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Merge branch 'exp-highpass'

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This commit is contained in:
Jean-Marc Valin 2011-09-01 21:59:50 -04:00
commit bafbd08db1
8 changed files with 152 additions and 111 deletions
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103
silk/silk_HP_variable_cutoff.c
View file

@ -41,85 +41,38 @@ void silk_HP_variable_cutoff(
const opus_int nChannels /* I Number of channels */ const opus_int nChannels /* I Number of channels */
) )
{ {
opus_int quality_Q15, cutoff_Hz; opus_int quality_Q15;
opus_int32 B_Q28[ 3 ], A_Q28[ 2 ]; opus_int32 pitch_freq_Hz_Q16, pitch_freq_log_Q7, delta_freq_Q7;
opus_int32 Fc_Q19, r_Q28, r_Q22; silk_encoder_state *psEncC1 = &state_Fxx[ 0 ].sCmn;
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 */
/* Adaptive cutoff frequency: estimate low end of pitch frequency range */ if( psEncC1->prevSignalType == TYPE_VOICED ) {
if( psEncC1->prevSignalType == TYPE_VOICED ) { /* difference, in log domain */
/* difference, in log domain */ pitch_freq_Hz_Q16 = SKP_DIV32_16( SKP_LSHIFT( SKP_MUL( psEncC1->fs_kHz, 1000 ), 16 ), psEncC1->prevLag );
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 );
pitch_freq_log_Q7 = silk_lin2log( pitch_freq_Hz_Q16 ) - ( 16 << 7 );
/* adjustment based on quality */ /* adjustment based on quality */
quality_Q15 = psEncC1->input_quality_bands_Q15[ 0 ]; 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 = 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 ) ) ); 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 = pitch_freq_log - psEnc->variable_HP_smth1; */
delta_freq_Q7 = pitch_freq_log_Q7 - SKP_RSHIFT( psEncC1->variable_HP_smth1_Q15, 8 ); delta_freq_Q7 = pitch_freq_log_Q7 - SKP_RSHIFT( psEncC1->variable_HP_smth1_Q15, 8 );
if( delta_freq_Q7 < 0 ) { if( delta_freq_Q7 < 0 ) {
/* less smoothing for decreasing pitch frequency, to track something close to the minimum */ /* less smoothing for decreasing pitch frequency, to track something close to the minimum */
delta_freq_Q7 = SKP_MUL( delta_freq_Q7, 3 ); delta_freq_Q7 = SKP_MUL( delta_freq_Q7, 3 );
} }
/* limit delta, to reduce impact of outliers in pitch estimation */ /* 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 ) ); 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 */ /* update smoother */
psEncC1->variable_HP_smth1_Q15 = SKP_SMLAWB( psEncC1->variable_HP_smth1_Q15, 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 ) ); SKP_SMULBB( psEncC1->speech_activity_Q8, delta_freq_Q7 ), SILK_FIX_CONST( VARIABLE_HP_SMTH_COEF1, 16 ) );
/* limit frequency range */ /* limit frequency range */
psEncC1->variable_HP_smth1_Q15 = SKP_LIMIT_32( psEncC1->variable_HP_smth1_Q15, 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_MIN_CUTOFF_HZ ), 8 ),
SKP_LSHIFT( silk_lin2log( VARIABLE_HP_MAX_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 );
}
} }

2
silk/silk_LP_variable_cutoff.c
View file

@ -131,6 +131,6 @@ void silk_LP_variable_cutoff(
/* ARMA low-pass filtering */ /* ARMA low-pass filtering */
SKP_assert( TRANSITION_NB == 3 && TRANSITION_NA == 2 ); 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);
} }
} }

3
silk/silk_SigProc_FIX.h
View file

@ -126,7 +126,8 @@ void silk_biquad_alt(
const opus_int32 *A_Q28, /* I: AR coefficients [2] */ const opus_int32 *A_Q28, /* I: AR coefficients [2] */
opus_int32 *S, /* I/O: State vector [2] */ opus_int32 *S, /* I/O: State vector [2] */
opus_int16 *out, /* O: output signal */ 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. */ /* Variable order MA prediction error filter. */

