mirror of
https://github.com/FFmpeg/FFmpeg.git
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275 lines
8.8 KiB
C
275 lines
8.8 KiB
C
/*
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* LATM/LOAS muxer
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* Copyright (c) 2011 Kieran Kunhya <kieran@kunhya.com>
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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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#include "libavcodec/get_bits.h"
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#include "libavcodec/put_bits.h"
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#include "libavcodec/codec_id.h"
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#include "libavcodec/codec_par.h"
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#include "libavcodec/packet.h"
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#include "libavcodec/mpeg4audio.h"
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#include "libavutil/opt.h"
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#include "avformat.h"
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#include "internal.h"
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#include "rawenc.h"
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#define MAX_EXTRADATA_SIZE 1024
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typedef struct LATMContext {
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AVClass *av_class;
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int off;
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int channel_conf;
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int object_type;
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int counter;
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int mod;
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uint8_t buffer[0x1fff + MAX_EXTRADATA_SIZE + 1024];
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} LATMContext;
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static const AVOption options[] = {
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{"smc-interval", "StreamMuxConfig interval.",
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offsetof(LATMContext, mod), AV_OPT_TYPE_INT, {.i64 = 0x0014}, 0x0001, 0xffff, AV_OPT_FLAG_ENCODING_PARAM},
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{NULL},
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};
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static const AVClass latm_muxer_class = {
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.class_name = "LATM/LOAS muxer",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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static int latm_decode_extradata(AVFormatContext *s, uint8_t *buf, int size)
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{
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LATMContext *ctx = s->priv_data;
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MPEG4AudioConfig m4ac;
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if (size > MAX_EXTRADATA_SIZE) {
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av_log(s, AV_LOG_ERROR, "Extradata is larger than currently supported.\n");
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return AVERROR_INVALIDDATA;
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}
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ctx->off = avpriv_mpeg4audio_get_config2(&m4ac, buf, size, 1, s);
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if (ctx->off < 0)
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return ctx->off;
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if (ctx->object_type == AOT_ALS && (ctx->off & 7)) {
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// as long as avpriv_mpeg4audio_get_config works correctly this is impossible
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av_log(s, AV_LOG_ERROR, "BUG: ALS offset is not byte-aligned\n");
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return AVERROR_INVALIDDATA;
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}
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/* FIXME: are any formats not allowed in LATM? */
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if (m4ac.object_type > AOT_SBR && m4ac.object_type != AOT_ALS) {
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av_log(s, AV_LOG_ERROR, "Muxing MPEG-4 AOT %d in LATM is not supported\n", m4ac.object_type);
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return AVERROR_INVALIDDATA;
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}
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ctx->channel_conf = m4ac.chan_config;
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ctx->object_type = m4ac.object_type;
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return 0;
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}
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static int latm_write_header(AVFormatContext *s)
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{
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AVCodecParameters *par = s->streams[0]->codecpar;
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if (par->codec_id == AV_CODEC_ID_AAC_LATM)
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return 0;
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if (par->codec_id != AV_CODEC_ID_AAC && par->codec_id != AV_CODEC_ID_MP4ALS) {
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av_log(s, AV_LOG_ERROR, "Only AAC, LATM and ALS are supported\n");
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return AVERROR(EINVAL);
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}
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if (par->extradata_size > 0 &&
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latm_decode_extradata(s, par->extradata, par->extradata_size) < 0)
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return AVERROR_INVALIDDATA;
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return 0;
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}
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static void copy_bits(PutBitContext *pb, const uint8_t *src, int length)
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{
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int words = length >> 4;
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int bits = length & 15;
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int i;
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for (i = 0; i < words; i++)
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put_bits(pb, 16, AV_RB16(src + 2 * i));
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if (bits)
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put_bits(pb, bits, AV_RB16(src + 2 * words) >> (16 - bits));
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}
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static void latm_write_frame_header(AVFormatContext *s, PutBitContext *bs)
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{
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LATMContext *ctx = s->priv_data;
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AVCodecParameters *par = s->streams[0]->codecpar;
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int header_size;
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/* AudioMuxElement */
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put_bits(bs, 1, !!ctx->counter);
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if (!ctx->counter) {
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/* StreamMuxConfig */
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put_bits(bs, 1, 0); /* audioMuxVersion */
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put_bits(bs, 1, 1); /* allStreamsSameTimeFraming */
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put_bits(bs, 6, 0); /* numSubFrames */
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put_bits(bs, 4, 0); /* numProgram */
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put_bits(bs, 3, 0); /* numLayer */
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/* AudioSpecificConfig */
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if (ctx->object_type == AOT_ALS) {
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header_size = (par->extradata_size - (ctx->off >> 3)) * 8;
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copy_bits(bs, &par->extradata[ctx->off >> 3], header_size);
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} else {
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// + 3 assumes not scalable and dependsOnCoreCoder == 0,
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// see decode_ga_specific_config in libavcodec/aacdec.c
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copy_bits(bs, par->extradata, ctx->off + 3);
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if (!ctx->channel_conf) {
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GetBitContext gb;
