/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
+ * Copyright (C) 2015 Centricular Ltd
+ * Author: Sebastian Dröge <sebastian@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* Last reviewed on 2013-07-11 (1.0.0)
*/
+#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#include <gst/app/gstappsrc.h>
#include <gst/app/gstappsink.h>
+#include <gst/sdp/gstmikey.h>
+#include <gst/rtp/gstrtppayloads.h>
+
+#define AES_128_KEY_LEN 16
+#define AES_256_KEY_LEN 32
+
+#define HMAC_32_KEY_LEN 4
+#define HMAC_80_KEY_LEN 10
+
#include "rtsp-media.h"
#define GST_RTSP_MEDIA_GET_PRIVATE(obj) \
guint buffer_size;
GstRTSPAddressPool *pool;
gboolean blocked;
+ GstRTSPTransportMode transport_mode;
GstElement *element;
GRecMutex state_lock; /* locking order: state lock, lock */
GstRTSPTimeRange range; /* protected by lock */
GstClockTime range_start;
GstClockTime range_stop;
+
+ GList *payloads; /* protected by lock */
+ GstClockTime rtx_time; /* protected by lock */
+ guint latency; /* protected by lock */
};
#define DEFAULT_SHARED FALSE
#define DEFAULT_EOS_SHUTDOWN FALSE
#define DEFAULT_BUFFER_SIZE 0x80000
#define DEFAULT_TIME_PROVIDER FALSE
+#define DEFAULT_LATENCY 200
+#define DEFAULT_TRANSPORT_MODE GST_RTSP_TRANSPORT_MODE_PLAY
/* define to dump received RTCP packets */
#undef DUMP_STATS
PROP_BUFFER_SIZE,
PROP_ELEMENT,
PROP_TIME_PROVIDER,
+ PROP_LATENCY,
+ PROP_TRANSPORT_MODE,
PROP_LAST
};
static GstElement *default_create_rtpbin (GstRTSPMedia * media);
static gboolean default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
GstSDPInfo * info);
+static gboolean default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
static gboolean wait_preroll (GstRTSPMedia * media);
#define C_ENUM(v) ((gint) v)
-#define GST_TYPE_RTSP_SUSPEND_MODE (gst_rtsp_suspend_mode_get_type())
GType
gst_rtsp_suspend_mode_get_type (void)
{
return (GType) id;
}
+#define C_FLAGS(v) ((guint) v)
+
+GType
+gst_rtsp_transport_mode_get_type (void)
+{
+ static gsize id = 0;
+ static const GFlagsValue values[] = {
+ {C_FLAGS (GST_RTSP_TRANSPORT_MODE_PLAY), "GST_RTSP_TRANSPORT_MODE_PLAY",
+ "play"},
+ {C_FLAGS (GST_RTSP_TRANSPORT_MODE_RECORD), "GST_RTSP_TRANSPORT_MODE_RECORD",
+ "record"},
+ {0, NULL, NULL}
+ };
+
+ if (g_once_init_enter (&id)) {
+ GType tmp = g_flags_register_static ("GstRTSPTransportMode", values);
+ g_once_init_leave (&id, tmp);
+ }
+ return (GType) id;
+}
+
G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
static void
"Use a NetTimeProvider for clients",
DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_LATENCY,
+ g_param_spec_uint ("latency", "Latency",
+ "Latency used for receiving media in milliseconds", 0, G_MAXUINT,
+ DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+ g_object_class_install_property (gobject_class, PROP_TRANSPORT_MODE,
+ g_param_spec_flags ("transport-mode", "Transport Mode",
+ "If this media pipeline can be used for PLAY or RECORD",
+ GST_TYPE_RTSP_TRANSPORT_MODE, DEFAULT_TRANSPORT_MODE,
+ G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
klass->query_stop = default_query_stop;
klass->create_rtpbin = default_create_rtpbin;
klass->setup_sdp = default_setup_sdp;
+ klass->handle_sdp = default_handle_sdp;
}
static void
priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
priv->buffer_size = DEFAULT_BUFFER_SIZE;
priv->time_provider = DEFAULT_TIME_PROVIDER;
+ priv->transport_mode = DEFAULT_TRANSPORT_MODE;
}
static void
gst_object_unref (priv->element);
if (priv->pool)
g_object_unref (priv->pool);
+ if (priv->payloads)
+ g_list_free (priv->payloads);
g_mutex_clear (&priv->lock);
g_cond_clear (&priv->cond);
g_rec_mutex_clear (&priv->state_lock);
case PROP_TIME_PROVIDER:
g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
