diff options
Diffstat (limited to 'media/libopus/silk/stereo_LR_to_MS.c')
| -rw-r--r-- | media/libopus/silk/stereo_LR_to_MS.c | 229 |
1 files changed, 229 insertions, 0 deletions
diff --git a/media/libopus/silk/stereo_LR_to_MS.c b/media/libopus/silk/stereo_LR_to_MS.c new file mode 100644 index 000000000..dda0298de --- /dev/null +++ b/media/libopus/silk/stereo_LR_to_MS.c @@ -0,0 +1,229 @@ +/*********************************************************************** +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_LSHIFT( 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_LSHIFT( 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, 900 ); + 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_LSHIFT( 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_LSHIFT( 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; +} |