shithub: opus

ref: 4cc9a459e84d23bd9a96560e1e7397cd78dd38ab
dir: /silk/dec_API.c/

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/***********************************************************************
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:
- 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
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OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
***********************************************************************/

#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "API.h"
#include "main.h"

/************************/
/* Decoder Super Struct */
/************************/
typedef struct {
    silk_decoder_state          channel_state[ DECODER_NUM_CHANNELS ];
    stereo_dec_state                sStereo;
    opus_int                         nChannelsAPI;
    opus_int                         nChannelsInternal;
} silk_decoder;

/*********************/
/* Decoder functions */
/*********************/

opus_int silk_Get_Decoder_Size( int *decSizeBytes )
{
    opus_int ret = SILK_NO_ERROR;

    *decSizeBytes = sizeof( silk_decoder );

    return ret;
}

/* Reset decoder state */
opus_int silk_InitDecoder(
    void* decState                                      /* I/O: State                                          */
)
{
    opus_int n, ret = SILK_NO_ERROR;
    silk_decoder_state *channel_state = ((silk_decoder *)decState)->channel_state;

    for( n = 0; n < DECODER_NUM_CHANNELS; n++ ) {
        ret  = silk_init_decoder( &channel_state[ n ] );
    }

    return ret;
}

/* Decode a frame */
opus_int silk_Decode(
    void*                               decState,       /* I/O: State                                           */
    silk_DecControlStruct*      decControl,     /* I/O: Control Structure                               */
    opus_int                             lostFlag,       /* I:   0: no loss, 1 loss, 2 decode FEC                */
    opus_int                             newPacketFlag,  /* I:   Indicates first decoder call for this packet    */
    ec_dec                              *psRangeDec,    /* I/O  Compressor data structure                       */
    opus_int16                           *samplesOut,    /* O:   Decoded output speech vector                    */
    opus_int32                           *nSamplesOut    /* O:   Number of samples decoded                       */
)
{
    opus_int   i, n, prev_fs_kHz, decode_only_middle = 0, ret = SILK_NO_ERROR;
    opus_int32 nSamplesOutDec, LBRR_symbol;
    opus_int16 samplesOut1_tmp[ 2 ][ MAX_FS_KHZ * MAX_FRAME_LENGTH_MS + 2 ];
    opus_int16 samplesOut2_tmp[ MAX_API_FS_KHZ * MAX_FRAME_LENGTH_MS ];
    opus_int32 MS_pred_Q13[ 2 ] = { 0 };
    opus_int16 *resample_out_ptr;
    silk_decoder *psDec = ( silk_decoder * )decState;
    silk_decoder_state *channel_state = psDec->channel_state;

    /**********************************/
    /* Test if first frame in payload */
    /**********************************/
    if( newPacketFlag ) {
        for( n = 0; n < decControl->nChannelsInternal; n++ ) {
            channel_state[ n ].nFramesDecoded = 0;  /* Used to count frames in packet */
        }
    }

    /* Save previous sample frequency */
    prev_fs_kHz = channel_state[ 0 ].fs_kHz;

    /* If Mono -> Stereo transition in bitstream: init state of second channel */
    if( decControl->nChannelsInternal > psDec->nChannelsInternal ) {
        ret += silk_init_decoder( &channel_state[ 1 ] );
        if( psDec->nChannelsAPI == 2 ) {
            silk_memcpy( &channel_state[ 1 ].resampler_state, &channel_state[ 0 ].resampler_state, sizeof( silk_resampler_state_struct ) );
        }
    }

    for( n = 0; n < decControl->nChannelsInternal; n++ ) {
        if( channel_state[ n ].nFramesDecoded == 0 ) {
            opus_int fs_kHz_dec;
            if( decControl->payloadSize_ms == 0 ) {
                /* Assuming packet loss, use 10 ms */
                channel_state[ n ].nFramesPerPacket = 1;
                channel_state[ n ].nb_subfr = 2;
            } else if( decControl->payloadSize_ms == 10 ) {
                channel_state[ n ].nFramesPerPacket = 1;
                channel_state[ n ].nb_subfr = 2;
            } else if( decControl->payloadSize_ms == 20 ) {
                channel_state[ n ].nFramesPerPacket = 1;
                channel_state[ n ].nb_subfr = 4;
            } else if( decControl->payloadSize_ms == 40 ) {
                channel_state[ n ].nFramesPerPacket = 2;
                channel_state[ n ].nb_subfr = 4;
            } else if( decControl->payloadSize_ms == 60 ) {
                channel_state[ n ].nFramesPerPacket = 3;
                channel_state[ n ].nb_subfr = 4;
            } else {
                silk_assert( 0 );
                return SILK_DEC_INVALID_FRAME_SIZE;
            }
            fs_kHz_dec = ( decControl->internalSampleRate >> 10 ) + 1;
            if( fs_kHz_dec != 8 && fs_kHz_dec != 12 && fs_kHz_dec != 16 ) {
                silk_assert( 0 );
                return SILK_DEC_INVALID_SAMPLING_FREQUENCY;
            }
            silk_decoder_set_fs( &channel_state[ n ], fs_kHz_dec );
        }
    }

