gstreamer/gst/asfmux/gstrtpasfpay.c
Thiago Santos 2641cd9d94 asfmux: Adds new plugin asfmux
Adds the brand new asfmux plugin, containing 3 elements:
asfmux, rtpasfpay and asfparse. This plugin was developed
as a GSoC 2009 project, with David Schleef as the mentor and
Thiago Santos as the student.
2009-07-24 14:52:28 -03:00

446 lines
15 KiB
C

/* ASF RTP Payloader plugin for GStreamer
* Copyright (C) 2009 Thiago Santos <thiagoss@embedded.ufcg.edu.br>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* 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.
*/
/* FIXME
* - this element doesn't follow (max/min) time properties,
* is it possible to do it with a container format?
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <gst/rtp/gstrtpbuffer.h>
#include <string.h>
#include "gstrtpasfpay.h"
GST_DEBUG_CATEGORY_STATIC (rtpasfpay_debug);
#define GST_CAT_DEFAULT (rtpasfpay_debug)
/* elementfactory information */
static const GstElementDetails gst_rtp_asf_pay_details =
GST_ELEMENT_DETAILS ("RTP ASF payloader",
"Codec/Payloader/Network",
"Payload-encodes ASF into RTP packets (MS_RTSP)",
"Thiago Santos <thiagoss@embedded.ufcg.edu.br>");
static GstStaticPadTemplate gst_rtp_asf_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/x-ms-asf, " "parsed = (boolean) true")
);
static GstStaticPadTemplate gst_rtp_asf_pay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) {\"audio\", \"video\", \"application\"}, "
"clock-rate = (int) 1000, " "encoding-name = (string) \"X-ASF-PF\"")
);
static GstFlowReturn
gst_rtp_asf_pay_handle_buffer (GstBaseRTPPayload * rtppay, GstBuffer * buffer);
static gboolean
gst_rtp_asf_pay_set_caps (GstBaseRTPPayload * rtppay, GstCaps * caps);
GST_BOILERPLATE (GstRtpAsfPay, gst_rtp_asf_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
static void
gst_rtp_asf_pay_init (GstRtpAsfPay * rtpasfpay, GstRtpAsfPayClass * klass)
{
rtpasfpay->first_ts = 0;
rtpasfpay->config = NULL;
rtpasfpay->packets_count = 0;
rtpasfpay->state = ASF_NOT_STARTED;
rtpasfpay->headers = NULL;
rtpasfpay->current = NULL;
}
static void
gst_rtp_asf_pay_finalize (GObject * object)
{
GstRtpAsfPay *rtpasfpay;
rtpasfpay = GST_RTP_ASF_PAY (object);
g_free (rtpasfpay->config);
if (rtpasfpay->headers)
gst_buffer_unref (rtpasfpay->headers);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_rtp_asf_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_asf_pay_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_asf_pay_src_template));
gst_element_class_set_details (element_class, &gst_rtp_asf_pay_details);
}
static void
gst_rtp_asf_pay_class_init (GstRtpAsfPayClass * klass)
{
GObjectClass *gobject_class;
GstBaseRTPPayloadClass *gstbasertppayload_class;
gobject_class = (GObjectClass *) klass;
gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
gobject_class->finalize = gst_rtp_asf_pay_finalize;
gstbasertppayload_class->handle_buffer = gst_rtp_asf_pay_handle_buffer;
gstbasertppayload_class->set_caps = gst_rtp_asf_pay_set_caps;
GST_DEBUG_CATEGORY_INIT (rtpasfpay_debug, "rtpasfpay", 0,
"ASF RTP Payloader");
}
static gboolean
gst_rtp_asf_pay_set_caps (GstBaseRTPPayload * rtppay, GstCaps * caps)
{
/* FIXME change application for the actual content */
gst_basertppayload_set_options (rtppay, "application", TRUE, "X-ASF-PF",
1000);
return TRUE;
}
static GstFlowReturn
gst_rtp_asf_pay_handle_packet (GstRtpAsfPay * rtpasfpay, GstBuffer * buffer)
{
GstBaseRTPPayload *rtppay;
GstAsfPacketInfo *packetinfo;
guint8 flags;
guint8 *data;
guint32 packet_util_size;
guint32 packet_offset;
guint32 size_left;
GstFlowReturn ret = GST_FLOW_OK;
rtppay = GST_BASE_RTP_PAYLOAD (rtpasfpay);
packetinfo = &rtpasfpay->packetinfo;
if (!gst_asf_parse_packet (buffer, packetinfo, TRUE)) {
GST_ERROR_OBJECT (rtpasfpay, "Error while parsing asf packet");
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
if (packetinfo->packet_size == 0)
packetinfo->packet_size = rtpasfpay->asfinfo.packet_size;
GST_LOG_OBJECT (rtpasfpay, "Packet size: %" G_GUINT32_FORMAT
", padding: %" G_GUINT32_FORMAT, packetinfo->packet_size,
packetinfo->padding);
/* FIXME - should update the padding field to 0 */
packet_util_size = packetinfo->packet_size - packetinfo->padding;
packet_offset = 0;
while (packet_util_size > 0) {
/* Even if we don't fill completely an output buffer we
* push it when we add an fragment. Because it seems that
* it is not possible to determine where a asf packet
* fragment ends inside a rtp packet payload.
* This flag tells us to push the packet.
*/
gboolean force_push = FALSE;
/* we have no output buffer pending, create one */
if (rtpasfpay->current == NULL) {
GST_LOG_OBJECT (rtpasfpay, "Creating new output buffer");
rtpasfpay->current =
gst_rtp_buffer_new_allocate_len (GST_BASE_RTP_PAYLOAD_MTU (rtpasfpay),
0, 0);
rtpasfpay->cur_off = gst_rtp_buffer_get_header_len (rtpasfpay->current);
rtpasfpay->has_ts = FALSE;
rtpasfpay->marker = FALSE;
}
data = GST_BUFFER_DATA (rtpasfpay->current) + rtpasfpay->cur_off;
size_left = GST_BUFFER_SIZE (rtpasfpay->current) - rtpasfpay->cur_off;
GST_DEBUG_OBJECT (rtpasfpay, "Input buffer bytes consumed: %"
G_GUINT32_FORMAT "/%" G_GUINT32_FORMAT, packet_offset,
GST_BUFFER_SIZE (buffer));
GST_DEBUG_OBJECT (rtpasfpay, "Output rtpbuffer status");
GST_DEBUG_OBJECT (rtpasfpay, "Current offset: %" G_GUINT32_FORMAT,
rtpasfpay->cur_off);
GST_DEBUG_OBJECT (rtpasfpay, "Size left: %" G_GUINT32_FORMAT, size_left);
GST_DEBUG_OBJECT (rtpasfpay, "Has ts: %s",
rtpasfpay->has_ts ? "yes" : "no");
if (rtpasfpay->has_ts) {
GST_DEBUG_OBJECT (rtpasfpay, "Ts: %" G_GUINT32_FORMAT, rtpasfpay->ts);
}
flags = 0;
if (packetinfo->has_keyframe) {
flags = flags | 0x80;
}
flags = flags | 0x20; /* Relative timestamp is present */
if (!rtpasfpay->has_ts) {
/* this is the first asf packet, its send time is the
* rtp packet timestamp */
rtpasfpay->has_ts = TRUE;
rtpasfpay->ts = packetinfo->send_time;
}
if (GST_BUFFER_SIZE (rtpasfpay->current) - rtpasfpay->cur_off >=
packet_util_size + 8) {
/* enough space for the rest of the packet */
if (packet_offset == 0) {
flags = flags | 0x40;
GST_WRITE_UINT24_BE (data + 1, packet_util_size);
} else {
GST_WRITE_UINT24_BE (data + 1, packet_offset);
force_push = TRUE;
}
data[0] = flags;
GST_WRITE_UINT32_BE (data + 4,
(gint32) (packetinfo->send_time) - (gint32) rtpasfpay->ts);
memcpy (data + 8, GST_BUFFER_DATA (buffer) + packet_offset,
packet_util_size);
/* updating status variables */
rtpasfpay->cur_off += 8 + packet_util_size;
size_left -= packet_util_size + 8;
packet_offset += packet_util_size;
packet_util_size = 0;
rtpasfpay->marker = TRUE;
} else {
/* fragment packet */
data[0] = flags;
GST_WRITE_UINT24_BE (data + 1, packet_offset);
GST_WRITE_UINT32_BE (data + 4,
(gint32) (packetinfo->send_time) - (gint32) rtpasfpay->ts);
memcpy (data + 8, GST_BUFFER_DATA (buffer) + packet_offset,
size_left - 8);
/* updating status variables */
rtpasfpay->cur_off += size_left;
packet_offset += size_left - 8;
packet_util_size -= size_left - 8;
size_left = 0;
force_push = TRUE;
}
/* there is not enough room for any more buffers */
if (force_push || size_left <= 8) {
if (size_left != 0) {
/* trim remaining bytes not used */
GstBuffer *aux = gst_buffer_create_sub (rtpasfpay->current, 0,
GST_BUFFER_SIZE (rtpasfpay->current) - size_left);
gst_buffer_unref (rtpasfpay->current);
rtpasfpay->current = aux;
}
gst_rtp_buffer_set_ssrc (rtpasfpay->current, rtppay->current_ssrc);
gst_rtp_buffer_set_marker (rtpasfpay->current, rtpasfpay->marker);
gst_rtp_buffer_set_payload_type (rtpasfpay->current,
GST_BASE_RTP_PAYLOAD_PT (rtppay));
gst_rtp_buffer_set_seq (rtpasfpay->current, rtppay->seqnum + 1);
gst_rtp_buffer_set_timestamp (rtpasfpay->current, packetinfo->send_time);
GST_BUFFER_TIMESTAMP (rtpasfpay->current) = GST_BUFFER_TIMESTAMP (buffer);
gst_buffer_set_caps (rtpasfpay->current,
GST_PAD_CAPS (GST_BASE_RTP_PAYLOAD_SRCPAD (rtppay)));
rtppay->seqnum++;
rtppay->timestamp = packetinfo->send_time;
GST_DEBUG_OBJECT (rtpasfpay, "Pushing rtp buffer");
ret =
gst_pad_push (GST_BASE_RTP_PAYLOAD_SRCPAD (rtppay),
rtpasfpay->current);
rtpasfpay->current = NULL;
if (ret != GST_FLOW_OK) {
gst_buffer_unref (buffer);
return ret;
}
}
}
gst_buffer_unref (buffer);
return ret;
}
static GstFlowReturn
gst_rtp_asf_pay_parse_headers (GstRtpAsfPay * rtpasfpay)
{
GstFlowReturn ret = GST_FLOW_OK;
gchar *maxps;
g_return_val_if_fail (rtpasfpay->headers, GST_FLOW_ERROR);
if (!gst_asf_parse_headers (rtpasfpay->headers, &rtpasfpay->asfinfo))
goto error;
GST_DEBUG_OBJECT (rtpasfpay, "Packets number: %" G_GUINT64_FORMAT,
rtpasfpay->asfinfo.packets_count);
GST_DEBUG_OBJECT (rtpasfpay, "Packets size: %" G_GUINT32_FORMAT,
rtpasfpay->asfinfo.packet_size);
GST_DEBUG_OBJECT (rtpasfpay, "Broadcast mode: %s",
rtpasfpay->asfinfo.broadcast ? "true" : "false");
/* get the config for caps */
g_free (rtpasfpay->config);
rtpasfpay->config = g_base64_encode (GST_BUFFER_DATA (rtpasfpay->headers),
GST_BUFFER_SIZE (rtpasfpay->headers));
GST_DEBUG_OBJECT (rtpasfpay, "Serialized headers to base64 string %s",
rtpasfpay->config);
g_assert (rtpasfpay->config != NULL);
GST_DEBUG_OBJECT (rtpasfpay, "Setting optional caps values: maxps=%"
G_GUINT32_FORMAT " and config=%s", rtpasfpay->asfinfo.packet_size,
rtpasfpay->config);
maxps =
g_strdup_printf ("%" G_GUINT32_FORMAT, rtpasfpay->asfinfo.packet_size);
gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpasfpay), "maxps",
G_TYPE_STRING, maxps, "config", G_TYPE_STRING, rtpasfpay->config, NULL);
g_free (maxps);
return GST_FLOW_OK;
error:
ret = GST_FLOW_ERROR;
GST_ERROR_OBJECT (rtpasfpay, "Error while parsing headers");
return GST_FLOW_ERROR;
}
static GstFlowReturn
gst_rtp_asf_pay_handle_buffer (GstBaseRTPPayload * rtppay, GstBuffer * buffer)
{
GstRtpAsfPay *rtpasfpay = GST_RTP_ASF_PAY_CAST (rtppay);
if (G_UNLIKELY (rtpasfpay->state == ASF_END)) {
GST_LOG_OBJECT (rtpasfpay,
"Dropping buffer as we already pushed all packets");
gst_buffer_unref (buffer);
return GST_FLOW_UNEXPECTED; /* we already finished our job */
}
/* receive headers
* we only accept if they are in a single buffer */
if (G_UNLIKELY (rtpasfpay->state == ASF_NOT_STARTED)) {
guint64 header_size;
if (GST_BUFFER_SIZE (buffer) < 24) { /* guid+object size size */
GST_ERROR_OBJECT (rtpasfpay,
"Buffer too small, smaller than a Guid and object size");
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
header_size = gst_asf_match_and_peek_obj_size (GST_BUFFER_DATA (buffer),
&(guids[ASF_HEADER_OBJECT_INDEX]));
if (header_size > 0) {
GST_DEBUG_OBJECT (rtpasfpay, "ASF header guid received, size %"
G_GUINT64_FORMAT, header_size);
if (GST_BUFFER_SIZE (buffer) < header_size) {
GST_ERROR_OBJECT (rtpasfpay, "Headers should be contained in a single"
" buffer");
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
} else {
rtpasfpay->state = ASF_DATA_OBJECT;
/* clear previous headers, if any */
if (rtpasfpay->headers) {
gst_buffer_unref (rtpasfpay->headers);
}
GST_DEBUG_OBJECT (rtpasfpay, "Storing headers");
if (GST_BUFFER_SIZE (buffer) == header_size) {
rtpasfpay->headers = buffer;
return GST_FLOW_OK;
} else {
/* headers are a subbuffer of thie buffer */
GstBuffer *aux = gst_buffer_create_sub (buffer, header_size,
GST_BUFFER_SIZE (buffer) - header_size);
rtpasfpay->headers = gst_buffer_create_sub (buffer, 0, header_size);
gst_buffer_replace (&buffer, aux);
}
}
} else {
GST_ERROR_OBJECT (rtpasfpay, "Missing ASF header start");
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
}
if (G_UNLIKELY (rtpasfpay->state == ASF_DATA_OBJECT)) {
if (GST_BUFFER_SIZE (buffer) != ASF_DATA_OBJECT_SIZE) {
GST_ERROR_OBJECT (rtpasfpay, "Received buffer of different size of "
"the data object header");
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
if (gst_asf_match_guid (GST_BUFFER_DATA (buffer),
&(guids[ASF_DATA_OBJECT_INDEX]))) {
GST_DEBUG_OBJECT (rtpasfpay, "Received data object header");
rtpasfpay->headers = gst_buffer_join (rtpasfpay->headers, buffer);
rtpasfpay->state = ASF_PACKETS;
return gst_rtp_asf_pay_parse_headers (rtpasfpay);
} else {
GST_ERROR_OBJECT (rtpasfpay, "Unexpected object received (was expecting "
"data object)");
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
}
if (G_LIKELY (rtpasfpay->state == ASF_PACKETS)) {
/* in broadcast mode we can't trust the packets count information
* from the headers
* We assume that if this is on broadcast mode it is a live stream
* and we are going to keep receiving packets indefinitely
*/
if (rtpasfpay->asfinfo.broadcast ||
rtpasfpay->packets_count < rtpasfpay->asfinfo.packets_count) {
GST_DEBUG_OBJECT (rtpasfpay, "Received packet %"
G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT,
rtpasfpay->packets_count, rtpasfpay->asfinfo.packets_count);
rtpasfpay->packets_count++;
return gst_rtp_asf_pay_handle_packet (rtpasfpay, buffer);
} else {
GST_INFO_OBJECT (rtpasfpay, "Packets ended");
rtpasfpay->state = ASF_END;
gst_buffer_unref (buffer);
return GST_FLOW_UNEXPECTED;
}
}
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
gboolean
gst_rtp_asf_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpasfpay",
GST_RANK_NONE, GST_TYPE_RTP_ASF_PAY);
}