Proper SILK delay compensation for resampling

Adds SILK delay compensation that depends on encode and decode sampling
rate, as well as SILK internal coding rate. This ensures that the SILK
part of Opus is always in sync with the CELT part no matter what the
sampling rates are. It also increases the resampling delay to 1.15 ms
(was previously 0.48 ms).

1 files changed, 24 insertions, 6 deletions

diff --git a/silk/dec_API.c b/silk/dec_API.c
index 675bfb99..5e676932 100644
--- a/silk/dec_API.c
+++ b/silk/dec_API.c
@@ -31,6 +31,14 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
 #include "API.h"
 #include "main.h"
 
+static const int dec_delay_matrix[3][5] = {
+/*SILK API 8  12  16  24  48 */
+/* 8 */   {3, 0, 2, 0, 0},
+/*12 */   {0, 8, 5, 7, 5},
+/*16 */   {0, 0, 8, 5, 5}
+};
+
+
 /************************/
 /* Decoder Super Struct */
 /************************/
@@ -82,12 +90,15 @@ opus_int silk_Decode(
 {
     opus_int   i, n, prev_fs_kHz, decode_only_middle = 0, ret = SILK_NO_ERROR;
     opus_int32 nSamplesOutDec, LBRR_symbol;
-    opus_int16 samplesOut1_tmp[ 2 ][ MAX_FS_KHZ * MAX_FRAME_LENGTH_MS + 2 ];
+    opus_int16 samplesOut1_tmp[ 2 ][ MAX_FS_KHZ * MAX_FRAME_LENGTH_MS + 2 + MAX_DECODER_DELAY ];
     opus_int16 samplesOut2_tmp[ MAX_API_FS_KHZ * MAX_FRAME_LENGTH_MS ];
     opus_int32 MS_pred_Q13[ 2 ] = { 0 };
     opus_int16 *resample_out_ptr;
     silk_decoder *psDec = ( silk_decoder * )decState;
     silk_decoder_state *channel_state = psDec->channel_state;
+    int delay;
+
+    delay = channel_state[ 0 ].delay;
 
     /**********************************/
     /* Test if first frame in payload */
@@ -106,6 +117,7 @@ opus_int silk_Decode(
         ret += silk_init_decoder( &channel_state[ 1 ] );
         if( psDec->nChannelsAPI == 2 ) {
             silk_memcpy( &channel_state[ 1 ].resampler_state, &channel_state[ 0 ].resampler_state, sizeof( silk_resampler_state_struct ) );
+            silk_memcpy( &channel_state[ 1 ].delayBuf, &channel_state[ 0 ].delayBuf, MAX_DECODER_DELAY*sizeof(opus_int16));
         }
     }
 
@@ -143,9 +155,12 @@ opus_int silk_Decode(
 
     /* Initialize resampler when switching internal or external sampling frequency */
     if( prev_fs_kHz != channel_state[ 0 ].fs_kHz || channel_state[ 0 ].prev_API_sampleRate != decControl->API_sampleRate ) {
+        channel_state[ 0 ].delay = dec_delay_matrix[rateID(silk_SMULBB( channel_state[ 0 ].fs_kHz, 1000 ))][rateID(decControl->API_sampleRate)];
+        silk_assert(channel_state[ 0 ].delay <= MAX_DECODER_DELAY);
         ret = silk_resampler_init( &channel_state[ 0 ].resampler_state, silk_SMULBB( channel_state[ 0 ].fs_kHz, 1000 ), decControl->API_sampleRate );
         if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 2 ) {
             silk_memcpy( &channel_state[ 1 ].resampler_state, &channel_state[ 0 ].resampler_state, sizeof( silk_resampler_state_struct ) );
+            channel_state[ 1 ].delay = channel_state[ 0 ].delay;
         }
     }
     channel_state[ 0 ].prev_API_sampleRate = decControl->API_sampleRate;
@@ -230,19 +245,19 @@ opus_int silk_Decode(
     /* Call decoder for one frame */
     for( n = 0; n < decControl->nChannelsInternal; n++ ) {
         if( n == 0 || decode_only_middle == 0 ) {
-            ret += silk_decode_frame( &channel_state[ n ], psRangeDec, &samplesOut1_tmp[ n ][ 2 ], &nSamplesOutDec, lostFlag );
+            ret += silk_decode_frame( &channel_state[ n ], psRangeDec, &samplesOut1_tmp[ n ][ 2 + delay ], &nSamplesOutDec, lostFlag );
         } else {
-            silk_memset( &samplesOut1_tmp[ n ][ 2 ], 0, nSamplesOutDec * sizeof( opus_int16 ) );
+            silk_memset( &samplesOut1_tmp[ n ][ 2 + delay ], 0, nSamplesOutDec * sizeof( opus_int16 ) );
         }
     }
 
     if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 2 ) {
         /* Convert Mid/Side to Left/Right */
-        silk_stereo_MS_to_LR( &psDec->sStereo, samplesOut1_tmp[ 0 ], samplesOut1_tmp[ 1 ], MS_pred_Q13, channel_state[ 0 ].fs_kHz, nSamplesOutDec );
+        silk_stereo_MS_to_LR( &psDec->sStereo, &samplesOut1_tmp[ 0 ][delay], &samplesOut1_tmp[ 1 ][delay], MS_pred_Q13, channel_state[ 0 ].fs_kHz, nSamplesOutDec );
     } else {
         /* Buffering */
-        silk_memcpy( samplesOut1_tmp[ 0 ], psDec->sStereo.sMid, 2 * sizeof( opus_int16 ) );
-        silk_memcpy( psDec->sStereo.sMid, &samplesOut1_tmp[ 0 ][ nSamplesOutDec ], 2 * sizeof( opus_int16 ) );
+        silk_memcpy( &samplesOut1_tmp[ 0 ][delay], psDec->sStereo.sMid, 2 * sizeof( opus_int16 ) );
+        silk_memcpy( psDec->sStereo.sMid, &samplesOut1_tmp[ 0 ][ nSamplesOutDec + delay ], 2 * sizeof( opus_int16 ) );
     }
 
     /* Number of output samples */
@@ -256,8 +271,11 @@ opus_int silk_Decode(
     }
 
     for( n = 0; n < silk_min( decControl->nChannelsAPI, decControl->nChannelsInternal ); n++ ) {
+
+        silk_memcpy(&samplesOut1_tmp[ n ][ 1 ], &channel_state[ n ].delayBuf[ MAX_DECODER_DELAY-delay ], delay*sizeof(opus_int16));
         /* Resample decoded signal to API_sampleRate */
         ret += silk_resampler( &channel_state[ n ].resampler_state, resample_out_ptr, &samplesOut1_tmp[ n ][ 1 ], nSamplesOutDec );
+        silk_memcpy(channel_state[ n ].delayBuf, &samplesOut1_tmp[ n ][ 1 + nSamplesOutDec + delay - MAX_DECODER_DELAY ], MAX_DECODER_DELAY*sizeof(opus_int16));
 
         /* Interleave if stereo output and stereo stream */
         if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 2 ) {