    /* Initialize resampler when switching internal or external sampling frequency */
    if( prev_fs_kHz != channel_state[ 0 ].fs_kHz || channel_state[ 0 ].prev_API_sampleRate != decControl->API_sampleRate ) {
        ret = silk_resampler_init( &channel_state[ 0 ].resampler_state, silk_SMULBB( channel_state[ 0 ].fs_kHz, 1000 ), decControl->API_sampleRate );
        if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 2 ) {
            silk_memcpy( &channel_state[ 1 ].resampler_state, &channel_state[ 0 ].resampler_state, sizeof( silk_resampler_state_struct ) );
        }
    }
    channel_state[ 0 ].prev_API_sampleRate = decControl->API_sampleRate;
    if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 2 && ( psDec->nChannelsAPI == 1 || psDec->nChannelsInternal == 1 ) ) {
        silk_memset( psDec->sStereo.pred_prev_Q13, 0, sizeof( psDec->sStereo.pred_prev_Q13 ) );
        silk_memset( psDec->sStereo.sSide, 0, sizeof( psDec->sStereo.sSide ) );
    }
    psDec->nChannelsAPI      = decControl->nChannelsAPI;
    psDec->nChannelsInternal = decControl->nChannelsInternal;

    if( decControl->API_sampleRate > MAX_API_FS_KHZ * 1000 || decControl->API_sampleRate < 8000 ) {
        ret = SILK_DEC_INVALID_SAMPLING_FREQUENCY;
        return( ret );
    }

    if( lostFlag != FLAG_PACKET_LOST && channel_state[ 0 ].nFramesDecoded == 0 ) {
        /* First decoder call for this payload */
        /* Decode VAD flags and LBRR flag */
        for( n = 0; n < decControl->nChannelsInternal; n++ ) {
            for( i = 0; i < channel_state[ n ].nFramesPerPacket; i++ ) {
                channel_state[ n ].VAD_flags[ i ] = ec_dec_bit_logp(psRangeDec, 1);
            }
            channel_state[ n ].LBRR_flag = ec_dec_bit_logp(psRangeDec, 1);
        }
        /* Decode LBRR flags */
        for( n = 0; n < decControl->nChannelsInternal; n++ ) {
            silk_memset( channel_state[ n ].LBRR_flags, 0, sizeof( channel_state[ n ].LBRR_flags ) );
            if( channel_state[ n ].LBRR_flag ) {
                if( channel_state[ n ].nFramesPerPacket == 1 ) {
                    channel_state[ n ].LBRR_flags[ 0 ] = 1;
                } else {
                    LBRR_symbol = ec_dec_icdf( psRangeDec, silk_LBRR_flags_iCDF_ptr[ channel_state[ n ].nFramesPerPacket - 2 ], 8 ) + 1;
                    for( i = 0; i < channel_state[ n ].nFramesPerPacket; i++ ) {
                        channel_state[ n ].LBRR_flags[ i ] = silk_RSHIFT( LBRR_symbol, i ) & 1;
                    }
                }
            }
        }

        if( lostFlag == FLAG_DECODE_NORMAL ) {
            /* Regular decoding: skip all LBRR data */
            for( i = 0; i < channel_state[ 0 ].nFramesPerPacket; i++ ) {
                for( n = 0; n < decControl->nChannelsInternal; n++ ) {
                    if( channel_state[ n ].LBRR_flags[ i ] ) {
                        opus_int pulses[ MAX_FRAME_LENGTH ];
                        if( decControl->nChannelsInternal == 2 && n == 0 ) {
                            silk_stereo_decode_pred( psRangeDec, MS_pred_Q13 );
                            if( channel_state[ 1 ].LBRR_flags[ i ] == 0 ) {
                                silk_stereo_decode_mid_only( psRangeDec, &decode_only_middle );
                            }
                        }
                        silk_decode_indices( &channel_state[ n ], psRangeDec, i, 1 );
                        silk_decode_pulses( psRangeDec, pulses, channel_state[ n ].indices.signalType,
                            channel_state[ n ].indices.quantOffsetType, channel_state[ n ].frame_length );
                    }
                }
            }
        }
    }

