@scrypted/libav 1.0.166 → 1.0.168

package/opus/silk/control_audio_bandwidth.c

@@ -0,0 +1,132 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "main.h"
+#include "tuning_parameters.h"
+
+/* Control internal sampling rate */
+opus_int silk_control_audio_bandwidth(
+    silk_encoder_state          *psEncC,                        /* I/O  Pointer to Silk encoder state               */
+    silk_EncControlStruct       *encControl                     /* I    Control structure                           */
+)
+{
+    opus_int   fs_kHz;
+    opus_int   orig_kHz;
+    opus_int32 fs_Hz;
+
+    orig_kHz = psEncC->fs_kHz;
+    /* Handle a bandwidth-switching reset where we need to be aware what the last sampling rate was. */
+    if( orig_kHz == 0 ) {
+        orig_kHz = psEncC->sLP.saved_fs_kHz;
+    }
+    fs_kHz = orig_kHz;
+    fs_Hz = silk_SMULBB( fs_kHz, 1000 );
+    if( fs_Hz == 0 ) {
+        /* Encoder has just been initialized */
+        fs_Hz  = silk_min( psEncC->desiredInternal_fs_Hz, psEncC->API_fs_Hz );
+        fs_kHz = silk_DIV32_16( fs_Hz, 1000 );
+    } else if( fs_Hz > psEncC->API_fs_Hz || fs_Hz > psEncC->maxInternal_fs_Hz || fs_Hz < psEncC->minInternal_fs_Hz ) {
+        /* Make sure internal rate is not higher than external rate or maximum allowed, or lower than minimum allowed */
+        fs_Hz  = psEncC->API_fs_Hz;
+        fs_Hz  = silk_min( fs_Hz, psEncC->maxInternal_fs_Hz );
+        fs_Hz  = silk_max( fs_Hz, psEncC->minInternal_fs_Hz );
+        fs_kHz = silk_DIV32_16( fs_Hz, 1000 );
+    } else {
+        /* State machine for the internal sampling rate switching */
+        if( psEncC->sLP.transition_frame_no >= TRANSITION_FRAMES ) {
+            /* Stop transition phase */
+            psEncC->sLP.mode = 0;
+        }
+        if( psEncC->allow_bandwidth_switch || encControl->opusCanSwitch ) {
+            /* Check if we should switch down */
+            if( silk_SMULBB( orig_kHz, 1000 ) > psEncC->desiredInternal_fs_Hz )
+            {
+                /* Switch down */
+                if( psEncC->sLP.mode == 0 ) {
+                    /* New transition */
+                    psEncC->sLP.transition_frame_no = TRANSITION_FRAMES;
+
+                    /* Reset transition filter state */
+                    silk_memset( psEncC->sLP.In_LP_State, 0, sizeof( psEncC->sLP.In_LP_State ) );
+                }
+                if( encControl->opusCanSwitch ) {
+                    /* Stop transition phase */
+                    psEncC->sLP.mode = 0;
+
+                    /* Switch to a lower sample frequency */
+                    fs_kHz = orig_kHz == 16 ? 12 : 8;
+                } else {
+                   if( psEncC->sLP.transition_frame_no <= 0 ) {
+                       encControl->switchReady = 1;
+                       /* Make room for redundancy */
+                       encControl->maxBits -= encControl->maxBits * 5 / ( encControl->payloadSize_ms + 5 );
+                   } else {
+                       /* Direction: down (at double speed) */
+                       psEncC->sLP.mode = -2;
+                   }
+                }
+            }
+            else
+            /* Check if we should switch up */
+            if( silk_SMULBB( orig_kHz, 1000 ) < psEncC->desiredInternal_fs_Hz )
+            {
+                /* Switch up */
+                if( encControl->opusCanSwitch ) {
+                    /* Switch to a higher sample frequency */
+                    fs_kHz = orig_kHz == 8 ? 12 : 16;
+
+                    /* New transition */
+                    psEncC->sLP.transition_frame_no = 0;
+
+                    /* Reset transition filter state */
+                    silk_memset( psEncC->sLP.In_LP_State, 0, sizeof( psEncC->sLP.In_LP_State ) );
+
+                    /* Direction: up */
+                    psEncC->sLP.mode = 1;
+                } else {
+                   if( psEncC->sLP.mode == 0 ) {
+                       encControl->switchReady = 1;
+                       /* Make room for redundancy */
+                       encControl->maxBits -= encControl->maxBits * 5 / ( encControl->payloadSize_ms + 5 );
+                   } else {
+                       /* Direction: up */
+                       psEncC->sLP.mode = 1;
+                   }
+                }
+            } else {
+               if (psEncC->sLP.mode<0)
+                  psEncC->sLP.mode = 1;
+            }
+        }
+    }
+
+    return fs_kHz;
+}

package/opus/silk/control_codec.c

@@ -0,0 +1,423 @@
+/***********************************************************************
+Copyright (c) 2006-2011, Skype Limited. All rights reserved.
