Index: webrtc/modules/audio_processing/aec3/render_delay_buffer.h |
diff --git a/webrtc/modules/audio_processing/aec3/render_delay_buffer.h b/webrtc/modules/audio_processing/aec3/render_delay_buffer.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..0fe2c9e677b8d5eeda95e91dfe2d17b6726785f6 |
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+++ b/webrtc/modules/audio_processing/aec3/render_delay_buffer.h |
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+/* |
+ * 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_BUFFER_H_ |
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_ |
+ |
+#include <stddef.h> |
+#include <vector> |
+ |
+namespace webrtc { |
+ |
+// Class for buffering the incoming render blocks such that these may be |
+// extracted with a specified delay. |
+class RenderDelayBuffer { |
+ public: |
+ static RenderDelayBuffer* Create(size_t size_blocks, |
+ size_t num_bands, |
+ size_t max_api_jitter_blocks); |
+ virtual ~RenderDelayBuffer() = default; |
+ |
+ // Swaps a block into the buffer (the content of block is destroyed) and |
+ // returns true if the insert is successful. |
+ virtual bool Insert(std::vector<std::vector<float>>* block) = 0; |
+ |
+ // Gets a reference to the next block (having the specified buffer delay) to |
+ // read in the buffer. This method can only be called if a block is |
+ // available which means that that must be checked prior to the call using |
+ // the method IsBlockAvailable(). |
+ virtual const std::vector<std::vector<float>>& GetNext() = 0; |
+ |
+ // Sets the buffer delay. The delay set must be lower than the delay reported |
+ // by MaxDelay(). |
+ virtual void SetDelay(size_t delay) = 0; |
+ |
+ // Gets the buffer delay. |
+ virtual size_t Delay() const = 0; |
+ |
+ // Returns the maximum allowed buffer delay increase. |
+ virtual size_t MaxDelay() const = 0; |
+ |
+ // Returns whether a block is available for reading. |
+ virtual bool IsBlockAvailable() const = 0; |
+ |
+ // Returns the maximum allowed api call jitter in blocks. |
+ virtual size_t MaxApiJitter() const = 0; |
+}; |
+ |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_ |