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Add ST2110-30 sender to the gstreamer and demo
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136 changes: 136 additions & 0 deletions
136
cpp/libs/ossrf_gstreamer_api/lib/src/sender/st2110_30_sender_plugin.cpp
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| // 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. | ||
|
|
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| #include "st2110_30_sender_plugin.h" | ||
| #include "bisect/expected/macros.h" | ||
| #include "bisect/pipeline.h" | ||
| #include <gst/gst.h> | ||
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| using namespace bisect; | ||
| using namespace ossrf::gst::sender; | ||
| using namespace ossrf::gst::plugins; | ||
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| namespace | ||
| { | ||
| constexpr auto queue_max_size_time = 200000; | ||
| constexpr auto queue_max_size_buffers = 0; | ||
| constexpr auto queue_max_size_bytes = 0; | ||
| }; // namespace | ||
|
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| struct gst_st2110_30_sender_impl : gst_sender_plugin_t | ||
| { | ||
| sender_settings s_; | ||
| audio_info_t f_; | ||
| gst::pipeline pipeline_; | ||
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| gst_st2110_30_sender_impl(sender_settings settings, audio_info_t format) : s_(settings), f_(format) {} | ||
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| ~gst_st2110_30_sender_impl() { stop(); } | ||
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| 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(); | ||
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| // 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"); | ||
|
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| // 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"); | ||
|
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| // 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"); | ||
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| // 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"); | ||
|
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| // 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"); | ||
|
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| // Add pipeline capsfilter | ||
| auto* capsfilter = gst_element_factory_make("capsfilter", NULL); | ||
| BST_ENFORCE(capsfilter != nullptr, "Failed creating capsfilter"); | ||
|
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| // 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); | ||
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| // 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"); | ||
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| // 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"); | ||
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| // 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"); | ||
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| // Link elements | ||
| BST_ENFORCE(gst_element_link_many(source, queue1, audioconvert, queue2, audioresample, capsfilter, queue3, | ||
| rtpL24pay, udpsink, NULL), | ||
| "Failed linking GStreamer audio pipeline"); | ||
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| // Setup runner | ||
| pipeline_.run_loop(); | ||
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| return {}; | ||
| } | ||
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| void stop() noexcept override | ||
| { | ||
| pipeline_.stop(); | ||
| pipeline_ = {}; | ||
| } | ||
| }; | ||
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| 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); | ||
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| BST_CHECK(i->create_gstreamer_pipeline()); | ||
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| return i; | ||
| } |
24 changes: 24 additions & 0 deletions
24
cpp/libs/ossrf_gstreamer_api/lib/src/sender/st2110_30_sender_plugin.h
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| Original file line number | Diff line number | Diff line change |
|---|---|---|
| @@ -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. | ||
|
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| #pragma once | ||
| #include "ossrf/gstreamer/api/sender/sender_plugin.h" | ||
| #include "ossrf/gstreamer/api/sender/sender_configuration.h" | ||
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| 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; | ||
| } |