| /* |
| * 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_CELT_H_ |
| #define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_ |
| |
| #include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h" |
| |
| // forward declaration |
| struct CELT_encinst_t_; |
| struct CELT_decinst_t_; |
| |
| namespace webrtc { |
| |
| class ACMCELT : public ACMGenericCodec { |
| public: |
| explicit ACMCELT(int16_t codec_id); |
| ~ACMCELT(); |
| |
| ACMGenericCodec* CreateInstance(void); |
| |
| int16_t InternalEncode(uint8_t* bitstream, int16_t* bitstream_len_byte); |
| |
| int16_t InternalInitEncoder(WebRtcACMCodecParams *codec_params); |
| |
| int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params); |
| |
| protected: |
| WebRtc_Word16 DecodeSafe( |
| uint8_t* /* bitstream */, |
| int16_t /* bitstream_len_byte */, |
| int16_t* /* audio */, |
| int16_t* /* audio_samples */, |
| // TODO(leozwang): use int8_t here when WebRtc_Word8 is properly typed. |
| // http://code.google.com/p/webrtc/issues/detail?id=311 |
| WebRtc_Word8* /* speech_type */); |
| |
| int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def, |
| const CodecInst& codec_inst); |
| |
| void DestructEncoderSafe(); |
| |
| void DestructDecoderSafe(); |
| |
| int16_t InternalCreateEncoder(); |
| |
| int16_t InternalCreateDecoder(); |
| |
| void InternalDestructEncoderInst(void* ptr_inst); |
| |
| bool IsTrueStereoCodec(); |
| |
| int16_t SetBitRateSafe(const int32_t rate); |
| |
| void SplitStereoPacket(uint8_t* payload, int32_t* payload_length); |
| |
| CELT_encinst_t_* enc_inst_ptr_; |
| CELT_decinst_t_* dec_inst_ptr_; |
| uint16_t sampling_freq_; |
| int32_t bitrate_; |
| uint16_t channels_; |
| uint16_t dec_channels_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_ |