Int get_prog_config(
tDec_Int_File *pVars,
ProgConfig *pScratchPCE)
{
Int i;
UInt tag;
Int numChars;
UInt temp;
Bool flag;
Int status = SUCCESS;
BITS *pInputStream = &(pVars->inputStream);
/*
* The tag is used at the very end to see if this PCE is
* the one to be used. Otherwise it does not need to be saved for the
* the simple configurations to be used in this version of an AAC
* decoder.
*
* All of the bits of this PCE must be read even if this PCE will not
* be used. They are read into a temporary PCE, then later it is decided
* whether to keep this PCE.
*
* To allow quick removal of the fields from the ProgConfig structure
* that will probably not be used at a later date,
* while still advancing the bitstream pointer,the return value of
* getbits is saved into a temporary variable, then transfered to
* the structure item.
*/
tag = get9_n_lessbits(LEN_TAG, pInputStream);
pScratchPCE->profile = get9_n_lessbits(LEN_PROFILE, pInputStream);
pScratchPCE->sampling_rate_idx = get9_n_lessbits(LEN_SAMP_IDX, pInputStream);
if (!pVars->adif_test && (pScratchPCE->sampling_rate_idx != pVars->prog_config.sampling_rate_idx))
{
#ifdef AAC_PLUS
/*
* PCE carries the baseline frequency, if SBR or PS are used, the frequencies will not match
* so check for this unique case, and let decoding continue if this is a redundant PCE
*/
if ((pScratchPCE->sampling_rate_idx != (pVars->prog_config.sampling_rate_idx + 3)) ||
(pVars->mc_info.upsamplingFactor != 2))
#endif
{
/* rewind the pointer as implicit channel configuration maybe the case */
pInputStream->usedBits -= (LEN_TAG + LEN_PROFILE + LEN_SAMP_IDX);
return (1); /* mismatch cannot happen */
}
}
/*
* Retrieve the number of element lists for each of
* front, side, back, lfe, data, and coupling.
*
* For two-channel stereo or mono, only the data in the front needs
* to be saved. However, ALL fields need to be skipped over in some
* fashion. Also, the number of elements needs to be temporarily saved
* to call get_ele_list(). If that function was changed to pass in
* the number of points to be read, the memory set aside inside the
* ProgConfig structure could be removed.
*/
/*
* The next six function calls could be combined into one, then use
* shifts and masks to retrieve the individual fields.
*/
temp = get9_n_lessbits(LEN_NUM_ELE, pInputStream);
pScratchPCE->front.num_ele = temp;
/* Needed only to read in the element list. */
temp = get9_n_lessbits(LEN_NUM_ELE, pInputStream);
pScratchPCE->side.num_ele = temp;
/* Needed only to read in the element list. */
temp = get9_n_lessbits(LEN_NUM_ELE, pInputStream);
pScratchPCE->back.num_ele = temp;
/* Needed only to read in the element list. */
temp = get9_n_lessbits(LEN_NUM_LFE, pInputStream);
pScratchPCE->lfe.num_ele = temp;
/* Needed only to read in the element list. */
temp = get9_n_lessbits(LEN_NUM_DAT, pInputStream);
pScratchPCE->data.num_ele = temp;
/* Needed only to read in the element list. */
temp = get9_n_lessbits(LEN_NUM_CCE, pInputStream);
pScratchPCE->coupling.num_ele = temp;
/*
* Read in mix down data.
*
* Whether these fields can be removed and have proper operation
* will be determined at a later date.
*/
/* Read presence of mono_mix */
flag = get1bits(pInputStream);/* LEN_MIX_PRES,*/
pScratchPCE->mono_mix.present = flag;
if (flag != FALSE)
{
temp = get9_n_lessbits(LEN_TAG, pInputStream);
pScratchPCE->mono_mix.ele_tag = temp;
} /* end if (flag != FALSE) */
/* Read presence of stereo mix */
flag = get1bits(pInputStream); /* LEN_MIX_PRES,*/
pScratchPCE->stereo_mix.present = flag;
if (flag != FALSE)
{
temp = get9_n_lessbits(LEN_TAG, pInputStream);
pScratchPCE->stereo_mix.ele_tag = temp;
} /* end if (flag != FALSE) */
/* Read presence of matrix mix */
flag = get1bits(pInputStream); /* LEN_MIX_PRES,*/
pScratchPCE->matrix_mix.present = flag;
if (flag != FALSE)
{
temp = get9_n_lessbits(LEN_MMIX_IDX, pInputStream);
pScratchPCE->matrix_mix.ele_tag = temp;
temp = get1bits(pInputStream); /* LEN_PSUR_ENAB,*/
pScratchPCE->matrix_mix.pseudo_enab = temp;
} /* end if (flag != FALSE) */
/*
* Get each of the element lists. Only the front information will be
* used for the PV decoder, but the usedBits field of pInputStream must
* be advanced appropriately.
