| |
| /* ----------------------------------------------------------------------------------------------------------- |
| 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; |
| } |