Create local copy of Opus v1.1.2

webrtc/modules/audio_coding/codecs/opus/opus/src/src/opus_encoder.c

Index: webrtc/modules/audio_coding/codecs/opus/opus/src/src/opus_encoder.c
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus/src/src/opus_encoder.c b/webrtc/modules/audio_coding/codecs/opus/opus/src/src/opus_encoder.c
new file mode 100644
index 0000000000000000000000000000000000000000..a7e19127d69d11611ce14187e023e2e4c115ad14
--- /dev/null
+++ b/webrtc/modules/audio_coding/codecs/opus/opus/src/src/opus_encoder.c
@@ -0,0 +1,2527 @@
+/* Copyright (c) 2010-2011 Xiph.Org Foundation, Skype Limited
+   Written by Jean-Marc Valin and Koen Vos */
+/*
+   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 COPYRIGHT OWNER
+   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
+
+#include <stdarg.h>
+#include "celt.h"
+#include "entenc.h"
+#include "modes.h"
+#include "API.h"
+#include "stack_alloc.h"
+#include "float_cast.h"
+#include "opus.h"
+#include "arch.h"
+#include "pitch.h"
+#include "opus_private.h"
+#include "os_support.h"
+#include "cpu_support.h"
+#include "analysis.h"
+#include "mathops.h"
+#include "tuning_parameters.h"
+#ifdef FIXED_POINT
+#include "fixed/structs_FIX.h"
+#else
+#include "float/structs_FLP.h"
+#endif
+
+#define MAX_ENCODER_BUFFER 480
+
+typedef struct {
+   opus_val32 XX, XY, YY;
+   opus_val16 smoothed_width;
+   opus_val16 max_follower;
+} StereoWidthState;
+
+struct OpusEncoder {
+    int          celt_enc_offset;
+    int          silk_enc_offset;
+    silk_EncControlStruct silk_mode;
+    int          application;
+    int          channels;
+    int          delay_compensation;
+    int          force_channels;
+    int          signal_type;
+    int          user_bandwidth;
+    int          max_bandwidth;
+    int          user_forced_mode;
+    int          voice_ratio;
+    opus_int32   Fs;
+    int          use_vbr;
+    int          vbr_constraint;
+    int          variable_duration;
+    opus_int32   bitrate_bps;
+    opus_int32   user_bitrate_bps;
+    int          lsb_depth;
+    int          encoder_buffer;
+    int          lfe;
+    int          arch;
+#ifndef DISABLE_FLOAT_API
+    TonalityAnalysisState analysis;
+#endif
+
+#define OPUS_ENCODER_RESET_START stream_channels
+    int          stream_channels;
+    opus_int16   hybrid_stereo_width_Q14;
+    opus_int32   variable_HP_smth2_Q15;
+    opus_val16   prev_HB_gain;
+    opus_val32   hp_mem[4];
+    int          mode;
+    int          prev_mode;
+    int          prev_channels;
+    int          prev_framesize;
+    int          bandwidth;
+    int          silk_bw_switch;
+    /* Sampling rate (at the API level) */
+    int          first;
+    opus_val16 * energy_masking;
+    StereoWidthState width_mem;
+    opus_val16   delay_buffer[MAX_ENCODER_BUFFER*2];
+#ifndef DISABLE_FLOAT_API
+    int          detected_bandwidth;
+#endif
+    opus_uint32  rangeFinal;
+};
+
+/* Transition tables for the voice and music. First column is the
+   middle (memoriless) threshold. The second column is the hysteresis
+   (difference with the middle) */
+static const opus_int32 mono_voice_bandwidth_thresholds[8] = {
+        11000, 1000, /* NB<->MB */
+        14000, 1000, /* MB<->WB */
+        17000, 1000, /* WB<->SWB */
+        21000, 2000, /* SWB<->FB */
+};
+static const opus_int32 mono_music_bandwidth_thresholds[8] = {
+        12000, 1000, /* NB<->MB */
+        15000, 1000, /* MB<->WB */
+        18000, 2000, /* WB<->SWB */
+        22000, 2000, /* SWB<->FB */
+};
+static const opus_int32 stereo_voice_bandwidth_thresholds[8] = {
+        11000, 1000, /* NB<->MB */
+        14000, 1000, /* MB<->WB */
+        21000, 2000, /* WB<->SWB */
+        28000, 2000, /* SWB<->FB */
+};
+static const opus_int32 stereo_music_bandwidth_thresholds[8] = {
+        12000, 1000, /* NB<->MB */
+        18000, 2000, /* MB<->WB */
+        21000, 2000, /* WB<->SWB */
+        30000, 2000, /* SWB<->FB */
+};
+/* Threshold bit-rates for switching between mono and stereo */
+static const opus_int32 stereo_voice_threshold = 30000;
+static const opus_int32 stereo_music_threshold = 30000;
+
+/* Threshold bit-rate for switching between SILK/hybrid and CELT-only */
+static const opus_int32 mode_thresholds[2][2] = {
+      /* voice */ /* music */
+      {  64000,      16000}, /* mono */
+      {  36000,      16000}, /* stereo */
+};
+
+int opus_encoder_get_size(int channels)
+{
+    int silkEncSizeBytes, celtEncSizeBytes;
+    int ret;
+    if (channels<1 || channels > 2)
+        return 0;
+    ret = silk_Get_Encoder_Size( &silkEncSizeBytes );
+    if (ret)
+        return 0;
+    silkEncSizeBytes = align(silkEncSizeBytes);
+    celtEncSizeBytes = celt_encoder_get_size(channels);
+    return align(sizeof(OpusEncoder))+silkEncSizeBytes+celtEncSizeBytes;
+}
+
+int opus_encoder_init(OpusEncoder* st, opus_int32 Fs, int channels, int application)
+{
+    void *silk_enc;
+    CELTEncoder *celt_enc;
+    int err;
+    int ret, silkEncSizeBytes;
+
+   if((Fs!=48000&&Fs!=24000&&Fs!=16000&&Fs!=12000&&Fs!=8000)||(channels!=1&&channels!=2)||
+        (application != OPUS_APPLICATION_VOIP && application != OPUS_APPLICATION_AUDIO
+        && application != OPUS_APPLICATION_RESTRICTED_LOWDELAY))
+        return OPUS_BAD_ARG;
+
+    OPUS_CLEAR((char*)st, opus_encoder_get_size(channels));
+    /* Create SILK encoder */
+    ret = silk_Get_Encoder_Size( &silkEncSizeBytes );
+    if (ret)
+        return OPUS_BAD_ARG;
+    silkEncSizeBytes = align(silkEncSizeBytes);
+    st->silk_enc_offset = align(sizeof(OpusEncoder));
+    st->celt_enc_offset = st->silk_enc_offset+silkEncSizeBytes;
+    silk_enc = (char*)st+st->silk_enc_offset;
+    celt_enc = (CELTEncoder*)((char*)st+st->celt_enc_offset);
+
+    st->stream_channels = st->channels = channels;
+
+    st->Fs = Fs;
+
+    st->arch = opus_select_arch();
+
+    ret = silk_InitEncoder( silk_enc, st->arch, &st->silk_mode );
+    if(ret)return OPUS_INTERNAL_ERROR;
+
+    /* default SILK parameters */
+    st->silk_mode.nChannelsAPI              = channels;
+    st->silk_mode.nChannelsInternal         = channels;
+    st->silk_mode.API_sampleRate            = st->Fs;
+    st->silk_mode.maxInternalSampleRate     = 16000;
+    st->silk_mode.minInternalSampleRate     = 8000;
+    st->silk_mode.desiredInternalSampleRate = 16000;
+    st->silk_mode.payloadSize_ms            = 20;
+    st->silk_mode.bitRate                   = 25000;
+    st->silk_mode.packetLossPercentage      = 0;
+    st->silk_mode.complexity                = 9;
+    st->silk_mode.useInBandFEC              = 0;
+    st->silk_mode.useDTX                    = 0;
+    st->silk_mode.useCBR                    = 0;
+    st->silk_mode.reducedDependency         = 0;
+
+    /* Create CELT encoder */
+    /* Initialize CELT encoder */
+    err = celt_encoder_init(celt_enc, Fs, channels, st->arch);
+    if(err!=OPUS_OK)return OPUS_INTERNAL_ERROR;
+
+    celt_encoder_ctl(celt_enc, CELT_SET_SIGNALLING(0));
+    celt_encoder_ctl(celt_enc, OPUS_SET_COMPLEXITY(st->silk_mode.complexity));
+
+    st->use_vbr = 1;
+    /* Makes constrained VBR the default (safer for real-time use) */
+    st->vbr_constraint = 1;
+    st->user_bitrate_bps = OPUS_AUTO;
+    st->bitrate_bps = 3000+Fs*channels;
+    st->application = application;
+    st->signal_type = OPUS_AUTO;
+    st->user_bandwidth = OPUS_AUTO;
+    st->max_bandwidth = OPUS_BANDWIDTH_FULLBAND;
+    st->force_channels = OPUS_AUTO;
+    st->user_forced_mode = OPUS_AUTO;
+    st->voice_ratio = -1;
+    st->encoder_buffer = st->Fs/100;
+    st->lsb_depth = 24;
+    st->variable_duration = OPUS_FRAMESIZE_ARG;
+
+    /* Delay compensation of 4 ms (2.5 ms for SILK's extra look-ahead
+       + 1.5 ms for SILK resamplers and stereo prediction) */
+    st->delay_compensation = st->Fs/250;
+
+    st->hybrid_stereo_width_Q14 = 1 << 14;
+    st->prev_HB_gain = Q15ONE;
+    st->variable_HP_smth2_Q15 = silk_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 );
+    st->first = 1;
+    st->mode = MODE_HYBRID;
+    st->bandwidth = OPUS_BANDWIDTH_FULLBAND;
+
+#ifndef DISABLE_FLOAT_API
+    tonality_analysis_init(&st->analysis);
+#endif
+
+    return OPUS_OK;
+}
+
+static unsigned char gen_toc(int mode, int framerate, int bandwidth, int channels)
+{
+   int period;
+   unsigned char toc;
+   period = 0;
+   while (framerate < 400)
+   {
+       framerate <<= 1;
+       period++;
+   }
+   if (mode == MODE_SILK_ONLY)
+   {
+       toc = (bandwidth-OPUS_BANDWIDTH_NARROWBAND)<<5;
+       toc |= (period-2)<<3;
+   } else if (mode == MODE_CELT_ONLY)
+   {
+       int tmp = bandwidth-OPUS_BANDWIDTH_MEDIUMBAND;
+       if (tmp < 0)
+           tmp = 0;
+       toc = 0x80;
+       toc |= tmp << 5;
+       toc |= period<<3;
+   } else /* Hybrid */
+   {
+       toc = 0x60;
+       toc |= (bandwidth-OPUS_BANDWIDTH_SUPERWIDEBAND)<<4;
+       toc |= (period-2)<<3;
+   }
+   toc |= (channels==2)<<2;
+   return toc;
+}
+
+#ifndef FIXED_POINT
+static 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] = (opus_val32)(A_Q28[0] * (1.f/((opus_int32)1<<28)));
+    A[1] = (opus_val32)(A_Q28[1] * (1.f/((opus_int32)1<<28)));
+    B[0] = (opus_val32)(B_Q28[0] * (1.f/((opus_int32)1<<28)));
+    B[1] = (opus_val32)(B_Q28[1] * (1.f/((opus_int32)1<<28)));
+    B[2] = (opus_val32)(B_Q28[2] * (1.f/((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 + VERY_SMALL;
+
+        /* 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;
+
+   silk_assert( cutoff_Hz <= silk_int32_MAX / SILK_FIX_CONST( 1.5 * 3.14159 / 1000, 19 ) );
+   Fc_Q19 = silk_DIV32_16( silk_SMULBB( SILK_FIX_CONST( 1.5 * 3.14159 / 1000, 19 ), cutoff_Hz ), Fs/1000 );
+   silk_assert( Fc_Q19 > 0 && Fc_Q19 < 32768 );
+
+   r_Q28 = SILK_FIX_CONST( 1.0, 28 ) - silk_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 ] = silk_LSHIFT( -r_Q28, 1 );
+   B_Q28[ 2 ] = r_Q28;
+
+   /* -r * ( 2 - Fc * Fc ); */
+   r_Q22  = silk_RSHIFT( r_Q28, 6 );
+   A_Q28[ 0 ] = silk_SMULWW( r_Q22, silk_SMULWW( Fc_Q19, Fc_Q19 ) - SILK_FIX_CONST( 2.0,  22 ) );
+   A_Q28[ 1 ] = silk_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
+}
+
+#ifdef FIXED_POINT
+static void dc_reject(const opus_val16 *in, opus_int32 cutoff_Hz, opus_val16 *out, opus_val32 *hp_mem, int len, int channels, opus_int32 Fs)
+{
+   int c, i;
+   int shift;
+
+   /* Approximates -round(log2(4.*cutoff_Hz/Fs)) */
+   shift=celt_ilog2(Fs/(cutoff_Hz*3));
+   for (c=0;c<channels;c++)
