2012-07-11 19:15:24 +02:00
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/* -----------------------------------------------------------------------------------------------------------
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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2016-04-08 19:52:42 +02:00
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<EFBFBD> Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur F<EFBFBD>rderung der angewandten Forschung e.V.
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2012-07-11 19:15:24 +02:00
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All rights reserved.
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1. INTRODUCTION
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The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
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the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
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This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
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AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
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audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
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independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
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of the MPEG specifications.
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Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
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may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
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individually for the purpose of encoding or decoding bit streams in products that are compliant with
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the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
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these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
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software may already be covered under those patent licenses when it is used for those licensed purposes only.
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Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
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are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
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applications information and documentation.
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2. COPYRIGHT LICENSE
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Redistribution and use in source and binary forms, with or without modification, are permitted without
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payment of copyright license fees provided that you satisfy the following conditions:
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You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
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your modifications thereto in source code form.
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You must retain the complete text of this software license in the documentation and/or other materials
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provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
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You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
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modifications thereto to recipients of copies in binary form.
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The name of Fraunhofer may not be used to endorse or promote products derived from this library without
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prior written permission.
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You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
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software or your modifications thereto.
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Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
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and the date of any change. For modified versions of the FDK AAC Codec, the term
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"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
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"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
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3. NO PATENT LICENSE
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NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
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ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
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respect to this software.
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You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
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by appropriate patent licenses.
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4. DISCLAIMER
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This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
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"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
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of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
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CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
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including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
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or business interruption, however caused and on any theory of liability, whether in contract, strict
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liability, or tort (including negligence), arising in any way out of the use of this software, even if
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advised of the possibility of such damage.
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5. CONTACT INFORMATION
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Fraunhofer Institute for Integrated Circuits IIS
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Attention: Audio and Multimedia Departments - FDK AAC LL
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Am Wolfsmantel 33
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91058 Erlangen, Germany
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www.iis.fraunhofer.de/amm
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amm-info@iis.fraunhofer.de
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----------------------------------------------------------------------------------------------------------- */
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/*************************** Fraunhofer IIS ***********************
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Author(s):
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Description: SBR encoder top level processing prototype
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******************************************************************************/
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#ifndef __SBR_ENCODER_H
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#define __SBR_ENCODER_H
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#include "common_fix.h"
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#include "FDK_audio.h"
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#include "FDK_bitstream.h"
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/* core coder helpers */
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#define MAX_TRANS_FAC 8
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#define MAX_CODEC_FRAME_RATIO 2
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#define MAX_PAYLOAD_SIZE 256
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2013-08-29 21:24:04 +02:00
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typedef enum codecType
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{
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CODEC_AAC=0,
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CODEC_AACLD=1,
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CODEC_UNSPECIFIED=99
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} CODEC_TYPE;
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2012-07-11 19:15:24 +02:00
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typedef struct
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{
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INT bitRate;
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INT nChannels;
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INT sampleFreq;
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INT transFac;
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INT standardBitrate;
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} CODEC_PARAM;
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typedef enum
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{
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SBR_MONO,
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SBR_LEFT_RIGHT,
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SBR_COUPLING,
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SBR_SWITCH_LRC
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} SBR_STEREO_MODE;
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/* bitstream syntax flags */
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enum
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{
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SBR_SYNTAX_LOW_DELAY = 0x0001,
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SBR_SYNTAX_SCALABLE = 0x0002,
