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Update opus from 1.1 to 1.1.4

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Define FLOAT_APPROX in Makefile and misc/msvc12/quake3.vcxproj.
This commit is contained in:
Zack Middleton committed 2017-05-23 11:47:26 -05:00
1 parent 67bfe8f8d6
commit 853110d5d4
146 files changed
+11917 -2109
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code/opus-1.1/silk/dec_API.c
+32 -10
View File
@@ -31,6 +31,7 @@ POSSIBILITY OF SUCH DAMAGE.
#include "API.h"
#include "main.h"
#include "stack_alloc.h"
#include "os_support.h"
/************************/
/* Decoder Super Struct */
@@ -84,13 +85,15 @@ opus_int silk_Decode( /* O Returns error co
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_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_storage );
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;
@@ -98,6 +101,7 @@ opus_int silk_Decode( /* O Returns error co
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 );
@@ -196,7 +200,7 @@ opus_int silk_Decode( /* O Returns error co
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_int16 pulses[ MAX_FRAME_LENGTH ];
opus_int condCoding;
if( decControl->nChannelsInternal == 2 && n == 0 ) {
@@ -251,13 +255,22 @@ opus_int silk_Decode( /* O Returns error co
psDec->channel_state[ 1 ].first_frame_after_reset = 1;
}
ALLOC( samplesOut1_tmp_storage,
decControl->nChannelsInternal*(
channel_state[ 0 ].frame_length + 2 ),
/* 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 );
samplesOut1_tmp[ 0 ] = samplesOut1_tmp_storage;
samplesOut1_tmp[ 1 ] = samplesOut1_tmp_storage
+ channel_state[ 0 ].frame_length + 2;
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;
@@ -284,7 +297,7 @@ opus_int silk_Decode( /* O Returns error co
} else {
condCoding = CODE_CONDITIONALLY;
}
ret += silk_decode_frame( &channel_state[ n ], psRangeDec, &samplesOut1_tmp[ n ][ 2 ], &nSamplesOutDec, lostFlag, condCoding);
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 ) );
}
@@ -312,6 +325,15 @@ opus_int silk_Decode( /* O Returns error co
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 */