[SCM] libav/experimental: Move the RTP packetization code for MPEG12 video in its own file (rtp_mpv.c)
siretart at users.alioth.debian.org
siretart at users.alioth.debian.org
Sun Jun 30 16:05:55 UTC 2013
The following commit has been merged in the experimental branch:
commit 98561024ac791d9aa5caee0700f9cb55188fa266
Author: Luca Abeni <lucabe72 at email.it>
Date: Fri Aug 24 07:13:34 2007 +0000
Move the RTP packetization code for MPEG12 video in its own file (rtp_mpv.c)
Originally committed as revision 10201 to svn://svn.ffmpeg.org/ffmpeg/trunk
diff --git a/libavformat/Makefile b/libavformat/Makefile
index 794d212..8accc8a 100644
--- a/libavformat/Makefile
+++ b/libavformat/Makefile
@@ -119,7 +119,7 @@ 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-$(CONFIG_RTP_MUXER) += rtp.o rtp_h264.o
+OBJS-$(CONFIG_RTP_MUXER) += rtp.o rtp_h264.o rtp_mpv.o
OBJS-$(CONFIG_RTSP_DEMUXER) += rtsp.o
OBJS-$(CONFIG_SDP_DEMUXER) += rtsp.o
OBJS-$(CONFIG_SEGAFILM_DEMUXER) += segafilm.o
diff --git a/libavformat/rtp.c b/libavformat/rtp.c
index e42cfb1..f45ad37 100644
--- a/libavformat/rtp.c
+++ b/libavformat/rtp.c
@@ -27,6 +27,7 @@
#include "rtp_internal.h"
#include "rtp_h264.h"
+#include "rtp_mpv.h"
//#define DEBUG
@@ -788,7 +789,7 @@ static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time)
/* send an rtp packet. sequence number is incremented, but the caller
must update the timestamp itself */
-static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
+void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
{
RTPDemuxContext *s = s1->priv_data;
@@ -836,7 +837,7 @@ static void rtp_send_samples(AVFormatContext *s1,
n = (s->buf_ptr - s->buf);
/* if buffer full, then send it */
if (n >= max_packet_size) {
- rtp_send_data(s1, s->buf, n, 0);
+ ff_rtp_send_data(s1, s->buf, n, 0);
s->buf_ptr = s->buf;
/* update timestamp */
s->timestamp += n / sample_size;
@@ -859,7 +860,7 @@ static void rtp_send_mpegaudio(AVFormatContext *s1,
len = (s->buf_ptr - s->buf);
if ((len + size) > max_packet_size) {
if (len > 4) {
- rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
+ ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0);
s->buf_ptr = s->buf + 4;
/* 90 KHz time stamp */
s->timestamp = s->base_timestamp +
@@ -881,7 +882,7 @@ static void rtp_send_mpegaudio(AVFormatContext *s1,
s->buf[2] = count >> 8;
s->buf[3] = count;
memcpy(s->buf + 4, buf1, len);
- rtp_send_data(s1, s->buf, len + 4, 0);
+ ff_rtp_send_data(s1, s->buf, len + 4, 0);
size -= len;
buf1 += len;
count += len;
@@ -900,55 +901,6 @@ static void rtp_send_mpegaudio(AVFormatContext *s1,
s->cur_timestamp += st->codec->frame_size;
}
-/* NOTE: a single frame must be passed with sequence header if
- needed. XXX: use slices. */
-static void rtp_send_mpegvideo(AVFormatContext *s1,
- const uint8_t *buf1, int size)
-{
- RTPDemuxContext *s = s1->priv_data;
- AVStream *st = s1->streams[0];
- int len, h, max_packet_size;
- uint8_t *q;
-
- max_packet_size = s->max_payload_size;
-
- while (size > 0) {
- /* XXX: more correct headers */
- h = 0;
- if (st->codec->sub_id == 2)
- h |= 1 << 26; /* mpeg 2 indicator */
- q = s->buf;
- *q++ = h >> 24;
- *q++ = h >> 16;
- *q++ = h >> 8;
- *q++ = h;
-
- if (st->codec->sub_id == 2) {
- h = 0;
- *q++ = h >> 24;
- *q++ = h >> 16;
- *q++ = h >> 8;
- *q++ = h;
- }
-
- len = max_packet_size - (q - s->buf);
- if (len > size)
- len = size;
-
- memcpy(q, buf1, len);
- q += len;
-
- /* 90 KHz time stamp */
- s->timestamp = s->base_timestamp +
- av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
- rtp_send_data(s1, s->buf, q - s->buf, (len == size));
-
- buf1 += len;
- size -= len;
- }
- s->cur_timestamp++;
-}
-
