Move src/ -> webrtc/

TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-10-22 18:19:23 +00:00
parent 24a419c0c7
commit 14b43beb7c
1888 changed files with 23 additions and 23 deletions
--- a/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
+++ b/webrtc/modules/audio_coding/codecs/opus/interface/opus_interface.h
@ -0,0 +1,123 @@
+/*
+ *  Copyright (c) 2012 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_CODING_CODECS_OPUS_INTERFACE_OPUS_INTERFACE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_OPUS_INTERFACE_H_
+
+#include "typedefs.h"
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+// Opaque wrapper types for the codec state.
+typedef struct WebRtcOpusEncInst OpusEncInst;
+typedef struct WebRtcOpusDecInst OpusDecInst;
+
+int16_t WebRtcOpus_EncoderCreate(OpusEncInst** inst, int32_t channels);
+int16_t WebRtcOpus_EncoderFree(OpusEncInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_Encode(...)
+ *
+ * This function encodes audio as a series of Opus frames and inserts
+ * it into a packet. Input buffer can be any length.
+ *
+ * Input:
+ *      - inst                  : Encoder context
+ *      - audio_in              : Input speech data buffer
+ *      - samples               : Samples in audio_in
+ *      - length_encoded_buffer : Output buffer size
+ *
+ * Output:
+ *      - encoded               : Output compressed data buffer
+ *
+ * Return value                 : >0 - Length (in bytes) of coded data
+ *                                -1 - Error
+ */
+int16_t WebRtcOpus_Encode(OpusEncInst* inst, int16_t* audio_in, int16_t samples,
+                          int16_t length_encoded_buffer, uint8_t* encoded);
+
+/****************************************************************************
+ * WebRtcOpus_SetBitRate(...)
+ *
+ * This function adjusts the target bitrate of the encoder.
+ *
+ * Input:
+ *      - inst               : Encoder context
+ *      - rate               : New target bitrate
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate);
+
+int16_t WebRtcOpus_DecoderCreate(OpusDecInst** inst, int channels);
+int16_t WebRtcOpus_DecoderFree(OpusDecInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_DecoderInit(...)
+ *
+ * This function resets state of the decoder.
+ *
+ * Input:
+ *      - inst               : Decoder context
+ *
+ * Return value              :  0 - Success
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_DecoderInit(OpusDecInst* inst);
+
+/****************************************************************************
+ * WebRtcOpus_Decode(...)
+ *
+ * This function decodes an Opus packet into one or more audio frames at the
+ * ACM interface's sampling rate (32 kHz).
+ *
+ * Input:
+ *      - inst               : Decoder context
+ *      - encoded            : Encoded data
+ *      - encoded_bytes      : Bytes in encoded vector
+ *
+ * Output:
+ *      - decoded            : The decoded vector
+ *      - audio_type         : 1 normal, 2 CNG (for Opus it should
+ *                             always return 1 since we're not using Opus's
+ *                             built-in DTX/CNG scheme)
+ *
+ * Return value              : >0 - Samples in decoded vector
+ *                             -1 - Error
+ */
+int16_t WebRtcOpus_Decode(OpusDecInst* inst, int16_t* encoded,
+                          int16_t encoded_bytes, int16_t* decoded,
+                          int16_t* audio_type);
+
+/****************************************************************************
+ * WebRtcOpus_DecodePlc(...)
+ *
+ * This function precesses PLC for opus frame(s).
+ * Input:
+ *        - inst                  : Decoder context
+ *        - number_of_lost_frames : Number of PLC frames to produce
+ *
+ * Output:
+ *        - decoded               : The decoded vector
+ *
+ * Return value                   : >0 - number of samples in decoded PLC vector
+ *                                  -1 - Error
+ */
+int16_t WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
+                             int16_t number_of_lost_frames);
+
+#ifdef __cplusplus
+}  // extern "C"
+#endif
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_OPUS_INTERFACE_H_