@scrypted/libav 1.0.166 → 1.0.168

package/opus/silk/enc_API.c

@@ -0,0 +1,593 @@
+/***********************************************************************
+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 "define.h"
+#include "API.h"
+#include "control.h"
+#include "typedef.h"
+#include "stack_alloc.h"
+#include "structs.h"
+#include "tuning_parameters.h"
+#ifdef FIXED_POINT
+#include "main_FIX.h"
+#else
+#include "main_FLP.h"
+#endif
+
+#ifdef ENABLE_DRED
+#include "dred_encoder.h"
+#endif
+
+/***************************************/
+/* Read control structure from encoder */
+/***************************************/
+static opus_int silk_QueryEncoder(                      /* O    Returns error code                              */
+    const void                      *encState,          /* I    State                                           */
+    silk_EncControlStruct           *encStatus          /* O    Encoder Status                                  */
+);
+
+/****************************************/
+/* Encoder functions                    */
+/****************************************/
+
+opus_int silk_Get_Encoder_Size(                         /* O    Returns error code                              */
+    opus_int                        *encSizeBytes       /* O    Number of bytes in SILK encoder state           */
+)
+{
+    opus_int ret = SILK_NO_ERROR;
+
+    *encSizeBytes = sizeof( silk_encoder );
+
+    return ret;
+}
+
+/*************************/
+/* Init or Reset encoder */
+/*************************/
+opus_int silk_InitEncoder(                              /* O    Returns error code                              */
+    void                            *encState,          /* I/O  State                                           */
+    int                              arch,              /* I    Run-time architecture                           */
+    silk_EncControlStruct           *encStatus          /* O    Encoder Status                                  */
+)
+{
+    silk_encoder *psEnc;
+    opus_int n, ret = SILK_NO_ERROR;
+
+    psEnc = (silk_encoder *)encState;
+
+    /* Reset encoder */
+    silk_memset( psEnc, 0, sizeof( silk_encoder ) );
+    for( n = 0; n < ENCODER_NUM_CHANNELS; n++ ) {
+        if( ret += silk_init_encoder( &psEnc->state_Fxx[ n ], arch ) ) {
+            celt_assert( 0 );
+        }
+    }
+
+    psEnc->nChannelsAPI = 1;
+    psEnc->nChannelsInternal = 1;
+
+    /* Read control structure */
+    if( ret += silk_QueryEncoder( encState, encStatus ) ) {
+        celt_assert( 0 );
+    }
+
+    return ret;
+}
+
+/***************************************/
+/* Read control structure from encoder */
+/***************************************/
+static opus_int silk_QueryEncoder(                      /* O    Returns error code                              */
+    const void                      *encState,          /* I    State                                           */
+    silk_EncControlStruct           *encStatus          /* O    Encoder Status                                  */
+)
+{
+    opus_int ret = SILK_NO_ERROR;
+    silk_encoder_state_Fxx *state_Fxx;
+    silk_encoder *psEnc = (silk_encoder *)encState;
+
+    state_Fxx = psEnc->state_Fxx;
+
+    encStatus->nChannelsAPI              = psEnc->nChannelsAPI;
+    encStatus->nChannelsInternal         = psEnc->nChannelsInternal;
+    encStatus->API_sampleRate            = state_Fxx[ 0 ].sCmn.API_fs_Hz;
+    encStatus->maxInternalSampleRate     = state_Fxx[ 0 ].sCmn.maxInternal_fs_Hz;
+    encStatus->minInternalSampleRate     = state_Fxx[ 0 ].sCmn.minInternal_fs_Hz;
+    encStatus->desiredInternalSampleRate = state_Fxx[ 0 ].sCmn.desiredInternal_fs_Hz;
+    encStatus->payloadSize_ms            = state_Fxx[ 0 ].sCmn.PacketSize_ms;
+    encStatus->bitRate                   = state_Fxx[ 0 ].sCmn.TargetRate_bps;
+    encStatus->packetLossPercentage      = state_Fxx[ 0 ].sCmn.PacketLoss_perc;
+    encStatus->complexity                = state_Fxx[ 0 ].sCmn.Complexity;
+    encStatus->useInBandFEC              = state_Fxx[ 0 ].sCmn.useInBandFEC;
+    encStatus->useDTX                    = state_Fxx[ 0 ].sCmn.useDTX;
