Add ST2110-30 sender to the gstreamer and demo

cpp/demos/ossrf-nmos-api/config/nmos_config.json

@@ -5,7 +5,7 @@
             "label": "BISECT OSSRF Node",
             "description": "BISECT OSSRF node",
             "host_addresses": [
-                "192.168.1.79"
+                "192.168.1.15"
             ],
             "interfaces": [
                 {
@@ -24,10 +24,10 @@
                     "locked": true
                 }
             ],
-            "registry_address": "192.168.1.79",
+            "registry_address": "192.168.1.15",
             "registry_version": "v1.3",
             "registration_port": 8010,
-            "system_address": "192.168.1.79",
+            "system_address": "192.168.1.15",
             "system_version": "v1.0",
             "system_port": 8010
         }
@@ -44,7 +44,7 @@
             "description": "BISECT OSSRF receiver video",
             "network": {
                 "primary": {
-                    "interface_address": "192.168.1.79",
+                    "interface_address": "192.168.1.15",
                     "interface_name": "wlp1s0"
                 }
             },
@@ -60,7 +60,7 @@
             "description": "BISECT OSSRF sender video 1",
             "network": {
                 "primary": {
-                    "source_address": "192.168.1.79",
+                    "source_address": "192.168.1.15",
                     "interface_name": "wlp1s0",
                     "destination_address": "239.10.10.10",
                     "destination_port": 5004
@@ -81,27 +81,22 @@
         },
         {
             "id": "f2aa5651-c673-448c-bd02-5e8475898c7f",
-            "label": "BISECT OSSRF sender video 2",
-            "description": "BISECT OSSRF sender video 2",
+            "label": "BISECT OSSRF sender audio",
+            "description": "BISECT OSSRF sender audio",
             "network": {
                 "primary": {
-                    "source_address": "192.168.1.79",
+                    "source_address": "192.168.1.15",
                     "interface_name": "wlp1s0",
                     "destination_address": "239.10.10.11",
                     "destination_port": 5005
                 }
             },
             "payload_type": 97,
-            "media_type": "video/raw",
+            "media_type": "audio/L24",
             "media": {
-                "width": 640,
-                "height": 480,
-                "frame_rate": {
-                    "num": 50,
-                    "den": 1
-                },
-                "sampling": "RGB_444",
-                "structure": "progressive"
+                "number_of_channels": 2,
+                "sampling_rate": 48000,
+                "packet_time": 1.000
             }
         }
     ]

cpp/libs/ossrf_gstreamer_api/lib/src/sender/sender_plugin.cpp

@@ -18,6 +18,7 @@
 #include "bisect/expected/match.h"
 #include "bisect/json.h"
 #include "st2110_20_sender_plugin.h"
+#include "st2110_30_sender_plugin.h"
 #include <nlohmann/json.hpp>
 
 using namespace bisect;
@@ -95,7 +96,7 @@ namespace
         {
             BST_CHECK_ASSIGN(s.format, video_sender_info_from_json(media));
         }
-        else if(media_type == "audio/raw")
+        else if(media_type == "audio/L24")
         {
             BST_CHECK_ASSIGN(s.format, audio_l24_sender_info_from_json(media));
         }
@@ -118,6 +119,12 @@ namespace
     {
         return create_gst_st2110_20_plugin(settings, format, pattern);
     }
+
+    expected<gst_sender_plugin_uptr> do_create_plugin(const audio_info_t& format, const sender_settings& settings,
+                                                      int pattern)
+    {
+        return create_gst_st2110_30_plugin(settings, format);
+    }
 } // namespace
 
 expected<gst_sender_plugin_uptr> plugins::create_gst_sender_plugin(const std::string& config, int pattern) noexcept

