mirror of
https://github.com/Q3Rally-Team/q3rally.git
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0d5fb492cd
Fix GCC 6 misleading-indentation warning add SECURITY.md OpenGL2: Restore adding fixed ambient light when HDR is enabled Few LCC memory fixes. fix a few potential buffer overwrite in Game VM Enable compiler optimization on all macOS architectures Don't allow qagame module to create "botlib.log" at ANY filesystem location Make FS_BuildOSPath for botlib.log consistent with typical usage tiny readme thing Remove extra plus sign from Huff_Compress() Fix VMs being able to change CVAR_PROTECTED cvars Don't register fs_game cvar everywhere just to get the value Don't let VMs change engine latch cvars immediately Fix fs_game '..' reading outside of home and base path Fix VMs forcing engine latch cvar to update to latched value Revert my recent cvar latch changes Revert "Don't let VMs change engine latch cvars immediately" Partially revert "Fix fs_game '..' reading outside of home and base path" Revert "Fix VMs forcing engine latch cvar to update to latched value" Fix exploit to bypass filename restrictions on Windows Changes to systemd q3a.service Fix Q_vsnprintf for mingw-w64 Fix timelimit causing an infinite map ending loop Fix invalid access to cluster 0 in AAS_AreaRouteToGoalArea() Fix negative frag/capturelimit causing an infinite map end loop OpenGL2: Fix dark lightmap on shader in mpteam6 Make FS_InvalidGameDir() consider subdirectories invalid [qcommon] Remove dead serialization code [qcommon] Make several zone variables and functions static. Fix MAC_OS_X_VERSION_MIN_REQUIRED for macOS 10.10 and later Increase q3_ui .arena filename list buffer size to 4096 bytes OpenGL2: Fix crash when BSP has deluxe maps and vertex lit surfaces Support Unicode characters greater than 0xFF in cl_consoleKeys Fix macOS app bundle with space in name OpenGL1: Use glGenTextures instead of hardcoded values Remove CON_FlushIn function and where STDIN needs flushing, use tcflush POSIX function Update libogg from 1.3.2 to 1.3.3 Rename (already updated) libogg-1.3.2 to libogg-1.3.3 Update libvorbis from 1.3.5 to 1.3.6 * Fix CVE-2018-5146 - out-of-bounds write on codebook decoding. * Fix CVE-2017-14632 - free() on unitialized data * Fix CVE-2017-14633 - out-of-bounds read Rename (already updated) libvorbis-1.3.5 to libvorbis-1.3.6 Update opus from 1.1.4 to 1.2.1 Rename (already updated) opus-1.1.4 to opus-1.2.1 Update opusfile from 0.8 to 0.9 Rename (already updated) opusfile-0.8 to opusfile-0.9 First swing at a CONTRIBUTING.md Allow loading system OpenAL library on macOS again Remove duplicate setting of FREETYPE_CFLAGS in Makefile Fix exploit to reset player by sending wrong serverId Fix "Going to CS_ZOMBIE for [clientname]" developer message Fix MSG_Read*String*() functions not being able to read last byte from message
419 lines
19 KiB
C
419 lines
19 KiB
C
/***********************************************************************
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Copyright (c) 2006-2011, Skype Limited. All rights reserved.
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Redistribution and use in source and binary forms, with or without
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modification, are permitted provided that the following conditions
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are met:
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- Redistributions of source code must retain the above copyright notice,
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this list of conditions and the following disclaimer.
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- Redistributions in binary form must reproduce the above copyright
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notice, this list of conditions and the following disclaimer in the
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documentation and/or other materials provided with the distribution.
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- Neither the name of Internet Society, IETF or IETF Trust, nor the
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names of specific contributors, may be used to endorse or promote
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products derived from this software without specific prior written
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permission.
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THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
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AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
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LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
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CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
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SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
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INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
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CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
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ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
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POSSIBILITY OF SUCH DAMAGE.
