* RTP input/output format
* Copyright (c) 2002 Fabrice Bellard.
*
- * This library is free software; you can redistribute it and/or
+ * 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 of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
*
- * This library is distributed in the hope that it will be useful,
+ * 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 this library; if not, write to the Free Software
- * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ * 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 "mpegts.h"
buffer to 'rtp_write_packet' contains all the packets for ONE
frame. Each packet should have a four byte header containing
the length in big endian format (same trick as
- 'url_open_dyn_packet_buf')
+ 'url_open_dyn_packet_buf')
*/
/* from http://www.iana.org/assignments/rtp-parameters last updated 05 January 2005 */
{9, "G722", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
{10, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 2},
{11, "L16", CODEC_TYPE_AUDIO, CODEC_ID_PCM_S16BE, 44100, 1},
- {12, "QCELP", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
+ {12, "QCELP", CODEC_TYPE_AUDIO, CODEC_ID_QCELP, 8000, 1},
{13, "CN", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
{14, "MPA", CODEC_TYPE_AUDIO, CODEC_ID_MP2, 90000, -1},
{15, "G728", CODEC_TYPE_AUDIO, CODEC_ID_NONE, 8000, 1},
MpegTSContext *ts; /* only used for MP2T payloads */
int read_buf_index;
int read_buf_size;
-
+
/* rtcp sender statistics receive */
int64_t last_rtcp_ntp_time;
int64_t first_rtcp_ntp_time;
{
if (AVRtpPayloadTypes[payload_type].codec_id != CODEC_ID_NONE) {
codec->codec_type = AVRtpPayloadTypes[payload_type].codec_type;
- codec->codec_id = AVRtpPayloadTypes[payload_type].codec_type;
+ codec->codec_id = AVRtpPayloadTypes[payload_type].codec_id;
if (AVRtpPayloadTypes[payload_type].audio_channels > 0)
codec->channels = AVRtpPayloadTypes[payload_type].audio_channels;
if (AVRtpPayloadTypes[payload_type].clock_rate > 0)
/**
* open a new RTP parse context for stream 'st'. 'st' can be NULL for
* MPEG2TS streams to indicate that they should be demuxed inside the
- * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
+ * rtp demux (otherwise CODEC_ID_MPEG2TS packets are returned)
*/
RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, int payload_type, rtp_payload_data_t *rtp_payload_data)
{
return NULL;
}
} else {
- switch(st->codec.codec_id) {
+ switch(st->codec->codec_id) {
case CODEC_ID_MPEG1VIDEO:
case CODEC_ID_MPEG2VIDEO:
case CODEC_ID_MP2:
static int rtp_parse_mp4_au(RTPDemuxContext *s, const uint8_t *buf)
{
- AVCodecContext codec;
int au_headers_length, au_header_size, i;
GetBitContext getbitcontext;
rtp_payload_data_t *infos;
if (infos == NULL)
return -1;
- codec = s->st->codec;
-
/* decode the first 2 bytes where are stored the AUHeader sections
length in bits */
au_headers_length = BE_16(buf);
}
/**
- * Parse an RTP or RTCP packet directly sent as a buffer.
+ * Parse an RTP or RTCP packet directly sent as a buffer.
* @param s RTP parse context.
* @param pkt returned packet
* @param buf input buffer or NULL to read the next packets
* @param len buffer len
- * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
+ * @return 0 if a packet is returned, 1 if a packet is returned and more can follow
* (use buf as NULL to read the next). -1 if no packet (error or no more packet).
*/
-int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
+int rtp_parse_packet(RTPDemuxContext *s, AVPacket *pkt,
const uint8_t *buf, int len)
{
unsigned int ssrc, h;
int payload_type, seq, delta_timestamp, ret;
AVStream *st;
uint32_t timestamp;
-
+
if (!buf) {
/* return the next packets, if any */
if (s->read_buf_index >= s->read_buf_size)
return -1;
- ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
+ ret = mpegts_parse_packet(s->ts, pkt, s->buf + s->read_buf_index,
s->read_buf_size - s->read_buf_index);
if (ret < 0)
return -1;
seq = (buf[2] << 8) | buf[3];
timestamp = decode_be32(buf + 4);
ssrc = decode_be32(buf + 8);
-
+
/* NOTE: we can handle only one payload type */
if (s->payload_type != payload_type)
return -1;
+
+ st = s->st;
#if defined(DEBUG) || 1
if (seq != ((s->seq + 1) & 0xffff)) {
- av_log(&s->st->codec, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
+ av_log(st?st->codec:NULL, AV_LOG_ERROR, "RTP: PT=%02x: bad cseq %04x expected=%04x\n",
payload_type, seq, ((s->seq + 1) & 0xffff));
}
#endif
len -= 12;
buf += 12;
- st = s->st;
if (!st) {
/* specific MPEG2TS demux support */
ret = mpegts_parse_packet(s->ts, pkt, buf, len);
return 1;
}
} else {
- switch(st->codec.codec_id) {
+ switch(st->codec->codec_id) {
case CODEC_ID_MP2:
/* better than nothing: skip mpeg audio RTP header */
if (len <= 4)
memcpy(pkt->data, buf, len);
break;
}
-
- switch(st->codec.codec_id) {
+
+ switch(st->codec->codec_id) {
case CODEC_ID_MP2:
