tina.legrand@webrtc.org f7fa6276e2 Reformating files in audio coding module.
This CL format the ramaining files on the audio coding module. No other changes are done, except for fixing a few long lines and TODOs without owner.

BUG=issue1024

Review URL: https://webrtc-codereview.appspot.com/928012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3042 4adac7df-926f-26a2-2b94-8c16560cd09d
2012-11-05 09:35:51 +00:00

67 lines
1.9 KiB
C++

/*
* 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 "acm_generic_codec.h"
// forward declaration
struct GSMFR_encinst_t_;
struct GSMFR_decinst_t_;
namespace webrtc {
class ACMGSMFR : public ACMGenericCodec {
public:
ACMGSMFR(WebRtc_Word16 codecID);
~ACMGSMFR();
// for FEC
ACMGenericCodec* CreateInstance(void);
WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
WebRtc_Word16* bitStreamLenByte);
WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
protected:
WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
WebRtc_Word16 bitStreamLenByte,
WebRtc_Word16* audio,
WebRtc_Word16* audioSamples,
WebRtc_Word8* speechType);
WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
const CodecInst& codecInst);
void DestructEncoderSafe();
void DestructDecoderSafe();
WebRtc_Word16 InternalCreateEncoder();
WebRtc_Word16 InternalCreateDecoder();
void InternalDestructEncoderInst(void* ptrInst);
WebRtc_Word16 EnableDTX();
WebRtc_Word16 DisableDTX();
GSMFR_encinst_t_* _encoderInstPtr;
GSMFR_decinst_t_* _decoderInstPtr;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_