| |
| /* ----------------------------------------------------------------------------------------------------------- |
| 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 */ |
