voxl-streamer does not handle "teardown-request" signal correctly
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See https://forum.modalai.com/topic/1615/issue-with-voxl-streamer-not-reconnecting?_=1672332432641 for more context. It seems that the problem is that when a client initially disconnects, it returns a "teardown-request" not a "closed" signal, rtsp-client signal specifically, which never closes the client correctly. What happens then is that when the client tries to reconnect, it does not and disconnects automatically. Is there a better way of handling this, is there a way of closing the pipeline when a "teardown-request" is received. I have tried running the
gst_rtsp_client_closeand the
static gboolean remove_sessions (GstRTSPServer * server) { GstRTSPSessionPool *pool; pool = gst_rtsp_server_get_session_pool (server); guint removed = gst_rtsp_session_pool_cleanup (pool); g_object_unref (pool); M_PRINT("Removed %d sessions\n", removed); return TRUE; }but it seems not to work and just gives me the following error of pthread, is there something I am not taking into consideration?
Section of code that was modified from the original main.c file can be below:static gboolean remove_sessions (GstRTSPServer * server) { GstRTSPSessionPool *pool; pool = gst_rtsp_server_get_session_pool (server); guint removed = gst_rtsp_session_pool_cleanup (pool); g_object_unref (pool); M_PRINT("Removed %d sessions\n", removed); return TRUE; } static void rtsp_client_tear_down_request(GstRTSPClient* self, context_data *data) { M_PRINT("An existing client has requested tear down from the RTSP server\n"); // remove_sessions(data->rtsp_server); gst_rtsp_client_close(self); pthread_mutex_lock(&data->lock); data->num_rtsp_clients--; if ( ! data->num_rtsp_clients) { data->input_frame_number = 0; data->output_frame_number = 0; data->need_data = 0; data->initial_timestamp = 0; data->last_timestamp = 0; } pthread_mutex_unlock(&data->lock); } // This is called by the RTSP server when a client has connected. static void rtsp_client_connected(GstRTSPServer* self, GstRTSPClient* object, context_data *data) { M_PRINT("A new client has connected to the RTSP server\n"); pthread_mutex_lock(&data->lock); data->num_rtsp_clients++; // Install the disconnect callback with the client. g_signal_connect(object, "closed", G_CALLBACK(rtsp_client_disconnected), data); // Install tear down callback with the client g_signal_connect(object, "teardown-request", G_CALLBACK(rtsp_client_tear_down_request), data); pthread_mutex_unlock(&data->lock); }
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It seems that the problem is the first ip port never gets closed which causes a conflict.
ec 30 14:50:04 robot133 voxl-streamer[5967]: Stream available at rtsp://127.0.0.1:8900/live Dec 30 14:50:04 robot133 voxl-streamer[5967]: A new client has connected to the RTSP server Dec 30 14:50:04 robot133 voxl-streamer[5967]: [1]A new client rtsp://92.185.19.128:56804(null) has connected Dec 30 14:50:04 robot133 voxl-streamer[5967]: gbm_create_device(156): Info: backend name is: msm_drm Dec 30 14:50:04 robot133 voxl-streamer[5967]: gbm_create_device(156): Info: backend name is: msm_drm Dec 30 14:50:44 robot133 voxl-streamer[5967]: A new client has connected to the RTSP server Dec 30 14:50:44 robot133 voxl-streamer[5967]: [2]A new client rtsp://92.185.19.128:38300(null) has connected Dec 30 14:51:46 robot133 voxl-streamer[5967]: A new client has connected to the RTSP server
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For clarification, if you lose your stream via losing wifi connection or similar, the stream does not return when wifi returns?
This is RB5 drone and what does voxl-version say for full software listing?
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@RichieRich said in voxl-streamer does not handle "teardown-request" signal correctly:
listing
we were able to find a workaround.
