int lfe_channel;
int bit_rate;
unsigned int sample_rate;
- unsigned int bsid;
+ unsigned int bitstream_id;
unsigned int frame_size_min; /* minimum frame size in case rounding is necessary */
unsigned int frame_size; /* current frame size in words */
unsigned int bits_written;
unsigned int samples_written;
int sr_shift;
- unsigned int frmsizecod;
+ unsigned int frame_size_code;
unsigned int sr_code; /* frequency */
- unsigned int acmod;
+ unsigned int channel_mode;
int lfe;
- unsigned int bsmod;
+ unsigned int bitstream_mode;
short last_samples[AC3_MAX_CHANNELS][256];
unsigned int chbwcod[AC3_MAX_CHANNELS];
int nb_coefs[AC3_MAX_CHANNELS];
static int16_t costab[64];
static int16_t sintab[64];
-static int16_t fft_rev[512];
static int16_t xcos1[128];
static int16_t xsin1[128];
/* new exponents are sent if their Norm 1 exceed this number */
#define EXP_DIFF_THRESHOLD 1000
-static void fft_init(int ln);
-
static inline int16_t fix15(float a)
{
int v;
static void fft_init(int ln)
{
- int i, j, m, n;
+ int i, n;
float alpha;
n = 1 << ln;
costab[i] = fix15(cos(alpha));
sintab[i] = fix15(sin(alpha));
}
-
- for(i=0;i<n;i++) {
- m=0;
- for(j=0;j<ln;j++) {
- m |= ((i >> j) & 1) << (ln-j-1);
- }
- fft_rev[i]=m;
- }
}
/* butter fly op */
/* reverse */
for(j=0;j<np;j++) {
- int k;
- IComplex tmp;
- k = fft_rev[j];
- if (k < j) {
- tmp = z[k];
- z[k] = z[j];
- z[j] = tmp;
- }
+ int k = ff_reverse[j] >> (8 - ln);
+ if (k < j)
+ FFSWAP(IComplex, z[k], z[j]);
}
/* pass 0 */
/* header size */
frame_bits += 65;
- // if (s->acmod == 2)
+ // if (s->channel_mode == 2)
// frame_bits += 2;
- frame_bits += frame_bits_inc[s->acmod];
+ frame_bits += frame_bits_inc[s->channel_mode];
/* audio blocks */
for(i=0;i<NB_BLOCKS;i++) {
frame_bits += s->nb_channels * 2 + 2; /* blksw * c, dithflag * c, dynrnge, cplstre */
- if (s->acmod == AC3_ACMOD_STEREO) {
+ if (s->channel_mode == AC3_CHMODE_STEREO) {
frame_bits++; /* rematstr */
if(i==0) frame_bits += 4;
}
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, -ab 384k for example!\n");
+ av_log(NULL, AV_LOG_ERROR, "Bit allocation failed. Try increasing the bitrate.\n");
return -1;
}
while ((coarse_snr_offset + SNR_INC1) <= 63 &&
AC3EncodeContext *s = avctx->priv_data;
int i, j, ch;
float alpha;
- static const uint8_t acmod_defs[6] = {
+ int bw_code;
+ static const uint8_t channel_mode_defs[6] = {
0x01, /* C */
0x02, /* L R */
0x03, /* L C R */
/* number of channels */
if (channels < 1 || channels > 6)
return -1;
- s->acmod = acmod_defs[channels - 1];
+ s->channel_mode = channel_mode_defs[channels - 1];
s->lfe = (channels == 6) ? 1 : 0;
s->nb_all_channels = channels;
s->nb_channels = channels > 5 ? 5 : channels;
s->sample_rate = freq;
s->sr_shift = i;
s->sr_code = j;
- s->bsid = 8 + s->sr_shift;
- s->bsmod = 0; /* complete main audio service */
+ s->bitstream_id = 8 + s->sr_shift;
+ s->bitstream_mode = 0; /* complete main audio service */
/* bitrate & frame size */
- bitrate /= 1000;
for(i=0;i<19;i++) {
- if ((ff_ac3_bitrate_tab[i] >> s->sr_shift) == bitrate)
+ if ((ff_ac3_bitrate_tab[i] >> s->sr_shift)*1000 == bitrate)
break;
}
if (i == 19)
return -1;
s->bit_rate = bitrate;
- s->frmsizecod = i << 1;
- s->frame_size_min = ff_ac3_frame_size_tab[s->frmsizecod][s->sr_code];
+ s->frame_size_code = i << 1;
