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| author | Fraunhofer IIS FDK <audio-fdk@iis.fraunhofer.de> | 2019-12-19 17:28:15 +0100 |
|---|---|---|
| committer | Jean-Michel Trivi <jmtrivi@google.com> | 2020-02-14 10:53:51 -0800 |
| commit | e016635f0d3a5c7532b00711ce461f97a13f7bc2 (patch) | |
| tree | 44d6676c2975eec965bb3e6c2562e1632eaf4385 /libPCMutils/include | |
| parent | 57c9355de0269afb462ad4a8aa8814f6a6486ff1 (diff) | |
| download | fdk-aac-e016635f0d3a5c7532b00711ce461f97a13f7bc2.tar.gz fdk-aac-e016635f0d3a5c7532b00711ce461f97a13f7bc2.tar.bz2 fdk-aac-e016635f0d3a5c7532b00711ce461f97a13f7bc2.zip | |
Avoid decoder internal clipping by converting the whole audio sample data path from 16 to 32 bit data width (FDKdec v3.2.0).
Bug: 149514474
Test: atest DecoderTestXheAac DecoderTestAacDrc
Change-Id: I8a504ab709e42e27a61fe29840212953742283a5
Diffstat (limited to 'libPCMutils/include')
| -rw-r--r-- | libPCMutils/include/limiter.h | 36 |
1 files changed, 10 insertions, 26 deletions
diff --git a/libPCMutils/include/limiter.h b/libPCMutils/include/limiter.h index fab7226..419e891 100644 --- a/libPCMutils/include/limiter.h +++ b/libPCMutils/include/limiter.h @@ -1,7 +1,7 @@ /* ----------------------------------------------------------------------------- Software License for The Fraunhofer FDK AAC Codec Library for Android -© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +© Copyright 1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. All rights reserved. 1. INTRODUCTION @@ -109,8 +109,6 @@ amm-info@iis.fraunhofer.de #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 @@ -128,10 +126,7 @@ struct TDLimiter { unsigned int maxBufIdx, delayBufIdx; FIXP_DBL smoothState0; FIXP_DBL minGain; - - FIXP_DBL additionalGainPrev; - FIXP_DBL additionalGainFilterState; - FIXP_DBL additionalGainFilterState1; + INT scaling; }; typedef enum { @@ -255,27 +250,16 @@ TDLIMITER_ERROR pcmLimiter_SetThreshold(TDLimiterPtr limiter, /****************************************************************************** * pcmLimiter_Apply * - * 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 * + * limiter: limiter handle * + * samplesIn: pointer to input buffer containing interleaved samples * + * samplesOut: pointer to output buffer containing interleaved samples * + * pGainPerSample: pointer to gains for each sample * + * scaling: scaling of output samples * + * nSamples: number of samples per channel * * returns: error code * ******************************************************************************/ TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn, - INT_PCM* samplesOut, FIXP_DBL* pGain, - const INT* gain_scale, const UINT gain_size, - const UINT gain_delay, const UINT nSamples); + INT_PCM* samplesOut, FIXP_DBL* pGainPerSample, + const INT scaling, const UINT nSamples); #endif /* #ifndef LIMITER_H */ |