The voxl version is the following.-------------------------------------------------------------------------------- system-image: 1.4.1-M0052-14.1a-perf kernel: #1 SMP PREEMPT Thu Oct 13 18:59:44 UTC 2022 4.19.125 -------------------------------------------------------------------------------- hw version: M0052 -------------------------------------------------------------------------------- voxl-suite: 0.9.4 -------------------------------------------------------------------------------- Packages: Repo: http://voxl-packages.modalai.com/ ./dists/qrb5165/sdk-0.9/binary-arm64/ Last Updated: 2022-12-27 18:53:03 List: libmodal-cv 0.2.3 libmodal-exposure 0.0.7 libmodal-journal 0.2.1 libmodal-json 0.4.3 libmodal-pipe 2.6.0 libqrb5165-io 0.1.0 libvoxl-cci-direct 0.1.5 libvoxl-cutils 0.1.1 mv-voxl 0.1-r0 qrb5165-bind 0.1-r0 qrb5165-dfs-server 0.1.0 qrb5165-imu-server 0.5.0 qrb5165-slpi-test-sig 01-r0 qrb5165-system-tweaks 0.1.5 qrb5165-tflite 2.8.0-2 voxl-bind 0.0.1 voxl-camera-calibration 0.2.3 voxl-camera-server 1.3.5 voxl-cpu-monitor 0.3.0 voxl-docker-support 1.2.4 voxl-gphoto2-server 0.0.10 voxl-jpeg-turbo 2.1.3-4 voxl-libgphoto2 0.0.4 voxl-libuvc 1.0.7 voxl-logger 0.3.4 voxl-mavlink 0.1.0 voxl-mavlink-server 0.2.0 voxl-modem 0.16.1 voxl-mongoose 7.7.0-1 voxl-mpa-to-ros 0.3.6 voxl-mpa-tools 0.7.6 voxl-opencv 4.5.5-1 voxl-portal 0.4.2 voxl-px4 1.12.31 voxl-px4-imu-server 0.1.2 voxl-qvio-server 0.8.2 voxl-remote-id 0.0.5 voxl-streamer 0.4.1 voxl-suite 0.9.4 voxl-tag-detector 0.0.4 voxl-tflite-server 0.3.1 voxl-utils 1.2.2 voxl-uvc-server 0.1.3 voxl-vision-px4 1.4.0We were able to solve the problem. @RichieRich . I will make a branch in voxl-streamer branch with the solution.
Since I cant push the changes to the voxl-streamer gitlab repo, I will add the files changed.
cmake_minimum_required(VERSION 3.3) project(voxl-streamer) add_definitions(-D_GNU_SOURCE) set(CMAKE_C_FLAGS "${CMAKE_C_FLAGS} -O0 -g -std=c11 -Wall -pedantic") set(CMAKE_C_FLAGS "${CMAKE_C_FLAGS} -D_XOPEN_SOURCE=500 -fno-strict-aliasing") set(CMAKE_C_FLAGS "${CMAKE_C_FLAGS} -fdata-sections -fno-zero-initialized-in-bss") set(CMAKE_C_FLAGS "${CMAKE_C_FLAGS} -Wall -Wextra -Wno-unused-parameter") set(CMAKE_C_FLAGS "${CMAKE_C_FLAGS} -Wno-unused-function") set(CMAKE_C_FLAGS "${CMAKE_C_FLAGS} -Wno-cast-align") set(CMAKE_C_FLAGS "${CMAKE_C_FLAGS} -Wno-missing-braces -Wno-strict-aliasing") set(CMAKE_C_FLAGS "${CMAKE_C_FLAGS} -D__FILENAME__='\"$(subst ${CMAKE_SOURCE_DIR}/,,$(abspath $<))\"'") include_directories( SYSTEM /usr/include/gstreamer-1.0 ) include_directories( SYSTEM /usr/include/glib-2.0 ) include_directories( include/ ) if(${BUILDSIZE} MATCHES 64) message("BUILDSIZE matches 64") include_directories( SYSTEM /usr/lib/aarch64-linux-gnu/glib-2.0/include ) find_library(MODAL_JSON modal_json HINTS /usr/lib64) find_library(MODAL_PIPE modal_pipe HINTS /usr/lib64) find_library(MODAL_JOURNAL modal_journal HINTS /usr/lib64) else() message("BUILDSIZE does not match 64") include_directories( SYSTEM /usr/lib/glib-2.0/include ) find_library(MODAL_JSON modal_json HINTS /usr/lib) find_library(MODAL_PIPE modal_pipe HINTS /usr/lib) find_library(MODAL_JOURNAL modal_journal HINTS /usr/lib) endif() add_executable( voxl-streamer src/util.c src/pipeline.c src/configuration.c src/main.c ) target_link_libraries(voxl-streamer pthread gstreamer-1.0 gstvideo-1.0 gstrtspserver-1.0 gobject-2.0 gstapp-1.0 glib-2.0 gstrtsp-1.0 ${MODAL_JSON} ${MODAL_PIPE} ${MODAL_JOURNAL} ) add_executable( voxl-configure-streamer config/voxl-configure-streamer.c ) target_link_libraries(voxl-configure-streamer ${MODAL_JSON} ) install( TARGETS voxl-streamer voxl-configure-streamer DESTINATION /usr/bin )main.c
