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dfdf73eb1a
ffmpeg/libavcodec/ac3enc.c
Justin Ruggles dfdf73eb1a Split exponent processing into separate functions. 
Originally committed as revision 25991 to svn://svn.ffmpeg.org/ffmpeg/trunk
2010-12-14 14:53:02 +00:00

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/*
* The simplest AC-3 encoder
* Copyright (c) 2000 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* The simplest AC-3 encoder.
*/
//#define DEBUG
#include "libavcore/audioconvert.h"
#include "libavutil/crc.h"
#include "avcodec.h"
#include "put_bits.h"
#include "ac3.h"
#include "audioconvert.h"
#define MDCT_NBITS 9
#define MDCT_SAMPLES (1 << MDCT_NBITS)
/** Scale a float value by 2^bits and convert to an integer. */
#define SCALE_FLOAT(a, bits) lrintf((a) * (float)(1 << (bits)))
/** Scale a float value by 2^15, convert to an integer, and clip to int16_t range. */
#define FIX15(a) av_clip_int16(SCALE_FLOAT(a, 15))
/**
* Compex number.
* Used in fixed-point MDCT calculation.
*/
typedef struct IComplex {
int16_t re,im;
} IComplex;
/**
* AC-3 encoder private context.
*/
typedef struct AC3EncodeContext {
PutBitContext pb; ///< bitstream writer context
int bitstream_id; ///< bitstream id (bsid)
int bitstream_mode; ///< bitstream mode (bsmod)
int bit_rate; ///< target bit rate, in bits-per-second
int sample_rate; ///< sampling frequency, in Hz
int frame_size_min; ///< minimum frame size in case rounding is necessary
int frame_size; ///< current frame size in bytes
int frame_size_code; ///< frame size code (frmsizecod)
int bits_written; ///< bit count (used to avg. bitrate)
int samples_written; ///< sample count (used to avg. bitrate)
int fbw_channels; ///< number of full-bandwidth channels (nfchans)
int channels; ///< total number of channels (nchans)
int lfe_on; ///< indicates if there is an LFE channel (lfeon)
int lfe_channel; ///< channel index of the LFE channel
int channel_mode; ///< channel mode (acmod)
const uint8_t *channel_map; ///< channel map used to reorder channels
int bandwidth_code[AC3_MAX_CHANNELS]; ///< bandwidth code (0 to 60) (chbwcod)
int nb_coefs[AC3_MAX_CHANNELS];
/* bitrate allocation control */
int slow_gain_code; ///< slow gain code (sgaincod)
int slow_decay_code; ///< slow decay code (sdcycod)
int fast_decay_code; ///< fast decay code (fdcycod)
int db_per_bit_code; ///< dB/bit code (dbpbcod)
int floor_code; ///< floor code (floorcod)
AC3BitAllocParameters bit_alloc; ///< bit allocation parameters
int coarse_snr_offset; ///< coarse SNR offsets (csnroffst)
int fast_gain_code[AC3_MAX_CHANNELS]; ///< fast gain codes (signal-to-mask ratio) (fgaincod)
int fine_snr_offset[AC3_MAX_CHANNELS]; ///< fine SNR offsets (fsnroffst)
/* mantissa encoding */
int mant1_cnt, mant2_cnt, mant4_cnt; ///< mantissa counts for bap=1,2,4
int16_t last_samples[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]; ///< last 256 samples from previous frame
} AC3EncodeContext;
/** MDCT and FFT tables */
static int16_t costab[64];
static int16_t sintab[64];
static int16_t xcos1[128];
static int16_t xsin1[128];
/**
* Deinterleave input samples.
* Channels are reordered from FFmpeg's default order to AC-3 order.
*/
static void deinterleave_input_samples(AC3EncodeContext *s,
const int16_t *samples,
int16_t planar_samples[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE+AC3_FRAME_SIZE])
{
int ch, i;
/* deinterleave and remap input samples */
for (ch = 0; ch < s->channels; ch++) {
const int16_t *sptr;
int sinc;
/* copy last 256 samples of previous frame to the start of the current frame */
memcpy(&planar_samples[ch][0], s->last_samples[ch],
AC3_BLOCK_SIZE * sizeof(planar_samples[0][0]));
/* deinterleave */
sinc = s->channels;
sptr = samples + s->channel_map[ch];
for (i = AC3_BLOCK_SIZE; i < AC3_FRAME_SIZE+AC3_BLOCK_SIZE; i++) {
planar_samples[ch][i] = *sptr;
sptr += sinc;
}
/* save last 256 samples for next frame */
memcpy(s->last_samples[ch], &planar_samples[ch][6* AC3_BLOCK_SIZE],
AC3_BLOCK_SIZE * sizeof(planar_samples[0][0]));
}
}
/**
* Initialize FFT tables.
* @param ln log2(FFT size)
*/
static av_cold void fft_init(int ln)
{
int i, n, n2;
float alpha;
n = 1 << ln;
n2 = n >> 1;
for (i = 0; i < n2; i++) {
alpha = 2.0 * M_PI * i / n;
costab[i] = FIX15(cos(alpha));
sintab[i] = FIX15(sin(alpha));
}
}
/**
* Initialize MDCT tables.
* @param nbits log2(MDCT size)
*/
static av_cold void mdct_init(int nbits)
{
int i, n, n4;
n = 1 << nbits;
n4 = n >> 2;
fft_init(nbits - 2);
for (i = 0; i < n4; i++) {
float alpha = 2.0 * M_PI * (i + 1.0 / 8.0) / n;
xcos1[i] = FIX15(-cos(alpha));
xsin1[i] = FIX15(-sin(alpha));
}
}
/** Butterfly op */
#define BF(pre, pim, qre, qim, pre1, pim1, qre1, qim1) \
{ \
int ax, ay, bx, by; \
bx = pre1; \
by = pim1; \
ax = qre1; \
ay = qim1; \
pre = (bx + ax) >> 1; \
pim = (by + ay) >> 1; \
qre = (bx - ax) >> 1; \
qim = (by - ay) >> 1; \
}
/** Complex multiply */
#define CMUL(pre, pim, are, aim, bre, bim) \
{ \
pre = (MUL16(are, bre) - MUL16(aim, bim)) >> 15; \
pim = (MUL16(are, bim) + MUL16(bre, aim)) >> 15; \
}
/**
* Calculate a 2^n point complex FFT on 2^ln points.