+Redistribution and use in source and binary forms, with or without
+modification, are permitted provided that the following conditions
+are met:
+- Redistributions of source code must retain the above copyright notice,
+this list of conditions and the following disclaimer.
+- Redistributions in binary form must reproduce the above copyright
+notice, this list of conditions and the following disclaimer in the
+documentation and/or other materials provided with the distribution.
+- Neither the name of Internet Society, IETF or IETF Trust, nor the
+names of specific contributors, may be used to endorse or promote
+products derived from this software without specific prior written
+permission.
+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
+AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
+LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
+CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
+SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
+INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
+CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
+ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
+POSSIBILITY OF SUCH DAMAGE.
+***********************************************************************/
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+#ifdef FIXED_POINT
+#include "main_FIX.h"
+#define silk_encoder_state_Fxx      silk_encoder_state_FIX
+#else
+#include "main_FLP.h"
+#define silk_encoder_state_Fxx      silk_encoder_state_FLP
+#endif
+#include "stack_alloc.h"
+#include "tuning_parameters.h"
+#include "pitch_est_defines.h"
+
+static opus_int silk_setup_resamplers(
+    silk_encoder_state_Fxx          *psEnc,             /* I/O                      */
+    opus_int                        fs_kHz              /* I                        */
+);
+
+static opus_int silk_setup_fs(
+    silk_encoder_state_Fxx          *psEnc,             /* I/O                      */
+    opus_int                        fs_kHz,             /* I                        */
+    opus_int                        PacketSize_ms       /* I                        */
+);
+
+static opus_int silk_setup_complexity(
+    silk_encoder_state              *psEncC,            /* I/O                      */
+    opus_int                        Complexity          /* I                        */
+);
+
+static OPUS_INLINE opus_int silk_setup_LBRR(
+    silk_encoder_state              *psEncC,            /* I/O                      */
+    const silk_EncControlStruct     *encControl         /* I                        */
+);
+
+
+/* Control encoder */
+opus_int silk_control_encoder(
+    silk_encoder_state_Fxx          *psEnc,                                 /* I/O  Pointer to Silk encoder state                                               */
+    silk_EncControlStruct           *encControl,                            /* I    Control structure                                                           */
+    const opus_int                  allow_bw_switch,                        /* I    Flag to allow switching audio bandwidth                                     */
+    const opus_int                  channelNb,                              /* I    Channel number                                                              */
+    const opus_int                  force_fs_kHz
+)
+{