/******************************************************************************* * Copyright 2020 ModalAI Inc. * * Redistribution and use in source and binary forms, with or without * modification, are permitted provided that the following conditions are met: * * 1. Redistributions of source code must retain the above copyright notice, * this list of conditions and the following disclaimer. * * 2. Redistributions in binary form must reproduce the above copyright notice, * this list of conditions and the following disclaimer in the documentation * and/or other materials provided with the distribution. * * 3. Neither the name of the copyright holder nor the names of its contributors * may be used to endorse or promote products derived from this software * without specific prior written permission. * * 4. The Software is used solely in conjunction with devices provided by * ModalAI Inc. * * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS" * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE * LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE * POSSIBILITY OF SUCH DAMAGE. ******************************************************************************/ #include <stdio.h> #include <stdint.h> #include <getopt.h> #include <string.h> #include <inttypes.h> #include <sys/types.h> #include <sys/stat.h> #include <fcntl.h> #include <unistd.h> #include <gst/gst.h> #include <gst/video/video.h> #include <gst/app/gstappsrc.h> #include <glib-object.h> #include <modal_pipe.h> #include <modal_journal.h> #include "context.h" #include "pipeline.h" #include "configuration.h" #include "gst/rtsp/gstrtspconnection.h" // This is the main data structure for the application. It is passed / shared // with other modules as needed. static context_data context; // called whenever we connect or reconnect to the server static void _cam_connect_cb(__attribute__((unused)) int ch, __attribute__((unused)) void* context) { M_PRINT("Camera server Connected\n"); } // called whenever we disconnect from the server static void _cam_disconnect_cb(__attribute__((unused)) int ch, __attribute__((unused)) void* context) { M_PRINT("Camera server Disconnected\n"); } // camera helper callback whenever a frame arrives static void _cam_helper_cb( __attribute__((unused))int ch, camera_image_metadata_t meta, char* frame, void* context) { static int dump_meta_data = 1; context_data *ctx = (context_data*) context; GstMapInfo info; GstFlowReturn status; if (dump_meta_data) { M_DEBUG("Meta data from incoming frame:\n"); M_DEBUG("\tmagic_number 0x%X \n", meta.magic_number); M_DEBUG("\ttimestamp_ns: %" PRIu64 "\n", meta.timestamp_ns); M_DEBUG("\tframe_id: %d\n", meta.frame_id); M_DEBUG("\twidth: %d\n", meta.width); M_DEBUG("\theight: %d\n", meta.height); M_DEBUG("\tsize_bytes: %d\n", meta.size_bytes); M_DEBUG("\tstride: %d\n", meta.stride); M_DEBUG("\texposure_ns: %d\n", meta.exposure_ns); M_DEBUG("\tgain: %d\n", meta.gain); M_DEBUG("\tformat: %d\n", meta.format); dump_meta_data = 0; } // Initialize the input parameters based on the incoming meta data if ( ! ctx->input_parameters_initialized) { // We need to estimate our frame rate. So get a timestamp from the // very first frame and then on the second frame figure out the delta time // between the two frames. Estimate frame rate based on that delta. if (ctx->last_timestamp == 0) { ctx->last_timestamp = (guint64) meta.timestamp_ns; M_DEBUG("First frame timestamp: %" PRIu64 "\n", ctx->last_timestamp); return; } else { M_DEBUG("Second frame timestamp: %" PRIu64 "\n", (guint64) meta.timestamp_ns); guint64 delta_frame_time_ns = (guint64) meta.timestamp_ns - ctx->last_timestamp; M_DEBUG("Calculated frame delta in ns: %" PRIu64 "\n", delta_frame_time_ns); uint32_t delta_frame_time_100us = delta_frame_time_ns / 100000; M_DEBUG("Calculated frame delta in 100us: %u\n", delta_frame_time_100us); ctx->input_frame_rate = (uint32_t) ((10000.0 / (double) delta_frame_time_100us) + 0.5); M_DEBUG("Calculated input frame rate is: %u\n", ctx->input_frame_rate); ctx->input_frame_rate /= ctx->output_frame_decimator; ctx->output_frame_rate = ctx->input_frame_rate; M_DEBUG("Output frame rate will be: %u\n", ctx->input_frame_rate); } ctx->last_timestamp = (guint64) meta.timestamp_ns; if(ctx->output_stream_rotation == 90 || ctx->output_stream_rotation == 270){ ctx->output_stream_height = meta.width; ctx->output_stream_width = meta.height; } else { ctx->output_stream_height = meta.height; ctx->output_stream_width = meta.width; } switch (meta.format) { case IMAGE_FORMAT_RAW8: case IMAGE_FORMAT_STEREO_RAW8: strncpy(ctx->input_frame_format, "gray8", MAX_IMAGE_FORMAT_STRING_LENGTH); break; //For stereo color images we only send the first one, treat it as if only one came through case IMAGE_FORMAT_STEREO_NV12: meta.size_bytes /= 2; case IMAGE_FORMAT_NV12: strncpy(ctx->input_frame_format, "nv12", MAX_IMAGE_FORMAT_STRING_LENGTH); break; //For stereo color images we only send the first one, treat it as if only one came through case IMAGE_FORMAT_STEREO_NV21: meta.size_bytes /= 2; case IMAGE_FORMAT_NV21: strncpy(ctx->input_frame_format, "nv21", MAX_IMAGE_FORMAT_STRING_LENGTH); break; case IMAGE_FORMAT_YUV422: strncpy(ctx->input_frame_format, "yuyv", MAX_IMAGE_FORMAT_STRING_LENGTH); break; case IMAGE_FORMAT_YUV422_UYVY: strncpy(ctx->input_frame_format, "uyvy", MAX_IMAGE_FORMAT_STRING_LENGTH); break; case IMAGE_FORMAT_YUV420: strncpy(ctx->input_frame_format, "yuv420", MAX_IMAGE_FORMAT_STRING_LENGTH); break; case IMAGE_FORMAT_RGB: strncpy(ctx->input_frame_format, "rgb", MAX_IMAGE_FORMAT_STRING_LENGTH); break; case IMAGE_FORMAT_RAW16: strncpy(ctx->input_frame_format, "gray16", MAX_IMAGE_FORMAT_STRING_LENGTH); break; default: M_ERROR("Unsupported frame format %d\n", meta.format); main_running = 0; break; } if ( ! main_running) return; ctx->input_frame_width = meta.width; if (meta.format == IMAGE_FORMAT_STEREO_RAW8) { // The 2 stereo images are stacked on top of each other // so have to double the height to account for both of them M_DEBUG("Got stereo raw8 as input format\n"); ctx->input_frame_height = meta.height * 2; } else { ctx->input_frame_height = meta.height; } configure_frame_format(ctx->input_frame_format, ctx); if (ctx->input_frame_size != (uint32_t) meta.size_bytes) { M_ERROR("Frame size mismatch %d %d\n", meta.size_bytes, ctx->input_frame_size); main_running = 0; return; } ctx->input_parameters_initialized = 1; } // Allocate a gstreamer buffer to hold the frame data GstBuffer *gst_buffer = gst_buffer_new_and_alloc(ctx->input_frame_size); gst_buffer_map(gst_buffer, &info, GST_MAP_WRITE); if (info.size < ctx->input_frame_size) { M_ERROR("Not enough memory for the frame buffer\n"); main_running = 0; return; } memcpy(info.data, frame, ctx->input_frame_size); // The need_data flag is set by the pipeline callback asking for // more data. if (ctx->need_data) { // If the input frame rate is higher than the output frame rate then // we ignore some of the frames. if ( ! (ctx->input_frame_number % ctx->output_frame_decimator)) { pthread_mutex_lock(&ctx->lock); // To get minimal latency make sure to set this to the timestamp of // the very first frame that we will be sending through the pipeline. if (ctx->initial_timestamp == 0) ctx->initial_timestamp = (guint64) meta.timestamp_ns; // Do the timestamp calculations. // It is very important to set these up accurately. // Otherwise, the stream can look bad or just not work at all. // TODO: Experiment with taking some time off of pts??? GST_BUFFER_TIMESTAMP(gst_buffer) = ((guint64) meta.timestamp_ns - ctx->initial_timestamp); GST_BUFFER_DURATION(gst_buffer) = ((guint64) meta.timestamp_ns) - ctx->last_timestamp; ctx->last_timestamp = (guint64) meta.timestamp_ns; pthread_mutex_unlock(&ctx->lock); M_VERBOSE("Output frame %d %" PRIu64 " %" PRIu64 "\n", ctx->output_frame_number, GST_BUFFER_TIMESTAMP(gst_buffer), GST_BUFFER_DURATION(gst_buffer)); ctx->output_frame_number++; // Signal that the frame is ready for use g_signal_emit_by_name(ctx->app_source, "push-buffer", gst_buffer, &status); if (status == GST_FLOW_OK) { M_VERBOSE("Frame %d accepted\n", ctx->output_frame_number); } else { M_ERROR("New frame rejected\n"); } } ctx->input_frame_number++; } else { M_VERBOSE("*** Skipping buffer ***\n"); } // Release the buffer so that we don't have a memory leak gst_buffer_unmap(gst_buffer, &info); gst_buffer_unref(gst_buffer); } // This callback lets us know when an RTSP client has disconnected so that // we can stop trying to feed video frames to the pipeline and reset everything // for the next connection. static void rtsp_client_disconnected(GstRTSPClient* self, context_data *data) { M_PRINT("An existing client has disconnected from the RTSP server\n"); pthread_mutex_lock(&data->lock); data->num_rtsp_clients--; if ( ! data->num_rtsp_clients) { data->input_frame_number = 0; data->output_frame_number = 0; data->need_data = 0; data->initial_timestamp = 0; data->last_timestamp = 0; } pthread_mutex_unlock(&data->lock); } // Gts client information static void print_client_info(GstRTSPClient* object) { GstRTSPConnection *connection = gst_rtsp_client_get_connection(object); if (connection == NULL) { M_PRINT("Could not get RTSP connection\n"); // DEBUG_EXIT; return; } GstRTSPUrl *url = gst_rtsp_connection_get_url(connection); if (url == NULL) { M_PRINT("Could not get RTSP connection URL\n"); // DEBUG_EXIT; return; } gchar* uri = gst_rtsp_url_get_request_uri(url); M_PRINT("A new client %s has connected\n",uri); g_free(uri); } // This is called by the RTSP server when a client has connected. static void rtsp_client_connected(GstRTSPServer* self, GstRTSPClient* object, context_data *data) { M_PRINT("A new client has connected to the RTSP server\n"); pthread_mutex_lock(&data->lock); print_client_info(object); data->num_rtsp_clients++; // Install the disconnect callback with the client. g_signal_connect(object, "closed", G_CALLBACK(rtsp_client_disconnected), data); pthread_mutex_unlock(&data->lock); } /* this timeout is periodically run to clean up the expired sessions from the * pool. This needs to be run explicitly currently but might be done * automatically as part of the mainloop. */ static gboolean timeout (gpointer data) { context_data *context_data_local = (context_data *)data; GstRTSPSessionPool *pool; pool = gst_rtsp_server_get_session_pool (context_data_local->rtsp_server); guint removed = gst_rtsp_session_pool_cleanup (pool); g_object_unref (pool); // M_PRINT("Removed sessions: %d\n", removed); if (removed > 0) { M_PRINT("Removed %d sessions\n", removed); pthread_mutex_lock(&context_data_local->lock); context_data_local->num_rtsp_clients--; if ( ! context_data_local->num_rtsp_clients) { context_data_local->input_frame_number = 0; context_data_local->output_frame_number = 0; context_data_local->need_data = 0; context_data_local->initial_timestamp = 0; context_data_local->last_timestamp = 0; } pthread_mutex_unlock(&context_data_local->lock); } return TRUE; } // This callback is setup to happen at 1 second intervals so that we can // monitor when the program is ending and exit the main loop. gboolean loop_callback(gpointer data) { if ( ! main_running) { M_PRINT("Trying to stop loop\n"); g_main_loop_quit((GMainLoop*) data); } return TRUE; } // This will cause all remaining RTSP clients to be removed GstRTSPFilterResult stop_rtsp_clients(GstRTSPServer* server, GstRTSPClient* client, gpointer user_data) { return GST_RTSP_FILTER_REMOVE; } static void PrintHelpMessage() { M_PRINT("\nCommand line arguments are as follows:\n\n"); M_PRINT("-b --bitrate <#> | Use specified bitrate instead of one in config file\n"); M_PRINT("-d --decimator <#> | Use specified decimator instead of one in config file\n"); M_PRINT("-h --help | Print this help message\n"); M_PRINT("-i --input-pipe <name> | Use specified input pipe instead of one in config file\n"); M_PRINT("-p --port <#> | Use specified port number for the RTSP URI instead of config file\n"); M_PRINT("-r --rotation <#> | Use specified rotation instead of one in config file\n"); M_PRINT("-s --software-encode | Use software h264 encoder instead of OMX hardware encoder.\n"); M_PRINT("-v --verbosity <#> | Log verbosity level (Default 2)\n"); M_PRINT(" 0 | Print verbose logs\n"); M_PRINT(" 1 | Print >= info logs\n"); M_PRINT(" 2 | Print >= warning logs\n"); M_PRINT(" 3 | Print only fatal logs\n"); } static int ParseArgs(int argc, ///< Number of arguments char* const argv[]) ///< Argument list { static struct option LongOptions[] = { {"bitrate", required_argument, 0, 'b'}, {"decimator", required_argument, 0, 'd'}, {"help", no_argument, 0, 'h'}, {"input-pipe", required_argument, 0, 'i'}, {"port", required_argument, 0, 'p'}, {"rotation", required_argument, 0, 'r'}, {"software-encode", no_argument, 0, 's'}, {"verbosity", required_argument, 0, 'v'}, }; int optionIndex = 0; int option; while ((option = getopt_long (argc, argv, ":b:d:hi:p:r:sv:", &LongOptions[0], &optionIndex)) != -1) { switch (option) { case 'v':{ int verbosity; if (sscanf(optarg, "%d", &verbosity) != 1) { M_ERROR("Failed to parse debug level specified after -d flag\n"); return -1; } if (verbosity >= PRINT || verbosity < VERBOSE) { M_ERROR("Invalid debug level specified: %d\n", verbosity); return -1; } M_JournalSetLevel((M_JournalLevel) verbosity); break; } case 'b': if(sscanf(optarg, "%u", &context.output_stream_bitrate) != 1){ M_ERROR("Failed to get valid integer for bitrate from: %s\n", optarg); return -1; } break; case 'd': if(sscanf(optarg, "%u", &context.output_frame_decimator) != 1){ M_ERROR("Failed to get valid integer for decimator from: %s\n", optarg); return -1; } break; case 'r': if(sscanf(optarg, "%u", &context.output_stream_rotation) != 1){ M_ERROR("Failed to get valid integer for rotation from: %s\n", optarg); return -1; } if( context.output_stream_rotation != 0 && context.output_stream_rotation != 90 && context.output_stream_rotation != 180 && context.output_stream_rotation != 270 ) { M_ERROR("Invalid rotation: %u, must be 0, 90, 180, or 270\n", context.output_stream_rotation); return -1; } break; case 'i': strncpy(context.input_pipe_name, optarg, MODAL_PIPE_MAX_PATH_LEN); break; case 'p': strncpy(context.rtsp_server_port, optarg, MAX_RTSP_PORT_SIZE); break; case 's': M_PRINT("Enabling software h264 encoder instead of OMX\n"); context.use_sw_h264 = 1; break; case 'h': PrintHelpMessage(); return -1; case ':': M_ERROR("Option %c needs a value\n\n", optopt); PrintHelpMessage(); return -1; case '?': M_ERROR("Unknown option: %c\n\n", optopt); PrintHelpMessage(); return -1; } } // optind is for the extra arguments which are not parsed if(optind < argc){ M_WARN("extra arguments:\n"); for (; optind < argc; optind++) { M_WARN("\t%s\n", argv[optind]); } } return 0; } //-------- // Main //-------- int main(int argc, char *argv[]) { GstRTSPMountPoints *mounts; GstRTSPMediaFactory *factory; GMainContext* loop_context; GMainLoop *loop; GSource *loop_source; GSource *time_loop_source; strncpy(context.rtsp_server_port, DEFAULT_RTSP_PORT, MAX_RTSP_PORT_SIZE); // Have the configuration module fill in the context data structure // with all of the required parameters to support the given configuration // in the given configuration file. if (prepare_configuration(&context)) { M_ERROR("Could not parse the configuration data\n"); return -1; } if(ParseArgs(argc, argv)){ M_ERROR("Failed to parse args\n"); return -1; } // start signal handler so we can exit cleanly main_running = 1; if(enable_signal_handler()==-1){ M_ERROR("Failed to start signal handler\n"); return -1; } M_DEBUG("Using input: %s\n", context.input_pipe_name); M_DEBUG("Using RTSP port: %s\n", context.rtsp_server_port); M_DEBUG("Using bitrate: %d\n", context.output_stream_bitrate); M_DEBUG("Using roatation: %d\n", context.output_stream_rotation); M_DEBUG("Using decimator: %d\n", context.output_frame_decimator); // Pass a pointer to the context to the pipeline module pipeline_init(&context); // Initialize Gstreamer gst_init(NULL, NULL); pipe_client_set_connect_cb(0, _cam_connect_cb, NULL); pipe_client_set_disconnect_cb(0, _cam_disconnect_cb, NULL); pipe_client_set_camera_helper_cb(0, _cam_helper_cb, &context); pipe_client_open(0, context.input_pipe_name, "voxl-streamer", EN_PIPE_CLIENT_CAMERA_HELPER, 0); while (main_running) { if (context.input_parameters_initialized) break; usleep(500000); } if (context.input_parameters_initialized) { M_DEBUG("Input parameters initialized\n"); } else { M_ERROR("Timeout on input parameter initialization\n"); return -1; } // Create the RTSP server context.rtsp_server = gst_rtsp_server_new(); if (context.rtsp_server) { M_DEBUG("Made rtsp_server\n"); } else { M_ERROR("couldn't make rtsp_server\n"); return -1; } // Configure the RTSP server port g_object_set(context.rtsp_server, "service", context.rtsp_server_port, NULL); // Setup the callback to alert when a new connection has come in g_signal_connect(context.rtsp_server, "client-connected", G_CALLBACK(rtsp_client_connected), &context); // Setup the glib loop for the RTSP server to use loop_context = g_main_context_new(); if (loop_context) { M_DEBUG("Created RTSP server main loop context\n"); } else { M_ERROR("Couldn't create loop context\n"); return -1; } loop = g_main_loop_new(loop_context, FALSE); if (loop) { M_DEBUG("Created RTSP server main loop\n"); } else { M_ERROR("Couldn't create loop\n"); return -1; } // Add an event source