Moves channel-dependent audio input processing to separate encoder task queue

webrtc/voice_engine/audio_frame_pool.cc

Index: webrtc/voice_engine/audio_frame_pool.cc
diff --git a/webrtc/voice_engine/audio_frame_pool.cc b/webrtc/voice_engine/audio_frame_pool.cc
new file mode 100644
index 0000000000000000000000000000000000000000..fd4b1ed66a593a33f9b786f8e42a591ccca6c332
--- /dev/null
+++ b/webrtc/voice_engine/audio_frame_pool.cc
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2017 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.
+ */
+
+#include "webrtc/voice_engine/audio_frame_pool.h"
+
+namespace webrtc {
+
+AudioFramePool::AudioFramePool(size_t capacity) : audio_frame_queue_(capacity) {
+  for (size_t i = 0; i < capacity; ++i) {
+    std::unique_ptr<AudioFrame> audio_frame(new AudioFrame());
+    Push(std::move(audio_frame));
+  }
+  RTC_DCHECK_EQ(audio_frame_queue_.Capacity(), capacity);
+  RTC_DCHECK_EQ(audio_frame_queue_.Size(), capacity);
+}
+
+AudioFramePool::~AudioFramePool() = default;
+
+void AudioFramePool::Push(std::unique_ptr<AudioFrame> audio_frame) {
+  // Swap "full" audio frame with "empty" in queue.
+  bool result = audio_frame_queue_.Insert(&audio_frame);
+  RTC_DCHECK(result) << "Audio frame pool is full";
+  // TODO(henrika): shall we verify that we swapped with an "empty" frame?
+  RTC_DCHECK(!audio_frame);
+}
+
+std::unique_ptr<AudioFrame> AudioFramePool::Pop() {
+  // Swap "empty" audio frame with "full" from queue.
+  std::unique_ptr<AudioFrame> audio_frame;
+  bool result = audio_frame_queue_.Remove(&audio_frame);
+  if (!result) {
+    // TODO(henrika): if we ever enter this state, it means that audio has been
+    // recorded but we can't ask the pool for frames to place the audio in.
+    // Hence, recorded audio samples will be dropped.
+    LOG(LS_ERROR) << "Audio frame pool is empty";
+  }
+  if (size() < min_size_) {
+    min_size_ = size();
+    LOG(INFO) << "min size: " << min_size_;
+  }
+  return audio_frame;
+}
+
+}  // namespace webrtc