diff options
author | Fraunhofer IIS FDK <audio-fdk@iis.fraunhofer.de> | 2018-02-26 20:17:00 +0100 |
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committer | Jean-Michel Trivi <jmtrivi@google.com> | 2018-04-19 11:21:15 -0700 |
commit | 6cfabd35363c3ef5e3b209b867169a500b3ccc3c (patch) | |
tree | 01c0a19f2735e8b5d2407555fe992d4230d089eb /libSBRenc/include/sbr_encoder.h | |
parent | 6288a1e34c4dede4c2806beb1736ece6580558c7 (diff) | |
download | fdk-aac-6cfabd35363c3ef5e3b209b867169a500b3ccc3c.tar.gz fdk-aac-6cfabd35363c3ef5e3b209b867169a500b3ccc3c.tar.bz2 fdk-aac-6cfabd35363c3ef5e3b209b867169a500b3ccc3c.zip |
Upgrade to FDKv2
Bug: 71430241
Test: CTS DecoderTest and DecoderTestAacDrc
original-Change-Id: Iaa20f749b8a04d553b20247cfe1a8930ebbabe30
Apply clang-format also on header files.
original-Change-Id: I14de1ef16bbc79ec0283e745f98356a10efeb2e4
Fixes for MPEG-D DRC
original-Change-Id: If1de2d74bbbac84b3f67de3b88b83f6a23b8a15c
Catch unsupported tw_mdct at an early stage
original-Change-Id: Ied9dd00d754162a0e3ca1ae3e6b854315d818afe
Fixing PVC transition frames
original-Change-Id: Ib75725abe39252806c32d71176308f2c03547a4e
Move qmf bands sanity check
original-Change-Id: Iab540c3013c174d9490d2ae100a4576f51d8dbc4
Initialize scaling variable
original-Change-Id: I3c4087101b70e998c71c1689b122b0d7762e0f9e
Add 16 qmf band configuration to getSlotNrgHQ()
original-Change-Id: I49a5d30f703a1b126ff163df9656db2540df21f1
Always apply byte alignment at the end of the AudioMuxElement
original-Change-Id: I42d560287506d65d4c3de8bfe3eb9a4ebeb4efc7
Setup SBR element only if no parse error exists
original-Change-Id: I1915b73704bc80ab882b9173d6bec59cbd073676
Additional array index check in HCR
original-Change-Id: I18cc6e501ea683b5009f1bbee26de8ddd04d8267
Fix fade-in index selection in concealment module
original-Change-Id: Ibf802ed6ed8c05e9257e1f3b6d0ac1162e9b81c1
Enable explicit backward compatible parser for AAC_LD
original-Change-Id: I27e9c678dcb5d40ed760a6d1e06609563d02482d
Skip spatial specific config in explicit backward compatible ASC
original-Change-Id: Iff7cc365561319e886090cedf30533f562ea4d6e
Update flags description in decoder API
original-Change-Id: I9a5b4f8da76bb652f5580cbd3ba9760425c43830
Add QMF domain reset function
original-Change-Id: I4f89a8a2c0277d18103380134e4ed86996e9d8d6
DRC upgrade v2.1.0
original-Change-Id: I5731c0540139dab220094cd978ef42099fc45b74
Fix integer overflow in sqrtFixp_lookup()
original-Change-Id: I429a6f0d19aa2cc957e0f181066f0ca73968c914
Fix integer overflow in invSqrtNorm2()
original-Change-Id: I84de5cbf9fb3adeb611db203fe492fabf4eb6155
Fix integer overflow in GenerateRandomVector()
original-Change-Id: I3118a641008bd9484d479e5b0b1ee2b5d7d44d74
Fix integer overflow in adjustTimeSlot_EldGrid()
original-Change-Id: I29d503c247c5c8282349b79df940416a512fb9d5
Fix integer overflow in FDKsbrEnc_codeEnvelope()
original-Change-Id: I6b34b61ebb9d525b0c651ed08de2befc1f801449
Follow-up on: Fix integer overflow in adjustTimeSlot_EldGrid()
original-Change-Id: I6f8f578cc7089e5eb7c7b93e580b72ca35ad689a
Fix integer overflow in get_pk_v2()
original-Change-Id: I63375bed40d45867f6eeaa72b20b1f33e815938c
Fix integer overflow in Syn_filt_zero()
original-Change-Id: Ie0c02fdfbe03988f9d3b20d10cd9fe4c002d1279
Fix integer overflow in CFac_CalcFacSignal()
original-Change-Id: Id2d767c40066c591b51768e978eb8af3b803f0c5
Fix integer overflow in FDKaacEnc_FDKaacEnc_calcPeNoAH()
original-Change-Id: Idcbd0f4a51ae2550ed106aa6f3d678d1f9724841
Fix integer overflow in sbrDecoder_calculateGainVec()
