enhancement #23: make mic a seperate pipeline since it is not sync with

AV

Author: FabienDanieau

Plugin/CMakeLists.txt

@@ -22,6 +22,8 @@ set(SOURCE_FILES
 	src/GstBasePipeline.cpp
 	src/GstAVPipeline.cpp
 	src/GstAVPipeline.h
+	src/GstMicPipeline.cpp
+	src/GstMicPipeline.h
 	src/GstDataPipeline.cpp
 	src/GstDataPipeline.h
 

Plugin/src/GstAVPipeline.cpp

@@ -354,157 +354,6 @@ GstElement* GstAVPipeline::add_webrtcsrc(GstElement* pipeline, const std::string
     return webrtcsrc;
 }
 
-GstElement* GstAVPipeline::add_wasapi2src(GstElement* pipeline)
-{
-    GstElement* wasapi2src = gst_element_factory_make("wasapi2src", nullptr);
-    if (!wasapi2src)
-    {
-        Debug::Log("Failed to create wasapi2src", Level::Error);
-        return nullptr;
-    }
-    g_object_set(wasapi2src, "low-latency", true, "provide-clock", false, nullptr);
-
-    gst_bin_add(GST_BIN(pipeline), wasapi2src);
-    return wasapi2src;
-}
-
-GstElement* GstAVPipeline::add_opusenc(GstElement* pipeline)
-{
-    GstElement* opusenc = gst_element_factory_make("opusenc", nullptr);
-    if (!opusenc)
-    {
-        Debug::Log("Failed to create opusenc", Level::Error);
-        return nullptr;
-    }
-
-    g_object_set(opusenc, "audio-type", "restricted-lowdelay", /* "frame-size", 10,*/ nullptr);
-
-    gst_bin_add(GST_BIN(pipeline), opusenc);
-    return opusenc;
-}
-
-GstElement* GstAVPipeline::add_audio_caps_capsfilter(GstElement* pipeline)
-{
-    GstElement* audio_caps_capsfilter = gst_element_factory_make("capsfilter", nullptr);
-    if (!audio_caps_capsfilter)
-    {
-        Debug::Log("Failed to create capsfilter", Level::Error);
-        return nullptr;
-    }
-
-    GstCaps* audio_caps = gst_caps_from_string("audio/x-opus");
-    gst_caps_set_simple(audio_caps, "channels", G_TYPE_INT, 1, "rate", G_TYPE_INT, 48000, nullptr);
-    g_object_set(audio_caps_capsfilter, "caps", audio_caps, nullptr);
-
-    gst_bin_add(GST_BIN(pipeline), audio_caps_capsfilter);
-    gst_caps_unref(audio_caps);
-    return audio_caps_capsfilter;
-}
-
-void GstAVPipeline::consumer_added_callback(GstElement* consumer_id, gchararray webrtcbin, GstElement* arg1, gpointer udata)
-{
-    Debug::Log("Consumer added");
-    GstIterator* sinks = gst_bin_iterate_sinks(GST_BIN(consumer_id));
-    gboolean done = FALSE;
-    while (!done)
-    {
-        GValue item = G_VALUE_INIT;
-        switch (gst_iterator_next(sinks, &item))
-        {
-            case GST_ITERATOR_OK:
-            {
-                GstElement* sink = GST_ELEMENT(g_value_get_object(&item));
-
-                // Log a message indicating that the processing deadline is being set for the sink
-                std::string name = GST_ELEMENT_NAME(sink);
-                Debug::Log("Setting processing deadline for " + name);
-
-                // Set the processing deadline for the sink
-                g_object_set(sink, "processing-deadline", 1000000, nullptr);
-
-                // Unref the sink to free its resources
-                g_object_unref(sink);
-
-                // Free the item value
-                g_value_unset(&item);
-
-                break;
-            }
-            case GST_ITERATOR_RESYNC:
-                // Resync the iterator
-                gst_iterator_resync(sinks);
-                break;
-            case GST_ITERATOR_ERROR:
-                // Handle the error
-                g_warning("Error iterating sinks");
-                done = TRUE;
-                break;
-            case GST_ITERATOR_DONE:
-                // We're done iterating
-                done = TRUE;
-                break;
-        }
-    }
-
-    // Free the iterator
-    gst_iterator_free(sinks);
-}
-
-GstElement* GstAVPipeline::add_webrtcsink(GstElement* pipeline, const std::string& uri)
-{
-    GstElement* webrtcsink = gst_element_factory_make("webrtcsink", nullptr);
-    if (!webrtcsink)
-    {
-        Debug::Log("Failed to create webrtcsink", Level::Error);
-        return nullptr;
-    }
-
-    GObject* signaller;
-    g_object_get(webrtcsink, "signaller", &signaller, nullptr);
-    if (signaller)
-    {
-        g_object_set(signaller, "uri", uri.c_str(), nullptr);
-        g_object_unref(signaller); // Unref signaller when done
-    }
-    else
-    {
-        Debug::Log("Failed to get signaller property from webrtcsink.", Level::Error);
-    }
-
-    GstStructure* meta_structure = gst_structure_new_empty("meta");
-    gst_structure_set(meta_structure, "name", G_TYPE_STRING, "UnityClient", nullptr);
-    GValue meta_value = G_VALUE_INIT;
-    g_value_init(&meta_value, GST_TYPE_STRUCTURE);
-    gst_value_set_structure(&meta_value, meta_structure);
-    g_object_set_property(G_OBJECT(webrtcsink), "meta", &meta_value);
-    gst_structure_free(meta_structure);
-    g_value_unset(&meta_value);
-
-    g_object_set(webrtcsink, "stun-server", nullptr, "do-restransmission", false, nullptr);
-
-    //g_signal_connect(webrtcsink, "consumer-added", G_CALLBACK(consumer_added_callback), nullptr);
-
-    gst_bin_add(GST_BIN(pipeline), webrtcsink);
-    return webrtcsink;
-}
-
-
-GstElement* GstAVPipeline::add_webrtcdsp(GstElement* pipeline)
-{
-    GstElement* webrtcdsp = gst_element_factory_make("webrtcdsp", nullptr);
-    if (!webrtcdsp)
-    {
-        Debug::Log("Failed to create webrtcdsp", Level::Error);
-        return nullptr;
-    }
-
-    // Echo cancel done on the robot side
-    g_object_set(webrtcdsp, "echo-cancel", false, nullptr);
-
-    gst_bin_add(GST_BIN(pipeline), webrtcdsp);
-    return webrtcdsp;
-}
-
 void GstAVPipeline::on_pad_added(GstElement* src, GstPad* new_pad, gpointer data)
 {
     GstAVPipeline* avpipeline = static_cast<GstAVPipeline*>(data);
@@ -671,19 +520,6 @@ void GstAVPipeline::CreatePipeline(const char* uri, const char* remote_peer_id)
     GstBasePipeline::CreatePipeline();
 
