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simpler bandwidth allocation for RTSP streaming - use av_read_frame() - initial support for raw MPEG2 transport stream streaming using RTSP

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Originally committed as revision 2506 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Fabrice Bellard 2003-11-10 19:02:56 +00:00
parent 72ea344bd1
commit e240a0bbe0
1 changed files with 138 additions and 266 deletions
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ffserver.c
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@ -51,9 +51,6 @@ enum HTTPState {
HTTPSTATE_SEND_DATA_TRAILER,
HTTPSTATE_RECEIVE_DATA,
HTTPSTATE_WAIT_FEED, /* wait for data from the feed */
HTTPSTATE_WAIT, /* wait before sending next packets */
HTTPSTATE_WAIT_SHORT, /* short wait for short term
bandwidth limitation */
HTTPSTATE_READY,
RTSPSTATE_WAIT_REQUEST,
@ -70,8 +67,6 @@ const char *http_state[] = {
"SEND_DATA_TRAILER",
"RECEIVE_DATA",
"WAIT_FEED",
"WAIT",
"WAIT_SHORT",
"READY",
"RTSP_WAIT_REQUEST",
@ -113,8 +108,13 @@ typedef struct HTTPContext {
AVFormatContext *fmt_in;
long start_time; /* In milliseconds - this wraps fairly often */
int64_t first_pts; /* initial pts value */
int64_t cur_pts; /* current pts value */
int pts_stream_index; /* stream we choose as clock reference */
int64_t cur_pts; /* current pts value from the stream in us */
int64_t cur_frame_duration; /* duration of the current frame in us */
int cur_frame_bytes; /* output frame size, needed to compute
the time at which we send each
packet */
int pts_stream_index; /* stream we choose as clock reference */
int64_t cur_clock; /* current clock reference value in us */
/* output format handling */
struct FFStream *stream;
/* -1 is invalid stream */
@ -138,15 +138,12 @@ typedef struct HTTPContext {
uint8_t *pb_buffer; /* XXX: use that in all the code */
ByteIOContext *pb;
int seq; /* RTSP sequence number */
/* RTP state specific */
enum RTSPProtocol rtp_protocol;
char session_id[32]; /* session id */
AVFormatContext *rtp_ctx[MAX_STREAMS];
/* RTP short term bandwidth limitation */
int packet_byte_count;
int packet_start_time_us; /* used for short durations (a few
seconds max) */
/* RTP/UDP specific */
URLContext *rtp_handles[MAX_STREAMS];
@ -183,6 +180,8 @@ typedef struct FFStream {
char filename[1024]; /* stream filename */
struct FFStream *feed; /* feed we are using (can be null if
coming from file) */
AVFormatParameters *ap_in; /* input parameters */
AVInputFormat *ifmt; /* if non NULL, force input format */
AVOutputFormat *fmt;
IPAddressACL *acl;
int nb_streams;
@ -247,7 +246,6 @@ static void compute_stats(HTTPContext *c);
static int open_input_stream(HTTPContext *c, const char *info);
static int http_start_receive_data(HTTPContext *c);
static int http_receive_data(HTTPContext *c);
static int compute_send_delay(HTTPContext *c);
/* RTSP handling */
static int rtsp_parse_request(HTTPContext *c);
@ -572,9 +570,12 @@ static int http_server(void)
poll_entry->events = POLLOUT;
poll_entry++;
} else {
/* not strictly correct, but currently cannot add
