46ae64cd60
This reverts commit e00426c512.
The way we handle platform-specific intrinsics is not good, so the
current state will not compile on armv8. This commit also requires
SSE4.1 support, which is likely not a good idea for portable binaries.
We'll have to redo this with more caution after 3.2 is released, or
we might simply drop opus as we're only using it as dependency for
theora right now.
Fixes #33606.
229 lines
12 KiB
C
229 lines
12 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 "main.h"
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#include "stack_alloc.h"
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/* Convert Left/Right stereo signal to adaptive Mid/Side representation */
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void silk_stereo_LR_to_MS(
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stereo_enc_state *state, /* I/O State */
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opus_int16 x1[], /* I/O Left input signal, becomes mid signal */
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opus_int16 x2[], /* I/O Right input signal, becomes side signal */
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opus_int8 ix[ 2 ][ 3 ], /* O Quantization indices */
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opus_int8 *mid_only_flag, /* O Flag: only mid signal coded */
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opus_int32 mid_side_rates_bps[], /* O Bitrates for mid and side signals */
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opus_int32 total_rate_bps, /* I Total bitrate */
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opus_int prev_speech_act_Q8, /* I Speech activity level in previous frame */
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opus_int toMono, /* I Last frame before a stereo->mono transition */
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opus_int fs_kHz, /* I Sample rate (kHz) */
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opus_int frame_length /* I Number of samples */
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)
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{
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opus_int n, is10msFrame, denom_Q16, delta0_Q13, delta1_Q13;
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opus_int32 sum, diff, smooth_coef_Q16, pred_Q13[ 2 ], pred0_Q13, pred1_Q13;
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opus_int32 LP_ratio_Q14, HP_ratio_Q14, frac_Q16, frac_3_Q16, min_mid_rate_bps, width_Q14, w_Q24, deltaw_Q24;
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VARDECL( opus_int16, side );
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VARDECL( opus_int16, LP_mid );
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VARDECL( opus_int16, HP_mid );
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VARDECL( opus_int16, LP_side );
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VARDECL( opus_int16, HP_side );
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opus_int16 *mid = &x1[ -2 ];
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SAVE_STACK;
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ALLOC( side, frame_length + 2, opus_int16 );
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/* Convert to basic mid/side signals */
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for( n = 0; n < frame_length + 2; n++ ) {
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sum = x1[ n - 2 ] + (opus_int32)x2[ n - 2 ];
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diff = x1[ n - 2 ] - (opus_int32)x2[ n - 2 ];
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mid[ n ] = (opus_int16)silk_RSHIFT_ROUND( sum, 1 );
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side[ n ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( diff, 1 ) );
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}
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/* Buffering */
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silk_memcpy( mid, state->sMid, 2 * sizeof( opus_int16 ) );
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silk_memcpy( side, state->sSide, 2 * sizeof( opus_int16 ) );
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silk_memcpy( state->sMid, &mid[ frame_length ], 2 * sizeof( opus_int16 ) );
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silk_memcpy( state->sSide, &side[ frame_length ], 2 * sizeof( opus_int16 ) );
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/* LP and HP filter mid signal */
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ALLOC( LP_mid, frame_length, opus_int16 );
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ALLOC( HP_mid, frame_length, opus_int16 );
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for( n = 0; n < frame_length; n++ ) {
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sum = silk_RSHIFT_ROUND( silk_ADD_LSHIFT( mid[ n ] + (opus_int32)mid[ n + 2 ], mid[ n + 1 ], 1 ), 2 );
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LP_mid[ n ] = sum;
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HP_mid[ n ] = mid[ n + 1 ] - sum;
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}
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/* LP and HP filter side signal */
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ALLOC( LP_side, frame_length, opus_int16 );
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ALLOC( HP_side, frame_length, opus_int16 );
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for( n = 0; n < frame_length; n++ ) {
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sum = silk_RSHIFT_ROUND( silk_ADD_LSHIFT( side[ n ] + (opus_int32)side[ n + 2 ], side[ n + 1 ], 1 ), 2 );
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LP_side[ n ] = sum;
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HP_side[ n ] = side[ n + 1 ] - sum;
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}
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/* Find energies and predictors */
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is10msFrame = frame_length == 10 * fs_kHz;
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smooth_coef_Q16 = is10msFrame ?