break;
+ case PROP_LATENCY:
+ g_value_set_uint (value, gst_rtsp_media_get_latency (media));
+ break;
+ case PROP_TRANSPORT_MODE:
+ g_value_set_flags (value, gst_rtsp_media_get_transport_mode (media));
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
case PROP_TIME_PROVIDER:
gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
break;
+ case PROP_LATENCY:
+ gst_rtsp_media_set_latency (media, g_value_get_uint (value));
+ break;
+ case PROP_TRANSPORT_MODE:
+ gst_rtsp_media_set_transport_mode (media, g_value_get_flags (value));
+ break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
/**
+ * gst_rtsp_media_set_retransmission_time:
+ * @media: a #GstRTSPMedia
+ * @time: the new value
+ *
+ * Set the amount of time to store retransmission packets.
+ */
+void
+gst_rtsp_media_set_retransmission_time (GstRTSPMedia * media, GstClockTime time)
+{
+ GstRTSPMediaPrivate *priv;
+ guint i;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ GST_LOG_OBJECT (media, "set retransmission time %" G_GUINT64_FORMAT, time);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->rtx_time = time;
+ for (i = 0; i < priv->streams->len; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+
+ gst_rtsp_stream_set_retransmission_time (stream, time);
+ }
+
+ if (priv->rtpbin)
+ g_object_set (priv->rtpbin, "do-retransmission", time > 0, NULL);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_retransmission_time:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the amount of time to store retransmission data.
+ *
+ * Returns: the amount of time to store retransmission data.
+ */
+GstClockTime
+gst_rtsp_media_get_retransmission_time (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstClockTime res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_unlock (&priv->lock);
+ res = priv->rtx_time;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
+ * gst_rtsp_media_set_latncy:
+ * @media: a #GstRTSPMedia
+ * @latency: latency in milliseconds
+ *
+ * Configure the latency used for receiving media.
+ */
+void
+gst_rtsp_media_set_latency (GstRTSPMedia * media, guint latency)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ GST_LOG_OBJECT (media, "set latency %ums", latency);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->latency = latency;
+ if (priv->rtpbin)
+ g_object_set (priv->rtpbin, "latency", latency, NULL);
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_latency:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the latency that is used for receiving media.
+ *
+ * Returns: latency in milliseconds
+ */
+guint
+gst_rtsp_media_get_latency (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ guint res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_unlock (&priv->lock);
+ res = priv->latency;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
+
+/**
* gst_rtsp_media_use_time_provider:
* @media: a #GstRTSPMedia
* @time_provider: if a #GstNetTimeProvider should be used
return result;
}
+static GList *
+_find_payload_types (GstRTSPMedia * media)
+{
+ gint i, n;
+ GQueue queue = G_QUEUE_INIT;
+
+ n = media->priv->streams->len;
+ for (i = 0; i < n; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (media->priv->streams, i);
+ guint pt = gst_rtsp_stream_get_pt (stream);
+
+ g_queue_push_tail (&queue, GUINT_TO_POINTER (pt));
+ }
+
+ return queue.head;
+}
+
+static guint
+_next_available_pt (GList * payloads)
+{
+ guint i;
+
+ for (i = 96; i <= 127; i++) {
+ GList *iter = g_list_find (payloads, GINT_TO_POINTER (i));
+ if (!iter)
+ return GPOINTER_TO_UINT (i);
+ }
+
+ return 0;
+}
+
/**
* gst_rtsp_media_collect_streams:
* @media: a #GstRTSPMedia
*
* Collect all dynamic elements, named dynpay\%d, and add them to
* the list of dynamic elements.
