From 2228e360595641dd906bf1773307f43d304f5b2e Mon Sep 17 00:00:00 2001 From: The Android Open Source Project Date: Wed, 11 Jul 2012 10:15:24 -0700 Subject: Snapshot 2bda038c163298531d47394bc2c09e1409c5d0db Change-Id: If584e579464f28b97d50e51fc76ba654a5536c54 --- libSBRenc/include/sbr_encoder.h | 394 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 394 insertions(+) create mode 100644 libSBRenc/include/sbr_encoder.h (limited to 'libSBRenc/include/sbr_encoder.h') diff --git a/libSBRenc/include/sbr_encoder.h b/libSBRenc/include/sbr_encoder.h new file mode 100644 index 0000000..992c20c --- /dev/null +++ b/libSBRenc/include/sbr_encoder.h @@ -0,0 +1,394 @@ + +/* ----------------------------------------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2012 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 +----------------------------------------------------------------------------------------------------------- */ + +/*************************** 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 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 struct +{ + UINT bitrateFrom; /*!< inclusive */ + UINT bitrateTo; /*!< exclusive */ + + USHORT 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 freq_res_fixfix[3]; /*!< Frequency resolution of envelopes in frame class FIXFIX + 0=1 Env; 1=2 Env; 2=4 Env; */ + /* + 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. */ + INT bDownSampledSbr; /*!< Signal downsampled SBR is used. */ + + /* + 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; +} sbrConfiguration, *sbrConfigurationPtr ; + +typedef struct +{ + 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 coeefs 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; +} SBR_CONFIG_DATA; + +typedef 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 bit rate to specified desired bit rate 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 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 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 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). + * \return 0 on success, and non-zero if failed. + */ +INT sbrEncoder_Init( HANDLE_SBR_ENCODER hSbrEncoder, + SBR_ELEMENT_INFO elInfo[(6)], + int noElements, + INT_PCM *inputBuffer, + INT *bandwidth, + INT *bufferOffset, + INT *numChannels, + INT *sampleRate, + INT *frameLength, + AUDIO_OBJECT_TYPE *aot, + int *delay, + int transformFactor, + 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[(6)], + UCHAR sbrData[(6)][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(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 */ -- cgit v1.2.3