+    opus_int   fs_kHz, ret = 0;
+
+    psEnc->sCmn.useDTX                 = encControl->useDTX;
+    psEnc->sCmn.useCBR                 = encControl->useCBR;
+    psEnc->sCmn.API_fs_Hz              = encControl->API_sampleRate;
+    psEnc->sCmn.maxInternal_fs_Hz      = encControl->maxInternalSampleRate;
+    psEnc->sCmn.minInternal_fs_Hz      = encControl->minInternalSampleRate;
+    psEnc->sCmn.desiredInternal_fs_Hz  = encControl->desiredInternalSampleRate;
+    psEnc->sCmn.useInBandFEC           = encControl->useInBandFEC;
+    psEnc->sCmn.nChannelsAPI           = encControl->nChannelsAPI;
+    psEnc->sCmn.nChannelsInternal      = encControl->nChannelsInternal;
+    psEnc->sCmn.allow_bandwidth_switch = allow_bw_switch;
+    psEnc->sCmn.channelNb              = channelNb;
+
+    if( psEnc->sCmn.controlled_since_last_payload != 0 && psEnc->sCmn.prefillFlag == 0 ) {
+        if( psEnc->sCmn.API_fs_Hz != psEnc->sCmn.prev_API_fs_Hz && psEnc->sCmn.fs_kHz > 0 ) {
+            /* Change in API sampling rate in the middle of encoding a packet */
+            ret += silk_setup_resamplers( psEnc, psEnc->sCmn.fs_kHz );
+        }
+        return ret;
+    }
+
+    /* Beyond this point we know that there are no previously coded frames in the payload buffer */
+
+    /********************************************/
+    /* Determine internal sampling rate         */
+    /********************************************/
+    fs_kHz = silk_control_audio_bandwidth( &psEnc->sCmn, encControl );
+    if( force_fs_kHz ) {
+       fs_kHz = force_fs_kHz;
+    }
+    /********************************************/
+    /* Prepare resampler and buffered data      */
+    /********************************************/
+    ret += silk_setup_resamplers( psEnc, fs_kHz );
+
+    /********************************************/
+    /* Set internal sampling frequency          */
+    /********************************************/
+    ret += silk_setup_fs( psEnc, fs_kHz, encControl->payloadSize_ms );
+
+    /********************************************/
+    /* Set encoding complexity                  */
+    /********************************************/
+    ret += silk_setup_complexity( &psEnc->sCmn, encControl->complexity  );
+
+    /********************************************/
+    /* Set packet loss rate measured by farend  */
+    /********************************************/
+    psEnc->sCmn.PacketLoss_perc = encControl->packetLossPercentage;
+
+    /********************************************/
+    /* Set LBRR usage                           */
+    /********************************************/
+    ret += silk_setup_LBRR( &psEnc->sCmn, encControl );
+
+    psEnc->sCmn.controlled_since_last_payload = 1;
+
+    return ret;
+}
+
+static opus_int silk_setup_resamplers(
+    silk_encoder_state_Fxx          *psEnc,             /* I/O                      */
+    opus_int                         fs_kHz              /* I                        */
+)
+{
+    opus_int   ret = SILK_NO_ERROR;
+    SAVE_STACK;
+
+    if( psEnc->sCmn.fs_kHz != fs_kHz || psEnc->sCmn.prev_API_fs_Hz != psEnc->sCmn.API_fs_Hz )
+    {
+        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, 1 );
+        } else {
+            VARDECL( opus_int16, x_buf_API_fs_Hz );
+            VARDECL( silk_resampler_state_struct, temp_resampler_state );
+#ifdef FIXED_POINT