original-Change-Id: I7081bcbe29c5cede9821b38d93de07c7add2d507
Fix integer overflow in CLpc_SynthesisLattice()
original-Change-Id: I4a95ddc18de150102352d4a1845f06094764c881
Fix integer overflow in Pred_Lt4()
original-Change-Id: I4dbd012b2de7d07c3e70a47b92e3bfae8dbc750a
Fix integer overflow in FDKsbrEnc_InitSbrFastTransientDetector()
original-Change-Id: I788cbec1a4a00f44c2f3a72ad7a4afa219807d04
Fix unsigned integer overflow in FDKaacEnc_WriteBitstream()
original-Change-Id: I68fc75166e7d2cd5cd45b18dbe3d8c2a92f1822a
Fix unsigned integer overflow in FDK_MetadataEnc_Init()
original-Change-Id: Ie8d025f9bcdb2442c704bd196e61065c03c10af4
Fix overflow in pseudo random number generators
original-Change-Id: I3e2551ee01356297ca14e3788436ede80bd5513c
Fix unsigned integer overflow in sbrDecoder_Parse()
original-Change-Id: I3f231b2f437e9c37db4d5b964164686710eee971
Fix unsigned integer overflow in longsub()
original-Change-Id: I73c2bc50415cac26f1f5a29e125bbe75f9180a6e
Fix unsigned integer overflow in CAacDecoder_DecodeFrame()
original-Change-Id: Ifce2db4b1454b46fa5f887e9d383f1cc43b291e4
Fix overflow at CLpdChannelStream_Read()
original-Change-Id: Idb9d822ce3a4272e4794b643644f5434e2d4bf3f
Fix unsigned integer overflow in Hcr_State_BODY_SIGN_ESC__ESC_WORD()
original-Change-Id: I1ccf77c0015684b85534c5eb97162740a870b71c
Fix unsigned integer overflow in UsacConfig_Parse()
original-Change-Id: Ie6d27f84b6ae7eef092ecbff4447941c77864d9f
Fix unsigned integer overflow in aacDecoder_drcParse()
original-Change-Id: I713f28e883eea3d70b6fa56a7b8f8c22bcf66ca0
Fix unsigned integer overflow in aacDecoder_drcReadCompression()
original-Change-Id: Ia34dfeb88c4705c558bce34314f584965cafcf7a
Fix unsigned integer overflow in CDataStreamElement_Read()
original-Change-Id: Iae896cc1d11f0a893d21be6aa90bd3e60a2c25f0
Fix unsigned integer overflow in transportDec_AdjustEndOfAccessUnit()
original-Change-Id: I64cf29a153ee784bb4a16fdc088baabebc0007dc
Fix unsigned integer overflow in transportDec_GetAuBitsRemaining()
original-Change-Id: I975b3420faa9c16a041874ba0db82e92035962e4
Fix unsigned integer overflow in extractExtendedData()
original-Change-Id: I2a59eb09e2053cfb58dfb75fcecfad6b85a80a8f
Fix signed integer overflow in CAacDecoder_ExtPayloadParse()
original-Change-Id: I4ad5ca4e3b83b5d964f1c2f8c5e7b17c477c7929
Fix unsigned integer overflow in CAacDecoder_DecodeFrame()
original-Change-Id: I29a39df77d45c52a0c9c5c83c1ba81f8d0f25090
Follow-up on: Fix integer overflow in CLpc_SynthesisLattice()
original-Change-Id: I8fb194ffc073a3432a380845be71036a272d388f
Fix signed integer overflow in _interpolateDrcGain()
original-Change-Id: I879ec9ab14005069a7c47faf80e8bc6e03d22e60
Fix unsigned integer overflow in FDKreadBits()
original-Change-Id: I1f47a6a8037ff70375aa8844947d5681bb4287ad
Fix unsigned integer overflow in FDKbyteAlign()
original-Change-Id: Id5f3a11a0c9e50fc6f76ed6c572dbd4e9f2af766
Fix unsigned integer overflow in FDK_get32()
original-Change-Id: I9d33b8e97e3d38cbb80629cb859266ca0acdce96
Fix unsigned integer overflow in FDK_pushBack()
original-Change-Id: Ic87f899bc8c6acf7a377a8ca7f3ba74c3a1e1c19
Fix unsigned integer overflow in FDK_pushForward()
original-Change-Id: I3b754382f6776a34be1602e66694ede8e0b8effc
Fix unsigned integer overflow in ReadPsData()
original-Change-Id: I25361664ba8139e32bbbef2ca8c106a606ce9c37
Fix signed integer overflow in E_UTIL_residu()
original-Change-Id: I8c3abd1f437ee869caa8fb5903ce7d3d641b6aad
REVERT: Follow-up on: Integer overflow in CLpc_SynthesisLattice().
