SILK fixes following last codec WG meeting

decoder:
- fixed incorrect scaling of filter states for the smallest quantization
  step sizes
- NLSF2A now limits the prediction gain of LPC filters

encoder:
- increased damping of LTP coefficients in LTP analysis
- increased white noise fraction in noise shaping LPC analysis
- introduced maximum total prediction gain.  Used by Burg's method to
  exit early if prediction gain is exceeded.  This improves packet
  loss robustness and numerical robustness in Burg's method
- Prefiltered signal is now in int32 Q10 domain, from int16 Q0
- Increased max number of iterations in CBR gain control loop from 5 to 6
- Removed useless code from LTP scaling control
- Optimization: smarter LPC loop unrolling
- Switched default win32 compile mode to be floating-point

resampler:
- made resampler have constant delay of 0.75 ms; removed delay
  compensation from silk code.
- removed obsolete table entries (~850 Bytes)
- increased downsampling filter order from 16 to 18/24/36 (depending on
  frequency ratio)
- reoptimized filter coefficients

1 files changed, 9 insertions, 23 deletions

diff --git a/silk/enc_API.c b/silk/enc_API.c
index 02a8fd01..b993ed59 100644
--- a/silk/enc_API.c
+++ b/silk/enc_API.c
@@ -140,10 +140,10 @@ opus_int silk_Encode(                                   /* O    Returns error co
     opus_int   n, i, nBits, flags, tmp_payloadSize_ms = 0, tmp_complexity = 0, ret = 0;
     opus_int   nSamplesToBuffer, nBlocksOf10ms, nSamplesFromInput = 0;
     opus_int   speech_act_thr_for_switch_Q8;
-    opus_int32 TargetRate_bps, MStargetRates_bps[ 2 ], channelRate_bps, LBRR_symbol;
+    opus_int32 TargetRate_bps, MStargetRates_bps[ 2 ], channelRate_bps, LBRR_symbol, sum;
     silk_encoder *psEnc = ( silk_encoder * )encState;
-    opus_int16 buf[ MAX_FRAME_LENGTH_MS * MAX_API_FS_KHZ + MAX_ENCODER_DELAY];
-    opus_int transition, delay, curr_block, tot_blocks;
+    opus_int16 buf[ MAX_FRAME_LENGTH_MS * MAX_API_FS_KHZ ];
+    opus_int transition, curr_block, tot_blocks;
 
     psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded = psEnc->state_Fxx[ 1 ].sCmn.nFramesEncoded = 0;
 
@@ -233,7 +233,6 @@ opus_int silk_Encode(                                   /* O    Returns error co
     }
     silk_assert( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 0 ].sCmn.fs_kHz == psEnc->state_Fxx[ 1 ].sCmn.fs_kHz );
 
-    delay = psEnc->state_Fxx[ 0 ].sCmn.delay;
     /* Input buffering/resampling and encoding */
     while( 1 ) {
         nSamplesToBuffer  = psEnc->state_Fxx[ 0 ].sCmn.frame_length - psEnc->state_Fxx[ 0 ].sCmn.inputBufIx;
@@ -243,15 +242,12 @@ opus_int silk_Encode(                                   /* O    Returns error co
         if( encControl->nChannelsAPI == 2 && encControl->nChannelsInternal == 2 ) {
             opus_int id = psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded;
             for( n = 0; n < nSamplesFromInput; n++ ) {
-                buf[ n+delay ] = samplesIn[ 2 * n ];
+                buf[ n ] = samplesIn[ 2 * n ];
             }
-            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) {
+            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));
 
             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 );
@@ -260,25 +256,18 @@ opus_int silk_Encode(                                   /* O    Returns error co
             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 ] = samplesIn[ 2 * n + 1 ];
             }
-            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));
 
             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 );
+                sum = samplesIn[ 2 * n ] + samplesIn[ 2 * n + 1 ];
+                buf[ n ] = (opus_int16)silk_RSHIFT_ROUND( sum,  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));
             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  */
@@ -291,15 +280,12 @@ opus_int silk_Encode(                                   /* O    Returns error co
                                   + 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));
             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, samplesIn, nSamplesFromInput*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));
             psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer;
         }