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- /* -----------------------------------------------------------------------------------------------------------
- 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.
- 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
- [email protected]
- ----------------------------------------------------------------------------------------------------------- */
- /*************************** Fraunhofer IIS ***********************
- 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
- };
- 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 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 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 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,
- 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
- );
- /**
- * \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.
- * \return 0 on success, and non-zero if failed.
- */
- INT sbrEncoder_UpdateBuffers(HANDLE_SBR_ENCODER hEnvEnc,
- INT_PCM *timeBuffer
- );
- /**
- * \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, 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,
- 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 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 */
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