Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

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 */