ref: bafbd08db132d387bcef0637258f90e862a8e4ea
parent: e335065a1ba72c474b2bf9324e789d5e1dc1e884
parent: ec2802210cd0d62514f4cb9bc4db2f7df1259ad4
author: Jean-Marc Valin <[email protected]>
date: Thu Sep 1 17:59:50 EDT 2011
Merge branch 'exp-highpass'
--- a/silk/silk_HP_variable_cutoff.c
+++ b/silk/silk_HP_variable_cutoff.c
@@ -41,85 +41,38 @@
const opus_int nChannels /* I Number of channels */
)
{- opus_int quality_Q15, cutoff_Hz;
- opus_int32 B_Q28[ 3 ], A_Q28[ 2 ];
- opus_int32 Fc_Q19, r_Q28, r_Q22;
- opus_int32 pitch_freq_Hz_Q16, pitch_freq_log_Q7, delta_freq_Q7;
- silk_encoder_state *psEncC1 = &state_Fxx[ 0 ].sCmn;
+ opus_int quality_Q15;
+ opus_int32 pitch_freq_Hz_Q16, pitch_freq_log_Q7, delta_freq_Q7;
+ silk_encoder_state *psEncC1 = &state_Fxx[ 0 ].sCmn;
- if( psEncC1->HP_cutoff_Hz == 0 ) {- /* Adaptive cutoff frequency: estimate low end of pitch frequency range */
- if( psEncC1->prevSignalType == TYPE_VOICED ) {- /* difference, in log domain */
- pitch_freq_Hz_Q16 = SKP_DIV32_16( SKP_LSHIFT( SKP_MUL( psEncC1->fs_kHz, 1000 ), 16 ), psEncC1->prevLag );
- pitch_freq_log_Q7 = silk_lin2log( pitch_freq_Hz_Q16 ) - ( 16 << 7 );
+ /* Adaptive cutoff frequency: estimate low end of pitch frequency range */
+ if( psEncC1->prevSignalType == TYPE_VOICED ) {+ /* difference, in log domain */
+ pitch_freq_Hz_Q16 = SKP_DIV32_16( SKP_LSHIFT( SKP_MUL( psEncC1->fs_kHz, 1000 ), 16 ), psEncC1->prevLag );
+ pitch_freq_log_Q7 = silk_lin2log( pitch_freq_Hz_Q16 ) - ( 16 << 7 );
- /* adjustment based on quality */
- quality_Q15 = psEncC1->input_quality_bands_Q15[ 0 ];
- pitch_freq_log_Q7 = SKP_SMLAWB( pitch_freq_log_Q7, SKP_SMULWB( SKP_LSHIFT( -quality_Q15, 2 ), quality_Q15 ),
- pitch_freq_log_Q7 - ( silk_lin2log( SILK_FIX_CONST( VARIABLE_HP_MIN_CUTOFF_HZ, 16 ) ) - ( 16 << 7 ) ) );
+ /* adjustment based on quality */
+ quality_Q15 = psEncC1->input_quality_bands_Q15[ 0 ];
+ pitch_freq_log_Q7 = SKP_SMLAWB( pitch_freq_log_Q7, SKP_SMULWB( SKP_LSHIFT( -quality_Q15, 2 ), quality_Q15 ),
+ pitch_freq_log_Q7 - ( silk_lin2log( SILK_FIX_CONST( VARIABLE_HP_MIN_CUTOFF_HZ, 16 ) ) - ( 16 << 7 ) ) );
- /* delta_freq = pitch_freq_log - psEnc->variable_HP_smth1; */
- delta_freq_Q7 = pitch_freq_log_Q7 - SKP_RSHIFT( psEncC1->variable_HP_smth1_Q15, 8 );
- if( delta_freq_Q7 < 0 ) {- /* less smoothing for decreasing pitch frequency, to track something close to the minimum */
- delta_freq_Q7 = SKP_MUL( delta_freq_Q7, 3 );
- }
+ /* delta_freq = pitch_freq_log - psEnc->variable_HP_smth1; */
+ delta_freq_Q7 = pitch_freq_log_Q7 - SKP_RSHIFT( psEncC1->variable_HP_smth1_Q15, 8 );
+ if( delta_freq_Q7 < 0 ) {+ /* less smoothing for decreasing pitch frequency, to track something close to the minimum */
+ delta_freq_Q7 = SKP_MUL( delta_freq_Q7, 3 );
+ }
