/* GStreamer
- * Copyright (C) <2005> Wim Taymans <wim@fluendo.com>
+ * Copyright (C) <2005> Wim Taymans <wim.taymans@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
GST_DEBUG_CATEGORY_STATIC (rtph263ppay_debug);
#define GST_CAT_DEFAULT rtph263ppay_debug
-/* elementfactory information */
-static const GstElementDetails gst_rtp_h263ppay_details =
-GST_ELEMENT_DETAILS ("RTP packet payloader",
- "Codec/Payloader/Network",
- "Payload-encodes H263+ video in RTP packets (RFC 2429)",
- "Wim Taymans <wim@fluendo.com>");
-
static GstStaticPadTemplate gst_rtp_h263p_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("video/x-h263, "
- "variant = (string) \"itu\", " "h263version = (string) \"h263p\"")
+ GST_STATIC_CAPS ("video/x-h263, " "variant = (string) \"itu\" ")
);
static GstStaticPadTemplate gst_rtp_h263p_pay_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
+ GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"video\", "
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
- "clock-rate = (int) 90000, " "encoding-name = (string) \"H263-1998\"")
+ "clock-rate = (int) 90000, " "encoding-name = (string) \"H263-1998\"; "
+ "application/x-rtp, "
+ "media = (string) \"video\", "
+ "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+ "clock-rate = (int) 90000, " "encoding-name = (string) \"H263-2000\"")
);
-static void gst_rtp_h263p_pay_class_init (GstRtpH263PPayClass * klass);
-static void gst_rtp_h263p_pay_base_init (GstRtpH263PPayClass * klass);
-static void gst_rtp_h263p_pay_init (GstRtpH263PPay * rtph263ppay);
static void gst_rtp_h263p_pay_finalize (GObject * object);
static void gst_rtp_h263p_pay_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
-
static void gst_rtp_h263p_pay_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstFlowReturn gst_rtp_h263p_pay_handle_buffer (GstBaseRTPPayload *
payload, GstBuffer * buffer);
-static GstBaseRTPPayloadClass *parent_class = NULL;
-
-static GType
-gst_rtp_h263p_pay_get_type (void)
-{
- static GType rtph263ppay_type = 0;
-
- if (!rtph263ppay_type) {
- static const GTypeInfo rtph263ppay_info = {
- sizeof (GstRtpH263PPayClass),
- (GBaseInitFunc) gst_rtp_h263p_pay_base_init,
- NULL,
- (GClassInitFunc) gst_rtp_h263p_pay_class_init,
- NULL,
- NULL,
- sizeof (GstRtpH263PPay),
- 0,
- (GInstanceInitFunc) gst_rtp_h263p_pay_init,
- };
-
- rtph263ppay_type =
- g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpH263PPay",
- &rtph263ppay_info, 0);
- }
- return rtph263ppay_type;
-}
+GST_BOILERPLATE (GstRtpH263PPay, gst_rtp_h263p_pay, GstBaseRTPPayload,
+ GST_TYPE_BASE_RTP_PAYLOAD);
static void
-gst_rtp_h263p_pay_base_init (GstRtpH263PPayClass * klass)
+gst_rtp_h263p_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_h263p_pay_sink_template));
- gst_element_class_set_details (element_class, &gst_rtp_h263ppay_details);
+ gst_element_class_set_details_simple (element_class, "RTP H263 payloader",
+ "Codec/Payloader/Network",
+ "Payload-encodes H263/+/++ video in RTP packets (RFC 4629)",
+ "Wim Taymans <wim.taymans@gmail.com>");
}
static void
gst_rtp_h263p_pay_class_init (GstRtpH263PPayClass * klass)
{
GObjectClass *gobject_class;
- GstElementClass *gstelement_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
- gstelement_class = (GstElementClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
- parent_class = g_type_class_peek_parent (klass);
-
gobject_class->finalize = gst_rtp_h263p_pay_finalize;
gobject_class->set_property = gst_rtp_h263p_pay_set_property;
gobject_class->get_property = gst_rtp_h263p_pay_get_property;
PROP_FRAGMENTATION_MODE, g_param_spec_enum ("fragmentation-mode",
"Fragmentation Mode",
"Packet Fragmentation Mode", GST_TYPE_FRAGMENTATION_MODE,
- DEFAULT_FRAGMENTATION_MODE, G_PARAM_READWRITE));
+ DEFAULT_FRAGMENTATION_MODE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
GST_DEBUG_CATEGORY_INIT (rtph263ppay_debug, "rtph263ppay",
- 0, "rtph263ppay (RFC 2429)");
-
+ 0, "rtph263ppay (RFC 4629)");
}
static void
-gst_rtp_h263p_pay_init (GstRtpH263PPay * rtph263ppay)
+gst_rtp_h263p_pay_init (GstRtpH263PPay * rtph263ppay,
+ GstRtpH263PPayClass * klass)
{
rtph263ppay->adapter = gst_adapter_new ();
static gboolean
gst_rtp_h263p_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
+ gboolean res;
+
gst_basertppayload_set_options (payload, "video", TRUE, "H263-1998", 90000);
- gst_basertppayload_set_outcaps (payload, NULL);
+ res = gst_basertppayload_set_outcaps (payload, NULL);
- return TRUE;
+ return res;
}
static void
}
}
-
-
static GstFlowReturn
gst_rtp_h263p_pay_flush (GstRtpH263PPay * rtph263ppay)
{
return GST_FLOW_OK;
fragmented = FALSE;
- /* This algorithm assumes the H263+ encoder sends complete frames in each
+ /* This algorithm assumes the H263/+/++ encoder sends complete frames in each
* buffer */
/* With Fragmentation Mode at GST_FRAGMENTATION_MODE_NORMAL:
* This algorithm implements the Follow-on packets method for packetization.
* This assumes low packet loss network.
* With Fragmentation Mode at GST_FRAGMENTATION_MODE_SYNC:
* This algorithm separates large frames at synchronisation points (Segments)
- * (See RFC 2429 section 6). It would be interesting to have a property such as network
+ * (See RFC 4629 section 6). It would be interesting to have a property such as network
* quality to select between both packetization methods */
- /* TODO Add VRC supprt (See RFC 2429 section 4.2) */
+ /* TODO Add VRC supprt (See RFC 4629 section 5.2) */
while (avail > 0) {
guint towrite;
guint8 *payload;
- guint8 *data;
guint payload_len;
gint header_len;
guint next_gop = 0;
parse_data = gst_adapter_peek (rtph263ppay->adapter, avail);
/* Check if we have a gob or eos , eossbs */
+ /* FIXME EOS and EOSSBS packets should never contain any gobs and vice-versa */
if (avail >= 3 && *parse_data == 0 && *(parse_data + 1) == 0
&& *(parse_data + 2) >= 0x80) {
GST_DEBUG_OBJECT (rtph263ppay, " Found GOB header");
found_gob = TRUE;
}
/* Find next and cut the packet accordingly */
+ /* TODO we should get as many gobs as possible until MTU is reached, this
+ * code seems to just get one GOB per packet */
while (parsed_len + 2 < avail) {
if (parse_data[parsed_len] == 0 && parse_data[parsed_len + 1] == 0
&& parse_data[parsed_len + 2] >= 0x80) {
payload = gst_rtp_buffer_get_payload (outbuf);
- data = (guint8 *) gst_adapter_peek (rtph263ppay->adapter, towrite);
- memcpy (&payload[header_len], data, towrite);
+ gst_adapter_copy (rtph263ppay->adapter, &payload[header_len], 0, towrite);
/* 0 1
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5
payload[0] = (fragmented && !found_gob) ? 0x00 : 0x04;
payload[1] = 0;
- GST_BUFFER_TIMESTAMP (outbuf) = rtph263ppay->first_ts;
+ GST_BUFFER_TIMESTAMP (outbuf) = rtph263ppay->first_timestamp;
+ GST_BUFFER_DURATION (outbuf) = rtph263ppay->first_duration;
+
gst_adapter_flush (rtph263ppay->adapter, towrite);
ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtph263ppay), outbuf);
{
GstRtpH263PPay *rtph263ppay;
GstFlowReturn ret;
- guint size;
rtph263ppay = GST_RTP_H263P_PAY (payload);
- size = GST_BUFFER_SIZE (buffer);
- rtph263ppay->first_ts = GST_BUFFER_TIMESTAMP (buffer);
+ rtph263ppay->first_timestamp = GST_BUFFER_TIMESTAMP (buffer);
+ rtph263ppay->first_duration = GST_BUFFER_DURATION (buffer);
/* we always encode and flush a full picture */
gst_adapter_push (rtph263ppay->adapter, buffer);