original-Change-Id: I3d340099acb0414795c8dfbe6362bc0a8f045f9b
Follow-up on: Fix integer overflow in CLpc_SynthesisLattice()
original-Change-Id: I4aedb8b3a187064e9f4d985175aa55bb99cc7590
Follow-up on: Fix unsigned integer overflow in aacDecoder_drcParse()
original-Change-Id: I2aa2e13916213bf52a67e8b0518e7bf7e57fb37d
Fix integer overflow in acelp
original-Change-Id: Ie6390c136d84055f8b728aefbe4ebef6e029dc77
Fix unsigned integer overflow in aacDecoder_UpdateBitStreamCounters()
original-Change-Id: I391ffd97ddb0b2c184cba76139bfb356a3b4d2e2
Adjust concealment default settings
original-Change-Id: I6a95db935a327c47df348030bcceafcb29f54b21
Saturate estimatedStartPos
original-Change-Id: I27be2085e0ae83ec9501409f65e003f6bcba1ab6
Negative shift exponent in _interpolateDrcGain()
original-Change-Id: I18edb26b26d002aafd5e633d4914960f7a359c29
Negative shift exponent in calculateICC()
original-Change-Id: I3dcd2ae98d2eb70ee0d59750863cbb2a6f4f8aba
Too large shift exponent in FDK_put()
original-Change-Id: Ib7d9aaa434d2d8de4a13b720ca0464b31ca9b671
Too large shift exponent in CalcInvLdData()
original-Change-Id: I43e6e78d4cd12daeb1dcd5d82d1798bdc2550262
Member access within null pointer of type SBR_CHANNEL
original-Change-Id: Idc5e4ea8997810376d2f36bbdf628923b135b097
Member access within null pointer of type CpePersistentData
original-Change-Id: Ib6c91cb0d37882768e5baf63324e429589de0d9d
Member access within null pointer FDKaacEnc_psyMain()
original-Change-Id: I7729b7f4479970531d9dc823abff63ca52e01997
Member access within null pointer FDKaacEnc_GetPnsParam()
original-Change-Id: I9aa3b9f3456ae2e0f7483dbd5b3dde95fc62da39
Member access within null pointer FDKsbrEnc_EnvEncodeFrame()
original-Change-Id: I67936f90ea714e90b3e81bc0dd1472cc713eb23a
Add HCR sanity check
original-Change-Id: I6c1d9732ebcf6af12f50b7641400752f74be39f7
Fix memory issue for HBE edge case with 8:3 SBR
original-Change-Id: I11ea58a61e69fbe8bf75034b640baee3011e63e9
Additional SBR parametrization sanity check for ELD
original-Change-Id: Ie26026fbfe174c2c7b3691f6218b5ce63e322140
Add MPEG-D DRC channel layout check
original-Change-Id: Iea70a74f171b227cce636a9eac4ba662777a2f72
Additional out-of-bounds checks in MPEG-D DRC
original-Change-Id: Ife4a8c3452c6fde8a0a09e941154a39a769777d4
Change-Id: Ic63cb2f628720f54fe9b572b0cb528e2599c624e
Diffstat (limited to 'libSBRenc/include/sbr_encoder.h')
-rw-r--r-- | libSBRenc/include/sbr_encoder.h | 489 |
1 files changed, 271 insertions, 218 deletions
diff --git a/libSBRenc/include/sbr_encoder.h b/libSBRenc/include/sbr_encoder.h index aec0398..d979ba6 100644 --- a/libSBRenc/include/sbr_encoder.h +++ b/libSBRenc/include/sbr_encoder.h @@ -1,74 +1,85 @@ - -/* ----------------------------------------------------------------------------------------------------------- +/* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. - All rights reserved. +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. 1. INTRODUCTION -The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements -the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. -This FDK AAC Codec software is intended to be used on a wide variety of Android devices. - -AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual -audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by -independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part -of the MPEG specifications. - -Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) -may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners -individually for the purpose of encoding or decoding bit streams in products that are compliant with -the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license -these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec -software may already be covered under those patent licenses when it is used for those licensed purposes only. - -Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, -are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional -applications information and documentation. +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. 