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Diffstat (limited to 'media/libopus/src/analysis.c')
| -rw-r--r-- | media/libopus/src/analysis.c | 983 |
1 files changed, 983 insertions, 0 deletions
diff --git a/media/libopus/src/analysis.c b/media/libopus/src/analysis.c new file mode 100644 index 0000000000..058328f0fd --- /dev/null +++ b/media/libopus/src/analysis.c @@ -0,0 +1,983 @@ +/* Copyright (c) 2011 Xiph.Org Foundation + Written by Jean-Marc Valin */ +/* + 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. + + 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 FOUNDATION 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 + +#define ANALYSIS_C + +#ifdef MLP_TRAINING +#include <stdio.h> +#endif + +#include "mathops.h" +#include "kiss_fft.h" +#include "celt.h" +#include "modes.h" +#include "arch.h" +#include "quant_bands.h" +#include "analysis.h" +#include "mlp.h" +#include "stack_alloc.h" +#include "float_cast.h" + +#ifndef M_PI +#define M_PI 3.141592653 +#endif + +#ifndef DISABLE_FLOAT_API + +#define TRANSITION_PENALTY 10 + +static const float dct_table[128] = { + 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, + 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, 0.250000f, + 0.351851f, 0.338330f, 0.311806f, 0.273300f, 0.224292f, 0.166664f, 0.102631f, 0.034654f, + -0.034654f,-0.102631f,-0.166664f,-0.224292f,-0.273300f,-0.311806f,-0.338330f,-0.351851f, + 0.346760f, 0.293969f, 0.196424f, 0.068975f,-0.068975f,-0.196424f,-0.293969f,-0.346760f, + -0.346760f,-0.293969f,-0.196424f,-0.068975f, 0.068975f, 0.196424f, 0.293969f, 0.346760f, + 0.338330f, 0.224292f, 0.034654f,-0.166664f,-0.311806f,-0.351851f,-0.273300f,-0.102631f, + 0.102631f, 0.273300f, 0.351851f, 0.311806f, 0.166664f,-0.034654f,-0.224292f,-0.338330f, + 0.326641f, 0.135299f,-0.135299f,-0.326641f,-0.326641f,-0.135299f, 0.135299f, 0.326641f, + 0.326641f, 0.135299f,-0.135299f,-0.326641f,-0.326641f,-0.135299f, 0.135299f, 0.326641f, + 0.311806f, 0.034654f,-0.273300f,-0.338330f,-0.102631f, 0.224292f, 0.351851f, 0.166664f, + -0.166664f,-0.351851f,-0.224292f, 0.102631f, 0.338330f, 0.273300f,-0.034654f,-0.311806f, + 0.293969f,-0.068975f,-0.346760f,-0.196424f, 0.196424f, 0.346760f, 0.068975f,-0.293969f, + -0.293969f, 0.068975f, 0.346760f, 0.196424f,-0.196424f,-0.346760f,-0.068975f, 0.293969f, + 0.273300f,-0.166664f,-0.338330f, 0.034654f, 0.351851f, 0.102631f,-0.311806f,-0.224292f, + 0.224292f, 0.311806f,-0.102631f,-0.351851f,-0.034654f, 0.338330f, 0.166664f,-0.273300f, +}; + +static const float analysis_window[240] = { + 0.000043f, 0.000171f, 0.000385f, 0.000685f, 0.001071f, 0.001541f, 0.002098f, 0.002739f, + 0.003466f, 0.004278f, 0.005174f, 0.006156f, 0.007222f, 0.008373f, 0.009607f, 0.010926f, + 0.012329f, 0.013815f, 0.015385f, 0.017037f, 0.018772f, 0.020590f, 0.022490f, 0.024472f, + 0.026535f, 0.028679f, 0.030904f, 0.033210f, 0.035595f, 0.038060f, 0.040604f, 0.043227f, + 0.045928f, 0.048707f, 0.051564f, 0.054497f, 0.057506f, 0.060591f, 0.063752f, 0.066987f, + 0.070297f, 0.073680f, 0.077136f, 0.080665f, 0.084265f, 0.087937f, 0.091679f, 0.095492f, + 0.099373f, 0.103323f, 0.107342f, 0.111427f, 0.115579f, 0.119797f, 0.124080f, 0.128428f, + 0.132839f, 0.137313f, 0.141849f, 0.146447f, 0.151105f, 0.155823f, 0.160600f, 0.165435f, + 0.170327f, 0.175276f, 0.180280f, 0.185340f, 0.190453f, 0.195619f, 0.200838f, 0.206107f, + 0.211427f, 0.216797f, 0.222215f, 0.227680f, 0.233193f, 0.238751f, 0.244353f, 0.250000f, + 0.255689f, 0.261421f, 0.267193f, 0.273005f, 0.278856f, 0.284744f, 0.290670f, 0.296632f, + 0.302628f, 0.308658f, 0.314721f, 0.320816f, 0.326941f, 0.333097f, 0.339280f, 0.345492f, + 0.351729f, 0.357992f, 0.364280f, 0.370590f, 0.376923f, 0.383277f, 0.389651f, 0.396044f, + 0.402455f, 0.408882f, 0.415325f, 0.421783f, 0.428254f, 0.434737f, 0.441231f, 0.447736f, + 0.454249f, 0.460770f, 0.467298f, 0.473832f, 0.480370f, 0.486912f, 0.493455f, 0.500000f, + 0.506545f, 0.513088f, 0.519630f, 0.526168f, 0.532702f, 0.539230f, 0.545751f, 0.552264f, + 0.558769f, 0.565263f, 0.571746f, 