more than one fd in poll entry */
delay = 0;
/* when ffserver is doing the timing, we work by
looking at which packet need to be sent every
10 ms */
delay1 = 10; /* one tick wait XXX: 10 ms assumed */
if (delay1 < delay)
delay = delay1;
}
break;
case HTTPSTATE_WAIT_REQUEST:
@ -587,18 +588,6 @@ static int http_server(void)
poll_entry->events = POLLIN;/* Maybe this will work */
poll_entry++;
break;
case HTTPSTATE_WAIT:
c->poll_entry = NULL;
delay1 = compute_send_delay(c);
if (delay1 < delay)
delay = delay1;
break;
case HTTPSTATE_WAIT_SHORT:
c->poll_entry = NULL;
delay1 = 10; /* one tick wait XXX: 10 ms assumed */
if (delay1 < delay)
delay = delay1;
break;
default:
c->poll_entry = NULL;
break;
@ -896,16 +885,6 @@ static int handle_connection(HTTPContext *c)
/* nothing to do, we'll be waken up by incoming feed packets */
break;
case HTTPSTATE_WAIT:
/* if the delay expired, we can send new packets */
if (compute_send_delay(c) <= 0)
c->state = HTTPSTATE_SEND_DATA;
break;
case HTTPSTATE_WAIT_SHORT:
/* just return back to send data */
c->state = HTTPSTATE_SEND_DATA;
break;
case RTSPSTATE_SEND_REPLY:
if (c->poll_entry->revents & (POLLERR | POLLHUP)) {
av_freep(&c->pb_buffer);
@ -1695,6 +1674,9 @@ static void compute_stats(HTTPContext *c)
video_codec_name = codec->name;
}
break;
case CODEC_TYPE_DATA:
video_bit_rate += st->codec.bit_rate;
break;
default:
av_abort();
}
@ -1934,7 +1916,8 @@ static int open_input_stream(HTTPContext *c, const char *info)
#endif
/* open stream */
if (av_open_input_file(&s, input_filename, NULL, buf_size, NULL) < 0) {
if (av_open_input_file(&s, input_filename, c->stream->ifmt,
buf_size, c->stream->ap_in) < 0) {
http_log("%s not found", input_filename);
return -1;
}
@ -1954,191 +1937,41 @@ static int open_input_stream(HTTPContext *c, const char *info)
}
}
#if 0
if (c->fmt_in->iformat->read_seek) {
c->fmt_in->iformat->read_seek(c->fmt_in, stream_pos);
}
#endif
/* set the start time (needed for maxtime and RTP packet timing) */
c->start_time = cur_time;
c->first_pts = AV_NOPTS_VALUE;
return 0;
}
/* currently desactivated because the new PTS handling is not
satisfactory yet */
//#define AV_READ_FRAME
#ifdef AV_READ_FRAME
/* XXX: generalize that in ffmpeg for picture/audio/data. Currently
the return packet MUST NOT be freed */
int av_read_frame(AVFormatContext *s, AVPacket *pkt)
/* return the server clock (in us) */
static int64_t get_server_clock(HTTPContext *c)
{
AVStream *st;
int len, ret, old_nb_streams, i;
/* see if remaining frames must be parsed */
for(;;) {
if (s->cur_len > 0) {
st = s->streams[s->cur_pkt.stream_index];
len = avcodec_parse_frame(&st->codec, &pkt->data, &pkt->size,
s->cur_ptr, s->cur_len);
if (len < 0) {
/* error: get next packet */
s->cur_len = 0;
} else {
s->cur_ptr += len;
s->cur_len -= len;
if (pkt->size) {
/* init pts counter if not done */
if (st->pts.den == 0) {
switch(st->codec.codec_type) {
case CODEC_TYPE_AUDIO:
st->pts_incr = (int64_t)s->pts_den;
av_frac_init(&st->pts, st->pts.val, 0,
(int64_t)s->pts_num * st->codec.sample_rate);
break;
case CODEC_TYPE_VIDEO:
st->pts_incr = (int64_t)s->pts_den * st->codec.frame_rate_base;
av_frac_init(&st->pts, st->pts.val, 0,
(int64_t)s->pts_num * st->codec.frame_rate);
break;
default:
av_abort();
}
}
/* a frame was read: return it */
pkt->pts = st->pts.val;
#if 0
printf("add pts=%Lx num=%Lx den=%Lx incr=%Lx\n",
st->pts.val, st->pts.num, st->pts.den, st->pts_incr);
#endif
switch(st->codec.codec_type) {
case CODEC_TYPE_AUDIO:
av_frac_add(&st->pts, st->pts_incr * st->codec.frame_size);
break;
case CODEC_TYPE_VIDEO:
av_frac_add(&st->pts, st->pts_incr);
break;
default:
av_abort();
}
pkt->stream_index = s->cur_pkt.stream_index;
/* we use the codec indication because it is
more accurate than the demux flags */
pkt->flags = 0;
if (st->codec.coded_frame->key_frame)
pkt->flags |= PKT_FLAG_KEY;
return 0;
}
}
} else {
/* free previous packet */
av_free_packet(&s->cur_pkt);
old_nb_streams = s->nb_streams;
ret = av_read_packet(s, &s->cur_pkt);
if (ret)
return ret;
/* open parsers for each new streams */
for(i = old_nb_streams; i < s->nb_streams; i++)
open_parser(s, i);
st = s->streams[s->cur_pkt.stream_index];
/* update current pts (XXX: dts handling) from packet, or
use current pts if none given */
if (s->cur_pkt.pts != AV_NOPTS_VALUE) {
av_frac_set(&st->pts, s->cur_pkt.pts);
} else {
s->cur_pkt.pts = st->pts.val;
}
if (!st->codec.codec) {
/* no codec opened: just return the raw packet */
*pkt = s->cur_pkt;
/* no codec opened: just update the pts by considering we
have one frame and free the packet */
if (st->pts.den == 0) {
switch(st->codec.codec_type) {
case CODEC_TYPE_AUDIO:
st->pts_incr = (int64_t)s->pts_den * st->codec.frame_size;
av_frac_init(&st->pts, st->pts.val, 0,
(int64_t)s->pts_num * st->codec.sample_rate);
break;
case CODEC_TYPE_VIDEO:
st->pts_incr = (int64_t)s->pts_den * st->codec.frame_rate_base;
av_frac_init(&st->pts, st->pts.val, 0,
(int64_t)s->pts_num * st->codec.frame_rate);
break;
default:
av_abort();
}
}
av_frac_add(&st->pts, st->pts_incr);
return 0;
} else {
s->cur_ptr = s->cur_pkt.data;
s->cur_len = s->cur_pkt.size;
}
}
}
}
static int compute_send_delay(HTTPContext *c)
{
int64_t cur_pts, delta_pts, next_pts;
int delay1;
/* compute current pts value from system time */
cur_pts = ((int64_t)(cur_time - c->start_time) * c->fmt_in->pts_den) /
(c->fmt_in->pts_num * 1000LL);
/* compute the delta from the stream we choose as
main clock (we do that to avoid using explicit
buffers to do exact packet reordering for each
stream */
/* XXX: really need to fix the number of streams */
if (c->pts_stream_index >= c->fmt_in->nb_streams)
next_pts = cur_pts;
else
next_pts = c->fmt_in->streams[c->pts_stream_index]->pts.val;
delta_pts = next_pts - cur_pts;
if (delta_pts <= 0) {
delay1 = 0;
} else {
delay1 = (delta_pts * 1000 * c->fmt_in->pts_num) / c->fmt_in->pts_den;
}
return delay1;
return (int64_t)(cur_time - c->start_time) * 1000LL;
}
#else
/* just fall backs */
static int av_read_frame(AVFormatContext *s, AVPacket *pkt)
/* return the estimated time at which the current packet must be sent
(in us) */
static int64_t get_packet_send_clock(HTTPContext *c)
{
return av_read_packet(s, pkt);
}
static int compute_send_delay(HTTPContext *c)
{
int datarate = 8 * get_longterm_datarate(&c->datarate, c->data_count);
int64_t delta_pts;
int64_t time_pts;
int m_delay;
if (datarate > c->stream->bandwidth * 2000) {
return 1000;
}
if (!c->stream->feed && c->first_pts!=AV_NOPTS_VALUE) {
time_pts = ((int64_t)(cur_time - c->start_time) * c->fmt_in->pts_den) /
((int64_t) c->fmt_in->pts_num*1000);
delta_pts = c->cur_pts - time_pts;
m_delay = (delta_pts * 1000 * c->fmt_in->pts_num) / c->fmt_in->pts_den;
return m_delay>0 ? m_delay : 0;
} else {
return 0;
}
}
#endif
int bytes_left, bytes_sent, frame_bytes;
frame_bytes = c->cur_frame_bytes;
if (frame_bytes <= 0) {
return c->cur_pts;
} else {
bytes_left = c->buffer_end - c->buffer_ptr;
bytes_sent = frame_bytes - bytes_left;
return c->cur_pts + (c->cur_frame_duration * bytes_sent) / frame_bytes;
}
}
static int http_prepare_data(HTTPContext *c)
{
int i, len, ret;
@ -2214,12 +2047,6 @@ static int http_prepare_data(HTTPContext *c)
/* We have timed out */
c->state = HTTPSTATE_SEND_DATA_TRAILER;
} else {
if (1 || c->is_packetized) {
if (compute_send_delay(c) > 0) {
c->state = HTTPSTATE_WAIT;
return 1; /* state changed */
}
}
redo:
if (av_read_frame(c->fmt_in, &pkt) < 0) {
if (c->stream->feed && c->stream->feed->feed_opened) {
@ -2243,10 +2070,9 @@ static int http_prepare_data(HTTPContext *c)
} else {
/* update first pts if needed */
if (c->first_pts == AV_NOPTS_VALUE) {
c->first_pts = pkt.pts;
c->first_pts = pkt.dts;
c->start_time = cur_time;
}
c->cur_pts = pkt.pts;
/* send it to the appropriate stream */
if (c->stream->feed) {
/* if coming from a feed, select the right stream */
@ -2290,6 +2116,22 @@ static int http_prepare_data(HTTPContext *c)
output stream (one for each RTP
connection). XXX: need more abstract handling */
if (c->is_packetized) {
AVStream *st;
/* compute send time and duration */
st = c->fmt_in->streams[pkt.stream_index];
c->cur_pts = pkt.dts;
if (st->start_time != AV_NOPTS_VALUE)
c->cur_pts -= st->start_time;
c->cur_frame_duration = pkt.duration;
#if 0
printf("index=%d pts=%0.3f duration=%0.6f\n",
pkt.stream_index,
(double)c->cur_pts /
AV_TIME_BASE,
(double)c->cur_frame_duration /
AV_TIME_BASE);
#endif
/* find RTP context */
c->packet_stream_index = pkt.stream_index;
ctx = c->rtp_ctx[c->packet_stream_index];
if(!ctx) {
@ -2306,14 +2148,6 @@ static int http_prepare_data(HTTPContext *c)
}
codec->coded_frame->key_frame = ((pkt.flags & PKT_FLAG_KEY) != 0);
#ifdef PJSG
if (codec->codec_type == CODEC_TYPE_AUDIO) {
codec->frame_size = (codec->sample_rate * pkt.duration + 500000) / 1000000;
/* printf("Calculated size %d, from sr %d, duration %d\n", codec->frame_size, codec->sample_rate, pkt.duration); */
}
#endif
if (c->is_packetized) {
int max_packet_size;
if (c->rtp_protocol == RTSP_PROTOCOL_RTP_TCP)
@ -2321,8 +2155,6 @@ static int http_prepare_data(HTTPContext *c)
else
max_packet_size = url_get_max_packet_size(c->rtp_handles[c->packet_stream_index]);
ret = url_open_dyn_packet_buf(&ctx->pb, max_packet_size);
c->packet_byte_count = 0;
c->packet_start_time_us = av_gettime();
} else {
ret = url_open_dyn_buf(&ctx->pb);
}
@ -2335,14 +2167,15 @@ static int http_prepare_data(HTTPContext *c)
}
len = url_close_dyn_buf(&ctx->pb, &c->pb_buffer);
c->cur_frame_bytes = len;
c->buffer_ptr = c->pb_buffer;
c->buffer_end = c->pb_buffer + len;
codec->frame_number++;
if (len == 0)
goto redo;
}
#ifndef AV_READ_FRAME
av_free_packet(&pkt);
#endif
}
}
}
@ -2377,7 +2210,7 @@ static int http_prepare_data(HTTPContext *c)
(either UDP or TCP connection) */
static int http_send_data(HTTPContext *c)
{
int len, ret, dt;
int len, ret;
for(;;) {
if (c->buffer_ptr >= c->buffer_end) {
@ -2404,7 +2237,16 @@ static int http_send_data(HTTPContext *c)
(c->buffer_ptr[3]);
if (len > (c->buffer_end - c->buffer_ptr))
goto fail1;
if ((get_packet_send_clock(c) - get_server_clock(c)) > 0) {
/* nothing to send yet: we can wait */
return 0;
}
c->data_count += len;
update_datarate(&c->datarate, c->data_count);
if (c->stream)
c->stream->bytes_served += len;
if (c->rtp_protocol == RTSP_PROTOCOL_RTP_TCP) {
/* RTP packets are sent inside the RTSP TCP connection */
ByteIOContext pb1, *pb = &pb1;
@ -2439,28 +2281,32 @@ static int http_send_data(HTTPContext *c)
/* prepare asynchronous TCP sending */
rtsp_c->packet_buffer_ptr = c->packet_buffer;
rtsp_c->packet_buffer_end = c->packet_buffer + size;
rtsp_c->state = RTSPSTATE_SEND_PACKET;
c->buffer_ptr += len;
/* send everything we can NOW */
len = write(rtsp_c->fd, rtsp_c->packet_buffer_ptr,
rtsp_c->packet_buffer_end - rtsp_c->packet_buffer_ptr);
if (len > 0) {
rtsp_c->packet_buffer_ptr += len;
}
if (rtsp_c->packet_buffer_ptr < rtsp_c->packet_buffer_end) {
/* if we could not send all the data, we will
send it later, so a new state is needed to
"lock" the RTSP TCP connection */
rtsp_c->state = RTSPSTATE_SEND_PACKET;
break;
} else {
/* all data has been sent */
av_freep(&c->packet_buffer);
}
} else {
/* send RTP packet directly in UDP */
/* short term bandwidth limitation */
dt = av_gettime() - c->packet_start_time_us;
if (dt < 1)
dt = 1;
if ((c->packet_byte_count + len) * (int64_t)1000000 >=
(SHORT_TERM_BANDWIDTH / 8) * (int64_t)dt) {
/* bandwidth overflow : wait at most one tick and retry */
c->state = HTTPSTATE_WAIT_SHORT;
return 0;
}
c->buffer_ptr += 4;
url_write(c->rtp_handles[c->packet_stream_index],
c->buffer_ptr, len);
c->buffer_ptr += len;
/* here we continue as we can send several packets per 10 ms slot */
}
c->buffer_ptr += len;
c->packet_byte_count += len;
} else {
/* TCP data output */
len = write(c->fd, c->buffer_ptr, c->buffer_end - c->buffer_ptr);
@ -2474,12 +2320,12 @@ static int http_send_data(HTTPContext *c)
} else {
c->buffer_ptr += len;
}
c->data_count += len;
update_datarate(&c->datarate, c->data_count);
if (c->stream)
c->stream->bytes_served += len;
break;
}
c->data_count += len;
update_datarate(&c->datarate, c->data_count);
if (c->stream)
c->stream->bytes_served += len;
break;
}
} /* for(;;) */
return 0;
@ -2775,19 +2621,23 @@ static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer,
url_fprintf(pb, "c=IN IP4 %s\n", inet_ntoa(stream->multicast_ip));
}
/* for each stream, we output the necessary info */
private_payload_type = 96;
private_payload_type = RTP_PT_PRIVATE;
