diff options
Diffstat (limited to 'media/libstagefright/codecs/aacdec/pvmp4audiodecoderframe.cpp')
| -rw-r--r-- | media/libstagefright/codecs/aacdec/pvmp4audiodecoderframe.cpp | 1458 |
1 files changed, 0 insertions, 1458 deletions
diff --git a/media/libstagefright/codecs/aacdec/pvmp4audiodecoderframe.cpp b/media/libstagefright/codecs/aacdec/pvmp4audiodecoderframe.cpp deleted file mode 100644 index 7a279dc..0000000 --- a/media/libstagefright/codecs/aacdec/pvmp4audiodecoderframe.cpp +++ /dev/null @@ -1,1458 +0,0 @@ -/* ------------------------------------------------------------------ - * Copyright (C) 1998-2009 PacketVideo - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either - * express or implied. - * See the License for the specific language governing permissions - * and limitations under the License. - * ------------------------------------------------------------------- - */ -/* - - Pathname: pvmp4audiodecodeframe - ------------------------------------------------------------------------------- - REVISION HISTORY - - Description: Modified from original shareware code - - Description: Pulled in loop structure from console.c, so that this function - now decodes all frames in the file. - - Original program used several global variables. These have been - eliminated, except for situations in which the global variables - could be converted into const types. Otherwise, they are passed - by reference through the functions. - - Description: Begin mods for file I/O removal - - Description: Merged trans4m_freq_2_time, trans4m_time_2_freq, etc. - - Description: Removing commented out sections of code. This includes the - removal of unneeded functions init_lt_pred, reset_mc_info, - - Description: Copied from aac_decode_frame.c and renamed file, - Made many changes. - - Description: Prepare for code review - - Description: Update per review comments: - 1) Add comment about leaveGetLoop - 2) Remove inverseTNSCoef array - 3) fix wnd_shape_this_bk to wnd_shape_prev_bk in F to T - 4) Clean up comments - 5) Change call to long_term_synthesis - - Description: Remove division for calculation of bitrate. - - Description: Remove update of LTP buffers if not LTP audio object type. - - Description: Add hasmask to call to right_ch_sfb_tools_ms - - Description: - Modified to call ltp related routines on the left channel - before intensity is called on the right channel. The previous version - was causing a problem when IS was used on the right channel and LTP - on the left channel for the same scalefactor band. - - This fix required creating a new function, apply_ms_synt, deleting another - function (right_ch_sfb_tools_noms.c), and modifying the calling order of - the other functions. - - Description: Made changes per review comments. - - Description: Changed name of right_ch_sfb_tools_ms to pns_intensity_right - - Description: Added cast, since pVars->inputStream.usedBits is UInt, and - pExt->remainderBits is Int. - - pExt->remainderBits = - (Int)(pVars->inputStream.usedBits & INBUF_BIT_MODULO_MASK); - - Description: Modified to pass a pointer to scratch memory into - tns_setup_filter.c - - Description: Removed include of "s_TNSInfo.h" - - Description: Removed call to "tns_setup_filter" which has been eliminated - by merging its functionality into "get_tns" - - Description: Passing in a pointer to a q-format array, rather than - the address of a single q-format, for the inverse filter case for - apply_tns. - - Description: - (1) Added #include of "e_ElementId.h" - Previously, this function was relying on another include file - to include "e_ElementId.h" - - (2) Updated the copyright header. - - Description: - Per review comments, declared two temporary variables - - pChLeftShare = pChVars[LEFT]->pShareWfxpCoef; - pChRightShare = pChVars[RIGHT]->pShareWfxpCoef; - - Description: - long_term_synthesis should have been invoked with max_sfb - as the 2nd parameter, rather than pFrameInfo->sfb_per_win[0]. - - Old - long_term_synthesis( - pChVars[ch]->wnd, - pFrameInfo->sfb_per_win[0] ... - - Correction - long_term_synthesis( - pChVars[ch]->wnd, - pChVars[ch]->pShareWfxpCoef->max_sfb ... - - This problem caused long_term_synthesis to read memory which - was not initialized in get_ics_info.c - - Description: - (1) Utilize scratch memory for the scratch Prog_Config. - - Description: (1) Modified to decode ID_END syntactic element after header - - Description: - (1) Reconfigured LTP buffer as a circular buffer. This saves - 2048 Int16->Int16 copies per frame. - - Description: Updated so ltp buffers are not used as a wasteful - intermediate buffer for LC streams. Data is transferred directly - from the filterbank to the output stream. - - Description: Decode ADIF header if frame count