lavc/opus*: move to opus/ subdir

24 files changed, 8415 insertions, 0 deletions

diff --git a/libavcodec/opus/Makefile b/libavcodec/opus/Makefile
new file mode 100644
index 0000000000..53cb98e28d
--- /dev/null
+++ b/libavcodec/opus/Makefile
@@ -0,0 +1,30 @@
+clean::
+	$(RM) $(CLEANSUFFIXES:%=libavcodec/opus/%)
+
+OBJS-$(CONFIG_OPUS_DECODER) +=  \
+    opus/dec.o                  \
+    opus/dec_celt.o             \
+    opus/celt.o                 \
+    opus/pvq.o                  \
+    opus/silk.o                 \
+    opus/tab.o                  \
+    opus/dsp.o                  \
+    opus/parse.o                \
+    opus/rc.o                   \
+
+
+OBJS-$(CONFIG_OPUS_PARSER) +=   \
+    opus/parser.o               \
+    opus/parse.o                \
+
+
+OBJS-$(CONFIG_OPUS_ENCODER) +=  \
+    opus/enc.o                  \
+    opus/enc_psy.o              \
+    opus/celt.o                 \
+    opus/pvq.o                  \
+    opus/rc.o                   \
+    opus/tab.o                  \
+
+
+libavcodec/opus/%.o: CPPFLAGS += -I$(SRC_PATH)/libavcodec/
diff --git a/libavcodec/opus/celt.c b/libavcodec/opus/celt.c
new file mode 100644
index 0000000000..3b9c633702
--- /dev/null
+++ b/libavcodec/opus/celt.c
@@ -0,0 +1,484 @@
+/*
+ * Copyright (c) 2012 Andrew D'Addesio
+ * Copyright (c) 2013-2014 Mozilla Corporation
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <stdint.h>
+
+#include "celt.h"
+#include "pvq.h"
+#include "tab.h"
+
+void ff_celt_quant_bands(CeltFrame *f, OpusRangeCoder *rc)
+{
+    float lowband_scratch[8 * 22];
+    float norm1[2 * 8 * 100];
+    float *norm2 = norm1 + 8 * 100;
+
+    int totalbits = (f->framebits << 3) - f->anticollapse_needed;
+
+    int update_lowband = 1;
+    int lowband_offset = 0;
+
+    int i, j;
+
+    for (i = f->start_band; i < f->end_band; i++) {
+        uint32_t cm[2] = { (1 << f->blocks) - 1, (1 << f->blocks) - 1 };
+        int band_offset = ff_celt_freq_bands[i] << f->size;
+        int band_size   = ff_celt_freq_range[i] << f->size;
+        float *X = f->block[0].coeffs + band_offset;
+        float *Y = (f->channels == 2) ? f->block[1].coeffs + band_offset : NULL;
+        float *norm_loc1, *norm_loc2;
+
+        int consumed = opus_rc_tell_frac(rc);
+        int effective_lowband = -1;
+        int b = 0;
+
+        /* Compute how many bits we want to allocate to this band */
+        if (i != f->start_band)
+            f->remaining -= consumed;
+        f->remaining2 = totalbits - consumed - 1;
+        if (i <= f->coded_bands - 1) {
+            int curr_balance = f->remaining / FFMIN(3, f->coded_bands-i);
+            b = av_clip_uintp2(FFMIN(f->remaining2 + 1, f->pulses[i] + curr_balance), 14);
+        }
+
+        if ((ff_celt_freq_bands[i] - ff_celt_freq_range[i] >= ff_celt_freq_bands[f->start_band] ||
+            i == f->start_band + 1) && (update_lowband || lowband_offset == 0))
+            lowband_offset = i;
+
+        if (i == f->start_band + 1) {
+            /* Special Hybrid Folding (RFC 8251 section 9). Copy the first band into
+            the second to ensure the second band never has to use the LCG. */
+            int count = (ff_celt_freq_range[i] - ff_celt_freq_range[i-1]) << f->size;
+
+            memcpy(&norm1[band_offset], &norm1[band_offset - count], count * sizeof(float));
+
+            if (f->channels == 2)
+                memcpy(&norm2[band_offset], &norm2[band_offset - count], count * sizeof(float));
+        }
+
+        /* Get a conservative estimate of the collapse_mask's for the bands we're
+           going to be folding from. */
+        if (lowband_offset != 0 && (f->spread != CELT_SPREAD_AGGRESSIVE ||
+                                    f->blocks > 1 || f->tf_change[i] < 0)) {
+            int foldstart, foldend;
+
+            /* This ensures we never repeat spectral content within one band */
+            effective_lowband = FFMAX(ff_celt_freq_bands[f->start_band],
+                                      ff_celt_freq_bands[lowband_offset] - ff_celt_freq_range[i]);
+            foldstart = lowband_offset;
+            while (ff_celt_freq_bands[--foldstart] > effective_lowband);
+            foldend = lowband_offset - 1;
+            while (++foldend < i && ff_celt_freq_bands[foldend] < effective_lowband + ff_celt_freq_range[i]);
+
+            cm[0] = cm[1] = 0;
+            for (j = foldstart; j < foldend; j++) {
+                cm[0] |= f->block[0].collapse_masks[j];
+                cm[1] |= f->block[f->channels - 1].collapse_masks[j];
+            }
+        }
+
+        if (f->dual_stereo && i == f->intensity_stereo) {
+            /* Switch off dual stereo to do intensity */
+            f->dual_stereo = 0;
+            for (j = ff_celt_freq_bands[f->start_band] << f->size; j < band_offset; j++)
+                norm1[j] = (norm1[j] + norm2[j]) / 2;
+        }
+
+        norm_loc1 = effective_lowband != -1 ? norm1 + (effective_lowband << f->size) : NULL;
+        norm_loc2 = effective_lowband != -1 ? norm2 + (effective_lowband << f->size) : NULL;
+
+        if (f->dual_stereo) {
+            cm[0] = f->pvq->quant_band(f->pvq, f, rc, i, X, NULL, band_size, b >> 1,
+                                       f->blocks, norm_loc1, f->size,
+                                       norm1 + band_offset, 0, 1.0f,
+                                       lowband_scratch, cm[0]);
+
+            cm[1] = f->pvq->quant_band(f->pvq, f, rc, i, Y, NULL, band_size, b >> 1,
+                                       f->blocks, norm_loc2, f->size,
+                                       norm2 + band_offset, 0, 1.0f,
+                                       lowband_scratch, cm[1]);
+        } else {
+            cm[0] = f->pvq->quant_band(f->pvq, f, rc, i, X,    Y, band_size, b >> 0,
+                                       f->blocks, norm_loc1, f->size,
+                                       norm1 + band_offset, 0, 1.0f,
+                                       lowband_scratch, cm[0] | cm[1]);
+            cm[1] = cm[0];
+        }
+
+        f->block[0].collapse_masks[i]               = (uint8_t)cm[0];
+        f->block[f->channels - 1].collapse_masks[i] = (uint8_t)cm[1];
+        f->remaining += f->pulses[i] + consumed;
+
+        /* Update the folding position only as long as we have 1 bit/sample depth */
+        update_lowband = (b > band_size << 3);
+    }
+}
+
+#define NORMC(bits) ((bits) << (f->channels - 1) << f->size >> 2)
+
+void ff_celt_bitalloc(CeltFrame *f, OpusRangeCoder *rc, int encode)
+{
+    int i, j, low, high, total, done, bandbits, remaining, tbits_8ths;
+    int skip_startband      = f->start_band;
+    int skip_bit            = 0;
+    int intensitystereo_bit = 0;
+    int dualstereo_bit      = 0;
+    int dynalloc            = 6;
+    int extrabits           = 0;
+
+    int boost[CELT_MAX_BANDS] = { 0 };
+    int trim_offset[CELT_MAX_BANDS];
+    int threshold[CELT_MAX_BANDS];
+    int bits1[CELT_MAX_BANDS];
+    int bits2[CELT_MAX_BANDS];
+
+    /* Spread */
+    if (opus_rc_tell(rc) + 4 <= f->framebits) {
+        if (encode)
+            ff_opus_rc_enc_cdf(rc, f->spread, ff_celt_model_spread);
+        else
+            f->spread = ff_opus_rc_dec_cdf(rc, ff_celt_model_spread);
+    } else {
+        f->spread = CELT_SPREAD_NORMAL;
+    }
+
+    /* Initialize static allocation caps */
+    for (i = 0; i < CELT_MAX_BANDS; i++)
+        f->caps[i] = NORMC((ff_celt_static_caps[f->size][f->channels - 1][i] + 64) * ff_celt_freq_range[i]);
+
+    /* Band boosts */
+    tbits_8ths = f->framebits << 3;
+    for (i = f->start_band; i < f->end_band; i++) {
+        int quanta = ff_celt_freq_range[i] << (f->channels - 1) << f->size;
+        int b_dynalloc = dynalloc;
+        int boost_amount = f->alloc_boost[i];
+        quanta = FFMIN(quanta << 3, FFMAX(6 << 3, quanta));
+
+        while (opus_rc_tell_frac(rc) + (b_dynalloc << 3) < tbits_8ths && boost[i] < f->caps[i]) {
+            int is_boost;
+            if (encode) {
+                is_boost = boost_amount--;
+                ff_opus_rc_enc_log(rc, is_boost, b_dynalloc);
+            } else {
+                is_boost = ff_opus_rc_dec_log(rc, b_dynalloc);
+            }
+
+            if (!is_boost)
+                break;
+
+            boost[i]   += quanta;
+            tbits_8ths -= quanta;
+
+            b_dynalloc = 1;
+        }
+
+        if (boost[i])
+            dynalloc = FFMAX(dynalloc - 1, 2);
+    }
+
+    /* Allocation trim */
+    if (!encode)
+        f->alloc_trim = 5;
+    if (opus_rc_tell_frac(rc) + (6 << 3) <= tbits_8ths)
+        if (encode)
+            ff_opus_rc_enc_cdf(rc, f->alloc_trim, ff_celt_model_alloc_trim);
+        else
+            f->alloc_trim = ff_opus_rc_dec_cdf(rc, ff_celt_model_alloc_trim);
+
+    /* Anti-collapse bit reservation */
+    tbits_8ths = (f->framebits << 3) - opus_rc_tell_frac(rc) - 1;
+    f->anticollapse_needed = 0;
+    if (f->transient && f->size >= 2 && tbits_8ths >= ((f->size + 2) << 3))
+        f->anticollapse_needed = 1 << 3;
+    tbits_8ths -= f->anticollapse_needed;
+
+    /* Band skip bit reservation */
+    if (tbits_8ths >= 1 << 3)
+        skip_bit = 1 << 3;
+    tbits_8ths -= skip_bit;
+
+    /* Intensity/dual stereo bit reservation */
+    if (f->channels == 2) {
+        intensitystereo_bit = ff_celt_log2_frac[f->end_band - f->start_band];
+        if (intensitystereo_bit <= tbits_8ths) {
+            tbits_8ths -= intensitystereo_bit;
+            if (tbits_8ths >= 1 << 3) {
+                dualstereo_bit = 1 << 3;
+                tbits_8ths -= 1 << 3;
+            }
+        } else {
+            intensitystereo_bit = 0;
+        }
+    }
+
+    /* Trim offsets */
+    for (i = f->start_band; i < f->end_band; i++) {
+        int trim     = f->alloc_trim - 5 - f->size;
+        int band     = ff_celt_freq_range[i] * (f->end_band - i - 1);
+        int duration = f->size + 3;
+        int scale    = duration + f->channels - 1;
+
+        /* PVQ minimum allocation threshold, below this value the band is
+         * skipped */
+        threshold[i] = FFMAX(3 * ff_celt_freq_range[i] << duration >> 4,
+                             f->channels << 3);
+
+        trim_offset[i] = trim * (band << scale) >> 6;
+
+        if (ff_celt_freq_range[i] << f->size == 1)
+            trim_offset[i] -= f->channels << 3;
+    }
+
+    /* Bisection */
+    low  = 1;
+    high = CELT_VECTORS - 1;
+    while (low <= high) {
+        int center = (low + high) >> 1;
+        done = total = 0;
+
+        for (i = f->end_band - 1; i >= f->start_band; i--) {
+            bandbits = NORMC(ff_celt_freq_range[i] * ff_celt_static_alloc[center][i]);
+
+            if (bandbits)
+                bandbits = FFMAX(bandbits + trim_offset[i], 0);
+            bandbits += boost[i];
+
+            if (bandbits >= threshold[i] || done) {
+                done = 1;
+                total += FFMIN(bandbits, f->caps[i]);
+            } else if (bandbits >= f->channels << 3) {
+                total += f->channels << 3;
+            }
+        }
+
+        if (total > tbits_8ths)
+            high = center - 1;
+        else
+            low = center + 1;
+    }
+    high = low--;
+
+    /* Bisection */
+    for (i = f->start_band; i < f->end_band; i++) {
+        bits1[i] = NORMC(ff_celt_freq_range[i] * ff_celt_static_alloc[low][i]);
+        bits2[i] = high >= CELT_VECTORS ? f->caps[i] :
+                   NORMC(ff_celt_freq_range[i] * ff_celt_static_alloc[high][i]);
+
+        if (bits1[i])
+            bits1[i] = FFMAX(bits1[i] + trim_offset[i], 0);
+        if (bits2[i])
+            bits2[i] = FFMAX(bits2[i] + trim_offset[i], 0);
+
+        if (low)
+            bits1[i] += boost[i];
+        bits2[i] += boost[i];
+
+        if (boost[i])
+            skip_startband = i;
+        bits2[i] = FFMAX(bits2[i] - bits1[i], 0);
+    }
+
+    /* Bisection */
+    low  = 0;
+    high = 1 << CELT_ALLOC_STEPS;
+    for (i = 0; i < CELT_ALLOC_STEPS; i++) {
+        int center = (low + high) >> 1;
+        done = total = 0;
+
+        for (j = f->end_band - 1; j >= f->start_band; j--) {
+            bandbits = bits1[j] + (center * bits2[j] >> CELT_ALLOC_STEPS);
+
+            if (bandbits >= threshold[j] || done) {
+                done = 1;
+                total += FFMIN(bandbits, f->caps[j]);
+            } else if (bandbits >= f->channels << 3)
+                total += f->channels << 3;
+        }
+        if (total > tbits_8ths)
+            high = center;
+        else
+            low = center;
+    }
+
+    /* Bisection */
+    done = total = 0;
+    for (i = f->end_band - 1; i >= f->start_band; i--) {
+        bandbits = bits1[i] + (low * bits2[i] >> CELT_ALLOC_STEPS);
+
+        if (bandbits >= threshold[i] || done)
+            done = 1;
+        else
+            bandbits = (bandbits >= f->channels << 3) ?
+            f->channels << 3 : 0;
+
+        bandbits     = FFMIN(bandbits, f->caps[i]);
+        f->pulses[i] = bandbits;
+        total      += bandbits;
+    }
+
+    /* Band skipping */
+    for (f->coded_bands = f->end_band; ; f->coded_bands--) {
+        int allocation;
+        j = f->coded_bands - 1;
+
+        if (j == skip_startband) {
+            /* all remaining bands are not skipped */
+            tbits_8ths += skip_bit;
+            break;
+        }
+
+        /* determine the number of bits available for coding "do not skip" markers */
+        remaining   = tbits_8ths - total;
+        bandbits    = remaining / (ff_celt_freq_bands[j+1] - ff_celt_freq_bands[f->start_band]);
+        remaining  -= bandbits  * (ff_celt_freq_bands[j+1] - ff_celt_freq_bands[f->start_band]);
+        allocation  = f->pulses[j] + bandbits * ff_celt_freq_range[j];
+        allocation += FFMAX(remaining - (ff_celt_freq_bands[j] - ff_celt_freq_bands[f->start_band]), 0);
+
+        /* a "do not skip" marker is only coded if the allocation is
+         * above the chosen threshold */
+        if (allocation >= FFMAX(threshold[j], (f->channels + 1) << 3)) {
+            int do_not_skip;
+            if (encode) {
+                do_not_skip = f->coded_bands <= f->skip_band_floor;
+                ff_opus_rc_enc_log(rc, do_not_skip, 1);
+            } else {
+                do_not_skip = ff_opus_rc_dec_log(rc, 1);
+            }
+
+            if (do_not_skip)
+                break;
+
+            total      += 1 << 3;
+            allocation -= 1 << 3;
+        }
+
+        /* the band is skipped, so reclaim its bits */
+        total -= f->pulses[j];
+        if (intensitystereo_bit) {
+            total -= intensitystereo_bit;
+            intensitystereo_bit = ff_celt_log2_frac[j - f->start_band];
+            total += intensitystereo_bit;
+        }
+
+        total += f->pulses[j] = (allocation >= f->channels << 3) ? f->channels << 3 : 0;
+    }
+
+    /* IS start band */
+    if (encode) {
+        if (intensitystereo_bit) {
+            f->intensity_stereo = FFMIN(f->intensity_stereo, f->coded_bands);
+            ff_opus_rc_enc_uint(rc, f->intensity_stereo, f->coded_bands + 1 - f->start_band);
+        }
+    } else {
+        f->intensity_stereo = f->dual_stereo = 0;
+        if (intensitystereo_bit)
+            f->intensity_stereo = f->start_band + ff_opus_rc_dec_uint(rc, f->coded_bands + 1 - f->start_band);
+    }
+
+    /* DS flag */
+    if (f->intensity_stereo <= f->start_band)
+        tbits_8ths += dualstereo_bit; /* no intensity stereo means no dual stereo */
+    else if (dualstereo_bit)
+        if (encode)
+            ff_opus_rc_enc_log(rc, f->dual_stereo, 1);
+        else
+            f->dual_stereo = ff_opus_rc_dec_log(rc, 1);
+
+    /* Supply the remaining bits in this frame to lower bands */
+    remaining = tbits_8ths - total;
+    bandbits  = remaining / (ff_celt_freq_bands[f->coded_bands] - ff_celt_freq_bands[f->start_band]);
+    remaining -= bandbits * (ff_celt_freq_bands[f->coded_bands] - ff_celt_freq_bands[f->start_band]);
+    for (i = f->start_band; i < f->coded_bands; i++) {
+        const int bits = FFMIN(remaining, ff_celt_freq_range[i]);
+        f->pulses[i] += bits + bandbits * ff_celt_freq_range[i];
+        remaining    -= bits;
+    }
+
+    /* Finally determine the allocation */
+    for (i = f->start_band; i < f->coded_bands; i++) {
+        int N = ff_celt_freq_range[i] << f->size;
+        int prev_extra = extrabits;
+        f->pulses[i] += extrabits;
+
+        if (N > 1) {
+            int dof;        /* degrees of freedom */
+            int temp;       /* dof * channels * log(dof) */
+            int fine_bits;
+            int max_bits;
+            int offset;     /* fine energy quantization offset, i.e.
+                             * extra bits assigned over the standard
+                             * totalbits/dof */
+
+            extrabits = FFMAX(f->pulses[i] - f->caps[i], 0);
+            f->pulses[i] -= extrabits;
+
+            /* intensity stereo makes use of an extra degree of freedom */
+            dof = N * f->channels + (f->channels == 2 && N > 2 && !f->dual_stereo && i < f->intensity_stereo);
+            temp = dof * (ff_celt_log_freq_range[i] + (f->size << 3));
+            offset = (temp >> 1) - dof * CELT_FINE_OFFSET;
+            if (N == 2) /* dof=2 is the only case that doesn't fit the model */
+                offset += dof << 1;
+
+            /* grant an additional bias for the first and second pulses */
+            if (f->pulses[i] + offset < 2 * (dof << 3))
+                offset += temp >> 2;
+            else if (f->pulses[i] + offset < 3 * (dof << 3))
+                offset += temp >> 3;
+
+            fine_bits = (f->pulses[i] + offset + (dof << 2)) / (dof << 3);
+            max_bits  = FFMIN((f->pulses[i] >> 3) >> (f->channels - 1), CELT_MAX_FINE_BITS);
+            max_bits  = FFMAX(max_bits, 0);
+            f->fine_bits[i] = av_clip(fine_bits, 0, max_bits);
+
+            /* If fine_bits was rounded down or capped,
+             * give priority for the final fine energy pass */
+            f->fine_priority[i] = (f->fine_bits[i] * (dof << 3) >= f->pulses[i] + offset);
+
+            /* the remaining bits are assigned to PVQ */
+            f->pulses[i] -= f->fine_bits[i] << (f->channels - 1) << 3;
+        } else {
+            /* all bits go to fine energy except for the sign bit */
+            extrabits = FFMAX(f->pulses[i] - (f->channels << 3), 0);
+            f->pulses[i] -= extrabits;
+            f->fine_bits[i] = 0;
+            f->fine_priority[i] = 1;
+        }
+
+        /* hand back a limited number of extra fine energy bits to this band */
+        if (extrabits > 0) {
+            int fineextra = FFMIN(extrabits >> (f->channels + 2),
+                                  CELT_MAX_FINE_BITS - f->fine_bits[i]);
+            f->fine_bits[i] += fineextra;
+
+            fineextra <<= f->channels + 2;
+            f->fine_priority[i] = (fineextra >= extrabits - prev_extra);
+            extrabits -= fineextra;
+        }
+    }
+    f->remaining = extrabits;
+
+    /* skipped bands dedicate all of their bits for fine energy */
+    for (; i < f->end_band; i++) {
+        f->fine_bits[i]     = f->pulses[i] >> (f->channels - 1) >> 3;
+        f->pulses[i]        = 0;
+        f->fine_priority[i] = f->fine_bits[i] < 1;
+    }
+}
diff --git a/libavcodec/opus/celt.h b/libavcodec/opus/celt.h
new file mode 100644
index 0000000000..e957f2c123
--- /dev/null
+++ b/libavcodec/opus/celt.h
@@ -0,0 +1,184 @@
+/*
+ * Opus decoder/encoder CELT functions
+ * Copyright (c) 2012 Andrew D'Addesio
+ * Copyright (c) 2013-2014 Mozilla Corporation
+ * Copyright (c) 2016 Rostislav Pehlivanov <atomnuker@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_OPUS_CELT_H
+#define AVCODEC_OPUS_CELT_H
+
+#include <stdint.h>
+
+#include "libavcodec/avcodec.h"
+
+#include "dsp.h"
+#include "rc.h"
+
+#include "libavutil/float_dsp.h"
+#include "libavutil/libm.h"
+#include "libavutil/mem_internal.h"
+#include "libavutil/tx.h"
+
+#define CELT_SHORT_BLOCKSIZE         120
+#define CELT_OVERLAP                 CELT_SHORT_BLOCKSIZE
+#define CELT_MAX_LOG_BLOCKS          3
+#define CELT_MAX_FRAME_SIZE          (CELT_SHORT_BLOCKSIZE * (1 << CELT_MAX_LOG_BLOCKS))
+#define CELT_MAX_BANDS               21
+
+#define CELT_VECTORS                 11
+#define CELT_ALLOC_STEPS             6
+#define CELT_FINE_OFFSET             21
+#define CELT_MAX_FINE_BITS           8
+#define CELT_NORM_SCALE              16384
+#define CELT_QTHETA_OFFSET           4
+#define CELT_QTHETA_OFFSET_TWOPHASE  16
+#define CELT_POSTFILTER_MINPERIOD    15
+#define CELT_ENERGY_SILENCE          (-28.0f)
+
+enum CeltSpread {
+    CELT_SPREAD_NONE,
+    CELT_SPREAD_LIGHT,
+    CELT_SPREAD_NORMAL,
+    CELT_SPREAD_AGGRESSIVE
+};
+
+enum CeltBlockSize {
+    CELT_BLOCK_120,
+    CELT_BLOCK_240,
+    CELT_BLOCK_480,
+    CELT_BLOCK_960,
+
+    CELT_BLOCK_NB
+};
+
+typedef struct CeltBlock {
+    float energy[CELT_MAX_BANDS];
+    float lin_energy[CELT_MAX_BANDS];
+    float error_energy[CELT_MAX_BANDS];
+    float prev_energy[2][CELT_MAX_BANDS];
+
+    uint8_t collapse_masks[CELT_MAX_BANDS];
+
+    /* buffer for mdct output + postfilter */
+    DECLARE_ALIGNED(32, float, buf)[2048];
+    DECLARE_ALIGNED(32, float, coeffs)[CELT_MAX_FRAME_SIZE];
+
+    /* Used by the encoder */
+    DECLARE_ALIGNED(32, float, overlap)[FFALIGN(CELT_OVERLAP, 16)];
+    DECLARE_ALIGNED(32, float, samples)[FFALIGN(CELT_MAX_FRAME_SIZE, 16)];
+
+    /* postfilter parameters */
+    int   pf_period_new;
+    float pf_gains_new[3];
+    int   pf_period;
+    float pf_gains[3];
+    int   pf_period_old;
+    float pf_gains_old[3];
+
+    float emph_coeff;
+} CeltBlock;
+
+typedef struct CeltFrame {
+    // constant values that do not change during context lifetime
+    AVCodecContext      *avctx;
+    AVTXContext        *tx[4];
+    av_tx_fn            tx_fn[4];
+    AVFloatDSPContext   *dsp;
+    CeltBlock           block[2];
+    struct CeltPVQ      *pvq;
+    OpusDSP             opusdsp;
+    int channels;
+    int output_channels;
+    int apply_phase_inv;
+
+    enum CeltBlockSize size;
+    int start_band;
+    int end_band;
+    int coded_bands;
+    int transient;
+    int pfilter;
+    int skip_band_floor;
+    int tf_select;
+    int alloc_trim;
+    int alloc_boost[CELT_MAX_BANDS];
+    int blocks;        /* number of iMDCT blocks in the frame, depends on transient */
+    int blocksize;     /* size of each block */
+    int silence;       /* Frame is filled with silence */
+    int anticollapse_needed; /* Whether to expect an anticollapse bit */
+    int anticollapse;  /* Encoded anticollapse bit */
+    int intensity_stereo;
+    int dual_stereo;
+    int flushed;
+    uint32_t seed;
+    enum CeltSpread spread;
+
+    /* Encoder PF coeffs */
+    int pf_octave;
+    int pf_period;
+    int pf_tapset;
+    float pf_gain;
+
+    /* Bit allocation */
+    int framebits;
+    int remaining;
+    int remaining2;
+    int caps         [CELT_MAX_BANDS];
+    int fine_bits    [CELT_MAX_BANDS];
+    int fine_priority[CELT_MAX_BANDS];
+    int pulses       [CELT_MAX_BANDS];
+    int tf_change    [CELT_MAX_BANDS];
+} CeltFrame;
+
+/* LCG for noise generation */
+static av_always_inline uint32_t celt_rng(CeltFrame *f)
+{
+    f->seed = 1664525 * f->seed + 1013904223;
+    return f->seed;
+}
+
+static av_always_inline void celt_renormalize_vector(float *X, int N, float gain)
+{
+    int i;
+    float g = 1e-15f;
+    for (i = 0; i < N; i++)
+        g += X[i] * X[i];
+    g = gain / sqrtf(g);
+
+    for (i = 0; i < N; i++)
+        X[i] *= g;
+}
+
+int ff_celt_init(AVCodecContext *avctx, CeltFrame **f, int output_channels,
+                 int apply_phase_inv);
+
+void ff_celt_free(CeltFrame **f);
+
+void ff_celt_flush(CeltFrame *f);
+
+int ff_celt_decode_frame(CeltFrame *f, OpusRangeCoder *rc, float **output,
+                         int coded_channels, int frame_size, int startband, int endband);
+
+/* Encode or decode CELT bands */
+void ff_celt_quant_bands(CeltFrame *f, OpusRangeCoder *rc);
+
+/* Encode or decode CELT bitallocation */
+void ff_celt_bitalloc(CeltFrame *f, OpusRangeCoder *rc, int encode);
+
+#endif /* AVCODEC_OPUS_CELT_H */
diff --git a/libavcodec/opus/dec.c b/libavcodec/opus/dec.c
new file mode 100644
index 0000000000..6c59dc1f46
--- /dev/null
+++ b/libavcodec/opus/dec.c
@@ -0,0 +1,784 @@
+/*
+ * Opus decoder
+ * Copyright (c) 2012 Andrew D'Addesio
+ * Copyright (c) 2013-2014 Mozilla Corporation
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Opus decoder
+ * @author Andrew D'Addesio, Anton Khirnov
+ *
+ * Codec homepage: http://opus-codec.org/
+ * Specification: http://tools.ietf.org/html/rfc6716
+ * Ogg Opus specification: https://tools.ietf.org/html/draft-ietf-codec-oggopus-03
+ *
+ * Ogg-contained .opus files can be produced with opus-tools:
+ * http://git.xiph.org/?p=opus-tools.git
+ */
+
+#include <stdint.h>
+
+#include "libavutil/attributes.h"
+#include "libavutil/audio_fifo.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/ffmath.h"
+#include "libavutil/float_dsp.h"
+#include "libavutil/frame.h"
+#include "libavutil/mem.h"
+#include "libavutil/mem_internal.h"
+#include "libavutil/opt.h"
+
+#include "libswresample/swresample.h"
+
+#include "avcodec.h"
+#include "codec_internal.h"
+#include "decode.h"
+#include "opus.h"
+#include "tab.h"
+#include "celt.h"
+#include "parse.h"
+#include "rc.h"
+#include "silk.h"
+
+static const uint16_t silk_frame_duration_ms[16] = {
+    10, 20, 40, 60,
+    10, 20, 40, 60,
+    10, 20, 40, 60,
+    10, 20,
+    10, 20,
+};
+
+/* number of samples of silence to feed to the resampler
+ * at the beginning */
+static const int silk_resample_delay[] = {
+    4, 8, 11, 11, 11
+};
+
+typedef struct OpusStreamContext {
+    AVCodecContext *avctx;
+    int output_channels;
+
+    /* number of decoded samples for this stream */
+    int decoded_samples;
+    /* current output buffers for this stream */
+    float *out[2];
+    int out_size;
+    /* Buffer with samples from this stream for synchronizing
+     * the streams when they have different resampling delays */
+    AVAudioFifo *sync_buffer;
+
+    OpusRangeCoder rc;
+    OpusRangeCoder redundancy_rc;
+    SilkContext *silk;
+    CeltFrame *celt;
+    AVFloatDSPContext *fdsp;
+
+    float silk_buf[2][960];
+    float *silk_output[2];
+    DECLARE_ALIGNED(32, float, celt_buf)[2][960];
+    float *celt_output[2];
+
+    DECLARE_ALIGNED(32, float, redundancy_buf)[2][960];
+    float *redundancy_output[2];
+
+    /* buffers for the next samples to be decoded */
+    float *cur_out[2];
+    int remaining_out_size;
+
+    float *out_dummy;
+    int    out_dummy_allocated_size;
+
+    SwrContext *swr;
+    AVAudioFifo *celt_delay;
+    int silk_samplerate;
+    /* number of samples we still want to get from the resampler */
+    int delayed_samples;
+
+    OpusPacket packet;
+
+    int redundancy_idx;
+} OpusStreamContext;
+
+typedef struct OpusContext {
+    AVClass *av_class;
+
+    struct OpusStreamContext *streams;
+    int apply_phase_inv;
+
+    AVFloatDSPContext *fdsp;
+    float   gain;
+
+    OpusParseContext p;
+} OpusContext;
+
+static int get_silk_samplerate(int config)
+{
+    if (config < 4)
+        return 8000;
+    else if (config < 8)
+        return 12000;
+    return 16000;
+}
+
+static void opus_fade(float *out,
+                      const float *in1, const float *in2,
+                      const float *window, int len)
+{
+    int i;
+    for (i = 0; i < len; i++)
+        out[i] = in2[i] * window[i] + in1[i] * (1.0 - window[i]);
+}
+
+static int opus_flush_resample(OpusStreamContext *s, int nb_samples)
+{
+    int celt_size = av_audio_fifo_size(s->celt_delay);
+    int ret, i;
+    ret = swr_convert(s->swr,
+                      (uint8_t**)s->cur_out, nb_samples,
+                      NULL, 0);
+    if (ret < 0)
+        return ret;
+    else if (ret != nb_samples) {
+        av_log(s->avctx, AV_LOG_ERROR, "Wrong number of flushed samples: %d\n",
+               ret);
+        return AVERROR_BUG;
+    }
+
+    if (celt_size) {
+        if (celt_size != nb_samples) {
+            av_log(s->avctx, AV_LOG_ERROR, "Wrong number of CELT delay samples.\n");
+            return AVERROR_BUG;
+        }
+        av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, nb_samples);
+        for (i = 0; i < s->output_channels; i++) {
+            s->fdsp->vector_fmac_scalar(s->cur_out[i],
+                                        s->celt_output[i], 1.0,
+                                        nb_samples);
+        }
+    }
+
+    if (s->redundancy_idx) {
+        for (i = 0; i < s->output_channels; i++)
+            opus_fade(s->cur_out[i], s->cur_out[i],
+                      s->redundancy_output[i] + 120 + s->redundancy_idx,
+                      ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
+        s->redundancy_idx = 0;
+    }
+
+    s->cur_out[0]         += nb_samples;
+    s->cur_out[1]         += nb_samples;
+    s->remaining_out_size -= nb_samples * sizeof(float);
+
+    return 0;
+}
+
+static int opus_init_resample(OpusStreamContext *s)
+{
+    static const float delay[16] = { 0.0 };
+    const uint8_t *delayptr[2] = { (uint8_t*)delay, (uint8_t*)delay };
+    int ret;
+
+    av_opt_set_int(s->swr, "in_sample_rate", s->silk_samplerate, 0);
+    ret = swr_init(s->swr);
+    if (ret < 0) {
+        av_log(s->avctx, AV_LOG_ERROR, "Error opening the resampler.\n");
+        return ret;
+    }
+
+    ret = swr_convert(s->swr,
+                      NULL, 0,
+                      delayptr, silk_resample_delay[s->packet.bandwidth]);
+    if (ret < 0) {
+        av_log(s->avctx, AV_LOG_ERROR,
+               "Error feeding initial silence to the resampler.\n");
+        return ret;
+    }
+
+    return 0;
+}
+
+static int opus_decode_redundancy(OpusStreamContext *s, const uint8_t *data, int size)
+{
+    int ret = ff_opus_rc_dec_init(&s->redundancy_rc, data, size);
+    if (ret < 0)
+        goto fail;
+    ff_opus_rc_dec_raw_init(&s->redundancy_rc, data + size, size);
+
+    ret = ff_celt_decode_frame(s->celt, &s->redundancy_rc,
+                               s->redundancy_output,
+                               s->packet.stereo + 1, 240,
+                               0, ff_celt_band_end[s->packet.bandwidth]);
+    if (ret < 0)
+        goto fail;
+
+    return 0;
+fail:
+    av_log(s->avctx, AV_LOG_ERROR, "Error decoding the redundancy frame.\n");
+    return ret;
+}
+
+static int opus_decode_frame(OpusStreamContext *s, const uint8_t *data, int size)
+{
+    int samples    = s->packet.frame_duration;
+    int redundancy = 0;
+    int redundancy_size, redundancy_pos;
+    int ret, i, consumed;
+    int delayed_samples = s->delayed_samples;
+
+    ret = ff_opus_rc_dec_init(&s->rc, data, size);
+    if (ret < 0)
+        return ret;
+
+    /* decode the silk frame */
+    if (s->packet.mode == OPUS_MODE_SILK || s->packet.mode == OPUS_MODE_HYBRID) {
+        if (!swr_is_initialized(s->swr)) {
+            ret = opus_init_resample(s);
+            if (ret < 0)
+                return ret;
+        }
+
+        samples = ff_silk_decode_superframe(s->silk, &s->rc, s->silk_output,
+                                            FFMIN(s->packet.bandwidth, OPUS_BANDWIDTH_WIDEBAND),
+                                            s->packet.stereo + 1,
+                                            silk_frame_duration_ms[s->packet.config]);
+        if (samples < 0) {
+            av_log(s->avctx, AV_LOG_ERROR, "Error decoding a SILK frame.\n");
+            return samples;
+        }
+        samples = swr_convert(s->swr,
+                              (uint8_t**)s->cur_out, s->packet.frame_duration,
+                              (const uint8_t**)s->silk_output, samples);
+        if (samples < 0) {
+            av_log(s->avctx, AV_LOG_ERROR, "Error resampling SILK data.\n");
+            return samples;
+        }
+        av_assert2((samples & 7) == 0);
+        s->delayed_samples += s->packet.frame_duration - samples;
+    } else
+        ff_silk_flush(s->silk);
+
+    // decode redundancy information
+    consumed = opus_rc_tell(&s->rc);
+    if (s->packet.mode == OPUS_MODE_HYBRID && consumed + 37 <= size * 8)
+        redundancy = ff_opus_rc_dec_log(&s->rc, 12);
+    else if (s->packet.mode == OPUS_MODE_SILK && consumed + 17 <= size * 8)
+        redundancy = 1;
+
+    if (redundancy) {
+        redundancy_pos = ff_opus_rc_dec_log(&s->rc, 1);
+
+        if (s->packet.mode == OPUS_MODE_HYBRID)
+            redundancy_size = ff_opus_rc_dec_uint(&s->rc, 256) + 2;
+        else
+            redundancy_size = size - (consumed + 7) / 8;
+        size -= redundancy_size;
+        if (size < 0) {
+            av_log(s->avctx, AV_LOG_ERROR, "Invalid redundancy frame size.\n");
+            return AVERROR_INVALIDDATA;
+        }
+
+        if (redundancy_pos) {
+            ret = opus_decode_redundancy(s, data + size, redundancy_size);
+            if (ret < 0)
+                return ret;
+            ff_celt_flush(s->celt);
+        }
+    }
+
+    /* decode the CELT frame */
+    if (s->packet.mode == OPUS_MODE_CELT || s->packet.mode == OPUS_MODE_HYBRID) {
+        float *out_tmp[2] = { s->cur_out[0], s->cur_out[1] };
+        float **dst = (s->packet.mode == OPUS_MODE_CELT) ?
