AAC Decoder: introduce time domain limiter
Introduce time domain limiter. The module is per default enabled for all AAC-LC and HE-AAC v1/2 streams. For all ER-AAC-LD and ER-AAC-ELD streams the limiter is disabled per default. The feature can be en- or disabled via dynamic API parameter. Note that the limiter introduces an additional output delay which depends on the module parameters and the streams sampling rate. Bug 9428126 Change-Id: I299a072340b33e2c324facbd347a72c8de3d380e
This commit is contained in:
Jean-Michel Trivi
parent
629f60c0e7
commit
603f48ab99
File diff suppressed because it is too large
Load Diff
@ -436,6 +436,16 @@ typedef enum
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2: Create a dual mono output signal from channel 2. \n
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3: Create a dual mono output signal by mixing both channels (L' = R' = 0.5*Ch1 + 0.5*Ch2). */
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AAC_PCM_OUTPUT_CHANNEL_MAPPING = 0x0003, /*!< Output buffer channel ordering. 0: MPEG PCE style order, 1: WAV file channel order (default). */
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AAC_PCM_LIMITER_ENABLE = 0x0004, /*!< Enable signal level limiting. \n
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-1: Auto-config. Enable limiter for all non-lowdelay configurations by default. \n
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0: Disable limiter in general. \n
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1: Enable limiter always.
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It is recommended to call the decoder with a AACDEC_CLRHIST flag to reset all states when
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the limiter switch is changed explicitly. */
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AAC_PCM_LIMITER_ATTACK_TIME = 0x0005, /*!< Signal level limiting attack time in ms.
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Default confguration is 15 ms. Adjustable range from 1 ms to 15 ms. */
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AAC_PCM_LIMITER_RELEAS_TIME = 0x0006, /*!< Signal level limiting release time in ms.
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Default configuration is 50 ms. Adjustable time must be larger than 0 ms. */
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AAC_PCM_MIN_OUTPUT_CHANNELS = 0x0011, /*!< Minimum number of PCM output channels. If higher than the number of encoded audio channels,
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a simple channel extension is applied. \n
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-1, 0: Disable channel extenstion feature. The decoder output contains the same number of
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@ -130,7 +130,6 @@ void aacDecoder_drcInit (
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/* init control fields */
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self->enable = 0;
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self->numThreads = 0;
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self->digitalNorm = 0;
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/* init params */
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pParams = &self->params;
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@ -139,8 +138,9 @@ void aacDecoder_drcInit (
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pParams->usrCut = FL2FXCONST_DBL(0.0f);
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pParams->boost = FL2FXCONST_DBL(0.0f);
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pParams->usrBoost = FL2FXCONST_DBL(0.0f);
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pParams->targetRefLevel = AACDEC_DRC_DEFAULT_REF_LEVEL;
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pParams->targetRefLevel = -1;
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pParams->expiryFrame = AACDEC_DRC_DFLT_EXPIRY_FRAMES;
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pParams->applyDigitalNorm = 0;
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pParams->applyHeavyCompression = 0;
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/* initial program ref level = target ref level */
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@ -222,11 +222,12 @@ AAC_DECODER_ERROR aacDecoder_drcSetParam (
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return AAC_DEC_INVALID_HANDLE;
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}
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if (value < 0) {
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self->digitalNorm = 0;
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self->params.applyDigitalNorm = 0;
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self->params.targetRefLevel = -1;
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}
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else {
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/* ref_level must be between 0 and MAX_REFERENCE_LEVEL, inclusive */
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self->digitalNorm = 1;
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self->params.applyDigitalNorm = 1;
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if (self->params.targetRefLevel != (SCHAR)value) {
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self->params.targetRefLevel = (SCHAR)value;
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self->progRefLevel = (SCHAR)value; /* Always set the program reference level equal to the
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@ -234,6 +235,16 @@ AAC_DECODER_ERROR aacDecoder_drcSetParam (
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}
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}
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break;
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case APPLY_NORMALIZATION:
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if (value < 0 || value > 1) {
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return AAC_DEC_SET_PARAM_FAIL;
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}
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if (self == NULL) {
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return AAC_DEC_INVALID_HANDLE;
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}
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/* Store new parameter value */
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self->params.applyDigitalNorm = (UCHAR)value;
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break;
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case APPLY_HEAVY_COMPRESSION:
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if (value < 0 || value > 1) {
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return AAC_DEC_SET_PARAM_FAIL;
