diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index 8080e9127d..b8c05773d5 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -436,6 +436,7 @@ OBJS-$(CONFIG_NUV_DECODER)             += nuv.o rtjpeg.o
 OBJS-$(CONFIG_ON2AVC_DECODER)          += on2avc.o on2avcdata.o
 OBJS-$(CONFIG_OPUS_DECODER)            += opusdec.o opus.o opus_celt.o opus_rc.o \
                                           opus_pvq.o opus_silk.o opustab.o vorbis_data.o
+OBJS-$(CONFIG_OPUS_ENCODER)            += opusenc.o opustab.o opus_pvq.o
 OBJS-$(CONFIG_PAF_AUDIO_DECODER)       += pafaudio.o
 OBJS-$(CONFIG_PAF_VIDEO_DECODER)       += pafvideo.o
 OBJS-$(CONFIG_PAM_DECODER)             += pnmdec.o pnm.o
diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c
index f92b2b7496..bea32f8072 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -446,7 +446,7 @@ void avcodec_register_all(void)
     REGISTER_DECODER(MPC8,              mpc8);
     REGISTER_ENCDEC (NELLYMOSER,        nellymoser);
     REGISTER_DECODER(ON2AVC,            on2avc);
-    REGISTER_DECODER(OPUS,              opus);
+    REGISTER_ENCDEC (OPUS,              opus);
     REGISTER_DECODER(PAF_AUDIO,         paf_audio);
     REGISTER_DECODER(QCELP,             qcelp);
     REGISTER_DECODER(QDM2,              qdm2);
diff --git a/libavcodec/opus_celt.h b/libavcodec/opus_celt.h
index 1378d85df3..a54cab68c6 100644
--- a/libavcodec/opus_celt.h
+++ b/libavcodec/opus_celt.h
@@ -61,14 +61,23 @@ enum CeltBlockSize {
 
 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];
 
+    int band_bins[CELT_MAX_BANDS]; /* MDCT bins per band */
+    float *band_coeffs[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)[120];
+    DECLARE_ALIGNED(32, float, samples)[CELT_MAX_FRAME_SIZE];
+
     /* postfilter parameters */
     int   pf_period_new;
     float pf_gains_new[3];
@@ -94,6 +103,12 @@ typedef struct CeltFrame {
     int end_band;
     int coded_bands;
     int transient;
+    int intra;
+    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 */
@@ -109,6 +124,7 @@ typedef struct CeltFrame {
     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];
diff --git a/libavcodec/opus_pvq.c b/libavcodec/opus_pvq.c
index a39330b7e4..adf6f44c8d 100644
--- a/libavcodec/opus_pvq.c
+++ b/libavcodec/opus_pvq.c
@@ -1,7 +1,7 @@
 /*
  * Copyright (c) 2012 Andrew D'Addesio
  * Copyright (c) 2013-2014 Mozilla Corporation
- * Copyright (c) 2016 Rostislav Pehlivanov <atomnuker@gmail.com>
+ * Copyright (c) 2017 Rostislav Pehlivanov <atomnuker@gmail.com>
  *
  * This file is part of FFmpeg.
  *
@@ -78,8 +78,8 @@ static inline void celt_normalize_residual(const int * av_restrict iy, float * a
         X[i] = g * iy[i];
 }
 
-static void celt_exp_rotation1(float *X, uint32_t len, uint32_t stride,
-                               float c, float s)
+static void celt_exp_rotation_impl(float *X, uint32_t len, uint32_t stride,
+                                   float c, float s)
 {
     float *Xptr;
     int i;
@@ -105,7 +105,7 @@ static void celt_exp_rotation1(float *X, uint32_t len, uint32_t stride,
 
 static inline void celt_exp_rotation(float *X, uint32_t len,
                                      uint32_t stride, uint32_t K,
-                                     enum CeltSpread spread)
+                                     enum CeltSpread spread, const int encode)
 {
     uint32_t stride2 = 0;
     float c, s;
@@ -118,8 +118,8 @@ static inline void celt_exp_rotation(float *X, uint32_t len,
     gain = (float)len / (len + (20 - 5*spread) * K);
     theta = M_PI * gain * gain / 4;
 
-    c = cos(theta);
-    s = sin(theta);
+    c = cosf(theta);
+    s = sinf(theta);
 
     if (len >= stride << 3) {
         stride2 = 1;
@@ -133,9 +133,15 @@ static inline void celt_exp_rotation(float *X, uint32_t len,
     extract_collapse_mask().*/
     len /= stride;
     for (i = 0; i < stride; i++) {
-        if (stride2)
-            celt_exp_rotation1(X + i * len, len, stride2, s, c);
-        celt_exp_rotation1(X + i * len, len, 1, c, s);
+        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);
+        }
     }
 }
 
@@ -270,6 +276,18 @@ static inline int celt_compute_qn(int N, int b, int offset, int pulse_cap,
     return qn;
 }
 
+/* Convert the quantized vector to an index */
+static inline uint32_t celt_icwrsi(uint32_t N, 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)
 {
@@ -356,12 +374,85 @@ static inline uint64_t celt_cwrsi(uint32_t N, uint32_t K, uint32_t i, int *y)
     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, 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);
 }
 
+/*
+ * Placehoder, quite a lot slower than libopus's code, and its also similar,
+ * though we always work in the signed domain and always do a pre-projection
+ * (fancy saying for scale and round to the nearest smallest integer) and we always
+ * calculate the full distortion. I hate it.