+                      out_tmp : s->celt_output;
+        int celt_output_samples = samples;
+        int delay_samples = av_audio_fifo_size(s->celt_delay);
+
+        if (delay_samples) {
+            if (s->packet.mode == OPUS_MODE_HYBRID) {
+                av_audio_fifo_read(s->celt_delay, (void**)s->celt_output, delay_samples);
+
+                for (i = 0; i < s->output_channels; i++) {
+                    s->fdsp->vector_fmac_scalar(out_tmp[i], s->celt_output[i], 1.0,
+                                                delay_samples);
+                    out_tmp[i] += delay_samples;
+                }
+                celt_output_samples -= delay_samples;
+            } else {
+                av_log(s->avctx, AV_LOG_WARNING,
+                       "Spurious CELT delay samples present.\n");
+                av_audio_fifo_drain(s->celt_delay, delay_samples);
+                if (s->avctx->err_recognition & AV_EF_EXPLODE)
+                    return AVERROR_BUG;
+            }
+        }
+
+        ff_opus_rc_dec_raw_init(&s->rc, data + size, size);
+
+        ret = ff_celt_decode_frame(s->celt, &s->rc, dst,
+                                   s->packet.stereo + 1,
+                                   s->packet.frame_duration,
+                                   (s->packet.mode == OPUS_MODE_HYBRID) ? 17 : 0,
+                                   ff_celt_band_end[s->packet.bandwidth]);
+        if (ret < 0)
+            return ret;
+
+        if (s->packet.mode == OPUS_MODE_HYBRID) {
+            int celt_delay = s->packet.frame_duration - celt_output_samples;
+            void *delaybuf[2] = { s->celt_output[0] + celt_output_samples,
+                                  s->celt_output[1] + celt_output_samples };
+
+            for (i = 0; i < s->output_channels; i++) {
+                s->fdsp->vector_fmac_scalar(out_tmp[i],
+                                            s->celt_output[i], 1.0,
+                                            celt_output_samples);
+            }
+
+            ret = av_audio_fifo_write(s->celt_delay, delaybuf, celt_delay);
+            if (ret < 0)
+                return ret;
+        }
+    } else
+        ff_celt_flush(s->celt);
+
+    if (s->redundancy_idx) {
+        for (i = 0; i < s->output_channels; i++)
+            opus_fade(s->cur_out[i], s->cur_out[i],
+                      s->redundancy_output[i] + 120 + s->redundancy_idx,
+                      ff_celt_window2 + s->redundancy_idx, 120 - s->redundancy_idx);
+        s->redundancy_idx = 0;
+    }
+    if (redundancy) {
+        if (!redundancy_pos) {
+            ff_celt_flush(s->celt);
+            ret = opus_decode_redundancy(s, data + size, redundancy_size);
+            if (ret < 0)
+                return ret;
+
+            for (i = 0; i < s->output_channels; i++) {
+                opus_fade(s->cur_out[i] + samples - 120 + delayed_samples,
+                          s->cur_out[i] + samples - 120 + delayed_samples,
+                          s->redundancy_output[i] + 120,
+                          ff_celt_window2, 120 - delayed_samples);
+                if (delayed_samples)
+                    s->redundancy_idx = 120 - delayed_samples;
+            }
+        } else {
+            for (i = 0; i < s->output_channels; i++) {
+                memcpy(s->cur_out[i] + delayed_samples, s->redundancy_output[i], 120 * sizeof(float));
+                opus_fade(s->cur_out[i] + 120 + delayed_samples,
+                          s->redundancy_output[i] + 120,
+                          s->cur_out[i] + 120 + delayed_samples,
+                          ff_celt_window2, 120);
+            }
+        }
+    }
+
+    return samples;
+}
+
+static int opus_decode_subpacket(OpusStreamContext *s,
+                                 const uint8_t *buf, int buf_size,
+                                 int nb_samples)
+{
+    int output_samples = 0;
+    int flush_needed   = 0;
+    int i, j, ret;
+
+    s->cur_out[0]         = s->out[0];
+    s->cur_out[1]         = s->out[1];
+    s->remaining_out_size = s->out_size;
+
+    /* check if we need to flush the resampler */
+    if (swr_is_initialized(s->swr)) {
+        if (buf) {
+            int64_t cur_samplerate;
+            av_opt_get_int(s->swr, "in_sample_rate", 0, &cur_samplerate);
+            flush_needed = (s->packet.mode == OPUS_MODE_CELT) || (cur_samplerate != s->silk_samplerate);
+        } else {
+            flush_needed = !!s->delayed_samples;
+        }
+    }
+
+    if (!buf && !flush_needed)
+        return 0;
+
+    /* use dummy output buffers if the channel is not mapped to anything */
+    if (!s->cur_out[0] ||
+        (s->output_channels == 2 && !s->cur_out[1])) {
+        av_fast_malloc(&s->out_dummy, &s->out_dummy_allocated_size,
+                       s->remaining_out_size);
+        if (!s->out_dummy)
+            return AVERROR(ENOMEM);
+        if (!s->cur_out[0])
+            s->cur_out[0] = s->out_dummy;
+        if (!s->cur_out[1])
+            s->cur_out[1] = s->out_dummy;
+    }
+
+    /* flush the resampler if necessary */
+    if (flush_needed) {
+        ret = opus_flush_resample(s, s->delayed_samples);
+        if (ret < 0) {
+            av_log(s->avctx, AV_LOG_ERROR, "Error flushing the resampler.\n");
+            return ret;
+        }
+        swr_close(s->swr);
+        output_samples += s->delayed_samples;
+        s->delayed_samples = 0;
+
+        if (!buf)
+            goto finish;
+    }
+
+    /* decode all the frames in the packet */
+    for (i = 0; i < s->packet.frame_count; i++) {
+        int size = s->packet.frame_size[i];
+        int samples = opus_decode_frame(s, buf + s->packet.frame_offset[i], size);
+
+        if (samples < 0) {
+            av_log(s->avctx, AV_LOG_ERROR, "Error decoding an Opus frame.\n");
+            if (s->avctx->err_recognition & AV_EF_EXPLODE)
+                return samples;
+
+            for (j = 0; j < s->output_channels; j++)
+                memset(s->cur_out[j], 0, s->packet.frame_duration * sizeof(float));
+            samples = s->packet.frame_duration;
+        }
+        output_samples += samples;
+
+        for (j = 0; j < s->output_channels; j++)
+            s->cur_out[j] += samples;
+        s->remaining_out_size -= samples * sizeof(float);
+    }
+
+finish:
+    s->cur_out[0] = s->cur_out[1] = NULL;
+    s->remaining_out_size = 0;
+
+    return output_samples;
+}
+
+static int opus_decode_packet(AVCodecContext *avctx, AVFrame *frame,
+                              int *got_frame_ptr, AVPacket *avpkt)
+{
+    OpusContext *c      = avctx->priv_data;
+    const uint8_t *buf  = avpkt->data;
+    int buf_size        = avpkt->size;
+    int coded_samples   = 0;
+    int decoded_samples = INT_MAX;
+    int delayed_samples = 0;
+    int i, ret;
+
+    /* calculate the number of delayed samples */
+    for (int i = 0; i < c->p.nb_streams; i++) {
+        OpusStreamContext *s = &c->streams[i];
+        s->out[0] =
+        s->out[1] = NULL;
+        delayed_samples = FFMAX(delayed_samples,
+                                s->delayed_samples + av_audio_fifo_size(s->sync_buffer));
+    }
+
+    /* decode the header of the first sub-packet to find out the sample count */
+    if (buf) {
+        OpusPacket *pkt = &c->streams[0].packet;
+        ret = ff_opus_parse_packet(pkt, buf, buf_size, c->p.nb_streams > 1);
+        if (ret < 0) {
+            av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
+            return ret;
+        }
+        coded_samples += pkt->frame_count * pkt->frame_duration;
+        c->streams[0].silk_samplerate = get_silk_samplerate(pkt->config);
+    }
+
+    frame->nb_samples = coded_samples + delayed_samples;
+
+    /* no input or buffered data => nothing to do */
+    if (!frame->nb_samples) {
+        *got_frame_ptr = 0;
+        return 0;
+    }
+
+    /* setup the data buffers */
+    ret = ff_get_buffer(avctx, frame, 0);
+    if (ret < 0)
+        return ret;
+    frame->nb_samples = 0;
+
+    for (i = 0; i < avctx->ch_layout.nb_channels; i++) {
+        ChannelMap *map = &c->p.channel_maps[i];
+        if (!map->copy)
+            c->streams[map->stream_idx].out[map->channel_idx] = (float*)frame->extended_data[i];
+    }
+
+    /* read the data from the sync buffers */
+    for (int i = 0; i < c->p.nb_streams; i++) {
+        OpusStreamContext *s = &c->streams[i];
+        float          **out = s->out;
+        int sync_size = av_audio_fifo_size(s->sync_buffer);
+
+        float sync_dummy[32];
+        int out_dummy = (!out[0]) | ((!out[1]) << 1);
+
+        if (!out[0])
+            out[0] = sync_dummy;
+        if (!out[1])
+            out[1] = sync_dummy;
+        if (out_dummy && sync_size > FF_ARRAY_ELEMS(sync_dummy))
+            return AVERROR_BUG;
+
+        ret = av_audio_fifo_read(s->sync_buffer, (void**)out, sync_size);
+        if (ret < 0)
+            return ret;
+
+        if (out_dummy & 1)
+            out[0] = NULL;
+        else
+            out[0] += ret;
+        if (out_dummy & 2)
+            out[1] = NULL;
+        else
+            out[1] += ret;
+
+        s->out_size = frame->linesize[0] - ret * sizeof(float);
+    }
+
+    /* decode each sub-packet */
+    for (int i = 0; i < c->p.nb_streams; i++) {
+        OpusStreamContext *s = &c->streams[i];
+
+        if (i && buf) {
+            ret = ff_opus_parse_packet(&s->packet, buf, buf_size, i != c->p.nb_streams - 1);
+            if (ret < 0) {
+                av_log(avctx, AV_LOG_ERROR, "Error parsing the packet header.\n");
+                return ret;
+            }
+            if (coded_samples != s->packet.frame_count * s->packet.frame_duration) {
+                av_log(avctx, AV_LOG_ERROR,
+                       "Mismatching coded sample count in substream %d.\n", i);
+                return AVERROR_INVALIDDATA;
+            }
+
+            s->silk_samplerate = get_silk_samplerate(s->packet.config);
+        }
+
+        ret = opus_decode_subpacket(&c->streams[i], buf, s->packet.data_size,
+                                    coded_samples);
+        if (ret < 0)
+            return ret;
+        s->decoded_samples = ret;
+        decoded_samples       = FFMIN(decoded_samples, ret);
+
+        buf      += s->packet.packet_size;
+        buf_size -= s->packet.packet_size;
+    }
+
+    /* buffer the extra samples */
+    for (int i = 0; i < c->p.nb_streams; i++) {
+        OpusStreamContext *s = &c->streams[i];
+        int   buffer_samples = s->decoded_samples - decoded_samples;
+        if (buffer_samples) {
+            float *buf[2] = { s->out[0] ? s->out[0] : (float*)frame->extended_data[0],
+                              s->out[1] ? s->out[1] : (float*)frame->extended_data[0] };
+            buf[0] += decoded_samples;
+            buf[1] += decoded_samples;
+            ret = av_audio_fifo_write(s->sync_buffer, (void**)buf, buffer_samples);
+            if (ret < 0)
+                return ret;
+        }
+    }
+
+    for (i = 0; i < avctx->ch_layout.nb_channels; i++) {
+        ChannelMap *map = &c->p.channel_maps[i];
+
+        /* handle copied channels */
+        if (map->copy) {
+            memcpy(frame->extended_data[i],
+                   frame->extended_data[map->copy_idx],
+                   frame->linesize[0]);
+        } else if (map->silence) {
+            memset(frame->extended_data[i], 0, frame->linesize[0]);
+        }
+
+        if (c->p.gain_i && decoded_samples > 0) {
+            c->fdsp->vector_fmul_scalar((float*)frame->extended_data[i],
+                                       (float*)frame->extended_data[i],
+                                       c->gain, FFALIGN(decoded_samples, 8));
+        }
+    }
+
+    frame->nb_samples = decoded_samples;
+    *got_frame_ptr    = !!decoded_samples;
+
+    return avpkt->size;
+}
+
+static av_cold void opus_decode_flush(AVCodecContext *ctx)
+{
+    OpusContext *c = ctx->priv_data;
+
+    for (int i = 0; i < c->p.nb_streams; i++) {
+        OpusStreamContext *s = &c->streams[i];
+
+        memset(&s->packet, 0, sizeof(s->packet));
+        s->delayed_samples = 0;
+
+        av_audio_fifo_drain(s->celt_delay, av_audio_fifo_size(s->celt_delay));
+        swr_close(s->swr);
+
+        av_audio_fifo_drain(s->sync_buffer, av_audio_fifo_size(s->sync_buffer));
+
+        ff_silk_flush(s->silk);
+        ff_celt_flush(s->celt);
+    }
+}
+
+static av_cold int opus_decode_close(AVCodecContext *avctx)
+{
+    OpusContext *c = avctx->priv_data;
+
+    for (int i = 0; i < c->p.nb_streams; i++) {
+        OpusStreamContext *s = &c->streams[i];
+
+        ff_silk_free(&s->silk);
+        ff_celt_free(&s->celt);
+
+        av_freep(&s->out_dummy);
+        s->out_dummy_allocated_size = 0;
+
+        av_audio_fifo_free(s->sync_buffer);
+        av_audio_fifo_free(s->celt_delay);
+        swr_free(&s->swr);
+    }
+
+    av_freep(&c->streams);
+
+    c->p.nb_streams = 0;
+
+    av_freep(&c->p.channel_maps);
+    av_freep(&c->fdsp);
+
+    return 0;
+}
+
+static av_cold int opus_decode_init(AVCodecContext *avctx)
+{
+    OpusContext *c = avctx->priv_data;
+    int ret;
+
+    avctx->sample_fmt  = AV_SAMPLE_FMT_FLTP;
+    avctx->sample_rate = 48000;
+
+    c->fdsp = avpriv_float_dsp_alloc(0);
+    if (!c->fdsp)
+        return AVERROR(ENOMEM);
+
+    /* find out the channel configuration */
+    ret = ff_opus_parse_extradata(avctx, &c->p);
+    if (ret < 0)
+        return ret;
+    if (c->p.gain_i)
+        c->gain = ff_exp10(c->p.gain_i / (20.0 * 256));
+
+    /* allocate and init each independent decoder */
+    c->streams = av_calloc(c->p.nb_streams, sizeof(*c->streams));
+    if (!c->streams) {
+        c->p.nb_streams = 0;
+        return AVERROR(ENOMEM);
+    }
+
+    for (int i = 0; i < c->p.nb_streams; i++) {
+        OpusStreamContext *s = &c->streams[i];
+        AVChannelLayout layout;
+
+        s->output_channels = (i < c->p.nb_stereo_streams) ? 2 : 1;
+
+        s->avctx = avctx;
+
+        for (int j = 0; j < s->output_channels; j++) {
+            s->silk_output[j]       = s->silk_buf[j];
+            s->celt_output[j]       = s->celt_buf[j];
+            s->redundancy_output[j] = s->redundancy_buf[j];
+        }
+
+        s->fdsp = c->fdsp;
+
+        s->swr =swr_alloc();
+        if (!s->swr)
+            return AVERROR(ENOMEM);
+
+        layout = (s->output_channels == 1) ? (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO :
+                                             (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO;
+        av_opt_set_int(s->swr, "in_sample_fmt",      avctx->sample_fmt,  0);
+        av_opt_set_int(s->swr, "out_sample_fmt",     avctx->sample_fmt,  0);
+        av_opt_set_chlayout(s->swr, "in_chlayout",   &layout,            0);
+        av_opt_set_chlayout(s->swr, "out_chlayout",  &layout,            0);
+        av_opt_set_int(s->swr, "out_sample_rate",    avctx->sample_rate, 0);
+        av_opt_set_int(s->swr, "filter_size",        16,                 0);
+
+        ret = ff_silk_init(avctx, &s->silk, s->output_channels);
+        if (ret < 0)
+            return ret;
+
+        ret = ff_celt_init(avctx, &s->celt, s->output_channels, c->apply_phase_inv);
+        if (ret < 0)
+            return ret;
+
+        s->celt_delay = av_audio_fifo_alloc(avctx->sample_fmt,
+                                            s->output_channels, 1024);
+        if (!s->celt_delay)
+            return AVERROR(ENOMEM);
+
+        s->sync_buffer = av_audio_fifo_alloc(avctx->sample_fmt,
+                                             s->output_channels, 32);
+        if (!s->sync_buffer)
+            return AVERROR(ENOMEM);
+    }
+
+    return 0;
+}
+
+#define OFFSET(x) offsetof(OpusContext, x)
+#define AD AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_DECODING_PARAM
+static const AVOption opus_options[] = {
+    { "apply_phase_inv", "Apply intensity stereo phase inversion", OFFSET(apply_phase_inv), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, AD },
+    { NULL },
+};
+
+static const AVClass opus_class = {
+    .class_name = "Opus Decoder",
+    .item_name  = av_default_item_name,
+    .option     = opus_options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
+const FFCodec ff_opus_decoder = {
+    .p.name          = "opus",
+    CODEC_LONG_NAME("Opus"),
+    .p.priv_class    = &opus_class,
+    .p.type          = AVMEDIA_TYPE_AUDIO,
+    .p.id            = AV_CODEC_ID_OPUS,
+    .priv_data_size  = sizeof(OpusContext),
+    .init            = opus_decode_init,
+    .close           = opus_decode_close,
+    FF_CODEC_DECODE_CB(opus_decode_packet),
+    .flush           = opus_decode_flush,
+    .p.capabilities  = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY | AV_CODEC_CAP_CHANNEL_CONF,
+    .caps_internal   = FF_CODEC_CAP_INIT_CLEANUP,
+};
diff --git a/libavcodec/opus/dec_celt.c b/libavcodec/opus/dec_celt.c
new file mode 100644
index 0000000000..3feb4a4e47
--- /dev/null
+++ b/libavcodec/opus/dec_celt.c
@@ -0,0 +1,589 @@
+/*
+ * Copyright (c) 2012 Andrew D'Addesio
+ * Copyright (c) 2013-2014 Mozilla Corporation
+ * Copyright (c) 2016 Rostislav Pehlivanov <atomnuker@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Opus CELT decoder
+ */
+
+#include <float.h>
+
+#include "libavutil/mem.h"
+#include "celt.h"
+#include "tab.h"
+#include "pvq.h"
+
+/* Use the 2D z-transform to apply prediction in both the time domain (alpha)
+ * and the frequency domain (beta) */
+static void celt_decode_coarse_energy(CeltFrame *f, OpusRangeCoder *rc)
+{
+    int i, j;
+    float prev[2] = { 0 };
+    float alpha = ff_celt_alpha_coef[f->size];
+    float beta  = ff_celt_beta_coef[f->size];
+    const uint8_t *model = ff_celt_coarse_energy_dist[f->size][0];
+
+    /* intra frame */
+    if (opus_rc_tell(rc) + 3 <= f->framebits && ff_opus_rc_dec_log(rc, 3)) {
+        alpha = 0.0f;
+        beta  = 1.0f - (4915.0f/32768.0f);
+        model = ff_celt_coarse_energy_dist[f->size][1];
+    }
+
+    for (i = 0; i < CELT_MAX_BANDS; i++) {
+        for (j = 0; j < f->channels; j++) {
+            CeltBlock *block = &f->block[j];
+            float value;
+            int available;
+
+            if (i < f->start_band || i >= f->end_band) {
+                block->energy[i] = 0.0;
+                continue;
+            }
+
+            available = f->framebits - opus_rc_tell(rc);
+            if (available >= 15) {
+                /* decode using a Laplace distribution */
+                int k = FFMIN(i, 20) << 1;
+                value = ff_opus_rc_dec_laplace(rc, model[k] << 7, model[k+1] << 6);
+            } else if (available >= 2) {
+                int x = ff_opus_rc_dec_cdf(rc, ff_celt_model_energy_small);
+                value = (x>>1) ^ -(x&1);
+            } else if (available >= 1) {
+                value = -(float)ff_opus_rc_dec_log(rc, 1);
+            } else value = -1;
+
+            block->energy[i] = FFMAX(-9.0f, block->energy[i]) * alpha + prev[j] + value;
+            prev[j] += beta * value;
+        }
+    }
+}
+
+static void celt_decode_fine_energy(CeltFrame *f, OpusRangeCoder *rc)
+{
+    int i;
+    for (i = f->start_band; i < f->end_band; i++) {
+        int j;
+        if (!f->fine_bits[i])
+            continue;
+
+        for (j = 0; j < f->channels; j++) {
+            CeltBlock *block = &f->block[j];
+            int q2;
+            float offset;
+            q2 = ff_opus_rc_get_raw(rc, f->fine_bits[i]);
+            offset = (q2 + 0.5f) * (1 << (14 - f->fine_bits[i])) / 16384.0f - 0.5f;
+            block->energy[i] += offset;
+        }
+    }
+}
+
+static void celt_decode_final_energy(CeltFrame *f, OpusRangeCoder *rc)
+{
+    int priority, i, j;
+    int bits_left = f->framebits - opus_rc_tell(rc);
+
+    for (priority = 0; priority < 2; priority++) {
+        for (i = f->start_band; i < f->end_band && bits_left >= f->channels; i++) {
+            if (f->fine_priority[i] != priority || f->fine_bits[i] >= CELT_MAX_FINE_BITS)
+                continue;
+
+            for (j = 0; j < f->channels; j++) {
+                int q2;
+                float offset;
+                q2 = ff_opus_rc_get_raw(rc, 1);
+                offset = (q2 - 0.5f) * (1 << (14 - f->fine_bits[i] - 1)) / 16384.0f;
+                f->block[j].energy[i] += offset;
+                bits_left--;
+            }
+        }
+    }
+}
+
+static void celt_decode_tf_changes(CeltFrame *f, OpusRangeCoder *rc)
+{
+    int i, diff = 0, tf_select = 0, tf_changed = 0, tf_select_bit;
+    int consumed, bits = f->transient ? 2 : 4;
+
+    consumed = opus_rc_tell(rc);
+    tf_select_bit = (f->size != 0 && consumed+bits+1 <= f->framebits);
+
+    for (i = f->start_band; i < f->end_band; i++) {
+        if (consumed+bits+tf_select_bit <= f->framebits) {
+            diff ^= ff_opus_rc_dec_log(rc, bits);
+            consumed = opus_rc_tell(rc);
+            tf_changed |= diff;
+        }
+        f->tf_change[i] = diff;
+        bits = f->transient ? 4 : 5;
+    }
+
+    if (tf_select_bit && ff_celt_tf_select[f->size][f->transient][0][tf_changed] !=
+                         ff_celt_tf_select[f->size][f->transient][1][tf_changed])
+        tf_select = ff_opus_rc_dec_log(rc, 1);
+
+    for (i = f->start_band; i < f->end_band; i++) {
+        f->tf_change[i] = ff_celt_tf_select[f->size][f->transient][tf_select][f->tf_change[i]];
+    }
+}
+
+static void celt_denormalize(CeltFrame *f, CeltBlock *block, float *data)
+{
+    int i, j;
+
+    for (i = f->start_band; i < f->end_band; i++) {
+        float *dst = data + (ff_celt_freq_bands[i] << f->size);
+        float log_norm = block->energy[i] + ff_celt_mean_energy[i];
+        float norm = exp2f(FFMIN(log_norm, 32.0f));
+
+        for (j = 0; j < ff_celt_freq_range[i] << f->size; j++)
+            dst[j] *= norm;
+    }
+}
+
+static void celt_postfilter_apply_transition(CeltBlock *block, float *data)
+{
+    const int T0 = block->pf_period_old;
+    const int T1 = block->pf_period;
+
+    float g00, g01, g02;
+    float g10, g11, g12;
+
+    float x0, x1, x2, x3, x4;
+
+    int i;
+
+    if (block->pf_gains[0]     == 0.0 &&
+        block->pf_gains_old[0] == 0.0)
+        return;
+
+    g00 = block->pf_gains_old[0];
+    g01 = block->pf_gains_old[1];
+    g02 = block->pf_gains_old[2];
+    g10 = block->pf_gains[0];
+    g11 = block->pf_gains[1];
+    g12 = block->pf_gains[2];
+
+    x1 = data[-T1 + 1];
+    x2 = data[-T1];
+    x3 = data[-T1 - 1];
+    x4 = data[-T1 - 2];
+
+    for (i = 0; i < CELT_OVERLAP; i++) {
+        float w = ff_celt_window2[i];
+        x0 = data[i - T1 + 2];
+
+        data[i] +=  (1.0 - w) * g00 * data[i - T0]                          +
+                    (1.0 - w) * g01 * (data[i - T0 - 1] + data[i - T0 + 1]) +
+                    (1.0 - w) * g02 * (data[i - T0 - 2] + data[i - T0 + 2]) +
+                    w         * g10 * x2                                    +
+                    w         * g11 * (x1 + x3)                             +
+                    w         * g12 * (x0 + x4);
+        x4 = x3;
+        x3 = x2;
+        x2 = x1;
+        x1 = x0;
+    }
+}
+
+static void celt_postfilter(CeltFrame *f, CeltBlock *block)
+{
+    int len = f->blocksize * f->blocks;
+    const int filter_len = len - 2 * CELT_OVERLAP;
+
+    celt_postfilter_apply_transition(block, block->buf + 1024);
+
+    block->pf_period_old = block->pf_period;
+    memcpy(block->pf_gains_old, block->pf_gains, sizeof(block->pf_gains));
+
+    block->pf_period = block->pf_period_new;
+    memcpy(block->pf_gains, block->pf_gains_new, sizeof(block->pf_gains));
+
+    if (len > CELT_OVERLAP) {
+        celt_postfilter_apply_transition(block, block->buf + 1024 + CELT_OVERLAP);
+
+        if (block->pf_gains[0] > FLT_EPSILON && filter_len > 0)
+            f->opusdsp.postfilter(block->buf + 1024 + 2 * CELT_OVERLAP,
+                                  block->pf_period, block->pf_gains,
+                                  filter_len);
+
+        block->pf_period_old = block->pf_period;
+        memcpy(block->pf_gains_old, block->pf_gains, sizeof(block->pf_gains));
+    }
+
+    memmove(block->buf, block->buf + len, (1024 + CELT_OVERLAP / 2) * sizeof(float));
+}
+
+static int parse_postfilter(CeltFrame *f, OpusRangeCoder *rc, int consumed)
+{
+    int i;
+
+    memset(f->block[0].pf_gains_new, 0, sizeof(f->block[0].pf_gains_new));
+    memset(f->block[1].pf_gains_new, 0, sizeof(f->block[1].pf_gains_new));
+
+    if (f->start_band == 0 && consumed + 16 <= f->framebits) {
+        int has_postfilter = ff_opus_rc_dec_log(rc, 1);
+        if (has_postfilter) {
+            float gain;
+            int tapset, octave, period;
+
+            octave = ff_opus_rc_dec_uint(rc, 6);
+            period = (16 << octave) + ff_opus_rc_get_raw(rc, 4 + octave) - 1;
+            gain   = 0.09375f * (ff_opus_rc_get_raw(rc, 3) + 1);
+            tapset = (opus_rc_tell(rc) + 2 <= f->framebits) ?
+                     ff_opus_rc_dec_cdf(rc, ff_celt_model_tapset) : 0;
+
+            for (i = 0; i < 2; i++) {
+                CeltBlock *block = &f->block[i];
+
+                block->pf_period_new = FFMAX(period, CELT_POSTFILTER_MINPERIOD);
+                block->pf_gains_new[0] = gain * ff_celt_postfilter_taps[tapset][0];
+                block->pf_gains_new[1] = gain * ff_celt_postfilter_taps[tapset][1];
+                block->pf_gains_new[2] = gain * ff_celt_postfilter_taps[tapset][2];
+            }
+        }
+
+        consumed = opus_rc_tell(rc);
+    }
+
+    return consumed;
+}
+
+static void process_anticollapse(CeltFrame *f, CeltBlock *block, float *X)
+{
+    int i, j, k;
+
+    for (i = f->start_band; i < f->end_band; i++) {
+        int renormalize = 0;
+        float *xptr;
+        float prev[2];
+        float Ediff, r;
+        float thresh, sqrt_1;
+        int depth;
+
+        /* depth in 1/8 bits */
+        depth = (1 + f->pulses[i]) / (ff_celt_freq_range[i] << f->size);
+        thresh = exp2f(-1.0 - 0.125f * depth);
+        sqrt_1 = 1.0f / sqrtf(ff_celt_freq_range[i] << f->size);
+
+        xptr = X + (ff_celt_freq_bands[i] << f->size);
+
+        prev[0] = block->prev_energy[0][i];
+        prev[1] = block->prev_energy[1][i];
+        if (f->channels == 1) {
+            CeltBlock *block1 = &f->block[1];
+
+            prev[0] = FFMAX(prev[0], block1->prev_energy[0][i]);
+            prev[1] = FFMAX(prev[1], block1->prev_energy[1][i]);
+        }
+        Ediff = block->energy[i] - FFMIN(prev[0], prev[1]);
+        Ediff = FFMAX(0, Ediff);
+
+        /* r needs to be multiplied by 2 or 2*sqrt(2) depending on LM because
+        short blocks don't have the same energy as long */
+        r = exp2f(1 - Ediff);
+        if (f->size == 3)
+            r *= M_SQRT2;
+        r = FFMIN(thresh, r) * sqrt_1;
+        for (k = 0; k < 1 << f->size; k++) {
+            /* Detect collapse */
+            if (!(block->collapse_masks[i] & 1 << k)) {
+                /* Fill with noise */
+                for (j = 0; j < ff_celt_freq_range[i]; j++)
+                    xptr[(j << f->size) + k] = (celt_rng(f) & 0x8000) ? r : -r;
+                renormalize = 1;
+            }
+        }
+
+        /* We just added some energy, so we need to renormalize */
+        if (renormalize)
+            celt_renormalize_vector(xptr, ff_celt_freq_range[i] << f->size, 1.0f);
+    }
+}
+
+int ff_celt_decode_frame(CeltFrame *f, OpusRangeCoder *rc,
+                         float **output, int channels, int frame_size,
+                         int start_band,  int end_band)
+{
+    int i, j, downmix = 0;
+    int consumed;           // bits of entropy consumed thus far for this frame
+    AVTXContext *imdct;
+    av_tx_fn imdct_fn;
+
+    if (channels != 1 && channels != 2) {
+        av_log(f->avctx, AV_LOG_ERROR, "Invalid number of coded channels: %d\n",
+               channels);
+        return AVERROR_INVALIDDATA;
+    }
+    if (start_band < 0 || start_band > end_band || end_band > CELT_MAX_BANDS) {
+        av_log(f->avctx, AV_LOG_ERROR, "Invalid start/end band: %d %d\n",
+               start_band, end_band);
+        return AVERROR_INVALIDDATA;
+    }
+
+    f->silence        = 0;
+    f->transient      = 0;
+    f->anticollapse   = 0;
+    f->flushed        = 0;
+    f->channels       = channels;
+    f->start_band     = start_band;
+    f->end_band       = end_band;
+    f->framebits      = rc->rb.bytes * 8;
+
+    f->size = av_log2(frame_size / CELT_SHORT_BLOCKSIZE);
+    if (f->size > CELT_MAX_LOG_BLOCKS ||
+        frame_size != CELT_SHORT_BLOCKSIZE * (1 << f->size)) {
+        av_log(f->avctx, AV_LOG_ERROR, "Invalid CELT frame size: %d\n",
+               frame_size);
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (!f->output_channels)
+        f->output_channels = channels;
+
+    for (i = 0; i < f->channels; i++) {
+        memset(f->block[i].coeffs,         0, sizeof(f->block[i].coeffs));
+        memset(f->block[i].collapse_masks, 0, sizeof(f->block[i].collapse_masks));
+    }
+
+    consumed = opus_rc_tell(rc);
+
+    /* obtain silence flag */
+    if (consumed >= f->framebits)
+        f->silence = 1;
+    else if (consumed == 1)
+        f->silence = ff_opus_rc_dec_log(rc, 15);
+
+
+    if (f->silence) {
+        consumed = f->framebits;
+        rc->total_bits += f->framebits - opus_rc_tell(rc);
+    }
+
+    /* obtain post-filter options */
+    consumed = parse_postfilter(f, rc, consumed);
+
+    /* obtain transient flag */
+    if (f->size != 0 && consumed+3 <= f->framebits)
+        f->transient = ff_opus_rc_dec_log(rc, 3);
+
+    f->blocks    = f->transient ? 1 << f->size : 1;
+    f->blocksize = frame_size / f->blocks;
+
+    imdct = f->tx[f->transient ? 0 : f->size];
+    imdct_fn = f->tx_fn[f->transient ? 0 : f->size];
+
+    if (channels == 1) {
+        for (i = 0; i < CELT_MAX_BANDS; i++)
+            f->block[0].energy[i] = FFMAX(f->block[0].energy[i], f->block[1].energy[i]);
+    }
+
+    celt_decode_coarse_energy(f, rc);
+    celt_decode_tf_changes   (f, rc);
+    ff_celt_bitalloc         (f, rc, 0);
+    celt_decode_fine_energy  (f, rc);
+    ff_celt_quant_bands      (f, rc);
+
+    if (f->anticollapse_needed)
+        f->anticollapse = ff_opus_rc_get_raw(rc, 1);
+
+    celt_decode_final_energy(f, rc);
+
+    /* apply anti-collapse processing and denormalization to
+     * each coded channel */
+    for (i = 0; i < f->channels; i++) {
+        CeltBlock *block = &f->block[i];
+
+        if (f->anticollapse)
+            process_anticollapse(f, block, f->block[i].coeffs);
+
+        celt_denormalize(f, block, f->block[i].coeffs);
+    }
+
+    /* stereo -> mono downmix */
+    if (f->output_channels < f->channels) {
+        f->dsp->vector_fmac_scalar(f->block[0].coeffs, f->block[1].coeffs, 1.0, FFALIGN(frame_size, 16));
+        downmix = 1;
+    } else if (f->output_channels > f->channels)
+        memcpy(f->block[1].coeffs, f->block[0].coeffs, frame_size * sizeof(float));
+
+    if (f->silence) {
+        for (i = 0; i < 2; i++) {
+            CeltBlock *block = &f->block[i];
+
+            for (j = 0; j < FF_ARRAY_ELEMS(block->energy); j++)
+                block->energy[j] = CELT_ENERGY_SILENCE;
+        }
+        memset(f->block[0].coeffs, 0, sizeof(f->block[0].coeffs));
+        memset(f->block[1].coeffs, 0, sizeof(f->block[1].coeffs));
+    }
+
+    /* transform and output for each output channel */
+    for (i = 0; i < f->output_channels; i++) {
+        CeltBlock *block = &f->block[i];
+
+        /* iMDCT and overlap-add */
+        for (j = 0; j < f->blocks; j++) {
+            float *dst  = block->buf + 1024 + j * f->blocksize;
+
+            imdct_fn(imdct, dst + CELT_OVERLAP / 2, f->block[i].coeffs + j,
+                     sizeof(float)*f->blocks);
+            f->dsp->vector_fmul_window(dst, dst, dst + CELT_OVERLAP / 2,
+                                       ff_celt_window, CELT_OVERLAP / 2);
+        }
+
+        if (downmix)
+            f->dsp->vector_fmul_scalar(&block->buf[1024], &block->buf[1024], 0.5f, frame_size);
+
+        /* postfilter */
+        celt_postfilter(f, block);
+
+        /* deemphasis */
+        block->emph_coeff = f->opusdsp.deemphasis(output[i],
+                                                  &block->buf[1024 - frame_size],
+                                                  block->emph_coeff,
+                                                  ff_opus_deemph_weights,
+                                                  frame_size);
+    }
+
+    if (channels == 1)
+        memcpy(f->block[1].energy, f->block[0].energy, sizeof(f->block[0].energy));
+
+    for (i = 0; i < 2; i++ ) {
+        CeltBlock *block = &f->block[i];
+
+        if (!f->transient) {
+            memcpy(block->prev_energy[1], block->prev_energy[0], sizeof(block->prev_energy[0]));
+            memcpy(block->prev_energy[0], block->energy,         sizeof(block->prev_energy[0]));
+        } else {
+            for (j = 0; j < CELT_MAX_BANDS; j++)
+                block->prev_energy[0][j] = FFMIN(block->prev_energy[0][j], block->energy[j]);
+        }
+
+        for (j = 0; j < f->start_band; j++) {
+            block->prev_energy[0][j] = CELT_ENERGY_SILENCE;
+            block->energy[j]         = 0.0;
+        }
+        for (j = f->end_band; j < CELT_MAX_BANDS; j++) {
+            block->prev_energy[0][j] = CELT_ENERGY_SILENCE;
+            block->energy[j]         = 0.0;
+        }
+    }
+
+    f->seed = rc->range;
+
+    return 0;
+}
+
+void ff_celt_flush(CeltFrame *f)
+{
+    int i, j;
+
+    if (f->flushed)
+        return;
+
+    for (i = 0; i < 2; i++) {
+        CeltBlock *block = &f->block[i];
+
+        for (j = 0; j < CELT_MAX_BANDS; j++)
+            block->prev_energy[0][j] = block->prev_energy[1][j] = CELT_ENERGY_SILENCE;
+
+        memset(block->energy, 0, sizeof(block->energy));
+        memset(block->buf,    0, sizeof(block->buf));
+
+        memset(block->pf_gains,     0, sizeof(block->pf_gains));
+        memset(block->pf_gains_old, 0, sizeof(block->pf_gains_old));
+        memset(block->pf_gains_new, 0, sizeof(block->pf_gains_new));
+
+        /* libopus uses CELT_EMPH_COEFF on init, but 0 is better since there's
+         * a lesser discontinuity when seeking.
