diff options
Diffstat (limited to 'libavcodec/mpegaudioenc_template.c')
-rw-r--r-- | libavcodec/mpegaudioenc_template.c | 785 |
1 files changed, 785 insertions, 0 deletions
diff --git a/libavcodec/mpegaudioenc_template.c b/libavcodec/mpegaudioenc_template.c new file mode 100644 index 0000000..1bf9471 --- /dev/null +++ b/libavcodec/mpegaudioenc_template.c @@ -0,0 +1,785 @@ +/* + * The simplest mpeg audio layer 2 encoder + * Copyright (c) 2000, 2001 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 mpeg audio layer 2 encoder. + */ + +#include "libavutil/channel_layout.h" + +#include "avcodec.h" +#include "internal.h" +#include "put_bits.h" + +#define FRAC_BITS 15 /* fractional bits for sb_samples and dct */ +#define WFRAC_BITS 14 /* fractional bits for window */ + +#include "mpegaudio.h" +#include "mpegaudiodsp.h" +#include "mpegaudiodata.h" +#include "mpegaudiotab.h" + +/* currently, cannot change these constants (need to modify + quantization stage) */ +#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS) + +#define SAMPLES_BUF_SIZE 4096 + +typedef struct MpegAudioContext { + PutBitContext pb; + int nb_channels; + int lsf; /* 1 if mpeg2 low bitrate selected */ + int bitrate_index; /* bit rate */ + int freq_index; + int frame_size; /* frame size, in bits, without padding */ + /* padding computation */ + int frame_frac, frame_frac_incr, do_padding; + short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */ + int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */ + int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT]; + unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */ + /* code to group 3 scale factors */ + unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT]; + int sblimit; /* number of used subbands */ + const unsigned char *alloc_table; + int16_t filter_bank[512]; + int scale_factor_table[64]; + unsigned char scale_diff_table[128]; +#if USE_FLOATS + float scale_factor_inv_table[64]; +#else + int8_t scale_factor_shift[64]; + unsigned short scale_factor_mult[64]; +#endif + unsigned short total_quant_bits[17]; /* total number of bits per allocation group */ +} MpegAudioContext; + +static av_cold int MPA_encode_init(AVCodecContext *avctx) +{ + MpegAudioContext *s = avctx->priv_data; + int freq = avctx->sample_rate; + int bitrate = avctx->bit_rate; + int channels = avctx->channels; + int i, v, table; + float a; + + if (channels <= 0 || channels > 2){ + av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels); + return AVERROR(EINVAL); + } + bitrate = bitrate / 1000; + s->nb_channels = channels; + avctx->frame_size = MPA_FRAME_SIZE; + avctx->initial_padding = 512 - 32 + 1; + + /* encoding freq */ + s->lsf = 0; + for(i=0;i<3;i++) { + if (avpriv_mpa_freq_tab[i] == freq) + break; + if ((avpriv_mpa_freq_tab[i] / 2) == freq) { + s->lsf = 1; + break; + } + } + if (i == 3){ + av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq); + return AVERROR(EINVAL); + } + s->freq_index = i; + + /* encoding bitrate & frequency */ + for(i=1;i<15;i++) { + if (avpriv_mpa_bitrate_tab[s->lsf][1][i] == bitrate) + break; + } + if (i == 15 && !avctx->bit_rate) { + i = 14; + bitrate = avpriv_mpa_bitrate_tab[s->lsf][1][i]; + avctx->bit_rate = bitrate * 1000; + } + if (i == 15){ + av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate); + return AVERROR(EINVAL); + } + s->bitrate_index = i; + + /* compute total header size & pad bit */ + + a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0); + s->frame_size = ((int)a) * 8; + + /* frame fractional size to compute padding */ + s->frame_frac = 0; + s->frame_frac_incr = (int)((a - floor(a)) * 65536.0); + + /* select the right allocation