+   {
+      for (i=0;i<len;i++)
+      {
+         opus_val32 x, tmp, y;
+         x = SHL32(EXTEND32(in[channels*i+c]), 15);
+         /* First stage */
+         tmp = x-hp_mem[2*c];
+         hp_mem[2*c] = hp_mem[2*c] + PSHR32(x - hp_mem[2*c], shift);
+         /* Second stage */
+         y = tmp - hp_mem[2*c+1];
+         hp_mem[2*c+1] = hp_mem[2*c+1] + PSHR32(tmp - hp_mem[2*c+1], shift);
+         out[channels*i+c] = EXTRACT16(SATURATE(PSHR32(y, 15), 32767));
+      }
+   }
+}
+
+#else
+static void dc_reject(const opus_val16 *in, opus_int32 cutoff_Hz, opus_val16 *out, opus_val32 *hp_mem, int len, int channels, opus_int32 Fs)
+{
+   int c, i;
+   float coef;
+
+   coef = 4.0f*cutoff_Hz/Fs;
+   for (c=0;c<channels;c++)
+   {
+      for (i=0;i<len;i++)
+      {
+         opus_val32 x, tmp, y;
+         x = in[channels*i+c];
+         /* First stage */
+         tmp = x-hp_mem[2*c];
+         hp_mem[2*c] = hp_mem[2*c] + coef*(x - hp_mem[2*c]) + VERY_SMALL;
+         /* Second stage */
+         y = tmp - hp_mem[2*c+1];
+         hp_mem[2*c+1] = hp_mem[2*c+1] + coef*(tmp - hp_mem[2*c+1]) + VERY_SMALL;
+         out[channels*i+c] = y;
+      }
+   }
+}
+#endif
+
+static void stereo_fade(const opus_val16 *in, opus_val16 *out, opus_val16 g1, opus_val16 g2,
+        int overlap48, int frame_size, int channels, const opus_val16 *window, opus_int32 Fs)
+{
+    int i;
+    int overlap;
+    int inc;
+    inc = 48000/Fs;
+    overlap=overlap48/inc;
+    g1 = Q15ONE-g1;
+    g2 = Q15ONE-g2;
+    for (i=0;i<overlap;i++)
+    {
+       opus_val32 diff;
+       opus_val16 g, w;
+       w = MULT16_16_Q15(window[i*inc], window[i*inc]);
+       g = SHR32(MAC16_16(MULT16_16(w,g2),
+             Q15ONE-w, g1), 15);
+       diff = EXTRACT16(HALF32((opus_val32)in[i*channels] - (opus_val32)in[i*channels+1]));
+       diff = MULT16_16_Q15(g, diff);
+       out[i*channels] = out[i*channels] - diff;
+       out[i*channels+1] = out[i*channels+1] + diff;
+    }
+    for (;i<frame_size;i++)
+    {
+       opus_val32 diff;
+       diff = EXTRACT16(HALF32((opus_val32)in[i*channels] - (opus_val32)in[i*channels+1]));
+       diff = MULT16_16_Q15(g2, diff);
+       out[i*channels] = out[i*channels] - diff;
+       out[i*channels+1] = out[i*channels+1] + diff;
+    }
+}
+
+static void gain_fade(const opus_val16 *in, opus_val16 *out, opus_val16 g1, opus_val16 g2,
+        int overlap48, int frame_size, int channels, const opus_val16 *window, opus_int32 Fs)
+{
+    int i;
+    int inc;
+    int overlap;
+    int c;
+    inc = 48000/Fs;
+    overlap=overlap48/inc;
+    if (channels==1)
+    {
+       for (i=0;i<overlap;i++)
+       {
+          opus_val16 g, w;
+          w = MULT16_16_Q15(window[i*inc], window[i*inc]);
+          g = SHR32(MAC16_16(MULT16_16(w,g2),
+                Q15ONE-w, g1), 15);
+          out[i] = MULT16_16_Q15(g, in[i]);
+       }
+    } else {
+       for (i=0;i<overlap;i++)
+       {
+          opus_val16 g, w;
+          w = MULT16_16_Q15(window[i*inc], window[i*inc]);
+          g = SHR32(MAC16_16(MULT16_16(w,g2),
+                Q15ONE-w, g1), 15);
+          out[i*2] = MULT16_16_Q15(g, in[i*2]);
+          out[i*2+1] = MULT16_16_Q15(g, in[i*2+1]);
+       }
+    }
+    c=0;do {
+       for (i=overlap;i<frame_size;i++)
+       {
+          out[i*channels+c] = MULT16_16_Q15(g2, in[i*channels+c]);
+       }
+    }
+    while (++c<channels);
+}
+
+OpusEncoder *opus_encoder_create(opus_int32 Fs, int channels, int application, int *error)
+{
+   int ret;
+   OpusEncoder *st;
+   if((Fs!=48000&&Fs!=24000&&Fs!=16000&&Fs!=12000&&Fs!=8000)||(channels!=1&&channels!=2)||
+       (application != OPUS_APPLICATION_VOIP && application != OPUS_APPLICATION_AUDIO
+       && application != OPUS_APPLICATION_RESTRICTED_LOWDELAY))
+   {
+      if (error)
+         *error = OPUS_BAD_ARG;
+      return NULL;
+   }
+   st = (OpusEncoder *)opus_alloc(opus_encoder_get_size(channels));
+   if (st == NULL)
+   {
+      if (error)
+         *error = OPUS_ALLOC_FAIL;
+      return NULL;
+   }
+   ret = opus_encoder_init(st, Fs, channels, application);
+   if (error)
+      *error = ret;
+   if (ret != OPUS_OK)
+   {
+      opus_free(st);
+      st = NULL;
+   }
+   return st;
+}
+
+static opus_int32 user_bitrate_to_bitrate(OpusEncoder *st, int frame_size, int max_data_bytes)
+{
+  if(!frame_size)frame_size=st->Fs/400;
+  if (st->user_bitrate_bps==OPUS_AUTO)
+    return 60*st->Fs/frame_size + st->Fs*st->channels;
+  else if (st->user_bitrate_bps==OPUS_BITRATE_MAX)
+    return max_data_bytes*8*st->Fs/frame_size;
+  else
+    return st->user_bitrate_bps;
+}
+
+#ifndef DISABLE_FLOAT_API
+/* Don't use more than 60 ms for the frame size analysis */
+#define MAX_DYNAMIC_FRAMESIZE 24
+/* Estimates how much the bitrate will be boosted based on the sub-frame energy */
+static float transient_boost(const float *E, const float *E_1, int LM, int maxM)
+{
+   int i;
+   int M;
+   float sumE=0, sumE_1=0;
+   float metric;
+
+   M = IMIN(maxM, (1<<LM)+1);
+   for (i=0;i<M;i++)
+   {
+      sumE += E[i];
+      sumE_1 += E_1[i];
+   }
+   metric = sumE*sumE_1/(M*M);
+   /*if (LM==3)
+      printf("%f\n", metric);*/
+   /*return metric>10 ? 1 : 0;*/
+   /*return MAX16(0,1-exp(-.25*(metric-2.)));*/
+   return MIN16(1,(float)sqrt(MAX16(0,.05f*(metric-2))));
+}
+
+/* Viterbi decoding trying to find the best frame size combination using look-ahead
+
+   State numbering:
+    0: unused
+    1:  2.5 ms
+    2:  5 ms (#1)
+    3:  5 ms (#2)
+    4: 10 ms (#1)
+    5: 10 ms (#2)
+    6: 10 ms (#3)
+    7: 10 ms (#4)
+    8: 20 ms (#1)
+    9: 20 ms (#2)
+   10: 20 ms (#3)
+   11: 20 ms (#4)
+   12: 20 ms (#5)
+   13: 20 ms (#6)
+   14: 20 ms (#7)
+   15: 20 ms (#8)
+*/
+static int transient_viterbi(const float *E, const float *E_1, int N, int frame_cost, int rate)
+{
+   int i;
+   float cost[MAX_DYNAMIC_FRAMESIZE][16];
+   int states[MAX_DYNAMIC_FRAMESIZE][16];
+   float best_cost;
+   int best_state;
+   float factor;
+   /* Take into account that we damp VBR in the 32 kb/s to 64 kb/s range. */
+   if (rate<80)
+      factor=0;
+   else if (rate>160)
+      factor=1;
+   else
+      factor = (rate-80.f)/80.f;
+   /* Makes variable framesize less aggressive at lower bitrates, but I can't
+      find any valid theoretical justification for this (other than it seems
+      to help) */
+   for (i=0;i<16;i++)
+   {
+      /* Impossible state */
+      states[0][i] = -1;
+      cost[0][i] = 1e10;
+   }
+   for (i=0;i<4;i++)
+   {
+      cost[0][1<<i] = (frame_cost + rate*(1<<i))*(1+factor*transient_boost(E, E_1, i, N+1));
+      states[0][1<<i] = i;
+   }
+   for (i=1;i<N;i++)
+   {
+      int j;
+
+      /* Follow continuations */
+      for (j=2;j<16;j++)
+      {
+         cost[i][j] = cost[i-1][j-1];
+         states[i][j] = j-1;
+      }
+
+      /* New frames */
+      for(j=0;j<4;j++)
+      {
+         int k;
+         float min_cost;
+         float curr_cost;
+         states[i][1<<j] = 1;
+         min_cost = cost[i-1][1];
+         for(k=1;k<4;k++)
+         {
+            float tmp = cost[i-1][(1<<(k+1))-1];
+            if (tmp < min_cost)
+            {
+               states[i][1<<j] = (1<<(k+1))-1;
+               min_cost = tmp;
+            }
+         }
+         curr_cost = (frame_cost + rate*(1<<j))*(1+factor*transient_boost(E+i, E_1+i, j, N-i+1));
+         cost[i][1<<j] = min_cost;
+         /* If part of the frame is outside the analysis window, only count part of the cost */
+         if (N-i < (1<<j))
+            cost[i][1<<j] += curr_cost*(float)(N-i)/(1<<j);
+         else
+            cost[i][1<<j] += curr_cost;
+      }
+   }
+
+   best_state=1;
+   best_cost = cost[N-1][1];
+   /* Find best end state (doesn't force a frame to end at N-1) */
+   for (i=2;i<16;i++)
+   {
+      if (cost[N-1][i]<best_cost)
+      {
+         best_cost = cost[N-1][i];
+         best_state = i;
+      }
+   }
+
+   /* Follow transitions back */
+   for (i=N-1;i>=0;i--)
+   {
+      /*printf("%d ", best_state);*/
+      best_state = states[i][best_state];
+   }
+   /*printf("%d\n", best_state);*/
+   return best_state;
+}
+
+static int optimize_framesize(const void *x, int len, int C, opus_int32 Fs,
+                int bitrate, opus_val16 tonality, float *mem, int buffering,
+                downmix_func downmix)
+{
+   int N;
+   int i;
+   float e[MAX_DYNAMIC_FRAMESIZE+4];
+   float e_1[MAX_DYNAMIC_FRAMESIZE+3];
+   opus_val32 memx;
+   int bestLM=0;
+   int subframe;
+   int pos;
+   int offset;
+   VARDECL(opus_val32, sub);
+
+   subframe = Fs/400;
+   ALLOC(sub, subframe, opus_val32);
+   e[0]=mem[0];
+   e_1[0]=1.f/(EPSILON+mem[0]);
+   if (buffering)
+   {
+      /* Consider the CELT delay when not in restricted-lowdelay */
+      /* We assume the buffering is between 2.5 and 5 ms */
+      offset = 2*subframe - buffering;
+      celt_assert(offset>=0 && offset <= subframe);
+      len -= offset;
+      e[1]=mem[1];
+      e_1[1]=1.f/(EPSILON+mem[1]);
+      e[2]=mem[2];
+      e_1[2]=1.f/(EPSILON+mem[2]);
+      pos = 3;
+   } else {
+      pos=1;
+      offset=0;
+   }
+   N=IMIN(len/subframe, MAX_DYNAMIC_FRAMESIZE);
+   /* Just silencing a warning, it's really initialized later */
+   memx = 0;
+   for (i=0;i<N;i++)
+   {
+      float tmp;
+      opus_val32 tmpx;
+      int j;
+      tmp=EPSILON;
+
+      downmix(x, sub, subframe, i*subframe+offset, 0, -2, C);
+      if (i==0)
+         memx = sub[0];
+      for (j=0;j<subframe;j++)
+      {
+         tmpx = sub[j];
+         tmp += (tmpx-memx)*(float)(tmpx-memx);
+         memx = tmpx;
+      }
+      e[i+pos] = tmp;
+      e_1[i+pos] = 1.f/tmp;
+   }
+   /* Hack to get 20 ms working with APPLICATION_AUDIO
+      The real problem is that the corresponding memory needs to use 1.5 ms
+      from this frame and 1 ms from the next frame */
+   e[i+pos] = e[i+pos-1];
+   if (buffering)
+      N=IMIN(MAX_DYNAMIC_FRAMESIZE, N+2);
+   bestLM = transient_viterbi(e, e_1, N, (int)((1.f+.5f*tonality)*(60*C+40)), bitrate/400);
+   mem[0] = e[1<<bestLM];
+   if (buffering)
+   {
+      mem[1] = e[(1<<bestLM)+1];
+      mem[2] = e[(1<<bestLM)+2];
+   }
+   return bestLM;
+}
+
+#endif
+
+#ifndef DISABLE_FLOAT_API
+#ifdef FIXED_POINT
+#define PCM2VAL(x) FLOAT2INT16(x)
+#else
+#define PCM2VAL(x) SCALEIN(x)
+#endif
+void downmix_float(const void *_x, opus_val32 *sub, int subframe, int offset, int c1, int c2, int C)
+{
