From 2f0ca7618d8af91439921ddd99f09c9869a1cf9b Mon Sep 17 00:00:00 2001 From: Jean-Marc Valin Date: Thu, 1 Sep 2011 16:56:40 -0400 Subject: [PATCH 1/3] Simplifying buffering to make an Opus-level highpass possible --- src/opus_encoder.c | 29 +++++++++++------------------ 1 file changed, 11 insertions(+), 18 deletions(-) diff --git a/src/opus_encoder.c b/src/opus_encoder.c index 4f3d39f4..ec866725 100644 --- a/src/opus_encoder.c +++ b/src/opus_encoder.c @@ -467,6 +467,12 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size, ec_enc_init(&enc, data, max_data_bytes-1); + ALLOC(pcm_buf, (st->delay_compensation+frame_size)*st->channels, opus_val16); + for (i=0;idelay_compensation*st->channels;i++) + pcm_buf[i] = st->delay_buffer[(st->encoder_buffer-st->delay_compensation)*st->channels+i]; + for (i=0;ichannels;i++) + pcm_buf[st->delay_compensation*st->channels + i] = pcm[i]; + /* SILK processing */ if (st->mode != MODE_CELT_ONLY) { @@ -537,10 +543,10 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size, } #ifdef FIXED_POINT - pcm_silk = pcm; + pcm_silk = pcm_buf+st->delay_compensation*st->channels; #else for (i=0;ichannels;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 ) { @@ -634,11 +640,9 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size, nb_compr_bytes = 0; } - ALLOC(pcm_buf, IMAX(frame_size, st->Fs/200)*st->channels, opus_val16); - for (i=0;idelay_compensation)*st->channels;i++) - pcm_buf[i] = st->delay_buffer[(st->encoder_buffer-st->delay_compensation)*st->channels+i]; - for (;ichannels;i++) - pcm_buf[i] = pcm[i-st->delay_compensation*st->channels]; + for (i=0;iencoder_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) */ @@ -735,17 +739,6 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size, } - if (frame_size>st->encoder_buffer) - { - for (i=0;iencoder_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;ichannels;i++) - st->delay_buffer[i] = st->delay_buffer[i+frame_size*st->channels]; - for (i=0;ichannels;i++) - st->delay_buffer[tmp*st->channels+i] = pcm[i]; - } /* Signalling the mode in the first byte */ data--; From 957f7f169b1d834875cd0b154c5459a108c5e75e Mon Sep 17 00:00:00 2001 From: Jean-Marc Valin Date: Thu, 1 Sep 2011 18:02:43 -0400 Subject: [PATCH 2/3] First attempt at global high-pass filter Doesn't work for fixed-point for some unknown reason --- silk/silk_HP_variable_cutoff.c | 6 +- silk/silk_LP_variable_cutoff.c | 2 +- silk/silk_SigProc_FIX.h | 3 +- silk/silk_biquad_alt.c | 7 ++- src/opus_encoder.c | 109 ++++++++++++++++++++++++++++++++- 5 files changed, 118 insertions(+), 9 deletions(-) diff --git a/silk/silk_HP_variable_cutoff.c b/silk/silk_HP_variable_cutoff.c index fb6c3665..9bcd27ef 100644 --- a/silk/silk_HP_variable_cutoff.c +++ b/silk/silk_HP_variable_cutoff.c @@ -84,6 +84,7 @@ void silk_HP_variable_cutoff( psEncC1->variable_HP_smth1_Q15 = SKP_LSHIFT( silk_lin2log( cutoff_Hz ), 8 ); } +#if 0 /* 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 ) ); @@ -117,9 +118,10 @@ void silk_HP_variable_cutoff( /********************************/ /* High-Pass Filter */ /********************************/ - silk_biquad_alt( psEncC1->inputBuf, B_Q28, A_Q28, psEncC1->In_HP_State, psEncC1->inputBuf, psEncC1->frame_length ); + silk_biquad_alt( psEncC1->inputBuf, B_Q28, A_Q28, psEncC1->In_HP_State, psEncC1->inputBuf, psEncC1->frame_length, 1 ); 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 ); + state_Fxx[ 1 ].sCmn.inputBuf, state_Fxx[ 1 ].sCmn.frame_length, 1 ); } +#endif } diff --git a/silk/silk_LP_variable_cutoff.c b/silk/silk_LP_variable_cutoff.c index d679f92e..0c758a03 100644 --- a/silk/silk_LP_variable_cutoff.c +++ b/silk/silk_LP_variable_cutoff.c @@ -131,6 +131,6 @@ void silk_LP_variable_cutoff( /* 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); } } diff --git a/silk/silk_SigProc_FIX.h b/silk/silk_SigProc_FIX.h index b30ead80..222bc312 100644 --- a/silk/silk_SigProc_FIX.h +++ b/silk/silk_SigProc_FIX.h @@ -126,7 +126,8 @@ void silk_biquad_alt( 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. */ diff --git a/silk/silk_biquad_alt.c b/silk/silk_biquad_alt.c