- /* limit delta, to reduce impact of outliers in pitch estimation */
- delta_freq_Q7 = SKP_LIMIT_32( delta_freq_Q7, -SILK_FIX_CONST( VARIABLE_HP_MAX_DELTA_FREQ, 7 ), SILK_FIX_CONST( VARIABLE_HP_MAX_DELTA_FREQ, 7 ) );
+ /* limit delta, to reduce impact of outliers in pitch estimation */
+ delta_freq_Q7 = SKP_LIMIT_32( delta_freq_Q7, -SILK_FIX_CONST( VARIABLE_HP_MAX_DELTA_FREQ, 7 ), SILK_FIX_CONST( VARIABLE_HP_MAX_DELTA_FREQ, 7 ) );
- /* update smoother */
- psEncC1->variable_HP_smth1_Q15 = SKP_SMLAWB( psEncC1->variable_HP_smth1_Q15,
- SKP_SMULBB( psEncC1->speech_activity_Q8, delta_freq_Q7 ), SILK_FIX_CONST( VARIABLE_HP_SMTH_COEF1, 16 ) );
+ /* update smoother */
+ psEncC1->variable_HP_smth1_Q15 = SKP_SMLAWB( psEncC1->variable_HP_smth1_Q15,
+ SKP_SMULBB( psEncC1->speech_activity_Q8, delta_freq_Q7 ), SILK_FIX_CONST( VARIABLE_HP_SMTH_COEF1, 16 ) );
- /* limit frequency range */
- psEncC1->variable_HP_smth1_Q15 = SKP_LIMIT_32( psEncC1->variable_HP_smth1_Q15,
- SKP_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 ),
- SKP_LSHIFT( silk_lin2log( VARIABLE_HP_MAX_CUTOFF_HZ ), 8 ) );
- }
- } else {- /* Externally-controlled cutoff frequency */
- cutoff_Hz = SKP_LIMIT( psEncC1->HP_cutoff_Hz, 10, 500 );
- psEncC1->variable_HP_smth1_Q15 = SKP_LSHIFT( silk_lin2log( cutoff_Hz ), 8 );
- }
-
- /* second smoother */
- psEncC1->variable_HP_smth2_Q15 = SKP_SMLAWB( psEncC1->variable_HP_smth2_Q15,
- psEncC1->variable_HP_smth1_Q15 - psEncC1->variable_HP_smth2_Q15, SILK_FIX_CONST( VARIABLE_HP_SMTH_COEF2, 16 ) );
-
- /* convert from log scale to Hertz */
- cutoff_Hz = silk_log2lin( SKP_RSHIFT( psEncC1->variable_HP_smth2_Q15, 8 ) );
-
- /********************************/
- /* Compute Filter Coefficients */
- /********************************/
- /* compute cut-off frequency, in radians */
- /* Fc_num = 1.5 * 3.14159 * cutoff_Hz */
- /* Fc_denom = 1e3f * psEncC1->fs_kHz */
- SKP_assert( cutoff_Hz <= SKP_int32_MAX / SILK_FIX_CONST( 1.5 * 3.14159 / 1000, 19 ) );
- Fc_Q19 = SKP_DIV32_16( SKP_SMULBB( SILK_FIX_CONST( 1.5 * 3.14159 / 1000, 19 ), cutoff_Hz ), psEncC1->fs_kHz );
- SKP_assert( Fc_Q19 > 0 && Fc_Q19 < 32768 );
-
- r_Q28 = SILK_FIX_CONST( 1.0, 28 ) - SKP_MUL( SILK_FIX_CONST( 0.92, 9 ), Fc_Q19 );
-
- /* b = r * [ 1; -2; 1 ]; */
- /* a = [ 1; -2 * r * ( 1 - 0.5 * Fc^2 ); r^2 ]; */
- B_Q28[ 0 ] = r_Q28;
- B_Q28[ 1 ] = SKP_LSHIFT( -r_Q28, 1 );
- B_Q28[ 2 ] = r_Q28;
-
- /* -r * ( 2 - Fc * Fc ); */
- r_Q22 = SKP_RSHIFT( r_Q28, 6 );
- A_Q28[ 0 ] = SKP_SMULWW( r_Q22, SKP_SMULWW( Fc_Q19, Fc_Q19 ) - SILK_FIX_CONST( 2.0, 22 ) );
- A_Q28[ 1 ] = SKP_SMULWW( r_Q22, r_Q22 );
-
- /********************************/
- /* High-Pass Filter */
- /********************************/
- silk_biquad_alt( psEncC1->inputBuf, B_Q28, A_Q28, psEncC1->In_HP_State, psEncC1->inputBuf, psEncC1->frame_length );