+ *
+ * Find all depayloader elements, they should be named depay\%d in the
+ * element of @media, and create #GstRTSPStreams for them.
*/
void
gst_rtsp_media_collect_streams (GstRTSPMedia * media)
GstPad *pad;
gint i;
gboolean have_elem;
+ GstRTSPTransportMode mode = 0;
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
gst_object_unref (elem);
have_elem = TRUE;
+ mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
}
g_free (name);
g_mutex_unlock (&priv->lock);
have_elem = TRUE;
+ mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
}
g_free (name);
+
+ name = g_strdup_printf ("depay%d", i);
+ if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
+ GST_INFO ("found stream %d with depayloader %p", i, elem);
+
+ /* take the pad of the payloader */
+ pad = gst_element_get_static_pad (elem, "sink");
+ /* create the stream */
+ gst_rtsp_media_create_stream (media, elem, pad);
+ gst_object_unref (pad);
+ gst_object_unref (elem);
+
+ have_elem = TRUE;
+ mode |= GST_RTSP_TRANSPORT_MODE_RECORD;
+ }
+ g_free (name);
+ }
+
+ if (have_elem) {
+ if (priv->transport_mode != mode)
+ GST_WARNING ("found different mode than expected (0x%02x != 0x%02d)",
+ priv->transport_mode, mode);
}
}
* gst_rtsp_media_create_stream:
* @media: a #GstRTSPMedia
* @payloader: a #GstElement
- * @srcpad: a source #GstPad
+ * @pad: a #GstPad
*
- * Create a new stream in @media that provides RTP data on @srcpad.
- * @srcpad should be a pad of an element inside @media->element.
+ * Create a new stream in @media that provides RTP data on @pad.
+ * @pad should be a pad of an element inside @media->element.
*
* Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
* as @media exists.
{
GstRTSPMediaPrivate *priv;
GstRTSPStream *stream;
- GstPad *srcpad;
+ GstPad *ghostpad;
gchar *name;
gint idx;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
g_return_val_if_fail (GST_IS_PAD (pad), NULL);
- g_return_val_if_fail (GST_PAD_IS_SRC (pad), NULL);
priv = media->priv;
GST_DEBUG ("media %p: creating stream with index %d", media, idx);
- name = g_strdup_printf ("src_%u", idx);
- srcpad = gst_ghost_pad_new (name, pad);
- gst_pad_set_active (srcpad, TRUE);
- gst_element_add_pad (priv->element, srcpad);
+ if (GST_PAD_IS_SRC (pad))
+ name = g_strdup_printf ("src_%u", idx);
+ else
+ name = g_strdup_printf ("sink_%u", idx);
+
+ ghostpad = gst_ghost_pad_new (name, pad);
+ gst_pad_set_active (ghostpad, TRUE);
+ gst_element_add_pad (priv->element, ghostpad);
g_free (name);
- stream = gst_rtsp_stream_new (idx, payloader, srcpad);
+ stream = gst_rtsp_stream_new (idx, payloader, ghostpad);
if (priv->pool)
gst_rtsp_stream_set_address_pool (stream, priv->pool);
gst_rtsp_stream_set_profiles (stream, priv->profiles);
gst_rtsp_stream_set_protocols (stream, priv->protocols);
+ gst_rtsp_stream_set_retransmission_time (stream, priv->rtx_time);
g_ptr_array_add (priv->streams, stream);
+
+ if (GST_PAD_IS_SRC (pad)) {
+ gint i, n;
+
+ if (priv->payloads)
+ g_list_free (priv->payloads);
+ priv->payloads = _find_payload_types (media);
+
+ n = priv->streams->len;
+ for (i = 0; i < n; i++) {
+ GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
+ guint rtx_pt = _next_available_pt (priv->payloads);
+
+ if (rtx_pt == 0) {
+ GST_WARNING ("Ran out of space of dynamic payload types");
