c454fab03b
TEST=VoE auto-test, audio_coding_module_test; only 15 ms of teststereo_out_1.pcm is not bit-exact with output file of the head revision Review URL: https://webrtc-codereview.appspot.com/937035 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3287 4adac7df-926f-26a2-2b94-8c16560cd09d
75 lines
2.1 KiB
C++
75 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_CELT_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_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 CELT_encinst_t_;
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struct CELT_decinst_t_;
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namespace webrtc {
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class ACMCELT : public ACMGenericCodec {
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public:
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explicit ACMCELT(int16_t codec_id);
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~ACMCELT();
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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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int16_t InternalInitDecoder(WebRtcACMCodecParams *codec_params);
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protected:
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WebRtc_Word16 DecodeSafe(
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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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// TODO(leozwang): use int8_t here when WebRtc_Word8 is properly typed.
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// http://code.google.com/p/webrtc/issues/detail?id=311
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WebRtc_Word8* /* 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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bool IsTrueStereoCodec();
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int16_t SetBitRateSafe(const int32_t rate);
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void SplitStereoPacket(uint8_t* payload, int32_t* payload_length);
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CELT_encinst_t_* enc_inst_ptr_;
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CELT_decinst_t_* dec_inst_ptr_;
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uint16_t sampling_freq_;
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int32_t bitrate_;
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uint16_t channels_;
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uint16_t dec_channels_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_
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