*
* This could be optimized by advancing the bit stream for the
* elements that do not need to be read.
*/
get_ele_list(&pScratchPCE->front,
pInputStream,
TRUE);
get_ele_list(&pScratchPCE->side,
pInputStream,
TRUE);
get_ele_list(&pScratchPCE->back,
pInputStream,
TRUE);
get_ele_list(&pScratchPCE->lfe,
pInputStream,
FALSE);
get_ele_list(&pScratchPCE->data,
pInputStream,
FALSE);
get_ele_list(&pScratchPCE->coupling,
pInputStream,
TRUE);
/*
* The standard requests a byte alignment before reading in the
* comment. This can be done because LEN_COMMENT_BYTES == 8.
*/
byte_align(pInputStream);
numChars = get9_n_lessbits(LEN_COMMENT_BYTES, pInputStream);
/*
* Ignore the comment - it requires 65 bytes to store (or worse on DSP).
* If this field is restored, make sure to append a trailing '\0'
*/
for (i = numChars; i > 0; i--)
{
pScratchPCE->comments[i] = (Char) get9_n_lessbits(LEN_BYTE,
pInputStream);
} /* end for */
if (pVars->current_program < 0)
{
/*
* If this is the first PCE, it becomes the current, regardless of
* its tag number.
*/
pVars->current_program = tag;
pVars->mc_info.ch_info[0].tag = 0;
} /* end if (pVars->current_program < 0) */
if (tag == (UInt)pVars->current_program)
{
/*
* This branch is reached under two conditions:
* 1) This is the first PCE found, it was selected in the above if
* block. In all encoders found thus far, the tag value has been
* zero.
* 2) A PCE has been sent by the encoder with a tag that matches the
* the first one sent. It will then be re-read. No encoder found
*
* Regardless, the temporary PCE will now be copied into the
* the one official program configuration.
*/
/*
* Keep adts setting in case of a redundant PCE (only applicable when
* using aac-lib own adts parser)
*/
pScratchPCE->file_is_adts = pVars->prog_config.file_is_adts;
pScratchPCE->headerless_frames = pVars->prog_config.headerless_frames;
pv_memcpy(&pVars->prog_config,
pScratchPCE,
sizeof(ProgConfig));
tag = 0;
/*
* Check that dual-mono does not carry more than 2 tracks, otherwise flag an non-supported error
*/
if ((pVars->prog_config.front.num_ele > 2) && !(pVars->prog_config.front.ele_is_cpe[tag]))
{
status = 1;
}
/*
* Check that stereo does not carry more than 1 track, otherwise flag an non-supported error
*/
if ((pVars->prog_config.front.num_ele > 1) && (pVars->prog_config.front.ele_is_cpe[tag]))
{
status = 1;
}
if (!status)
{
/* enter configuration into MC_Info structure */
status = set_mc_info(&pVars->mc_info,
(tMP4AudioObjectType)(pVars->prog_config.profile + 1),
pVars->prog_config.sampling_rate_idx,
pVars->prog_config.front.ele_tag[tag],
pVars->prog_config.front.ele_is_cpe[tag],
pVars->winmap,
pVars->SFBWidth128);
if (pVars->mc_info.upsamplingFactor == 2)
{
/*
* prog_config.sampling_rate_idx corresponds to the aac base layer,
* if the upsampling factor is active, then the output frequency needs
* to be adjusted accordingly
*/
pVars->prog_config.sampling_rate_idx -= 3;
}
}
} /* end if (tag == pVars->current_program) */
return (status);
}
/*----------------------------------------------------------------------------
; FUNCTION CODE
----------------------------------------------------------------------------*/
Int get_GA_specific_config(
tDec_Int_File * const pVars,
BITS *pInputStream,
UInt channel_config,
const tMP4AudioObjectType audioObjectType
)
{
Int status = SUCCESS;
UInt dependsOnCoreCoder;
/* Int coreCoderDelay; */
UInt extFlag;
/* These variables are left for future implementation */
/* UInt layer_num; */
/* UInt numOfSubFrame; */
/* UInt layer_len; */
/* UInt aacSectionDataResilienceFlag; */