for(i = 0; i < stream->nb_streams; i++) {
st = stream->streams[i];
switch(st->codec.codec_type) {
case CODEC_TYPE_AUDIO:
mediatype = "audio";
break;
case CODEC_TYPE_VIDEO:
if (st->codec.codec_id == CODEC_ID_MPEG2TS) {
mediatype = "video";
break;
default:
mediatype = "application";
break;
} else {
switch(st->codec.codec_type) {
case CODEC_TYPE_AUDIO:
mediatype = "audio";
break;
case CODEC_TYPE_VIDEO:
mediatype = "video";
break;
default:
mediatype = "application";
break;
}
}
/* NOTE: the port indication is not correct in case of
unicast. It is not an issue because RTSP gives it */
@ -2801,7 +2651,7 @@ static int prepare_sdp_description(FFStream *stream, uint8_t **pbuffer,
}
url_fprintf(pb, "m=%s %d RTP/AVP %d\n",
mediatype, port, payload_type);
if (payload_type >= 96) {
if (payload_type >= RTP_PT_PRIVATE) {
/* for private payload type, we need to give more info */
switch(st->codec.codec_id) {
case CODEC_ID_MPEG4:
@ -2874,7 +2724,6 @@ static void rtsp_cmd_describe(HTTPContext *c, const char *url)
/* get the host IP */
len = sizeof(my_addr);
getsockname(c->fd, (struct sockaddr *)&my_addr, &len);
content_length = prepare_sdp_description(stream, &content, my_addr.sin_addr);
if (content_length < 0) {
rtsp_reply_error(c, RTSP_STATUS_INTERNAL);
@ -3116,6 +2965,14 @@ static void rtsp_cmd_play(HTTPContext *c, const char *url, RTSPHeader *h)
return;
}
#if 0
/* XXX: seek in stream */
if (h->range_start != AV_NOPTS_VALUE) {
printf("range_start=%0.3f\n", (double)h->range_start / AV_TIME_BASE);
av_seek_frame(rtp_c->fmt_in, -1, h->range_start);
}
#endif
rtp_c->state = HTTPSTATE_SEND_DATA;
/* now everything is OK, so we can send the connection parameters */
@ -3477,8 +3334,16 @@ static void build_file_streams(void)
/* the stream comes from a file */
/* try to open the file */
/* open stream */
stream->ap_in = av_mallocz(sizeof(AVFormatParameters));
if (stream->fmt == &rtp_mux) {
/* specific case : if transport stream output to RTP,
we use a raw transport stream reader */
stream->ap_in->mpeg2ts_raw = 1;
stream->ap_in->mpeg2ts_compute_pcr = 1;
}
if (av_open_input_file(&infile, stream->feed_filename,
NULL, 0, NULL) < 0) {
stream->ifmt, 0, stream->ap_in) < 0) {
http_log("%s not found", stream->feed_filename);
/* remove stream (no need to spend more time on it) */
fail:
@ -3554,7 +3419,8 @@ static void build_feed_streams(void)
if (sf->index != ss->index ||
sf->id != ss->id) {
printf("Index & Id do not match for stream %d\n", i);
printf("Index & Id do not match for stream %d (%s)\n",
i, feed->feed_filename);
matches = 0;
} else {
AVCodecContext *ccf, *ccs;
@ -4091,6 +3957,12 @@ static int parse_ffconfig(const char *filename)
audio_id = stream->fmt->audio_codec;
video_id = stream->fmt->video_codec;
}
} else if (!strcasecmp(cmd, "InputFormat")) {
stream->ifmt = av_find_input_format(arg);
if (!stream->ifmt) {
fprintf(stderr, "%s:%d: Unknown input format: %s\n",
filename, line_num, arg);
}
} else if (!strcasecmp(cmd, "FaviconURL")) {
if (stream && stream->stream_type == STREAM_TYPE_STATUS) {
get_arg(stream->feed_filename, sizeof(stream->feed_filename), &p);