+ s->frame_size_min = ff_ac3_frame_size_tab[s->frame_size_code][s->sr_code];
s->bits_written = 0;
s->samples_written = 0;
s->frame_size = s->frame_size_min;
/* bit allocation init */
- for(ch=0;ch<s->nb_channels;ch++) {
- /* bandwidth for each channel */
+ if(avctx->cutoff) {
+ /* calculate bandwidth based on user-specified cutoff frequency */
+ int cutoff = av_clip(avctx->cutoff, 1, s->sample_rate >> 1);
+ int fbw_coeffs = cutoff * 512 / 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 anoying high freq artefacts */
- s->chbwcod[ch] = 50; /* sample bandwidth as mpeg audio layer 2 table 0 */
- s->nb_coefs[ch] = ((s->chbwcod[ch] + 12) * 3) + 37;
+ bw_code = 50;
+ }
+ for(ch=0;ch<s->nb_channels;ch++) {
+ /* bandwidth for each channel */
+ s->chbwcod[ch] = bw_code;
+ s->nb_coefs[ch] = bw_code * 3 + 73;
}
if (s->lfe) {
s->nb_coefs[s->lfe_channel] = 7; /* fixed */
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->sr_code);
- put_bits(&s->pb, 6, s->frmsizecod + (s->frame_size - s->frame_size_min));
- put_bits(&s->pb, 5, s->bsid);
- put_bits(&s->pb, 3, s->bsmod);
- put_bits(&s->pb, 3, s->acmod);
- if ((s->acmod & 0x01) && s->acmod != AC3_ACMOD_MONO)
+ put_bits(&s->pb, 6, s->frame_size_code + (s->frame_size - s->frame_size_min));
+ 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->acmod & 0x04)
+ if (s->channel_mode & 0x04)
put_bits(&s->pb, 2, 1); /* XXX -6 dB */
- if (s->acmod == AC3_ACMOD_STEREO)
+ if (s->channel_mode == AC3_CHMODE_STEREO)
put_bits(&s->pb, 2, 0); /* surround not indicated */
put_bits(&s->pb, 1, s->lfe); /* LFE */
put_bits(&s->pb, 5, 31); /* dialog norm: -31 db */
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 addtional bit stream info */
+ put_bits(&s->pb, 1, 0); /* no additional bit stream info */
}
/* symetric quantization on 'levels' levels */
put_bits(&s->pb, 1, 0); /* no new coupling strategy */
}
- if (s->acmod == AC3_ACMOD_STEREO)
+ if (s->channel_mode == AC3_CHMODE_STEREO)
{
if(block_num==0)
{
/* 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 >> 1) + (frame_size >> 3);
- crc1 = bswap_16(av_crc(av_crc8005, 0, frame + 4, 2 * frame_size_58 - 4));
+ crc1 = bswap_16(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0,
+ frame + 4, 2 * frame_size_58 - 4));
/* XXX: could precompute crc_inv */
crc_inv = pow_poly((CRC16_POLY >> 1), (16 * frame_size_58) - 16, CRC16_POLY);
crc1 = mul_poly(crc_inv, crc1, CRC16_POLY);
AV_WB16(frame+2,crc1);
- crc2 = bswap_16(av_crc(av_crc8005, 0, frame + 2 * frame_size_58, (frame_size - frame_size_58) * 2 - 2));
+ crc2 = bswap_16(av_crc(av_crc_get_table(AV_CRC_16_ANSI), 0,
+ frame + 2 * frame_size_58,
+ (frame_size - frame_size_58) * 2 - 2));
AV_WB16(frame+2*frame_size-2,crc2);
// printf("n=%d frame_size=%d\n", n, frame_size);
}
/* adjust for fractional frame sizes */
- while(s->bits_written >= s->bit_rate*1000 && s->samples_written >= s->sample_rate) {
- s->bits_written -= s->bit_rate*1000;
+ 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 + (s->bits_written * s->sample_rate < s->samples_written * s->bit_rate*1000);
+ s->frame_size = s->frame_size_min + (s->bits_written * s->sample_rate < s->samples_written * s->bit_rate);
s->bits_written += s->frame_size * 16;
s->samples_written += AC3_FRAME_SIZE;
projects / ffmpeg / blobdiff