+         * The deemphasis functions differ from libopus in that they require
+         * an initial state divided by the coefficient. */
+        block->emph_coeff = 0.0f / ff_opus_deemph_weights[0];
+    }
+    f->seed = 0;
+
+    f->flushed = 1;
+}
+
+void ff_celt_free(CeltFrame **f)
+{
+    CeltFrame *frm = *f;
+    int i;
+
+    if (!frm)
+        return;
+
+    for (i = 0; i < FF_ARRAY_ELEMS(frm->tx); i++)
+        av_tx_uninit(&frm->tx[i]);
+
+    ff_celt_pvq_uninit(&frm->pvq);
+
+    av_freep(&frm->dsp);
+    av_freep(f);
+}
+
+int ff_celt_init(AVCodecContext *avctx, CeltFrame **f, int output_channels,
+                 int apply_phase_inv)
+{
+    CeltFrame *frm;
+    int i, ret;
+
+    if (output_channels != 1 && output_channels != 2) {
+        av_log(avctx, AV_LOG_ERROR, "Invalid number of output channels: %d\n",
+               output_channels);
+        return AVERROR(EINVAL);
+    }
+
+    frm = av_mallocz(sizeof(*frm));
+    if (!frm)
+        return AVERROR(ENOMEM);
+
+    frm->avctx           = avctx;
+    frm->output_channels = output_channels;
+    frm->apply_phase_inv = apply_phase_inv;
+
+    for (i = 0; i < FF_ARRAY_ELEMS(frm->tx); i++) {
+        const float scale = -1.0f/32768;
+        if ((ret = av_tx_init(&frm->tx[i], &frm->tx_fn[i], AV_TX_FLOAT_MDCT, 1, 15 << (i + 3), &scale, 0)) < 0)
+            goto fail;
+    }
+
+    if ((ret = ff_celt_pvq_init(&frm->pvq, 0)) < 0)
+        goto fail;
+
+    frm->dsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
+    if (!frm->dsp) {
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+
+    ff_opus_dsp_init(&frm->opusdsp);
+    ff_celt_flush(frm);
+
+    *f = frm;
+
+    return 0;
+fail:
+    ff_celt_free(&frm);
+    return ret;
+}
diff --git a/libavcodec/opus/dsp.c b/libavcodec/opus/dsp.c
new file mode 100644
index 0000000000..6cd76ceceb
--- /dev/null
+++ b/libavcodec/opus/dsp.c
@@ -0,0 +1,68 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "config.h"
+#include "libavutil/attributes.h"
+#include "libavutil/mem_internal.h"
+#include "dsp.h"
+
+static void postfilter_c(float *data, int period, float *gains, int len)
+{
+    const float g0 = gains[0];
+    const float g1 = gains[1];
+    const float g2 = gains[2];
+
+    float x4 = data[-period - 2];
+    float x3 = data[-period - 1];
+    float x2 = data[-period + 0];
+    float x1 = data[-period + 1];
+
+    for (int i = 0; i < len; i++) {
+        float x0 = data[i - period + 2];
+        data[i] += g0 * x2        +
+                   g1 * (x1 + x3) +
+                   g2 * (x0 + x4);
+        x4 = x3;
+        x3 = x2;
+        x2 = x1;
+        x1 = x0;
+    }
+}
+
+static float deemphasis_c(float *y, float *x, float coeff, const float *weights, int len)
+{
+    const float c = weights[0];
+    for (int i = 0; i < len; i++)
+        coeff = y[i] = x[i] + coeff*c;
+
+    return coeff;
+}
+
+av_cold void ff_opus_dsp_init(OpusDSP *ctx)
+{
+    ctx->postfilter = postfilter_c;
+    ctx->deemphasis = deemphasis_c;
+
+#if ARCH_AARCH64
+    ff_opus_dsp_init_aarch64(ctx);
+#elif ARCH_RISCV
+    ff_opus_dsp_init_riscv(ctx);
+#elif ARCH_X86
+    ff_opus_dsp_init_x86(ctx);
+#endif
+}
diff --git a/libavcodec/opus/dsp.h b/libavcodec/opus/dsp.h
new file mode 100644
index 0000000000..2179ee6953
--- /dev/null
+++ b/libavcodec/opus/dsp.h
@@ -0,0 +1,33 @@
+/*
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_OPUS_DSP_H
+#define AVCODEC_OPUS_DSP_H
+
+typedef struct OpusDSP {
+    void (*postfilter)(float *data, int period, float *gains, int len);
+    float (*deemphasis)(float *out, float *in, float coeff, const float *weights, int len);
+} OpusDSP;
+
+void ff_opus_dsp_init(OpusDSP *ctx);
+
+void ff_opus_dsp_init_x86(OpusDSP *ctx);
+void ff_opus_dsp_init_aarch64(OpusDSP *ctx);
+void ff_opus_dsp_init_riscv(OpusDSP *ctx);
+
+#endif /* AVCODEC_OPUS_DSP_H */
diff --git a/libavcodec/opus/enc.c b/libavcodec/opus/enc.c
new file mode 100644
index 0000000000..5398263119
--- /dev/null
+++ b/libavcodec/opus/enc.c
@@ -0,0 +1,754 @@
+/*
+ * Opus encoder
+ * Copyright (c) 2017 Rostislav Pehlivanov <atomnuker@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <float.h>
+
+#include "encode.h"
+#include "enc.h"
+#include "pvq.h"
+#include "enc_psy.h"
+#include "tab.h"
+
+#include "libavutil/channel_layout.h"
+#include "libavutil/float_dsp.h"
+#include "libavutil/mem.h"
+#include "libavutil/mem_internal.h"
+#include "libavutil/opt.h"
+#include "bytestream.h"
+#include "audio_frame_queue.h"
+#include "codec_internal.h"
+
+typedef struct OpusEncContext {
+    AVClass *av_class;
+    OpusEncOptions options;
+    OpusPsyContext psyctx;
+    AVCodecContext *avctx;
+    AudioFrameQueue afq;
+    AVFloatDSPContext *dsp;
+    AVTXContext *tx[CELT_BLOCK_NB];
+    av_tx_fn tx_fn[CELT_BLOCK_NB];
+    CeltPVQ *pvq;
+    struct FFBufQueue bufqueue;
+
+    uint8_t enc_id[64];
+    int enc_id_bits;
+
+    OpusPacketInfo packet;
+
+    int channels;
+
+    CeltFrame *frame;
+    OpusRangeCoder *rc;
+
+    /* Actual energy the decoder will have */
+    float last_quantized_energy[OPUS_MAX_CHANNELS][CELT_MAX_BANDS];
+
+    DECLARE_ALIGNED(32, float, scratch)[2048];
+} OpusEncContext;
+
+static void opus_write_extradata(AVCodecContext *avctx)
+{
+    uint8_t *bs = avctx->extradata;
+
+    bytestream_put_buffer(&bs, "OpusHead", 8);
+    bytestream_put_byte  (&bs, 0x1);
+    bytestream_put_byte  (&bs, avctx->ch_layout.nb_channels);
+    bytestream_put_le16  (&bs, avctx->initial_padding);
+    bytestream_put_le32  (&bs, avctx->sample_rate);
+    bytestream_put_le16  (&bs, 0x0);
+    bytestream_put_byte  (&bs, 0x0); /* Default layout */
+}
+
+static int opus_gen_toc(OpusEncContext *s, uint8_t *toc, int *size, int *fsize_needed)
+{
+    int tmp = 0x0, extended_toc = 0;
+    static const int toc_cfg[][OPUS_MODE_NB][OPUS_BANDWITH_NB] = {
+        /*  Silk                    Hybrid                  Celt                    Layer     */
+        /*  NB  MB  WB SWB  FB      NB  MB  WB SWB  FB      NB  MB  WB SWB  FB      Bandwidth */
+        { {  0,  0,  0,  0,  0 }, {  0,  0,  0,  0,  0 }, { 17,  0, 21, 25, 29 } }, /* 2.5 ms */
+        { {  0,  0,  0,  0,  0 }, {  0,  0,  0,  0,  0 }, { 18,  0, 22, 26, 30 } }, /*   5 ms */
+        { {  1,  5,  9,  0,  0 }, {  0,  0,  0, 13, 15 }, { 19,  0, 23, 27, 31 } }, /*  10 ms */
+        { {  2,  6, 10,  0,  0 }, {  0,  0,  0, 14, 16 }, { 20,  0, 24, 28, 32 } }, /*  20 ms */
+        { {  3,  7, 11,  0,  0 }, {  0,  0,  0,  0,  0 }, {  0,  0,  0,  0,  0 } }, /*  40 ms */
+        { {  4,  8, 12,  0,  0 }, {  0,  0,  0,  0,  0 }, {  0,  0,  0,  0,  0 } }, /*  60 ms */
+    };
+    int cfg = toc_cfg[s->packet.framesize][s->packet.mode][s->packet.bandwidth];
+    *fsize_needed = 0;
+    if (!cfg)
+        return 1;
+    if (s->packet.frames == 2) {                                       /* 2 packets */
+        if (s->frame[0].framebits == s->frame[1].framebits) {          /* same size */
+            tmp = 0x1;
+        } else {                                                  /* different size */
+            tmp = 0x2;
+            *fsize_needed = 1;                     /* put frame sizes in the packet */
+        }
+    } else if (s->packet.frames > 2) {
+        tmp = 0x3;
+        extended_toc = 1;
+    }
+    tmp |= (s->channels > 1) << 2;                                /* Stereo or mono */
+    tmp |= (cfg - 1)         << 3;                           /* codec configuration */
+    *toc++ = tmp;
+    if (extended_toc) {
+        for (int i = 0; i < (s->packet.frames - 1); i++)
+            *fsize_needed |= (s->frame[i].framebits != s->frame[i + 1].framebits);
+        tmp = (*fsize_needed) << 7;                                /* vbr flag */
+        tmp |= (0) << 6;                                       /* padding flag */
+        tmp |= s->packet.frames;
+        *toc++ = tmp;
+    }
+    *size = 1 + extended_toc;
+    return 0;
+}
+
+static void celt_frame_setup_input(OpusEncContext *s, CeltFrame *f)
+{
+    AVFrame *cur = NULL;
+    const int subframesize = s->avctx->frame_size;
+    int subframes = OPUS_BLOCK_SIZE(s->packet.framesize) / subframesize;
+
+    cur = ff_bufqueue_get(&s->bufqueue);
+
+    for (int ch = 0; ch < f->channels; ch++) {
+        CeltBlock *b = &f->block[ch];
+        const void *input = cur->extended_data[ch];
+        size_t bps = av_get_bytes_per_sample(cur->format);
+        memcpy(b->overlap, input, bps*cur->nb_samples);
+    }
+
+    av_frame_free(&cur);
+
+    for (int sf = 0; sf < subframes; sf++) {
+        if (sf != (subframes - 1))
+            cur = ff_bufqueue_get(&s->bufqueue);
+        else
+            cur = ff_bufqueue_peek(&s->bufqueue, 0);
+
+        for (int ch = 0; ch < f->channels; ch++) {
+            CeltBlock *b = &f->block[ch];
+            const void *input = cur->extended_data[ch];
+            const size_t bps  = av_get_bytes_per_sample(cur->format);
+            const size_t left = (subframesize - cur->nb_samples)*bps;
+            const size_t len  = FFMIN(subframesize, cur->nb_samples)*bps;
+            memcpy(&b->samples[sf*subframesize], input, len);
+            memset(&b->samples[cur->nb_samples], 0, left);
+        }
+
+        /* Last frame isn't popped off and freed yet - we need it for overlap */
+        if (sf != (subframes - 1))
+            av_frame_free(&cur);
+    }
+}
+
+/* Apply the pre emphasis filter */
+static void celt_apply_preemph_filter(OpusEncContext *s, CeltFrame *f)
+{
+    const int subframesize = s->avctx->frame_size;
+    const int subframes = OPUS_BLOCK_SIZE(s->packet.framesize) / subframesize;
+    const float c = ff_opus_deemph_weights[0];
+
+    /* Filter overlap */
+    for (int ch = 0; ch < f->channels; ch++) {
+        CeltBlock *b = &f->block[ch];
+        float m = b->emph_coeff;
+        for (int i = 0; i < CELT_OVERLAP; i++) {
+            float sample = b->overlap[i];
+            b->overlap[i] = sample - m;
+            m = sample * c;
+        }
+        b->emph_coeff = m;
+    }
+
+    /* Filter the samples but do not update the last subframe's coeff - overlap ^^^ */
+    for (int sf = 0; sf < subframes; sf++) {
+        for (int ch = 0; ch < f->channels; ch++) {
+            CeltBlock *b = &f->block[ch];
+            float m = b->emph_coeff;
+            for (int i = 0; i < subframesize; i++) {
+                float sample = b->samples[sf*subframesize + i];
+                b->samples[sf*subframesize + i] = sample - m;
+                m = sample * c;
+            }
+            if (sf != (subframes - 1))
+                b->emph_coeff = m;
+        }
+    }
+}
+
+/* Create the window and do the mdct */
+static void celt_frame_mdct(OpusEncContext *s, CeltFrame *f)
+{
+    float *win = s->scratch, *temp = s->scratch + 1920;
+
+    if (f->transient) {
+        for (int ch = 0; ch < f->channels; ch++) {
+            CeltBlock *b = &f->block[ch];
+            float *src1 = b->overlap;
+            for (int t = 0; t < f->blocks; t++) {
+                float *src2 = &b->samples[CELT_OVERLAP*t];
+                s->dsp->vector_fmul(win, src1, ff_celt_window, 128);
+                s->dsp->vector_fmul_reverse(&win[CELT_OVERLAP], src2,
+                                            ff_celt_window_padded, 128);
+                src1 = src2;
+                s->tx_fn[0](s->tx[0], b->coeffs + t, win, sizeof(float)*f->blocks);
+            }
+        }
+    } else {
+        int blk_len = OPUS_BLOCK_SIZE(f->size), wlen = OPUS_BLOCK_SIZE(f->size + 1);
+        int rwin = blk_len - CELT_OVERLAP, lap_dst = (wlen - blk_len - CELT_OVERLAP) >> 1;
+        memset(win, 0, wlen*sizeof(float));
+        for (int ch = 0; ch < f->channels; ch++) {
+            CeltBlock *b = &f->block[ch];
+
+            /* Overlap */
+            s->dsp->vector_fmul(temp, b->overlap, ff_celt_window, 128);
+            memcpy(win + lap_dst, temp, CELT_OVERLAP*sizeof(float));
+
+            /* Samples, flat top window */
+            memcpy(&win[lap_dst + CELT_OVERLAP], b->samples, rwin*sizeof(float));
+
+            /* Samples, windowed */
+            s->dsp->vector_fmul_reverse(temp, b->samples + rwin,
+                                        ff_celt_window_padded, 128);
+            memcpy(win + lap_dst + blk_len, temp, CELT_OVERLAP*sizeof(float));
+
+            s->tx_fn[f->size](s->tx[f->size], b->coeffs, win, sizeof(float));
+        }
+    }
+
+    for (int ch = 0; ch < f->channels; ch++) {
+        CeltBlock *block = &f->block[ch];
+        for (int i = 0; i < CELT_MAX_BANDS; i++) {
+            float ener = 0.0f;
+            int band_offset = ff_celt_freq_bands[i] << f->size;
+            int band_size   = ff_celt_freq_range[i] << f->size;
+            float *coeffs   = &block->coeffs[band_offset];
+
+            for (int j = 0; j < band_size; j++)
+                ener += coeffs[j]*coeffs[j];
+
+            block->lin_energy[i] = sqrtf(ener) + FLT_EPSILON;
+            ener = 1.0f/block->lin_energy[i];
+
+            for (int j = 0; j < band_size; j++)
+                coeffs[j] *= ener;
+
+            block->energy[i] = log2f(block->lin_energy[i]) - ff_celt_mean_energy[i];
+
+            /* CELT_ENERGY_SILENCE is what the decoder uses and its not -infinity */
+            block->energy[i] = FFMAX(block->energy[i], CELT_ENERGY_SILENCE);
+        }
+    }
+}
+
+static void celt_enc_tf(CeltFrame *f, OpusRangeCoder *rc)
+{
+    int tf_select = 0, diff = 0, tf_changed = 0, tf_select_needed;
+    int bits = f->transient ? 2 : 4;
+
+    tf_select_needed = ((f->size && (opus_rc_tell(rc) + bits + 1) <= f->framebits));
+
+    for (int i = f->start_band; i < f->end_band; i++) {
+        if ((opus_rc_tell(rc) + bits + tf_select_needed) <= f->framebits) {
+            const int tbit = (diff ^ 1) == f->tf_change[i];
+            ff_opus_rc_enc_log(rc, tbit, bits);
+            diff ^= tbit;
+            tf_changed |= diff;
+        }
+        bits = f->transient ? 4 : 5;
+    }
+
+    if (tf_select_needed && ff_celt_tf_select[f->size][f->transient][0][tf_changed] !=
+                            ff_celt_tf_select[f->size][f->transient][1][tf_changed]) {
+        ff_opus_rc_enc_log(rc, f->tf_select, 1);
+        tf_select = f->tf_select;
+    }
+
+    for (int i = f->start_band; i < f->end_band; i++)
+        f->tf_change[i] = ff_celt_tf_select[f->size][f->transient][tf_select][f->tf_change[i]];
+}
+
+static void celt_enc_quant_pfilter(OpusRangeCoder *rc, CeltFrame *f)
+{
+    float gain = f->pf_gain;
+    int txval, octave = f->pf_octave, period = f->pf_period, tapset = f->pf_tapset;
+
+    ff_opus_rc_enc_log(rc, f->pfilter, 1);
+    if (!f->pfilter)
+        return;
+
+    /* Octave */
+    txval = FFMIN(octave, 6);
+    ff_opus_rc_enc_uint(rc, txval, 6);
+    octave = txval;
+    /* Period */
+    txval = av_clip(period - (16 << octave) + 1, 0, (1 << (4 + octave)) - 1);
+    ff_opus_rc_put_raw(rc, period, 4 + octave);
+    period = txval + (16 << octave) - 1;
+    /* Gain */
+    txval = FFMIN(((int)(gain / 0.09375f)) - 1, 7);
+    ff_opus_rc_put_raw(rc, txval, 3);
+    gain   = 0.09375f * (txval + 1);
+    /* Tapset */
+    if ((opus_rc_tell(rc) + 2) <= f->framebits)
+        ff_opus_rc_enc_cdf(rc, tapset, ff_celt_model_tapset);
+    else
+        tapset = 0;
+    /* Finally create the coeffs */
+    for (int i = 0; i < 2; i++) {
+        CeltBlock *block = &f->block[i];
+
+        block->pf_period_new = FFMAX(period, CELT_POSTFILTER_MINPERIOD);
+        block->pf_gains_new[0] = gain * ff_celt_postfilter_taps[tapset][0];
+        block->pf_gains_new[1] = gain * ff_celt_postfilter_taps[tapset][1];
+        block->pf_gains_new[2] = gain * ff_celt_postfilter_taps[tapset][2];
+    }
+}
+
+static void exp_quant_coarse(OpusRangeCoder *rc, CeltFrame *f,
+                             float last_energy[][CELT_MAX_BANDS], int intra)
+{
+    float alpha, beta, prev[2] = { 0, 0 };
+    const uint8_t *pmod = ff_celt_coarse_energy_dist[f->size][intra];
+
+    /* Inter is really just differential coding */
+    if (opus_rc_tell(rc) + 3 <= f->framebits)
+        ff_opus_rc_enc_log(rc, intra, 3);
+    else
+        intra = 0;
+
+    if (intra) {
+        alpha = 0.0f;
+        beta  = 1.0f - (4915.0f/32768.0f);
+    } else {
+        alpha = ff_celt_alpha_coef[f->size];
+        beta  = ff_celt_beta_coef[f->size];
+    }
+
+    for (int i = f->start_band; i < f->end_band; i++) {
+        for (int ch = 0; ch < f->channels; ch++) {
+            CeltBlock *block = &f->block[ch];
+            const int left = f->framebits - opus_rc_tell(rc);
+            const float last = FFMAX(-9.0f, last_energy[ch][i]);
+            float diff = block->energy[i] - prev[ch] - last*alpha;
+            int q_en = lrintf(diff);
+            if (left >= 15) {
+                ff_opus_rc_enc_laplace(rc, &q_en, pmod[i << 1] << 7, pmod[(i << 1) + 1] << 6);
+            } else if (left >= 2) {
+                q_en = av_clip(q_en, -1, 1);
+                ff_opus_rc_enc_cdf(rc, 2*q_en + 3*(q_en < 0), ff_celt_model_energy_small);
+            } else if (left >= 1) {
+                q_en = av_clip(q_en, -1, 0);
+                ff_opus_rc_enc_log(rc, (q_en & 1), 1);
+            } else q_en = -1;
+
+            block->error_energy[i] = q_en - diff;
+            prev[ch] += beta * q_en;
+        }
+    }
+}
+
+static void celt_quant_coarse(CeltFrame *f, OpusRangeCoder *rc,
+                              float last_energy[][CELT_MAX_BANDS])
+{
+    uint32_t inter, intra;
+    OPUS_RC_CHECKPOINT_SPAWN(rc);
+
+    exp_quant_coarse(rc, f, last_energy, 1);
+    intra = OPUS_RC_CHECKPOINT_BITS(rc);
+
+    OPUS_RC_CHECKPOINT_ROLLBACK(rc);
+
+    exp_quant_coarse(rc, f, last_energy, 0);
+    inter = OPUS_RC_CHECKPOINT_BITS(rc);
+
+    if (inter > intra) { /* Unlikely */
+        OPUS_RC_CHECKPOINT_ROLLBACK(rc);
+        exp_quant_coarse(rc, f, last_energy, 1);
+    }
+}
+
+static void celt_quant_fine(CeltFrame *f, OpusRangeCoder *rc)
+{
+    for (int i = f->start_band; i < f->end_band; i++) {
+        if (!f->fine_bits[i])
+            continue;
+        for (int ch = 0; ch < f->channels; ch++) {
+            CeltBlock *block = &f->block[ch];
+            int quant, lim = (1 << f->fine_bits[i]);
+            float offset, diff = 0.5f - block->error_energy[i];
+            quant = av_clip(floor(diff*lim), 0, lim - 1);
+            ff_opus_rc_put_raw(rc, quant, f->fine_bits[i]);
+            offset = 0.5f - ((quant + 0.5f) * (1 << (14 - f->fine_bits[i])) / 16384.0f);
+            block->error_energy[i] -= offset;
+        }
+    }
+}
+
+static void celt_quant_final(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
+{
+    for (int priority = 0; priority < 2; priority++) {
+        for (int i = f->start_band; i < f->end_band && (f->framebits - opus_rc_tell(rc)) >= f->channels; i++) {
+            if (f->fine_priority[i] != priority || f->fine_bits[i] >= CELT_MAX_FINE_BITS)
+                continue;
+            for (int ch = 0; ch < f->channels; ch++) {
+                CeltBlock *block = &f->block[ch];
+                const float err = block->error_energy[i];
+                const float offset = 0.5f * (1 << (14 - f->fine_bits[i] - 1)) / 16384.0f;
+                const int sign = FFABS(err + offset) < FFABS(err - offset);
+                ff_opus_rc_put_raw(rc, sign, 1);
+                block->error_energy[i] -= offset*(1 - 2*sign);
+            }
+        }
+    }
+}
+
+static void celt_encode_frame(OpusEncContext *s, OpusRangeCoder *rc,
+                              CeltFrame *f, int index)
+{
+    ff_opus_rc_enc_init(rc);
+
+    ff_opus_psy_celt_frame_init(&s->psyctx, f, index);
+
+    celt_frame_setup_input(s, f);
+
+    if (f->silence) {
+        if (f->framebits >= 16)
+            ff_opus_rc_enc_log(rc, 1, 15); /* Silence (if using explicit singalling) */
+        for (int ch = 0; ch < s->channels; ch++)
+            memset(s->last_quantized_energy[ch], 0.0f, sizeof(float)*CELT_MAX_BANDS);
+        return;
+    }
+
+    /* Filters */
+    celt_apply_preemph_filter(s, f);
+    if (f->pfilter) {
+        ff_opus_rc_enc_log(rc, 0, 15);
+        celt_enc_quant_pfilter(rc, f);
+    }
+
+    /* Transform */
+    celt_frame_mdct(s, f);
+
+    /* Need to handle transient/non-transient switches at any point during analysis */
+    while (ff_opus_psy_celt_frame_process(&s->psyctx, f, index))
+        celt_frame_mdct(s, f);
+
+    ff_opus_rc_enc_init(rc);
+
+    /* Silence */
+    ff_opus_rc_enc_log(rc, 0, 15);
+
+    /* Pitch filter */
+    if (!f->start_band && opus_rc_tell(rc) + 16 <= f->framebits)
+        celt_enc_quant_pfilter(rc, f);
+
+    /* Transient flag */
+    if (f->size && opus_rc_tell(rc) + 3 <= f->framebits)
+        ff_opus_rc_enc_log(rc, f->transient, 3);
+
+    /* Main encoding */
+    celt_quant_coarse  (f, rc, s->last_quantized_energy);
+    celt_enc_tf        (f, rc);
+    ff_celt_bitalloc   (f, rc, 1);
+    celt_quant_fine    (f, rc);
+    ff_celt_quant_bands(f, rc);
+
+    /* Anticollapse bit */
+    if (f->anticollapse_needed)
+        ff_opus_rc_put_raw(rc, f->anticollapse, 1);
+
+    /* Final per-band energy adjustments from leftover bits */
+    celt_quant_final(s, rc, f);
+
+    for (int ch = 0; ch < f->channels; ch++) {
+        CeltBlock *block = &f->block[ch];
+        for (int i = 0; i < CELT_MAX_BANDS; i++)
+            s->last_quantized_energy[ch][i] = block->energy[i] + block->error_energy[i];
+    }
+}
+
+static inline int write_opuslacing(uint8_t *dst, int v)
+{
+    dst[0] = FFMIN(v - FFALIGN(v - 255, 4), v);
+    dst[1] = v - dst[0] >> 2;
+    return 1 + (v >= 252);
+}
+
+static void opus_packet_assembler(OpusEncContext *s, AVPacket *avpkt)
+{
+    int offset, fsize_needed;
+
+    /* Write toc */
+    opus_gen_toc(s, avpkt->data, &offset, &fsize_needed);
+
+    /* Frame sizes if needed */
+    if (fsize_needed) {
+        for (int i = 0; i < s->packet.frames - 1; i++) {
+            offset += write_opuslacing(avpkt->data + offset,
+                                       s->frame[i].framebits >> 3);
+        }
+    }
+
+    /* Packets */
+    for (int i = 0; i < s->packet.frames; i++) {
+        ff_opus_rc_enc_end(&s->rc[i], avpkt->data + offset,
+                           s->frame[i].framebits >> 3);
+        offset += s->frame[i].framebits >> 3;
+    }
+
+    avpkt->size = offset;
+}
+
+/* Used as overlap for the first frame and padding for the last encoded packet */
+static AVFrame *spawn_empty_frame(OpusEncContext *s)
+{
+    AVFrame *f = av_frame_alloc();
+    int ret;
+    if (!f)
+        return NULL;
+    f->format         = s->avctx->sample_fmt;
+    f->nb_samples     = s->avctx->frame_size;
+    ret = av_channel_layout_copy(&f->ch_layout, &s->avctx->ch_layout);
+    if (ret < 0) {
+        av_frame_free(&f);
+        return NULL;
+    }
+    if (av_frame_get_buffer(f, 4)) {
+        av_frame_free(&f);
+        return NULL;
+    }
+    for (int i = 0; i < s->channels; i++) {
+        size_t bps = av_get_bytes_per_sample(f->format);
+        memset(f->extended_data[i], 0, bps*f->nb_samples);
+    }
+    return f;
+}
+
+static int opus_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
+                             const AVFrame *frame, int *got_packet_ptr)
+{
+    OpusEncContext *s = avctx->priv_data;
+    int ret, frame_size, alloc_size = 0;
+
+    if (frame) { /* Add new frame to queue */
+        if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
+            return ret;
+        ff_bufqueue_add(avctx, &s->bufqueue, av_frame_clone(frame));
+    } else {
+        ff_opus_psy_signal_eof(&s->psyctx);
+        if (!s->afq.remaining_samples || !avctx->frame_num)
+            return 0; /* We've been flushed and there's nothing left to encode */
+    }
+
+    /* Run the psychoacoustic system */
+    if (ff_opus_psy_process(&s->psyctx, &s->packet))
+        return 0;
+
+    frame_size = OPUS_BLOCK_SIZE(s->packet.framesize);
+
+    if (!frame) {
+        /* This can go negative, that's not a problem, we only pad if positive */
+        int pad_empty = s->packet.frames*(frame_size/s->avctx->frame_size) - s->bufqueue.available + 1;
+        /* Pad with empty 2.5 ms frames to whatever framesize was decided,
+         * this should only happen at the very last flush frame. The frames
+         * allocated here will be freed (because they have no other references)
+         * after they get used by celt_frame_setup_input() */
+        for (int i = 0; i < pad_empty; i++) {
+            AVFrame *empty = spawn_empty_frame(s);
+            if (!empty)
+                return AVERROR(ENOMEM);
+            ff_bufqueue_add(avctx, &s->bufqueue, empty);
+        }
+    }
+
+    for (int i = 0; i < s->packet.frames; i++) {
+        celt_encode_frame(s, &s->rc[i], &s->frame[i], i);
+        alloc_size += s->frame[i].framebits >> 3;
+    }
+
+    /* Worst case toc + the frame lengths if needed */
+    alloc_size += 2 + s->packet.frames*2;
+
+    if ((ret = ff_alloc_packet(avctx, avpkt, alloc_size)) < 0)
+        return ret;
+
+    /* Assemble packet */
+    opus_packet_assembler(s, avpkt);
+
+    /* Update the psychoacoustic system */
+    ff_opus_psy_postencode_update(&s->psyctx, s->frame);
+
+    /* Remove samples from queue and skip if needed */
+    ff_af_queue_remove(&s->afq, s->packet.frames*frame_size, &avpkt->pts, &avpkt->duration);
+    if (s->packet.frames*frame_size > avpkt->duration) {
+        uint8_t *side = av_packet_new_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, 10);
+        if (!side)
+            return AVERROR(ENOMEM);
+        AV_WL32(&side[4], s->packet.frames*frame_size - avpkt->duration + 120);
+    }
+
+    *got_packet_ptr = 1;
+
+    return 0;
+}
+
+static av_cold int opus_encode_end(AVCodecContext *avctx)
+{
+    OpusEncContext *s = avctx->priv_data;
+
+    for (int i = 0; i < CELT_BLOCK_NB; i++)
+        av_tx_uninit(&s->tx[i]);
+
+    ff_celt_pvq_uninit(&s->pvq);
+    av_freep(&s->dsp);
+    av_freep(&s->frame);
+    av_freep(&s->rc);
+    ff_af_queue_close(&s->afq);
+    ff_opus_psy_end(&s->psyctx);
+    ff_bufqueue_discard_all(&s->bufqueue);
+
+    return 0;
+}
+
+static av_cold int opus_encode_init(AVCodecContext *avctx)
+{
+    int ret, max_frames;
+    OpusEncContext *s = avctx->priv_data;
+
+    s->avctx = avctx;
+    s->channels = avctx->ch_layout.nb_channels;
+
+    /* Opus allows us to change the framesize on each packet (and each packet may
+     * have multiple frames in it) but we can't change the codec's frame size on
+     * runtime, so fix it to the lowest possible number of samples and use a queue
+     * to accumulate AVFrames until we have enough to encode whatever the encoder
+     * decides is the best */
+    avctx->frame_size = 120;
+    /* Initial padding will change if SILK is ever supported */
+    avctx->initial_padding = 120;
+
+    if (!avctx->bit_rate) {
+        int coupled = ff_opus_default_coupled_streams[s->channels - 1];
+        avctx->bit_rate = coupled*(96000) + (s->channels - coupled*2)*(48000);
+    } else if (avctx->bit_rate < 6000 || avctx->bit_rate > 255000 * s->channels) {
+        int64_t clipped_rate = av_clip(avctx->bit_rate, 6000, 255000 * s->channels);
+        av_log(avctx, AV_LOG_ERROR, "Unsupported bitrate %"PRId64" kbps, clipping to %"PRId64" kbps\n",
+               avctx->bit_rate/1000, clipped_rate/1000);
+        avctx->bit_rate = clipped_rate;
+    }
+
+    /* Extradata */
+    avctx->extradata_size = 19;
+    avctx->extradata = av_malloc(avctx->extradata_size + AV_INPUT_BUFFER_PADDING_SIZE);
+    if (!avctx->extradata)
+        return AVERROR(ENOMEM);
+    opus_write_extradata(avctx);
+
+    ff_af_queue_init(avctx, &s->afq);
+
+    if ((ret = ff_celt_pvq_init(&s->pvq, 1)) < 0)
+        return ret;
+
+    if (!(s->dsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT)))
+        return AVERROR(ENOMEM);
+
+    /* I have no idea why a base scaling factor of 68 works, could be the twiddles */
+    for (int i = 0; i < CELT_BLOCK_NB; i++) {
+        const float scale = 68 << (CELT_BLOCK_NB - 1 - i);
+        if ((ret = av_tx_init(&s->tx[i], &s->tx_fn[i], AV_TX_FLOAT_MDCT, 0, 15 << (i + 3), &scale, 0)))
+            return AVERROR(ENOMEM);
+    }
+
+    /* Zero out previous energy (matters for inter first frame) */
+    for (int ch = 0; ch < s->channels; ch++)
+        memset(s->last_quantized_energy[ch], 0.0f, sizeof(float)*CELT_MAX_BANDS);
+
+    /* Allocate an empty frame to use as overlap for the first frame of audio */
+    ff_bufqueue_add(avctx, &s->bufqueue, spawn_empty_frame(s));
+    if (!ff_bufqueue_peek(&s->bufqueue, 0))
+        return AVERROR(ENOMEM);
+
+    if ((ret = ff_opus_psy_init(&s->psyctx, s->avctx, &s->bufqueue, &s->options)))
+        return ret;
+
+    /* Frame structs and range coder buffers */
+    max_frames = ceilf(FFMIN(s->options.max_delay_ms, 120.0f)/2.5f);
+    s->frame = av_malloc(max_frames*sizeof(CeltFrame));
+    if (!s->frame)
+        return AVERROR(ENOMEM);
+    s->rc = av_malloc(max_frames*sizeof(OpusRangeCoder));
+    if (!s->rc)
+        return AVERROR(ENOMEM);
+
+    for (int i = 0; i < max_frames; i++) {
+        s->frame[i].dsp = s->dsp;
+        s->frame[i].avctx = s->avctx;
+        s->frame[i].seed = 0;
+        s->frame[i].pvq = s->pvq;
+        s->frame[i].apply_phase_inv = s->options.apply_phase_inv;
+        s->frame[i].block[0].emph_coeff = s->frame[i].block[1].emph_coeff = 0.0f;
+    }
+
+    return 0;
+}
+
+#define OPUSENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
+static const AVOption opusenc_options[] = {
+    { "opus_delay", "Maximum delay in milliseconds", offsetof(OpusEncContext, options.max_delay_ms), AV_OPT_TYPE_FLOAT, { .dbl = OPUS_MAX_LOOKAHEAD }, 2.5f, OPUS_MAX_LOOKAHEAD, OPUSENC_FLAGS, .unit = "max_delay_ms" },
+    { "apply_phase_inv", "Apply intensity stereo phase inversion", offsetof(OpusEncContext, options.apply_phase_inv), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, OPUSENC_FLAGS, .unit = "apply_phase_inv" },
+    { NULL },
+};
+
+static const AVClass opusenc_class = {
+    .class_name = "Opus encoder",
+    .item_name  = av_default_item_name,
+    .option     = opusenc_options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
+static const FFCodecDefault opusenc_defaults[] = {
+    { "b", "0" },
+    { "compression_level", "10" },
+    { NULL },
+};
+
+const FFCodec ff_opus_encoder = {
+    .p.name         = "opus",
+    CODEC_LONG_NAME("Opus"),
+    .p.type         = AVMEDIA_TYPE_AUDIO,
+    .p.id           = AV_CODEC_ID_OPUS,
+    .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY |
+                      AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_EXPERIMENTAL,
+    .defaults       = opusenc_defaults,
+    .p.priv_class   = &opusenc_class,
+    .priv_data_size = sizeof(OpusEncContext),
+    .init           = opus_encode_init,
+    FF_CODEC_ENCODE_CB(opus_encode_frame),
+    .close          = opus_encode_end,
+    .caps_internal  = FF_CODEC_CAP_INIT_CLEANUP,
+    .p.supported_samplerates = (const int []){ 48000, 0 },
+    .p.ch_layouts    = (const AVChannelLayout []){ AV_CHANNEL_LAYOUT_MONO,
+                                                   AV_CHANNEL_LAYOUT_STEREO, { 0 } },
+    .p.sample_fmts  = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
+                                                     AV_SAMPLE_FMT_NONE },
+};
diff --git a/libavcodec/opus/enc.h b/libavcodec/opus/enc.h
new file mode 100644
index 0000000000..d56f91e625
--- /dev/null
+++ b/libavcodec/opus/enc.h
@@ -0,0 +1,55 @@
+/*
+ * Opus encoder
+ * Copyright (c) 2017 Rostislav Pehlivanov <atomnuker@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_OPUS_ENC_H
+#define AVCODEC_OPUS_ENC_H
+
+#include "libavutil/intmath.h"
+#include "opus.h"
+
+/* Determines the maximum delay the psychoacoustic system will use for lookahead */
+#define FF_BUFQUEUE_SIZE 145
+#include "libavfilter/bufferqueue.h"
+
+#define OPUS_MAX_LOOKAHEAD ((FF_BUFQUEUE_SIZE - 1)*2.5f)
+
+#define OPUS_MAX_CHANNELS 2
+
+/* 120 ms / 2.5 ms = 48 frames (extremely improbable, but the encoder'll work) */
+#define OPUS_MAX_FRAMES_PER_PACKET 48
+
+#define OPUS_BLOCK_SIZE(x) (2 * 15 * (1 << ((x) + 2)))
+
+#define OPUS_SAMPLES_TO_BLOCK_SIZE(x) (ff_log2((x) / (2 * 15)) - 2)
+
+typedef struct OpusEncOptions {
+    float max_delay_ms;
+    int apply_phase_inv;
+} OpusEncOptions;
+
+typedef struct OpusPacketInfo {
+    enum OpusMode mode;
+    enum OpusBandwidth bandwidth;
+    int framesize;
+    int frames;
+} OpusPacketInfo;
+
+#endif /* AVCODEC_OPUS_ENC_H */
diff --git a/libavcodec/opus/enc_psy.c b/libavcodec/opus/enc_psy.c
new file mode 100644
index 0000000000..250cfb567a
--- /dev/null
+++ b/libavcodec/opus/enc_psy.c
@@ -0,0 +1,614 @@
+/*
+ * Opus encoder
+ * Copyright (c) 2017 Rostislav Pehlivanov <atomnuker@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <float.h>
+
+#include "libavutil/mem.h"
+#include "enc_psy.h"
+#include "celt.h"
+#include "pvq.h"
+#include "tab.h"
+#include "libavfilter/window_func.h"
+
+static float pvq_band_cost(CeltPVQ *pvq, CeltFrame *f, OpusRangeCoder *rc, int band,
+                           float *bits, float lambda)
+{
+    int i, b = 0;
+    uint32_t cm[2] = { (1 << f->blocks) - 1, (1 << f->blocks) - 1 };
+    const int band_size = ff_celt_freq_range[band] << f->size;
+    float buf[176 * 2], lowband_scratch[176], norm1[176], norm2[176];
+    float dist, cost, err_x = 0.0f, err_y = 0.0f;
+    float *X = buf;
+    float *X_orig = f->block[0].coeffs + (ff_celt_freq_bands[band] << f->size);
+    float *Y = (f->channels == 2) ? &buf[176] : NULL;
+    float *Y_orig = f->block[1].coeffs + (ff_celt_freq_bands[band] << f->size);
+    OPUS_RC_CHECKPOINT_SPAWN(rc);
+
+    memcpy(X, X_orig, band_size*sizeof(float));
+    if (Y)
+        memcpy(Y, Y_orig, band_size*sizeof(float));
+
+    f->remaining2 = ((f->framebits << 3) - f->anticollapse_needed) - opus_rc_tell_frac(rc) - 1;
+    if (band <= f->coded_bands - 1) {
+        int curr_balance = f->remaining / FFMIN(3, f->coded_bands - band);
+        b = av_clip_uintp2(FFMIN(f->remaining2 + 1, f->pulses[band] + curr_balance), 14);
+    }
+
+    if (f->dual_stereo) {
+        pvq->quant_band(pvq, f, rc, band, X, NULL, band_size, b / 2, f->blocks, NULL,
+                        f->size, norm1, 0, 1.0f, lowband_scratch, cm[0]);
+
+        pvq->quant_band(pvq, f, rc, band, Y, NULL, band_size, b / 2, f->blocks, NULL,
+                        f->size, norm2, 0, 1.0f, lowband_scratch, cm[1]);
+    } else {
+        pvq->quant_band(pvq, f, rc, band, X, Y, band_size, b, f->blocks, NULL, f->size,
+                        norm1, 0, 1.0f, lowband_scratch, cm[0] | cm[1]);
+    }
+
+    for (i = 0; i < band_size; i++) {
+        err_x += (X[i] - X_orig[i])*(X[i] - X_orig[i]);
+        if (Y)
+            err_y += (Y[i] - Y_orig[i])*(Y[i] - Y_orig[i]);
+    }
+
+    dist = sqrtf(err_x) + sqrtf(err_y);
+    cost = OPUS_RC_CHECKPOINT_BITS(rc)/8.0f;
+    *bits += cost;
+
+    OPUS_RC_CHECKPOINT_ROLLBACK(rc);
+
+    return lambda*dist*cost;
+}
+
+/* Populate metrics without taking into consideration neighbouring steps */
+static void step_collect_psy_metrics(OpusPsyContext *s, int index)
+{
+    int silence = 0, ch, i, j;
+    OpusPsyStep *st = s->steps[index];
+
+    st->index = index;
+
+    for (ch = 0; ch < s->avctx->ch_layout.nb_channels; ch++) {
+        const int lap_size = (1 << s->bsize_analysis);
+        for (i = 1; i <= FFMIN(lap_size, index); i++) {
+            const int offset = i*120;
+            AVFrame *cur = ff_bufqueue_peek(s->bufqueue, index - i);
+            memcpy(&s->scratch[offset], cur->extended_data[ch], cur->nb_samples*sizeof(float));
+        }
+        for (i = 0; i < lap_size; i++) {
+            const int offset = i*120 + lap_size;
+            AVFrame *cur = ff_bufqueue_peek(s->bufqueue, index + i);
+            memcpy(&s->scratch[offset], cur->extended_data[ch], cur->nb_samples*sizeof(float));
+        }
+
+        s->dsp->vector_fmul(s->scratch, s->scratch, s->window[s->bsize_analysis],
+                            (OPUS_BLOCK_SIZE(s->bsize_analysis) << 1));
+
+        s->mdct_fn[s->bsize_analysis](s->mdct[s->bsize_analysis], st->coeffs[ch],
+                                      s->scratch, sizeof(float));
+
+        for (i = 0; i < CELT_MAX_BANDS; i++)
+            st->bands[ch][i] = &st->coeffs[ch][ff_celt_freq_bands[i] << s->bsize_analysis];
+    }
+
+    for (ch = 0; ch < s->avctx->ch_layout.nb_channels; ch++) {
+        for (i = 0; i < CELT_MAX_BANDS; i++) {
+            float avg_c_s, energy = 0.0f, dist_dev = 0.0f;
+            const int range = ff_celt_freq_range[i] << s->bsize_analysis;
+            const float *coeffs = st->bands[ch][i];
+            for (j = 0; j < range; j++)
+                energy += coeffs[j]*coeffs[j];
+
+            st->energy[ch][i] += sqrtf(energy);
+            silence |= !!st->energy[ch][i];
+            avg_c_s = energy / range;
+
+            for (j = 0; j < range; j++) {
+                const float c_s = coeffs[j]*coeffs[j];
+                dist_dev += (avg_c_s - c_s)*(avg_c_s - c_s);
+            }
+
+            st->tone[ch][i] += sqrtf(dist_dev);
+        }
+    }
+
+    st->silence = !silence;
+
+    if (s->avctx->ch_layout.nb_channels > 1) {
+        for (i = 0; i < CELT_MAX_BANDS; i++) {
+            float incompat = 0.0f;
+            const float *coeffs1 = st->bands[0][i];
+            const float *coeffs2 = st->bands[1][i];
+            const int range = ff_celt_freq_range[i] << s->bsize_analysis;
+            for (j = 0; j < range; j++)
+                incompat += (coeffs1[j] - coeffs2[j])*(coeffs1[j] - coeffs2[j]);
+            st->stereo[i] = sqrtf(incompat);
+        }
+    }
+
+    for (ch = 0; ch < s->avctx->ch_layout.nb_channels; ch++) {
+        for (i = 0; i < CELT_MAX_BANDS; i++) {
+            OpusBandExcitation *ex = &s->ex[ch][i];
+            float bp_e = bessel_filter(&s->bfilter_lo[ch][i], st->energy[ch][i]);
+            bp_e = bessel_filter(&s->bfilter_hi[ch][i], bp_e);
+            bp_e *= bp_e;
+            if (bp_e > ex->excitation) {
+                st->change_amp[ch][i] = bp_e - ex->excitation;
+                st->total_change += st->change_amp[ch][i];
+                ex->excitation = ex->excitation_init = bp_e;
+                ex->excitation_dist = 0.0f;
+            }
+            if (ex->excitation > 0.0f) {
+                ex->excitation -= av_clipf((1/expf(ex->excitation_dist)), ex->excitation_init/20, ex->excitation_init/1.09);
+                ex->excitation = FFMAX(ex->excitation, 0.0f);
+                ex->excitation_dist += 1.0f;
+            }
+        }
+    }
+}
+
+static void search_for_change_points(OpusPsyContext *s, float tgt_change,
+                                     int offset_s, int offset_e, int resolution,
+                                     int level)
+{
+    int i;
+    float c_change = 0.0f;
+    if ((offset_e - offset_s) <= resolution)
+        return;
+    for (i = offset_s; i < offset_e; i++) {
+        c_change += s->steps[i]->total_change;
+        if (c_change > tgt_change)
+            break;
+    }
+    if (i == offset_e)
+        return;
+    search_for_change_points(s, tgt_change / 2.0f, offset_s, i + 0, resolution, level + 1);
+    s->inflection_points[s->inflection_points_count++] = i;
+    search_for_change_points(s, tgt_change / 2.0f, i + 1, offset_e, resolution, level + 1);
+}
+
+static int flush_silent_frames(OpusPsyContext *s)
+{
+    int fsize, silent_frames;
+
+    for (silent_frames = 0; silent_frames < s->buffered_steps; silent_frames++)
+        if (!s->steps[silent_frames]->silence)
+            break;
+    if (--silent_frames < 0)
+        return 0;
+
+    for (fsize = CELT_BLOCK_960; fsize > CELT_BLOCK_120; fsize--) {
+        if ((1 << fsize) > silent_frames)
+            continue;
+        s->p.frames = FFMIN(silent_frames / (1 << fsize), 48 >> fsize);
+        s->p.framesize = fsize;
+        return 1;
+    }
+
+    return 0;
+}
+
+/* Main function which decides frame size and frames per current packet */
+static void psy_output_groups(OpusPsyContext *s)
+{
+    int max_delay_samples = (s->options->max_delay_ms*s->avctx->sample_rate)/1000;
+    int max_bsize = FFMIN(OPUS_SAMPLES_TO_BLOCK_SIZE(max_delay_samples), CELT_BLOCK_960);
+
+    /* These don't change for now */
+    s->p.mode      = OPUS_MODE_CELT;
+    s->p.bandwidth = OPUS_BANDWIDTH_FULLBAND;
+
+    /* Flush silent frames ASAP */
+    if (s->steps[0]->silence && flush_silent_frames(s))
+        return;
+
+    s->p.framesize = FFMIN(max_bsize, CELT_BLOCK_960);
+    s->p.frames    = 1;
+}
+
+int ff_opus_psy_process(OpusPsyContext *s, OpusPacketInfo *p)
+{
+    int i;
+    float total_energy_change = 0.0f;
+
+    if (s->buffered_steps < s->max_steps && !s->eof) {
+        const int awin = (1 << s->bsize_analysis);
+        if (++s->steps_to_process >= awin) {
+            step_collect_psy_metrics(s, s->buffered_steps - awin + 1);
+            s->steps_to_process = 0;
+        }
+        if ((++s->buffered_steps) < s->max_steps)
+            return 1;
+    }
+
+    for (i = 0; i < s->buffered_steps; i++)
+        total_energy_change += s->steps[i]->total_change;
+
+    search_for_change_points(s, total_energy_change / 2.0f, 0,
+                             s->buffered_steps, 1, 0);
+
+    psy_output_groups(s);
+
+    p->frames    = s->p.frames;
+    p->framesize = s->p.framesize;
+    p->mode      = s->p.mode;
+    p->bandwidth = s->p.bandwidth;
+
+    return 0;
+}
+
+void ff_opus_psy_celt_frame_init(OpusPsyContext *s, CeltFrame *f, int index)
+{
+    int i, neighbouring_points = 0, start_offset = 0;
+    int radius = (1 << s->p.framesize), step_offset = radius*index;
+    int silence = 1;
+
+    f->start_band = (s->p.mode == OPUS_MODE_HYBRID) ? 17 : 0;
+    f->end_band   = ff_celt_band_end[s->p.bandwidth];
+    f->channels   = s->avctx->ch_layout.nb_channels;
+    f->size       = s->p.framesize;
+
+    for (i = 0; i < (1 << f->size); i++)
+        silence &= s->steps[index*(1 << f->size) + i]->silence;
+
+    f->silence = silence;
+    if (f->silence) {
+        f->framebits = 0; /* Otherwise the silence flag eats up 16(!) bits */
+        return;
+    }
+
+    for (i = 0; i < s->inflection_points_count; i++) {
+        if (s->inflection_points[i] >= step_offset) {
+            start_offset = i;
+            break;
+        }
+    }
+