* @param z complex input/output samples
* @param ln log2(FFT size)
*/
static void fft(IComplex *z, int ln)
{
int j, l, np, np2;
int nblocks, nloops;
register IComplex *p,*q;
int tmp_re, tmp_im;
np = 1 << ln;
/* reverse */
for (j = 0; j < np; j++) {
int k = av_reverse[j] >> (8 - ln);
if (k < j)
FFSWAP(IComplex, z[k], z[j]);
}
/* pass 0 */
p = &z[0];
j = np >> 1;
do {
BF(p[0].re, p[0].im, p[1].re, p[1].im,
p[0].re, p[0].im, p[1].re, p[1].im);
p += 2;
} while (--j);
/* pass 1 */
p = &z[0];
j = np >> 2;
do {
BF(p[0].re, p[0].im, p[2].re, p[2].im,
p[0].re, p[0].im, p[2].re, p[2].im);
BF(p[1].re, p[1].im, p[3].re, p[3].im,
p[1].re, p[1].im, p[3].im, -p[3].re);
p+=4;
} while (--j);
/* pass 2 .. ln-1 */
nblocks = np >> 3;
nloops = 1 << 2;
np2 = np >> 1;
do {
p = z;
q = z + nloops;
for (j = 0; j < nblocks; j++) {
BF(p->re, p->im, q->re, q->im,
p->re, p->im, q->re, q->im);
p++;
q++;
for(l = nblocks; l < np2; l += nblocks) {
CMUL(tmp_re, tmp_im, costab[l], -sintab[l], q->re, q->im);
BF(p->re, p->im, q->re, q->im,
p->re, p->im, tmp_re, tmp_im);
p++;
q++;
}
p += nloops;
q += nloops;
}
nblocks = nblocks >> 1;
nloops = nloops << 1;
} while (nblocks);
}
/**
* Calculate a 512-point MDCT
* @param out 256 output frequency coefficients
* @param in 512 windowed input audio samples
*/
static void mdct512(int32_t *out, int16_t *in)
{
int i, re, im, re1, im1;
int16_t rot[MDCT_SAMPLES];
IComplex x[MDCT_SAMPLES/4];
/* shift to simplify computations */
for (i = 0; i < MDCT_SAMPLES/4; i++)
rot[i] = -in[i + 3*MDCT_SAMPLES/4];
for (;i < MDCT_SAMPLES; i++)
rot[i] = in[i - MDCT_SAMPLES/4];
/* pre rotation */
for (i = 0; i < MDCT_SAMPLES/4; i++) {
re = ((int)rot[ 2*i] - (int)rot[MDCT_SAMPLES -1-2*i]) >> 1;
im = -((int)rot[MDCT_SAMPLES/2+2*i] - (int)rot[MDCT_SAMPLES/2-1-2*i]) >> 1;
CMUL(x[i].re, x[i].im, re, im, -xcos1[i], xsin1[i]);
}
fft(x, MDCT_NBITS - 2);
/* post rotation */
for (i = 0; i < MDCT_SAMPLES/4; i++) {
re = x[i].re;
im = x[i].im;
CMUL(re1, im1, re, im, xsin1[i], xcos1[i]);
out[ 2*i] = im1;
out[MDCT_SAMPLES/2-1-2*i] = re1;
}
}
/**
* Apply KBD window to input samples prior to MDCT.
*/
static void apply_window(int16_t *output, const int16_t *input,
const int16_t *window, int n)
{
int i;
int n2 = n >> 1;
for (i = 0; i < n2; i++) {
output[i] = MUL16(input[i], window[i]) >> 15;
output[n-i-1] = MUL16(input[n-i-1], window[i]) >> 15;
}
}
/**
* Calculate the log2() of the maximum absolute value in an array.
* @param tab input array
* @param n number of values in the array
* @return log2(max(abs(tab[])))
*/
static int log2_tab(int16_t *tab, int n)
{
int i, v;
v = 0;
for (i = 0; i < n; i++)
v |= abs(tab[i]);
return av_log2(v);
}
/**
* Left-shift each value in an array by a specified amount.
* @param tab input array
* @param n number of values in the array
* @param lshift left shift amount. a negative value means right shift.
*/
static void lshift_tab(int16_t *tab, int n, int lshift)
{
int i;
if (lshift > 0) {
for(i = 0; i < n; i++)
tab[i] <<= lshift;
} else if (lshift < 0) {
lshift = -lshift;
for (i = 0; i < n; i++)
tab[i] >>= lshift;
}
}
/**
* Normalize the input samples to use the maximum available precision.
* This assumes signed 16-bit input samples. Exponents are reduced by 9 to
* match the 24-bit internal precision for MDCT coefficients.
*
* @return exponent shift
*/
static int normalize_samples(AC3EncodeContext *s,
int16_t windowed_samples[AC3_WINDOW_SIZE])
{
int v = 14 - log2_tab(windowed_samples, AC3_WINDOW_SIZE);
v = FFMAX(0, v);
lshift_tab(windowed_samples, AC3_WINDOW_SIZE, v);
return v - 9;
}
/**
* Apply the MDCT to input samples to generate frequency coefficients.
* This applies the KBD window and normalizes the input to reduce precision
* loss due to fixed-point calculations.
*/
static void apply_mdct(AC3EncodeContext *s,
int16_t planar_samples[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE+AC3_FRAME_SIZE],
int8_t exp_shift[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
int32_t mdct_coef[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS])
{
int blk, ch;
int16_t windowed_samples[AC3_WINDOW_SIZE];
for (ch = 0; ch < s->channels; ch++) {
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
const int16_t *input_samples = &planar_samples[ch][blk * AC3_BLOCK_SIZE];
apply_window(windowed_samples, input_samples, ff_ac3_window, AC3_WINDOW_SIZE);
exp_shift[blk][ch] = normalize_samples(s, windowed_samples);
mdct512(mdct_coef[blk][ch], windowed_samples);
}
}
}
/**
* Extract exponents from the MDCT coefficients.
* This takes into account the normalization that was done to the input samples
* by adjusting the exponents by the exponent shift values.
*/
static void extract_exponents(AC3EncodeContext *s,
int32_t mdct_coef[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
int8_t exp_shift[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
uint8_t exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS])
{
int blk, ch, i;
/* extract exponents */
for (ch = 0; ch < s->channels; ch++) {
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
/* compute "exponents". We take into account the normalization there */
for (i = 0; i < AC3_MAX_COEFS; i++) {
int e;
int v = abs(mdct_coef[blk][ch][i]);
if (v == 0)
e = 24;
else {
e = 23 - av_log2(v) + exp_shift[blk][ch];
if (e >= 24) {
e = 24;
mdct_coef[blk][ch][i] = 0;
}
}
exp[blk][ch][i] = e;
}
}
}
}
/**
* Calculate the sum of absolute differences (SAD) between 2 sets of exponents.
