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diff --git a/media/libopus/silk/stereo_LR_to_MS.c b/media/libopus/silk/stereo_LR_to_MS.c
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+/***********************************************************************
+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 "main.h"
+#include "stack_alloc.h"
+
+/* Convert Left/Right stereo signal to adaptive Mid/Side representation */
+void silk_stereo_LR_to_MS(
+ stereo_enc_state *state, /* I/O State */
+ opus_int16 x1[], /* I/O Left input signal, becomes mid signal */
+ opus_int16 x2[], /* I/O Right input signal, becomes side signal */
+ opus_int8 ix[ 2 ][ 3 ], /* O Quantization indices */
+ opus_int8 *mid_only_flag, /* O Flag: only mid signal coded */
+ opus_int32 mid_side_rates_bps[], /* O Bitrates for mid and side signals */
+ opus_int32 total_rate_bps, /* I Total bitrate */
+ opus_int prev_speech_act_Q8, /* I Speech activity level in previous frame */
+ opus_int toMono, /* I Last frame before a stereo->mono transition */
+ opus_int fs_kHz, /* I Sample rate (kHz) */
+ opus_int frame_length /* I Number of samples */
+)
+{
+ opus_int n, is10msFrame, denom_Q16, delta0_Q13, delta1_Q13;
+ opus_int32 sum, diff, smooth_coef_Q16, pred_Q13[ 2 ], pred0_Q13, pred1_Q13;
+ opus_int32 LP_ratio_Q14, HP_ratio_Q14, frac_Q16, frac_3_Q16, min_mid_rate_bps, width_Q14, w_Q24, deltaw_Q24;
+ VARDECL( opus_int16, side );
+ VARDECL( opus_int16, LP_mid );
+ VARDECL( opus_int16, HP_mid );
+ VARDECL( opus_int16, LP_side );
+ VARDECL( opus_int16, HP_side );
+ opus_int16 *mid = &x1[ -2 ];
+ SAVE_STACK;
+
+ ALLOC( side, frame_length + 2, opus_int16 );
+ /* Convert to basic mid/side signals */
+ for( n = 0; n < frame_length + 2; n++ ) {
+ sum = x1[ n - 2 ] + (opus_int32)x2[ n - 2 ];
+ diff = x1[ n - 2 ] - (opus_int32)x2[ n - 2 ];
+ mid[ n ] = (opus_int16)silk_RSHIFT_ROUND( sum, 1 );
+ side[ n ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( diff, 1 ) );
+ }
+
+ /* Buffering */
+ silk_memcpy( mid, state->sMid, 2 * sizeof( opus_int16 ) );
+ silk_memcpy( side, state->sSide, 2 * sizeof( opus_int16 ) );
+ silk_memcpy( state->sMid, &mid[ frame_length ], 2 * sizeof( opus_int16 ) );
+ silk_memcpy( state->sSide, &side[ frame_length ], 2 * sizeof( opus_int16 ) );
+
+ /* LP and HP filter mid signal */
+ ALLOC( LP_mid, frame_length, opus_int16 );
+ ALLOC( HP_mid, frame_length, opus_int16 );
+ for( n = 0; n < frame_length; n++ ) {
+ sum = silk_RSHIFT_ROUND( silk_ADD_LSHIFT32( mid[ n ] + (opus_int32)mid[ n + 2 ], mid[ n + 1 ], 1 ), 2 );
+ LP_mid[ n ] = sum;
+ HP_mid[ n ] = mid[ n + 1 ] - sum;
+ }
+
+ /* LP and HP filter side signal */
+ ALLOC( LP_side, frame_length, opus_int16 );
+ ALLOC( HP_side, frame_length, opus_int16 );
+ for( n = 0; n < frame_length; n++ ) {
+ sum = silk_RSHIFT_ROUND( silk_ADD_LSHIFT32( side[ n ] + (opus_int32)side[ n + 2 ], side[ n + 1 ], 1 ), 2 );
+ LP_side[ n ] = sum;
+ HP_side[ n ] = side[ n + 1 ] - sum;
+ }
+
+ /* Find energies and predictors */
+ is10msFrame = frame_length == 10 * fs_kHz;
+ smooth_coef_Q16 = is10msFrame ?
