turaj@webrtc.org 48af652ea5 Prepare to compile ACM1 and ACM2.
ACM1 code is wrapped in namespace acm1. Inculde paths and define guards of ACM2 source codes are corrected. gypi file of ACM2 is changed so that ACM1 will later on depends on ACM2.

BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2206004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4743 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-09-13 23:06:59 +00:00

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1.8 KiB
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/*
* 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_MAIN_SOURCE_ACM_GSMFR_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_
#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
// forward declaration
struct GSMFR_encinst_t_;
struct GSMFR_decinst_t_;
namespace webrtc {
namespace acm1 {
class ACMGSMFR : public ACMGenericCodec {
public:
explicit ACMGSMFR(int16_t codec_id);
~ACMGSMFR();
// for FEC
ACMGenericCodec* CreateInstance(void);
int16_t InternalEncode(uint8_t* bitstream,
int16_t* bitstream_len_byte);
int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
protected:
int16_t DecodeSafe(uint8_t* bitstream,
int16_t bitstream_len_byte,
int16_t* audio,
int16_t* audio_samples,
int8_t* speech_type);
int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
const CodecInst& codec_inst);
void DestructEncoderSafe();
void DestructDecoderSafe();
int16_t InternalCreateEncoder();
int16_t InternalCreateDecoder();
void InternalDestructEncoderInst(void* ptr_inst);
int16_t EnableDTX();
int16_t DisableDTX();
GSMFR_encinst_t_* encoder_inst_ptr_;
GSMFR_decinst_t_* decoder_inst_ptr_;
};
} // namespace acm1
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_