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* qatar/master: Handle unicode file names on windows rtp: Rename the open/close functions to alloc/free Lowercase all ff* program names. Refer to ff* tools by their lowercase names. NOT Pulled Replace more FFmpeg instances by Libav or ffmpeg. Replace `` by $() syntax in shell scripts. patcheck: Allow overiding grep program(s) through environment variables. NOT Pulled Remove stray libavcore and _g binary references. vorbis: Rename decoder/encoder files to follow general file naming scheme. aacenc: Fix whitespace after last commit. cook: Fix small typo in av_log_ask_for_sample message. aacenc: Finish 3GPP psymodel analysis for non mid/side cases. Remove RDFT dependency from AAC decoder. Add some debug log messages to AAC extradata Fix mov debug (u)int64_t format strings. bswap: use native types for av_bwap16(). doc: FLV muxing is supported. applehttp: Handle AES-128 encrypted streams Add a protocol handler for AES CBC decryption with PKCS7 padding doc: Mention that DragonFly BSD requires __BSD_VISIBLE set Conflicts: ffplay.c ffprobe.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
252 lines
7.7 KiB
C
252 lines
7.7 KiB
C
/**
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* Common code for the RTP depacketization of MPEG-4 formats.
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* Copyright (c) 2010 Fabrice Bellard
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* Romain Degez
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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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* @brief MPEG4 / RTP Code
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* @author Fabrice Bellard
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* @author Romain Degez
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*/
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#include "rtpdec_formats.h"
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#include "internal.h"
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#include "libavutil/avstring.h"
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#include "libavcodec/get_bits.h"
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#include <strings.h>
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/** Structure listing useful vars to parse RTP packet payload*/
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struct PayloadContext
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{
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int sizelength;
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int indexlength;
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int indexdeltalength;
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int profile_level_id;
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int streamtype;
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int objecttype;
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char *mode;
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/** mpeg 4 AU headers */
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struct AUHeaders {
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int size;
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int index;
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int cts_flag;
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int cts;
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int dts_flag;
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int dts;
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int rap_flag;
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int streamstate;
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} *au_headers;
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int au_headers_allocated;
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int nb_au_headers;
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int au_headers_length_bytes;
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int cur_au_index;
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};
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typedef struct {
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const char *str;
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uint16_t type;
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uint32_t offset;
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} AttrNameMap;
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/* All known fmtp parameters and the corresponding RTPAttrTypeEnum */
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#define ATTR_NAME_TYPE_INT 0
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#define ATTR_NAME_TYPE_STR 1
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static const AttrNameMap attr_names[]=
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{
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{ "SizeLength", ATTR_NAME_TYPE_INT,
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offsetof(PayloadContext, sizelength) },
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{ "IndexLength", ATTR_NAME_TYPE_INT,
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offsetof(PayloadContext, indexlength) },
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{ "IndexDeltaLength", ATTR_NAME_TYPE_INT,
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offsetof(PayloadContext, indexdeltalength) },
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{ "profile-level-id", ATTR_NAME_TYPE_INT,
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offsetof(PayloadContext, profile_level_id) },
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{ "StreamType", ATTR_NAME_TYPE_INT,
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offsetof(PayloadContext, streamtype) },
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{ "mode", ATTR_NAME_TYPE_STR,
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offsetof(PayloadContext, mode) },
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{ NULL, -1, -1 },
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};
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static PayloadContext *new_context(void)
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{
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return av_mallocz(sizeof(PayloadContext));
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}
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static void free_context(PayloadContext * data)
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{
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int i;
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for (i = 0; i < data->nb_au_headers; i++) {
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/* according to rtp_parse_mp4_au, we treat multiple
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* au headers as one, so nb_au_headers is always 1.
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* loop anyway in case this changes.