+    for (i = start_offset; i < FFMIN(radius, s->inflection_points_count - start_offset); i++) {
+        if (s->inflection_points[i] < (step_offset + radius)) {
+            neighbouring_points++;
+        }
+    }
+
+    /* Transient flagging */
+    f->transient = neighbouring_points > 0;
+    f->blocks = f->transient ? OPUS_BLOCK_SIZE(s->p.framesize)/CELT_OVERLAP : 1;
+
+    /* Some sane defaults */
+    f->pfilter   = 0;
+    f->pf_gain   = 0.5f;
+    f->pf_octave = 2;
+    f->pf_period = 1;
+    f->pf_tapset = 2;
+
+    /* More sane defaults */
+    f->tf_select = 0;
+    f->anticollapse = 1;
+    f->alloc_trim = 5;
+    f->skip_band_floor = f->end_band;
+    f->intensity_stereo = f->end_band;
+    f->dual_stereo = 0;
+    f->spread = CELT_SPREAD_NORMAL;
+    memset(f->tf_change, 0, sizeof(int)*CELT_MAX_BANDS);
+    memset(f->alloc_boost, 0, sizeof(int)*CELT_MAX_BANDS);
+}
+
+static void celt_gauge_psy_weight(OpusPsyContext *s, OpusPsyStep **start,
+                                  CeltFrame *f_out)
+{
+    int i, f, ch;
+    int frame_size = OPUS_BLOCK_SIZE(s->p.framesize);
+    float rate, frame_bits = 0;
+
+    /* Used for the global ROTATE flag */
+    float tonal = 0.0f;
+
+    /* Pseudo-weights */
+    float band_score[CELT_MAX_BANDS] = { 0 };
+    float max_score = 1.0f;
+
+    /* Pass one - one loop around each band, computing unquant stuff */
+    for (i = 0; i < CELT_MAX_BANDS; i++) {
+        float weight = 0.0f;
+        float tonal_contrib = 0.0f;
+        for (f = 0; f < (1 << s->p.framesize); f++) {
+            weight = start[f]->stereo[i];
+            for (ch = 0; ch < s->avctx->ch_layout.nb_channels; ch++) {
+                weight += start[f]->change_amp[ch][i] + start[f]->tone[ch][i] + start[f]->energy[ch][i];
+                tonal_contrib += start[f]->tone[ch][i];
+            }
+        }
+        tonal += tonal_contrib;
+        band_score[i] = weight;
+    }
+
+    tonal /= (float)CELT_MAX_BANDS;
+
+    for (i = 0; i < CELT_MAX_BANDS; i++) {
+        if (band_score[i] > max_score)
+            max_score = band_score[i];
+    }
+
+    for (i = 0; i < CELT_MAX_BANDS; i++) {
+        f_out->alloc_boost[i] = (int)((band_score[i]/max_score)*3.0f);
+        frame_bits += band_score[i]*8.0f;
+    }
+
+    tonal /= 1333136.0f;
+    f_out->spread = av_clip_uintp2(lrintf(tonal), 2);
+
+    rate = ((float)s->avctx->bit_rate) + frame_bits*frame_size*16;
+    rate *= s->lambda;
+    rate /= s->avctx->sample_rate/frame_size;
+
+    f_out->framebits = lrintf(rate);
+    f_out->framebits = FFMIN(f_out->framebits, OPUS_MAX_FRAME_SIZE * 8);
+    f_out->framebits = FFALIGN(f_out->framebits, 8);
+}
+
+static int bands_dist(OpusPsyContext *s, CeltFrame *f, float *total_dist)
+{
+    int i, tdist = 0.0f;
+    OpusRangeCoder dump;
+
+    ff_opus_rc_enc_init(&dump);
+    ff_celt_bitalloc(f, &dump, 1);
+
+    for (i = 0; i < CELT_MAX_BANDS; i++) {
+        float bits = 0.0f;
+        float dist = pvq_band_cost(f->pvq, f, &dump, i, &bits, s->lambda);
+        tdist += dist;
+    }
+
+    *total_dist = tdist;
+
+    return 0;
+}
+
+static void celt_search_for_dual_stereo(OpusPsyContext *s, CeltFrame *f)
+{
+    float td1, td2;
+    f->dual_stereo = 0;
+
+    if (s->avctx->ch_layout.nb_channels < 2)
+        return;
+
+    bands_dist(s, f, &td1);
+    f->dual_stereo = 1;
+    bands_dist(s, f, &td2);
+
+    f->dual_stereo = td2 < td1;
+    s->dual_stereo_used += td2 < td1;
+}
+
+static void celt_search_for_intensity(OpusPsyContext *s, CeltFrame *f)
+{
+    int i, best_band = CELT_MAX_BANDS - 1;
+    float dist, best_dist = FLT_MAX;
+    /* TODO: fix, make some heuristic up here using the lambda value */
+    float end_band = 0;
+
+    if (s->avctx->ch_layout.nb_channels < 2)
+        return;
+
+    for (i = f->end_band; i >= end_band; i--) {
+        f->intensity_stereo = i;
+        bands_dist(s, f, &dist);
+        if (best_dist > dist) {
+            best_dist = dist;
+            best_band = i;
+        }
+    }
+
+    f->intensity_stereo = best_band;
+    s->avg_is_band = (s->avg_is_band + f->intensity_stereo)/2.0f;
+}
+
+static int celt_search_for_tf(OpusPsyContext *s, OpusPsyStep **start, CeltFrame *f)
+{
+    int i, j, k, cway, config[2][CELT_MAX_BANDS] = { { 0 } };
+    float score[2] = { 0 };
+
+    for (cway = 0; cway < 2; cway++) {
+        int mag[2];
+        int base = f->transient ? 120 : 960;
+
+        for (i = 0; i < 2; i++) {
+            int c = ff_celt_tf_select[f->size][f->transient][cway][i];
+            mag[i] = c < 0 ? base >> FFABS(c) : base << FFABS(c);
+        }
+
+        for (i = 0; i < CELT_MAX_BANDS; i++) {
+            float iscore0 = 0.0f;
+            float iscore1 = 0.0f;
+            for (j = 0; j < (1 << f->size); j++) {
+                for (k = 0; k < s->avctx->ch_layout.nb_channels; k++) {
+                    iscore0 += start[j]->tone[k][i]*start[j]->change_amp[k][i]/mag[0];
+                    iscore1 += start[j]->tone[k][i]*start[j]->change_amp[k][i]/mag[1];
+                }
+            }
+            config[cway][i] = FFABS(iscore0 - 1.0f) < FFABS(iscore1 - 1.0f);
+            score[cway] += config[cway][i] ? iscore1 : iscore0;
+        }
+    }
+
+    f->tf_select = score[0] < score[1];
+    memcpy(f->tf_change, config[f->tf_select], sizeof(int)*CELT_MAX_BANDS);
+
+    return 0;
+}
+
+int ff_opus_psy_celt_frame_process(OpusPsyContext *s, CeltFrame *f, int index)
+{
+    int start_transient_flag = f->transient;
+    OpusPsyStep **start = &s->steps[index * (1 << s->p.framesize)];
+
+    if (f->silence)
+        return 0;
+
+    celt_gauge_psy_weight(s, start, f);
+    celt_search_for_intensity(s, f);
+    celt_search_for_dual_stereo(s, f);
+    celt_search_for_tf(s, start, f);
+
+    if (f->transient != start_transient_flag) {
+        f->blocks = f->transient ? OPUS_BLOCK_SIZE(s->p.framesize)/CELT_OVERLAP : 1;
+        return 1;
+    }
+
+    return 0;
+}
+
+void ff_opus_psy_postencode_update(OpusPsyContext *s, CeltFrame *f)
+{
+    int i, frame_size = OPUS_BLOCK_SIZE(s->p.framesize);
+    int steps_out = s->p.frames*(frame_size/120);
+    void *tmp[FF_BUFQUEUE_SIZE];
+    float ideal_fbits;
+
+    for (i = 0; i < steps_out; i++)
+        memset(s->steps[i], 0, sizeof(OpusPsyStep));
+
+    for (i = 0; i < s->max_steps; i++)
+        tmp[i] = s->steps[i];
+
+    for (i = 0; i < s->max_steps; i++) {
+        const int i_new = i - steps_out;
+        s->steps[i_new < 0 ? s->max_steps + i_new : i_new] = tmp[i];
+    }
+
+    for (i = steps_out; i < s->buffered_steps; i++)
+        s->steps[i]->index -= steps_out;
+
+    ideal_fbits = s->avctx->bit_rate/(s->avctx->sample_rate/frame_size);
+
+    for (i = 0; i < s->p.frames; i++) {
+        s->avg_is_band += f[i].intensity_stereo;
+        s->lambda *= ideal_fbits / f[i].framebits;
+    }
+
+    s->avg_is_band /= (s->p.frames + 1);
+
+    s->steps_to_process = 0;
+    s->buffered_steps -= steps_out;
+    s->total_packets_out += s->p.frames;
+    s->inflection_points_count = 0;
+}
+
+av_cold int ff_opus_psy_init(OpusPsyContext *s, AVCodecContext *avctx,
+                             struct FFBufQueue *bufqueue, OpusEncOptions *options)
+{
+    int i, ch, ret;
+
+    s->lambda = 1.0f;
+    s->options = options;
+    s->avctx = avctx;
+    s->bufqueue = bufqueue;
+    s->max_steps = ceilf(s->options->max_delay_ms/2.5f);
+    s->bsize_analysis = CELT_BLOCK_960;
+    s->avg_is_band = CELT_MAX_BANDS - 1;
+    s->inflection_points_count = 0;
+
+    s->inflection_points = av_mallocz(sizeof(*s->inflection_points)*s->max_steps);
+    if (!s->inflection_points) {
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+
+    s->dsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
+    if (!s->dsp) {
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+
+    for (ch = 0; ch < s->avctx->ch_layout.nb_channels; ch++) {
+        for (i = 0; i < CELT_MAX_BANDS; i++) {
+            bessel_init(&s->bfilter_hi[ch][i], 1.0f, 19.0f, 100.0f, 1);
+            bessel_init(&s->bfilter_lo[ch][i], 1.0f, 20.0f, 100.0f, 0);
+        }
+    }
+
+    for (i = 0; i < s->max_steps; i++) {
+        s->steps[i] = av_mallocz(sizeof(OpusPsyStep));
+        if (!s->steps[i]) {
+            ret = AVERROR(ENOMEM);
+            goto fail;
+        }
+    }
+
+    for (i = 0; i < CELT_BLOCK_NB; i++) {
+        float tmp;
+        const int len = OPUS_BLOCK_SIZE(i);
+        const float scale = 68 << (CELT_BLOCK_NB - 1 - i);
+        s->window[i] = av_malloc(2*len*sizeof(float));
+        if (!s->window[i]) {
+            ret = AVERROR(ENOMEM);
+            goto fail;
+        }
+        generate_window_func(s->window[i], 2*len, WFUNC_SINE, &tmp);
+        ret = av_tx_init(&s->mdct[i], &s->mdct_fn[i], AV_TX_FLOAT_MDCT,
+                         0, 15 << (i + 3), &scale, 0);
+        if (ret < 0)
+            goto fail;
+    }
+
+    return 0;
+
+fail:
+    av_freep(&s->inflection_points);
+    av_freep(&s->dsp);
+
+    for (i = 0; i < CELT_BLOCK_NB; i++) {
+        av_tx_uninit(&s->mdct[i]);
+        av_freep(&s->window[i]);
+    }
+
+    for (i = 0; i < s->max_steps; i++)
+        av_freep(&s->steps[i]);
+
+    return ret;
+}
+
+void ff_opus_psy_signal_eof(OpusPsyContext *s)
+{
+    s->eof = 1;
+}
+
+av_cold int ff_opus_psy_end(OpusPsyContext *s)
+{
+    int i;
+
+    av_freep(&s->inflection_points);
+    av_freep(&s->dsp);
+
+    for (i = 0; i < CELT_BLOCK_NB; i++) {
+        av_tx_uninit(&s->mdct[i]);
+        av_freep(&s->window[i]);
+    }
+
+    for (i = 0; i < s->max_steps; i++)
+        av_freep(&s->steps[i]);
+
+    av_log(s->avctx, AV_LOG_INFO, "Average Intensity Stereo band: %0.1f\n", s->avg_is_band);
+    av_log(s->avctx, AV_LOG_INFO, "Dual Stereo used: %0.2f%%\n", ((float)s->dual_stereo_used/s->total_packets_out)*100.0f);
+
+    return 0;
+}
diff --git a/libavcodec/opus/enc_psy.h b/libavcodec/opus/enc_psy.h
new file mode 100644
index 0000000000..569a33c03f
--- /dev/null
+++ b/libavcodec/opus/enc_psy.h
@@ -0,0 +1,97 @@
+/*
+ * Opus encoder
+ * Copyright (c) 2017 Rostislav Pehlivanov <atomnuker@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_OPUS_ENC_PSY_H
+#define AVCODEC_OPUS_ENC_PSY_H
+
+#include "libavutil/tx.h"
+#include "libavutil/mem_internal.h"
+
+#include "enc.h"
+#include "celt.h"
+#include "enc_utils.h"
+
+/* Each step is 2.5ms */
+typedef struct OpusPsyStep {
+    int   index; /* Current index */
+    int   silence;
+    float energy[OPUS_MAX_CHANNELS][CELT_MAX_BANDS]; /* Masking effects included */
+    float tone[OPUS_MAX_CHANNELS][CELT_MAX_BANDS];   /* Tonality */
+    float stereo[CELT_MAX_BANDS];                    /* IS/MS compatibility */
+    float change_amp[OPUS_MAX_CHANNELS][CELT_MAX_BANDS]; /* Jump over last frame */
+    float total_change; /* Total change */
+
+    float *bands[OPUS_MAX_CHANNELS][CELT_MAX_BANDS];
+    float coeffs[OPUS_MAX_CHANNELS][OPUS_BLOCK_SIZE(CELT_BLOCK_960)];
+} OpusPsyStep;
+
+typedef struct OpusBandExcitation {
+    float excitation;
+    float excitation_dist;
+    float excitation_init;
+} OpusBandExcitation;
+
+typedef struct OpusPsyContext {
+    AVCodecContext *avctx;
+    AVFloatDSPContext *dsp;
+    struct FFBufQueue *bufqueue;
+    OpusEncOptions *options;
+
+    OpusBandExcitation ex[OPUS_MAX_CHANNELS][CELT_MAX_BANDS];
+    FFBesselFilter bfilter_lo[OPUS_MAX_CHANNELS][CELT_MAX_BANDS];
+    FFBesselFilter bfilter_hi[OPUS_MAX_CHANNELS][CELT_MAX_BANDS];
+
+    OpusPsyStep *steps[FF_BUFQUEUE_SIZE + 1];
+    int max_steps;
+
+    float *window[CELT_BLOCK_NB];
+    AVTXContext *mdct[CELT_BLOCK_NB];
+    av_tx_fn mdct_fn[CELT_BLOCK_NB];
+    int bsize_analysis;
+
+    DECLARE_ALIGNED(32, float, scratch)[2048];
+
+    /* Stats */
+    float avg_is_band;
+    int64_t dual_stereo_used;
+    int64_t total_packets_out;
+
+    /* State */
+    OpusPacketInfo p;
+    int buffered_steps;
+    int steps_to_process;
+    int eof;
+    float lambda;
+    int *inflection_points;
+    int inflection_points_count;
+} OpusPsyContext;
+
+int  ff_opus_psy_process           (OpusPsyContext *s, OpusPacketInfo *p);
+void ff_opus_psy_celt_frame_init   (OpusPsyContext *s, CeltFrame *f, int index);
+int  ff_opus_psy_celt_frame_process(OpusPsyContext *s, CeltFrame *f, int index);
+void ff_opus_psy_postencode_update (OpusPsyContext *s, CeltFrame *f);
+
+int  ff_opus_psy_init(OpusPsyContext *s, AVCodecContext *avctx,
+                      struct FFBufQueue *bufqueue, OpusEncOptions *options);
+void ff_opus_psy_signal_eof(OpusPsyContext *s);
+int  ff_opus_psy_end(OpusPsyContext *s);
+
+#endif /* AVCODEC_OPUS_ENC_PSY_H */
diff --git a/libavcodec/opus/enc_utils.h b/libavcodec/opus/enc_utils.h
new file mode 100644
index 0000000000..3ebcdbbb6f
--- /dev/null
+++ b/libavcodec/opus/enc_utils.h
@@ -0,0 +1,90 @@
+/*
+ * Opus encoder
+ * Copyright (c) 2017 Rostislav Pehlivanov <atomnuker@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_OPUS_ENC_UTILS_H
+#define AVCODEC_OPUS_ENC_UTILS_H
+
+#include <math.h>
+#include <string.h>
+
+#include "opus.h"
+
+typedef struct FFBesselFilter {
+    float a[3];
+    float b[2];
+    float x[3];
+    float y[3];
+} FFBesselFilter;
+
+/* Fills the coefficients, returns 1 if filter will be unstable */
+static inline int bessel_reinit(FFBesselFilter *s, float n, float f0, float fs,
+                                int highpass)
+{
+    int unstable;
+    float c, cfreq, w0, k1, k2;
+
+    if (!highpass) {
+        c = (1.0f/sqrtf(sqrtf(pow(2.0f, 1.0f/n) - 3.0f/4.0f) - 0.5f))/sqrtf(3.0f);
+        cfreq = c*f0/fs;
+        unstable = (cfreq <= 0.0f || cfreq >= 1.0f/4.0f);
+    } else {
+        c = sqrtf(3.0f)*sqrtf(sqrtf(pow(2.0f, 1.0f/n) - 3.0f/4.0f) - 0.5f);
+        cfreq = 0.5f - c*f0/fs;
+        unstable = (cfreq <= 3.0f/8.0f || cfreq >= 1.0f/2.0f);
+    }
+
+    w0 = tanf(M_PI*cfreq);
+    k1 = 3.0f * w0;
+    k2 = 3.0f * w0;
+
+    s->a[0] = k2/(1.0f + k1 + k2);
+    s->a[1] = 2.0f * s->a[0];
+    s->a[2] = s->a[0];
+    s->b[0] = 2.0f * s->a[0] * (1.0f/k2 - 1.0f);
+    s->b[1] = 1.0f - (s->a[0] + s->a[1] + s->a[2] + s->b[0]);
+
+    if (highpass) {
+        s->a[1] *= -1;
+        s->b[0] *= -1;
+    }
+
+    return unstable;
+}
+
+static inline int bessel_init(FFBesselFilter *s, float n, float f0, float fs,
+                              int highpass)
+{
+    memset(s, 0, sizeof(FFBesselFilter));
+    return bessel_reinit(s, n, f0, fs, highpass);
+}
+
+static inline float bessel_filter(FFBesselFilter *s, float x)
+{
+    s->x[2] = s->x[1];
+    s->x[1] = s->x[0];
+    s->x[0] = x;
+    s->y[2] = s->y[1];
+    s->y[1] = s->y[0];
+    s->y[0] = s->a[0]*s->x[0] + s->a[1]*s->x[1] + s->a[2]*s->x[2] + s->b[0]*s->y[1] + s->b[1]*s->y[2];
+    return s->y[0];
+}
+
+#endif /* AVCODEC_OPUS_ENC_UTILS_H */
diff --git a/libavcodec/opus/opus.h b/libavcodec/opus/opus.h
new file mode 100644
index 0000000000..9b1693329e
--- /dev/null
+++ b/libavcodec/opus/opus.h
@@ -0,0 +1,59 @@
+/*
+ * Opus common header
+ * Copyright (c) 2012 Andrew D'Addesio
+ * Copyright (c) 2013-2014 Mozilla Corporation
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_OPUS_OPUS_H
+#define AVCODEC_OPUS_OPUS_H
+
+#include <stdint.h>
+
+#define OPUS_MAX_FRAME_SIZE          1275
+#define OPUS_MAX_FRAMES                48
+#define OPUS_MAX_PACKET_DUR          5760
+
+#define OPUS_TS_HEADER     0x7FE0        // 0x3ff (11 bits)
+#define OPUS_TS_MASK       0xFFE0        // top 11 bits
+
+static const uint8_t opus_default_extradata[30] = {
+    'O', 'p', 'u', 's', 'H', 'e', 'a', 'd',
+    1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+};
+
+enum OpusMode {
+    OPUS_MODE_SILK,
+    OPUS_MODE_HYBRID,
+    OPUS_MODE_CELT,
+
+    OPUS_MODE_NB
+};
+
+enum OpusBandwidth {
+    OPUS_BANDWIDTH_NARROWBAND,
+    OPUS_BANDWIDTH_MEDIUMBAND,
+    OPUS_BANDWIDTH_WIDEBAND,
+    OPUS_BANDWIDTH_SUPERWIDEBAND,
+    OPUS_BANDWIDTH_FULLBAND,
+
+    OPUS_BANDWITH_NB
+};
+
+#endif /* AVCODEC_OPUS_OPUS_H */
diff --git a/libavcodec/opus/parse.c b/libavcodec/opus/parse.c
new file mode 100644
index 0000000000..78a2a75fc7
--- /dev/null
+++ b/libavcodec/opus/parse.c
@@ -0,0 +1,469 @@
+/*
+ * Copyright (c) 2012 Andrew D'Addesio
+ * Copyright (c) 2013-2014 Mozilla Corporation
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Opus decoder/parser shared code
+ */
+
+#include "libavutil/attributes.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/error.h"
+#include "libavutil/intreadwrite.h"
+#include "libavutil/log.h"
+#include "libavutil/mem.h"
+
+#include "avcodec.h"
+#include "internal.h"
+#include "mathops.h"
+#include "opus.h"
+#include "parse.h"
+#include "vorbis_data.h"
+
+static const uint16_t opus_frame_duration[32] = {
+    480, 960, 1920, 2880,
+    480, 960, 1920, 2880,
+    480, 960, 1920, 2880,
+    480, 960,
+    480, 960,
+    120, 240,  480,  960,
+    120, 240,  480,  960,
+    120, 240,  480,  960,
+    120, 240,  480,  960,
+};
+
+/**
+ * Read a 1- or 2-byte frame length
+ */
+static inline int xiph_lacing_16bit(const uint8_t **ptr, const uint8_t *end)
+{
+    int val;
+
+    if (*ptr >= end)
+        return AVERROR_INVALIDDATA;
+    val = *(*ptr)++;
+    if (val >= 252) {
+        if (*ptr >= end)
+            return AVERROR_INVALIDDATA;
+        val += 4 * *(*ptr)++;
+    }
+    return val;
+}
+
+/**
+ * Read a multi-byte length (used for code 3 packet padding size)
+ */
+static inline int xiph_lacing_full(const uint8_t **ptr, const uint8_t *end)
+{
+    int val = 0;
+    int next;
+
+    while (1) {
+        if (*ptr >= end || val > INT_MAX - 254)
+            return AVERROR_INVALIDDATA;
+        next = *(*ptr)++;
+        val += next;
+        if (next < 255)
+            break;
+        else
+            val--;
+    }
+    return val;
+}
+
+/**
+ * Parse Opus packet info from raw packet data
+ */
+int ff_opus_parse_packet(OpusPacket *pkt, const uint8_t *buf, int buf_size,
+                         int self_delimiting)
+{
+    const uint8_t *ptr = buf;
+    const uint8_t *end = buf + buf_size;
+    int padding = 0;
+    int frame_bytes, i;
+
+    if (buf_size < 1)
+        goto fail;
+
+    /* TOC byte */
+    i = *ptr++;
+    pkt->code   = (i     ) & 0x3;
+    pkt->stereo = (i >> 2) & 0x1;
+    pkt->config = (i >> 3) & 0x1F;
+
+    /* code 2 and code 3 packets have at least 1 byte after the TOC */
+    if (pkt->code >= 2 && buf_size < 2)
+        goto fail;
+
+    switch (pkt->code) {
+    case 0:
+        /* 1 frame */
+        pkt->frame_count = 1;
+        pkt->vbr         = 0;
+
+        if (self_delimiting) {
+            int len = xiph_lacing_16bit(&ptr, end);
+            if (len < 0 || len > end - ptr)
+                goto fail;
+            end      = ptr + len;
+            buf_size = end - buf;
+        }
+
+        frame_bytes = end - ptr;
+        if (frame_bytes > OPUS_MAX_FRAME_SIZE)
+            goto fail;
+        pkt->frame_offset[0] = ptr - buf;
+        pkt->frame_size[0]   = frame_bytes;
+        break;
+    case 1:
+        /* 2 frames, equal size */
+        pkt->frame_count = 2;
+        pkt->vbr         = 0;
+
+        if (self_delimiting) {
+            int len = xiph_lacing_16bit(&ptr, end);
+            if (len < 0 || 2 * len > end - ptr)
+                goto fail;
+            end      = ptr + 2 * len;
+            buf_size = end - buf;
+        }
+
+        frame_bytes = end - ptr;
+        if (frame_bytes & 1 || frame_bytes >> 1 > OPUS_MAX_FRAME_SIZE)
+            goto fail;
+        pkt->frame_offset[0] = ptr - buf;
+        pkt->frame_size[0]   = frame_bytes >> 1;
+        pkt->frame_offset[1] = pkt->frame_offset[0] + pkt->frame_size[0];
+        pkt->frame_size[1]   = frame_bytes >> 1;
+        break;
+    case 2:
+        /* 2 frames, different sizes */
+        pkt->frame_count = 2;
+        pkt->vbr         = 1;
+
+        /* read 1st frame size */
+        frame_bytes = xiph_lacing_16bit(&ptr, end);
+        if (frame_bytes < 0)
+            goto fail;
+
+        if (self_delimiting) {
+            int len = xiph_lacing_16bit(&ptr, end);
+            if (len < 0 || len + frame_bytes > end - ptr)
+                goto fail;
+            end      = ptr + frame_bytes + len;
+            buf_size = end - buf;
+        }
+
+        pkt->frame_offset[0] = ptr - buf;
+        pkt->frame_size[0]   = frame_bytes;
+
+        /* calculate 2nd frame size */
+        frame_bytes = end - ptr - pkt->frame_size[0];
+        if (frame_bytes < 0 || frame_bytes > OPUS_MAX_FRAME_SIZE)
+            goto fail;
+        pkt->frame_offset[1] = pkt->frame_offset[0] + pkt->frame_size[0];
+        pkt->frame_size[1]   = frame_bytes;
+        break;
+    case 3:
+        /* 1 to 48 frames, can be different sizes */
+        i = *ptr++;
+        pkt->frame_count = (i     ) & 0x3F;
+        padding          = (i >> 6) & 0x01;
+        pkt->vbr         = (i >> 7) & 0x01;
+
+        if (pkt->frame_count == 0 || pkt->frame_count > OPUS_MAX_FRAMES)
+            goto fail;
+
+        /* read padding size */
+        if (padding) {
+            padding = xiph_lacing_full(&ptr, end);
+            if (padding < 0)
+                goto fail;
+        }
+
+        /* read frame sizes */
+        if (pkt->vbr) {
+            /* for VBR, all frames except the final one have their size coded
+               in the bitstream. the last frame size is implicit. */
+            int total_bytes = 0;
+            for (i = 0; i < pkt->frame_count - 1; i++) {
+                frame_bytes = xiph_lacing_16bit(&ptr, end);
+                if (frame_bytes < 0)
+                    goto fail;
+                pkt->frame_size[i] = frame_bytes;
+                total_bytes += frame_bytes;
+            }
+
+            if (self_delimiting) {
+                int len = xiph_lacing_16bit(&ptr, end);
+                if (len < 0 || len + total_bytes + padding > end - ptr)
+                    goto fail;
+                end      = ptr + total_bytes + len + padding;
+                buf_size = end - buf;
+            }
+
+            frame_bytes = end - ptr - padding;
+            if (total_bytes > frame_bytes)
+                goto fail;
+            pkt->frame_offset[0] = ptr - buf;
+            for (i = 1; i < pkt->frame_count; i++)
+                pkt->frame_offset[i] = pkt->frame_offset[i-1] + pkt->frame_size[i-1];
+            pkt->frame_size[pkt->frame_count-1] = frame_bytes - total_bytes;
+        } else {
+            /* for CBR, the remaining packet bytes are divided evenly between
+               the frames */
+            if (self_delimiting) {
+                frame_bytes = xiph_lacing_16bit(&ptr, end);
+                if (frame_bytes < 0 || pkt->frame_count * frame_bytes + padding > end - ptr)
+                    goto fail;
+                end      = ptr + pkt->frame_count * frame_bytes + padding;
+                buf_size = end - buf;
+            } else {
+                frame_bytes = end - ptr - padding;
+                if (frame_bytes % pkt->frame_count ||
+                    frame_bytes / pkt->frame_count > OPUS_MAX_FRAME_SIZE)
+                    goto fail;
+                frame_bytes /= pkt->frame_count;
+            }
+
+            pkt->frame_offset[0] = ptr - buf;
+            pkt->frame_size[0]   = frame_bytes;
+            for (i = 1; i < pkt->frame_count; i++) {
+                pkt->frame_offset[i] = pkt->frame_offset[i-1] + pkt->frame_size[i-1];
+                pkt->frame_size[i]   = frame_bytes;
+            }
+        }
+    }
+
+    pkt->packet_size = buf_size;
+    pkt->data_size   = pkt->packet_size - padding;
+
+    /* total packet duration cannot be larger than 120ms */
+    pkt->frame_duration = opus_frame_duration[pkt->config];
+    if (pkt->frame_duration * pkt->frame_count > OPUS_MAX_PACKET_DUR)
+        goto fail;
+
+    /* set mode and bandwidth */
+    if (pkt->config < 12) {
+        pkt->mode = OPUS_MODE_SILK;
+        pkt->bandwidth = pkt->config >> 2;
+    } else if (pkt->config < 16) {
+        pkt->mode = OPUS_MODE_HYBRID;
+        pkt->bandwidth = OPUS_BANDWIDTH_SUPERWIDEBAND + (pkt->config >= 14);
+    } else {
+        pkt->mode = OPUS_MODE_CELT;
+        pkt->bandwidth = (pkt->config - 16) >> 2;
+        /* skip medium band */
+        if (pkt->bandwidth)
+            pkt->bandwidth++;
+    }
+
+    return 0;
+
+fail:
+    memset(pkt, 0, sizeof(*pkt));
+    return AVERROR_INVALIDDATA;
+}
+
+static int channel_reorder_vorbis(int nb_channels, int channel_idx)
+{
+    return ff_vorbis_channel_layout_offsets[nb_channels - 1][channel_idx];
+}
+
+static int channel_reorder_unknown(int nb_channels, int channel_idx)
+{
+    return channel_idx;
+}
+
+av_cold int ff_opus_parse_extradata(AVCodecContext *avctx,
+                                    OpusParseContext *s)
+{
+    static const uint8_t default_channel_map[2] = { 0, 1 };
+
+    int (*channel_reorder)(int, int) = channel_reorder_unknown;
+    int channels = avctx->ch_layout.nb_channels;
+
+    const uint8_t *extradata, *channel_map;
+    int extradata_size;
+    int version, map_type, streams, stereo_streams, i, j, ret;
+    AVChannelLayout layout = { 0 };
+
+    if (!avctx->extradata) {
+        if (channels > 2) {
+            av_log(avctx, AV_LOG_ERROR,
+                   "Multichannel configuration without extradata.\n");
+            return AVERROR(EINVAL);
+        }
+        extradata      = opus_default_extradata;
+        extradata_size = sizeof(opus_default_extradata);
+    } else {
+        extradata = avctx->extradata;
+        extradata_size = avctx->extradata_size;
+    }
+
+    if (extradata_size < 19) {
+        av_log(avctx, AV_LOG_ERROR, "Invalid extradata size: %d\n",
+               extradata_size);
+        return AVERROR_INVALIDDATA;
+    }
+
+    version = extradata[8];
+    if (version > 15) {
+        avpriv_request_sample(avctx, "Extradata version %d", version);
+        return AVERROR_PATCHWELCOME;
+    }
+
+    avctx->delay = AV_RL16(extradata + 10);
+    if (avctx->internal)
+        avctx->internal->skip_samples = avctx->delay;
+
+    channels = avctx->extradata ? extradata[9] : (channels == 1) ? 1 : 2;
+    if (!channels) {
+        av_log(avctx, AV_LOG_ERROR, "Zero channel count specified in the extradata\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    s->gain_i = AV_RL16(extradata + 16);
+
+    map_type = extradata[18];
+    if (!map_type) {
+        if (channels > 2) {
+            av_log(avctx, AV_LOG_ERROR,
+                   "Channel mapping 0 is only specified for up to 2 channels\n");
+            ret = AVERROR_INVALIDDATA;
+            goto fail;
+        }
+        layout         = (channels == 1) ? (AVChannelLayout)AV_CHANNEL_LAYOUT_MONO :
+                                           (AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO;
+        streams        = 1;
+        stereo_streams = channels - 1;
+        channel_map    = default_channel_map;
+    } else if (map_type == 1 || map_type == 2 || map_type == 255) {
+        if (extradata_size < 21 + channels) {
+            av_log(avctx, AV_LOG_ERROR, "Invalid extradata size: %d\n",
+                   extradata_size);
+            ret = AVERROR_INVALIDDATA;
+            goto fail;
+        }
+
+        streams        = extradata[19];
+        stereo_streams = extradata[20];
+        if (!streams || stereo_streams > streams ||
+            streams + stereo_streams > 255) {
+            av_log(avctx, AV_LOG_ERROR,
+                   "Invalid stream/stereo stream count: %d/%d\n", streams, stereo_streams);
+            ret = AVERROR_INVALIDDATA;
+            goto fail;
+        }
+
+        if (map_type == 1) {
+            if (channels > 8) {
+                av_log(avctx, AV_LOG_ERROR,
+                       "Channel mapping 1 is only specified for up to 8 channels\n");
+                ret = AVERROR_INVALIDDATA;
+                goto fail;
+            }
+            av_channel_layout_copy(&layout, &ff_vorbis_ch_layouts[channels - 1]);
+            channel_reorder = channel_reorder_vorbis;
+        } else if (map_type == 2) {
+            int ambisonic_order = ff_sqrt(channels) - 1;
+            if (channels != ((ambisonic_order + 1) * (ambisonic_order + 1)) &&
+                channels != ((ambisonic_order + 1) * (ambisonic_order + 1) + 2)) {
+                av_log(avctx, AV_LOG_ERROR,
+                       "Channel mapping 2 is only specified for channel counts"
+                       " which can be written as (n + 1)^2 or (n + 1)^2 + 2"
+                       " for nonnegative integer n\n");
+                ret = AVERROR_INVALIDDATA;
+                goto fail;
+            }
+            if (channels > 227) {
+                av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
+                ret = AVERROR_INVALIDDATA;
+                goto fail;
+            }
+
+            layout.order = AV_CHANNEL_ORDER_AMBISONIC;
+            layout.nb_channels = channels;
+            if (channels != ((ambisonic_order + 1) * (ambisonic_order + 1)))
+                layout.u.mask = AV_CH_LAYOUT_STEREO;
+        } else {
+            layout.order       = AV_CHANNEL_ORDER_UNSPEC;
+            layout.nb_channels = channels;
+        }
+
+        channel_map = extradata + 21;
+    } else {
+        avpriv_request_sample(avctx, "Mapping type %d", map_type);
+        return AVERROR_PATCHWELCOME;
+    }
+
+    s->channel_maps = av_calloc(channels, sizeof(*s->channel_maps));
+    if (!s->channel_maps) {
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+
+    for (i = 0; i < channels; i++) {
+        ChannelMap *map = &s->channel_maps[i];
+        uint8_t     idx = channel_map[channel_reorder(channels, i)];
+
+        if (idx == 255) {
+            map->silence = 1;
+            continue;
+        } else if (idx >= streams + stereo_streams) {
+            av_log(avctx, AV_LOG_ERROR,
+                   "Invalid channel map for output channel %d: %d\n", i, idx);
+            av_freep(&s->channel_maps);
+            ret = AVERROR_INVALIDDATA;
+            goto fail;
+        }
+
+        /* check that we did not see this index yet */
+        map->copy = 0;
+        for (j = 0; j < i; j++)
+            if (channel_map[channel_reorder(channels, j)] == idx) {
+                map->copy     = 1;
+                map->copy_idx = j;
+                break;
+            }
+
+        if (idx < 2 * stereo_streams) {
+            map->stream_idx  = idx / 2;
+            map->channel_idx = idx & 1;
+        } else {
+            map->stream_idx  = idx - stereo_streams;
+            map->channel_idx = 0;
+        }
+    }
+
+    ret = av_channel_layout_copy(&avctx->ch_layout, &layout);
+    if (ret < 0)
+        goto fail;
+
+    s->nb_streams         = streams;
+    s->nb_stereo_streams  = stereo_streams;
+
+    return 0;
+fail:
+    av_channel_layout_uninit(&layout);
+    return ret;
+}
+
diff --git a/libavcodec/opus/parse.h b/libavcodec/opus/parse.h
new file mode 100644
index 0000000000..467957364f
--- /dev/null
+++ b/libavcodec/opus/parse.h
@@ -0,0 +1,77 @@
+/*
+ * Opus decoder/parser common functions and structures
+ * Copyright (c) 2012 Andrew D'Addesio
+ * Copyright (c) 2013-2014 Mozilla Corporation
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_OPUS_PARSE_H
+#define AVCODEC_OPUS_PARSE_H
+
+#include <stdint.h>
+
+#include "libavcodec/avcodec.h"
+#include "opus.h"
+
+typedef struct OpusPacket {
+    int packet_size;                /**< packet size */
+    int data_size;                  /**< size of the useful data -- packet size - padding */
+    int code;                       /**< packet code: specifies the frame layout */
+    int stereo;                     /**< whether this packet is mono or stereo */
+    int vbr;                        /**< vbr flag */
+    int config;                     /**< configuration: tells the audio mode,
+                                     **                bandwidth, and frame duration */
+    int frame_count;                /**< frame count */
+    int frame_offset[OPUS_MAX_FRAMES]; /**< frame offsets */
+    int frame_size[OPUS_MAX_FRAMES]; /**< frame sizes */
+    int frame_duration;             /**< frame duration, in samples @ 48kHz */
+    enum OpusMode mode;             /**< mode */
+    enum OpusBandwidth bandwidth;   /**< bandwidth */
+} OpusPacket;
+
+// a mapping between an opus stream and an output channel
+typedef struct ChannelMap {
+    int stream_idx;
+    int channel_idx;
+
+    // when a single decoded channel is mapped to multiple output channels, we
+    // write to the first output directly and copy from it to the others
+    // this field is set to 1 for those copied output channels
+    int copy;
+    // this is the index of the output channel to copy from
+    int copy_idx;
+
+    // this channel is silent
+    int silence;
+} ChannelMap;
+
+typedef struct OpusParseContext {
+    int             nb_streams;
+    int      nb_stereo_streams;
+
+    int16_t gain_i;
+
+    ChannelMap *channel_maps;
+} OpusParseContext;
+
+int ff_opus_parse_packet(OpusPacket *pkt, const uint8_t *buf, int buf_size,
+                         int self_delimited);
+
+int ff_opus_parse_extradata(AVCodecContext *avctx, OpusParseContext *s);
+
+#endif /* AVCODEC_OPUS_PARSE_H */
diff --git a/libavcodec/opus/parser.c b/libavcodec/opus/parser.c
new file mode 100644
index 0000000000..41665e68f9
--- /dev/null
+++ b/libavcodec/opus/parser.c
@@ -0,0 +1,200 @@
+/*
+ * Copyright (c) 2013-2014 Mozilla Corporation
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Opus parser
+ *
+ * Determines the duration for each packet.
+ */
+
+#include "libavutil/mem.h"
+#include "avcodec.h"
+#include "bytestream.h"
+#include "opus.h"
+#include "parse.h"
+#include "parser.h"
+
+typedef struct OpusParserContext {
+    ParseContext pc;
+    OpusParseContext ctx;
+    OpusPacket pkt;
+    int extradata_parsed;
+    int ts_framing;
+} OpusParserContext;
+
+static const uint8_t *parse_opus_ts_header(const uint8_t *start, int *payload_len, int buf_len)
+{
+    const uint8_t *buf = start + 1;
+    int start_trim_flag, end_trim_flag, control_extension_flag, control_extension_length;
+    uint8_t flags;
+    uint64_t payload_len_tmp;
+
+    GetByteContext gb;
+    bytestream2_init(&gb, buf, buf_len);
+
+    flags = bytestream2_get_byte(&gb);
+    start_trim_flag        = (flags >> 4) & 1;
+    end_trim_flag          = (flags >> 3) & 1;
+    control_extension_flag = (flags >> 2) & 1;
+
+    payload_len_tmp = *payload_len = 0;
+    while (bytestream2_peek_byte(&gb) == 0xff)
+        payload_len_tmp += bytestream2_get_byte(&gb);
+
+    payload_len_tmp += bytestream2_get_byte(&gb);
+
+    if (start_trim_flag)
+        bytestream2_skip(&gb, 2);
+    if (end_trim_flag)
+        bytestream2_skip(&gb, 2);
+    if (control_extension_flag) {
+        control_extension_length = bytestream2_get_byte(&gb);
+        bytestream2_skip(&gb, control_extension_length);
+    }
+
+    if (bytestream2_tell(&gb) + payload_len_tmp > buf_len)
+        return NULL;
+
+    *payload_len = payload_len_tmp;
+
+    return buf + bytestream2_tell(&gb);
+}
+
+/**
+ * Find the end of the current frame in the bitstream.
+ * @return the position of the first byte of the next frame, or -1
+ */
+static int opus_find_frame_end(AVCodecParserContext *ctx, AVCodecContext *avctx,
+                               const uint8_t *buf, int buf_size, int *header_len)
+{
+    OpusParserContext *s = ctx->priv_data;
+    ParseContext *pc    = &s->pc;
+    int ret, start_found, i = 0, payload_len = 0;
+    const uint8_t *payload;
+    uint32_t state;
+    uint16_t hdr;
+    *header_len = 0;
+
+    if (!buf_size)
+        return 0;
+
+    start_found = pc->frame_start_found;
+    state = pc->state;
+    payload = buf;
+
+    /* Check if we're using Opus in MPEG-TS framing */
+    if (!s->ts_framing && buf_size > 2) {
+        hdr = AV_RB16(buf);
+        if ((hdr & OPUS_TS_MASK) == OPUS_TS_HEADER)
+            s->ts_framing = 1;
+    }
+
+    if (s->ts_framing && !start_found) {
+        for (i = 0; i < buf_size-2; i++) {
+            state = (state << 8) | payload[i];
+            if ((state & OPUS_TS_MASK) == OPUS_TS_HEADER) {
+                payload = parse_opus_ts_header(payload, &payload_len, buf_size - i);
+                if (!payload) {
+                    av_log(avctx, AV_LOG_ERROR, "Error parsing Ogg TS header.\n");
+                    return AVERROR_INVALIDDATA;
+                }
+                *header_len = payload - buf;
+                start_found = 1;
+                break;
+            }
+        }
+    }
+
+    if (!s->ts_framing)
+        payload_len = buf_size;
+
+    if (avctx->extradata && !s->extradata_parsed) {
+        ret = ff_opus_parse_extradata(avctx, &s->ctx);
+        if (ret < 0) {
+            av_log(avctx, AV_LOG_ERROR, "Error parsing Ogg extradata.\n");
+            return AVERROR_INVALIDDATA;
+        }
+        av_freep(&s->ctx.channel_maps);
+        s->extradata_parsed = 1;
+    }
+
+    if (payload_len <= buf_size && (!s->ts_framing || start_found)) {
+        ret = ff_opus_parse_packet(&s->pkt, payload, payload_len, s->ctx.nb_streams > 1);
+        if (ret < 0) {
+            av_log(avctx, AV_LOG_ERROR, "Error parsing Opus packet header.\n");
+            pc->frame_start_found = 0;
+            return AVERROR_INVALIDDATA;
+        }
+
+        ctx->duration = s->pkt.frame_count * s->pkt.frame_duration;
+    }
+
+    if (s->ts_framing) {
+        if (start_found) {
+            if (payload_len + *header_len <= buf_size) {
+                pc->frame_start_found = 0;
+                pc->state             = -1;
+                return payload_len + *header_len;
+            }
+        }
+
+        pc->frame_start_found = start_found;
+        pc->state = state;
+        return END_NOT_FOUND;
+    }
+
+    return buf_size;
+}
+
+static int opus_parse(AVCodecParserContext *ctx, AVCodecContext *avctx,
+                       const uint8_t **poutbuf, int *poutbuf_size,
+                       const uint8_t *buf, int buf_size)
+{
+    OpusParserContext *s = ctx->priv_data;
+    ParseContext *pc    = &s->pc;
+    int next, header_len;
+
+    next = opus_find_frame_end(ctx, avctx, buf, buf_size, &header_len);
+
+    if (s->ts_framing && next != AVERROR_INVALIDDATA &&
+        ff_combine_frame(pc, next, &buf, &buf_size) < 0) {
+        *poutbuf      = NULL;
+        *poutbuf_size = 0;
+        return buf_size;
+    }
+
+    if (next == AVERROR_INVALIDDATA){
+        *poutbuf      = NULL;
+        *poutbuf_size = 0;
+        return buf_size;
+    }
+
+    *poutbuf      = buf + header_len;
+    *poutbuf_size = buf_size - header_len;
+    return next;
+}
+
+const AVCodecParser ff_opus_parser = {
+    .codec_ids      = { AV_CODEC_ID_OPUS },
+    .priv_data_size = sizeof(OpusParserContext),
+    .parser_parse   = opus_parse,
+    .parser_close   = ff_parse_close
+};
diff --git a/libavcodec/opus/pvq.c b/libavcodec/opus/pvq.c
new file mode 100644
index 0000000000..fe57ab02ce
--- /dev/null
+++ b/libavcodec/opus/pvq.c
@@ -0,0 +1,930 @@
+/*
+ * Copyright (c) 2007-2008 CSIRO
+ * Copyright (c) 2007-2009 Xiph.Org Foundation
+ * Copyright (c) 2008-2009 Gregory Maxwell
+ * Copyright (c) 2012 Andrew D'Addesio
+ * Copyright (c) 2013-2014 Mozilla Corporation
+ * Copyright (c) 2017 Rostislav Pehlivanov <atomnuker@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include <float.h>
+
+#include "config_components.h"
+
+#include "libavutil/mem.h"
+#include "mathops.h"
+#include "tab.h"
+#include "pvq.h"
+
+#define ROUND_MUL16(a,b)  ((MUL16(a, b) + 16384) >> 15)
+
+#define CELT_PVQ_U(n, k) (ff_celt_pvq_u_row[FFMIN(n, k)][FFMAX(n, k)])
+#define CELT_PVQ_V(n, k) (CELT_PVQ_U(n, k) + CELT_PVQ_U(n, (k) + 1))
+
+static inline int16_t celt_cos(int16_t x)
+{
+    x = (MUL16(x, x) + 4096) >> 13;
+    x = (32767-x) + ROUND_MUL16(x, (-7651 + ROUND_MUL16(x, (8277 + ROUND_MUL16(-626, x)))));
+    return x + 1;
+}
+
+static inline int celt_log2tan(int isin, int icos)
+{
+    int lc, ls;
+    lc = opus_ilog(icos);
+    ls = opus_ilog(isin);
+    icos <<= 15 - lc;
+    isin <<= 15 - ls;
+    return (ls << 11) - (lc << 11) +
+           ROUND_MUL16(isin, ROUND_MUL16(isin, -2597) + 7932) -
+           ROUND_MUL16(icos, ROUND_MUL16(icos, -2597) + 7932);
+}
+
+static inline int celt_bits2pulses(const uint8_t *cache, int bits)
+{
+    // TODO: Find the size of cache and make it into an array in the parameters list
+    int i, low = 0, high;
+
+    high = cache[0];
+    bits--;
+
+    for (i = 0; i < 6; i++) {
+        int center = (low + high + 1) >> 1;
+        if (cache[center] >= bits)
+            high = center;
+        else
+            low = center;
+    }
+
+    return (bits - (low == 0 ? -1 : cache[low]) <= cache[high] - bits) ? low : high;
+}
+
+static inline int celt_pulses2bits(const uint8_t *cache, int pulses)
+{
+    // TODO: Find the size of cache and make it into an array in the parameters list
+   return (pulses == 0) ? 0 : cache[pulses] + 1;
+}
+
+static inline void celt_normalize_residual(const int * restrict iy, float * restrict X,
+                                           int N, float g)
+{
+    int i;
+    for (i = 0; i < N; i++)
+        X[i] = g * iy[i];
+}
+
+static void celt_exp_rotation_impl(float *X, uint32_t len, uint32_t stride,
+                                   float c, float s)
+{
+    float *Xptr;
+    int i;
+
+    Xptr = X;
+    for (i = 0; i < len - stride; i++) {
+        float x1     = Xptr[0];
+        float x2     = Xptr[stride];
+        Xptr[stride] = c * x2 + s * x1;
+        *Xptr++      = c * x1 - s * x2;
+    }
+
+    Xptr = &X[len - 2 * stride - 1];
+    for (i = len - 2 * stride - 1; i >= 0; i--) {
+        float x1     = Xptr[0];
+        float x2     = Xptr[stride];
+        Xptr[stride] = c * x2 + s * x1;
+        *Xptr--      = c * x1 - s * x2;
+    }
+}
+
+static inline void celt_exp_rotation(float *X, uint32_t len,
+                                     uint32_t stride, uint32_t K,
+                                     enum CeltSpread spread, const int encode)
+{
+    uint32_t stride2 = 0;
+    float c, s;
+    float gain, theta;
+    int i;
+
+    if (2*K >= len || spread == CELT_SPREAD_NONE)
+        return;
+
+    gain = (float)len / (len + (20 - 5*spread) * K);
+    theta = M_PI * gain * gain / 4;
+
+    c = cosf(theta);
+    s = sinf(theta);
+
+    if (len >= stride << 3) {
+        stride2 = 1;
+        /* This is just a simple (equivalent) way of computing sqrt(len/stride) with rounding.
