git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d

modules/audio_processing/utility/ring_buffer.h

@@ -0,0 +1,41 @@
+/*
+ *  Copyright (c) 2011 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.
+ */
+
+/*
+ * Specifies the interface for the AEC generic buffer.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
+
+// Determines buffer datatype
+typedef short bufdata_t;
+
+// Unless otherwise specified, functions return 0 on success and -1 on error
+int WebRtcApm_CreateBuffer(void **bufInst, int size);
+int WebRtcApm_InitBuffer(void *bufInst);
+int WebRtcApm_FreeBuffer(void *bufInst);
+
+// Returns number of samples read
+int WebRtcApm_ReadBuffer(void *bufInst, bufdata_t *data, int size);
+
+// Returns number of samples written
+int WebRtcApm_WriteBuffer(void *bufInst, const bufdata_t *data, int size);
+
+// Returns number of samples flushed
+int WebRtcApm_FlushBuffer(void *bufInst, int size);
+
+// Returns number of samples stuffed
+int WebRtcApm_StuffBuffer(void *bufInst, int size);
+
+// Returns number of samples in buffer
+int WebRtcApm_get_buffer_size(const void *bufInst);
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_