Commit f8201b4c authored by multrus's avatar multrus
Browse files

[cleanup] accept FIX_BASOP_2326_WRONG_SIG_LENGTH

parent cd6e0986

lib_com/options.h

+0 −1
Original line number Diff line number Diff line
@@ -87,7 +87,6 @@ 
#define FIX_2255_ISAR_RENDER_POSES                      /* VA: issue 2255: fix missing check in isar_render_poses() */

#define FIX_1904_HARM_GSC_ENC                           /* VA: #1904 Harmonization of EVS and IVAS GSC code */

#define FIX_1478_UNINIT_ON_BFI                          /* VA: Fix issue 1478 where a vector is partly un-initialized during bfi */

#define FIX_BASOP_2326_WRONG_SIG_LENGTH                 /* Nokia: basop issue 2326: Fix issue by using correct signal length in multiple places */

#define FIX_BASOP_2327_WRONG_LOOP_END                   /* Nokia: basop inssue 2327: Correct loop end to solve MSAN error */

#define FIX_BASOP_2328_MSAN_WRONG_ASSIGNMENT            /* Nokia: basop issue 2328: Fix MSAN error by correcting the variable assignment */

#define FIX_BASOP_2329_UNINIT_VALUE                     /* Nokia: basop issue 2329: Fix issue by checking for structure instead of length */

lib_rend/lib_rend_fx.c

+0 −17
Original line number Diff line number Diff line
@@ -9175,15 +9175,10 @@ static ivas_error renderInputMasa( 
            }



            copyMasaMetadataToDiracRenderer_fx( &masaInput->masaMetadata, masaInput->hMasaExtRend->hSpatParamRendCom, maxBin );

#ifdef FIX_BASOP_2326_WRONG_SIG_LENGTH

            FOR( ch = 0; ch < inAudio.config.numChannels; ch++ )

            {

                Scale_sig32( tmpBuffer_buff_fx[ch], inAudio.config.numSamplesPerChannel, sub( Q11, *outAudio.pq_fact ) ); /* Q11 */

            }

#else

            Scale_sig32( tmpBuffer_buff_fx[0], L_FRAME48k, sub( Q11, *outAudio.pq_fact ) ); /* Q11 */

            Scale_sig32( tmpBuffer_buff_fx[1], L_FRAME48k, sub( Q11, *outAudio.pq_fact ) ); /* Q11 */

#endif

            // scale_sig32( outAudio.data_fx, i_mult( outAudio.config.numChannels, outAudio.config.numSamplesPerChannel ), sub( Q11, *outAudio.pq_fact ) );



            ivas_masa_ext_rend_parambin_render_fx( masaInput->hMasaExtRend, *masaInput->base.ctx.pCombinedOrientationData, tmpBuffer_fx, num_subframes, masaInput->base.ctx.pSplitRendWrapper, Cldfb_RealBuffer_Binaural, Cldfb_ImagBuffer_Binaural );
@@ -9207,17 +9202,10 @@ static ivas_error renderInputMasa( 
                        masaInput->hMasaExtRend->cldfbAnaRend[0]->Q_cldfb_state = Q11;

                        move16();

                    }

#ifdef FIX_BASOP_2326_WRONG_SIG_LENGTH

                    FOR( ch = 0; ch < inAudio.config.numChannels; ch++ )

                    {

                        Scale_sig32( tmpBuffer_buff_fx[ch], inAudio.config.numSamplesPerChannel, sub( Q11, *outAudio.pq_fact ) ); /* Q11 */

                    }

#else

                    FOR( ch = 0; ch < MAX_OUTPUT_CHANNELS; ch++ )

                    {

                        Scale_sig32( tmpBuffer_buff_fx[ch], L_FRAME48k, sub( Q11, *outAudio.pq_fact ) ); /* Q11 */

                    }

#endif



                    scale_sig32( outAudio.data_fx, i_mult( outAudio.config.numChannels, outAudio.config.numSamplesPerChannel ), sub( Q11, *outAudio.pq_fact ) ); /* Q11 */
@@ -9241,15 +9229,10 @@ static ivas_error renderInputMasa( 


                    copyMasaMetadataToDiracRenderer_fx( &masaInput->masaMetadata, masaInput->hMasaExtRend->hSpatParamRendCom, maxBin );



#ifdef FIX_BASOP_2326_WRONG_SIG_LENGTH

                    FOR( ch = 0; ch < inAudio.config.numChannels; ch++ )

                    {

                        Scale_sig32( tmpBuffer_buff_fx[ch], inAudio.config.numSamplesPerChannel, sub( Q11, *outAudio.pq_fact ) ); /* Q11 */

                    }

#else

                    Scale_sig32( tmpBuffer_buff_fx[0], L_FRAME48k, sub( Q11, *outAudio.pq_fact ) ); /* Q11 */

                    Scale_sig32( tmpBuffer_buff_fx[1], L_FRAME48k, sub( Q11, *outAudio.pq_fact ) ); /* Q11 */

#endif



                    scale_sig32( outAudio.data_fx, i_mult( outAudio.config.numChannels, outAudio.config.numSamplesPerChannel ), sub( Q11, *outAudio.pq_fact ) );