+            opus_int16 *x_bufFIX = psEnc->x_buf;
+#else
+            VARDECL( opus_int16, x_bufFIX );
+            opus_int32 new_buf_samples;
+#endif
+            opus_int32 api_buf_samples;
+            opus_int32 old_buf_samples;
+            opus_int32 buf_length_ms;
+
+            buf_length_ms = silk_LSHIFT( psEnc->sCmn.nb_subfr * 5, 1 ) + LA_SHAPE_MS;
+            old_buf_samples = buf_length_ms * psEnc->sCmn.fs_kHz;
+
+#ifndef FIXED_POINT
+            new_buf_samples = buf_length_ms * fs_kHz;
+            ALLOC( x_bufFIX, silk_max( old_buf_samples, new_buf_samples ),
+                   opus_int16 );
+            silk_float2short_array( x_bufFIX, psEnc->x_buf, old_buf_samples );
+#endif
+
+            /* Initialize resampler for temporary resampling of x_buf data to API_fs_Hz */
+            ALLOC( temp_resampler_state, 1, silk_resampler_state_struct );
+            ret += silk_resampler_init( temp_resampler_state, silk_SMULBB( psEnc->sCmn.fs_kHz, 1000 ), psEnc->sCmn.API_fs_Hz, 0 );
+
+            /* Calculate number of samples to temporarily upsample */
+            api_buf_samples = buf_length_ms * silk_DIV32_16( psEnc->sCmn.API_fs_Hz, 1000 );
+
+            /* Temporary resampling of x_buf data to API_fs_Hz */
+            ALLOC( x_buf_API_fs_Hz, api_buf_samples, opus_int16 );
+            ret += silk_resampler( temp_resampler_state, x_buf_API_fs_Hz, x_bufFIX, old_buf_samples );
+
+            /* 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 ), 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, api_buf_samples );
+
+#ifndef FIXED_POINT
+            silk_short2float_array( psEnc->x_buf, x_bufFIX, new_buf_samples);
+#endif
+        }
+    }
+
+    psEnc->sCmn.prev_API_fs_Hz = psEnc->sCmn.API_fs_Hz;
+
+    RESTORE_STACK;
+    return ret;
+}
+
+static opus_int silk_setup_fs(
+    silk_encoder_state_Fxx          *psEnc,             /* I/O                      */
+    opus_int                        fs_kHz,             /* I                        */
+    opus_int                        PacketSize_ms       /* I                        */
+)
+{
+    opus_int ret = SILK_NO_ERROR;
+
+    /* Set packet size */
+    if( PacketSize_ms != psEnc->sCmn.PacketSize_ms ) {
+        if( ( PacketSize_ms !=  10 ) &&
+            ( PacketSize_ms !=  20 ) &&
+            ( PacketSize_ms !=  40 ) &&
+            ( PacketSize_ms !=  60 ) ) {
+            ret = SILK_ENC_PACKET_SIZE_NOT_SUPPORTED;
+        }
+        if( PacketSize_ms <= 10 ) {
+            psEnc->sCmn.nFramesPerPacket = 1;
+            psEnc->sCmn.nb_subfr = PacketSize_ms == 10 ? 2 : 1;
+            psEnc->sCmn.frame_length = silk_SMULBB( PacketSize_ms, fs_kHz );
+            psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS_2_SF, fs_kHz );
+            if( psEnc->sCmn.fs_kHz == 8 ) {
+                psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_NB_iCDF;
+            } else {
+                psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_iCDF;
+            }
+        } else {
+            psEnc->sCmn.nFramesPerPacket = silk_DIV32_16( PacketSize_ms, MAX_FRAME_LENGTH_MS );
+            psEnc->sCmn.nb_subfr = MAX_NB_SUBFR;
+            psEnc->sCmn.frame_length = silk_SMULBB( 20, fs_kHz );
+            psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS, fs_kHz );
+            if( psEnc->sCmn.fs_kHz == 8 ) {