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SBR_SYNTAX_CRC = 0x0004,
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SBR_SYNTAX_DRM_CRC = 0x0008
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};
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2016-04-08 19:52:42 +02:00
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typedef enum
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{
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FREQ_RES_LOW = 0,
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FREQ_RES_HIGH
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} FREQ_RES;
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2012-07-11 19:15:24 +02:00
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typedef struct
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{
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2013-08-29 21:24:04 +02:00
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CODEC_TYPE coreCoder; /*!< LC or ELD */
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2012-07-11 19:15:24 +02:00
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UINT bitrateFrom; /*!< inclusive */
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UINT bitrateTo; /*!< exclusive */
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2013-08-29 21:24:04 +02:00
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UINT sampleRate; /*!< */
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2012-07-11 19:15:24 +02:00
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UCHAR numChannels; /*!< */
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UCHAR startFreq; /*!< bs_start_freq */
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UCHAR startFreqSpeech; /*!< bs_start_freq for speech config flag */
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UCHAR stopFreq; /*!< bs_stop_freq */
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UCHAR stopFreqSpeech; /*!< bs_stop_freq for speech config flag */
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UCHAR numNoiseBands; /*!< */
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UCHAR noiseFloorOffset; /*!< */
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SCHAR noiseMaxLevel; /*!< */
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SBR_STEREO_MODE stereoMode; /*!< */
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UCHAR freqScale; /*!< */
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} sbrTuningTable_t;
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typedef struct sbrConfiguration
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{
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/*
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core coder dependent configurations
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*/
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CODEC_PARAM codecSettings; /*!< Core coder settings. To be set from core coder. */
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INT SendHeaderDataTime; /*!< SBR header send update frequency in ms. */
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INT useWaveCoding; /*!< Flag: usage of wavecoding tool. */
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INT crcSbr; /*!< Flag: usage of SBR-CRC. */
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INT dynBwSupported; /*!< Flag: support for dynamic bandwidth in this combination. */
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INT parametricCoding; /*!< Flag: usage of parametric coding tool. */
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INT downSampleFactor; /*!< Sampling rate relation between the SBR and the core encoder. */
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FREQ_RES freq_res_fixfix[2];/*!< Frequency resolution of envelopes in frame class FIXFIX, for non-split case and split case */
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UCHAR fResTransIsLow; /*!< Frequency resolution of envelopes in transient frames: low (0) or variable (1) */
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2012-07-11 19:15:24 +02:00
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/*
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core coder dependent tuning parameters
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*/
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INT tran_thr; /*!< SBR transient detector threshold (* 100). */
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INT noiseFloorOffset; /*!< Noise floor offset. */
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UINT useSpeechConfig; /*!< Flag: adapt tuning parameters according to speech. */
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/*
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core coder independent configurations
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*/
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INT sbrFrameSize; /*!< SBR frame size in samples. Will be calculated from core coder settings. */
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INT sbr_data_extra; /*!< Flag usage of data extra. */
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INT amp_res; /*!< Amplitude resolution. */
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INT ana_max_level; /*!< Noise insertion maximum level. */
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INT tran_fc; /*!< Transient detector start frequency. */
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INT tran_det_mode; /*!< Transient detector mode. */
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INT spread; /*!< Flag: usage of SBR spread. */
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INT stat; /*!< Flag: usage of static framing. */
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INT e; /*!< Number of envelopes when static framing is chosen. */
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SBR_STEREO_MODE stereoMode; /*!< SBR stereo mode. */
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INT deltaTAcrossFrames; /*!< Flag: allow time-delta coding. */
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FIXP_DBL dF_edge_1stEnv; /*!< Extra fraction delta-F coding is allowed to be more expensive. */
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FIXP_DBL dF_edge_incr; /*!< Increment dF_edge_1stEnv this much if dT-coding was used this frame. */
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INT sbr_invf_mode; /*!< Inverse filtering mode. */
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INT sbr_xpos_mode; /*!< Transposer mode. */
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INT sbr_xpos_ctrl; /*!< Transposer control. */
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INT sbr_xpos_level; /*!< Transposer 3rd order level. */
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INT startFreq; /*!< The start frequency table index. */
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INT stopFreq; /*!< The stop frequency table index. */
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INT useSaPan; /*!< Flag: usage of SAPAN stereo. */
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INT dynBwEnabled; /*!< Flag: usage of dynamic bandwidth. */
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INT bParametricStereo; /*!< Flag: usage of parametric stereo coding tool. */
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/*
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header_extra1 configuration
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*/
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UCHAR freqScale; /*!< Frequency grouping. */
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INT alterScale; /*!< Scale resolution. */
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INT sbr_noise_bands; /*!< Number of noise bands. */
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/*
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header_extra2 configuration
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*/
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INT sbr_limiter_bands; /*!< Number of limiter bands. */
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INT sbr_limiter_gains; /*!< Gain of limiter. */
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INT sbr_interpol_freq; /*!< Flag: use interpolation in freq. direction. */
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INT sbr_smoothing_length; /*!< Flag: choose length 4 or 0 (=on, off). */
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UCHAR init_amp_res_FF;
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FIXP_DBL threshold_AmpRes_FF_m;
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SCHAR threshold_AmpRes_FF_e;
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2012-07-11 19:15:24 +02:00
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} sbrConfiguration, *sbrConfigurationPtr ;
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2013-08-29 21:24:04 +02:00
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typedef struct SBR_CONFIG_DATA
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{
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UINT sbrSyntaxFlags; /**< SBR syntax flags derived from AOT. */
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INT nChannels; /**< Number of channels. */
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INT nSfb[2]; /**< Number of SBR scalefactor bands for LO_RES and HI_RES (?) */
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INT num_Master; /**< Number of elements in v_k_master. */
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INT sampleFreq; /**< SBR sampling frequency. */
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INT frameSize;
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INT xOverFreq; /**< The SBR start frequency. */
