Upgrade opus 1.0.2 -> 1.1

1 files changed, 0 insertions, 392 deletions

diff --git a/src/opus-1.0.2/silk/dec_API.c b/src/opus-1.0.2/silk/dec_API.c
deleted file mode 100644
index 68403b7c..00000000
--- a/src/opus-1.0.2/silk/dec_API.c
+++ /dev/null
@@ -1,392 +0,0 @@
-/***********************************************************************
-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 "API.h"
-#include "main.h"
-#include "stack_alloc.h"
-
-/************************/
-/* Decoder Super Struct */
-/************************/
-typedef struct {
-    silk_decoder_state          channel_state[ DECODER_NUM_CHANNELS ];
-    stereo_dec_state                sStereo;
-    opus_int                         nChannelsAPI;
-    opus_int                         nChannelsInternal;
-    opus_int                         prev_decode_only_middle;
-} silk_decoder;
-
-/*********************/
-/* Decoder functions */
-/*********************/
-
-opus_int silk_Get_Decoder_Size(                         /* O    Returns error code                              */
-    opus_int                        *decSizeBytes       /* O    Number of bytes in SILK decoder state           */
-)
-{
-    opus_int ret = SILK_NO_ERROR;
-
-    *decSizeBytes = sizeof( silk_decoder );
-
-    return ret;
-}
-
-/* Reset decoder state */
-opus_int silk_InitDecoder(                              /* O    Returns error code                              */
-    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(                                   /* O    Returns error code                              */
-    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, decode_only_middle = 0, ret = SILK_NO_ERROR;
-    opus_int32 nSamplesOutDec, LBRR_symbol;
-    opus_int16 *samplesOut1_tmp[ 2 ];
-    VARDECL( opus_int16, samplesOut1_tmp_storage );
-    VARDECL( opus_int16, samplesOut2_tmp );
-    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;
-    opus_int has_side;
-    opus_int stereo_to_mono;
-    SAVE_STACK;
-
-    /**********************************/
-    /* 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 */
-        }
-    }
-
-    /* If Mono -> Stereo transition in bitstream: init state of second channel */
-    if( decControl->nChannelsInternal > psDec->nChannelsInternal ) {
-        ret += silk_init_decoder( &channel_state[ 1 ] );
-    }
-
-    stereo_to_mono = decControl->nChannelsInternal == 1 && psDec->nChannelsInternal == 2 &&
-                     ( decControl->internalSampleRate == 1000*channel_state[ 0 ].fs_kHz );
-
-    if( channel_state[ 0 ].nFramesDecoded == 0 ) {
-        for( n = 0; n < decControl->nChannelsInternal; n++ ) {
-            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 );
-                RESTORE_STACK;
-                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 );
-                RESTORE_STACK;
-                return SILK_DEC_INVALID_SAMPLING_FREQUENCY;
-            }
-            ret += silk_decoder_set_fs( &channel_state[ n ], fs_kHz_dec, 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 ) );
-        silk_memcpy( &channel_state[ 1 ].resampler_state, &channel_state[ 0 ].resampler_state, sizeof( silk_resampler_state_struct ) );
-    }
-    psDec->nChannelsAPI      = decControl->nChannelsAPI;
-    psDec->nChannelsInternal = decControl->nChannelsInternal;
-
-    if( decControl->API_sampleRate > (opus_int32)MAX_API_FS_KHZ * 1000 || decControl->API_sampleRate < 8000 ) {
-        ret = SILK_DEC_INVALID_SAMPLING_FREQUENCY;
-        RESTORE_STACK;
-        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 ];
-                        opus_int condCoding;
-
-                        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 );
-                            }
-                        }
-                        /* Use conditional coding if previous frame available */
-                        if( i > 0 && channel_state[ n ].LBRR_flags[ i - 1 ] ) {
-                            condCoding = CODE_CONDITIONALLY;
-                        } else {
-                            condCoding = CODE_INDEPENDENTLY;
-                        }
-                        silk_decode_indices( &channel_state[ n ], psRangeDec, i, 1, condCoding );
-                        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, decode mid-only flag only if side-channel's LBRR flag is false */
