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, 6 insertions, 16 deletions

diff --git a/silk/control_codec.c b/silk/control_codec.c
index 829b08f7..16666d68 100644
--- a/silk/control_codec.c
+++ b/silk/control_codec.c
@@ -38,13 +38,6 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
 #include "tuning_parameters.h"
 #include "pitch_est_defines.h"
 
-static const opus_int enc_delay_matrix[3][5] = {
-/*SILK API 8  12  16  24  48 */
-/* 8 */   {5,  0,  3,  4,  8},
-/*12 */   {0,  6,  0,  0,  0},
-/*16 */   {4,  5, 11,  5, 18}
-};
-
 opus_int silk_setup_resamplers(
     silk_encoder_state_Fxx          *psEnc,             /* I/O                      */
     opus_int                        fs_kHz              /* I                        */
@@ -150,7 +143,7 @@ opus_int silk_setup_resamplers(
     {
         if( psEnc->sCmn.fs_kHz == 0 ) {
             /* Initialize the resampler for enc_API.c preparing resampling from API_fs_Hz to fs_kHz */
-            ret += silk_resampler_init( &psEnc->sCmn.resampler_state, psEnc->sCmn.API_fs_Hz, fs_kHz * 1000 );
+            ret += silk_resampler_init( &psEnc->sCmn.resampler_state, psEnc->sCmn.API_fs_Hz, fs_kHz * 1000, 1 );
         } else {
             /* Allocate worst case space for temporary upsampling, 8 to 48 kHz, so a factor 6 */
             opus_int16 x_buf_API_fs_Hz[ ( 2 * MAX_FRAME_LENGTH_MS + LA_SHAPE_MS ) * MAX_API_FS_KHZ ];
@@ -168,7 +161,7 @@ opus_int silk_setup_resamplers(
 #endif
 
             /* Initialize resampler for temporary resampling of x_buf data to API_fs_Hz */
-            ret += silk_resampler_init( &temp_resampler_state, silk_SMULBB( psEnc->sCmn.fs_kHz, 1000 ), psEnc->sCmn.API_fs_Hz );
+            ret += silk_resampler_init( &temp_resampler_state, silk_SMULBB( psEnc->sCmn.fs_kHz, 1000 ), psEnc->sCmn.API_fs_Hz, 0 );
 
             /* Temporary resampling of x_buf data to API_fs_Hz */
             ret += silk_resampler( &temp_resampler_state, x_buf_API_fs_Hz, x_bufFIX, nSamples_temp );
@@ -177,7 +170,7 @@ opus_int silk_setup_resamplers(
             nSamples_temp = silk_DIV32_16( nSamples_temp * psEnc->sCmn.API_fs_Hz, silk_SMULBB( psEnc->sCmn.fs_kHz, 1000 ) );
 
             /* Initialize the resampler for enc_API.c preparing resampling from API_fs_Hz to fs_kHz */
-            ret += silk_resampler_init( &psEnc->sCmn.resampler_state, psEnc->sCmn.API_fs_Hz, silk_SMULBB( fs_kHz, 1000 ) );
+            ret += silk_resampler_init( &psEnc->sCmn.resampler_state, psEnc->sCmn.API_fs_Hz, silk_SMULBB( fs_kHz, 1000 ), 1 );
 
             /* Correct resampler state by resampling buffered data from API_fs_Hz to fs_kHz */
             ret += silk_resampler( &psEnc->sCmn.resampler_state, x_bufFIX, x_buf_API_fs_Hz, nSamples_temp );
@@ -234,9 +227,6 @@ opus_int silk_setup_fs(
         psEnc->sCmn.TargetRate_bps = 0;         /* trigger new SNR computation */
     }
 
-    psEnc->sCmn.delay = enc_delay_matrix[rateID(fs_kHz*1000)][rateID(psEnc->sCmn.API_fs_Hz)];
-    silk_assert(psEnc->sCmn.delay <= MAX_ENCODER_DELAY);
-
     /* Set internal sampling frequency */
     silk_assert( fs_kHz == 8 || fs_kHz == 12 || fs_kHz == 16 );
     silk_assert( psEnc->sCmn.nb_subfr == 2 || psEnc->sCmn.nb_subfr == 4 );
@@ -336,7 +326,7 @@ opus_int silk_setup_complexity(
         psEncC->shapingLPCOrder                 = 10;
         psEncC->la_shape                        = 5 * psEncC->fs_kHz;
         psEncC->nStatesDelayedDecision          = 1;
-        psEncC->useInterpolatedNLSFs            = 1;
+        psEncC->useInterpolatedNLSFs            = 0;
         psEncC->LTPQuantLowComplexity           = 0;
         psEncC->NLSF_MSVQ_Survivors             = 4;
         psEncC->warping_Q16                     = 0;
@@ -347,7 +337,7 @@ opus_int silk_setup_complexity(
         psEncC->shapingLPCOrder                 = 12;
         psEncC->la_shape                        = 5 * psEncC->fs_kHz;
         psEncC->nStatesDelayedDecision          = 2;
-        psEncC->useInterpolatedNLSFs            = 0;
+        psEncC->useInterpolatedNLSFs            = 1;
         psEncC->LTPQuantLowComplexity           = 0;
         psEncC->NLSF_MSVQ_Survivors             = 8;
         psEncC->warping_Q16                     = psEncC->fs_kHz * SILK_FIX_CONST( WARPING_MULTIPLIER, 16 );
@@ -358,7 +348,7 @@ opus_int silk_setup_complexity(
         psEncC->shapingLPCOrder                 = 14;
         psEncC->la_shape                        = 5 * psEncC->fs_kHz;
         psEncC->nStatesDelayedDecision          = 3;
-        psEncC->useInterpolatedNLSFs            = 0;
+        psEncC->useInterpolatedNLSFs            = 1;
         psEncC->LTPQuantLowComplexity           = 0;
         psEncC->NLSF_MSVQ_Survivors             = 16;
         psEncC->warping_Q16                     = psEncC->fs_kHz * SILK_FIX_CONST( WARPING_MULTIPLIER, 16 );