Index: webrtc/modules/audio_processing/residual_echo_detector.h |
diff --git a/webrtc/modules/audio_processing/residual_echo_detector.h b/webrtc/modules/audio_processing/residual_echo_detector.h |
new file mode 100644 |
index 0000000000000000000000000000000000000000..ec9515bf0a5eb873ec07089605705d7991f1e480 |
--- /dev/null |
+++ b/webrtc/modules/audio_processing/residual_echo_detector.h |
@@ -0,0 +1,64 @@ |
+/* |
+ * Copyright (c) 2016 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_RESIDUAL_ECHO_DETECTOR_H_ |
+#define WEBRTC_MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_ |
+ |
+#include <memory> |
+#include <vector> |
+ |
+#include "webrtc/base/constructormagic.h" |
+#include "webrtc/base/swap_queue.h" |
+#include "webrtc/modules/audio_processing/include/audio_processing.h" |
+#include "webrtc/modules/audio_processing/render_queue_item_verifier.h" |
+ |
+namespace webrtc { |
+ |
+class AudioBuffer; |
+class EchoDetector; |
+ |
+class ResidualEchoDetector { |
+ public: |
+ ResidualEchoDetector(); |
+ ~ResidualEchoDetector(); |
+ |
+ // This function should be called while holding the render lock. |
+ // Returns 0 on success and -1 when the render buffer is full. In that case, |
+ // ReadQueuedRenderData() should be called before calling this function again. |
+ int AnalyzeRenderAudio(const AudioBuffer* audio) const; |
hlundin-webrtc
2016/10/14 12:36:21
Make this return true or false instead.
ivoc
2016/10/14 14:45:35
Good idea, done.
|
+ |
+ // This function should be called while holding the capture lock. |
+ void AnalyzeCaptureAudio(const AudioBuffer* audio); |
+ |
+ // This function should be called while holding the capture lock. |
+ void Initialize(int sample_rate_hz); |
+ |
+ // Reads render side data that has been queued on the render call. |
+ // This function should be called while holding the capture lock. |
+ void ReadQueuedRenderData(); |
+ |
+ // This function should be called while holding the capture lock. |
+ float get_echo_likelihood() const; |
+ |
+ private: |
+ mutable std::vector<float> render_queue_buffer_; |
+ std::vector<float> capture_queue_buffer_; |
+ |
+ // Lock protection not needed. |
+ mutable std::unique_ptr< |
+ SwapQueue<std::vector<float>, RenderQueueItemVerifier<float>>> |
+ render_signal_queue_; |
+ |
+ std::unique_ptr<EchoDetector> detector_; |
+}; |
+ |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_RESIDUAL_ECHO_DETECTOR_H_ |