2. COPYRIGHT LICENSE -Redistribution and use in source and binary forms, with or without modification, are permitted without -payment of copyright license fees provided that you satisfy the following conditions: +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: -You must retain the complete text of this software license in redistributions of the FDK AAC Codec or -your modifications thereto in source code form. +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. -You must retain the complete text of this software license in the documentation and/or other materials -provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. -You must make available free of charge copies of the complete source code of the FDK AAC Codec and your +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your modifications thereto to recipients of copies in binary form. -The name of Fraunhofer may not be used to endorse or promote products derived from this library without -prior written permission. +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. -You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec -software or your modifications thereto. +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. -Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software -and the date of any change. For modified versions of the FDK AAC Codec, the term -"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term -"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." 3. NO PATENT LICENSE -NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, -ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with -respect to this software. +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. -You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized -by appropriate patent licenses. +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. 4. DISCLAIMER -This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors -"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties -of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR -CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, -including but not limited to procurement of substitute goods or services; loss of use, data, or profits, -or business interruption, however caused and on any theory of liability, whether in contract, strict -liability, or tort (including negligence), arising in any way out of the use of this software, even if -advised of the possibility of such damage. +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. 5. CONTACT INFORMATION @@ -79,17 +90,18 @@ Am Wolfsmantel 33 www.iis.fraunhofer.de/amm amm-info@iis.fraunhofer.de ------------------------------------------------------------------------------------------------------------ */ +----------------------------------------------------------------------------- */ -/*************************** Fraunhofer IIS *********************** +/**************************** SBR encoder library ****************************** Author(s): + Description: SBR encoder top level processing prototype -******************************************************************************/ +*******************************************************************************/ -#ifndef __SBR_ENCODER_H -#define __SBR_ENCODER_H +#ifndef SBR_ENCODER_H +#define SBR_ENCODER_H #include "common_fix.h" #include "FDK_audio.h" @@ -97,20 +109,17 @@ amm-info@iis.fraunhofer.de #include "FDK_bitstream.h" /* core coder helpers */ -#define MAX_TRANS_FAC 8 +#define MAX_TRANS_FAC 8 #define MAX_CODEC_FRAME_RATIO 2 -#define MAX_PAYLOAD_SIZE 256 +#define MAX_PAYLOAD_SIZE 256 -typedef enum codecType -{ - CODEC_AAC=0, - CODEC_AACLD=1, - CODEC_UNSPECIFIED=99 +typedef enum codecType { + CODEC_AAC = 0, + CODEC_AACLD = 1, + CODEC_UNSPECIFIED = 99 } CODEC_TYPE; - -typedef struct -{ +typedef struct { INT bitRate; INT nChannels; INT sampleFreq; @@ -118,8 +127,7 @@ typedef struct INT standardBitrate; } CODEC_PARAM; -typedef enum -{ +typedef enum { SBR_MONO, SBR_LEFT_RIGHT, SBR_COUPLING, @@ -127,136 +135,143 @@ typedef enum } SBR_STEREO_MODE; /* bitstream syntax flags */ -enum -{ +enum { SBR_SYNTAX_LOW_DELAY = 0x0001, - SBR_SYNTAX_SCALABLE = 0x0002, - SBR_SYNTAX_CRC = 0x0004, - SBR_SYNTAX_DRM_CRC = 0x0008 + SBR_SYNTAX_SCALABLE = 0x0002, + SBR_SYNTAX_CRC = 0x0004, + SBR_SYNTAX_DRM_CRC = 0x0008, + SBR_SYNTAX_ELD_REDUCED_DELAY = 0x0010 }; -typedef enum -{ - FREQ_RES_LOW = 0, - FREQ_RES_HIGH -} FREQ_RES; - -typedef struct -{ - CODEC_TYPE coreCoder; /*!< LC or ELD */ - UINT bitrateFrom; /*!< inclusive */ - UINT bitrateTo; /*!