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/* |
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* ATRAC1 compatible decoder |
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* Copyright (c) 2009 Maxim Poliakovski |
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* Copyright (c) 2009 Benjamin Larsson |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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/** |
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* @file |
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* ATRAC1 compatible decoder. |
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* This decoder handles raw ATRAC1 data and probably SDDS data. |
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*/ |
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/* Many thanks to Tim Craig for all the help! */ |
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#include <math.h> |
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#include <stddef.h> |
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#include <stdio.h> |
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#include "libavutil/float_dsp.h" |
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#include "libavutil/mem_internal.h" |
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#include "avcodec.h" |
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#include "get_bits.h" |
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#include "fft.h" |
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#include "internal.h" |
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#include "sinewin.h" |
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#include "atrac.h" |
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#include "atrac1data.h" |
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#define AT1_MAX_BFU 52 ///< max number of block floating units in a sound unit |
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#define AT1_SU_SIZE 212 ///< number of bytes in a sound unit |
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#define AT1_SU_SAMPLES 512 ///< number of samples in a sound unit |
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#define AT1_FRAME_SIZE AT1_SU_SIZE * 2 |
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#define AT1_SU_MAX_BITS AT1_SU_SIZE * 8 |
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#define AT1_MAX_CHANNELS 2 |
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#define AT1_QMF_BANDS 3 |
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#define IDX_LOW_BAND 0 |
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#define IDX_MID_BAND 1 |
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#define IDX_HIGH_BAND 2 |
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/** |
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* Sound unit struct, one unit is used per channel |
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*/ |
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typedef struct AT1SUCtx { |
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int log2_block_count[AT1_QMF_BANDS]; ///< log2 number of blocks in a band |
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int num_bfus; ///< number of Block Floating Units |
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float* spectrum[2]; |
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DECLARE_ALIGNED(32, float, spec1)[AT1_SU_SAMPLES]; ///< mdct buffer |
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DECLARE_ALIGNED(32, float, spec2)[AT1_SU_SAMPLES]; ///< mdct buffer |
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DECLARE_ALIGNED(32, float, fst_qmf_delay)[46]; ///< delay line for the 1st stacked QMF filter |
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DECLARE_ALIGNED(32, float, snd_qmf_delay)[46]; ///< delay line for the 2nd stacked QMF filter |
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DECLARE_ALIGNED(32, float, last_qmf_delay)[256+39]; ///< delay line for the last stacked QMF filter |
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} AT1SUCtx; |
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/** |
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* The atrac1 context, holds all needed parameters for decoding |
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*/ |
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typedef struct AT1Ctx { |
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AT1SUCtx SUs[AT1_MAX_CHANNELS]; ///< channel sound unit |
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DECLARE_ALIGNED(32, float, spec)[AT1_SU_SAMPLES]; ///< the mdct spectrum buffer |
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DECLARE_ALIGNED(32, float, low)[256]; |
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DECLARE_ALIGNED(32, float, mid)[256]; |
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DECLARE_ALIGNED(32, float, high)[512]; |
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float* bands[3]; |
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FFTContext mdct_ctx[3]; |
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void (*vector_fmul_window)(float *dst, const float *src0, |
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const float *src1, const float *win, int len); |
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} AT1Ctx; |
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/** size of the transform in samples in the long mode for each QMF band */ |
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static const uint16_t samples_per_band[3] = {128, 128, 256}; |
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static const uint8_t mdct_long_nbits[3] = {7, 7, 8}; |
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static void at1_imdct(AT1Ctx *q, float *spec, float *out, int nbits, |
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int rev_spec) |
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{ |
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FFTContext* mdct_context = &q->mdct_ctx[nbits - 5 - (nbits > 6)]; |
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int transf_size = 1 << nbits; |
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✓✓ |
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if (rev_spec) { |
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int i; |
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✓✓ |
459048 |
for (i = 0; i < transf_size / 2; i++) |
