Remove the extra directories to include that were needed by the Opus library. Now the lib includes more specific paths to avoid those.
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 OPUS_HAVE_CONFIG_H
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#include "opus/opus_config.h"
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#endif
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#include "opus/silk/silk_main.h"
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#include "opus/celt/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 ] + 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 ] + 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 ] + 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 ] + 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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