    /* Get MS predictor index */
    if( decControl->nChannelsInternal == 2 ) {
        if(   lostFlag == FLAG_DECODE_NORMAL ||
            ( lostFlag == FLAG_DECODE_LBRR && channel_state[ 0 ].LBRR_flags[ channel_state[ 0 ].nFramesDecoded ] == 1 ) )
        {
            silk_stereo_decode_pred( psRangeDec, MS_pred_Q13 );
            /* For LBRR data, only decode mid-only flag if side-channel's LBRR flag is false */
            if(   lostFlag == FLAG_DECODE_NORMAL ||
                ( lostFlag == FLAG_DECODE_LBRR && channel_state[ 1 ].LBRR_flags[ channel_state[ 0 ].nFramesDecoded ] == 0 ) )
            {
                silk_stereo_decode_mid_only( psRangeDec, &decode_only_middle );
            } else {
                decode_only_middle = 0;
            }
        } else {
            for( n = 0; n < 2; n++ ) {
                MS_pred_Q13[n] = psDec->sStereo.pred_prev_Q13[n];
            }
        }
    }

    /* Call decoder for one frame */
    for( n = 0; n < decControl->nChannelsInternal; n++ ) {
        if( n == 0 || decode_only_middle == 0 ) {
            ret += silk_decode_frame( &channel_state[ n ], psRangeDec, &samplesOut1_tmp[ n ][ 2 ], &nSamplesOutDec, lostFlag );
        } else {
            silk_memset( &samplesOut1_tmp[ n ][ 2 ], 0, nSamplesOutDec * sizeof( opus_int16 ) );
        }
    }

    if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 2 ) {
        /* Convert Mid/Side to Left/Right */
        silk_stereo_MS_to_LR( &psDec->sStereo, samplesOut1_tmp[ 0 ], samplesOut1_tmp[ 1 ], MS_pred_Q13, channel_state[ 0 ].fs_kHz, nSamplesOutDec );
    } else {
        /* Buffering */
        silk_memcpy( samplesOut1_tmp[ 0 ], psDec->sStereo.sMid, 2 * sizeof( opus_int16 ) );
        silk_memcpy( psDec->sStereo.sMid, &samplesOut1_tmp[ 0 ][ nSamplesOutDec ], 2 * sizeof( opus_int16 ) );
    }

    /* Number of output samples */
    *nSamplesOut = silk_DIV32( nSamplesOutDec * decControl->API_sampleRate, silk_SMULBB( channel_state[ 0 ].fs_kHz, 1000 ) );

    /* Set up pointers to temp buffers */
    if( decControl->nChannelsAPI == 2 ) {
        resample_out_ptr = samplesOut2_tmp;
    } else {
        resample_out_ptr = samplesOut;
    }

    for( n = 0; n < silk_min( decControl->nChannelsAPI, decControl->nChannelsInternal ); n++ ) {
        /* Resample decoded signal to API_sampleRate */
        ret += silk_resampler( &channel_state[ n ].resampler_state, resample_out_ptr, &samplesOut1_tmp[ n ][ 1 ], nSamplesOutDec );

        /* Interleave if stereo output and stereo stream */
        if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 2 ) {
            for( i = 0; i < *nSamplesOut; i++ ) {
                samplesOut[ n + 2 * i ] = resample_out_ptr[ i ];
            }
        }
    }

    /* Create two channel output from mono stream */
    if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 1 ) {
        for( i = 0; i < *nSamplesOut; i++ ) {
            samplesOut[ 0 + 2 * i ] = samplesOut[ 1 + 2 * i ] = resample_out_ptr[ i ];
        }
    }

    return ret;
}

/* Getting table of contents for a packet */
opus_int silk_get_TOC(
    const opus_uint8                     *payload,           /* I    Payload data                                */
    const opus_int                       nBytesIn,           /* I:   Number of input bytes                       */
    const opus_int                       nFramesPerPayload,  /* I:   Number of SILK frames per payload           */
    silk_TOC_struct                 *Silk_TOC           /* O:   Type of content                             */
)
{
    opus_int i, flags, ret = SILK_NO_ERROR;

    if( nBytesIn < 1 ) {
        return -1;
    }
    if( nFramesPerPayload < 0 || nFramesPerPayload > 3 ) {
        return -1;
    }

    silk_memset( Silk_TOC, 0, sizeof( Silk_TOC ) );

    /* For stereo, extract the flags for the mid channel */
    flags = silk_RSHIFT( payload[ 0 ], 7 - nFramesPerPayload ) & ( silk_LSHIFT( 1, nFramesPerPayload + 1 ) - 1 );

    Silk_TOC->inbandFECFlag = flags & 1;
    for( i = nFramesPerPayload - 1; i >= 0 ; i-- ) {
        flags = silk_RSHIFT( flags, 1 );
        Silk_TOC->VADFlags[ i ] = flags & 1;
        Silk_TOC->VADFlag |= flags & 1;
    }

    return ret;
}