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FFSWAP(float, spec[i], spec[transf_size - 1 - i]); |
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} |
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mdct_context->imdct_half(mdct_context, out, spec); |
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} |
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static int at1_imdct_block(AT1SUCtx* su, AT1Ctx *q) |
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{ |
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int band_num, band_samples, log2_block_count, nbits, num_blocks, block_size; |
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unsigned int start_pos, ref_pos = 0, pos = 0; |
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✓✓ |
9464 |
for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) { |
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float *prev_buf; |
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int j; |
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band_samples = samples_per_band[band_num]; |
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log2_block_count = su->log2_block_count[band_num]; |
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/* number of mdct blocks in the current QMF band: 1 - for long mode */ |
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/* 4 for short mode(low/middle bands) and 8 for short mode(high band)*/ |
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num_blocks = 1 << log2_block_count; |
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✓✓ |
7098 |
if (num_blocks == 1) { |
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/* mdct block size in samples: 128 (long mode, low & mid bands), */ |
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/* 256 (long mode, high band) and 32 (short mode, all bands) */ |
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block_size = band_samples >> log2_block_count; |
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/* calc transform size in bits according to the block_size_mode */ |
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nbits = mdct_long_nbits[band_num] - log2_block_count; |
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✓✗✓✓ ✗✓ |
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if (nbits != 5 && nbits != 7 && nbits != 8) |
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return AVERROR_INVALIDDATA; |
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} else { |
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block_size = 32; |
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nbits = 5; |
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} |
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start_pos = 0; |
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prev_buf = &su->spectrum[1][ref_pos + band_samples - 16]; |
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✓✓ |
14255 |
for (j=0; j < num_blocks; j++) { |
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at1_imdct(q, &q->spec[pos], &su->spectrum[0][ref_pos + start_pos], nbits, band_num); |
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/* overlap and window */ |
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q->vector_fmul_window(&q->bands[band_num][start_pos], prev_buf, |
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&su->spectrum[0][ref_pos + start_pos], ff_sine_32, 16); |
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prev_buf = &su->spectrum[0][ref_pos+start_pos + 16]; |
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start_pos += block_size; |
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pos += block_size; |
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} |
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✓✓ |
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if (num_blocks == 1) |
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memcpy(q->bands[band_num] + 32, &su->spectrum[0][ref_pos + 16], 240 * sizeof(float)); |
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ref_pos += band_samples; |
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} |
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/* Swap buffers so the mdct overlap works */ |
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FFSWAP(float*, su->spectrum[0], su->spectrum[1]); |
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return 0; |
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} |
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/** |
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* Parse the block size mode byte |
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*/ |
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static int at1_parse_bsm(GetBitContext* gb, int log2_block_cnt[AT1_QMF_BANDS]) |
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{ |
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int log2_block_count_tmp, i; |
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✓✓ |
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for (i = 0; i < 2; i++) { |
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/* low and mid band */ |
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log2_block_count_tmp = get_bits(gb, 2); |
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✗✓ |
4732 |
if (log2_block_count_tmp & 1) |
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return AVERROR_INVALIDDATA; |
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log2_block_cnt[i] = 2 - log2_block_count_tmp; |
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} |
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/* high band */ |
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log2_block_count_tmp = get_bits(gb, 2); |
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✓✓✗✓
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if (log2_block_count_tmp != 0 && log2_block_count_tmp != 3) |
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return AVERROR_INVALIDDATA; |
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log2_block_cnt[IDX_HIGH_BAND] = 3 - log2_block_count_tmp; |
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skip_bits(gb, 2); |
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return 0; |
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} |
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static int at1_unpack_dequant(GetBitContext* gb, AT1SUCtx* su, |