     GstElement* webrtcsrc = add_webrtcsrc(pipeline_, remote_peer_id, uri, this);
-    GstElement* wasapi2src = add_wasapi2src(pipeline_);
-    GstElement* webrtcdsp = add_webrtcdsp(pipeline_);
-    GstElement* audioconvert = add_audioconvert(pipeline_);
-    GstElement* queue = add_queue(pipeline_);
-    GstElement* opusenc = add_opusenc(pipeline_);
-    GstElement* audio_caps_capsfilter = add_audio_caps_capsfilter(pipeline_);
-    GstElement* webrtcsink = add_webrtcsink(pipeline_, uri);
-
-    if (!gst_element_link_many(wasapi2src, audioconvert, webrtcdsp, queue, opusenc, audio_caps_capsfilter, webrtcsink,
-                               nullptr))
-    {
-        Debug::Log("Audio sending elements could not be linked.", Level::Error);
-    }
 
     CreateBusThread();
 }

Plugin/src/GstAVPipeline.h

@@ -52,7 +52,6 @@ class GstAVPipeline : GstBasePipeline
 private:
     static void on_pad_added(GstElement* src, GstPad* new_pad, gpointer data);
     static void webrtcbin_ready(GstElement* self, gchararray peer_id, GstElement* webrtcbin, gpointer udata);
-    static void consumer_added_callback(GstElement* consumer_id, gchararray webrtcbin, GstElement* arg1, gpointer udata);
 
     static GstFlowReturn GstAVPipeline::on_new_sample(GstAppSink* appsink, gpointer user_data);
 
@@ -71,9 +70,4 @@ class GstAVPipeline : GstBasePipeline
     static GstElement* add_wasapi2sink(GstElement* pipeline);
     static GstElement* add_webrtcsrc(GstElement* pipeline, const std::string& remote_peer_id, const std::string& uri,
                                      GstAVPipeline* self);
-    static GstElement* add_wasapi2src(GstElement* pipeline);
-    static GstElement* add_opusenc(GstElement* pipeline);
-    static GstElement* add_audio_caps_capsfilter(GstElement* pipeline);
-    static GstElement* add_webrtcsink(GstElement* pipeline, const std::string& uri);
-    static GstElement* add_webrtcdsp(GstElement* pipeline);
 };