gstreamer/gst/rtp/gstrtpdvpay.c
Sebastian Dröge dc059efa60 rtp: Use GST_BUFFER_PTS() instead of GST_BUFFER_TIMESTAMP()
The mix between all these in the RTP code is confusing, let's try to be
consistent.
2015-06-10 14:34:47 +02:00

395 lines
12 KiB
C

/* Farsight
* Copyright (C) 2006 Marcel Moreaux <marcelm@spacelabs.nl>
* (C) 2008 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
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpdvpay.h"
GST_DEBUG_CATEGORY (rtpdvpay_debug);
#define GST_CAT_DEFAULT (rtpdvpay_debug)
#define DEFAULT_MODE GST_DV_PAY_MODE_VIDEO
enum
{
PROP_0,
PROP_MODE
};
/* takes both system and non-system streams */
static GstStaticPadTemplate gst_rtp_dv_pay_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/x-dv")
);
static GstStaticPadTemplate gst_rtp_dv_pay_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) { \"video\", \"audio\" } ,"
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"encoding-name = (string) \"DV\", "
"clock-rate = (int) 90000,"
"encode = (string) { \"SD-VCR/525-60\", \"SD-VCR/625-50\", \"HD-VCR/1125-60\","
"\"HD-VCR/1250-50\", \"SDL-VCR/525-60\", \"SDL-VCR/625-50\","
"\"306M/525-60\", \"306M/625-50\", \"314M-25/525-60\","
"\"314M-25/625-50\", \"314M-50/525-60\", \"314M-50/625-50\" }"
/* optional parameters can't go in the template
* "audio = (string) { \"bundled\", \"none\" }"
*/
)
);
static gboolean gst_rtp_dv_pay_setcaps (GstRTPBasePayload * payload,
GstCaps * caps);
static GstFlowReturn gst_rtp_dv_pay_handle_buffer (GstRTPBasePayload * payload,
GstBuffer * buffer);
#define GST_TYPE_DV_PAY_MODE (gst_dv_pay_mode_get_type())
static GType
gst_dv_pay_mode_get_type (void)
{
static GType dv_pay_mode_type = 0;
static const GEnumValue dv_pay_modes[] = {
{GST_DV_PAY_MODE_VIDEO, "Video only", "video"},
{GST_DV_PAY_MODE_BUNDLED, "Video and Audio bundled", "bundled"},
{GST_DV_PAY_MODE_AUDIO, "Audio only", "audio"},
{0, NULL, NULL},
};
if (!dv_pay_mode_type) {
dv_pay_mode_type = g_enum_register_static ("GstDVPayMode", dv_pay_modes);
}
return dv_pay_mode_type;
}
static void gst_dv_pay_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_dv_pay_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
#define gst_rtp_dv_pay_parent_class parent_class
G_DEFINE_TYPE (GstRTPDVPay, gst_rtp_dv_pay, GST_TYPE_RTP_BASE_PAYLOAD);
static void
gst_rtp_dv_pay_class_init (GstRTPDVPayClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstRTPBasePayloadClass *gstrtpbasepayload_class;
GST_DEBUG_CATEGORY_INIT (rtpdvpay_debug, "rtpdvpay", 0, "DV RTP Payloader");
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;
gobject_class->set_property = gst_dv_pay_set_property;
gobject_class->get_property = gst_dv_pay_get_property;
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"The payload mode of payloading",
GST_TYPE_DV_PAY_MODE, DEFAULT_MODE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_dv_pay_sink_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_dv_pay_src_template));
gst_element_class_set_static_metadata (gstelement_class, "RTP DV Payloader",
"Codec/Payloader/Network/RTP",
"Payloads DV into RTP packets (RFC 3189)",
"Marcel Moreaux <marcelm@spacelabs.nl>, Wim Taymans <wim.taymans@gmail.com>");
gstrtpbasepayload_class->set_caps = gst_rtp_dv_pay_setcaps;
gstrtpbasepayload_class->handle_buffer = gst_rtp_dv_pay_handle_buffer;
}
static void
gst_rtp_dv_pay_init (GstRTPDVPay * rtpdvpay)
{
}
static void
gst_dv_pay_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstRTPDVPay *rtpdvpay = GST_RTP_DV_PAY (object);
switch (prop_id) {
case PROP_MODE:
rtpdvpay->mode = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_dv_pay_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstRTPDVPay *rtpdvpay = GST_RTP_DV_PAY (object);
switch (prop_id) {
case PROP_MODE:
g_value_set_enum (value, rtpdvpay->mode);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_rtp_dv_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
{
/* We don't do anything here, but we could check if it's a system stream and if
* it's not, default to sending the video only. We will negotiate downstream
* caps when we get to see the first frame. */
return TRUE;
}
static gboolean
gst_dv_pay_negotiate (GstRTPDVPay * rtpdvpay, guint8 * data, gsize size)
{
const gchar *encode, *media;
gboolean audio_bundled, res;
if ((data[3] & 0x80) == 0) { /* DSF flag */
/* it's an NTSC format */
if ((data[80 * 5 + 48 + 3] & 0x4) && (data[80 * 5 + 48] == 0x60)) { /* 4:2:2 sampling */
/* NTSC 50Mbps */
encode = "314M-25/525-60";
} else { /* 4:1:1 sampling */
/* NTSC 25Mbps */
encode = "SD-VCR/525-60";
}
} else {
/* it's a PAL format */
if ((data[80 * 5 + 48 + 3] & 0x4) && (data[80 * 5 + 48] == 0x60)) { /* 4:2:2 sampling */
/* PAL 50Mbps */
encode = "314M-50/625-50";
} else if ((data[5] & 0x07) == 0) { /* APT flag */
/* PAL 25Mbps 4:2:0 */
encode = "SD-VCR/625-50";
} else
/* PAL 25Mbps 4:1:1 */
encode = "314M-25/625-50";
}
media = "video";
audio_bundled = FALSE;
switch (rtpdvpay->mode) {
case GST_DV_PAY_MODE_AUDIO:
media = "audio";
break;
case GST_DV_PAY_MODE_BUNDLED:
audio_bundled = TRUE;
break;
default:
break;
}
gst_rtp_base_payload_set_options (GST_RTP_BASE_PAYLOAD (rtpdvpay), media,
TRUE, "DV", 90000);
if (audio_bundled) {
res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpdvpay),
"encode", G_TYPE_STRING, encode,
"audio", G_TYPE_STRING, "bundled", NULL);
} else {
res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpdvpay),
"encode", G_TYPE_STRING, encode, NULL);
}
return res;
}
static gboolean
include_dif (GstRTPDVPay * rtpdvpay, guint8 * data)
{
gint block_type;
gboolean res;
block_type = data[0] >> 5;
switch (block_type) {
case 0: /* Header block */
case 1: /* Subcode block */
case 2: /* VAUX block */
/* always include these blocks */
res = TRUE;
break;
case 3: /* Audio block */
/* never include audio if we are doing video only */
if (rtpdvpay->mode == GST_DV_PAY_MODE_VIDEO)
res = FALSE;
else
res = TRUE;
break;
case 4: /* Video block */
/* never include video if we are doing audio only */
if (rtpdvpay->mode == GST_DV_PAY_MODE_AUDIO)
res = FALSE;
else
res = TRUE;
break;
default: /* Something bogus, just ignore */
res = FALSE;
break;
}
return res;
}
/* Get a DV frame, chop it up in pieces, and push the pieces to the RTP layer.
*/
static GstFlowReturn
gst_rtp_dv_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRTPDVPay *rtpdvpay;
guint max_payload_size;
GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK;
gint hdrlen;
gsize size;
GstMapInfo map;
guint8 *data;
guint8 *dest;
guint filled;
GstRTPBuffer rtp = { NULL, };
rtpdvpay = GST_RTP_DV_PAY (basepayload);
hdrlen = gst_rtp_buffer_calc_header_len (0);
/* DV frames are made up from a bunch of DIF blocks. DIF blocks are 80 bytes
* each, and we should put an integral number of them in each RTP packet.
* Therefore, we round the available room down to the nearest multiple of 80.
*
* The available room is just the packet MTU, minus the RTP header length. */
max_payload_size = ((GST_RTP_BASE_PAYLOAD_MTU (rtpdvpay) - hdrlen) / 80) * 80;
/* The length of the buffer to transmit. */
if (!gst_buffer_map (buffer, &map, GST_MAP_READ)) {
GST_ELEMENT_ERROR (rtpdvpay, CORE, FAILED,
(NULL), ("Failed to map buffer"));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
}
data = map.data;
size = map.size;
GST_DEBUG_OBJECT (rtpdvpay,
"DV RTP payloader got buffer of %" G_GSIZE_FORMAT
" bytes, splitting in %u byte " "payload fragments, at time %"
GST_TIME_FORMAT, size, max_payload_size,
GST_TIME_ARGS (GST_BUFFER_PTS (buffer)));
if (!rtpdvpay->negotiated) {
gst_dv_pay_negotiate (rtpdvpay, data, size);
/* if we have not yet scanned the stream for its type, do so now */
rtpdvpay->negotiated = TRUE;
}
outbuf = NULL;
dest = NULL;
filled = 0;
/* while we have a complete DIF chunks left */
while (size >= 80) {
/* Allocate a new buffer, set the timestamp */
if (outbuf == NULL) {
outbuf = gst_rtp_buffer_new_allocate (max_payload_size, 0, 0);
GST_BUFFER_PTS (outbuf) = GST_BUFFER_PTS (buffer);
if (!gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp)) {
gst_buffer_unref (outbuf);
GST_ELEMENT_ERROR (rtpdvpay, CORE, FAILED,
(NULL), ("Failed to map RTP buffer"));
ret = GST_FLOW_ERROR;
goto beach;
}
dest = gst_rtp_buffer_get_payload (&rtp);
filled = 0;
}
/* inspect the DIF chunk, if we don't need to include it, skip to the next one. */
if (include_dif (rtpdvpay, data)) {
/* copy data in packet */
memcpy (dest, data, 80);
dest += 80;
filled += 80;
}
/* go to next dif chunk */
size -= 80;
data += 80;
/* push out the buffer if the next one would exceed the max packet size or
* when we are pushing the last packet */
if (filled + 80 > max_payload_size || size < 80) {
if (size < 160) {
guint hlen;
/* set marker */
gst_rtp_buffer_set_marker (&rtp, TRUE);
/* shrink buffer to last packet */
hlen = gst_rtp_buffer_get_header_len (&rtp);
gst_rtp_buffer_set_packet_len (&rtp, hlen + filled);
}
/* Push out the created piece, and check for errors. */
gst_rtp_buffer_unmap (&rtp);
ret = gst_rtp_base_payload_push (basepayload, outbuf);
if (ret != GST_FLOW_OK)
break;
outbuf = NULL;
}
}
beach:
gst_buffer_unmap (buffer, &map);
gst_buffer_unref (buffer);
return ret;
}
gboolean
gst_rtp_dv_pay_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rtpdvpay",
GST_RANK_SECONDARY, GST_TYPE_RTP_DV_PAY);
}