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float spec[AT1_SU_SAMPLES]) |
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{ |
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int bits_used, band_num, bfu_num, i; |
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uint8_t idwls[AT1_MAX_BFU]; ///< the word length indexes for each BFU |
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uint8_t idsfs[AT1_MAX_BFU]; ///< the scalefactor indexes for each BFU |
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/* parse the info byte (2nd byte) telling how much BFUs were coded */ |
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su->num_bfus = bfu_amount_tab1[get_bits(gb, 3)]; |
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/* calc number of consumed bits: |
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num_BFUs * (idwl(4bits) + idsf(6bits)) + log2_block_count(8bits) + info_byte(8bits) |
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+ info_byte_copy(8bits) + log2_block_count_copy(8bits) */ |
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bits_used = su->num_bfus * 10 + 32 + |
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bfu_amount_tab2[get_bits(gb, 2)] + |
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(bfu_amount_tab3[get_bits(gb, 3)] << 1); |
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/* get word length index (idwl) for each BFU */ |
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✓✓ |
107674 |
for (i = 0; i < su->num_bfus; i++) |
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105308 |
idwls[i] = get_bits(gb, 4); |
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/* get scalefactor index (idsf) for each BFU */ |
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✓✓ |
107674 |
for (i = 0; i < su->num_bfus; i++) |
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105308 |
idsfs[i] = get_bits(gb, 6); |
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/* zero idwl/idsf for empty BFUs */ |
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✓✓ |
20090 |
for (i = su->num_bfus; i < AT1_MAX_BFU; i++) |
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17724 |
idwls[i] = idsfs[i] = 0; |
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/* read in the spectral data and reconstruct MDCT spectrum of this channel */ |
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✓✓ |
9464 |
for (band_num = 0; band_num < AT1_QMF_BANDS; band_num++) { |
224 |
✓✓ |
130130 |
for (bfu_num = bfu_bands_t[band_num]; bfu_num < bfu_bands_t[band_num+1]; bfu_num++) { |
225 |
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int pos; |
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123032 |
int num_specs = specs_per_bfu[bfu_num]; |
228 |
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123032 |
int word_len = !!idwls[bfu_num] + idwls[bfu_num]; |
229 |
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123032 |
float scale_factor = ff_atrac_sf_table[idsfs[bfu_num]]; |
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123032 |
bits_used += word_len * num_specs; /* add number of bits consumed by current BFU */ |
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232 |
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/* check for bitstream overflow */ |
233 |
✗✓ |
123032 |
if (bits_used > AT1_SU_MAX_BITS) |
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return AVERROR_INVALIDDATA; |
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236 |
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/* get the position of the 1st spec according to the block size mode */ |
237 |
✓✓ |
123032 |
pos = su->log2_block_count[band_num] ? bfu_start_short[bfu_num] : bfu_start_long[bfu_num]; |
238 |
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239 |
✓✓ |
123032 |
if (word_len) { |
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57252 |
float max_quant = 1.0 / (float)((1 << (word_len - 1)) - 1); |
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242 |
✓✓ |
555280 |
for (i = 0; i < num_specs; i++) { |
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/* read in a quantized spec and convert it to |
244 |
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* signed int and then inverse quantization |
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*/ |
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498028 |
spec[pos+i] = get_sbits(gb, word_len) * scale_factor * max_quant; |
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} |
248 |
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} else { /* word_len = 0 -> empty BFU, zero all specs in the empty BFU */ |
249 |
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65780 |
memset(&spec[pos], 0, num_specs * sizeof(float)); |
250 |
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} |
251 |
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} |
252 |
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} |
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2366 |
return 0; |
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} |
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2366 |
static void at1_subband_synthesis(AT1Ctx *q, AT1SUCtx* su, float *pOut) |
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{ |
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float temp[256]; |
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float iqmf_temp[512 + 46]; |
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263 |
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/* combine low and middle bands */ |
264 |
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2366 |
ff_atrac_iqmf(q->bands[0], q->bands[1], 128, temp, su->fst_qmf_delay, iqmf_temp); |
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266 |
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/* delay the signal of the high band by 39 samples */ |
267 |
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2366 |
memcpy( su->last_qmf_delay, &su->last_qmf_delay[256], sizeof(float) * 39); |
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2366 |
memcpy(&su->last_qmf_delay[39], q->bands[2], sizeof(float) * 256); |
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/* combine (low + middle) and high bands */ |
271 |
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2366 |
ff_atrac_iqmf(temp, su->last_qmf_delay, 256, pOut, su->snd_qmf_delay, iqmf_temp); |
272 |
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2366 |
} |
