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server: use appsink and appsrc with the API

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Use the appsink/appsrc API instead of the signals for higher
performance.
This commit is contained in:
Wim Taymans 2009-04-14 23:38:58 +02:00 committed by Wim Taymans
parent 5a074c81dd
commit 3f1f38f479
3 changed files with 40 additions and 15 deletions
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4
gst/rtsp-server/Makefile.am
View file

@ -29,7 +29,9 @@ libgstrtspserver_@GST_MAJORMINOR@_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstrtspserver_@GST_MAJORMINOR@_la_LIBADD = \
$(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) \
-lgstrtp-@GST_MAJORMINOR@ -lgstrtsp-@GST_MAJORMINOR@ \
-lgstsdp-@GST_MAJORMINOR@ $(GST_LIBS) $(LIBM)
-lgstsdp-@GST_MAJORMINOR@ \
-lgstapp-@GST_MAJORMINOR@ \
$(GST_LIBS) $(LIBM)
libgstrtspserver_@GST_MAJORMINOR@_la_LIBTOOLFLAGS = --tag=disable-static
libgstrtspserver_@GST_MAJORMINOR@includedir = $(includedir)/gstreamer-@GST_MAJORMINOR@/gst/rtsp-server

9
gst/rtsp-server/rtsp-client.c
View file

@ -1017,6 +1017,7 @@ handle_data (GstRTSPClient *client, GstRTSPMessage *message)
guint8 *data;
guint size;
GstBuffer *buffer;
gboolean handled;
/* find the stream for this message */
res = gst_rtsp_message_parse_data (message, &channel);
@ -1030,6 +1031,7 @@ handle_data (GstRTSPClient *client, GstRTSPMessage *message)
GST_BUFFER_MALLOCDATA (buffer) = data;
GST_BUFFER_SIZE (buffer) = size;
handled = FALSE;
for (walk = client->streams; walk; walk = g_list_next (walk)) {
GstRTSPSessionStream *stream = (GstRTSPSessionStream *) walk->data;
GstRTSPMediaStream *mstream;
@ -1048,12 +1050,17 @@ handle_data (GstRTSPClient *client, GstRTSPMessage *message)
/* dispatch to the stream based on the channel number */
if (tr->interleaved.min == channel) {
gst_rtsp_media_stream_rtp (mstream, buffer);
handled = TRUE;
break;
} else if (tr->interleaved.max == channel) {
gst_rtsp_media_stream_rtcp (mstream, buffer);
handled = TRUE;
break;
}
}
}
gst_buffer_unref (buffer);
if (!handled)
gst_buffer_unref (buffer);
}
/**

42
gst/rtsp-server/rtsp-media.c
View file

@ -17,6 +17,9 @@
* Boston, MA 02111-1307, USA.
*/
#include <gst/app/gstappsrc.h>
#include <gst/app/gstappsink.h>
#include "rtsp-media.h"
#define DEFAULT_SHARED FALSE
@ -468,6 +471,8 @@ weird_type:
* Handle an RTP buffer for the stream. This method is usually called when a
* message has been received from a client using the TCP transport.
*
* This function takes ownership of @buffer.
*
* Returns: a GstFlowReturn.
*/
GstFlowReturn
@ -475,7 +480,7 @@ gst_rtsp_media_stream_rtp (GstRTSPMediaStream *stream, GstBuffer *buffer)
{
GstFlowReturn ret;
g_signal_emit_by_name (stream->appsrc[0], "push-buffer", buffer, &ret);
ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[0]), buffer);
return ret;
}
@ -488,6 +493,8 @@ gst_rtsp_media_stream_rtp (GstRTSPMediaStream *stream, GstBuffer *buffer)
* Handle an RTCP buffer for the stream. This method is usually called when a
* message has been received from a client using the TCP transport.
*
* This function takes ownership of @buffer.
*
* Returns: a GstFlowReturn.
*/
GstFlowReturn
@ -495,9 +502,9 @@ gst_rtsp_media_stream_rtcp (GstRTSPMediaStream *stream, GstBuffer *buffer)
{
GstFlowReturn ret;
g_signal_emit_by_name (stream->appsrc[1], "push-buffer", buffer, &ret);
ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[1]), buffer);
return GST_FLOW_ERROR;
return ret;
}
/* Allocate the udp ports and sockets */
@ -710,20 +717,23 @@ on_timeout (GObject *session, GObject *source, GstRTSPMedia *media)
g_message ("%p: source %p timeout", media, source);
}
static void
handle_new_buffer (GstElement *sink, GstRTSPMediaStream *stream)
static GstFlowReturn
handle_new_buffer (GstAppSink *sink, gpointer user_data)
{
GList *walk;
GstBuffer *buffer;
GstRTSPMediaStream *stream;
g_signal_emit_by_name (sink, "pull-buffer", &buffer);
buffer = gst_app_sink_pull_buffer (sink);
if (!buffer)
return;
return GST_FLOW_OK;
stream = (GstRTSPMediaStream *) user_data;
for (walk = stream->transports; walk; walk = g_list_next (walk)) {
GstRTSPMediaTrans *tr = (GstRTSPMediaTrans *) walk->data;
if (sink == stream->appsink[0]) {
if (GST_ELEMENT_CAST (sink) == stream->appsink[0]) {
if (tr->send_rtp)
tr->send_rtp (buffer, tr->transport->interleaved.min, tr->user_data);
}
@ -733,8 +743,16 @@ handle_new_buffer (GstElement *sink, GstRTSPMediaStream *stream)
}
}
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
static GstAppSinkCallbacks sink_cb = {
NULL, /* not interested in EOS */
NULL, /* not interested in preroll buffers */
handle_new_buffer
};
/* prepare the pipeline objects to handle @stream in @media */
static gboolean
setup_stream (GstRTSPMediaStream *stream, guint idx, GstRTSPMedia *media)
@ -762,15 +780,13 @@ setup_stream (GstRTSPMediaStream *stream, guint idx, GstRTSPMedia *media)
stream->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
stream->appsink[i] = gst_element_factory_make ("appsink", NULL);
g_object_set (stream->appsink[i], "async", FALSE, "sync", FALSE, NULL);
g_object_set (stream->appsink[i], "emit-signals", TRUE, NULL);
g_object_set (stream->appsink[i], "emit-signals", FALSE, NULL);
g_object_set (stream->appsink[i], "preroll-queue-len", 1, NULL);
gst_bin_add (GST_BIN_CAST (media->pipeline), stream->appsink[i]);
gst_bin_add (GST_BIN_CAST (media->pipeline), stream->appsrc[i]);
gst_app_sink_set_callbacks (GST_APP_SINK_CAST (stream->appsink[i]),
&sink_cb, stream, NULL);
}
g_signal_connect (stream->appsink[0], "new-buffer",
(GCallback) handle_new_buffer, stream);
g_signal_connect (stream->appsink[1], "new-buffer",
(GCallback) handle_new_buffer, stream);
/* hook up the stream to the RTP session elements. */
name = g_strdup_printf ("send_rtp_sink_%d", idx);