cpp/libs/ossrf_gstreamer_api/lib/src/sender/st2110_30_sender_plugin.cpp

@@ -0,0 +1,136 @@
+// Copyright (C) 2024 Advanced Media Workflow Association
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+#include "st2110_30_sender_plugin.h"
+#include "bisect/expected/macros.h"
+#include "bisect/pipeline.h"
+#include <gst/gst.h>
+
+using namespace bisect;
+using namespace ossrf::gst::sender;
+using namespace ossrf::gst::plugins;
+
+namespace
+{
+    constexpr auto queue_max_size_time    = 200000;
+    constexpr auto queue_max_size_buffers = 0;
+    constexpr auto queue_max_size_bytes   = 0;
+}; // namespace
+
+struct gst_st2110_30_sender_impl : gst_sender_plugin_t
+{
+    sender_settings s_;
+    audio_info_t f_;
+    gst::pipeline pipeline_;
+
+    gst_st2110_30_sender_impl(sender_settings settings, audio_info_t format) : s_(settings), f_(format) {}
+
+    ~gst_st2110_30_sender_impl() { stop(); }
+
+    maybe_ok create_gstreamer_pipeline()
+    {
+        // Create pipeline and check if all elements are created successfully
+        BST_CHECK_ASSIGN(pipeline_, bisect::gst::pipeline::create(NULL));
+        auto* pipeline = pipeline_.get();
+
+        // Add pipeline pulsesrc (audio source)
+        auto* source = gst_element_factory_make("pulsesrc", NULL);
+        BST_ENFORCE(source != nullptr, "Failed creating GStreamer element pulsesrc");
+        BST_ENFORCE(gst_bin_add(GST_BIN(pipeline), source), "Failed adding pulsesrc to the pipeline");
+
+        // Add pipeline queue1
+        auto* queue1 = gst_element_factory_make("queue", NULL);
+        BST_ENFORCE(queue1 != nullptr, "Failed creating GStreamer element queue");
+        g_object_set(G_OBJECT(queue1), "max-size-time", queue_max_size_time, "max-size-buffers", queue_max_size_buffers,
+                     "max-size-bytes", queue_max_size_bytes, NULL);
+        BST_ENFORCE(gst_bin_add(GST_BIN(pipeline), queue1), "Failed adding queue to the pipeline");
+
+        // Add pipeline audioconvert
+        auto* audioconvert = gst_element_factory_make("audioconvert", NULL);
+        BST_ENFORCE(audioconvert != nullptr, "Failed creating GStreamer element audioconvert");
+        BST_ENFORCE(gst_bin_add(GST_BIN(pipeline), audioconvert), "Failed adding audioconvert to the pipeline");
+
+        // Add pipeline queue2
+        auto* queue2 = gst_element_factory_make("queue", NULL);
+        BST_ENFORCE(queue2 != nullptr, "Failed creating GStreamer element queue");
+        g_object_set(G_OBJECT(queue1), "max-size-time", queue_max_size_time, "max-size-buffers", queue_max_size_buffers,
+                     "max-size-bytes", queue_max_size_bytes, NULL);
+        BST_ENFORCE(gst_bin_add(GST_BIN(pipeline), queue2), "Failed adding queue to the pipeline");
+
+        // Add pipeline audioresample
+        auto* audioresample = gst_element_factory_make("audioresample", NULL);
+        BST_ENFORCE(audioresample != nullptr, "Failed creating GStreamer element audioresample");
+        BST_ENFORCE(gst_bin_add(GST_BIN(pipeline), audioresample), "Failed adding audioresample to the pipeline");
+
+        // Add pipeline capsfilter
+        auto* capsfilter = gst_element_factory_make("capsfilter", NULL);
+        BST_ENFORCE(capsfilter != nullptr, "Failed creating capsfilter");
+
+        // Create caps for capsfilter
+        auto* caps = gst_caps_new_simple("audio/x-raw", "channels", G_TYPE_INT, f_.number_of_channels, "rate",
+                                         G_TYPE_INT, f_.sampling_rate, NULL);
+        BST_ENFORCE(caps != nullptr, "Failed creating GStreamer audio caps");
+        g_object_set(G_OBJECT(capsfilter), "caps", caps, NULL);
+        BST_ENFORCE(gst_bin_add(GST_BIN(pipeline), capsfilter), "Failed adding capsfilter to the pipeline");
+        gst_caps_unref(caps);
+
+        // Add pipeline queue1
+        auto* queue3 = gst_element_factory_make("queue", NULL);
+        BST_ENFORCE(queue3 != nullptr, "Failed creating GStreamer element queue");
+        g_object_set(G_OBJECT(queue1), "max-size-time", queue_max_size_time, "max-size-buffers", queue_max_size_buffers,
+                     "max-size-bytes", queue_max_size_bytes, NULL);
+        BST_ENFORCE(gst_bin_add(GST_BIN(pipeline), queue3), "Failed adding queue to the pipeline");
+
+        // Add pipeline rtpL24pay (audio payload)
+        auto* rtpL24pay = gst_element_factory_make("rtpL24pay", NULL);
+        BST_ENFORCE(rtpL24pay != nullptr, "Failed creating GStreamer element rtpL24pay");
+        BST_ENFORCE(gst_bin_add(GST_BIN(pipeline), rtpL24pay), "Failed adding rtpL24pay to the pipeline");
+
+        // Add pipeline udpsink
+        auto* udpsink = gst_element_factory_make("udpsink", NULL);
+        BST_ENFORCE(udpsink != nullptr, "Failed creating GStreamer element udpsink");
+        // Set properties
+        g_object_set(G_OBJECT(udpsink), "host", s_.primary.destination_ip_address.c_str(), NULL);
+        g_object_set(G_OBJECT(udpsink), "port", s_.primary.destination_port, NULL);
+        g_object_set(G_OBJECT(udpsink), "auto-multicast", TRUE, NULL);
+        g_object_set(G_OBJECT(udpsink), "multicast-iface", s_.primary.interface_name.c_str(), NULL);
+        BST_ENFORCE(gst_bin_add(GST_BIN(pipeline), udpsink), "Failed adding udpsink to the pipeline");
+
+        // Link elements
+        BST_ENFORCE(gst_element_link_many(source, queue1, audioconvert, queue2, audioresample, capsfilter, queue3,
+                                          rtpL24pay, udpsink, NULL),
+                    "Failed linking GStreamer audio pipeline");
+
+        // Setup runner
+        pipeline_.run_loop();
+
+        return {};
+    }
+
+    void stop() noexcept override
+    {
+        pipeline_.stop();
+        pipeline_ = {};
+    }
+};
+
+expected<gst_sender_plugin_uptr> ossrf::gst::plugins::create_gst_st2110_30_plugin(sender_settings settings,
+                                                                                  audio_info_t format) noexcept
+{
+    auto i = std::make_unique<gst_st2110_30_sender_impl>(settings, format);
+
+    BST_CHECK(i->create_gstreamer_pipeline());
+
+    return i;
+}

cpp/libs/ossrf_gstreamer_api/lib/src/sender/st2110_30_sender_plugin.h

@@ -0,0 +1,24 @@
+// Copyright (C) 2024 Advanced Media Workflow Association
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+#pragma once
+#include "ossrf/gstreamer/api/sender/sender_plugin.h"
+#include "ossrf/gstreamer/api/sender/sender_configuration.h"
+
+namespace ossrf::gst::plugins
+{
+    bisect::expected<gst_sender_plugin_uptr>
+    create_gst_st2110_30_plugin(ossrf::gst::sender::sender_settings settings,
+                                ossrf::gst::sender::audio_info_t audio_info_t) noexcept;
+}