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standards - rtsp with new audio and video refactors and sync standard working

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This commit is contained in:
Gerard CL
2014-05-20 16:51:35 +02:00
parent 4361eef2d1
commit e322ffdbde
23 changed files with 906 additions and 636 deletions
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37
src/main.cpp
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@@ -210,7 +210,7 @@ static void usage(void)
printf("\n");
printf("\t-c <cfg> \tcompress video (see '-c help')\n");
printf("\n");
printf("\t--rtsp-server \t\tRTSP server: dynamically serving H264 RTP standard transport\n");
printf("\t--rtsp-server \tRTSP server: dynamically serving H264 RTP standard transport (use '--rtps-server=help' to see usage)\n");
printf("\n");
printf("\t-i|--sage[=<opts>] \tiHDTV compatibility mode / SAGE TX\n");
printf("\n");
@@ -434,6 +434,8 @@ int main(int argc, char *argv[])
const char *requested_audio_fec = DEFAULT_AUDIO_FEC;
char *audio_channel_map = NULL;
const char *audio_scale = "mixauto";
rtsp_serv_t* rtsp_server = NULL;
int rtsp_port = 0;
bool isStd = FALSE;
int recv_port_number = PORT_BASE;
int send_port_number = PORT_BASE;
@@ -464,6 +466,7 @@ int main(int argc, char *argv[])
bool receiver_thread_started = false,
capture_thread_started = false;
unsigned display_flags = 0;
int compressed_audio_sample_rate = 48000;
int ret;
struct vidcap_params *audio_cap_dev;
long packet_rate;
@@ -500,7 +503,7 @@ int main(int argc, char *argv[])
{"compress", required_argument, 0, 'c'},
{"ihdtv", no_argument, 0, 'i'},
{"sage", optional_argument, 0, 'S'},
{"rtsp-server", no_argument, 0, 'H'},
{"rtsp-server", optional_argument, 0, 'H'},
{"receive", required_argument, 0, 'r'},
{"send", required_argument, 0, 's'},
{"help", no_argument, 0, 'h'},
@@ -608,7 +611,16 @@ int main(int argc, char *argv[])
break;
case 'H':
video_protocol = H264_STD;
//h264_opts = optarg;
if (optarg == NULL) {
rtsp_port = 0;
} else {
if (!strcmp(optarg, "help")) {
rtps_server_usage();
return 0;
}
rtsp_port = get_rtsp_server_port(optarg);
if (rtsp_port == -1) return 0;
}
break;
case 'r':
audio_recv = optarg;
@@ -956,20 +968,17 @@ int main(int argc, char *argv[])
display_device, requested_mtu,
argc, argv);
}else if (video_protocol == H264_STD) {
rtps_types_t avType;
if(strcmp("none", vidcap_params_get_driver(vidcap_params_head)) != 0 && (strcmp("none",audio_send) != 0)) avType = avStdDyn; //AVStream
else if((strcmp("none",audio_send) != 0)) avType = audioPCMUdyn; //AStream
else if(strcmp("none", vidcap_params_get_driver(vidcap_params_head))) avType = videoH264; //VStream
else {
printf("[RTSP SERVER CHECK] no stream type... check capture devices input...\n");
return EXIT_FAIL_USAGE;
}
rtps_types_t avType;
if(strcmp("none", vidcap_params_get_driver(vidcap_params_head)) != 0 && (strcmp("none",audio_send) != 0)) avType = av; //AVStream
else if((strcmp("none",audio_send) != 0)) avType = audio; //AStream
else if(strcmp("none", vidcap_params_get_driver(vidcap_params_head))) avType = video; //VStream
else printf("[RTSP SERVER CHECK] no stream type... check capture devices input...\n");
uv->state_video_rxtx = new h264_rtp_video_rxtx(&root_mod, video_exporter,
requested_compression, requested_encryption,
requested_receiver, recv_port_number, send_port_number,
ipv6, requested_mcast_if, requested_video_fec, requested_mtu,
packet_rate, avType);
packet_rate, avType, get_audio_codec(audio_codec), compressed_audio_sample_rate, audio_capture_channels, 2 /*bps*/, rtsp_port);
} else if (video_protocol == ULTRAGRID_RTP) {
uv->state_video_rxtx = new ultragrid_rtp_video_rxtx(&root_mod, video_exporter,
requested_compression, requested_encryption,
@@ -1065,6 +1074,10 @@ cleanup:
vidcap_params_head = next;
}
#ifdef HAVE_RTSP_SERVER
if(rtsp_server) c_stop_server(rtsp_server);
#endif
module_done(&root_mod);
free(uv);