0.578217f, 0.584675f, 0.591118f, 0.597545f, 0.603956f, + 0.610349f, 0.616723f, 0.623077f, 0.629410f, 0.635720f, 0.642008f, 0.648271f, 0.654508f, + 0.660720f, 0.666903f, 0.673059f, 0.679184f, 0.685279f, 0.691342f, 0.697372f, 0.703368f, + 0.709330f, 0.715256f, 0.721144f, 0.726995f, 0.732807f, 0.738579f, 0.744311f, 0.750000f, + 0.755647f, 0.761249f, 0.766807f, 0.772320f, 0.777785f, 0.783203f, 0.788573f, 0.793893f, + 0.799162f, 0.804381f, 0.809547f, 0.814660f, 0.819720f, 0.824724f, 0.829673f, 0.834565f, + 0.839400f, 0.844177f, 0.848895f, 0.853553f, 0.858151f, 0.862687f, 0.867161f, 0.871572f, + 0.875920f, 0.880203f, 0.884421f, 0.888573f, 0.892658f, 0.896677f, 0.900627f, 0.904508f, + 0.908321f, 0.912063f, 0.915735f, 0.919335f, 0.922864f, 0.926320f, 0.929703f, 0.933013f, + 0.936248f, 0.939409f, 0.942494f, 0.945503f, 0.948436f, 0.951293f, 0.954072f, 0.956773f, + 0.959396f, 0.961940f, 0.964405f, 0.966790f, 0.969096f, 0.971321f, 0.973465f, 0.975528f, + 0.977510f, 0.979410f, 0.981228f, 0.982963f, 0.984615f, 0.986185f, 0.987671f, 0.989074f, + 0.990393f, 0.991627f, 0.992778f, 0.993844f, 0.994826f, 0.995722f, 0.996534f, 0.997261f, + 0.997902f, 0.998459f, 0.998929f, 0.999315f, 0.999615f, 0.999829f, 0.999957f, 1.000000f, +}; + +static const int tbands[NB_TBANDS+1] = { + 4, 8, 12, 16, 20, 24, 28, 32, 40, 48, 56, 64, 80, 96, 112, 136, 160, 192, 240 +}; + +#define NB_TONAL_SKIP_BANDS 9 + +static opus_val32 silk_resampler_down2_hp( + opus_val32 *S, /* I/O State vector [ 2 ] */ + opus_val32 *out, /* O Output signal [ floor(len/2) ] */ + const opus_val32 *in, /* I Input signal [ len ] */ + int inLen /* I Number of input samples */ +) +{ + int k, len2 = inLen/2; + opus_val32 in32, out32, out32_hp, Y, X; + opus_val64 hp_ener = 0; + /* Internal variables and state are in Q10 format */ + for( k = 0; k < len2; k++ ) { + /* Convert to Q10 */ + in32 = in[ 2 * k ]; + + /* All-pass section for even input sample */ + Y = SUB32( in32, S[ 0 ] ); + X = MULT16_32_Q15(QCONST16(0.6074371f, 15), Y); + out32 = ADD32( S[ 0 ], X ); + S[ 0 ] = ADD32( in32, X ); + out32_hp = out32; + /* Convert to Q10 */ + in32 = in[ 2 * k + 1 ]; + + /* All-pass section for odd input sample, and add to output of previous section */ + Y = SUB32( in32, S[ 1 ] ); + X = MULT16_32_Q15(QCONST16(0.15063f, 15), Y); + out32 = ADD32( out32, S[ 1 ] ); + out32 = ADD32( out32, X ); + S[ 1 ] = ADD32( in32, X ); + + Y = SUB32( -in32, S[ 2 ] ); + X = MULT16_32_Q15(QCONST16(0.15063f, 15), Y); + out32_hp = ADD32( out32_hp, S[ 2 ] ); + out32_hp = ADD32( out32_hp, X ); + S[ 2 ] = ADD32( -in32, X ); + + hp_ener += out32_hp*(opus_val64)out32_hp; + /* Add, convert back to int16 and store to output */ + out[ k ] = HALF32(out32); + } +#ifdef FIXED_POINT + /* len2 can be up to 480, so we shift by 8 more to make it fit. */ + hp_ener = hp_ener >> (2*SIG_SHIFT + 8); +#endif + return (opus_val32)hp_ener; +} + +static opus_val32 downmix_and_resample(downmix_func downmix, const void *_x, opus_val32 *y, opus_val32 S[3], int subframe, int offset, int c1, int c2, int C, int Fs) +{ + VARDECL(opus_val32, tmp); + opus_val32 scale; + int j; + opus_val32 ret = 0; + SAVE_STACK; + + if (subframe==0) return 0; + if (Fs == 48000) + { + subframe *= 2; + offset *= 2; + } else if (Fs == 16000) { + subframe = subframe*2/3; + offset = offset*2/3; + } + ALLOC(tmp, subframe, opus_val32); + + downmix(_x, tmp, subframe, offset, c1, c2, C); +#ifdef FIXED_POINT + scale = (1<<SIG_SHIFT); +#else + scale = 1.f/32768; +#endif + if (c2==-2) + scale /= C; + else if (c2>-1) + scale /= 2; + for (j=0;j<subframe;j++) + tmp[j] *= scale; + if (Fs == 48000) + { + ret = silk_resampler_down2_hp(S, y, tmp, subframe); + } else if (Fs == 24000) { + OPUS_COPY(y, tmp, subframe); + } else if (Fs == 16000) { + VARDECL(opus_val32, tmp3x); + ALLOC(tmp3x, 3*subframe, opus_val32); + /* Don't do this at home! This resampler is horrible and it's only (barely) + usable for the purpose of the analysis because we don't care about all + the aliasing between 8 kHz and 12 kHz. */ + for (j=0;j<subframe;j++) + { + tmp3x[3*j] = tmp[j]; + tmp3x[3*j+1] = tmp[j]; + tmp3x[3*j+2] = tmp[j]; + } + silk_resampler_down2_hp(S, y, tmp3x, 3*subframe); + } + RESTORE_STACK; + return ret; +} + +void tonality_analysis_init(TonalityAnalysisState *tonal, opus_int32 Fs) +{ + /* Initialize reusable fields. */ + tonal->arch = opus_select_arch(); + tonal->Fs = Fs; + /* Clear remaining fields. */ + tonality_analysis_reset(tonal); +} + +void tonality_analysis_reset(TonalityAnalysisState *tonal) +{ + /* Clear non-reusable fields. */ + char *start = (char*)&tonal->TONALITY_ANALYSIS_RESET_START; + OPUS_CLEAR(start, sizeof(TonalityAnalysisState) - (start - (char*)tonal)); +} + +void tonality_get_info(TonalityAnalysisState *tonal, AnalysisInfo *info_out, int len) +{ + int pos; + int curr_lookahead; + float tonality_max; + float tonality_avg; + int tonality_count; + int i; + int pos0; + float prob_avg; + float prob_count; + float prob_min, prob_max; + float vad_prob; + int mpos, vpos; + int bandwidth_span; + + pos = tonal->read_pos; + curr_lookahead = tonal->write_pos-tonal->read_pos; + if (curr_lookahead<0) + curr_lookahead += DETECT_SIZE; + + tonal->read_subframe += len/(tonal->Fs/400); + while (tonal->read_subframe>=8) + { + tonal->read_subframe -= 8; + tonal->read_pos++; + } + if (tonal->read_pos>=DETECT_SIZE) + tonal->read_pos-=DETECT_SIZE; + + /* On long frames, look at the second analysis window rather than the first. */ + if (len > tonal->Fs/50 && pos != tonal->write_pos) + { + pos++; + if (pos==DETECT_SIZE) + pos=0; + } + if (pos == tonal->write_pos) + pos--; + if (pos<0) + pos = DETECT_SIZE-1; + pos0 = pos; + OPUS_COPY(info_out, &tonal->info[pos], 1); + if (!info_out->valid) + return; + tonality_max = tonality_avg = info_out->tonality; + tonality_count = 1; + /* Look at the neighbouring frames and pick largest bandwidth found (to be safe). */ + bandwidth_span = 6; + /* If possible, look ahead for a tone to compensate for the delay in the tone detector. */ + for (i=0;i<3;i++) + { + pos++; + if (pos==DETECT_SIZE) + pos = 0; + if (pos == tonal->write_pos) + break; + tonality_max = MAX32(tonality_max, tonal->info[pos].tonality); + tonality_avg += tonal->info[pos].tonality; + tonality_count++; + info_out->bandwidth = IMAX(info_out->bandwidth, tonal->info[pos].bandwidth); + bandwidth_span--; + } + pos = pos0; + /* Look back in time to see if any has a wider bandwidth than the current frame. */ + for (i=0;i<bandwidth_span;i++) + { + pos--; + if (pos < 0) + pos = DETECT_SIZE-1; + if (pos == tonal->write_pos) + break; + info_out->bandwidth = IMAX(info_out->bandwidth, tonal->info[pos].bandwidth); + } + info_out->tonality = MAX32(tonality_avg/tonality_count, tonality_max-.2f); + + mpos = vpos = pos0; + /* If we have enough look-ahead, compensate for the ~5-frame delay in the music prob and + ~1 frame delay in the VAD prob. */ + if (curr_lookahead > 15) + { + mpos += 5; + if (mpos>=DETECT_SIZE) + mpos -= DETECT_SIZE; + vpos += 1; + if (vpos>=DETECT_SIZE) + vpos -= DETECT_SIZE; + } + + /* The following calculations attempt to minimize a "badness function" + for the transition. When switching from speech to music, the badness + of switching at frame k is + b_k = S*v_k + \sum_{i=0}^{k-1} v_i*(p_i - T) + where + v_i is the activity probability (VAD) at frame i, + p_i is the music probability at frame i + T is the probability threshold for switching + S is the penalty for switching during active audio rather than silence + the current frame has index i=0 + + Rather than apply badness to directly decide when to switch, what we compute + instead is the threshold for which the optimal switching point is now. When + considering whether to switch now (frame 0) or at frame k, we have: + S*v_0 = S*v_k + \sum_{i=0}^{k-1} v_i*(p_i - T) + which gives us: + T = ( \sum_{i=0}^{k-1} v_i*p_i + S*(v_k-v_0) ) / ( \sum_{i=0}^{k-1} v_i ) + We take the min threshold across all positive values of k (up to the maximum + amount of lookahead we have) to give us the threshold for which the