is zero. - The AudioSpecificConfig is decoded by a separate API. - - Description: Added comments explaining how the ltp_buffer_state - variable is updated. - - - Description: Modified code to take advantage of new trans4m_freq_2_time_fxp, - which writes the output directly into a 16-bit output buffer. This - improvement allows faster operation by reducing the amount of memory - transfers. Speed can be further improved on most platforms via use of a - DMA transfer in the function write_output.c - - Description: perChan[] is an array of structures in tDec_Int_File. Made - corresponding changes. - - Description: Included changes in interface for q_normalize() and - trans4m_freq_2_time_fxp. - - Description: Included changes in interface for long_term_prediction. - - Description: Added support for DSE (Data Streaming Channel). Added - function get_dse() and included file get_dse.h - - Description: Added support for the ill-case when a raw data block contains - only a terminator <ID_END>. This is illegal but is added - for convinience - - Description: Added support for empty audio frames, such the one containing - only DSE or FILL elements. A trap was added to stop processing - when no audio information was sent. - - Description: Added support for adts format files. Added saturation to - floating point version of aac+ decoding - - Description: - ------------------------------------------------------------------------------- - INPUT AND OUTPUT DEFINITIONS - - Inputs: - pExt = pointer to the external interface structure. See the file - PVMP4AudioDecoder_API.h for a description of each field. - Data type of pointer to a tPVMP4AudioDecoderExternal - structure. - - pMem = void pointer to hide the internal implementation of the library - It is cast back to a tDec_Int_File structure. This structure - contains information that needs to persist between calls to - this function, or is too big to be placed on the stack, even - though the data is only needed during execution of this function - Data type void pointer, internally pointer to a tDec_Int_File - structure. - - Local Stores/Buffers/Pointers Needed: None - (The memory set aside in pMem performs this task) - - Global Stores/Buffers/Pointers Needed: None - - Outputs: - status = 0 if no error occurred - MP4AUDEC_NONRECOVERABLE if a non-recoverable error occurred - MP4AUDEC_RECOVERABLE if a recoverable error occurred. - Presently a recoverable error does not exist, but this - was a requirement. - - - Pointers and Buffers Modified: - pMem contents are modified. - pExt: (more detail in the file PVMP4AudioDecoder_API.h) - inputBufferUsedLength - number of array elements used up by the stream. - remainderBits - remaining bits in the next UInt32 buffer - samplingRate - sampling rate in samples per sec - bitRate - bit rate in bits per second, varies frame to frame. - encodedChannels - channels found on the file (informative) - frameLength - length of the frame - - Local Stores Modified: None. - - Global Stores Modified: None. - ------------------------------------------------------------------------------- - FUNCTION DESCRIPTION - - Decodes one frame of an MPEG-2/MPEG-4 encoded audio bitstream. - - This function calls the various components of the decoder in the proper order. - - - Left Channel Right Channel - | | - | | - | | - \|/ \|/ - #1 ____________________ #2 ____________________ - | | | | - | Huffman Decoding | | Huffman Decoding | - |__________________| |__________________| - | | - | | - | | - \|/ | - #3 ____________________ | - | | | - | PNS LEFT | | - |__________________| | - | | - | | - | | - \|/ \|/ - #4 ______________________________________________________________________ - | | - | Apply MS_Synt | - |____________________________________________________________________| - | | - | | - \|/ | - #5 ____________________ | - | | W - | LTP | A - |__________________| I - | T - | | - | F - \|/ O - #6 ____________________ R - | | | - | Time -> Freq | L - |__________________| E - | F - | T - | | - \|/ C - #7 ____________________ H - | | A - | TNS Inverse | N - |__________________| N - | E - | L - | | - \|/ | - #8 ____________________ | - | | | - | Long Term Synth | | - |__________________| | - | | - | \|/ - | #9 ____________________ - | | | - |--DATA ON LEFT CHANNEL MAY BE USED----->| PNS/Intensity Rt | - | |__________________| - | | - | | - | \|/ - | #10 ____________________ - W | | - A | LTP | - I |__________________| - T | - | | - F | - O \|/ - R #11 ____________________ - | | | - R | Time -> Freq | - I |__________________| - G | - H | - T | - | \|/ - C #12 ____________________ - H | | - A | TNS Inverse | - N |__________________| - N | - E | - L | - | \|/ - | #13 ____________________ - | | | - | | Long Term Synth | - | |__________________| - | | - | | - | | - \|/ \|/ -#14 ____________________ #18 ____________________ - | | | | - | TNS | | TNS | - |__________________| |__________________| - | | - | | - | | - \|/ \|/ -#15 ____________________ #19 ____________________ - | | | | - | qFormatNorm | | qFormatNorm | - |__________________| |__________________| - | | - | | - | | - \|/ \|/ -#16 ____________________ #20 ____________________ - | | | | - | Freq / Time | | Freq / Time | - |__________________| |__________________| - | | - | | - | | - \|/ \|/ -#17 ____________________ #21 ____________________ - | | | | - | Limit Buffer | | Limit Buffer | - |__________________| |__________________| - | | - | | - | | - \|/ \|/ -#22 ______________________________________________________________________ - | | - | Write Output | - |____________________________________________________________________| - - ------------------------------------------------------------------------------- - REQUIREMENTS - - PacketVideo Document # CCC-AUD-AAC-ERS-0003 - ------------------------------------------------------------------------------- - REFERENCES - - (1) MPEG-2 NBC Audio Decoder - "This software module was originally developed by AT&T, Dolby - Laboratories, Fraunhofer Gesellschaft IIS in the course of development - of the MPEG-2 NBC/MPEG-4 Audio standard ISO/IEC 13818-7, 14496-1,2 and - 3. This software module is an implementation of a part of one or more - MPEG-2 NBC/MPEG-4 Audio tools as specified by the MPEG-2 NBC/MPEG-4 - Audio standard. ISO/IEC gives users of the MPEG-2 NBC/MPEG-4 Audio - standards free license to this software module or modifications thereof - for use in hardware or software products claiming conformance to the - MPEG-2 NBC/MPEG-4 Audio standards. Those intending to use this software - module in hardware or software products are advised that this use may - infringe existing patents. The original developer of this software - module and his/her company, the subsequent editors and their companies, - and ISO/IEC have no liability for use of this software module or - modifications thereof in an implementation. Copyright is not released - for non MPEG-2 NBC/MPEG-4 Audio conforming products.The original - developer retains full right to use the code for his/her own purpose, - assign or donate the code to a third party and to inhibit third party - from using the code for non MPEG-2 NBC/MPEG-4 Audio conforming products. - This copyright notice must be included in all copies or derivative - works." - Copyright(c)1996. - ------------------------------------------------------------------------------- - RESOURCES USED - When the code is written for a specific target processor the - the resources used should be documented below. - - STACK USAGE: [stack count for this module] + [variable to represent - stack usage for each subroutine called] - - where: [stack usage variable] = stack usage for [subroutine - name] (see [filename].ext) - - DATA MEMORY USED: x words - - PROGRAM MEMORY USED: x words - - CLOCK CYCLES: [cycle count equation for this module] + [variable - used to represent cycle count for each subroutine - called] - - where: [cycle count variable] = cycle count for [subroutine - name] (see [filename].ext) - ------------------------------------------------------------------------------- -*/ - - -/*---------------------------------------------------------------------------- -; INCLUDES -----------------------------------------------------------------------------*/ -#include "pv_audio_type_defs.h" - -#include "s_tdec_int_chan.h" -#include "s_tdec_int_file.h" -#include "aac_mem_funcs.h" -#include "sfb.h" /* Where samp_rate_info[] is declared */ -#include "e_tmp4audioobjecttype.h" -#include "e_elementid.h" - - -#include "get_adif_header.h" -#include "get_adts_header.h" -#include "get_audio_specific_config.h" -#include "ibstream.h" /* where getbits is declared */ - -#include "huffman.h" /* where huffdecode is declared */ -#include "get_prog_config.h" -#include "getfill.h" -#include "pns_left.h" - -#include "apply_ms_synt.h" -#include "pns_intensity_right.h" -#include "q_normalize.h" -#include "long_term_prediction.h" -#include "long_term_synthesis.h" -#include "ltp_common_internal.h" -#include "apply_tns.h" - -#include "window_block_fxp.h" - -#include "write_output.h" - -#include "pvmp4audiodecoder_api.h" /* Where this function is declared */ -#include "get_dse.h" - -#include "sbr_applied.h" -#include "sbr_open.h" -#include "get_sbr_bitstream.h" -#include "e_sbr_element_id.h" - - - -/*---------------------------------------------------------------------------- -; MACROS -; Define module specific macros here -----------------------------------------------------------------------------*/ - -/*---------------------------------------------------------------------------- -; DEFINES -; Include all pre-processor statements here. Include conditional -; compile variables also. -----------------------------------------------------------------------------*/ - -#define LEFT (0) -#define RIGHT (1) - - -/*---------------------------------------------------------------------------- -; LOCAL FUNCTION DEFINITIONS -; Function Prototype declaration -----------------------------------------------------------------------------*/ - -/*---------------------------------------------------------------------------- -; LOCAL STORE/BUFFER/POINTER DEFINITIONS -; Variable declaration - defined here and used outside this module -----------------------------------------------------------------------------*/ - -/*---------------------------------------------------------------------------- -; EXTERNAL FUNCTION REFERENCES -; Declare functions defined elsewhere and referenced in this module -----------------------------------------------------------------------------*/ - -void InitSbrSynFilterbank(bool bDownSampleSBR); - - - -/*---------------------------------------------------------------------------- -; EXTERNAL GLOBAL STORE/BUFFER/POINTER REFERENCES -; Declare variables used in this module but defined elsewhere -----------------------------------------------------------------------------*/ - -/*---------------------------------------------------------------------------- -; FUNCTION CODE -----------------------------------------------------------------------------*/ - - -OSCL_EXPORT_REF Int PVMP4AudioDecodeFrame( - tPVMP4AudioDecoderExternal *pExt, - void *pMem) -{ - Int frameLength; /* Helper variable */ - Int ch; - Int id_syn_ele; - UInt initialUsedBits; /* Unsigned for C55x */ - Int qFormatNorm; - Int qPredictedSamples; - Bool leaveGetLoop; - MC_Info *pMC_Info; /* Helper pointer */ - FrameInfo *pFrameInfo; /* Helper pointer */ - tDec_Int_File *pVars; /* Helper pointer */ - tDec_Int_Chan *pChVars[Chans]; /* Helper pointer */ - - per_chan_share_w_fxpCoef *pChLeftShare; /* Helper pointer */ - per_chan_share_w_fxpCoef *pChRightShare; /* Helper pointer */ - - Int status = MP4AUDEC_SUCCESS; - - - Bool empty_frame; - -#ifdef AAC_PLUS - - SBRDECODER_DATA *sbrDecoderData; - SBR_DEC *sbrDec; - SBRBITSTREAM *sbrBitStream; - -#endif - /* - * Initialize "helper" pointers to existing memory. - */ - pVars = (tDec_Int_File *)pMem; - - pMC_Info = &pVars->mc_info; - - pChVars[LEFT] = &pVars->perChan[LEFT]; - pChVars[RIGHT] = &pVars->perChan[RIGHT]; - - pChLeftShare = pChVars[LEFT]->pShareWfxpCoef; - pChRightShare = pChVars[RIGHT]->pShareWfxpCoef; - - -#ifdef AAC_PLUS - - sbrDecoderData = (SBRDECODER_DATA *) & pVars->sbrDecoderData; - sbrDec = (SBR_DEC *) & pVars->sbrDec; - sbrBitStream = (SBRBITSTREAM *) & pVars->sbrBitStr; - -#ifdef PARAMETRICSTEREO - sbrDecoderData->hParametricStereoDec = (HANDLE_PS_DEC) & pVars->sbrDecoderData.ParametricStereoDec; -#endif - -#endif - /* - * Translate input buffer variables. - */ - pVars->inputStream.pBuffer = pExt->pInputBuffer; - - pVars->inputStream.inputBufferCurrentLength = (UInt)pExt->inputBufferCurrentLength; - - pVars->inputStream.availableBits = - (UInt)(pExt->inputBufferCurrentLength << INBUF_ARRAY_INDEX_SHIFT); - - initialUsedBits = - (UInt)((pExt->inputBufferUsedLength << INBUF_ARRAY_INDEX_SHIFT) + - pExt->remainderBits); - - pVars->inputStream.usedBits = initialUsedBits; - - if (initialUsedBits > pVars->inputStream.availableBits) - { - status = MP4AUDEC_INVALID_FRAME; - } - else if (pVars->bno == 0) - { - /* - * Attempt to read in ADIF format first because it is easily identified. - * If its not an ADIF bitstream, get_adif_header rewinds the "pointer" - * (actually usedBits). - */ - status = - get_adif_header( - pVars, - &(pVars->scratch.scratch_prog_config)); - - byte_align(&pVars->inputStream); - - if (status == SUCCESS) - { - pVars->prog_config.file_is_adts = FALSE; - } - else /* we've tried simple audio config, adif, then it should be adts */ - { - pVars->prog_config.file_is_adts = TRUE; - } - } - else if ((pVars->bno == 1) && (pVars->prog_config.file_is_adts == FALSE)) - { - - /* - * There might be an ID_END element following immediately after the - * AudioSpecificConfig header. This syntactic element should be read - * and byte_aligned before proceeds to decode "real" AAC raw data. - */ - id_syn_ele = (Int)getbits(LEN_SE_ID, &pVars->inputStream) ; - - if (id_syn_ele == ID_END) - { - - byte_align(&pVars->inputStream); - - pExt->inputBufferUsedLength = - pVars->inputStream.usedBits >> INBUF_ARRAY_INDEX_SHIFT; - - pExt->remainderBits = pVars->inputStream.usedBits & INBUF_BIT_MODULO_MASK; - - pVars->bno++; - - return(status); - } - else - { - /* - * Rewind bitstream pointer so that the syntactic element can be - * read when decoding raw bitstream - */ - pVars->inputStream.usedBits -= LEN_SE_ID; - } - - } - - if (pVars->prog_config.file_is_adts == TRUE) - { - /* - * If file is adts format, let the decoder handle only on data raw - * block at the time, once the last (or only) data block has been - * processed, then synch on the next header - */ - if (pVars->prog_config.headerless_frames) - { - pVars->prog_config.headerless_frames--; /* raw data block counter */ - } - else - { - status = get_adts_header(pVars, - &(pVars->syncword), - &(pVars->invoke), - 3); /* CorrectlyReadFramesCount */ - - if (status != SUCCESS) - { - status = MP4AUDEC_LOST_FRAME_SYNC; /* we lost track of header */ - } - } - } - else - { - byte_align(&pVars->inputStream); - } - -#ifdef AAC_PLUS - sbrBitStream->NrElements = 0; - sbrBitStream->NrElementsCore = 0; - -#endif - - /* - * The variable leaveGetLoop is used to signal that the following - * loop can be left, which retrieves audio syntatic elements until - * an ID_END is found, or an error occurs. - */ - leaveGetLoop = FALSE; - empty_frame = TRUE; - - while ((leaveGetLoop == FALSE) && (status == SUCCESS)) - { - /* get audio syntactic element */ - id_syn_ele = (Int)get9_n_lessbits(LEN_SE_ID, &pVars->inputStream); - - /* - * As fractional frames are a possible input, check that parsing does not - * go beyond the available bits before parsing the syntax. - */ - if (pVars->inputStream.usedBits > pVars->inputStream.availableBits) - { - status = MP4AUDEC_INCOMPLETE_FRAME; /* possible EOF or fractional frame */ - id_syn_ele = ID_END; /* quit while-loop */ - } - - switch (id_syn_ele) - { - case ID_END: /* terminator field */ - leaveGetLoop = TRUE; - break; - - case ID_SCE: /* single channel */ - case ID_CPE: /* channel pair */ - empty_frame = FALSE; - status = - huffdecode( - id_syn_ele, - &(pVars->inputStream), - pVars, - pChVars); - -#ifdef AAC_PLUS - if (id_syn_ele == ID_SCE) - { - sbrBitStream->sbrElement[sbrBitStream->NrElements].ElementID = SBR_ID_SCE; - } - else if (id_syn_ele == ID_CPE) - { - sbrBitStream->sbrElement[sbrBitStream->NrElements].ElementID = SBR_ID_CPE; - } - sbrBitStream->NrElementsCore++; - - -#endif - - break; - - case ID_PCE: /* program config element */ - /* - * PCE are not accepted in the middle of a - * raw_data_block. If found, a possible error may happen - * If a PCE is encountered during the first 2 frames, - * it will be read and accepted - * if its tag matches the first, with no error checking - * (inside of get_prog_config) - */ - - if (pVars->bno <= 1) - { - status = get_prog_config(pVars, - &(pVars->scratch.scratch_prog_config)); - } - else - { - status = MP4AUDEC_INVALID_FRAME; - } - break; - - case ID_FIL: /* fill element */ -#ifdef AAC_PLUS - get_sbr_bitstream(sbrBitStream, &pVars->inputStream); - -#else - getfill(&pVars->inputStream); -#endif - - break; - - case ID_DSE: /* Data Streaming element */ - get_dse(pVars->share.data_stream_bytes, - &pVars->inputStream); - break; - - default: /* Unsupported element, including ID_LFE */ - status = -1; /* ERROR CODE needs to be updated */ - break; - - } /* end switch() */ - - } /* end while() */ - - byte_align(&pVars->inputStream); - - /* - * After parsing the first frame ( bno=0 (adif), bno=1 (raw)) - * verify if implicit signalling is forcing to upsample AAC with - * no AAC+/eAAC+ content. If so, disable upsampling - */ - -#ifdef AAC_PLUS - if (pVars->bno <= 1) - { - if ((pVars->mc_info.ExtendedAudioObjectType == MP4AUDIO_AAC_LC) && - (!sbrBitStream->NrElements)) - { - PVMP4AudioDecoderDisableAacPlus(pExt, pMem); - } - } -#endif - - /* - * There might be an empty raw data block with only a - * ID_END element or non audio ID_DSE, ID_FIL - * This is an "illegal" condition but this trap - * avoids any further processing - */ - - if (empty_frame == TRUE) - { - pExt->inputBufferUsedLength = - pVars->inputStream.usedBits >> INBUF_ARRAY_INDEX_SHIFT; - - pExt->remainderBits = pVars->inputStream.usedBits & INBUF_BIT_MODULO_MASK; - - pVars->bno++; - - return(status); - - } - -#ifdef AAC_PLUS - - if (sbrBitStream->NrElements) - { - /* for every core SCE or CPE there must be an SBR element, otherwise sths. wrong */ - if (sbrBitStream->NrElements != sbrBitStream->NrElementsCore) - { - status = MP4AUDEC_INVALID_FRAME; - } - - if (pExt->aacPlusEnabled == false) - { - sbrBitStream->NrElements = 0; /* disable aac processing */ - } - } - else - { - /* - * This is AAC, but if aac+/eaac+ was declared in the stream, and there is not sbr content - * something is wrong - */ - if (pMC_Info->sbrPresentFlag || pMC_Info->psPresentFlag) - { - status = MP4AUDEC_INVALID_FRAME; - } - } -#endif - - - - - /* - * Signal processing