simplecall: Add shm version of simple-call

2 files changed, 384 insertions, 1 deletions

diff --git a/examples/commandline/Makefile.am b/examples/commandline/Makefile.am
index d374e674..f1cfa994 100644
--- a/examples/commandline/Makefile.am
+++ b/examples/commandline/Makefile.am
@@ -1,5 +1,5 @@
 
-noinst_PROGRAMS = simple-call
+noinst_PROGRAMS = simple-call simple-call-shm
 
 
 AM_CFLAGS = \
@@ -12,3 +12,12 @@ AM_CFLAGS = \
 LDADD = \
 	$(top_builddir)/gst-libs/gst/farsight/libgstfarsight-0.10.la \
 	$(GST_LIBS) 
+
+
+simple_call_shm_CFLAGS = \
+	$(AM_CFLAGS) \
+	$(GIO_UNIX_CFLAGS)
+
+simple_call_shm_LDADD = \
+	$(LDADD) \
+	$(GIO_UNIX_LIBS)
diff --git a/examples/commandline/simple-call-shm.c b/examples/commandline/simple-call-shm.c
new file mode 100644
index 00000000..4ed10029
--- /dev/null
+++ b/examples/commandline/simple-call-shm.c
@@ -0,0 +1,374 @@
+/* Farsight 2 ad-hoc test for simple calls.
+ *
+ * Copyright (C) 2008 Collabora, Nokia
+ * @author: Olivier Crete <olivier.crete@collabora.co.uk>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA  02110-1301 USA
+ */
+
+/*
+ * WARNING:
+ *
+ * Do not use this as an example of a proper use of farsight, it assumes that
+ * both ends have the EXACT same list of codec installed in the EXACT same order
+ */
+
+
+#include <stdlib.h>
+#include <unistd.h>
+
+#include <glib.h>
+#include <gio/gio.h>
+#include <gio/gunixinputstream.h>
+#include <gst/gst.h>
+
+#include <gst/farsight/fs-conference-iface.h>
+
+#define DEFAULT_AUDIOSRC       "audiotestsrc is-live=1 ! audio/x-raw-int, rate=8000 ! identity"
+#define DEFAULT_AUDIOSINK      "alsasink sync=false async=false"
+
+typedef struct _TestSession
+{
+  FsSession *session;
+  FsStream *stream;
+
+  gchar *send_socket;
+  gchar *recv_socket;
+} TestSession;
+
+
+static void
+print_error (GError *error)
+{
+  if (error)
+  {
+    g_error ("Error: %s:%d : %s", g_quark_to_string (error->domain),
+        error->code, error->message);
+  }
+}
+
+static void
+src_pad_added_cb (FsStream *stream, GstPad *pad, FsCodec *codec,
+    gpointer user_data)
+{
+  GstElement *pipeline = GST_ELEMENT_CAST (user_data);
+  GstElement *sink = NULL;
+  GError *error = NULL;
+  GstPad *pad2;
+
+  g_print ("Adding receive pipeline\n");
+
+  if (g_getenv ("AUDIOSINK"))
+    sink = gst_parse_bin_from_description (g_getenv ("AUDIOSINK"), TRUE,
+        &error);
+  else
+    sink = gst_parse_bin_from_description (DEFAULT_AUDIOSINK, TRUE,
+        &error);
+  print_error (error);
+  g_assert (sink);
+
+  g_assert (gst_bin_add (GST_BIN (pipeline), sink));
+
+
+  pad2 = gst_element_get_static_pad (sink, "sink");
+  g_assert (pad2);
+
+  g_assert (GST_PAD_LINK_SUCCESSFUL (gst_pad_link (pad, pad2)));
+
+  g_assert (gst_element_set_state (sink, GST_STATE_PLAYING) !=
+      GST_STATE_CHANGE_FAILURE);
+
+  gst_object_unref (pad2);
+}
+
+static TestSession*
+add_audio_session (GstElement *pipeline, FsConference *conf, guint id,
+    FsParticipant *part, gchar *send_socket, gchar *recv_socket)
+{
+  TestSession *ses = g_slice_new0 (TestSession);
+  GError *error = NULL;
+  GstPad *pad = NULL, *pad2 = NULL;
+  GstElement *src = NULL;
+  GList *cands = NULL;
+  GParameter param = {0};
+  gboolean res;
+  FsCandidate *cand;
+  GList *codecs = NULL;
+
+  ses->send_socket = send_socket;
+  ses->recv_socket = recv_socket;
+
+  ses->session = fs_conference_new_session (conf, FS_MEDIA_TYPE_AUDIO, &error);
+  print_error (error);
+  g_assert (ses->session);
+
+  g_object_get (ses->session, "sink-pad", &pad, NULL);
+
+  if (g_getenv ("AUDIOSRC"))
+    src = gst_parse_bin_from_description (g_getenv ("AUDIOSRC"), TRUE,
+        &error);
+  else
+    src = gst_parse_bin_from_description (DEFAULT_AUDIOSRC, TRUE,
+        &error);
+  print_error (error);
+  g_assert (src);
+
+  g_assert (gst_bin_add (GST_BIN (pipeline), src));
+
+  pad2 = gst_element_get_static_pad (src, "src");
+  g_assert (pad2);
+
+  g_assert (GST_PAD_LINK_SUCCESSFUL (gst_pad_link (pad2, pad)));
+
+  gst_object_unref (pad2);
+  gst_object_unref (pad);
+
+  cand = fs_candidate_new ("", FS_COMPONENT_RTP,
+      FS_CANDIDATE_TYPE_HOST, FS_NETWORK_PROTOCOL_UDP, send_socket, 0);
+  cands = g_list_prepend (NULL, cand);
+
+  param.name = "preferred-local-candidates";
+  g_value_init (&param.value, FS_TYPE_CANDIDATE_LIST);
+  g_value_take_boxed (&param.value, cands);
+
+  ses->stream = fs_session_new_stream (ses->session, part, FS_DIRECTION_BOTH,
+      "shm", 1, &param, &error);
+  print_error (error);
+  g_assert (ses->stream);
+
+  g_value_unset (&param.value);
+
+  g_signal_connect (ses->stream, "src-pad-added",
+      G_CALLBACK (src_pad_added_cb), pipeline);
+
+  codecs = g_list_prepend (NULL,
+      fs_codec_new (FS_CODEC_ID_ANY, "PCMA", FS_MEDIA_TYPE_AUDIO, 0));
+  codecs = g_list_prepend (codecs,
+      fs_codec_new (FS_CODEC_ID_ANY, "PCMU", FS_MEDIA_TYPE_AUDIO, 0));
+
+  res = fs_session_set_codec_preferences (ses->session, codecs, &error);
+  print_error (error);
+  fs_codec_list_destroy (codecs);
+
+
+  g_object_get (ses->session, "codecs", &codecs, NULL);
+  res = fs_stream_set_remote_codecs (ses->stream, codecs, &error);
+  print_error (error);
+  g_assert (res);
+
+
+  return ses;
+}
+
+static gboolean
+async_bus_cb (GstBus *bus, GstMessage *message, gpointer user_data)
+{
+  switch (GST_MESSAGE_TYPE(message))
+  {
+    case GST_MESSAGE_ERROR:
+      {
+        GError *error = NULL;
+        gchar *debug_str = NULL;
+
+        gst_message_parse_error (message, &error, &debug_str);
+        g_print ("Got gst message: %s %s", error->message, debug_str);
+      }
+      break;
+    case GST_MESSAGE_WARNING:
+      {
+        GError *error = NULL;
+        gchar *debug_str = NULL;
+
+        gst_message_parse_warning (message, &error, &debug_str);
+        g_warning ("Got gst message: %s %s", error->message, debug_str);
+      }
+      break;
+    case GST_MESSAGE_ELEMENT:
+      {
+        const GstStructure *s = gst_message_get_structure (message);
+
+        if (gst_structure_has_name (s, "farsight-error"))
+        {
+          gint error;
+          const gchar *error_msg = gst_structure_get_string (s, "error-msg");
+          const gchar *debug_msg = gst_structure_get_string (s, "debug-msg");
+
+          g_assert (gst_structure_get_enum (s, "error-no", FS_TYPE_ERROR,
+                  &error));
+
+          if (FS_ERROR_IS_FATAL (error))
+            g_error ("Farsight fatal error: %d %s %s", error, error_msg,
+                debug_msg);
+          else
+            g_warning ("Farsight non-fatal error: %d %s %s", error, error_msg,
+                debug_msg);
+        }
+        else if (gst_structure_has_name (s, "farsight-new-local-candidate"))
+        {
+          const GValue *val = gst_structure_get_value (s, "candidate");
+          FsCandidate *cand = NULL;
+
+          g_assert (val);
+          cand = g_value_get_boxed (val);
+
+          g_print ("New candidate: socket %s\n", cand->ip);
+          g_print ("You can press ENTER on the other side\n");
+        }
+        else if (gst_structure_has_name (s,
+                "farsight-local-candidates-prepared"))
+        {
+          g_print ("Local candidates prepared\n");
+        }
+        else if (gst_structure_has_name (s, "farsight-recv-codecs-changed"))
+        {
+          const GValue *val = gst_structure_get_value (s, "codecs");
+          GList *codecs = NULL;
+
+          g_assert (val);
+          codecs = g_value_get_boxed (val);
+
+          g_print ("Recv codecs changed:\n");
+          for (; codecs; codecs = g_list_next (codecs))
+          {
+            FsCodec *codec = codecs->data;
+            gchar *tmp = fs_codec_to_string (codec);
+            g_print ("%s\n", tmp);
+            g_free (tmp);
+          }
+        }
+        else if (gst_structure_has_name (s, "farsight-send-codec-changed"))
+        {
+          const GValue *val = gst_structure_get_value (s, "codec");
+          FsCodec *codec = NULL;
+          gchar *tmp;
+          g_assert (val);
+          codec = g_value_get_boxed (val);
+          tmp = fs_codec_to_string (codec);
+
+          g_print ("Send codec changed: %s\n", tmp);
+          g_free (tmp);
+        }
+      }
+      break;
+    default:
+      break;
+  }
+
+  return TRUE;
+}
+
+static void
+free_session (TestSession *ses)
+{
+  g_object_unref (ses->stream);
+  g_object_unref (ses->session);
+  g_slice_free (TestSession, ses);
+}
+
+static void
+skipped_cb (GObject *istream, GAsyncResult *result, gpointer user_data)
+{
+  TestSession *ses = user_data;
+  FsCandidate *cand;
+  GList *cands = NULL;
+  GError *error = NULL;
+  gboolean res;
+
+  cand = fs_candidate_new ("", FS_COMPONENT_RTP,
+      FS_CANDIDATE_TYPE_HOST, FS_NETWORK_PROTOCOL_UDP, ses->send_socket, 0);
+  cand->base_ip = g_strdup (ses->recv_socket);
+  cands = g_list_prepend (NULL, cand);
+
+  res = fs_stream_set_remote_candidates (ses->stream, cands, &error);
+  print_error (error);
+  g_assert (res);
+
+  fs_candidate_list_destroy (cands);
+}
+
+int main (int argc, char **argv)
+{
+  GMainLoop *loop = NULL;
+  GstElement *pipeline = NULL;
+  GstBus *bus = NULL;
+  GstElement *conf = NULL;
+  FsParticipant *part = NULL;
+  GError *error = NULL;
+  GInputStream *istream = NULL;
+  gchar *send_socket, *recv_socket;
+  TestSession *ses;
+
+  gst_init (&argc, &argv);
+
+  if (argc != 3)
+  {
+    g_print ("Usage: %s <send socket> <recv_socket>\n", argv[0]);
+    return 1;
+  }
+
+  send_socket = argv[1];
+  recv_socket = argv[2];
+
+  if (unlink (send_socket) < 0 && errno != ENOENT)
+  {
+    g_print ("Could not delete send or recv sockets");
+    return 2;
+  }
+
+  g_print ("Press ENTER when the other side is ready\n");
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  pipeline = gst_pipeline_new (NULL);
+
+  bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
+  gst_bus_add_watch (bus, async_bus_cb, pipeline);
+  gst_object_unref (bus);
+
+  conf = gst_element_factory_make ("fsrtpconference", NULL);
+  g_assert (conf);
+
+  part = fs_conference_new_participant (FS_CONFERENCE (conf), "test@ignore",
+      &error);
+  print_error (error);
+  g_assert (part);
+
+  g_assert (gst_bin_add (GST_BIN (pipeline), conf));
+
+  istream = g_unix_input_stream_new (0, FALSE);
+
+  ses = add_audio_session (pipeline, FS_CONFERENCE (conf), 1, part, send_socket,
+      recv_socket);
+
+  g_input_stream_skip_async (istream, 1, G_PRIORITY_DEFAULT, NULL, skipped_cb,
+      ses);
+
+  g_assert (gst_element_set_state (pipeline, GST_STATE_PLAYING) !=
+      GST_STATE_CHANGE_FAILURE);
+  g_main_loop_run (loop);
+
+  g_assert (gst_element_set_state (pipeline, GST_STATE_NULL) !=
+      GST_STATE_CHANGE_FAILURE);
+
+  g_object_unref (part);
+  g_object_unref (istream);
+
+  free_session (ses);
+
+  gst_object_unref (pipeline);
+  g_main_loop_unref (loop);
+
+  return 0;
+}