+ *
+ * The problem this function solves is very simple: you have a K integer amount of
+ * "pulses", and you have to give them to the y[] array such that when you
+ * normalize y[] to the L2 norm (just do y_norm += y[i]*y[i], then divide
+ * every y[i]/sqrtf(y_norm)) the difference between X[] (which is already normalized)
+ * is the smallest. Summing up y[i] should return K.
+ *
+ * Note that this code still returns identical results to libopus's search. Usually
+ * any increase in distortion, even the slightest, will result in worse quality.
+ */
+static void celt_pvq_search(float *X, int *y, int K, int N)
+{
+    int i, j, k;
+    float res = K, x_sum = 0.0f, y_norm = 0.0f;
+
+    for (i = 0; i < N; i++)
+        x_sum += FFABS(X[i]);
+
+    res /= x_sum;
+
+    for (i = 0; i < N; i++) {
+        y[i] = floor(res*X[i]);
+        K -= FFABS(y[i]);
+        y_norm += y[i]*y[i];
+    }
+
+    if (K) {
+        const int phase = FFSIGN(K);
+        for (j = 0; j < FFABS(K); j++) {
+            int min_idx = 0;
+            float min_dist = FLT_MAX;
+            for (i = 0; i < N; i++) {
+                float new_norm, dist = 0.0f, o_yn = y_norm - y[i]*y[i];
+                y[i] += FFSIGN(X[i])*phase;
+                new_norm = 1.0f/(sqrtf(o_yn + y[i]*y[i]) + FLT_MIN);
+                for (k = 0; k < N; k++) {
+                    const float deq_v = y[k]*new_norm;
+                    dist += (deq_v - X[k])*(deq_v - X[k]);
+                }
+                y[i] -= FFSIGN(X[i])*phase;
+                if (dist < min_dist) {
+                    min_idx = i;
+                    min_dist = dist;
+                }
+            }
+            y_norm -= y[min_idx]*y[min_idx];
+            y[min_idx] += FFSIGN(X[min_idx])*phase;
+            y_norm += y[min_idx]*y[min_idx];
+        }
+    }
+}
+
+static uint32_t celt_alg_quant(OpusRangeCoder *rc, float *X, uint32_t N, uint32_t K,
+                               enum CeltSpread spread, uint32_t blocks, float gain)
+{
+    int y[176];
+
+    celt_exp_rotation(X, N, blocks, K, spread, 1);
+    celt_pvq_search(X, y, K, N);
+    celt_encode_pulses(rc, y,  N, K);
+    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,
@@ -371,7 +462,7 @@ static uint32_t celt_alg_unquant(OpusRangeCoder *rc, float *X, uint32_t N, uint3
 
     gain /= sqrtf(celt_decode_pulses(rc, y, N, K));
     celt_normalize_residual(y, X, N, gain);
-    celt_exp_rotation(X, N, blocks, K, spread);
+    celt_exp_rotation(X, N, blocks, K, spread, 0);
     return celt_extract_collapse_mask(y, N, blocks);
 }
 
@@ -725,5 +816,353 @@ uint32_t ff_celt_decode_band(CeltFrame *f, OpusRangeCoder *rc, const int band,
         }
         cm = av_mod_uintp2(cm, blocks);
     }
+
+    return cm;
+}
+
+/* This has to be, AND MUST BE done by the psychoacoustic system, this has a very
+ * big impact on the entire quantization and especially huge on transients */
+static int celt_calc_theta(const float *X, const float *Y, int coupling, int N)
+{
+    int j;
+    float e[2] = { 0.0f, 0.0f };
+    for (j = 0; j < N; j++) {
+        if (coupling) { /* Coupling case */
+            e[0] += (X[j] + Y[j])*(X[j] + Y[j]);
+            e[1] += (X[j] - Y[j])*(X[j] - Y[j]);
+        } else {
+            e[0] += X[j]*X[j];
+            e[1] += Y[j]*Y[j];
+        }
+    }
+    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_MIN);
+    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;
+    const float decouple_norm = 1.0f/sqrtf(2.0f);
+    for (i = 0; i < N; i++) {
+        const float Xret = X[i];
+        X[i] = (X[i] + Y[i])*decouple_norm;
+        Y[i] = (Y[i] - Xret)*decouple_norm;
+    }
+}
+
+uint32_t ff_celt_encode_band(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)
+{
+    const uint8_t *cache;
+    int dualstereo, split;
+    int imid = 0, iside = 0;
+    //uint32_t N0 = N;
+    int N_B;
+    //int N_B0;
+    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;
+
+    //N_B0 = N_B = N / blocks;
+    split = dualstereo = (Y != NULL);
+
+    if (N == 1) {
+        /* special case for one sample - the decoder's output will be +- 1.0f!!! */
+        int i;
+        float *x = X;
+        for (i = 0; i <= dualstereo; i++) {
+            if (f->remaining2 >= 1<<3) {
+                ff_opus_rc_put_raw(rc, x[0] < 0, 1);
+                f->remaining2 -= 1 << 3;
+                b             -= 1 << 3;
+            }
+            x = Y;
+        }
+        if (lowband_out)
+            lowband_out[0] = X[0];
+        return 1;
+    }
+
+    if (!dualstereo && 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)) {
+            int j;
