godotengine/godot
1
0
Fork 0
mirror of https://github.com/godotengine/godot.git synced 2025-11-17 14:11:06 +00:00
Code Releases Wiki Activity

Update Opus driver to 1.1.2

Browse Source 
And opusfile to 0.7.
This commit is contained in:
George Marques
2016-05-01 12:48:46 -03:00
parent a3d81cab8a
commit 7c59d819a7
270 changed files with 12814 additions and 5061 deletions
Download Patch File Download Diff File Expand all files Collapse all files

47
drivers/opus/silk/dec_API.c
View File

@@ -24,13 +24,11 @@ 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 OPUS_ENABLED
#include "opus/opus_config.h"
#endif
#include "opus/silk/API.h"
#include "opus/silk/silk_main.h"
#include "opus/silk/main.h"
#include "opus/celt/stack_alloc.h"
#include "opus/celt/os_support.h"
/************************/
/* Decoder Super Struct */
@@ -84,13 +82,15 @@ 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 +98,7 @@ 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 +197,7 @@ 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 +252,22 @@ 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 +294,7 @@ 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 +322,15 @@ 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 */