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SILK_FIX_CONST( STEREO_RATIO_SMOOTH_COEF / 2, 16 ) :
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SILK_FIX_CONST( STEREO_RATIO_SMOOTH_COEF, 16 );
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smooth_coef_Q16 = silk_SMULWB( silk_SMULBB( prev_speech_act_Q8, prev_speech_act_Q8 ), smooth_coef_Q16 );
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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 );
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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 );
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/* Ratio of the norms of residual and mid signals */
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frac_Q16 = silk_SMLABB( HP_ratio_Q14, LP_ratio_Q14, 3 );
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frac_Q16 = silk_min( frac_Q16, SILK_FIX_CONST( 1, 16 ) );
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/* Determine bitrate distribution between mid and side, and possibly reduce stereo width */
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total_rate_bps -= is10msFrame ? 1200 : 600; /* Subtract approximate bitrate for coding stereo parameters */
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if( total_rate_bps < 1 ) {
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total_rate_bps = 1;
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}
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min_mid_rate_bps = silk_SMLABB( 2000, fs_kHz, 900 );
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silk_assert( min_mid_rate_bps < 32767 );
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/* Default bitrate distribution: 8 parts for Mid and (5+3*frac) parts for Side. so: mid_rate = ( 8 / ( 13 + 3 * frac ) ) * total_ rate */
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frac_3_Q16 = silk_MUL( 3, frac_Q16 );
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mid_side_rates_bps[ 0 ] = silk_DIV32_varQ( total_rate_bps, SILK_FIX_CONST( 8 + 5, 16 ) + frac_3_Q16, 16+3 );
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/* If Mid bitrate below minimum, reduce stereo width */
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if( mid_side_rates_bps[ 0 ] < min_mid_rate_bps ) {
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mid_side_rates_bps[ 0 ] = min_mid_rate_bps;
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mid_side_rates_bps[ 1 ] = total_rate_bps - mid_side_rates_bps[ 0 ];
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/* width = 4 * ( 2 * side_rate - min_rate ) / ( ( 1 + 3 * frac ) * min_rate ) */
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width_Q14 = silk_DIV32_varQ( silk_LSHIFT( mid_side_rates_bps[ 1 ], 1 ) - min_mid_rate_bps,
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silk_SMULWB( SILK_FIX_CONST( 1, 16 ) + frac_3_Q16, min_mid_rate_bps ), 14+2 );
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width_Q14 = silk_LIMIT( width_Q14, 0, SILK_FIX_CONST( 1, 14 ) );
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} else {
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mid_side_rates_bps[ 1 ] = total_rate_bps - mid_side_rates_bps[ 0 ];
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width_Q14 = SILK_FIX_CONST( 1, 14 );
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}
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/* Smoother */
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state->smth_width_Q14 = (opus_int16)silk_SMLAWB( state->smth_width_Q14, width_Q14 - state->smth_width_Q14, smooth_coef_Q16 );
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/* At very low bitrates or for inputs that are nearly amplitude panned, switch to panned-mono coding */
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*mid_only_flag = 0;
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if( toMono ) {
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/* Last frame before stereo->mono transition; collapse stereo width */
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width_Q14 = 0;
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pred_Q13[ 0 ] = 0;
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pred_Q13[ 1 ] = 0;
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silk_stereo_quant_pred( pred_Q13, ix );
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} else if( state->width_prev_Q14 == 0 &&
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( 8 * total_rate_bps < 13 * min_mid_rate_bps || silk_SMULWB( frac_Q16, state->smth_width_Q14 ) < SILK_FIX_CONST( 0.05, 14 ) ) )
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{
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/* Code as panned-mono; previous frame already had zero width */
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/* Scale down and quantize predictors */
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pred_Q13[ 0 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 0 ] ), 14 );
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pred_Q13[ 1 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 1 ] ), 14 );
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silk_stereo_quant_pred( pred_Q13, ix );
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/* Collapse stereo width */
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width_Q14 = 0;
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pred_Q13[ 0 ] = 0;
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pred_Q13[ 1 ] = 0;
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mid_side_rates_bps[ 0 ] = total_rate_bps;
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mid_side_rates_bps[ 1 ] = 0;
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*mid_only_flag = 1;
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} else if( state->width_prev_Q14 != 0 &&
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( 8 * total_rate_bps < 11 * min_mid_rate_bps || silk_SMULWB( frac_Q16, state->smth_width_Q14 ) < SILK_FIX_CONST( 0.02, 14 ) ) )
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{
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/* Transition to zero-width stereo */
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/* Scale down and quantize predictors */
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pred_Q13[ 0 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 0 ] ), 14 );