+ break;
+ }
+
+ gst_rtsp_stream_set_retransmission_pt (stream, rtx_pt);
+
+ priv->payloads =
+ g_list_append (priv->payloads, GUINT_TO_POINTER (rtx_pt));
+ }
+ }
g_mutex_unlock (&priv->lock);
g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
goto not_prepared;
/* Update the seekable state of the pipeline in case it changed */
- query = gst_query_new_seeking (GST_FORMAT_TIME);
- if (gst_element_query (priv->pipeline, query)) {
- GstFormat format;
- gboolean seekable;
- gint64 start, end;
-
- gst_query_parse_seeking (query, &format, &seekable, &start, &end);
- priv->seekable = seekable;
+ if ((priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD)) {
+ /* TODO: Seeking for RECORD? */
+ priv->seekable = FALSE;
+ } else {
+ query = gst_query_new_seeking (GST_FORMAT_TIME);
+ if (gst_element_query (priv->pipeline, query)) {
+ GstFormat format;
+ gboolean seekable;
+ gint64 start, end;
+
+ gst_query_parse_seeking (query, &format, &seekable, &start, &end);
+ priv->seekable = seekable;
+ }
+ gst_query_unref (query);
}
- gst_query_unref (query);
if (!priv->seekable)
goto not_seekable;
switch (type) {
case GST_MESSAGE_STATE_CHANGED:
+ {
+ GstState old, new, pending;
+
+ if (GST_MESSAGE_SRC (message) != GST_OBJECT (priv->pipeline))
+ break;
+
+ gst_message_parse_state_changed (message, &old, &new, &pending);
+
+ GST_DEBUG ("%p: went from %s to %s (pending %s)", media,
+ gst_element_state_get_name (old), gst_element_state_get_name (new),
+ gst_element_state_get_name (pending));
+ if ((priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD)
+ && old == GST_STATE_READY && new == GST_STATE_PAUSED) {
+ GST_INFO ("%p: went to PAUSED, prepared now", media);
+ collect_media_stats (media);
+
+ if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+ }
+
break;
+ }
case GST_MESSAGE_BUFFERING:
{
gint percent;
* seeking query in preroll instead */
priv->seekable = FALSE;
priv->is_live = TRUE;
- /* start blocked to make sure nothing goes to the sink */
- media_streams_set_blocked (media, TRUE);
+ if (!(priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD)) {
+ /* start blocked to make sure nothing goes to the sink */
+ media_streams_set_blocked (media, TRUE);
+ }
ret = set_state (media, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE)
goto state_failed;
}
}
+ if (priv->rtpbin)
+ g_object_set (priv->rtpbin, "do-retransmission", priv->rtx_time > 0, NULL);
+
for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
GstElement *elem = walk->data;
DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
if (priv->rtpbin != NULL) {
gboolean success = TRUE;
+ g_object_set (priv->rtpbin, "latency", priv->latency, NULL);
+
if (klass->setup_rtpbin)
success = klass->setup_rtpbin (media, priv->rtpbin);
{
GstRTSPMediaPrivate *priv = media->priv;
+ gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
+
if (priv->eos_shutdown) {
GST_DEBUG ("sending EOS for shutdown");
/* ref so that we don't disappear */
set_target_state (media, GST_STATE_NULL, FALSE);
success = TRUE;
- gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
-
if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
GstRTSPMediaClass *klass;
}
}
+static const gchar *
+rtsp_get_attribute_for_pt (const GstSDPMedia * media, const gchar * name,
