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Unified Diff: webrtc/modules/audio_processing/aec3/render_delay_controller.h

Issue 2611223003: Adding second layer of the echo canceller 3 functionality. (Closed)
Patch Set: Disabled DEATH tests that were causing memory leakage reports on test bots Created 3 years, 11 months ago
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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_

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