- if( nChannels == 2 ) {- silk_biquad_alt( state_Fxx[ 1 ].sCmn.inputBuf, B_Q28, A_Q28, state_Fxx[ 1 ].sCmn.In_HP_State,
- state_Fxx[ 1 ].sCmn.inputBuf, state_Fxx[ 1 ].sCmn.frame_length );
- }
+ /* limit frequency range */
+ psEncC1->variable_HP_smth1_Q15 = SKP_LIMIT_32( psEncC1->variable_HP_smth1_Q15,
+ SKP_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 ),
+ SKP_LSHIFT( silk_lin2log( VARIABLE_HP_MAX_CUTOFF_HZ ), 8 ) );
+ }
}
--- a/silk/silk_LP_variable_cutoff.c
+++ b/silk/silk_LP_variable_cutoff.c
@@ -131,6 +131,6 @@
/* ARMA low-pass filtering */
SKP_assert( TRANSITION_NB == 3 && TRANSITION_NA == 2 );
- silk_biquad_alt( frame, B_Q28, A_Q28, psLP->In_LP_State, frame, frame_length );
+ silk_biquad_alt( frame, B_Q28, A_Q28, psLP->In_LP_State, frame, frame_length, 1);
}
}
--- a/silk/silk_SigProc_FIX.h
+++ b/silk/silk_SigProc_FIX.h
@@ -126,7 +126,8 @@
const opus_int32 *A_Q28, /* I: AR coefficients [2] */
opus_int32 *S, /* I/O: State vector [2] */
opus_int16 *out, /* O: output signal */
- const opus_int32 len /* I: signal length (must be even) */
+ const opus_int32 len, /* I: signal length (must be even) */
+ int stride
);
/* Variable order MA prediction error filter. */
--- a/silk/silk_biquad_alt.c
+++ b/silk/silk_biquad_alt.c
@@ -46,7 +46,8 @@
const opus_int32 *A_Q28, /* I: AR coefficients [2] */
opus_int32 *S, /* I/O: State vector [2] */
opus_int16 *out, /* O: Output signal */
- const opus_int32 len /* I: Signal length (must be even) */
+ const opus_int32 len, /* I: Signal length (must be even) */
+ int stride
)
{/* DIRECT FORM II TRANSPOSED (uses 2 element state vector) */
@@ -61,7 +62,7 @@
for( k = 0; k < len; k++ ) {/* S[ 0 ], S[ 1 ]: Q12 */
- inval = in[ k ];
+ inval = in[ k*stride ];
out32_Q14 = SKP_LSHIFT( SKP_SMLAWB( S[ 0 ], B_Q28[ 0 ], inval ), 2 );
S[ 0 ] = S[1] + SKP_RSHIFT_ROUND( SKP_SMULWB( out32_Q14, A0_L_Q28 ), 14 );
@@ -73,6 +74,6 @@
S[ 1 ] = SKP_SMLAWB( S[ 1 ], B_Q28[ 2 ], inval );
/* Scale back to Q0 and saturate */
- out[ k ] = (opus_int16)SKP_SAT16( SKP_RSHIFT( out32_Q14 + (1<<14) - 1, 14 ) );
+ out[ k*stride ] = (opus_int16)SKP_SAT16( SKP_RSHIFT( out32_Q14 + (1<<14) - 1, 14 ) );
}
}
--- a/silk/silk_control.h
+++ b/silk/silk_control.h
@@ -83,9 +83,6 @@
/* I: Flag to use constant bitrate */
opus_int useCBR;
- /* I: Cutoff frequency of input HP filter (of zero: adaptive) */
- opus_int HP_cutoff_Hz;
-
/* O: Internal sampling rate used, in Hertz; 8000/12000/16000 */
opus_int32 internalSampleRate;
--- a/silk/silk_enc_API.c
+++ b/silk/silk_enc_API.c
@@ -112,7 +112,6 @@
encStatus->useInBandFEC = state_Fxx[ 0 ].sCmn.useInBandFEC;
encStatus->useDTX = state_Fxx[ 0 ].sCmn.useDTX;
encStatus->useCBR = state_Fxx[ 0 ].sCmn.useCBR;
- encStatus->HP_cutoff_Hz = state_Fxx[ 0 ].sCmn.HP_cutoff_Hz;