+    encStatus->useCBR                    = state_Fxx[ 0 ].sCmn.useCBR;
+    encStatus->internalSampleRate        = silk_SMULBB( state_Fxx[ 0 ].sCmn.fs_kHz, 1000 );
+    encStatus->allowBandwidthSwitch      = state_Fxx[ 0 ].sCmn.allow_bandwidth_switch;
+    encStatus->inWBmodeWithoutVariableLP = state_Fxx[ 0 ].sCmn.fs_kHz == 16 && state_Fxx[ 0 ].sCmn.sLP.mode == 0;
+
+    return ret;
+}
+
+
+/**************************/
+/* Encode frame with Silk */
+/**************************/
+/* Note: if prefillFlag is set, the input must contain 10 ms of audio, irrespective of what                     */
+/* encControl->payloadSize_ms is set to                                                                         */
+opus_int silk_Encode(                                   /* O    Returns error code                              */
+    void                            *encState,          /* I/O  State                                           */
+    silk_EncControlStruct           *encControl,        /* I    Control status                                  */
+    const opus_res                  *samplesIn,         /* I    Speech sample input vector                      */
+    opus_int                        nSamplesIn,         /* I    Number of samples in input vector               */
+    ec_enc                          *psRangeEnc,        /* I/O  Compressor data structure                       */
+    opus_int32                      *nBytesOut,         /* I/O  Number of bytes in payload (input: Max bytes)   */
+    const opus_int                  prefillFlag,        /* I    Flag to indicate prefilling buffers no coding   */
+    opus_int                        activity            /* I    Decision of Opus voice activity detector        */
+)
+{
+    opus_int   n, i, nBits, flags, tmp_payloadSize_ms = 0, tmp_complexity = 0, ret = 0;
+    opus_int   nSamplesToBuffer, nSamplesToBufferMax, nBlocksOf10ms;
+    opus_int   nSamplesFromInput = 0, nSamplesFromInputMax;
+    opus_int   speech_act_thr_for_switch_Q8;
+    opus_int32 TargetRate_bps, MStargetRates_bps[ 2 ], channelRate_bps, LBRR_symbol, sum;
+    silk_encoder *psEnc = ( silk_encoder * )encState;
+    VARDECL( opus_int16, buf );
+    opus_int transition, curr_block, tot_blocks;
+    SAVE_STACK;
+
+    if (encControl->reducedDependency)
+    {
+       psEnc->state_Fxx[0].sCmn.first_frame_after_reset = 1;
+       psEnc->state_Fxx[1].sCmn.first_frame_after_reset = 1;
+    }
+    psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded = psEnc->state_Fxx[ 1 ].sCmn.nFramesEncoded = 0;
+
+    /* Check values in encoder control structure */
+    if( ( ret = check_control_input( encControl ) ) != 0 ) {
+        celt_assert( 0 );
+        RESTORE_STACK;
+        return ret;
+    }
+
+    encControl->switchReady = 0;
+
+    if( encControl->nChannelsInternal > psEnc->nChannelsInternal ) {
+        /* Mono -> Stereo transition: init state of second channel and stereo state */
+        ret += silk_init_encoder( &psEnc->state_Fxx[ 1 ], psEnc->state_Fxx[ 0 ].sCmn.arch );
+        silk_memset( psEnc->sStereo.pred_prev_Q13, 0, sizeof( psEnc->sStereo.pred_prev_Q13 ) );
+        silk_memset( psEnc->sStereo.sSide, 0, sizeof( psEnc->sStereo.sSide ) );
+        psEnc->sStereo.mid_side_amp_Q0[ 0 ] = 0;
+        psEnc->sStereo.mid_side_amp_Q0[ 1 ] = 1;
+        psEnc->sStereo.mid_side_amp_Q0[ 2 ] = 0;
+        psEnc->sStereo.mid_side_amp_Q0[ 3 ] = 1;
+        psEnc->sStereo.width_prev_Q14 = 0;
+        psEnc->sStereo.smth_width_Q14 = SILK_FIX_CONST( 1, 14 );
+        if( psEnc->nChannelsAPI == 2 ) {
+            silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.resampler_state, &psEnc->state_Fxx[ 0 ].sCmn.resampler_state, sizeof( silk_resampler_state_struct ) );
+            silk_memcpy( &psEnc->state_Fxx[ 1 ].sCmn.In_HP_State,     &psEnc->state_Fxx[ 0 ].sCmn.In_HP_State,     sizeof( psEnc->state_Fxx[ 1 ].sCmn.In_HP_State ) );
+        }
+    }
+
+    transition = (encControl->payloadSize_ms != psEnc->state_Fxx[ 0 ].sCmn.PacketSize_ms) || (psEnc->nChannelsInternal != encControl->nChannelsInternal);
+
+    psEnc->nChannelsAPI = encControl->nChannelsAPI;
+    psEnc->nChannelsInternal = encControl->nChannelsInternal;
+
+    nBlocksOf10ms = silk_DIV32( 100 * nSamplesIn, encControl->API_sampleRate );
+    tot_blocks = ( nBlocksOf10ms > 1 ) ? nBlocksOf10ms >> 1 : 1;
+    curr_block = 0;
+    if( prefillFlag ) {
+        silk_LP_state save_LP;
+        /* Only accept input length of 10 ms */
+        if( nBlocksOf10ms != 1 ) {
+            celt_assert( 0 );
+            RESTORE_STACK;
+            return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
+        }
+        if ( prefillFlag == 2 ) {
+            save_LP = psEnc->state_Fxx[ 0 ].sCmn.sLP;
+            /* Save the sampling rate so the bandwidth switching code can keep handling transitions. */
+            save_LP.saved_fs_kHz = psEnc->state_Fxx[ 0 ].sCmn.fs_kHz;
+        }
+        /* Reset Encoder */
+        for( n = 0; n < encControl->nChannelsInternal; n++ ) {
+            ret = silk_init_encoder( &psEnc->state_Fxx[ n ], psEnc->state_Fxx[ n ].sCmn.arch );
+            /* Restore the variable LP state. */
+            if ( prefillFlag == 2 ) {
+                psEnc->state_Fxx[ n ].sCmn.sLP = save_LP;
+            }
+            celt_assert( !ret );
+        }
+        tmp_payloadSize_ms = encControl->payloadSize_ms;
+        encControl->payloadSize_ms = 10;
+        tmp_complexity = encControl->complexity;
+        encControl->complexity = 0;
+        for( n = 0; n < encControl->nChannelsInternal; n++ ) {
+            psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0;
+            psEnc->state_Fxx[ n ].sCmn.prefillFlag = 1;
+        }
+    } else {
+        /* Only accept input lengths that are a multiple of 10 ms */
+        if( nBlocksOf10ms * encControl->API_sampleRate != 100 * nSamplesIn || nSamplesIn < 0 ) {
+            celt_assert( 0 );
+            RESTORE_STACK;
+            return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
+        }
+        /* Make sure no more than one packet can be produced */
+        if( 1000 * (opus_int32)nSamplesIn > encControl->payloadSize_ms * encControl->API_sampleRate ) {
+            celt_assert( 0 );
+            RESTORE_STACK;
+            return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
+        }
+    }
+
+    for( n = 0; n < encControl->nChannelsInternal; n++ ) {
+        /* Force the side channel to the same rate as the mid */
+        opus_int force_fs_kHz = (n==1) ? psEnc->state_Fxx[0].sCmn.fs_kHz : 0;
+        if( ( ret = silk_control_encoder( &psEnc->state_Fxx[ n ], encControl, psEnc->allowBandwidthSwitch, n, force_fs_kHz ) ) != 0 ) {
+            silk_assert( 0 );
+            RESTORE_STACK;
+            return ret;
+        }
+        if( psEnc->state_Fxx[n].sCmn.first_frame_after_reset || transition ) {
+            for( i = 0; i < psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket; i++ ) {
+                psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ] = 0;
+            }
+        }
+        psEnc->state_Fxx[ n ].sCmn.inDTX = psEnc->state_Fxx[ n ].sCmn.useDTX;
+    }
+    celt_assert( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 0 ].sCmn.fs_kHz == psEnc->state_Fxx[ 1 ].sCmn.fs_kHz );
+
+    /* Input buffering/resampling and encoding */
+    nSamplesToBufferMax =
+        10 * nBlocksOf10ms * psEnc->state_Fxx[ 0 ].sCmn.fs_kHz;
+    nSamplesFromInputMax =
+        silk_DIV32_16( nSamplesToBufferMax *
+                           psEnc->state_Fxx[ 0 ].sCmn.API_fs_Hz,
+                       psEnc->state_Fxx[ 0 ].sCmn.fs_kHz * 1000 );
+    ALLOC( buf, nSamplesFromInputMax, opus_int16 );
+    while( 1 ) {
+        int curr_nBitsUsedLBRR = 0;
+        nSamplesToBuffer  = psEnc->state_Fxx[ 0 ].sCmn.frame_length - psEnc->state_Fxx[ 0 ].sCmn.inputBufIx;
+        nSamplesToBuffer  = silk_min( nSamplesToBuffer, nSamplesToBufferMax );
+        nSamplesFromInput = silk_DIV32_16( nSamplesToBuffer * psEnc->state_Fxx[ 0 ].sCmn.API_fs_Hz, psEnc->state_Fxx[ 0 ].sCmn.fs_kHz * 1000 );
+        /* Resample and write to buffer */
+        if( encControl->nChannelsAPI == 2 && encControl->nChannelsInternal == 2 ) {
+            opus_int id = psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded;
+            for( n = 0; n < nSamplesFromInput; n++ ) {
+                buf[ n ] = RES2INT16(samplesIn[ 2 * n ]);
+            }
+            /* 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));
+            }
+
+            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 );
+            psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer;
+
+            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 ] = RES2INT16(samplesIn[ 2 * n + 1 ]);
+            }
+            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 );
+
+            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++ ) {
+                sum = RES2INT16(samplesIn[ 2 * n ] + samplesIn[ 2 * n + 1 ]);
+                buf[ n ] = (opus_int16)silk_RSHIFT_ROUND( sum,  1 );
+            }
+            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  */
+            if( psEnc->nPrevChannelsInternal == 2 && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == 0 ) {
+               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 );
+               for( n = 0; n < psEnc->state_Fxx[ 0 ].sCmn.frame_length; n++ ) {
+                  psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.inputBufIx+n+2 ] =
+                        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);
+               }
+            }
+            psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer;
+        } else {
+            celt_assert( encControl->nChannelsAPI == 1 && encControl->nChannelsInternal == 1 );
+            for( n = 0; n < nSamplesFromInput; n++ ) {
+                buf[n] = RES2INT16(samplesIn[n]);
+            }
+            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 );
+            psEnc->state_Fxx[ 0 ].sCmn.inputBufIx += nSamplesToBuffer;
+        }
+
+        samplesIn  += nSamplesFromInput * encControl->nChannelsAPI;
+        nSamplesIn -= nSamplesFromInput;
+
+        /* Default */
+        psEnc->allowBandwidthSwitch = 0;
+
+        /* Silk encoder */
+        if( psEnc->state_Fxx[ 0 ].sCmn.inputBufIx >= psEnc->state_Fxx[ 0 ].sCmn.frame_length ) {
+            /* Enough data in input buffer, so encode */
+            celt_assert( psEnc->state_Fxx[ 0 ].sCmn.inputBufIx == psEnc->state_Fxx[ 0 ].sCmn.frame_length );
+            celt_assert( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 1 ].sCmn.inputBufIx == psEnc->state_Fxx[ 1 ].sCmn.frame_length );
+
+            /* Deal with LBRR data */
+            if( psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == 0 && !prefillFlag ) {
+                /* Create space at start of payload for VAD and FEC flags */
+                opus_uint8 iCDF[ 2 ] = { 0, 0 };
+                iCDF[ 0 ] = 256 - silk_RSHIFT( 256, ( psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket + 1 ) * encControl->nChannelsInternal );
+                ec_enc_icdf( psRangeEnc, 0, iCDF, 8 );
+                curr_nBitsUsedLBRR = ec_tell( psRangeEnc );
+
+                /* Encode any LBRR data from previous packet */
+                /* Encode LBRR flags */
+                for( n = 0; n < encControl->nChannelsInternal; n++ ) {
+                    LBRR_symbol = 0;
+                    for( i = 0; i < psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket; i++ ) {
+                        LBRR_symbol |= silk_LSHIFT( psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ], i );
+                    }
+                    psEnc->state_Fxx[ n ].sCmn.LBRR_flag = LBRR_symbol > 0 ? 1 : 0;
+                    if( LBRR_symbol && psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket > 1 ) {
+                        ec_enc_icdf( psRangeEnc, LBRR_symbol - 1, silk_LBRR_flags_iCDF_ptr[ psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket - 2 ], 8 );
+                    }
+                }
+
+                /* Code LBRR indices and excitation signals */
+                for( i = 0; i < psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket; i++ ) {
+                    for( n = 0; n < encControl->nChannelsInternal; n++ ) {
+                        if( psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i ] ) {
+                            opus_int condCoding;
+
+                            if( encControl->nChannelsInternal == 2 && n == 0 ) {
+                                silk_stereo_encode_pred( psRangeEnc, psEnc->sStereo.predIx[ i ] );
+                                /* For LBRR data there's no need to code the mid-only flag if the side-channel LBRR flag is set */
+                                if( psEnc->state_Fxx[ 1 ].sCmn.LBRR_flags[ i ] == 0 ) {
+                                    silk_stereo_encode_mid_only( psRangeEnc, psEnc->sStereo.mid_only_flags[ i ] );
+                                }
+                            }
+                            /* Use conditional coding if previous frame available */
+                            if( i > 0 && psEnc->state_Fxx[ n ].sCmn.LBRR_flags[ i - 1 ] ) {
+                                condCoding = CODE_CONDITIONALLY;
+                            } else {
+                                condCoding = CODE_INDEPENDENTLY;
+                            }
+                            silk_encode_indices( &psEnc->state_Fxx[ n ].sCmn, psRangeEnc, i, 1, condCoding );
+                            silk_encode_pulses( psRangeEnc, psEnc->state_Fxx[ n ].sCmn.indices_LBRR[i].signalType, psEnc->state_Fxx[ n ].sCmn.indices_LBRR[i].quantOffsetType,
+                                psEnc->state_Fxx[ n ].sCmn.pulses_LBRR[ i ], psEnc->state_Fxx[ n ].sCmn.frame_length );
+                        }
+                    }
+                }
+
+                /* Reset LBRR flags */
+                for( n = 0; n < encControl->nChannelsInternal; n++ ) {
+                    silk_memset( psEnc->state_Fxx[ n ].sCmn.LBRR_flags, 0, sizeof( psEnc->state_Fxx[ n ].sCmn.LBRR_flags ) );
+                }
+                curr_nBitsUsedLBRR = ec_tell( psRangeEnc ) - curr_nBitsUsedLBRR;
+            }
+
+            silk_HP_variable_cutoff( psEnc->state_Fxx );
+
+            /* Total target bits for packet */
+            nBits = silk_DIV32_16( silk_MUL( encControl->bitRate, encControl->payloadSize_ms ), 1000 );
+            /* Subtract bits used for LBRR */
+            if( !prefillFlag ) {
+                /* psEnc->nBitsUsedLBRR is an exponential moving average of the LBRR usage,
+                   except that for the first LBRR frame it does no averaging and for the first
+                   frame after after LBRR, it goes back to zero immediately. */
+                if ( curr_nBitsUsedLBRR < 10 ) {
+                    psEnc->nBitsUsedLBRR = 0;
+                } else if ( psEnc->nBitsUsedLBRR < 10) {
+                    psEnc->nBitsUsedLBRR = curr_nBitsUsedLBRR;
+                } else {
+                    psEnc->nBitsUsedLBRR = ( psEnc->nBitsUsedLBRR + curr_nBitsUsedLBRR ) / 2;
+                }
+                nBits -= psEnc->nBitsUsedLBRR;
+            }
+            /* Divide by number of uncoded frames left in packet */
+            nBits = silk_DIV32_16( nBits, psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket );
+            /* Convert to bits/second */
+            if( encControl->payloadSize_ms == 10 ) {
+                TargetRate_bps = silk_SMULBB( nBits, 100 );
+            } else {
+                TargetRate_bps = silk_SMULBB( nBits, 50 );
+            }
+            /* Subtract fraction of bits in excess of target in previous frames and packets */
+            TargetRate_bps -= silk_DIV32_16( silk_MUL( psEnc->nBitsExceeded, 1000 ), BITRESERVOIR_DECAY_TIME_MS );
+            if( !prefillFlag && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded > 0 ) {
+                /* Compare actual vs target bits so far in this packet */
+                opus_int32 bitsBalance = ec_tell( psRangeEnc ) - psEnc->nBitsUsedLBRR - nBits * psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded;
+                TargetRate_bps -= silk_DIV32_16( silk_MUL( bitsBalance, 1000 ), BITRESERVOIR_DECAY_TIME_MS );
+            }
+            /* Never exceed input bitrate */
+            TargetRate_bps = silk_LIMIT( TargetRate_bps, encControl->bitRate, 5000 );
+
+            /* Convert Left/Right to Mid/Side */
+            if( encControl->nChannelsInternal == 2 ) {
+                silk_stereo_LR_to_MS( &psEnc->sStereo, &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ 2 ], &psEnc->state_Fxx[ 1 ].sCmn.inputBuf[ 2 ],
+                    psEnc->sStereo.predIx[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ], &psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ],
+                    MStargetRates_bps, TargetRate_bps, psEnc->state_Fxx[ 0 ].sCmn.speech_activity_Q8, encControl->toMono,
+                    psEnc->state_Fxx[ 0 ].sCmn.fs_kHz, psEnc->state_Fxx[ 0 ].sCmn.frame_length );
+                if( psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] == 0 ) {
+                    /* Reset side channel encoder memory for first frame with side coding */
+                    if( 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 ].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 ) );
+                        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_gain_Q16      = 65536;
+                        psEnc->state_Fxx[ 1 ].sCmn.first_frame_after_reset = 1;
+                    }
+                    silk_encode_do_VAD_Fxx( &psEnc->state_Fxx[ 1 ], activity );
+                } else {
+                    psEnc->state_Fxx[ 1 ].sCmn.VAD_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] = 0;
+                }
+                if( !prefillFlag ) {
+                    silk_stereo_encode_pred( psRangeEnc, psEnc->sStereo.predIx[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] );
+                    if( psEnc->state_Fxx[ 1 ].sCmn.VAD_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] == 0 ) {
+                        silk_stereo_encode_mid_only( psRangeEnc, psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded ] );
+                    }
+                }
+            } else {
+                /* Buffering */
+                silk_memcpy( psEnc->state_Fxx[ 0 ].sCmn.inputBuf, psEnc->sStereo.sMid, 2 * sizeof( opus_int16 ) );
+                silk_memcpy( psEnc->sStereo.sMid, &psEnc->state_Fxx[ 0 ].sCmn.inputBuf[ psEnc->state_Fxx[ 0 ].sCmn.frame_length ], 2 * sizeof( opus_int16 ) );
+            }
+            silk_encode_do_VAD_Fxx( &psEnc->state_Fxx[ 0 ], activity );
+
+            /* Encode */
+            for( n = 0; n < encControl->nChannelsInternal; n++ ) {
+                opus_int maxBits, useCBR;
+
+                /* Handling rate constraints */
+                maxBits = encControl->maxBits;
+                if( tot_blocks == 2 && curr_block == 0 ) {
+                    maxBits = maxBits * 3 / 5;
+                } else if( tot_blocks == 3 ) {
+                    if( curr_block == 0 ) {
+                        maxBits = maxBits * 2 / 5;
+                    } else if( curr_block == 1 ) {
+                        maxBits = maxBits * 3 / 4;
+                    }
+                }
+                useCBR = encControl->useCBR && curr_block == tot_blocks - 1;
+
+                if( encControl->nChannelsInternal == 1 ) {
+                    channelRate_bps = TargetRate_bps;
+                } else {
+                    channelRate_bps = MStargetRates_bps[ n ];
+                    if( n == 0 && MStargetRates_bps[ 1 ] > 0 ) {
+                        useCBR = 0;
+                        /* Give mid up to 1/2 of the max bits for that frame */
+                        maxBits -= encControl->maxBits / ( tot_blocks * 2 );
+                    }
+                }
+
+                if( channelRate_bps > 0 ) {
+                    opus_int condCoding;
+
+                    silk_control_SNR( &psEnc->state_Fxx[ n ].sCmn, channelRate_bps );
+
+                    /* Use independent coding if no previous frame available */
+                    if( psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded - n <= 0 ) {
+                        condCoding = CODE_INDEPENDENTLY;
+                    } else if( n > 0 && psEnc->prev_decode_only_middle ) {
+                        /* If we skipped a side frame in this packet, we don't
+                           need LTP scaling; the LTP state is well-defined. */
+                        condCoding = CODE_INDEPENDENTLY_NO_LTP_SCALING;
+                    } else {
+                        condCoding = CODE_CONDITIONALLY;
+                    }
+                    if( ( ret = silk_encode_frame_Fxx( &psEnc->state_Fxx[ n ], nBytesOut, psRangeEnc, condCoding, maxBits, useCBR ) ) != 0 ) {
+                        silk_assert( 0 );
+                    }
+                }
+                psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0;
+                psEnc->state_Fxx[ n ].sCmn.inputBufIx = 0;
+                psEnc->state_Fxx[ n ].sCmn.nFramesEncoded++;
+            }
+            psEnc->prev_decode_only_middle = psEnc->sStereo.mid_only_flags[ psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded - 1 ];
+
+            /* Insert VAD and FEC flags at beginning of bitstream */
+            if( *nBytesOut > 0 && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded == psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket) {
+                flags = 0;
+                for( n = 0; n < encControl->nChannelsInternal; n++ ) {
+                    for( i = 0; i < psEnc->state_Fxx[ n ].sCmn.nFramesPerPacket; i++ ) {
+                        flags  = silk_LSHIFT( flags, 1 );
+                        flags |= psEnc->state_Fxx[ n ].sCmn.VAD_flags[ i ];
+                    }
+                    flags  = silk_LSHIFT( flags, 1 );
+                    flags |= psEnc->state_Fxx[ n ].sCmn.LBRR_flag;
+                }
+                if( !prefillFlag ) {
+                    ec_enc_patch_initial_bits( psRangeEnc, flags, ( psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket + 1 ) * encControl->nChannelsInternal );
+                }
+
+                /* Return zero bytes if all channels DTXed */
+                if( psEnc->state_Fxx[ 0 ].sCmn.inDTX && ( encControl->nChannelsInternal == 1 || psEnc->state_Fxx[ 1 ].sCmn.inDTX ) ) {
+                    *nBytesOut = 0;
+                }
+
+                psEnc->nBitsExceeded += *nBytesOut * 8;
+                psEnc->nBitsExceeded -= silk_DIV32_16( silk_MUL( encControl->bitRate, encControl->payloadSize_ms ), 1000 );
+                psEnc->nBitsExceeded  = silk_LIMIT( psEnc->nBitsExceeded, 0, 10000 );
+
+                /* Update flag indicating if bandwidth switching is allowed */
+                speech_act_thr_for_switch_Q8 = silk_SMLAWB( SILK_FIX_CONST( SPEECH_ACTIVITY_DTX_THRES, 8 ),
+                    SILK_FIX_CONST( ( 1 - SPEECH_ACTIVITY_DTX_THRES ) / MAX_BANDWIDTH_SWITCH_DELAY_MS, 16 + 8 ), psEnc->timeSinceSwitchAllowed_ms );
+                if( psEnc->state_Fxx[ 0 ].sCmn.speech_activity_Q8 < speech_act_thr_for_switch_Q8 ) {
+                    psEnc->allowBandwidthSwitch = 1;
+                    psEnc->timeSinceSwitchAllowed_ms = 0;
+                } else {
+                    psEnc->allowBandwidthSwitch = 0;
+                    psEnc->timeSinceSwitchAllowed_ms += encControl->payloadSize_ms;
+                }
+            }
+
+            if( nSamplesIn == 0 ) {
+                break;
+            }
+        } else {
+            break;
+        }
+        curr_block++;
+    }
+
+    psEnc->nPrevChannelsInternal = encControl->nChannelsInternal;
+
+    encControl->allowBandwidthSwitch = psEnc->allowBandwidthSwitch;
+    encControl->inWBmodeWithoutVariableLP = psEnc->state_Fxx[ 0 ].sCmn.fs_kHz == 16 && psEnc->state_Fxx[ 0 ].sCmn.sLP.mode == 0;
+    encControl->internalSampleRate = silk_SMULBB( psEnc->state_Fxx[ 0 ].sCmn.fs_kHz, 1000 );
+    encControl->stereoWidth_Q14 = encControl->toMono ? 0 : psEnc->sStereo.smth_width_Q14;
+    if( prefillFlag ) {
+        encControl->payloadSize_ms = tmp_payloadSize_ms;
+        encControl->complexity = tmp_complexity;
+        for( n = 0; n < encControl->nChannelsInternal; n++ ) {
+            psEnc->state_Fxx[ n ].sCmn.controlled_since_last_payload = 0;
+            psEnc->state_Fxx[ n ].sCmn.prefillFlag = 0;
+        }
+    }
+
+    encControl->signalType = psEnc->state_Fxx[0].sCmn.indices.signalType;
+    encControl->offset = silk_Quantization_Offsets_Q10
+                         [ psEnc->state_Fxx[0].sCmn.indices.signalType >> 1 ]
+                         [ psEnc->state_Fxx[0].sCmn.indices.quantOffsetType ];
+    RESTORE_STACK;
+    return ret;
+}
+