+            for (j = 0; j < N; j++)
+                lowband_scratch[j] = lowband[j];
+            lowband = lowband_scratch;
+        }
+
+        for (k = 0; k < recombine; k++) {
+            celt_haar1(X, 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) {
+            celt_haar1(X, 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)
+            celt_deinterleave_hadamard(f->scratch, X, 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 (!dualstereo && 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 = celt_calc_theta(X, Y, dualstereo, N);
+        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) - (dualstereo && N == 2 ? CELT_QTHETA_OFFSET_TWOPHASE :
+                                                          CELT_QTHETA_OFFSET);
+        qn = (dualstereo && band >= f->intensity_stereo) ? 1 :
+             celt_compute_qn(N, b, offset, pulse_cap, dualstereo);
+        tell = opus_rc_tell_frac(rc);
+
+        if (qn != 1) {
+
+            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 (dualstereo && N > 2)
+                ff_opus_rc_enc_uint_step(rc, itheta, qn / 2);
+            else if (dualstereo || 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 (dualstereo) {
+                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 (dualstereo) {
+             inv = itheta > 8192;
+             if (inv)
+             {
+                int j;
+                for (j=0;j<N;j++)
+                   Y[j] = -Y[j];
+             }
+             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;
+            }
+
+            itheta = 0;
+        }
+        qalloc = opus_rc_tell_frac(rc) - tell;
+        b -= qalloc;
+
+        orig_fill = fill;
+        if (itheta == 0) {
+            imid = 32767;
+            iside = 0;
+            fill = av_mod_uintp2(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 && dualstereo) {
+            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) {
+                sign = x2[0]*y2[1] - x2[1]*y2[0] < 0;
+                ff_opus_rc_put_raw(rc, sign, 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 = ff_celt_encode_band(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;
+
+            /* Give more bits to low-energy MDCTs than they would
+             * otherwise deserve */
+            if (B0 > 1 && !dualstereo && (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 && !dualstereo)
+                next_lowband2 = lowband + N; /* >32-bit split case */
+
+            /* Only stereo needs to pass on lowband_out.
+             * Otherwise, it's handled at the end */
+            if (dualstereo)
+                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 = ff_celt_encode_band(f, rc, band, X, NULL, N, mbits, blocks,
+                                         lowband, duration, next_lowband_out1,
+                                         next_level, dualstereo ? 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. */
+                cm |= ff_celt_encode_band(f, rc, band, Y, NULL, N, sbits, blocks,
+                                          next_lowband2, duration, NULL,
+                                          next_level, gain * side, NULL,
+                                          fill >> blocks) << ((B0 >> 1) & (dualstereo - 1));
+            } else {
+                /* For a stereo split, the high bits of fill are always zero,
+                 * so no folding will be done to the side. */
+                cm = ff_celt_encode_band(f, rc, band, Y, NULL, N, sbits, blocks,
+                                         next_lowband2, duration, NULL,
+                                         next_level, gain * side, NULL,
+                                         fill >> blocks) << ((B0 >> 1) & (dualstereo - 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 |= ff_celt_encode_band(f, rc, band, X, NULL, N, mbits, blocks,
+                                          lowband, duration, next_lowband_out1,
+                                          next_level, dualstereo ? 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 quantization */
+            cm = celt_alg_quant(rc, X, N, (q < 8) ? q : (8 + (q & 7)) << ((q >> 3) - 1),
+                                f->spread, blocks, gain);
+        }
+    }
+
     return cm;
 }
diff --git a/libavcodec/opus_pvq.h b/libavcodec/opus_pvq.h
index 0354a3c960..d414b47a42 100644
--- a/libavcodec/opus_pvq.h
+++ b/libavcodec/opus_pvq.h
@@ -32,4 +32,10 @@ uint32_t ff_celt_decode_band(CeltFrame *f, OpusRangeCoder *rc, const int band,
                              float *lowband, int duration, float *lowband_out, int level,