encStatus->internalSampleRate = SKP_SMULBB( state_Fxx[ 0 ].sCmn.fs_kHz, 1000 );
encStatus->allowBandwidthSwitch = state_Fxx[ 0 ].sCmn.allow_bandwidth_switch;
encStatus->inWBmodeWithoutVariableLP = state_Fxx[ 0 ].sCmn.fs_kHz == 16 && state_Fxx[ 0 ].sCmn.sLP.mode == 0;
@@ -302,11 +301,7 @@
}
}
- /* High-pass filter, deactivated if less than zero */
- if(encControl->HP_cutoff_Hz>=0) {- psEnc->state_Fxx[ 0 ].sCmn.HP_cutoff_Hz = encControl->HP_cutoff_Hz;
- silk_HP_variable_cutoff( psEnc->state_Fxx, psEnc->nChannelsInternal );
- }
+ silk_HP_variable_cutoff( psEnc->state_Fxx, psEnc->nChannelsInternal );
/* Total target bits for packet */
nBits = SKP_DIV32_16( SKP_MUL( encControl->bitRate, encControl->payloadSize_ms ), 1000 );
--- a/silk/silk_structs.h
+++ b/silk/silk_structs.h
@@ -132,7 +132,6 @@
opus_int32 In_HP_State[ 2 ]; /* High pass filter state */
opus_int32 variable_HP_smth1_Q15; /* State of first smoother */
opus_int32 variable_HP_smth2_Q15; /* State of second smoother */
- opus_int HP_cutoff_Hz; /* Fixed cutoff frequency (if zero: adaptive) */
silk_LP_state sLP; /* Low pass filter state */
silk_VAD_state sVAD; /* Voice activity detector state */
silk_nsq_state sNSQ; /* Noise Shape Quantizer State */
--- a/src/opus_encoder.c
+++ b/src/opus_encoder.c
@@ -41,6 +41,13 @@
#include "opus_private.h"
#include "os_support.h"
+#include "silk_tuning_parameters.h"
+#ifdef FIXED_POINT
+#include "fixed/silk_structs_FIX.h"
+#else
+#include "float/silk_structs_FLP.h"
+#endif
+
#define MAX_ENCODER_BUFFER 480
struct OpusEncoder {@@ -64,6 +71,8 @@
#define OPUS_ENCODER_RESET_START stream_channels
int stream_channels;
int hybrid_stereo_width_Q14;
+ opus_int32 variable_HP_smth2_Q15;
+ opus_val32 hp_mem[4];
int mode;
int prev_mode;
int bandwidth;
@@ -150,7 +159,6 @@
st->silk_mode.useInBandFEC = 0;
st->silk_mode.useDTX = 0;
st->silk_mode.useCBR = 0;
- st->silk_mode.HP_cutoff_Hz = 0;
/* Create CELT encoder */
/* Initialize CELT encoder */
@@ -179,6 +187,7 @@
st->delay_compensation += 2;
st->hybrid_stereo_width_Q14 = 1 << 14;
+ st->variable_HP_smth2_Q15 = SKP_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 );
st->first = 1;
st->mode = MODE_HYBRID;
st->bandwidth = OPUS_BANDWIDTH_FULLBAND;
@@ -221,6 +230,82 @@
toc |= (channels==2)<<2;
return toc;
}
+
+#ifndef FIXED_POINT
+void silk_biquad_float(
+ const opus_val16 *in, /* I: Input signal */
+ const opus_int32 *B_Q28, /* I: MA coefficients [3] */
+ const opus_int32 *A_Q28, /* I: AR coefficients [2] */
+ opus_val32 *S, /* I/O: State vector [2] */
+ opus_val16 *out, /* O: Output signal */
+ const opus_int32 len, /* I: Signal length (must be even) */
+ int stride
+)
+{+ /* DIRECT FORM II TRANSPOSED (uses 2 element state vector) */
+ opus_int k;
+ opus_val32 vout;
+ opus_val32 inval;
+ opus_val32 A[2], B[3];
+
