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Delaying allocation of the SILK temporary output buffer to reduce peak stack

--- a/silk/dec_API.c

+++ b/silk/dec_API.c

@@ -31,6 +31,7 @@

 #include "API.h"

 #include "main.h"

 #include "stack_alloc.h"

+#include "os_support.h"

 /************************/

 /* Decoder Super Struct */

@@ -90,7 +91,8 @@

     opus_int   i, n, decode_only_middle = 0, ret = SILK_NO_ERROR;

     opus_int32 nSamplesOutDec, LBRR_symbol;

     opus_int16 *samplesOut1_tmp[ 2 ];

-    VARDECL( opus_int16, samplesOut1_tmp_storage );

+    VARDECL( opus_int16, samplesOut1_tmp_storage1 );

+    VARDECL( opus_int16, samplesOut1_tmp_storage2 );

     VARDECL( opus_int16, samplesOut2_tmp );

     opus_int32 MS_pred_Q13[ 2 ] = { 0 };

     opus_int16 *resample_out_ptr;

@@ -98,6 +100,7 @@

     silk_decoder_state *channel_state = psDec->channel_state;

     opus_int has_side;

     opus_int stereo_to_mono;

+    int delay_stack_alloc;

     SAVE_STACK;

     silk_assert( decControl->nChannelsInternal == 1 || decControl->nChannelsInternal == 2 );

@@ -251,13 +254,22 @@

         psDec->channel_state[ 1 ].first_frame_after_reset = 1;

     }

-    ALLOC( samplesOut1_tmp_storage,

-           decControl->nChannelsInternal*(

-               channel_state[ 0 ].frame_length + 2 ),

+    /* Check if the temp buffer fits into the output PCM buffer. If it fits,

+       we can delay allocating the temp buffer until after the SILK peak stack

+       usage. We need to use a < and not a <= because of the two extra samples. */

+    delay_stack_alloc = decControl->internalSampleRate*decControl->nChannelsInternal

+          < decControl->API_sampleRate*decControl->nChannelsAPI;

+    ALLOC( samplesOut1_tmp_storage1, delay_stack_alloc ? ALLOC_NONE

+           : decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2 ),

            opus_int16 );

-    samplesOut1_tmp[ 0 ] = samplesOut1_tmp_storage;

-    samplesOut1_tmp[ 1 ] = samplesOut1_tmp_storage

-                           + channel_state[ 0 ].frame_length + 2;

+    if ( delay_stack_alloc )

+    {

+       samplesOut1_tmp[ 0 ] = samplesOut;

+       samplesOut1_tmp[ 1 ] = samplesOut + channel_state[ 0 ].frame_length + 2;

+    } else {

+       samplesOut1_tmp[ 0 ] = samplesOut1_tmp_storage1;

+       samplesOut1_tmp[ 1 ] = samplesOut1_tmp_storage1 + channel_state[ 0 ].frame_length + 2;

+    }

     if( lostFlag == FLAG_DECODE_NORMAL ) {

         has_side = !decode_only_middle;

@@ -312,6 +324,15 @@

         resample_out_ptr = samplesOut;

     }

+    ALLOC( samplesOut1_tmp_storage2, delay_stack_alloc

+           ? decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2 )

+           : ALLOC_NONE,

+           opus_int16 );

+    if ( delay_stack_alloc ) {

+       OPUS_COPY(samplesOut1_tmp_storage2, samplesOut, decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2));

+       samplesOut1_tmp[ 0 ] = samplesOut1_tmp_storage2;

+       samplesOut1_tmp[ 1 ] = samplesOut1_tmp_storage2 + channel_state[ 0 ].frame_length + 2;

+    }

     for( n = 0; n < silk_min( decControl->nChannelsAPI, decControl->nChannelsInternal ); n++ ) {

         /* Resample decoded signal to API_sampleRate */