diff options
Diffstat (limited to 'gst-libs/gst/rtp/gstbasertpdepayload.c')
| -rw-r--r-- | gst-libs/gst/rtp/gstbasertpdepayload.c | 696 |
1 files changed, 0 insertions, 696 deletions
diff --git a/gst-libs/gst/rtp/gstbasertpdepayload.c b/gst-libs/gst/rtp/gstbasertpdepayload.c deleted file mode 100644 index 7afa5dde..00000000 --- a/gst-libs/gst/rtp/gstbasertpdepayload.c +++ /dev/null @@ -1,696 +0,0 @@ -/* GStreamer - * Copyright (C) <2005> Philippe Khalaf <burger@speedy.org> - * Copyright (C) <2005> Nokia Corporation <kai.vehmanen@nokia.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., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -/** - * SECTION:gstbasertpdepayload - * @short_description: Base class for RTP depayloader - * - * <refsect2> - * <para> - * Provides a base class for RTP depayloaders - * </para> - * </refsect2> - */ - -#include "gstbasertpdepayload.h" - -#ifdef GST_DISABLE_DEPRECATED -#define QUEUE_LOCK_INIT(base) (g_static_rec_mutex_init(&base->queuelock)) -#define QUEUE_LOCK_FREE(base) (g_static_rec_mutex_free(&base->queuelock)) -#define QUEUE_LOCK(base) (g_static_rec_mutex_lock(&base->queuelock)) -#define QUEUE_UNLOCK(base) (g_static_rec_mutex_unlock(&base->queuelock)) -#else -/* otherwise it's already been defined in the header (FIXME 0.11)*/ -#endif - -GST_DEBUG_CATEGORY_STATIC (basertpdepayload_debug); -#define GST_CAT_DEFAULT (basertpdepayload_debug) - -#define GST_BASE_RTP_DEPAYLOAD_GET_PRIVATE(obj) \ - (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_RTP_DEPAYLOAD, GstBaseRTPDepayloadPrivate)) - -struct _GstBaseRTPDepayloadPrivate -{ - GstClockTime npt_start; - GstClockTime npt_stop; - gdouble play_speed; - gdouble play_scale; - - gboolean discont; - GstClockTime timestamp; - GstClockTime duration; - - guint32 next_seqnum; - - gboolean negotiated; -}; - -/* Filter signals and args */ -enum -{ - /* FILL ME */ - LAST_SIGNAL -}; - -#define DEFAULT_QUEUE_DELAY 0 - -enum -{ - PROP_0, - PROP_QUEUE_DELAY, - PROP_LAST -}; - -static void gst_base_rtp_depayload_finalize (GObject * object); -static void gst_base_rtp_depayload_set_property (GObject * object, - guint prop_id, const GValue * value, GParamSpec * pspec); -static void gst_base_rtp_depayload_get_property (GObject * object, - guint prop_id, GValue * value, GParamSpec * pspec); - -static gboolean gst_base_rtp_depayload_setcaps (GstPad * pad, GstCaps * caps); -static GstFlowReturn gst_base_rtp_depayload_chain (GstPad * pad, - GstBuffer * in); -static gboolean gst_base_rtp_depayload_handle_sink_event (GstPad * pad, - GstEvent * event); - -static GstStateChangeReturn gst_base_rtp_depayload_change_state (GstElement * - element, GstStateChange transition); - -static void gst_base_rtp_depayload_set_gst_timestamp - (GstBaseRTPDepayload * filter, guint32 rtptime, GstBuffer * buf); -static gboolean gst_base_rtp_depayload_packet_lost (GstBaseRTPDepayload * - filter, GstEvent * event); - -GST_BOILERPLATE (GstBaseRTPDepayload, gst_base_rtp_depayload, GstElement, - GST_TYPE_ELEMENT); - -static void -gst_base_rtp_depayload_base_init (gpointer klass) -{ - /*GstElementClass *element_class = GST_ELEMENT_CLASS (klass); */ -} - -static void -gst_base_rtp_depayload_class_init (GstBaseRTPDepayloadClass * klass) -{ - GObjectClass *gobject_class; - GstElementClass *gstelement_class; - - gobject_class = G_OBJECT_CLASS (klass); - gstelement_class = (GstElementClass *) klass; - parent_class = g_type_class_peek_parent (klass); - - g_type_class_add_private (klass, sizeof (GstBaseRTPDepayloadPrivate)); - - gobject_class->finalize = gst_base_rtp_depayload_finalize; - gobject_class->set_property = gst_base_rtp_depayload_set_property; - gobject_class->get_property = gst_base_rtp_depayload_get_property; - - /** - * GstBaseRTPDepayload::queue-delay - * - * Control the amount of packets to buffer. - * - * Deprecated: Use a jitterbuffer or RTP session manager to delay packet - * playback. This property has no effect anymore since 0.10.15. - */ -#ifndef GST_REMOVE_DEPRECATED - g_object_class_install_property (gobject_class, PROP_QUEUE_DELAY, - g_param_spec_uint ("queue-delay", "Queue Delay", - "Amount of ms to queue/buffer, deprecated", 0, G_MAXUINT, - DEFAULT_QUEUE_DELAY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); -#endif - - gstelement_class->change_state = gst_base_rtp_depayload_change_state; - - klass->set_gst_timestamp = gst_base_rtp_depayload_set_gst_timestamp; - klass->packet_lost = gst_base_rtp_depayload_packet_lost; - - GST_DEBUG_CATEGORY_INIT (basertpdepayload_debug, "basertpdepayload", 0, - "Base class for RTP Depayloaders"); -} - -static void -gst_base_rtp_depayload_init (GstBaseRTPDepayload * filter, - GstBaseRTPDepayloadClass * klass) -{ - GstPadTemplate *pad_template; - GstBaseRTPDepayloadPrivate *priv; - - priv = GST_BASE_RTP_DEPAYLOAD_GET_PRIVATE (filter); - filter->priv = priv; - - GST_DEBUG_OBJECT (filter, "init"); - - pad_template = - gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "sink"); - g_return_if_fail (pad_template != NULL); - filter->sinkpad = gst_pad_new_from_template (pad_template, "sink"); - gst_pad_set_setcaps_function (filter->sinkpad, - gst_base_rtp_depayload_setcaps); - gst_pad_set_chain_function (filter->sinkpad, gst_base_rtp_depayload_chain); - gst_pad_set_event_function (filter->sinkpad, - gst_base_rtp_depayload_handle_sink_event); - gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad); - - pad_template = - gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "src"); - g_return_if_fail (pad_template != NULL); - filter->srcpad = gst_pad_new_from_template (pad_template, "src"); - gst_pad_use_fixed_caps (filter->srcpad); - gst_element_add_pad (GST_ELEMENT (filter), filter->srcpad); - - filter->queue = g_queue_new (); - filter->queue_delay = DEFAULT_QUEUE_DELAY; - - gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED); -} - -static void -gst_base_rtp_depayload_finalize (GObject * object) -{ - GstBaseRTPDepayload *filter = GST_BASE_RTP_DEPAYLOAD (object); - - g_queue_free (filter->queue); - - G_OBJECT_CLASS (parent_class)->finalize (object); -} - -static gboolean -gst_base_rtp_depayload_setcaps (GstPad * pad, GstCaps * caps) -{ - GstBaseRTPDepayload *filter; - GstBaseRTPDepayloadClass *bclass; - GstBaseRTPDepayloadPrivate *priv; - gboolean res; - GstStructure *caps_struct; - const GValue *value; - - filter = GST_BASE_RTP_DEPAYLOAD (gst_pad_get_parent (pad)); - priv = filter->priv; - - bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter); - - GST_DEBUG_OBJECT (filter, "Set caps"); - - caps_struct = gst_caps_get_structure (caps, 0); - - /* get other values for newsegment */ - value = gst_structure_get_value (caps_struct, "npt-start"); - if (value && G_VALUE_HOLDS_UINT64 (value)) - priv->npt_start = g_value_get_uint64 (value); - else - priv->npt_start = 0; - GST_DEBUG_OBJECT (filter, "NPT start %" G_GUINT64_FORMAT, priv->npt_start); - - value = gst_structure_get_value (caps_struct, "npt-stop"); - if (value && G_VALUE_HOLDS_UINT64 (value)) - priv->npt_stop = g_value_get_uint64 (value); - else - priv->npt_stop = -1; - - GST_DEBUG_OBJECT (filter, "NPT stop %" G_GUINT64_FORMAT, priv->npt_stop); - - value = gst_structure_get_value (caps_struct, "play-speed"); - if (value && G_VALUE_HOLDS_DOUBLE (value)) - priv->play_speed = g_value_get_double (value); - else - priv->play_speed = 1.0; - - value = gst_structure_get_value (caps_struct, "play-scale"); - if (value && G_VALUE_HOLDS_DOUBLE (value)) - priv->play_scale = g_value_get_double (value); - else - priv->play_scale = 1.0; - - if (bclass->set_caps) - res = bclass->set_caps (filter, caps); - else - res = TRUE; - - priv->negotiated = res; - - gst_object_unref (filter); - - return res; -} - -static GstFlowReturn -gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in) -{ - GstBaseRTPDepayload *filter; - GstBaseRTPDepayloadPrivate *priv; - GstBaseRTPDepayloadClass *bclass; - GstFlowReturn ret = GST_FLOW_OK; - GstBuffer *out_buf; - GstClockTime timestamp; - guint16 seqnum; - guint32 rtptime; - gboolean reset_seq, discont; - gint gap; - - filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad)); - priv = filter->priv; - - /* we must have a setcaps first */ - if (G_UNLIKELY (!priv->negotiated)) - goto not_negotiated; - - /* we must validate, it's possible that this element is plugged right after a - * network receiver and we don't want to operate on invalid data */ - if (G_UNLIKELY (!gst_rtp_buffer_validate (in))) - goto invalid_buffer; - - priv->discont = GST_BUFFER_IS_DISCONT (in); - - timestamp = GST_BUFFER_TIMESTAMP (in); - /* convert to running_time and save the timestamp, this is the timestamp - * we put on outgoing buffers. */ - timestamp = gst_segment_to_running_time (&filter->segment, GST_FORMAT_TIME, - timestamp); - priv->timestamp = timestamp; - priv->duration = GST_BUFFER_DURATION (in); - - seqnum = gst_rtp_buffer_get_seq (in); - rtptime = gst_rtp_buffer_get_timestamp (in); - reset_seq = TRUE; - discont = FALSE; - - GST_LOG_OBJECT (filter, "discont %d, seqnum %u, rtptime %u, timestamp %" - GST_TIME_FORMAT, priv->discont, seqnum, rtptime, - GST_TIME_ARGS (timestamp)); - - /* Check seqnum. This is a very simple check that makes sure that the seqnums - * are striclty increasing, dropping anything that is out of the ordinary. We - * can only do this when the next_seqnum is known. */ - if (G_LIKELY (priv->next_seqnum != -1)) { - gap = gst_rtp_buffer_compare_seqnum (seqnum, priv->next_seqnum); - - /* if we have no gap, all is fine */ - if (G_UNLIKELY (gap != 0)) { - GST_LOG_OBJECT (filter, "got packet %u, expected %u, gap %d", seqnum, - priv->next_seqnum, gap); - if (gap < 0) { - /* seqnum > next_seqnum, we are missing some packets, this is always a - * DISCONT. */ - GST_LOG_OBJECT (filter, "%d missing packets", gap); - discont = TRUE; - } else { - /* seqnum < next_seqnum, we have seen this packet before or the sender - * could be restarted. If the packet is not too old, we throw it away as - * a duplicate, otherwise we mark discont and continue. 100 misordered - * packets is a good threshold. See also RFC 4737. */ - if (gap < 100) - goto dropping; - - GST_LOG_OBJECT (filter, - "%d > 100, packet too old, sender likely restarted", gap); - discont = TRUE; - } - } - } - priv->next_seqnum = (seqnum + 1) & 0xffff; - - if (G_UNLIKELY (discont && !priv->discont)) { - GST_LOG_OBJECT (filter, "mark DISCONT on input buffer"); - /* we detected a seqnum discont but the buffer was not flagged with a discont, - * set the discont flag so that the subclass can throw away old data. */ - priv->discont = TRUE; - GST_BUFFER_FLAG_SET (in, GST_BUFFER_FLAG_DISCONT); - } - - bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter); - - if (G_UNLIKELY (bclass->process == NULL)) - goto no_process; - - /* let's send it out to processing */ - out_buf = bclass->process (filter, in); - if (out_buf) { - /* we pass rtptime as backward compatibility, in reality, the incomming - * buffer timestamp is always applied to the outgoing packet. */ - ret = gst_base_rtp_depayload_push_ts (filter, rtptime, out_buf); - } - gst_buffer_unref (in); - - return ret; - - /* ERRORS */ -not_negotiated: - { - /* this is not fatal but should be filtered earlier */ - GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION, (NULL), - ("Not RTP format was negotiated")); - gst_buffer_unref (in); - return GST_FLOW_NOT_NEGOTIATED; - } -invalid_buffer: - { - /* this is not fatal but should be filtered earlier */ - GST_ELEMENT_WARNING (filter, STREAM, DECODE, (NULL), - ("Received invalid RTP payload, dropping")); - gst_buffer_unref (in); - return GST_FLOW_OK; - } -dropping: - { - GST_WARNING_OBJECT (filter, "%d <= 100, dropping old packet", gap); - gst_buffer_unref (in); - return GST_FLOW_OK; - } -no_process: - { - /* this is not fatal but should be filtered earlier */ - GST_ELEMENT_ERROR (filter, STREAM, NOT_IMPLEMENTED, (NULL), - ("The subclass does not have a process method")); - gst_buffer_unref (in); - return GST_FLOW_ERROR; - } -} - -static gboolean -gst_base_rtp_depayload_handle_sink_event (GstPad * pad, GstEvent * event) -{ - GstBaseRTPDepayload *filter; - gboolean res = TRUE; - gboolean forward = TRUE; - - filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad)); - - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_FLUSH_STOP: - gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED); - filter->need_newsegment = TRUE; - filter->priv->next_seqnum = -1; - break; - case GST_EVENT_NEWSEGMENT: - { - gboolean update; - gdouble rate; - GstFormat fmt; - gint64 start, stop, position; - - gst_event_parse_new_segment (event, &update, &rate, &fmt, &start, &stop, - &position); - - gst_segment_set_newsegment (&filter->segment, update, rate, fmt, - start, stop, position); - - /* don't pass the event downstream, we generate our own segment including - * the NTP time and other things we receive in caps */ - forward = FALSE; - break; - } - case GST_EVENT_CUSTOM_DOWNSTREAM: - { - GstBaseRTPDepayloadClass *bclass; - - bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter); - - if (gst_event_has_name (event, "GstRTPPacketLost")) { - /* we get this event from the jitterbuffer when it considers a packet as - * being lost. We send it to our packet_lost vmethod. The default - * implementation will make time progress by pushing out a NEWSEGMENT - * update event. Subclasses can override and to one of the following: - * - Adjust timestamp/duration to something more accurate before - * calling the parent (default) packet_lost method. - * - do some more advanced error concealing on the already received - * (fragmented) packets. - * - ignore the packet lost. - */ - if (bclass->packet_lost) - res = bclass->packet_lost (filter, event); - forward = FALSE; - } - break; - } - default: - break; - } - - if (forward) - res = gst_pad_push_event (filter->srcpad, event); - else - gst_event_unref (event); - - return res; -} - -static GstEvent * -create_segment_event (GstBaseRTPDepayload * filter, gboolean update, - GstClockTime position) -{ - GstEvent *event; - GstClockTime stop; - GstBaseRTPDepayloadPrivate *priv; - - priv = filter->priv; - - if (priv->npt_stop != -1) - stop = priv->npt_stop - priv->npt_start; - else - stop = -1; - - event = gst_event_new_new_segment_full (update, priv->play_speed, - priv->play_scale, GST_FORMAT_TIME, position, stop, - position + priv->npt_start); - - return event; -} - -static GstFlowReturn -gst_base_rtp_depayload_push_full (GstBaseRTPDepayload * filter, - gboolean do_ts, guint32 rtptime, GstBuffer * out_buf) -{ - GstFlowReturn ret; - GstCaps *srccaps; - GstBaseRTPDepayloadClass *bclass; - GstBaseRTPDepayloadPrivate *priv; - - priv = filter->priv; - - /* set the caps if any */ - srccaps = GST_PAD_CAPS (filter->srcpad); - if (G_LIKELY (srccaps)) - gst_buffer_set_caps (out_buf, srccaps); - - bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter); - - /* set the timestamp if we must and can */ - if (bclass->set_gst_timestamp && do_ts) - bclass->set_gst_timestamp (filter, rtptime, out_buf); - - /* if this is the first buffer send a NEWSEGMENT */ - if (G_UNLIKELY (filter->need_newsegment)) { - GstEvent *event; - - event = create_segment_event (filter, FALSE, 0); - - gst_pad_push_event (filter->srcpad, event); - - filter->need_newsegment = FALSE; - GST_DEBUG_OBJECT (filter, "Pushed newsegment event on this first buffer"); - } - - if (G_UNLIKELY (priv->discont)) { - GST_LOG_OBJECT (filter, "Marking DISCONT on output buffer"); - GST_BUFFER_FLAG_SET (out_buf, GST_BUFFER_FLAG_DISCONT); - priv->discont = FALSE; - } - - /* push it */ - GST_LOG_OBJECT (filter, "Pushing buffer size %d, timestamp %" GST_TIME_FORMAT, - GST_BUFFER_SIZE (out_buf), - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (out_buf))); - - ret = gst_pad_push (filter->srcpad, out_buf); - - return ret; -} - -/** - * gst_base_rtp_depayload_push_ts: - * @filter: a #GstBaseRTPDepayload - * @timestamp: an RTP timestamp to apply - * @out_buf: a #GstBuffer - * - * Push @out_buf to the peer of @filter. This function takes ownership of - * @out_buf. - * - * Unlike gst_base_rtp_depayload_push(), this function will apply @timestamp - * on the outgoing buffer, using the configured clock_rate to convert the - * timestamp to a valid GStreamer clock time. - * - * Returns: a #GstFlowReturn. - */ -GstFlowReturn -gst_base_rtp_depayload_push_ts (GstBaseRTPDepayload * filter, guint32 timestamp, - GstBuffer * out_buf) -{ - return gst_base_rtp_depayload_push_full (filter, TRUE, timestamp, out_buf); -} - -/** - * gst_base_rtp_depayload_push: - * @filter: a #GstBaseRTPDepayload - * @out_buf: a #GstBuffer - * - * Push @out_buf to the peer of @filter. This function takes ownership of - * @out_buf. - * - * Unlike gst_base_rtp_depayload_push_ts(), this function will not apply - * any timestamp on the outgoing buffer. - * - * Returns: a #GstFlowReturn. - */ -GstFlowReturn -gst_base_rtp_depayload_push (GstBaseRTPDepayload * filter, GstBuffer * out_buf) -{ - return gst_base_rtp_depayload_push_full (filter, FALSE, 0, out_buf); -} - -/* convert the PacketLost event form a jitterbuffer to a segment update. - * subclasses can override this. */ -static gboolean -gst_base_rtp_depayload_packet_lost (GstBaseRTPDepayload * filter, - GstEvent * event) -{ - GstClockTime timestamp, duration, position; - GstEvent *sevent; - const GstStructure *s; - - s = gst_event_get_structure (event); - - /* first start by parsing the timestamp and duration */ - timestamp = -1; - duration = -1; - - gst_structure_get_clock_time (s, "timestamp", ×tamp); - gst_structure_get_clock_time (s, "duration", &duration); - - position = timestamp; - if (duration != -1) - position += duration; - - /* update the current segment with the elapsed time */ - sevent = create_segment_event (filter, TRUE, position); - - return gst_pad_push_event (filter->srcpad, sevent); -} - -static void -gst_base_rtp_depayload_set_gst_timestamp (GstBaseRTPDepayload * filter, - guint32 rtptime, GstBuffer * buf) -{ - GstBaseRTPDepayloadPrivate *priv; - GstClockTime timestamp, duration; - - priv = filter->priv; - - timestamp = GST_BUFFER_TIMESTAMP (buf); - duration = GST_BUFFER_DURATION (buf); - - /* apply last incomming timestamp and duration to outgoing buffer if - * not otherwise set. */ - if (!GST_CLOCK_TIME_IS_VALID (timestamp)) - GST_BUFFER_TIMESTAMP (buf) = priv->timestamp; - if (!GST_CLOCK_TIME_IS_VALID (duration)) - GST_BUFFER_DURATION (buf) = priv->duration; -} - -static GstStateChangeReturn -gst_base_rtp_depayload_change_state (GstElement * element, - GstStateChange transition) -{ - GstBaseRTPDepayload *filter; - GstBaseRTPDepayloadPrivate *priv; - GstStateChangeReturn ret; - - filter = GST_BASE_RTP_DEPAYLOAD (element); - priv = filter->priv; - - switch (transition) { - case GST_STATE_CHANGE_NULL_TO_READY: - break; - case GST_STATE_CHANGE_READY_TO_PAUSED: - filter->need_newsegment = TRUE; - priv->npt_start = 0; - priv->npt_stop = -1; - priv->play_speed = 1.0; - priv->play_scale = 1.0; - priv->next_seqnum = -1; - priv->negotiated = FALSE; - break; - case GST_STATE_CHANGE_PAUSED_TO_PLAYING: - break; - default: - break; - } - - ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); - - switch (transition) { - case GST_STATE_CHANGE_PLAYING_TO_PAUSED: - break; - case GST_STATE_CHANGE_PAUSED_TO_READY: - break; - case GST_STATE_CHANGE_READY_TO_NULL: - break; - default: - break; - } - return ret; -} - -static void -gst_base_rtp_depayload_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstBaseRTPDepayload *filter; - - filter = GST_BASE_RTP_DEPAYLOAD (object); - - switch (prop_id) { - case PROP_QUEUE_DELAY: - filter->queue_delay = g_value_get_uint (value); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_base_rtp_depayload_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstBaseRTPDepayload *filter; - - filter = GST_BASE_RTP_DEPAYLOAD (object); - - switch (prop_id) { - case PROP_QUEUE_DELAY: - g_value_set_uint (value, filter->queue_delay); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} |