Index: webrtc/modules/audio_processing/aec3/render_delay_controller.h |
diff --git a/webrtc/modules/audio_processing/aec3/render_delay_controller.h b/webrtc/modules/audio_processing/aec3/render_delay_controller.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..b22a5fde16d3503c8976108e96e285bc9cc0ea81 |
--- /dev/null |
+++ b/webrtc/modules/audio_processing/aec3/render_delay_controller.h |
@@ -0,0 +1,40 @@ |
+/* |
+ * 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. |
+ */ |
+ |
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_ |
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_ |
+ |
+#include "webrtc/base/array_view.h" |
+#include "webrtc/base/optional.h" |
+#include "webrtc/modules/audio_processing/aec3/render_delay_buffer.h" |
+#include "webrtc/modules/audio_processing/logging/apm_data_dumper.h" |
+ |
+namespace webrtc { |
+ |
+// Class for aligning the render and capture signal using a RenderDelayBuffer. |
+class RenderDelayController { |
+ public: |
+ static RenderDelayController* Create( |
+ int sample_rate_hz, |
+ const RenderDelayBuffer& render_delay_buffer); |
+ virtual ~RenderDelayController() = default; |
+ |
+ // Aligns the render buffer content with the capture signal. |
+ virtual size_t GetDelay(rtc::ArrayView<const float> capture) = 0; |
+ |
+ // Analyzes the render signal and returns false if there is a buffer overrun. |
+ virtual bool AnalyzeRender(rtc::ArrayView<const float> render) = 0; |
+ |
+ // Returns an approximate value for the headroom in the buffer alignment. |
+ virtual rtc::Optional<size_t> AlignmentHeadroomSamples() const = 0; |
+}; |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_ |