< exclusive */ - - UINT sampleRate; /*!< */ - UCHAR numChannels; /*!< */ - - UCHAR startFreq; /*!< bs_start_freq */ - UCHAR startFreqSpeech; /*!< bs_start_freq for speech config flag */ - UCHAR stopFreq; /*!< bs_stop_freq */ - UCHAR stopFreqSpeech; /*!< bs_stop_freq for speech config flag */ - - UCHAR numNoiseBands; /*!< */ - UCHAR noiseFloorOffset; /*!< */ - SCHAR noiseMaxLevel; /*!< */ - SBR_STEREO_MODE stereoMode; /*!< */ - UCHAR freqScale; /*!< */ +typedef enum { FREQ_RES_LOW = 0, FREQ_RES_HIGH } FREQ_RES; + +typedef struct { + CODEC_TYPE coreCoder; /*!< LC or ELD */ + UINT bitrateFrom; /*!< inclusive */ + UINT bitrateTo; /*!< exclusive */ + + UINT sampleRate; /*!< */ + UCHAR numChannels; /*!< */ + + UCHAR startFreq; /*!< bs_start_freq */ + UCHAR startFreqSpeech; /*!< bs_start_freq for speech config flag */ + UCHAR stopFreq; /*!< bs_stop_freq */ + UCHAR stopFreqSpeech; /*!< bs_stop_freq for speech config flag */ + + UCHAR numNoiseBands; /*!< */ + UCHAR noiseFloorOffset; /*!< */ + SCHAR noiseMaxLevel; /*!< */ + SBR_STEREO_MODE stereoMode; /*!< */ + UCHAR freqScale; /*!< */ } sbrTuningTable_t; -typedef struct sbrConfiguration -{ +typedef struct sbrConfiguration { /* core coder dependent configurations */ - CODEC_PARAM codecSettings; /*!< Core coder settings. To be set from core coder. */ - INT SendHeaderDataTime; /*!< SBR header send update frequency in ms. */ - INT useWaveCoding; /*!< Flag: usage of wavecoding tool. */ - INT crcSbr; /*!< Flag: usage of SBR-CRC. */ - INT dynBwSupported; /*!< Flag: support for dynamic bandwidth in this combination. */ - INT parametricCoding; /*!< Flag: usage of parametric coding tool. */ - INT downSampleFactor; /*!< Sampling rate relation between the SBR and the core encoder. */ - FREQ_RES freq_res_fixfix[2];/*!< Frequency resolution of envelopes in frame class FIXFIX, for non-split case and split case */ - UCHAR fResTransIsLow; /*!< Frequency resolution of envelopes in transient frames: low (0) or variable (1) */ + CODEC_PARAM + codecSettings; /*!< Core coder settings. To be set from core coder. */ + INT SendHeaderDataTime; /*!< SBR header send update frequency in ms. */ + INT useWaveCoding; /*!< Flag: usage of wavecoding tool. */ + INT crcSbr; /*!< Flag: usage of SBR-CRC. */ + INT dynBwSupported; /*!< Flag: support for dynamic bandwidth in this + combination. */ + INT parametricCoding; /*!< Flag: usage of parametric coding tool. */ + INT downSampleFactor; /*!< Sampling rate relation between the SBR and the core + encoder. */ + FREQ_RES freq_res_fixfix[2]; /*!< Frequency resolution of envelopes in frame + class FIXFIX, for non-split case and split + case */ + UCHAR fResTransIsLow; /*!< Frequency resolution of envelopes in transient + frames: low (0) or variable (1) */ /* core coder dependent tuning parameters */ - INT tran_thr; /*!< SBR transient detector threshold (* 100). */ - INT noiseFloorOffset; /*!< Noise floor offset. */ - UINT useSpeechConfig; /*!< Flag: adapt tuning parameters according to speech. */ - - + INT tran_thr; /*!< SBR transient detector threshold (* 100). */ + INT noiseFloorOffset; /*!< Noise floor offset. */ + UINT useSpeechConfig; /*!< Flag: adapt tuning parameters according to speech. + */ /* core coder independent configurations */ - INT sbrFrameSize; /*!< SBR frame size in samples. Will be calculated from core coder settings. */ - INT sbr_data_extra; /*!< Flag usage of data extra. */ - INT amp_res; /*!< Amplitude resolution. */ - INT ana_max_level; /*!< Noise insertion maximum level. */ - INT tran_fc; /*!< Transient detector start frequency. */ - INT tran_det_mode; /*!< Transient detector mode. */ - INT spread; /*!< Flag: usage of SBR spread. */ - INT stat; /*!< Flag: usage of static framing. */ - INT e; /*!< Number of envelopes when static framing is chosen. */ + INT sbrFrameSize; /*!< SBR frame size in samples. Will be calculated from core + coder settings. */ + INT sbr_data_extra; /*!< Flag usage of data extra. */ + INT amp_res; /*!< Amplitude resolution. */ + INT ana_max_level; /*!< Noise insertion maximum level. */ + INT tran_fc; /*!< Transient detector start frequency. */ + INT tran_det_mode; /*!< Transient detector mode. */ + INT spread; /*!< Flag: usage of SBR spread. */ + INT stat; /*!< Flag: usage of static framing. */ + INT e; /*!