+   const float *x;
+   opus_val32 scale;
+   int j;
+   x = (const float *)_x;
+   for (j=0;j<subframe;j++)
+      sub[j] = PCM2VAL(x[(j+offset)*C+c1]);
+   if (c2>-1)
+   {
+      for (j=0;j<subframe;j++)
+         sub[j] += PCM2VAL(x[(j+offset)*C+c2]);
+   } else if (c2==-2)
+   {
+      int c;
+      for (c=1;c<C;c++)
+      {
+         for (j=0;j<subframe;j++)
+            sub[j] += PCM2VAL(x[(j+offset)*C+c]);
+      }
+   }
+#ifdef FIXED_POINT
+   scale = (1<<SIG_SHIFT);
+#else
+   scale = 1.f;
+#endif
+   if (C==-2)
+      scale /= C;
+   else
+      scale /= 2;
+   for (j=0;j<subframe;j++)
+      sub[j] *= scale;
+}
+#endif
+
+void downmix_int(const void *_x, opus_val32 *sub, int subframe, int offset, int c1, int c2, int C)
+{
+   const opus_int16 *x;
+   opus_val32 scale;
+   int j;
+   x = (const opus_int16 *)_x;
+   for (j=0;j<subframe;j++)
+      sub[j] = x[(j+offset)*C+c1];
+   if (c2>-1)
+   {
+      for (j=0;j<subframe;j++)
+         sub[j] += x[(j+offset)*C+c2];
+   } else if (c2==-2)
+   {
+      int c;
+      for (c=1;c<C;c++)
+      {
+         for (j=0;j<subframe;j++)
+            sub[j] += x[(j+offset)*C+c];
+      }
+   }
+#ifdef FIXED_POINT
+   scale = (1<<SIG_SHIFT);
+#else
+   scale = 1.f/32768;
+#endif
+   if (C==-2)
+      scale /= C;
+   else
+      scale /= 2;
+   for (j=0;j<subframe;j++)
+      sub[j] *= scale;
+}
+
+opus_int32 frame_size_select(opus_int32 frame_size, int variable_duration, opus_int32 Fs)
+{
+   int new_size;
+   if (frame_size<Fs/400)
+      return -1;
+   if (variable_duration == OPUS_FRAMESIZE_ARG)
+      new_size = frame_size;
+   else if (variable_duration == OPUS_FRAMESIZE_VARIABLE)
+      new_size = Fs/50;
+   else if (variable_duration >= OPUS_FRAMESIZE_2_5_MS && variable_duration <= OPUS_FRAMESIZE_60_MS)
+      new_size = IMIN(3*Fs/50, (Fs/400)<<(variable_duration-OPUS_FRAMESIZE_2_5_MS));
+   else
+      return -1;
+   if (new_size>frame_size)
+      return -1;
+   if (400*new_size!=Fs && 200*new_size!=Fs && 100*new_size!=Fs &&
+            50*new_size!=Fs && 25*new_size!=Fs && 50*new_size!=3*Fs)
+      return -1;
+   return new_size;
+}
+
+opus_int32 compute_frame_size(const void *analysis_pcm, int frame_size,
+      int variable_duration, int C, opus_int32 Fs, int bitrate_bps,
+      int delay_compensation, downmix_func downmix
+#ifndef DISABLE_FLOAT_API
+      , float *subframe_mem
+#endif
+      )
+{
+#ifndef DISABLE_FLOAT_API
+   if (variable_duration == OPUS_FRAMESIZE_VARIABLE && frame_size >= Fs/200)
+   {
+      int LM = 3;
+      LM = optimize_framesize(analysis_pcm, frame_size, C, Fs, bitrate_bps,
+            0, subframe_mem, delay_compensation, downmix);
+      while ((Fs/400<<LM)>frame_size)
+         LM--;
+      frame_size = (Fs/400<<LM);
+   } else
+#else
+   (void)analysis_pcm;
+   (void)C;
+   (void)bitrate_bps;
+   (void)delay_compensation;
+   (void)downmix;
+#endif
+   {
+      frame_size = frame_size_select(frame_size, variable_duration, Fs);
+   }
+   if (frame_size<0)
+      return -1;
+   return frame_size;
+}
+
+opus_val16 compute_stereo_width(const opus_val16 *pcm, int frame_size, opus_int32 Fs, StereoWidthState *mem)
+{
+   opus_val16 corr;
+   opus_val16 ldiff;
+   opus_val16 width;
+   opus_val32 xx, xy, yy;
+   opus_val16 sqrt_xx, sqrt_yy;
+   opus_val16 qrrt_xx, qrrt_yy;
+   int frame_rate;
+   int i;
+   opus_val16 short_alpha;
+
+   frame_rate = Fs/frame_size;
+   short_alpha = Q15ONE - 25*Q15ONE/IMAX(50,frame_rate);
+   xx=xy=yy=0;
+   for (i=0;i<frame_size;i+=4)
+   {
+      opus_val32 pxx=0;
+      opus_val32 pxy=0;
+      opus_val32 pyy=0;
+      opus_val16 x, y;
+      x = pcm[2*i];
+      y = pcm[2*i+1];
+      pxx = SHR32(MULT16_16(x,x),2);
+      pxy = SHR32(MULT16_16(x,y),2);
+      pyy = SHR32(MULT16_16(y,y),2);
+      x = pcm[2*i+2];
+      y = pcm[2*i+3];
+      pxx += SHR32(MULT16_16(x,x),2);
+      pxy += SHR32(MULT16_16(x,y),2);
+      pyy += SHR32(MULT16_16(y,y),2);
+      x = pcm[2*i+4];
+      y = pcm[2*i+5];
+      pxx += SHR32(MULT16_16(x,x),2);
+      pxy += SHR32(MULT16_16(x,y),2);
+      pyy += SHR32(MULT16_16(y,y),2);
+      x = pcm[2*i+6];
+      y = pcm[2*i+7];
+      pxx += SHR32(MULT16_16(x,x),2);
+      pxy += SHR32(MULT16_16(x,y),2);
+      pyy += SHR32(MULT16_16(y,y),2);
+
+      xx += SHR32(pxx, 10);
+      xy += SHR32(pxy, 10);
+      yy += SHR32(pyy, 10);
+   }
+   mem->XX += MULT16_32_Q15(short_alpha, xx-mem->XX);
+   mem->XY += MULT16_32_Q15(short_alpha, xy-mem->XY);
+   mem->YY += MULT16_32_Q15(short_alpha, yy-mem->YY);
+   mem->XX = MAX32(0, mem->XX);
+   mem->XY = MAX32(0, mem->XY);
+   mem->YY = MAX32(0, mem->YY);
+   if (MAX32(mem->XX, mem->YY)>QCONST16(8e-4f, 18))
+   {
+      sqrt_xx = celt_sqrt(mem->XX);
+      sqrt_yy = celt_sqrt(mem->YY);
+      qrrt_xx = celt_sqrt(sqrt_xx);
+      qrrt_yy = celt_sqrt(sqrt_yy);
+      /* Inter-channel correlation */
+      mem->XY = MIN32(mem->XY, sqrt_xx*sqrt_yy);
+      corr = SHR32(frac_div32(mem->XY,EPSILON+MULT16_16(sqrt_xx,sqrt_yy)),16);
+      /* Approximate loudness difference */
+      ldiff = Q15ONE*ABS16(qrrt_xx-qrrt_yy)/(EPSILON+qrrt_xx+qrrt_yy);
+      width = MULT16_16_Q15(celt_sqrt(QCONST32(1.f,30)-MULT16_16(corr,corr)), ldiff);
+      /* Smoothing over one second */
+      mem->smoothed_width += (width-mem->smoothed_width)/frame_rate;
+      /* Peak follower */
+      mem->max_follower = MAX16(mem->max_follower-QCONST16(.02f,15)/frame_rate, mem->smoothed_width);
+   } else {
+      width = 0;
+      corr=Q15ONE;
+      ldiff=0;
+   }
+   /*printf("%f %f %f %f %f ", corr/(float)Q15ONE, ldiff/(float)Q15ONE, width/(float)Q15ONE, mem->smoothed_width/(float)Q15ONE, mem->max_follower/(float)Q15ONE);*/
+   return EXTRACT16(MIN32(Q15ONE,20*mem->max_follower));
+}
+
+opus_int32 opus_encode_native(OpusEncoder *st, const opus_val16 *pcm, int frame_size,
+                unsigned char *data, opus_int32 out_data_bytes, int lsb_depth,
+                const void *analysis_pcm, opus_int32 analysis_size, int c1, int c2,
+                int analysis_channels, downmix_func downmix, int float_api)
+{
+    void *silk_enc;
+    CELTEncoder *celt_enc;
+    int i;
+    int ret=0;
+    opus_int32 nBytes;
+    ec_enc enc;
+    int bytes_target;
+    int prefill=0;
+    int start_band = 0;
+    int redundancy = 0;
+    int redundancy_bytes = 0; /* Number of bytes to use for redundancy frame */
+    int celt_to_silk = 0;
+    VARDECL(opus_val16, pcm_buf);
+    int nb_compr_bytes;
+    int to_celt = 0;
+    opus_uint32 redundant_rng = 0;
+    int cutoff_Hz, hp_freq_smth1;
+    int voice_est; /* Probability of voice in Q7 */
+    opus_int32 equiv_rate;
+    int delay_compensation;
+    int frame_rate;
+    opus_int32 max_rate; /* Max bitrate we're allowed to use */
+    int curr_bandwidth;
+    opus_val16 HB_gain;
+    opus_int32 max_data_bytes; /* Max number of bytes we're allowed to use */
+    int total_buffer;
+    opus_val16 stereo_width;
+    const CELTMode *celt_mode;
+#ifndef DISABLE_FLOAT_API
+    AnalysisInfo analysis_info;
+    int analysis_read_pos_bak=-1;
+    int analysis_read_subframe_bak=-1;
+#endif
+    VARDECL(opus_val16, tmp_prefill);
+
+    ALLOC_STACK;
+
+    max_data_bytes = IMIN(1276, out_data_bytes);
+
+    st->rangeFinal = 0;
+    if ((!st->variable_duration && 400*frame_size != st->Fs && 200*frame_size != st->Fs && 100*frame_size != st->Fs &&
+         50*frame_size != st->Fs &&  25*frame_size != st->Fs &&  50*frame_size != 3*st->Fs)
+         || (400*frame_size < st->Fs)
+         || max_data_bytes<=0
+         )
+    {
+       RESTORE_STACK;
+       return OPUS_BAD_ARG;
+    }
+    silk_enc = (char*)st+st->silk_enc_offset;
+    celt_enc = (CELTEncoder*)((char*)st+st->celt_enc_offset);
+    if (st->application == OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+       delay_compensation = 0;
+    else
+       delay_compensation = st->delay_compensation;
+
+    lsb_depth = IMIN(lsb_depth, st->lsb_depth);
+
+    celt_encoder_ctl(celt_enc, CELT_GET_MODE(&celt_mode));
+#ifndef DISABLE_FLOAT_API
+    analysis_info.valid = 0;
+#ifdef FIXED_POINT
+    if (st->silk_mode.complexity >= 10 && st->Fs==48000)
+#else
+    if (st->silk_mode.complexity >= 7 && st->Fs==48000)
+#endif
+    {
+       analysis_read_pos_bak = st->analysis.read_pos;
+       analysis_read_subframe_bak = st->analysis.read_subframe;
+       run_analysis(&st->analysis, celt_mode, analysis_pcm, analysis_size, frame_size,
+             c1, c2, analysis_channels, st->Fs,
+             lsb_depth, downmix, &analysis_info);
+    }
+#else
+    (void)analysis_pcm;
+    (void)analysis_size;
+#endif
+
+    st->voice_ratio = -1;
+
+#ifndef DISABLE_FLOAT_API
+    st->detected_bandwidth = 0;
+    if (analysis_info.valid)
+    {
+       int analysis_bandwidth;
+       if (st->signal_type == OPUS_AUTO)
+          st->voice_ratio = (int)floor(.5+100*(1-analysis_info.music_prob));
+
+       analysis_bandwidth = analysis_info.bandwidth;
+       if (analysis_bandwidth<=12)
+          st->detected_bandwidth = OPUS_BANDWIDTH_NARROWBAND;
+       else if (analysis_bandwidth<=14)
+          st->detected_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND;
+       else if (analysis_bandwidth<=16)
+          st->detected_bandwidth = OPUS_BANDWIDTH_WIDEBAND;
+       else if (analysis_bandwidth<=18)
+          st->detected_bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND;
+       else
+          st->detected_bandwidth = OPUS_BANDWIDTH_FULLBAND;
+    }
+#endif
+
+    if (st->channels==2 && st->force_channels!=1)
+       stereo_width = compute_stereo_width(pcm, frame_size, st->Fs, &st->width_mem);
+    else
+       stereo_width = 0;
+    total_buffer = delay_compensation;
+    st->bitrate_bps = user_bitrate_to_bitrate(st, frame_size, max_data_bytes);
+
+    frame_rate = st->Fs/frame_size;
+    if (max_data_bytes<3 || st->bitrate_bps < 3*frame_rate*8
+       || (frame_rate<50 && (max_data_bytes*frame_rate<300 || st->bitrate_bps < 2400)))
+    {
+       /*If the space is too low to do something useful, emit 'PLC' frames.*/
+       int tocmode = st->mode;
+       int bw = st->bandwidth == 0 ? OPUS_BANDWIDTH_NARROWBAND : st->bandwidth;
+       if (tocmode==0)
+          tocmode = MODE_SILK_ONLY;
+       if (frame_rate>100)
+          tocmode = MODE_CELT_ONLY;
+       if (frame_rate < 50)
+          tocmode = MODE_SILK_ONLY;
+       if(tocmode==MODE_SILK_ONLY&&bw>OPUS_BANDWIDTH_WIDEBAND)
+          bw=OPUS_BANDWIDTH_WIDEBAND;
