
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
87 lines
2.1 KiB
C++
87 lines
2.1 KiB
C++
/*
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* Copyright (c) 2012 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 "webrtc/modules/audio_coding/main/source/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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namespace acm1 {
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class ACMSPEEX : public ACMGenericCodec {
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public:
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explicit ACMSPEEX(int16_t codec_id);
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~ACMSPEEX();
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// for FEC
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ACMGenericCodec* CreateInstance(void);
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int16_t InternalEncode(uint8_t* bitstream,
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int16_t* bitstream_len_byte);
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int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params);
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int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
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protected:
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int16_t DecodeSafe(uint8_t* bitstream,
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int16_t bitstream_len_byte,
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int16_t* audio,
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int16_t* audio_samples,
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int8_t* speech_type);
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int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
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const CodecInst& codec_inst);
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void DestructEncoderSafe();
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void DestructDecoderSafe();
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int16_t InternalCreateEncoder();
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int16_t InternalCreateDecoder();
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void InternalDestructEncoderInst(void* ptr_inst);
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int16_t SetBitRateSafe(const int32_t rate);
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int16_t EnableDTX();
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int16_t DisableDTX();
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#ifdef UNUSEDSPEEX
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int16_t EnableVBR();
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int16_t DisableVBR();
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int16_t SetComplMode(int16_t mode);
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#endif
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SPEEX_encinst_t_* encoder_inst_ptr_;
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SPEEX_decinst_t_* decoder_inst_ptr_;
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int16_t compl_mode_;
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bool vbr_enabled_;
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int32_t encoding_rate_;
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int16_t sampling_frequency_;
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uint16_t samples_in_20ms_audio_;
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};
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} // namespace acm1
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_
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