< Number of envelopes when static framing is chosen. */ SBR_STEREO_MODE stereoMode; /*!< SBR stereo mode. */ INT deltaTAcrossFrames; /*!< Flag: allow time-delta coding. */ - FIXP_DBL dF_edge_1stEnv; /*!< Extra fraction delta-F coding is allowed to be more expensive. */ - FIXP_DBL dF_edge_incr; /*!< Increment dF_edge_1stEnv this much if dT-coding was used this frame. */ - INT sbr_invf_mode; /*!< Inverse filtering mode. */ - INT sbr_xpos_mode; /*!< Transposer mode. */ - INT sbr_xpos_ctrl; /*!< Transposer control. */ - INT sbr_xpos_level; /*!< Transposer 3rd order level. */ - INT startFreq; /*!< The start frequency table index. */ - INT stopFreq; /*!< The stop frequency table index. */ - INT useSaPan; /*!< Flag: usage of SAPAN stereo. */ - INT dynBwEnabled; /*!< Flag: usage of dynamic bandwidth. */ - INT bParametricStereo; /*!< Flag: usage of parametric stereo coding tool. */ + FIXP_DBL dF_edge_1stEnv; /*!< Extra fraction delta-F coding is allowed to be + more expensive. */ + FIXP_DBL dF_edge_incr; /*!< Increment dF_edge_1stEnv this much if dT-coding + was used this frame. */ + INT sbr_invf_mode; /*!< Inverse filtering mode. */ + INT sbr_xpos_mode; /*!< Transposer mode. */ + INT sbr_xpos_ctrl; /*!< Transposer control. */ + INT sbr_xpos_level; /*!< Transposer 3rd order level. */ + INT startFreq; /*!< The start frequency table index. */ + INT stopFreq; /*!< The stop frequency table index. */ + INT useSaPan; /*!< Flag: usage of SAPAN stereo. */ + INT dynBwEnabled; /*!< Flag: usage of dynamic bandwidth. */ + INT bParametricStereo; /*!< Flag: usage of parametric stereo coding tool. */ /* header_extra1 configuration */ - UCHAR freqScale; /*!< Frequency grouping. */ - INT alterScale; /*!< Scale resolution. */ - INT sbr_noise_bands; /*!< Number of noise bands. */ - + UCHAR freqScale; /*!< Frequency grouping. */ + INT alterScale; /*!< Scale resolution. */ + INT sbr_noise_bands; /*!< Number of noise bands. */ /* header_extra2 configuration */ - INT sbr_limiter_bands; /*!< Number of limiter bands. */ - INT sbr_limiter_gains; /*!< Gain of limiter. */ - INT sbr_interpol_freq; /*!< Flag: use interpolation in freq. direction. */ - INT sbr_smoothing_length; /*!< Flag: choose length 4 or 0 (=on, off). */ + INT sbr_limiter_bands; /*!< Number of limiter bands. */ + INT sbr_limiter_gains; /*!< Gain of limiter. */ + INT sbr_interpol_freq; /*!< Flag: use interpolation in freq. direction. */ + INT sbr_smoothing_length; /*!< Flag: choose length 4 or 0 (=on, off). */ UCHAR init_amp_res_FF; FIXP_DBL threshold_AmpRes_FF_m; SCHAR threshold_AmpRes_FF_e; -} sbrConfiguration, *sbrConfigurationPtr ; +} sbrConfiguration, *sbrConfigurationPtr; -typedef struct SBR_CONFIG_DATA -{ - UINT sbrSyntaxFlags; /**< SBR syntax flags derived from AOT. */ - INT nChannels; /**< Number of channels. */ +typedef struct SBR_CONFIG_DATA { + UINT sbrSyntaxFlags; /**< SBR syntax flags derived from AOT. */ + INT nChannels; /**< Number of channels. */ - INT nSfb[2]; /**< Number of SBR scalefactor bands for LO_RES and HI_RES (?) */ - INT num_Master; /**< Number of elements in v_k_master. */ - INT sampleFreq; /**< SBR sampling frequency. */ + INT nSfb[2]; /**< Number of SBR scalefactor bands for LO_RES and HI_RES (?) */ + INT num_Master; /**< Number of elements in v_k_master. */ + INT sampleFreq; /**< SBR sampling frequency. */ INT frameSize; - INT xOverFreq; /**< The SBR start frequency. */ - INT dynXOverFreq; /**< Used crossover frequency when dynamic bandwidth is enabled. */ - INT noQmfBands; /**< Number of QMF frequency bands. */ - INT noQmfSlots; /**< Number of QMF slots. */ + INT xOverFreq; /**< The SBR start frequency. */ + INT dynXOverFreq; /**< Used crossover frequency when dynamic bandwidth is + enabled. */ - UCHAR *freqBandTable[2]; /**< Frequency table for low and hires, only MAX_FREQ_COEFFS/2 +1 coeffs actually needed for lowres. */ - UCHAR *v_k_master; /**< Master BandTable where freqBandTable is derived from. */ + INT noQmfBands; /**< Number of QMF frequency bands. */ + INT noQmfSlots; /**< Number of QMF slots. */ + UCHAR *freqBandTable[2]; /**< Frequency table for low and hires, only + MAX_FREQ_COEFFS/2 +1 coeffs actually needed for + lowres. */ + UCHAR + *v_k_master; /**< Master BandTable where freqBandTable is derived from. */ SBR_STEREO_MODE stereoMode; - INT