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-rw-r--r--libAACdec/src/usacdec_lpd.h19
1 files changed, 11 insertions, 8 deletions
diff --git a/libAACdec/src/usacdec_lpd.h b/libAACdec/src/usacdec_lpd.h
index 3e7938d..448dc55 100644
--- a/libAACdec/src/usacdec_lpd.h
+++ b/libAACdec/src/usacdec_lpd.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
@@ -140,13 +140,14 @@ void CLpdChannelStream_Decode(
* \param pTimeData pointer to output buffer
* \param samplesPerFrame amount of output samples
* \param pSamplingRateInfo holds the sampling rate information
- * \param pWorkBuffer1 pointer to work buffer for temporal data
+ * \param aacOutDataHeadroom headroom of the output time signal to prevent
+ * clipping
*/
AAC_DECODER_ERROR CLpd_RenderTimeSignal(
CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo,
- CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM *pTimeData,
+ CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC *pTimeData,
INT samplesPerFrame, SamplingRateInfo *pSamplingRateInfo, UINT frameOk,
- UINT flags, UINT strmFlags);
+ const INT aacOutDataHeadroom, UINT flags, UINT strmFlags);
static inline INT CLpd_FAC_getLength(int fNotShortBlock, int fac_length_long) {
if (fNotShortBlock) {
@@ -156,8 +157,9 @@ static inline INT CLpd_FAC_getLength(int fNotShortBlock, int fac_length_long) {
}
}
-void filtLP(const FIXP_DBL *syn, FIXP_PCM *syn_out, FIXP_DBL *noise,
- const FIXP_SGL *filt, INT stop, int len);
+void filtLP(const FIXP_DBL *syn, PCM_DEC *syn_out, FIXP_DBL *noise,
+ const FIXP_SGL *filt, const INT aacOutDataHeadroom, INT stop,
+ int len);
/**
* \brief perform a low-frequency pitch enhancement on time domain signal
@@ -171,13 +173,14 @@ void filtLP(const FIXP_DBL *syn, FIXP_PCM *syn_out, FIXP_DBL *noise,
* \param[in] l_frame length of filtering, must be multiple of L_SUBFR
* \param[in] l_next length of allowed look ahead on syn[i], i < l_frame+l_next
* \param[out] synth_out pointer to time domain output signal
+ * \param[in] headroom of the output time signal to prevent clipping
* \param[in,out] mem_bpf pointer to filter memory (L_FILT+L_SUBFR)
*/
void bass_pf_1sf_delay(FIXP_DBL syn[], const INT T_sf[], FIXP_DBL *pit_gain,
const int frame_length, const INT l_frame,
- const INT l_next, FIXP_PCM *synth_out,
- FIXP_DBL mem_bpf[]);
+ const INT l_next, PCM_DEC *synth_out,
+ const INT aacOutDataHeadroom, FIXP_DBL mem_bpf[]);
/**
* \brief random sign generator for FD and TCX noise filling