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https://github.com/FFmpeg/FFmpeg.git
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aacdec: move LATM decode functions into a separate file
This commit is contained in:
Lynne
parent
e93793bf3c
commit
ce740618d1
350
libavcodec/aac/aacdec_latm.h
Normal file
350
libavcodec/aac/aacdec_latm.h
Normal file
@ -0,0 +1,350 @@
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/*
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* AAC decoder
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* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
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* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
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* Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
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*
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* AAC LATM decoder
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* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
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* Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
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*
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* AAC decoder fixed-point implementation
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* Copyright (c) 2013
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* MIPS Technologies, Inc., California.
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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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#ifndef AVCODEC_AAC_AACDEC_LATM_H
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#define AVCODEC_AAC_AACDEC_LATM_H
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#define LOAS_SYNC_WORD 0x2b7 ///< 11 bits LOAS sync word
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struct LATMContext {
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AACDecContext aac_ctx; ///< containing AACContext
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int initialized; ///< initialized after a valid extradata was seen
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// parser data
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int audio_mux_version_A; ///< LATM syntax version
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int frame_length_type; ///< 0/1 variable/fixed frame length
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int frame_length; ///< frame length for fixed frame length
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};
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static inline uint32_t latm_get_value(GetBitContext *b)
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{
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int length = get_bits(b, 2);
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return get_bits_long(b, (length+1)*8);
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}
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static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
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GetBitContext *gb, int asclen)
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{
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AACDecContext *ac = &latmctx->aac_ctx;
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AVCodecContext *avctx = ac->avctx;
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MPEG4AudioConfig m4ac = { 0 };
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GetBitContext gbc;
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int config_start_bit = get_bits_count(gb);
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int sync_extension = 0;
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int bits_consumed, esize, i;
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if (asclen > 0) {
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sync_extension = 1;
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asclen = FFMIN(asclen, get_bits_left(gb));
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init_get_bits(&gbc, gb->buffer, config_start_bit + asclen);
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skip_bits_long(&gbc, config_start_bit);
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} else if (asclen == 0) {
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gbc = *gb;
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} else {
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return AVERROR_INVALIDDATA;
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}
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if (get_bits_left(gb) <= 0)
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return AVERROR_INVALIDDATA;
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bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &m4ac,
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&gbc, config_start_bit,
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sync_extension);
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if (bits_consumed < config_start_bit)
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return AVERROR_INVALIDDATA;
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bits_consumed -= config_start_bit;
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if (asclen == 0)
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asclen = bits_consumed;
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if (!latmctx->initialized ||
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ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
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ac->oc[1].m4ac.chan_config != m4ac.chan_config) {
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if (latmctx->initialized) {
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av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", m4ac.sample_rate, m4ac.chan_config);
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} else {
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av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n");
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}
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latmctx->initialized = 0;
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esize = (asclen + 7) / 8;
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if (avctx->extradata_size < esize) {
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av_free(avctx->extradata);
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avctx->extradata = av_malloc(esize + AV_INPUT_BUFFER_PADDING_SIZE);
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if (!avctx->extradata)
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return AVERROR(ENOMEM);
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}
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avctx->extradata_size = esize;
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gbc = *gb;
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for (i = 0; i < esize; i++) {
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avctx->extradata[i] = get_bits(&gbc, 8);
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}
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memset(avctx->extradata+esize, 0, AV_INPUT_BUFFER_PADDING_SIZE);
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}
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skip_bits_long(gb, asclen);
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return 0;
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}
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static int read_stream_mux_config(struct LATMContext *latmctx,
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GetBitContext *gb)
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{
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int ret, audio_mux_version = get_bits(gb, 1);
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latmctx->audio_mux_version_A = 0;
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if (audio_mux_version)
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latmctx->audio_mux_version_A = get_bits(gb, 1);
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if (!latmctx->audio_mux_version_A) {
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if (audio_mux_version)
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latm_get_value(gb); // taraFullness
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skip_bits(gb, 1); // allStreamSameTimeFraming
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skip_bits(gb, 6); // numSubFrames
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// numPrograms
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if (get_bits(gb, 4)) { // numPrograms
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avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs");
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return AVERROR_PATCHWELCOME;
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}
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// for each program (which there is only one in DVB)
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// for each layer (which there is only one in DVB)
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if (get_bits(gb, 3)) { // numLayer
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avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
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return AVERROR_PATCHWELCOME;
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}
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// for all but first stream: use_same_config = get_bits(gb, 1);
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if (!audio_mux_version) {
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if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)
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return ret;
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} else {
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int ascLen = latm_get_value(gb);
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if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)
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return ret;
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}
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latmctx->frame_length_type = get_bits(gb, 3);
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switch (latmctx->frame_length_type) {
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case 0:
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skip_bits(gb, 8); // latmBufferFullness
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break;
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case 1:
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latmctx->frame_length = get_bits(gb, 9);
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break;