                              float gain, float *lowband_scratch, int fill);
 
+/* Encodes a band using PVQ */
+uint32_t ff_celt_encode_band(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);
+
 #endif /* AVCODEC_OPUS_PVQ_H */
diff --git a/libavcodec/opusenc.c b/libavcodec/opusenc.c
new file mode 100644
index 0000000000..da05c70927
--- /dev/null
+++ b/libavcodec/opusenc.c
@@ -0,0 +1,1132 @@
+/*
+ * 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 "opus_celt.h"
+#include "opus_pvq.h"
+#include "opustab.h"
+
+#include "libavutil/float_dsp.h"
+#include "libavutil/opt.h"
+#include "internal.h"
+#include "bytestream.h"
+#include "audio_frame_queue.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
+
+/* Just a practical limit, actual limit is 120ms of total audio length */
+#define OPUS_MAX_FRAMES 6
+
+#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;
+} OpusEncOptions;
+
+typedef struct OpusEncContext {
+    AVClass *av_class;
+    OpusEncOptions options;
+    AVCodecContext *avctx;
+    AVFrame *empty;
+    AudioFrameQueue afq;
+    AVFloatDSPContext *dsp;
+    MDCT15Context *mdct[CELT_BLOCK_NB];
+    struct FFBufQueue bufqueue;
+
+    enum OpusMode mode;
+    enum OpusBandwidth bandwidth;
+    int pkt_framesize;
+    int pkt_frames;
+    int insane_padding;
+
+    int channels;
+
+    CeltFrame frame[OPUS_MAX_FRAMES];
+    OpusRangeCoder rc[OPUS_MAX_FRAMES];
+
+    /* 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->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 i, 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->pkt_framesize][s->mode][s->bandwidth];
+    *fsize_needed = 0;
+    if (!cfg)
+        return 1;
+    if (s->pkt_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->pkt_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 (i = 0; i < (s->pkt_frames - 1); i++)
+            *fsize_needed |= (s->frame[i].framebits != s->frame[i + 1].framebits);
+        tmp = (*fsize_needed) << 7;                                     /* vbr flag */
+        tmp |= s->pkt_frames;                    /* frame number - can be 0 as well */
+        *toc++ = tmp;
+    }
+    *size = 1 + extended_toc;
+    return 0;
+}
+
+static void celt_frame_setup_input(OpusEncContext *s, CeltFrame *f)
+{
+    int sf, ch;
+    AVFrame *cur = NULL;
+    const int subframesize = s->avctx->frame_size;
+    int subframes = OPUS_BLOCK_SIZE(s->pkt_framesize) / subframesize;
+
+    cur = ff_bufqueue_get(&s->bufqueue);
+
+    for (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 (sf = 0; sf < subframes; sf++) {
+        if (sf != (subframes - 1))
+            cur = ff_bufqueue_get(&s->bufqueue);
+        else
+            cur = ff_bufqueue_peek(&s->bufqueue, 0);
+
+        for (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)
+{
+    int i, sf, ch;
+    const int subframesize = s->avctx->frame_size;
+    const int subframes = OPUS_BLOCK_SIZE(s->pkt_framesize) / subframesize;
+
+    /* Filter overlap */
+    for (ch = 0; ch < f->channels; ch++) {
+        CeltBlock *b = &f->block[ch];
+        float m = b->emph_coeff;
+        for (i = 0; i < CELT_OVERLAP; i++) {
+            float sample = b->overlap[i];
+            b->overlap[i] = sample - m;
+            m = sample * CELT_EMPH_COEFF;
+        }
+        b->emph_coeff = m;
+    }
+
+    /* Filter the samples but do not update the last subframe's coeff - overlap ^^^ */
+    for (sf = 0; sf < subframes; sf++) {
+        for (ch = 0; ch < f->channels; ch++) {
+            CeltBlock *b = &f->block[ch];
+            float m = b->emph_coeff;
+            for (i = 0; i < subframesize; i++) {
+                float sample = b->samples[sf*subframesize + i];
+                b->samples[sf*subframesize + i] = sample - m;
+                m = sample * CELT_EMPH_COEFF;
+            }
+            if (sf != (subframes - 1))
+                b->emph_coeff = m;
+        }
+    }
+}
+
+/* Create the window and mdct it to get the coefficient */
+static void celt_frame_mdct(OpusEncContext *s, CeltFrame *f)
+{
+    float *win = s->scratch;
+    int i, t, ch, t_transforms = OPUS_BLOCK_SIZE(f->size)/CELT_OVERLAP;
+    const 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;
+
+    /* I think I can use s->dsp->vector_fmul_window for transients at least */
+    if (f->transient) {
+        for (ch = 0; ch < f->channels; ch++) {
+            CeltBlock *b = &f->block[ch];
+            float *src1 = b->overlap;
+            for (t = 0; t < t_transforms; t++) {
+                float tmp_c[120];
+                float *src2 = &b->samples[120*t];