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@ -278,7 +289,7 @@ AAC_DECODER_ERROR aacDecoder_drcSetParam (
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self->enable = ( (self->params.boost > (FIXP_DBL)0)
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|| (self->params.cut > (FIXP_DBL)0)
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|| (self->params.applyHeavyCompression != 0)
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|| (self->digitalNorm == 1) );
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|| (self->params.targetRefLevel >= 0) );
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return ErrorStatus;
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@ -827,6 +838,7 @@ void aacDecoder_drcApply (
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void *pSbrDec,
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CAacDecoderChannelInfo *pAacDecoderChannelInfo,
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CDrcChannelData *pDrcChData,
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FIXP_DBL *extGain,
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int ch, /* needed only for SBR */
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int aacFrameSize,
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int bSbrPresent )
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@ -838,8 +850,8 @@ void aacDecoder_drcApply (
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FIXP_DBL max_mantissa;
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INT max_exponent;
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FIXP_DBL norm_mantissa = FL2FXCONST_DBL(0.0f);
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INT norm_exponent = 0;
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FIXP_DBL norm_mantissa = FL2FXCONST_DBL(0.5f);
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INT norm_exponent = 1;
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FIXP_DBL fact_mantissa[MAX_DRC_BANDS];
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INT fact_exponent[MAX_DRC_BANDS];
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@ -861,6 +873,15 @@ void aacDecoder_drcApply (
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if (!self->enable) {
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sbrDecoder_drcDisable( (HANDLE_SBRDECODER)pSbrDec, ch );
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if (extGain != NULL) {
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INT gainScale = (INT)*extGain;
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/* The gain scaling must be passed to the function in the buffer pointed on by extGain. */
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if (gainScale >= 0 && gainScale <= DFRACT_BITS) {
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*extGain = scaleValue(norm_mantissa, norm_exponent-gainScale);
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} else {
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FDK_ASSERT(0);
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}
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}
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return;
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}
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@ -876,7 +897,7 @@ void aacDecoder_drcApply (
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reduced DAC SNR (if signal is attenuated) or clipping (if signal is
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boosted) */
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if (self->digitalNorm == 1)
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if (pParams->targetRefLevel >= 0)
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{
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/* 0.5^((targetRefLevel - progRefLevel)/24) */
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norm_mantissa = fLdPow(
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@ -886,7 +907,18 @@ void aacDecoder_drcApply (
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3,
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&norm_exponent );
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}
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else {
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/* Always export the normalization gain (if possible). */
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if (extGain != NULL) {
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INT gainScale = (INT)*extGain;
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/* The gain scaling must be passed to the function in the buffer pointed on by extGain. */
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if (gainScale >= 0 && gainScale <= DFRACT_BITS) {
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*extGain = scaleValue(norm_mantissa, norm_exponent-gainScale);
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} else {
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FDK_ASSERT(0);
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}
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}
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if (self->params.applyDigitalNorm == 0) {
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/* Reset normalization gain since this module must not apply it */
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norm_mantissa = FL2FXCONST_DBL(0.5f);
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norm_exponent = 1;
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}
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@ -98,7 +98,6 @@ amm-info@iis.fraunhofer.de
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#include "channel.h"
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#include "FDK_bitstream.h"
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#define AACDEC_DRC_DEFAULT_REF_LEVEL ( 108 ) /* -27 dB below full scale (typical for movies) */
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#define AACDEC_DRC_DFLT_EXPIRY_FRAMES ( 50 ) /* Default DRC data expiry time in AAC frames */
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/**
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@ -111,6 +110,7 @@ typedef enum
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TARGET_REF_LEVEL,
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DRC_BS_DELAY,
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DRC_DATA_EXPIRY_FRAME,
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APPLY_NORMALIZATION,
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APPLY_HEAVY_COMPRESSION
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} AACDEC_DRC_PARAM;
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@ -149,6 +149,8 @@ int aacDecoder_drcProlog (
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* \param pSbrDec pointer to SBR decoder instance