7
silk/silk_biquad_alt.c
View file

@ -46,7 +46,8 @@ void silk_biquad_alt(
const opus_int32 *A_Q28, /* I: AR coefficients [2] */ const opus_int32 *A_Q28, /* I: AR coefficients [2] */
opus_int32 *S, /* I/O: State vector [2] */ opus_int32 *S, /* I/O: State vector [2] */
opus_int16 *out, /* O: Output signal */ 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) */ /* DIRECT FORM II TRANSPOSED (uses 2 element state vector) */
@ -61,7 +62,7 @@ void silk_biquad_alt(
for( k = 0; k < len; k++ ) { for( k = 0; k < len; k++ ) {
/* S[ 0 ], S[ 1 ]: Q12 */ /* 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 ); 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 ); 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 ); S[ 1 ] = SKP_SMLAWB( S[ 1 ], B_Q28[ 2 ], inval );
/* Scale back to Q0 and saturate */ /* 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 ) );
} }
} }

3
silk/silk_control.h
View file

@ -83,9 +83,6 @@ typedef struct {
/* I: Flag to use constant bitrate */ /* I: Flag to use constant bitrate */
opus_int useCBR; 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 */ /* O: Internal sampling rate used, in Hertz; 8000/12000/16000 */
opus_int32 internalSampleRate; opus_int32 internalSampleRate;

7
silk/silk_enc_API.c
View file

@ -112,7 +112,6 @@ opus_int silk_QueryEncoder(
encStatus->useInBandFEC = state_Fxx[ 0 ].sCmn.useInBandFEC; encStatus->useInBandFEC = state_Fxx[ 0 ].sCmn.useInBandFEC;
encStatus->useDTX = state_Fxx[ 0 ].sCmn.useDTX; encStatus->useDTX = state_Fxx[ 0 ].sCmn.useDTX;
encStatus->useCBR = state_Fxx[ 0 ].sCmn.useCBR; 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->internalSampleRate = SKP_SMULBB( state_Fxx[ 0 ].sCmn.fs_kHz, 1000 );
encStatus->allowBandwidthSwitch = state_Fxx[ 0 ].sCmn.allow_bandwidth_switch; 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; 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 */ silk_HP_variable_cutoff( psEnc->state_Fxx, psEnc->nChannelsInternal );
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 );
}
/* Total target bits for packet */ /* Total target bits for packet */
nBits = SKP_DIV32_16( SKP_MUL( encControl->bitRate, encControl->payloadSize_ms ), 1000 ); nBits = SKP_DIV32_16( SKP_MUL( encControl->bitRate, encControl->payloadSize_ms ), 1000 );

1
silk/silk_structs.h
View file

@ -132,7 +132,6 @@ typedef struct {
opus_int32 In_HP_State[ 2 ]; /* High pass filter state */ opus_int32 In_HP_State[ 2 ]; /* High pass filter state */
opus_int32 variable_HP_smth1_Q15; /* State of first smoother */ opus_int32 variable_HP_smth1_Q15; /* State of first smoother */
opus_int32 variable_HP_smth2_Q15; /* State of second 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_LP_state sLP; /* Low pass filter state */
silk_VAD_state sVAD; /* Voice activity detector state */ silk_VAD_state sVAD; /* Voice activity detector state */
silk_nsq_state sNSQ; /* Noise Shape Quantizer State */ silk_nsq_state sNSQ; /* Noise Shape Quantizer State */