+                for (i = 0; i < CELT_OVERLAP; i++) {
+                    win[               i] = src1[i]*ff_celt_window[i];
+                    win[CELT_OVERLAP + i] = src2[i]*ff_celt_window[CELT_OVERLAP - i - 1];
+                }
+                src1 = src2;
+                s->mdct[0]->mdct(s->mdct[0], tmp_c, win, t_transforms);
+                /* This can and should be done in place */
+                for (i = 0; i < CELT_OVERLAP; i++)
+                    b->coeffs[i*t_transforms + t] = tmp_c[i];
+            }
+        }
+    } else {
+        for (ch = 0; ch < f->channels; ch++) {
+            CeltBlock *b = &f->block[ch];
+
+            memset(win, 0, wlen*sizeof(float));
+
+            memcpy(&win[lap_dst + CELT_OVERLAP], b->samples, rwin*sizeof(float));
+
+            /* You know what's aligned? NOT. FUCKING. THIS. */
+            //s->dsp->vector_fmul(&dst[lap_dst], b->overlap, ff_celt_window, CELT_OVERLAP);
+            //s->dsp->vector_fmul_reverse(&dst[lap_dst + blk_len - CELT_OVERLAP], b->samples, ff_celt_window, CELT_OVERLAP);
+
+            for (i = 0; i < CELT_OVERLAP; i++) {
+                win[lap_dst           + i] = b->overlap[i]       *ff_celt_window[i];
+                win[lap_dst + blk_len + i] = b->samples[rwin + i]*ff_celt_window[CELT_OVERLAP - i - 1];
+            }
+
+            s->mdct[f->size]->mdct(s->mdct[f->size], b->coeffs, win, 1);
+        }
+    }
+}
+
+/* Fills the bands and normalizes them */
+static int celt_frame_map_norm_bands(OpusEncContext *s, CeltFrame *f)
+{
+    int i, j, ch, noise = 0;
+
+    for (ch = 0; ch < f->channels; ch++) {
+        CeltBlock *block = &f->block[ch];
+        float *start = block->coeffs;
+        for (i = 0; i < CELT_MAX_BANDS; i++) {
+            float ener = 0.0f;
+
+            /* Calculate band bins */
+            block->band_bins[i] = ff_celt_freq_range[i] << f->size;
+            block->band_coeffs[i] = start;
+            start += block->band_bins[i];
+
+            /* Normalize band energy */
+            for (j = 0; j < block->band_bins[i]; j++)
+                ener += block->band_coeffs[i][j]*block->band_coeffs[i][j];
+
+            block->lin_energy[i] = sqrtf(ener) + FLT_MIN;
+            ener = 1.0f/block->lin_energy[i];
+
+            for (j = 0; j < block->band_bins[i]; j++)
+                block->band_coeffs[i][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);
+            noise |= block->energy[i] > CELT_ENERGY_SILENCE;
+        }
+    }
+    return !noise;
+}
+
+static void celt_enc_tf(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
+{
+    int i, 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 (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 (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_bitalloc(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
+{
+    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 *cap = f->caps;
+    int boost[CELT_MAX_BANDS];
+    int trim_offset[CELT_MAX_BANDS];
+    int threshold[CELT_MAX_BANDS];
+    int bits1[CELT_MAX_BANDS];
+    int bits2[CELT_MAX_BANDS];
+
+    /* Tell the spread to the decoder */
+    if (opus_rc_tell(rc) + 4 <= f->framebits)
+        ff_opus_rc_enc_cdf(rc, f->spread, ff_celt_model_spread);
+
+    /* Generate static allocation caps */
+    for (i = 0; i < CELT_MAX_BANDS; i++) {
+        cap[i] = (ff_celt_static_caps[f->size][f->channels - 1][i] + 64)
+                 * ff_celt_freq_range[i] << (f->channels - 1) << f->size >> 2;
+    }
+
+    /* Band boosts */
+    tbits_8ths = f->framebits << 3;
+    for (i = f->start_band; i < f->end_band; i++) {
+        int quanta, b_dynalloc, boost_amount = f->alloc_boost[i];
+
+        boost[i] = 0;
+
+        quanta = ff_celt_freq_range[i] << (f->channels - 1) << f->size;
+        quanta = FFMIN(quanta << 3, FFMAX(6 << 3, quanta));
+        b_dynalloc = dynalloc;
+
+        while (opus_rc_tell_frac(rc) + (b_dynalloc << 3) < tbits_8ths && boost[i] < cap[i]) {
+            int is_boost = boost_amount--;
+
+            ff_opus_rc_enc_log(rc, is_boost, b_dynalloc);
+            if (!is_boost)
+                break;
+
+            boost[i]   += quanta;
+            tbits_8ths -= quanta;
+
+            b_dynalloc = 1;
+        }
+
+        if (boost[i])
+            dynalloc = FFMAX(2, dynalloc - 1);
+    }
+
+    /* Put allocation trim */
+    if (opus_rc_tell_frac(rc) + (6 << 3) <= tbits_8ths)
+        ff_opus_rc_enc_cdf(rc, f->alloc_trim, 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 = ff_celt_freq_range[i] * ff_celt_static_alloc[center][i]
+                       << (f->channels - 1) << f->size >> 2;
+
+            if (bandbits)
