  }
-
-  // Insert the samples into the queue.
-  if (!render_signal_queue_->Insert(&render_queue_buffer_)) {
-    // The data queue is full and needs to be emptied.
-    ReadQueuedRenderData();
-
-    // Retry the insert (should always work).
-    RTC_DCHECK_EQ(render_signal_queue_->Insert(&render_queue_buffer_), true);
-  }
-
-  return AudioProcessing::kNoError;
-}
-
-// Read chunks of data that were received and queued on the render side from
-// a queue. All the data chunks are buffered into the farend signal of the AGC.
-void GainControlImpl::ReadQueuedRenderData() {
-  rtc::CritScope cs(crit_capture_);
-
+void GainControlImpl::ProcessRenderAudio(
+    rtc::ArrayView<const int16_t> packed_render_audio) {
+  rtc::CritScope cs_capture(crit_capture_);
   if (!enabled_) {
     return;
   }
 
-  while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
-    size_t buffer_index = 0;
-    RTC_DCHECK(num_proc_channels_);
-    RTC_DCHECK_LT(0ul, *num_proc_channels_);
-    const size_t num_frames_per_band =
-        capture_queue_buffer_.size() / (*num_proc_channels_);
-    for (auto& gain_controller : gain_controllers_) {
-      WebRtcAgc_AddFarend(gain_controller->state(),
-                          &capture_queue_buffer_[buffer_index],
-                          num_frames_per_band);
-
-      buffer_index += num_frames_per_band;
-    }
+  for (auto& gain_controller : gain_controllers_) {
+    WebRtcAgc_AddFarend(gain_controller->state(), packed_render_audio.data(),
+                        packed_render_audio.size());
   }
 }
 
+void GainControlImpl::PackRenderAudioBuffer(
+    AudioBuffer* audio,
+    std::vector<int16_t>* packed_buffer) {
+  RTC_DCHECK_GE(160u, audio->num_frames_per_band());
+
+  packed_buffer->clear();
+  packed_buffer->insert(
+      packed_buffer->end(), audio->mixed_low_pass_data(),
+      (audio->mixed_low_pass_data() + audio->num_frames_per_band()));
+}
+
 int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
   rtc::CritScope cs(crit_capture_);
 
   if (!enabled_) {
     return AudioProcessing::kNoError;
   }
 
   RTC_DCHECK(num_proc_channels_);
   RTC_DCHECK_GE(160u, audio->num_frames_per_band());
   RTC_DCHECK_EQ(audio->num_channels(), *num_proc_channels_);
@@ skipping 256 matching lines @@
   gain_controllers_.resize(*num_proc_channels_);
   for (auto& gain_controller : gain_controllers_) {
     if (!gain_controller) {
       gain_controller.reset(new GainController());
     }
     gain_controller->Initialize(minimum_capture_level_, maximum_capture_level_,
                                 mode_, *sample_rate_hz_, analog_capture_level_);
   }
 
   Configure();
-
-  AllocateRenderQueue();
-}
-
-void GainControlImpl::AllocateRenderQueue() {
-  rtc::CritScope cs_render(crit_render_);
-  rtc::CritScope cs_capture(crit_capture_);
-
-  RTC_DCHECK(num_proc_channels_);
-  const size_t new_render_queue_element_max_size = std::max<size_t>(
-      static_cast<size_t>(1),
-      kMaxAllowedValuesOfSamplesPerFrame * (*num_proc_channels_));
-
-  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();
-  }
 }
 
 int GainControlImpl::Configure() {
   rtc::CritScope cs_render(crit_render_);
   rtc::CritScope cs_capture(crit_capture_);
   WebRtcAgcConfig config;
   // TODO(ajm): Flip the sign here (since AGC expects a positive value) if we
   //            change the interface.
   //RTC_DCHECK_LE(target_level_dbfs_, 0);
   //config.targetLevelDbfs = static_cast<int16_t>(-target_level_dbfs_);
   config.targetLevelDbfs = static_cast<int16_t>(target_level_dbfs_);
   config.compressionGaindB =
       static_cast<int16_t>(compression_gain_db_);
   config.limiterEnable = limiter_enabled_;
 
   int error = AudioProcessing::kNoError;
   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