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case 3:
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case 4:
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case 5:
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skip_bits(gb, 6); // CELP frame length table index
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break;
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case 6:
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case 7:
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skip_bits(gb, 1); // HVXC frame length table index
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break;
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}
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if (get_bits(gb, 1)) { // other data
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if (audio_mux_version) {
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latm_get_value(gb); // other_data_bits
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} else {
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int esc;
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do {
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if (get_bits_left(gb) < 9)
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return AVERROR_INVALIDDATA;
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esc = get_bits(gb, 1);
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skip_bits(gb, 8);
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} while (esc);
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}
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}
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if (get_bits(gb, 1)) // crc present
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skip_bits(gb, 8); // config_crc
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}
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return 0;
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}
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static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
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{
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uint8_t tmp;
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if (ctx->frame_length_type == 0) {
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int mux_slot_length = 0;
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do {
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if (get_bits_left(gb) < 8)
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return AVERROR_INVALIDDATA;
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tmp = get_bits(gb, 8);
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mux_slot_length += tmp;
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} while (tmp == 255);
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return mux_slot_length;
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} else if (ctx->frame_length_type == 1) {
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return ctx->frame_length;
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} else if (ctx->frame_length_type == 3 ||
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ctx->frame_length_type == 5 ||
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ctx->frame_length_type == 7) {
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skip_bits(gb, 2); // mux_slot_length_coded
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}
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return 0;
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}
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static int read_audio_mux_element(struct LATMContext *latmctx,
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GetBitContext *gb)
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{
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int err;
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uint8_t use_same_mux = get_bits(gb, 1);
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if (!use_same_mux) {
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if ((err = read_stream_mux_config(latmctx, gb)) < 0)
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return err;
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} else if (!latmctx->aac_ctx.avctx->extradata) {
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av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
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"no decoder config found\n");
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return 1;
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}
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if (latmctx->audio_mux_version_A == 0) {
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int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
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if (mux_slot_length_bytes < 0 || mux_slot_length_bytes * 8LL > get_bits_left(gb)) {
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av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
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return AVERROR_INVALIDDATA;
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} else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
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av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
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"frame length mismatch %d << %d\n",
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mux_slot_length_bytes * 8, get_bits_left(gb));
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return AVERROR_INVALIDDATA;
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}
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}
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return 0;
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}
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static int latm_decode_frame(AVCodecContext *avctx, AVFrame *out,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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struct LATMContext *latmctx = avctx->priv_data;
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int muxlength, err;
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GetBitContext gb;
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if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
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return err;
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// check for LOAS sync word
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if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
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return AVERROR_INVALIDDATA;
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muxlength = get_bits(&gb, 13) + 3;
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// not enough data, the parser should have sorted this out
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if (muxlength > avpkt->size)
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return AVERROR_INVALIDDATA;
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if ((err = read_audio_mux_element(latmctx, &gb)))
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return (err < 0) ? err : avpkt->size;
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if (!latmctx->initialized) {
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if (!avctx->extradata) {
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*got_frame_ptr = 0;
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return avpkt->size;
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} else {
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push_output_configuration(&latmctx->aac_ctx);
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if ((err = decode_audio_specific_config(
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&latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac,
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avctx->extradata, avctx->extradata_size*8LL, 1)) < 0) {
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pop_output_configuration(&latmctx->aac_ctx);
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return err;
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}
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latmctx->initialized = 1;
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}
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}
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if (show_bits(&gb, 12) == 0xfff) {
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av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
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"ADTS header detected, probably as result of configuration "
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"misparsing\n");
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return AVERROR_INVALIDDATA;
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}
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switch (latmctx->aac_ctx.oc[1].m4ac.object_type) {
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case AOT_ER_AAC_LC:
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case AOT_ER_AAC_LTP:
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case AOT_ER_AAC_LD:
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case AOT_ER_AAC_ELD:
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err = aac_decode_er_frame(avctx, out, got_frame_ptr, &gb);
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break;
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default:
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err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt);
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}
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if (err < 0)
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return err;
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return muxlength;
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}
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static av_cold int latm_decode_init(AVCodecContext *avctx)
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{
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struct LATMContext *latmctx = avctx->priv_data;
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int ret = aac_decode_init(avctx);
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if (avctx->extradata_size > 0)
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latmctx->initialized = !ret;
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return ret;
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}
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/*
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Note: This decoder filter is intended to decode LATM streams transferred
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in MPEG transport streams which only contain one program.
