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Diffstat (limited to 'media/libstagefright/codecs/aacdec/pvmp4audiodecoderframe.cpp')
| -rw-r--r-- | media/libstagefright/codecs/aacdec/pvmp4audiodecoderframe.cpp | 1458 |
1 files changed, 1458 insertions, 0 deletions
diff --git a/media/libstagefright/codecs/aacdec/pvmp4audiodecoderframe.cpp b/media/libstagefright/codecs/aacdec/pvmp4audiodecoderframe.cpp new file mode 100644 index 0000000..7a279dc --- /dev/null +++ b/media/libstagefright/codecs/aacdec/pvmp4audiodecoderframe.cpp @@ -0,0 +1,1458 @@ +/* ------------------------------------------------------------------ + * 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 */ + |