Misc bug fixes

- There was a bug where the decoder resampler was not properly initialized
when fs_kHz == API_fs_kHz. In that case the resampler would continue to
upsample, and the output was corrupt.

- The delay value in the decoder was taken from the state before it was
potentially updated. This caused the decoder to apply the new dalay value one
frame late

- The encoder and decoder states are now updated more consistently, when
the sampling rate changes (pesq liked these changes)

- Properly resetting the side channel encoder and decoder for the first
frame with side coding active again

- Faster updating the "ratio" value in the LR_to_MS() code for large
prediction values means that for certain extreme/artificial input
signals the output looks better

1 files changed, 28 insertions, 12 deletions

diff --git a/silk/enc_API.c b/silk/enc_API.c
index fb27437e..339dafc4 100644
--- a/silk/enc_API.c
+++ b/silk/enc_API.c
@@ -237,13 +237,13 @@ opus_int silk_Encode(
             for( n = 0; n < nSamplesFromInput; n++ ) {
                 buf[ n+delay ] = samplesIn[ 2 * n ];
             }
-            silk_memcpy(buf, &psEnc->state_Fxx[ 0 ].sCmn.delayBuf[MAX_ENCODER_DELAY-delay], delay*sizeof(opus_int16));
+            silk_memcpy(buf, &psEnc->state_Fxx[ 0 ].sCmn.delayBuf[ MAX_ENCODER_DELAY - delay ], delay * sizeof(opus_int16));
             /* Making sure to start both resamplers from the same state when switching from mono to stereo */
             if(psEnc->nPrevChannelsInternal == 1 && id==0) {
                silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, sizeof(psEnc->state_Fxx[ 1 ].sCmn.resampler_state));
                silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.delayBuf, &psEnc->state_Fxx[ 0 ].sCmn.delayBuf, MAX_ENCODER_DELAY*sizeof(opus_int16));
             }
-            silk_memcpy(psEnc->state_Fxx[ 0 ].sCmn.delayBuf, buf+nSamplesFromInput+delay-MAX_ENCODER_DELAY, MAX_ENCODER_DELAY*sizeof(opus_int16));
+            silk_memcpy(psEnc->state_Fxx[ 0 ].sCmn.delayBuf, buf + nSamplesFromInput + delay - MAX_ENCODER_DELAY, MAX_ENCODER_DELAY*sizeof(opus_int16));
 
             ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state,
                 &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput );
@@ -252,24 +252,24 @@ opus_int silk_Encode(
             nSamplesToBuffer  = psEnc->state_Fxx[ 1 ].sCmn.frame_length - psEnc->state_Fxx[ 1 ].sCmn.inputBufIx;
             nSamplesToBuffer  = silk_min( nSamplesToBuffer, 10 * nBlocksOf10ms * psEnc->state_Fxx[ 1 ].sCmn.fs_kHz );
             for( n = 0; n < nSamplesFromInput; n++ ) {
-                buf[ n+delay ] = samplesIn[ 2 * n + 1 ];
+                buf[ n + delay ] = samplesIn[ 2 * n + 1 ];
             }
-            silk_memcpy(buf, &psEnc->state_Fxx[ 1 ].sCmn.delayBuf[MAX_ENCODER_DELAY-delay], delay*sizeof(opus_int16));
+            silk_memcpy(buf, &psEnc->state_Fxx[ 1 ].sCmn.delayBuf[ MAX_ENCODER_DELAY - delay ], delay * sizeof(opus_int16));
             ret += silk_resampler( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state,
                 &psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput );
-            silk_memcpy(psEnc->state_Fxx[ 1 ].sCmn.delayBuf, buf+nSamplesFromInput+delay-MAX_ENCODER_DELAY, MAX_ENCODER_DELAY*sizeof(opus_int16));
+            silk_memcpy(psEnc->state_Fxx[ 1 ].sCmn.delayBuf, buf + nSamplesFromInput + delay - MAX_ENCODER_DELAY, MAX_ENCODER_DELAY*sizeof(opus_int16));
 