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INT dynXOverFreq; /**< Used crossover frequency when dynamic bandwidth is enabled. */
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INT noQmfBands; /**< Number of QMF frequency bands. */
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INT noQmfSlots; /**< Number of QMF slots. */
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2016-04-08 19:52:42 +02:00
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UCHAR *freqBandTable[2]; /**< Frequency table for low and hires, only MAX_FREQ_COEFFS/2 +1 coeffs actually needed for lowres. */
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UCHAR *v_k_master; /**< Master BandTable where freqBandTable is derived from. */
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SBR_STEREO_MODE stereoMode;
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INT noEnvChannels; /**< Number of envelope channels. */
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INT useWaveCoding; /**< Flag indicates whether to use wave coding at all. */
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INT useParametricCoding; /**< Flag indicates whether to use para coding at all. */
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INT xposCtrlSwitch; /**< Flag indicates whether to switch xpos ctrl on the fly. */
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INT switchTransposers; /**< Flag indicates whether to switch xpos on the fly . */
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UCHAR initAmpResFF;
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FIXP_DBL thresholdAmpResFF_m;
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SCHAR thresholdAmpResFF_e;
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} SBR_CONFIG_DATA, *HANDLE_SBR_CONFIG_DATA;
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typedef struct {
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MP4_ELEMENT_ID elType;
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INT bitRate;
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int instanceTag;
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UCHAR fParametricStereo;
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UCHAR nChannelsInEl;
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UCHAR ChannelIndex[2];
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} SBR_ELEMENT_INFO;
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#ifdef __cplusplus
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extern "C" {
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#endif
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typedef struct SBR_ENCODER *HANDLE_SBR_ENCODER;
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/**
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* \brief Get the max required input buffer size including delay balancing space
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* for N audio channels.
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* \param noChannels Number of audio channels.
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* \return Max required input buffer size in bytes.
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*/
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INT sbrEncoder_GetInBufferSize(int noChannels);
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INT sbrEncoder_Open(
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HANDLE_SBR_ENCODER *phSbrEncoder,
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INT nElements,
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INT nChannels,
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INT supportPS
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);
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/**
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* \brief Get closest working bitrate to specified desired
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* bitrate for a single SBR element.
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* \param bitRate The desired target bit rate
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* \param numChannels The amount of audio channels
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* \param coreSampleRate The sample rate of the core coder
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* \param aot The current Audio Object Type
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* \return Closest working bit rate to bitRate value
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2012-07-11 19:15:24 +02:00
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*/
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UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate, AUDIO_OBJECT_TYPE aot);
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2013-08-29 21:24:04 +02:00
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/**
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* \brief Check whether downsampled SBR single rate is possible
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* with given audio object type.
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* \param aot The Audio object type.
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* \return 0 when downsampled SBR is not possible,
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* 1 when downsampled SBR is possible.
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*/
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UINT sbrEncoder_IsSingleRatePossible(AUDIO_OBJECT_TYPE aot);
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2012-07-11 19:15:24 +02:00
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/**
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* \brief Initialize SBR Encoder instance.
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* \param phSbrEncoder Pointer to a SBR Encoder instance.
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* \param elInfo Structure that describes the element/channel arrangement.
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* \param noElements Amount of elements described in elInfo.
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* \param inputBuffer Pointer to the encoder audio buffer
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* \param bandwidth Returns the core audio encoder bandwidth (output)
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* \param bufferOffset Returns the offset for the audio input data in order to do delay balancing.
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* \param numChannels Input: Encoder input channels. output: core encoder channels.
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* \param sampleRate Input: Encoder samplerate. output core encoder samplerate.
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2013-08-29 21:24:04 +02:00
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* \param downSampleFactor Input: Relation between SBR and core coder sampling rate;
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2012-07-11 19:15:24 +02:00
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* \param frameLength Input: Encoder frameLength. output core encoder frameLength.
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* \param aot Input: Desired AOT. output AOT to be used after parameter checking.
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* \param delay Input: core encoder delay. Output: total delay because of SBR.
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* \param transformFactor The core encoder transform factor (blockswitching).
|
2013-08-09 03:25:28 +02:00
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* \param headerPeriod Repetition rate of the SBR header:
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* - (-1) means intern configuration.
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* - (1-10) corresponds to header repetition rate in frames.
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2012-07-11 19:15:24 +02:00
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* \return 0 on success, and non-zero if failed.
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*/
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2013-08-29 21:24:04 +02:00
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INT sbrEncoder_Init(
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HANDLE_SBR_ENCODER hSbrEncoder,
|
2013-08-30 01:05:22 +02:00
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SBR_ELEMENT_INFO elInfo[(8)],
|
2013-08-29 21:24:04 +02:00
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int noElements,
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INT_PCM *inputBuffer,
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INT *coreBandwidth,
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INT *inputBufferOffset,
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INT *numChannels,
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INT *sampleRate,
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UINT *downSampleFactor,
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INT *frameLength,
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AUDIO_OBJECT_TYPE aot,
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int *delay,
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int transformFactor,
|
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|
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|
const int headerPeriod,
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|
|
ULONG statesInitFlag
|
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|
|
|
);
|
2012-07-11 19:15:24 +02:00
|
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|
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|
|
|
/**
|
|
|
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|
* \brief Do delay line buffers housekeeping. To be called after each encoded audio frame.