+       else if (tocmode==MODE_CELT_ONLY&&bw==OPUS_BANDWIDTH_MEDIUMBAND)
+          bw=OPUS_BANDWIDTH_NARROWBAND;
+       else if (bw<=OPUS_BANDWIDTH_SUPERWIDEBAND)
+          bw=OPUS_BANDWIDTH_SUPERWIDEBAND;
+       data[0] = gen_toc(tocmode, frame_rate, bw, st->stream_channels);
+       RESTORE_STACK;
+       return 1;
+    }
+    if (!st->use_vbr)
+    {
+       int cbrBytes;
+       cbrBytes = IMIN( (st->bitrate_bps + 4*frame_rate)/(8*frame_rate) , max_data_bytes);
+       st->bitrate_bps = cbrBytes * (8*frame_rate);
+       max_data_bytes = cbrBytes;
+    }
+    max_rate = frame_rate*max_data_bytes*8;
+
+    /* Equivalent 20-ms rate for mode/channel/bandwidth decisions */
+    equiv_rate = st->bitrate_bps - (40*st->channels+20)*(st->Fs/frame_size - 50);
+
+    if (st->signal_type == OPUS_SIGNAL_VOICE)
+       voice_est = 127;
+    else if (st->signal_type == OPUS_SIGNAL_MUSIC)
+       voice_est = 0;
+    else if (st->voice_ratio >= 0)
+    {
+       voice_est = st->voice_ratio*327>>8;
+       /* For AUDIO, never be more than 90% confident of having speech */
+       if (st->application == OPUS_APPLICATION_AUDIO)
+          voice_est = IMIN(voice_est, 115);
+    } else if (st->application == OPUS_APPLICATION_VOIP)
+       voice_est = 115;
+    else
+       voice_est = 48;
+
+    if (st->force_channels!=OPUS_AUTO && st->channels == 2)
+    {
+        st->stream_channels = st->force_channels;
+    } else {
+#ifdef FUZZING
+       /* Random mono/stereo decision */
+       if (st->channels == 2 && (rand()&0x1F)==0)
+          st->stream_channels = 3-st->stream_channels;
+#else
+       /* Rate-dependent mono-stereo decision */
+       if (st->channels == 2)
+       {
+          opus_int32 stereo_threshold;
+          stereo_threshold = stereo_music_threshold + ((voice_est*voice_est*(stereo_voice_threshold-stereo_music_threshold))>>14);
+          if (st->stream_channels == 2)
+             stereo_threshold -= 1000;
+          else
+             stereo_threshold += 1000;
+          st->stream_channels = (equiv_rate > stereo_threshold) ? 2 : 1;
+       } else {
+          st->stream_channels = st->channels;
+       }
+#endif
+    }
+    equiv_rate = st->bitrate_bps - (40*st->stream_channels+20)*(st->Fs/frame_size - 50);
+
+    /* Mode selection depending on application and signal type */
+    if (st->application == OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+    {
+       st->mode = MODE_CELT_ONLY;
+    } else if (st->user_forced_mode == OPUS_AUTO)
+    {
+#ifdef FUZZING
+       /* Random mode switching */
+       if ((rand()&0xF)==0)
+       {
+          if ((rand()&0x1)==0)
+             st->mode = MODE_CELT_ONLY;
+          else
+             st->mode = MODE_SILK_ONLY;
+       } else {
+          if (st->prev_mode==MODE_CELT_ONLY)
+             st->mode = MODE_CELT_ONLY;
+          else
+             st->mode = MODE_SILK_ONLY;
+       }
+#else
+       opus_int32 mode_voice, mode_music;
+       opus_int32 threshold;
+
+       /* Interpolate based on stereo width */
+       mode_voice = (opus_int32)(MULT16_32_Q15(Q15ONE-stereo_width,mode_thresholds[0][0])
+             + MULT16_32_Q15(stereo_width,mode_thresholds[1][0]));
+       mode_music = (opus_int32)(MULT16_32_Q15(Q15ONE-stereo_width,mode_thresholds[1][1])
+             + MULT16_32_Q15(stereo_width,mode_thresholds[1][1]));
+       /* Interpolate based on speech/music probability */
+       threshold = mode_music + ((voice_est*voice_est*(mode_voice-mode_music))>>14);
+       /* Bias towards SILK for VoIP because of some useful features */
+       if (st->application == OPUS_APPLICATION_VOIP)
+          threshold += 8000;
+
+       /*printf("%f %d\n", stereo_width/(float)Q15ONE, threshold);*/
+       /* Hysteresis */
+       if (st->prev_mode == MODE_CELT_ONLY)
+           threshold -= 4000;
+       else if (st->prev_mode>0)
+           threshold += 4000;
+
+       st->mode = (equiv_rate >= threshold) ? MODE_CELT_ONLY: MODE_SILK_ONLY;
+
+       /* When FEC is enabled and there's enough packet loss, use SILK */
+       if (st->silk_mode.useInBandFEC && st->silk_mode.packetLossPercentage > (128-voice_est)>>4)
+          st->mode = MODE_SILK_ONLY;
+       /* When encoding voice and DTX is enabled, set the encoder to SILK mode (at least for now) */
+       if (st->silk_mode.useDTX && voice_est > 100)
+          st->mode = MODE_SILK_ONLY;
+#endif
+    } else {
+       st->mode = st->user_forced_mode;
+    }
+
+    /* Override the chosen mode to make sure we meet the requested frame size */
+    if (st->mode != MODE_CELT_ONLY && frame_size < st->Fs/100)
+       st->mode = MODE_CELT_ONLY;
+    if (st->lfe)
+       st->mode = MODE_CELT_ONLY;
+    /* If max_data_bytes represents less than 8 kb/s, switch to CELT-only mode */
+    if (max_data_bytes < (frame_rate > 50 ? 12000 : 8000)*frame_size / (st->Fs * 8))
+       st->mode = MODE_CELT_ONLY;
+
+    if (st->stream_channels == 1 && st->prev_channels ==2 && st->silk_mode.toMono==0
+          && st->mode != MODE_CELT_ONLY && st->prev_mode != MODE_CELT_ONLY)
+    {
+       /* Delay stereo->mono transition by two frames so that SILK can do a smooth downmix */
+       st->silk_mode.toMono = 1;
+       st->stream_channels = 2;
+    } else {
+       st->silk_mode.toMono = 0;
+    }
+
+    if (st->prev_mode > 0 &&
+        ((st->mode != MODE_CELT_ONLY && st->prev_mode == MODE_CELT_ONLY) ||
+    (st->mode == MODE_CELT_ONLY && st->prev_mode != MODE_CELT_ONLY)))
+    {
+        redundancy = 1;
+        celt_to_silk = (st->mode != MODE_CELT_ONLY);
+        if (!celt_to_silk)
+        {
+            /* Switch to SILK/hybrid if frame size is 10 ms or more*/
+            if (frame_size >= st->Fs/100)
+            {
+                st->mode = st->prev_mode;
+                to_celt = 1;
+            } else {
+                redundancy=0;
+            }
+        }
+    }
+    /* For the first frame at a new SILK bandwidth */
+    if (st->silk_bw_switch)
+    {
+       redundancy = 1;
+       celt_to_silk = 1;
+       st->silk_bw_switch = 0;
+       prefill=1;
+    }
+
+    if (redundancy)
+    {
+       /* Fair share of the max size allowed */
+       redundancy_bytes = IMIN(257, max_data_bytes*(opus_int32)(st->Fs/200)/(frame_size+st->Fs/200));
+       /* For VBR, target the actual bitrate (subject to the limit above) */
+       if (st->use_vbr)
+          redundancy_bytes = IMIN(redundancy_bytes, st->bitrate_bps/1600);
+    }
+
+    if (st->mode != MODE_CELT_ONLY && st->prev_mode == MODE_CELT_ONLY)
+    {
+        silk_EncControlStruct dummy;
+        silk_InitEncoder( silk_enc, st->arch, &dummy);
+        prefill=1;
+    }
+
+    /* Automatic (rate-dependent) bandwidth selection */
+    if (st->mode == MODE_CELT_ONLY || st->first || st->silk_mode.allowBandwidthSwitch)
+    {
+        const opus_int32 *voice_bandwidth_thresholds, *music_bandwidth_thresholds;
+        opus_int32 bandwidth_thresholds[8];
+        int bandwidth = OPUS_BANDWIDTH_FULLBAND;
+        opus_int32 equiv_rate2;
+
+        equiv_rate2 = equiv_rate;
+        if (st->mode != MODE_CELT_ONLY)
+        {
+           /* Adjust the threshold +/- 10% depending on complexity */
+           equiv_rate2 = equiv_rate2 * (45+st->silk_mode.complexity)/50;
+           /* CBR is less efficient by ~1 kb/s */
+           if (!st->use_vbr)
+              equiv_rate2 -= 1000;
+        }
+        if (st->channels==2 && st->force_channels!=1)
+        {
+           voice_bandwidth_thresholds = stereo_voice_bandwidth_thresholds;
+           music_bandwidth_thresholds = stereo_music_bandwidth_thresholds;
+        } else {
+           voice_bandwidth_thresholds = mono_voice_bandwidth_thresholds;
+           music_bandwidth_thresholds = mono_music_bandwidth_thresholds;
+        }
+        /* Interpolate bandwidth thresholds depending on voice estimation */
+        for (i=0;i<8;i++)
+        {
+           bandwidth_thresholds[i] = music_bandwidth_thresholds[i]
+                    + ((voice_est*voice_est*(voice_bandwidth_thresholds[i]-music_bandwidth_thresholds[i]))>>14);
+        }
+        do {
+            int threshold, hysteresis;
+            threshold = bandwidth_thresholds[2*(bandwidth-OPUS_BANDWIDTH_MEDIUMBAND)];
+            hysteresis = bandwidth_thresholds[2*(bandwidth-OPUS_BANDWIDTH_MEDIUMBAND)+1];
+            if (!st->first)
+            {
+                if (st->bandwidth >= bandwidth)
+                    threshold -= hysteresis;
+                else
+                    threshold += hysteresis;
+            }
+            if (equiv_rate2 >= threshold)
+                break;
+        } while (--bandwidth>OPUS_BANDWIDTH_NARROWBAND);
+        st->bandwidth = bandwidth;
+        /* Prevents any transition to SWB/FB until the SILK layer has fully
+           switched to WB mode and turned the variable LP filter off */
+        if (!st->first && st->mode != MODE_CELT_ONLY && !st->silk_mode.inWBmodeWithoutVariableLP && st->bandwidth > OPUS_BANDWIDTH_WIDEBAND)
+            st->bandwidth = OPUS_BANDWIDTH_WIDEBAND;
+    }
+
+    if (st->bandwidth>st->max_bandwidth)
+       st->bandwidth = st->max_bandwidth;
+
+    if (st->user_bandwidth != OPUS_AUTO)
+        st->bandwidth = st->user_bandwidth;
+
+    /* This prevents us from using hybrid at unsafe CBR/max rates */
+    if (st->mode != MODE_CELT_ONLY && max_rate < 15000)
+    {
+       st->bandwidth = IMIN(st->bandwidth, OPUS_BANDWIDTH_WIDEBAND);
+    }
+
+    /* Prevents Opus from wasting bits on frequencies that are above
+       the Nyquist rate of the input signal */
+    if (st->Fs <= 24000 && st->bandwidth > OPUS_BANDWIDTH_SUPERWIDEBAND)
+        st->bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND;
+    if (st->Fs <= 16000 && st->bandwidth > OPUS_BANDWIDTH_WIDEBAND)
+        st->bandwidth = OPUS_BANDWIDTH_WIDEBAND;
+    if (st->Fs <= 12000 && st->bandwidth > OPUS_BANDWIDTH_MEDIUMBAND)
+        st->bandwidth = OPUS_BANDWIDTH_MEDIUMBAND;
+    if (st->Fs <= 8000 && st->bandwidth > OPUS_BANDWIDTH_NARROWBAND)
+        st->bandwidth = OPUS_BANDWIDTH_NARROWBAND;
+#ifndef DISABLE_FLOAT_API
+    /* Use detected bandwidth to reduce the encoded bandwidth. */
+    if (st->detected_bandwidth && st->user_bandwidth == OPUS_AUTO)
+    {
+       int min_detected_bandwidth;
+       /* Makes bandwidth detection more conservative just in case the detector
+          gets it wrong when we could have coded a high bandwidth transparently.