static void rtp_send_raw(AVFormatContext *s1,
const uint8_t *buf1, int size)
{
@@ -966,7 +918,7 @@ static void rtp_send_raw(AVFormatContext *s1,
/* 90 KHz time stamp */
s->timestamp = s->base_timestamp +
av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
- rtp_send_data(s1, buf1, len, (len == size));
+ ff_rtp_send_data(s1, buf1, len, (len == size));
buf1 += len;
size -= len;
@@ -992,7 +944,7 @@ static void rtp_send_mpegts_raw(AVFormatContext *s1,
out_len = s->buf_ptr - s->buf;
if (out_len >= s->max_payload_size) {
- rtp_send_data(s1, s->buf, out_len, 0);
+ ff_rtp_send_data(s1, s->buf, out_len, 0);
s->buf_ptr = s->buf;
}
}
@@ -1042,7 +994,7 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
rtp_send_mpegaudio(s1, buf1, size);
break;
case CODEC_ID_MPEG1VIDEO:
- rtp_send_mpegvideo(s1, buf1, size);
+ ff_rtp_send_mpegvideo(s1, buf1, size);
break;
case CODEC_ID_MPEG2TS:
rtp_send_mpegts_raw(s1, buf1, size);
diff --git a/libavformat/rtp_internal.h b/libavformat/rtp_internal.h
index 882098f..b03f74e 100644
--- a/libavformat/rtp_internal.h
+++ b/libavformat/rtp_internal.h
@@ -110,5 +110,7 @@ struct RTPDemuxContext {
extern RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler;
int rtsp_next_attr_and_value(const char **p, char *attr, int attr_size, char *value, int value_size); ///< from rtsp.c, but used by rtp dynamic protocol handlers.
+
+void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m);
#endif /* RTP_INTERNAL_H */
diff --git a/libavformat/rtp_mpv.c b/libavformat/rtp_mpv.c
new file mode 100644
index 0000000..0db6663
--- /dev/null
+++ b/libavformat/rtp_mpv.c
@@ -0,0 +1,72 @@
+/*
+ * RTP packetization for MPEG video
+ * Copyright (c) 2002 Fabrice Bellard.
+ *
+ * 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_internal.h"
+
+/* NOTE: a single frame must be passed with sequence header if
+ needed. XXX: use slices. */
+void ff_rtp_send_mpegvideo(AVFormatContext *s1, const uint8_t *buf1, int size)
+{
+ RTPDemuxContext *s = s1->priv_data;
+ AVStream *st = s1->streams[0];
+ int len, h, max_packet_size;
+ uint8_t *q;
+
+ max_packet_size = s->max_payload_size;
+
+ while (size > 0) {
+ /* XXX: more correct headers */
+ h = 0;
+ if (st->codec->sub_id == 2)
+ h |= 1 << 26; /* mpeg 2 indicator */
+ q = s->buf;
+ *q++ = h >> 24;
+ *q++ = h >> 16;
+ *q++ = h >> 8;
+ *q++ = h;
+
+ if (st->codec->sub_id == 2) {
+ h = 0;
+ *q++ = h >> 24;
+ *q++ = h >> 16;
+ *q++ = h >> 8;
+ *q++ = h;
+ }
+
+ len = max_packet_size - (q - s->buf);
+ if (len > size)
+ len = size;
+
+ memcpy(q, buf1, len);
+ q += len;
+
+ /* 90 KHz time stamp */
+ s->timestamp = s->base_timestamp +
+ av_rescale((int64_t)s->cur_timestamp * st->codec->time_base.num, 90000, st->codec->time_base.den); //FIXME pass timestamps
+ ff_rtp_send_data(s1, s->buf, q - s->buf, (len == size));
+
+ buf1 += len;
+ size -= len;
+ }
+ s->cur_timestamp++;
+}
+
+
diff --git a/ffserver.h b/libavformat/rtp_mpv.h
similarity index 79%
copy from ffserver.h
copy to libavformat/rtp_mpv.h
index a1b95e8..fb09a25 100644
--- a/ffserver.h
+++ b/libavformat/rtp_mpv.h
@@ -1,6 +1,6 @@
/*
- * Multiple format streaming server
- * copyright (c) 2002 Fabrice Bellard
+ * RTP definitions
+ * Copyright (c) 2002 Fabrice Bellard.
*
* This file is part of FFmpeg.
*
@@ -18,11 +18,9 @@
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
-#ifndef FFSERVER_H
-#define FFSERVER_H
+#ifndef RTP_MPV_H
+#define RTP_MPV_H
-/* interface between ffserver and modules */
+void ff_rtp_send_mpegvideo(AVFormatContext *s1, const uint8_t *buf1, int size);
-void ffserver_module_init(void);
-
-#endif
+#endif /* RTP_MPV_H */
--
Libav/FFmpeg packaging
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