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-rwxr-xr-xconfigure1
-rw-r--r--libavcodec/Makefile1
-rw-r--r--libavcodec/allcodecs.c2
-rw-r--r--libavcodec/opus_celt.h16
-rw-r--r--libavcodec/opus_pvq.c444
-rw-r--r--libavcodec/opus_pvq.h6
-rw-r--r--libavcodec/opusenc.c1130
7 files changed, 1591 insertions, 9 deletions
diff --git a/configure b/configure
index a7cd3a2244..2f761c39c2 100755
--- a/configure
+++ b/configure
@@ -2492,6 +2492,7 @@ nuv_decoder_select="idctdsp lzo"
on2avc_decoder_select="mdct"
opus_decoder_deps="swresample"
opus_decoder_select="mdct15"
+opus_encoder_select="audio_frame_queue audiodsp mdct15"
png_decoder_select="zlib"
png_encoder_select="llvidencdsp zlib"
prores_decoder_select="blockdsp idctdsp"
diff --git a/libavcodec/Makefile b/libavcodec/Makefile
index e1f83a0404..a1ce264f25 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -444,6 +444,7 @@ 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 opus_rc.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 368012992b..f12a54df50 100644
--- a/libavcodec/allcodecs.c
+++ b/libavcodec/allcodecs.c
@@ -449,7 +449,7 @@ 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 7b46a5116b..23c20891cd 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 @@ 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 @@ 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 ddc5fc2895..a2f5088aa9 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;
@@ -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 +374,74 @@ 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);
}
+/*
+ * Faster than libopus's search, operates entirely in the signed domain.
+ * Slightly worse/better depending on N, K and the input vector.
+ */
+static void celt_pvq_search(float *X, int *y, int K, int N)
+{
+ int i;
+ float res = 0.0f, y_norm = 0.0f, xy_norm = 0.0f;
+
+ for (i = 0; i < N; i++)
+ res += FFABS(X[i]);
+
+ res = K/res;
+
+ 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_den = 1.0f, max_num = 0.0f;
+ y_norm += 1.0f;
+
+ for (i = 0; i < N; i++) {
+ float xy_new = xy_norm + 1*phase*FFABS(X[i]);
+ float y_new = y_norm + 2*phase*FFABS(y[i]);
+ xy_new = xy_new * xy_new;
+ if ((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;
+ }
+}
+
+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 +451,7 @@ 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 +805,353 @@ 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_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;
+ 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..985b41ccc0
--- /dev/null
+++ b/libavcodec/opusenc.c
@@ -0,0 +1,1130 @@
+/*
+ * 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
+
+/* 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;
+} OpusEncOptions;
+
+typedef struct OpusEncContext {
+ AVClass *av_class;
+ OpusEncOptions options;
+ AVCodecContext *avctx;
+ 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 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->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 do the mdct */
+static void celt_frame_mdct(OpusEncContext *s, CeltFrame *f)
+{
+ int i, t, ch;
+ float *win = s->scratch;
+
+ /* 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 < f->blocks; t++) {
+ float *src2 = &b->samples[CELT_OVERLAP*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], b->coeffs + t, win, 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;
+ 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));
+
+ /* Alignment fucks me over */
+ //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_EPSILON;
+ 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 = 0;
+ 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);
+ av_freep(&s->frame);
+ av_freep(&s->rc);
+ 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*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 %li kbps, clipping to %li kbps\n",
+ avctx->bit_rate/1000, clipped_rate/1000);
+ avctx->bit_rate = clipped_rate;
+ }
+
+ /* Frame structs and range coder buffers */
+ s->frame = av_malloc(OPUS_MAX_FRAMES_PER_PACKET*sizeof(CeltFrame));
+ if (!s->frame)
+ return AVERROR(ENOMEM);
+ s->rc = av_malloc(OPUS_MAX_FRAMES_PER_PACKET*sizeof(OpusRangeCoder));
+ if (!s->rc)
+ return AVERROR(ENOMEM);
+
+ /* 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 (!(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))))
+ return AVERROR(ENOMEM);
+
+ /* 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));
+ if (!ff_bufqueue_peek(&s->bufqueue, 0))
+ return AVERROR(ENOMEM);
+
+ return 0;
+}
+
+#define OPUSENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
+static const AVOption opusenc_options[] = {
+ { "opus_delay", "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 | FF_CODEC_CAP_INIT_CLEANUP,
+ .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 },
+};
generated by cgit v1.2.3 (git 2.39.1) at 2024-06-30 06:01:49 +0000