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To do a more complex LATM demuxing a separate LATM demuxer should be used.
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*/
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const FFCodec ff_aac_latm_decoder = {
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.p.name = "aac_latm",
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CODEC_LONG_NAME("AAC LATM (Advanced Audio Coding LATM syntax)"),
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.p.type = AVMEDIA_TYPE_AUDIO,
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.p.id = AV_CODEC_ID_AAC_LATM,
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.priv_data_size = sizeof(struct LATMContext),
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.init = latm_decode_init,
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.close = ff_aac_decode_close,
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FF_CODEC_DECODE_CB(latm_decode_frame),
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.p.sample_fmts = (const enum AVSampleFormat[]) {
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AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
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},
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.p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
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.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
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.p.ch_layouts = ff_aac_ch_layout,
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.flush = flush,
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.p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
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};
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#endif /* AVCODEC_AAC_AACDEC_LATM_H */
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@ -65,296 +65,7 @@
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#include "aacdec_template.c"
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#define LOAS_SYNC_WORD 0x2b7 ///< 11 bits LOAS sync word
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struct LATMContext {
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AACDecContext aac_ctx; ///< containing AACContext
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int initialized; ///< initialized after a valid extradata was seen
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// parser data
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int audio_mux_version_A; ///< LATM syntax version
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int frame_length_type; ///< 0/1 variable/fixed frame length
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int frame_length; ///< frame length for fixed frame length
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};
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static inline uint32_t latm_get_value(GetBitContext *b)
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{
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int length = get_bits(b, 2);
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return get_bits_long(b, (length+1)*8);
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}
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static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
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GetBitContext *gb, int asclen)
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{
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AACDecContext *ac = &latmctx->aac_ctx;
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AVCodecContext *avctx = ac->avctx;
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MPEG4AudioConfig m4ac = { 0 };
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GetBitContext gbc;
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int config_start_bit = get_bits_count(gb);
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int sync_extension = 0;
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int bits_consumed, esize, i;
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if (asclen > 0) {
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sync_extension = 1;
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asclen = FFMIN(asclen, get_bits_left(gb));
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init_get_bits(&gbc, gb->buffer, config_start_bit + asclen);
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skip_bits_long(&gbc, config_start_bit);
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} else if (asclen == 0) {
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gbc = *gb;
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} else {
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return AVERROR_INVALIDDATA;
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}
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if (get_bits_left(gb) <= 0)
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return AVERROR_INVALIDDATA;
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bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &m4ac,
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&gbc, config_start_bit,
