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/*
* 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.
*/
#ifndef MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
#define MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
#include <stdint.h>
/*
* Solution to support multiple instances
*/
typedef struct WebRtcG722EncInst G722EncInst;
typedef struct WebRtcG722DecInst G722DecInst;
/*
* Comfort noise constants
*/
#define G722_WEBRTC_SPEECH 1
#define G722_WEBRTC_CNG 2
#ifdef __cplusplus
extern "C" {
#endif
/****************************************************************************
* WebRtcG722_CreateEncoder(...)
*
* Create memory used for G722 encoder
*
* Input:
* - G722enc_inst : G722 instance for encoder
*
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcG722_CreateEncoder(G722EncInst** G722enc_inst);
/****************************************************************************
* WebRtcG722_EncoderInit(...)
*
* This function initializes a G722 instance
*
* Input:
* - G722enc_inst : G722 instance, i.e. the user that should receive
* be initialized
*
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcG722_EncoderInit(G722EncInst* G722enc_inst);
/****************************************************************************
* WebRtcG722_FreeEncoder(...)
*
* Free the memory used for G722 encoder
*
* Input:
* - G722enc_inst : G722 instance for encoder
*
* Return value : 0 - Ok
* -1 - Error
*/
int WebRtcG722_FreeEncoder(G722EncInst* G722enc_inst);
/****************************************************************************
* WebRtcG722_Encode(...)
*
* This function encodes G722 encoded data.
*
* Input:
* - G722enc_inst : G722 instance, i.e. the user that should encode
* a packet
* - speechIn : Input speech vector
* - len : Samples in speechIn
*
* Output:
* - encoded : The encoded data vector
*
* Return value : Length (in bytes) of coded data
*/
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
size_t WebRtcG722_Encode(G722EncInst* G722enc_inst,
const int16_t* speechIn,
size_t len,
uint8_t* encoded);
/****************************************************************************
* WebRtcG722_CreateDecoder(...)
*
* Create memory used for G722 encoder
*
* Input:
* - G722dec_inst : G722 instance for decoder
*
* Return value : 0 - Ok
* -1 - Error
*/
int16_t WebRtcG722_CreateDecoder(G722DecInst** G722dec_inst);
/****************************************************************************
* WebRtcG722_DecoderInit(...)
*
* This function initializes a G722 instance
*
* Input:
* - inst : G722 instance
*/
void WebRtcG722_DecoderInit(G722DecInst* inst);
/****************************************************************************
* WebRtcG722_FreeDecoder(...)
*
* Free the memory used for G722 decoder
*
* Input:
* - G722dec_inst : G722 instance for decoder
*
* Return value : 0 - Ok
* -1 - Error
*/
int WebRtcG722_FreeDecoder(G722DecInst* G722dec_inst);
/****************************************************************************
* WebRtcG722_Decode(...)
*
* This function decodes a packet with G729 frame(s). Output speech length
* will be a multiple of 80 samples (80*frames/packet).
*
* Input:
* - G722dec_inst : G722 instance, i.e. the user that should decode
* a packet
* - encoded : Encoded G722 frame(s)
* - len : Bytes in encoded vector
*
* Output:
* - decoded : The decoded vector
* - speechType : 1 normal, 2 CNG (Since G722 does not have its own
* DTX/CNG scheme it should always return 1)
*
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
* Return value : Samples in decoded vector
*/
Update a ton of audio code to use size_t more correctly and in general reduce use of int16_t/uint16_t. This is the upshot of a recommendation by henrik.lundin and kwiberg on an original small change ( https://webrtc-codereview.appspot.com/42569004/#ps1 ) to stop using int16_t just because values could fit in it, and is similar in nature to a previous "mass change to use size_t more" ( https://webrtc-codereview.appspot.com/23129004/ ) which also needed to be split up for review but to land all at once, since, like adding "const", such changes tend to cause a lot of transitive effects. This was be reviewed and approved in pieces: https://codereview.webrtc.org/1224093003 https://codereview.webrtc.org/1224123002 https://codereview.webrtc.org/1224163002 https://codereview.webrtc.org/1225133003 https://codereview.webrtc.org/1225173002 https://codereview.webrtc.org/1227163003 https://codereview.webrtc.org/1227203003 https://codereview.webrtc.org/1227213002 https://codereview.webrtc.org/1227893002 https://codereview.webrtc.org/1228793004 https://codereview.webrtc.org/1228803003 https://codereview.webrtc.org/1228823002 https://codereview.webrtc.org/1228823003 https://codereview.webrtc.org/1228843002 https://codereview.webrtc.org/1230693002 https://codereview.webrtc.org/1231713002 The change is being landed as TBR to all the folks who reviewed the above. BUG=chromium:81439 TEST=none R=andrew@webrtc.org, pbos@webrtc.org TBR=aluebs, andrew, asapersson, henrika, hlundin, jan.skoglund, kwiberg, minyue, pbos, pthatcher Review URL: https://codereview.webrtc.org/1230503003 . Cr-Commit-Position: refs/heads/master@{#9768}
2015-08-24 14:52:23 -07:00
size_t WebRtcG722_Decode(G722DecInst* G722dec_inst,
const uint8_t* encoded,
size_t len,
int16_t* decoded,
int16_t* speechType);
/****************************************************************************
* WebRtcG722_Version(...)
*
* Get a string with the current version of the codec
*/
int16_t WebRtcG722_Version(char* versionStr, short len);
#ifdef __cplusplus
}
#endif
#endif /* MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_ */