+                bandbits = FFMAX(0, bandbits + trim_offset[i]);
+            bandbits += boost[i];
+
+            if (bandbits >= threshold[i] || done) {
+                done = 1;
+                total += FFMIN(bandbits, cap[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] = ff_celt_freq_range[i] * ff_celt_static_alloc[low][i]
+                   << (f->channels - 1) << f->size >> 2;
+        bits2[i] = high >= CELT_VECTORS ? cap[i] :
+                   ff_celt_freq_range[i] * ff_celt_static_alloc[high][i]
+                   << (f->channels - 1) << f->size >> 2;
+
+        if (bits1[i])
+            bits1[i] = FFMAX(0, bits1[i] + trim_offset[i]);
+        if (bits2[i])
+            bits2[i] = FFMAX(0, bits2[i] + trim_offset[i]);
+        if (low)
+            bits1[i] += boost[i];
+        bits2[i] += boost[i];
+
+        if (boost[i])
+            skip_startband = i;
+        bits2[i] = FFMAX(0, bits2[i] - bits1[i]);
+    }
+
+    /* 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, cap[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, cap[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]
+                      + FFMAX(0, remaining - (ff_celt_freq_bands[j] - ff_celt_freq_bands[f->start_band]));
+
+        /* 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)) {
+            const int do_not_skip = f->coded_bands <= f->skip_band_floor;
+            ff_opus_rc_enc_log(rc, do_not_skip, 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;
+    }
+
+    /* Encode stereo flags */
+    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);
+    }
+    if (f->intensity_stereo <= f->start_band)
+        tbits_8ths += dualstereo_bit; /* no intensity stereo means no dual stereo */
+    else if (dualstereo_bit)
+        ff_opus_rc_enc_log(rc, f->dual_stereo, 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++) {
+        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 offset;     // fine energy quantization offset, i.e.
+                            // extra bits assigned over the standard
+                            // totalbits/dof
+            int fine_bits, max_bits;
+
+            extrabits = FFMAX(0, f->pulses[i] - cap[i]);
+            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(0, f->pulses[i] - (f->channels << 3));
+            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;
+    }
+}
+
+static void celt_quant_coarse(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
+{
+    int i, ch;
+    float alpha, beta, prev[2] = { 0, 0 };
+    const uint8_t *pmod = ff_celt_coarse_energy_dist[f->size][f->intra];
+
+    /* Inter is really just differential coding */
+    if (opus_rc_tell(rc) + 3 <= f->framebits)
+        ff_opus_rc_enc_log(rc, f->intra, 3);
+    else
+        f->intra = 0;
+
+    if (f->intra) {
+        alpha = 0.0f;
+        beta  = 1.0f - 4915.0f/32768.0f;
+    } else {
+        alpha = ff_celt_alpha_coef[f->size];
+        beta  = 1.0f - ff_celt_beta_coef[f->size];
+    }
+
+    for (i = f->start_band; i < f->end_band; i++) {
+        for (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, s->last_quantized_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, ((q_en & 1) << 1) | (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_fine(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
+{
+    int i, ch;
+    for (i = f->start_band; i < f->end_band; i++) {
+        if (!f->fine_bits[i])
+            continue;
+        for (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)
+{
+    int i, ch, priority;
+    for (priority = 0; priority < 2; priority++) {
+        for (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 (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_quant_bands(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
+{
+    float lowband_scratch[8 * 22];
+    float norm[2 * 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++) {
+        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;
+
+        int consumed = opus_rc_tell_frac(rc);
+        float *norm2 = norm + 8 * 100;
+        int effective_lowband = -1;
+        unsigned int cm[2];
+        int b;
+
+        /* 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);
+        } else
+            b = 0;
+
+        if (ff_celt_freq_bands[i] - ff_celt_freq_range[i] >= ff_celt_freq_bands[f->start_band] &&
+            (update_lowband || lowband_offset == 0))
+            lowband_offset = i;
+
+        /* 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 (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];
+            }
+        } else
