*
* 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/rtp/gstrtpbuffer.h>
#include "gstrtpmp4gpay.h"
+#include "gstrtputils.h"
GST_DEBUG_CATEGORY_STATIC (rtpmp4gpay_debug);
#define GST_CAT_DEFAULT (rtpmp4gpay_debug)
static GstStateChangeReturn gst_rtp_mp4g_pay_change_state (GstElement * element,
GstStateChange transition);
-static gboolean gst_rtp_mp4g_pay_setcaps (GstBaseRTPPayload * payload,
+static gboolean gst_rtp_mp4g_pay_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
-static GstFlowReturn gst_rtp_mp4g_pay_handle_buffer (GstBaseRTPPayload *
+static GstFlowReturn gst_rtp_mp4g_pay_handle_buffer (GstRTPBasePayload *
payload, GstBuffer * buffer);
+static gboolean gst_rtp_mp4g_pay_sink_event (GstRTPBasePayload * payload,
+ GstEvent * event);
-GST_BOILERPLATE (GstRtpMP4GPay, gst_rtp_mp4g_pay, GstBaseRTPPayload,
- GST_TYPE_BASE_RTP_PAYLOAD)
-
- static void gst_rtp_mp4g_pay_base_init (gpointer klass)
-{
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_rtp_mp4g_pay_src_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&gst_rtp_mp4g_pay_sink_template));
-
- gst_element_class_set_details_simple (element_class, "RTP MPEG4 ES payloader",
- "Codec/Payloader/Network",
- "Payload MPEG4 elementary streams as RTP packets (RFC 3640)",
- "Wim Taymans <wim.taymans@gmail.com>");
-}
+#define gst_rtp_mp4g_pay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpMP4GPay, gst_rtp_mp4g_pay, GST_TYPE_RTP_BASE_PAYLOAD);
static void
gst_rtp_mp4g_pay_class_init (GstRtpMP4GPayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
- GstBaseRTPPayloadClass *gstbasertppayload_class;
+ GstRTPBasePayloadClass *gstrtpbasepayload_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
- gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+ gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gobject_class->finalize = gst_rtp_mp4g_pay_finalize;
gstelement_class->change_state = gst_rtp_mp4g_pay_change_state;
- gstbasertppayload_class->set_caps = gst_rtp_mp4g_pay_setcaps;
- gstbasertppayload_class->handle_buffer = gst_rtp_mp4g_pay_handle_buffer;
+ gstrtpbasepayload_class->set_caps = gst_rtp_mp4g_pay_setcaps;
+ gstrtpbasepayload_class->handle_buffer = gst_rtp_mp4g_pay_handle_buffer;
+ gstrtpbasepayload_class->sink_event = gst_rtp_mp4g_pay_sink_event;
+
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_mp4g_pay_src_template);
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_mp4g_pay_sink_template);
+
+ gst_element_class_set_static_metadata (gstelement_class,
+ "RTP MPEG4 ES payloader",
+ "Codec/Payloader/Network/RTP",
+ "Payload MPEG4 elementary streams as RTP packets (RFC 3640)",
+ "Wim Taymans <wim.taymans@gmail.com>");
GST_DEBUG_CATEGORY_INIT (rtpmp4gpay_debug, "rtpmp4gpay", 0,
"MP4-generic RTP Payloader");
}
static void
-gst_rtp_mp4g_pay_init (GstRtpMP4GPay * rtpmp4gpay, GstRtpMP4GPayClass * klass)
+gst_rtp_mp4g_pay_init (GstRtpMP4GPay * rtpmp4gpay)
{
rtpmp4gpay->adapter = gst_adapter_new ();
}
GST_DEBUG_OBJECT (rtpmp4gpay, "reset");
gst_adapter_clear (rtpmp4gpay->adapter);
+}
+
+static void
+gst_rtp_mp4g_pay_cleanup (GstRtpMP4GPay * rtpmp4gpay)
+{
+ gst_rtp_mp4g_pay_reset (rtpmp4gpay);
g_free (rtpmp4gpay->params);
rtpmp4gpay->params = NULL;
rtpmp4gpay->mode = NULL;
rtpmp4gpay->frame_len = 0;
- rtpmp4gpay->offset = 0;
}
static void
rtpmp4gpay = GST_RTP_MP4G_PAY (object);
- gst_rtp_mp4g_pay_reset (rtpmp4gpay);
+ gst_rtp_mp4g_pay_cleanup (rtpmp4gpay);
g_object_unref (rtpmp4gpay->adapter);
rtpmp4gpay->adapter = NULL;
gst_rtp_mp4g_pay_parse_audio_config (GstRtpMP4GPay * rtpmp4gpay,
GstBuffer * buffer)
{
- guint8 *data;
- guint size;
- guint8 objectType;
- guint8 samplingIdx;
- guint8 channelCfg;
+ GstMapInfo map;
+ guint8 objectType = 0;
+ guint8 samplingIdx = 0;
+ guint8 channelCfg = 0;
GstBitReader br;
- data = GST_BUFFER_DATA (buffer);
- size = GST_BUFFER_SIZE (buffer);
+ gst_buffer_map (buffer, &map, GST_MAP_READ);
- gst_bit_reader_init (&br, data, size);
+ gst_bit_reader_init (&br, map.data, map.size);
/* any object type is fine, we need to copy it to the profile-level-id field. */
if (!gst_bit_reader_get_bits_uint8 (&br, &objectType, 5))
/* rtp rate depends on sampling rate of the audio */
if (samplingIdx == 15) {
- guint32 rate;
+ guint32 rate = 0;
/* index of 15 means we get the rate in the next 24 bits */
if (!gst_bit_reader_get_bits_uint32 (&br, &rate, 24))
case 6:
case 7:
{
- guint8 frameLenFlag;
+ guint8 frameLenFlag = 0;
if (gst_bit_reader_get_bits_uint8 (&br, &frameLenFlag, 1))
if (frameLenFlag)
objectType, samplingIdx, rtpmp4gpay->rate, channelCfg,
rtpmp4gpay->frame_len);
+ gst_buffer_unmap (buffer, &map);
return TRUE;
/* ERROR */
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
(NULL), ("config string too short"));
+ gst_buffer_unmap (buffer, &map);
return FALSE;
}
invalid_object:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
(NULL), ("invalid object type"));
+ gst_buffer_unmap (buffer, &map);
return FALSE;
}
wrong_freq:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED,
(NULL), ("unsupported frequency index %d", samplingIdx));
+ gst_buffer_unmap (buffer, &map);
return FALSE;
}
wrong_channels:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED,
(NULL), ("unsupported number of channels %d, must < 8", channelCfg));
+ gst_buffer_unmap (buffer, &map);
return FALSE;
}
}
gst_rtp_mp4g_pay_parse_video_config (GstRtpMP4GPay * rtpmp4gpay,
GstBuffer * buffer)
{
- guint8 *data;
- guint size;
+ GstMapInfo map;
guint32 code;
- data = GST_BUFFER_DATA (buffer);
- size = GST_BUFFER_SIZE (buffer);
+ gst_buffer_map (buffer, &map, GST_MAP_READ);
- if (size < 5)
+ if (map.size < 5)
goto too_short;
- code = GST_READ_UINT32_BE (data);
+ code = GST_READ_UINT32_BE (map.data);
g_free (rtpmp4gpay->profile);
if (code == VOS_STARTCODE) {
/* get profile */
- rtpmp4gpay->profile = g_strdup_printf ("%d", (gint) data[4]);
+ rtpmp4gpay->profile = g_strdup_printf ("%d", (gint) map.data[4]);
} else {
GST_ELEMENT_WARNING (rtpmp4gpay, STREAM, FORMAT,
(NULL), ("profile not found in config string, assuming \'1\'"));
GST_LOG_OBJECT (rtpmp4gpay, "profile %s", rtpmp4gpay->profile);
+ gst_buffer_unmap (buffer, &map);
+
return TRUE;
/* ERROR */
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
(NULL), ("config string too short"));
+ gst_buffer_unmap (buffer, &map);
return FALSE;
}
}
/* hmm, silly */
if (rtpmp4gpay->params) {
- res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
+ res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpmp4gpay),
"encoding-params", G_TYPE_STRING, rtpmp4gpay->params, MP4GCAPS);
} else {
- res = gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
+ res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpmp4gpay),
MP4GCAPS);
}
}
static gboolean
-gst_rtp_mp4g_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
+gst_rtp_mp4g_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
GstRtpMP4GPay *rtpmp4gpay;
GstStructure *structure;
if (media_type == NULL)
goto config_failed;
- gst_basertppayload_set_options (payload, media_type, TRUE, "MPEG4-GENERIC",
+ gst_rtp_base_payload_set_options (payload, media_type, TRUE, "MPEG4-GENERIC",
rtpmp4gpay->rate);
res = gst_rtp_mp4g_pay_new_caps (rtpmp4gpay);
guint avail, total;
GstBuffer *outbuf;
GstFlowReturn ret;
- gboolean fragmented;
guint mtu;
- fragmented = FALSE;
-
/* the data available in the adapter is either smaller
* than the MTU or bigger. In the case it is smaller, the complete
* adapter contents can be put in one packet. In the case the
total = avail = gst_adapter_available (rtpmp4gpay->adapter);
ret = GST_FLOW_OK;
- mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpmp4gpay);
+ mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpmp4gpay);
while (avail > 0) {
guint towrite;
guint8 *payload;
guint payload_len;
guint packet_len;
+ GstRTPBuffer rtp = { NULL };
+ GstBuffer *paybuf;
- /* this will be the total lenght of the packet */
+ /* this will be the total length of the packet */
packet_len = gst_rtp_buffer_calc_packet_len (avail, 0, 0);
/* fill one MTU or all available bytes, we need 4 spare bytes for
packet_len, payload_len);
/* create buffer to hold the payload, also make room for the 4 header bytes. */
- outbuf = gst_rtp_buffer_new_allocate (payload_len + 4, 0, 0);
+ outbuf =
+ gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD
+ (rtpmp4gpay), 4, 0, 0);
+ gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
/* copy payload */
- payload = gst_rtp_buffer_get_payload (outbuf);
+ payload = gst_rtp_buffer_get_payload (&rtp);
/* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+- .. -+-+-+-+-+-+-+-+-+-+
* |AU-headers-length|AU-header|AU-header| |AU-header|padding|
payload[2] = (total & 0x1fe0) >> 5;
payload[3] = (total & 0x1f) << 3; /* we use 13 bits for the size, 3 bits index */
- /* copy stuff from adapter to payload */
- gst_adapter_copy (rtpmp4gpay->adapter, &payload[4], 0, payload_len);
- gst_adapter_flush (rtpmp4gpay->adapter, payload_len);
-
/* marker only if the packet is complete */
- gst_rtp_buffer_set_marker (outbuf, avail <= payload_len);
+ gst_rtp_buffer_set_marker (&rtp, avail <= payload_len);
+
+ gst_rtp_buffer_unmap (&rtp);
- GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4gpay->first_timestamp;
+ paybuf = gst_adapter_take_buffer_fast (rtpmp4gpay->adapter, payload_len);
+ gst_rtp_copy_meta (GST_ELEMENT_CAST (rtpmp4gpay), outbuf, paybuf, 0);
+ outbuf = gst_buffer_append (outbuf, paybuf);
+
+ GST_BUFFER_PTS (outbuf) = rtpmp4gpay->first_timestamp;
GST_BUFFER_DURATION (outbuf) = rtpmp4gpay->first_duration;
- if (rtpmp4gpay->frame_len) {
- GST_BUFFER_OFFSET (outbuf) = rtpmp4gpay->offset;
- rtpmp4gpay->offset += rtpmp4gpay->frame_len;
+ GST_BUFFER_OFFSET (outbuf) = GST_BUFFER_OFFSET_NONE;
+
+ if (rtpmp4gpay->discont) {
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ /* Only the first outputted buffer has the DISCONT flag */
+ rtpmp4gpay->discont = FALSE;
}
- ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4gpay), outbuf);
+ ret = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (rtpmp4gpay), outbuf);
avail -= payload_len;
- fragmented = TRUE;
}
return ret;
/* we expect buffers as exactly one complete AU
*/
static GstFlowReturn
-gst_rtp_mp4g_pay_handle_buffer (GstBaseRTPPayload * basepayload,
+gst_rtp_mp4g_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRtpMP4GPay *rtpmp4gpay;
rtpmp4gpay = GST_RTP_MP4G_PAY (basepayload);
- rtpmp4gpay->first_timestamp = GST_BUFFER_TIMESTAMP (buffer);
+ rtpmp4gpay->first_timestamp = GST_BUFFER_PTS (buffer);
rtpmp4gpay->first_duration = GST_BUFFER_DURATION (buffer);
+ rtpmp4gpay->discont = GST_BUFFER_IS_DISCONT (buffer);
/* we always encode and flush a full AU */
gst_adapter_push (rtpmp4gpay->adapter, buffer);
return gst_rtp_mp4g_pay_flush (rtpmp4gpay);
}
+static gboolean
+gst_rtp_mp4g_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
+{
+ GstRtpMP4GPay *rtpmp4gpay;
+
+ rtpmp4gpay = GST_RTP_MP4G_PAY (payload);
+
+ GST_DEBUG ("Got event: %s", GST_EVENT_TYPE_NAME (event));
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_SEGMENT:
+ case GST_EVENT_EOS:
+ /* This flush call makes sure that the last buffer is always pushed
+ * to the base payloader */
+ gst_rtp_mp4g_pay_flush (rtpmp4gpay);
+ break;
+ case GST_EVENT_FLUSH_STOP:
+ gst_rtp_mp4g_pay_reset (rtpmp4gpay);
+ break;
+ default:
+ break;
+ }
+
+ /* let parent handle event too */
+ return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
+}
+
static GstStateChangeReturn
gst_rtp_mp4g_pay_change_state (GstElement * element, GstStateChange transition)
{
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
- gst_rtp_mp4g_pay_reset (rtpmp4gpay);
+ gst_rtp_mp4g_pay_cleanup (rtpmp4gpay);
break;
default:
break;
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+ switch (transition) {
+ case GST_STATE_CHANGE_PAUSED_TO_READY:
+ gst_rtp_mp4g_pay_cleanup (rtpmp4gpay);
+ break;
+ default:
+ break;
+ }
+
return ret;
}
gst_rtp_mp4g_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpmp4gpay",
- GST_RANK_NONE, GST_TYPE_RTP_MP4G_PAY);
+ GST_RANK_SECONDARY, GST_TYPE_RTP_MP4G_PAY);
}
projects / platform / upstream / gst-plugins-good.git / blobdiff