commit 764e364933da2a6ba1629b066d0a897cd5e5c10c
author peah <peah@webrtc.org> Sat Oct 22 05:04:30 2016 -0700
committer Commit bot <commit-bot@chromium.org> Sat Oct 22 12:04:35 2016 +0000
tree a16f508afa382b545f4ec3593901efa87ec4b642
parent 897497361e18c604ad46b949c7b478db9709734f

Several subcomponents inside APM copy render audio from
the render side to the capture side using the same
pattern. Currently this is done independently for the
submodules.

This CL moves the the AEC functionality for this into
APM.

BUG=webrtc:5298, webrtc:6540

Review-Url: https://codereview.webrtc.org/2427553003
Cr-Commit-Position: refs/heads/master@{#14726}

webrtc/modules/audio_processing/audio_processing_impl.h

diff --git a/webrtc/modules/audio_processing/audio_processing_impl.h b/webrtc/modules/audio_processing/audio_processing_impl.h
index 21bc588..d6af64d 100644
--- a/webrtc/modules/audio_processing/audio_processing_impl.h
+++ b/webrtc/modules/audio_processing/audio_processing_impl.h
@@ -19,9 +19,11 @@
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/gtest_prod_util.h"
 #include "webrtc/base/ignore_wundef.h"
+#include "webrtc/base/swap_queue.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_processing/audio_buffer.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/render_queue_item_verifier.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
@@ -233,6 +235,12 @@
       EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
   void InitializeLevelController() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
 
+  void EmptyQueuedRenderAudio();
+  void AllocateRenderQueue()
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
+  void QueueRenderAudio(const AudioBuffer* audio)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
+
   // Capture-side exclusive methods possibly running APM in a multi-threaded
   // manner that are called with the render lock already acquired.
   int ProcessCaptureStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
@@ -362,6 +370,15 @@
     std::unique_ptr<AudioConverter> render_converter;
     std::unique_ptr<AudioBuffer> render_audio;
   } render_ GUARDED_BY(crit_render_);
+
+  size_t render_queue_element_max_size_ GUARDED_BY(crit_render_)
+      GUARDED_BY(crit_capture_) = 0;
+  std::vector<float> render_queue_buffer_ GUARDED_BY(crit_render_);
+  std::vector<float> capture_queue_buffer_ GUARDED_BY(crit_capture_);
+
+  // Lock protection not needed.
+  std::unique_ptr<SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>>
+      render_signal_queue_;
 };
 
 }  // namespace webrtc