current + frame is the optimal switch point. + + The last step is that we need to consider whether we want to switch at all. + For that we use the average of the music probability over the entire window. + If the threshold is higher than that average we're not going to + switch, so we compute a min with the average as well. The result of all these + min operations is music_prob_min, which gives the threshold for switching to music + if we're currently encoding for speech. + + We do the exact opposite to compute music_prob_max which is used for switching + from music to speech. + */ + prob_min = 1.f; + prob_max = 0.f; + vad_prob = tonal->info[vpos].activity_probability; + prob_count = MAX16(.1f, vad_prob); + prob_avg = MAX16(.1f, vad_prob)*tonal->info[mpos].music_prob; + while (1) + { + float pos_vad; + mpos++; + if (mpos==DETECT_SIZE) + mpos = 0; + if (mpos == tonal->write_pos) + break; + vpos++; + if (vpos==DETECT_SIZE) + vpos = 0; + if (vpos == tonal->write_pos) + break; + pos_vad = tonal->info[vpos].activity_probability; + prob_min = MIN16((prob_avg - TRANSITION_PENALTY*(vad_prob - pos_vad))/prob_count, prob_min); + prob_max = MAX16((prob_avg + TRANSITION_PENALTY*(vad_prob - pos_vad))/prob_count, prob_max); + prob_count += MAX16(.1f, pos_vad); + prob_avg += MAX16(.1f, pos_vad)*tonal->info[mpos].music_prob; + } + info_out->music_prob = prob_avg/prob_count; + prob_min = MIN16(prob_avg/prob_count, prob_min); + prob_max = MAX16(prob_avg/prob_count, prob_max); + prob_min = MAX16(prob_min, 0.f); + prob_max = MIN16(prob_max, 1.f); + + /* If we don't have enough look-ahead, do our best to make a decent decision. */ + if (curr_lookahead < 10) + { + float pmin, pmax; + pmin = prob_min; + pmax = prob_max; + pos = pos0; + /* Look for min/max in the past. */ + for (i=0;i<IMIN(tonal->count-1, 15);i++) + { + pos--; + if (pos < 0) + pos = DETECT_SIZE-1; + pmin = MIN16(pmin, tonal->info[pos].music_prob); + pmax = MAX16(pmax, tonal->info[pos].music_prob); + } + /* Bias against switching on active audio. */ + pmin = MAX16(0.f, pmin - .1f*vad_prob); + pmax = MIN16(1.f, pmax + .1f*vad_prob); + prob_min += (1.f-.1f*curr_lookahead)*(pmin - prob_min); + prob_max += (1.f-.1f*curr_lookahead)*(pmax - prob_max); + } + info_out->music_prob_min = prob_min; + info_out->music_prob_max = prob_max; + + /* printf("%f %f %f %f %f\n", prob_min, prob_max, prob_avg/prob_count, vad_prob, info_out->music_prob); */ +} + +static const float std_feature_bias[9] = { + 5.684947f, 3.475288f, 1.770634f, 1.599784f, 3.773215f, + 2.163313f, 1.260756f, 1.116868f, 1.918795f +}; + +#define LEAKAGE_OFFSET 2.5f +#define LEAKAGE_SLOPE 2.f + +#ifdef FIXED_POINT +/* For fixed-point, the input is +/-2^15 shifted up by SIG_SHIFT, so we need to + compensate for that in the energy. */ +#define SCALE_COMPENS (1.f/((opus_int32)1<<(15+SIG_SHIFT))) +#define SCALE_ENER(e) ((SCALE_COMPENS*SCALE_COMPENS)*(e)) +#else +#define SCALE_ENER(e) (e) +#endif + +#ifdef FIXED_POINT +static int is_digital_silence32(const opus_val32* pcm, int frame_size, int channels, int lsb_depth) +{ + int silence = 0; + opus_val32 sample_max = 0; +#ifdef MLP_TRAINING + return 0; +#endif + sample_max = celt_maxabs32(pcm, frame_size*channels); + + silence = (sample_max == 0); + (void)lsb_depth; + return silence; +} +#else +#define is_digital_silence32(pcm, frame_size, channels, lsb_depth) is_digital_silence(pcm, frame_size, channels, lsb_depth) +#endif + +static void tonality_analysis(TonalityAnalysisState *tonal, const CELTMode *celt_mode, const void *x, int len, int offset, int c1, int c2, int C, int lsb_depth, downmix_func downmix) +{ + int i, b; + const kiss_fft_state *kfft; + VARDECL(kiss_fft_cpx, in); + VARDECL(kiss_fft_cpx, out); + int N = 480, N2=240; + float * OPUS_RESTRICT A = tonal->angle; + float * OPUS_RESTRICT dA = tonal->d_angle; + float * OPUS_RESTRICT d2A = tonal->d2_angle; + VARDECL(float, tonality); + VARDECL(float, noisiness); + float band_tonality[NB_TBANDS]; + float logE[NB_TBANDS]; + float BFCC[8]; + float