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int ret = init_get_bits8(&gb, par->extradata, par->extradata_size);
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av_assert0(ret >= 0); // extradata size has been checked already, so this should not fail
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skip_bits_long(&gb, ctx->off + 3);
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ff_copy_pce_data(bs, &gb);
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}
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}
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put_bits(bs, 3, 0); /* frameLengthType */
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put_bits(bs, 8, 0xff); /* latmBufferFullness */
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put_bits(bs, 1, 0); /* otherDataPresent */
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put_bits(bs, 1, 0); /* crcCheckPresent */
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}
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ctx->counter++;
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ctx->counter %= ctx->mod;
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}
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static int latm_write_packet(AVFormatContext *s, AVPacket *pkt)
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{
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LATMContext *ctx = s->priv_data;
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AVCodecParameters *par = s->streams[0]->codecpar;
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AVIOContext *pb = s->pb;
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PutBitContext bs;
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int i, len;
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uint8_t loas_header[] = "\x56\xe0\x00";
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if (par->codec_id == AV_CODEC_ID_AAC_LATM)
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return ff_raw_write_packet(s, pkt);
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if (!par->extradata) {
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if(pkt->size > 2 && pkt->data[0] == 0x56 && (pkt->data[1] >> 4) == 0xe &&
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(AV_RB16(pkt->data + 1) & 0x1FFF) + 3 == pkt->size)
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return ff_raw_write_packet(s, pkt);
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else {
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uint8_t *side_data;
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size_t side_data_size;
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int ret;
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side_data = av_packet_get_side_data(pkt, AV_PKT_DATA_NEW_EXTRADATA,
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&side_data_size);
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if (side_data_size) {
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if (latm_decode_extradata(s, side_data, side_data_size) < 0)
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return AVERROR_INVALIDDATA;
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ret = ff_alloc_extradata(par, side_data_size);
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if (ret < 0)
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return ret;
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memcpy(par->extradata, side_data, side_data_size);
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} else
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return AVERROR_INVALIDDATA;
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}
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}
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if (pkt->size > 0x1fff)
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goto too_large;
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init_put_bits(&bs, ctx->buffer, pkt->size+1024+MAX_EXTRADATA_SIZE);
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latm_write_frame_header(s, &bs);
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/* PayloadLengthInfo() */
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for (i = 0; i <= pkt->size-255; i+=255)
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put_bits(&bs, 8, 255);
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put_bits(&bs, 8, pkt->size-i);
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/* The LATM payload is written unaligned */
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/* PayloadMux() */
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if (pkt->size && (pkt->data[0] & 0xe1) == 0x81) {
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// Convert byte-aligned DSE to non-aligned.
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// Due to the input format encoding we know that
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// it is naturally byte-aligned in the input stream,
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// so there are no padding bits to account for.
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// To avoid having to add padding bits and rearrange
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// the whole stream we just remove the byte-align flag.
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// This allows us to remux our FATE AAC samples into latm
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// files that are still playable with minimal effort.
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put_bits(&bs, 8, pkt->data[0] & 0xfe);
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copy_bits(&bs, pkt->data + 1, 8*pkt->size - 8);
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} else
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copy_bits(&bs, pkt->data, 8*pkt->size);
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flush_put_bits(&bs);
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len = put_bytes_output(&bs);
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if (len > 0x1fff)
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goto too_large;
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loas_header[1] |= (len >> 8) & 0x1f;
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loas_header[2] |= len & 0xff;
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avio_write(pb, loas_header, 3);
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avio_write(pb, ctx->buffer, len);
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return 0;
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too_large:
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av_log(s, AV_LOG_ERROR, "LATM packet size larger than maximum size 0x1fff\n");
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return AVERROR_INVALIDDATA;
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}
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static int latm_check_bitstream(struct AVFormatContext *s, const AVPacket *pkt)
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{
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int ret = 1;
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AVStream *st = s->streams[pkt->stream_index];
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if (st->codecpar->codec_id == AV_CODEC_ID_AAC) {
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if (pkt->size > 2 && (AV_RB16(pkt->data) & 0xfff0) == 0xfff0)
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ret = ff_stream_add_bitstream_filter(st, "aac_adtstoasc", NULL);
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}
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return ret;
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}
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const AVOutputFormat ff_latm_muxer = {
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.name = "latm",
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.long_name = NULL_IF_CONFIG_SMALL("LOAS/LATM"),
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.mime_type = "audio/MP4A-LATM",
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.extensions = "latm,loas",
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.priv_data_size = sizeof(LATMContext),
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.audio_codec = AV_CODEC_ID_AAC,
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.video_codec = AV_CODEC_ID_NONE,
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.write_header = latm_write_header,
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.write_packet = latm_write_packet,
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.priv_class = &latm_muxer_class,
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.check_bitstream= latm_check_bitstream,
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.flags = AVFMT_NOTIMESTAMPS,
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};
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