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sync_extension);
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if (bits_consumed < config_start_bit)
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return AVERROR_INVALIDDATA;
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bits_consumed -= config_start_bit;
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if (asclen == 0)
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asclen = bits_consumed;
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if (!latmctx->initialized ||
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ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
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ac->oc[1].m4ac.chan_config != m4ac.chan_config) {
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if (latmctx->initialized) {
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av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", m4ac.sample_rate, m4ac.chan_config);
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} else {
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av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n");
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}
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latmctx->initialized = 0;
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esize = (asclen + 7) / 8;
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if (avctx->extradata_size < esize) {
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av_free(avctx->extradata);
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avctx->extradata = av_malloc(esize + AV_INPUT_BUFFER_PADDING_SIZE);
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if (!avctx->extradata)
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return AVERROR(ENOMEM);
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}
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avctx->extradata_size = esize;
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gbc = *gb;
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for (i = 0; i < esize; i++) {
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avctx->extradata[i] = get_bits(&gbc, 8);
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}
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memset(avctx->extradata+esize, 0, AV_INPUT_BUFFER_PADDING_SIZE);
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}
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skip_bits_long(gb, asclen);
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return 0;
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}
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static int read_stream_mux_config(struct LATMContext *latmctx,
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GetBitContext *gb)
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{
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int ret, audio_mux_version = get_bits(gb, 1);
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|
||||
latmctx->audio_mux_version_A = 0;
|
||||
if (audio_mux_version)
|
||||
latmctx->audio_mux_version_A = get_bits(gb, 1);
|
||||
|
||||
if (!latmctx->audio_mux_version_A) {
|
||||
|
||||
if (audio_mux_version)
|
||||
latm_get_value(gb); // taraFullness
|
||||
|
||||
skip_bits(gb, 1); // allStreamSameTimeFraming
|
||||
skip_bits(gb, 6); // numSubFrames
|
||||
// numPrograms
|
||||
if (get_bits(gb, 4)) { // numPrograms
|
||||
avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs");
|
||||
return AVERROR_PATCHWELCOME;
|
||||
}
|
||||
|
||||
// for each program (which there is only one in DVB)
|
||||
|
||||
// for each layer (which there is only one in DVB)
|
||||
if (get_bits(gb, 3)) { // numLayer
|
||||
avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
|
||||
return AVERROR_PATCHWELCOME;
|
||||
}
|
||||
|
||||
// for all but first stream: use_same_config = get_bits(gb, 1);
|
||||
if (!audio_mux_version) {
|
||||
if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)
|
||||
return ret;
|
||||
} else {
|
||||
int ascLen = latm_get_value(gb);
|
||||
if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)
|
||||
return ret;
|
||||
}
|
||||
|
||||
latmctx->frame_length_type = get_bits(gb, 3);
|
||||
switch (latmctx->frame_length_type) {
|
||||
case 0:
|
||||
skip_bits(gb, 8); // latmBufferFullness
|
||||
break;
|
||||
case 1:
|
||||
latmctx->frame_length = get_bits(gb, 9);
|
||||
break;
|
||||
case 3:
|
||||
case 4:
|
||||
case 5:
|
||||
skip_bits(gb, 6); // CELP frame length table index
|
||||
break;
|
||||
case 6:
|
||||
case 7:
|
||||
skip_bits(gb, 1); // HVXC frame length table index
|
||||
break;
|
||||
}
|
||||
|
||||
if (get_bits(gb, 1)) { // other data
|
||||
if (audio_mux_version) {
|
||||
latm_get_value(gb); // other_data_bits
|
||||
} else {
|
||||
int esc;
|
||||
do {
|
||||
if (get_bits_left(gb) < 9)
|
||||
return AVERROR_INVALIDDATA;
|
||||
esc = get_bits(gb, 1);
|
||||
skip_bits(gb, 8);
|
||||
} while (esc);
|
||||
}
|
||||
}
|
||||
|
||||
if (get_bits(gb, 1)) // crc present
|
||||
skip_bits(gb, 8); // config_crc
|
||||
}
|
||||
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
|
||||
{
|
||||
uint8_t tmp;
|
||||
|
||||
if (ctx->frame_length_type == 0) {
|
||||
int mux_slot_length = 0;
|
||||