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* (note: changes done carelessly might lead to a double free)
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*/
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av_free(&data->au_headers[i]);
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}
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av_free(data->mode);
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av_free(data);
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}
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static int parse_fmtp_config(AVCodecContext * codec, char *value)
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{
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/* decode the hexa encoded parameter */
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int len = ff_hex_to_data(NULL, value);
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av_free(codec->extradata);
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codec->extradata = av_mallocz(len + FF_INPUT_BUFFER_PADDING_SIZE);
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if (!codec->extradata)
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return AVERROR(ENOMEM);
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codec->extradata_size = len;
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ff_hex_to_data(codec->extradata, value);
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return 0;
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}
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static int rtp_parse_mp4_au(PayloadContext *data, const uint8_t *buf)
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{
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int au_headers_length, au_header_size, i;
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GetBitContext getbitcontext;
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/* decode the first 2 bytes where the AUHeader sections are stored
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length in bits */
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au_headers_length = AV_RB16(buf);
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if (au_headers_length > RTP_MAX_PACKET_LENGTH)
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return -1;
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data->au_headers_length_bytes = (au_headers_length + 7) / 8;
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/* skip AU headers length section (2 bytes) */
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buf += 2;
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init_get_bits(&getbitcontext, buf, data->au_headers_length_bytes * 8);
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/* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
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au_header_size = data->sizelength + data->indexlength;
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if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
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return -1;
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data->nb_au_headers = au_headers_length / au_header_size;
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if (!data->au_headers || data->au_headers_allocated < data->nb_au_headers) {
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av_free(data->au_headers);
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data->au_headers = av_malloc(sizeof(struct AUHeaders) * data->nb_au_headers);
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data->au_headers_allocated = data->nb_au_headers;
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}
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/* XXX: We handle multiple AU Section as only one (need to fix this for interleaving)
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In my test, the FAAD decoder does not behave correctly when sending each AU one by one
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but does when sending the whole as one big packet... */
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data->au_headers[0].size = 0;
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data->au_headers[0].index = 0;
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for (i = 0; i < data->nb_au_headers; ++i) {
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data->au_headers[0].size += get_bits_long(&getbitcontext, data->sizelength);
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data->au_headers[0].index = get_bits_long(&getbitcontext, data->indexlength);
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}
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data->nb_au_headers = 1;
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return 0;
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}
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/* Follows RFC 3640 */
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static int aac_parse_packet(AVFormatContext *ctx,
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PayloadContext *data,
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AVStream *st,
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AVPacket *pkt,
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uint32_t *timestamp,
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const uint8_t *buf, int len, int flags)
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{
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if (rtp_parse_mp4_au(data, buf))
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return -1;
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buf += data->au_headers_length_bytes + 2;
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len -= data->au_headers_length_bytes + 2;
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/* XXX: Fixme we only handle the case where rtp_parse_mp4_au define
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one au_header */
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av_new_packet(pkt, data->au_headers[0].size);
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memcpy(pkt->data, buf, data->au_headers[0].size);
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pkt->stream_index = st->index;
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return 0;
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}
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static int parse_fmtp(AVStream *stream, PayloadContext *data,
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char *attr, char *value)
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{
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AVCodecContext *codec = stream->codec;
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int res, i;
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if (!strcmp(attr, "config")) {
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res = parse_fmtp_config(codec, value);
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if (res < 0)
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return res;
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}
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if (codec->codec_id == CODEC_ID_AAC) {
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/* Looking for a known attribute */
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for (i = 0; attr_names[i].str; ++i) {
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if (!strcasecmp(attr, attr_names[i].str)) {
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if (attr_names[i].type == ATTR_NAME_TYPE_INT) {
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*(int *)((char *)data+
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attr_names[i].offset) = atoi(value);
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} else if (attr_names[i].type == ATTR_NAME_TYPE_STR)
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*(char **)((char *)data+
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attr_names[i].offset) = av_strdup(value);
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}
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}
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}
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return 0;
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}
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static int parse_sdp_line(AVFormatContext *s, int st_index,
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PayloadContext *data, const char *line)
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{
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const char *p;
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if (av_strstart(line, "fmtp:", &p))
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return ff_parse_fmtp(s->streams[st_index], data, p, parse_fmtp);
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return 0;
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}
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RTPDynamicProtocolHandler ff_mp4v_es_dynamic_handler = {
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.enc_name = "MP4V-ES",
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.codec_type = AVMEDIA_TYPE_VIDEO,
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.codec_id = CODEC_ID_MPEG4,
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.parse_sdp_a_line = parse_sdp_line,
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.alloc = NULL,
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.free = NULL,
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.parse_packet = NULL
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};
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RTPDynamicProtocolHandler ff_mpeg4_generic_dynamic_handler = {
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.enc_name = "mpeg4-generic",
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.codec_type = AVMEDIA_TYPE_AUDIO,
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.codec_id = CODEC_ID_AAC,
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.parse_sdp_a_line = parse_sdp_line,
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.alloc = new_context,
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.free = free_context,
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.parse_packet = aac_parse_packet
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};
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