|
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|
|
|
* \param hEnvEnc SBR Encoder handle.
|
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|
* \param timeBuffer Pointer to the encoder audio buffer.
|
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|
* \return 0 on success, and non-zero if failed.
|
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|
|
|
*/
|
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|
|
|
INT sbrEncoder_UpdateBuffers(HANDLE_SBR_ENCODER hEnvEnc,
|
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|
|
INT_PCM *timeBuffer
|
|
|
|
|
);
|
|
|
|
|
|
|
|
|
|
/**
|
|
|
|
|
* \brief Close SBR encoder instance.
|
|
|
|
|
* \param phEbrEncoder Handle of SBR encoder instance to be closed.
|
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|
|
|
* \return void
|
|
|
|
|
*/
|
|
|
|
|
void sbrEncoder_Close(HANDLE_SBR_ENCODER *phEbrEncoder);
|
|
|
|
|
|
|
|
|
|
/**
|
|
|
|
|
* \brief Encode SBR data of one complete audio frame.
|
|
|
|
|
* \param hEnvEncoder Handle of SBR encoder instance.
|
|
|
|
|
* \param samples Time samples, always interleaved.
|
|
|
|
|
* \param timeInStride Channel stride factor of samples buffer.
|
|
|
|
|
* \param sbrDataBits Size of SBR payload in bits.
|
|
|
|
|
* \param sbrData SBR payload.
|
|
|
|
|
* \return 0 on success, and non-zero if failed.
|
|
|
|
|
*/
|
|
|
|
|
INT sbrEncoder_EncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder,
|
|
|
|
|
INT_PCM *samples,
|
|
|
|
|
UINT timeInStride,
|
2013-08-30 01:05:22 +02:00
|
|
|
|
UINT sbrDataBits[(8)],
|
|
|
|
|
UCHAR sbrData[(8)][MAX_PAYLOAD_SIZE]
|
2012-07-11 19:15:24 +02:00
|
|
|
|
);
|
|
|
|
|
|
|
|
|
|
/**
|
|
|
|
|
* \brief Write SBR headers of one SBR element.
|
|
|
|
|
* \param sbrEncoder Handle of the SBR encoder instance.
|
|
|
|
|
* \param hBs Handle of bit stream handle to write SBR header to.
|
|
|
|
|
* \param element_index Index of the SBR element which header should be written.
|
|
|
|
|
* \param fSendHeaders Flag indicating that the SBR encoder should send more headers in the SBR payload or not.
|
|
|
|
|
* \return void
|
|
|
|
|
*/
|
2013-08-29 21:24:04 +02:00
|
|
|
|
void sbrEncoder_GetHeader(HANDLE_SBR_ENCODER sbrEncoder,
|
2012-07-11 19:15:24 +02:00
|
|
|
|
HANDLE_FDK_BITSTREAM hBs,
|
|
|
|
|
INT element_index,
|
|
|
|
|
int fSendHeaders);
|
|
|
|
|
|
|
|
|
|
/**
|
|
|
|
|
* \brief SBR encoder bitrate estimation.
|
|
|
|
|
* \param hSbrEncoder SBR encoder handle.
|
|
|
|
|
* \return Estimated bitrate.
|
|
|
|
|
*/
|
|
|
|
|
INT sbrEncoder_GetEstimateBitrate(HANDLE_SBR_ENCODER hSbrEncoder);
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
/**
|
|
|
|
|
* \brief Delay between input data and downsampled output data.
|
|
|
|
|
* \param hSbrEncoder SBR encoder handle.
|
|
|
|
|
* \return Delay.
|
|
|
|
|
*/
|
|
|
|
|
INT sbrEncoder_GetInputDataDelay(HANDLE_SBR_ENCODER hSbrEncoder);
|
|
|
|
|
|
|
|
|
|
/**
|
|
|
|
|
* \brief Get decoder library version info.
|
|
|
|
|
* \param info Pointer to an allocated LIB_INFO struct, where library info is written to.
|
|
|
|
|
* \return 0 on sucess.
|
|
|
|
|
*/
|
|
|
|
|
INT sbrEncoder_GetLibInfo(LIB_INFO *info);
|
|
|
|
|
|
|
|
|
|
void sbrPrintRAM(void);
|
|
|
|
|
|
|
|
|
|
void sbrPrintROM(void);
|
|
|
|
|
|
|
|
|
|
#ifdef __cplusplus
|
|
|
|
|
}
|
|
|
|
|
#endif
|
|
|
|
|
|
|
|
|
|
#endif /* ifndef __SBR_MAIN_H */
|