case CODEC_ID_MPEG1VIDEO:
if (s->last_rtcp_ntp_time != AV_NOPTS_VALUE) {
len -= infos->au_headers[0].size;
}
s->read_buf_size = len;
- s->buf_ptr = (char *)buf;
+ s->buf_ptr = buf;
pkt->stream_index = s->st->index;
return 0;
default:
return -1;
st = s1->streams[0];
- payload_type = rtp_get_payload_type(&st->codec);
+ payload_type = rtp_get_payload_type(st->codec);
if (payload_type < 0)
payload_type = RTP_PT_PRIVATE; /* private payload type */
s->payload_type = payload_type;
- s->base_timestamp = random();
+// following 2 FIXMies could be set based on the current time, theres normaly no info leak, as rtp will likely be transmitted immedeatly
+ s->base_timestamp = 0; /* FIXME: was random(), what should this be? */
s->timestamp = s->base_timestamp;
- s->ssrc = random();
+ s->ssrc = 0; /* FIXME: was random(), what should this be? */
s->first_packet = 1;
max_packet_size = url_fget_max_packet_size(&s1->pb);
return AVERROR_IO;
s->max_payload_size = max_packet_size - 12;
- switch(st->codec.codec_id) {
+ switch(st->codec->codec_id) {
case CODEC_ID_MP2:
case CODEC_ID_MP3:
s->buf_ptr = s->buf + 4;
/* 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)
+static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
{
RTPDemuxContext *s = s1->priv_data;
/* build the RTP header */
put_byte(&s1->pb, (RTP_VERSION << 6));
- put_byte(&s1->pb, s->payload_type & 0x7f);
+ put_byte(&s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7));
put_be16(&s1->pb, s->seq);
put_be32(&s1->pb, s->timestamp);
put_be32(&s1->pb, s->ssrc);
-
+
put_buffer(&s1->pb, buf1, len);
put_flush_packet(&s1->pb);
-
+
s->seq++;
s->octet_count += len;
s->packet_count++;
n = (s->buf_ptr - s->buf);
/* if buffer full, then send it */
if (n >= max_packet_size) {
- rtp_send_data(s1, s->buf, n);
+ rtp_send_data(s1, s->buf, n, 0);
s->buf_ptr = s->buf;
/* update timestamp */
s->timestamp += n / sample_size;
}
}
-}
+}
/* NOTE: we suppose that exactly one frame is given as argument here */
/* XXX: test it */
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);
+ 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 +
- (s->cur_timestamp * 90000LL) / st->codec.sample_rate;
+ s->timestamp = s->base_timestamp +
+ (s->cur_timestamp * 90000LL) / st->codec->sample_rate;
}
}
s->buf[2] = count >> 8;
s->buf[3] = count;
memcpy(s->buf + 4, buf1, len);
- rtp_send_data(s1, s->buf, len + 4);
+ rtp_send_data(s1, s->buf, len + 4, 0);
size -= len;
buf1 += len;
count += len;
memcpy(s->buf_ptr, buf1, size);
s->buf_ptr += size;
}
- s->cur_timestamp += st->codec.frame_size;
+ s->cur_timestamp += st->codec->frame_size;
}
/* NOTE: a single frame must be passed with sequence header if
while (size > 0) {
/* XXX: more correct headers */
h = 0;
- if (st->codec.sub_id == 2)
+ if (st->codec->sub_id == 2)
h |= 1 << 26; /* mpeg 2 indicator */
q = s->buf;
*q++ = h >> 24;
*q++ = h >> 8;
*q++ = h;
- if (st->codec.sub_id == 2) {
+ 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;
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);
+ 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;
len = size;
/* 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);
+ 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));
buf1 += len;
size -= len;
buf1 += len;
size -= len;
s->buf_ptr += len;
-
+
out_len = s->buf_ptr - s->buf;
if (out_len >= s->max_payload_size) {
- rtp_send_data(s1, s->buf, out_len);
+ rtp_send_data(s1, s->buf, out_len, 0);
s->buf_ptr = s->buf;
}
}
int64_t ntp_time;
int size= pkt->size;
uint8_t *buf1= pkt->data;
-
+
#ifdef DEBUG
printf("%d: write len=%d\n", pkt->stream_index, size);
#endif
/* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */
- rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
+ rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) /
RTCP_TX_RATIO_DEN;
if (s->first_packet || rtcp_bytes >= 28) {
/* compute NTP time */
/* XXX: 90 kHz timestamp hardcoded */
ntp_time = (pkt->pts << 28) / 5625;
- rtcp_send_sr(s1, ntp_time);
+ rtcp_send_sr(s1, ntp_time);
s->last_octet_count = s->octet_count;
s->first_packet = 0;
}
- switch(st->codec.codec_id) {
+ switch(st->codec->codec_id) {
case CODEC_ID_PCM_MULAW:
case CODEC_ID_PCM_ALAW:
case CODEC_ID_PCM_U8:
case CODEC_ID_PCM_S8:
- rtp_send_samples(s1, buf1, size, 1 * st->codec.channels);
+ rtp_send_samples(s1, buf1, size, 1 * st->codec->channels);
break;
case CODEC_ID_PCM_U16BE:
case CODEC_ID_PCM_U16LE:
case CODEC_ID_PCM_S16BE:
case CODEC_ID_PCM_S16LE:
- rtp_send_samples(s1, buf1, size, 2 * st->codec.channels);
+ rtp_send_samples(s1, buf1, size, 2 * st->codec->channels);
break;
case CODEC_ID_MP2:
case CODEC_ID_MP3:
return 0;
}
-AVOutputFormat rtp_mux = {
+AVOutputFormat rtp_muxer = {
"rtp",
"RTP output format",
NULL,
rtp_write_packet,
rtp_write_trailer,
};
-
-int rtp_init(void)
-{
- av_register_output_format(&rtp_mux);
- return 0;
-}
projects / ffmpeg / blobdiff