Index: webrtc/modules/audio_coding/codecs/opus/opus/src/silk/stereo_LR_to_MS.c |
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus/src/silk/stereo_LR_to_MS.c b/webrtc/modules/audio_coding/codecs/opus/opus/src/silk/stereo_LR_to_MS.c |
new file mode 100644 |
index 0000000000000000000000000000000000000000..42906e6f6769a7ab24b98f65409e395acabd1273 |
--- /dev/null |
+++ b/webrtc/modules/audio_coding/codecs/opus/opus/src/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 ] + 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 ] + 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 ] + 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 ] + 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; |
+} |