*/
static int calc_exp_diff(uint8_t *exp1, uint8_t *exp2, int n)
{
int sum, i;
sum = 0;
for (i = 0; i < n; i++)
sum += abs(exp1[i] - exp2[i]);
return sum;
}
/**
* Exponent Difference Threshold.
* New exponents are sent if their SAD exceed this number.
*/
#define EXP_DIFF_THRESHOLD 1000
/**
* Calculate exponent strategies for all blocks in a single channel.
*/
static void compute_exp_strategy_ch(uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
uint8_t exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
int ch, int is_lfe)
{
int blk, blk1;
int exp_diff;
/* estimate if the exponent variation & decide if they should be
reused in the next frame */
exp_strategy[0][ch] = EXP_NEW;
for (blk = 1; blk < AC3_MAX_BLOCKS; blk++) {
exp_diff = calc_exp_diff(exp[blk][ch], exp[blk-1][ch], AC3_MAX_COEFS);
if (exp_diff > EXP_DIFF_THRESHOLD)
exp_strategy[blk][ch] = EXP_NEW;
else
exp_strategy[blk][ch] = EXP_REUSE;
}
if (is_lfe)
return;
/* now select the encoding strategy type : if exponents are often
recoded, we use a coarse encoding */
blk = 0;
while (blk < AC3_MAX_BLOCKS) {
blk1 = blk + 1;
while (blk1 < AC3_MAX_BLOCKS && exp_strategy[blk1][ch] == EXP_REUSE)
blk1++;
switch (blk1 - blk) {
case 1: exp_strategy[blk][ch] = EXP_D45; break;
case 2:
case 3: exp_strategy[blk][ch] = EXP_D25; break;
default: exp_strategy[blk][ch] = EXP_D15; break;
}
blk = blk1;
}
}
/**
* Calculate exponent strategies for all channels.
*/
static void compute_exp_strategy(AC3EncodeContext *s,
uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
uint8_t exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS])
{
int ch;
for (ch = 0; ch < s->channels; ch++) {
compute_exp_strategy_ch(exp_strategy, exp, ch, ch == s->lfe_channel);
}
}
/**
* Set each encoded exponent in a block to the minimum of itself and the
* exponent in the same frequency bin of a following block.
* exp[i] = min(exp[i], exp1[i]
*/
static void exponent_min(uint8_t exp[AC3_MAX_COEFS], uint8_t exp1[AC3_MAX_COEFS], int n)
{
int i;
for (i = 0; i < n; i++) {
if (exp1[i] < exp[i])
exp[i] = exp1[i];
}
}
/**
* Update the exponents so that they are the ones the decoder will decode.
* @return the number of bits used to encode the exponents.
*/
static int encode_exponents_blk_ch(uint8_t encoded_exp[AC3_MAX_COEFS],
uint8_t exp[AC3_MAX_COEFS],
int nb_exps, int exp_strategy)
{
int group_size, nb_groups, i, j, k, exp_min;
uint8_t exp1[AC3_MAX_COEFS];
group_size = exp_strategy + (exp_strategy == EXP_D45);
nb_groups = ((nb_exps + (group_size * 3) - 4) / (3 * group_size)) * 3;
/* for each group, compute the minimum exponent */
exp1[0] = exp[0]; /* DC exponent is handled separately */
k = 1;
for (i = 1; i <= nb_groups; i++) {
exp_min = exp[k];
assert(exp_min >= 0 && exp_min <= 24);
for (j = 1; j < group_size; j++) {
if (exp[k+j] < exp_min)
exp_min = exp[k+j];
}
exp1[i] = exp_min;
k += group_size;
}
/* constraint for DC exponent */
if (exp1[0] > 15)
exp1[0] = 15;
/* decrease the delta between each groups to within 2 so that they can be
differentially encoded */
for (i = 1; i <= nb_groups; i++)
exp1[i] = FFMIN(exp1[i], exp1[i-1] + 2);
for (i = nb_groups-1; i >= 0; i--)
exp1[i] = FFMIN(exp1[i], exp1[i+1] + 2);
/* now we have the exponent values the decoder will see */
encoded_exp[0] = exp1[0];
k = 1;
for (i = 1; i <= nb_groups; i++) {
for (j = 0; j < group_size; j++)
encoded_exp[k+j] = exp1[i];
k += group_size;
}
return 4 + (nb_groups / 3) * 7;
}
/**
* Encode exponents from original extracted form to what the decoder will see.
* This copies and groups exponents based on exponent strategy and reduces
* deltas between adjacent exponent groups so that they can be differentially
* encoded.
* @return bits needed to encode the exponents
*/
static int encode_exponents(AC3EncodeContext *s,
uint8_t exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
uint8_t encoded_exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS])
{
int blk, blk1, blk2, ch;
int frame_bits;
frame_bits = 0;
for (ch = 0; ch < s->channels; ch++) {
/* for the EXP_REUSE case we select the min of the exponents */
blk = 0;
while (blk < AC3_MAX_BLOCKS) {
blk1 = blk + 1;
while (blk1 < AC3_MAX_BLOCKS && exp_strategy[blk1][ch] == EXP_REUSE) {
exponent_min(exp[blk][ch], exp[blk1][ch], s->nb_coefs[ch]);
blk1++;
}
frame_bits += encode_exponents_blk_ch(encoded_exp[blk][ch],
exp[blk][ch], s->nb_coefs[ch],
exp_strategy[blk][ch]);
/* copy encoded exponents for reuse case */
for (blk2 = blk+1; blk2 < blk1; blk2++) {
memcpy(encoded_exp[blk2][ch], encoded_exp[blk][ch],
s->nb_coefs[ch] * sizeof(uint8_t));
}
blk = blk1;
}
}
return frame_bits;
}
/**
* Calculate final exponents from the supplied MDCT coefficients and exponent shift.
* Extract exponents from MDCT coefficients, calculate exponent strategies,
* and encode final exponents.
* @return bits needed to encode the exponents
*/
static int process_exponents(AC3EncodeContext *s,
int32_t mdct_coef[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
int8_t exp_shift[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
uint8_t exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
uint8_t encoded_exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS])
{
extract_exponents(s, mdct_coef, exp_shift, exp);
compute_exp_strategy(s, exp_strategy, exp);
return encode_exponents(s, exp, exp_strategy, encoded_exp);
}
/**
* Calculate the number of bits needed to encode a set of mantissas.