/* UInt aacScalefactorDataResilienceFlag; */
/* UInt aacSpectralDataResilienceFlag; */
Int extFlag3;
/*
* frame length flag == 0, 1024 samples/frame
* frame length flag == 1, 960 samples/frame
*/
get1bits(/* LEN_FRAME_LEN_FLAG,*/
pInputStream);
/*
* dependsOnCoreCoder == 1, core coder has different sampling rate
* in a scalable bitstream
*/
dependsOnCoreCoder =
get1bits(/* LEN_DEPEND_ON_CORE,*/
pInputStream);
if (dependsOnCoreCoder != FALSE)
{
/*coreCoderDelay =
* getbits(
* LEN_CORE_DELAY,
* pInputStream);
*/
status = 1; /* do not support scalable coding in this release */
}
/*
* extension flag indicates if Amendment 1 objects are used or not
* extension flag == 0 objects = 1, 2, 3, 4, 6, 7
* extension flag == 1 objects = 17, 19, 20, 21, 22, 23
*/
extFlag = get1bits(pInputStream); /* LEN_EXT_FLAG,*/
/* Force checks for implicit channel configuration */
pVars->mc_info.implicit_channeling = 1;
if (status == SUCCESS)
{
if (channel_config == 0)
{
status = get_prog_config(pVars,
&pVars->scratch.scratch_prog_config);
if (status != SUCCESS)
{
pVars->prog_config.front.ele_is_cpe[0] = 0; /* default to mono */
pVars->mc_info.nch = 1;
pVars->prog_config.front.ele_tag[0] = 0;
status = SUCCESS;
}
}
else
{
/*
* dummy tag = 0 and
* set up decoding configurations
*/
channel_config--;
pVars->prog_config.front.ele_is_cpe[0] = channel_config;
pVars->prog_config.front.ele_tag[0] = 0;
status =
set_mc_info(
&(pVars->mc_info),
audioObjectType, /* previously profile */
pVars->prog_config.sampling_rate_idx,
pVars->prog_config.front.ele_tag[0],
pVars->prog_config.front.ele_is_cpe[0],
pVars->winmap, /*pVars->pWinSeqInfo,*/
pVars->SFBWidth128);
} /* if (channel_config) */
} /* if(status) */
/*
* This layer_num is not found in ISO/IEC specs,
* but it is defined in San Diego spec for scalable bitstream
*/
if ((audioObjectType == MP4AUDIO_AAC_SCALABLE) ||
(audioObjectType == MP4AUDIO_ER_AAC_SCALABLE))
{
/*layer_num =
* getbits(
* LEN_LAYER_NUM,
* pInputStream);
*/
status = 1; /* for this release only */
}
if (extFlag)
{
/*
* currently do not implement these functionalities
* defined in Amendment 1
* keep it here for future release
*/
if (audioObjectType == MP4AUDIO_ER_BSAC)
{
status = 1; /* NOT SUPPORTED */
/*
numOfSubFrame = getbits( LEN_SUB_FRAME, pInputStream);
layer_len = getbits( LEN_LAYER_LEN, pInputStream);
*/
}
/*
* The following code is equivalent to
* if ((audioObjectType == 17) || (audioObjectType == 18) ||
* (audioObjectType == 19) || (audioObjectType == 20) ||
* (audioObjectType == 21) || (audioObjectType == 23))
*/
if (((audioObjectType > 16) && (audioObjectType < 22)) ||
(audioObjectType == 23))
{
status = 1; /* NOT SUPPORTED */
/*
aacSectionDataResilienceFlag = getbits( LEN_SECT_RES_FLAG,
pInputStream);
aacScalefactorDataResilienceFlag = getbits( LEN_SCF_RES_FLAG,
pInputStream);
aacSpectralDataResilienceFlag = getbits( LEN_SPEC_RES_FLAG,
pInputStream);
*/
}
/*
* this flag is tbd in version 3 of ISO/IEC spec
* if the encoder generates this bit, then it has to be read
* current adif2mp4ff does not write this bit. If this bit is to
* be read, it can be done by the following code:
*/
extFlag3 = get1bits(pInputStream); /* LEN_EXT_FLAG3 */
if (extFlag3)
{
status = 1; /* NOT SUPPORTED */
}
}
return status;
}
Int PVMP4SetAudioConfig(
tPVMP4AudioDecoderExternal *pExt,
void *pMem,
Int upsamplingFactor,
Int samp_rate,
Int num_ch,
tMP4AudioObjectType audioObjectType)
{
tDec_Int_File *pVars; /* Helper pointer */
Int status = MP4AUDEC_INCOMPLETE_FRAME;
/*
* Initialize "helper" pointers to existing memory.