+ gint pt)
+{
+ guint i;
+
+ for (i = 0;; i++) {
+ const gchar *attr;
+ gint val;
+
+ if ((attr = gst_sdp_media_get_attribute_val_n (media, name, i)) == NULL)
+ break;
+
+ if (sscanf (attr, "%d ", &val) != 1)
+ continue;
+
+ if (val == pt)
+ return attr;
+ }
+ return NULL;
+}
+
+#define PARSE_INT(p, del, res) \
+G_STMT_START { \
+ gchar *t = p; \
+ p = strstr (p, del); \
+ if (p == NULL) \
+ res = -1; \
+ else { \
+ *p = '\0'; \
+ p++; \
+ res = atoi (t); \
+ } \
+} G_STMT_END
+
+#define PARSE_STRING(p, del, res) \
+G_STMT_START { \
+ gchar *t = p; \
+ p = strstr (p, del); \
+ if (p == NULL) { \
+ res = NULL; \
+ p = t; \
+ } \
+ else { \
+ *p = '\0'; \
+ p++; \
+ res = t; \
+ } \
+} G_STMT_END
+
+#define SKIP_SPACES(p) \
+ while (*p && g_ascii_isspace (*p)) \
+ p++;
+
+/* rtpmap contains:
+ *
+ * <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
+ */
+static gboolean
+parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
+ gint * rate, gchar ** params)
+{
+ gchar *p, *t;
+
+ p = (gchar *) rtpmap;
+
+ PARSE_INT (p, " ", *payload);
+ if (*payload == -1)
+ return FALSE;
+
+ SKIP_SPACES (p);
+ if (*p == '\0')
+ return FALSE;
+
+ PARSE_STRING (p, "/", *name);
+ if (*name == NULL) {
+ GST_DEBUG ("no rate, name %s", p);
+ /* no rate, assume -1 then, this is not supposed to happen but RealMedia
+ * streams seem to omit the rate. */
+ *name = p;
+ *rate = -1;
+ return TRUE;
+ }
+
+ t = p;
+ p = strstr (p, "/");
+ if (p == NULL) {
+ *rate = atoi (t);
+ return TRUE;
+ }
+ *p = '\0';
+ p++;
+ *rate = atoi (t);
+
+ t = p;
+ if (*p == '\0')
+ return TRUE;
+ *params = t;
+
+ return TRUE;
+}
+
+/*
+ * Mapping of caps to and from SDP fields:
+ *
+ * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
+ * a=fmtp:<payload> <param>[=<value>];...
+ */
+static GstCaps *
+media_to_caps (gint pt, const GstSDPMedia * media)
+{
+ GstCaps *caps;
+ const gchar *rtpmap;
+ const gchar *fmtp;
+ gchar *name = NULL;
+ gint rate = -1;
+ gchar *params = NULL;
+ gchar *tmp;
+ GstStructure *s;
+ gint payload = 0;
+ gboolean ret;
+
+ /* get and parse rtpmap */
+ rtpmap = rtsp_get_attribute_for_pt (media, "rtpmap", pt);
+
+ if (rtpmap) {
+ ret = parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
+ if (!ret) {
+ g_warning ("error parsing rtpmap, ignoring");
+ rtpmap = NULL;
+ }
+ }
+ /* dynamic payloads need rtpmap or we fail */
+ if (rtpmap == NULL && pt >= 96)
+ goto no_rtpmap;
+
+ /* check if we have a rate, if not, we need to look up the rate from the
+ * default rates based on the payload types. */
+ if (rate == -1) {
+ const GstRTPPayloadInfo *info;
+
+ if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
+ /* dynamic types, use media and encoding_name */
+ tmp = g_ascii_strdown (media->media, -1);
+ info = gst_rtp_payload_info_for_name (tmp, name);
+ g_free (tmp);
+ } else {
+ /* static types, use payload type */
+ info = gst_rtp_payload_info_for_pt (pt);
+ }
+
+ if (info) {
+ if ((rate = info->clock_rate) == 0)
+ rate = -1;
+ }
+ /* we fail if we cannot find one */
+ if (rate == -1)
+ goto no_rate;
+ }
+
+ tmp = g_ascii_strdown (media->media, -1);
+ caps = gst_caps_new_simple ("application/x-unknown",
+ "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
+ g_free (tmp);