+            /* Otherwise, we'll be using the LCG to fold, so all blocks will (almost
+            always) be non-zero.*/
+            cm[0] = cm[1] = (1 << f->blocks) - 1;
+
+        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++)
+                norm[j] = (norm[j] + norm2[j]) / 2;
+        }
+
+        if (f->dual_stereo) {
+            cm[0] = ff_celt_encode_band(f, rc, i, X, NULL, band_size, b / 2, f->blocks,
+                                        effective_lowband != -1 ? norm + (effective_lowband << f->size) : NULL, f->size,
+                                        norm + band_offset, 0, 1.0f, lowband_scratch, cm[0]);
+
+            cm[1] = ff_celt_encode_band(f, rc, i, Y, NULL, band_size, b/2, f->blocks,
+                                        effective_lowband != -1 ? norm2 + (effective_lowband << f->size) : NULL, f->size,
+                                        norm2 + band_offset, 0, 1.0f, lowband_scratch, cm[1]);
+        } else {
+            cm[0] = ff_celt_encode_band(f, rc, i, X, Y, band_size, b, f->blocks,
+                                        effective_lowband != -1 ? norm + (effective_lowband << f->size) : NULL, f->size,
+                                        norm + 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);
+    }
+}
+
+static void celt_encode_frame(OpusEncContext *s, OpusRangeCoder *rc, CeltFrame *f)
+{
+    int i, ch;
+
+    celt_frame_setup_input(s, f);
+    celt_apply_preemph_filter(s, f);
+    if (f->pfilter) {
+        /* Not implemented */
+    }
+    celt_frame_mdct(s, f);
+    f->silence = celt_frame_map_norm_bands(s, f);
+    if (f->silence) {
+        f->framebits = 1;
+        return;
+    }
+
+    ff_opus_rc_enc_log(rc, f->silence, 15);
+
+    if (!f->start_band && opus_rc_tell(rc) + 16 <= f->framebits)
+        ff_opus_rc_enc_log(rc, f->pfilter, 1);
+
+    if (f->pfilter) {
+        /* Not implemented */
+    }
+
+    if (f->size && opus_rc_tell(rc) + 3 <= f->framebits)
+        ff_opus_rc_enc_log(rc, f->transient, 3);
+
+    celt_quant_coarse (s, rc, f);
+    celt_enc_tf       (s, rc, f);
+    celt_bitalloc     (s, rc, f);
+    celt_quant_fine   (s, rc, f);
+    celt_quant_bands  (s, rc, f);
+
+    if (f->anticollapse_needed)
+        ff_opus_rc_put_raw(rc, f->anticollapse, 1);
+
+    celt_quant_final(s, rc, f);
+
+    for (ch = 0; ch < f->channels; ch++) {
+        CeltBlock *block = &f->block[ch];
+        for (i = 0; i < CELT_MAX_BANDS; i++)
+            s->last_quantized_energy[ch][i] = block->energy[i] + block->error_energy[i];
+    }
+}
+
+static void ff_opus_psy_process(OpusEncContext *s, int end, int *need_more)
+{
+    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);
+
+    s->pkt_frames = 1;
+    s->pkt_framesize = max_bsize;
+    s->mode = OPUS_MODE_CELT;
+    s->bandwidth = OPUS_BANDWIDTH_FULLBAND;
+
+    *need_more = s->bufqueue.available*s->avctx->frame_size < (max_delay_samples + CELT_OVERLAP);
+    /* Don't request more if we start being flushed with NULL frames */
+    *need_more = !end && *need_more;
+}
+
+static void ff_opus_psy_celt_frame_setup(OpusEncContext *s, CeltFrame *f, int index)
+{
+    int frame_size = OPUS_BLOCK_SIZE(s->pkt_framesize);
+
+    f->avctx = s->avctx;
+    f->dsp = s->dsp;
+    f->start_band = (s->mode == OPUS_MODE_HYBRID) ? 17 : 0;
+    f->end_band = ff_celt_band_end[s->bandwidth];
+    f->channels = s->channels;
+    f->size = s->pkt_framesize;
+
+    /* Decisions */
+    f->silence = 0;
+    f->pfilter = 0;
+    f->transient = 0;
+    f->intra = 1;
+    f->tf_select = 1;
+    f->anticollapse = 0;
+    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);
+
+    f->blocks = f->transient ? frame_size/CELT_OVERLAP : 1;
+    f->framebits = FFALIGN(lrintf((double)s->avctx->bit_rate/(s->avctx->sample_rate/frame_size)), 8);
+}
+
+static void opus_packet_assembler(OpusEncContext *s, AVPacket *avpkt)
+{
+    int i, offset, fsize_needed;
+
+    /* Write toc */
+    opus_gen_toc(s, avpkt->data, &offset, &fsize_needed);
+
+    for (i = 0; i < s->pkt_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)
+{
+    int i;
+    AVFrame *f = av_frame_alloc();
+    if (!f)
+        return NULL;
+    f->format = s->avctx->sample_fmt;
+    f->nb_samples = s->avctx->frame_size;
+    f->channel_layout = s->avctx->channel_layout;
+    if (av_frame_get_buffer(f, 4)) {
+        av_frame_free(&f);
+        return NULL;
+    }
+    for (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 i, ret, frame_size, need_more, 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 {
+        if (!s->afq.remaining_samples)
+            return 0; /* We've been flushed and there's nothing left to encode */
