1 files changed, 25 insertions, 0 deletions
diff --git a/webrtc/modules/audio_processing/aec3/downsampled_render_buffer.cc b/webrtc/modules/audio_processing/aec3/downsampled_render_buffer.cc
new file mode 100644
index 0000000..c105911
--- /dev/null
+++ b/webrtc/modules/audio_processing/aec3/downsampled_render_buffer.cc
@@ -0,0 +1,25 @@
+/*
+ *  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 "modules/audio_processing/aec3/downsampled_render_buffer.h"
+
+#include <algorithm>
+
+namespace webrtc {
+
+DownsampledRenderBuffer::DownsampledRenderBuffer(size_t downsampled_buffer_size)
+    : size(static_cast<int>(downsampled_buffer_size)),
+      buffer(downsampled_buffer_size, 0.f) {
+  std::fill(buffer.begin(), buffer.end(), 0.f);
+}
+
+DownsampledRenderBuffer::~DownsampledRenderBuffer() = default;
+
+}  // namespace webrtc