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***********************************************************************/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "API.h"
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#include "main.h"
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#include "stack_alloc.h"
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#include "os_support.h"
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/************************/
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/* Decoder Super Struct */
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/************************/
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typedef struct {
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silk_decoder_state channel_state[ DECODER_NUM_CHANNELS ];
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stereo_dec_state sStereo;
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opus_int nChannelsAPI;
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opus_int nChannelsInternal;
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opus_int prev_decode_only_middle;
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} silk_decoder;
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/*********************/
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/* Decoder functions */
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/*********************/
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opus_int silk_Get_Decoder_Size( /* O Returns error code */
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opus_int *decSizeBytes /* O Number of bytes in SILK decoder state */
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)
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{
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opus_int ret = SILK_NO_ERROR;
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*decSizeBytes = sizeof( silk_decoder );
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return ret;
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}
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/* Reset decoder state */
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opus_int silk_InitDecoder( /* O Returns error code */
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void *decState /* I/O State */
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)
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{
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opus_int n, ret = SILK_NO_ERROR;
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silk_decoder_state *channel_state = ((silk_decoder *)decState)->channel_state;
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for( n = 0; n < DECODER_NUM_CHANNELS; n++ ) {
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ret = silk_init_decoder( &channel_state[ n ] );
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}
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silk_memset(&((silk_decoder *)decState)->sStereo, 0, sizeof(((silk_decoder *)decState)->sStereo));
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/* Not strictly needed, but it's cleaner that way */
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((silk_decoder *)decState)->prev_decode_only_middle = 0;
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return ret;
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}
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/* Decode a frame */
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opus_int silk_Decode( /* O Returns error code */
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void* decState, /* I/O State */
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silk_DecControlStruct* decControl, /* I/O Control Structure */
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opus_int lostFlag, /* I 0: no loss, 1 loss, 2 decode fec */
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opus_int newPacketFlag, /* I Indicates first decoder call for this packet */
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ec_dec *psRangeDec, /* I/O Compressor data structure */
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opus_int16 *samplesOut, /* O Decoded output speech vector */
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opus_int32 *nSamplesOut, /* O Number of samples decoded */
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int arch /* I Run-time architecture */
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)
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{
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opus_int i, n, decode_only_middle = 0, ret = SILK_NO_ERROR;
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opus_int32 nSamplesOutDec, LBRR_symbol;
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opus_int16 *samplesOut1_tmp[ 2 ];
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VARDECL( opus_int16, samplesOut1_tmp_storage1 );
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VARDECL( opus_int16, samplesOut1_tmp_storage2 );
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VARDECL( opus_int16, samplesOut2_tmp );
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opus_int32 MS_pred_Q13[ 2 ] = { 0 };
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opus_int16 *resample_out_ptr;
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silk_decoder *psDec = ( silk_decoder * )decState;
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silk_decoder_state *channel_state = psDec->channel_state;
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opus_int has_side;
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opus_int stereo_to_mono;
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int delay_stack_alloc;
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SAVE_STACK;
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silk_assert( decControl->nChannelsInternal == 1 || decControl->nChannelsInternal == 2 );
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/**********************************/
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/* Test if first frame in payload */
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/**********************************/
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if( newPacketFlag ) {
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for( n = 0; n < decControl->nChannelsInternal; n++ ) {
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channel_state[ n ].nFramesDecoded = 0; /* Used to count frames in packet */
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}
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}
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/* If Mono -> Stereo transition in bitstream: init state of second channel */
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if( decControl->nChannelsInternal > psDec->nChannelsInternal ) {
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ret += silk_init_decoder( &channel_state[ 1 ] );
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}