*/
static int compute_mantissa_size(AC3EncodeContext *s, uint8_t *m, int nb_coefs)
{
int bits, mant, i;
bits = 0;
for (i = 0; i < nb_coefs; i++) {
mant = m[i];
switch (mant) {
case 0:
/* nothing */
break;
case 1:
/* 3 mantissa in 5 bits */
if (s->mant1_cnt == 0)
bits += 5;
if (++s->mant1_cnt == 3)
s->mant1_cnt = 0;
break;
case 2:
/* 3 mantissa in 7 bits */
if (s->mant2_cnt == 0)
bits += 7;
if (++s->mant2_cnt == 3)
s->mant2_cnt = 0;
break;
case 3:
bits += 3;
break;
case 4:
/* 2 mantissa in 7 bits */
if (s->mant4_cnt == 0)
bits += 7;
if (++s->mant4_cnt == 2)
s->mant4_cnt = 0;
break;
case 14:
bits += 14;
break;
case 15:
bits += 16;
break;
default:
bits += mant - 1;
break;
}
}
return bits;
}
/**
* Calculate masking curve based on the final exponents.
* Also calculate the power spectral densities to use in future calculations.
*/
static void bit_alloc_masking(AC3EncodeContext *s,
uint8_t encoded_exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
int16_t psd[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
int16_t mask[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_CRITICAL_BANDS])
{
int blk, ch;
int16_t band_psd[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_CRITICAL_BANDS];
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
for (ch = 0; ch < s->channels; ch++) {
if(exp_strategy[blk][ch] == EXP_REUSE) {
memcpy(psd[blk][ch], psd[blk-1][ch], AC3_MAX_COEFS*sizeof(psd[0][0][0]));
memcpy(mask[blk][ch], mask[blk-1][ch], AC3_CRITICAL_BANDS*sizeof(mask[0][0][0]));
} else {
ff_ac3_bit_alloc_calc_psd(encoded_exp[blk][ch], 0,
s->nb_coefs[ch],
psd[blk][ch], band_psd[blk][ch]);
ff_ac3_bit_alloc_calc_mask(&s->bit_alloc, band_psd[blk][ch],
0, s->nb_coefs[ch],
ff_ac3_fast_gain_tab[s->fast_gain_code[ch]],
ch == s->lfe_channel,
DBA_NONE, 0, NULL, NULL, NULL,
mask[blk][ch]);
}
}
}
}
/**
* Run the bit allocation with a given SNR offset.
* This calculates the bit allocation pointers that will be used to determine
* the quantization of each mantissa.
* @return the number of remaining bits (positive or negative) if the given
* SNR offset is used to quantize the mantissas.
*/
static int bit_alloc(AC3EncodeContext *s,
int16_t mask[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_CRITICAL_BANDS],
int16_t psd[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
uint8_t bap[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
int frame_bits, int coarse_snr_offset, int fine_snr_offset)
{
int blk, ch;
int snr_offset;
snr_offset = (((coarse_snr_offset - 15) << 4) + fine_snr_offset) << 2;
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
s->mant1_cnt = 0;
s->mant2_cnt = 0;
s->mant4_cnt = 0;
for (ch = 0; ch < s->channels; ch++) {
ff_ac3_bit_alloc_calc_bap(mask[blk][ch], psd[blk][ch], 0,
s->nb_coefs[ch], snr_offset,
s->bit_alloc.floor, ff_ac3_bap_tab,
bap[blk][ch]);
frame_bits += compute_mantissa_size(s, bap[blk][ch], s->nb_coefs[ch]);
}
}
return 8 * s->frame_size - frame_bits;
}
#define SNR_INC1 4
/**
* Perform bit allocation search.
* Finds the SNR offset value that maximizes quality and fits in the specified
* frame size. Output is the SNR offset and a set of bit allocation pointers
* used to quantize the mantissas.
*/
static int compute_bit_allocation(AC3EncodeContext *s,
uint8_t bap[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
uint8_t encoded_exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS],
uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS],
int frame_bits)
{
int blk, ch;
int coarse_snr_offset, fine_snr_offset;
uint8_t bap1[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS];
int16_t psd[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS];
int16_t mask[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_CRITICAL_BANDS];
static const int frame_bits_inc[8] = { 0, 0, 2, 2, 2, 4, 2, 4 };
/* init default parameters */
s->slow_decay_code = 2;
s->fast_decay_code = 1;
s->slow_gain_code = 1;
s->db_per_bit_code = 2;
s->floor_code = 4;
for (ch = 0; ch < s->channels; ch++)
s->fast_gain_code[ch] = 4;
/* compute real values */
s->bit_alloc.slow_decay = ff_ac3_slow_decay_tab[s->slow_decay_code] >> s->bit_alloc.sr_shift;
s->bit_alloc.fast_decay = ff_ac3_fast_decay_tab[s->fast_decay_code] >> s->bit_alloc.sr_shift;
s->bit_alloc.slow_gain = ff_ac3_slow_gain_tab[s->slow_gain_code];
s->bit_alloc.db_per_bit = ff_ac3_db_per_bit_tab[s->db_per_bit_code];
s->bit_alloc.floor = ff_ac3_floor_tab[s->floor_code];
/* header size */
frame_bits += 65;
// if (s->channel_mode == 2)