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* \param pAacDecoderChannelInfo AAC decoder channel instance to be processed
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* \param pDrcDat DRC channel data
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* \param extGain Pointer to a FIXP_DBL where a externally applyable gain will be stored into (independently on whether it will be apply internally or not).
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* At function call the buffer must hold the scale (0 >= scale < DFRACT_BITS) to be applied on the gain value.
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* \param ch channel index
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* \param aacFrameSize AAC frame size
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* \param bSbrPresent flag indicating that SBR is present, in which case DRC is handed over to the SBR instance pSbrDec
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@ -158,6 +160,7 @@ void aacDecoder_drcApply (
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void *pSbrDec,
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CAacDecoderChannelInfo *pAacDecoderChannelInfo,
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CDrcChannelData *pDrcDat,
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FIXP_DBL *extGain,
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int ch,
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int aacFrameSize,
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int bSbrPresent );
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@ -140,6 +140,7 @@ typedef struct
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UINT expiryFrame;
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SCHAR targetRefLevel;
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UCHAR bsDelayEnable;
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UCHAR applyDigitalNorm;
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UCHAR applyHeavyCompression;
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} CDrcParams;
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@ -1653,6 +1653,8 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
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{
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int stride, offset, c;
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/* Turn on/off DRC modules level normalization in digital domain depending on the limiter status. */
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aacDecoder_drcSetParam( self->hDrcInfo, APPLY_NORMALIZATION, (self->limiterEnableCurr) ? 0 : 1 );
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/* Extract DRC control data and map it to channels (without bitstream delay) */
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aacDecoder_drcProlog (
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self->hDrcInfo,
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@ -1703,12 +1705,15 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
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/* Reset DRC control data for this channel */
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aacDecoder_drcInitChannelData ( &self->pAacDecoderStaticChannelInfo[c]->drcData );
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}
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/* The DRC module demands to be called with the gain field holding the gain scale. */
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self->extGain[0] = (FIXP_DBL)TDL_GAIN_SCALING;
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/* DRC processing */
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aacDecoder_drcApply (
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self->hDrcInfo,
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self->hSbrDecoder,
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pAacDecoderChannelInfo,
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&self->pAacDecoderStaticChannelInfo[c]->drcData,
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self->extGain,
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c,
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self->streamInfo.aacSamplesPerFrame,
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self->sbrEnabled
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@ -1726,6 +1731,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
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(self->frameOK && !(flags&AACDEC_CONCEAL)),
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self->aacCommonData.workBufferCore1->mdctOutTemp
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);
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self->extGainDelay = self->streamInfo.aacSamplesPerFrame;
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break;
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case AACDEC_RENDER_ELDFB:
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CBlock_FrequencyToTimeLowDelay(
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@ -1735,6 +1741,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
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self->streamInfo.aacSamplesPerFrame,
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stride
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);
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self->extGainDelay = (self->streamInfo.aacSamplesPerFrame*2 - self->streamInfo.aacSamplesPerFrame/2 - 1)/2;
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break;
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default:
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ErrorStatus = AAC_DEC_UNKNOWN;
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@ -111,6 +111,7 @@ amm-info@iis.fraunhofer.de
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#include "aacdec_drc.h"
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#include "pcmutils_lib.h"
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#include "limiter.h"
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/* Capabilities flags */
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@ -215,6 +216,12 @@ struct AAC_DECODER_INSTANCE {
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CAncData ancData; /*!< structure to handle ancillary data */
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HANDLE_PCM_DOWNMIX hPcmUtils; /*!< privat data for the PCM utils. */
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TDLimiterPtr hLimiter; /*!< Handle of time domain limiter. */
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UCHAR limiterEnableUser; /*!< The limiter configuration requested by the library user */
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UCHAR limiterEnableCurr; /*!< The current limiter configuration. */
