static GstElement *
fs_rtp_dtmf_sound_source_build (FsRtpSpecialSource *source,
GList *negotiated_codecs,
FsCodec *selected_codec)
{
FsCodec *telephony_codec = NULL;
GstCaps *caps = NULL;
GstPad *pad = NULL;
GstElement *dtmfsrc = NULL;
GstElement *capsfilter = NULL;
GstPad *ghostpad = NULL;
GstElement *bin = NULL;
GstElement *encoder = NULL;
GstElement *payloader = NULL;
gchar *encoder_name = NULL;
gchar *payloader_name = NULL;
telephony_codec = get_pcm_law_sound_codec (negotiated_codecs,
&encoder_name, &payloader_name);
g_return_val_if_fail (telephony_codec, NULL);
source->codec = fs_codec_copy (telephony_codec);
GST_DEBUG ("Creating dtmf sound source for " FS_CODEC_FORMAT,
FS_CODEC_ARGS (telephony_codec));
bin = gst_bin_new (NULL);
dtmfsrc = gst_element_factory_make ("dtmfsrc", NULL);
if (!dtmfsrc)
{
GST_ERROR ("Could not make rtpdtmfsrc");
goto error;
}
if (!gst_bin_add (GST_BIN (bin), dtmfsrc))
{
GST_ERROR ("Could not add rtpdtmfsrc to bin");
gst_object_unref (dtmfsrc);
goto error;
}
encoder = gst_element_factory_make (encoder_name, NULL);
if (!encoder)
{
GST_ERROR ("Could not make %s", encoder_name);
goto error;
}
if (!gst_bin_add (GST_BIN (bin), encoder))
{
GST_ERROR ("Could not add %s to bin", encoder_name);
gst_object_unref (dtmfsrc);
goto error;
}
if (!gst_element_link_pads (dtmfsrc, "src", encoder, "sink"))
{
GST_ERROR ("Could not link the rtpdtmfsrc and %s", encoder_name);
goto error;
}
payloader = gst_element_factory_make (payloader_name, NULL);
if (!payloader)
{
GST_ERROR ("Could not make %s", payloader_name);
goto error;
}
if (!gst_bin_add (GST_BIN (bin), payloader))
{
GST_ERROR ("Could not add %s to bin", payloader_name);
gst_object_unref (dtmfsrc);
goto error;
}
if (!gst_element_link_pads (encoder, "src", payloader, "sink"))
{
GST_ERROR ("Could not link the %s and %s", encoder_name, payloader_name);
goto error;
}
capsfilter = gst_element_factory_make ("capsfilter", NULL);
if (!capsfilter)
{
GST_ERROR ("Could not make capsfilter");
goto error;
}
if (!gst_bin_add (GST_BIN (bin), capsfilter))
{
GST_ERROR ("Could not add capsfilter to bin");
gst_object_unref (capsfilter);
goto error;
}
caps = fs_codec_to_gst_caps (telephony_codec);
g_object_set (capsfilter, "caps", caps, NULL);
{
gchar *str = gst_caps_to_string (caps);
GST_DEBUG ("Using caps %s for dtmf", str);
g_free (str);
}
gst_caps_unref (caps);
if (!gst_element_link_pads (payloader, "src", capsfilter, "sink"))
{
GST_ERROR ("Could not link the %s and its capsfilter", payloader_name);
goto error;
}
pad = gst_element_get_static_pad (capsfilter, "src");
if (!pad)
{
GST_ERROR ("Could not get \"src\" pad from capsfilter");
goto error;
}
ghostpad = gst_ghost_pad_new ("src", pad);
if (!ghostpad)
{
GST_ERROR ("Could not create a ghostpad for capsfilter src pad"
" for dtmfsrc");
goto error;
}
if (!gst_element_add_pad (bin, ghostpad))
{
GST_ERROR ("Could not get \"src\" ghostpad to dtmf sound source bin");
gst_object_unref (pad);
goto error;
}
gst_object_unref (pad);
return bin;
error:
gst_object_unref (bin);
return NULL;
}
gboolean
fs_rtp_sub_stream_add_output_ghostpad_unlock (FsRtpSubStream *substream,
GError **error)
{
GstPad *valve_srcpad;
gchar *padname = NULL;
GstPad *ghostpad = NULL;
FsCodec *codec = NULL;
if (fs_rtp_sub_stream_has_stopped_enter (substream))
{
FS_RTP_SESSION_UNLOCK (substream->priv->session);
return TRUE;
}
if (substream->priv->adding_output_ghostpad)
{
FS_RTP_SESSION_UNLOCK (substream->priv->session);
goto out;
}
g_assert (substream->priv->output_ghostpad == NULL);
substream->priv->adding_output_ghostpad = TRUE;
padname = g_strdup_printf ("src_%u_%u_%d", substream->priv->session->id,
substream->ssrc,
substream->pt);
FS_RTP_SESSION_UNLOCK (substream->priv->session);
valve_srcpad = gst_element_get_static_pad (substream->priv->output_valve,
"src");
g_assert (valve_srcpad);
ghostpad = gst_ghost_pad_new_from_template (padname, valve_srcpad,
gst_element_class_get_pad_template (
GST_ELEMENT_GET_CLASS (substream->priv->conference),
"src_%d_%d_%d"));
gst_object_unref (valve_srcpad);
g_free (padname);
if (!ghostpad)
{
g_set_error (error, FS_ERROR, FS_ERROR_CONSTRUCTION,
"Could not build ghostpad src_%u_%u_%d", substream->priv->session->id,
substream->ssrc, substream->pt);
goto error;
}
if (!gst_pad_set_active (ghostpad, TRUE))
{
g_set_error (error, FS_ERROR, FS_ERROR_CONSTRUCTION,
"Could not activate the src_%u_%u_%d", substream->priv->session->id,
substream->ssrc, substream->pt);
gst_object_unref (ghostpad);
goto error;
}
if (!gst_element_add_pad (GST_ELEMENT (substream->priv->conference),
ghostpad))
{
g_set_error (error, FS_ERROR, FS_ERROR_CONSTRUCTION,
"Could add build ghostpad src_%u_%u_%d to the conference",
substream->priv->session->id, substream->ssrc, substream->pt);
gst_object_unref (ghostpad);
goto error;
}
FS_RTP_SESSION_LOCK (substream->priv->session);
substream->priv->output_ghostpad = ghostpad;
GST_DEBUG ("Src pad added on substream for ssrc:%X pt:%u " FS_CODEC_FORMAT,
substream->ssrc, substream->pt,
FS_CODEC_ARGS (substream->codec));
codec = fs_codec_copy (substream->codec);
FS_RTP_SESSION_UNLOCK (substream->priv->session);
g_signal_emit (substream, signals[SRC_PAD_ADDED], 0,
ghostpad, codec);
g_signal_emit (substream, signals[CODEC_CHANGED], 0);
fs_codec_destroy (codec);
g_object_set (substream->priv->output_valve, "drop", FALSE, NULL);
out:
fs_rtp_sub_stream_has_stopped_exit (substream);
return TRUE;
error:
substream->priv->adding_output_ghostpad = FALSE;
fs_rtp_sub_stream_has_stopped_exit (substream);
return FALSE;
}
static GstElement *
fs_rtp_dtmf_event_source_build (FsRtpSpecialSource *source,
GList *negotiated_codec_associations,
FsCodec *selected_codec)
{
FsCodec *telephony_codec = NULL;
GstCaps *caps = NULL;
GstPad *pad = NULL;
GstElement *dtmfsrc = NULL;
GstElement *capsfilter = NULL;
GstPad *ghostpad = NULL;
GstElement *bin = NULL;
telephony_codec = fs_rtp_dtmf_event_source_get_codec (
FS_RTP_SPECIAL_SOURCE_GET_CLASS(source), negotiated_codec_associations,
selected_codec);
g_return_val_if_fail (telephony_codec, NULL);
source->codec = fs_codec_copy (telephony_codec);
bin = gst_bin_new (NULL);
GST_DEBUG ("Creating telephone-event source for " FS_CODEC_FORMAT,
FS_CODEC_ARGS (telephony_codec));
dtmfsrc = gst_element_factory_make ("rtpdtmfsrc", NULL);
if (!dtmfsrc)
{
GST_ERROR ("Could not make rtpdtmfsrc");
goto error;
}
if (!gst_bin_add (GST_BIN (bin), dtmfsrc))
{
GST_ERROR ("Could not add rtpdtmfsrc to bin");
gst_object_unref (dtmfsrc);
goto error;
}
capsfilter = gst_element_factory_make ("capsfilter", NULL);
if (!capsfilter)
{
GST_ERROR ("Could not make capsfilter");
goto error;
}
if (!gst_bin_add (GST_BIN (bin), capsfilter))
{
GST_ERROR ("Could not add capsfilter to bin");
gst_object_unref (capsfilter);
goto error;
}
caps = fs_codec_to_gst_caps (telephony_codec);
g_object_set (capsfilter, "caps", caps, NULL);
{
gchar *str = gst_caps_to_string (caps);
GST_DEBUG ("Using caps %s for dtmf", str);
g_free (str);
}
gst_caps_unref (caps);
if (!gst_element_link_pads (dtmfsrc, "src", capsfilter, "sink"))
{
GST_ERROR ("Could not link the rtpdtmfsrc and its capsfilter");
goto error;
}
pad = gst_element_get_static_pad (capsfilter, "src");
if (!pad)
{
GST_ERROR ("Could not get \"src\" pad from capsfilter");
goto error;
}
ghostpad = gst_ghost_pad_new ("src", pad);
if (!ghostpad)
{
GST_ERROR ("Could not create a ghostpad for capsfilter src pad for"
" rtpdtmfsrc");
goto error;
}
if (!gst_element_add_pad (bin, ghostpad))
{
GST_ERROR ("Could not get \"src\" ghostpad to dtmf source bin");
gst_object_unref (pad);
goto error;
}
gst_object_unref (pad);
return bin;
error:
gst_object_unref (bin);
return NULL;
}