table */ + table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf); + + /* number of used subbands */ + s->sblimit = ff_mpa_sblimit_table[table]; + s->alloc_table = ff_mpa_alloc_tables[table]; + + av_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n", + bitrate, freq, s->frame_size, table, s->frame_frac_incr); + + for(i=0;i<s->nb_channels;i++) + s->samples_offset[i] = 0; + + for(i=0;i<257;i++) { + int v; + v = ff_mpa_enwindow[i]; +#if WFRAC_BITS != 16 + v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS); +#endif + s->filter_bank[i] = v; + if ((i & 63) != 0) + v = -v; + if (i != 0) + s->filter_bank[512 - i] = v; + } + + for(i=0;i<64;i++) { + v = (int)(exp2((3 - i) / 3.0) * (1 << 20)); + if (v <= 0) + v = 1; + s->scale_factor_table[i] = v; +#if USE_FLOATS + s->scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20); +#else +#define P 15 + s->scale_factor_shift[i] = 21 - P - (i / 3); + s->scale_factor_mult[i] = (1 << P) * exp2((i % 3) / 3.0); +#endif + } + for(i=0;i<128;i++) { + v = i - 64; + if (v <= -3) + v = 0; + else if (v < 0) + v = 1; + else if (v == 0) + v = 2; + else if (v < 3) + v = 3; + else + v = 4; + s->scale_diff_table[i] = v; + } + + for(i=0;i<17;i++) { + v = ff_mpa_quant_bits[i]; + if (v < 0) + v = -v; + else + v = v * 3; + s->total_quant_bits[i] = 12 * v; + } + + return 0; +} + +/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */ +static void idct32(int *out, int *tab) +{ + int i, j; + int *t, *t1, xr; + const int *xp = costab32; + + for(j=31;j>=3;j-=2) tab[j] += tab[j - 2]; + + t = tab + 30; + t1 = tab + 2; + do { + t[0] += t[-4]; + t[1] += t[1 - 4]; + t -= 4; + } while (t != t1); + + t = tab + 28; + t1 = tab + 4; + do { + t[0] += t[-8]; + t[1] += t[1-8]; + t[2] += t[2-8]; + t[3] += t[3-8]; + t -= 8; + } while (t != t1); + + t = tab; + t1 = tab + 32; + do { + t[ 3] = -t[ 3]; + t[ 6] = -t[ 6]; + + t[11] = -t[11]; + t[12] = -t[12]; + t[13] = -t[13]; + t[15] = -t[15]; + t += 16; + } while (t != t1); + + + t = tab; + t1 = tab + 8; + do { + int x1, x2, x3, x4; + + x3 = MUL(t[16], FIX(SQRT2*0.5)); + x4 = t[0] - x3; + x3 = t[0] + x3; + + x2 = MUL(-(t[24] + t[8]), FIX(SQRT2*0.5)); + x1 = MUL((t[8] - x2), xp[0]); + x2 = MUL((t[8] + x2), xp[1]); + + t[ 0] = x3 + x1; + t[ 8] = x4 - x2; + t[16] = x4 + x2; + t[24] = x3 - x1; + t++; + } while (t != t1); + + xp += 2; + t = tab; + t1 = tab + 4; + do { + xr = MUL(t[28],xp[0]); + t[28] = (t[0] - xr); + t[0] = (t[0] + xr); + + xr = MUL(t[4],xp[1]); + t[ 4] = (t[24] - xr); + t[24] = (t[24] + xr); + + xr = MUL(t[20],xp[2]); + t[20] = (t[8] - xr); + t[ 8] = (t[8] + xr); + + xr = MUL(t[12],xp[3]); + t[12] = (t[16] - xr); + t[16] = (t[16] + xr); + t++; + } while (t != t1); + xp += 4; + + for (i = 0; i < 4; i++) { + xr = MUL(tab[30-i*4],xp[0]); + tab[30-i*4] = (tab[i*4] - xr); + tab[ i*4] = (tab[i*4] + xr); + + xr = MUL(tab[ 2+i*4],xp[1]); + tab[ 2+i*4] = (tab[28-i*4] - xr); + tab[28-i*4] = (tab[28-i*4] + xr); + + xr = MUL(tab[31-i*4],xp[0]); + tab[31-i*4] = (tab[1+i*4] - xr); + tab[ 1+i*4] = (tab[1+i*4] + xr); + + xr = MUL(tab[ 3+i*4],xp[1]); + tab[ 3+i*4] = (tab[29-i*4] - xr); + tab[29-i*4] = (tab[29-i*4] + xr); + + xp += 2; + } + + t = tab + 30; + t1 = tab + 1; + do { + xr = MUL(t1[0], *xp); + t1[0] = (t[0] - xr); + t[0] = (t[0] + xr); + t -= 2; + t1 += 2; + xp++; + } while (t >= tab); + + for(i=0;i<32;i++) { + out[i] = tab[bitinv32[i]]; + } +} + +#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS) + +static void filter(MpegAudioContext *s, int ch, const short *samples, int incr) +{ + short *p, *q; + int sum, offset, i, j; + int tmp[64]; + int tmp1[32]; + int *out; + + offset = s->samples_offset[ch]; + out = &s->sb_samples[ch][0][0][0]; + for(j=0;j<36;j++) { + /* 32 samples at once */ + for(i=0;i<32;i++) { + s->samples_buf[ch][offset + (31 - i)] = samples[0]; + samples += incr; + } + + /* filter */ + p = s->samples_buf[ch] + offset; + q = s->filter_bank; + /* maxsum = 23169 */ + for(i=0;i<64;i++) { + sum = p[0*64] * q[0*64]; + sum += p[1*64] * q[1*64]; + sum += p[2*64] * q[2*64]; + sum += p[3*64] * q[3*64]; + sum += p[4*64] * q[4*64]; + sum += p[5*64] * q[5*64]; + sum += p[6*64] * q[6*64]; + sum += p[7*64] * q[7*64]; + tmp[i] = sum; + p++; + q++; + } + tmp1[0] = tmp[16] >> WSHIFT; + for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT; + for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT; + + idct32(out, tmp1); + + /* advance of 32 samples */ + offset -= 32; + out += 32; + /* handle the wrap around */ + if (offset < 0) { + memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32), + s->samples_buf[ch], (512 - 32) * 2); + offset = SAMPLES_BUF_SIZE - 512; + } + } + s->samples_offset[ch] = offset; +} + +static void compute_scale_factors(MpegAudioContext *s, + unsigned char scale_code[SBLIMIT], + unsigned char scale_factors[SBLIMIT][3], + int sb_samples[3][12][SBLIMIT], + int sblimit) +{ + int *p, vmax, v, n, i, j, k, code; + int index, d1, d2; + unsigned char *sf = &scale_factors[0][0]; + + for(j=0;j<sblimit;j++) { + for(i=0;i<3;i++) { + /* find the max absolute value */ + p = &sb_samples[i][0][j]; + vmax = abs(*p); + for(k=1;k<12;k++) { + p += SBLIMIT; + v = abs(*p); + if (v > vmax) + vmax = v; + } + /* compute the scale factor index using log 2 computations */ + if (vmax > 1) { + n = av_log2(vmax); + /* n is the position of the MSB of vmax. now + use at most 2 compares to find the index */ + index = (21 - n) * 3 - 3; + if (index >= 0) { + while (vmax <= s->scale_factor_table[index+1]) + index++; + } else { + index = 0; /* very unlikely case of overflow */ + } + } else { + index = 62; /* value 63 is not allowed */ + } + + av_dlog(NULL, "%2d:%d in=%x %x %d\n", + j, i, vmax, s->scale_factor_table[index], index); + /* store the scale factor */ + av_assert2(index >=0 && index <= 63); + sf[i] = index; + } + + /* compute the transmission factor : look if the scale factors + are close enough to each other */ + d1 = s->scale_diff_table[sf[0] - sf[1] + 64]; + d2 = s->scale_diff_table[sf[1] - sf[2] + 64]; + + /* handle the 25 cases */ + switch(d1 * 5 + d2) { + case 0*5+0: + case 0*5+4: + case 3*5+4: + case 4*5+0: + case 4*5+4: + code = 0; + break; + case 0*5+1: + case 0*5+2: + case 4*5+1: + case 4*5+2: + code = 3; + sf[2] = sf[1]; + break; + case 0*5+3: + case 4*5+3: + code = 3; + sf[1] = sf[2]; + break; + case 1*5+0: + case 1*5+4: + case 2*5+4: + code = 1; + sf[1] = sf[0]; + break; + case 1*5+1: + case 1*5+2: + case 2*5+0: + case 2*5+1: + case 2*5+2: + code = 2; + sf[1] = sf[2] = sf[0]; + break; + case 2*5+3: + case 3*5+3: + code = 2; + sf[0] = sf[1] = sf[2]; + break; + case 3*5+0: + case 3*5+1: + case 3*5+2: + code = 2; + sf[0] = sf[2] = sf[1]; + break; + case 1*5+3: + code = 2; + if (sf[0] > sf[2]) + sf[0] = sf[2]; + sf[1] = sf[2] = sf[0]; + break; + default: + av_assert2(0); //cannot happen + code = 0; /* kill warning */ + } + + av_dlog(NULL, "%d: %2d %2d %2d %d %d -> %d\n", j, + sf[0], sf[1], sf[2], d1, d2, code); + scale_code[j] = code; + sf += 3; + } +} + +/* The most important function : psycho acoustic module. In this + encoder there is basically none, so this is the worst you can do, + but also this is the simpler. */ +static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT]) +{ + int i; + + for(i=0;i<s->sblimit;i++) { + smr[i] = (int)(fixed_smr[i] * 10); + } +} + + +#define SB_NOTALLOCATED 0 +#define SB_ALLOCATED 