section. - */ - frameLength = pVars->frameLength; - - if (status == SUCCESS) - { - /* - * PNS and INTENSITY STEREO and MS - */ - - pFrameInfo = pVars->winmap[pChVars[LEFT]->wnd]; - - pns_left( - pFrameInfo, - pChLeftShare->group, - pChLeftShare->cb_map, - pChLeftShare->factors, - pChLeftShare->lt_status.sfb_prediction_used, - pChLeftShare->lt_status.ltp_data_present, - pChVars[LEFT]->fxpCoef, - pChLeftShare->qFormat, - &(pVars->pns_cur_noise_state)); - - /* - * apply_ms_synt can only be ran for common windows. - * (where both the left and right channel share the - * same grouping, window length, etc. - * - * pVars->hasmask will be > 0 only if - * common windows are enabled for this frame. - */ - - if (pVars->hasmask > 0) - { - apply_ms_synt( - pFrameInfo, - pChLeftShare->group, - pVars->mask, - pChLeftShare->cb_map, - pChVars[LEFT]->fxpCoef, - pChVars[RIGHT]->fxpCoef, - pChLeftShare->qFormat, - pChRightShare->qFormat); - } - - for (ch = 0; (ch < pMC_Info->nch); ch++) - { - pFrameInfo = pVars->winmap[pChVars[ch]->wnd]; - - /* - * Note: This MP4 library assumes that if there are two channels, - * then the second channel is right AND it was a coupled channel, - * therefore there is no need to check the "is_cpe" flag. - */ - - if (ch > 0) - { - pns_intensity_right( - pVars->hasmask, - pFrameInfo, - pChRightShare->group, - pVars->mask, - pChRightShare->cb_map, - pChLeftShare->factors, - pChRightShare->factors, - pChRightShare->lt_status.sfb_prediction_used, - pChRightShare->lt_status.ltp_data_present, - pChVars[LEFT]->fxpCoef, - pChVars[RIGHT]->fxpCoef, - pChLeftShare->qFormat, - pChRightShare->qFormat, - &(pVars->pns_cur_noise_state)); - } - - if (pChVars[ch]->pShareWfxpCoef->lt_status.ltp_data_present != FALSE) - { - /* - * LTP - Long Term Prediction - */ - - qPredictedSamples = long_term_prediction( - pChVars[ch]->wnd, - pChVars[ch]->pShareWfxpCoef->lt_status. - weight_index, - pChVars[ch]->pShareWfxpCoef->lt_status. - delay, - pChVars[ch]->ltp_buffer, - pVars->ltp_buffer_state, - pChVars[ch]->time_quant, - pVars->share.predictedSamples, /* Scratch */ - frameLength); - - trans4m_time_2_freq_fxp( - pVars->share.predictedSamples, - pChVars[ch]->wnd, - pChVars[ch]->wnd_shape_prev_bk, - pChVars[ch]->wnd_shape_this_bk, - &qPredictedSamples, - pVars->scratch.fft); /* scratch memory for FFT */ - - - /* - * To solve a potential problem where a pointer tied to - * the qFormat was being incremented, a pointer to - * pChVars[ch]->qFormat is passed in here rather than - * the address of qPredictedSamples. - * - * Neither values are actually needed in the case of - * inverse filtering, but the pointer was being - * passed (and incremented) regardless. - * - * So, the solution is to pass a space of memory - * that a pointer can happily point to. - */ - - /* This is the inverse filter */ - apply_tns( - pVars->share.predictedSamples, /* scratch re-used for each ch */ - pChVars[ch]->pShareWfxpCoef->qFormat, /* Not used by the inv_filter */ - pFrameInfo, - &(pChVars[ch]->pShareWfxpCoef->tns), - TRUE, /* TRUE is FIR */ - pVars->scratch.tns_inv_filter); - - /* - * For the next function long_term_synthesis, - * the third param win_sfb_top[], and - * the tenth param coef_per_win, - * are used differently that in the rest of the project. This - * is because originally the ISO code was going to have - * these parameters change as the "short window" changed. - * These are all now the same value for each of the eight - * windows. This is why there is a [0] at the - * end of each of theses parameters. - * Note in particular that win_sfb_top was originally an - * array of pointers to arrays, but inside long_term_synthesis - * it is now a simple array. - * When the rest of the project functions are changed, the - * structure FrameInfo changes, and the [0]'s are removed, - * this comment could go away. - */ - long_term_synthesis( - pChVars[ch]->wnd, - pChVars[ch]->pShareWfxpCoef->max_sfb, - pFrameInfo->win_sfb_top[0], /* Look above */ - pChVars[ch]->pShareWfxpCoef->lt_status.win_prediction_used, - pChVars[ch]->pShareWfxpCoef->lt_status.sfb_prediction_used, - pChVars[ch]->fxpCoef, /* input and output */ - pChVars[ch]->pShareWfxpCoef->qFormat, /* input and output */ - pVars->share.predictedSamples, - qPredictedSamples, /* q format for previous aray */ - pFrameInfo->coef_per_win[0], /* Look above */ - NUM_SHORT_WINDOWS, - NUM_RECONSTRUCTED_SFB); - - } /* end if (pChVars[ch]->lt_status.ltp_data_present != FALSE) */ - - } /* for(ch) */ - - for (ch = 0; (ch < pMC_Info->nch); ch++) - { - - pFrameInfo = pVars->winmap[pChVars[ch]->wnd]; - - /* - * TNS - Temporal Noise Shaping - */ - - /* This is the forward filter - * - * A special note: Scratch memory is not used by - * the forward filter, but is