+        It's basically incrementing long as (stride2+0.5)^2 < len/stride. */
+        while ((stride2 * stride2 + stride2) * stride + (stride >> 2) < len)
+            stride2++;
+    }
+
+    len /= stride;
+    for (i = 0; i < stride; i++) {
+        if (encode) {
+            celt_exp_rotation_impl(X + i * len, len, 1, c, -s);
+            if (stride2)
+                celt_exp_rotation_impl(X + i * len, len, stride2, s, -c);
+        } else {
+            if (stride2)
+                celt_exp_rotation_impl(X + i * len, len, stride2, s, c);
+            celt_exp_rotation_impl(X + i * len, len, 1, c, s);
+        }
+    }
+}
+
+static inline uint32_t celt_extract_collapse_mask(const int *iy, uint32_t N, uint32_t B)
+{
+    int i, j, N0 = N / B;
+    uint32_t collapse_mask = 0;
+
+    if (B <= 1)
+        return 1;
+
+    for (i = 0; i < B; i++)
+        for (j = 0; j < N0; j++)
+            collapse_mask |= (!!iy[i*N0+j]) << i;
+    return collapse_mask;
+}
+
+static inline void celt_stereo_merge(float *X, float *Y, float mid, int N)
+{
+    int i;
+    float xp = 0, side = 0;
+    float E[2];
+    float mid2;
+    float gain[2];
+
+    /* Compute the norm of X+Y and X-Y as |X|^2 + |Y|^2 +/- sum(xy) */
+    for (i = 0; i < N; i++) {
+        xp   += X[i] * Y[i];
+        side += Y[i] * Y[i];
+    }
+
+    /* Compensating for the mid normalization */
+    xp *= mid;
+    mid2 = mid;
+    E[0] = mid2 * mid2 + side - 2 * xp;
+    E[1] = mid2 * mid2 + side + 2 * xp;
+    if (E[0] < 6e-4f || E[1] < 6e-4f) {
+        for (i = 0; i < N; i++)
+            Y[i] = X[i];
+        return;
+    }
+
+    gain[0] = 1.0f / sqrtf(E[0]);
+    gain[1] = 1.0f / sqrtf(E[1]);
+
+    for (i = 0; i < N; i++) {
+        float value[2];
+        /* Apply mid scaling (side is already scaled) */
+        value[0] = mid * X[i];
+        value[1] = Y[i];
+        X[i] = gain[0] * (value[0] - value[1]);
+        Y[i] = gain[1] * (value[0] + value[1]);
+    }
+}
+
+static void celt_interleave_hadamard(float *tmp, float *X, int N0,
+                                     int stride, int hadamard)
+{
+    int i, j, N = N0*stride;
+    const uint8_t *order = &ff_celt_hadamard_order[hadamard ? stride - 2 : 30];
+
+    for (i = 0; i < stride; i++)
+        for (j = 0; j < N0; j++)
+            tmp[j*stride+i] = X[order[i]*N0+j];
+
+    memcpy(X, tmp, N*sizeof(float));
+}
+
+static void celt_deinterleave_hadamard(float *tmp, float *X, int N0,
+                                       int stride, int hadamard)
+{
+    int i, j, N = N0*stride;
+    const uint8_t *order = &ff_celt_hadamard_order[hadamard ? stride - 2 : 30];
+
+    for (i = 0; i < stride; i++)
+        for (j = 0; j < N0; j++)
+            tmp[order[i]*N0+j] = X[j*stride+i];
+
+    memcpy(X, tmp, N*sizeof(float));
+}
+
+static void celt_haar1(float *X, int N0, int stride)
+{
+    int i, j;
+    N0 >>= 1;
+    for (i = 0; i < stride; i++) {
+        for (j = 0; j < N0; j++) {
+            float x0 = X[stride * (2 * j + 0) + i];
+            float x1 = X[stride * (2 * j + 1) + i];
+            X[stride * (2 * j + 0) + i] = (x0 + x1) * M_SQRT1_2;
+            X[stride * (2 * j + 1) + i] = (x0 - x1) * M_SQRT1_2;
+        }
+    }
+}
+
+static inline int celt_compute_qn(int N, int b, int offset, int pulse_cap,
+                                  int stereo)
+{
+    int qn, qb;
+    int N2 = 2 * N - 1;
+    if (stereo && N == 2)
+        N2--;
+
+    /* The upper limit ensures that in a stereo split with itheta==16384, we'll
+     * always have enough bits left over to code at least one pulse in the
+     * side; otherwise it would collapse, since it doesn't get folded. */
+    qb = FFMIN3(b - pulse_cap - (4 << 3), (b + N2 * offset) / N2, 8 << 3);
+    qn = (qb < (1 << 3 >> 1)) ? 1 : ((ff_celt_qn_exp2[qb & 0x7] >> (14 - (qb >> 3))) + 1) >> 1 << 1;
+    return qn;
+}
+
+/* Convert the quantized vector to an index */
+static inline uint32_t celt_icwrsi(uint32_t N, uint32_t K, const int *y)
+{
+    int i, idx = 0, sum = 0;
+    for (i = N - 1; i >= 0; i--) {
+        const uint32_t i_s = CELT_PVQ_U(N - i, sum + FFABS(y[i]) + 1);
+        idx += CELT_PVQ_U(N - i, sum) + (y[i] < 0)*i_s;
+        sum += FFABS(y[i]);
+    }
+    return idx;
+}
+
+// this code was adapted from libopus
+static inline uint64_t celt_cwrsi(uint32_t N, uint32_t K, uint32_t i, int *y)
+{
+    uint64_t norm = 0;
+    uint32_t q, p;
+    int s, val;
+    int k0;
+
+    while (N > 2) {
+        /*Lots of pulses case:*/
+        if (K >= N) {
+            const uint32_t *row = ff_celt_pvq_u_row[N];
+
+            /* Are the pulses in this dimension negative? */
+            p  = row[K + 1];
+            s  = -(i >= p);
+            i -= p & s;
+
+            /*Count how many pulses were placed in this dimension.*/
+            k0 = K;
+            q = row[N];
+            if (q > i) {
+                K = N;
+                do {
+                    p = ff_celt_pvq_u_row[--K][N];
+                } while (p > i);
+            } else
+                for (p = row[K]; p > i; p = row[K])
+                    K--;
+
+            i    -= p;
+            val   = (k0 - K + s) ^ s;
+            norm += val * val;
+            *y++  = val;
+        } else { /*Lots of dimensions case:*/
+            /*Are there any pulses in this dimension at all?*/
+            p = ff_celt_pvq_u_row[K    ][N];
+            q = ff_celt_pvq_u_row[K + 1][N];
+
+            if (p <= i && i < q) {
+                i -= p;
+                *y++ = 0;
+            } else {
+                /*Are the pulses in this dimension negative?*/
+                s  = -(i >= q);
+                i -= q & s;
+
+                /*Count how many pulses were placed in this dimension.*/
+                k0 = K;
+                do p = ff_celt_pvq_u_row[--K][N];
+                while (p > i);
+
+                i    -= p;
+                val   = (k0 - K + s) ^ s;
+                norm += val * val;
+                *y++  = val;
+            }
+        }
+        N--;
+    }
+
+    /* N == 2 */
+    p  = 2 * K + 1;
+    s  = -(i >= p);
+    i -= p & s;
+    k0 = K;
+    K  = (i + 1) / 2;
+
+    if (K)
+        i -= 2 * K - 1;
+
+    val   = (k0 - K + s) ^ s;
+    norm += val * val;
+    *y++  = val;
+
+    /* N==1 */
+    s     = -i;
+    val   = (K + s) ^ s;
+    norm += val * val;
+    *y    = val;
+
+    return norm;
+}
+
+static inline void celt_encode_pulses(OpusRangeCoder *rc, int *y, uint32_t N, uint32_t K)
+{
+    ff_opus_rc_enc_uint(rc, celt_icwrsi(N, K, y), CELT_PVQ_V(N, K));
+}
+
+static inline float celt_decode_pulses(OpusRangeCoder *rc, int *y, uint32_t N, uint32_t K)
+{
+    const uint32_t idx = ff_opus_rc_dec_uint(rc, CELT_PVQ_V(N, K));
+    return celt_cwrsi(N, K, idx, y);
+}
+
+#if CONFIG_OPUS_ENCODER
+/*
+ * Faster than libopus's search, operates entirely in the signed domain.
+ * Slightly worse/better depending on N, K and the input vector.
+ */
+static float ppp_pvq_search_c(float *X, int *y, int K, int N)
+{
+    int i, y_norm = 0;
+    float res = 0.0f, xy_norm = 0.0f;
+
+    for (i = 0; i < N; i++)
+        res += FFABS(X[i]);
+
+    res = K/(res + FLT_EPSILON);
+
+    for (i = 0; i < N; i++) {
+        y[i] = lrintf(res*X[i]);
+        y_norm  += y[i]*y[i];
+        xy_norm += y[i]*X[i];
+        K -= FFABS(y[i]);
+    }
+
+    while (K) {
+        int max_idx = 0, phase = FFSIGN(K);
+        float max_num = 0.0f;
+        float max_den = 1.0f;
+        y_norm += 1.0f;
+
+        for (i = 0; i < N; i++) {
+            /* If the sum has been overshot and the best place has 0 pulses allocated
+             * to it, attempting to decrease it further will actually increase the
+             * sum. Prevent this by disregarding any 0 positions when decrementing. */
+            const int ca = 1 ^ ((y[i] == 0) & (phase < 0));
+            const int y_new = y_norm  + 2*phase*FFABS(y[i]);
+            float xy_new = xy_norm + 1*phase*FFABS(X[i]);
+            xy_new = xy_new * xy_new;
+            if (ca && (max_den*xy_new) > (y_new*max_num)) {
+                max_den = y_new;
+                max_num = xy_new;
+                max_idx = i;
+            }
+        }
+
+        K -= phase;
+
+        phase *= FFSIGN(X[max_idx]);
+        xy_norm += 1*phase*X[max_idx];
+        y_norm  += 2*phase*y[max_idx];
+        y[max_idx] += phase;
+    }
+
+    return (float)y_norm;
+}
+#endif
+
+static uint32_t celt_alg_quant(OpusRangeCoder *rc, float *X, uint32_t N, uint32_t K,
+                               enum CeltSpread spread, uint32_t blocks, float gain,
+                               CeltPVQ *pvq)
+{
+    int *y = pvq->qcoeff;
+
+    celt_exp_rotation(X, N, blocks, K, spread, 1);
+    gain /= sqrtf(pvq->pvq_search(X, y, K, N));
+    celt_encode_pulses(rc, y,  N, K);
+    celt_normalize_residual(y, X, N, gain);
+    celt_exp_rotation(X, N, blocks, K, spread, 0);
+    return celt_extract_collapse_mask(y, N, blocks);
+}
+
+/** Decode pulse vector and combine the result with the pitch vector to produce
+    the final normalised signal in the current band. */
+static uint32_t celt_alg_unquant(OpusRangeCoder *rc, float *X, uint32_t N, uint32_t K,
+                                 enum CeltSpread spread, uint32_t blocks, float gain,
+                                 CeltPVQ *pvq)
+{
+    int *y = pvq->qcoeff;
+
+    gain /= sqrtf(celt_decode_pulses(rc, y, N, K));
+    celt_normalize_residual(y, X, N, gain);
+    celt_exp_rotation(X, N, blocks, K, spread, 0);
+    return celt_extract_collapse_mask(y, N, blocks);
+}
+
+static int celt_calc_theta(const float *X, const float *Y, int coupling, int N)
+{
+    int i;
+    float e[2] = { 0.0f, 0.0f };
+    if (coupling) { /* Coupling case */
+        for (i = 0; i < N; i++) {
+            e[0] += (X[i] + Y[i])*(X[i] + Y[i]);
+            e[1] += (X[i] - Y[i])*(X[i] - Y[i]);
+        }
+    } else {
+        for (i = 0; i < N; i++) {
+            e[0] += X[i]*X[i];
+            e[1] += Y[i]*Y[i];
+        }
+    }
+    return lrintf(32768.0f*atan2f(sqrtf(e[1]), sqrtf(e[0]))/M_PI);
+}
+
+static void celt_stereo_is_decouple(float *X, float *Y, float e_l, float e_r, int N)
+{
+    int i;
+    const float energy_n = 1.0f/(sqrtf(e_l*e_l + e_r*e_r) + FLT_EPSILON);
+    e_l *= energy_n;
+    e_r *= energy_n;
+    for (i = 0; i < N; i++)
+        X[i] = e_l*X[i] + e_r*Y[i];
+}
+
+static void celt_stereo_ms_decouple(float *X, float *Y, int N)
+{
+    int i;
+    for (i = 0; i < N; i++) {
+        const float Xret = X[i];
+        X[i] = (X[i] + Y[i])*M_SQRT1_2;
+        Y[i] = (Y[i] - Xret)*M_SQRT1_2;
+    }
+}
+
+static av_always_inline uint32_t quant_band_template(CeltPVQ *pvq, CeltFrame *f,
+                                                     OpusRangeCoder *rc,
+                                                     const int band, float *X,
+                                                     float *Y, int N, int b,
+                                                     uint32_t blocks, float *lowband,
+                                                     int duration, float *lowband_out,
+                                                     int level, float gain,
+                                                     float *lowband_scratch,
+                                                     int fill, int quant)
+{
+    int i;
+    const uint8_t *cache;
+    int stereo = !!Y, split = stereo;
+    int imid = 0, iside = 0;
+    uint32_t N0 = N;
+    int N_B = N / blocks;
+    int N_B0 = N_B;
+    int B0 = blocks;
+    int time_divide = 0;
+    int recombine = 0;
+    int inv = 0;
+    float mid = 0, side = 0;
+    int longblocks = (B0 == 1);
+    uint32_t cm = 0;
+
+    if (N == 1) {
+        float *x = X;
+        for (i = 0; i <= stereo; i++) {
+            int sign = 0;
+            if (f->remaining2 >= 1 << 3) {
+                if (quant) {
+                    sign = x[0] < 0;
+                    ff_opus_rc_put_raw(rc, sign, 1);
+                } else {
+                    sign = ff_opus_rc_get_raw(rc, 1);
+                }
+                f->remaining2 -= 1 << 3;
+            }
+            x[0] = 1.0f - 2.0f*sign;
+            x = Y;
+        }
+        if (lowband_out)
+            lowband_out[0] = X[0];
+        return 1;
+    }
+
+    if (!stereo && level == 0) {
+        int tf_change = f->tf_change[band];
+        int k;
+        if (tf_change > 0)
+            recombine = tf_change;
+        /* Band recombining to increase frequency resolution */
+
+        if (lowband &&
+            (recombine || ((N_B & 1) == 0 && tf_change < 0) || B0 > 1)) {
+            for (i = 0; i < N; i++)
+                lowband_scratch[i] = lowband[i];
+            lowband = lowband_scratch;
+        }
+
+        for (k = 0; k < recombine; k++) {
+            if (quant || lowband)
+                celt_haar1(quant ? X : lowband, N >> k, 1 << k);
+            fill = ff_celt_bit_interleave[fill & 0xF] | ff_celt_bit_interleave[fill >> 4] << 2;
+        }
+        blocks >>= recombine;
+        N_B <<= recombine;
+
+        /* Increasing the time resolution */
+        while ((N_B & 1) == 0 && tf_change < 0) {
+            if (quant || lowband)
+                celt_haar1(quant ? X : lowband, N_B, blocks);
+            fill |= fill << blocks;
+            blocks <<= 1;
+            N_B >>= 1;
+            time_divide++;
+            tf_change++;
+        }
+        B0 = blocks;
+        N_B0 = N_B;
+
+        /* Reorganize the samples in time order instead of frequency order */
+        if (B0 > 1 && (quant || lowband))
+            celt_deinterleave_hadamard(pvq->hadamard_tmp, quant ? X : lowband,
+                                       N_B >> recombine, B0 << recombine,
+                                       longblocks);
+    }
+
+    /* If we need 1.5 more bit than we can produce, split the band in two. */
+    cache = ff_celt_cache_bits +
+            ff_celt_cache_index[(duration + 1) * CELT_MAX_BANDS + band];
+    if (!stereo && duration >= 0 && b > cache[cache[0]] + 12 && N > 2) {
+        N >>= 1;
+        Y = X + N;
+        split = 1;
+        duration -= 1;
+        if (blocks == 1)
+            fill = (fill & 1) | (fill << 1);
+        blocks = (blocks + 1) >> 1;
+    }
+
+    if (split) {
+        int qn;
+        int itheta = quant ? celt_calc_theta(X, Y, stereo, N) : 0;
+        int mbits, sbits, delta;
+        int qalloc;
+        int pulse_cap;
+        int offset;
+        int orig_fill;
+        int tell;
+
+        /* Decide on the resolution to give to the split parameter theta */
+        pulse_cap = ff_celt_log_freq_range[band] + duration * 8;
+        offset = (pulse_cap >> 1) - (stereo && N == 2 ? CELT_QTHETA_OFFSET_TWOPHASE :
+                                                          CELT_QTHETA_OFFSET);
+        qn = (stereo && band >= f->intensity_stereo) ? 1 :
+             celt_compute_qn(N, b, offset, pulse_cap, stereo);
+        tell = opus_rc_tell_frac(rc);
+        if (qn != 1) {
+            if (quant)
+                itheta = (itheta*qn + 8192) >> 14;
+            /* Entropy coding of the angle. We use a uniform pdf for the
+             * time split, a step for stereo, and a triangular one for the rest. */
+            if (quant) {
+                if (stereo && N > 2)
+                    ff_opus_rc_enc_uint_step(rc, itheta, qn / 2);
+                else if (stereo || B0 > 1)
+                    ff_opus_rc_enc_uint(rc, itheta, qn + 1);
+                else
+                    ff_opus_rc_enc_uint_tri(rc, itheta, qn);
+                itheta = itheta * 16384 / qn;
+                if (stereo) {
+                    if (itheta == 0)
+                        celt_stereo_is_decouple(X, Y, f->block[0].lin_energy[band],
+                                                f->block[1].lin_energy[band], N);
+                    else
+                        celt_stereo_ms_decouple(X, Y, N);
+                }
+            } else {
+                if (stereo && N > 2)
+                    itheta = ff_opus_rc_dec_uint_step(rc, qn / 2);
+                else if (stereo || B0 > 1)
+                    itheta = ff_opus_rc_dec_uint(rc, qn+1);
+                else
+                    itheta = ff_opus_rc_dec_uint_tri(rc, qn);
+                itheta = itheta * 16384 / qn;
+            }
+        } else if (stereo) {
+            if (quant) {
+                inv = f->apply_phase_inv ? itheta > 8192 : 0;
+                 if (inv) {
+                    for (i = 0; i < N; i++)
+                       Y[i] *= -1;
+                 }
+                 celt_stereo_is_decouple(X, Y, f->block[0].lin_energy[band],
+                                         f->block[1].lin_energy[band], N);
+
+                if (b > 2 << 3 && f->remaining2 > 2 << 3) {
+                    ff_opus_rc_enc_log(rc, inv, 2);
+                } else {
+                    inv = 0;
+                }
+            } else {
+                inv = (b > 2 << 3 && f->remaining2 > 2 << 3) ? ff_opus_rc_dec_log(rc, 2) : 0;
+                inv = f->apply_phase_inv ? inv : 0;
+            }
+            itheta = 0;
+        }
+        qalloc = opus_rc_tell_frac(rc) - tell;
+        b -= qalloc;
+
+        orig_fill = fill;
+        if (itheta == 0) {
+            imid = 32767;
+            iside = 0;
+            fill = av_zero_extend(fill, blocks);
+            delta = -16384;
+        } else if (itheta == 16384) {
+            imid = 0;
+            iside = 32767;
+            fill &= ((1 << blocks) - 1) << blocks;
+            delta = 16384;
+        } else {
+            imid = celt_cos(itheta);
+            iside = celt_cos(16384-itheta);
+            /* This is the mid vs side allocation that minimizes squared error
+            in that band. */
+            delta = ROUND_MUL16((N - 1) << 7, celt_log2tan(iside, imid));
+        }
+
+        mid  = imid  / 32768.0f;
+        side = iside / 32768.0f;
+
+        /* This is a special case for N=2 that only works for stereo and takes
+        advantage of the fact that mid and side are orthogonal to encode
+        the side with just one bit. */
+        if (N == 2 && stereo) {
+            int c;
+            int sign = 0;
+            float tmp;
+            float *x2, *y2;
+            mbits = b;
+            /* Only need one bit for the side */
+            sbits = (itheta != 0 && itheta != 16384) ? 1 << 3 : 0;
+            mbits -= sbits;
+            c = (itheta > 8192);
+            f->remaining2 -= qalloc+sbits;
+
+            x2 = c ? Y : X;
+            y2 = c ? X : Y;
+            if (sbits) {
+                if (quant) {
+                    sign = x2[0]*y2[1] - x2[1]*y2[0] < 0;
+                    ff_opus_rc_put_raw(rc, sign, 1);
+                } else {
+                    sign = ff_opus_rc_get_raw(rc, 1);
+                }
+            }
+            sign = 1 - 2 * sign;
+            /* We use orig_fill here because we want to fold the side, but if
+            itheta==16384, we'll have cleared the low bits of fill. */
+            cm = pvq->quant_band(pvq, f, rc, band, x2, NULL, N, mbits, blocks, lowband, duration,
+                                 lowband_out, level, gain, lowband_scratch, orig_fill);
+            /* We don't split N=2 bands, so cm is either 1 or 0 (for a fold-collapse),
+            and there's no need to worry about mixing with the other channel. */
+            y2[0] = -sign * x2[1];
+            y2[1] =  sign * x2[0];
+            X[0] *= mid;
+            X[1] *= mid;
+            Y[0] *= side;
+            Y[1] *= side;
+            tmp = X[0];
+            X[0] = tmp - Y[0];
+            Y[0] = tmp + Y[0];
+            tmp = X[1];
+            X[1] = tmp - Y[1];
+            Y[1] = tmp + Y[1];
+        } else {
+            /* "Normal" split code */
+            float *next_lowband2     = NULL;
+            float *next_lowband_out1 = NULL;
+            int next_level = 0;
+            int rebalance;
+            uint32_t cmt;
+
+            /* Give more bits to low-energy MDCTs than they would
+             * otherwise deserve */
+            if (B0 > 1 && !stereo && (itheta & 0x3fff)) {
+                if (itheta > 8192)
+                    /* Rough approximation for pre-echo masking */
+                    delta -= delta >> (4 - duration);
+                else
+                    /* Corresponds to a forward-masking slope of
+                     * 1.5 dB per 10 ms */
+                    delta = FFMIN(0, delta + (N << 3 >> (5 - duration)));
+            }
+            mbits = av_clip((b - delta) / 2, 0, b);
+            sbits = b - mbits;
+            f->remaining2 -= qalloc;
+
+            if (lowband && !stereo)
+                next_lowband2 = lowband + N; /* >32-bit split case */
+
+            /* Only stereo needs to pass on lowband_out.
+             * Otherwise, it's handled at the end */
+            if (stereo)
+                next_lowband_out1 = lowband_out;
+            else
+                next_level = level + 1;
+
+            rebalance = f->remaining2;
+            if (mbits >= sbits) {
+                /* In stereo mode, we do not apply a scaling to the mid
+                 * because we need the normalized mid for folding later */
+                cm = pvq->quant_band(pvq, f, rc, band, X, NULL, N, mbits, blocks,
+                                     lowband, duration, next_lowband_out1, next_level,
+                                     stereo ? 1.0f : (gain * mid), lowband_scratch, fill);
+                rebalance = mbits - (rebalance - f->remaining2);
+                if (rebalance > 3 << 3 && itheta != 0)
+                    sbits += rebalance - (3 << 3);
+
+                /* For a stereo split, the high bits of fill are always zero,
+                 * so no folding will be done to the side. */
+                cmt = pvq->quant_band(pvq, f, rc, band, Y, NULL, N, sbits, blocks,
+                                      next_lowband2, duration, NULL, next_level,
+                                      gain * side, NULL, fill >> blocks);
+                cm |= cmt << ((B0 >> 1) & (stereo - 1));
+            } else {
+                /* For a stereo split, the high bits of fill are always zero,
+                 * so no folding will be done to the side. */
+                cm = pvq->quant_band(pvq, f, rc, band, Y, NULL, N, sbits, blocks,
+                                     next_lowband2, duration, NULL, next_level,
+                                     gain * side, NULL, fill >> blocks);
+                cm <<= ((B0 >> 1) & (stereo - 1));
+                rebalance = sbits - (rebalance - f->remaining2);
+                if (rebalance > 3 << 3 && itheta != 16384)
+                    mbits += rebalance - (3 << 3);
+
+                /* In stereo mode, we do not apply a scaling to the mid because
+                 * we need the normalized mid for folding later */
+                cm |= pvq->quant_band(pvq, f, rc, band, X, NULL, N, mbits, blocks,
+                                      lowband, duration, next_lowband_out1, next_level,
+                                      stereo ? 1.0f : (gain * mid), lowband_scratch, fill);
+            }
+        }
+    } else {
+        /* This is the basic no-split case */
+        uint32_t q         = celt_bits2pulses(cache, b);
+        uint32_t curr_bits = celt_pulses2bits(cache, q);
+        f->remaining2 -= curr_bits;
+
+        /* Ensures we can never bust the budget */
+        while (f->remaining2 < 0 && q > 0) {
+            f->remaining2 += curr_bits;
+            curr_bits      = celt_pulses2bits(cache, --q);
+            f->remaining2 -= curr_bits;
+        }
+
+        if (q != 0) {
+            /* Finally do the actual (de)quantization */
+            if (quant) {
+                cm = celt_alg_quant(rc, X, N, (q < 8) ? q : (8 + (q & 7)) << ((q >> 3) - 1),
+                                    f->spread, blocks, gain, pvq);
+            } else {
+                cm = celt_alg_unquant(rc, X, N, (q < 8) ? q : (8 + (q & 7)) << ((q >> 3) - 1),
+                                      f->spread, blocks, gain, pvq);
+            }
+        } else {
+            /* If there's no pulse, fill the band anyway */
+            uint32_t cm_mask = (1 << blocks) - 1;
+            fill &= cm_mask;
+            if (fill) {
+                if (!lowband) {
+                    /* Noise */
+                    for (i = 0; i < N; i++)
+                        X[i] = (((int32_t)celt_rng(f)) >> 20);
+                    cm = cm_mask;
+                } else {
+                    /* Folded spectrum */
+                    for (i = 0; i < N; i++) {
+                        /* About 48 dB below the "normal" folding level */
+                        X[i] = lowband[i] + (((celt_rng(f)) & 0x8000) ? 1.0f / 256 : -1.0f / 256);
+                    }
+                    cm = fill;
+                }
+                celt_renormalize_vector(X, N, gain);
+            } else {
+                memset(X, 0, N*sizeof(float));
+            }
+        }
+    }
+
+    /* This code is used by the decoder and by the resynthesis-enabled encoder */
+    if (stereo) {
+        if (N > 2)
+            celt_stereo_merge(X, Y, mid, N);
+        if (inv) {
+            for (i = 0; i < N; i++)
+                Y[i] *= -1;
+        }
+    } else if (level == 0) {
+        int k;
+
+        /* Undo the sample reorganization going from time order to frequency order */
+        if (B0 > 1)
+            celt_interleave_hadamard(pvq->hadamard_tmp, X, N_B >> recombine,
+                                     B0 << recombine, longblocks);
+
+        /* Undo time-freq changes that we did earlier */
+        N_B = N_B0;
+        blocks = B0;
+        for (k = 0; k < time_divide; k++) {
+            blocks >>= 1;
+            N_B <<= 1;
+            cm |= cm >> blocks;
+            celt_haar1(X, N_B, blocks);
+        }
+
+        for (k = 0; k < recombine; k++) {
+            cm = ff_celt_bit_deinterleave[cm];
+            celt_haar1(X, N0>>k, 1<<k);
+        }
+        blocks <<= recombine;
+
+        /* Scale output for later folding */
+        if (lowband_out) {
+            float n = sqrtf(N0);
+            for (i = 0; i < N0; i++)
+                lowband_out[i] = n * X[i];
+        }
+        cm = av_zero_extend(cm, blocks);
+    }
+
+    return cm;
+}
+
+static QUANT_FN(pvq_decode_band)
+{
+#if CONFIG_OPUS_DECODER
+    return quant_band_template(pvq, f, rc, band, X, Y, N, b, blocks, lowband, duration,
+                               lowband_out, level, gain, lowband_scratch, fill, 0);
+#else
+    return 0;
+#endif
+}
+
+static QUANT_FN(pvq_encode_band)
+{
+#if CONFIG_OPUS_ENCODER
+    return quant_band_template(pvq, f, rc, band, X, Y, N, b, blocks, lowband, duration,
+                               lowband_out, level, gain, lowband_scratch, fill, 1);
+#else
+    return 0;
+#endif
+}
+
+int av_cold ff_celt_pvq_init(CeltPVQ **pvq, int encode)
+{
+    CeltPVQ *s = av_malloc(sizeof(CeltPVQ));
+    if (!s)
+        return AVERROR(ENOMEM);
+
+    s->quant_band = encode ? pvq_encode_band : pvq_decode_band;
+
+#if CONFIG_OPUS_ENCODER
+    s->pvq_search = ppp_pvq_search_c;
+#if ARCH_X86
+    ff_celt_pvq_init_x86(s);
+#endif
+#endif
+
+    *pvq = s;
+
+    return 0;
+}
+
+void av_cold ff_celt_pvq_uninit(CeltPVQ **pvq)
+{
+    av_freep(pvq);
+}
diff --git a/libavcodec/opus/pvq.h b/libavcodec/opus/pvq.h
new file mode 100644
index 0000000000..07f568f6c0
--- /dev/null
+++ b/libavcodec/opus/pvq.h
@@ -0,0 +1,50 @@
+/*
+ * Copyright (c) 2012 Andrew D'Addesio
+ * Copyright (c) 2013-2014 Mozilla Corporation
+ * Copyright (c) 2016 Rostislav Pehlivanov <atomnuker@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_OPUS_PVQ_H
+#define AVCODEC_OPUS_PVQ_H
+
+#include "libavutil/mem_internal.h"
+
+#include "celt.h"
+
+#define QUANT_FN(name) uint32_t (name)(struct CeltPVQ *pvq, CeltFrame *f,            \
+                                       OpusRangeCoder *rc, const int band, float *X, \
+                                       float *Y, int N, int b, uint32_t blocks,      \
+                                       float *lowband, int duration,                 \
+                                       float *lowband_out, int level, float gain,    \
+                                       float *lowband_scratch, int fill)
+
+typedef struct CeltPVQ {
+    DECLARE_ALIGNED(32, int,   qcoeff      )[256];
+    DECLARE_ALIGNED(32, float, hadamard_tmp)[256];
+
+    float (*pvq_search)(float *X, int *y, int K, int N);
+    QUANT_FN(*quant_band);
+} CeltPVQ;
+
+void ff_celt_pvq_init_x86(struct CeltPVQ *s);
+
+int  ff_celt_pvq_init(struct CeltPVQ **pvq, int encode);
+void ff_celt_pvq_uninit(struct CeltPVQ **pvq);
+
+#endif /* AVCODEC_OPUS_PVQ_H */
diff --git a/libavcodec/opus/rc.c b/libavcodec/opus/rc.c
new file mode 100644
index 0000000000..8e58a52b85
--- /dev/null
+++ b/libavcodec/opus/rc.c
@@ -0,0 +1,411 @@
+/*
+ * Copyright (c) 2012 Andrew D'Addesio
+ * Copyright (c) 2013-2014 Mozilla Corporation
+ * Copyright (c) 2017 Rostislav Pehlivanov <atomnuker@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "rc.h"
+
+#define OPUS_RC_BITS 32
+#define OPUS_RC_SYM  8
+#define OPUS_RC_CEIL ((1 << OPUS_RC_SYM) - 1)
+#define OPUS_RC_TOP (1u << 31)
+#define OPUS_RC_BOT (OPUS_RC_TOP >> OPUS_RC_SYM)
+#define OPUS_RC_SHIFT (OPUS_RC_BITS - OPUS_RC_SYM - 1)
+
+static av_always_inline void opus_rc_enc_carryout(OpusRangeCoder *rc, int cbuf)
+{
+    const int cb = cbuf >> OPUS_RC_SYM, mb = (OPUS_RC_CEIL + cb) & OPUS_RC_CEIL;
+    if (cbuf == OPUS_RC_CEIL) {
+        rc->ext++;
+        return;
+    }
+    rc->rng_cur[0] = rc->rem + cb;
+    rc->rng_cur += (rc->rem >= 0);
+    for (; rc->ext > 0; rc->ext--)
+        *rc->rng_cur++ = mb;
+    av_assert0(rc->rng_cur < rc->rb.position);
+    rc->rem = cbuf & OPUS_RC_CEIL; /* Propagate */
+}
+
+static av_always_inline void opus_rc_dec_normalize(OpusRangeCoder *rc)
+{
+    while (rc->range <= OPUS_RC_BOT) {
+        rc->value = ((rc->value << OPUS_RC_SYM) | (get_bits(&rc->gb, OPUS_RC_SYM) ^ OPUS_RC_CEIL)) & (OPUS_RC_TOP - 1);
+        rc->range     <<= OPUS_RC_SYM;
+        rc->total_bits += OPUS_RC_SYM;
+    }
+}
+
+static av_always_inline void opus_rc_enc_normalize(OpusRangeCoder *rc)
+{
+    while (rc->range <= OPUS_RC_BOT) {
+        opus_rc_enc_carryout(rc, rc->value >> OPUS_RC_SHIFT);
+        rc->value = (rc->value << OPUS_RC_SYM) & (OPUS_RC_TOP - 1);
+        rc->range     <<= OPUS_RC_SYM;
+        rc->total_bits += OPUS_RC_SYM;
+    }
+}
+
+static av_always_inline void opus_rc_dec_update(OpusRangeCoder *rc, uint32_t scale,
+                                                uint32_t low, uint32_t high,
+                                                uint32_t total)
+{
+    rc->value -= scale * (total - high);
+    rc->range  = low ? scale * (high - low)
+                      : rc->range - scale * (total - high);
+    opus_rc_dec_normalize(rc);
+}
+
+/* Main encoding function, this needs to go fast */
+static av_always_inline void opus_rc_enc_update(OpusRangeCoder *rc, uint32_t b, uint32_t p,
+                                                uint32_t p_tot, const int ptwo)
+{
+    uint32_t rscaled, cnd = !!b;
+    if (ptwo) /* Whole function is inlined so hopefully branch is optimized out */
+        rscaled = rc->range >> ff_log2(p_tot);
+    else
+        rscaled = rc->range/p_tot;
+    rc->value +=    cnd*(rc->range - rscaled*(p_tot - b));
+    rc->range  = (!cnd)*(rc->range - rscaled*(p_tot - p)) + cnd*rscaled*(p - b);
+    opus_rc_enc_normalize(rc);
+}
+
+uint32_t ff_opus_rc_dec_cdf(OpusRangeCoder *rc, const uint16_t *cdf)
+{
+    unsigned int k, scale, total, symbol, low, high;
+
+    total = *cdf++;
+
+    scale   = rc->range / total;
+    symbol = rc->value / scale + 1;
+    symbol = total - FFMIN(symbol, total);
+
+    for (k = 0; cdf[k] <= symbol; k++);
+    high = cdf[k];
+    low  = k ? cdf[k-1] : 0;
+
+    opus_rc_dec_update(rc, scale, low, high, total);
+
+    return k;
+}
+
+void ff_opus_rc_enc_cdf(OpusRangeCoder *rc, int val, const uint16_t *cdf)
+{
+    opus_rc_enc_update(rc, (!!val)*cdf[val], cdf[val + 1], cdf[0], 1);
+}
+
+uint32_t ff_opus_rc_dec_log(OpusRangeCoder *rc, uint32_t bits)
+{
+    uint32_t k, scale;
+    scale = rc->range >> bits; // in this case, scale = symbol
+
+    if (rc->value >= scale) {
+        rc->value -= scale;
+        rc->range -= scale;
+        k = 0;
+    } else {
+        rc->range = scale;
+        k = 1;
+    }
+    opus_rc_dec_normalize(rc);
+    return k;
+}
+
+void ff_opus_rc_enc_log(OpusRangeCoder *rc, int val, uint32_t bits)
+{
+    bits = (1 << bits) - 1;
+    opus_rc_enc_update(rc, (!!val)*bits, bits + !!val, bits + 1, 1);
+}
+
+/**
+ * CELT: read 1-25 raw bits at the end of the frame, backwards byte-wise
+ */
+uint32_t ff_opus_rc_get_raw(OpusRangeCoder *rc, uint32_t count)
+{
+    uint32_t value = 0;
+
+    while (rc->rb.bytes && rc->rb.cachelen < count) {
+        rc->rb.cacheval |= *--rc->rb.position << rc->rb.cachelen;
+        rc->rb.cachelen += 8;
+        rc->rb.bytes--;
+    }
+
+    value = av_zero_extend(rc->rb.cacheval, count);
+    rc->rb.cacheval    >>= count;
+    rc->rb.cachelen     -= count;
+    rc->total_bits      += count;
+
+    return value;
+}
+
+/**
+ * CELT: write 0 - 31 bits to the rawbits buffer
+ */
+void ff_opus_rc_put_raw(OpusRangeCoder *rc, uint32_t val, uint32_t count)
+{
+    const int to_write = FFMIN(32 - rc->rb.cachelen, count);
+
+    rc->total_bits += count;
+    rc->rb.cacheval |= av_zero_extend(val, to_write) << rc->rb.cachelen;
+    rc->rb.cachelen = (rc->rb.cachelen + to_write) % 32;
+
+    if (!rc->rb.cachelen && count) {
+        AV_WB32((uint8_t *)rc->rb.position, rc->rb.cacheval);
+        rc->rb.bytes    += 4;
+        rc->rb.position -= 4;
+        rc->rb.cachelen = count - to_write;
+        rc->rb.cacheval = av_zero_extend(val >> to_write, rc->rb.cachelen);
+        av_assert0(rc->rng_cur < rc->rb.position);
+    }
+}
+
+/**
+ * CELT: read a uniform distribution
+ */
+uint32_t ff_opus_rc_dec_uint(OpusRangeCoder *rc, uint32_t size)
+{
+    uint32_t bits, k, scale, total;
+
+    bits  = opus_ilog(size - 1);
+    total = (bits > 8) ? ((size - 1) >> (bits - 8)) + 1 : size;
+
+    scale  = rc->range / total;
+    k      = rc->value / scale + 1;
+    k      = total - FFMIN(k, total);
+    opus_rc_dec_update(rc, scale, k, k + 1, total);
+
+    if (bits > 8) {
+        k = k << (bits - 8) | ff_opus_rc_get_raw(rc, bits - 8);
+        return FFMIN(k, size - 1);
+    } else
+        return k;
+}
+
+/**
+ * CELT: write a uniformly distributed integer
+ */
+void ff_opus_rc_enc_uint(OpusRangeCoder *rc, uint32_t val, uint32_t size)
+{
+    const int ps = FFMAX(opus_ilog(size - 1) - 8, 0);
+    opus_rc_enc_update(rc, val >> ps, (val >> ps) + 1, ((size - 1) >> ps) + 1, 0);
+    ff_opus_rc_put_raw(rc, val, ps);
+}
+
+uint32_t ff_opus_rc_dec_uint_step(OpusRangeCoder *rc, int k0)
+{
+    /* Use a probability of 3 up to itheta=8192 and then use 1 after */
+    uint32_t k, scale, symbol, total = (k0+1)*3 + k0;
+    scale  = rc->range / total;
+    symbol = rc->value / scale + 1;
+    symbol = total - FFMIN(symbol, total);
+
+    k = (symbol < (k0+1)*3) ? symbol/3 : symbol - (k0+1)*2;
+
+    opus_rc_dec_update(rc, scale, (k <= k0) ? 3*(k+0) : (k-1-k0) + 3*(k0+1),
+                       (k <= k0) ? 3*(k+1) : (k-0-k0) + 3*(k0+1), total);
+    return k;
+}
+
+void ff_opus_rc_enc_uint_step(OpusRangeCoder *rc, uint32_t val, int k0)
+{
+    const uint32_t a = val <= k0, b = 2*a + 1;
+    k0 = (k0 + 1) << 1;
+    val = b*(val + k0) - 3*a*k0;
+    opus_rc_enc_update(rc, val, val + b, (k0 << 1) - 1, 0);
+}
+
+uint32_t ff_opus_rc_dec_uint_tri(OpusRangeCoder *rc, int qn)
+{
+    uint32_t k, scale, symbol, total, low, center;
+
+    total = ((qn>>1) + 1) * ((qn>>1) + 1);
+    scale   = rc->range / total;
+    center = rc->value / scale + 1;
+    center = total - FFMIN(center, total);
+
+    if (center < total >> 1) {
+        k      = (ff_sqrt(8 * center + 1) - 1) >> 1;
+        low    = k * (k + 1) >> 1;
+        symbol = k + 1;
+    } else {
+        k      = (2*(qn + 1) - ff_sqrt(8*(total - center - 1) + 1)) >> 1;
+        low    = total - ((qn + 1 - k) * (qn + 2 - k) >> 1);
+        symbol = qn + 1 - k;
+    }
+
+    opus_rc_dec_update(rc, scale, low, low + symbol, total);
+
+    return k;
+}
+
+void ff_opus_rc_enc_uint_tri(OpusRangeCoder *rc, uint32_t k, int qn)
+{
+    uint32_t symbol, low, total;
+
+    total = ((qn>>1) + 1) * ((qn>>1) + 1);
+
+    if (k <= qn >> 1) {
+        low    = k * (k + 1) >> 1;
+        symbol = k + 1;
+    } else {
+        low    = total - ((qn + 1 - k) * (qn + 2 - k) >> 1);
+        symbol = qn + 1 - k;
+    }
+
+    opus_rc_enc_update(rc, low, low + symbol, total, 0);
+}
+
+int ff_opus_rc_dec_laplace(OpusRangeCoder *rc, uint32_t symbol, int decay)
+{
+    /* extends the range coder to model a Laplace distribution */
+    int value = 0;
+    uint32_t scale, low = 0, center;
+
+    scale  = rc->range >> 15;
+    center = rc->value / scale + 1;
+    center = (1 << 15) - FFMIN(center, 1 << 15);
+
+    if (center >= symbol) {
+        value++;
+        low = symbol;
+        symbol = 1 + ((32768 - 32 - symbol) * (16384-decay) >> 15);
+
+        while (symbol > 1 && center >= low + 2 * symbol) {
+            value++;
+            symbol *= 2;
+            low    += symbol;
+            symbol  = (((symbol - 2) * decay) >> 15) + 1;
+        }
+
+        if (symbol <= 1) {
+            int distance = (center - low) >> 1;
