Lock scheme #5: Applied the render queueing to the agc

webrtc/modules/audio_processing/gain_control_impl.h

Index: webrtc/modules/audio_processing/gain_control_impl.h
diff --git a/webrtc/modules/audio_processing/gain_control_impl.h b/webrtc/modules/audio_processing/gain_control_impl.h
index f24d200cf2216a05d4c162a655bcf8170d5bea37..1573aea3d1bfbbca7291408b4b40739e2bc06d64 100644
--- a/webrtc/modules/audio_processing/gain_control_impl.h
+++ b/webrtc/modules/audio_processing/gain_control_impl.h
@@ -13,11 +13,34 @@
 
 #include <vector>
 
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_audio/swap_queue.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/audio_processing/processing_component.h"
 
 namespace webrtc {
 
+namespace {
+// Functor to use when supplying a verifier function for the queue item
+// verifcation.
+class AgcRenderQueueItemVerifier {
+ public:
+  explicit AgcRenderQueueItemVerifier(size_t allowed_size_1,
+                                      size_t allowed_size_2)
+      : allowed_size_1_(allowed_size_1), allowed_size_2_(allowed_size_2) {}
+
+  bool operator()(const std::vector<int16_t>& v) const {
+    return (((v.size() == allowed_size_1_) || (v.size() == allowed_size_2_)) &&

[the sun, 2015/10/27 16:42:49]

Like said in another review: you only need to assert the capacity, not the size.

Besides, do you really need the two different sizes? They don't mean much unless
matched to the sample rate. You could just verify each vector has capacity for
the larger buffers.

[peah-webrtc, 2015/10/29 14:05:23]

Good find!  This has now been refactored.

Done.

+            (v.capacity() >= std::max(allowed_size_1_, allowed_size_2_)));
+  }
+
+ private:
+  size_t allowed_size_1_;
+  size_t allowed_size_2_;
+};
+
+}  // namespace anonymous
+
 class AudioBuffer;
 class CriticalSectionWrapper;
 
@@ -41,7 +64,16 @@ class GainControlImpl : public GainControl,
   bool is_limiter_enabled() const override;
   Mode mode() const override;
 
+  // Reads render side data that has been queued on the render call.
+  void ReadQueuedRenderData();
+
  private:
+  static const size_t kAllowedValuesOfSamplesPerFrame1 = 80;
+  static const size_t kAllowedValuesOfSamplesPerFrame2 = 160;
+  // TODO(peah): Decrease this once we properly handle hugely unbalanced
+  // reverse and forward call numbers.
+  static const size_t kMaxNumFramesToBuffer = 100;
+
   // GainControl implementation.
   int Enable(bool enable) override;
   int set_stream_analog_level(int level) override;
@@ -64,6 +96,8 @@ class GainControlImpl : public GainControl,
   int num_handles_required() const override;
   int GetHandleError(void* handle) const override;
 
+  void AllocateRenderQueue();
+
   const AudioProcessing* apm_;
   CriticalSectionWrapper* crit_;
   Mode mode_;
@@ -76,6 +110,12 @@ class GainControlImpl : public GainControl,
   int analog_capture_level_;
   bool was_analog_level_set_;
   bool stream_is_saturated_;
+
+  size_t render_queue_element_max_size_;
+  std::vector<int16_t> render_queue_buffer_;
+  std::vector<int16_t> capture_queue_buffer_;
+  rtc::scoped_ptr<SwapQueue<std::vector<int16_t>, AgcRenderQueueItemVerifier> >
+      render_signal_queue_;
 };
 }  // namespace webrtc