1 +#define SB_NOMORE 2 + +/* Try to maximize the smr while using a number of bits inferior to + the frame size. I tried to make the code simpler, faster and + smaller than other encoders :-) */ +static void compute_bit_allocation(MpegAudioContext *s, + short smr1[MPA_MAX_CHANNELS][SBLIMIT], + unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], + int *padding) +{ + int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size; + int incr; + short smr[MPA_MAX_CHANNELS][SBLIMIT]; + unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT]; + const unsigned char *alloc; + + memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT); + memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT); + memset(bit_alloc, 0, s->nb_channels * SBLIMIT); + + /* compute frame size and padding */ + max_frame_size = s->frame_size; + s->frame_frac += s->frame_frac_incr; + if (s->frame_frac >= 65536) { + s->frame_frac -= 65536; + s->do_padding = 1; + max_frame_size += 8; + } else { + s->do_padding = 0; + } + + /* compute the header + bit alloc size */ + current_frame_size = 32; + alloc = s->alloc_table; + for(i=0;i<s->sblimit;i++) { + incr = alloc[0]; + current_frame_size += incr * s->nb_channels; + alloc += 1 << incr; + } + for(;;) { + /* look for the subband with the largest signal to mask ratio */ + max_sb = -1; + max_ch = -1; + max_smr = INT_MIN; + for(ch=0;ch<s->nb_channels;ch++) { + for(i=0;i<s->sblimit;i++) { + if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) { + max_smr = smr[ch][i]; + max_sb = i; + max_ch = ch; + } + } + } + if (max_sb < 0) + break; + av_dlog(NULL, "current=%d max=%d max_sb=%d max_ch=%d alloc=%d\n", + current_frame_size, max_frame_size, max_sb, max_ch, + bit_alloc[max_ch][max_sb]); + + /* find alloc table entry (XXX: not optimal, should use + pointer table) */ + alloc = s->alloc_table; + for(i=0;i<max_sb;i++) { + alloc += 1 << alloc[0]; + } + + if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) { + /* nothing was coded for this band: add the necessary bits */ + incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6; + incr += s->total_quant_bits[alloc[1]]; + } else { + /* increments bit allocation */ + b = bit_alloc[max_ch][max_sb]; + incr = s->total_quant_bits[alloc[b + 1]] - + s->total_quant_bits[alloc[b]]; + } + + if (current_frame_size + incr <= max_frame_size) { + /* can increase size */ + b = ++bit_alloc[max_ch][max_sb]; + current_frame_size += incr; + /* decrease smr by the resolution we added */ + smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]]; + /* max allocation size reached ? */ + if (b == ((1 << alloc[0]) - 1)) + subband_status[max_ch][max_sb] = SB_NOMORE; + else + subband_status[max_ch][max_sb] = SB_ALLOCATED; + } else { + /* cannot increase the size of this subband */ + subband_status[max_ch][max_sb] = SB_NOMORE; + } + } + *padding = max_frame_size - current_frame_size; + av_assert0(*padding >= 0); +} + +/* + * Output the mpeg audio layer 2 frame. Note how the code is small + * compared to other encoders :-) + */ +static void encode_frame(MpegAudioContext *s, + unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT], + int padding) +{ + int i, j, k, l, bit_alloc_bits, b, ch; + unsigned char *sf; + int q[3]; + PutBitContext *p = &s->pb; + + /* header */ + + put_bits(p, 12, 0xfff); + put_bits(p, 1, 1 - s->lsf); /* 1 = mpeg1 ID, 0 = mpeg2 lsf ID */ + put_bits(p, 2, 4-2); /* layer 2 */ + put_bits(p, 1, 1); /* no error protection */ + put_bits(p, 4, s->bitrate_index); + put_bits(p, 2, s->freq_index); + put_bits(p, 1, s->do_padding); /* use padding */ + put_bits(p, 1, 0); /* private_bit */ + put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO); + put_bits(p, 