+                psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_NB_iCDF;
+            } else {
+                psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_iCDF;
+            }
+        }
+        psEnc->sCmn.PacketSize_ms  = PacketSize_ms;
+        psEnc->sCmn.TargetRate_bps = 0;         /* trigger new SNR computation */
+    }
+
+    /* Set internal sampling frequency */
+    celt_assert( fs_kHz == 8 || fs_kHz == 12 || fs_kHz == 16 );
+    celt_assert( psEnc->sCmn.nb_subfr == 2 || psEnc->sCmn.nb_subfr == 4 );
+    if( psEnc->sCmn.fs_kHz != fs_kHz ) {
+        /* reset part of the state */
+        silk_memset( &psEnc->sShape,               0, sizeof( psEnc->sShape ) );
+        silk_memset( &psEnc->sCmn.sNSQ,            0, sizeof( psEnc->sCmn.sNSQ ) );
+        silk_memset( psEnc->sCmn.prev_NLSFq_Q15,   0, sizeof( psEnc->sCmn.prev_NLSFq_Q15 ) );
+        silk_memset( &psEnc->sCmn.sLP.In_LP_State, 0, sizeof( psEnc->sCmn.sLP.In_LP_State ) );
+        psEnc->sCmn.inputBufIx                  = 0;
+        psEnc->sCmn.nFramesEncoded              = 0;
+        psEnc->sCmn.TargetRate_bps              = 0;     /* trigger new SNR computation */
+
+        /* Initialize non-zero parameters */
+        psEnc->sCmn.prevLag                     = 100;
+        psEnc->sCmn.first_frame_after_reset     = 1;
+        psEnc->sShape.LastGainIndex             = 10;
+        psEnc->sCmn.sNSQ.lagPrev                = 100;
+        psEnc->sCmn.sNSQ.prev_gain_Q16          = 65536;
+        psEnc->sCmn.prevSignalType              = TYPE_NO_VOICE_ACTIVITY;
+
+        psEnc->sCmn.fs_kHz = fs_kHz;
+        if( psEnc->sCmn.fs_kHz == 8 ) {
+            if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) {
+                psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_NB_iCDF;
+            } else {
+                psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_NB_iCDF;
+            }
+        } else {
+            if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) {
+                psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_iCDF;
+            } else {
+                psEnc->sCmn.pitch_contour_iCDF = silk_pitch_contour_10_ms_iCDF;
+            }
+        }
+        if( psEnc->sCmn.fs_kHz == 8 || psEnc->sCmn.fs_kHz == 12 ) {
+            psEnc->sCmn.predictLPCOrder = MIN_LPC_ORDER;
+            psEnc->sCmn.psNLSF_CB  = &silk_NLSF_CB_NB_MB;
+        } else {
+            psEnc->sCmn.predictLPCOrder = MAX_LPC_ORDER;
+            psEnc->sCmn.psNLSF_CB  = &silk_NLSF_CB_WB;
+        }
+        psEnc->sCmn.subfr_length   = SUB_FRAME_LENGTH_MS * fs_kHz;
+        psEnc->sCmn.frame_length   = silk_SMULBB( psEnc->sCmn.subfr_length, psEnc->sCmn.nb_subfr );
+        psEnc->sCmn.ltp_mem_length = silk_SMULBB( LTP_MEM_LENGTH_MS, fs_kHz );
+        psEnc->sCmn.la_pitch       = silk_SMULBB( LA_PITCH_MS, fs_kHz );
+        psEnc->sCmn.max_pitch_lag  = silk_SMULBB( 18, fs_kHz );
+        if( psEnc->sCmn.nb_subfr == MAX_NB_SUBFR ) {
+            psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS, fs_kHz );
+        } else {
+            psEnc->sCmn.pitch_LPC_win_length = silk_SMULBB( FIND_PITCH_LPC_WIN_MS_2_SF, fs_kHz );
+        }
+        if( psEnc->sCmn.fs_kHz == 16 ) {
+            psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform8_iCDF;