+ A[0] = A_Q28[0] * (1./((opus_int32)1<<28));
+ A[1] = A_Q28[1] * (1./((opus_int32)1<<28));
+ B[0] = B_Q28[0] * (1./((opus_int32)1<<28));
+ B[1] = B_Q28[1] * (1./((opus_int32)1<<28));
+ B[2] = B_Q28[2] * (1./((opus_int32)1<<28));
+
+ /* Negate A_Q28 values and split in two parts */
+
+ for( k = 0; k < len; k++ ) {+ /* S[ 0 ], S[ 1 ]: Q12 */
+ inval = in[ k*stride ];
+ vout = S[ 0 ] + B[0]*inval;
+
+ S[ 0 ] = S[1] - vout*A[0] + B[1]*inval;
+
+ S[ 1 ] = - vout*A[1] + B[2]*inval;
+
+ /* Scale back to Q0 and saturate */
+ out[ k*stride ] = vout;
+ }
+}
+#endif
+
+static void hp_cutoff(const opus_val16 *in, opus_int32 cutoff_Hz, opus_val16 *out, opus_val32 *hp_mem, int len, int channels, opus_int32 Fs)
+{+ opus_int32 B_Q28[ 3 ], A_Q28[ 2 ];
+ opus_int32 Fc_Q19, r_Q28, r_Q22;
+
+ SKP_assert( cutoff_Hz <= SKP_int32_MAX / SILK_FIX_CONST( 1.5 * 3.14159 / 1000, 19 ) );
+ Fc_Q19 = SKP_DIV32_16( SKP_SMULBB( SILK_FIX_CONST( 1.5 * 3.14159 / 1000, 19 ), cutoff_Hz ), Fs/1000 );
+ SKP_assert( Fc_Q19 > 0 && Fc_Q19 < 32768 );
+
+ r_Q28 = SILK_FIX_CONST( 1.0, 28 ) - SKP_MUL( SILK_FIX_CONST( 0.92, 9 ), Fc_Q19 );
+
+ /* b = r * [ 1; -2; 1 ]; */
+ /* a = [ 1; -2 * r * ( 1 - 0.5 * Fc^2 ); r^2 ]; */
+ B_Q28[ 0 ] = r_Q28;
+ B_Q28[ 1 ] = SKP_LSHIFT( -r_Q28, 1 );
+ B_Q28[ 2 ] = r_Q28;
+
+ /* -r * ( 2 - Fc * Fc ); */
+ r_Q22 = SKP_RSHIFT( r_Q28, 6 );
+ A_Q28[ 0 ] = SKP_SMULWW( r_Q22, SKP_SMULWW( Fc_Q19, Fc_Q19 ) - SILK_FIX_CONST( 2.0, 22 ) );
+ A_Q28[ 1 ] = SKP_SMULWW( r_Q22, r_Q22 );
+
+#ifdef FIXED_POINT
+ silk_biquad_alt( in, B_Q28, A_Q28, hp_mem, out, len, channels );
+ if( channels == 2 ) {+ silk_biquad_alt( in+1, B_Q28, A_Q28, hp_mem+2, out+1, len, channels );
+ }
+#else
+ silk_biquad_float( in, B_Q28, A_Q28, hp_mem, out, len, channels );
+ if( channels == 2 ) {+ silk_biquad_float( in+1, B_Q28, A_Q28, hp_mem+2, out+1, len, channels );
+ }
+#endif
+}
+
OpusEncoder *opus_encoder_create(int Fs, int channels, int mode, int *error)
{int ret;
@@ -267,6 +352,7 @@
int to_celt = 0;
opus_int32 mono_rate;
opus_uint32 redundant_rng = 0;
+ int cutoff_Hz, hp_freq_smth1;
ALLOC_STACK;
st->rangeFinal = 0;
@@ -338,7 +424,6 @@
opus_int32 threshold;
threshold = 20000;
/* OPUS_APPLICATION_VOIP default to auto high-pass */
- st->silk_mode.HP_cutoff_Hz=0;
/* Hysteresis */
if (st->prev_mode == MODE_CELT_ONLY)
threshold -= 4000;
@@ -355,7 +440,6 @@
/* SILK/CELT threshold is higher for voice than for music */
threshold = 36000;
/* OPUS_APPLICATION_AUDIO disables the high-pass */
- st->silk_mode.HP_cutoff_Hz=-1;
if (st->signal_type == OPUS_SIGNAL_MUSIC)
threshold -= 20000;
else if (st->signal_type == OPUS_SIGNAL_VOICE)
@@ -467,6 +551,29 @@
ec_enc_init(&enc, data, max_data_bytes-1);
+ ALLOC(pcm_buf, (st->delay_compensation+frame_size)*st->channels, opus_val16);
+ for (i=0;i<st->delay_compensation*st->channels;i++)