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FIXP_DBL extGain[1]; /*!< Gain that must be applied to the output signal. */
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UINT extGainDelay; /*!< Delay that must be accounted for extGain. */
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};
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@ -110,7 +110,7 @@ amm-info@iis.fraunhofer.de
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/* Decoder library info */
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#define AACDECODER_LIB_VL0 2
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#define AACDECODER_LIB_VL1 5
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#define AACDECODER_LIB_VL2 6
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#define AACDECODER_LIB_VL2 7
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#define AACDECODER_LIB_TITLE "AAC Decoder Lib"
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#define AACDECODER_LIB_BUILD_DATE __DATE__
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#define AACDECODER_LIB_BUILD_TIME __TIME__
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@ -397,12 +397,14 @@ aacDecoder_SetParam ( const HANDLE_AACDECODER self, /*!< Handle of the decode
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CConcealParams *pConcealData = NULL;
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HANDLE_AAC_DRC hDrcInfo = NULL;
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HANDLE_PCM_DOWNMIX hPcmDmx = NULL;
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TDLimiterPtr hPcmTdl = NULL;
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/* check decoder handle */
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if (self != NULL) {
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pConcealData = &self->concealCommonData;
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hDrcInfo = self->hDrcInfo;
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hPcmDmx = self->hPcmUtils;
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hPcmTdl = self->hLimiter;
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} else {
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errorStatus = AAC_DEC_INVALID_HANDLE;
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}
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@ -486,6 +488,47 @@ aacDecoder_SetParam ( const HANDLE_AACDECODER self, /*!< Handle of the decode
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}
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break;
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case AAC_PCM_LIMITER_ENABLE:
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if (value < -1 || value > 1) {
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return AAC_DEC_SET_PARAM_FAIL;
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}
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if (self == NULL) {
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return AAC_DEC_INVALID_HANDLE;
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}
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self->limiterEnableUser = value;
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break;
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case AAC_PCM_LIMITER_ATTACK_TIME:
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if (value <= 0) { /* module function converts value to unsigned */
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return AAC_DEC_SET_PARAM_FAIL;
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}
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switch (setLimiterAttack(hPcmTdl, value)) {
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case TDLIMIT_OK:
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break;
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case TDLIMIT_INVALID_HANDLE:
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return AAC_DEC_INVALID_HANDLE;
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case TDLIMIT_INVALID_PARAMETER:
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default:
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return AAC_DEC_SET_PARAM_FAIL;
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}
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break;
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case AAC_PCM_LIMITER_RELEAS_TIME:
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if (value <= 0) { /* module function converts value to unsigned */
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return AAC_DEC_SET_PARAM_FAIL;
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}
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switch (setLimiterRelease(hPcmTdl, value)) {
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case TDLIMIT_OK:
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break;
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case TDLIMIT_INVALID_HANDLE:
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return AAC_DEC_INVALID_HANDLE;
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case TDLIMIT_INVALID_PARAMETER:
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default:
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return AAC_DEC_SET_PARAM_FAIL;
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}
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break;
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case AAC_PCM_OUTPUT_CHANNEL_MAPPING:
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switch (value) {
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case 0:
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@ -632,6 +675,14 @@ LINKSPEC_CPP HANDLE_AACDECODER aacDecoder_Open(TRANSPORT_TYPE transportFmt, UINT
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goto bail;
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}
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aacDec->hLimiter = createLimiter(TDL_ATTACK_DEFAULT_MS, TDL_RELEASE_DEFAULT_MS, SAMPLE_MAX, (8), 96000);
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if (NULL == aacDec->hLimiter) {
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err = -1;
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goto bail;
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}
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aacDec->limiterEnableUser = (UCHAR)-1;
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aacDec->limiterEnableCurr = 0;
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/* Assure that all modules have same delay */
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@ -807,6 +858,17 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
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self->streamInfo.numTotalBytes = 0;
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}
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if (self->limiterEnableUser==(UCHAR)-1) {
