fb389e3b02
Patch Set 1: Removing blanks at end of lines. Patch Set 2: Removing tabs. Patch Set 3: Fixing include-guards. Patch Set 4-7: Formatting files in the list. Patch Set 8: Formatting CNG. Patch Set 9: * Fixing comments from code review * Fixing formating in acm_dtmf_playout.cc * Started fixing formating of acm_g7221.cc. More work needed, so don't spend too much time reviewing. * Refactored constructor of ACMGenericCodec. Rest of file still to be fixed. * Fixing break; after return ...; in several files. Patch Set 10: * Chaning from reintepret_cast to static_cast in three files, acm_amr.cc, acm_cng.cc and acm_g722.cc NOTE! Not all files have the right format. That work will continue in separate CLs. Review URL: http://webrtc-codereview.appspot.com/175002 git-svn-id: http://webrtc.googlecode.com/svn/trunk@881 4adac7df-926f-26a2-2b94-8c16560cd09d
95 lines
2.4 KiB
C++
95 lines
2.4 KiB
C++
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_
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#include "acm_generic_codec.h"
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// forward declaration
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struct SPEEX_encinst_t_;
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struct SPEEX_decinst_t_;
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namespace webrtc {
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class ACMSPEEX : public ACMGenericCodec
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{
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public:
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ACMSPEEX(WebRtc_Word16 codecID);
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~ACMSPEEX();
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// for FEC
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ACMGenericCodec* CreateInstance(void);
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WebRtc_Word16 InternalEncode(
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WebRtc_UWord8* bitstream,
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WebRtc_Word16* bitStreamLenByte);
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WebRtc_Word16 InternalInitEncoder(
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WebRtcACMCodecParams *codecParams);
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WebRtc_Word16 InternalInitDecoder(
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WebRtcACMCodecParams *codecParams);
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protected:
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WebRtc_Word16 DecodeSafe(
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WebRtc_UWord8* bitStream,
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WebRtc_Word16 bitStreamLenByte,
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WebRtc_Word16* audio,
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WebRtc_Word16* audioSamples,
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WebRtc_Word8* speechType);
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WebRtc_Word32 CodecDef(
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WebRtcNetEQ_CodecDef& codecDef,
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const CodecInst& codecInst);
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void DestructEncoderSafe();
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void DestructDecoderSafe();
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WebRtc_Word16 InternalCreateEncoder();
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WebRtc_Word16 InternalCreateDecoder();
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void InternalDestructEncoderInst(
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void* ptrInst);
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WebRtc_Word16 SetBitRateSafe(
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const WebRtc_Word32 rate);
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WebRtc_Word16 EnableDTX();
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WebRtc_Word16 DisableDTX();
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#ifdef UNUSEDSPEEX
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WebRtc_Word16 EnableVBR();
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WebRtc_Word16 DisableVBR();
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WebRtc_Word16 SetComplMode(
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WebRtc_Word16 mode);
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#endif
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WebRtc_Word16 UnregisterFromNetEqSafe(
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ACMNetEQ* netEq,
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WebRtc_Word16 payloadType);
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SPEEX_encinst_t_* _encoderInstPtr;
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SPEEX_decinst_t_* _decoderInstPtr;
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WebRtc_Word16 _complMode;
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bool _vbrEnabled;
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WebRtc_Word32 _encodingRate;
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WebRtc_Word16 _samplingFrequency;
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WebRtc_UWord16 _samplesIn20MsAudio;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_
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