noEnvChannels; /**< Number of envelope channels. */ - - INT useWaveCoding; /**< Flag indicates whether to use wave coding at all. */ - INT useParametricCoding; /**< Flag indicates whether to use para coding at all. */ - INT xposCtrlSwitch; /**< Flag indicates whether to switch xpos ctrl on the fly. */ - INT switchTransposers; /**< Flag indicates whether to switch xpos on the fly . */ + INT noEnvChannels; /**< Number of envelope channels. */ + + INT useWaveCoding; /**< Flag indicates whether to use wave coding at all. */ + INT useParametricCoding; /**< Flag indicates whether to use para coding at + all. */ + INT xposCtrlSwitch; /**< Flag indicates whether to switch xpos ctrl on the + fly. */ + INT switchTransposers; /**< Flag indicates whether to switch xpos on the fly . + */ UCHAR initAmpResFF; FIXP_DBL thresholdAmpResFF_m; SCHAR thresholdAmpResFF_e; @@ -267,6 +282,8 @@ typedef struct { INT bitRate; int instanceTag; UCHAR fParametricStereo; + UCHAR fDualMono; /**< This flags allows to disable coupling in sbr channel + pair element */ UCHAR nChannelsInEl; UCHAR ChannelIndex[2]; } SBR_ELEMENT_INFO; @@ -278,19 +295,15 @@ extern "C" { typedef struct SBR_ENCODER *HANDLE_SBR_ENCODER; /** - * \brief Get the max required input buffer size including delay balancing space - * for N audio channels. + * \brief Get the max required input buffer size including delay balancing + * space for N audio channels. * \param noChannels Number of audio channels. * \return Max required input buffer size in bytes. */ INT sbrEncoder_GetInBufferSize(int noChannels); -INT sbrEncoder_Open( - HANDLE_SBR_ENCODER *phSbrEncoder, - INT nElements, - INT nChannels, - INT supportPS - ); +INT sbrEncoder_Open(HANDLE_SBR_ENCODER *phSbrEncoder, INT nElements, + INT nChannels, INT supportPS); /** * \brief Get closest working bitrate to specified desired @@ -301,8 +314,8 @@ INT sbrEncoder_Open( * \param aot The current Audio Object Type * \return Closest working bit rate to bitRate value */ -UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate, AUDIO_OBJECT_TYPE aot); - +UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, + UINT coreSampleRate, AUDIO_OBJECT_TYPE aot); /** * \brief Check whether downsampled SBR single rate is possible @@ -316,50 +329,51 @@ UINT sbrEncoder_IsSingleRatePossible(AUDIO_OBJECT_TYPE aot); /** * \brief Initialize SBR Encoder instance. * \param phSbrEncoder Pointer to a SBR Encoder instance. - * \param elInfo Structure that describes the element/channel arrangement. + * \param elInfo Structure that describes the element/channel + * arrangement. * \param noElements Amount of elements described in elInfo. * \param inputBuffer Pointer to the encoder audio buffer + * \param inputBufferBufSize Buffer offset of one channel (frameSize + delay) * \param bandwidth Returns the core audio encoder bandwidth (output) - * \param bufferOffset Returns the offset for the audio input data in order to do delay balancing. - * \param numChannels Input: Encoder input channels. output: core encoder channels. - * \param sampleRate Input: Encoder samplerate. output core encoder samplerate. - * \param downSampleFactor Input: Relation between SBR and core coder sampling rate; - * \param frameLength Input: Encoder frameLength. output core encoder frameLength. - * \param aot Input: Desired AOT. output AOT to be used after parameter checking. - * \param delay Input: core encoder delay. Output: total delay because of SBR. + * \param bufferOffset Returns the offset for the audio input data in order + * to do delay balancing. + * \param numChannels Input: Encoder input channels. output: core encoder + * channels. + * \param sampleRate Input: Encoder samplerate. output core encoder + * samplerate. + * \param downSampleFactor Input: Relation between SBR and core coder sampling + * rate; + * \param frameLength Input: Encoder frameLength. output core encoder + * frameLength. + * \param aot Input: AOT.. + * \param delay Input: core encoder delay. Output: total delay + * because of SBR. * \param transformFactor The core encoder transform factor (blockswitching). * \param headerPeriod Repetition rate of the SBR header: * - (-1) means intern configuration. - * - (1-10) corresponds to header repetition rate in frames. + * - (1-10) corresponds to header