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/* Enbale limiter for all non-lowdelay AOT's. */
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self->limiterEnableCurr = ( self->flags & (AC_LD|AC_ELD) ) ? 0 : 1;
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}
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else {
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/* Use limiter configuration as requested. */
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self->limiterEnableCurr = self->limiterEnableUser;
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}
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/* reset limiter gain on a per frame basis */
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self->extGain[0] = FL2FXCONST_DBL(1.0f/(float)(1<<TDL_GAIN_SCALING));
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ErrorStatus = CAacDecoder_DecodeFrame(self,
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flags | (fTpConceal ? AACDEC_CONCEAL : 0),
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@ -909,6 +971,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
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{
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INT pcmLimiterScale = 0;
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PCMDMX_ERROR dmxErr = PCMDMX_OK;
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if ( flags & (AACDEC_INTR | AACDEC_CLRHIST) ) {
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/* delete data from the past (e.g. mixdown coeficients) */
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@ -924,13 +987,34 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
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self->channelType,
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self->channelIndices,
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self->channelOutputMapping,
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NULL
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(self->limiterEnableCurr) ? &pcmLimiterScale : NULL
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);
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if (dmxErr == PCMDMX_INVALID_MODE) {
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/* Announce the framework that the current combination of channel configuration and downmix
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* settings are not know to produce a predictable behavior and thus maybe produce strange output. */
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ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR;
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}
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if ( flags & AACDEC_CLRHIST ) {
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/* Delete the delayed signal. */
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resetLimiter(self->hLimiter);
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}
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if (self->limiterEnableCurr)
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{
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/* Set actual signal parameters */
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setLimiterNChannels(self->hLimiter, self->streamInfo.numChannels);
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setLimiterSampleRate(self->hLimiter, self->streamInfo.sampleRate);
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applyLimiter(
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self->hLimiter,
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pTimeData,
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self->extGain,
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&pcmLimiterScale,
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1,
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self->extGainDelay,
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self->streamInfo.frameSize
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);
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}
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}
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@ -956,6 +1040,9 @@ LINKSPEC_CPP void aacDecoder_Close ( HANDLE_AACDECODER self )
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return;
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if (self->hLimiter != NULL) {
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destroyLimiter(self->hLimiter);
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}
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if (self->hPcmUtils != NULL) {
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pcmDmx_Close( &self->hPcmUtils );
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233
libPCMutils/include/limiter.h
Normal file
233
libPCMutils/include/limiter.h
Normal file
@ -0,0 +1,233 @@
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/* -----------------------------------------------------------------------------------------------------------
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Software License for The Fraunhofer FDK AAC Codec Library for Android
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|
||||
© Copyright 1995 - 2013 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
|
||||
----------------------------------------------------------------------------------------------------------- */
|
||||
|
||||
/************************ FDK PCM postprocessor module *********************
|
||||
|
||||
Author(s): Matthias Neusinger
|
||||
Description: Hard limiter for clipping prevention
|
||||
|
||||
*******************************************************************************/
|
||||
|
||||
#ifndef _LIMITER_H_
|
||||
#define _LIMITER_H_
|
||||
|
||||
|
||||
#include "common_fix.h"
|
||||
|
||||
#define TDL_ATTACK_DEFAULT_MS (15) /* default attack time in ms */
|
||||
#define TDL_RELEASE_DEFAULT_MS (50) /* default release time in ms */
|
||||
|
||||
#define TDL_GAIN_SCALING (15) /* scaling of gain value. */
|
||||
|
||||
|
||||
#ifdef __cplusplus
|
||||
extern "C" {
|
||||
#endif
|
||||
|
||||
|
||||
typedef enum {
|
||||
TDLIMIT_OK = 0,
|
||||
|
||||
__error_codes_start = -100,
|
||||
|
||||
TDLIMIT_INVALID_HANDLE,
|
||||
TDLIMIT_INVALID_PARAMETER,
|
||||
|
||||
__error_codes_end
|
||||
} TDLIMITER_ERROR;
|
||||
|
||||
struct TDLimiter;
|
||||
typedef struct TDLimiter* TDLimiterPtr;
|
||||
|
||||
/******************************************************************************
|
||||
* createLimiter *
|
||||
* maxAttackMs: maximum and initial attack/lookahead time in milliseconds *
|
||||
* releaseMs: release time in milliseconds (90% time constant) *
|
||||
* threshold: limiting threshold *
|
||||
* maxChannels: maximum and initial number of channels *
|
||||
* maxSampleRate: maximum and initial sampling rate in Hz *