// frame_bits += 2;
frame_bits += frame_bits_inc[s->channel_mode];
/* audio blocks */
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
frame_bits += s->fbw_channels * 2 + 2; /* blksw * c, dithflag * c, dynrnge, cplstre */
if (s->channel_mode == AC3_CHMODE_STEREO) {
frame_bits++; /* rematstr */
if (!blk)
frame_bits += 4;
}
frame_bits += 2 * s->fbw_channels; /* chexpstr[2] * c */
if (s->lfe_on)
frame_bits++; /* lfeexpstr */
for (ch = 0; ch < s->fbw_channels; ch++) {
if (exp_strategy[blk][ch] != EXP_REUSE)
frame_bits += 6 + 2; /* chbwcod[6], gainrng[2] */
}
frame_bits++; /* baie */
frame_bits++; /* snr */
frame_bits += 2; /* delta / skip */
}
frame_bits++; /* cplinu for block 0 */
/* bit alloc info */
/* sdcycod[2], fdcycod[2], sgaincod[2], dbpbcod[2], floorcod[3] */
/* csnroffset[6] */
/* (fsnoffset[4] + fgaincod[4]) * c */
frame_bits += 2*4 + 3 + 6 + s->channels * (4 + 3);
/* auxdatae, crcrsv */
frame_bits += 2;
/* CRC */
frame_bits += 16;
/* calculate psd and masking curve before doing bit allocation */
bit_alloc_masking(s, encoded_exp, exp_strategy, psd, mask);
/* now the big work begins : do the bit allocation. Modify the snr
offset until we can pack everything in the requested frame size */
coarse_snr_offset = s->coarse_snr_offset;
while (coarse_snr_offset >= 0 &&
bit_alloc(s, mask, psd, bap, frame_bits, coarse_snr_offset, 0) < 0)
coarse_snr_offset -= SNR_INC1;
if (coarse_snr_offset < 0) {
av_log(NULL, AV_LOG_ERROR, "Bit allocation failed. Try increasing the bitrate.\n");
return -1;
}
while (coarse_snr_offset + SNR_INC1 <= 63 &&
bit_alloc(s, mask, psd, bap1, frame_bits,
coarse_snr_offset + SNR_INC1, 0) >= 0) {
coarse_snr_offset += SNR_INC1;
memcpy(bap, bap1, sizeof(bap1));
}
while (coarse_snr_offset + 1 <= 63 &&
bit_alloc(s, mask, psd, bap1, frame_bits, coarse_snr_offset + 1, 0) >= 0) {
coarse_snr_offset++;
memcpy(bap, bap1, sizeof(bap1));
}
fine_snr_offset = 0;
while (fine_snr_offset + SNR_INC1 <= 15 &&
bit_alloc(s, mask, psd, bap1, frame_bits,
coarse_snr_offset, fine_snr_offset + SNR_INC1) >= 0) {
fine_snr_offset += SNR_INC1;
memcpy(bap, bap1, sizeof(bap1));
}
while (fine_snr_offset + 1 <= 15 &&
bit_alloc(s, mask, psd, bap1, frame_bits,
coarse_snr_offset, fine_snr_offset + 1) >= 0) {
fine_snr_offset++;
memcpy(bap, bap1, sizeof(bap1));
}
s->coarse_snr_offset = coarse_snr_offset;
for (ch = 0; ch < s->channels; ch++)
s->fine_snr_offset[ch] = fine_snr_offset;
return 0;
}
/**
* Write the AC-3 frame header to the output bitstream.
*/
static void output_frame_header(AC3EncodeContext *s, unsigned char *frame)
{
init_put_bits(&s->pb, frame, AC3_MAX_CODED_FRAME_SIZE);
put_bits(&s->pb, 16, 0x0b77); /* frame header */
put_bits(&s->pb, 16, 0); /* crc1: will be filled later */
put_bits(&s->pb, 2, s->bit_alloc.sr_code);
put_bits(&s->pb, 6, s->frame_size_code + (s->frame_size - s->frame_size_min) / 2);
put_bits(&s->pb, 5, s->bitstream_id);
put_bits(&s->pb, 3, s->bitstream_mode);
put_bits(&s->pb, 3, s->channel_mode);
if ((s->channel_mode & 0x01) && s->channel_mode != AC3_CHMODE_MONO)
put_bits(&s->pb, 2, 1); /* XXX -4.5 dB */
if (s->channel_mode & 0x04)
put_bits(&s->pb, 2, 1); /* XXX -6 dB */
if (s->channel_mode == AC3_CHMODE_STEREO)
put_bits(&s->pb, 2, 0); /* surround not indicated */
put_bits(&s->pb, 1, s->lfe_on); /* LFE */
put_bits(&s->pb, 5, 31); /* dialog norm: -31 db */
put_bits(&s->pb, 1, 0); /* no compression control word */
put_bits(&s->pb, 1, 0); /* no lang code */
put_bits(&s->pb, 1, 0); /* no audio production info */
put_bits(&s->pb, 1, 0); /* no copyright */
put_bits(&s->pb, 1, 1); /* original bitstream */
put_bits(&s->pb, 1, 0); /* no time code 1 */
put_bits(&s->pb, 1, 0); /* no time code 2 */
put_bits(&s->pb, 1, 0); /* no additional bit stream info */
}
/**
* Symmetric quantization on 'levels' levels.
*/
static inline int sym_quant(int c, int e, int levels)
{
int v;
if (c >= 0) {
v = (levels * (c << e)) >> 24;
v = (v + 1) >> 1;
v = (levels >> 1) + v;
} else {
v = (levels * ((-c) << e)) >> 24;
v = (v + 1) >> 1;
v = (levels >> 1) - v;
}
assert (v >= 0 && v < levels);
return v;
}
/**
* Asymmetric quantization on 2^qbits levels.
*/
static inline int asym_quant(int c, int e, int qbits)
{
int lshift, m, v;
lshift = e + qbits - 24;
if (lshift >= 0)
v = c << lshift;
else
v = c >> (-lshift);
/* rounding */
v = (v + 1) >> 1;
m = (1 << (qbits-1));
if (v >= m)
v = m - 1;
assert(v >= -m);
return v & ((1 << qbits)-1);
}
/**
* Write one audio block to the output bitstream.