2, 0); /* mode_ext */ + put_bits(p, 1, 0); /* no copyright */ + put_bits(p, 1, 1); /* original */ + put_bits(p, 2, 0); /* no emphasis */ + + /* bit allocation */ + j = 0; + for(i=0;i<s->sblimit;i++) { + bit_alloc_bits = s->alloc_table[j]; + for(ch=0;ch<s->nb_channels;ch++) { + put_bits(p, bit_alloc_bits, bit_alloc[ch][i]); + } + j += 1 << bit_alloc_bits; + } + + /* scale codes */ + for(i=0;i<s->sblimit;i++) { + for(ch=0;ch<s->nb_channels;ch++) { + if (bit_alloc[ch][i]) + put_bits(p, 2, s->scale_code[ch][i]); + } + } + + /* scale factors */ + for(i=0;i<s->sblimit;i++) { + for(ch=0;ch<s->nb_channels;ch++) { + if (bit_alloc[ch][i]) { + sf = &s->scale_factors[ch][i][0]; + switch(s->scale_code[ch][i]) { + case 0: + put_bits(p, 6, sf[0]); + put_bits(p, 6, sf[1]); + put_bits(p, 6, sf[2]); + break; + case 3: + case 1: + put_bits(p, 6, sf[0]); + put_bits(p, 6, sf[2]); + break; + case 2: + put_bits(p, 6, sf[0]); + break; + } + } + } + } + + /* quantization & write sub band samples */ + + for(k=0;k<3;k++) { + for(l=0;l<12;l+=3) { + j = 0; + for(i=0;i<s->sblimit;i++) { + bit_alloc_bits = s->alloc_table[j]; + for(ch=0;ch<s->nb_channels;ch++) { + b = bit_alloc[ch][i]; + if (b) { + int qindex, steps, m, sample, bits; + /* we encode 3 sub band samples of the same sub band at a time */ + qindex = s->alloc_table[j+b]; + steps = ff_mpa_quant_steps[qindex]; + for(m=0;m<3;m++) { + sample = s->sb_samples[ch][k][l + m][i]; + /* divide by scale factor */ +#if USE_FLOATS + { + float a; + a = (float)sample * s->scale_factor_inv_table[s->scale_factors[ch][i][k]]; + q[m] = (int)((a + 1.0) * steps * 0.5); + } +#else + { + int q1, e, shift, mult; + e = s->scale_factors[ch][i][k]; + shift = s->scale_factor_shift[e]; + mult = s->scale_factor_mult[e]; + + /* normalize to P bits */ + if (shift < 0) + q1 = sample << (-shift); + else + q1 = sample >> shift; + q1 = (q1 * mult) >> P; + q1 += 1 << P; + if (q1 < 0) + q1 = 0; + q[m] = (q1 * (unsigned)steps) >> (P + 1); + } +#endif + if (q[m] >= steps) + q[m] = steps - 1; + av_assert2(q[m] >= 0 && q[m] < steps); + } + bits = ff_mpa_quant_bits[qindex]; + if (bits < 0) { + /* group the 3 values to save bits */ + put_bits(p, -bits, + q[0] + steps * (q[1] + steps * q[2])); + } else { + put_bits(p, bits, q[0]); + put_bits(p, bits, q[1]); + put_bits(p, bits, q[2]); + } + } + } + /* next subband in alloc table */ + j += 1 << bit_alloc_bits; + } + } + } + + /* padding */ + for(i=0;i<padding;i++) + put_bits(p, 1, 0); + + /* flush */ + flush_put_bits(p); +} + +static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, + const AVFrame *frame, int *got_packet_ptr) +{ + MpegAudioContext *s = avctx->priv_data; + const int16_t *samples = (const int16_t *)frame->data[0]; + short smr[MPA_MAX_CHANNELS][SBLIMIT]; + unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT]; + int padding, i, ret; + + for(i=0;i<s->nb_channels;i++) { + filter(s, i, samples + i, s->nb_channels); + } + + for(i=0;i<s->nb_channels;i++) { + compute_scale_factors(s, s->scale_code[i], s->scale_factors[i], + s->sb_samples[i], s->sblimit); + } + for(i=0;i<s->nb_channels;i++) { + psycho_acoustic_model(s, smr[i]); + } + compute_bit_allocation(s, smr, bit_alloc, &padding); + + if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE)) < 0) + return ret; + + init_put_bits(&s->pb, avpkt->data, avpkt->size); + + encode_frame(s, bit_alloc, padding); + + if (frame->pts != AV_NOPTS_VALUE) + avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->initial_padding); + + avpkt->size = put_bits_count(&s->pb) / 8; + *got_packet_ptr = 1; + return 0; +} + +static const AVCodecDefault mp2_defaults[] = { + { "b", "0" }, + { NULL }, +}; + |