repetition rate in + * frames. * \return 0 on success, and non-zero if failed. */ -INT sbrEncoder_Init( - HANDLE_SBR_ENCODER hSbrEncoder, - SBR_ELEMENT_INFO elInfo[(8)], - int noElements, - INT_PCM *inputBuffer, - INT *coreBandwidth, - INT *inputBufferOffset, - INT *numChannels, - INT *sampleRate, - UINT *downSampleFactor, - INT *frameLength, - AUDIO_OBJECT_TYPE aot, - int *delay, - int transformFactor, - const int headerPeriod, - ULONG statesInitFlag - ); +INT sbrEncoder_Init(HANDLE_SBR_ENCODER hSbrEncoder, + SBR_ELEMENT_INFO elInfo[(8)], int noElements, + INT_PCM *inputBuffer, UINT inputBufferBufSize, + INT *coreBandwidth, INT *inputBufferOffset, + INT *numChannels, const UINT syntaxFlags, INT *sampleRate, + UINT *downSampleFactor, INT *frameLength, + AUDIO_OBJECT_TYPE aot, int *delay, int transformFactor, + const int headerPeriod, ULONG statesInitFlag); /** - * \brief Do delay line buffers housekeeping. To be called after each encoded audio frame. + * \brief Do delay line buffers housekeeping. To be called after + * each encoded audio frame. * \param hEnvEnc SBR Encoder handle. * \param timeBuffer Pointer to the encoder audio buffer. + * \param timeBufferBufSIze buffer size for one channel * \return 0 on success, and non-zero if failed. */ -INT sbrEncoder_UpdateBuffers(HANDLE_SBR_ENCODER hEnvEnc, - INT_PCM *timeBuffer - ); +INT sbrEncoder_UpdateBuffers(HANDLE_SBR_ENCODER hEnvEnc, INT_PCM *timeBuffer, + UINT timeBufferBufSIze); /** * \brief Close SBR encoder instance. @@ -371,31 +385,63 @@ 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 samples Time samples, not interleaved. + * \param timeInStride Channel offset 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, - UINT sbrDataBits[(8)], - UCHAR sbrData[(8)][MAX_PAYLOAD_SIZE] - ); +INT sbrEncoder_EncodeFrame(HANDLE_SBR_ENCODER hEnvEncoder, INT_PCM *samples, + UINT samplesBufSize, UINT sbrDataBits[(8)], + UCHAR sbrData[(8)][MAX_PAYLOAD_SIZE]); /** * \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. + * \param fSendHeaders Flag indicating that the SBR encoder should send more + * headers in the SBR payload or not. * \return void */ -void sbrEncoder_GetHeader(HANDLE_SBR_ENCODER sbrEncoder, - HANDLE_FDK_BITSTREAM hBs, - INT element_index, - int fSendHeaders); +void sbrEncoder_GetHeader(HANDLE_SBR_ENCODER sbrEncoder, + HANDLE_FDK_BITSTREAM hBs, INT element_index, + int fSendHeaders); + +/** + * \brief Request to write SBR header. + * \param hSbrEncoder SBR encoder handle. + * \return 0 on success, and non-zero if failed. + */ +INT sbrEncoder_SendHeader(HANDLE_SBR_ENCODER hSbrEncoder); + +/** + * \brief Request if last sbr payload contains an SBR header. + * \param hSbrEncoder SBR encoder handle. + * \return 1 contains sbr header, 0 without sbr header. + */ +INT sbrEncoder_ContainsHeader(HANDLE_SBR_ENCODER hSbrEncoder); + +/** + * \brief SBR header delay in frames. + * \param hSbrEncoder SBR encoder handle. + * \return Delay in frames, -1 on failure. + */ +INT sbrEncoder_GetHeaderDelay(HANDLE_SBR_ENCODER hSbrEncoder); + +/** + * \brief Bitstrem delay in SBR frames. + * \param hSbrEncoder SBR encoder handle. + * \return Delay in frames, -1 on failure. + */ +INT sbrEncoder_GetBsDelay(HANDLE_SBR_ENCODER hSbrEncoder); + +/** + * \brief Prepare SBR payload for SAP. + * \param hSbrEncoder SBR encoder handle. + * \return 0 on success, and non-zero if failed. + */ +INT sbrEncoder_SAPPrepare(HANDLE_SBR_ENCODER hSbrEncoder); /** * \brief SBR encoder bitrate estimation. @@ -404,7 +450,6 @@ void sbrEncoder_GetHeader(HANDLE_SBR_ENCODER sbrEncoder, */ INT sbrEncoder_GetEstimateBitrate(HANDLE_SBR_ENCODER hSbrEncoder); - /** * \brief Delay between input data and downsampled output data. * \param hSbrEncoder SBR encoder handle. @@ -413,8 +458,16 @@ INT sbrEncoder_GetEstimateBitrate(HANDLE_SBR_ENCODER hSbrEncoder); INT sbrEncoder_GetInputDataDelay(HANDLE_SBR_ENCODER hSbrEncoder); /** + * \brief Delay caused by the SBR decoder. + * \param hSbrEncoder SBR encoder handle. + * \return Delay. + */ +INT sbrEncoder_GetSbrDecDelay(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. + * \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); @@ -424,7 +477,7 @@ void sbrPrintRAM(void); void sbrPrintROM(void); #ifdef __cplusplus - } +} #endif #endif /* ifndef __SBR_MAIN_H */ |