features[25]; + float frame_tonality; + float max_frame_tonality; + /*float tw_sum=0;*/ + float frame_noisiness; + const float pi4 = (float)(M_PI*M_PI*M_PI*M_PI); + float slope=0; + float frame_stationarity; + float relativeE; + float frame_probs[2]; + float alpha, alphaE, alphaE2; + float frame_loudness; + float bandwidth_mask; + int is_masked[NB_TBANDS+1]; + int bandwidth=0; + float maxE = 0; + float noise_floor; + int remaining; + AnalysisInfo *info; + float hp_ener; + float tonality2[240]; + float midE[8]; + float spec_variability=0; + float band_log2[NB_TBANDS+1]; + float leakage_from[NB_TBANDS+1]; + float leakage_to[NB_TBANDS+1]; + float layer_out[MAX_NEURONS]; + float below_max_pitch; + float above_max_pitch; + int is_silence; + SAVE_STACK; + + if (!tonal->initialized) + { + tonal->mem_fill = 240; + tonal->initialized = 1; + } + alpha = 1.f/IMIN(10, 1+tonal->count); + alphaE = 1.f/IMIN(25, 1+tonal->count); + /* Noise floor related decay for bandwidth detection: -2.2 dB/second */ + alphaE2 = 1.f/IMIN(100, 1+tonal->count); + if (tonal->count <= 1) alphaE2 = 1; + + if (tonal->Fs == 48000) + { + /* len and offset are now at 24 kHz. */ + len/= 2; + offset /= 2; + } else if (tonal->Fs == 16000) { + len = 3*len/2; + offset = 3*offset/2; + } + + kfft = celt_mode->mdct.kfft[0]; + tonal->hp_ener_accum += (float)downmix_and_resample(downmix, x, + &tonal->inmem[tonal->mem_fill], tonal->downmix_state, + IMIN(len, ANALYSIS_BUF_SIZE-tonal->mem_fill), offset, c1, c2, C, tonal->Fs); + if (tonal->mem_fill+len < ANALYSIS_BUF_SIZE) + { + tonal->mem_fill += len; + /* Don't have enough to update the analysis */ + RESTORE_STACK; + return; + } + hp_ener = tonal->hp_ener_accum; + info = &tonal->info[tonal->write_pos++]; + if (tonal->write_pos>=DETECT_SIZE) + tonal->write_pos-=DETECT_SIZE; + + is_silence = is_digital_silence32(tonal->inmem, ANALYSIS_BUF_SIZE, 1, lsb_depth); + + ALLOC(in, 480, kiss_fft_cpx); + ALLOC(out, 480, kiss_fft_cpx); + ALLOC(tonality, 240, float); + ALLOC(noisiness, 240, float); + for (i=0;i<N2;i++) + { + float w = analysis_window[i]; + in[i].r = (kiss_fft_scalar)(w*tonal->inmem[i]); + in[i].i = (kiss_fft_scalar)(w*tonal->inmem[N2+i]); + in[N-i-1].r = (kiss_fft_scalar)(w*tonal->inmem[N-i-1]); + in[N-i-1].i = (kiss_fft_scalar)(w*tonal->inmem[N+N2-i-1]); + } + OPUS_MOVE(tonal->inmem, tonal->inmem+ANALYSIS_BUF_SIZE-240, 240); + remaining = len - (ANALYSIS_BUF_SIZE-tonal->mem_fill); + tonal->hp_ener_accum = (float)downmix_and_resample(downmix, x, + &tonal->inmem[240], tonal->downmix_state, remaining, + offset+ANALYSIS_BUF_SIZE-tonal->mem_fill, c1, c2, C, tonal->Fs); + tonal->mem_fill = 240 + remaining; + if (is_silence) + { + /* On silence, copy the previous analysis. */ + int prev_pos = tonal->write_pos-2; + if (prev_pos < 0) + prev_pos += DETECT_SIZE; + OPUS_COPY(info, &tonal->info[prev_pos], 1); + RESTORE_STACK; + return; + } + opus_fft(kfft, in, out, tonal->arch); +#ifndef FIXED_POINT + /* If there's any NaN on the input, the entire output will be NaN, so we only need to check one value. */ + if (celt_isnan(out[0].r)) + { + info->valid = 0; + RESTORE_STACK; + return; + } +#endif + + for (i=1;i<N2;i++) + { + float X1r, X2r, X1i, X2i; + float angle, d_angle, d2_angle; + float angle2, d_angle2, d2_angle2; + float mod1, mod2, avg_mod; + X1r = (float)out[i].r+out[N-i].r; + X1i = (float)out[i].i-out[N-i].i; + X2r = (float)out[i].i+out[N-i].i; + X2i = (float)out[N-i].r-out[i].r; + + angle = (float)(.5f/M_PI)*fast_atan2f(X1i, X1r); + d_angle = angle - A[i]; + d2_angle = d_angle - dA[i]; + + angle2 = (float)(.5f/M_PI)*fast_atan2f(X2i, X2r); + d_angle2 = angle2 - angle; + d2_angle2 = d_angle2 - d_angle; + + mod1 = d2_angle - (float)float2int(d2_angle); + noisiness[i] = ABS16(mod1); + mod1 *= mod1; + mod1 *= mod1; + + mod2 = d2_angle2 - (float)float2int(d2_angle2); + noisiness[i] += ABS16(mod2); + mod2 *= mod2; + mod2 *= mod2; + + avg_mod = .25f*(d2A[i]+mod1+2*mod2); + /* This introduces an extra delay of 2 frames in the detection. */ + tonality[i] = 