Commit 171dce48 authored by Luca Abeni's avatar Luca Abeni

Support for AAC streaming over RTP. Fragmentation is not implemented yet

Originally committed as revision 10491 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent f0dd9d45
......@@ -122,7 +122,7 @@ OBJS-$(CONFIG_RM_DEMUXER) += rmdec.o
OBJS-$(CONFIG_RM_MUXER) += rmenc.o
OBJS-$(CONFIG_ROQ_DEMUXER) += idroq.o
OBJS-$(CONFIG_ROQ_MUXER) += raw.o
OBJS-$(CONFIG_RTP_MUXER) += rtp.o rtp_h264.o rtp_mpv.o
OBJS-$(CONFIG_RTP_MUXER) += rtp.o rtp_h264.o rtp_mpv.o rtp_aac.o
OBJS-$(CONFIG_RTSP_DEMUXER) += rtsp.o
OBJS-$(CONFIG_SDP_DEMUXER) += rtsp.o
OBJS-$(CONFIG_SEGAFILM_DEMUXER) += segafilm.o
......
......@@ -28,6 +28,7 @@
#include "rtp_internal.h"
#include "rtp_h264.h"
#include "rtp_mpv.h"
#include "rtp_aac.h"
//#define DEBUG
......@@ -762,6 +763,8 @@ static int rtp_write_header(AVFormatContext *s1)
s->max_payload_size = n * TS_PACKET_SIZE;
s->buf_ptr = s->buf;
break;
case CODEC_ID_AAC:
s->read_buf_index = 0;
default:
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
av_set_pts_info(st, 32, 1, st->codec->sample_rate);
......@@ -993,6 +996,9 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
case CODEC_ID_MPEG1VIDEO:
ff_rtp_send_mpegvideo(s1, buf1, size);
break;
case CODEC_ID_AAC:
ff_rtp_send_aac(s1, buf1, size);
break;
case CODEC_ID_MPEG2TS:
rtp_send_mpegts_raw(s1, buf1, size);
break;
......
/*
* copyright (c) 2007 Luca Abeni
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "avformat.h"
#include "rtp_aac.h"
#include "rtp_internal.h"
#define MAX_FRAMES_PER_PACKET 5
#define MAX_AU_HEADERS_SIZE (2 + 2 * MAX_FRAMES_PER_PACKET)
void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size)
{
RTPDemuxContext *s = s1->priv_data;
int len, max_packet_size;
uint8_t *p;
/* skip ADTS header, if present */
if ((s1->streams[0]->codec->extradata_size) == 0) {
size -= 7;
buff += 7;
}
max_packet_size = s->max_payload_size - MAX_AU_HEADERS_SIZE;
/* test if the packet must be sent */
len = (s->buf_ptr - s->buf);
if ((s->read_buf_index == MAX_FRAMES_PER_PACKET) || (len && (len + size) > max_packet_size)) {
int au_size = s->read_buf_index * 2;
p = s->buf + MAX_AU_HEADERS_SIZE - au_size - 2;
if (p != s->buf) {
memmove(p + 2, s->buf + 2, au_size);
}
/* Write the AU header size */
p[0] = ((au_size * 8) & 0xFF) >> 8;
p[1] = (au_size * 8) & 0xFF;
ff_rtp_send_data(s1, p, s->buf_ptr - p, 1);
s->read_buf_index = 0;
}
if (s->read_buf_index == 0) {
s->buf_ptr = s->buf + MAX_AU_HEADERS_SIZE;
s->timestamp = s->cur_timestamp;
}
if (size < max_packet_size) {
p = s->buf + s->read_buf_index++ * 2 + 2;
*p++ = size >> 5;
*p = (size & 0x1F) << 3;
memcpy(s->buf_ptr, buff, size);
s->buf_ptr += size;
} else {
av_log(s1, AV_LOG_ERROR, "Unsupported!\n");
}
}
/*
* RTP definitions
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef RTP_AAC_H
#define RTP_AAC_H
void ff_rtp_send_aac(AVFormatContext *s1, const uint8_t *buff, int size);
#endif /* RTP_AAC_H */
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