+    }
+
+    /* Run the psychoacoustic system */
+    ff_opus_psy_process(s, !frame, &need_more);
+
+    /* Get more samples for lookahead/encoding */
+    if (need_more)
+        return 0;
+
+    frame_size = OPUS_BLOCK_SIZE(s->pkt_framesize);
+
+    if (!frame) {
+        /* This can go negative, that's not a problem, we only pad if positive */
+        int pad_empty = s->pkt_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 (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 (i = 0; i < s->pkt_frames; i++) {
+        ff_opus_rc_enc_init(&s->rc[i]);
+        ff_opus_psy_celt_frame_setup(s, &s->frame[i], i);
+        celt_encode_frame(s, &s->rc[i], &s->frame[i]);
+        alloc_size += s->frame[i].framebits >> 3;
+    }
+
+    /* Worst case toc + the frame lengths if needed */
+    alloc_size += 2 + s->pkt_frames*2;
+
+    if ((ret = ff_alloc_packet2(avctx, avpkt, alloc_size, 0)) < 0)
+        return ret;
+
+    /* Assemble packet */
+    opus_packet_assembler(s, avpkt);
+
+    /* Remove samples from queue and skip if needed */
+    ff_af_queue_remove(&s->afq, s->pkt_frames*frame_size, &avpkt->pts, &avpkt->duration);
+    if (s->pkt_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->pkt_frames*frame_size - avpkt->duration + 120);
+    }
+
+    *got_packet_ptr = 1;
+
+    return 0;
+}
+
+static av_cold int opus_encode_end(AVCodecContext *avctx)
+{
+    int i;
+    OpusEncContext *s = avctx->priv_data;
+
+    for (i = 0; i < CELT_BLOCK_NB; i++)
+        ff_mdct15_uninit(&s->mdct[i]);
+
+    av_freep(&s->dsp);
+    ff_af_queue_close(&s->afq);
+    ff_bufqueue_discard_all(&s->bufqueue);
+    av_freep(&avctx->extradata);
+
+    return 0;
+}
+
+static av_cold int opus_encode_init(AVCodecContext *avctx)
+{
+    int i, ch, ret;
+    OpusEncContext *s = avctx->priv_data;
+
+    s->avctx = avctx;
+    s->channels = avctx->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;
+
+    avctx->cutoff = !avctx->cutoff ? 20000 : avctx->cutoff;
+
+    if (!avctx->bit_rate) {
+        int coupled = ff_opus_default_coupled_streams[s->channels - 1];
+        avctx->bit_rate = coupled*(96000) + (s->channels - coupled)*(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 %li kbps, clipping to %li 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) {
+        av_log(avctx, AV_LOG_ERROR, "Failed to allocate extradata.\n");
+        ret = AVERROR(ENOMEM);
+        goto fail;
+    }
+    opus_write_extradata(avctx);
+
+    ff_af_queue_init(avctx, &s->afq);
+
+    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 (i = 0; i < CELT_BLOCK_NB; i++)
+        if ((ret = ff_mdct15_init(&s->mdct[i], 0, i + 3, 68 << (CELT_BLOCK_NB - 1 - i))))
+            goto fail;
+
+    /* Zero out previous energy (matters for inter first frame) */
+    for (ch = 0; ch < s->channels; ch++)
+        for (i = 0; i < CELT_MAX_BANDS; i++)
+            s->last_quantized_energy[ch][i] = 0.0f;
+
+    /* Allocate an empty frame to use as overlap for the first frame of audio */
+    ff_bufqueue_add(avctx, &s->bufqueue, spawn_empty_frame(s));
+
+    return 0;
+
+fail:
+    av_frame_free(&s->empty);
+    opus_encode_end(avctx);
+    return ret;
+}
+
+#define OPUSENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
+static const AVOption opusenc_options[] = {
+    { "opus_max_delay_ms", "Maximum delay (and lookahead) in milliseconds", offsetof(OpusEncContext, options.max_delay_ms), AV_OPT_TYPE_FLOAT, { .dbl = OPUS_MAX_LOOKAHEAD }, 2.5f, OPUS_MAX_LOOKAHEAD, OPUSENC_FLAGS },
+    { 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 AVCodecDefault opusenc_defaults[] = {
+    { "b", "0" },
+    { "compression_level", "10" },
+    { NULL },
+};
+
+AVCodec ff_opus_encoder = {
+    .name           = "opus",
+    .long_name      = NULL_IF_CONFIG_SMALL("Opus"),
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_OPUS,
+    .defaults       = opusenc_defaults,
+    .priv_class     = &opusenc_class,
+    .priv_data_size = sizeof(OpusEncContext),
+    .init           = opus_encode_init,
+    .encode2        = opus_encode_frame,
+    .close          = opus_encode_end,
+    .caps_internal  = FF_CODEC_CAP_INIT_THREADSAFE,
+    .capabilities   = AV_CODEC_CAP_EXPERIMENTAL | AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
+    .supported_samplerates = (const int []){ 48000, 0 },
+    .channel_layouts = (const uint64_t []){ AV_CH_LAYOUT_MONO,
+                                            AV_CH_LAYOUT_STEREO, 0 },
+    .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
+                                                     AV_SAMPLE_FMT_NONE },
+};