passed in to maintain - * common interface for inverse and forward filter - */ - apply_tns( - pChVars[ch]->fxpCoef, - pChVars[ch]->pShareWfxpCoef->qFormat, - pFrameInfo, - &(pChVars[ch]->pShareWfxpCoef->tns), - FALSE, /* FALSE is IIR */ - pVars->scratch.tns_inv_filter); - - /* - * Normalize the q format across all scale factor bands - * to one value. - */ - qFormatNorm = - q_normalize( - pChVars[ch]->pShareWfxpCoef->qFormat, - pFrameInfo, - pChVars[ch]->abs_max_per_window, - pChVars[ch]->fxpCoef); - - /* - * filterbank - converts frequency coeficients to time domain. - */ - -#ifdef AAC_PLUS - if (sbrBitStream->NrElements == 0 && pMC_Info->upsamplingFactor == 1) - { - trans4m_freq_2_time_fxp_2( - pChVars[ch]->fxpCoef, - pChVars[ch]->time_quant, - pChVars[ch]->wnd, /* window sequence */ - pChVars[ch]->wnd_shape_prev_bk, - pChVars[ch]->wnd_shape_this_bk, - qFormatNorm, - pChVars[ch]->abs_max_per_window, - pVars->scratch.fft, - &pExt->pOutputBuffer[ch]); - /* - * Update LTP buffers if needed - */ - - if (pVars->mc_info.audioObjectType == MP4AUDIO_LTP) - { - Int16 * pt = &pExt->pOutputBuffer[ch]; - Int16 * ptr = &(pChVars[ch]->ltp_buffer[pVars->ltp_buffer_state]); - Int16 x, y; - for (Int32 i = HALF_LONG_WINDOW; i != 0; i--) - { - x = *pt; - pt += 2; - y = *pt; - pt += 2; - *(ptr++) = x; - *(ptr++) = y; - } - } - } - else - { - trans4m_freq_2_time_fxp_1( - pChVars[ch]->fxpCoef, - pChVars[ch]->time_quant, - &(pChVars[ch]->ltp_buffer[pVars->ltp_buffer_state + 288]), - pChVars[ch]->wnd, /* window sequence */ - pChVars[ch]->wnd_shape_prev_bk, - pChVars[ch]->wnd_shape_this_bk, - qFormatNorm, - pChVars[ch]->abs_max_per_window, - pVars->scratch.fft); - - } -#else - - trans4m_freq_2_time_fxp_2( - pChVars[ch]->fxpCoef, - pChVars[ch]->time_quant, - pChVars[ch]->wnd, /* window sequence */ - pChVars[ch]->wnd_shape_prev_bk, - pChVars[ch]->wnd_shape_this_bk, - qFormatNorm, - pChVars[ch]->abs_max_per_window, - pVars->scratch.fft, - &pExt->pOutputBuffer[ch]); - /* - * Update LTP buffers only if needed - */ - - if (pVars->mc_info.audioObjectType == MP4AUDIO_LTP) - { - Int16 * pt = &pExt->pOutputBuffer[ch]; - Int16 * ptr = &(pChVars[ch]->ltp_buffer[pVars->ltp_buffer_state]); - Int16 x, y; - for (Int32 i = HALF_LONG_WINDOW; i != 0; i--) - { - x = *pt; - pt += 2; - y = *pt; - pt += 2; - *(ptr++) = x; - *(ptr++) = y; - } - - } - - -#endif - - - /* Update the window shape */ - pChVars[ch]->wnd_shape_prev_bk = pChVars[ch]->wnd_shape_this_bk; - - } /* end for() */ - - - /* - * Copy to the final output buffer, taking into account the desired - * channels from the calling environment, the actual channels, and - * whether the data should be interleaved or not. - * - * If the stream had only one channel, write_output will not use - * the right channel data. - * - */ - - - /* CONSIDER USE OF DMA OPTIMIZATIONS WITHIN THE write_output FUNCTION. - * - * It is presumed that the ltp_buffer will reside in internal (fast) - * memory, while the pExt->pOutputBuffer will reside in external - * (slow) memory. - * - */ - - -#ifdef AAC_PLUS - - if (sbrBitStream->NrElements || pMC_Info->upsamplingFactor == 2) - { - - if (pVars->bno <= 1) /* allows console to operate with ADIF and audio config */ - { - if (sbrDec->outSampleRate == 0) /* do it only once (disregarding of signaling type) */ - { - sbr_open(samp_rate_info[pVars->mc_info.sampling_rate_idx].samp_rate, - sbrDec, - sbrDecoderData, - pVars->mc_info.bDownSampledSbr); - } - - } - pMC_Info->upsamplingFactor = - sbrDecoderData->SbrChannel[0].frameData.sbr_header.sampleRateMode; - - - /* reuse right aac spectrum channel */ - { - Int16 *pt_left = &(pChVars[LEFT ]->ltp_buffer[pVars->ltp_buffer_state]); - Int16 *pt_right = &(pChVars[RIGHT]->ltp_buffer[pVars->ltp_buffer_state]); - - if (sbr_applied(sbrDecoderData, - sbrBitStream, - pt_left, - pt_right, - pExt->pOutputBuffer, - sbrDec, - pVars, - pMC_Info->nch) != SBRDEC_OK) - { - status = MP4AUDEC_INVALID_FRAME; - } - } - - - } /* if( pExt->aacPlusEnabled == FALSE) */ -#endif - - /* - * Copied mono data in both channels or just leave it as mono, - * according with desiredChannels (default is 2) - */ - - if (pExt->desiredChannels == 2) - { - -#if defined(AAC_PLUS) -#if defined(PARAMETRICSTEREO)&&defined(HQ_SBR) - if (pMC_Info->nch != 2 && pMC_Info->psPresentFlag != 1) -#else - if (pMC_Info->nch != 2) -#endif -#else - if (pMC_Info->nch != 2) -#endif - { - /* mono */ - - - Int16 * pt = &pExt->pOutputBuffer[0]; - Int16 * pt2 = &pExt->pOutputBuffer[1]; - Int i; - if (pMC_Info->upsamplingFactor == 2) - { - for (i = 0; i < 1024; i++) - { - *pt2 = *pt; - pt += 2; - pt2 += 2; - } - pt = &pExt->pOutputBuffer_plus[0]; - pt2 = &pExt->pOutputBuffer_plus[1]; - - for (i = 0; i < 1024; i++) - { - *pt2 = *pt; - pt += 2; - pt2 += 2; - } - } - else - { - for (i = 0; i < 1024; i++) - { - *pt2 = *pt; - pt += 2; - pt2 += 2; - } - } - - } - -#if defined(AAC_PLUS) -#if defined(PARAMETRICSTEREO)&&defined(HQ_SBR) - - else if (pMC_Info->psPresentFlag == 1) - { - Int32 frameSize = 0; - if (pExt->aacPlusEnabled == false) - { - /* - * Decoding eaac+ when only aac is enabled, copy L into R - */ - frameSize = 1024; - } - else if (sbrDecoderData->SbrChannel[0].syncState != SBR_ACTIVE) - { - /* - * Decoding eaac+ when no PS data was found, copy upsampled L into R - */ - frameSize = 2048; - } - - Int16 * pt = &pExt->pOutputBuffer[0]; - Int16 * pt2 = &pExt->pOutputBuffer[1]; - Int i; - for (i = 0; i < frameSize; i++) - { - *pt2 = *pt; - pt += 2; - pt2 += 2; - } - } -#endif -#endif - - } - else - { - -#if defined(AAC_PLUS) -#if defined(PARAMETRICSTEREO)&&defined(HQ_SBR) - if (pMC_Info->nch != 2 && pMC_Info->psPresentFlag != 1) -#else - if (pMC_Info->nch != 2) -#endif -#else - if (pMC_Info->nch != 2) -#endif - { - /* mono */ - Int16 * pt = &pExt->pOutputBuffer[0]; - Int16 * pt2 = &pExt->pOutputBuffer[0]; - Int i; - - if (pMC_Info->upsamplingFactor == 2) - { - for (i = 0; i < 1024; i++) - { - *pt2++ = *pt; - pt += 2; - } - - pt = &pExt->pOutputBuffer_plus[0]; - pt2 = &pExt->pOutputBuffer_plus[0]; - - for (i = 0; i < 1024; i++) - { - *pt2++ = *pt; - pt += 2; - } - } - else - { - for (i = 0; i < 1024; i++) - { - *pt2++ = *pt; - pt += 2; - } - } - - } - - } - - - - - /* pVars->ltp_buffer_state cycles between 0 and 1024. The value - * indicates the location of the data corresponding to t == -2. - * - * | t == -2 | t == -1 | pVars->ltp_buffer_state == 0 - * - * | t == -1 | t == -2 | pVars->ltp_buffer_state == 1024 - * - */ - -#ifdef AAC_PLUS - if (sbrBitStream->NrElements == 0 && pMC_Info->upsamplingFactor == 1) - { - pVars->ltp_buffer_state ^= frameLength; - } - else - { - pVars->ltp_buffer_state ^= (frameLength + 288); - } -#else - pVars->ltp_buffer_state ^= frameLength; -#endif - - - if (pVars->bno <= 1) - { - /* - * to set these values only during the second call - * when they change. - */ - pExt->samplingRate = - samp_rate_info[pVars->mc_info.sampling_rate_idx].samp_rate; - - pVars->mc_info.implicit_channeling = 0; /* disable flag, as this is allowed - * only the first time - */ - - -#ifdef AAC_PLUS - - if (pMC_Info->upsamplingFactor == 2) - { - pExt->samplingRate *= pMC_Info->upsamplingFactor; - pExt->aacPlusUpsamplingFactor = pMC_Info->upsamplingFactor; - } - -#endif - - pExt->extendedAudioObjectType = pMC_Info->ExtendedAudioObjectType; - pExt->audioObjectType = pMC_Info->audioObjectType; - - pExt->encodedChannels = pMC_Info->nch; - pExt->frameLength = pVars->frameLength; - } - - pVars->bno++; - - - /* - * Using unit analysis, the bitrate is a function of the sampling rate, bits, - * points in a frame - * - * bits samples frame - * ---- = --------- * bits * ------- - * sec sec sample - * - * To save a divide, a shift is used. Presently only the value of - * 1024 is used by this library, so make it the most accurate for that - * value. This may need to be updated later. - */ - - pExt->bitRate = (pExt->samplingRate * - (pVars->inputStream.usedBits - initialUsedBits)) >> 10; /* LONG_WINDOW 1024 */ - - pExt->bitRate >>= (pMC_Info->upsamplingFactor - 1); - - - } /* end if (status == SUCCESS) */ - - - if (status != MP4AUDEC_SUCCESS) - { - /* - * A non-SUCCESS decoding could be due to an error on the bitstream or - * an incomplete frame. As access to the bitstream beyond frame boundaries - * are not allowed, in those cases the bitstream reading routine return a 0 - * Zero values guarantees that the data structures are filled in with values - * that eventually will signal an error (like invalid parameters) or that allow - * completion of the parsing routine. Either way, the partial frame condition - * is verified at this time. - */ - if (pVars->prog_config.file_is_adts == TRUE) - { - status = MP4AUDEC_LOST_FRAME_SYNC; - pVars->prog_config.headerless_frames = 0; /* synchronization forced */ - } - else - { - /* - * Check if the decoding error was due to buffer overrun, if it was, - * update status - */ - if (pVars->inputStream.usedBits > pVars->inputStream.availableBits) - { - /* all bits were used but were not enough to complete decoding */ - pVars->inputStream.usedBits = pVars->inputStream.availableBits; - - status = MP4AUDEC_INCOMPLETE_FRAME; /* possible EOF or fractional frame */ - } - } - } - - /* - * Translate from units of bits back into units of words. - */ - - pExt->inputBufferUsedLength = - pVars->inputStream.usedBits >> INBUF_ARRAY_INDEX_SHIFT; - - pExt->remainderBits = (Int)(pVars->inputStream.usedBits & INBUF_BIT_MODULO_MASK); - - - - return (status); - -} /* PVMP4AudioDecoderDecodeFrame */ - |