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Move the RTP packetization code for MPEG12 video in its own file (rtp_mpv.c)

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Originally committed as revision 10201 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Luca Abeni 2007-08-24 07:13:34 +00:00
parent b75c8d16e7
commit 98561024ac
5 changed files with 109 additions and 57 deletions
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2
libavformat/Makefile
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@ -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

64
libavformat/rtp.c
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@ -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);

2
libavformat/rtp_internal.h
View File

@ -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 */

72
libavformat/rtp_mpv.c Normal file
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@ -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++;
}

26
libavformat/rtp_mpv.h Normal file
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@ -0,0 +1,26 @@
/*
* RTP definitions
* 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
*/
#ifndef RTP_MPV_H
#define RTP_MPV_H
void ff_rtp_send_mpegvideo(AVFormatContext *s1, const uint8_t *buf1, int size);
#endif /* RTP_MPV_H */