+            value += distance;
+            low   += 2 * distance;
+        }
+
+        if (center < low + symbol)
+            value *= -1;
+        else
+            low += symbol;
+    }
+
+    opus_rc_dec_update(rc, scale, low, FFMIN(low + symbol, 32768), 32768);
+
+    return value;
+}
+
+void ff_opus_rc_enc_laplace(OpusRangeCoder *rc, int *value, uint32_t symbol, int decay)
+{
+    uint32_t low = symbol;
+    int i = 1, val = FFABS(*value), pos = *value > 0;
+    if (!val) {
+        opus_rc_enc_update(rc, 0, symbol, 1 << 15, 1);
+        return;
+    }
+    symbol = ((32768 - 32 - symbol)*(16384 - decay)) >> 15;
+    for (; i < val && symbol; i++) {
+        low   += (symbol << 1) + 2;
+        symbol = (symbol*decay) >> 14;
+    }
+    if (symbol) {
+        low += (++symbol)*pos;
+    } else {
+        const int distance = FFMIN(val - i, (((32768 - low) - !pos) >> 1) - 1);
+        low   += pos + (distance << 1);
+        symbol = FFMIN(1, 32768 - low);
+        *value = FFSIGN(*value)*(distance + i);
+    }
+    opus_rc_enc_update(rc, low, low + symbol, 1 << 15, 1);
+}
+
+int ff_opus_rc_dec_init(OpusRangeCoder *rc, const uint8_t *data, int size)
+{
+    int ret = init_get_bits8(&rc->gb, data, size);
+    if (ret < 0)
+        return ret;
+
+    rc->range = 128;
+    rc->value = 127 - get_bits(&rc->gb, 7);
+    rc->total_bits = 9;
+    opus_rc_dec_normalize(rc);
+
+    return 0;
+}
+
+void ff_opus_rc_dec_raw_init(OpusRangeCoder *rc, const uint8_t *rightend, uint32_t bytes)
+{
+    rc->rb.position = rightend;
+    rc->rb.bytes    = bytes;
+    rc->rb.cachelen = 0;
+    rc->rb.cacheval = 0;
+}
+
+void ff_opus_rc_enc_end(OpusRangeCoder *rc, uint8_t *dst, int size)
+{
+    int rng_bytes, bits = OPUS_RC_BITS - opus_ilog(rc->range);
+    uint32_t mask = (OPUS_RC_TOP - 1) >> bits;
+    uint32_t end = (rc->value + mask) & ~mask;
+
+    if ((end | mask) >= rc->value + rc->range) {
+        bits++;
+        mask >>= 1;
+        end = (rc->value + mask) & ~mask;
+    }
+
+    /* Finish what's left */
+    while (bits > 0) {
+        opus_rc_enc_carryout(rc, end >> OPUS_RC_SHIFT);
+        end = (end << OPUS_RC_SYM) & (OPUS_RC_TOP - 1);
+        bits -= OPUS_RC_SYM;
+    }
+
+    /* Flush out anything left or marked */
+    if (rc->rem >= 0 || rc->ext > 0)
+        opus_rc_enc_carryout(rc, 0);
+
+    rng_bytes = rc->rng_cur - rc->buf;
+    memcpy(dst, rc->buf, rng_bytes);
+
+    rc->waste = size*8 - (rc->rb.bytes*8 + rc->rb.cachelen) - rng_bytes*8;
+
+    /* Put the rawbits part, if any */
+    if (rc->rb.bytes || rc->rb.cachelen) {
+        int i, lap;
+        uint8_t *rb_src, *rb_dst;
+        ff_opus_rc_put_raw(rc, 0, 32 - rc->rb.cachelen);
+        rb_src = rc->buf + OPUS_MAX_FRAME_SIZE + 12 - rc->rb.bytes;
+        rb_dst = dst + FFMAX(size - rc->rb.bytes, 0);
+        lap = &dst[rng_bytes] - rb_dst;
+        for (i = 0; i < lap; i++)
+            rb_dst[i] |= rb_src[i];
+        memcpy(&rb_dst[lap], &rb_src[lap], FFMAX(rc->rb.bytes - lap, 0));
+    }
+}
+
+void ff_opus_rc_enc_init(OpusRangeCoder *rc)
+{
+    rc->value = 0;
+    rc->range = OPUS_RC_TOP;
+    rc->total_bits = OPUS_RC_BITS + 1;
+    rc->rem = -1;
+    rc->ext =  0;
+    rc->rng_cur = rc->buf;
+    ff_opus_rc_dec_raw_init(rc, rc->buf + OPUS_MAX_FRAME_SIZE + 8, 0);
+}
diff --git a/libavcodec/opus/rc.h b/libavcodec/opus/rc.h
new file mode 100644
index 0000000000..e9407f2ca4
--- /dev/null
+++ b/libavcodec/opus/rc.h
@@ -0,0 +1,127 @@
+/*
+ * Copyright (c) 2012 Andrew D'Addesio
+ * Copyright (c) 2013-2014 Mozilla Corporation
+ * Copyright (c) 2017 Rostislav Pehlivanov <atomnuker@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_OPUS_RC_H
+#define AVCODEC_OPUS_RC_H
+
+#include <stdint.h>
+
+#include "libavcodec/get_bits.h"
+
+#include "opus.h"
+
+#define opus_ilog(i) (av_log2(i) + !!(i))
+
+typedef struct RawBitsContext {
+    const uint8_t *position;
+    uint32_t bytes;
+    uint32_t cachelen;
+    uint32_t cacheval;
+} RawBitsContext;
+
+typedef struct OpusRangeCoder {
+    GetBitContext gb;
+    RawBitsContext rb;
+    uint32_t range;
+    uint32_t value;
+    uint32_t total_bits;
+
+    /* Encoder */
+    uint8_t buf[OPUS_MAX_FRAME_SIZE + 12]; /* memcpy vs (memmove + overreading) */
+    uint8_t *rng_cur;                      /* Current range coded byte */
+    int ext;                               /* Awaiting propagation */
+    int rem;                               /* Carryout flag */
+
+    /* Encoding stats */
+    int waste;
+} OpusRangeCoder;
+
+/**
+ * CELT: estimate bits of entropy that have thus far been consumed for the
+ *       current CELT frame, to integer and fractional (1/8th bit) precision
+ */
+static av_always_inline uint32_t opus_rc_tell(const OpusRangeCoder *rc)
+{
+    return rc->total_bits - av_log2(rc->range) - 1;
+}
+
+static av_always_inline uint32_t opus_rc_tell_frac(const OpusRangeCoder *rc)
+{
+    uint32_t i, total_bits, rcbuffer, range;
+
+    total_bits = rc->total_bits << 3;
+    rcbuffer   = av_log2(rc->range) + 1;
+    range      = rc->range >> (rcbuffer-16);
+
+    for (i = 0; i < 3; i++) {
+        int bit;
+        range = range * range >> 15;
+        bit = range >> 16;
+        rcbuffer = rcbuffer << 1 | bit;
+        range >>= bit;
+    }
+
+    return total_bits - rcbuffer;
+}
+
+uint32_t ff_opus_rc_dec_cdf(OpusRangeCoder *rc, const uint16_t *cdf);
+void     ff_opus_rc_enc_cdf(OpusRangeCoder *rc, int val, const uint16_t *cdf);
+
+uint32_t ff_opus_rc_dec_log(OpusRangeCoder *rc, uint32_t bits);
+void     ff_opus_rc_enc_log(OpusRangeCoder *rc, int val, uint32_t bits);
+
+uint32_t ff_opus_rc_dec_uint_step(OpusRangeCoder *rc, int k0);
+void     ff_opus_rc_enc_uint_step(OpusRangeCoder *rc, uint32_t val, int k0);
+
+uint32_t ff_opus_rc_dec_uint_tri(OpusRangeCoder *rc, int qn);
+void     ff_opus_rc_enc_uint_tri(OpusRangeCoder *rc, uint32_t k, int qn);
+
+uint32_t ff_opus_rc_dec_uint(OpusRangeCoder *rc, uint32_t size);
+void     ff_opus_rc_enc_uint(OpusRangeCoder *rc, uint32_t val, uint32_t size);
+
+uint32_t ff_opus_rc_get_raw(OpusRangeCoder *rc, uint32_t count);
+void     ff_opus_rc_put_raw(OpusRangeCoder *rc, uint32_t val, uint32_t count);
+
+int      ff_opus_rc_dec_laplace(OpusRangeCoder *rc, uint32_t symbol, int decay);
+void     ff_opus_rc_enc_laplace(OpusRangeCoder *rc, int *value, uint32_t symbol, int decay);
+
+int      ff_opus_rc_dec_init(OpusRangeCoder *rc, const uint8_t *data, int size);
+void     ff_opus_rc_dec_raw_init(OpusRangeCoder *rc, const uint8_t *rightend, uint32_t bytes);
+
+void     ff_opus_rc_enc_end(OpusRangeCoder *rc, uint8_t *dst, int size);
+void     ff_opus_rc_enc_init(OpusRangeCoder *rc);
+
+#define OPUS_RC_CHECKPOINT_UPDATE(rc) \
+    rc_rollback_bits = opus_rc_tell_frac(rc); \
+    rc_rollback_ctx  = *rc
+
+#define OPUS_RC_CHECKPOINT_SPAWN(rc) \
+    uint32_t rc_rollback_bits = opus_rc_tell_frac(rc); \
+    OpusRangeCoder rc_rollback_ctx = *rc \
+
+#define OPUS_RC_CHECKPOINT_BITS(rc) \
+    (opus_rc_tell_frac(rc) - rc_rollback_bits)
+
+#define OPUS_RC_CHECKPOINT_ROLLBACK(rc) \
+    memcpy(rc, &rc_rollback_ctx, sizeof(OpusRangeCoder)); \
+
+#endif /* AVCODEC_OPUS_RC_H */
diff --git a/libavcodec/opus/silk.c b/libavcodec/opus/silk.c
new file mode 100644
index 0000000000..97bb95037c
--- /dev/null
+++ b/libavcodec/opus/silk.c
@@ -0,0 +1,905 @@
+/*
+ * Copyright (c) 2012 Andrew D'Addesio
+ * Copyright (c) 2013-2014 Mozilla Corporation
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Opus SILK decoder
+ */
+
+#include <stdint.h>
+
+#include "libavutil/mem.h"
+#include "mathops.h"
+#include "opus.h"
+#include "rc.h"
+#include "silk.h"
+#include "tab.h"
+
+#define ROUND_MULL(a,b,s) (((MUL64(a, b) >> ((s) - 1)) + 1) >> 1)
+
+typedef struct SilkFrame {
+    int coded;
+    int log_gain;
+    int16_t nlsf[16];
+    float    lpc[16];
+
+    float output     [2 * SILK_HISTORY];
+    float lpc_history[2 * SILK_HISTORY];
+    int primarylag;
+
+    int prev_voiced;
+} SilkFrame;
+
+struct SilkContext {
+    void *logctx;
+    int output_channels;
+
+    int midonly;
+    int subframes;
+    int sflength;
+    int flength;
+    int nlsf_interp_factor;
+
+    enum OpusBandwidth bandwidth;
+    int wb;
+
+    SilkFrame frame[2];
+    float prev_stereo_weights[2];
+    float stereo_weights[2];
+
+    int prev_coded_channels;
+};
+
+static inline void silk_stabilize_lsf(int16_t nlsf[16], int order, const uint16_t min_delta[17])
+{
+    int pass, i;
+    for (pass = 0; pass < 20; pass++) {
+        int k, min_diff = 0;
+        for (i = 0; i < order+1; i++) {
+            int low  = i != 0     ? nlsf[i-1] : 0;
+            int high = i != order ? nlsf[i]   : 32768;
+            int diff = (high - low) - (min_delta[i]);
+
+            if (diff < min_diff) {
+                min_diff = diff;
+                k = i;
+
+                if (pass == 20)
+                    break;
+            }
+        }
+        if (min_diff == 0) /* no issues; stabilized */
+            return;
+
+        /* wiggle one or two LSFs */
+        if (k == 0) {
+            /* repel away from lower bound */
+            nlsf[0] = min_delta[0];
+        } else if (k == order) {
+            /* repel away from higher bound */
+            nlsf[order-1] = 32768 - min_delta[order];
+        } else {
+            /* repel away from current position */
+            int min_center = 0, max_center = 32768, center_val;
+
+            /* lower extent */
+            for (i = 0; i < k; i++)
+                min_center += min_delta[i];
+            min_center += min_delta[k] >> 1;
+
+            /* upper extent */
+            for (i = order; i > k; i--)
+                max_center -= min_delta[i];
+            max_center -= min_delta[k] >> 1;
+
+            /* move apart */
+            center_val = nlsf[k - 1] + nlsf[k];
+            center_val = (center_val >> 1) + (center_val & 1); // rounded divide by 2
+            center_val = FFMIN(max_center, FFMAX(min_center, center_val));
+
+            nlsf[k - 1] = center_val - (min_delta[k] >> 1);
+            nlsf[k]     = nlsf[k - 1] + min_delta[k];
+        }
+    }
+
+    /* resort to the fall-back method, the standard method for LSF stabilization */
+
+    /* sort; as the LSFs should be nearly sorted, use insertion sort */
+    for (i = 1; i < order; i++) {
+        int j, value = nlsf[i];
+        for (j = i - 1; j >= 0 && nlsf[j] > value; j--)
+            nlsf[j + 1] = nlsf[j];
+        nlsf[j + 1] = value;
+    }
+
+    /* push forwards to increase distance */
+    if (nlsf[0] < min_delta[0])
+        nlsf[0] = min_delta[0];
+    for (i = 1; i < order; i++)
+        nlsf[i] = FFMAX(nlsf[i], FFMIN(nlsf[i - 1] + min_delta[i], 32767));
+
+    /* push backwards to increase distance */
+    if (nlsf[order-1] > 32768 - min_delta[order])
+        nlsf[order-1] = 32768 - min_delta[order];
+    for (i = order-2; i >= 0; i--)
+        if (nlsf[i] > nlsf[i + 1] - min_delta[i+1])
+            nlsf[i] = nlsf[i + 1] - min_delta[i+1];
+
+    return;
+}
+
+static inline int silk_is_lpc_stable(const int16_t lpc[16], int order)
+{
+    int k, j, DC_resp = 0;
+    int32_t lpc32[2][16];       // Q24
+    int totalinvgain = 1 << 30; // 1.0 in Q30
+    int32_t *row = lpc32[0], *prevrow;
+
+    /* initialize the first row for the Levinson recursion */
+    for (k = 0; k < order; k++) {
+        DC_resp += lpc[k];
+        row[k] = lpc[k] * 4096;
+    }
+
+    if (DC_resp >= 4096)
+        return 0;
+
+    /* check if prediction gain pushes any coefficients too far */
+    for (k = order - 1; 1; k--) {
+        int rc;      // Q31; reflection coefficient
+        int gaindiv; // Q30; inverse of the gain (the divisor)
+        int gain;    // gain for this reflection coefficient
+        int fbits;   // fractional bits used for the gain
+        int error;   // Q29; estimate of the error of our partial estimate of 1/gaindiv
+
+        if (FFABS(row[k]) > 16773022)
+            return 0;
+
+        rc      = -(row[k] * 128);
+        gaindiv = (1 << 30) - MULH(rc, rc);
+
+        totalinvgain = MULH(totalinvgain, gaindiv) << 2;
+        if (k == 0)
+            return (totalinvgain >= 107374);
+
+        /* approximate 1.0/gaindiv */
+        fbits = opus_ilog(gaindiv);
+        gain  = ((1 << 29) - 1) / (gaindiv >> (fbits + 1 - 16)); // Q<fbits-16>
+        error = (1 << 29) - MULL(gaindiv << (15 + 16 - fbits), gain, 16);
+        gain  = ((gain << 16) + (error * gain >> 13));
+
+        /* switch to the next row of the LPC coefficients */
+        prevrow = row;
+        row = lpc32[k & 1];
+
+        for (j = 0; j < k; j++) {
+            int x = av_sat_sub32(prevrow[j], ROUND_MULL(prevrow[k - j - 1], rc, 31));
+            int64_t tmp = ROUND_MULL(x, gain, fbits);
+
+            /* per RFC 8251 section 6, if this calculation overflows, the filter
+               is considered unstable. */
+            if (tmp < INT32_MIN || tmp > INT32_MAX)
+                return 0;
+
+            row[j] = (int32_t)tmp;
+        }
+    }
+}
+
+static void silk_lsp2poly(const int32_t lsp[/* 2 * half_order - 1 */],
+                          int32_t pol[/* half_order + 1 */], int half_order)
+{
+    int i, j;
+
+    pol[0] = 65536; // 1.0 in Q16
+    pol[1] = -lsp[0];
+
+    for (i = 1; i < half_order; i++) {
+        pol[i + 1] = pol[i - 1] * 2 - ROUND_MULL(lsp[2 * i], pol[i], 16);
+        for (j = i; j > 1; j--)
+            pol[j] += pol[j - 2] - ROUND_MULL(lsp[2 * i], pol[j - 1], 16);
+
+        pol[1] -= lsp[2 * i];
+    }
+}
+
+static void silk_lsf2lpc(const int16_t nlsf[16], float lpcf[16], int order)
+{
+    int i, k;
+    int32_t lsp[16];     // Q17; 2*cos(LSF)
+    int32_t p[9], q[9];  // Q16
+    int32_t lpc32[16];   // Q17
+    int16_t lpc[16];     // Q12
+
+    /* convert the LSFs to LSPs, i.e. 2*cos(LSF) */
+    for (k = 0; k < order; k++) {
+        int index = nlsf[k] >> 8;
+        int offset = nlsf[k] & 255;
+        int k2 = (order == 10) ? ff_silk_lsf_ordering_nbmb[k] : ff_silk_lsf_ordering_wb[k];
+
+        /* interpolate and round */
+        lsp[k2]  = ff_silk_cosine[index] * 256;
+        lsp[k2] += (ff_silk_cosine[index + 1] - ff_silk_cosine[index]) * offset;
+        lsp[k2]  = (lsp[k2] + 4) >> 3;
+    }
+
+    silk_lsp2poly(lsp    , p, order >> 1);
+    silk_lsp2poly(lsp + 1, q, order >> 1);
+
+    /* reconstruct A(z) */
+    for (k = 0; k < order>>1; k++) {
+        int32_t p_tmp = p[k + 1] + p[k];
+        int32_t q_tmp = q[k + 1] - q[k];
+        lpc32[k]         = -q_tmp - p_tmp;
+        lpc32[order-k-1] =  q_tmp - p_tmp;
+    }
+
+    /* limit the range of the LPC coefficients to each fit within an int16_t */
+    for (i = 0; i < 10; i++) {
+        int j;
+        unsigned int maxabs = 0;
+        for (j = 0, k = 0; j < order; j++) {
+            unsigned int x = FFABS(lpc32[k]);
+            if (x > maxabs) {
+                maxabs = x; // Q17
+                k      = j;
+            }
+        }
+
+        maxabs = (maxabs + 16) >> 5; // convert to Q12
+
+        if (maxabs > 32767) {
+            /* perform bandwidth expansion */
+            unsigned int chirp, chirp_base; // Q16
+            maxabs = FFMIN(maxabs, 163838); // anything above this overflows chirp's numerator
+            chirp_base = chirp = 65470 - ((maxabs - 32767) << 14) / ((maxabs * (k+1)) >> 2);
+
+            for (k = 0; k < order; k++) {
+                lpc32[k] = ROUND_MULL(lpc32[k], chirp, 16);
+                chirp    = (chirp_base * chirp + 32768) >> 16;
+            }
+        } else break;
+    }
+
+    if (i == 10) {
+        /* time's up: just clamp */
+        for (k = 0; k < order; k++) {
+            int x = (lpc32[k] + 16) >> 5;
+            lpc[k] = av_clip_int16(x);
+            lpc32[k] = lpc[k] << 5; // shortcut mandated by the spec; drops lower 5 bits
+        }
+    } else {
+        for (k = 0; k < order; k++)
+            lpc[k] = (lpc32[k] + 16) >> 5;
+    }
+
+    /* if the prediction gain causes the LPC filter to become unstable,
+       apply further bandwidth expansion on the Q17 coefficients */
+    for (i = 1; i <= 16 && !silk_is_lpc_stable(lpc, order); i++) {
+        unsigned int chirp, chirp_base;
+        chirp_base = chirp = 65536 - (1 << i);
+
+        for (k = 0; k < order; k++) {
+            lpc32[k] = ROUND_MULL(lpc32[k], chirp, 16);
+            lpc[k]   = (lpc32[k] + 16) >> 5;
+            chirp    = (chirp_base * chirp + 32768) >> 16;
+        }
+    }
+
+    for (i = 0; i < order; i++)
+        lpcf[i] = lpc[i] / 4096.0f;
+}
+
+static inline void silk_decode_lpc(SilkContext *s, SilkFrame *frame,
+                                   OpusRangeCoder *rc,
+                                   float lpc_leadin[16], float lpc[16],
+                                   int *lpc_order, int *has_lpc_leadin, int voiced)
+{
+    int i;
+    int order;                   // order of the LP polynomial; 10 for NB/MB and 16 for WB
+    int8_t  lsf_i1, lsf_i2[16];  // stage-1 and stage-2 codebook indices
+    int16_t lsf_res[16];         // residual as a Q10 value
+    int16_t nlsf[16];            // Q15
+
+    *lpc_order = order = s->wb ? 16 : 10;
+
+    /* obtain LSF stage-1 and stage-2 indices */
+    lsf_i1 = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s1[s->wb][voiced]);
+    for (i = 0; i < order; i++) {
+        int index = s->wb ? ff_silk_lsf_s2_model_sel_wb  [lsf_i1][i] :
+                            ff_silk_lsf_s2_model_sel_nbmb[lsf_i1][i];
+        lsf_i2[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s2[index]) - 4;
+        if (lsf_i2[i] == -4)
+            lsf_i2[i] -= ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s2_ext);
+        else if (lsf_i2[i] == 4)
+            lsf_i2[i] += ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_s2_ext);
+    }
+
+    /* reverse the backwards-prediction step */
+    for (i = order - 1; i >= 0; i--) {
+        int qstep = s->wb ? 9830 : 11796;
+
+        lsf_res[i] = lsf_i2[i] * 1024;
+        if (lsf_i2[i] < 0)      lsf_res[i] += 102;
+        else if (lsf_i2[i] > 0) lsf_res[i] -= 102;
+        lsf_res[i] = (lsf_res[i] * qstep) >> 16;
+
+        if (i + 1 < order) {
+            int weight = s->wb ? ff_silk_lsf_pred_weights_wb  [ff_silk_lsf_weight_sel_wb  [lsf_i1][i]][i] :
+                                 ff_silk_lsf_pred_weights_nbmb[ff_silk_lsf_weight_sel_nbmb[lsf_i1][i]][i];
+            lsf_res[i] += (lsf_res[i+1] * weight) >> 8;
+        }
+    }
+
+    /* reconstruct the NLSF coefficients from the supplied indices */
+    for (i = 0; i < order; i++) {
+        const uint8_t * codebook = s->wb ? ff_silk_lsf_codebook_wb  [lsf_i1] :
+                                           ff_silk_lsf_codebook_nbmb[lsf_i1];
+        int cur, prev, next, weight_sq, weight, ipart, fpart, y, value;
+
+        /* find the weight of the residual */
+        /* TODO: precompute */
+        cur = codebook[i];
+        prev = i ? codebook[i - 1] : 0;
+        next = i + 1 < order ? codebook[i + 1] : 256;
+        weight_sq = (1024 / (cur - prev) + 1024 / (next - cur)) << 16;
+
+        /* approximate square-root with mandated fixed-point arithmetic */
+        ipart = opus_ilog(weight_sq);
+        fpart = (weight_sq >> (ipart-8)) & 127;
+        y = ((ipart & 1) ? 32768 : 46214) >> ((32 - ipart)>>1);
+        weight = y + ((213 * fpart * y) >> 16);
+
+        value = cur * 128 + (lsf_res[i] * 16384) / weight;
+        nlsf[i] = av_clip_uintp2(value, 15);
+    }
+
+    /* stabilize the NLSF coefficients */
+    silk_stabilize_lsf(nlsf, order, s->wb ? ff_silk_lsf_min_spacing_wb :
+                                            ff_silk_lsf_min_spacing_nbmb);
+
+    /* produce an interpolation for the first 2 subframes, */
+    /* and then convert both sets of NLSFs to LPC coefficients */
+    *has_lpc_leadin = 0;
+    if (s->subframes == 4) {
+        int offset = ff_opus_rc_dec_cdf(rc, ff_silk_model_lsf_interpolation_offset);
+        if (offset != 4 && frame->coded) {
+            *has_lpc_leadin = 1;
+            if (offset != 0) {
+                int16_t nlsf_leadin[16];
+                for (i = 0; i < order; i++)
+                    nlsf_leadin[i] = frame->nlsf[i] +
+                        ((nlsf[i] - frame->nlsf[i]) * offset >> 2);
+                silk_lsf2lpc(nlsf_leadin, lpc_leadin, order);
+            } else  /* avoid re-computation for a (roughly) 1-in-4 occurrence */
+                memcpy(lpc_leadin, frame->lpc, 16 * sizeof(float));
+        } else
+            offset = 4;
+        s->nlsf_interp_factor = offset;
+
+        silk_lsf2lpc(nlsf, lpc, order);
+    } else {
+        s->nlsf_interp_factor = 4;
+        silk_lsf2lpc(nlsf, lpc, order);
+    }
+
+    memcpy(frame->nlsf, nlsf, order * sizeof(nlsf[0]));
+    memcpy(frame->lpc,  lpc,  order * sizeof(lpc[0]));
+}
+
+static inline void silk_count_children(OpusRangeCoder *rc, int model, int32_t total,
+                                       int32_t child[2])
+{
+    if (total != 0) {
+        child[0] = ff_opus_rc_dec_cdf(rc,
+                       ff_silk_model_pulse_location[model] + (((total - 1 + 5) * (total - 1)) >> 1));
+        child[1] = total - child[0];
+    } else {
+        child[0] = 0;
+        child[1] = 0;
+    }
+}
+
+static inline void silk_decode_excitation(SilkContext *s, OpusRangeCoder *rc,
+                                          float* excitationf,
+                                          int qoffset_high, int active, int voiced)
+{
+    int i;
+    uint32_t seed;
+    int shellblocks;
+    int ratelevel;
+    uint8_t pulsecount[20];     // total pulses in each shell block
+    uint8_t lsbcount[20] = {0}; // raw lsbits defined for each pulse in each shell block
+    int32_t excitation[320];    // Q23
+
+    /* excitation parameters */
+    seed = ff_opus_rc_dec_cdf(rc, ff_silk_model_lcg_seed);
+    shellblocks = ff_silk_shell_blocks[s->bandwidth][s->subframes >> 2];
+    ratelevel = ff_opus_rc_dec_cdf(rc, ff_silk_model_exc_rate[voiced]);
+
+    for (i = 0; i < shellblocks; i++) {
+        pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[ratelevel]);
+        if (pulsecount[i] == 17) {
+            while (pulsecount[i] == 17 && ++lsbcount[i] != 10)
+                pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[9]);
+            if (lsbcount[i] == 10)
+                pulsecount[i] = ff_opus_rc_dec_cdf(rc, ff_silk_model_pulse_count[10]);
+        }
+    }
+
+    /* decode pulse locations using PVQ */
+    for (i = 0; i < shellblocks; i++) {
+        if (pulsecount[i] != 0) {
+            int a, b, c, d;
+            int32_t * location = excitation + 16*i;
+            int32_t branch[4][2];
+            branch[0][0] = pulsecount[i];
+
+            /* unrolled tail recursion */
+            for (a = 0; a < 1; a++) {
+                silk_count_children(rc, 0, branch[0][a], branch[1]);
+                for (b = 0; b < 2; b++) {
+                    silk_count_children(rc, 1, branch[1][b], branch[2]);
+                    for (c = 0; c < 2; c++) {
+                        silk_count_children(rc, 2, branch[2][c], branch[3]);
+                        for (d = 0; d < 2; d++) {
+                            silk_count_children(rc, 3, branch[3][d], location);
+                            location += 2;
+                        }
+                    }
+                }
+            }
+        } else
+            memset(excitation + 16*i, 0, 16*sizeof(int32_t));
+    }
+
+    /* decode least significant bits */
+    for (i = 0; i < shellblocks << 4; i++) {
+        int bit;
+        for (bit = 0; bit < lsbcount[i >> 4]; bit++)
+            excitation[i] = (excitation[i] << 1) |
+                            ff_opus_rc_dec_cdf(rc, ff_silk_model_excitation_lsb);
+    }
+
+    /* decode signs */
+    for (i = 0; i < shellblocks << 4; i++) {
+        if (excitation[i] != 0) {
+            int sign = ff_opus_rc_dec_cdf(rc, ff_silk_model_excitation_sign[active +
+                                         voiced][qoffset_high][FFMIN(pulsecount[i >> 4], 6)]);
+            if (sign == 0)
+                excitation[i] *= -1;
+        }
+    }
+
+    /* assemble the excitation */
+    for (i = 0; i < shellblocks << 4; i++) {
+        int value = excitation[i];
+        excitation[i] = value * 256 | ff_silk_quant_offset[voiced][qoffset_high];
+        if (value < 0)      excitation[i] += 20;
+        else if (value > 0) excitation[i] -= 20;
+
+        /* invert samples pseudorandomly */
+        seed = 196314165 * seed + 907633515;
+        if (seed & 0x80000000)
+            excitation[i] *= -1;
+        seed += value;
+
+        excitationf[i] = excitation[i] / 8388608.0f;
+    }
+}
+
+/** Maximum residual history according to 4.2.7.6.1 */
+#define SILK_MAX_LAG  (288 + LTP_ORDER / 2)
+
+/** Order of the LTP filter */
+#define LTP_ORDER 5
+
+static void silk_decode_frame(SilkContext *s, OpusRangeCoder *rc,
+                              int frame_num, int channel, int coded_channels,
+                              int active, int active1, int redundant)
+{
+    /* per frame */
+    int voiced;       // combines with active to indicate inactive, active, or active+voiced
+    int qoffset_high;
+    int order;                             // order of the LPC coefficients
+    float lpc_leadin[16], lpc_body[16], residual[SILK_MAX_LAG + SILK_HISTORY];
+    int has_lpc_leadin;
+    float ltpscale;
+
+    /* per subframe */
+    struct {
+        float gain;
+        int pitchlag;
+        float ltptaps[5];
+    } sf[4];
+
+    SilkFrame * const frame = s->frame + channel;
+
+    int i;
+
+    /* obtain stereo weights */
+    if (coded_channels == 2 && channel == 0) {
+        int n, wi[2], ws[2], w[2];
+        n     = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s1);
+        wi[0] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s2) + 3 * (n / 5);
+        ws[0] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s3);
+        wi[1] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s2) + 3 * (n % 5);
+        ws[1] = ff_opus_rc_dec_cdf(rc, ff_silk_model_stereo_s3);
+
+        for (i = 0; i < 2; i++)
+            w[i] = ff_silk_stereo_weights[wi[i]] +
+                   (((ff_silk_stereo_weights[wi[i] + 1] - ff_silk_stereo_weights[wi[i]]) * 6554) >> 16)
+                    * (ws[i]*2 + 1);
+
+        s->stereo_weights[0] = (w[0] - w[1]) / 8192.0;
+        s->stereo_weights[1] = w[1]          / 8192.0;
+
+        /* and read the mid-only flag */
+        s->midonly = active1 ? 0 : ff_opus_rc_dec_cdf(rc, ff_silk_model_mid_only);
+    }
+
+    /* obtain frame type */
+    if (!active) {
+        qoffset_high = ff_opus_rc_dec_cdf(rc, ff_silk_model_frame_type_inactive);
+        voiced = 0;
+    } else {
+        int type = ff_opus_rc_dec_cdf(rc, ff_silk_model_frame_type_active);
+        qoffset_high = type & 1;
+        voiced = type >> 1;
+    }
+
+    /* obtain subframe quantization gains */
+    for (i = 0; i < s->subframes; i++) {
+        int log_gain;     //Q7
+        int ipart, fpart, lingain;
+
+        if (i == 0 && (frame_num == 0 || !frame->coded)) {
+            /* gain is coded absolute */
+            int x = ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_highbits[active + voiced]);
+            log_gain = (x<<3) | ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_lowbits);
+
+            if (frame->coded)
+                log_gain = FFMAX(log_gain, frame->log_gain - 16);
+        } else {
+            /* gain is coded relative */
+            int delta_gain = ff_opus_rc_dec_cdf(rc, ff_silk_model_gain_delta);
+            log_gain = av_clip_uintp2(FFMAX((delta_gain<<1) - 16,
+                                     frame->log_gain + delta_gain - 4), 6);
+        }
+
+        frame->log_gain = log_gain;
+
+        /* approximate 2**(x/128) with a Q7 (i.e. non-integer) input */
+        log_gain = (log_gain * 0x1D1C71 >> 16) + 2090;
+        ipart = log_gain >> 7;
+        fpart = log_gain & 127;
+        lingain = (1 << ipart) + ((-174 * fpart * (128-fpart) >>16) + fpart) * ((1<<ipart) >> 7);
+        sf[i].gain = lingain / 65536.0f;
+    }
+
+    /* obtain LPC filter coefficients */
+    silk_decode_lpc(s, frame, rc, lpc_leadin, lpc_body, &order, &has_lpc_leadin, voiced);
+
+    /* obtain pitch lags, if this is a voiced frame */
+    if (voiced) {
+        int lag_absolute = (!frame_num || !frame->prev_voiced);
+        int primarylag;         // primary pitch lag for the entire SILK frame
+        int ltpfilter;
+        const int8_t * offsets;
+
+        if (!lag_absolute) {
+            int delta = ff_opus_rc_dec_cdf(rc, ff_silk_model_pitch_delta);
+            if (delta)
+                primarylag = frame->primarylag + delta - 9;
+            else
+                lag_absolute = 1;
+        }
+
+        if (lag_absolute) {
+            /* primary lag is coded absolute */
+            int highbits, lowbits;
+            static const uint16_t * const model[] = {
+                ff_silk_model_pitch_lowbits_nb, ff_silk_model_pitch_lowbits_mb,
+                ff_silk_model_pitch_lowbits_wb
+            };
+            highbits = ff_opus_rc_dec_cdf(rc, ff_silk_model_pitch_highbits);
+            lowbits  = ff_opus_rc_dec_cdf(rc, model[s->bandwidth]);
+
+            primarylag = ff_silk_pitch_min_lag[s->bandwidth] +
+                         highbits*ff_silk_pitch_scale[s->bandwidth] + lowbits;
+        }
+        frame->primarylag = primarylag;
+
+        if (s->subframes == 2)
+            offsets = (s->bandwidth == OPUS_BANDWIDTH_NARROWBAND)
+                     ? ff_silk_pitch_offset_nb10ms[ff_opus_rc_dec_cdf(rc,
+                                                ff_silk_model_pitch_contour_nb10ms)]
+                     : ff_silk_pitch_offset_mbwb10ms[ff_opus_rc_dec_cdf(rc,
+                                                ff_silk_model_pitch_contour_mbwb10ms)];
+        else
+            offsets = (s->bandwidth == OPUS_BANDWIDTH_NARROWBAND)
+                     ? ff_silk_pitch_offset_nb20ms[ff_opus_rc_dec_cdf(rc,
+                                                ff_silk_model_pitch_contour_nb20ms)]
+                     : ff_silk_pitch_offset_mbwb20ms[ff_opus_rc_dec_cdf(rc,
+                                                ff_silk_model_pitch_contour_mbwb20ms)];
+
+        for (i = 0; i < s->subframes; i++)
+            sf[i].pitchlag = av_clip(primarylag + offsets[i],
+                                     ff_silk_pitch_min_lag[s->bandwidth],
+                                     ff_silk_pitch_max_lag[s->bandwidth]);
+
+        /* obtain LTP filter coefficients */
+        ltpfilter = ff_opus_rc_dec_cdf(rc, ff_silk_model_ltp_filter);
+        for (i = 0; i < s->subframes; i++) {
+            int index, j;
+            static const uint16_t * const filter_sel[] = {
+                ff_silk_model_ltp_filter0_sel, ff_silk_model_ltp_filter1_sel,
+                ff_silk_model_ltp_filter2_sel
+            };
+            static const int8_t (* const filter_taps[])[5] = {
+                ff_silk_ltp_filter0_taps, ff_silk_ltp_filter1_taps, ff_silk_ltp_filter2_taps
+            };
+            index = ff_opus_rc_dec_cdf(rc, filter_sel[ltpfilter]);
+            for (j = 0; j < 5; j++)
+                sf[i].ltptaps[j] = filter_taps[ltpfilter][index][j] / 128.0f;
+        }
+    }
+
+    /* obtain LTP scale factor */
+    if (voiced && frame_num == 0)
+        ltpscale = ff_silk_ltp_scale_factor[ff_opus_rc_dec_cdf(rc,
+                                         ff_silk_model_ltp_scale_index)] / 16384.0f;
+    else ltpscale = 15565.0f/16384.0f;
+
+    /* generate the excitation signal for the entire frame */
+    silk_decode_excitation(s, rc, residual + SILK_MAX_LAG, qoffset_high,
+                           active, voiced);
+
+    /* skip synthesising the output if we do not need it */
+    // TODO: implement error recovery
+    if (s->output_channels == channel || redundant)
+        return;
+
+    /* generate the output signal */
+    for (i = 0; i < s->subframes; i++) {
+        const float * lpc_coeff = (i < 2 && has_lpc_leadin) ? lpc_leadin : lpc_body;
+        float *dst    = frame->output      + SILK_HISTORY + i * s->sflength;
+        float *resptr = residual           + SILK_MAX_LAG + i * s->sflength;
+        float *lpc    = frame->lpc_history + SILK_HISTORY + i * s->sflength;
+        float sum;
+        int j, k;
+
+        if (voiced) {
+            int out_end;
+            float scale;
+
+            if (i < 2 || s->nlsf_interp_factor == 4) {
+                out_end = -i * s->sflength;
+                scale   = ltpscale;
+            } else {
+                out_end = -(i - 2) * s->sflength;
+                scale   = 1.0f;
+            }
+
+            /* when the LPC coefficients change, a re-whitening filter is used */
+            /* to produce a residual that accounts for the change */
+            for (j = - sf[i].pitchlag - LTP_ORDER/2; j < out_end; j++) {
+                sum = dst[j];
+                for (k = 0; k < order; k++)
+                    sum -= lpc_coeff[k] * dst[j - k - 1];
+                resptr[j] = av_clipf(sum, -1.0f, 1.0f) * scale / sf[i].gain;
+            }
+
+            if (out_end) {
+                float rescale = sf[i-1].gain / sf[i].gain;
+                for (j = out_end; j < 0; j++)
+                    resptr[j] *= rescale;
+            }
+
+            /* LTP synthesis */
+            for (j = 0; j < s->sflength; j++) {
+                sum = resptr[j];
+                for (k = 0; k < LTP_ORDER; k++)
+                    sum += sf[i].ltptaps[k] * resptr[j - sf[i].pitchlag + LTP_ORDER/2 - k];
+                resptr[j] = sum;
+            }
+        }
+
+        /* LPC synthesis */
+        for (j = 0; j < s->sflength; j++) {
+            sum = resptr[j] * sf[i].gain;
+            for (k = 1; k <= order; k++)
+                sum += lpc_coeff[k - 1] * lpc[j - k];
+
+            lpc[j] = sum;
+            dst[j] = av_clipf(sum, -1.0f, 1.0f);
+        }
+    }
+
+    frame->prev_voiced = voiced;
+    memmove(frame->lpc_history, frame->lpc_history + s->flength, SILK_HISTORY * sizeof(float));
+    memmove(frame->output,      frame->output      + s->flength, SILK_HISTORY * sizeof(float));
+
+    frame->coded = 1;
+}
+
+static void silk_unmix_ms(SilkContext *s, float *l, float *r)
+{
+    float *mid    = s->frame[0].output + SILK_HISTORY - s->flength;
+    float *side   = s->frame[1].output + SILK_HISTORY - s->flength;
+    float w0_prev = s->prev_stereo_weights[0];
+    float w1_prev = s->prev_stereo_weights[1];
+    float w0      = s->stereo_weights[0];
+    float w1      = s->stereo_weights[1];
+    int n1        = ff_silk_stereo_interp_len[s->bandwidth];
+    int i;
+
+    for (i = 0; i < n1; i++) {
+        float interp0 = w0_prev + i * (w0 - w0_prev) / n1;
+        float interp1 = w1_prev + i * (w1 - w1_prev) / n1;
+        float p0      = 0.25 * (mid[i - 2] + 2 * mid[i - 1] + mid[i]);
+
+        l[i] = av_clipf((1 + interp1) * mid[i - 1] + side[i - 1] + interp0 * p0, -1.0, 1.0);
+        r[i] = av_clipf((1 - interp1) * mid[i - 1] - side[i - 1] - interp0 * p0, -1.0, 1.0);
+    }
+
+    for (; i < s->flength; i++) {
+        float p0 = 0.25 * (mid[i - 2] + 2 * mid[i - 1] + mid[i]);
+
+        l[i] = av_clipf((1 + w1) * mid[i - 1] + side[i - 1] + w0 * p0, -1.0, 1.0);
+        r[i] = av_clipf((1 - w1) * mid[i - 1] - side[i - 1] - w0 * p0, -1.0, 1.0);
+    }
+
+    memcpy(s->prev_stereo_weights, s->stereo_weights, sizeof(s->stereo_weights));
+}
+
+static void silk_flush_frame(SilkFrame *frame)
+{
+    if (!frame->coded)
+        return;
+
+    memset(frame->output,      0, sizeof(frame->output));
+    memset(frame->lpc_history, 0, sizeof(frame->lpc_history));
+
+    memset(frame->lpc,  0, sizeof(frame->lpc));
+    memset(frame->nlsf, 0, sizeof(frame->nlsf));
+
+    frame->log_gain = 0;
+
+    frame->primarylag  = 0;
+    frame->prev_voiced = 0;
+    frame->coded       = 0;
+}
+
+int ff_silk_decode_superframe(SilkContext *s, OpusRangeCoder *rc,
+                              float *output[2],
+                              enum OpusBandwidth bandwidth,
+                              int coded_channels,
+                              int duration_ms)
+{
+    int active[2][6], redundancy[2];
+    int nb_frames, i, j;
+
+    if (bandwidth > OPUS_BANDWIDTH_WIDEBAND ||
+        coded_channels > 2 || duration_ms > 60) {
+        av_log(s->logctx, AV_LOG_ERROR, "Invalid parameters passed "
+               "to the SILK decoder.\n");
+        return AVERROR(EINVAL);
+    }
+
+    nb_frames = 1 + (duration_ms > 20) + (duration_ms > 40);
+    s->subframes = duration_ms / nb_frames / 5;         // 5ms subframes
+    s->sflength  = 20 * (bandwidth + 2);
+    s->flength   = s->sflength * s->subframes;
+    s->bandwidth = bandwidth;
+    s->wb        = bandwidth == OPUS_BANDWIDTH_WIDEBAND;
+
+    /* make sure to flush the side channel when switching from mono to stereo */
+    if (coded_channels > s->prev_coded_channels)
+        silk_flush_frame(&s->frame[1]);
+    s->prev_coded_channels = coded_channels;
+
+    /* read the LP-layer header bits */
+    for (i = 0; i < coded_channels; i++) {
+        for (j = 0; j < nb_frames; j++)
+            active[i][j] = ff_opus_rc_dec_log(rc, 1);
+
+        redundancy[i] = ff_opus_rc_dec_log(rc, 1);
+    }
+
+    /* read the per-frame LBRR flags */
+    for (i = 0; i < coded_channels; i++)
+        if (redundancy[i] && duration_ms > 20) {
+            redundancy[i] = ff_opus_rc_dec_cdf(rc, duration_ms == 40 ?