+          When operating in SILK/hybrid mode, we don't go below wideband to avoid
+          more complicated switches that require redundancy. */
+       if (equiv_rate <= 18000*st->stream_channels && st->mode == MODE_CELT_ONLY)
+          min_detected_bandwidth = OPUS_BANDWIDTH_NARROWBAND;
+       else if (equiv_rate <= 24000*st->stream_channels && st->mode == MODE_CELT_ONLY)
+          min_detected_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND;
+       else if (equiv_rate <= 30000*st->stream_channels)
+          min_detected_bandwidth = OPUS_BANDWIDTH_WIDEBAND;
+       else if (equiv_rate <= 44000*st->stream_channels)
+          min_detected_bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND;
+       else
+          min_detected_bandwidth = OPUS_BANDWIDTH_FULLBAND;
+
+       st->detected_bandwidth = IMAX(st->detected_bandwidth, min_detected_bandwidth);
+       st->bandwidth = IMIN(st->bandwidth, st->detected_bandwidth);
+    }
+#endif
+    celt_encoder_ctl(celt_enc, OPUS_SET_LSB_DEPTH(lsb_depth));
+
+    /* CELT mode doesn't support mediumband, use wideband instead */
+    if (st->mode == MODE_CELT_ONLY && st->bandwidth == OPUS_BANDWIDTH_MEDIUMBAND)
+        st->bandwidth = OPUS_BANDWIDTH_WIDEBAND;
+    if (st->lfe)
+       st->bandwidth = OPUS_BANDWIDTH_NARROWBAND;
+
+    /* Can't support higher than wideband for >20 ms frames */
+    if (frame_size > st->Fs/50 && (st->mode == MODE_CELT_ONLY || st->bandwidth > OPUS_BANDWIDTH_WIDEBAND))
+    {
+       VARDECL(unsigned char, tmp_data);
+       int nb_frames;
+       int bak_mode, bak_bandwidth, bak_channels, bak_to_mono;
+       VARDECL(OpusRepacketizer, rp);
+       opus_int32 bytes_per_frame;
+       opus_int32 repacketize_len;
+
+#ifndef DISABLE_FLOAT_API
+       if (analysis_read_pos_bak!= -1)
+       {
+          st->analysis.read_pos = analysis_read_pos_bak;
+          st->analysis.read_subframe = analysis_read_subframe_bak;
+       }
+#endif
+
+       nb_frames = frame_size > st->Fs/25 ? 3 : 2;
+       bytes_per_frame = IMIN(1276,(out_data_bytes-3)/nb_frames);
+
+       ALLOC(tmp_data, nb_frames*bytes_per_frame, unsigned char);
+
+       ALLOC(rp, 1, OpusRepacketizer);
+       opus_repacketizer_init(rp);
+
+       bak_mode = st->user_forced_mode;
+       bak_bandwidth = st->user_bandwidth;
+       bak_channels = st->force_channels;
+
+       st->user_forced_mode = st->mode;
+       st->user_bandwidth = st->bandwidth;
+       st->force_channels = st->stream_channels;
+       bak_to_mono = st->silk_mode.toMono;
+
+       if (bak_to_mono)
+          st->force_channels = 1;
+       else
+          st->prev_channels = st->stream_channels;
+       for (i=0;i<nb_frames;i++)
+       {
+          int tmp_len;
+          st->silk_mode.toMono = 0;
+          /* When switching from SILK/Hybrid to CELT, only ask for a switch at the last frame */
+          if (to_celt && i==nb_frames-1)
+             st->user_forced_mode = MODE_CELT_ONLY;
+          tmp_len = opus_encode_native(st, pcm+i*(st->channels*st->Fs/50), st->Fs/50,
+                tmp_data+i*bytes_per_frame, bytes_per_frame, lsb_depth,
+                NULL, 0, c1, c2, analysis_channels, downmix, float_api);
+          if (tmp_len<0)
+          {
+             RESTORE_STACK;
+             return OPUS_INTERNAL_ERROR;
+          }
+          ret = opus_repacketizer_cat(rp, tmp_data+i*bytes_per_frame, tmp_len);
+          if (ret<0)
+          {
+             RESTORE_STACK;
+             return OPUS_INTERNAL_ERROR;
+          }
+       }
+       if (st->use_vbr)
+          repacketize_len = out_data_bytes;
+       else
+          repacketize_len = IMIN(3*st->bitrate_bps/(3*8*50/nb_frames), out_data_bytes);
+       ret = opus_repacketizer_out_range_impl(rp, 0, nb_frames, data, repacketize_len, 0, !st->use_vbr);
+       if (ret<0)
+       {
+          RESTORE_STACK;
+          return OPUS_INTERNAL_ERROR;
+       }
+       st->user_forced_mode = bak_mode;
+       st->user_bandwidth = bak_bandwidth;
+       st->force_channels = bak_channels;
+       st->silk_mode.toMono = bak_to_mono;
+       RESTORE_STACK;
+       return ret;
+    }
+    curr_bandwidth = st->bandwidth;
+
+    /* Chooses the appropriate mode for speech
+       *NEVER* switch to/from CELT-only mode here as this will invalidate some assumptions */
+    if (st->mode == MODE_SILK_ONLY && curr_bandwidth > OPUS_BANDWIDTH_WIDEBAND)
+        st->mode = MODE_HYBRID;
+    if (st->mode == MODE_HYBRID && curr_bandwidth <= OPUS_BANDWIDTH_WIDEBAND)
+        st->mode = MODE_SILK_ONLY;
+
+    /* printf("%d %d %d %d\n", st->bitrate_bps, st->stream_channels, st->mode, curr_bandwidth); */
+    bytes_target = IMIN(max_data_bytes-redundancy_bytes, st->bitrate_bps * frame_size / (st->Fs * 8)) - 1;
+
+    data += 1;
+
+    ec_enc_init(&enc, data, max_data_bytes-1);
+
+    ALLOC(pcm_buf, (total_buffer+frame_size)*st->channels, opus_val16);
+    OPUS_COPY(pcm_buf, &st->delay_buffer[(st->encoder_buffer-total_buffer)*st->channels], total_buffer*st->channels);
+
+    if (st->mode == MODE_CELT_ONLY)
+       hp_freq_smth1 = silk_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 = silk_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( silk_RSHIFT( st->variable_HP_smth2_Q15, 8 ) );
+
+    if (st->application == OPUS_APPLICATION_VOIP)
+    {
+       hp_cutoff(pcm, cutoff_Hz, &pcm_buf[total_buffer*st->channels], st->hp_mem, frame_size, st->channels, st->Fs);
+    } else {
+       dc_reject(pcm, 3, &pcm_buf[total_buffer*st->channels], st->hp_mem, frame_size, st->channels, st->Fs);
+    }
+#ifndef FIXED_POINT
+    if (float_api)
+    {
+       opus_val32 sum;
+       sum = celt_inner_prod(&pcm_buf[total_buffer*st->channels], &pcm_buf[total_buffer*st->channels], frame_size*st->channels, st->arch);
+       /* This should filter out both NaNs and ridiculous signals that could
+          cause NaNs further down. */
+       if (!(sum < 1e9f) || celt_isnan(sum))
+       {
+          OPUS_CLEAR(&pcm_buf[total_buffer*st->channels], frame_size*st->channels);
+          st->hp_mem[0] = st->hp_mem[1] = st->hp_mem[2] = st->hp_mem[3] = 0;
+       }
+    }
+#endif
+
+
+    /* SILK processing */
+    HB_gain = Q15ONE;
+    if (st->mode != MODE_CELT_ONLY)
+    {
+        opus_int32 total_bitRate, celt_rate;
+#ifdef FIXED_POINT
+       const opus_int16 *pcm_silk;
+#else
+       VARDECL(opus_int16, pcm_silk);
+       ALLOC(pcm_silk, st->channels*frame_size, opus_int16);
+#endif
+
+        /* Distribute bits between SILK and CELT */
+        total_bitRate = 8 * bytes_target * frame_rate;
+        if( st->mode == MODE_HYBRID ) {
+            int HB_gain_ref;
+            /* Base rate for SILK */
+            st->silk_mode.bitRate = st->stream_channels * ( 5000 + 1000 * ( st->Fs == 100 * frame_size ) );
+            if( curr_bandwidth == OPUS_BANDWIDTH_SUPERWIDEBAND ) {
+                /* SILK gets 2/3 of the remaining bits */
+                st->silk_mode.bitRate += ( total_bitRate - st->silk_mode.bitRate ) * 2 / 3;
+            } else { /* FULLBAND */
+                /* SILK gets 3/5 of the remaining bits */
+                st->silk_mode.bitRate += ( total_bitRate - st->silk_mode.bitRate ) * 3 / 5;
+            }
+            /* Don't let SILK use more than 80% */
+            if( st->silk_mode.bitRate > total_bitRate * 4/5 ) {
+                st->silk_mode.bitRate = total_bitRate * 4/5;
+            }
+            if (!st->energy_masking)
+            {
+               /* Increasingly attenuate high band when it gets allocated fewer bits */
+               celt_rate = total_bitRate - st->silk_mode.bitRate;
+               HB_gain_ref = (curr_bandwidth == OPUS_BANDWIDTH_SUPERWIDEBAND) ? 3000 : 3600;
+               HB_gain = SHL32((opus_val32)celt_rate, 9) / SHR32((opus_val32)celt_rate + st->stream_channels * HB_gain_ref, 6);
+               HB_gain = HB_gain < Q15ONE*6/7 ? HB_gain + Q15ONE/7 : Q15ONE;
+            }
+        } else {
+            /* SILK gets all bits */
+            st->silk_mode.bitRate = total_bitRate;
+        }
+
+        /* Surround masking for SILK */
+        if (st->energy_masking && st->use_vbr && !st->lfe)
+        {
+           opus_val32 mask_sum=0;
+           opus_val16 masking_depth;
+           opus_int32 rate_offset;
+           int c;
+           int end = 17;
+           opus_int16 srate = 16000;
+           if (st->bandwidth == OPUS_BANDWIDTH_NARROWBAND)
+           {
+              end = 13;
+              srate = 8000;
+           } else if (st->bandwidth == OPUS_BANDWIDTH_MEDIUMBAND)
+           {
+              end = 15;
+              srate = 12000;
+           }
+           for (c=0;c<st->channels;c++)
+           {
+              for(i=0;i<end;i++)
+              {
+                 opus_val16 mask;
+                 mask = MAX16(MIN16(st->energy_masking[21*c+i],
+                        QCONST16(.5f, DB_SHIFT)), -QCONST16(2.0f, DB_SHIFT));
+                 if (mask > 0)
+                    mask = HALF16(mask);
+                 mask_sum += mask;
+              }
+           }
+           /* Conservative rate reduction, we cut the masking in half */
+           masking_depth = mask_sum / end*st->channels;
+           masking_depth += QCONST16(.2f, DB_SHIFT);
+           rate_offset = (opus_int32)PSHR32(MULT16_16(srate, masking_depth), DB_SHIFT);
+           rate_offset = MAX32(rate_offset, -2*st->silk_mode.bitRate/3);
+           /* Split the rate change between the SILK and CELT part for hybrid. */
+           if (st->bandwidth==OPUS_BANDWIDTH_SUPERWIDEBAND || st->bandwidth==OPUS_BANDWIDTH_FULLBAND)
+              st->silk_mode.bitRate += 3*rate_offset/5;
+           else
+              st->silk_mode.bitRate += rate_offset;
+           bytes_target += rate_offset * frame_size / (8 * st->Fs);
