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author | Matthias P. Braendli <matthias.braendli@mpb.li> | 2019-11-11 11:38:02 +0100 |
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committer | Matthias P. Braendli <matthias.braendli@mpb.li> | 2019-11-11 11:38:02 +0100 |
commit | 0e5af65c467b2423a0b857ae3ad98c91acc1e190 (patch) | |
tree | d07f69550d8886271e44fe79c4dcfb299cafbd38 /fdk-aac/libSBRenc/include | |
parent | efe406d9724f959c8bc2a31802559ca6d41fd897 (diff) | |
download | ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.tar.gz ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.tar.bz2 ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.zip |
Include patched FDK-AAC in the repository
The initial idea was to get the DAB+ patch into upstream, but since
that follows the android source releases, there is no place for a custom
DAB+ patch there.
So instead of having to maintain a patched fdk-aac that has to have the
same .so version as the distribution package on which it is installed,
we prefer having a separate fdk-aac-dab library to avoid collision.
At that point, there's no reason to keep fdk-aac in a separate
repository, as odr-audioenc is the only tool that needs DAB+ encoding
support. Including it here simplifies installation, and makes it
consistent with toolame-dab, also shipped in this repository.
DAB+ decoding support (needed by ODR-SourceCompanion, dablin, etisnoop,
welle.io and others) can be done using upstream FDK-AAC.
Diffstat (limited to 'fdk-aac/libSBRenc/include')
-rw-r--r-- | fdk-aac/libSBRenc/include/sbr_encoder.h | 483 |
1 files changed, 483 insertions, 0 deletions
diff --git a/fdk-aac/libSBRenc/include/sbr_encoder.h b/fdk-aac/libSBRenc/include/sbr_encoder.h new file mode 100644 index 0000000..d979ba6 --- /dev/null +++ b/fdk-aac/libSBRenc/include/sbr_encoder.h @@ -0,0 +1,483 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© 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. + +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: + +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 +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. + +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." + +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. + +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. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** SBR encoder library ****************************** + + Author(s): + + Description: SBR encoder top level processing prototype + +*******************************************************************************/ + +#ifndef SBR_ENCODER_H +#define SBR_ENCODER_H + +#include "common_fix.h" +#include "FDK_audio.h" + +#include "FDK_bitstream.h" + +/* core coder helpers */ +#define MAX_TRANS_FAC 8 +#define MAX_CODEC_FRAME_RATIO 2 +#define MAX_PAYLOAD_SIZE 256 + +typedef enum codecType { + CODEC_AAC = 0, + CODEC_AACLD = 1, + CODEC_UNSPECIFIED = 99 +} CODEC_TYPE; + +typedef struct { + INT bitRate; + INT nChannels; + INT sampleFreq; + INT transFac; + INT standardBitrate; +} CODEC_PARAM; + +typedef enum { + SBR_MONO, + SBR_LEFT_RIGHT, + SBR_COUPLING, + SBR_SWITCH_LRC +} SBR_STEREO_MODE; + +/* bitstream syntax flags */ +enum { + SBR_SYNTAX_LOW_DELAY = 0x0001, + 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; /*!< */ +} sbrTuningTable_t; + +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) */ + + /* + 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. + */ + + /* + 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. */ + 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. */ + + /* + header_extra1 configuration + */ + 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). */ + UCHAR init_amp_res_FF; + FIXP_DBL threshold_AmpRes_FF_m; + SCHAR threshold_AmpRes_FF_e; +} sbrConfiguration, *sbrConfigurationPtr; + +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 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. */ + + 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 . + */ + UCHAR initAmpResFF; + FIXP_DBL thresholdAmpResFF_m; + SCHAR thresholdAmpResFF_e; +} SBR_CONFIG_DATA, *HANDLE_SBR_CONFIG_DATA; + +typedef struct { + MP4_ELEMENT_ID elType; + 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; + +#ifdef __cplusplus +extern "C" { +#endif + +typedef struct SBR_ENCODER *HANDLE_SBR_ENCODER; + +/** + * \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); + +/** + * \brief Get closest working bitrate to specified desired + * bitrate for a single SBR element. + * \param bitRate The desired target bit rate + * \param numChannels The amount of audio channels + * \param coreSampleRate The sample rate of the core coder + * \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); + +/** + * \brief Check whether downsampled SBR single rate is possible + * with given audio object type. + * \param aot The Audio object type. + * \return 0 when downsampled SBR is not possible, + * 1 when downsampled SBR is possible. + */ +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 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: 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. + * \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, 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. + * \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, + UINT timeBufferBufSIze); + +/** + * \brief Close SBR encoder instance. + * \param phEbrEncoder Handle of SBR encoder instance to be closed. + * \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, 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 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. + * \return void + */ +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. + * \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 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. + * \return 0 on sucess. + */ +INT sbrEncoder_GetLibInfo(LIB_INFO *info); + +void sbrPrintRAM(void); + +void sbrPrintROM(void); + +#ifdef __cplusplus +} +#endif + +#endif /* ifndef __SBR_MAIN_H */ |