137
src/opus_encoder.c
View file

@ -41,6 +41,13 @@
#include "opus_private.h" #include "opus_private.h"
#include "os_support.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 #define MAX_ENCODER_BUFFER 480
struct OpusEncoder { struct OpusEncoder {
@ -64,6 +71,8 @@ struct OpusEncoder {
#define OPUS_ENCODER_RESET_START stream_channels #define OPUS_ENCODER_RESET_START stream_channels
int stream_channels; int stream_channels;
int hybrid_stereo_width_Q14; int hybrid_stereo_width_Q14;
opus_int32 variable_HP_smth2_Q15;
opus_val32 hp_mem[4];
int mode; int mode;
int prev_mode; int prev_mode;
int bandwidth; int bandwidth;
@ -150,7 +159,6 @@ int opus_encoder_init(OpusEncoder* st, int Fs, int channels, int application)
st->silk_mode.useInBandFEC = 0; st->silk_mode.useInBandFEC = 0;
st->silk_mode.useDTX = 0; st->silk_mode.useDTX = 0;
st->silk_mode.useCBR = 0; st->silk_mode.useCBR = 0;
st->silk_mode.HP_cutoff_Hz = 0;
/* Create CELT encoder */ /* Create CELT encoder */
/* Initialize CELT encoder */ /* Initialize CELT encoder */
@ -179,6 +187,7 @@ int opus_encoder_init(OpusEncoder* st, int Fs, int channels, int application)
st->delay_compensation += 2; st->delay_compensation += 2;
st->hybrid_stereo_width_Q14 = 1 << 14; 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->first = 1;
st->mode = MODE_HYBRID; st->mode = MODE_HYBRID;
st->bandwidth = OPUS_BANDWIDTH_FULLBAND; st->bandwidth = OPUS_BANDWIDTH_FULLBAND;
@ -221,6 +230,82 @@ static unsigned char gen_toc(int mode, int framerate, int bandwidth, int silk_ba
toc |= (channels==2)<<2; toc |= (channels==2)<<2;
return toc; 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) OpusEncoder *opus_encoder_create(int Fs, int channels, int mode, int *error)
{ {
int ret; int ret;
@ -267,6 +352,7 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
int to_celt = 0; int to_celt = 0;
opus_int32 mono_rate; opus_int32 mono_rate;
opus_uint32 redundant_rng = 0; opus_uint32 redundant_rng = 0;
int cutoff_Hz, hp_freq_smth1;
ALLOC_STACK; ALLOC_STACK;
st->rangeFinal = 0; st->rangeFinal = 0;
@ -338,7 +424,6 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
opus_int32 threshold; opus_int32 threshold;
threshold = 20000; threshold = 20000;
/* OPUS_APPLICATION_VOIP default to auto high-pass */ /* OPUS_APPLICATION_VOIP default to auto high-pass */
st->silk_mode.HP_cutoff_Hz=0;
/* Hysteresis */ /* Hysteresis */
if (st->prev_mode == MODE_CELT_ONLY) if (st->prev_mode == MODE_CELT_ONLY)
threshold -= 4000; threshold -= 4000;
@ -355,7 +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 */ /* SILK/CELT threshold is higher for voice than for music */
threshold = 36000; threshold = 36000;
/* OPUS_APPLICATION_AUDIO disables the high-pass */ /* OPUS_APPLICATION_AUDIO disables the high-pass */
st->silk_mode.HP_cutoff_Hz=-1;
if (st->signal_type == OPUS_SIGNAL_MUSIC) if (st->signal_type == OPUS_SIGNAL_MUSIC)
threshold -= 20000; threshold -= 20000;
else if (st->signal_type == OPUS_SIGNAL_VOICE) else if (st->signal_type == OPUS_SIGNAL_VOICE)
@ -467,6 +551,29 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
ec_enc_init(&enc, data, max_data_bytes-1); 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 */ /* SILK processing */
if (st->mode != MODE_CELT_ONLY) if (st->mode != MODE_CELT_ONLY)
{ {
@ -537,10 +644,10 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
} }
#ifdef FIXED_POINT #ifdef FIXED_POINT
pcm_silk = pcm; pcm_silk = pcm_buf+st->delay_compensation*st->channels;
#else #else
for (i=0;i<frame_size*st->channels;i++) 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 #endif
ret = silk_Encode( silk_enc, &st->silk_mode, pcm_silk, frame_size, &enc, &nBytes, 0 ); ret = silk_Encode( silk_enc, &st->silk_mode, pcm_silk, frame_size, &enc, &nBytes, 0 );
if( ret ) { if( ret ) {
@ -634,11 +741,9 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
nb_compr_bytes = 0; nb_compr_bytes = 0;
} }
ALLOC(pcm_buf, IMAX(frame_size, st->Fs/200)*st->channels, opus_val16); for (i=0;i<st->encoder_buffer*st->channels;i++)
for (i=0;i<IMIN(frame_size, st->delay_compensation)*st->channels;i++) st->delay_buffer[i] = pcm_buf[(frame_size+st->delay_compensation-st->encoder_buffer)*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];
if( st->mode == MODE_HYBRID && st->stream_channels == 2 ) { if( st->mode == MODE_HYBRID && st->stream_channels == 2 ) {
/* Apply stereo width reduction (at low bitrates) */ /* Apply stereo width reduction (at low bitrates) */
@ -735,17 +840,6 @@ int opus_encode_float(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
} }
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 */ /* Signalling the mode in the first byte */
data--; data--;
@ -1010,6 +1104,7 @@ int opus_encoder_ctl(OpusEncoder *st, int request, ...)
st->first = 1; st->first = 1;
st->mode = MODE_HYBRID; st->mode = MODE_HYBRID;
st->bandwidth = OPUS_BANDWIDTH_FULLBAND; st->bandwidth = OPUS_BANDWIDTH_FULLBAND;
st->variable_HP_smth2_Q15 = SKP_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 );
} }
break; break;
default: default: