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MPEG4 streaming over RTP patch by (Luca Abeni: lucabe72, email it)

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Originally committed as revision 4469 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Luca Abeni 2005-07-23 21:48:58 +00:00 committed by Michael Niedermayer
parent b5bc8591ee
commit 003640635a
1 changed files with 8 additions and 8 deletions
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16
libavformat/rtp.c
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@ -585,7 +585,7 @@ 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)
static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m)
{
RTPDemuxContext *s = s1->priv_data;
@ -595,7 +595,7 @@ static void rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len)
/* 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);
@ -633,7 +633,7 @@ 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);
rtp_send_data(s1, s->buf, n, 0);
s->buf_ptr = s->buf;
/* update timestamp */
s->timestamp += n / sample_size;
@ -656,7 +656,7 @@ 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);
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 +
@ -678,7 +678,7 @@ 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);
rtp_send_data(s1, s->buf, len + 4, 0);
size -= len;
buf1 += len;
count += len;
@ -738,7 +738,7 @@ static void rtp_send_mpegvideo(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, s->buf, q - s->buf);
rtp_send_data(s1, s->buf, q - s->buf, 0);
buf1 += len;
size -= len;
@ -763,7 +763,7 @@ 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);
rtp_send_data(s1, buf1, len, (len == size));
buf1 += len;
size -= len;
@ -789,7 +789,7 @@ 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);
rtp_send_data(s1, s->buf, out_len, 0);
s->buf_ptr = s->buf;
}
}