*
* 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.
*/
#ifdef HAVE_CONFIG_H
#include <gst/base/gstbitreader.h>
#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/audio/audio.h>
#include <string.h>
#include "gstrtpmp4adepay.h"
+#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpmp4adepay_debug);
#define GST_CAT_DEFAULT (rtpmp4adepay_debug)
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg,"
- "mpegversion = (int) 4," "framed = (boolean) true, "
+ "mpegversion = (int) 4," "framed = (boolean) { false, true }, "
"stream-format = (string) raw")
);
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\", "
- "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) [1, MAX ], "
"encoding-name = (string) \"MP4A-LATM\""
/* All optional parameters
static gboolean gst_rtp_mp4a_depay_setcaps (GstRTPBaseDepayload * depayload,
GstCaps * caps);
static GstBuffer *gst_rtp_mp4a_depay_process (GstRTPBaseDepayload * depayload,
- GstBuffer * buf);
+ GstRTPBuffer * rtp);
static GstStateChangeReturn gst_rtp_mp4a_depay_change_state (GstElement *
element, GstStateChange transition);
gstelement_class->change_state = gst_rtp_mp4a_depay_change_state;
- gstrtpbasedepayload_class->process = gst_rtp_mp4a_depay_process;
+ gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_mp4a_depay_process;
gstrtpbasedepayload_class->set_caps = gst_rtp_mp4a_depay_setcaps;
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_rtp_mp4a_depay_src_template));
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&gst_rtp_mp4a_depay_sink_template));
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_mp4a_depay_src_template);
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_mp4a_depay_sink_template);
- gst_element_class_set_details_simple (gstelement_class,
+ gst_element_class_set_static_metadata (gstelement_class,
"RTP MPEG4 audio depayloader", "Codec/Depayloader/Network/RTP",
"Extracts MPEG4 audio from RTP packets (RFC 3016)",
"Nokia Corporation (contact <stefan.kost@nokia.com>), "
gst_rtp_mp4a_depay_init (GstRtpMP4ADepay * rtpmp4adepay)
{
rtpmp4adepay->adapter = gst_adapter_new ();
+ rtpmp4adepay->framed = FALSE;
}
static void
rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
+ rtpmp4adepay->framed = FALSE;
+
structure = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
srccaps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, 4,
- "framed", G_TYPE_BOOLEAN, TRUE, "channels", G_TYPE_INT, channels,
+ "framed", G_TYPE_BOOLEAN, FALSE, "channels", G_TYPE_INT, channels,
"stream-format", G_TYPE_STRING, "raw", NULL);
if ((str = gst_structure_get_string (structure, "config"))) {
}
static GstBuffer *
-gst_rtp_mp4a_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
+gst_rtp_mp4a_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
{
GstRtpMP4ADepay *rtpmp4adepay;
GstBuffer *outbuf;
- GstRTPBuffer rtp = { NULL };
GstMapInfo map;
rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
/* flush remaining data on discont */
- if (GST_BUFFER_IS_DISCONT (buf)) {
+ if (GST_BUFFER_IS_DISCONT (rtp->buffer)) {
gst_adapter_clear (rtpmp4adepay->adapter);
}
- gst_rtp_buffer_map (buf, GST_MAP_READ, &rtp);
- outbuf = gst_rtp_buffer_get_payload_buffer (&rtp);
+ outbuf = gst_rtp_buffer_get_payload_buffer (rtp);
+
+ if (!rtpmp4adepay->framed) {
+ if (gst_rtp_buffer_get_marker (rtp)) {
+ GstCaps *caps;
+
+ rtpmp4adepay->framed = TRUE;
+
+ gst_rtp_base_depayload_push (depayload, outbuf);
+
+ caps = gst_pad_get_current_caps (depayload->srcpad);
+ caps = gst_caps_make_writable (caps);
+ gst_caps_set_simple (caps, "framed", G_TYPE_BOOLEAN, TRUE, NULL);
+ gst_pad_set_caps (depayload->srcpad, caps);
+ gst_caps_unref (caps);
+ return NULL;
+ } else {
+ return outbuf;
+ }
+ }
outbuf = gst_buffer_make_writable (outbuf);
- GST_BUFFER_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buf);
+ GST_BUFFER_PTS (outbuf) = GST_BUFFER_PTS (rtp->buffer);
gst_adapter_push (rtpmp4adepay->adapter, outbuf);
/* RTP marker bit indicates the last packet of the AudioMuxElement => create
* and push a buffer */
- if (gst_rtp_buffer_get_marker (&rtp)) {
+ if (gst_rtp_buffer_get_marker (rtp)) {
guint avail;
guint i;
guint8 *data;
GstClockTime timestamp;
avail = gst_adapter_available (rtpmp4adepay->adapter);
- timestamp = gst_adapter_prev_timestamp (rtpmp4adepay->adapter, NULL);
+ timestamp = gst_adapter_prev_pts (rtpmp4adepay->adapter, NULL);
GST_LOG_OBJECT (rtpmp4adepay, "have marker and %u available", avail);
/* take data out, skip the header */
pos += skip;
- tmp = gst_buffer_copy_region (outbuf, GST_BUFFER_COPY_MEMORY, pos,
- data_len);
+ tmp = gst_buffer_copy_region (outbuf, GST_BUFFER_COPY_ALL, pos, data_len);
/* skip data too */
skip += data_len;
pos += data_len;
- /* update our pointers whith what we consumed */
+ /* update our pointers with what we consumed */
data += skip;
avail -= skip;
- GST_BUFFER_TIMESTAMP (tmp) = timestamp;
+ GST_BUFFER_PTS (tmp) = timestamp;
+ gst_rtp_drop_non_audio_meta (depayload, tmp);
gst_rtp_base_depayload_push (depayload, tmp);
/* shift ts for next buffers */
gst_buffer_unmap (outbuf, &map);
gst_buffer_unref (outbuf);
}
- gst_rtp_buffer_unmap (&rtp);
return NULL;
/* ERRORS */
("Packet did not validate"), ("wrong packet size"));
gst_buffer_unmap (outbuf, &map);
gst_buffer_unref (outbuf);
- gst_rtp_buffer_unmap (&rtp);
return NULL;
}
}
gst_adapter_clear (rtpmp4adepay->adapter);
rtpmp4adepay->frame_len = 0;
rtpmp4adepay->numSubFrames = 0;
+ rtpmp4adepay->framed = FALSE;
break;
default:
break;
projects / platform / upstream / gst-plugins-good.git / blobdiff