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

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