/* Farsight * Copyright (C) 2006 Marcel Moreaux * (C) 2008 Wim Taymans * * 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. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #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_details_simple (gstelement_class, "RTP DV Payloader", "Codec/Payloader/Network/RTP", "Payloads DV into RTP packets (RFC 3189)", "Marcel Moreaux , Wim Taymans "); 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, osize; guint8 *data, *odata; 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. */ data = gst_buffer_map (buffer, &size, NULL, GST_MAP_READ); odata = data; osize = size; GST_DEBUG_OBJECT (rtpdvpay, "DV RTP payloader got buffer of %u bytes, splitting in %u byte " "payload fragments, at time %" GST_TIME_FORMAT, size, max_payload_size, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (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_TIMESTAMP (outbuf) = GST_BUFFER_TIMESTAMP (buffer); gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp); 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; } } gst_buffer_unmap (buffer, odata, osize); 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); }