|
||||
* returns: limiter handle *
|
||||
******************************************************************************/
|
||||
TDLimiterPtr createLimiter(unsigned int maxAttackMs,
|
||||
unsigned int releaseMs,
|
||||
INT_PCM threshold,
|
||||
unsigned int maxChannels,
|
||||
unsigned int maxSampleRate);
|
||||
|
||||
|
||||
/******************************************************************************
|
||||
* resetLimiter *
|
||||
* limiter: limiter handle *
|
||||
* returns: error code *
|
||||
******************************************************************************/
|
||||
TDLIMITER_ERROR resetLimiter(TDLimiterPtr limiter);
|
||||
|
||||
|
||||
/******************************************************************************
|
||||
* destroyLimiter *
|
||||
* limiter: limiter handle *
|
||||
* returns: error code *
|
||||
******************************************************************************/
|
||||
TDLIMITER_ERROR destroyLimiter(TDLimiterPtr limiter);
|
||||
|
||||
/******************************************************************************
|
||||
* applyLimiter *
|
||||
* limiter: limiter handle *
|
||||
* pGain : pointer to gains to be applied to the signal before limiting, *
|
||||
* which are downscaled by TDL_GAIN_SCALING bit. *
|
||||
* These gains are delayed by gain_delay, and smoothed. *
|
||||
* Smoothing is done by a butterworth lowpass filter with a cutoff *
|
||||
* frequency which is fixed with respect to the sampling rate. *
|
||||
* It is a substitute for the smoothing due to windowing and *
|
||||
* overlap/add, if a gain is applied in frequency domain. *
|
||||
* gain_scale: pointer to scaling exponents to be applied to the signal before *
|
||||
* limiting, without delay and without smoothing *
|
||||
* gain_size: number of elements in pGain, currently restricted to 1 *
|
||||
* gain_delay: delay [samples] with which the gains in pGain shall be applied *
|
||||
* gain_delay <= nSamples *
|
||||
* samples: input/output buffer containing interleaved samples *
|
||||
* precision of output will be DFRACT_BITS-TDL_GAIN_SCALING bits *
|
||||
* nSamples: number of samples per channel *
|
||||
* returns: error code *
|
||||
******************************************************************************/
|
||||
TDLIMITER_ERROR applyLimiter(TDLimiterPtr limiter,
|
||||
INT_PCM* samples,
|
||||
FIXP_DBL* pGain,
|
||||
const INT* gain_scale,
|
||||
const UINT gain_size,
|
||||
const UINT gain_delay,
|
||||
const UINT nSamples);
|
||||
|
||||
/******************************************************************************
|
||||
* getLimiterDelay *
|
||||
* limiter: limiter handle *
|
||||
* returns: exact delay caused by the limiter in samples *
|
||||
******************************************************************************/
|
||||
unsigned int getLimiterDelay(TDLimiterPtr limiter);
|
||||
|
||||
/******************************************************************************
|
||||
* setLimiterNChannels *
|
||||
* limiter: limiter handle *
|
||||
* nChannels: number of channels ( <= maxChannels specified on create) *
|
||||
* returns: error code *
|
||||
******************************************************************************/
|
||||
TDLIMITER_ERROR setLimiterNChannels(TDLimiterPtr limiter, unsigned int nChannels);
|
||||
|
||||
/******************************************************************************
|
||||
* setLimiterSampleRate *
|
||||
* limiter: limiter handle *
|
||||
* sampleRate: sampling rate in Hz ( <= maxSampleRate specified on create) *
|
||||
* returns: error code *
|
||||
******************************************************************************/
|
||||
TDLIMITER_ERROR setLimiterSampleRate(TDLimiterPtr limiter, unsigned int sampleRate);
|
||||
|
||||
/******************************************************************************
|
||||
* setLimiterAttack *
|
||||
* limiter: limiter handle *
|
||||
* attackMs: attack time in ms ( <= maxAttackMs specified on create) *
|
||||
* returns: error code *
|
||||
******************************************************************************/
|
||||
TDLIMITER_ERROR setLimiterAttack(TDLimiterPtr limiter, unsigned int attackMs);
|
||||
|
||||
/******************************************************************************
|
||||
* setLimiterRelease *
|
||||
* limiter: limiter handle *
|
||||
* releaseMs: release time in ms *
|
||||
* returns: error code *
|
||||
******************************************************************************/
|
||||
TDLIMITER_ERROR setLimiterRelease(TDLimiterPtr limiter, unsigned int releaseMs);
|
||||
|
||||
/******************************************************************************
|
||||
* setLimiterThreshold *
|
||||
* limiter: limiter handle *
|
||||
* threshold: limiter threshold *
|
||||
* returns: error code *
|
||||
******************************************************************************/
|
||||
TDLIMITER_ERROR setLimiterThreshold(TDLimiterPtr limiter, INT_PCM threshold);
|
||||
|
||||
#ifdef __cplusplus
|
||||
}
|
||||
#endif
|
||||
|
||||
|
||||
#endif //#ifndef _LIMITER_H_
|
||||
498
libPCMutils/src/limiter.cpp
Normal file
498
libPCMutils/src/limiter.cpp
Normal file
@ -0,0 +1,498 @@
|
||||
|
||||
/* -----------------------------------------------------------------------------------------------------------
|
||||
Software License for The Fraunhofer FDK AAC Codec Library for Android
|
||||
|
||||
© Copyright 1995 - 2013 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
|
||||
----------------------------------------------------------------------------------------------------------- */
|
||||
|
||||
/************************ FDK PCM postprocessor module *********************
|
||||
|
||||
Author(s): Matthias Neusinger
|
||||
Description: Hard limiter for clipping prevention
|
||||
|
||||
*******************************************************************************/
|
||||
|
||||
#include "limiter.h"
|
||||
|
||||
|
||||
struct TDLimiter {
|
||||
unsigned int attack;
|
||||
FIXP_DBL attackConst, releaseConst;
|
||||
unsigned int attackMs, releaseMs, maxAttackMs;
|
||||
FIXP_PCM threshold;
|
||||
unsigned int channels, maxChannels;
|
||||
unsigned int sampleRate, maxSampleRate;
|
||||
FIXP_DBL cor, max;
|
||||
FIXP_DBL* maxBuf;
|
||||
FIXP_DBL* delayBuf;
|
||||
unsigned int maxBufIdx, delayBufIdx;
|
||||
FIXP_DBL smoothState0;
|
||||
FIXP_DBL minGain;
|
||||
|
||||
FIXP_DBL additionalGainPrev;
|
||||
FIXP_DBL additionalGainFilterState;
|
||||
FIXP_DBL additionalGainFilterState1;
|
||||
};
|
||||
|
||||
/* create limiter */
|
||||
TDLimiterPtr createLimiter(
|
||||
unsigned int maxAttackMs,
|
||||
unsigned int releaseMs,
|
||||
INT_PCM threshold,
|
||||
unsigned int maxChannels,
|
||||
unsigned int maxSampleRate
|
||||
)
|
||||
{
|
||||
TDLimiterPtr limiter = NULL;
|
||||
unsigned int attack, release;
