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https://github.com/oxen-io/session-android.git
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d83a3d71bc
Merge in RedPhone // FREEBIE
55 lines
1.3 KiB
C++
55 lines
1.3 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_ACM2_ACM_GSMFR_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GSMFR_H_
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#include "webrtc/modules/audio_coding/main/acm2/acm_generic_codec.h"
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// forward declaration
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struct GSMFR_encinst_t_;
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struct GSMFR_decinst_t_;
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namespace webrtc {
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namespace acm2 {
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class ACMGSMFR : public ACMGenericCodec {
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public:
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explicit ACMGSMFR(int16_t codec_id);
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~ACMGSMFR();
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// for FEC
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ACMGenericCodec* CreateInstance(void);
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int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte);
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int16_t InternalInitEncoder(WebRtcACMCodecParams* codec_params);
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protected:
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void DestructEncoderSafe();
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int16_t InternalCreateEncoder();
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void InternalDestructEncoderInst(void* ptr_inst);
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int16_t EnableDTX();
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int16_t DisableDTX();
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GSMFR_encinst_t_* encoder_inst_ptr_;
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};
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} // namespace acm2
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_GSMFR_H_
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