messenger-gtk

commit d8b35ac19fe6bfff949a915fd0603a5dd1a38dfd
parent d5d9c3378e1cd461249db071624869fc34cf9144
Author: Jacki <jacki@thejackimonster.de>
Date:   Thu, 20 Jun 2024 21:41:32 +0200

Get rtp streaming from audio input working

Signed-off-by: Jacki <jacki@thejackimonster.de>

Diffstat:
M
src/application.c
 | 
2
+-
M
src/discourse.c
 | 
74
++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++

2 files changed, 75 insertions(+), 1 deletion(-)
diff --git a/src/application.c b/src/application.c
@@ -26,8 +26,8 @@
 #include "request.h"
 #include "resources.h"
 
-#include <gnunet/gnunet_chat_lib.h>
 #include <gnunet/gnunet_common.h>
+#include <gnunet/gnunet_chat_lib.h>
 #include <gstreamer-1.0/gst/gst.h>
 #include <gtk-3.0/gtk/gtk.h>
 #include <libhandy-1/handy.h>
diff --git a/src/discourse.c b/src/discourse.c
@@ -26,6 +26,7 @@
 
 #include <gnunet/gnunet_common.h>
 #include <gnunet/gnunet_chat_lib.h>
+#include <gstreamer-1.0/gst/gst.h>
 
 static void
 _setup_gst_pipelines_of_subscription(MESSENGER_DiscourseSubscriptionInfo *info)
@@ -47,6 +48,55 @@ _setup_gst_pipelines_of_subscription(MESSENGER_DiscourseSubscriptionInfo *info)
   // https://gstreamer.freedesktop.org/documentation/audiomixer/audiomixer.html?gi-language=c
 }
 
+static GstFlowReturn
+_new_audio_sample(GstElement *sink, gpointer data)
+{
+  g_assert((sink) && (data));
+
+  MESSENGER_DiscourseInfo *info = (MESSENGER_DiscourseInfo*) data;
+
+  GstSample *sample;
+  g_signal_emit_by_name(sink, "pull-sample", &sample);
+
+  if (!sample)
+    return GST_FLOW_ERROR;
+
+  GstBuffer *buffer = gst_sample_get_buffer(sample);
+
+  if (!buffer)
+    goto cleanup_sample;
+
+  gsize size = gst_buffer_get_size(buffer);
+  if (!size)
+    goto cleanup_sample;
+
+  GstMapInfo mapping;
+  if (gst_buffer_map(buffer, &mapping, GST_MAP_READ))
+  {
+    GNUNET_CHAT_discourse_write(info->discourse, (const char*) mapping.data, mapping.size);
+    gst_buffer_unmap(buffer, &mapping);
+  }
+
+cleanup_sample:
+  gst_sample_unref(sample);
+  return GST_FLOW_OK;
+}
+
+/* This function is called when an error message is posted on the bus */
+static void
+error_cb(GstBus *bus, GstMessage *msg, gpointer data)
+{
+  GError *err;
+  gchar *debug_info;
+
+  /* Print error details on the screen */
+  gst_message_parse_error (msg, &err, &debug_info);
+  g_printerr ("Error received from element %s: %s\n", GST_OBJECT_NAME (msg->src), err->message);
+  g_printerr ("Debugging information: %s\n", debug_info ? debug_info : "none");
+  g_clear_error (&err);
+  g_free (debug_info);
+}
+
 static void
 _setup_gst_pipelines(MESSENGER_DiscourseInfo *info)
 {
@@ -65,6 +115,30 @@ _setup_gst_pipelines(MESSENGER_DiscourseInfo *info)
     GST_BIN(info->record_pipeline), "sink"
   );
 
+  {
+    GstBus *bus = gst_element_get_bus(info->record_pipeline);
+    gst_bus_add_signal_watch(bus);
+    g_signal_connect(G_OBJECT(bus), "message::error", (GCallback)error_cb, info);
+    gst_object_unref(bus);
+
+    GstCaps *caps = gst_caps_new_simple (
+      "application/x-rtp",
+      "media", G_TYPE_STRING, "audio",
+      "payload", G_TYPE_INT, 96,
+      "clock-rate", G_TYPE_INT, 44100,
+      "encoding-name", G_TYPE_STRING, "L16",
+      "channels", G_TYPE_INT, 1,
+      NULL
+    );
+
+    g_object_set(info->record_sink, "emit-signals", TRUE, "caps", caps, NULL);
+    g_signal_connect(info->record_sink, "new-sample", G_CALLBACK(_new_audio_sample), info);
+    
+    gst_caps_unref(caps);
+
+    gst_element_set_state(info->record_pipeline, GST_STATE_PLAYING);
+  }
+
   // TODO: Have mix_pipeline in background while being subscribed to discourse!
 
   info->mix_pipeline = gst_parse_launch(