*/
static void output_audio_block(AC3EncodeContext *s,
uint8_t exp_strategy[AC3_MAX_CHANNELS],
uint8_t encoded_exp[AC3_MAX_CHANNELS][AC3_MAX_COEFS],
uint8_t bap[AC3_MAX_CHANNELS][AC3_MAX_COEFS],
int32_t mdct_coef[AC3_MAX_CHANNELS][AC3_MAX_COEFS],
int8_t exp_shift[AC3_MAX_CHANNELS],
int block_num)
{
int ch, nb_groups, group_size, i, baie, rbnd;
uint8_t *p;
uint16_t qmant[AC3_MAX_CHANNELS][AC3_MAX_COEFS];
int exp0, exp1;
int mant1_cnt, mant2_cnt, mant4_cnt;
uint16_t *qmant1_ptr, *qmant2_ptr, *qmant4_ptr;
int delta0, delta1, delta2;
for (ch = 0; ch < s->fbw_channels; ch++)
put_bits(&s->pb, 1, 0); /* no block switching */
for (ch = 0; ch < s->fbw_channels; ch++)
put_bits(&s->pb, 1, 1); /* no dither */
put_bits(&s->pb, 1, 0); /* no dynamic range */
if (!block_num) {
put_bits(&s->pb, 1, 1); /* coupling strategy present */
put_bits(&s->pb, 1, 0); /* no coupling strategy */
} else {
put_bits(&s->pb, 1, 0); /* no new coupling strategy */
}
if (s->channel_mode == AC3_CHMODE_STEREO) {
if (!block_num) {
/* first block must define rematrixing (rematstr) */
put_bits(&s->pb, 1, 1);
/* dummy rematrixing rematflg(1:4)=0 */
for (rbnd = 0; rbnd < 4; rbnd++)
put_bits(&s->pb, 1, 0);
} else {
/* no matrixing (but should be used in the future) */
put_bits(&s->pb, 1, 0);
}
}
/* exponent strategy */
for (ch = 0; ch < s->fbw_channels; ch++)
put_bits(&s->pb, 2, exp_strategy[ch]);
if (s->lfe_on)
put_bits(&s->pb, 1, exp_strategy[s->lfe_channel]);
/* bandwidth */
for (ch = 0; ch < s->fbw_channels; ch++) {
if (exp_strategy[ch] != EXP_REUSE)
put_bits(&s->pb, 6, s->bandwidth_code[ch]);
}
/* exponents */
for (ch = 0; ch < s->channels; ch++) {
if (exp_strategy[ch] == EXP_REUSE)
continue;
group_size = exp_strategy[ch] + (exp_strategy[ch] == EXP_D45);
nb_groups = (s->nb_coefs[ch] + (group_size * 3) - 4) / (3 * group_size);
p = encoded_exp[ch];
/* first exponent */
exp1 = *p++;
put_bits(&s->pb, 4, exp1);
/* next ones are delta encoded */
for (i = 0; i < nb_groups; i++) {
/* merge three delta in one code */
exp0 = exp1;
exp1 = p[0];
p += group_size;
delta0 = exp1 - exp0 + 2;
exp0 = exp1;
exp1 = p[0];
p += group_size;
delta1 = exp1 - exp0 + 2;
exp0 = exp1;
exp1 = p[0];
p += group_size;
delta2 = exp1 - exp0 + 2;
put_bits(&s->pb, 7, ((delta0 * 5 + delta1) * 5) + delta2);
}
if (ch != s->lfe_channel)
put_bits(&s->pb, 2, 0); /* no gain range info */
}
/* bit allocation info */
baie = (block_num == 0);
put_bits(&s->pb, 1, baie);
if (baie) {
put_bits(&s->pb, 2, s->slow_decay_code);
put_bits(&s->pb, 2, s->fast_decay_code);
put_bits(&s->pb, 2, s->slow_gain_code);
put_bits(&s->pb, 2, s->db_per_bit_code);
put_bits(&s->pb, 3, s->floor_code);
}
/* snr offset */
put_bits(&s->pb, 1, baie);
if (baie) {
put_bits(&s->pb, 6, s->coarse_snr_offset);
for (ch = 0; ch < s->channels; ch++) {
put_bits(&s->pb, 4, s->fine_snr_offset[ch]);
put_bits(&s->pb, 3, s->fast_gain_code[ch]);
}
}
put_bits(&s->pb, 1, 0); /* no delta bit allocation */
put_bits(&s->pb, 1, 0); /* no data to skip */
/* mantissa encoding : we use two passes to handle the grouping. A
one pass method may be faster, but it would necessitate to
modify the output stream. */
/* first pass: quantize */
mant1_cnt = mant2_cnt = mant4_cnt = 0;
qmant1_ptr = qmant2_ptr = qmant4_ptr = NULL;
for (ch = 0; ch < s->channels; ch++) {
int b, c, e, v;
for (i = 0; i < s->nb_coefs[ch]; i++) {
c = mdct_coef[ch][i];
e = encoded_exp[ch][i] - exp_shift[ch];
b = bap[ch][i];
switch (b) {
case 0:
v = 0;
break;
case 1:
v = sym_quant(c, e, 3);
switch (mant1_cnt) {
case 0:
qmant1_ptr = &qmant[ch][i];
v = 9 * v;
mant1_cnt = 1;
break;
case 1:
*qmant1_ptr += 3 * v;
mant1_cnt = 2;
v = 128;
break;
default:
*qmant1_ptr += v;
mant1_cnt = 0;
v = 128;
break;
}
break;
case 2:
v = sym_quant(c, e, 5);
switch (mant2_cnt) {
case 0:
qmant2_ptr = &qmant[ch][i];
v = 25 * v;
mant2_cnt = 1;
break;
case 1:
*qmant2_ptr += 5 * v;
mant2_cnt = 2;
v = 128;
break;
default:
*qmant2_ptr += v;
mant2_cnt = 0;
v = 128;
break;
}
break;
case 3:
v = sym_quant(c, e, 7);
break;
case 4:
v = sym_quant(c, e, 11);
switch (mant4_cnt) {
case 0:
qmant4_ptr = &qmant[ch][i];
v = 11 * v;
mant4_cnt = 1;
break;
default:
*qmant4_ptr += v;
mant4_cnt = 0;
v = 128;
break;
}
break;
case 5:
v = sym_quant(c, e, 15);
break;
case 14:
v = asym_quant(c, e, 14);
break;
case 15:
v = asym_quant(c, e, 16);
break;
default:
v = asym_quant(c, e, b - 1);
break;
}
qmant[ch][i] = v;
}
}
/* second pass : output the values */
for (ch = 0; ch < s->channels; ch++) {
int b, q;
for (i = 0; i < s->nb_coefs[ch]; i++) {
q = qmant[ch][i];
b = bap[ch][i];
switch (b) {
case 0: break;
case 1: if (q != 128) put_bits(&s->pb, 5, q); break;
case 2: if (q != 128) put_bits(&s->pb, 7, q); break;
case 3: put_bits(&s->pb, 3, q); break;
case 4: if (q != 128) put_bits(&s->pb, 7, q); break;
case 14: put_bits(&s->pb, 14, q); break;
case 15: put_bits(&s->pb, 16, q); break;
default: put_bits(&s->pb, b-1, q); break;
}
}
}
}
/** CRC-16 Polynomial */
#define CRC16_POLY ((1 << 0) | (1 << 2) | (1 << 15) | (1 << 16))
static unsigned int mul_poly(unsigned int a, unsigned int b, unsigned int poly)
{
unsigned int c;
c = 0;
while (a) {
if (a & 1)
c ^= b;
a = a >> 1;
b = b << 1;
if (b & (1 << 16))
b ^= poly;
}
return c;
}
static unsigned int pow_poly(unsigned int a, unsigned int n, unsigned int poly)
{
unsigned int r;
r = 1;
while (n) {
if (n & 1)
r = mul_poly(r, a, poly);
a = mul_poly(a, a, poly);
n >>= 1;
}
return r;
}
/**
* Fill the end of the frame with 0's and compute the two CRCs.
