예제 #1
0
opus_int silk_setup_resamplers(
    silk_encoder_state_Fxx          *psEnc,             /* I/O                      */
    opus_int                         fs_kHz              /* I                        */
)
{
    opus_int   ret = SILK_NO_ERROR;
    opus_int32 nSamples_temp;

    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 {
            /* Allocate worst case space for temporary upsampling, 8 to 48 kHz, so a factor 6 */
            opus_int16 x_buf_API_fs_Hz[ ( 2 * MAX_FRAME_LENGTH_MS + LA_SHAPE_MS ) * MAX_API_FS_KHZ ];
            silk_resampler_state_struct  temp_resampler_state;
#ifdef FIXED_POINT
            opus_int16 *x_bufFIX = psEnc->x_buf;
#else
            opus_int16 x_bufFIX[ 2 * MAX_FRAME_LENGTH + LA_SHAPE_MAX ];
#endif

            nSamples_temp = silk_LSHIFT( psEnc->sCmn.frame_length, 1 ) + LA_SHAPE_MS * psEnc->sCmn.fs_kHz;

#ifndef FIXED_POINT
            silk_float2short_array( x_bufFIX, psEnc->x_buf, nSamples_temp );
#endif

            /* Initialize resampler for temporary resampling of x_buf data to API_fs_Hz */
            ret += silk_resampler_init( &temp_resampler_state, silk_SMULBB( psEnc->sCmn.fs_kHz, 1000 ), psEnc->sCmn.API_fs_Hz, 0 );

            /* Temporary resampling of x_buf data to API_fs_Hz */
            ret += silk_resampler( &temp_resampler_state, x_buf_API_fs_Hz, x_bufFIX, nSamples_temp );

            /* Calculate number of samples that has been temporarily upsampled */
            nSamples_temp = silk_DIV32_16( nSamples_temp * psEnc->sCmn.API_fs_Hz, silk_SMULBB( psEnc->sCmn.fs_kHz, 1000 ) );

            /* Initialize the resampler for enc_API.c preparing resampling from API_fs_Hz to fs_kHz */
            ret += silk_resampler_init( &psEnc->sCmn.resampler_state, psEnc->sCmn.API_fs_Hz, silk_SMULBB( fs_kHz, 1000 ), 1 );

            /* Correct resampler state by resampling buffered data from API_fs_Hz to fs_kHz */
            ret += silk_resampler( &psEnc->sCmn.resampler_state, x_bufFIX, x_buf_API_fs_Hz, nSamples_temp );

#ifndef FIXED_POINT
            silk_short2float_array( psEnc->x_buf, x_bufFIX, ( 2 * MAX_FRAME_LENGTH_MS + LA_SHAPE_MS ) * fs_kHz );
#endif
        }
    }

    psEnc->sCmn.prev_API_fs_Hz = psEnc->sCmn.API_fs_Hz;

    return ret;
}
예제 #2
0
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 OPUS_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 OPUS_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 OPUS_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;
}
예제 #3
0
파일: enc_API.c 프로젝트: kode54/Cog
/* 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, 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 ) {
        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 ], 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 ) {
        /* Only accept input length of 10 ms */
        if( nBlocksOf10ms != 1 ) {
            silk_assert( 0 );
            RESTORE_STACK;
            return SILK_ENC_INPUT_INVALID_NO_OF_SAMPLES;
        }
        /* Reset Encoder */
        for( n = 0; n < encControl->nChannelsInternal; n++ ) {
            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;
        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 ) {
            silk_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 ) {
            silk_assert( 0 );
            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++ ) {
        /* 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, TargetRate_bps, 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;
    }
    silk_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 ) {
        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 ] = 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 ] = 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 = 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 {
            silk_assert( encControl->nChannelsAPI == 1 && encControl->nChannelsInternal == 1 );
            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 );
            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 */
            silk_assert( psEnc->state_Fxx[ 0 ].sCmn.inputBufIx == psEnc->state_Fxx[ 0 ].sCmn.frame_length );
            silk_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 );

                /* 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 ) );
                }

                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 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 );
            /* 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 ].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 ) );
                        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 ] );
                    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 ] );

            /* 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 = (opus_int) 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;
        }
    }

    RESTORE_STACK;
    return ret;
}