summaryrefslogtreecommitdiffstats
path: root/media/libopus/silk/dec_API.c
diff options
context:
space:
mode:
authorMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
committerMatt A. Tobin <mattatobin@localhost.localdomain>2018-02-02 04:16:08 -0500
commit5f8de423f190bbb79a62f804151bc24824fa32d8 (patch)
tree10027f336435511475e392454359edea8e25895d /media/libopus/silk/dec_API.c
parent49ee0794b5d912db1f95dce6eb52d781dc210db5 (diff)
downloadUXP-5f8de423f190bbb79a62f804151bc24824fa32d8.tar
UXP-5f8de423f190bbb79a62f804151bc24824fa32d8.tar.gz
UXP-5f8de423f190bbb79a62f804151bc24824fa32d8.tar.lz
UXP-5f8de423f190bbb79a62f804151bc24824fa32d8.tar.xz
UXP-5f8de423f190bbb79a62f804151bc24824fa32d8.zip
Add m-esr52 at 52.6.0
Diffstat (limited to 'media/libopus/silk/dec_API.c')
-rw-r--r--media/libopus/silk/dec_API.c419
1 files changed, 419 insertions, 0 deletions
diff --git a/media/libopus/silk/dec_API.c b/media/libopus/silk/dec_API.c
new file mode 100644
index 000000000..b7d8ed48d
--- /dev/null
+++ b/media/libopus/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