*/
pVars = (tDec_Int_File *)pMem;
/*
* Translate input buffer variables.
*/
pVars->inputStream.pBuffer = pExt->pInputBuffer;
pVars->inputStream.availableBits = 0;
pVars->inputStream.usedBits = 0;
/*
* get sampling rate index
*/
switch (samp_rate)
{
case 96000:
pVars->prog_config.sampling_rate_idx = 0;
break;
case 88200:
pVars->prog_config.sampling_rate_idx = 1;
break;
case 64000:
pVars->prog_config.sampling_rate_idx = 2;
break;
case 48000:
pVars->prog_config.sampling_rate_idx = 3;
break;
case 44100:
pVars->prog_config.sampling_rate_idx = 4;
break;
case 32000:
pVars->prog_config.sampling_rate_idx = 5;
break;
case 24000:
pVars->prog_config.sampling_rate_idx = 6;
break;
case 22050:
pVars->prog_config.sampling_rate_idx = 7;
break;
case 16000:
pVars->prog_config.sampling_rate_idx = 8;
break;
case 12000:
pVars->prog_config.sampling_rate_idx = 9;
break;
case 11025:
pVars->prog_config.sampling_rate_idx = 10;
break;
case 8000:
pVars->prog_config.sampling_rate_idx = 11;
break;
case 7350:
pVars->prog_config.sampling_rate_idx = 12;
break;
default:
status = -1;
break;
}
pVars->mc_info.sbrPresentFlag = 0;
pVars->mc_info.psPresentFlag = 0;
#ifdef AAC_PLUS
pVars->mc_info.bDownSampledSbr = 0;
#endif
pVars->mc_info.implicit_channeling = 0;
pVars->mc_info.nch = num_ch;
pVars->mc_info.upsamplingFactor = upsamplingFactor;
/*
* Set number of channels
*/
if (num_ch == 2)
{
pVars->prog_config.front.ele_is_cpe[0] = 1;
}
else if (num_ch == 1)
{
pVars->prog_config.front.ele_is_cpe[0] = 0;
}
else
{
status = -1; /* do not support more than two channels */
pVars->status = status;
return (status);
}
/*
* Set AAC bitstream
*/
if ((audioObjectType == MP4AUDIO_AAC_LC) ||
(audioObjectType == MP4AUDIO_LTP))
{
pVars->aacPlusEnabled = false;
status = set_mc_info(&(pVars->mc_info),
audioObjectType, /* previously profile */
pVars->prog_config.sampling_rate_idx,
pVars->prog_config.front.ele_tag[0],
pVars->prog_config.front.ele_is_cpe[0],
pVars->winmap, /*pVars->pWinSeqInfo,*/
pVars->SFBWidth128);
}
else if ((audioObjectType == MP4AUDIO_SBR) ||
(audioObjectType == MP4AUDIO_PS))
{
pVars->aacPlusEnabled = true;
status = set_mc_info(&(pVars->mc_info),
MP4AUDIO_AAC_LC,
pVars->prog_config.sampling_rate_idx,
pVars->prog_config.front.ele_tag[0],
pVars->prog_config.front.ele_is_cpe[0],
pVars->winmap, /*pVars->pWinSeqInfo,*/
pVars->SFBWidth128);
pVars->mc_info.sbrPresentFlag = 1;
if (audioObjectType == MP4AUDIO_PS)
{
pVars->mc_info.psPresentFlag = 1;
}
if (upsamplingFactor == 1)
{
#ifdef AAC_PLUS
pVars->mc_info.bDownSampledSbr = 1;
#endif
/*
* Disable SBR decoding for any sbr-downsampled file whose SF is >= 24 KHz
*/
if (pVars->prog_config.sampling_rate_idx < 6)
{
pVars->aacPlusEnabled = false;
}
}
}
else
{
status = -1;
}
/*
* Translate from units of bits back into units of words.
*/
pExt->inputBufferUsedLength = 0;
pExt->remainderBits = 0;
pVars->bno++;
pExt->samplingRate = samp_rate * upsamplingFactor;
pExt->aacPlusEnabled = pVars->aacPlusEnabled;
/*
* we default to 2 channel, even for mono files, (where channels have same content)
* this is done to ensure support for enhanced aac+ with implicit signalling
*/
pExt->encodedChannels = 2;
pExt->frameLength = 1024;
#ifdef AAC_PLUS
pExt->aacPlusUpsamplingFactor = upsamplingFactor;
#endif
pVars->status = status;
return (status);
} /* PVMP4AudioDecoderDecodeFrame */