+ s = gst_caps_get_structure (caps, 0);
+
+ gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
+
+ /* encoding name must be upper case */
+ if (name != NULL) {
+ tmp = g_ascii_strup (name, -1);
+ gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
+ g_free (tmp);
+ }
+
+ /* params must be lower case */
+ if (params != NULL) {
+ tmp = g_ascii_strdown (params, -1);
+ gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
+ g_free (tmp);
+ }
+
+ /* parse optional fmtp: field */
+ if ((fmtp = rtsp_get_attribute_for_pt (media, "fmtp", pt))) {
+ gchar *p;
+ gint payload = 0;
+
+ p = (gchar *) fmtp;
+
+ /* p is now of the format <payload> <param>[=<value>];... */
+ PARSE_INT (p, " ", payload);
+ if (payload != -1 && payload == pt) {
+ gchar **pairs;
+ gint i;
+
+ /* <param>[=<value>] are separated with ';' */
+ pairs = g_strsplit (p, ";", 0);
+ for (i = 0; pairs[i]; i++) {
+ gchar *valpos;
+ const gchar *val, *key;
+
+ /* the key may not have a '=', the value can have other '='s */
+ valpos = strstr (pairs[i], "=");
+ if (valpos) {
+ /* we have a '=' and thus a value, remove the '=' with \0 */
+ *valpos = '\0';
+ /* value is everything between '=' and ';'. We split the pairs at ;
+ * boundaries so we can take the remainder of the value. Some servers
+ * put spaces around the value which we strip off here. Alternatively
+ * we could strip those spaces in the depayloaders should these spaces
+ * actually carry any meaning in the future. */
+ val = g_strstrip (valpos + 1);
+ } else {
+ /* simple <param>;.. is translated into <param>=1;... */
+ val = "1";
+ }
+ /* strip the key of spaces, convert key to lowercase but not the value. */
+ key = g_strstrip (pairs[i]);
+ if (strlen (key) > 1) {
+ tmp = g_ascii_strdown (key, -1);
+ gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
+ g_free (tmp);
+ }
+ }
+ g_strfreev (pairs);
+ }
+ }
+ return caps;
+
+ /* ERRORS */
+no_rtpmap:
+ {
+ g_warning ("rtpmap type not given for dynamic payload %d", pt);
+ return NULL;
+ }
+no_rate:
+ {
+ g_warning ("rate unknown for payload type %d", pt);
+ return NULL;
+ }
+}
+
+static gboolean
+parse_keymgmt (const gchar * keymgmt, GstCaps * caps)
+{
+ gboolean res = FALSE;
+ gchar *p, *kmpid;
+ gsize size;
+ guchar *data;
+ GstMIKEYMessage *msg;
+ const GstMIKEYPayload *payload;
+ const gchar *srtp_cipher;
+ const gchar *srtp_auth;
+
+ p = (gchar *) keymgmt;
+
+ SKIP_SPACES (p);
+ if (*p == '\0')
+ return FALSE;
+
+ PARSE_STRING (p, " ", kmpid);
+ if (!g_str_equal (kmpid, "mikey"))
+ return FALSE;
+
+ data = g_base64_decode (p, &size);
+ if (data == NULL)
+ return FALSE;
+
+ msg = gst_mikey_message_new_from_data (data, size, NULL, NULL);
+ g_free (data);
+ if (msg == NULL)
+ return FALSE;
+
+ srtp_cipher = "aes-128-icm";
+ srtp_auth = "hmac-sha1-80";
+
+ /* check the Security policy if any */
+ if ((payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, 0))) {
+ GstMIKEYPayloadSP *p = (GstMIKEYPayloadSP *) payload;
+ guint len, i;
+
+ if (p->proto != GST_MIKEY_SEC_PROTO_SRTP)
+ goto done;
+
+ len = gst_mikey_payload_sp_get_n_params (payload);
+ for (i = 0; i < len; i++) {
+ const GstMIKEYPayloadSPParam *param =
+ gst_mikey_payload_sp_get_param (payload, i);
+
+ switch (param->type) {
+ case GST_MIKEY_SP_SRTP_ENC_ALG:
+ switch (param->val[0]) {
+ case 0:
+ srtp_cipher = "null";
+ break;
+ case 2:
+ case 1:
+ srtp_cipher = "aes-128-icm";
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
+ switch (param->val[0]) {
+ case AES_128_KEY_LEN:
+ srtp_cipher = "aes-128-icm";
+ break;
+ case AES_256_KEY_LEN:
+ srtp_cipher = "aes-256-icm";
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_AUTH_ALG:
+ switch (param->val[0]) {
+ case 0:
+ srtp_auth = "null";
+ break;
+ case 2:
+ case 1:
+ srtp_auth = "hmac-sha1-80";
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
+ switch (param->val[0]) {
+ case HMAC_32_KEY_LEN:
+ srtp_auth = "hmac-sha1-32";
+ break;
+ case HMAC_80_KEY_LEN:
+ srtp_auth = "hmac-sha1-80";
+ break;
+ default:
+ break;
+ }
+ break;
+ case GST_MIKEY_SP_SRTP_SRTP_ENC:
+ break;
+ case GST_MIKEY_SP_SRTP_SRTCP_ENC:
+ break;
+ default:
+ break;
+ }
+ }
+ }
+
+ if (!(payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_KEMAC, 0)))
+ goto done;
+ else {
+ GstMIKEYPayloadKEMAC *p = (GstMIKEYPayloadKEMAC *) payload;
+ const GstMIKEYPayload *sub;
+ GstMIKEYPayloadKeyData *pkd;
+ GstBuffer *buf;
+
+ if (p->enc_alg != GST_MIKEY_ENC_NULL || p->mac_alg != GST_MIKEY_MAC_NULL)
+ goto done;
+
+ if (!(sub = gst_mikey_payload_kemac_get_sub (payload, 0)))
+ goto done;
+
+ if (sub->type != GST_MIKEY_PT_KEY_DATA)
+ goto done;
+
+ pkd = (GstMIKEYPayloadKeyData *) sub;
+ buf =
+ gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
+ pkd->key_len);
+ gst_caps_set_simple (caps, "srtp-key", GST_TYPE_BUFFER, buf, NULL);
+ }
+
+ gst_caps_set_simple (caps,
+ "srtp-cipher", G_TYPE_STRING, srtp_cipher,
+ "srtp-auth", G_TYPE_STRING, srtp_auth,
+ "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
+ "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
+
+ res = TRUE;
+done:
+ gst_mikey_message_unref (msg);
+
+ return res;
+}
+
+/*
+ * Mapping SDP attributes to caps
+ *
+ * prepend 'a-' to IANA registered sdp attributes names
+ * (ie: not prefixed with 'x-') in order to avoid
+ * collision with gstreamer standard caps properties names
+ */
+static void
+sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
+{
+ if (attributes->len > 0) {
+ GstStructure *s;
+ guint i;
+
+ s = gst_caps_get_structure (caps, 0);
+
+ for (i = 0; i < attributes->len; i++) {
+ GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
+ gchar *tofree, *key;
+
+ key = attr->key;
+
+ /* skip some of the attribute we already handle */
+ if (!strcmp (key, "fmtp"))
+ continue;
+ if (!strcmp (key, "rtpmap"))
+ continue;
+ if (!strcmp (key, "control"))
+ continue;
+ if (!strcmp (key, "range"))
+ continue;
+ if (g_str_equal (key, "key-mgmt")) {
+ parse_keymgmt (attr->value, caps);
+ continue;
+ }
+
+ /* string must be valid UTF8 */
+ if (!g_utf8_validate (attr->value, -1, NULL))
+ continue;
+
+ if (!g_str_has_prefix (key, "x-"))
+ tofree = key = g_strdup_printf ("a-%s", key);
+ else
+ tofree = NULL;
+
+ GST_DEBUG ("adding caps: %s=%s", key, attr->value);
+ gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
+ g_free (tofree);
+ }
+ }
+}
+
+static gboolean
+default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
+{
+ GstRTSPMediaPrivate *priv = media->priv;
+ gint i, medias_len;
+
+ medias_len = gst_sdp_message_medias_len (sdp);
+ if (medias_len != priv->streams->len) {