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stereo_to_mono = decControl->nChannelsInternal == 1 && psDec->nChannelsInternal == 2 &&
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( decControl->internalSampleRate == 1000*channel_state[ 0 ].fs_kHz );
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if( channel_state[ 0 ].nFramesDecoded == 0 ) {
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for( n = 0; n < decControl->nChannelsInternal; n++ ) {
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opus_int fs_kHz_dec;
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if( decControl->payloadSize_ms == 0 ) {
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/* Assuming packet loss, use 10 ms */
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channel_state[ n ].nFramesPerPacket = 1;
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channel_state[ n ].nb_subfr = 2;
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} else if( decControl->payloadSize_ms == 10 ) {
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channel_state[ n ].nFramesPerPacket = 1;
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channel_state[ n ].nb_subfr = 2;
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} else if( decControl->payloadSize_ms == 20 ) {
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channel_state[ n ].nFramesPerPacket = 1;
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channel_state[ n ].nb_subfr = 4;
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} else if( decControl->payloadSize_ms == 40 ) {
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channel_state[ n ].nFramesPerPacket = 2;
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channel_state[ n ].nb_subfr = 4;
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} else if( decControl->payloadSize_ms == 60 ) {
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channel_state[ n ].nFramesPerPacket = 3;
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channel_state[ n ].nb_subfr = 4;
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} else {
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silk_assert( 0 );
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RESTORE_STACK;
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return SILK_DEC_INVALID_FRAME_SIZE;
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}
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fs_kHz_dec = ( decControl->internalSampleRate >> 10 ) + 1;
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if( fs_kHz_dec != 8 && fs_kHz_dec != 12 && fs_kHz_dec != 16 ) {
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silk_assert( 0 );
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RESTORE_STACK;
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return SILK_DEC_INVALID_SAMPLING_FREQUENCY;
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}
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ret += silk_decoder_set_fs( &channel_state[ n ], fs_kHz_dec, decControl->API_sampleRate );
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}
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}
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if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 2 && ( psDec->nChannelsAPI == 1 || psDec->nChannelsInternal == 1 ) ) {
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silk_memset( psDec->sStereo.pred_prev_Q13, 0, sizeof( psDec->sStereo.pred_prev_Q13 ) );
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silk_memset( psDec->sStereo.sSide, 0, sizeof( psDec->sStereo.sSide ) );
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silk_memcpy( &channel_state[ 1 ].resampler_state, &channel_state[ 0 ].resampler_state, sizeof( silk_resampler_state_struct ) );
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}
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psDec->nChannelsAPI = decControl->nChannelsAPI;
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psDec->nChannelsInternal = decControl->nChannelsInternal;
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if( decControl->API_sampleRate > (opus_int32)MAX_API_FS_KHZ * 1000 || decControl->API_sampleRate < 8000 ) {
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ret = SILK_DEC_INVALID_SAMPLING_FREQUENCY;
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RESTORE_STACK;
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return( ret );
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}
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if( lostFlag != FLAG_PACKET_LOST && channel_state[ 0 ].nFramesDecoded == 0 ) {
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/* First decoder call for this payload */
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/* Decode VAD flags and LBRR flag */
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for( n = 0; n < decControl->nChannelsInternal; n++ ) {
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for( i = 0; i < channel_state[ n ].nFramesPerPacket; i++ ) {
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channel_state[ n ].VAD_flags[ i ] = ec_dec_bit_logp(psRangeDec, 1);
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}
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channel_state[ n ].LBRR_flag = ec_dec_bit_logp(psRangeDec, 1);
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}
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/* Decode LBRR flags */
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for( n = 0; n < decControl->nChannelsInternal; n++ ) {
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silk_memset( channel_state[ n ].LBRR_flags, 0, sizeof( channel_state[ n ].LBRR_flags ) );
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if( channel_state[ n ].LBRR_flag ) {
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if( channel_state[ n ].nFramesPerPacket == 1 ) {
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channel_state[ n ].LBRR_flags[ 0 ] = 1;
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} else {
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LBRR_symbol = ec_dec_icdf( psRangeDec, silk_LBRR_flags_iCDF_ptr[ channel_state[ n ].nFramesPerPacket - 2 ], 8 ) + 1;
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for( i = 0; i < channel_state[ n ].nFramesPerPacket; i++ ) {
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channel_state[ n ].LBRR_flags[ i ] = silk_RSHIFT( LBRR_symbol, i ) & 1;
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}
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}
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}
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}
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if( lostFlag == FLAG_DECODE_NORMAL ) {
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/* Regular decoding: skip all LBRR data */
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for( i = 0; i < channel_state[ 0 ].nFramesPerPacket; i++ ) {
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for( n = 0; n < decControl->nChannelsInternal; n++ ) {