to the loop context. This will generate a callback // at 1 second intervals. We can use this to detect when the program is // ending and exit the loop loop_source = g_timeout_source_new_seconds(1); if (loop_source) { M_DEBUG("Created RTSP server main loop source for callback\n"); } else { M_ERROR("Couldn't create loop source for callback\n"); return -1; } g_source_set_callback(loop_source, loop_callback, loop, NULL); g_source_attach(loop_source, loop_context); // attach timeout to out loop time_loop_source = g_timeout_source_new_seconds(2); if (time_loop_source) { M_DEBUG("Created timeout loop source for callback\n"); } else { M_ERROR("Couldn't create timeout loop source for callback\n"); return -1; } g_source_set_callback(time_loop_source, timeout, &context, NULL); g_source_attach(time_loop_source, loop_context); g_main_context_unref(loop_context); g_source_unref(loop_source); g_source_unref(time_loop_source); // get the mount points for the RTSP server, every server has a default object // that will be used to map uri mount points to media factories mounts = gst_rtsp_server_get_mount_points(context.rtsp_server); if ( ! mounts) { M_ERROR("Couldn't get mount points\n"); return -1; } // We override the create element function with our own so that we can use // our custom pipeline instead of a launch line. factory = gst_rtsp_media_factory_new(); if ( ! factory) { M_ERROR("Couldn't create new media factory\n"); return -1; } // Set as shared to consider video disconnects gst_rtsp_media_factory_set_shared (factory, TRUE); GstRTSPMediaFactoryClass *memberFunctions = GST_RTSP_MEDIA_FACTORY_GET_CLASS(factory); if ( ! memberFunctions) { M_ERROR("Couldn't get media factory class pointer\n"); return -1; } memberFunctions->create_element = create_custom_element; char *link_name = "/live"; gst_rtsp_mount_points_add_factory(mounts, link_name, factory); g_object_unref(mounts); // Attach the RTSP server to our loop int source_id = gst_rtsp_server_attach(context.rtsp_server, loop_context); M_DEBUG("Got %d from gst_rtsp_server_attach\n", source_id); if ( ! source_id) { M_ERROR("gst_rtsp_server_attach failed\n"); return -1; } // Indicate how to connect to the stream M_PRINT("Stream available at rtsp://127.0.0.1:%s%s\n", context.rtsp_server_port, link_name); // Start the main loop that the RTSP Server is attached to. This will not // exit until it is stopped. g_main_loop_run(loop); g_main_loop_unref(loop); // Main loop has exited, time to clean up and exit the program M_DEBUG("g_main_loop exited\n"); // Wait for the buffer processing thread to exit pipe_client_close_all(); // Stop any remaining RTSP clients (void) gst_rtsp_server_client_filter(context.rtsp_server, stop_rtsp_clients, NULL); // Clean up gstreamer gst_deinit(); M_PRINT("Exited Cleanly\n"); return 0; }
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this is very helpful, thank you. We will try to incorporate. We do accept pull requests on our gitlab for future reference
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@Chad-Sweet @RichieRich Just created a pull request, I though I couldn't. It is here https://gitlab.com/voxl-public/voxl-sdk/services/voxl-streamer/-/merge_requests/24. Would love to follow up on this, in case any changes are made from your side to incorporate in on our side.
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Everything looks good, gave it a glance and a quick test, it's been merged to our dev and staging branches in v0.4.2 and will be in the next release, thanks!