+        }
+
+        st->silk_mode.payloadSize_ms = 1000 * frame_size / st->Fs;
+        st->silk_mode.nChannelsAPI = st->channels;
+        st->silk_mode.nChannelsInternal = st->stream_channels;
+        if (curr_bandwidth == OPUS_BANDWIDTH_NARROWBAND) {
+            st->silk_mode.desiredInternalSampleRate = 8000;
+        } else if (curr_bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) {
+            st->silk_mode.desiredInternalSampleRate = 12000;
+        } else {
+            silk_assert( st->mode == MODE_HYBRID || curr_bandwidth == OPUS_BANDWIDTH_WIDEBAND );
+            st->silk_mode.desiredInternalSampleRate = 16000;
+        }
+        if( st->mode == MODE_HYBRID ) {
+            /* Don't allow bandwidth reduction at lowest bitrates in hybrid mode */
+            st->silk_mode.minInternalSampleRate = 16000;
+        } else {
+            st->silk_mode.minInternalSampleRate = 8000;
+        }
+
+        if (st->mode == MODE_SILK_ONLY)
+        {
+           opus_int32 effective_max_rate = max_rate;
+           st->silk_mode.maxInternalSampleRate = 16000;
+           if (frame_rate > 50)
+              effective_max_rate = effective_max_rate*2/3;
+           if (effective_max_rate < 13000)
+           {
+              st->silk_mode.maxInternalSampleRate = 12000;
+              st->silk_mode.desiredInternalSampleRate = IMIN(12000, st->silk_mode.desiredInternalSampleRate);
+           }
+           if (effective_max_rate < 9600)
+           {
+              st->silk_mode.maxInternalSampleRate = 8000;
+              st->silk_mode.desiredInternalSampleRate = IMIN(8000, st->silk_mode.desiredInternalSampleRate);
+           }
+        } else {
+           st->silk_mode.maxInternalSampleRate = 16000;
+        }
+
+        st->silk_mode.useCBR = !st->use_vbr;
+
+        /* Call SILK encoder for the low band */
+        nBytes = IMIN(1275, max_data_bytes-1-redundancy_bytes);
+
+        st->silk_mode.maxBits = nBytes*8;
+        /* Only allow up to 90% of the bits for hybrid mode*/
+        if (st->mode == MODE_HYBRID)
+           st->silk_mode.maxBits = (opus_int32)st->silk_mode.maxBits*9/10;
+        if (st->silk_mode.useCBR)
+        {
+           st->silk_mode.maxBits = (st->silk_mode.bitRate * frame_size / (st->Fs * 8))*8;
+           /* Reduce the initial target to make it easier to reach the CBR rate */
+           st->silk_mode.bitRate = IMAX(1, st->silk_mode.bitRate-2000);
+        }
+
+        if (prefill)
+        {
+            opus_int32 zero=0;
+            int prefill_offset;
+            /* Use a smooth onset for the SILK prefill to avoid the encoder trying to encode
+               a discontinuity. The exact location is what we need to avoid leaving any "gap"
+               in the audio when mixing with the redundant CELT frame. Here we can afford to
+               overwrite st->delay_buffer because the only thing that uses it before it gets
+               rewritten is tmp_prefill[] and even then only the part after the ramp really
+               gets used (rather than sent to the encoder and discarded) */
+            prefill_offset = st->channels*(st->encoder_buffer-st->delay_compensation-st->Fs/400);
+            gain_fade(st->delay_buffer+prefill_offset, st->delay_buffer+prefill_offset,
+                  0, Q15ONE, celt_mode->overlap, st->Fs/400, st->channels, celt_mode->window, st->Fs);
+            OPUS_CLEAR(st->delay_buffer, prefill_offset);
+#ifdef FIXED_POINT
+            pcm_silk = st->delay_buffer;
+#else
+            for (i=0;i<st->encoder_buffer*st->channels;i++)
+                pcm_silk[i] = FLOAT2INT16(st->delay_buffer[i]);
+#endif
+            silk_Encode( silk_enc, &st->silk_mode, pcm_silk, st->encoder_buffer, NULL, &zero, 1 );
+        }
+
+#ifdef FIXED_POINT
+        pcm_silk = pcm_buf+total_buffer*st->channels;
+#else
+        for (i=0;i<frame_size*st->channels;i++)
+            pcm_silk[i] = FLOAT2INT16(pcm_buf[total_buffer*st->channels + i]);
+#endif
+        ret = silk_Encode( silk_enc, &st->silk_mode, pcm_silk, frame_size, &enc, &nBytes, 0 );
+        if( ret ) {
+            /*fprintf (stderr, "SILK encode error: %d\n", ret);*/
+            /* Handle error */
+           RESTORE_STACK;
+           return OPUS_INTERNAL_ERROR;
+        }
+        if (nBytes==0)
+        {
+           st->rangeFinal = 0;
+           data[-1] = gen_toc(st->mode, st->Fs/frame_size, curr_bandwidth, st->stream_channels);
+           RESTORE_STACK;
+           return 1;
+        }
+        /* Extract SILK internal bandwidth for signaling in first byte */
+        if( st->mode == MODE_SILK_ONLY ) {
+            if( st->silk_mode.internalSampleRate == 8000 ) {
+               curr_bandwidth = OPUS_BANDWIDTH_NARROWBAND;
+            } else if( st->silk_mode.internalSampleRate == 12000 ) {
+               curr_bandwidth = OPUS_BANDWIDTH_MEDIUMBAND;
+            } else if( st->silk_mode.internalSampleRate == 16000 ) {
+               curr_bandwidth = OPUS_BANDWIDTH_WIDEBAND;
+            }
+        } else {
+            silk_assert( st->silk_mode.internalSampleRate == 16000 );
+        }
+
+        st->silk_mode.opusCanSwitch = st->silk_mode.switchReady;
+        /* FIXME: How do we allocate the redundancy for CBR? */
+        if (st->silk_mode.opusCanSwitch)
+        {
+           redundancy = 1;
+           celt_to_silk = 0;
+           st->silk_bw_switch = 1;
+        }
+    }
+
+    /* CELT processing */
+    {
+        int endband=21;
+
+        switch(curr_bandwidth)
+        {
+            case OPUS_BANDWIDTH_NARROWBAND:
+                endband = 13;
+                break;
+            case OPUS_BANDWIDTH_MEDIUMBAND:
+            case OPUS_BANDWIDTH_WIDEBAND:
+                endband = 17;
+                break;
+            case OPUS_BANDWIDTH_SUPERWIDEBAND:
+                endband = 19;
+                break;
+            case OPUS_BANDWIDTH_FULLBAND:
+                endband = 21;
+                break;
+        }
+        celt_encoder_ctl(celt_enc, CELT_SET_END_BAND(endband));
+        celt_encoder_ctl(celt_enc, CELT_SET_CHANNELS(st->stream_channels));
+    }
+    celt_encoder_ctl(celt_enc, OPUS_SET_BITRATE(OPUS_BITRATE_MAX));
+    if (st->mode != MODE_SILK_ONLY)
+    {
+        opus_val32 celt_pred=2;
+        celt_encoder_ctl(celt_enc, OPUS_SET_VBR(0));
+        /* We may still decide to disable prediction later */
+        if (st->silk_mode.reducedDependency)
+           celt_pred = 0;
+        celt_encoder_ctl(celt_enc, CELT_SET_PREDICTION(celt_pred));
+
+        if (st->mode == MODE_HYBRID)
+        {
+            int len;
+
+            len = (ec_tell(&enc)+7)>>3;
+            if (redundancy)
+               len += st->mode == MODE_HYBRID ? 3 : 1;
+            if( st->use_vbr ) {
+                nb_compr_bytes = len + bytes_target - (st->silk_mode.bitRate * frame_size) / (8 * st->Fs);
+            } else {
+                /* check if SILK used up too much */
+                nb_compr_bytes = len > bytes_target ? len : bytes_target;
+            }
+        } else {
+            if (st->use_vbr)
+            {
+                opus_int32 bonus=0;
+#ifndef DISABLE_FLOAT_API
+                if (st->variable_duration==OPUS_FRAMESIZE_VARIABLE && frame_size != st->Fs/50)
+                {
+                   bonus = (60*st->stream_channels+40)*(st->Fs/frame_size-50);
+                   if (analysis_info.valid)
+                      bonus = (opus_int32)(bonus*(1.f+.5f*analysis_info.tonality));
+                }
+#endif
+                celt_encoder_ctl(celt_enc, OPUS_SET_VBR(1));
+                celt_encoder_ctl(celt_enc, OPUS_SET_VBR_CONSTRAINT(st->vbr_constraint));
+                celt_encoder_ctl(celt_enc, OPUS_SET_BITRATE(st->bitrate_bps+bonus));
+                nb_compr_bytes = max_data_bytes-1-redundancy_bytes;
+            } else {
+                nb_compr_bytes = bytes_target;
+            }
+        }
+
+    } else {
+        nb_compr_bytes = 0;
+    }
+
+    ALLOC(tmp_prefill, st->channels*st->Fs/400, opus_val16);
+    if (st->mode != MODE_SILK_ONLY && st->mode != st->prev_mode && st->prev_mode > 0)
+    {
+       OPUS_COPY(tmp_prefill, &st->delay_buffer[(st->encoder_buffer-total_buffer-st->Fs/400)*st->channels], st->channels*st->Fs/400);
+    }
+
+    if (st->channels*(st->encoder_buffer-(frame_size+total_buffer)) > 0)
+    {
+       OPUS_MOVE(st->delay_buffer, &st->delay_buffer[st->channels*frame_size], st->channels*(st->encoder_buffer-frame_size-total_buffer));
+       OPUS_COPY(&st->delay_buffer[st->channels*(st->encoder_buffer-frame_size-total_buffer)],
+             &pcm_buf[0],
+             (frame_size+total_buffer)*st->channels);
+    } else {
+       OPUS_COPY(st->delay_buffer, &pcm_buf[(frame_size+total_buffer-st->encoder_buffer)*st->channels], st->encoder_buffer*st->channels);
+    }
+    /* gain_fade() and stereo_fade() need to be after the buffer copying
+       because we don't want any of this to affect the SILK part */
+    if( st->prev_HB_gain < Q15ONE || HB_gain < Q15ONE ) {
+       gain_fade(pcm_buf, pcm_buf,
+             st->prev_HB_gain, HB_gain, celt_mode->overlap, frame_size, st->channels, celt_mode->window, st->Fs);
+    }
+    st->prev_HB_gain = HB_gain;
+    if (st->mode != MODE_HYBRID || st->stream_channels==1)
+       st->silk_mode.stereoWidth_Q14 = IMIN((1<<14),2*IMAX(0,equiv_rate-30000));
+    if( !st->energy_masking && st->channels == 2 ) {
+        /* Apply stereo width reduction (at low bitrates) */
+        if( st->hybrid_stereo_width_Q14 < (1 << 14) || st->silk_mode.stereoWidth_Q14 < (1 << 14) ) {
+            opus_val16 g1, g2;
+            g1 = st->hybrid_stereo_width_Q14;
+            g2 = (opus_val16)(st->silk_mode.stereoWidth_Q14);
