*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/**
+ * SECTION:element-rtpac3pay
+ * @title: rtpac3pay
+ * @see_also: rtpac3depay
+ *
+ * Payload AC3 audio into RTP packets according to RFC 4184.
+ * For detailed information see: http://www.rfc-editor.org/rfc/rfc4184.txt
+ *
+ * ## Example pipeline
+ * |[
+ * gst-launch-1.0 -v audiotestsrc ! avenc_ac3 ! rtpac3pay ! udpsink
+ * ]| This example pipeline will encode and payload AC3 stream. Refer to
+ * the rtpac3depay example to depayload and decode the RTP stream.
+ *
*/
#ifdef HAVE_CONFIG_H
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/audio/audio.h>
#include "gstrtpac3pay.h"
+#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpac3pay_debug);
#define GST_CAT_DEFAULT (rtpac3pay_debug)
static GstStateChangeReturn gst_rtp_ac3_pay_change_state (GstElement * element,
GstStateChange transition);
-static gboolean gst_rtp_ac3_pay_setcaps (GstBaseRTPPayload * payload,
+static gboolean gst_rtp_ac3_pay_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
-static gboolean gst_rtp_ac3_pay_handle_event (GstBaseRTPPayload * payload,
+static gboolean gst_rtp_ac3_pay_sink_event (GstRTPBasePayload * payload,
GstEvent * event);
static GstFlowReturn gst_rtp_ac3_pay_flush (GstRtpAC3Pay * rtpac3pay);
-static GstFlowReturn gst_rtp_ac3_pay_handle_buffer (GstBaseRTPPayload * payload,
+static GstFlowReturn gst_rtp_ac3_pay_handle_buffer (GstRTPBasePayload * payload,
GstBuffer * buffer);
#define gst_rtp_ac3_pay_parent_class parent_class
-G_DEFINE_TYPE (GstRtpAC3Pay, gst_rtp_ac3_pay, GST_TYPE_BASE_RTP_PAYLOAD);
+G_DEFINE_TYPE (GstRtpAC3Pay, gst_rtp_ac3_pay, GST_TYPE_RTP_BASE_PAYLOAD);
static void
gst_rtp_ac3_pay_class_init (GstRtpAC3PayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
- GstBaseRTPPayloadClass *gstbasertppayload_class;
+ GstRTPBasePayloadClass *gstrtpbasepayload_class;
GST_DEBUG_CATEGORY_INIT (rtpac3pay_debug, "rtpac3pay", 0,
"AC3 Audio RTP Depayloader");
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
- gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+ gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gobject_class->finalize = gst_rtp_ac3_pay_finalize;
gstelement_class->change_state = gst_rtp_ac3_pay_change_state;
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_rtp_ac3_pay_src_template));
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_rtp_ac3_pay_sink_template));
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_ac3_pay_src_template);
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_ac3_pay_sink_template);
- gst_element_class_set_details_simple (gstelement_class,
+ gst_element_class_set_static_metadata (gstelement_class,
"RTP AC3 audio payloader", "Codec/Payloader/Network/RTP",
"Payload AC3 audio as RTP packets (RFC 4184)",
"Wim Taymans <wim.taymans@gmail.com>");
- gstbasertppayload_class->set_caps = gst_rtp_ac3_pay_setcaps;
- gstbasertppayload_class->handle_event = gst_rtp_ac3_pay_handle_event;
- gstbasertppayload_class->handle_buffer = gst_rtp_ac3_pay_handle_buffer;
+ gstrtpbasepayload_class->set_caps = gst_rtp_ac3_pay_setcaps;
+ gstrtpbasepayload_class->sink_event = gst_rtp_ac3_pay_sink_event;
+ gstrtpbasepayload_class->handle_buffer = gst_rtp_ac3_pay_handle_buffer;
}
static void
}
static gboolean
-gst_rtp_ac3_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
+gst_rtp_ac3_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
gboolean res;
gint rate;
if (!gst_structure_get_int (structure, "rate", &rate))
rate = 90000; /* default */
- gst_basertppayload_set_options (payload, "audio", TRUE, "AC3", rate);
- res = gst_basertppayload_set_outcaps (payload, NULL);
+ gst_rtp_base_payload_set_options (payload, "audio", TRUE, "AC3", rate);
+ res = gst_rtp_base_payload_set_outcaps (payload, NULL);
return res;
}
static gboolean
-gst_rtp_ac3_pay_handle_event (GstBaseRTPPayload * payload, GstEvent * event)
+gst_rtp_ac3_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
{
gboolean res;
GstRtpAC3Pay *rtpac3pay;
break;
}
- res =
- GST_BASE_RTP_PAYLOAD_CLASS (parent_class)->handle_event (payload, event);
+ res = GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
return res;
}
/* number of frames */
NF = rtpac3pay->NF;
- mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpac3pay);
+ mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpac3pay);
GST_LOG_OBJECT (rtpac3pay, "flushing %u bytes", avail);
guint payload_len;
guint packet_len;
GstRTPBuffer rtp = { NULL, };
+ GstBuffer *payload_buffer;
/* this will be the total length of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (2 + avail, 0, 0);
payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0);
/* create buffer to hold the payload */
- outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0);
+ outbuf =
+ gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
+ (rtpac3pay), 2, 0, 0);
if (FT == 0) {
/* check if it all fits */
payload[1] = NF;
payload_len -= 2;
- gst_adapter_copy (rtpac3pay->adapter, &payload[2], 0, payload_len);
- gst_adapter_flush (rtpac3pay->adapter, payload_len);
-
- avail -= payload_len;
- if (avail == 0)
+ if (avail == payload_len)
gst_rtp_buffer_set_marker (&rtp, TRUE);
gst_rtp_buffer_unmap (&rtp);
- GST_BUFFER_TIMESTAMP (outbuf) = rtpac3pay->first_ts;
+ payload_buffer =
+ gst_adapter_take_buffer_fast (rtpac3pay->adapter, payload_len);
+
+ gst_rtp_copy_audio_meta (rtpac3pay, outbuf, payload_buffer);
+
+ outbuf = gst_buffer_append (outbuf, payload_buffer);
+
+ avail -= payload_len;
+
+ GST_BUFFER_PTS (outbuf) = rtpac3pay->first_ts;
GST_BUFFER_DURATION (outbuf) = rtpac3pay->duration;
- ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpac3pay), outbuf);
+ ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpac3pay), outbuf);
}
return ret;
}
static GstFlowReturn
-gst_rtp_ac3_pay_handle_buffer (GstBaseRTPPayload * basepayload,
+gst_rtp_ac3_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRtpAC3Pay *rtpac3pay;
GstFlowReturn ret;
- gsize size, avail, left, NF;
- guint8 *data, *p;
+ gsize avail, left, NF;
+ GstMapInfo map;
+ guint8 *p;
guint packet_len;
GstClockTime duration, timestamp;
rtpac3pay = GST_RTP_AC3_PAY (basepayload);
- data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ);
+ gst_buffer_map (buffer, &map, GST_MAP_READ);
duration = GST_BUFFER_DURATION (buffer);
- timestamp = GST_BUFFER_TIMESTAMP (buffer);
+ timestamp = GST_BUFFER_PTS (buffer);
if (GST_BUFFER_IS_DISCONT (buffer)) {
GST_DEBUG_OBJECT (rtpac3pay, "DISCONT");
gst_rtp_ac3_pay_reset (rtpac3pay);
}
- /* count the amount of incomming packets */
+ /* count the amount of incoming packets */
NF = 0;
- left = size;
- p = data;
+ left = map.size;
+ p = map.data;
while (TRUE) {
guint bsid, fscod, frmsizecod, frame_size;
break;
NF++;
- GST_DEBUG_OBJECT (rtpac3pay, "found frame %u of size %u", NF, frame_size);
+ GST_DEBUG_OBJECT (rtpac3pay, "found frame %" G_GSIZE_FORMAT " of size %u",
+ NF, frame_size);
p += frame_size;
left -= frame_size;
}
- gst_buffer_unmap (buffer, data, size);
+ gst_buffer_unmap (buffer, &map);
if (NF == 0)
goto no_frames;
/* get packet length of previous data and this new data,
* payload length includes a 4 byte header */
- packet_len = gst_rtp_buffer_calc_packet_len (2 + avail + size, 0, 0);
+ packet_len = gst_rtp_buffer_calc_packet_len (2 + avail + map.size, 0, 0);
/* if this buffer is going to overflow the packet, flush what we
* have. */
- if (gst_basertppayload_is_filled (basepayload,
+ if (gst_rtp_base_payload_is_filled (basepayload,
packet_len, rtpac3pay->duration + duration)) {
ret = gst_rtp_ac3_pay_flush (rtpac3pay);
avail = 0;
projects / platform / upstream / gst-plugins-good.git / blobdiff