+                                                   ff_silk_model_lbrr_flags_40 : ff_silk_model_lbrr_flags_60);
+        }
+
+    /* decode the LBRR frames */
+    for (i = 0; i < nb_frames; i++) {
+        for (j = 0; j < coded_channels; j++)
+            if (redundancy[j] & (1 << i)) {
+                int active1 = (j == 0 && !(redundancy[1] & (1 << i))) ? 0 : 1;
+                silk_decode_frame(s, rc, i, j, coded_channels, 1, active1, 1);
+            }
+
+        s->midonly = 0;
+    }
+
+    for (i = 0; i < nb_frames; i++) {
+        for (j = 0; j < coded_channels && !s->midonly; j++)
+            silk_decode_frame(s, rc, i, j, coded_channels, active[j][i], active[1][i], 0);
+
+        /* reset the side channel if it is not coded */
+        if (s->midonly && s->frame[1].coded)
+            silk_flush_frame(&s->frame[1]);
+
+        if (coded_channels == 1 || s->output_channels == 1) {
+            for (j = 0; j < s->output_channels; j++) {
+                memcpy(output[j] + i * s->flength,
+                       s->frame[0].output + SILK_HISTORY - s->flength - 2,
+                       s->flength * sizeof(float));
+            }
+        } else {
+            silk_unmix_ms(s, output[0] + i * s->flength, output[1] + i * s->flength);
+        }
+
+        s->midonly        = 0;
+    }
+
+    return nb_frames * s->flength;
+}
+
+void ff_silk_free(SilkContext **ps)
+{
+    av_freep(ps);
+}
+
+void ff_silk_flush(SilkContext *s)
+{
+    silk_flush_frame(&s->frame[0]);
+    silk_flush_frame(&s->frame[1]);
+
+    memset(s->prev_stereo_weights, 0, sizeof(s->prev_stereo_weights));
+}
+
+int ff_silk_init(void *logctx, SilkContext **ps, int output_channels)
+{
+    SilkContext *s;
+
+    if (output_channels != 1 && output_channels != 2) {
+        av_log(logctx, AV_LOG_ERROR, "Invalid number of output channels: %d\n",
+               output_channels);
+        return AVERROR(EINVAL);
+    }
+
+    s = av_mallocz(sizeof(*s));
+    if (!s)
+        return AVERROR(ENOMEM);
+
+    s->logctx          = logctx;
+    s->output_channels = output_channels;
+
+    ff_silk_flush(s);
+
+    *ps = s;
+
+    return 0;
+}
diff --git a/libavcodec/opus/silk.h b/libavcodec/opus/silk.h
new file mode 100644
index 0000000000..824b492715
--- /dev/null
+++ b/libavcodec/opus/silk.h
@@ -0,0 +1,47 @@
+/*
+ * Opus Silk functions/definitions
+ * Copyright (c) 2012 Andrew D'Addesio
+ * Copyright (c) 2013-2014 Mozilla Corporation
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_OPUS_SILK_H
+#define AVCODEC_OPUS_SILK_H
+
+#include "opus.h"
+#include "rc.h"
+
+#define SILK_HISTORY                 322
+#define SILK_MAX_LPC                 16
+
+typedef struct SilkContext SilkContext;
+
+int ff_silk_init(void *logctx, SilkContext **ps, int output_channels);
+void ff_silk_free(SilkContext **ps);
+void ff_silk_flush(SilkContext *s);
+
+/**
+ * Decode the LP layer of one Opus frame (which may correspond to several SILK
+ * frames).
+ */
+int ff_silk_decode_superframe(SilkContext *s, OpusRangeCoder *rc,
+                              float *output[2],
+                              enum OpusBandwidth bandwidth, int coded_channels,
+                              int duration_ms);
+
+#endif /* AVCODEC_OPUS_SILK_H */
diff --git a/libavcodec/opus/tab.c b/libavcodec/opus/tab.c
new file mode 100644
index 0000000000..e7d20d1688
--- /dev/null
+++ b/libavcodec/opus/tab.c
@@ -0,0 +1,1189 @@
+/*
+ * Copyright (c) 2012 Andrew D'Addesio
+ * Copyright (c) 2013-2014 Mozilla Corporation
+ * Copyright (c) 2016 Rostislav Pehlivanov <atomnuker@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavutil/mem_internal.h"
+
+#include "tab.h"
+
+const uint8_t ff_opus_default_coupled_streams[] = { 0, 1, 1, 2, 2, 2, 2, 3 };
+
+const uint8_t ff_celt_band_end[] = { 13, 17, 17, 19, 21 };
+
+const uint16_t ff_silk_model_lbrr_flags_40[] = { 256, 0, 53, 106, 256 };
+const uint16_t ff_silk_model_lbrr_flags_60[] = { 256, 0, 41, 61, 90, 131, 146, 174, 256 };
+
+const uint16_t ff_silk_model_stereo_s1[] = {
+    256,   7,   9,  10,  11,  12,  22,  46,  54,  55,  56,  59,  82, 174, 197, 200,
+    201, 202, 210, 234, 244, 245, 246, 247, 249, 256
+};
+
+const uint16_t ff_silk_model_stereo_s2[] = {256, 85, 171, 256};
+
+const uint16_t ff_silk_model_stereo_s3[] = {256, 51, 102, 154, 205, 256};
+
+const uint16_t ff_silk_model_mid_only[] = {256, 192, 256};
+
+const uint16_t ff_silk_model_frame_type_inactive[] = {256, 26, 256};
+
+const uint16_t ff_silk_model_frame_type_active[] = {256, 24, 98, 246, 256};
+
+const uint16_t ff_silk_model_gain_highbits[3][9] = {
+    {256,  32, 144, 212, 241, 253, 254, 255, 256},
+    {256,   2,  19,  64, 124, 186, 233, 252, 256},
+    {256,   1,   4,  30, 101, 195, 245, 254, 256}
+};
+
+const uint16_t ff_silk_model_gain_lowbits[] = {256, 32, 64, 96, 128, 160, 192, 224, 256};
+
+const uint16_t ff_silk_model_gain_delta[] = {
+    256,   6,  11,  22,  53, 185, 206, 214, 218, 221, 223, 225, 227, 228, 229, 230,
+    231, 232, 233, 234, 235, 236, 237, 238, 239, 240, 241, 242, 243, 244, 245, 246,
+    247, 248, 249, 250, 251, 252, 253, 254, 255, 256
+};
+const uint16_t ff_silk_model_lsf_s1[2][2][33] = {
+    {
+        {    // NB or MB, unvoiced
+            256,  44,  78, 108, 127, 148, 160, 171, 174, 177, 179, 195, 197, 199, 200, 205,
+            207, 208, 211, 214, 215, 216, 218, 220, 222, 225, 226, 235, 244, 246, 253, 255, 256
+        }, { // NB or MB, voiced
+            256,   1,  11,  12,  20,  23,  31,  39,  53,  66,  80,  81,  95, 107, 120, 131,
+            142, 154, 165, 175, 185, 196, 204, 213, 221, 228, 236, 237, 238, 244, 245, 251, 256
+        }
+    }, {
+        {    // WB, unvoiced
+            256,  31,  52,  55,  72,  73,  81,  98, 102, 103, 121, 137, 141, 143, 146, 147,
+            157, 158, 161, 177, 188, 204, 206, 208, 211, 213, 224, 225, 229, 238, 246, 253, 256
+        }, { // WB, voiced
+            256,   1,   5,  21,  26,  44,  55,  60,  74,  89,  90,  93, 105, 118, 132, 146,
+            152, 166, 178, 180, 186, 187, 199, 211, 222, 232, 235, 245, 250, 251, 252, 253, 256
+        }
+    }
+};
+
+const uint16_t ff_silk_model_lsf_s2[32][10] = {
+    // NB, MB
+    { 256,   1,   2,   3,  18, 242, 253, 254, 255, 256 },
+    { 256,   1,   2,   4,  38, 221, 253, 254, 255, 256 },
+    { 256,   1,   2,   6,  48, 197, 252, 254, 255, 256 },
+    { 256,   1,   2,  10,  62, 185, 246, 254, 255, 256 },
+    { 256,   1,   4,  20,  73, 174, 248, 254, 255, 256 },
+    { 256,   1,   4,  21,  76, 166, 239, 254, 255, 256 },
+    { 256,   1,   8,  32,  85, 159, 226, 252, 255, 256 },
+    { 256,   1,   2,  20,  83, 161, 219, 249, 255, 256 },
+
+    // WB
+    { 256,   1,   2,   3,  12, 244, 253, 254, 255, 256 },
+    { 256,   1,   2,   4,  32, 218, 253, 254, 255, 256 },
+    { 256,   1,   2,   5,  47, 199, 252, 254, 255, 256 },
+    { 256,   1,   2,  12,  61, 187, 252, 254, 255, 256 },
+    { 256,   1,   5,  24,  72, 172, 249, 254, 255, 256 },
+    { 256,   1,   2,  16,  70, 170, 242, 254, 255, 256 },
+    { 256,   1,   2,  17,  78, 165, 226, 251, 255, 256 },
+    { 256,   1,   8,  29,  79, 156, 237, 254, 255, 256 }
+};
+
+const uint16_t ff_silk_model_lsf_s2_ext[] = { 256, 156, 216, 240, 249, 253, 255, 256 };
+
+const uint16_t ff_silk_model_lsf_interpolation_offset[] = { 256, 13, 35, 64, 75, 256 };
+
+const uint16_t ff_silk_model_pitch_highbits[] = {
+    256,   3,   6,  12,  23,  44,  74, 106, 125, 136, 146, 158, 171, 184, 196, 207,
+    216, 224, 231, 237, 241, 243, 245, 247, 248, 249, 250, 251, 252, 253, 254, 255, 256
+};
+
+const uint16_t ff_silk_model_pitch_lowbits_nb[] = { 256, 64, 128, 192, 256 };
+
+const uint16_t ff_silk_model_pitch_lowbits_mb[] = { 256, 43, 85, 128, 171, 213, 256 };
+
+const uint16_t ff_silk_model_pitch_lowbits_wb[] = { 256, 32, 64, 96, 128, 160, 192, 224, 256 };
+
+const uint16_t ff_silk_model_pitch_delta[] = {
+    256,  46,  48,  50,  53,  57,  63,  73,  88, 114, 152, 182, 204, 219, 229, 236,
+    242, 246, 250, 252, 254, 256
+};
+
+const uint16_t ff_silk_model_pitch_contour_nb10ms[] = { 256, 143, 193, 256 };
+
+const uint16_t ff_silk_model_pitch_contour_nb20ms[] = {
+    256,  68,  80, 101, 118, 137, 159, 189, 213, 230, 246, 256
+};
+
+const uint16_t ff_silk_model_pitch_contour_mbwb10ms[] = {
+    256,  91, 137, 176, 195, 209, 221, 229, 236, 242, 247, 252, 256
+};
+
+const uint16_t ff_silk_model_pitch_contour_mbwb20ms[] = {
+    256,  33,  55,  73,  89, 104, 118, 132, 145, 158, 168, 177, 186, 194, 200, 206,
+    212, 217, 221, 225, 229, 232, 235, 238, 240, 242, 244, 246, 248, 250, 252, 253,
+    254, 255, 256
+};
+
+const uint16_t ff_silk_model_ltp_filter[] = { 256, 77, 157, 256 };
+
+const uint16_t ff_silk_model_ltp_filter0_sel[] = {
+    256, 185, 200, 213, 226, 235, 244, 250, 256
+};
+
+const uint16_t ff_silk_model_ltp_filter1_sel[] = {
+    256,  57,  91, 112, 132, 147, 160, 172, 185, 195, 205, 214, 224, 233, 241, 248, 256
+};
+
+const uint16_t ff_silk_model_ltp_filter2_sel[] = {
+    256,  15,  31,  45,  57,  69,  81,  92, 103, 114, 124, 133, 142, 151, 160, 168,
+    176, 184, 192, 199, 206, 212, 218, 223, 227, 232, 236, 240, 244, 247, 251, 254, 256
+};
+
+const uint16_t ff_silk_model_ltp_scale_index[] = { 256, 128, 192, 256 };
+
+const uint16_t ff_silk_model_lcg_seed[] = { 256, 64, 128, 192, 256 };
+
+const uint16_t ff_silk_model_exc_rate[2][10] = {
+    { 256,  15,  66,  78, 124, 169, 182, 215, 242, 256 }, // unvoiced
+    { 256,  33,  63,  99, 116, 150, 199, 217, 238, 256 }  // voiced
+};
+
+const uint16_t ff_silk_model_pulse_count[11][19] = {
+    { 256, 131, 205, 230, 238, 241, 244, 245, 246,
+      247, 248, 249, 250, 251, 252, 253, 254, 255, 256 },
+    { 256,  58, 151, 211, 234, 241, 244, 245, 246,
+      247, 248, 249, 250, 251, 252, 253, 254, 255, 256 },
+    { 256,  43,  94, 140, 173, 197, 213, 224, 232,
+      238, 241, 244, 247, 249, 250, 251, 253, 254, 256 },
+    { 256,  17,  69, 140, 197, 228, 240, 245, 246,
+      247, 248, 249, 250, 251, 252, 253, 254, 255, 256 },
+    { 256,   6,  27,  68, 121, 170, 205, 226, 237,
+      243, 246, 248, 250, 251, 252, 253, 254, 255, 256 },
+    { 256,   7,  21,  43,  71, 100, 128, 153, 173,
+      190, 203, 214, 223, 230, 235, 239, 243, 246, 256 },
+    { 256,   2,   7,  21,  50,  92, 138, 179, 210,
+      229, 240, 246, 249, 251, 252, 253, 254, 255, 256 },
+    { 256,   1,   3,   7,  17,  36,  65, 100, 137,
+      171, 199, 219, 233, 241, 246, 250, 252, 254, 256 },
+    { 256,   1,   3,   5,  10,  19,  33,  53,  77,
+      104, 132, 158, 181, 201, 216, 227, 235, 241, 256 },
+    { 256,   1,   2,   3,   9,  36,  94, 150, 189,
+      214, 228, 238, 244, 247, 250, 252, 253, 254, 256 },
+    { 256,   2,   3,   9,  36,  94, 150, 189, 214,
+      228, 238, 244, 247, 250, 252, 253, 254, 256, 256 }
+};
+
+const uint16_t ff_silk_model_pulse_location[4][168] = {
+    {
+        256, 126, 256,
+        256, 56, 198, 256,
+        256, 25, 126, 230, 256,
+        256, 12, 72, 180, 244, 256,
+        256, 7, 42, 126, 213, 250, 256,
+        256, 4, 24, 83, 169, 232, 253, 256,
+        256, 3, 15, 53, 125, 200, 242, 254, 256,
+        256, 2, 10, 35, 89, 162, 221, 248, 255, 256,
+        256, 2, 7, 24, 63, 126, 191, 233, 251, 255, 256,
+        256, 1, 5, 17, 45, 94, 157, 211, 241, 252, 255, 256,
+        256, 1, 5, 13, 33, 70, 125, 182, 223, 245, 253, 255, 256,
+        256, 1, 4, 11, 26, 54, 98, 151, 199, 232, 248, 254, 255, 256,
+        256, 1, 3, 9, 21, 42, 77, 124, 172, 212, 237, 249, 254, 255, 256,
+        256, 1, 2, 6, 16, 33, 60, 97, 144, 187, 220, 241, 250, 254, 255, 256,
+        256, 1, 2, 3, 11, 25, 47, 80, 120, 163, 201, 229, 245, 253, 254, 255, 256,
+        256, 1, 2, 3, 4, 17, 35, 62, 98, 139, 180, 214, 238, 252, 253, 254, 255, 256
+    },{
+        256, 127, 256,
+        256, 53, 202, 256,
+        256, 22, 127, 233, 256,
+        256, 11, 72, 183, 246, 256,
+        256, 6, 41, 127, 215, 251, 256,
+        256, 4, 24, 83, 170, 232, 253, 256,
+        256, 3, 16, 56, 127, 200, 241, 254, 256,
+        256, 3, 12, 39, 92, 162, 218, 246, 255, 256,
+        256, 3, 11, 30, 67, 124, 185, 229, 249, 255, 256,
+        256, 3, 10, 25, 53, 97, 151, 200, 233, 250, 255, 256,
+        256, 1, 8, 21, 43, 77, 123, 171, 209, 237, 251, 255, 256,
+        256, 1, 2, 13, 35, 62, 97, 139, 186, 219, 244, 254, 255, 256,
+        256, 1, 2, 8, 22, 48, 85, 128, 171, 208, 234, 248, 254, 255, 256,
+        256, 1, 2, 6, 16, 36, 67, 107, 149, 189, 220, 240, 250, 254, 255, 256,
+        256, 1, 2, 5, 13, 29, 55, 90, 128, 166, 201, 227, 243, 251, 254, 255, 256,
+        256, 1, 2, 4, 10, 22, 43, 73, 109, 147, 183, 213, 234, 246, 252, 254, 255, 256
+    },{
+        256, 127, 256,
+        256, 49, 206, 256,
+        256, 20, 127, 236, 256,
+        256, 11, 71, 184, 246, 256,
+        256, 7, 43, 127, 214, 250, 256,
+        256, 6, 30, 87, 169, 229, 252, 256,
+        256, 5, 23, 62, 126, 194, 236, 252, 256,
+        256, 6, 20, 49, 96, 157, 209, 239, 253, 256,
+        256, 1, 16, 39, 74, 125, 175, 215, 245, 255, 256,
+        256, 1, 2, 23, 55, 97, 149, 195, 236, 254, 255, 256,
+        256, 1, 7, 23, 50, 86, 128, 170, 206, 233, 249, 255, 256,
+        256, 1, 6, 18, 39, 70, 108, 148, 186, 217, 238, 250, 255, 256,
+        256, 1, 4, 13, 30, 56, 90, 128, 166, 200, 226, 243, 252, 255, 256,
+        256, 1, 4, 11, 25, 47, 76, 110, 146, 180, 209, 231, 245, 252, 255, 256,
+        256, 1, 3, 8, 19, 37, 62, 93, 128, 163, 194, 219, 237, 248, 253, 255, 256,
+        256, 1, 2, 6, 15, 30, 51, 79, 111, 145, 177, 205, 226, 241, 250, 254, 255, 256
+    },{
+        256, 128, 256,
+        256, 42, 214, 256,
+        256, 21, 128, 235, 256,
+        256, 12, 72, 184, 245, 256,
+        256, 8, 42, 128, 214, 249, 256,
+        256, 8, 31, 86, 176, 231, 251, 256,
+        256, 5, 20, 58, 130, 202, 238, 253, 256,
+        256, 6, 18, 45, 97, 174, 221, 241, 251, 256,
+        256, 6, 25, 53, 88, 128, 168, 203, 231, 250, 256,
+        256, 4, 18, 40, 71, 108, 148, 185, 216, 238, 252, 256,
+        256, 3, 13, 31, 57, 90, 128, 166, 199, 225, 243, 253, 256,
+        256, 2, 10, 23, 44, 73, 109, 147, 183, 212, 233, 246, 254, 256,
+        256, 1, 6, 16, 33, 58, 90, 128, 166, 198, 223, 240, 250, 255, 256,
+        256, 1, 5, 12, 25, 46, 75, 110, 146, 181, 210, 231, 244, 251, 255, 256,
+        256, 1, 3, 8, 18, 35, 60, 92, 128, 164, 196, 221, 238, 248, 253, 255, 256,
+        256, 1, 3, 7, 14, 27, 48, 76, 110, 146, 180, 208, 229, 242, 249, 253, 255, 256
+    }
+};
+
+const uint16_t ff_silk_model_excitation_lsb[] = {256, 136, 256};
+
+const uint16_t ff_silk_model_excitation_sign[3][2][7][3] = {
+    {    // Inactive
+        {    // Low offset
+            {256,   2, 256},
+            {256, 207, 256},
+            {256, 189, 256},
+            {256, 179, 256},
+            {256, 174, 256},
+            {256, 163, 256},
+            {256, 157, 256}
+        }, { // High offset
+            {256,  58, 256},
+            {256, 245, 256},
+            {256, 238, 256},
+            {256, 232, 256},
+            {256, 225, 256},
+            {256, 220, 256},
+            {256, 211, 256}
+        }
+    }, { // Unvoiced
+        {    // Low offset
+            {256,   1, 256},
+            {256, 210, 256},
+            {256, 190, 256},
+            {256, 178, 256},
+            {256, 169, 256},
+            {256, 162, 256},
+            {256, 152, 256}
+        }, { // High offset
+            {256,  48, 256},
+            {256, 242, 256},
+            {256, 235, 256},
+            {256, 224, 256},
+            {256, 214, 256},
+            {256, 205, 256},
+            {256, 190, 256}
+        }
+    }, { // Voiced
+        {    // Low offset
+            {256,   1, 256},
+            {256, 162, 256},
+            {256, 152, 256},
+            {256, 147, 256},
+            {256, 144, 256},
+            {256, 141, 256},
+            {256, 138, 256}
+        }, { // High offset
+            {256,   8, 256},
+            {256, 203, 256},
+            {256, 187, 256},
+            {256, 176, 256},
+            {256, 168, 256},
+            {256, 161, 256},
+            {256, 154, 256}
+        }
+    }
+};
+
+const int16_t ff_silk_stereo_weights[] = {
+    -13732, -10050,  -8266,  -7526,  -6500,  -5000,  -2950,   -820,
+       820,   2950,   5000,   6500,   7526,   8266,  10050,  13732
+};
+
+const uint8_t ff_silk_lsf_s2_model_sel_nbmb[32][10] = {
+    { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+    { 1, 3, 1, 2, 2, 1, 2, 1, 1, 1 },
+    { 2, 1, 1, 1, 1, 1, 1, 1, 1, 1 },
+    { 1, 2, 2, 2, 2, 1, 2, 1, 1, 1 },
+    { 2, 3, 3, 3, 3, 2, 2, 2, 2, 2 },
+    { 0, 5, 3, 3, 2, 2, 2, 2, 1, 1 },
+    { 0, 2, 2, 2, 2, 2, 2, 2, 2, 1 },
+    { 2, 3, 6, 4, 4, 4, 5, 4, 5, 5 },
+    { 2, 4, 5, 5, 4, 5, 4, 6, 4, 4 },
+    { 2, 4, 4, 7, 4, 5, 4, 5, 5, 4 },
+    { 4, 3, 3, 3, 2, 3, 2, 2, 2, 2 },
+    { 1, 5, 5, 6, 4, 5, 4, 5, 5, 5 },
+    { 2, 7, 4, 6, 5, 5, 5, 5, 5, 5 },
+    { 2, 7, 5, 5, 5, 5, 5, 6, 5, 4 },
+    { 3, 3, 5, 4, 4, 5, 4, 5, 4, 4 },
+    { 2, 3, 3, 5, 5, 4, 4, 4, 4, 4 },
+    { 2, 4, 4, 6, 4, 5, 4, 5, 5, 5 },
+    { 2, 5, 4, 6, 5, 5, 5, 4, 5, 4 },
+    { 2, 7, 4, 5, 4, 5, 4, 5, 5, 5 },
+    { 2, 5, 4, 6, 7, 6, 5, 6, 5, 4 },
+    { 3, 6, 7, 4, 6, 5, 5, 6, 4, 5 },
+    { 2, 7, 6, 4, 4, 4, 5, 4, 5, 5 },
+    { 4, 5, 5, 4, 6, 6, 5, 6, 5, 4 },
+    { 2, 5, 5, 6, 5, 6, 4, 6, 4, 4 },
+    { 4, 5, 5, 5, 3, 7, 4, 5, 5, 4 },
+    { 2, 3, 4, 5, 5, 6, 4, 5, 5, 4 },
+    { 2, 3, 2, 3, 3, 4, 2, 3, 3, 3 },
+    { 1, 1, 2, 2, 2, 2, 2, 3, 2, 2 },
+    { 4, 5, 5, 6, 6, 6, 5, 6, 4, 5 },
+    { 3, 5, 5, 4, 4, 4, 4, 3, 3, 2 },
+    { 2, 5, 3, 7, 5, 5, 4, 4, 5, 4 },
+    { 4, 4, 5, 4, 5, 6, 5, 6, 5, 4 }
+};
+
+const uint8_t ff_silk_lsf_s2_model_sel_wb[32][16] = {
+    {  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8 },
+    { 10, 11, 11, 11, 11, 11, 10, 10, 10, 10, 10,  9,  9,  9,  8, 11 },
+    { 10, 13, 13, 11, 15, 12, 12, 13, 10, 13, 12, 13, 13, 12, 11, 11 },
+    {  8, 10,  9, 10, 10,  9,  9,  9,  9,  9,  8,  8,  8,  8,  8,  9 },
+    {  8, 14, 13, 12, 14, 12, 15, 13, 12, 12, 12, 13, 13, 12, 12, 11 },
+    {  8, 11, 13, 13, 12, 11, 11, 13, 11, 11, 11, 11, 11, 11, 10, 12 },
+    {  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8 },
+    {  8, 10, 14, 11, 15, 10, 13, 11, 12, 13, 13, 12, 11, 11, 10, 11 },
+    {  8, 14, 10, 14, 14, 12, 13, 12, 14, 13, 12, 12, 13, 11, 11, 11 },
+    { 10,  9,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8 },
+    {  8,  9,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  9 },
+    { 10, 10, 11, 12, 13, 11, 11, 11, 11, 11, 11, 11, 10, 10,  9, 11 },
+    { 10, 10, 11, 11, 12, 11, 11, 11, 11, 11, 11, 11, 11, 10,  9, 11 },
+    { 11, 12, 12, 12, 14, 12, 12, 13, 11, 13, 12, 12, 13, 12, 11, 12 },
+    {  8, 14, 12, 13, 12, 15, 13, 10, 14, 13, 15, 12, 12, 11, 13, 11 },
+    {  8,  9,  8,  9,  9,  9,  9,  9,  9,  9,  8,  8,  8,  8,  9,  8 },
+    {  9, 14, 13, 15, 13, 12, 13, 11, 12, 13, 12, 12, 12, 11, 11, 12 },
+    {  9, 11, 11, 12, 12, 11, 11, 13, 10, 11, 11, 13, 13, 13, 11, 12 },
+    { 10, 11, 11, 10, 10, 10, 11, 10,  9, 10,  9, 10,  9,  9,  9, 12 },
+    {  8, 10, 11, 13, 11, 11, 10, 10, 10,  9,  9,  8,  8,  8,  8,  8 },
+    { 11, 12, 11, 13, 11, 11, 10, 10,  9,  9,  9,  9,  9, 10, 10, 12 },
+    { 10, 14, 11, 15, 15, 12, 13, 12, 13, 11, 13, 11, 11, 10, 11, 11 },
+    { 10, 11, 13, 14, 14, 11, 13, 11, 12, 12, 11, 11, 11, 11, 10, 12 },
+    {  9, 11, 11, 12, 12, 12, 12, 11, 13, 13, 13, 11,  9,  9,  9,  9 },
+    { 10, 13, 11, 14, 14, 12, 15, 12, 12, 13, 11, 12, 12, 11, 11, 11 },
+    {  8, 14,  9,  9,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8,  8 },
+    {  8, 14, 14, 11, 13, 10, 13, 13, 11, 12, 12, 15, 15, 12, 12, 12 },
+    { 11, 11, 15, 11, 13, 12, 11, 11, 11, 10, 10, 11, 11, 11, 10, 11 },
+    {  8,  8,  9,  8,  8,  8, 10,  9, 10,  9,  9, 10, 10, 10,  9,  9 },
+    {  8, 11, 10, 13, 11, 11, 10, 11, 10,  9,  8,  8,  9,  8,  8,  9 },
+    { 11, 13, 13, 12, 15, 13, 11, 11, 10, 11, 10, 10,  9,  8,  9,  8 },
+    { 10, 11, 13, 11, 12, 11, 11, 11, 10,  9, 10, 14, 12,  8,  8,  8 }
+};
+
+const uint8_t ff_silk_lsf_pred_weights_nbmb[2][9] = {
+    {179, 138, 140, 148, 151, 149, 153, 151, 163},
+    {116,  67,  82,  59,  92,  72, 100,  89,  92}
+};
+
+const uint8_t ff_silk_lsf_pred_weights_wb[2][15] = {
+    {175, 148, 160, 176, 178, 173, 174, 164, 177, 174, 196, 182, 198, 192, 182},
+    { 68,  62,  66,  60,  72, 117,  85,  90, 118, 136, 151, 142, 160, 142, 155}
+};
+
+const uint8_t ff_silk_lsf_weight_sel_nbmb[32][9] = {
+    { 0, 1, 0, 0, 0, 0, 0, 0, 0 },
+    { 1, 0, 0, 0, 0, 0, 0, 0, 0 },
+    { 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+    { 1, 1, 1, 0, 0, 0, 0, 1, 0 },
+    { 0, 1, 0, 0, 0, 0, 0, 0, 0 },
+    { 0, 1, 0, 0, 0, 0, 0, 0, 0 },
+    { 1, 0, 1, 1, 0, 0, 0, 1, 0 },
+    { 0, 1, 1, 0, 0, 1, 1, 0, 0 },
+    { 0, 0, 1, 1, 0, 1, 0, 1, 1 },
+    { 0, 0, 1, 1, 0, 0, 1, 1, 1 },
+    { 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+    { 0, 1, 0, 1, 1, 1, 1, 1, 0 },
+    { 0, 1, 0, 1, 1, 1, 1, 1, 0 },
+    { 0, 1, 1, 1, 1, 1, 1, 1, 0 },
+    { 1, 0, 1, 1, 0, 1, 1, 1, 1 },
+    { 0, 1, 1, 1, 1, 1, 0, 1, 0 },
+    { 0, 0, 1, 1, 0, 1, 0, 1, 0 },
+    { 0, 0, 1, 1, 1, 0, 1, 1, 1 },
+    { 0, 1, 1, 0, 0, 1, 1, 1, 0 },
+    { 0, 0, 0, 1, 1, 1, 0, 1, 0 },
+    { 0, 1, 1, 0, 0, 1, 0, 1, 0 },
+    { 0, 1, 1, 0, 0, 0, 1, 1, 0 },
+    { 0, 0, 0, 0, 0, 1, 1, 1, 1 },
+    { 0, 0, 1, 1, 0, 0, 0, 1, 1 },
+    { 0, 0, 0, 1, 0, 1, 1, 1, 1 },
+    { 0, 1, 1, 1, 1, 1, 1, 1, 0 },
+    { 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+    { 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+    { 0, 0, 1, 0, 1, 1, 0, 1, 0 },
+    { 1, 0, 0, 1, 0, 0, 0, 0, 0 },
+    { 0, 0, 0, 1, 1, 0, 1, 0, 1 },
+    { 1, 0, 1, 1, 0, 1, 1, 1, 1 }
+};
+
+const uint8_t ff_silk_lsf_weight_sel_wb[32][15] = {
+    { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1 },
+    { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+    { 0, 0, 1, 0, 0, 1, 1, 1, 0, 1, 1, 1, 1, 0, 0 },
+    { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 0 },
+    { 0, 1, 1, 0, 1, 0, 1, 1, 0, 1, 1, 1, 1, 1, 0 },
+    { 0, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+    { 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 1, 0 },
+    { 0, 1, 1, 0, 0, 0, 1, 0, 1, 1, 1, 0, 1, 0, 1 },
+    { 0, 1, 0, 1, 1, 0, 1, 0, 1, 0, 1, 1, 1, 1, 1 },
+    { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1 },
+    { 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+    { 0, 0, 1, 0, 1, 1, 1, 1, 1, 1, 1, 0, 1, 0, 0 },
+    { 0, 0, 1, 0, 0, 1, 0, 1, 0, 1, 0, 0, 1, 0, 0 },
+    { 0, 0, 0, 0, 1, 1, 0, 1, 0, 1, 1, 1, 1, 0, 0 },
+    { 0, 1, 0, 0, 0, 1, 1, 0, 1, 1, 1, 0, 1, 1, 1 },
+    { 0, 0, 1, 1, 0, 0, 0, 0, 0, 0, 0, 0, 1, 1, 0 },
+    { 0, 1, 1, 0, 1, 0, 1, 1, 1, 1, 1, 0, 1, 0, 0 },
+    { 0, 0, 1, 0, 0, 0, 0, 1, 0, 0, 1, 1, 1, 0, 0 },
+    { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1 },
+    { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 0 },
+    { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
+    { 0, 1, 0, 1, 0, 1, 1, 0, 1, 0, 1, 0, 1, 1, 0 },
+    { 0, 0, 1, 1, 1, 1, 0, 1, 1, 0, 0, 1, 1, 0, 0 },
+    { 0, 1, 1, 0, 1, 0, 1, 0, 1, 0, 0, 0, 0, 1, 0 },
+    { 0, 0, 0, 1, 1, 0, 1, 0, 1, 1, 1, 1, 1, 1, 1 },
+    { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1 },
+    { 0, 1, 1, 0, 0, 0, 1, 1, 0, 0, 1, 1, 1, 1, 1 },
+    { 0, 0, 0, 0, 0, 1, 0, 1, 1, 1, 1, 0, 1, 1, 1 },
+    { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1 },
+    { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1 },
+    { 1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 0, 0, 0 },
+    { 0, 0, 1, 0, 0, 1, 1, 1, 0, 0, 1, 0, 0, 1, 0 }
+};
+
+const uint8_t ff_silk_lsf_codebook_nbmb[32][10] = {
+    { 12,  35,  60,  83, 108, 132, 157, 180, 206, 228 },
+    { 15,  32,  55,  77, 101, 125, 151, 175, 201, 225 },
+    { 19,  42,  66,  89, 114, 137, 162, 184, 209, 230 },
+    { 12,  25,  50,  72,  97, 120, 147, 172, 200, 223 },
+    { 26,  44,  69,  90, 114, 135, 159, 180, 205, 225 },
+    { 13,  22,  53,  80, 106, 130, 156, 180, 205, 228 },
+    { 15,  25,  44,  64,  90, 115, 142, 168, 196, 222 },
+    { 19,  24,  62,  82, 100, 120, 145, 168, 190, 214 },
+    { 22,  31,  50,  79, 103, 120, 151, 170, 203, 227 },
+    { 21,  29,  45,  65, 106, 124, 150, 171, 196, 224 },
+    { 30,  49,  75,  97, 121, 142, 165, 186, 209, 229 },
+    { 19,  25,  52,  70,  93, 116, 143, 166, 192, 219 },
+    { 26,  34,  62,  75,  97, 118, 145, 167, 194, 217 },
+    { 25,  33,  56,  70,  91, 113, 143, 165, 196, 223 },
+    { 21,  34,  51,  72,  97, 117, 145, 171, 196, 222 },
+    { 20,  29,  50,  67,  90, 117, 144, 168, 197, 221 },
+    { 22,  31,  48,  66,  95, 117, 146, 168, 196, 222 },
+    { 24,  33,  51,  77, 116, 134, 158, 180, 200, 224 },
+    { 21,  28,  70,  87, 106, 124, 149, 170, 194, 217 },
+    { 26,  33,  53,  64,  83, 117, 152, 173, 204, 225 },
+    { 27,  34,  65,  95, 108, 129, 155, 174, 210, 225 },
+    { 20,  26,  72,  99, 113, 131, 154, 176, 200, 219 },
+    { 34,  43,  61,  78,  93, 114, 155, 177, 205, 229 },
+    { 23,  29,  54,  97, 124, 138, 163, 179, 209, 229 },
+    { 30,  38,  56,  89, 118, 129, 158, 178, 200, 231 },
+    { 21,  29,  49,  63,  85, 111, 142, 163, 193, 222 },
+    { 27,  48,  77, 103, 133, 158, 179, 196, 215, 232 },
+    { 29,  47,  74,  99, 124, 151, 176, 198, 220, 237 },
+    { 33,  42,  61,  76,  93, 121, 155, 174, 207, 225 },
+    { 29,  53,  87, 112, 136, 154, 170, 188, 208, 227 },
+    { 24,  30,  52,  84, 131, 150, 166, 186, 203, 229 },
+    { 37,  48,  64,  84, 104, 118, 156, 177, 201, 230 }
+};
+
+const uint8_t ff_silk_lsf_codebook_wb[32][16] = {
+    {  7,  23,  38,  54,  69,  85, 100, 116, 131, 147, 162, 178, 193, 208, 223, 239 },
+    { 13,  25,  41,  55,  69,  83,  98, 112, 127, 142, 157, 171, 187, 203, 220, 236 },
+    { 15,  21,  34,  51,  61,  78,  92, 106, 126, 136, 152, 167, 185, 205, 225, 240 },
+    { 10,  21,  36,  50,  63,  79,  95, 110, 126, 141, 157, 173, 189, 205, 221, 237 },
+    { 17,  20,  37,  51,  59,  78,  89, 107, 123, 134, 150, 164, 184, 205, 224, 240 },
+    { 10,  15,  32,  51,  67,  81,  96, 112, 129, 142, 158, 173, 189, 204, 220, 236 },
+    {  8,  21,  37,  51,  65,  79,  98, 113, 126, 138, 155, 168, 179, 192, 209, 218 },
+    { 12,  15,  34,  55,  63,  78,  87, 108, 118, 131, 148, 167, 185, 203, 219, 236 },
+    { 16,  19,  32,  36,  56,  79,  91, 108, 118, 136, 154, 171, 186, 204, 220, 237 },
+    { 11,  28,  43,  58,  74,  89, 105, 120, 135, 150, 165, 180, 196, 211, 226, 241 },
+    {  6,  16,  33,  46,  60,  75,  92, 107, 123, 137, 156, 169, 185, 199, 214, 225 },
+    { 11,  19,  30,  44,  57,  74,  89, 105, 121, 135, 152, 169, 186, 202, 218, 234 },
+    { 12,  19,  29,  46,  57,  71,  88, 100, 120, 132, 148, 165, 182, 199, 216, 233 },
+    { 17,  23,  35,  46,  56,  77,  92, 106, 123, 134, 152, 167, 185, 204, 222, 237 },
+    { 14,  17,  45,  53,  63,  75,  89, 107, 115, 132, 151, 171, 188, 206, 221, 240 },
+    {  9,  16,  29,  40,  56,  71,  88, 103, 119, 137, 154, 171, 189, 205, 222, 237 },
+    { 16,  19,  36,  48,  57,  76,  87, 105, 118, 132, 150, 167, 185, 202, 218, 236 },
+    { 12,  17,  29,  54,  71,  81,  94, 104, 126, 136, 149, 164, 182, 201, 221, 237 },
+    { 15,  28,  47,  62,  79,  97, 115, 129, 142, 155, 168, 180, 194, 208, 223, 238 },
+    {  8,  14,  30,  45,  62,  78,  94, 111, 127, 143, 159, 175, 192, 207, 223, 239 },
+    { 17,  30,  49,  62,  79,  92, 107, 119, 132, 145, 160, 174, 190, 204, 220, 235 },
+    { 14,  19,  36,  45,  61,  76,  91, 108, 121, 138, 154, 172, 189, 205, 222, 238 },
+    { 12,  18,  31,  45,  60,  76,  91, 107, 123, 138, 154, 171, 187, 204, 221, 236 },
+    { 13,  17,  31,  43,  53,  70,  83, 103, 114, 131, 149, 167, 185, 203, 220, 237 },
+    { 17,  22,  35,  42,  58,  78,  93, 110, 125, 139, 155, 170, 188, 206, 224, 240 },
+    {  8,  15,  34,  50,  67,  83,  99, 115, 131, 146, 162, 178, 193, 209, 224, 239 },
+    { 13,  16,  41,  66,  73,  86,  95, 111, 128, 137, 150, 163, 183, 206, 225, 241 },
+    { 17,  25,  37,  52,  63,  75,  92, 102, 119, 132, 144, 160, 175, 191, 212, 231 },
+    { 19,  31,  49,  65,  83, 100, 117, 133, 147, 161, 174, 187, 200, 213, 227, 242 },
+    { 18,  31,  52,  68,  88, 103, 117, 126, 138, 149, 163, 177, 192, 207, 223, 239 },
+    { 16,  29,  47,  61,  76,  90, 106, 119, 133, 147, 161, 176, 193, 209, 224, 240 },
+    { 15,  21,  35,  50,  61,  73,  86,  97, 110, 119, 129, 141, 175, 198, 218, 237 }
+};
+
+const uint16_t ff_silk_lsf_min_spacing_nbmb[] = {
+    250, 3, 6, 3, 3, 3, 4, 3, 3, 3, 461
+};
+
+const uint16_t ff_silk_lsf_min_spacing_wb[] = {
+    100, 3, 40, 3, 3, 3, 5, 14, 14, 10, 11, 3, 8, 9, 7, 3, 347
+};
+
+const uint8_t ff_silk_lsf_ordering_nbmb[] = {
+    0, 9, 6, 3, 4, 5, 8, 1, 2, 7
+};
+
+const uint8_t ff_silk_lsf_ordering_wb[] = {
+    0, 15, 8, 7, 4, 11, 12, 3, 2, 13, 10, 5, 6, 9, 14, 1
+};
+
+const int16_t ff_silk_cosine[] = { /* (0.12) */
+     4096,  4095,  4091,  4085,
+     4076,  4065,  4052,  4036,
+     4017,  3997,  3973,  3948,
+     3920,  3889,  3857,  3822,
+     3784,  3745,  3703,  3659,
+     3613,  3564,  3513,  3461,
+     3406,  3349,  3290,  3229,
+     3166,  3102,  3035,  2967,
+     2896,  2824,  2751,  2676,
+     2599,  2520,  2440,  2359,
+     2276,  2191,  2106,  2019,
+     1931,  1842,  1751,  1660,
+     1568,  1474,  1380,  1285,
+     1189,  1093,   995,   897,
+      799,   700,   601,   501,
+      401,   301,   201,   101,
+        0,  -101,  -201,  -301,
+     -401,  -501,  -601,  -700,
+     -799,  -897,  -995, -1093,
+    -1189, -1285, -1380, -1474,
+    -1568, -1660, -1751, -1842,
+    -1931, -2019, -2106, -2191,
+    -2276, -2359, -2440, -2520,
+    -2599, -2676, -2751, -2824,
+    -2896, -2967, -3035, -3102,
+    -3166, -3229, -3290, -3349,
+    -3406, -3461, -3513, -3564,
+    -3613, -3659, -3703, -3745,
+    -3784, -3822, -3857, -3889,
+    -3920, -3948, -3973, -3997,
+    -4017, -4036, -4052, -4065,
+    -4076, -4085, -4091, -4095,
+    -4096
+};
+
+const uint16_t ff_silk_pitch_scale[]   = {  4,   6,   8};
+
+const uint16_t ff_silk_pitch_min_lag[] = { 16,  24,  32};
+
+const uint16_t ff_silk_pitch_max_lag[] = {144, 216, 288};
+
+const int8_t ff_silk_pitch_offset_nb10ms[3][2] = {
+    { 0,  0},
+    { 1,  0},
+    { 0,  1}
+};
+
+const int8_t ff_silk_pitch_offset_nb20ms[11][4] = {
+    { 0,  0,  0,  0},
+    { 2,  1,  0, -1},
+    {-1,  0,  1,  2},
+    {-1,  0,  0,  1},
+    {-1,  0,  0,  0},
+    { 0,  0,  0,  1},
+    { 0,  0,  1,  1},
+    { 1,  1,  0,  0},
+    { 1,  0,  0,  0},
+    { 0,  0,  0, -1},
+    { 1,  0,  0, -1}
+};
+
+const int8_t ff_silk_pitch_offset_mbwb10ms[12][2] = {
+    { 0,  0},
+    { 0,  1},
+    { 1,  0},
+    {-1,  1},
+    { 1, -1},
+    {-1,  2},
+    { 2, -1},
+    {-2,  2},
+    { 2, -2},
+    {-2,  3},
+    { 3, -2},
+    {-3,  3}
+};
+
+const int8_t ff_silk_pitch_offset_mbwb20ms[34][4] = {
+    { 0,  0,  0,  0},
+    { 0,  0,  1,  1},
+    { 1,  1,  0,  0},
+    {-1,  0,  0,  0},
+    { 0,  0,  0,  1},
+    { 1,  0,  0,  0},
+    {-1,  0,  0,  1},
+    { 0,  0,  0, -1},
+    {-1,  0,  1,  2},
+    { 1,  0,  0, -1},
+    {-2, -1,  1,  2},
+    { 2,  1,  0, -1},
+    {-2,  0,  0,  2},
+    {-2,  0,  1,  3},
+    { 2,  1, -1, -2},
+    {-3, -1,  1,  3},
+    { 2,  0,  0, -2},
+    { 3,  1,  0, -2},
+    {-3, -1,  2,  4},
+    {-4, -1,  1,  4},
+    { 3,  1, -1, -3},
+    {-4, -1,  2,  5},
+    { 4,  2, -1, -3},
+    { 4,  1, -1, -4},
+    {-5, -1,  2,  6},
+    { 5,  2, -1, -4},
+    {-6, -2,  2,  6},
+    {-5, -2,  2,  5},
+    { 6,  2, -1, -5},
+    {-7, -2,  3,  8},
+    { 6,  2, -2, -6},
+    { 5,  2, -2, -5},
+    { 8,  3, -2, -7},
+    {-9, -3,  3,  9}
+};
+
+const int8_t ff_silk_ltp_filter0_taps[8][5] = {
+    {  4,   6,  24,   7,   5},
+    {  0,   0,   2,   0,   0},
+    { 12,  28,  41,  13,  -4},
+    { -9,  15,  42,  25,  14},
+    {  1,  -2,  62,  41,  -9},