do {
|
||||
if (get_bits_left(gb) < 8)
|
||||
return AVERROR_INVALIDDATA;
|
||||
tmp = get_bits(gb, 8);
|
||||
mux_slot_length += tmp;
|
||||
} while (tmp == 255);
|
||||
return mux_slot_length;
|
||||
} else if (ctx->frame_length_type == 1) {
|
||||
return ctx->frame_length;
|
||||
} else if (ctx->frame_length_type == 3 ||
|
||||
ctx->frame_length_type == 5 ||
|
||||
ctx->frame_length_type == 7) {
|
||||
skip_bits(gb, 2); // mux_slot_length_coded
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int read_audio_mux_element(struct LATMContext *latmctx,
|
||||
GetBitContext *gb)
|
||||
{
|
||||
int err;
|
||||
uint8_t use_same_mux = get_bits(gb, 1);
|
||||
if (!use_same_mux) {
|
||||
if ((err = read_stream_mux_config(latmctx, gb)) < 0)
|
||||
return err;
|
||||
} else if (!latmctx->aac_ctx.avctx->extradata) {
|
||||
av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
|
||||
"no decoder config found\n");
|
||||
return 1;
|
||||
}
|
||||
if (latmctx->audio_mux_version_A == 0) {
|
||||
int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
|
||||
if (mux_slot_length_bytes < 0 || mux_slot_length_bytes * 8LL > get_bits_left(gb)) {
|
||||
av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
|
||||
return AVERROR_INVALIDDATA;
|
||||
} else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
|
||||
av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
|
||||
"frame length mismatch %d << %d\n",
|
||||
mux_slot_length_bytes * 8, get_bits_left(gb));
|
||||
return AVERROR_INVALIDDATA;
|
||||
}
|
||||
}
|
||||
return 0;
|
||||
}
|
||||
|
||||
|
||||
static int latm_decode_frame(AVCodecContext *avctx, AVFrame *out,
|
||||
int *got_frame_ptr, AVPacket *avpkt)
|
||||
{
|
||||
struct LATMContext *latmctx = avctx->priv_data;
|
||||
int muxlength, err;
|
||||
GetBitContext gb;
|
||||
|
||||
if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
|
||||
return err;
|
||||
|
||||
// check for LOAS sync word
|
||||
if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
|
||||
return AVERROR_INVALIDDATA;
|
||||
|
||||
muxlength = get_bits(&gb, 13) + 3;
|
||||
// not enough data, the parser should have sorted this out
|
||||
if (muxlength > avpkt->size)
|
||||
return AVERROR_INVALIDDATA;
|
||||
|
||||
if ((err = read_audio_mux_element(latmctx, &gb)))
|
||||
return (err < 0) ? err : avpkt->size;
|
||||
|
||||
if (!latmctx->initialized) {
|
||||
if (!avctx->extradata) {
|
||||
*got_frame_ptr = 0;
|
||||
return avpkt->size;
|
||||
} else {
|
||||
push_output_configuration(&latmctx->aac_ctx);
|
||||
if ((err = decode_audio_specific_config(
|
||||
&latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac,
|
||||
avctx->extradata, avctx->extradata_size*8LL, 1)) < 0) {
|
||||
pop_output_configuration(&latmctx->aac_ctx);
|
||||
return err;
|
||||
}
|
||||
latmctx->initialized = 1;
|
||||
}
|
||||
}
|
||||
|
||||
if (show_bits(&gb, 12) == 0xfff) {
|
||||
av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
|
||||
"ADTS header detected, probably as result of configuration "
|
||||
"misparsing\n");
|
||||
return AVERROR_INVALIDDATA;
|
||||
}
|
||||
|
||||
switch (latmctx->aac_ctx.oc[1].m4ac.object_type) {
|
||||
case AOT_ER_AAC_LC:
|
||||
case AOT_ER_AAC_LTP:
|
||||
case AOT_ER_AAC_LD:
|
||||
case AOT_ER_AAC_ELD:
|
||||
err = aac_decode_er_frame(avctx, out, got_frame_ptr, &gb);
|
||||
break;
|
||||
default:
|
||||
err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt);
|
||||
}
|
||||
if (err < 0)
|
||||
return err;
|
||||
|
||||
return muxlength;
|
||||
}
|
||||
|
||||
static av_cold int latm_decode_init(AVCodecContext *avctx)
|
||||
{
|
||||
struct LATMContext *latmctx = avctx->priv_data;
|
||||
int ret = aac_decode_init(avctx);
|
||||
|
||||
if (avctx->extradata_size > 0)
|
||||
latmctx->initialized = !ret;
|
||||
|
||||
return ret;
|
||||
}
|
||||
#include "libavcodec/aac/aacdec_latm.h"
|
||||
|
||||
const FFCodec ff_aac_decoder = {
|
||||
.p.name = "aac",
|
||||
@ -376,30 +87,6 @@ const FFCodec ff_aac_decoder = {
|
||||
.p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
|
||||
};
|
||||
|
||||
/*
|
||||
Note: This decoder filter is intended to decode LATM streams transferred
|
||||
in MPEG transport streams which only contain one program.
|
||||
To do a more complex LATM demuxing a separate LATM demuxer should be used.
|
||||
*/
|
||||
const FFCodec ff_aac_latm_decoder = {
|
||||
.p.name = "aac_latm",
|
||||
CODEC_LONG_NAME("AAC LATM (Advanced Audio Coding LATM syntax)"),
|
||||
.p.type = AVMEDIA_TYPE_AUDIO,
|
||||
.p.id = AV_CODEC_ID_AAC_LATM,
|
||||
.priv_data_size = sizeof(struct LATMContext),
|
||||
.init = latm_decode_init,
|
||||
.close = ff_aac_decode_close,
|
||||
FF_CODEC_DECODE_CB(latm_decode_frame),
|
||||
.p.sample_fmts = (const enum AVSampleFormat[]) {
|
||||
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
|
||||
},
|
||||
.p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
|
||||
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
|
||||
.p.ch_layouts = ff_aac_ch_layout,
|
||||
.flush = flush,
|
||||
.p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
|
||||
};
|
||||
|
||||
const FFCodec ff_aac_fixed_decoder = {
|
||||
.p.name = "aac_fixed",
|
||||
CODEC_LONG_NAME("AAC (Advanced Audio Coding)"),
|
||||
|
||||
Loading…
x
Reference in New Issue
Block a user