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if( channel_state[ n ].LBRR_flags[ i ] ) {
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opus_int16 pulses[ MAX_FRAME_LENGTH ];
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opus_int condCoding;
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if( decControl->nChannelsInternal == 2 && n == 0 ) {
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silk_stereo_decode_pred( psRangeDec, MS_pred_Q13 );
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if( channel_state[ 1 ].LBRR_flags[ i ] == 0 ) {
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silk_stereo_decode_mid_only( psRangeDec, &decode_only_middle );
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}
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}
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/* Use conditional coding if previous frame available */
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if( i > 0 && channel_state[ n ].LBRR_flags[ i - 1 ] ) {
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condCoding = CODE_CONDITIONALLY;
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} else {
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condCoding = CODE_INDEPENDENTLY;
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}
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silk_decode_indices( &channel_state[ n ], psRangeDec, i, 1, condCoding );
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silk_decode_pulses( psRangeDec, pulses, channel_state[ n ].indices.signalType,
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channel_state[ n ].indices.quantOffsetType, channel_state[ n ].frame_length );
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}
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}
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}
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}
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}
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/* Get MS predictor index */
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if( decControl->nChannelsInternal == 2 ) {
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if( lostFlag == FLAG_DECODE_NORMAL ||
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( lostFlag == FLAG_DECODE_LBRR && channel_state[ 0 ].LBRR_flags[ channel_state[ 0 ].nFramesDecoded ] == 1 ) )
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{
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silk_stereo_decode_pred( psRangeDec, MS_pred_Q13 );
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/* For LBRR data, decode mid-only flag only if side-channel's LBRR flag is false */
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if( ( lostFlag == FLAG_DECODE_NORMAL && channel_state[ 1 ].VAD_flags[ channel_state[ 0 ].nFramesDecoded ] == 0 ) ||
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( lostFlag == FLAG_DECODE_LBRR && channel_state[ 1 ].LBRR_flags[ channel_state[ 0 ].nFramesDecoded ] == 0 ) )
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{
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silk_stereo_decode_mid_only( psRangeDec, &decode_only_middle );
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} else {
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decode_only_middle = 0;
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}
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} else {
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for( n = 0; n < 2; n++ ) {
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MS_pred_Q13[ n ] = psDec->sStereo.pred_prev_Q13[ n ];
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}
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}
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}
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/* Reset side channel decoder prediction memory for first frame with side coding */
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if( decControl->nChannelsInternal == 2 && decode_only_middle == 0 && psDec->prev_decode_only_middle == 1 ) {
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silk_memset( psDec->channel_state[ 1 ].outBuf, 0, sizeof(psDec->channel_state[ 1 ].outBuf) );
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silk_memset( psDec->channel_state[ 1 ].sLPC_Q14_buf, 0, sizeof(psDec->channel_state[ 1 ].sLPC_Q14_buf) );
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psDec->channel_state[ 1 ].lagPrev = 100;
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psDec->channel_state[ 1 ].LastGainIndex = 10;
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psDec->channel_state[ 1 ].prevSignalType = TYPE_NO_VOICE_ACTIVITY;
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psDec->channel_state[ 1 ].first_frame_after_reset = 1;
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}
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/* Check if the temp buffer fits into the output PCM buffer. If it fits,
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we can delay allocating the temp buffer until after the SILK peak stack
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usage. We need to use a < and not a <= because of the two extra samples. */
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delay_stack_alloc = decControl->internalSampleRate*decControl->nChannelsInternal
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< decControl->API_sampleRate*decControl->nChannelsAPI;
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ALLOC( samplesOut1_tmp_storage1, delay_stack_alloc ? ALLOC_NONE
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: decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2 ),
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opus_int16 );
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if ( delay_stack_alloc )
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{
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samplesOut1_tmp[ 0 ] = samplesOut;
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samplesOut1_tmp[ 1 ] = samplesOut + channel_state[ 0 ].frame_length + 2;
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} else {
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samplesOut1_tmp[ 0 ] = samplesOut1_tmp_storage1;
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samplesOut1_tmp[ 1 ] = samplesOut1_tmp_storage1 + channel_state[ 0 ].frame_length + 2;
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}
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if( lostFlag == FLAG_DECODE_NORMAL ) {
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has_side = !decode_only_middle;
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} else {
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has_side = !psDec->prev_decode_only_middle
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|| (decControl->nChannelsInternal == 2 && lostFlag == FLAG_DECODE_LBRR && channel_state[1].LBRR_flags[ channel_state[1].nFramesDecoded ] == 1 );
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}