+        } else if( psEnc->sCmn.fs_kHz == 12 ) {
+            psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform6_iCDF;
+        } else {
+            psEnc->sCmn.pitch_lag_low_bits_iCDF = silk_uniform4_iCDF;
+        }
+    }
+
+    /* Check that settings are valid */
+    celt_assert( ( psEnc->sCmn.subfr_length * psEnc->sCmn.nb_subfr ) == psEnc->sCmn.frame_length );
+
+    return ret;
+}
+
+static opus_int silk_setup_complexity(
+    silk_encoder_state              *psEncC,            /* I/O                      */
+    opus_int                        Complexity          /* I                        */
+)
+{
+    opus_int ret = 0;
+
+    /* Set encoding complexity */
+    celt_assert( Complexity >= 0 && Complexity <= 10 );
+    if( Complexity < 1 ) {
+        psEncC->pitchEstimationComplexity       = SILK_PE_MIN_COMPLEX;
+        psEncC->pitchEstimationThreshold_Q16    = SILK_FIX_CONST( 0.8, 16 );
+        psEncC->pitchEstimationLPCOrder         = 6;
+        psEncC->shapingLPCOrder                 = 12;
+        psEncC->la_shape                        = 3 * psEncC->fs_kHz;
+        psEncC->nStatesDelayedDecision          = 1;
+        psEncC->useInterpolatedNLSFs            = 0;
+        psEncC->NLSF_MSVQ_Survivors             = 2;
+        psEncC->warping_Q16                     = 0;
+    } else if( Complexity < 2 ) {
+        psEncC->pitchEstimationComplexity       = SILK_PE_MID_COMPLEX;
+        psEncC->pitchEstimationThreshold_Q16    = SILK_FIX_CONST( 0.76, 16 );
+        psEncC->pitchEstimationLPCOrder         = 8;
+        psEncC->shapingLPCOrder                 = 14;
+        psEncC->la_shape                        = 5 * psEncC->fs_kHz;
+        psEncC->nStatesDelayedDecision          = 1;
+        psEncC->useInterpolatedNLSFs            = 0;
+        psEncC->NLSF_MSVQ_Survivors             = 3;
+        psEncC->warping_Q16                     = 0;
+    } else if( Complexity < 3 ) {
+        psEncC->pitchEstimationComplexity       = SILK_PE_MIN_COMPLEX;
+        psEncC->pitchEstimationThreshold_Q16    = SILK_FIX_CONST( 0.8, 16 );
+        psEncC->pitchEstimationLPCOrder         = 6;
+        psEncC->shapingLPCOrder                 = 12;
+        psEncC->la_shape                        = 3 * psEncC->fs_kHz;
+        psEncC->nStatesDelayedDecision          = 2;
+        psEncC->useInterpolatedNLSFs            = 0;
+        psEncC->NLSF_MSVQ_Survivors             = 2;
+        psEncC->warping_Q16                     = 0;
+    } else if( Complexity < 4 ) {
+        psEncC->pitchEstimationComplexity       = SILK_PE_MID_COMPLEX;
+        psEncC->pitchEstimationThreshold_Q16    = SILK_FIX_CONST( 0.76, 16 );
+        psEncC->pitchEstimationLPCOrder         = 8;
+        psEncC->shapingLPCOrder                 = 14;
+        psEncC->la_shape                        = 5 * psEncC->fs_kHz;
+        psEncC->nStatesDelayedDecision          = 2;
+        psEncC->useInterpolatedNLSFs            = 0;
+        psEncC->NLSF_MSVQ_Survivors             = 4;
+        psEncC->warping_Q16                     = 0;
+    } else if( Complexity < 6 ) {
+        psEncC->pitchEstimationComplexity       = SILK_PE_MID_COMPLEX;
+        psEncC->pitchEstimationThreshold_Q16    = SILK_FIX_CONST( 0.74, 16 );
+        psEncC->pitchEstimationLPCOrder         = 10;
+        psEncC->shapingLPCOrder                 = 16;