1.f/(1.f+40.f*16.f*pi4*avg_mod)-.015f; + /* No delay on this detection, but it's less reliable. */ + tonality2[i] = 1.f/(1.f+40.f*16.f*pi4*mod2)-.015f; + + A[i] = angle2; + dA[i] = d_angle2; + d2A[i] = mod2; + } + for (i=2;i<N2-1;i++) + { + float tt = MIN32(tonality2[i], MAX32(tonality2[i-1], tonality2[i+1])); + tonality[i] = .9f*MAX32(tonality[i], tt-.1f); + } + frame_tonality = 0; + max_frame_tonality = 0; + /*tw_sum = 0;*/ + info->activity = 0; + frame_noisiness = 0; + frame_stationarity = 0; + if (!tonal->count) + { + for (b=0;b<NB_TBANDS;b++) + { + tonal->lowE[b] = 1e10; + tonal->highE[b] = -1e10; + } + } + relativeE = 0; + frame_loudness = 0; + /* The energy of the very first band is special because of DC. */ + { + float E = 0; + float X1r, X2r; + X1r = 2*(float)out[0].r; + X2r = 2*(float)out[0].i; + E = X1r*X1r + X2r*X2r; + for (i=1;i<4;i++) + { + float binE = out[i].r*(float)out[i].r + out[N-i].r*(float)out[N-i].r + + out[i].i*(float)out[i].i + out[N-i].i*(float)out[N-i].i; + E += binE; + } + E = SCALE_ENER(E); + band_log2[0] = .5f*1.442695f*(float)log(E+1e-10f); + } + for (b=0;b<NB_TBANDS;b++) + { + float E=0, tE=0, nE=0; + float L1, L2; + float stationarity; + for (i=tbands[b];i<tbands[b+1];i++) + { + float binE = out[i].r*(float)out[i].r + out[N-i].r*(float)out[N-i].r + + out[i].i*(float)out[i].i + out[N-i].i*(float)out[N-i].i; + binE = SCALE_ENER(binE); + E += binE; + tE += binE*MAX32(0, tonality[i]); + nE += binE*2.f*(.5f-noisiness[i]); + } +#ifndef FIXED_POINT + /* Check for extreme band energies that could cause NaNs later. */ + if (!(E<1e9f) || celt_isnan(E)) + { + info->valid = 0; + RESTORE_STACK; + return; + } +#endif + + tonal->E[tonal->E_count][b] = E; + frame_noisiness += nE/(1e-15f+E); + + frame_loudness += (float)sqrt(E+1e-10f); + logE[b] = (float)log(E+1e-10f); + band_log2[b+1] = .5f*1.442695f*(float)log(E+1e-10f); + tonal->logE[tonal->E_count][b] = logE[b]; + if (tonal->count==0) + tonal->highE[b] = tonal->lowE[b] = logE[b]; + if (tonal->highE[b] > tonal->lowE[b] + 7.5) + { + if (tonal->highE[b] - logE[b] > logE[b] - tonal->lowE[b]) + tonal->highE[b] -= .01f; + else + tonal->lowE[b] += .01f; + } + if (logE[b] > tonal->highE[b]) + { + tonal->highE[b] = logE[b]; + tonal->lowE[b] = MAX32(tonal->highE[b]-15, tonal->lowE[b]); + } else if (logE[b] < tonal->lowE[b]) + { + tonal->lowE[b] = logE[b]; + tonal->highE[b] = MIN32(tonal->lowE[b]+15, tonal->highE[b]); + } + relativeE += (logE[b]-tonal->lowE[b])/(1e-5f + (tonal->highE[b]-tonal->lowE[b])); + + L1=L2=0; + for (i=0;i<NB_FRAMES;i++) + { + L1 += (float)sqrt(tonal->E[i][b]); + L2 += tonal->E[i][b]; + } + + stationarity = MIN16(0.99f,L1/(float)sqrt(1e-15+NB_FRAMES*L2)); + stationarity *= stationarity; + stationarity *= stationarity; + frame_stationarity += stationarity; + /*band_tonality[b] = tE/(1e-15+E)*/; + band_tonality[b] = MAX16(tE/(1e-15f+E), stationarity*tonal->prev_band_tonality[b]); +#if 0 + if (b>=NB_TONAL_SKIP_BANDS) + { + frame_tonality += tweight[b]*band_tonality[b]; + tw_sum += tweight[b]; + } +#else + frame_tonality += band_tonality[b]; + if (b>=NB_TBANDS-NB_TONAL_SKIP_BANDS) + frame_tonality -= band_tonality[b-NB_TBANDS+NB_TONAL_SKIP_BANDS]; +#endif + max_frame_tonality = MAX16(max_frame_tonality, (1.f+.03f*(b-NB_TBANDS))*frame_tonality); + slope += band_tonality[b]*(b-8); + /*printf("%f %f ", band_tonality[b], stationarity);*/ + tonal->prev_band_tonality[b] = band_tonality[b]; + } + + leakage_from[0] = band_log2[0]; + leakage_to[0] = band_log2[0] - LEAKAGE_OFFSET; + for (b=1;b<NB_TBANDS+1;b++) + { + float leak_slope = LEAKAGE_SLOPE*(tbands[b]-tbands[b-1])/4; + leakage_from[b] = MIN16(leakage_from[b-1]+leak_slope, band_log2[b]); + leakage_to[b] = MAX16(leakage_to[b-1]-leak_slope, band_log2[b]-LEAKAGE_OFFSET); + } + for (b=NB_TBANDS-2;b>=0;b--) + { + float leak_slope = LEAKAGE_SLOPE*(tbands[b+1]-tbands[b])/4; + leakage_from[b] = MIN16(leakage_from[b+1]+leak_slope, leakage_from[b]); + leakage_to[b] = MAX16(leakage_to[b+1]-leak_slope, leakage_to[b]); + } + celt_assert(NB_TBANDS+1 <= LEAK_BANDS); + for (b=0;b<NB_TBANDS+1;b++) + { + /* leak_boost[] is made up of two terms. The first, based on leakage_to[], + represents the boost needed to overcome the amount of analysis leakage + cause in a weaker band b by louder neighbouring bands. + The second, based on leakage_from[], applies to a loud band b for + which the quantization noise causes synthesis leakage to the weaker + neighbouring bands. */ + float boost = MAX16(0, leakage_to[b] - band_log2[b]) + + MAX16(0, band_log2[b] - (leakage_from[b]+LEAKAGE_OFFSET)); + info->leak_boost[b] = IMIN(255, (int)floor(.5 + 64.f*boost)); + } + for (;b<LEAK_BANDS;b++) info->leak_boost[b] = 0; + + for (i=0;i<NB_FRAMES;i++) + { + int j; + float mindist = 1e15f; + for (j=0;j<NB_FRAMES;j++) + { + int k; + float dist=0; + for (k=0;k<NB_TBANDS;k++) + { + float tmp; + tmp = tonal->logE[i][k] - tonal->logE[j][k]; + dist += tmp*tmp; + } + if (j!=i) + mindist = MIN32(mindist, dist); + } + spec_variability += mindist; + } + spec_variability = (float)sqrt(spec_variability/NB_FRAMES/NB_TBANDS); + bandwidth_mask = 0; + bandwidth = 0; + maxE = 0; + noise_floor = 5.7e-4f/(1<<(IMAX(0,lsb_depth-8))); + noise_floor *= noise_floor; + below_max_pitch=0; + above_max_pitch=0; + for (b=0;b<NB_TBANDS;b++) + { + float E=0; + float Em; + int band_start, band_end; + /* Keep a margin of 300 Hz for aliasing */ + band_start = tbands[b]; + band_end = tbands[b+1]; + for (i=band_start;i<band_end;i++) + { + float binE = out[i].r*(float)out[i].r + out[N-i].r*(float)out[N-i].r + + out[i].i*(float)out[i].i + out[N-i].i*(float)out[N-i].i; + E += binE; + } + E = SCALE_ENER(E); + maxE = MAX32(maxE, E); + if (band_start < 64) + { + below_max_pitch += E; + } else { + above_max_pitch += E; + } + tonal->meanE[b] = MAX32((1-alphaE2)*tonal->meanE[b], E); + Em = MAX32(E, tonal->meanE[b]); + /* Consider the band "active" only if all these conditions are met: + 1) less than 90 dB below the peak band (maximal masking possible considering + both the ATH and the loudness-dependent slope of the spreading function) + 2) above the PCM quantization noise floor + We use b+1 because the first CELT band isn't included in tbands[] + */ + if (E*1e9f > maxE && (Em > 3*noise_floor*(band_end-band_start) || E > noise_floor*(band_end-band_start))) + bandwidth = b+1; + /* Check if the band is masked (see below). */ + is_masked[b] = E < (tonal->prev_bandwidth >= b+1 ? .01f : .05f)*bandwidth_mask; + /* Use a simple follower with 13 dB/Bark slope for spreading function. */ + bandwidth_mask = MAX32(.05f*bandwidth_mask, E); + } + /* Special case for the last two bands, for which we don't have spectrum but only + the energy above 12 kHz. The difficulty here is that the high-pass we use + leaks some LF energy, so we need to increase the threshold without accidentally cutting + off the band. */ + if (tonal->Fs == 48000) { + float noise_ratio; + float Em; + float E = hp_ener*(1.f/(60*60)); + noise_ratio = tonal->prev_bandwidth==20 ? 10.f : 30.f; + +#ifdef FIXED_POINT + /* silk_resampler_down2_hp() shifted right by an extra 8 bits. */ + E *= 256.f*(1.f/Q15ONE)*(1.f/Q15ONE); +#endif + above_max_pitch += E; + tonal->meanE[b] = MAX32((1-alphaE2)*tonal->meanE[b], E); + Em = MAX32(E, tonal->meanE[b]); + if (Em > 3*noise_ratio*noise_floor*160 || E > noise_ratio*noise_floor*160) + bandwidth = 20; + /* Check if the band is masked (see below). */ + is_masked[b] = E < (tonal->prev_bandwidth == 20 ? .01f : .05f)*bandwidth_mask; + } + if (above_max_pitch > below_max_pitch) + info->max_pitch_ratio = below_max_pitch/above_max_pitch; + else + info->max_pitch_ratio = 1; + /* In some cases, resampling aliasing can create a small amount of energy in the first band + being cut. So if the last band is masked, we don't include it. */ + if (bandwidth == 20 && is_masked[NB_TBANDS]) + bandwidth-=2; + else if (bandwidth > 0 && bandwidth <= NB_TBANDS && is_masked[bandwidth-1]) + bandwidth--; + if (tonal->count<=2) + bandwidth = 20; + frame_loudness = 20*(float)log10(frame_loudness); + tonal->Etracker = MAX32(tonal->Etracker-.003f, frame_loudness); + tonal->lowECount *= (1-alphaE); + if (frame_loudness < tonal->Etracker-30) + tonal->lowECount += alphaE; + + for (i=0;i<8;i++) + { + float sum=0; + for (b=0;b<16;b++) + sum += dct_table[i*16+b]*logE[b]; + BFCC[i] = sum; + } + for (i=0;i<8;i++) + { + float sum=0; + for (b=0;b<16;b++) + sum += dct_table[i*16+b]*.5f*(tonal->highE[b]+tonal->lowE[b]); + midE[i] = sum; + } + + frame_stationarity /= NB_TBANDS; + relativeE /= NB_TBANDS; + if (tonal->count<10) + relativeE = .5f; + frame_noisiness /= NB_TBANDS; +#if 1 + info->activity = frame_noisiness + (1-frame_noisiness)*relativeE; +#else + info->activity = .5*(1+frame_noisiness-frame_stationarity); +#endif + frame_tonality = (max_frame_tonality/(NB_TBANDS-NB_TONAL_SKIP_BANDS)); + frame_tonality = MAX16(frame_tonality, tonal->prev_tonality*.8f); + tonal->prev_tonality = frame_tonality; + + slope /= 8*8; + info->tonality_slope = slope; + + tonal->E_count = (tonal->E_count+1)%NB_FRAMES; + tonal->count = IMIN(tonal->count+1, ANALYSIS_COUNT_MAX); + info->tonality = frame_tonality; + + for (i=0;i<4;i++) + features[i] = -0.12299f*(BFCC[i]+tonal->mem[i+24]) + 0.49195f*(tonal->mem[i]+tonal->mem[i+16]) + 0.69693f*tonal->mem[i+8] - 1.4349f*tonal->cmean[i]; + + for (i=0;i<4;i++) + tonal->cmean[i] = (1-alpha)*tonal->cmean[i] + alpha*BFCC[i]; + + for (i=0;i<4;i++) + features[4+i] = 0.63246f*(BFCC[i]-tonal->mem[i+24]) + 0.31623f*(tonal->mem[i]-tonal->mem[i+16]); + for (i=0;i<3;i++) + features[8+i] = 0.53452f*(BFCC[i]+tonal->mem[i+24]) - 0.26726f*(tonal->mem[i]+tonal->mem[i+16]) -0.53452f*tonal->mem[i+8]; + + if (tonal->count > 5) + { + for (i=0;i<9;i++) + tonal->std[i] = (1-alpha)*tonal->std[i] + alpha*features[i]*features[i]; + } + for (i=0;i<4;i++) + features[i] = BFCC[i]-midE[i]; + + for (i=0;i<8;i++) + { + tonal->mem[i+24] = tonal->mem[i+16]; + tonal->mem[i+16] = tonal->mem[i+8]; + tonal->mem[i+8] = tonal->mem[i]; + tonal->mem[i] = BFCC[i]; + } + for (i=0;i<9;i++) + features[11+i] = (float)sqrt(tonal->std[i]) - std_feature_bias[i]; + features[18] = spec_variability - 0.78f; + features[20] = info->tonality - 0.154723f; + features[21] = info->activity - 0.724643f; + features[22] = frame_stationarity - 0.743717f; + features[23] = info->tonality_slope + 0.069216f; + features[24] = tonal->lowECount - 0.067930f; + + compute_dense(&layer0, layer_out, features); + compute_gru(&layer1, tonal->rnn_state, layer_out); + compute_dense(&layer2, frame_probs, tonal->rnn_state); + + /* Probability of speech or music vs noise */ + info->activity_probability = frame_probs[1]; + info->music_prob = frame_probs[0]; + + /*printf("%f %f %f\n", frame_probs[0], frame_probs[1], info->music_prob);*/ +#ifdef MLP_TRAINING + for (i=0;i<25;i++) + printf("%f ", features[i]); + printf("\n"); +#endif + + info->bandwidth = bandwidth; + tonal->prev_bandwidth = bandwidth; + /*printf("%d %d\n", info->bandwidth, info->opus_bandwidth);*/ + info->noisiness = frame_noisiness; + info->valid = 1; + RESTORE_STACK; +} + +void run_analysis(TonalityAnalysisState *analysis, const CELTMode *celt_mode, const void *analysis_pcm, + int analysis_frame_size, int frame_size, int c1, int c2, int C, opus_int32 Fs, + int lsb_depth, downmix_func downmix, AnalysisInfo *analysis_info) +{ + int offset; + int pcm_len; + + analysis_frame_size -= analysis_frame_size&1; + if (analysis_pcm != NULL) + { + /* Avoid overflow/wrap-around of the analysis buffer */ + analysis_frame_size = IMIN((DETECT_SIZE-5)*Fs/50, analysis_frame_size); + + pcm_len = analysis_frame_size - analysis->analysis_offset; + offset = analysis->analysis_offset; + while (pcm_len>0) { + tonality_analysis(analysis, celt_mode, analysis_pcm, IMIN(Fs/50, pcm_len), offset, c1, c2, C, lsb_depth, downmix); + offset += Fs/50; + pcm_len -= Fs/50; + } + analysis->analysis_offset = analysis_frame_size; + + analysis->analysis_offset -= frame_size; + } + + tonality_get_info(analysis, analysis_info, frame_size); +} + +#endif /* DISABLE_FLOAT_API */ |