/* RTP Retransmission queue element for GStreamer * * gstrtprtxqueue.c: * * Copyright (C) 2013 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. */ /** * SECTION:element-rtprtxqueue * @title: rtprtxqueue * * rtprtxqueue maintains a queue of transmitted RTP packets, up to a * configurable limit (see #GstRTPRtxQueue:max-size-time, * #GstRTPRtxQueue:max-size-packets), and retransmits them upon request * from the downstream rtpsession (GstRTPRetransmissionRequest event). * * This element is similar to rtprtxsend, but it has differences: * - Retransmission from rtprtxqueue is not RFC 4588 compliant. The * retransmitted packets have the same ssrc and payload type as the original * stream. * - As a side-effect of the above, rtprtxqueue does not require the use of * rtprtxreceive on the receiving end. rtpjitterbuffer alone is able to * reconstruct the stream. * - Retransmission from rtprtxqueue happens as soon as the next regular flow * packet is chained, while rtprtxsend retransmits as soon as the retransmission * event is received, using a helper thread. * - rtprtxqueue can be used with rtpbin without the need of hooking to its * #GstRtpBin::request-aux-sender signal, which means it can be used with * rtpbin using gst-launch. * * See also #GstRtpRtxSend, #GstRtpRtxReceive * * # Example pipelines * * |[ * gst-launch-1.0 rtpbin name=b rtp-profile=avpf \ * audiotestsrc is-live=true ! opusenc ! rtpopuspay pt=96 ! rtprtxqueue ! b.send_rtp_sink_0 \ * b.send_rtp_src_0 ! identity drop-probability=0.01 ! udpsink host="127.0.0.1" port=5000 \ * udpsrc port=5001 ! b.recv_rtcp_sink_0 \ * b.send_rtcp_src_0 ! udpsink host="127.0.0.1" port=5002 sync=false async=false * ]| * Sender pipeline * * |[ * gst-launch-1.0 rtpbin name=b rtp-profile=avpf do-retransmission=true \ * udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)96" ! \ * b.recv_rtp_sink_0 \ * b. ! rtpopusdepay ! opusdec ! audioconvert ! audioresample ! autoaudiosink \ * udpsrc port=5002 ! b.recv_rtcp_sink_0 \ * b.send_rtcp_src_0 ! udpsink host="127.0.0.1" port=5001 sync=false async=false * ]| * Receiver pipeline */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include "gstrtprtxqueue.h" GST_DEBUG_CATEGORY_STATIC (gst_rtp_rtx_queue_debug); #define GST_CAT_DEFAULT gst_rtp_rtx_queue_debug #define DEFAULT_MAX_SIZE_TIME 0 #define DEFAULT_MAX_SIZE_PACKETS 100 enum { PROP_0, PROP_MAX_SIZE_TIME, PROP_MAX_SIZE_PACKETS, PROP_REQUESTS, PROP_FULFILLED_REQUESTS, }; static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp") ); static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp") ); static gboolean gst_rtp_rtx_queue_src_event (GstPad * pad, GstObject * parent, GstEvent * event); static gboolean gst_rtp_rtx_queue_sink_event (GstPad * pad, GstObject * parent, GstEvent * event); static GstFlowReturn gst_rtp_rtx_queue_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer); static GstFlowReturn gst_rtp_rtx_queue_chain_list (GstPad * pad, GstObject * parent, GstBufferList * list); static GstStateChangeReturn gst_rtp_rtx_queue_change_state (GstElement * element, GstStateChange transition); static void gst_rtp_rtx_queue_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtp_rtx_queue_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static void gst_rtp_rtx_queue_finalize (GObject * object); G_DEFINE_TYPE (GstRTPRtxQueue, gst_rtp_rtx_queue, GST_TYPE_ELEMENT); static void gst_rtp_rtx_queue_class_init (GstRTPRtxQueueClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gobject_class->get_property = gst_rtp_rtx_queue_get_property; gobject_class->set_property = gst_rtp_rtx_queue_set_property; gobject_class->finalize = gst_rtp_rtx_queue_finalize; g_object_class_install_property (gobject_class, PROP_MAX_SIZE_TIME, g_param_spec_uint ("max-size-time", "Max Size Times", "Amount of ms to queue (0 = unlimited)", 0, G_MAXUINT, DEFAULT_MAX_SIZE_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_MAX_SIZE_PACKETS, g_param_spec_uint ("max-size-packets", "Max Size Packets", "Amount of packets to queue (0 = unlimited)", 0, G_MAXUINT, DEFAULT_MAX_SIZE_PACKETS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_REQUESTS, g_param_spec_uint ("requests", "Requests", "Total number of retransmission requests", 0, G_MAXUINT, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_FULFILLED_REQUESTS, g_param_spec_uint ("fulfilled-requests", "Fulfilled Requests", "Number of fulfilled retransmission requests", 0, G_MAXUINT, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); gst_element_class_add_static_pad_template (gstelement_class, &src_factory); gst_element_class_add_static_pad_template (gstelement_class, &sink_factory); gst_element_class_set_static_metadata (gstelement_class, "RTP Retransmission Queue", "Codec", "Keep RTP packets in a queue for retransmission", "Wim Taymans "); gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_change_state); } static void gst_rtp_rtx_queue_reset (GstRTPRtxQueue * rtx, gboolean full) { g_mutex_lock (&rtx->lock); g_queue_foreach (rtx->queue, (GFunc) gst_buffer_unref, NULL); g_queue_clear (rtx->queue); g_list_foreach (rtx->pending, (GFunc) gst_buffer_unref, NULL); g_list_free (rtx->pending); rtx->pending = NULL; rtx->n_requests = 0; rtx->n_fulfilled_requests = 0; g_mutex_unlock (&rtx->lock); } static void gst_rtp_rtx_queue_finalize (GObject * object) { GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (object); gst_rtp_rtx_queue_reset (rtx, TRUE); g_queue_free (rtx->queue); g_mutex_clear (&rtx->lock); G_OBJECT_CLASS (gst_rtp_rtx_queue_parent_class)->finalize (object); } static void gst_rtp_rtx_queue_init (GstRTPRtxQueue * rtx) { GstElementClass *klass = GST_ELEMENT_GET_CLASS (rtx); rtx->srcpad = gst_pad_new_from_template (gst_element_class_get_pad_template (klass, "src"), "src"); GST_PAD_SET_PROXY_CAPS (rtx->srcpad); GST_PAD_SET_PROXY_ALLOCATION (rtx->srcpad); gst_pad_set_event_function (rtx->srcpad, GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_src_event)); gst_element_add_pad (GST_ELEMENT (rtx), rtx->srcpad); rtx->sinkpad = gst_pad_new_from_template (gst_element_class_get_pad_template (klass, "sink"), "sink"); GST_PAD_SET_PROXY_CAPS (rtx->sinkpad); GST_PAD_SET_PROXY_ALLOCATION (rtx->sinkpad); gst_pad_set_event_function (rtx->sinkpad, GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_sink_event)); gst_pad_set_chain_function (rtx->sinkpad, GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_chain)); gst_pad_set_chain_list_function (rtx->sinkpad, GST_DEBUG_FUNCPTR (gst_rtp_rtx_queue_chain_list)); gst_element_add_pad (GST_ELEMENT (rtx), rtx->sinkpad); rtx->queue = g_queue_new (); g_mutex_init (&rtx->lock); rtx->max_size_time = DEFAULT_MAX_SIZE_TIME; rtx->max_size_packets = DEFAULT_MAX_SIZE_PACKETS; } typedef struct { GstRTPRtxQueue *rtx; guint seqnum; gboolean found; } RTXData; static void push_seqnum (GstBuffer * buffer, RTXData * data) { GstRTPRtxQueue *rtx = data->rtx; GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT; guint16 seqnum; if (data->found) return; if (!GST_IS_BUFFER (buffer) || !gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpbuffer)) return; seqnum = gst_rtp_buffer_get_seq (&rtpbuffer); gst_rtp_buffer_unmap (&rtpbuffer); if (seqnum == data->seqnum) { data->found = TRUE; GST_DEBUG_OBJECT (rtx, "found %d", seqnum); rtx->pending = g_list_prepend (rtx->pending, gst_buffer_ref (buffer)); } } static gboolean gst_rtp_rtx_queue_src_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (parent); gboolean res; switch (GST_EVENT_TYPE (event)) { case GST_EVENT_CUSTOM_UPSTREAM: { const GstStructure *s; s = gst_event_get_structure (event); if (gst_structure_has_name (s, "GstRTPRetransmissionRequest")) { guint seqnum; RTXData data; if (!gst_structure_get_uint (s, "seqnum", &seqnum)) seqnum = -1; GST_DEBUG_OBJECT (rtx, "request %d", seqnum); g_mutex_lock (&rtx->lock); data.rtx = rtx; data.seqnum = seqnum; data.found = FALSE; rtx->n_requests += 1; g_queue_foreach (rtx->queue, (GFunc) push_seqnum, &data); g_mutex_unlock (&rtx->lock); gst_event_unref (event); res = TRUE; } else { res = gst_pad_event_default (pad, parent, event); } break; } default: res = gst_pad_event_default (pad, parent, event); break; } return res; } static gboolean gst_rtp_rtx_queue_sink_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (parent); gboolean res; switch (GST_EVENT_TYPE (event)) { case GST_EVENT_SEGMENT: { g_mutex_lock (&rtx->lock); gst_event_copy_segment (event, &rtx->head_segment); g_queue_push_head (rtx->queue, gst_event_ref (event)); g_mutex_unlock (&rtx->lock); /* fall through */ } default: res = gst_pad_event_default (pad, parent, event); break; } return res; } static void do_push (GstBuffer * buffer, GstRTPRtxQueue * rtx) { rtx->n_fulfilled_requests += 1; gst_pad_push (rtx->srcpad, buffer); } static guint32 get_ts_diff (GstRTPRtxQueue * rtx) { GstClockTime high_ts, low_ts; GstClockTimeDiff result; GstBuffer *high_buf, *low_buf; high_buf = g_queue_peek_head (rtx->queue); while (GST_IS_EVENT ((low_buf = g_queue_peek_tail (rtx->queue)))) { GstEvent *event = g_queue_pop_tail (rtx->queue); gst_event_copy_segment (event, &rtx->tail_segment); gst_event_unref (event); } if (!high_buf || !low_buf || high_buf == low_buf) return 0; high_ts = GST_BUFFER_TIMESTAMP (high_buf); low_ts = GST_BUFFER_TIMESTAMP (low_buf); high_ts = gst_segment_to_running_time (&rtx->head_segment, GST_FORMAT_TIME, high_ts); low_ts = gst_segment_to_running_time (&rtx->tail_segment, GST_FORMAT_TIME, low_ts); result = high_ts - low_ts; /* return value in ms instead of ns */ return (guint32) gst_util_uint64_scale_int (result, 1, GST_MSECOND); } /* Must be called with rtx->lock */ static void shrink_queue (GstRTPRtxQueue * rtx) { if (rtx->max_size_packets) { while (g_queue_get_length (rtx->queue) > rtx->max_size_packets) gst_buffer_unref (g_queue_pop_tail (rtx->queue)); } if (rtx->max_size_time) { while (get_ts_diff (rtx) > rtx->max_size_time) gst_buffer_unref (g_queue_pop_tail (rtx->queue)); } } static GstFlowReturn gst_rtp_rtx_queue_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer) { GstRTPRtxQueue *rtx; GstFlowReturn ret; GList *pending; rtx = GST_RTP_RTX_QUEUE (parent); g_mutex_lock (&rtx->lock); g_queue_push_head (rtx->queue, gst_buffer_ref (buffer)); shrink_queue (rtx); pending = rtx->pending; rtx->pending = NULL; g_mutex_unlock (&rtx->lock); pending = g_list_reverse (pending); g_list_foreach (pending, (GFunc) do_push, rtx); g_list_free (pending); ret = gst_pad_push (rtx->srcpad, buffer); return ret; } static gboolean push_to_queue (GstBuffer ** buffer, guint idx, gpointer user_data) { GQueue *queue = user_data; g_queue_push_head (queue, gst_buffer_ref (*buffer)); return TRUE; } static GstFlowReturn gst_rtp_rtx_queue_chain_list (GstPad * pad, GstObject * parent, GstBufferList * list) { GstRTPRtxQueue *rtx; GstFlowReturn ret; GList *pending; rtx = GST_RTP_RTX_QUEUE (parent); g_mutex_lock (&rtx->lock); gst_buffer_list_foreach (list, push_to_queue, rtx->queue); shrink_queue (rtx); pending = rtx->pending; rtx->pending = NULL; g_mutex_unlock (&rtx->lock); pending = g_list_reverse (pending); g_list_foreach (pending, (GFunc) do_push, rtx); g_list_free (pending); ret = gst_pad_push_list (rtx->srcpad, list); return ret; } static void gst_rtp_rtx_queue_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (object); switch (prop_id) { case PROP_MAX_SIZE_TIME: g_value_set_uint (value, rtx->max_size_time); break; case PROP_MAX_SIZE_PACKETS: g_value_set_uint (value, rtx->max_size_packets); break; case PROP_REQUESTS: g_value_set_uint (value, rtx->n_requests); break; case PROP_FULFILLED_REQUESTS: g_value_set_uint (value, rtx->n_fulfilled_requests); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtp_rtx_queue_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstRTPRtxQueue *rtx = GST_RTP_RTX_QUEUE (object); switch (prop_id) { case PROP_MAX_SIZE_TIME: rtx->max_size_time = g_value_get_uint (value); break; case PROP_MAX_SIZE_PACKETS: rtx->max_size_packets = g_value_get_uint (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstStateChangeReturn gst_rtp_rtx_queue_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn ret; GstRTPRtxQueue *rtx; rtx = GST_RTP_RTX_QUEUE (element); switch (transition) { default: break; } ret = GST_ELEMENT_CLASS (gst_rtp_rtx_queue_parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: gst_rtp_rtx_queue_reset (rtx, TRUE); break; default: break; } return ret; } gboolean gst_rtp_rtx_queue_plugin_init (GstPlugin * plugin) { GST_DEBUG_CATEGORY_INIT (gst_rtp_rtx_queue_debug, "rtprtxqueue", 0, "rtp retransmission queue"); return gst_element_register (plugin, "rtprtxqueue", GST_RANK_NONE, GST_TYPE_RTP_RTX_QUEUE); }