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/* Call decoder for one frame */
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for( n = 0; n < decControl->nChannelsInternal; n++ ) {
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if( n == 0 || has_side ) {
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opus_int FrameIndex;
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opus_int condCoding;
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FrameIndex = channel_state[ 0 ].nFramesDecoded - n;
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/* Use independent coding if no previous frame available */
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if( FrameIndex <= 0 ) {
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condCoding = CODE_INDEPENDENTLY;
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} else if( lostFlag == FLAG_DECODE_LBRR ) {
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condCoding = channel_state[ n ].LBRR_flags[ FrameIndex - 1 ] ? CODE_CONDITIONALLY : CODE_INDEPENDENTLY;
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} else if( n > 0 && psDec->prev_decode_only_middle ) {
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/* If we skipped a side frame in this packet, we don't
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need LTP scaling; the LTP state is well-defined. */
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condCoding = CODE_INDEPENDENTLY_NO_LTP_SCALING;
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} else {
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condCoding = CODE_CONDITIONALLY;
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}
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ret += silk_decode_frame( &channel_state[ n ], psRangeDec, &samplesOut1_tmp[ n ][ 2 ], &nSamplesOutDec, lostFlag, condCoding, arch);
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} else {
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silk_memset( &samplesOut1_tmp[ n ][ 2 ], 0, nSamplesOutDec * sizeof( opus_int16 ) );
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}
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channel_state[ n ].nFramesDecoded++;
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}
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if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 2 ) {
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/* Convert Mid/Side to Left/Right */
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silk_stereo_MS_to_LR( &psDec->sStereo, samplesOut1_tmp[ 0 ], samplesOut1_tmp[ 1 ], MS_pred_Q13, channel_state[ 0 ].fs_kHz, nSamplesOutDec );
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} else {
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/* Buffering */
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silk_memcpy( samplesOut1_tmp[ 0 ], psDec->sStereo.sMid, 2 * sizeof( opus_int16 ) );
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silk_memcpy( psDec->sStereo.sMid, &samplesOut1_tmp[ 0 ][ nSamplesOutDec ], 2 * sizeof( opus_int16 ) );
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}
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/* Number of output samples */
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*nSamplesOut = silk_DIV32( nSamplesOutDec * decControl->API_sampleRate, silk_SMULBB( channel_state[ 0 ].fs_kHz, 1000 ) );
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/* Set up pointers to temp buffers */
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ALLOC( samplesOut2_tmp,
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decControl->nChannelsAPI == 2 ? *nSamplesOut : ALLOC_NONE, opus_int16 );
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if( decControl->nChannelsAPI == 2 ) {
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resample_out_ptr = samplesOut2_tmp;
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} else {
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resample_out_ptr = samplesOut;
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}
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ALLOC( samplesOut1_tmp_storage2, delay_stack_alloc
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? decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2 )
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: ALLOC_NONE,
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opus_int16 );
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if ( delay_stack_alloc ) {
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OPUS_COPY(samplesOut1_tmp_storage2, samplesOut, decControl->nChannelsInternal*(channel_state[ 0 ].frame_length + 2));
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samplesOut1_tmp[ 0 ] = samplesOut1_tmp_storage2;
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samplesOut1_tmp[ 1 ] = samplesOut1_tmp_storage2 + channel_state[ 0 ].frame_length + 2;
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}
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for( n = 0; n < silk_min( decControl->nChannelsAPI, decControl->nChannelsInternal ); n++ ) {
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/* Resample decoded signal to API_sampleRate */
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ret += silk_resampler( &channel_state[ n ].resampler_state, resample_out_ptr, &samplesOut1_tmp[ n ][ 1 ], nSamplesOutDec );
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/* Interleave if stereo output and stereo stream */
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if( decControl->nChannelsAPI == 2 ) {
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for( i = 0; i < *nSamplesOut; i++ ) {
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samplesOut[ n + 2 * i ] = resample_out_ptr[ i ];
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}
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}
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}
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/* Create two channel output from mono stream */
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if( decControl->nChannelsAPI == 2 && decControl->nChannelsInternal == 1 ) {
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if ( stereo_to_mono ){
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/* Resample right channel for newly collapsed stereo just in case
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we weren't doing collapsing when switching to mono */
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ret += silk_resampler( &channel_state[ 1 ].resampler_state, resample_out_ptr, &samplesOut1_tmp[ 0 ][ 1 ], nSamplesOutDec );
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for( i = 0; i < *nSamplesOut; i++ ) {
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samplesOut[ 1 + 2 * i ] = resample_out_ptr[ i ];
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}
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} else {
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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
|