+        psEncC->la_shape                        = 5 * psEncC->fs_kHz;
+        psEncC->nStatesDelayedDecision          = 2;
+        psEncC->useInterpolatedNLSFs            = 1;
+        psEncC->NLSF_MSVQ_Survivors             = 6;
+        psEncC->warping_Q16                     = psEncC->fs_kHz * SILK_FIX_CONST( WARPING_MULTIPLIER, 16 );
+    } else if( Complexity < 8 ) {
+        psEncC->pitchEstimationComplexity       = SILK_PE_MID_COMPLEX;
+        psEncC->pitchEstimationThreshold_Q16    = SILK_FIX_CONST( 0.72, 16 );
+        psEncC->pitchEstimationLPCOrder         = 12;
+        psEncC->shapingLPCOrder                 = 20;
+        psEncC->la_shape                        = 5 * psEncC->fs_kHz;
+        psEncC->nStatesDelayedDecision          = 3;
+        psEncC->useInterpolatedNLSFs            = 1;
+        psEncC->NLSF_MSVQ_Survivors             = 8;
+        psEncC->warping_Q16                     = psEncC->fs_kHz * SILK_FIX_CONST( WARPING_MULTIPLIER, 16 );
+    } else {
+        psEncC->pitchEstimationComplexity       = SILK_PE_MAX_COMPLEX;
+        psEncC->pitchEstimationThreshold_Q16    = SILK_FIX_CONST( 0.7, 16 );
+        psEncC->pitchEstimationLPCOrder         = 16;
+        psEncC->shapingLPCOrder                 = 24;
+        psEncC->la_shape                        = 5 * psEncC->fs_kHz;
+        psEncC->nStatesDelayedDecision          = MAX_DEL_DEC_STATES;
+        psEncC->useInterpolatedNLSFs            = 1;
+        psEncC->NLSF_MSVQ_Survivors             = 16;
+        psEncC->warping_Q16                     = psEncC->fs_kHz * SILK_FIX_CONST( WARPING_MULTIPLIER, 16 );
+    }
+
+    /* Do not allow higher pitch estimation LPC order than predict LPC order */
+    psEncC->pitchEstimationLPCOrder = silk_min_int( psEncC->pitchEstimationLPCOrder, psEncC->predictLPCOrder );
+    psEncC->shapeWinLength          = SUB_FRAME_LENGTH_MS * psEncC->fs_kHz + 2 * psEncC->la_shape;
+    psEncC->Complexity              = Complexity;
+
+    celt_assert( psEncC->pitchEstimationLPCOrder <= MAX_FIND_PITCH_LPC_ORDER );
+    celt_assert( psEncC->shapingLPCOrder         <= MAX_SHAPE_LPC_ORDER      );
+    celt_assert( psEncC->nStatesDelayedDecision  <= MAX_DEL_DEC_STATES       );
+    celt_assert( psEncC->warping_Q16             <= 32767                    );
+    celt_assert( psEncC->la_shape                <= LA_SHAPE_MAX             );
+    celt_assert( psEncC->shapeWinLength          <= SHAPE_LPC_WIN_MAX        );
+
+    return ret;
+}
+
+static OPUS_INLINE opus_int silk_setup_LBRR(
+    silk_encoder_state          *psEncC,            /* I/O                      */
+    const silk_EncControlStruct *encControl         /* I                        */
+)
+{
+    opus_int   LBRR_in_previous_packet, ret = SILK_NO_ERROR;
+
+    LBRR_in_previous_packet = psEncC->LBRR_enabled;
+    psEncC->LBRR_enabled = encControl->LBRR_coded;
+    if( psEncC->LBRR_enabled ) {
+        /* Set gain increase for coding LBRR excitation */
+        if( LBRR_in_previous_packet == 0 ) {
+            /* Previous packet did not have LBRR, and was therefore coded at a higher bitrate */
+            psEncC->LBRR_GainIncreases = 7;
+        } else {
+            psEncC->LBRR_GainIncreases = silk_max_int( 7 - silk_SMULWB( (opus_int32)psEncC->PacketLoss_perc, SILK_FIX_CONST( 0.2, 16 ) ), 3 );
+        }
+    }
+
+    return ret;
+}