*/
static void output_frame_end(AC3EncodeContext *s)
{
int frame_size, frame_size_58, pad_bytes, crc1, crc2, crc_inv;
uint8_t *frame;
frame_size = s->frame_size; /* frame size in words */
/* align to 8 bits */
flush_put_bits(&s->pb);
/* add zero bytes to reach the frame size */
frame = s->pb.buf;
pad_bytes = s->frame_size - (put_bits_ptr(&s->pb) - frame) - 2;
assert(pad_bytes >= 0);
if (pad_bytes > 0)
memset(put_bits_ptr(&s->pb), 0, pad_bytes);
/* Now we must compute both crcs : this is not so easy for crc1
because it is at the beginning of the data... */
frame_size_58 = ((frame_size >> 2) + (frame_size >> 4)) << 1;
crc1 = av_bswap16(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0,
frame + 4, frame_size_58 - 4));
/* XXX: could precompute crc_inv */
crc_inv = pow_poly((CRC16_POLY >> 1), (8 * frame_size_58) - 16, CRC16_POLY);
crc1 = mul_poly(crc_inv, crc1, CRC16_POLY);
AV_WB16(frame + 2, crc1);
crc2 = av_bswap16(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0,
frame + frame_size_58,
frame_size - frame_size_58 - 2));
AV_WB16(frame + frame_size - 2, crc2);
}
/**
* Encode a single AC-3 frame.
*/
static int ac3_encode_frame(AVCodecContext *avctx,
unsigned char *frame, int buf_size, void *data)
{
AC3EncodeContext *s = avctx->priv_data;
const int16_t *samples = data;
int blk;
int16_t planar_samples[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE+AC3_FRAME_SIZE];
int32_t mdct_coef[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS];
uint8_t exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS];
uint8_t exp_strategy[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS];
uint8_t encoded_exp[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS];
uint8_t bap[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS][AC3_MAX_COEFS];
int8_t exp_shift[AC3_MAX_BLOCKS][AC3_MAX_CHANNELS];
int frame_bits;
deinterleave_input_samples(s, samples, planar_samples);
apply_mdct(s, planar_samples, exp_shift, mdct_coef);
frame_bits = process_exponents(s, mdct_coef, exp_shift, exp, exp_strategy, encoded_exp);
/* adjust for fractional frame sizes */
while (s->bits_written >= s->bit_rate && s->samples_written >= s->sample_rate) {
s->bits_written -= s->bit_rate;
s->samples_written -= s->sample_rate;
}
s->frame_size = s->frame_size_min + 2 * (s->bits_written * s->sample_rate < s->samples_written * s->bit_rate);
s->bits_written += s->frame_size * 8;
s->samples_written += AC3_FRAME_SIZE;
compute_bit_allocation(s, bap, encoded_exp, exp_strategy, frame_bits);
/* everything is known... let's output the frame */
output_frame_header(s, frame);
for (blk = 0; blk < AC3_MAX_BLOCKS; blk++) {
output_audio_block(s, exp_strategy[blk], encoded_exp[blk],
bap[blk], mdct_coef[blk], exp_shift[blk], blk);
}
output_frame_end(s);
return s->frame_size;
}
/**
* Finalize encoding and free any memory allocated by the encoder.
*/
static av_cold int ac3_encode_close(AVCodecContext *avctx)
{
av_freep(&avctx->coded_frame);
return 0;
}
/**
* Set channel information during initialization.
*/
static av_cold int set_channel_info(AC3EncodeContext *s, int channels,
int64_t *channel_layout)
{
int ch_layout;
if (channels < 1 || channels > AC3_MAX_CHANNELS)
return AVERROR(EINVAL);
if ((uint64_t)*channel_layout > 0x7FF)
return AVERROR(EINVAL);
ch_layout = *channel_layout;
if (!ch_layout)
ch_layout = avcodec_guess_channel_layout(channels, CODEC_ID_AC3, NULL);
if (av_get_channel_layout_nb_channels(ch_layout) != channels)
return AVERROR(EINVAL);
s->lfe_on = !!(ch_layout & AV_CH_LOW_FREQUENCY);
s->channels = channels;
s->fbw_channels = channels - s->lfe_on;
s->lfe_channel = s->lfe_on ? s->fbw_channels : -1;
if (s->lfe_on)
ch_layout -= AV_CH_LOW_FREQUENCY;
switch (ch_layout) {
case AV_CH_LAYOUT_MONO: s->channel_mode = AC3_CHMODE_MONO; break;
case AV_CH_LAYOUT_STEREO: s->channel_mode = AC3_CHMODE_STEREO; break;
case AV_CH_LAYOUT_SURROUND: s->channel_mode = AC3_CHMODE_3F; break;
case AV_CH_LAYOUT_2_1: s->channel_mode = AC3_CHMODE_2F1R; break;
case AV_CH_LAYOUT_4POINT0: s->channel_mode = AC3_CHMODE_3F1R; break;
case AV_CH_LAYOUT_QUAD:
case AV_CH_LAYOUT_2_2: s->channel_mode = AC3_CHMODE_2F2R; break;
case AV_CH_LAYOUT_5POINT0:
case AV_CH_LAYOUT_5POINT0_BACK: s->channel_mode = AC3_CHMODE_3F2R; break;
default:
return AVERROR(EINVAL);
}
s->channel_map = ff_ac3_enc_channel_map[s->channel_mode][s->lfe_on];
*channel_layout = ch_layout;
if (s->lfe_on)
*channel_layout |= AV_CH_LOW_FREQUENCY;
return 0;
}
static av_cold int validate_options(AVCodecContext *avctx, AC3EncodeContext *s)
{
int i, ret;
/* validate channel layout */
if (!avctx->channel_layout) {
av_log(avctx, AV_LOG_WARNING, "No channel layout specified. The "
"encoder will guess the layout, but it "
"might be incorrect.\n");
}
ret = set_channel_info(s, avctx->channels, &avctx->channel_layout);
if (ret) {
av_log(avctx, AV_LOG_ERROR, "invalid channel layout\n");
return ret;
}
/* validate sample rate */
for (i = 0; i < 9; i++) {
if ((ff_ac3_sample_rate_tab[i / 3] >> (i % 3)) == avctx->sample_rate)
break;
}
if (i == 9) {
av_log(avctx, AV_LOG_ERROR, "invalid sample rate\n");
return AVERROR(EINVAL);
}
s->sample_rate = avctx->sample_rate;
s->bit_alloc.sr_shift = i % 3;
s->bit_alloc.sr_code = i / 3;
/* validate bit rate */
for (i = 0; i < 19; i++) {
if ((ff_ac3_bitrate_tab[i] >> s->bit_alloc.sr_shift)*1000 == avctx->bit_rate)
break;
}
if (i == 19) {
av_log(avctx, AV_LOG_ERROR, "invalid bit rate\n");
return AVERROR(EINVAL);
}
s->bit_rate = avctx->bit_rate;
s->frame_size_code = i << 1;
return 0;
}
/**
* Set bandwidth for all channels.
* The user can optionally supply a cutoff frequency. Otherwise an appropriate
* default value will be used.
*/
static av_cold void set_bandwidth(AC3EncodeContext *s, int cutoff)
{
int ch, bw_code;
if (cutoff) {
/* calculate bandwidth based on user-specified cutoff frequency */
int fbw_coeffs;
cutoff = av_clip(cutoff, 1, s->sample_rate >> 1);
fbw_coeffs = cutoff * 2 * AC3_MAX_COEFS / s->sample_rate;
bw_code = av_clip((fbw_coeffs - 73) / 3, 0, 60);
} else {
/* use default bandwidth setting */
/* XXX: should compute the bandwidth according to the frame
size, so that we avoid annoying high frequency artifacts */
bw_code = 50;
}
/* set number of coefficients for each channel */
for (ch = 0; ch < s->fbw_channels; ch++) {
s->bandwidth_code[ch] = bw_code;
s->nb_coefs[ch] = bw_code * 3 + 73;
}
if (s->lfe_on)
s->nb_coefs[s->lfe_channel] = 7; /* LFE channel always has 7 coefs */
}
/**
* Initialize the encoder.
*/
static av_cold int ac3_encode_init(AVCodecContext *avctx)
{
AC3EncodeContext *s = avctx->priv_data;
int ret;
avctx->frame_size = AC3_FRAME_SIZE;
ac3_common_init();
ret = validate_options(avctx, s);
if (ret)
return ret;
s->bitstream_id = 8 + s->bit_alloc.sr_shift;
s->bitstream_mode = 0; /* complete main audio service */
s->frame_size_min = 2 * ff_ac3_frame_size_tab[s->frame_size_code][s->bit_alloc.sr_code];
s->bits_written = 0;
s->samples_written = 0;
s->frame_size = s->frame_size_min;
set_bandwidth(s, avctx->cutoff);
/* initial snr offset */
s->coarse_snr_offset = 40;
mdct_init(9);
avctx->coded_frame= avcodec_alloc_frame();
avctx->coded_frame->key_frame= 1;
return 0;
}
#ifdef TEST
/*************************************************************************/
/* TEST */
#include "libavutil/lfg.h"
#define FN (MDCT_SAMPLES/4)
static void fft_test(AVLFG *lfg)
{
IComplex in[FN], in1[FN];
int k, n, i;
float sum_re, sum_im, a;
for (i = 0; i < FN; i++) {
in[i].re = av_lfg_get(lfg) % 65535 - 32767;
in[i].im = av_lfg_get(lfg) % 65535 - 32767;
in1[i] = in[i];
}
fft(in, 7);
/* do it by hand */
for (k = 0; k < FN; k++) {
sum_re = 0;
sum_im = 0;
for (n = 0; n < FN; n++) {
a = -2 * M_PI * (n * k) / FN;
sum_re += in1[n].re * cos(a) - in1[n].im * sin(a);
sum_im += in1[n].re * sin(a) + in1[n].im * cos(a);
}
av_log(NULL, AV_LOG_DEBUG, "%3d: %6d,%6d %6.0f,%6.0f\n",
k, in[k].re, in[k].im, sum_re / FN, sum_im / FN);
}
}
static void mdct_test(AVLFG *lfg)
{
int16_t input[MDCT_SAMPLES];
int32_t output[AC3_MAX_COEFS];
float input1[MDCT_SAMPLES];
float output1[AC3_MAX_COEFS];
float s, a, err, e, emax;
int i, k, n;
for (i = 0; i < MDCT_SAMPLES; i++) {
input[i] = (av_lfg_get(lfg) % 65535 - 32767) * 9 / 10;
input1[i] = input[i];
}
mdct512(output, input);
/* do it by hand */
for (k = 0; k < AC3_MAX_COEFS; k++) {
s = 0;
for (n = 0; n < MDCT_SAMPLES; n++) {
a = (2*M_PI*(2*n+1+MDCT_SAMPLES/2)*(2*k+1) / (4 * MDCT_SAMPLES));
s += input1[n] * cos(a);
}
output1[k] = -2 * s / MDCT_SAMPLES;
}
err = 0;
emax = 0;
for (i = 0; i < AC3_MAX_COEFS; i++) {
av_log(NULL, AV_LOG_DEBUG, "%3d: %7d %7.0f\n", i, output[i], output1[i]);
e = output[i] - output1[i];
if (e > emax)
emax = e;
err += e * e;
}
av_log(NULL, AV_LOG_DEBUG, "err2=%f emax=%f\n", err / AC3_MAX_COEFS, emax);
}
int main(void)
{
AVLFG lfg;
av_log_set_level(AV_LOG_DEBUG);
mdct_init(9);
fft_test(&lfg);
mdct_test(&lfg);
return 0;
}
#endif /* TEST */
AVCodec ac3_encoder = {
"ac3",
AVMEDIA_TYPE_AUDIO,
CODEC_ID_AC3,
sizeof(AC3EncodeContext),
ac3_encode_init,
ac3_encode_frame,
ac3_encode_close,
NULL,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE},
.long_name = NULL_IF_CONFIG_SMALL("ATSC A/52A (AC-3)"),
.channel_layouts = (const int64_t[]){
AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
AV_CH_LAYOUT_2_1,
AV_CH_LAYOUT_SURROUND,
AV_CH_LAYOUT_2_2,
AV_CH_LAYOUT_QUAD,
AV_CH_LAYOUT_4POINT0,
AV_CH_LAYOUT_5POINT0,
AV_CH_LAYOUT_5POINT0_BACK,
(AV_CH_LAYOUT_MONO | AV_CH_LOW_FREQUENCY),
(AV_CH_LAYOUT_STEREO | AV_CH_LOW_FREQUENCY),
(AV_CH_LAYOUT_2_1 | AV_CH_LOW_FREQUENCY),
(AV_CH_LAYOUT_SURROUND | AV_CH_LOW_FREQUENCY),
(AV_CH_LAYOUT_2_2 | AV_CH_LOW_FREQUENCY),
(AV_CH_LAYOUT_QUAD | AV_CH_LOW_FREQUENCY),
(AV_CH_LAYOUT_4POINT0 | AV_CH_LOW_FREQUENCY),
AV_CH_LAYOUT_5POINT1,
AV_CH_LAYOUT_5POINT1_BACK,
0 },
};
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