static void
rtsp_cmd_play(RTSPContext *ctx, const char *url, RTSPMessageHeader *h) {
char path[4096];
//
av_url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path, sizeof(path), url);
if(strncmp(path, rtspconf->object, strlen(rtspconf->object)) != 0) {
rtsp_reply_error(ctx, RTSP_STATUS_SESSION);
return;
}
if(strcmp(ctx->session_id, h->session_id) != 0) {
rtsp_reply_error(ctx, RTSP_STATUS_SESSION);
return;
}
//
if(ctx->state != SERVER_STATE_READY
&& ctx->state != SERVER_STATE_PAUSE) {
rtsp_reply_error(ctx, RTSP_STATUS_STATE);
return;
}
// 2014-05-20: support only shared-encoder model
if(ff_server_register_client(ctx) < 0) {
ga_error("cannot register encoder client.\n");
rtsp_reply_error(ctx, RTSP_STATUS_INTERNAL);
return;
}
//
ctx->state = SERVER_STATE_PLAYING;
rtsp_reply_header(ctx, RTSP_STATUS_OK);
rtsp_printf(ctx, "Session: %s\r\n", ctx->session_id);
rtsp_printf(ctx, "\r\n");
return;
}
static void
rtsp_cmd_pause(RTSPContext *ctx, const char *url, RTSPMessageHeader *h) {
char path[4096];
//
av_url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path, sizeof(path), url);
if(strncmp(path, rtspconf->object, strlen(rtspconf->object)) != 0) {
rtsp_reply_error(ctx, RTSP_STATUS_SESSION);
return;
}
if(strcmp(ctx->session_id, h->session_id) != 0) {
rtsp_reply_error(ctx, RTSP_STATUS_SESSION);
return;
}
//
if(ctx->state != SERVER_STATE_PLAYING) {
rtsp_reply_error(ctx, RTSP_STATUS_STATE);
return;
}
//
ctx->state = SERVER_STATE_PAUSE;
rtsp_reply_header(ctx, RTSP_STATUS_OK);
rtsp_printf(ctx, "Session: %s\r\n", ctx->session_id);
rtsp_printf(ctx, "\r\n");
return;
}
static void
rtsp_cmd_describe(RTSPContext *ctx, const char *url) {
struct sockaddr_in myaddr;
#ifdef WIN32
int addrlen;
#else
socklen_t addrlen;
#endif
char path[4096];
char content[4096];
int content_length;
//
av_url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path, sizeof(path), url);
if(strcmp(path, rtspconf->object) != 0) {
rtsp_reply_error(ctx, RTSP_STATUS_SERVICE);
return;
}
//
addrlen = sizeof(myaddr);
getsockname(ctx->fd, (struct sockaddr*) &myaddr, &addrlen);
content_length = prepare_sdp_description(ctx, content, sizeof(content));
if(content_length < 0) {
rtsp_reply_error(ctx, RTSP_STATUS_INTERNAL);
return;
}
// state does not change
rtsp_reply_header(ctx, RTSP_STATUS_OK);
rtsp_printf(ctx, "Content-Base: %s/\r\n", url);
rtsp_printf(ctx, "Content-Type: application/sdp\r\n");
rtsp_printf(ctx, "Content-Length: %d\r\n", content_length);
rtsp_printf(ctx, "\r\n");
rtsp_write(ctx, content, content_length);
return;
}
static void rtsp_cmd_interrupt(RTSPContext *c, const char *url, RTSPMessageHeader *h, int pause_only)
{
RTSPContext *rtp_c;
rtp_c = find_rtp_session_with_url(url, h->session_id);
if (!rtp_c) {
rtsp_reply_error(c, RTSP_STATUS_SESSION);
return;
}
if (pause_only) {
if (rtp_c->state != RTPSTATE_SEND_DATA) {
rtsp_reply_error(c, RTSP_STATUS_STATE);
return;
}
rtp_c->state = HTTPSTATE_READY;
c->ntp_start_time = 0.0;
}
rtsp_reply_header(c, RTSP_STATUS_OK);
/* session ID */
avio_printf(c->pb, "Session: %s\r\n", rtp_c->session_id);
avio_printf(c->pb, "\r\n");
if (!pause_only)
close_connection(rtp_c);
}
/* RTSP Handle Function */
static void rtsp_cmd_options(RTSPContext *c, const char *url)
{
rtsp_reply_header(c, RTSP_STATUS_OK);
avio_printf(c->pb, "RTSP/1.0 %d %s\r\n", RTSP_STATUS_OK, "OK");
avio_printf(c->pb, "CSeq: %d\r\n", c->seq);
avio_printf(c->pb, "Public: %s\r\n", "OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE");
avio_printf(c->pb, "\r\n");
}
static void rtsp_cmd_play(RTSPContext *c, const char *url, RTSPMessageHeader *h)
{
RTSPContext *rtp_c;
rtp_c = find_rtp_session_with_url(url, h->session_id);
if (!rtp_c) {
printf("can't find rtp session %s with url %s \n", h->session_id, url);
rtsp_reply_error(c, RTSP_STATUS_SESSION);
return;
}
rtp_c->start_time = cur_time; // set start time
// pos * 188이 offset!
int64_t last_offset = rtp_c->cur_offset;
printf("cmd play! cur_offset = %lld\n", rtp_c->cur_offset);
if (c->ntp_start_time != 0.0) {
// set start time for seek!
printf("find start time by ntp time!!! seek request! seek time = %lfs \n", c->ntp_start_time);
rtp_c->ntp_start_time = c->ntp_start_time;
rtp_c->cur_idx = get_closest_iframe_by_time(rtp_c->stream->first_idx, rtp_c->ntp_start_time);
if (rtp_c->cur_idx != NULL)
rtp_c->cur_offset = rtp_c->cur_idx->pos * 188;
// play time이 바뀐 경우, buffer도 비워줘야 함.
if (rtp_c->buffer_ptr >= rtp_c->buffer_end) {
printf("no need to reset rtp_c->buffer ptr \n");
} else {
printf("need to reset rtp_c->buffer ptr \n");
rtp_c->buffer_ptr = rtp_c->buffer_end;
av_free(c->pb_buffer);
}
} else {
rtp_c->cur_idx = NULL;
}
printf("cmd play! after find iframe cur_offset = %lld\n", rtp_c->cur_offset);
// reset served bytes for restart from pause state
rtp_c->served_bytes = 0;
rtp_c->state = RTPSTATE_SEND_DATA;
/* now everything is OK, so we can send the connection parameters */
rtsp_reply_header(c, RTSP_STATUS_OK);
/* session ID */
avio_printf(c->pb, "Session: %s\r\n", rtp_c->session_id);
if (rtp_c->cur_idx != NULL) {
avio_printf(c->pb, "Range: npt=%.3lf-\r\n", get_iIndex_PCR(rtp_c->cur_idx));
printf("range response! Range: npt=%.3lf-\n", get_iIndex_PCR(rtp_c->cur_idx));
avio_printf(c->pb, "RTP-Info: seq=%d\r\n", rtp_c->cur_seq);
printf("RTP info: seq=%d\n", rtp_c->cur_seq);
}
avio_printf(c->pb, "\r\n");
}
static void
rtsp_cmd_play(RTSPContext *ctx, const char *url, RTSPMessageHeader *h) {
char path[4096];
//
av_url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path, sizeof(path), url);
if(strncmp(path, rtspconf->object, strlen(rtspconf->object)) != 0) {
rtsp_reply_error(ctx, RTSP_STATUS_SESSION);
return;
}
if(strcmp(ctx->session_id, h->session_id) != 0) {
rtsp_reply_error(ctx, RTSP_STATUS_SESSION);
return;
}
//
if(ctx->state != SERVER_STATE_READY
&& ctx->state != SERVER_STATE_PAUSE) {
rtsp_reply_error(ctx, RTSP_STATUS_STATE);
return;
}
// create threads
#ifndef SHARE_ENCODER
if(pthread_create(&ctx->vthread, NULL, vencoder_thread, ctx) != 0) {
ga_error("cannot create video thread\n");
rtsp_reply_error(ctx, RTSP_STATUS_INTERNAL);
return;
}
#ifdef ENABLE_AUDIO
if(pthread_create(&ctx->athread, NULL, aencoder_thread, ctx) != 0) {
ga_error("cannot create audio thread\n");
rtsp_reply_error(ctx, RTSP_STATUS_INTERNAL);
return;
}
#endif /* ENABLE_AUDIO */
#else
if(encoder_register_client(ctx) < 0) {
ga_error("cannot register encoder client.\n");
rtsp_reply_error(ctx, RTSP_STATUS_INTERNAL);
return;
}
#endif /* SHARE_ENCODER */
//
ctx->state = SERVER_STATE_PLAYING;
rtsp_reply_header(ctx, RTSP_STATUS_OK);
rtsp_printf(ctx, "Session: %s\r\n", ctx->session_id);
rtsp_printf(ctx, "\r\n");
return;
}
static void rtsp_cmd_describe(RTSPContext *c, const char *url)
{
iStream *stream;
char path1[1024];
const char *path;
uint8_t *content;
int content_length = 0;
socklen_t len;
struct sockaddr_in my_addr;
/* find which URL is asked */
av_url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path1, sizeof(path1), url);
int l = strlen(path1);
if (path1[l-1] == '/') {
path1[l-1] = '\0';
}
path = path1;
if (*path == '/')
path++;
for(stream = first_stream; stream != NULL; stream = stream->next) {
if (!strcmp(path, stream->name)) {
goto found;
}
}
/* no stream found */
rtsp_reply_error(c, RTSP_STATUS_SERVICE); /* XXX: right error ? */
return;
found:
/* prepare the media description in SDP format */
content = av_mallocz(1024);
/* get the host IP */
len = sizeof(my_addr);
getsockname(c->fd, (struct sockaddr *)&my_addr, &len);
rtsp_reply_header(c, RTSP_STATUS_OK);
avio_printf(c->pb, "Content-Base: %s/\r\n", url);
avio_printf(c->pb, "Content-Type: application/sdp\r\n");
int added_length = 0;
// 여기에 sdp정보 추가 해야 함.
content_length = generate_sdp_context(stream, &content);
avio_printf(c->pb, "Content-Length: %d\r\n", content_length + added_length);
avio_printf(c->pb, "\r\n");
// content added
avio_write(c->pb, content, content_length + added_length);
av_free(content);
}
static void
rtsp_cmd_teardown(RTSPContext *ctx, const char *url, RTSPMessageHeader *h, int bruteforce) {
char path[4096];
//
av_url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path, sizeof(path), url);
if(strncmp(path, rtspconf->object, strlen(rtspconf->object)) != 0) {
rtsp_reply_error(ctx, RTSP_STATUS_SESSION);
return;
}
if(strcmp(ctx->session_id, h->session_id) != 0) {
rtsp_reply_error(ctx, RTSP_STATUS_SESSION);
return;
}
//
ctx->state = SERVER_STATE_TEARDOWN;
if(bruteforce != 0)
return;
// XXX: well, gently response
rtsp_reply_header(ctx, RTSP_STATUS_OK);
rtsp_printf(ctx, "Session: %s\r\n", ctx->session_id);
rtsp_printf(ctx, "\r\n");
return;
}
static void
rtsp_cmd_setup(RTSPContext *ctx, const char *url, RTSPMessageHeader *h) {
int i;
RTSPTransportField *th;
struct sockaddr_in destaddr, myaddr;
#ifdef WIN32
int destaddrlen, myaddrlen;
#else
socklen_t destaddrlen, myaddrlen;
#endif
char path[4096];
char channelname[VIDEO_SOURCE_CHANNEL_MAX+1][RTSP_STREAM_FORMAT_MAXLEN];
int baselen = strlen(rtspconf->object);
int streamid;
int rtp_port, rtcp_port;
enum RTSPStatusCode errcode;
//
av_url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path, sizeof(path), url);
for(i = 0; i < VIDEO_SOURCE_CHANNEL_MAX+1; i++) {
snprintf(channelname[i], RTSP_STREAM_FORMAT_MAXLEN, RTSP_STREAM_FORMAT, i);
}
//
if(strncmp(path, rtspconf->object, baselen) != 0) {
ga_error("invalid object (path=%s)\n", path);
rtsp_reply_error(ctx, RTSP_STATUS_AGGREGATE);
return;
}
for(i = 0; i < VIDEO_SOURCE_CHANNEL_MAX+1; i++) {
if(strcmp(path+baselen+1, channelname[i]) == 0) {
streamid = i;
break;
}
}
if(i == VIDEO_SOURCE_CHANNEL_MAX+1) {
// not found
ga_error("invalid service (path=%s)\n", path);
rtsp_reply_error(ctx, RTSP_STATUS_SERVICE);
return;
}
//
if(ctx->state != SERVER_STATE_IDLE
&& ctx->state != SERVER_STATE_READY) {
rtsp_reply_error(ctx, RTSP_STATUS_STATE);
return;
}
// create session id?
if(ctx->session_id == NULL) {
if(h->session_id[0] == '\0') {
snprintf(h->session_id, sizeof(h->session_id), "%04x%04x",
rand()%0x0ffff, rand()%0x0ffff);
ctx->session_id = strdup(h->session_id);
ga_error("New session created (id = %s)\n", ctx->session_id);
}
}
// session id must match -- we have only one session
if(ctx->session_id == NULL
|| strcmp(ctx->session_id, h->session_id) != 0) {
ga_error("Bad session id %s != %s\n", h->session_id, ctx->session_id);
errcode = RTSP_STATUS_SESSION;
goto error_setup;
}
// find supported transport
if((th = find_transport(h, RTSP_LOWER_TRANSPORT_UDP)) == NULL) {
th = find_transport(h, RTSP_LOWER_TRANSPORT_TCP);
}
if(th == NULL) {
ga_error("Cannot find transport\n");
errcode = RTSP_STATUS_TRANSPORT;
goto error_setup;
}
//
destaddrlen = sizeof(destaddr);
bzero(&destaddr, destaddrlen);
if(getpeername(ctx->fd, (struct sockaddr*) &destaddr, &destaddrlen) < 0) {
ga_error("Cannot get peername\n");
errcode = RTSP_STATUS_INTERNAL;
goto error_setup;
}
destaddr.sin_port = htons(th->client_port_min);
//
myaddrlen = sizeof(myaddr);
bzero(&myaddr, myaddrlen);
if(getsockname(ctx->fd, (struct sockaddr*) &myaddr, &myaddrlen) < 0) {
ga_error("Cannot get sockname\n");
errcode = RTSP_STATUS_INTERNAL;
goto error_setup;
}
//
ctx->lower_transport[streamid] = th->lower_transport;
if(rtp_new_av_stream(ctx, &destaddr, streamid,
streamid == video_source_channels()/*rtspconf->audio_id*/ ?
rtspconf->audio_encoder_codec->id : rtspconf->video_encoder_codec->id) < 0) {
ga_error("Create AV stream %d failed.\n", streamid);
errcode = RTSP_STATUS_TRANSPORT;
goto error_setup;
}
//
ctx->state = SERVER_STATE_READY;
rtsp_reply_header(ctx, RTSP_STATUS_OK);
rtsp_printf(ctx, "Session: %s\r\n", ctx->session_id);
switch(th->lower_transport) {
case RTSP_LOWER_TRANSPORT_UDP:
#ifdef HOLE_PUNCHING
rtp_port = ctx->rtpLocalPort[streamid*2];
rtcp_port = ctx->rtpLocalPort[streamid*2+1];
ctx->rtpPeerPort[streamid*2] = htons(th->client_port_min);
ctx->rtpPeerPort[streamid*2+1] = htons(th->client_port_max);
#else
rtp_port = ff_rtp_get_local_rtp_port((URLContext*) ctx->fmtctx[streamid]->pb->opaque);
rtcp_port = ff_rtp_get_local_rtcp_port((URLContext*) ctx->fmtctx[streamid]->pb->opaque);
#endif
ga_error("RTP/UDP: streamid=%d; client=%d-%d; server=%d-%d\n",
streamid,
th->client_port_min, th->client_port_max,
rtp_port, rtcp_port);
rtsp_printf(ctx, "Transport: RTP/AVP/UDP;unicast;client_port=%d-%d;server_port=%d-%d\r\n",
th->client_port_min, th->client_port_max,
rtp_port, rtcp_port);
break;
case RTSP_LOWER_TRANSPORT_TCP:
ga_error("RTP/TCP: interleaved=%d-%d\n",
streamid*2, streamid*2+1);
rtsp_printf(ctx, "Transport: RTP/AVP/TCP;unicast;interleaved=%d-%d\r\n",
streamid*2, streamid*2+1, streamid*2);
break;
default:
// should not happen
break;
}
rtsp_printf(ctx, "\r\n");
return;
error_setup:
if(ctx->session_id != NULL) {
free(ctx->session_id);
ctx->session_id = NULL;
}
if(ctx->encoder[streamid] != NULL) {
ctx->encoder[streamid] = NULL;
}
if(ctx->stream[streamid] != NULL) {
ctx->stream[streamid] = NULL;
}