             psEnc->state_Fxx[ 1 ].sCmn.inputBufIx += nSamplesToBuffer;
         } else if( encControl->nChannelsAPI == 2 && encControl->nChannelsInternal == 1 ) {
             /* Combine left and right channels before resampling */
             for( n = 0; n < nSamplesFromInput; n++ ) {
-                buf[ n+delay ] = (opus_int16)silk_RSHIFT_ROUND( samplesIn[ 2 * n ] + samplesIn[ 2 * n + 1 ],  1 );
+                buf[ n + delay ] = (opus_int16)silk_RSHIFT_ROUND( samplesIn[ 2 * n ] + samplesIn[ 2 * n + 1 ],  1 );
             }
             if(psEnc->nPrevChannelsInternal == 2 && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded==0) {
                for ( n = 0; n<MAX_ENCODER_DELAY; n++ )
                   psEnc->state_Fxx[ 0 ].sCmn.delayBuf[ n ] = silk_RSHIFT(psEnc->state_Fxx[ 0 ].sCmn.delayBuf[ n ]+(opus_int32)psEnc->state_Fxx[ 1 ].sCmn.delayBuf[ n ], 1);
             }
-            silk_memcpy(buf, &psEnc->state_Fxx[ 0 ].sCmn.delayBuf[MAX_ENCODER_DELAY-delay], delay*sizeof(opus_int16));
+            silk_memcpy(buf, &psEnc->state_Fxx[ 0 ].sCmn.delayBuf[ MAX_ENCODER_DELAY - delay ], delay * sizeof(opus_int16));
             ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state,
                 &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput );
             /* On the first mono frame, average the results for the two resampler states  */
@@ -281,17 +281,16 @@ opus_int silk_Encode(
                         silk_RSHIFT(psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx+n+2 ]
                                   + psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ psEnc->state_Fxx[ 1 ].sCmn.inputBufIx+n+2 ], 1);
                }
-
             }
-            silk_memcpy(psEnc->state_Fxx[ 0 ].sCmn.delayBuf, buf+nSamplesFromInput+delay-MAX_ENCODER_DELAY, MAX_ENCODER_DELAY*sizeof(opus_int16));
+            silk_memcpy(psEnc->state_Fxx[ 0 ].sCmn.delayBuf, buf + nSamplesFromInput + delay - MAX_ENCODER_DELAY, MAX_ENCODER_DELAY*sizeof(opus_int16));
             psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer;
         } else {
             silk_assert( encControl->nChannelsAPI == 1 && encControl->nChannelsInternal == 1 );
-            silk_memcpy(buf+delay, samplesIn, nSamplesFromInput*sizeof(opus_int16));
-            silk_memcpy(buf, &psEnc->state_Fxx[ 0 ].sCmn.delayBuf[MAX_ENCODER_DELAY-delay], delay*sizeof(opus_int16));
+            silk_memcpy(buf + delay, samplesIn, nSamplesFromInput*sizeof(opus_int16));
+            silk_memcpy(buf, &psEnc->state_Fxx[ 0 ].sCmn.delayBuf[ MAX_ENCODER_DELAY - delay ], delay * sizeof(opus_int16));
             ret += silk_resampler( &psEnc->state_Fxx[ 0 ].sCmn.resampler_state,
                 &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx + 2 ], buf, nSamplesFromInput );
-            silk_memcpy(psEnc->state_Fxx[ 0 ].sCmn.delayBuf, buf+nSamplesFromInput+delay-MAX_ENCODER_DELAY, MAX_ENCODER_DELAY*sizeof(opus_int16));
+            silk_memcpy(psEnc->state_Fxx[ 0 ].sCmn.delayBuf, buf + nSamplesFromInput + delay - MAX_ENCODER_DELAY, MAX_ENCODER_DELAY*sizeof(opus_int16));
             psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer;
         }
 
@@ -387,6 +386,22 @@ opus_int silk_Encode(
                 silk_memcpy( psEnc->sStereo.sMid, &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.frame_length ], 2 * sizeof( opus_int16 ) );
             }
 
+            /* Reset side channel encoder memory for first frame with side coding */
+            if( encControl->nChannelsInternal == 2 && psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] == 0 && psEnc->prev_decode_only_middle == 1 ) {
+                silk_memset( &psEnc->state_Fxx[ 1 ].sShape,               0, sizeof( psEnc->state_Fxx[ 1 ].sShape ) );
+                silk_memset( &psEnc->state_Fxx[ 1 ].sPrefilt,             0, sizeof( psEnc->state_Fxx[ 1 ].sPrefilt ) );
+                silk_memset( &psEnc->state_Fxx[ 1 ].sCmn.sNSQ,            0, sizeof( psEnc->state_Fxx[ 1 ].sCmn.sNSQ ) );
+                silk_memset( psEnc->state_Fxx[ 1 ].sCmn.prev_NLSFq_Q15,   0, sizeof( psEnc->state_Fxx[ 1 ].sCmn.prev_NLSFq_Q15 ) );
+                silk_memset( &psEnc->state_Fxx[ 1 ].sCmn.sLP.In_LP_State, 0, sizeof( psEnc->state_Fxx[ 1 ].sCmn.sLP.In_LP_State ) );
+                silk_memset( &psEnc->state_Fxx[ 1 ].sCmn.inputBuf,        0, sizeof( psEnc->state_Fxx[ 1 ].sCmn.inputBuf ) );
+                psEnc->state_Fxx[ 1 ].sCmn.prevLag                = 100;
+                psEnc->state_Fxx[ 1 ].sCmn.sNSQ.lagPrev           = 100;
+                psEnc->state_Fxx[ 1 ].sShape.LastGainIndex        = 10;
+                psEnc->state_Fxx[ 1 ].sCmn.prevSignalType         = TYPE_NO_VOICE_ACTIVITY;
+                psEnc->state_Fxx[ 1 ].sCmn.sNSQ.prev_inv_gain_Q16 = 65536;
+            }
+            psEnc->prev_decode_only_middle = psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ];
+
             /* Encode */
             for( n = 0; n < encControl->nChannelsInternal; n++ ) {
                 if( encControl->nChannelsInternal == 1 ) {
@@ -450,6 +465,7 @@ opus_int silk_Encode(
             break;
         }
     }
+
     psEnc->nPrevChannelsInternal = encControl->nChannelsInternal;
 
     encControl->allowBandwidthSwitch = psEnc->allowBandwidthSwitch;