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pred_Q13[ 1 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 1 ] ), 14 );
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silk_stereo_quant_pred( pred_Q13, ix );
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/* Collapse stereo width */
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width_Q14 = 0;
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pred_Q13[ 0 ] = 0;
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pred_Q13[ 1 ] = 0;
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} else if( state->smth_width_Q14 > SILK_FIX_CONST( 0.95, 14 ) ) {
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/* Full-width stereo coding */
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silk_stereo_quant_pred( pred_Q13, ix );
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width_Q14 = SILK_FIX_CONST( 1, 14 );
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} else {
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/* Reduced-width stereo coding; scale down and quantize predictors */
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pred_Q13[ 0 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 0 ] ), 14 );
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pred_Q13[ 1 ] = silk_RSHIFT( silk_SMULBB( state->smth_width_Q14, pred_Q13[ 1 ] ), 14 );
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silk_stereo_quant_pred( pred_Q13, ix );
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width_Q14 = state->smth_width_Q14;
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}
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/* Make sure to keep on encoding until the tapered output has been transmitted */
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if( *mid_only_flag == 1 ) {
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state->silent_side_len += frame_length - STEREO_INTERP_LEN_MS * fs_kHz;
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if( state->silent_side_len < LA_SHAPE_MS * fs_kHz ) {
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*mid_only_flag = 0;
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} else {
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/* Limit to avoid wrapping around */
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state->silent_side_len = 10000;
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}
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} else {
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state->silent_side_len = 0;
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}
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if( *mid_only_flag == 0 && mid_side_rates_bps[ 1 ] < 1 ) {
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mid_side_rates_bps[ 1 ] = 1;
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mid_side_rates_bps[ 0 ] = silk_max_int( 1, total_rate_bps - mid_side_rates_bps[ 1 ]);
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}
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/* Interpolate predictors and subtract prediction from side channel */
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pred0_Q13 = -state->pred_prev_Q13[ 0 ];
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pred1_Q13 = -state->pred_prev_Q13[ 1 ];
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w_Q24 = silk_LSHIFT( state->width_prev_Q14, 10 );
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denom_Q16 = silk_DIV32_16( (opus_int32)1 << 16, STEREO_INTERP_LEN_MS * fs_kHz );
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delta0_Q13 = -silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 0 ] - state->pred_prev_Q13[ 0 ], denom_Q16 ), 16 );
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delta1_Q13 = -silk_RSHIFT_ROUND( silk_SMULBB( pred_Q13[ 1 ] - state->pred_prev_Q13[ 1 ], denom_Q16 ), 16 );
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deltaw_Q24 = silk_LSHIFT( silk_SMULWB( width_Q14 - state->width_prev_Q14, denom_Q16 ), 10 );
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for( n = 0; n < STEREO_INTERP_LEN_MS * fs_kHz; n++ ) {
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pred0_Q13 += delta0_Q13;
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pred1_Q13 += delta1_Q13;
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w_Q24 += deltaw_Q24;
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sum = silk_LSHIFT( silk_ADD_LSHIFT( mid[ n ] + (opus_int32)mid[ n + 2 ], mid[ n + 1 ], 1 ), 9 ); /* Q11 */
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sum = silk_SMLAWB( silk_SMULWB( w_Q24, side[ n + 1 ] ), sum, pred0_Q13 ); /* Q8 */
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sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)mid[ n + 1 ], 11 ), pred1_Q13 ); /* Q8 */
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x2[ n - 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) );
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}
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pred0_Q13 = -pred_Q13[ 0 ];
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pred1_Q13 = -pred_Q13[ 1 ];
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w_Q24 = silk_LSHIFT( width_Q14, 10 );
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for( n = STEREO_INTERP_LEN_MS * fs_kHz; n < frame_length; n++ ) {
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sum = silk_LSHIFT( silk_ADD_LSHIFT( mid[ n ] + (opus_int32)mid[ n + 2 ], mid[ n + 1 ], 1 ), 9 ); /* Q11 */
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sum = silk_SMLAWB( silk_SMULWB( w_Q24, side[ n + 1 ] ), sum, pred0_Q13 ); /* Q8 */
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sum = silk_SMLAWB( sum, silk_LSHIFT( (opus_int32)mid[ n + 1 ], 11 ), pred1_Q13 ); /* Q8 */
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x2[ n - 1 ] = (opus_int16)silk_SAT16( silk_RSHIFT_ROUND( sum, 8 ) );
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}
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state->pred_prev_Q13[ 0 ] = (opus_int16)pred_Q13[ 0 ];
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state->pred_prev_Q13[ 1 ] = (opus_int16)pred_Q13[ 1 ];
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state->width_prev_Q14 = (opus_int16)width_Q14;
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RESTORE_STACK;
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}
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