if(ctx->fmtctx[streamid] != NULL) {
avformat_free_context(ctx->fmtctx[streamid]);
ctx->fmtctx[streamid] = NULL;
}
rtsp_reply_error(ctx, errcode);
return;
}
static void
rtsp_reply_error(RTSPContext *c, enum RTSPStatusCode error_number) {
rtsp_reply_header(c, error_number);
rtsp_printf(c, "\r\n");
}
static void rtsp_cmd_setup(RTSPContext *c, const char *url, RTSPMessageHeader *h)
{
iStream *stream;
int stream_index, rtp_port, rtcp_port;
char buf[1024];
char path1[1024];
char *path;
RTSPContext *rtp_c;
RTSPTransportField *th;
struct sockaddr_in dest_addr;
RTSPActionServerSetup setup;
/* find which URL is asked */
av_url_split(NULL, 0, NULL, 0, NULL, 0, NULL, path1, sizeof(path1), url);
int len = strlen(path1);
if (path1[len-1] == '/') {
path1[len-1] = '\0';
}
// / 뒤에꺼 없애자!
char *removed_path = strrchr(path1, '/');
if (removed_path != NULL && removed_path != path1)
{
path = path1;
path[removed_path-path1] = '\0';
} else {
path = path1;
}
if (*path == '/')
path++;
printf("rtsp setup! find stream name = %s\n", path);
/* now check each stream */
for(stream = first_stream; stream != NULL; stream = stream->next) {
/* accept aggregate filenames only if single stream */
if (!strcmp(path, stream->name)) {
goto found;
}
}
/* no stream found */
rtsp_reply_error(c, RTSP_STATUS_SERVICE); /* XXX: right error ? */
return;
found:
/* generate session id if needed */
if (h->session_id[0] == '\0') {
unsigned random0 = av_lfg_get(&random_state);
unsigned random1 = av_lfg_get(&random_state);
snprintf(h->session_id, sizeof(h->session_id), "%08x%08x",
random0, random1);
}
/* find RTP session, and create it if none found */
rtp_c = find_rtp_session(h->session_id);
if (!rtp_c) {
rtp_c = rtp_new_connection(&c->from_addr, stream, h->session_id);
if (!rtp_c) {
rtsp_reply_error(c, RTSP_STATUS_BANDWIDTH);
return;
}
}
/* test if stream is OK (test needed because several SETUP needs
to be done for a given file) */
if (rtp_c->stream != stream) {
rtsp_reply_error(c, RTSP_STATUS_SERVICE);
return;
}
// udp 로 전송 가능한지 체크.
th = find_transport(h);
if (!th) {
printf("can't do udp transport \n");
rtsp_reply_error(c, RTSP_STATUS_INTERNAL);
return;
}
/* setup default options */
setup.transport_option[0] = '\0';
dest_addr = rtp_c->from_addr;
dest_addr.sin_port = htons(th->client_port_min);
// setup stream
// open new rtp/udp socket!
if (rtp_new_stream(rtp_c, &dest_addr) < 0) {
rtsp_reply_error(c, RTSP_STATUS_TRANSPORT);
return;
}
rtp_c->to_addr = dest_addr;
/* now everything is OK, so we can send the connection parameters */
rtsp_reply_header(c, RTSP_STATUS_OK);
/* session ID */
avio_printf(c->pb, "Session: %s\r\n", rtp_c->session_id);
// rtcp 도 열어야되나? -_-;
rtp_port = rtp_c->udp_port;
rtcp_port = rtp_port+1; // 실제로 만들지는 않음..
avio_printf(c->pb, "Transport: RTP/AVP/UDP;unicast;"
"client_port=%d-%d;server_port=%d-%d",
th->client_port_min, th->client_port_max,
rtp_port, rtcp_port);
if (setup.transport_option[0] != '\0')
avio_printf(c->pb, ";%s", setup.transport_option);
avio_printf(c->pb, "\r\n");
avio_printf(c->pb, "\r\n");
}