+#ifdef FIXED_POINT
+            g1 = g1==16384 ? Q15ONE : SHL16(g1,1);
+            g2 = g2==16384 ? Q15ONE : SHL16(g2,1);
+#else
+            g1 *= (1.f/16384);
+            g2 *= (1.f/16384);
+#endif
+            stereo_fade(pcm_buf, pcm_buf, g1, g2, celt_mode->overlap,
+                  frame_size, st->channels, celt_mode->window, st->Fs);
+            st->hybrid_stereo_width_Q14 = st->silk_mode.stereoWidth_Q14;
+        }
+    }
+
+    if ( st->mode != MODE_CELT_ONLY && ec_tell(&enc)+17+20*(st->mode == MODE_HYBRID) <= 8*(max_data_bytes-1))
+    {
+        /* For SILK mode, the redundancy is inferred from the length */
+        if (st->mode == MODE_HYBRID && (redundancy || ec_tell(&enc)+37 <= 8*nb_compr_bytes))
+           ec_enc_bit_logp(&enc, redundancy, 12);
+        if (redundancy)
+        {
+            int max_redundancy;
+            ec_enc_bit_logp(&enc, celt_to_silk, 1);
+            if (st->mode == MODE_HYBRID)
+               max_redundancy = (max_data_bytes-1)-nb_compr_bytes;
+            else
+               max_redundancy = (max_data_bytes-1)-((ec_tell(&enc)+7)>>3);
+            /* Target the same bit-rate for redundancy as for the rest,
+               up to a max of 257 bytes */
+            redundancy_bytes = IMIN(max_redundancy, st->bitrate_bps/1600);
+            redundancy_bytes = IMIN(257, IMAX(2, redundancy_bytes));
+            if (st->mode == MODE_HYBRID)
+                ec_enc_uint(&enc, redundancy_bytes-2, 256);
+        }
+    } else {
+        redundancy = 0;
+    }
+
+    if (!redundancy)
+    {
+       st->silk_bw_switch = 0;
+       redundancy_bytes = 0;
+    }
+    if (st->mode != MODE_CELT_ONLY)start_band=17;
+
+    if (st->mode == MODE_SILK_ONLY)
+    {
+        ret = (ec_tell(&enc)+7)>>3;
+        ec_enc_done(&enc);
+        nb_compr_bytes = ret;
+    } else {
+       nb_compr_bytes = IMIN((max_data_bytes-1)-redundancy_bytes, nb_compr_bytes);
+       ec_enc_shrink(&enc, nb_compr_bytes);
+    }
+
+#ifndef DISABLE_FLOAT_API
+    if (redundancy || st->mode != MODE_SILK_ONLY)
+       celt_encoder_ctl(celt_enc, CELT_SET_ANALYSIS(&analysis_info));
+#endif
+
+    /* 5 ms redundant frame for CELT->SILK */
+    if (redundancy && celt_to_silk)
+    {
+        int err;
+        celt_encoder_ctl(celt_enc, CELT_SET_START_BAND(0));
+        celt_encoder_ctl(celt_enc, OPUS_SET_VBR(0));
+        err = celt_encode_with_ec(celt_enc, pcm_buf, st->Fs/200, data+nb_compr_bytes, redundancy_bytes, NULL);
+        if (err < 0)
+        {
+           RESTORE_STACK;
+           return OPUS_INTERNAL_ERROR;
+        }
+        celt_encoder_ctl(celt_enc, OPUS_GET_FINAL_RANGE(&redundant_rng));
+        celt_encoder_ctl(celt_enc, OPUS_RESET_STATE);
+    }
+
+    celt_encoder_ctl(celt_enc, CELT_SET_START_BAND(start_band));
+
+    if (st->mode != MODE_SILK_ONLY)
+    {
+        if (st->mode != st->prev_mode && st->prev_mode > 0)
+        {
+           unsigned char dummy[2];
+           celt_encoder_ctl(celt_enc, OPUS_RESET_STATE);
+
+           /* Prefilling */
+           celt_encode_with_ec(celt_enc, tmp_prefill, st->Fs/400, dummy, 2, NULL);
+           celt_encoder_ctl(celt_enc, CELT_SET_PREDICTION(0));
+        }
+        /* If false, we already busted the budget and we'll end up with a "PLC packet" */
+        if (ec_tell(&enc) <= 8*nb_compr_bytes)
+        {
+           ret = celt_encode_with_ec(celt_enc, pcm_buf, frame_size, NULL, nb_compr_bytes, &enc);
+           if (ret < 0)
+           {
+              RESTORE_STACK;
+              return OPUS_INTERNAL_ERROR;
+           }
+        }
+    }
+
+    /* 5 ms redundant frame for SILK->CELT */
+    if (redundancy && !celt_to_silk)
+    {
+        int err;
+        unsigned char dummy[2];
+        int N2, N4;
+        N2 = st->Fs/200;
+        N4 = st->Fs/400;
+
+        celt_encoder_ctl(celt_enc, OPUS_RESET_STATE);
+        celt_encoder_ctl(celt_enc, CELT_SET_START_BAND(0));
+        celt_encoder_ctl(celt_enc, CELT_SET_PREDICTION(0));
+
+        /* NOTE: We could speed this up slightly (at the expense of code size) by just adding a function that prefills the buffer */
+        celt_encode_with_ec(celt_enc, pcm_buf+st->channels*(frame_size-N2-N4), N4, dummy, 2, NULL);
+
+        err = celt_encode_with_ec(celt_enc, pcm_buf+st->channels*(frame_size-N2), N2, data+nb_compr_bytes, redundancy_bytes, NULL);
+        if (err < 0)
+        {
+           RESTORE_STACK;
+           return OPUS_INTERNAL_ERROR;
+        }
+        celt_encoder_ctl(celt_enc, OPUS_GET_FINAL_RANGE(&redundant_rng));
+    }
+
+
+
+    /* Signalling the mode in the first byte */
+    data--;
+    data[0] = gen_toc(st->mode, st->Fs/frame_size, curr_bandwidth, st->stream_channels);
+
+    st->rangeFinal = enc.rng ^ redundant_rng;
+
+    if (to_celt)
+        st->prev_mode = MODE_CELT_ONLY;
+    else
+        st->prev_mode = st->mode;
+    st->prev_channels = st->stream_channels;
+    st->prev_framesize = frame_size;
+
+    st->first = 0;
+
+    /* In the unlikely case that the SILK encoder busted its target, tell
+       the decoder to call the PLC */
+    if (ec_tell(&enc) > (max_data_bytes-1)*8)
+    {
+       if (max_data_bytes < 2)
+       {
+          RESTORE_STACK;
+          return OPUS_BUFFER_TOO_SMALL;
+       }
+       data[1] = 0;
+       ret = 1;
+       st->rangeFinal = 0;
+    } else if (st->mode==MODE_SILK_ONLY&&!redundancy)
+    {
+       /*When in LPC only mode it's perfectly
+         reasonable to strip off trailing zero bytes as
+         the required range decoder behavior is to
+         fill these in. This can't be done when the MDCT
+         modes are used because the decoder needs to know
+         the actual length for allocation purposes.*/
+       while(ret>2&&data[ret]==0)ret--;
+    }
+    /* Count ToC and redundancy */
+    ret += 1+redundancy_bytes;
+    if (!st->use_vbr)
+    {
+       if (opus_packet_pad(data, ret, max_data_bytes) != OPUS_OK)
+
+       {
+          RESTORE_STACK;
+          return OPUS_INTERNAL_ERROR;
+       }
+       ret = max_data_bytes;
+    }
+    RESTORE_STACK;
+    return ret;
+}
+
+#ifdef FIXED_POINT
+
+#ifndef DISABLE_FLOAT_API
+opus_int32 opus_encode_float(OpusEncoder *st, const float *pcm, int analysis_frame_size,
+      unsigned char *data, opus_int32 max_data_bytes)
+{
+   int i, ret;
+   int frame_size;
+   int delay_compensation;
+   VARDECL(opus_int16, in);
+   ALLOC_STACK;
+
+   if (st->application == OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+      delay_compensation = 0;
+   else
+      delay_compensation = st->delay_compensation;
+   frame_size = compute_frame_size(pcm, analysis_frame_size,
+         st->variable_duration, st->channels, st->Fs, st->bitrate_bps,
+         delay_compensation, downmix_float, st->analysis.subframe_mem);
+
+   ALLOC(in, frame_size*st->channels, opus_int16);
+
+   for (i=0;i<frame_size*st->channels;i++)
+      in[i] = FLOAT2INT16(pcm[i]);
+   ret = opus_encode_native(st, in, frame_size, data, max_data_bytes, 16,
+                            pcm, analysis_frame_size, 0, -2, st->channels, downmix_float, 1);
+   RESTORE_STACK;
+   return ret;
+}
+#endif
+
+opus_int32 opus_encode(OpusEncoder *st, const opus_int16 *pcm, int analysis_frame_size,
+                unsigned char *data, opus_int32 out_data_bytes)
+{
+   int frame_size;
+   int delay_compensation;
+   if (st->application == OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+      delay_compensation = 0;
+   else
+      delay_compensation = st->delay_compensation;
+   frame_size = compute_frame_size(pcm, analysis_frame_size,
+         st->variable_duration, st->channels, st->Fs, st->bitrate_bps,
+         delay_compensation, downmix_int
+#ifndef DISABLE_FLOAT_API
+         , st->analysis.subframe_mem
+#endif
+         );
+   return opus_encode_native(st, pcm, frame_size, data, out_data_bytes, 16,
+                             pcm, analysis_frame_size, 0, -2, st->channels, downmix_int, 0);
+}
+
+#else
+opus_int32 opus_encode(OpusEncoder *st, const opus_int16 *pcm, int analysis_frame_size,
+      unsigned char *data, opus_int32 max_data_bytes)
+{
+   int i, ret;
+   int frame_size;
+   int delay_compensation;
+   VARDECL(float, in);
+   ALLOC_STACK;
+
+   if (st->application == OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+      delay_compensation = 0;
+   else
+      delay_compensation = st->delay_compensation;
+   frame_size = compute_frame_size(pcm, analysis_frame_size,
+         st->variable_duration, st->channels, st->Fs, st->bitrate_bps,
+         delay_compensation, downmix_int, st->analysis.subframe_mem);
+
+   ALLOC(in, frame_size*st->channels, float);
+
+   for (i=0;i<frame_size*st->channels;i++)
+      in[i] = (1.0f/32768)*pcm[i];
+   ret = opus_encode_native(st, in, frame_size, data, max_data_bytes, 16,
+                            pcm, analysis_frame_size, 0, -2, st->channels, downmix_int, 0);
+   RESTORE_STACK;
+   return ret;
+}
+opus_int32 opus_encode_float(OpusEncoder *st, const float *pcm, int analysis_frame_size,
+                      unsigned char *data, opus_int32 out_data_bytes)
+{
+   int frame_size;
+   int delay_compensation;
+   if (st->application == OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+      delay_compensation = 0;
+   else
+      delay_compensation = st->delay_compensation;
+   frame_size = compute_frame_size(pcm, analysis_frame_size,
+         st->variable_duration, st->channels, st->Fs, st->bitrate_bps,
+         delay_compensation, downmix_float, st->analysis.subframe_mem);
+   return opus_encode_native(st, pcm, frame_size, data, out_data_bytes, 24,
+                             pcm, analysis_frame_size, 0, -2, st->channels, downmix_float, 1);
+}
+#endif
+
+
+int opus_encoder_ctl(OpusEncoder *st, int request, ...)