+    {-10,  37,  65,  -4,   3},
+    { -6,   4,  66,   7,  -8},
+    { 16,  14,  38,  -3,  33}
+};
+
+const int8_t ff_silk_ltp_filter1_taps[16][5] = {
+    { 13,  22,  39,  23,  12},
+    { -1,  36,  64,  27,  -6},
+    { -7,  10,  55,  43,  17},
+    {  1,   1,   8,   1,   1},
+    {  6, -11,  74,  53,  -9},
+    {-12,  55,  76, -12,   8},
+    { -3,   3,  93,  27,  -4},
+    { 26,  39,  59,   3,  -8},
+    {  2,   0,  77,  11,   9},
+    { -8,  22,  44,  -6,   7},
+    { 40,   9,  26,   3,   9},
+    { -7,  20, 101,  -7,   4},
+    {  3,  -8,  42,  26,   0},
+    {-15,  33,  68,   2,  23},
+    { -2,  55,  46,  -2,  15},
+    {  3,  -1,  21,  16,  41}
+};
+
+const int8_t ff_silk_ltp_filter2_taps[32][5] = {
+    { -6,  27,  61,  39,   5},
+    {-11,  42,  88,   4,   1},
+    { -2,  60,  65,   6,  -4},
+    { -1,  -5,  73,  56,   1},
+    { -9,  19,  94,  29,  -9},
+    {  0,  12,  99,   6,   4},
+    {  8, -19, 102,  46, -13},
+    {  3,   2,  13,   3,   2},
+    {  9, -21,  84,  72, -18},
+    {-11,  46, 104, -22,   8},
+    { 18,  38,  48,  23,   0},
+    {-16,  70,  83, -21,  11},
+    {  5, -11, 117,  22,  -8},
+    { -6,  23, 117, -12,   3},
+    {  3,  -8,  95,  28,   4},
+    {-10,  15,  77,  60, -15},
+    { -1,   4, 124,   2,  -4},
+    {  3,  38,  84,  24, -25},
+    {  2,  13,  42,  13,  31},
+    { 21,  -4,  56,  46,  -1},
+    { -1,  35,  79, -13,  19},
+    { -7,  65,  88,  -9, -14},
+    { 20,   4,  81,  49, -29},
+    { 20,   0,  75,   3, -17},
+    {  5,  -9,  44,  92,  -8},
+    {  1,  -3,  22,  69,  31},
+    { -6,  95,  41, -12,   5},
+    { 39,  67,  16,  -4,   1},
+    {  0,  -6, 120,  55, -36},
+    {-13,  44, 122,   4, -24},
+    { 81,   5,  11,   3,   7},
+    {  2,   0,   9,  10,  88}
+};
+
+const uint16_t ff_silk_ltp_scale_factor[] = {15565, 12288, 8192};
+
+const uint8_t ff_silk_shell_blocks[3][2] = {
+    { 5, 10}, // NB
+    { 8, 15}, // MB
+    {10, 20}  // WB
+};
+
+const uint8_t ff_silk_quant_offset[2][2] = { /* (0.23) */
+    {25, 60}, // Inactive or Unvoiced
+    { 8, 25}  // Voiced
+};
+
+const int ff_silk_stereo_interp_len[3] = {
+    64, 96, 128
+};
+
+const uint16_t ff_celt_model_tapset[] = { 4, 2, 3, 4 };
+
+const uint16_t ff_celt_model_spread[] = { 32, 7, 9, 30, 32 };
+
+const uint16_t ff_celt_model_alloc_trim[] = {
+    128,   2,   4,   9,  19,  41,  87, 109, 119, 124, 126, 128
+};
+
+const uint16_t ff_celt_model_energy_small[] = { 4, 2, 3, 4 };
+
+const uint8_t ff_celt_freq_bands[] = { /* in steps of 200Hz */
+    0,  1,  2,  3,  4,  5,  6,  7,  8, 10, 12, 14, 16, 20, 24, 28, 34, 40, 48, 60, 78, 100
+};
+
+const uint8_t ff_celt_freq_range[] = {
+    1,  1,  1,  1,  1,  1,  1,  1,  2,  2,  2,  2,  4,  4,  4,  6,  6,  8, 12, 18, 22
+};
+
+const uint8_t ff_celt_log_freq_range[] = {
+    0,  0,  0,  0,  0,  0,  0,  0,  8,  8,  8,  8, 16, 16, 16, 21, 21, 24, 29, 34, 36
+};
+
+/* Positive - increased freqeuency resolution (only possible on transients)
+ * Negative - increased time resolution */
+const int8_t ff_celt_tf_select[4][2][2][2] = {
+    /*          OFF                        ON                Transient frame */
+    /*     OFF        ON             OFF        ON           TF select flag  */
+    /*   OFF  ON    OFF  ON        OFF  ON    OFF  ON        TF change flag  */
+    { { { 0, -1 }, { 0, -1 } }, { { 0, -1 }, { 0, -1 } } }, /* 120 */
+    { { { 0, -1 }, { 0, -2 } }, { { 1,  0 }, { 1, -1 } } }, /* 240 */
+    { { { 0, -2 }, { 0, -3 } }, { { 2,  0 }, { 1, -1 } } }, /* 480 */
+    { { { 0, -2 }, { 0, -3 } }, { { 3,  0 }, { 1, -1 } } }  /* 960 */
+};
+
+const float ff_celt_mean_energy[] = {
+    6.437500f, 6.250000f, 5.750000f, 5.312500f, 5.062500f,
+    4.812500f, 4.500000f, 4.375000f, 4.875000f, 4.687500f,
+    4.562500f, 4.437500f, 4.875000f, 4.625000f, 4.312500f,
+    4.500000f, 4.375000f, 4.625000f, 4.750000f, 4.437500f,
+    3.750000f, 3.750000f, 3.750000f, 3.750000f, 3.750000f
+};
+
+const float ff_celt_alpha_coef[] = {
+    29440.0f/32768.0f,    26112.0f/32768.0f,    21248.0f/32768.0f,    16384.0f/32768.0f
+};
+
+const float ff_celt_beta_coef[] = {
+    1.0f - (30147.0f/32768.0f), 1.0f - (22282.0f/32768.0f), 1.0f - (12124.0f/32768.0f), 1.0f - (6554.0f/32768.0f),
+};
+
+const uint8_t ff_celt_coarse_energy_dist[4][2][42] = {
+    {
+        {       // 120-sample inter
+             72, 127,  65, 129,  66, 128,  65, 128,  64, 128,  62, 128,  64, 128,
+             64, 128,  92,  78,  92,  79,  92,  78,  90,  79, 116,  41, 115,  40,
+            114,  40, 132,  26, 132,  26, 145,  17, 161,  12, 176,  10, 177,  11
+        }, {    // 120-sample intra
+             24, 179,  48, 138,  54, 135,  54, 132,  53, 134,  56, 133,  55, 132,
+             55, 132,  61, 114,  70,  96,  74,  88,  75,  88,  87,  74,  89,  66,
+             91,  67, 100,  59, 108,  50, 120,  40, 122,  37,  97,  43,  78,  50
+        }
+    }, {
+        {       // 240-sample inter
+             83,  78,  84,  81,  88,  75,  86,  74,  87,  71,  90,  73,  93,  74,
+             93,  74, 109,  40, 114,  36, 117,  34, 117,  34, 143,  17, 145,  18,
+            146,  19, 162,  12, 165,  10, 178,   7, 189,   6, 190,   8, 177,   9
+        }, {    // 240-sample intra
+             23, 178,  54, 115,  63, 102,  66,  98,  69,  99,  74,  89,  71,  91,
+             73,  91,  78,  89,  86,  80,  92,  66,  93,  64, 102,  59, 103,  60,
+            104,  60, 117,  52, 123,  44, 138,  35, 133,  31,  97,  38,  77,  45
+        }
+    }, {
+        {       // 480-sample inter
+             61,  90,  93,  60, 105,  42, 107,  41, 110,  45, 116,  38, 113,  38,
+            112,  38, 124,  26, 132,  27, 136,  19, 140,  20, 155,  14, 159,  16,
+            158,  18, 170,  13, 177,  10, 187,   8, 192,   6, 175,   9, 159,  10
+        }, {    // 480-sample intra
+             21, 178,  59, 110,  71,  86,  75,  85,  84,  83,  91,  66,  88,  73,
+             87,  72,  92,  75,  98,  72, 105,  58, 107,  54, 115,  52, 114,  55,
+            112,  56, 129,  51, 132,  40, 150,  33, 140,  29,  98,  35,  77,  42
+        }
+    }, {
+        {       // 960-sample inter
+             42, 121,  96,  66, 108,  43, 111,  40, 117,  44, 123,  32, 120,  36,
+            119,  33, 127,  33, 134,  34, 139,  21, 147,  23, 152,  20, 158,  25,
+            154,  26, 166,  21, 173,  16, 184,  13, 184,  10, 150,  13, 139,  15
+        }, {    // 960-sample intra
+             22, 178,  63, 114,  74,  82,  84,  83,  92,  82, 103,  62,  96,  72,
+             96,  67, 101,  73, 107,  72, 113,  55, 118,  52, 125,  52, 118,  52,
+            117,  55, 135,  49, 137,  39, 157,  32, 145,  29,  97,  33,  77,  40
+        }
+    }
+};
+
+const uint8_t ff_celt_static_alloc[11][21] = {  /* 1/32 bit/sample */
+    {   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0,   0 },
+    {  90,  80,  75,  69,  63,  56,  49,  40,  34,  29,  20,  18,  10,   0,   0,   0,   0,   0,   0,   0,   0 },
+    { 110, 100,  90,  84,  78,  71,  65,  58,  51,  45,  39,  32,  26,  20,  12,   0,   0,   0,   0,   0,   0 },
+    { 118, 110, 103,  93,  86,  80,  75,  70,  65,  59,  53,  47,  40,  31,  23,  15,   4,   0,   0,   0,   0 },
+    { 126, 119, 112, 104,  95,  89,  83,  78,  72,  66,  60,  54,  47,  39,  32,  25,  17,  12,   1,   0,   0 },
+    { 134, 127, 120, 114, 103,  97,  91,  85,  78,  72,  66,  60,  54,  47,  41,  35,  29,  23,  16,  10,   1 },
+    { 144, 137, 130, 124, 113, 107, 101,  95,  88,  82,  76,  70,  64,  57,  51,  45,  39,  33,  26,  15,   1 },
+    { 152, 145, 138, 132, 123, 117, 111, 105,  98,  92,  86,  80,  74,  67,  61,  55,  49,  43,  36,  20,   1 },
+    { 162, 155, 148, 142, 133, 127, 121, 115, 108, 102,  96,  90,  84,  77,  71,  65,  59,  53,  46,  30,   1 },
+    { 172, 165, 158, 152, 143, 137, 131, 125, 118, 112, 106, 100,  94,  87,  81,  75,  69,  63,  56,  45,  20 },
+    { 200, 200, 200, 200, 200, 200, 200, 200, 198, 193, 188, 183, 178, 173, 168, 163, 158, 153, 148, 129, 104 }
+};
+
+const uint8_t ff_celt_static_caps[4][2][21] = {
+    {       // 120-sample
+        {224, 224, 224, 224, 224, 224, 224, 224, 160, 160,
+         160, 160, 185, 185, 185, 178, 178, 168, 134,  61,  37},
+        {224, 224, 224, 224, 224, 224, 224, 224, 240, 240,
+         240, 240, 207, 207, 207, 198, 198, 183, 144,  66,  40},
+    }, {    // 240-sample
+        {160, 160, 160, 160, 160, 160, 160, 160, 185, 185,
+         185, 185, 193, 193, 193, 183, 183, 172, 138,  64,  38},
+        {240, 240, 240, 240, 240, 240, 240, 240, 207, 207,
+         207, 207, 204, 204, 204, 193, 193, 180, 143,  66,  40},
+    }, {    // 480-sample
+        {185, 185, 185, 185, 185, 185, 185, 185, 193, 193,
+         193, 193, 193, 193, 193, 183, 183, 172, 138,  65,  39},
+        {207, 207, 207, 207, 207, 207, 207, 207, 204, 204,
+         204, 204, 201, 201, 201, 188, 188, 176, 141,  66,  40},
+    }, {    // 960-sample
+        {193, 193, 193, 193, 193, 193, 193, 193, 193, 193,
+         193, 193, 194, 194, 194, 184, 184, 173, 139,  65,  39},
+        {204, 204, 204, 204, 204, 204, 204, 204, 201, 201,
+         201, 201, 198, 198, 198, 187, 187, 175, 140,  66,  40}
+    }
+};
+
+const uint8_t ff_celt_cache_bits[392] = {
+    40, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
+    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7,
+    7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 7, 40, 15, 23, 28,
+    31, 34, 36, 38, 39, 41, 42, 43, 44, 45, 46, 47, 47, 49, 50,
+    51, 52, 53, 54, 55, 55, 57, 58, 59, 60, 61, 62, 63, 63, 65,
+    66, 67, 68, 69, 70, 71, 71, 40, 20, 33, 41, 48, 53, 57, 61,
+    64, 66, 69, 71, 73, 75, 76, 78, 80, 82, 85, 87, 89, 91, 92,
+    94, 96, 98, 101, 103, 105, 107, 108, 110, 112, 114, 117, 119, 121, 123,
+    124, 126, 128, 40, 23, 39, 51, 60, 67, 73, 79, 83, 87, 91, 94,
+    97, 100, 102, 105, 107, 111, 115, 118, 121, 124, 126, 129, 131, 135, 139,
+    142, 145, 148, 150, 153, 155, 159, 163, 166, 169, 172, 174, 177, 179, 35,
+    28, 49, 65, 78, 89, 99, 107, 114, 120, 126, 132, 136, 141, 145, 149,
+    153, 159, 165, 171, 176, 180, 185, 189, 192, 199, 205, 211, 216, 220, 225,
+    229, 232, 239, 245, 251, 21, 33, 58, 79, 97, 112, 125, 137, 148, 157,
+    166, 174, 182, 189, 195, 201, 207, 217, 227, 235, 243, 251, 17, 35, 63,
+    86, 106, 123, 139, 152, 165, 177, 187, 197, 206, 214, 222, 230, 237, 250,
+    25, 31, 55, 75, 91, 105, 117, 128, 138, 146, 154, 161, 168, 174, 180,
+    185, 190, 200, 208, 215, 222, 229, 235, 240, 245, 255, 16, 36, 65, 89,
+    110, 128, 144, 159, 173, 185, 196, 207, 217, 226, 234, 242, 250, 11, 41,
+    74, 103, 128, 151, 172, 191, 209, 225, 241, 255, 9, 43, 79, 110, 138,
+    163, 186, 207, 227, 246, 12, 39, 71, 99, 123, 144, 164, 182, 198, 214,
+    228, 241, 253, 9, 44, 81, 113, 142, 168, 192, 214, 235, 255, 7, 49,
+    90, 127, 160, 191, 220, 247, 6, 51, 95, 134, 170, 203, 234, 7, 47,
+    87, 123, 155, 184, 212, 237, 6, 52, 97, 137, 174, 208, 240, 5, 57,
+    106, 151, 192, 231, 5, 59, 111, 158, 202, 243, 5, 55, 103, 147, 187,
+    224, 5, 60, 113, 161, 206, 248, 4, 65, 122, 175, 224, 4, 67, 127,
+    182, 234
+};
+
+const int16_t ff_celt_cache_index[105] = {
+    -1, -1, -1, -1, -1, -1, -1, -1, 0, 0, 0, 0, 41, 41, 41,
+    82, 82, 123, 164, 200, 222, 0, 0, 0, 0, 0, 0, 0, 0, 41,
+    41, 41, 41, 123, 123, 123, 164, 164, 240, 266, 283, 295, 41, 41, 41,
+    41, 41, 41, 41, 41, 123, 123, 123, 123, 240, 240, 240, 266, 266, 305,
+    318, 328, 336, 123, 123, 123, 123, 123, 123, 123, 123, 240, 240, 240, 240,
+    305, 305, 305, 318, 318, 343, 351, 358, 364, 240, 240, 240, 240, 240, 240,
+    240, 240, 305, 305, 305, 305, 343, 343, 343, 351, 351, 370, 376, 382, 387,
+};
+
+const uint8_t ff_celt_log2_frac[] = {
+    0, 8, 13, 16, 19, 21, 23, 24, 26, 27, 28, 29, 30, 31, 32, 32, 33, 34, 34, 35, 36, 36, 37, 37
+};
+
+const uint8_t ff_celt_bit_interleave[] = {
+    0, 1, 1, 1, 2, 3, 3, 3, 2, 3, 3, 3, 2, 3, 3, 3
+};
+
+const uint8_t ff_celt_bit_deinterleave[] = {
+    0x00, 0x03, 0x0C, 0x0F, 0x30, 0x33, 0x3C, 0x3F,
+    0xC0, 0xC3, 0xCC, 0xCF, 0xF0, 0xF3, 0xFC, 0xFF
+};
+
+const uint8_t ff_celt_hadamard_order[] = {
+    1,   0,
+    3,   0,  2,  1,
+    7,   0,  4,  3,  6,  1,  5,  2,
+    15,  0,  8,  7, 12,  3, 11,  4, 14,  1,  9,  6, 13,  2, 10,  5,
+    0,   1,  2,  3,  4,  5,  6,  7,  8,  9, 10, 11, 12, 13, 14, 15
+};
+
+const uint16_t ff_celt_qn_exp2[] = {
+    16384, 17866, 19483, 21247, 23170, 25267, 27554, 30048
+};
+
+static const uint32_t celt_pvq_u[1272] = {
+    /* N = 0, K = 0...176 */
+    1, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+    0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
+    /* N = 1, K = 1...176 */
+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,
+    /* N = 2, K = 2...176 */
+    3, 5, 7, 9, 11, 13, 15, 17, 19, 21, 23, 25, 27, 29, 31, 33, 35, 37, 39, 41,
+    43, 45, 47, 49, 51, 53, 55, 57, 59, 61, 63, 65, 67, 69, 71, 73, 75, 77, 79,
+    81, 83, 85, 87, 89, 91, 93, 95, 97, 99, 101, 103, 105, 107, 109, 111, 113,
+    115, 117, 119, 121, 123, 125, 127, 129, 131, 133, 135, 137, 139, 141, 143,
+    145, 147, 149, 151, 153, 155, 157, 159, 161, 163, 165, 167, 169, 171, 173,
+    175, 177, 179, 181, 183, 185, 187, 189, 191, 193, 195, 197, 199, 201, 203,
+    205, 207, 209, 211, 213, 215, 217, 219, 221, 223, 225, 227, 229, 231, 233,
+    235, 237, 239, 241, 243, 245, 247, 249, 251, 253, 255, 257, 259, 261, 263,
+    265, 267, 269, 271, 273, 275, 277, 279, 281, 283, 285, 287, 289, 291, 293,
+    295, 297, 299, 301, 303, 305, 307, 309, 311, 313, 315, 317, 319, 321, 323,
+    325, 327, 329, 331, 333, 335, 337, 339, 341, 343, 345, 347, 349, 351,
+    /* N = 3, K = 3...176 */
+    13, 25, 41, 61, 85, 113, 145, 181, 221, 265, 313, 365, 421, 481, 545, 613,
+    685, 761, 841, 925, 1013, 1105, 1201, 1301, 1405, 1513, 1625, 1741, 1861,
+    1985, 2113, 2245, 2381, 2521, 2665, 2813, 2965, 3121, 3281, 3445, 3613, 3785,
+    3961, 4141, 4325, 4513, 4705, 4901, 5101, 5305, 5513, 5725, 5941, 6161, 6385,
+    6613, 6845, 7081, 7321, 7565, 7813, 8065, 8321, 8581, 8845, 9113, 9385, 9661,
+    9941, 10225, 10513, 10805, 11101, 11401, 11705, 12013, 12325, 12641, 12961,
+    13285, 13613, 13945, 14281, 14621, 14965, 15313, 15665, 16021, 16381, 16745,
+    17113, 17485, 17861, 18241, 18625, 19013, 19405, 19801, 20201, 20605, 21013,
+    21425, 21841, 22261, 22685, 23113, 23545, 23981, 24421, 24865, 25313, 25765,
+    26221, 26681, 27145, 27613, 28085, 28561, 29041, 29525, 30013, 30505, 31001,
+    31501, 32005, 32513, 33025, 33541, 34061, 34585, 35113, 35645, 36181, 36721,
+    37265, 37813, 38365, 38921, 39481, 40045, 40613, 41185, 41761, 42341, 42925,
+    43513, 44105, 44701, 45301, 45905, 46513, 47125, 47741, 48361, 48985, 49613,
+    50245, 50881, 51521, 52165, 52813, 53465, 54121, 54781, 55445, 56113, 56785,
+    57461, 58141, 58825, 59513, 60205, 60901, 61601,
+    /* N = 4, K = 4...176 */
+    63, 129, 231, 377, 575, 833, 1159, 1561, 2047, 2625, 3303, 4089, 4991, 6017,
+    7175, 8473, 9919, 11521, 13287, 15225, 17343, 19649, 22151, 24857, 27775,
+    30913, 34279, 37881, 41727, 45825, 50183, 54809, 59711, 64897, 70375, 76153,
+    82239, 88641, 95367, 102425, 109823, 117569, 125671, 134137, 142975, 152193,
+    161799, 171801, 182207, 193025, 204263, 215929, 228031, 240577, 253575,
+    267033, 280959, 295361, 310247, 325625, 341503, 357889, 374791, 392217,
+    410175, 428673, 447719, 467321, 487487, 508225, 529543, 551449, 573951,
+    597057, 620775, 645113, 670079, 695681, 721927, 748825, 776383, 804609,
+    833511, 863097, 893375, 924353, 956039, 988441, 1021567, 1055425, 1090023,
+    1125369, 1161471, 1198337, 1235975, 1274393, 1313599, 1353601, 1394407,
+    1436025, 1478463, 1521729, 1565831, 1610777, 1656575, 1703233, 1750759,
+    1799161, 1848447, 1898625, 1949703, 2001689, 2054591, 2108417, 2163175,
+    2218873, 2275519, 2333121, 2391687, 2451225, 2511743, 2573249, 2635751,
+    2699257, 2763775, 2829313, 2895879, 2963481, 3032127, 3101825, 3172583,
+    3244409, 3317311, 3391297, 3466375, 3542553, 3619839, 3698241, 3777767,
+    3858425, 3940223, 4023169, 4107271, 4192537, 4278975, 4366593, 4455399,
+    4545401, 4636607, 4729025, 4822663, 4917529, 5013631, 5110977, 5209575,
+    5309433, 5410559, 5512961, 5616647, 5721625, 5827903, 5935489, 6044391,
+    6154617, 6266175, 6379073, 6493319, 6608921, 6725887, 6844225, 6963943,
+    7085049, 7207551,
+    /* N = 5, K = 5...176 */
+    321, 681, 1289, 2241, 3649, 5641, 8361, 11969, 16641, 22569, 29961, 39041,
+    50049, 63241, 78889, 97281, 118721, 143529, 172041, 204609, 241601, 283401,
+    330409, 383041, 441729, 506921, 579081, 658689, 746241, 842249, 947241,
+    1061761, 1186369, 1321641, 1468169, 1626561, 1797441, 1981449, 2179241,
+    2391489, 2618881, 2862121, 3121929, 3399041, 3694209, 4008201, 4341801,
+    4695809, 5071041, 5468329, 5888521, 6332481, 6801089, 7295241, 7815849,
+    8363841, 8940161, 9545769, 10181641, 10848769, 11548161, 12280841, 13047849,
+    13850241, 14689089, 15565481, 16480521, 17435329, 18431041, 19468809,
+    20549801, 21675201, 22846209, 24064041, 25329929, 26645121, 28010881,
+    29428489, 30899241, 32424449, 34005441, 35643561, 37340169, 39096641,
+    40914369, 42794761, 44739241, 46749249, 48826241, 50971689, 53187081,
+    55473921, 57833729, 60268041, 62778409, 65366401, 68033601, 70781609,
+    73612041, 76526529, 79526721, 82614281, 85790889, 89058241, 92418049,
+    95872041, 99421961, 103069569, 106816641, 110664969, 114616361, 118672641,
+    122835649, 127107241, 131489289, 135983681, 140592321, 145317129, 150160041,
+    155123009, 160208001, 165417001, 170752009, 176215041, 181808129, 187533321,
+    193392681, 199388289, 205522241, 211796649, 218213641, 224775361, 231483969,
+    238341641, 245350569, 252512961, 259831041, 267307049, 274943241, 282741889,
+    290705281, 298835721, 307135529, 315607041, 324252609, 333074601, 342075401,
+    351257409, 360623041, 370174729, 379914921, 389846081, 399970689, 410291241,
+    420810249, 431530241, 442453761, 453583369, 464921641, 476471169, 488234561,
+    500214441, 512413449, 524834241, 537479489, 550351881, 563454121, 576788929,
+    590359041, 604167209, 618216201, 632508801,
+    /* N = 6, K = 6...96 (technically V(109,5) fits in 32 bits, but that can't be
+     achieved by splitting an Opus band) */
+    1683, 3653, 7183, 13073, 22363, 36365, 56695, 85305, 124515, 177045, 246047,
+    335137, 448427, 590557, 766727, 982729, 1244979, 1560549, 1937199, 2383409,
+    2908411, 3522221, 4235671, 5060441, 6009091, 7095093, 8332863, 9737793,
+    11326283, 13115773, 15124775, 17372905, 19880915, 22670725, 25765455,
+    29189457, 32968347, 37129037, 41699767, 46710137, 52191139, 58175189,
+    64696159, 71789409, 79491819, 87841821, 96879431, 106646281, 117185651,
+    128542501, 140763503, 153897073, 167993403, 183104493, 199284183, 216588185,
+    235074115, 254801525, 275831935, 298228865, 322057867, 347386557, 374284647,
+    402823977, 433078547, 465124549, 499040399, 534906769, 572806619, 612825229,
+    655050231, 699571641, 746481891, 795875861, 847850911, 902506913, 959946283,
+    1020274013, 1083597703, 1150027593, 1219676595, 1292660325, 1369097135,
+    1449108145, 1532817275, 1620351277, 1711839767, 1807415257, 1907213187,
+    2011371957, 2120032959,
+    /* N = 7, K = 7...54 (technically V(60,6) fits in 32 bits, but that can't be
+     achieved by splitting an Opus band) */
+    8989, 19825, 40081, 75517, 134245, 227305, 369305, 579125, 880685, 1303777,
+    1884961, 2668525, 3707509, 5064793, 6814249, 9041957, 11847485, 15345233,
+    19665841, 24957661, 31388293, 39146185, 48442297, 59511829, 72616013,
+    88043969, 106114625, 127178701, 151620757, 179861305, 212358985, 249612805,
+    292164445, 340600625, 395555537, 457713341, 527810725, 606639529, 695049433,
+    793950709, 904317037, 1027188385, 1163673953, 1314955181, 1482288821,
+    1667010073, 1870535785, 2094367717,
+    /* N = 8, K = 8...37 (technically V(40,7) fits in 32 bits, but that can't be
+     achieved by splitting an Opus band) */
+    48639, 108545, 224143, 433905, 795455, 1392065, 2340495, 3800305, 5984767,
+    9173505, 13726991, 20103025, 28875327, 40754369, 56610575, 77500017,
+    104692735, 139703809, 184327311, 240673265, 311207743, 398796225, 506750351,
+    638878193, 799538175, 993696769, 1226990095, 1505789553, 1837271615,
+    2229491905,
+    /* N = 9, K = 9...28 (technically V(29,8) fits in 32 bits, but that can't be
+     achieved by splitting an Opus band) */
+    265729, 598417, 1256465, 2485825, 4673345, 8405905, 14546705, 24331777,
+    39490049, 62390545, 96220561, 145198913, 214828609, 312193553, 446304145,
+    628496897, 872893441, 1196924561, 1621925137, 2173806145,
+    /* N = 10, K = 10...24 */
+    1462563, 3317445, 7059735, 14218905, 27298155, 50250765, 89129247, 152951073,
+    254831667, 413442773, 654862247, 1014889769, 1541911931, 2300409629,
+    3375210671,
+    /* N = 11, K = 11...19 (technically V(20,10) fits in 32 bits, but that can't be
+     achieved by splitting an Opus band) */
+    8097453, 18474633, 39753273, 81270333, 158819253, 298199265, 540279585,
+    948062325, 1616336765,
+    /* N = 12, K = 12...18 */
+    45046719, 103274625, 224298231, 464387817, 921406335, 1759885185,
+    3248227095,
+    /* N = 13, K = 13...16 */
+    251595969, 579168825, 1267854873, 2653649025,
+    /* N = 14, K = 14 */
+    1409933619
+};
+
+const float ff_celt_postfilter_taps[3][3] = {
+    { 0.3066406250f, 0.2170410156f, 0.1296386719f },
+    { 0.4638671875f, 0.2680664062f, 0.0           },
+    { 0.7998046875f, 0.1000976562f, 0.0           }
+};
+
+DECLARE_ALIGNED(32, const float, ff_celt_window_padded)[136] = {
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+    0.00000000f, 0.00000000f, 0.00000000f, 0.00000000f,
+    6.7286966e-05f, 0.00060551348f, 0.0016815970f, 0.0032947962f, 0.0054439943f,
+    0.0081276923f, 0.011344001f, 0.015090633f, 0.019364886f, 0.024163635f,
+    0.029483315f, 0.035319905f, 0.041668911f, 0.048525347f, 0.055883718f,
+    0.063737999f, 0.072081616f, 0.080907428f, 0.090207705f, 0.099974111f,
+    0.11019769f, 0.12086883f, 0.13197729f, 0.14351214f, 0.15546177f,
+    0.16781389f, 0.18055550f, 0.19367290f, 0.20715171f, 0.22097682f,
+    0.23513243f, 0.24960208f, 0.26436860f, 0.27941419f, 0.29472040f,
+    0.31026818f, 0.32603788f, 0.34200931f, 0.35816177f, 0.37447407f,
+    0.39092462f, 0.40749142f, 0.42415215f, 0.44088423f, 0.45766484f,
+    0.47447104f, 0.49127978f, 0.50806798f, 0.52481261f, 0.54149077f,
+    0.55807973f, 0.57455701f, 0.59090049f, 0.60708841f, 0.62309951f,
+    0.63891306f, 0.65450896f, 0.66986776f, 0.68497077f, 0.69980010f,
+    0.71433873f, 0.72857055f, 0.74248043f, 0.75605424f, 0.76927895f,
+    0.78214257f, 0.79463430f, 0.80674445f, 0.81846456f, 0.82978733f,
+    0.84070669f, 0.85121779f, 0.86131698f, 0.87100183f, 0.88027111f,
+    0.88912479f, 0.89756398f, 0.90559094f, 0.91320904f, 0.92042270f,
+    0.92723738f, 0.93365955f, 0.93969656f, 0.94535671f, 0.95064907f,
+    0.95558353f, 0.96017067f, 0.96442171f, 0.96834849f, 0.97196334f,
+    0.97527906f, 0.97830883f, 0.98106616f, 0.98356480f, 0.98581869f,
+    0.98784191f, 0.98964856f, 0.99125274f, 0.99266849f, 0.99390969f,
+    0.99499004f, 0.99592297f, 0.99672162f, 0.99739874f, 0.99796667f,
+    0.99843728f, 0.99882195f, 0.99913147f, 0.99937606f, 0.99956527f,
+    0.99970802f, 0.99981248f, 0.99988613f, 0.99993565f, 0.99996697f,
+    0.99998518f, 0.99999457f, 0.99999859f, 0.99999982f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f, 1.00000000f,
+    1.00000000f, 1.00000000f, 1.00000000f,
+};
+
+/* square of the window, used for the postfilter */
+const float ff_celt_window2[120] = {
+    4.5275357e-09f, 3.66647e-07f, 2.82777e-06f, 1.08557e-05f, 2.96371e-05f, 6.60594e-05f,
+    0.000128686f, 0.000227727f, 0.000374999f, 0.000583881f, 0.000869266f, 0.0012475f,
+    0.0017363f, 0.00235471f, 0.00312299f, 0.00406253f, 0.00519576f, 0.00654601f,
+    0.00813743f, 0.00999482f, 0.0121435f, 0.0146093f, 0.017418f, 0.0205957f, 0.0241684f,
+    0.0281615f, 0.0326003f, 0.0375092f, 0.0429118f, 0.0488308f, 0.0552873f, 0.0623012f,
+    0.0698908f, 0.0780723f, 0.0868601f, 0.0962664f, 0.106301f, 0.11697f, 0.12828f,
+    0.140231f, 0.152822f, 0.166049f, 0.179905f, 0.194379f, 0.209457f, 0.225123f, 0.241356f,
+    0.258133f, 0.275428f, 0.293212f, 0.311453f, 0.330116f, 0.349163f, 0.368556f, 0.388253f,
+    0.40821f, 0.428382f, 0.448723f, 0.469185f, 0.48972f, 0.51028f, 0.530815f, 0.551277f,
+    0.571618f, 0.59179f, 0.611747f, 0.631444f, 0.650837f, 0.669884f, 0.688547f, 0.706788f,
+    0.724572f, 0.741867f, 0.758644f, 0.774877f, 0.790543f, 0.805621f, 0.820095f, 0.833951f,
+    0.847178f, 0.859769f, 0.87172f, 0.88303f, 0.893699f, 0.903734f, 0.91314f, 0.921928f,
+    0.930109f, 0.937699f, 0.944713f, 0.951169f, 0.957088f, 0.962491f, 0.9674f, 0.971838f,
+    0.975832f, 0.979404f, 0.982582f, 0.985391f, 0.987857f, 0.990005f, 0.991863f, 0.993454f,
+    0.994804f, 0.995937f, 0.996877f, 0.997645f, 0.998264f, 0.998753f, 0.999131f, 0.999416f,
+    0.999625f, 0.999772f, 0.999871f, 0.999934f, 0.99997f, 0.999989f, 0.999997f, 0.99999964f, 1.0f,
+};
+
+const uint32_t * const ff_celt_pvq_u_row[15] = {
+    celt_pvq_u +    0, celt_pvq_u +  176, celt_pvq_u +  351,
+    celt_pvq_u +  525, celt_pvq_u +  698, celt_pvq_u +  870,
+    celt_pvq_u + 1041, celt_pvq_u + 1131, celt_pvq_u + 1178,
+    celt_pvq_u + 1207, celt_pvq_u + 1226, celt_pvq_u + 1240,
+    celt_pvq_u + 1248, celt_pvq_u + 1254, celt_pvq_u + 1257
+};
+
+/* Deemphasis constant (alpha_p), as specified in RFC6716 as 0.8500061035.
+ * libopus uses a slighly rounded constant, set to 0.85 exactly,
+ * to simplify its fixed-point version, but it's not significant to impact
+ * compliance. */
+#define CELT_EMPH_COEFF 0.8500061035
+
+DECLARE_ALIGNED(16, const float, ff_opus_deemph_weights)[] = {
+    CELT_EMPH_COEFF,
+    CELT_EMPH_COEFF*CELT_EMPH_COEFF,
+    CELT_EMPH_COEFF*CELT_EMPH_COEFF*CELT_EMPH_COEFF,
+    CELT_EMPH_COEFF*CELT_EMPH_COEFF*CELT_EMPH_COEFF*CELT_EMPH_COEFF,
+
+    0,
+    CELT_EMPH_COEFF,
+    CELT_EMPH_COEFF*CELT_EMPH_COEFF,
+    CELT_EMPH_COEFF*CELT_EMPH_COEFF*CELT_EMPH_COEFF,
+
+    0,
+    0,
+    CELT_EMPH_COEFF,
+    CELT_EMPH_COEFF*CELT_EMPH_COEFF,
+
+    0,
+    0,
+    0,
+    CELT_EMPH_COEFF,
+};
diff --git a/libavcodec/opus/tab.h b/libavcodec/opus/tab.h
new file mode 100644
index 0000000000..109a422b9f
--- /dev/null
+++ b/libavcodec/opus/tab.h
@@ -0,0 +1,169 @@
+/*
+ * Copyright (c) 2012 Andrew D'Addesio
+ * Copyright (c) 2013-2014 Mozilla Corporation
+ * Copyright (c) 2016 Rostislav Pehlivanov <atomnuker@gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#ifndef AVCODEC_OPUS_TAB_H
+#define AVCODEC_OPUS_TAB_H
+
+#include <stdint.h>
+
+#include "libavutil/attributes_internal.h"
+
+FF_VISIBILITY_PUSH_HIDDEN
+extern const uint8_t  ff_celt_band_end[];
+
+extern const uint8_t  ff_opus_default_coupled_streams[];
+
+extern const uint16_t ff_silk_model_lbrr_flags_40[];
+extern const uint16_t ff_silk_model_lbrr_flags_60[];
+
+extern const uint16_t ff_silk_model_stereo_s1[];
+extern const uint16_t ff_silk_model_stereo_s2[];
+extern const uint16_t ff_silk_model_stereo_s3[];
+extern const uint16_t ff_silk_model_mid_only[];
+
+extern const uint16_t ff_silk_model_frame_type_inactive[];
+extern const uint16_t ff_silk_model_frame_type_active[];
+
+extern const uint16_t ff_silk_model_gain_highbits[3][9];
+extern const uint16_t ff_silk_model_gain_lowbits[];
+extern const uint16_t ff_silk_model_gain_delta[];
+
+extern const uint16_t ff_silk_model_lsf_s1[2][2][33];
+extern const uint16_t ff_silk_model_lsf_s2[32][10];
+extern const uint16_t ff_silk_model_lsf_s2_ext[];
+extern const uint16_t ff_silk_model_lsf_interpolation_offset[];
+
+extern const uint16_t ff_silk_model_pitch_highbits[];
+extern const uint16_t ff_silk_model_pitch_lowbits_nb[];
+extern const uint16_t ff_silk_model_pitch_lowbits_mb[];
+extern const uint16_t ff_silk_model_pitch_lowbits_wb[];
+extern const uint16_t ff_silk_model_pitch_delta[];
+extern const uint16_t ff_silk_model_pitch_contour_nb10ms[];
+extern const uint16_t ff_silk_model_pitch_contour_nb20ms[];
+extern const uint16_t ff_silk_model_pitch_contour_mbwb10ms[];
+extern const uint16_t ff_silk_model_pitch_contour_mbwb20ms[];
+
+extern const uint16_t ff_silk_model_ltp_filter[];
+extern const uint16_t ff_silk_model_ltp_filter0_sel[];
+extern const uint16_t ff_silk_model_ltp_filter1_sel[];
+extern const uint16_t ff_silk_model_ltp_filter2_sel[];
+extern const uint16_t ff_silk_model_ltp_scale_index[];
+
+extern const uint16_t ff_silk_model_lcg_seed[];
+
+extern const uint16_t ff_silk_model_exc_rate[2][10];
+
+extern const uint16_t ff_silk_model_pulse_count[11][19];
+extern const uint16_t ff_silk_model_pulse_location[4][168];
+
+extern const uint16_t ff_silk_model_excitation_lsb[];
+extern const uint16_t ff_silk_model_excitation_sign[3][2][7][3];
+
+extern const int16_t  ff_silk_stereo_weights[];
+
+extern const uint8_t  ff_silk_lsf_s2_model_sel_nbmb[32][10];
+extern const uint8_t  ff_silk_lsf_s2_model_sel_wb[32][16];
+
+extern const uint8_t  ff_silk_lsf_pred_weights_nbmb[2][9];
+extern const uint8_t  ff_silk_lsf_pred_weights_wb[2][15];
+
+extern const uint8_t  ff_silk_lsf_weight_sel_nbmb[32][9];
+extern const uint8_t  ff_silk_lsf_weight_sel_wb[32][15];
+
+extern const uint8_t  ff_silk_lsf_codebook_nbmb[32][10];
+extern const uint8_t  ff_silk_lsf_codebook_wb[32][16];
+
+extern const uint16_t ff_silk_lsf_min_spacing_nbmb[];
+extern const uint16_t ff_silk_lsf_min_spacing_wb[];
+
+extern const uint8_t  ff_silk_lsf_ordering_nbmb[];
+extern const uint8_t  ff_silk_lsf_ordering_wb[];
+
+extern const int16_t  ff_silk_cosine[];
+
+extern const uint16_t ff_silk_pitch_scale[];
+extern const uint16_t ff_silk_pitch_min_lag[];
+extern const uint16_t ff_silk_pitch_max_lag[];
+
+extern const int8_t   ff_silk_pitch_offset_nb10ms[3][2];
+extern const int8_t   ff_silk_pitch_offset_nb20ms[11][4];
+extern const int8_t   ff_silk_pitch_offset_mbwb10ms[12][2];
+extern const int8_t   ff_silk_pitch_offset_mbwb20ms[34][4];
+
+extern const int8_t   ff_silk_ltp_filter0_taps[8][5];
+extern const int8_t   ff_silk_ltp_filter1_taps[16][5];
+extern const int8_t   ff_silk_ltp_filter2_taps[32][5];
+
+extern const uint16_t ff_silk_ltp_scale_factor[];
+
+extern const uint8_t  ff_silk_shell_blocks[3][2];
+
+extern const uint8_t  ff_silk_quant_offset[2][2];
+
+extern const int      ff_silk_stereo_interp_len[3];
+
+extern const uint16_t ff_celt_model_tapset[];
+extern const uint16_t ff_celt_model_spread[];
+extern const uint16_t ff_celt_model_alloc_trim[];
+extern const uint16_t ff_celt_model_energy_small[];
+
+extern const uint8_t  ff_celt_freq_bands[];
+extern const uint8_t  ff_celt_freq_range[];
+extern const uint8_t  ff_celt_log_freq_range[];
+
+extern const int8_t   ff_celt_tf_select[4][2][2][2];
+
+extern const float    ff_celt_mean_energy[];
+
+extern const float    ff_celt_alpha_coef[];
+extern const float    ff_celt_beta_coef[];
+
+extern const uint8_t  ff_celt_coarse_energy_dist[4][2][42];
+
+extern const uint8_t  ff_celt_static_alloc[11][21];
+extern const uint8_t  ff_celt_static_caps[4][2][21];
+
+extern const uint8_t  ff_celt_cache_bits[392];
+extern const int16_t  ff_celt_cache_index[105];
+
+extern const uint8_t  ff_celt_log2_frac[];
+
+extern const uint8_t  ff_celt_bit_interleave[];
+extern const uint8_t  ff_celt_bit_deinterleave[];
+
+extern const uint8_t  ff_celt_hadamard_order[];
+
+extern const uint16_t ff_celt_qn_exp2[];
+
+extern const float    ff_celt_postfilter_taps[3][3];
+
+extern const float    ff_celt_window2[120];
+
+extern const float    ff_celt_window_padded[];
+static const float *const ff_celt_window = &ff_celt_window_padded[8];
+
+extern const float    ff_opus_deemph_weights[];
+
+extern const uint32_t * const ff_celt_pvq_u_row[15];
+FF_VISIBILITY_POP_HIDDEN
+
+#endif /* AVCODEC_OPUS_TAB_H */