| /* |
| * 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_OPUS_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_ |
| |
| #include "acm_generic_codec.h" |
| #include "opus_interface.h" |
| #include "resampler.h" |
| |
| namespace webrtc { |
| |
| class ACMOpus : public ACMGenericCodec { |
| public: |
| ACMOpus(int16_t codecID); |
| ~ACMOpus(); |
| |
| ACMGenericCodec* CreateInstance(void); |
| |
| int16_t InternalEncode(uint8_t* bitstream, int16_t* bitStreamLenByte); |
| |
| int16_t InternalInitEncoder(WebRtcACMCodecParams *codecParams); |
| |
| int16_t InternalInitDecoder(WebRtcACMCodecParams *codecParams); |
| |
| protected: |
| int16_t DecodeSafe(uint8_t* bitStream, int16_t bitStreamLenByte, |
| int16_t* audio, int16_t* audioSamples, int8_t* speechType); |
| |
| int32_t CodecDef(WebRtcNetEQ_CodecDef& codecDef, const CodecInst& codecInst); |
| |
| void DestructEncoderSafe(); |
| |
| void DestructDecoderSafe(); |
| |
| int16_t InternalCreateEncoder(); |
| |
| int16_t InternalCreateDecoder(); |
| |
| void InternalDestructEncoderInst(void* ptrInst); |
| |
| int16_t SetBitRateSafe(const int32_t rate); |
| |
| OpusEncInst* _encoderInstPtr; |
| OpusDecInst* _decoderInstPtr; |
| uint16_t _sampleFreq; |
| uint16_t _bitrate; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_ |