-            if( ( lostFlag == FLAG_DECODE_NORMAL && channel_state[ 1 ].VAD_flags[ channel_state[ 0 ].nFramesDecoded ] == 0 ) ||
-                ( 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 ];
-            }
-        }
-    }
-
-    /* Reset side channel decoder prediction memory for first frame with side coding */
-    if( decControl->nChannelsInternal == 2 && decode_only_middle == 0 && psDec->prev_decode_only_middle == 1 ) {
-        silk_memset( psDec->channel_state[ 1 ].outBuf, 0, sizeof(psDec->channel_state[ 1 ].outBuf) );
-        silk_memset( psDec->channel_state[ 1 ].sLPC_Q14_buf, 0, sizeof(psDec->channel_state[ 1 ].sLPC_Q14_buf) );
-        psDec->channel_state[ 1 ].lagPrev        = 100;
-        psDec->channel_state[ 1 ].LastGainIndex  = 10;
-        psDec->channel_state[ 1 ].prevSignalType = TYPE_NO_VOICE_ACTIVITY;
-        psDec->channel_state[ 1 ].first_frame_after_reset = 1;
-    }
-
-    ALLOC( samplesOut1_tmp_storage,
-           decControl->nChannelsInternal*(
-               channel_state[ 0 ].frame_length + 2 ),
-           opus_int16 );
-    samplesOut1_tmp[ 0 ] = samplesOut1_tmp_storage;
-    samplesOut1_tmp[ 1 ] = samplesOut1_tmp_storage
-                           + channel_state[ 0 ].frame_length + 2;
-
-    if( lostFlag == FLAG_DECODE_NORMAL ) {
-        has_side = !decode_only_middle;
-    } else {
-        has_side = !psDec->prev_decode_only_middle
-              || (decControl->nChannelsInternal == 2 && lostFlag == FLAG_DECODE_LBRR && channel_state[1].LBRR_flags[ channel_state[1].nFramesDecoded ] == 1 );
-    }
-    /* Call decoder for one frame */
-    for( n = 0; n < decControl->nChannelsInternal; n++ ) {
-        if( n == 0 || has_side ) {
-            opus_int FrameIndex;
-            opus_int condCoding;
-
-            FrameIndex = channel_state[ 0 ].nFramesDecoded - n;
-            /* Use independent coding if no previous frame available */
-            if( FrameIndex <= 0 ) {
-                condCoding = CODE_INDEPENDENTLY;
-            } else if( lostFlag == FLAG_DECODE_LBRR ) {
-                condCoding = channel_state[ n ].LBRR_flags[ FrameIndex - 1 ] ? CODE_CONDITIONALLY : CODE_INDEPENDENTLY;
-            } else if( n > 0 && psDec->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;
-            }
-            ret += silk_decode_frame( &channel_state[ n ], psRangeDec, &samplesOut1_tmp[ n ][ 2 ], &nSamplesOutDec, lostFlag, condCoding);
-        } else {
-            silk_memset( &samplesOut1_tmp[ n ][ 2 ], 0, nSamplesOutDec * sizeof( opus_int16 ) );
-        }
-        channel_state[ n ].nFramesDecoded++;
-    }
-
-    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 */
-    ALLOC( samplesOut2_tmp,
-           decControl->nChannelsAPI == 2 ? *nSamplesOut : 0, opus_int16 );
-    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 ) {
-            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 ) {
-        if ( stereo_to_mono ){
-            /* Resample right channel for newly collapsed stereo just in case
-               we weren't doing collapsing when switching to mono */
-            ret += silk_resampler( &channel_state[ 1 ].resampler_state, resample_out_ptr, &samplesOut1_tmp[ 0 ][ 1 ], nSamplesOutDec );
-
-            for( i = 0; i < *nSamplesOut; i++ ) {
-                samplesOut[ 1 + 2 * i ] = resample_out_ptr[ i ];
-            }
-        } else {
-            for( i = 0; i < *nSamplesOut; i++ ) {
-                samplesOut[ 1 + 2 * i ] = samplesOut[ 0 + 2 * i ];
-            }
-        }
-    }
-
-    /* Export pitch lag, measured at 48 kHz sampling rate */
-    if( channel_state[ 0 ].prevSignalType == TYPE_VOICED ) {
-        int mult_tab[ 3 ] = { 6, 4, 3 };
-        decControl->prevPitchLag = channel_state[ 0 ].lagPrev * mult_tab[ ( channel_state[ 0 ].fs_kHz - 8 ) >> 2 ];
-    } else {
-        decControl->prevPitchLag = 0;
-    }
-
-    if( lostFlag == FLAG_PACKET_LOST ) {
-       /* On packet loss, remove the gain clamping to prevent having the energy "bounce back"
-          if we lose packets when the energy is going down */
-       for ( i = 0; i < psDec->nChannelsInternal; i++ )
-          psDec->channel_state[ i ].LastGainIndex = 10;
-    } else {
-       psDec->prev_decode_only_middle = decode_only_middle;
-    }
-    RESTORE_STACK;
-    return ret;
-}
-
-#if 0
-/* 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;
-}
-#endif