This marks the first time anyone has written an Opus encoder without
using any libopus code. The aim of the encoder is to prove how far
the format can go by writing the craziest encoder for it.
Right now the encoder's basic, it only supports CBR encoding, however
internally every single feature the CELT layer has is implemented
(except the pitch pre-filter which needs to work well with the rest of
whatever gets implemented). Psychoacoustic and rate control systems are
under development.
The encoder takes in frames of 120 samples and depending on the value of
opus_delay the plan is to use the extra buffered frames as lookahead.
Right now the encoder will pick the nearest largest legal frame size and
won't use the lookahead, but that'll change once there's a
psychoacoustic system.
Even though its a pretty basic encoder its already outperforming
any other native encoder FFmpeg has by a huge amount.
The PVQ search algorithm is faster and more accurate than libopus's
algorithm so the encoder's performance is close to that of libopus
at zero complexity (libopus has more SIMD).
The algorithm might be ported to libopus or other codecs using PVQ in
the future.
The encoder still has a few minor bugs, like desyncs at ultra low
bitrates (below 9kbps with 20ms frames).
Signed-off-by: Rostislav Pehlivanov <atomnu...@gmail.com>
---
configure | 1 +
libavcodec/Makefile | 1 +
libavcodec/allcodecs.c | 2 +-
libavcodec/opus_celt.h | 16 +
libavcodec/opus_pvq.c | 444 ++++++++++++++++++-
libavcodec/opus_pvq.h | 6 +
libavcodec/opusenc.c | 1128 ++++++++++++++++++++++++++++++++++++++++++++++++
7 files changed, 1589 insertions(+), 9 deletions(-)
create mode 100644 libavcodec/opusenc.c
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 c61fd7e259..0af1d52b4f 100644
--- a/libavcodec/Makefile
+++ b/libavcodec/Makefile
@@ -440,6 +440,7 @@ 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 4d9a98374b..6e48e02b4f 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 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 <atomnu...@gmail.com>
+ * Copyright (c) 2017 Rostislav Pehlivanov <atomnu...@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..ab52b0d376
--- /dev/null
+++ b/libavcodec/opusenc.c
@@ -0,0 +1,1128 @@
+/*
+ * Opus encoder
+ * Copyright (c) 2017 Rostislav Pehlivanov <atomnu...@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);
+ 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 },
+};
--
2.11.0.483.g087da7b7c
_______________________________________________
ffmpeg-devel mailing list
ffmpeg-devel@ffmpeg.org
http://ffmpeg.org/mailman/listinfo/ffmpeg-devel