|
||||
FIXP_DBL attackConst, releaseConst, exponent;
|
||||
INT e_ans;
|
||||
|
||||
/* calc attack and release time in samples */
|
||||
attack = (unsigned int)(maxAttackMs * maxSampleRate / 1000);
|
||||
release = (unsigned int)(releaseMs * maxSampleRate / 1000);
|
||||
|
||||
/* alloc limiter struct */
|
||||
limiter = (TDLimiterPtr)FDKcalloc(1, sizeof(struct TDLimiter));
|
||||
if (!limiter) return NULL;
|
||||
|
||||
/* alloc max and delay buffers */
|
||||
limiter->maxBuf = (FIXP_DBL*)FDKcalloc(attack + 1, sizeof(FIXP_DBL));
|
||||
limiter->delayBuf = (FIXP_DBL*)FDKcalloc(attack * maxChannels, sizeof(FIXP_DBL));
|
||||
|
||||
if (!limiter->maxBuf || !limiter->delayBuf) {
|
||||
destroyLimiter(limiter);
|
||||
return NULL;
|
||||
}
|
||||
|
||||
/* attackConst = pow(0.1, 1.0 / (attack + 1)) */
|
||||
exponent = invFixp(attack+1);
|
||||
attackConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
|
||||
attackConst = scaleValue(attackConst, e_ans);
|
||||
|
||||
/* releaseConst = (float)pow(0.1, 1.0 / (release + 1)) */
|
||||
exponent = invFixp(release + 1);
|
||||
releaseConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
|
||||
releaseConst = scaleValue(releaseConst, e_ans);
|
||||
|
||||
/* init parameters */
|
||||
limiter->attackMs = maxAttackMs;
|
||||
limiter->maxAttackMs = maxAttackMs;
|
||||
limiter->releaseMs = releaseMs;
|
||||
limiter->attack = attack;
|
||||
limiter->attackConst = attackConst;
|
||||
limiter->releaseConst = releaseConst;
|
||||
limiter->threshold = (FIXP_PCM)threshold;
|
||||
limiter->channels = maxChannels;
|
||||
limiter->maxChannels = maxChannels;
|
||||
limiter->sampleRate = maxSampleRate;
|
||||
limiter->maxSampleRate = maxSampleRate;
|
||||
|
||||
resetLimiter(limiter);
|
||||
|
||||
return limiter;
|
||||
}
|
||||
|
||||
|
||||
/* reset limiter */
|
||||
TDLIMITER_ERROR resetLimiter(TDLimiterPtr limiter)
|
||||
{
|
||||
if (limiter != NULL) {
|
||||
|
||||
limiter->maxBufIdx = 0;
|
||||
limiter->delayBufIdx = 0;
|
||||
limiter->max = (FIXP_DBL)0;
|
||||
limiter->cor = FL2FXCONST_DBL(1.0f/(1<<1));
|
||||
limiter->smoothState0 = FL2FXCONST_DBL(1.0f/(1<<1));
|
||||
limiter->minGain = FL2FXCONST_DBL(1.0f/(1<<1));
|
||||
|
||||
limiter->additionalGainPrev = FL2FXCONST_DBL(1.0f/(1<<TDL_GAIN_SCALING));
|
||||
limiter->additionalGainFilterState = FL2FXCONST_DBL(1.0f/(1<<TDL_GAIN_SCALING));
|
||||
limiter->additionalGainFilterState1 = FL2FXCONST_DBL(1.0f/(1<<TDL_GAIN_SCALING));
|
||||
|
||||
FDKmemset(limiter->maxBuf, 0, (limiter->attack + 1) * sizeof(FIXP_DBL) );
|
||||
FDKmemset(limiter->delayBuf, 0, limiter->attack * limiter->channels * sizeof(FIXP_DBL) );
|
||||
}
|
||||
else {
|
||||
return TDLIMIT_INVALID_HANDLE;
|
||||
}
|
||||
|
||||
return TDLIMIT_OK;
|
||||
}
|
||||
|
||||
|
||||
/* destroy limiter */
|
||||
TDLIMITER_ERROR destroyLimiter(TDLimiterPtr limiter)
|
||||
{
|
||||
if (limiter != NULL) {
|
||||
FDKfree(limiter->maxBuf);
|
||||
FDKfree(limiter->delayBuf);
|
||||
|
||||
FDKfree(limiter);
|
||||
}
|
||||
else {
|
||||
return TDLIMIT_INVALID_HANDLE;
|
||||
}
|
||||
return TDLIMIT_OK;
|
||||
}
|
||||
|
||||
/* apply limiter */
|
||||
TDLIMITER_ERROR applyLimiter(TDLimiterPtr limiter,
|
||||
INT_PCM* samples,
|
||||
FIXP_DBL* pGain,
|
||||
const INT* gain_scale,
|
||||
const UINT gain_size,
|
||||
const UINT gain_delay,
|
||||
const UINT nSamples)
|
||||
{
|
||||
unsigned int i, j;
|
||||
FIXP_PCM tmp1, tmp2;
|
||||
FIXP_DBL tmp, old, gain, additionalGain, additionalGainUnfiltered;
|
||||
FIXP_DBL minGain = FL2FXCONST_DBL(1.0f/(1<<1));
|
||||
|
||||
FDK_ASSERT(gain_size == 1);
|
||||
FDK_ASSERT(gain_delay <= nSamples);
|
||||
|
||||
if ( limiter == NULL ) return TDLIMIT_INVALID_HANDLE;
|
||||
|
||||
{
|
||||
unsigned int channels = limiter->channels;
|
||||
unsigned int attack = limiter->attack;
|
||||
FIXP_DBL attackConst = limiter->attackConst;
|
||||
FIXP_DBL releaseConst = limiter->releaseConst;
|
||||
FIXP_DBL threshold = FX_PCM2FX_DBL(limiter->threshold)>>TDL_GAIN_SCALING;
|
||||
|
||||
FIXP_DBL max = limiter->max;
|
||||
FIXP_DBL* maxBuf = limiter->maxBuf;
|
||||
unsigned int maxBufIdx = limiter->maxBufIdx;
|
||||
FIXP_DBL cor = limiter->cor;
|
||||
FIXP_DBL* delayBuf = limiter->delayBuf;
|
||||
unsigned int delayBufIdx = limiter->delayBufIdx;
|
||||
|
||||
FIXP_DBL smoothState0 = limiter->smoothState0;
|
||||
FIXP_DBL additionalGainSmoothState = limiter->additionalGainFilterState;
|
||||
FIXP_DBL additionalGainSmoothState1 = limiter->additionalGainFilterState1;
|
||||
|
||||
for (i = 0; i < nSamples; i++) {
|
||||
|
||||
if (i < gain_delay) {
|
||||
additionalGainUnfiltered = limiter->additionalGainPrev;
|
||||
} else {
|
||||
additionalGainUnfiltered = pGain[0];
|
||||
}
|
||||
|
||||
/* Smooth additionalGain */
|
||||
/* [b,a] = butter(1, 0.01) */
|
||||
static const FIXP_SGL b[] = { FL2FXCONST_SGL(0.015466*2.0), FL2FXCONST_SGL( 0.015466*2.0) };
|
||||
static const FIXP_SGL a[] = { FL2FXCONST_SGL(1.000000), FL2FXCONST_SGL(-0.96907) };
|
||||
/* [b,a] = butter(1, 0.001) */
|
||||
//static const FIXP_SGL b[] = { FL2FXCONST_SGL(0.0015683*2.0), FL2FXCONST_SGL( 0.0015683*2.0) };
|
||||
//static const FIXP_SGL a[] = { FL2FXCONST_SGL(1.0000000), FL2FXCONST_SGL(-0.99686) };
|
||||
additionalGain = - fMult(additionalGainSmoothState, a[1]) + fMultDiv2( additionalGainUnfiltered, b[0]) + fMultDiv2(additionalGainSmoothState1, b[1]);
|
||||
additionalGainSmoothState1 = additionalGainUnfiltered;
|
||||
additionalGainSmoothState = additionalGain;
|
||||
|
||||
/* Apply the additional scaling that has no delay and no smoothing */
|
||||
if (gain_scale[0] > 0) {
|
||||
additionalGain <<= gain_scale[0];
|
||||
} else {
|
||||
additionalGain >>= gain_scale[0];
|
||||
}
|
||||
|
||||
/* get maximum absolute sample value of all channels, including the additional gain. */
|
||||
tmp1 = (FIXP_PCM)0;
|
||||
for (j = 0; j < channels; j++) {
|
||||
tmp2 = (FIXP_PCM)samples[i * channels + j];
|
||||
if (tmp2 == (FIXP_PCM)SAMPLE_MIN) /* protect fAbs from -1.0 value */
|
||||
tmp2 = (FIXP_PCM)(SAMPLE_MIN+1);
|
||||
tmp1 = fMax(tmp1, fAbs(tmp2));
|
||||
}
|
||||
tmp = SATURATE_LEFT_SHIFT(fMultDiv2(tmp1, additionalGain), 1, DFRACT_BITS);
|
||||
|
||||
/* set threshold as lower border to save calculations in running maximum algorithm */
|
||||
tmp = fMax(tmp, threshold);
|
||||
|
||||
/* running maximum */
|
||||
old = maxBuf[maxBufIdx];
|
||||
maxBuf[maxBufIdx] = tmp;
|
||||
|
||||
if (tmp >= max) {
|
||||
/* new sample is greater than old maximum, so it is the new maximum */
|
||||
max = tmp;
|
||||
}
|
||||
else if (old < max) {
|
||||
/* maximum does not change, as the sample, which has left the window was
|
||||
not the maximum */
|
||||
}
|
||||
else {
|
||||
/* the old maximum has left the window, we have to search the complete
|
||||
buffer for the new max */
|
||||
max = maxBuf[0];
|
||||
for (j = 1; j <= attack; j++) {