+{
+    int ret;
+    CELTEncoder *celt_enc;
+    va_list ap;
+
+    ret = OPUS_OK;
+    va_start(ap, request);
+
+    celt_enc = (CELTEncoder*)((char*)st+st->celt_enc_offset);
+
+    switch (request)
+    {
+        case OPUS_SET_APPLICATION_REQUEST:
+        {
+            opus_int32 value = va_arg(ap, opus_int32);
+            if (   (value != OPUS_APPLICATION_VOIP && value != OPUS_APPLICATION_AUDIO
+                 && value != OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+               || (!st->first && st->application != value))
+            {
+               ret = OPUS_BAD_ARG;
+               break;
+            }
+            st->application = value;
+        }
+        break;
+        case OPUS_GET_APPLICATION_REQUEST:
+        {
+            opus_int32 *value = va_arg(ap, opus_int32*);
+            if (!value)
+            {
+               goto bad_arg;
+            }
+            *value = st->application;
+        }
+        break;
+        case OPUS_SET_BITRATE_REQUEST:
+        {
+            opus_int32 value = va_arg(ap, opus_int32);
+            if (value != OPUS_AUTO && value != OPUS_BITRATE_MAX)
+            {
+                if (value <= 0)
+                    goto bad_arg;
+                else if (value <= 500)
+                    value = 500;
+                else if (value > (opus_int32)300000*st->channels)
+                    value = (opus_int32)300000*st->channels;
+            }
+            st->user_bitrate_bps = value;
+        }
+        break;
+        case OPUS_GET_BITRATE_REQUEST:
+        {
+            opus_int32 *value = va_arg(ap, opus_int32*);
+            if (!value)
+            {
+               goto bad_arg;
+            }
+            *value = user_bitrate_to_bitrate(st, st->prev_framesize, 1276);
+        }
+        break;
+        case OPUS_SET_FORCE_CHANNELS_REQUEST:
+        {
+            opus_int32 value = va_arg(ap, opus_int32);
+            if((value<1 || value>st->channels) && value != OPUS_AUTO)
+            {
+               goto bad_arg;
+            }
+            st->force_channels = value;
+        }
+        break;
+        case OPUS_GET_FORCE_CHANNELS_REQUEST:
+        {
+            opus_int32 *value = va_arg(ap, opus_int32*);
+            if (!value)
+            {
+               goto bad_arg;
+            }
+            *value = st->force_channels;
+        }
+        break;
+        case OPUS_SET_MAX_BANDWIDTH_REQUEST:
+        {
+            opus_int32 value = va_arg(ap, opus_int32);
+            if (value < OPUS_BANDWIDTH_NARROWBAND || value > OPUS_BANDWIDTH_FULLBAND)
+            {
+               goto bad_arg;
+            }
+            st->max_bandwidth = value;
+            if (st->max_bandwidth == OPUS_BANDWIDTH_NARROWBAND) {
+                st->silk_mode.maxInternalSampleRate = 8000;
+            } else if (st->max_bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) {
+                st->silk_mode.maxInternalSampleRate = 12000;
+            } else {
+                st->silk_mode.maxInternalSampleRate = 16000;
+            }
+        }
+        break;
+        case OPUS_GET_MAX_BANDWIDTH_REQUEST:
+        {
+            opus_int32 *value = va_arg(ap, opus_int32*);
+            if (!value)
+            {
+               goto bad_arg;
+            }
+            *value = st->max_bandwidth;
+        }
+        break;
+        case OPUS_SET_BANDWIDTH_REQUEST:
+        {
+            opus_int32 value = va_arg(ap, opus_int32);
+            if ((value < OPUS_BANDWIDTH_NARROWBAND || value > OPUS_BANDWIDTH_FULLBAND) && value != OPUS_AUTO)
+            {
+               goto bad_arg;
+            }
+            st->user_bandwidth = value;
+            if (st->user_bandwidth == OPUS_BANDWIDTH_NARROWBAND) {
+                st->silk_mode.maxInternalSampleRate = 8000;
+            } else if (st->user_bandwidth == OPUS_BANDWIDTH_MEDIUMBAND) {
+                st->silk_mode.maxInternalSampleRate = 12000;
+            } else {
+                st->silk_mode.maxInternalSampleRate = 16000;
+            }
+        }
+        break;
+        case OPUS_GET_BANDWIDTH_REQUEST:
+        {
+            opus_int32 *value = va_arg(ap, opus_int32*);
+            if (!value)
+            {
+               goto bad_arg;
+            }
+            *value = st->bandwidth;
+        }
+        break;
+        case OPUS_SET_DTX_REQUEST:
+        {
+            opus_int32 value = va_arg(ap, opus_int32);
+            if(value<0 || value>1)
+            {
+               goto bad_arg;
+            }
+            st->silk_mode.useDTX = value;
+        }
+        break;
+        case OPUS_GET_DTX_REQUEST:
+        {
+            opus_int32 *value = va_arg(ap, opus_int32*);
+            if (!value)
+            {
+               goto bad_arg;
+            }
+            *value = st->silk_mode.useDTX;
+        }
+        break;
+        case OPUS_SET_COMPLEXITY_REQUEST:
+        {
+            opus_int32 value = va_arg(ap, opus_int32);
+            if(value<0 || value>10)
+            {
+               goto bad_arg;
+            }
+            st->silk_mode.complexity = value;
+            celt_encoder_ctl(celt_enc, OPUS_SET_COMPLEXITY(value));
+        }
+        break;
+        case OPUS_GET_COMPLEXITY_REQUEST:
+        {
+            opus_int32 *value = va_arg(ap, opus_int32*);
+            if (!value)
+            {
+               goto bad_arg;
+            }
+            *value = st->silk_mode.complexity;
+        }
+        break;
+        case OPUS_SET_INBAND_FEC_REQUEST:
+        {
+            opus_int32 value = va_arg(ap, opus_int32);
+            if(value<0 || value>1)
+            {
+               goto bad_arg;
+            }
+            st->silk_mode.useInBandFEC = value;
+        }
+        break;
+        case OPUS_GET_INBAND_FEC_REQUEST:
+        {
+            opus_int32 *value = va_arg(ap, opus_int32*);
+            if (!value)
+            {
+               goto bad_arg;
+            }
+            *value = st->silk_mode.useInBandFEC;
+        }
+        break;
+        case OPUS_SET_PACKET_LOSS_PERC_REQUEST:
+        {
+            opus_int32 value = va_arg(ap, opus_int32);
+            if (value < 0 || value > 100)
+            {
+               goto bad_arg;
+            }
+            st->silk_mode.packetLossPercentage = value;
+            celt_encoder_ctl(celt_enc, OPUS_SET_PACKET_LOSS_PERC(value));
+        }
+        break;
+        case OPUS_GET_PACKET_LOSS_PERC_REQUEST:
+        {
+            opus_int32 *value = va_arg(ap, opus_int32*);
+            if (!value)
+            {
+               goto bad_arg;
+            }
+            *value = st->silk_mode.packetLossPercentage;
+        }
+        break;
+        case OPUS_SET_VBR_REQUEST:
+        {
+            opus_int32 value = va_arg(ap, opus_int32);
+            if(value<0 || value>1)
+            {
+               goto bad_arg;
+            }
+            st->use_vbr = value;
+            st->silk_mode.useCBR = 1-value;
+        }
+        break;
+        case OPUS_GET_VBR_REQUEST:
+        {
+            opus_int32 *value = va_arg(ap, opus_int32*);
+            if (!value)
+            {
+               goto bad_arg;
+            }
+            *value = st->use_vbr;
+        }
+        break;
+        case OPUS_SET_VOICE_RATIO_REQUEST:
+        {
+            opus_int32 value = va_arg(ap, opus_int32);
+            if (value<-1 || value>100)
+            {
+               goto bad_arg;
+            }
+            st->voice_ratio = value;
+        }
+        break;
+        case OPUS_GET_VOICE_RATIO_REQUEST:
+        {
+            opus_int32 *value = va_arg(ap, opus_int32*);
+            if (!value)
+            {
+               goto bad_arg;
+            }
+            *value = st->voice_ratio;
+        }
+        break;
+        case OPUS_SET_VBR_CONSTRAINT_REQUEST:
+        {
+            opus_int32 value = va_arg(ap, opus_int32);
+            if(value<0 || value>1)
+            {
+               goto bad_arg;
+            }
+            st->vbr_constraint = value;
+        }
+        break;
+        case OPUS_GET_VBR_CONSTRAINT_REQUEST:
+        {
+            opus_int32 *value = va_arg(ap, opus_int32*);
+            if (!value)
+            {
+               goto bad_arg;
+            }
+            *value = st->vbr_constraint;
+        }
+        break;
+        case OPUS_SET_SIGNAL_REQUEST:
+        {
+            opus_int32 value = va_arg(ap, opus_int32);
+            if(value!=OPUS_AUTO && value!=OPUS_SIGNAL_VOICE && value!=OPUS_SIGNAL_MUSIC)
+            {
+               goto bad_arg;
+            }
+            st->signal_type = value;
+        }
+        break;
+        case OPUS_GET_SIGNAL_REQUEST:
+        {
+            opus_int32 *value = va_arg(ap, opus_int32*);
+            if (!value)
+            {
+               goto bad_arg;
+            }
+            *value = st->signal_type;
+        }
+        break;
+        case OPUS_GET_LOOKAHEAD_REQUEST:
+        {
+            opus_int32 *value = va_arg(ap, opus_int32*);
+            if (!value)
+            {
+               goto bad_arg;
+            }
+            *value = st->Fs/400;
+            if (st->application != OPUS_APPLICATION_RESTRICTED_LOWDELAY)
+                *value += st->delay_compensation;
+        }
+        break;
+        case OPUS_GET_SAMPLE_RATE_REQUEST:
+        {
+            opus_int32 *value = va_arg(ap, opus_int32*);
+            if (!value)
+            {
+               goto bad_arg;
+            }
+            *value = st->Fs;
+        }
+        break;
+        case OPUS_GET_FINAL_RANGE_REQUEST:
+        {
+            opus_uint32 *value = va_arg(ap, opus_uint32*);
+            if (!value)
+            {
+               goto bad_arg;
+            }
+            *value = st->rangeFinal;
+        }
+        break;
+        case OPUS_SET_LSB_DEPTH_REQUEST:
+        {
+            opus_int32 value = va_arg(ap, opus_int32);
+            if (value<8 || value>24)
+            {
+               goto bad_arg;
+            }
+            st->lsb_depth=value;
+        }
+        break;
+        case OPUS_GET_LSB_DEPTH_REQUEST:
+        {
+            opus_int32 *value = va_arg(ap, opus_int32*);
+            if (!value)
+            {
+               goto bad_arg;
+            }
+            *value = st->lsb_depth;
+        }
+        break;
+        case OPUS_SET_EXPERT_FRAME_DURATION_REQUEST:
+        {
+            opus_int32 value = va_arg(ap, opus_int32);
+            if (value != OPUS_FRAMESIZE_ARG   && value != OPUS_FRAMESIZE_2_5_MS &&
+                value != OPUS_FRAMESIZE_5_MS  && value != OPUS_FRAMESIZE_10_MS  &&
+                value != OPUS_FRAMESIZE_20_MS && value != OPUS_FRAMESIZE_40_MS  &&
+                value != OPUS_FRAMESIZE_60_MS && value != OPUS_FRAMESIZE_VARIABLE)
+            {
+               goto bad_arg;
+            }
+            st->variable_duration = value;
+            celt_encoder_ctl(celt_enc, OPUS_SET_EXPERT_FRAME_DURATION(value));
+        }
+        break;
+        case OPUS_GET_EXPERT_FRAME_DURATION_REQUEST:
+        {
+            opus_int32 *value = va_arg(ap, opus_int32*);
+            if (!value)
+            {
+               goto bad_arg;
+            }
+            *value = st->variable_duration;
+        }
+        break;
+        case OPUS_SET_PREDICTION_DISABLED_REQUEST:
+        {
+           opus_int32 value = va_arg(ap, opus_int32);
+           if (value > 1 || value < 0)
+              goto bad_arg;
+           st->silk_mode.reducedDependency = value;
+        }
+        break;
+        case OPUS_GET_PREDICTION_DISABLED_REQUEST:
+        {
+           opus_int32 *value = va_arg(ap, opus_int32*);
+           if (!value)
+              goto bad_arg;
+           *value = st->silk_mode.reducedDependency;
+        }
+        break;
+        case OPUS_RESET_STATE:
+        {
+           void *silk_enc;
+           silk_EncControlStruct dummy;
+           char *start;
+           silk_enc = (char*)st+st->silk_enc_offset;
+#ifndef DISABLE_FLOAT_API
+           tonality_analysis_reset(&st->analysis);
+#endif
+
+           start = (char*)&st->OPUS_ENCODER_RESET_START;
+           OPUS_CLEAR(start, sizeof(OpusEncoder) - (start - (char*)st));
+
+           celt_encoder_ctl(celt_enc, OPUS_RESET_STATE);
+           silk_InitEncoder( silk_enc, st->arch, &dummy );
+           st->stream_channels = st->channels;
+           st->hybrid_stereo_width_Q14 = 1 << 14;
+           st->prev_HB_gain = Q15ONE;
+           st->first = 1;
+           st->mode = MODE_HYBRID;
+           st->bandwidth = OPUS_BANDWIDTH_FULLBAND;
+           st->variable_HP_smth2_Q15 = silk_LSHIFT( silk_lin2log( VARIABLE_HP_MIN_CUTOFF_HZ ), 8 );
+        }
+        break;
+        case OPUS_SET_FORCE_MODE_REQUEST:
+        {
+            opus_int32 value = va_arg(ap, opus_int32);
+            if ((value < MODE_SILK_ONLY || value > MODE_CELT_ONLY) && value != OPUS_AUTO)
+            {
+               goto bad_arg;
+            }
+            st->user_forced_mode = value;
+        }
+        break;
+        case OPUS_SET_LFE_REQUEST:
+        {
+            opus_int32 value = va_arg(ap, opus_int32);
+            st->lfe = value;
+            ret = celt_encoder_ctl(celt_enc, OPUS_SET_LFE(value));
+        }
+        break;
+        case OPUS_SET_ENERGY_MASK_REQUEST:
+        {
+            opus_val16 *value = va_arg(ap, opus_val16*);
+            st->energy_masking = value;
+            ret = celt_encoder_ctl(celt_enc, OPUS_SET_ENERGY_MASK(value));
+        }
+        break;
+
+        case CELT_GET_MODE_REQUEST:
+        {
+           const CELTMode ** value = va_arg(ap, const CELTMode**);
+           if (!value)
+           {
+              goto bad_arg;
+           }
+           ret = celt_encoder_ctl(celt_enc, CELT_GET_MODE(value));
+        }
+        break;
+        default:
+            /* fprintf(stderr, "unknown opus_encoder_ctl() request: %d", request);*/
+            ret = OPUS_UNIMPLEMENTED;
+            break;
+    }
+    va_end(ap);
+    return ret;
+bad_arg:
+    va_end(ap);
+    return OPUS_BAD_ARG;
+}
+
+void opus_encoder_destroy(OpusEncoder *st)
+{
+    opus_free(st);
+}