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1184 |
static int atrac1_decode_frame(AVCodecContext *avctx, void *data, |
276 |
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int *got_frame_ptr, AVPacket *avpkt) |
277 |
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{ |
278 |
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1184 |
AVFrame *frame = data; |
279 |
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1184 |
const uint8_t *buf = avpkt->data; |
280 |
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1184 |
int buf_size = avpkt->size; |
281 |
|
1184 |
AT1Ctx *q = avctx->priv_data; |
282 |
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int ch, ret; |
283 |
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GetBitContext gb; |
284 |
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285 |
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|
286 |
✓✓ |
1184 |
if (buf_size < 212 * avctx->channels) { |
287 |
|
1 |
av_log(avctx, AV_LOG_ERROR, "Not enough data to decode!\n"); |
288 |
|
1 |
return AVERROR_INVALIDDATA; |
289 |
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} |
290 |
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291 |
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/* get output buffer */ |
292 |
|
1183 |
frame->nb_samples = AT1_SU_SAMPLES; |
293 |
✗✓ |
1183 |
if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) |
294 |
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return ret; |
295 |
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296 |
✓✓ |
3549 |
for (ch = 0; ch < avctx->channels; ch++) { |
297 |
|
2366 |
AT1SUCtx* su = &q->SUs[ch]; |
298 |
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299 |
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2366 |
init_get_bits(&gb, &buf[212 * ch], 212 * 8); |
300 |
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301 |
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/* parse block_size_mode, 1st byte */ |
302 |
|
2366 |
ret = at1_parse_bsm(&gb, su->log2_block_count); |
303 |
✗✓ |
2366 |
if (ret < 0) |
304 |
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return ret; |
305 |
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306 |
|
2366 |
ret = at1_unpack_dequant(&gb, su, q->spec); |
307 |
✗✓ |
2366 |
if (ret < 0) |
308 |
|
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return ret; |
309 |
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310 |
|
2366 |
ret = at1_imdct_block(su, q); |
311 |
✗✓ |
2366 |
if (ret < 0) |
312 |
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return ret; |
313 |
|
2366 |
at1_subband_synthesis(q, su, (float *)frame->extended_data[ch]); |
314 |
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} |
315 |
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316 |
|
1183 |
*got_frame_ptr = 1; |
317 |
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318 |
|
1183 |
return avctx->block_align; |
319 |
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} |
320 |
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321 |
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322 |
|
5 |
static av_cold int atrac1_decode_end(AVCodecContext * avctx) |
323 |
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{ |
324 |
|
5 |
AT1Ctx *q = avctx->priv_data; |
325 |
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326 |
|
5 |
ff_mdct_end(&q->mdct_ctx[0]); |
327 |
|
5 |
ff_mdct_end(&q->mdct_ctx[1]); |
328 |
|
5 |
ff_mdct_end(&q->mdct_ctx[2]); |
329 |
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330 |
|
5 |
return 0; |
331 |
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} |
332 |
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333 |
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334 |
|
5 |
static av_cold int atrac1_decode_init(AVCodecContext *avctx) |
335 |
|
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{ |
336 |
|
5 |
AT1Ctx *q = avctx->priv_data; |
337 |
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AVFloatDSPContext *fdsp; |
338 |
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int ret; |
339 |
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340 |
|
5 |
avctx->sample_fmt = AV_SAMPLE_FMT_FLTP; |
341 |
|
|
|
342 |
✓✗✗✓
|
5 |
if (avctx->channels < 1 || avctx->channels > AT1_MAX_CHANNELS) { |
343 |
|
|
av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", |
344 |
|
|
avctx->channels); |
345 |
|
|
return AVERROR(EINVAL); |
346 |
|
|
} |
347 |
|
|
|
348 |
✗✓ |
5 |
if (avctx->block_align <= 0) { |
349 |
|
|
av_log(avctx, AV_LOG_ERROR, "Unsupported block align."); |
350 |
|
|
return AVERROR_PATCHWELCOME; |
351 |
|
|
} |
352 |
|
|
|
353 |
|
|
/* Init the mdct transforms */ |
354 |
✓✗✓✗
|
10 |
if ((ret = ff_mdct_init(&q->mdct_ctx[0], 6, 1, -1.0/ (1 << 15))) || |
355 |
✗✓ |
10 |
(ret = ff_mdct_init(&q->mdct_ctx[1], 8, 1, -1.0/ (1 << 15))) || |
356 |
|
5 |
(ret = ff_mdct_init(&q->mdct_ctx[2], 9, 1, -1.0/ (1 << 15)))) { |
357 |
|
|
av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n"); |
358 |
|
|
return ret; |
359 |
|
|
} |
360 |
|
|
|
361 |
|
5 |
ff_init_ff_sine_windows(5); |
362 |
|
|
|
363 |
|
5 |
ff_atrac_generate_tables(); |
364 |
|
|
|
365 |
|
5 |
fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT); |
366 |
✗✓ |
5 |
if (!fdsp) |
367 |
|
|
return AVERROR(ENOMEM); |
368 |
|
5 |
q->vector_fmul_window = fdsp->vector_fmul_window; |
369 |
|
5 |
av_free(fdsp); |
370 |
|
|
|
371 |
|
5 |
q->bands[0] = q->low; |
372 |
|
5 |
q->bands[1] = q->mid; |
373 |
|
5 |
q->bands[2] = q->high; |
374 |
|
|
|
375 |
|
|
/* Prepare the mdct overlap buffers */ |
376 |
|
5 |
q->SUs[0].spectrum[0] = q->SUs[0].spec1; |
377 |
|
5 |
q->SUs[0].spectrum[1] = q->SUs[0].spec2; |
378 |
|
5 |
q->SUs[1].spectrum[0] = q->SUs[1].spec1; |
379 |
|
5 |
q->SUs[1].spectrum[1] = q->SUs[1].spec2; |
380 |
|
|
|
381 |
|
5 |
return 0; |
382 |
|
|
} |
383 |
|
|
|
384 |
|
|
|
385 |
|
|
AVCodec ff_atrac1_decoder = { |
386 |
|
|
.name = "atrac1", |
387 |
|
|
.long_name = NULL_IF_CONFIG_SMALL("ATRAC1 (Adaptive TRansform Acoustic Coding)"), |
388 |
|
|
.type = AVMEDIA_TYPE_AUDIO, |
389 |
|
|
.id = AV_CODEC_ID_ATRAC1, |
390 |
|
|
.priv_data_size = sizeof(AT1Ctx), |
391 |
|
|
.init = atrac1_decode_init, |
392 |
|
|
.close = atrac1_decode_end, |
393 |
|
|
.decode = atrac1_decode_frame, |
394 |
|
|
.capabilities = AV_CODEC_CAP_DR1, |
395 |
|
|
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP, |
396 |
|
|
AV_SAMPLE_FMT_NONE }, |
397 |
|
|
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP, |
398 |
|
|
}; |