[FFmpeg-cvslog] r10491 - in trunk/libavformat: Makefile rtp.c rtp_aac.c rtp_aac.h
lucabe
subversion
Fri Sep 14 10:17:07 CEST 2007
Author: lucabe
Date: Fri Sep 14 10:17:06 2007
New Revision: 10491
Log:
Support for AAC streaming over RTP. Fragmentation is not implemented yet
Added:
trunk/libavformat/rtp_aac.c
trunk/libavformat/rtp_aac.h
Modified:
trunk/libavformat/Makefile
trunk/libavformat/rtp.c
Modified: trunk/libavformat/Makefile
==============================================================================
--- trunk/libavformat/Makefile (original)
+++ trunk/libavformat/Makefile Fri Sep 14 10:17:06 2007
@@ -122,7 +122,7 @@ OBJS-$(CONFIG_RM_DEMUXER)
OBJS-$(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
Modified: trunk/libavformat/rtp.c
==============================================================================
--- trunk/libavformat/rtp.c (original)
+++ trunk/libavformat/rtp.c Fri Sep 14 10:17:06 2007
@@ -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(AVFormatCont
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(AVFormatCont
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;
Added: trunk/libavformat/rtp_aac.c
==============================================================================
--- (empty file)
+++ trunk/libavformat/rtp_aac.c Fri Sep 14 10:17:06 2007
@@ -0,0 +1,72 @@
+/*
+ * 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");
+ }
+}
Added: trunk/libavformat/rtp_aac.h
==============================================================================
--- (empty file)
+++ trunk/libavformat/rtp_aac.h Fri Sep 14 10:17:06 2007
@@ -0,0 +1,25 @@
+/*
+ * 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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