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

1 files changed, 28 insertions, 7 deletions

diff --git a/silk/dec_API.c b/silk/dec_API.c
index 660f93db..1087c672 100644
--- a/silk/dec_API.c
+++ b/silk/dec_API.c
@@ -31,6 +31,7 @@ POSSIBILITY OF SUCH DAMAGE.
 #include "API.h"
 #include "main.h"
 #include "stack_alloc.h"
+#include "os_support.h"
 
 /************************/
 /* Decoder Super Struct */
@@ -90,7 +91,8 @@ opus_int silk_Decode(                                   /* O    Returns error co
     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 @@ opus_int silk_Decode(                                   /* O    Returns error co
     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 @@ opus_int silk_Decode(                                   /* O    Returns error co
         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 @@ opus_int silk_Decode(                                   /* O    Returns error co
         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 */