+ pcm_buf[i] = st->delay_buffer[(st->encoder_buffer-st->delay_compensation)*st->channels+i];
+
+ if (st->mode == MODE_CELT_ONLY)
+ hp_freq_smth1 = SKP_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 );
+ else
+ hp_freq_smth1 = ((silk_encoder*)silk_enc)->state_Fxx[0].sCmn.variable_HP_smth1_Q15;
+
+ st->variable_HP_smth2_Q15 = SKP_SMLAWB( st->variable_HP_smth2_Q15,
+ hp_freq_smth1 - st->variable_HP_smth2_Q15, SILK_FIX_CONST( VARIABLE_HP_SMTH_COEF2, 16 ) );
+
+ /* convert from log scale to Hertz */
+ cutoff_Hz = silk_log2lin( SKP_RSHIFT( st->variable_HP_smth2_Q15, 8 ) );
+
+ if (st->application == OPUS_APPLICATION_VOIP)
+ {+ hp_cutoff(pcm, cutoff_Hz, &pcm_buf[st->delay_compensation*st->channels], st->hp_mem, frame_size, st->channels, st->Fs);
+ } else {+ for (i=0;i<frame_size*st->channels;i++)
+ pcm_buf[st->delay_compensation*st->channels + i] = pcm[i];
+ }
+
/* SILK processing */
if (st->mode != MODE_CELT_ONLY)
{@@ -537,10 +644,10 @@
}
#ifdef FIXED_POINT
- pcm_silk = pcm;
+ pcm_silk = pcm_buf+st->delay_compensation*st->channels;
#else
for (i=0;i<frame_size*st->channels;i++)
- pcm_silk[i] = FLOAT2INT16(pcm[i]);
+ pcm_silk[i] = FLOAT2INT16(pcm_buf[st->delay_compensation*st->channels + i]);
#endif
ret = silk_Encode( silk_enc, &st->silk_mode, pcm_silk, frame_size, &enc, &nBytes, 0 );
if( ret ) {@@ -634,12 +741,10 @@
nb_compr_bytes = 0;
}
- ALLOC(pcm_buf, IMAX(frame_size, st->Fs/200)*st->channels, opus_val16);
- for (i=0;i<IMIN(frame_size, st->delay_compensation)*st->channels;i++)
- pcm_buf[i] = st->delay_buffer[(st->encoder_buffer-st->delay_compensation)*st->channels+i];
- for (;i<frame_size*st->channels;i++)
- pcm_buf[i] = pcm[i-st->delay_compensation*st->channels];
+ for (i=0;i<st->encoder_buffer*st->channels;i++)
+ st->delay_buffer[i] = pcm_buf[(frame_size+st->delay_compensation-st->encoder_buffer)*st->channels+i];
+
if( st->mode == MODE_HYBRID && st->stream_channels == 2 ) {/* Apply stereo width reduction (at low bitrates) */
if( st->hybrid_stereo_width_Q14 < (1 << 14) || st->silk_mode.stereoWidth_Q14 < (1 << 14) ) {@@ -735,17 +840,6 @@
}
- if (frame_size>st->encoder_buffer)
- {- for (i=0;i<st->encoder_buffer*st->channels;i++)
- st->delay_buffer[i] = pcm[(frame_size-st->encoder_buffer)*st->channels+i];
- } else {- int tmp = st->encoder_buffer-frame_size;
- for (i=0;i<tmp*st->channels;i++)
- st->delay_buffer[i] = st->delay_buffer[i+frame_size*st->channels];
- for (i=0;i<frame_size*st->channels;i++)
- st->delay_buffer[tmp*st->channels+i] = pcm[i];
- }
/* Signalling the mode in the first byte */
data--;
@@ -1010,6 +1104,7 @@
st->first = 1;
st->mode = MODE_HYBRID;
st->bandwidth = OPUS_BANDWIDTH_FULLBAND;
+ st->variable_HP_smth2_Q15 = SKP_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 );
}
break;
default: