/* GStreamer * Copyright (C) <2005> 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., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include #include #include #include "gstrtpelements.h" #include "gstrtpmp4vpay.h" #include "gstrtputils.h" GST_DEBUG_CATEGORY_STATIC (rtpmp4vpay_debug); #define GST_CAT_DEFAULT (rtpmp4vpay_debug) static GstStaticPadTemplate gst_rtp_mp4v_pay_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("video/mpeg," "mpegversion=(int) 4, systemstream=(boolean)false;" "video/x-divx") ); static GstStaticPadTemplate gst_rtp_mp4v_pay_src_template = 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) [1, MAX ], " "encoding-name = (string) \"MP4V-ES\"" /* two string params * "profile-level-id = (string) [1,MAX]" "config = (string) [1,MAX]" */ ) ); #define DEFAULT_CONFIG_INTERVAL 0 enum { PROP_0, PROP_CONFIG_INTERVAL }; static void gst_rtp_mp4v_pay_finalize (GObject * object); static void gst_rtp_mp4v_pay_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtp_mp4v_pay_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static gboolean gst_rtp_mp4v_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps); static GstFlowReturn gst_rtp_mp4v_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer); static gboolean gst_rtp_mp4v_pay_sink_event (GstRTPBasePayload * pay, GstEvent * event); #define gst_rtp_mp4v_pay_parent_class parent_class G_DEFINE_TYPE (GstRtpMP4VPay, gst_rtp_mp4v_pay, GST_TYPE_RTP_BASE_PAYLOAD); /* Note: This element is marked at a "+1" rank to make sure that * auto-plugging of payloaders for MPEG4 elementary streams don't * end up using the 'rtpmp4gpay' element (generic mpeg4) which isn't * as well supported as this RFC */ GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpmp4vpay, "rtpmp4vpay", GST_RANK_SECONDARY + 1, GST_TYPE_RTP_MP4V_PAY, rtp_element_init (plugin)); static void gst_rtp_mp4v_pay_class_init (GstRtpMP4VPayClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstRTPBasePayloadClass *gstrtpbasepayload_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass; gobject_class->set_property = gst_rtp_mp4v_pay_set_property; gobject_class->get_property = gst_rtp_mp4v_pay_get_property; gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_mp4v_pay_src_template); gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_mp4v_pay_sink_template); gst_element_class_set_static_metadata (gstelement_class, "RTP MPEG4 Video payloader", "Codec/Payloader/Network/RTP", "Payload MPEG-4 video as RTP packets (RFC 3016)", "Wim Taymans "); g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_CONFIG_INTERVAL, g_param_spec_int ("config-interval", "Config Send Interval", "Send Config Insertion Interval in seconds (configuration headers " "will be multiplexed in the data stream when detected.) " "(0 = disabled, -1 = send with every IDR frame)", -1, 3600, DEFAULT_CONFIG_INTERVAL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS) ); gobject_class->finalize = gst_rtp_mp4v_pay_finalize; gstrtpbasepayload_class->set_caps = gst_rtp_mp4v_pay_setcaps; gstrtpbasepayload_class->handle_buffer = gst_rtp_mp4v_pay_handle_buffer; gstrtpbasepayload_class->sink_event = gst_rtp_mp4v_pay_sink_event; GST_DEBUG_CATEGORY_INIT (rtpmp4vpay_debug, "rtpmp4vpay", 0, "MP4 video RTP Payloader"); } static void gst_rtp_mp4v_pay_init (GstRtpMP4VPay * rtpmp4vpay) { rtpmp4vpay->adapter = gst_adapter_new (); rtpmp4vpay->rate = 90000; rtpmp4vpay->profile = 1; rtpmp4vpay->need_config = TRUE; rtpmp4vpay->config_interval = DEFAULT_CONFIG_INTERVAL; rtpmp4vpay->last_config = -1; rtpmp4vpay->config = NULL; } static void gst_rtp_mp4v_pay_finalize (GObject * object) { GstRtpMP4VPay *rtpmp4vpay; rtpmp4vpay = GST_RTP_MP4V_PAY (object); if (rtpmp4vpay->config) { gst_buffer_unref (rtpmp4vpay->config); rtpmp4vpay->config = NULL; } g_object_unref (rtpmp4vpay->adapter); rtpmp4vpay->adapter = NULL; G_OBJECT_CLASS (parent_class)->finalize (object); } static gboolean gst_rtp_mp4v_pay_new_caps (GstRtpMP4VPay * rtpmp4vpay) { gchar *profile, *config; GValue v = { 0 }; gboolean res; profile = g_strdup_printf ("%d", rtpmp4vpay->profile); g_value_init (&v, GST_TYPE_BUFFER); gst_value_set_buffer (&v, rtpmp4vpay->config); config = gst_value_serialize (&v); res = gst_rtp_base_payload_set_outcaps (GST_RTP_BASE_PAYLOAD (rtpmp4vpay), "profile-level-id", G_TYPE_STRING, profile, "config", G_TYPE_STRING, config, NULL); g_value_unset (&v); g_free (profile); g_free (config); return res; } static gboolean gst_rtp_mp4v_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps) { GstRtpMP4VPay *rtpmp4vpay; GstStructure *structure; const GValue *codec_data; gboolean res; rtpmp4vpay = GST_RTP_MP4V_PAY (payload); gst_rtp_base_payload_set_options (payload, "video", TRUE, "MP4V-ES", rtpmp4vpay->rate); res = TRUE; structure = gst_caps_get_structure (caps, 0); codec_data = gst_structure_get_value (structure, "codec_data"); if (codec_data) { GST_LOG_OBJECT (rtpmp4vpay, "got codec_data"); if (G_VALUE_TYPE (codec_data) == GST_TYPE_BUFFER) { GstBuffer *buffer; buffer = gst_value_get_buffer (codec_data); if (gst_buffer_get_size (buffer) < 5) goto done; gst_buffer_extract (buffer, 4, &rtpmp4vpay->profile, 1); GST_LOG_OBJECT (rtpmp4vpay, "configuring codec_data, profile %d", rtpmp4vpay->profile); if (rtpmp4vpay->config) gst_buffer_unref (rtpmp4vpay->config); rtpmp4vpay->config = gst_buffer_copy (buffer); res = gst_rtp_mp4v_pay_new_caps (rtpmp4vpay); } } done: return res; } static void gst_rtp_mp4v_pay_empty (GstRtpMP4VPay * rtpmp4vpay) { gst_adapter_clear (rtpmp4vpay->adapter); } #define RTP_HEADER_LEN 12 static GstFlowReturn gst_rtp_mp4v_pay_flush (GstRtpMP4VPay * rtpmp4vpay) { guint avail, mtu; GstBuffer *outbuf; GstBuffer *outbuf_data = NULL; GstFlowReturn ret; GstBufferList *list = NULL; /* 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 * adapter has more than one MTU, we need to split the MP4V data * over multiple packets. */ avail = gst_adapter_available (rtpmp4vpay->adapter); if (rtpmp4vpay->config == NULL && rtpmp4vpay->need_config) { /* when we don't have a config yet, flush things out */ gst_adapter_flush (rtpmp4vpay->adapter, avail); avail = 0; } if (!avail) return GST_FLOW_OK; mtu = GST_RTP_BASE_PAYLOAD_MTU (rtpmp4vpay); /* Use buffer lists. Each frame will be put into a list * of buffers and the whole list will be pushed downstream * at once */ list = gst_buffer_list_new_sized ((avail / (mtu - RTP_HEADER_LEN)) + 1); while (avail > 0) { guint towrite; guint payload_len; guint packet_len; GstRTPBuffer rtp = { NULL }; /* 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 */ towrite = MIN (packet_len, mtu); /* this is the payload length */ payload_len = gst_rtp_buffer_calc_payload_len (towrite, 0, 0); /* create buffer without payload. The payload will be put * in next buffer instead. Both buffers will be merged */ outbuf = gst_rtp_base_payload_allocate_output_buffer (GST_RTP_BASE_PAYLOAD (rtpmp4vpay), 0, 0, 0); /* Take buffer with the payload from the adapter */ outbuf_data = gst_adapter_take_buffer_fast (rtpmp4vpay->adapter, payload_len); avail -= payload_len; gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp); gst_rtp_buffer_set_marker (&rtp, avail == 0); if (avail == 0) GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_MARKER); gst_rtp_buffer_unmap (&rtp); gst_rtp_copy_video_meta (rtpmp4vpay, outbuf, outbuf_data); outbuf = gst_buffer_append (outbuf, outbuf_data); GST_BUFFER_PTS (outbuf) = rtpmp4vpay->first_timestamp; /* add to list */ gst_buffer_list_insert (list, -1, outbuf); } /* push the whole buffer list at once */ ret = gst_rtp_base_payload_push_list (GST_RTP_BASE_PAYLOAD (rtpmp4vpay), list); return ret; } #define VOS_STARTCODE 0x000001B0 #define VOS_ENDCODE 0x000001B1 #define USER_DATA_STARTCODE 0x000001B2 #define GOP_STARTCODE 0x000001B3 #define VISUAL_OBJECT_STARTCODE 0x000001B5 #define VOP_STARTCODE 0x000001B6 static gboolean gst_rtp_mp4v_pay_depay_data (GstRtpMP4VPay * enc, guint8 * data, guint size, gint * strip, gboolean * vopi) { guint32 code; gboolean result; *vopi = FALSE; *strip = 0; if (size < 5) return FALSE; code = GST_READ_UINT32_BE (data); GST_DEBUG_OBJECT (enc, "start code 0x%08x", code); switch (code) { case VOS_STARTCODE: case 0x00000101: { gint i; guint8 profile; gboolean newprofile = FALSE; gboolean equal; if (code == VOS_STARTCODE) { /* profile_and_level_indication */ profile = data[4]; GST_DEBUG_OBJECT (enc, "VOS profile 0x%08x", profile); if (profile != enc->profile) { newprofile = TRUE; enc->profile = profile; } } /* up to the next GOP_STARTCODE or VOP_STARTCODE is * the config information */ code = 0xffffffff; for (i = 5; i < size - 4; i++) { code = (code << 8) | data[i]; if (code == GOP_STARTCODE || code == VOP_STARTCODE) break; } i -= 3; /* see if config changed */ equal = FALSE; if (enc->config) { if (gst_buffer_get_size (enc->config) == i) { equal = gst_buffer_memcmp (enc->config, 0, data, i) == 0; } } /* if config string changed or new profile, make new caps */ if (!equal || newprofile) { if (enc->config) gst_buffer_unref (enc->config); enc->config = gst_buffer_new_and_alloc (i); gst_buffer_fill (enc->config, 0, data, i); gst_rtp_mp4v_pay_new_caps (enc); } *strip = i; /* we need to flush out the current packet. */ result = TRUE; break; } case VOP_STARTCODE: GST_DEBUG_OBJECT (enc, "VOP"); /* VOP startcode, we don't have to flush the packet */ result = FALSE; /* vop-coding-type == I-frame */ if (size > 4 && (data[4] >> 6 == 0)) { GST_DEBUG_OBJECT (enc, "VOP-I"); *vopi = TRUE; } break; case GOP_STARTCODE: GST_DEBUG_OBJECT (enc, "GOP"); *vopi = TRUE; result = TRUE; break; case 0x00000100: enc->need_config = FALSE; result = TRUE; break; default: if (code >= 0x20 && code <= 0x2f) { GST_DEBUG_OBJECT (enc, "short header"); result = FALSE; } else { GST_DEBUG_OBJECT (enc, "other startcode"); /* all other startcodes need a flush */ result = TRUE; } break; } return result; } /* we expect buffers starting on startcodes. */ static GstFlowReturn gst_rtp_mp4v_pay_handle_buffer (GstRTPBasePayload * basepayload, GstBuffer * buffer) { GstRtpMP4VPay *rtpmp4vpay; GstFlowReturn ret; guint avail; guint packet_len; GstMapInfo map; gsize size; gboolean flush; gint strip; GstClockTime timestamp, duration; gboolean vopi; gboolean send_config; GstClockTime running_time = GST_CLOCK_TIME_NONE; ret = GST_FLOW_OK; send_config = FALSE; rtpmp4vpay = GST_RTP_MP4V_PAY (basepayload); gst_buffer_map (buffer, &map, GST_MAP_READ); size = map.size; timestamp = GST_BUFFER_PTS (buffer); duration = GST_BUFFER_DURATION (buffer); avail = gst_adapter_available (rtpmp4vpay->adapter); if (duration == -1) duration = 0; /* empty buffer, take timestamp */ if (avail == 0) { rtpmp4vpay->first_timestamp = timestamp; rtpmp4vpay->duration = 0; } /* depay incoming data and see if we need to start a new RTP * packet */ flush = gst_rtp_mp4v_pay_depay_data (rtpmp4vpay, map.data, size, &strip, &vopi); gst_buffer_unmap (buffer, &map); if (strip) { /* strip off config if requested, do not strip off if the * config_interval is set to -1 */ if (!(rtpmp4vpay->config_interval > 0) && !(rtpmp4vpay->config_interval == -1)) { GstBuffer *subbuf; GST_LOG_OBJECT (rtpmp4vpay, "stripping config at %d, size %d", strip, (gint) size - strip); /* strip off header */ subbuf = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, strip, size - strip); GST_BUFFER_PTS (subbuf) = timestamp; gst_buffer_unref (buffer); buffer = subbuf; size = gst_buffer_get_size (buffer); } else { running_time = gst_segment_to_running_time (&basepayload->segment, GST_FORMAT_TIME, timestamp); GST_LOG_OBJECT (rtpmp4vpay, "found config in stream"); rtpmp4vpay->last_config = running_time; } } /* there is a config request, see if we need to insert it */ if (vopi && (rtpmp4vpay->config_interval > 0) && rtpmp4vpay->config) { running_time = gst_segment_to_running_time (&basepayload->segment, GST_FORMAT_TIME, timestamp); if (rtpmp4vpay->last_config != -1) { guint64 diff; GST_LOG_OBJECT (rtpmp4vpay, "now %" GST_TIME_FORMAT ", last VOP-I %" GST_TIME_FORMAT, GST_TIME_ARGS (running_time), GST_TIME_ARGS (rtpmp4vpay->last_config)); /* calculate diff between last config in milliseconds */ if (running_time > rtpmp4vpay->last_config) { diff = running_time - rtpmp4vpay->last_config; } else { diff = 0; } GST_DEBUG_OBJECT (rtpmp4vpay, "interval since last config %" GST_TIME_FORMAT, GST_TIME_ARGS (diff)); /* bigger than interval, queue config */ if (GST_TIME_AS_SECONDS (diff) >= rtpmp4vpay->config_interval) { GST_DEBUG_OBJECT (rtpmp4vpay, "time to send config"); send_config = TRUE; } } else { /* no known previous config time, send now */ GST_DEBUG_OBJECT (rtpmp4vpay, "no previous config time, send now"); send_config = TRUE; } } if (vopi && (rtpmp4vpay->config_interval == -1)) { GST_DEBUG_OBJECT (rtpmp4vpay, "sending config before current IDR frame"); /* send config before every IDR frame */ send_config = TRUE; } if (send_config) { /* we need to send config now first */ GST_LOG_OBJECT (rtpmp4vpay, "inserting config in stream"); /* insert header */ buffer = gst_buffer_append (gst_buffer_ref (rtpmp4vpay->config), buffer); GST_BUFFER_PTS (buffer) = timestamp; size = gst_buffer_get_size (buffer); if (running_time != -1) { rtpmp4vpay->last_config = running_time; } } /* if we need to flush, do so now */ if (flush) { ret = gst_rtp_mp4v_pay_flush (rtpmp4vpay); rtpmp4vpay->first_timestamp = timestamp; rtpmp4vpay->duration = 0; avail = 0; } /* get packet length of data and see if we exceeded MTU. */ packet_len = gst_rtp_buffer_calc_packet_len (avail + size, 0, 0); if (gst_rtp_base_payload_is_filled (basepayload, packet_len, rtpmp4vpay->duration + duration)) { ret = gst_rtp_mp4v_pay_flush (rtpmp4vpay); rtpmp4vpay->first_timestamp = timestamp; rtpmp4vpay->duration = 0; } /* push new data */ gst_adapter_push (rtpmp4vpay->adapter, buffer); rtpmp4vpay->duration += duration; return ret; } static gboolean gst_rtp_mp4v_pay_sink_event (GstRTPBasePayload * pay, GstEvent * event) { GstRtpMP4VPay *rtpmp4vpay; rtpmp4vpay = GST_RTP_MP4V_PAY (pay); 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_mp4v_pay_flush (rtpmp4vpay); break; case GST_EVENT_FLUSH_STOP: gst_rtp_mp4v_pay_empty (rtpmp4vpay); break; default: break; } /* let parent handle event too */ return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (pay, event); } static void gst_rtp_mp4v_pay_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstRtpMP4VPay *rtpmp4vpay; rtpmp4vpay = GST_RTP_MP4V_PAY (object); switch (prop_id) { case PROP_CONFIG_INTERVAL: rtpmp4vpay->config_interval = g_value_get_int (value); break; default: break; } } static void gst_rtp_mp4v_pay_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstRtpMP4VPay *rtpmp4vpay; rtpmp4vpay = GST_RTP_MP4V_PAY (object); switch (prop_id) { case PROP_CONFIG_INTERVAL: g_value_set_int (value, rtpmp4vpay->config_interval); break; default: break; } }