a165d9c0a4
BUG= R=tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1979004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4500 4adac7df-926f-26a2-2b94-8c16560cd09d
75 lines
2.3 KiB
C++
75 lines
2.3 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_SOURCE_ACM_OPUS_H_
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#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_
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#include "webrtc/common_audio/resampler/include/resampler.h"
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#include "webrtc/modules/audio_coding/main/source/acm_generic_codec.h"
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struct WebRtcOpusEncInst;
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struct WebRtcOpusDecInst;
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namespace webrtc {
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class ACMOpus : public ACMGenericCodec {
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public:
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explicit ACMOpus(int16_t codec_id);
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virtual ~ACMOpus();
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virtual ACMGenericCodec* CreateInstance(void) OVERRIDE;
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virtual int16_t InternalEncode(uint8_t* bitstream,
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int16_t* bitstream_len_byte) OVERRIDE;
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virtual int16_t InternalInitEncoder(
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WebRtcACMCodecParams* codec_params) OVERRIDE;
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virtual int16_t InternalInitDecoder(
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WebRtcACMCodecParams* codec_params) OVERRIDE;
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protected:
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virtual 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) OVERRIDE;
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virtual int32_t CodecDef(WebRtcNetEQ_CodecDef& codec_def,
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const CodecInst& codec_inst) OVERRIDE;
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virtual void DestructEncoderSafe() OVERRIDE;
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virtual void DestructDecoderSafe() OVERRIDE;
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virtual int16_t InternalCreateEncoder() OVERRIDE;
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virtual int16_t InternalCreateDecoder() OVERRIDE;
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virtual void InternalDestructEncoderInst(void* ptr_inst) OVERRIDE;
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virtual int16_t SetBitRateSafe(const int32_t rate) OVERRIDE;
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virtual bool IsTrueStereoCodec() OVERRIDE;
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virtual void SplitStereoPacket(uint8_t* payload,
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int32_t* payload_length) OVERRIDE;
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WebRtcOpusEncInst* encoder_inst_ptr_;
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WebRtcOpusDecInst* decoder_inst_ptr_;
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uint16_t sample_freq_;
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uint32_t bitrate_;
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int channels_;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_
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