+ SILK_FIX_CONST( STEREO_RATIO_SMOOTH_COEF / 2, 16 ) :
+ SILK_FIX_CONST( STEREO_RATIO_SMOOTH_COEF, 16 );
+ smooth_coef_Q16 = silk_SMULWB( silk_SMULBB( prev_speech_act_Q8, prev_speech_act_Q8 ), smooth_coef_Q16 );
+
+ pred_Q13[ 0 ] = silk_stereo_find_predictor( &LP_ratio_Q14, LP_mid, LP_side, &state->mid_side_amp_Q0[ 0 ], frame_length, smooth_coef_Q16 );
+ pred_Q13[ 1 ] = silk_stereo_find_predictor( &HP_ratio_Q14, HP_mid, HP_side, &state->mid_side_amp_Q0[ 2 ], frame_length, smooth_coef_Q16 );
+ /* Ratio of the norms of residual and mid signals */
+ frac_Q16 = silk_SMLABB( HP_ratio_Q14, LP_ratio_Q14, 3 );
+ frac_Q16 = silk_min( frac_Q16, SILK_FIX_CONST( 1, 16 ) );
+
+ /* Determine bitrate distribution between mid and side, and possibly reduce stereo width */
+ total_rate_bps -= is10msFrame ? 1200 : 600; /* Subtract approximate bitrate for coding stereo parameters */
+ if( total_rate_bps < 1 ) {
+ total_rate_bps = 1;
+ }
+ min_mid_rate_bps = silk_SMLABB( 2000, fs_kHz, 600 );
+ silk_assert( min_mid_rate_bps < 32767 );
+ /* Default bitrate distribution: 8 parts for Mid and (5+3*frac) parts for Side. so: mid_rate = ( 8 / ( 13 + 3 * frac ) ) * total_ rate */
+ frac_3_Q16 = silk_MUL( 3, frac_Q16 );
+ mid_side_rates_bps[ 0 ] = silk_DIV32_varQ( total_rate_bps, SILK_FIX_CONST( 8 + 5, 16 ) + frac_3_Q16, 16+3 );
+ /* If Mid bitrate below minimum, reduce stereo width */
+ if( mid_side_rates_bps[ 0 ] < min_mid_rate_bps ) {
+ mid_side_rates_bps[ 0 ] = min_mid_rate_bps;
+ mid_side_rates_bps[ 1 ] = total_rate_bps - mid_side_rates_bps[ 0 ];
+ /* width = 4 * ( 2 * side_rate - min_rate ) / ( ( 1 + 3 * frac ) * min_rate ) */
+ width_Q14 = silk_DIV32_varQ( silk_LSHIFT( mid_side_rates_bps[ 1 ], 1 ) - min_mid_rate_bps,
+ silk_SMULWB( SILK_FIX_CONST( 1, 16 ) + frac_3_Q16, min_mid_rate_bps ), 14+2 );
+ width_Q14 = silk_LIMIT( width_Q14, 0, SILK_FIX_CONST( 1, 14 ) );
+ } else {
+ mid_side_rates_bps[ 1 ] = total_rate_bps - mid_side_rates_bps[ 0 ];
+ width_Q14 = SILK_FIX_CONST( 1, 14 );
+ }
+
+ /* Smoother */
+ state->smth_width_Q14 = (opus_int16)silk_SMLAWB( state->smth_width_Q14, width_Q14 - state->smth_width_Q14, smooth_coef_Q16 );
+
+ /* At very low bitrates or for inputs that are nearly amplitude panned, switch to panned-mono coding */
+ *mid_only_flag = 0;
+ if( toMono ) {
+ /* Last frame before stereo->mono transition; collapse stereo width */
+ width_Q14 = 0;
+ pred_Q13[ 0 ] = 0;
+ pred_Q13[ 1 ] = 0;
+ silk_stereo_quant_pred( pred_Q13, ix );
+ } else if( state->width_prev_Q14 == 0 &&
+ ( 8 * total_rate_bps < 13 * min_mid_rate_bps || silk_SMULWB( frac_Q16, state->smth_width_Q14 ) < SILK_FIX_CONST( 0.05, 14 ) ) )
+ {
+ /* Code as panned-mono; previous frame already had zero width */
+ /* Scale down and quantize predictors */
+ pred_Q13[ 0 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 0 ] ), 14 );
+ pred_Q13[ 1 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 1 ] ), 14 );
+ silk_stereo_quant_pred( pred_Q13, ix );
+ /* Collapse stereo width */
+ width_Q14 = 0;
+ pred_Q13[ 0 ] = 0;
+ pred_Q13[ 1 ] = 0;
+ mid_side_rates_bps[ 0 ] = total_rate_bps;
+ mid_side_rates_bps[ 1 ] = 0;
+ *mid_only_flag = 1;
+ } else if( state->width_prev_Q14 != 0 &&
+ ( 8 * total_rate_bps < 11 * min_mid_rate_bps || silk_SMULWB( frac_Q16, state->smth_width_Q14 ) < SILK_FIX_CONST( 0.02, 14 ) ) )
+ {
+ /* Transition to zero-width stereo */