|
||||
if (maxBuf[j] > max) max = maxBuf[j];
|
||||
}
|
||||
}
|
||||
maxBufIdx++;
|
||||
if (maxBufIdx >= attack+1) maxBufIdx = 0;
|
||||
|
||||
/* calc gain */
|
||||
/* gain is downscaled by one, so that gain = 1.0 can be represented */
|
||||
if (max > threshold) {
|
||||
gain = fDivNorm(threshold, max)>>1;
|
||||
}
|
||||
else {
|
||||
gain = FL2FXCONST_DBL(1.0f/(1<<1));
|
||||
}
|
||||
|
||||
/* gain smoothing, method: TDL_EXPONENTIAL */
|
||||
/* first order IIR filter with attack correction to avoid overshoots */
|
||||
|
||||
/* correct the 'aiming' value of the exponential attack to avoid the remaining overshoot */
|
||||
if (gain < smoothState0) {
|
||||
cor = fMin(cor, fMultDiv2((gain - fMultDiv2(FL2FXCONST_SGL(0.1f*(1<<1)),smoothState0)), FL2FXCONST_SGL(1.11111111f/(1<<1)))<<2);
|
||||
}
|
||||
else {
|
||||
cor = gain;
|
||||
}
|
||||
|
||||
/* smoothing filter */
|
||||
if (cor < smoothState0) {
|
||||
smoothState0 = fMult(attackConst,(smoothState0 - cor)) + cor; /* attack */
|
||||
smoothState0 = fMax(smoothState0, gain); /* avoid overshooting target */
|
||||
}
|
||||
else {
|
||||
/* sign inversion twice to round towards +infinity,
|
||||
so that gain can converge to 1.0 again,
|
||||
for bit-identical output when limiter is not active */
|
||||
smoothState0 = -fMult(releaseConst,-(smoothState0 - cor)) + cor; /* release */
|
||||
}
|
||||
|
||||
gain = smoothState0;
|
||||
|
||||
/* lookahead delay, apply gain */
|
||||
for (j = 0; j < channels; j++) {
|
||||
|
||||
tmp = delayBuf[delayBufIdx * channels + j];
|
||||
delayBuf[delayBufIdx * channels + j] = fMult((FIXP_PCM)samples[i * channels + j], additionalGain);
|
||||
|
||||
/* Apply gain to delayed signal */
|
||||
if (gain < FL2FXCONST_DBL(1.0f/(1<<1)))
|
||||
tmp = fMult(tmp,gain<<1);
|
||||
|
||||
samples[i * channels + j] = FX_DBL2FX_PCM((FIXP_DBL)SATURATE_LEFT_SHIFT(tmp,TDL_GAIN_SCALING,DFRACT_BITS));
|
||||
}
|
||||
delayBufIdx++;
|
||||
if (delayBufIdx >= attack) delayBufIdx = 0;
|
||||
|
||||
/* save minimum gain factor */
|
||||
if (gain < minGain) minGain = gain;
|
||||
}
|
||||
|
||||
|
||||
limiter->max = max;
|
||||
limiter->maxBufIdx = maxBufIdx;
|
||||
limiter->cor = cor;
|
||||
limiter->delayBufIdx = delayBufIdx;
|
||||
|
||||
limiter->smoothState0 = smoothState0;
|
||||
limiter->additionalGainFilterState = additionalGainSmoothState;
|
||||
limiter->additionalGainFilterState1 = additionalGainSmoothState1;
|
||||
|
||||
limiter->minGain = minGain;
|
||||
|
||||
limiter->additionalGainPrev = pGain[0];
|
||||
|
||||
return TDLIMIT_OK;
|
||||
}
|
||||
}
|
||||
|
||||
/* get delay in samples */
|
||||
unsigned int getLimiterDelay(TDLimiterPtr limiter)
|
||||
{
|
||||
FDK_ASSERT(limiter != NULL);
|
||||
return limiter->attack;
|
||||
}
|
||||
|
||||
/* set number of channels */
|
||||
TDLIMITER_ERROR setLimiterNChannels(TDLimiterPtr limiter, unsigned int nChannels)
|
||||
{
|
||||
if ( limiter == NULL ) return TDLIMIT_INVALID_HANDLE;
|
||||
|
||||
if (nChannels > limiter->maxChannels) return TDLIMIT_INVALID_PARAMETER;
|
||||
|
||||
limiter->channels = nChannels;
|
||||
//resetLimiter(limiter);
|
||||
|
||||
return TDLIMIT_OK;
|
||||
}
|
||||
|
||||
/* set sampling rate */
|
||||
TDLIMITER_ERROR setLimiterSampleRate(TDLimiterPtr limiter, unsigned int sampleRate)
|
||||
{
|
||||
unsigned int attack, release;
|
||||
FIXP_DBL attackConst, releaseConst, exponent;
|
||||
INT e_ans;
|
||||
|
||||
if ( limiter == NULL ) return TDLIMIT_INVALID_HANDLE;
|
||||
|
||||
if (sampleRate > limiter->maxSampleRate) return TDLIMIT_INVALID_PARAMETER;
|
||||
|
||||
/* update attack and release time in samples */
|
||||
attack = (unsigned int)(limiter->attackMs * sampleRate / 1000);
|
||||
release = (unsigned int)(limiter->releaseMs * sampleRate / 1000);
|
||||
|
||||
/* attackConst = pow(0.1, 1.0 / (attack + 1)) */
|
||||
exponent = invFixp(attack+1);
|
||||
attackConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
|
||||
attackConst = scaleValue(attackConst, e_ans);
|
||||
|
||||
/* releaseConst = (float)pow(0.1, 1.0 / (release + 1)) */
|
||||
exponent = invFixp(release + 1);
|
||||
releaseConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
|
||||
releaseConst = scaleValue(releaseConst, e_ans);
|
||||
|
||||
limiter->attack = attack;
|
||||
limiter->attackConst = attackConst;
|
||||
limiter->releaseConst = releaseConst;
|
||||
limiter->sampleRate = sampleRate;
|
||||
|
||||
/* reset */
|
||||
//resetLimiter(limiter);
|
||||
|
||||
return TDLIMIT_OK;
|
||||
}
|
||||
|
||||
/* set attack time */
|
||||
TDLIMITER_ERROR setLimiterAttack(TDLimiterPtr limiter, unsigned int attackMs)
|
||||
{
|
||||
unsigned int attack;
|
||||
FIXP_DBL attackConst, exponent;
|
||||
INT e_ans;
|
||||
|
||||
if ( limiter == NULL ) return TDLIMIT_INVALID_HANDLE;
|
||||
|
||||
if (attackMs > limiter->maxAttackMs) return TDLIMIT_INVALID_PARAMETER;
|
||||
|
||||
/* calculate attack time in samples */
|
||||
attack = (unsigned int)(attackMs * limiter->sampleRate / 1000);
|
||||
|
||||
/* attackConst = pow(0.1, 1.0 / (attack + 1)) */
|
||||
exponent = invFixp(attack+1);
|
||||
attackConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
|
||||
attackConst = scaleValue(attackConst, e_ans);
|
||||
|
||||
limiter->attack = attack;
|
||||
limiter->attackConst = attackConst;
|
||||
limiter->attackMs = attackMs;
|
||||
|
||||
return TDLIMIT_OK;
|
||||
}
|
||||
|
||||
/* set release time */
|
||||
TDLIMITER_ERROR setLimiterRelease(TDLimiterPtr limiter, unsigned int releaseMs)
|
||||
{
|
||||
unsigned int release;
|
||||
FIXP_DBL releaseConst, exponent;
|
||||
INT e_ans;
|
||||
|
||||
if ( limiter == NULL ) return TDLIMIT_INVALID_HANDLE;
|
||||
|
||||
/* calculate release time in samples */
|
||||
release = (unsigned int)(releaseMs * limiter->sampleRate / 1000);
|
||||
|
||||
/* releaseConst = (float)pow(0.1, 1.0 / (release + 1)) */
|
||||
exponent = invFixp(release + 1);
|
||||
releaseConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
|
||||
releaseConst = scaleValue(releaseConst, e_ans);
|
||||
|
||||
limiter->releaseConst = releaseConst;
|
||||
limiter->releaseMs = releaseMs;
|
||||
|
||||
return TDLIMIT_OK;
|
||||
}
|
||||
|
||||
/* set limiter threshold */
|
||||
TDLIMITER_ERROR setLimiterThreshold(TDLimiterPtr limiter, INT_PCM threshold)
|
||||
{
|
||||
if ( limiter == NULL ) return TDLIMIT_INVALID_HANDLE;
|
||||
|
||||
limiter->threshold = (FIXP_PCM)threshold;
|
||||
|
||||
return TDLIMIT_OK;
|
||||
}
|
||||
@ -148,7 +148,7 @@ amm-info@iis.fraunhofer.de
|
||||
/* Decoder library info */
|
||||
#define PCMDMX_LIB_VL0 2
|
||||
#define PCMDMX_LIB_VL1 4
|
||||
#define PCMDMX_LIB_VL2 1
|
||||
#define PCMDMX_LIB_VL2 2
|
||||
#define PCMDMX_LIB_TITLE "PCM Downmix Lib"
|
||||
#define PCMDMX_LIB_BUILD_DATE __DATE__
|
||||
#define PCMDMX_LIB_BUILD_TIME __TIME__
|
||||
|
||||
Loading…
Reference in New Issue
Block a user