Create local copy of Opus v1.1.2

webrtc/modules/audio_coding/codecs/opus/opus/src/silk/dec_API.c

Index: webrtc/modules/audio_coding/codecs/opus/opus/src/silk/dec_API.c
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus/src/silk/dec_API.c b/webrtc/modules/audio_coding/codecs/opus/opus/src/silk/dec_API.c
new file mode 100644
index 0000000000000000000000000000000000000000..b7d8ed48d88f5192860445ce92c3531b9f552cd6
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/opus/opus/src/silk/dec_API.c
@@ -0,0 +1,419 @@
+/***********************************************************************
+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"
+#include "os_support.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 ] );
+    }
+    silk_memset(&((silk_decoder *)decState)->sStereo, 0, sizeof(((silk_decoder *)decState)->sStereo));
+    /* Not strictly needed, but it's cleaner that way */
+    ((silk_decoder *)decState)->prev_decode_only_middle = 0;
+
+    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                       */
+    int                             arch                /* I    Run-time architecture                           */
+)
+{
+    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_storage1 );
+    VARDECL( opus_int16, samplesOut1_tmp_storage2 );
+    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;
+    int delay_stack_alloc;
+    SAVE_STACK;
+
+    silk_assert( decControl->nChannelsInternal == 1 || decControl->nChannelsInternal == 2 );
+
+    /**********************************/
+    /* 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_int16 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;
+    }
+
+    /* Check if the temp buffer fits into the output PCM buffer. If it fits,
+       we can delay allocating the temp buffer until after the SILK peak stack
+       usage. We need to use a < and not a <= because of the two extra samples. */
+    delay_stack_alloc = decControl->internalSampleRate*decControl->nChannelsInternal
+          < decControl->API_sampleRate*decControl->nChannelsAPI;
+    ALLOC( samplesOut1_tmp_storage1, delay_stack_alloc ? ALLOC_NONE
+           : decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2 ),
+           opus_int16 );
+    if ( delay_stack_alloc )
+    {
+       samplesOut1_tmp[ 0 ] = samplesOut;
+       samplesOut1_tmp[ 1 ] = samplesOut + channel_state[ 0 ].frame_length + 2;
+    } else {
+       samplesOut1_tmp[ 0 ] = samplesOut1_tmp_storage1;
+       samplesOut1_tmp[ 1 ] = samplesOut1_tmp_storage1 + 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, arch);
+        } 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 : ALLOC_NONE, opus_int16 );
+    if( decControl->nChannelsAPI == 2 ) {
+        resample_out_ptr = samplesOut2_tmp;
+    } else {
+        resample_out_ptr = samplesOut;
+    }
+
+    ALLOC( samplesOut1_tmp_storage2, delay_stack_alloc
+           ? decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2 )
+           : ALLOC_NONE,
+           opus_int16 );
+    if ( delay_stack_alloc ) {
+       OPUS_COPY(samplesOut1_tmp_storage2, samplesOut, decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2));
+       samplesOut1_tmp[ 0 ] = samplesOut1_tmp_storage2;
+       samplesOut1_tmp[ 1 ] = samplesOut1_tmp_storage2 + channel_state[ 0 ].frame_length + 2;
+    }
+    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