index 9d394c00..966ba28f 100644 --- a/silk/silk_biquad_alt.c +++ b/silk/silk_biquad_alt.c @@ -46,7 +46,8 @@ void silk_biquad_alt( 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 @@ void silk_biquad_alt( 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 @@ void silk_biquad_alt( 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 ) ); } } diff --git a/src/opus_encoder.c b/src/opus_encoder.c index ec866725..f8c83d1e 100644 --- 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 @@ struct OpusEncoder { #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; @@ -179,6 +188,7 @@ int opus_encoder_init(OpusEncoder* st, int Fs, int channels, int application) 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 +231,82 @@ static unsigned char gen_toc(int mode, int framerate, int bandwidth, int silk_ba 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 +353,7 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size, int to_celt = 0; opus_int32 mono_rate; opus_uint32 redundant_rng = 0; + int cutoff_Hz, hp_freq_smth1; ALLOC_STACK; st->rangeFinal = 0; @@ -470,8 +557,25 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size, ALLOC(pcm_buf, (st->delay_compensation+frame_size)*st->channels, opus_val16); for (i=0;idelay_compensation*st->channels;i++) pcm_buf[i] = st->delay_buffer[(st->encoder_buffer-st->delay_compensation)*st->channels+i]; - for (i=0;ichannels;i++) - pcm_buf[st->delay_compensation*st->channels + i] = pcm[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;ichannels;i++) + pcm_buf[st->delay_compensation*st->channels + i] = pcm[i]; + } /* SILK processing */ if (st->mode != MODE_CELT_ONLY) @@ -1003,6 +1107,7 @@ int opus_encoder_ctl(OpusEncoder *st, int request, ...) 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: From ec2802210cd0d62514f4cb9bc4db2f7df1259ad4 Mon Sep 17 00:00:00 2001 From: Jean-Marc Valin Date: Thu, 1 Sep 2011 21:47:38 -0400 Subject: [PATCH 3/3] Removes code that became useless with the Opus-level highpass --- silk/silk_HP_variable_cutoff.c | 105 +++++++++------------------------ silk/silk_control.h | 3 - silk/silk_enc_API.c | 7 +-- silk/silk_structs.h | 1 - src/opus_encoder.c | 3 - 5 files changed, 29 insertions(+), 90 deletions(-) diff --git a/silk/silk_HP_variable_cutoff.c b/silk/silk_HP_variable_cutoff.c index 9bcd27ef..f76767cf 100644 --- a/silk/silk_HP_variable_cutoff.c +++ b/silk/silk_HP_variable_cutoff.c @@ -41,87 +41,38 @@ void silk_HP_variable_cutoff( 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 ); - } - -#if 0 - /* 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, 1 ); - 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, 1 ); - } -#endif + /* 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 ) ); + } } diff --git a/silk/silk_control.h b/silk/silk_control.h index 58f8a7ae..0ebf5a33 100644 --- a/silk/silk_control.h +++ b/silk/silk_control.h @@ -83,9 +83,6 @@ typedef struct { /* 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; diff --git a/silk/silk_enc_API.c b/silk/silk_enc_API.c index 8e0315b8..bc65caa8 100644 --- a/silk/silk_enc_API.c +++ b/silk/silk_enc_API.c @@ -112,7 +112,6 @@ opus_int silk_QueryEncoder( 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 @@ opus_int silk_Encode( } } - /* 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 ); diff --git a/silk/silk_structs.h b/silk/silk_structs.h index 341a2a96..3044a02f 100644 --- a/silk/silk_structs.h +++ b/silk/silk_structs.h @@ -132,7 +132,6 @@ typedef struct { 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 */ diff --git a/src/opus_encoder.c b/src/opus_encoder.c index f8c83d1e..02c3ef72 100644 --- a/src/opus_encoder.c +++ b/src/opus_encoder.c @@ -159,7 +159,6 @@ int opus_encoder_init(OpusEncoder* st, int Fs, int channels, int application) 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 */ @@ -425,7 +424,6 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size, 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; @@ -442,7 +440,6 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size, /* 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)