Making it clearer to Coverity that nStates cannot exceed NLSF_QUANT_DEL_DEC_STATES
[opus.git] / silk / enc_API.c
index 339dafc..ba3db06 100644 (file)
@@ -1,28 +1,28 @@
 /***********************************************************************
 Copyright (c) 2006-2011, Skype Limited. All rights reserved.
 Redistribution and use in source and binary forms, with or without
-modification, (subject to the limitations in the disclaimer below)
-are permitted provided that the following conditions are met:
+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 Skype Limited, nor the names of specific
-contributors, may be used to endorse or promote products derived from
-this software without specific prior written permission.
-NO EXPRESS OR IMPLIED LICENSES TO ANY PARTY'S PATENT RIGHTS ARE GRANTED
-BY THIS LICENSE. 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.
+- 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
@@ -32,6 +32,7 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
 #include "API.h"
 #include "control.h"
 #include "typedef.h"
+#include "stack_alloc.h"
 #include "structs.h"
 #include "tuning_parameters.h"
 #ifdef FIXED_POINT
@@ -40,11 +41,21 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
 #include "main_FLP.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( int *encSizeBytes )
+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;
 
@@ -56,9 +67,10 @@ opus_int silk_Get_Encoder_Size( int *encSizeBytes )
 /*************************/
 /* Init or Reset encoder */
 /*************************/
-opus_int silk_InitEncoder(
-    void                            *encState,          /* I/O: State                                           */
-    silk_EncControlStruct           *encStatus          /* O:   Control structure                               */
+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;
@@ -69,7 +81,7 @@ opus_int silk_InitEncoder(
     /* 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 ] ) ) {
+        if( ret += silk_init_encoder( &psEnc->state_Fxx[ n ], arch ) ) {
             silk_assert( 0 );
         }
     }
@@ -88,9 +100,9 @@ opus_int silk_InitEncoder(
 /***************************************/
 /* Read control structure from encoder */
 /***************************************/
-opus_int silk_QueryEncoder(
-    const void *encState,                       /* I:   State Vector                                    */
-    silk_EncControlStruct *encStatus            /* O:   Control Structure                               */
+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;
@@ -123,35 +135,47 @@ opus_int silk_QueryEncoder(
 /**************************/
 /* Encode frame with Silk */
 /**************************/
-opus_int silk_Encode(
-    void                                *encState,      /* I/O: State                                           */
-    silk_EncControlStruct               *encControl,    /* I:   Control structure                               */
-    const opus_int16                     *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_int                             *nBytesOut,     /* I/O: Number of bytes in payload (input: Max bytes)   */
-    const opus_int                       prefillFlag     /* I:   Flag to indicate prefilling buffers; no coding  */
+/* 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_int16                *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   n, i, nBits, flags, tmp_payloadSize_ms = 0, tmp_complexity = 0, ret = 0;
-    opus_int   nSamplesToBuffer, nBlocksOf10ms, nSamplesFromInput = 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;
+    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;
-
+    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 ) ) {
+    if( ( ret = check_control_input( encControl ) ) != 0 ) {
         silk_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 ] );
+        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;
@@ -172,18 +196,19 @@ opus_int silk_Encode(
     psEnc->nChannelsInternal = encControl->nChannelsInternal;
 
     nBlocksOf10ms = silk_DIV32( 100 * nSamplesIn, encControl->API_sampleRate );
+    tot_blocks = ( nBlocksOf10ms > 1 ) ? nBlocksOf10ms >> 1 : 1;
+    curr_block = 0;
     if( prefillFlag ) {
         /* Only accept input length of 10 ms */
         if( nBlocksOf10ms != 1 ) {
-            ret = SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
             silk_assert( 0 );
-            return ret;
+            RESTORE_STACK;
+            return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
         }
         /* Reset Encoder */
         for( n = 0; n < encControl->nChannelsInternal; n++ ) {
-            if( (ret = silk_init_encoder( &psEnc->state_Fxx[ n ] ) ) != 0 ) {
-                silk_assert( 0 );
-            }
+            ret = silk_init_encoder( &psEnc->state_Fxx[ n ], psEnc->state_Fxx[ n ].sCmn.arch );
+            silk_assert( !ret );
         }
         tmp_payloadSize_ms = encControl->payloadSize_ms;
         encControl->payloadSize_ms = 10;
@@ -196,54 +221,58 @@ opus_int silk_Encode(
     } else {
         /* Only accept input lengths that are a multiple of 10 ms */
         if( nBlocksOf10ms * encControl->API_sampleRate != 100 * nSamplesIn || nSamplesIn < 0 ) {
-            ret = SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
             silk_assert( 0 );
-            return ret;
+            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 ) {
-            ret = SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
             silk_assert( 0 );
-            return ret;
+            RESTORE_STACK;
+            return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
         }
     }
 
     TargetRate_bps = silk_RSHIFT32( encControl->bitRate, encControl->nChannelsInternal - 1 );
     for( n = 0; n < encControl->nChannelsInternal; n++ ) {
-        /* JMV: Force the side channel to the same rate as the mid. Is this the right way? */
-        int force_fs_kHz = (n==1) ? psEnc->state_Fxx[0].sCmn.fs_kHz : 0;
-        if( ( ret = silk_control_encoder( &psEnc->state_Fxx[ n ], encControl, TargetRate_bps, psEnc->allowBandwidthSwitch, n, force_fs_kHz ) ) != 0 ) {
+        /* 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)
-        {
+        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;
     }
     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 */
+    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 ) {
         nSamplesToBuffer  = psEnc->state_Fxx[ 0 ].sCmn.frame_length - psEnc->state_Fxx[ 0 ].sCmn.inputBufIx;
-        nSamplesToBuffer  = silk_min( nSamplesToBuffer, 10 * nBlocksOf10ms * psEnc->state_Fxx[ 0 ].sCmn.fs_kHz );
+        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 ) {
-            int id = psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded;
+            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 );
@@ -252,45 +281,36 @@ 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 ] = 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  */
-            if (psEnc->nPrevChannelsInternal == 2 && psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded==0) {
+            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++ ) {
+               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);
                }
             }
-            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;
         }
 
@@ -330,6 +350,8 @@ opus_int silk_Encode(
                 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 */
@@ -337,7 +359,13 @@ opus_int silk_Encode(
                                     silk_stereo_encode_mid_only( psRangeEnc, psEnc->sStereo.mid_only_flags[ i ] );
                                 }
                             }
-                            silk_encode_indices( &psEnc->state_Fxx[ n ].sCmn, psRangeEnc, i, 1 );
+                            /* 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 );
                         }
@@ -348,25 +376,33 @@ opus_int silk_Encode(
                 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 ) );
                 }
+
+                psEnc->nBitsUsedLBRR = ec_tell( psRangeEnc );
             }
 
             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 half of the bits already used */
-            if (!prefillFlag)
-                nBits -= ec_tell( psRangeEnc ) >> 1;
+            /* Subtract bits used for LBRR */
+            if( !prefillFlag ) {
+                nBits -= psEnc->nBitsUsedLBRR;
+            }
             /* Divide by number of uncoded frames left in packet */
-            nBits = silk_DIV32_16( nBits, psEnc->state_Fxx[ 0 ].sCmn.nFramesPerPacket - psEnc->state_Fxx[ 0 ].sCmn.nFramesEncoded );
+            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 packets */
+            /* 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 );
 
@@ -376,51 +412,89 @@ opus_int silk_Encode(
                     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 ] );
+                } 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 ] );
-                    silk_stereo_encode_mid_only( psRangeEnc, psEnc->sStereo.mid_only_flags[ 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 ) );
             }
-
-            /* 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 ];
+            silk_encode_do_VAD_Fxx( &psEnc->state_Fxx[ 0 ] );
 
             /* 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 );
 
-                    if( ( ret = silk_encode_frame_Fxx( &psEnc->state_Fxx[ n ], nBytesOut, psRangeEnc ) ) != 0 ) {
+                    /* 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.nFramesEncoded++;
                 }
                 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) {
@@ -464,6 +538,7 @@ opus_int silk_Encode(
         } else {
             break;
         }
+        curr_block++;
     }
 
     psEnc->nPrevChannelsInternal = encControl->nChannelsInternal;
@@ -481,6 +556,11 @@ opus_int silk_Encode(
         }
     }
 
+    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;
 }