diff options
Diffstat (limited to 'libavcodec/mpegaudioenc.c')
-rw-r--r-- | libavcodec/mpegaudioenc.c | 767 |
1 files changed, 0 insertions, 767 deletions
diff --git a/libavcodec/mpegaudioenc.c b/libavcodec/mpegaudioenc.c deleted file mode 100644 index 4e074a5..0000000 --- a/libavcodec/mpegaudioenc.c +++ /dev/null @@ -1,767 +0,0 @@ -/* - * The simplest mpeg audio layer 2 encoder - * Copyright (c) 2000, 2001 Fabrice Bellard - * - * This file is part of Libav. - * - * Libav 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. - * - * Libav 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 Libav; 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]; - float scale_factor_inv_table[64]; - 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=0;i<15;i++) { - if (avpriv_mpa_bitrate_tab[s->lsf][1][i] == bitrate) - break; - } - 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)(pow(2.0, (3 - i) / 3.0) * (1 << 20)); - if (v <= 0) - v = 1; - s->scale_factor_table[i] = v; - s->scale_factor_inv_table[i] = pow(2.0, -(3 - i) / 3.0) / (float)(1 << 20); - } - 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 */ - assert(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: - assert(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; - assert(*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++) { - float a; - sample = s->sb_samples[ch][k][l + m][i]; - /* divide by scale factor */ - a = (float)sample * s->scale_factor_inv_table[s->scale_factors[ch][i][k]]; - q[m] = (int)((a + 1.0) * steps * 0.5); - if (q[m] >= steps) - q[m] = steps - 1; - assert(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_packet(avpkt, MPA_MAX_CODED_FRAME_SIZE))) { - av_log(avctx, AV_LOG_ERROR, "Error getting output packet\n"); - 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", "384000" }, - { NULL }, -}; - -AVCodec ff_mp2_encoder = { - .name = "mp2", - .long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"), - .type = AVMEDIA_TYPE_AUDIO, - .id = AV_CODEC_ID_MP2, - .priv_data_size = sizeof(MpegAudioContext), - .init = MPA_encode_init, - .encode2 = MPA_encode_frame, - .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16, - AV_SAMPLE_FMT_NONE }, - .supported_samplerates = (const int[]){ - 44100, 48000, 32000, 22050, 24000, 16000, 0 - }, - .channel_layouts = (const uint64_t[]){ AV_CH_LAYOUT_MONO, - AV_CH_LAYOUT_STEREO, - 0 }, - .defaults = mp2_defaults, -}; |