+ /* Scale down and quantize predictors */
+ pred_Q13[ 0 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 0 ] ), 14 );
+ pred_Q13[ 1 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 1 ] ), 14 );
+ silk_stereo_quant_pred( pred_Q13, ix );
+ /* Collapse stereo width */
+ width_Q14 = 0;
+ pred_Q13[ 0 ] = 0;
+ pred_Q13[ 1 ] = 0;
+ } else if( state->smth_width_Q14 > SILK_FIX_CONST( 0.95, 14 ) ) {
+ /* Full-width stereo coding */
+ silk_stereo_quant_pred( pred_Q13, ix );
+ width_Q14 = SILK_FIX_CONST( 1, 14 );
+ } else {
+ /* Reduced-width stereo coding; scale down and quantize predictors */
+ pred_Q13[ 0 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 0 ] ), 14 );
+ pred_Q13[ 1 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 1 ] ), 14 );
+ silk_stereo_quant_pred( pred_Q13, ix );
+ width_Q14 = state->smth_width_Q14;
+ }
+
+ /* Make sure to keep on encoding until the tapered output has been transmitted */
+ if( *mid_only_flag == 1 ) {
+ state->silent_side_len += frame_length - STEREO_INTERP_LEN_MS * fs_kHz;
+ if( state->silent_side_len < LA_SHAPE_MS * fs_kHz ) {
+ *mid_only_flag = 0;
+ } else {
+ /* Limit to avoid wrapping around */
+ state->silent_side_len = 10000;
+ }
+ } else {
+ state->silent_side_len = 0;
+ }
+
+ if( *mid_only_flag == 0 && mid_side_rates_bps[ 1 ] < 1 ) {
+ mid_side_rates_bps[ 1 ] = 1;
+ mid_side_rates_bps[ 0 ] = silk_max_int( 1, total_rate_bps - mid_side_rates_bps[ 1 ]);
+ }
+
+ /* Interpolate predictors and subtract prediction from side channel */
+ pred0_Q13 = -state->pred_prev_Q13[ 0 ];
+ pred1_Q13 = -state->pred_prev_Q13[ 1 ];
+ w_Q24 = silk_LSHIFT( state->width_prev_Q14, 10 );
+ denom_Q16 = silk_DIV32_16( (opus_int32)1 << 16, STEREO_INTERP_LEN_MS * fs_kHz );
+ delta0_Q13 = -silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 0 ] - state->pred_prev_Q13[ 0 ], denom_Q16 ), 16 );
+ delta1_Q13 = -silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 1 ] - state->pred_prev_Q13[ 1 ], denom_Q16 ), 16 );
+ deltaw_Q24 = silk_LSHIFT( silk_SMULWB( width_Q14 - state->width_prev_Q14, denom_Q16 ), 10 );
+ for( n = 0; n < STEREO_INTERP_LEN_MS * fs_kHz; n++ ) {
+ pred0_Q13 += delta0_Q13;
+ pred1_Q13 += delta1_Q13;
+ w_Q24 += deltaw_Q24;
+ sum = silk_LSHIFT( silk_ADD_LSHIFT32( mid[ n ] + (opus_int32)mid[ n + 2 ], mid[ n + 1 ], 1 ), 9 ); /* Q11 */
+ sum = silk_SMLAWB( silk_SMULWB( w_Q24, side[ n + 1 ] ), sum, pred0_Q13 ); /* Q8 */
+ sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)mid[ n + 1 ], 11 ), pred1_Q13 ); /* Q8 */
+ x2[ n - 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) );
+ }
+
+ pred0_Q13 = -pred_Q13[ 0 ];
+ pred1_Q13 = -pred_Q13[ 1 ];
+ w_Q24 = silk_LSHIFT( width_Q14, 10 );
+ for( n = STEREO_INTERP_LEN_MS * fs_kHz; n < frame_length; n++ ) {
+ sum = silk_LSHIFT( silk_ADD_LSHIFT32( mid[ n ] + (opus_int32)mid[ n + 2 ], mid[ n + 1 ], 1 ), 9 ); /* Q11 */
+ sum = silk_SMLAWB( silk_SMULWB( w_Q24, side[ n + 1 ] ), sum, pred0_Q13 ); /* Q8 */
+ sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)mid[ n + 1 ], 11 ), pred1_Q13 ); /* Q8 */
+ x2[ n - 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) );
+ }
+ state->pred_prev_Q13[ 0 ] = (opus_int16)pred_Q13[ 0 ];
+ state->pred_prev_Q13[ 1 ] = (opus_int16)pred_Q13[ 1 ];
+ state->width_prev_Q14 = (opus_int16)width_Q14;
+ RESTORE_STACK;
+}
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-11 09:15:14 +0000