+ GST_ERROR ("%p: Media has more or less streams than SDP (%d /= %d)", media,
+ priv->streams->len, medias_len);
+ return FALSE;
+ }
+
+ for (i = 0; i < medias_len; i++) {
+ const gchar *proto, *media_type;
+ const GstSDPMedia *sdp_media = gst_sdp_message_get_media (sdp, i);
+ GstRTSPStream *stream;
+ gint j, formats_len;
+ const gchar *control;
+ GstRTSPProfile profile, profiles;
+
+ stream = g_ptr_array_index (priv->streams, i);
+
+ /* TODO: Should we do something with the other SDP information? */
+
+ /* get proto */
+ proto = gst_sdp_media_get_proto (sdp_media);
+ if (proto == NULL) {
+ GST_ERROR ("%p: SDP media %d has no proto", media, i);
+ return FALSE;
+ }
+
+ if (g_str_equal (proto, "RTP/AVP")) {
+ media_type = "application/x-rtp";
+ profile = GST_RTSP_PROFILE_AVP;
+ } else if (g_str_equal (proto, "RTP/SAVP")) {
+ media_type = "application/x-srtp";
+ profile = GST_RTSP_PROFILE_SAVP;
+ } else if (g_str_equal (proto, "RTP/AVPF")) {
+ media_type = "application/x-rtp";
+ profile = GST_RTSP_PROFILE_AVPF;
+ } else if (g_str_equal (proto, "RTP/SAVPF")) {
+ media_type = "application/x-srtp";
+ profile = GST_RTSP_PROFILE_SAVPF;
+ } else {
+ GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
+ return FALSE;
+ }
+
+ profiles = gst_rtsp_stream_get_profiles (stream);
+ if ((profiles & profile) == 0) {
+ GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
+ return FALSE;
+ }
+
+ formats_len = gst_sdp_media_formats_len (sdp_media);
+ for (j = 0; j < formats_len; j++) {
+ gint pt;
+ GstCaps *caps;
+ GstStructure *s;
+
+ pt = atoi (gst_sdp_media_get_format (sdp_media, j));
+
+ GST_DEBUG (" looking at %d pt: %d", j, pt);
+
+ /* convert caps */
+ caps = media_to_caps (pt, sdp_media);
+ if (caps == NULL) {
+ GST_WARNING (" skipping pt %d without caps", pt);
+ continue;
+ }
+
+ /* do some tweaks */
+ GST_DEBUG ("mapping sdp session level attributes to caps");
+ sdp_attributes_to_caps (sdp->attributes, caps);
+ GST_DEBUG ("mapping sdp media level attributes to caps");
+ sdp_attributes_to_caps (sdp_media->attributes, caps);
+
+ s = gst_caps_get_structure (caps, 0);
+ gst_structure_set_name (s, media_type);
+
+ gst_rtsp_stream_set_pt_map (stream, pt, caps);
+ gst_caps_unref (caps);
+ }
+
+ control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+ if (control)
+ gst_rtsp_stream_set_control (stream, control);
+
+ }
+
+ return TRUE;
+}
+
+/**
+ * gst_rtsp_media_handle_sdp:
+ * @media: a #GstRTSPMedia
+ * @sdp: (transfer none): a #GstSDPMessage
+ *
+ * Configure an SDP on @media for receiving streams
+ *
+ * Returns: TRUE on success.
+ */
+gboolean
+gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPMediaClass *klass;
+ gboolean res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+ g_return_val_if_fail (sdp != NULL, FALSE);
+
+ priv = media->priv;
+
+ g_rec_mutex_lock (&priv->state_lock);
+
+ klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+ if (!klass->handle_sdp)
+ goto no_handle_sdp;
+
+ res = klass->handle_sdp (media, sdp);
+
+ g_rec_mutex_unlock (&priv->state_lock);
+
+ return res;
+
+ /* ERRORS */
+no_handle_sdp:
+ {
+ g_rec_mutex_unlock (&priv->state_lock);
+ GST_ERROR ("no handle_sdp function");
+ g_critical ("no handle_sdp vmethod function set");
+ return FALSE;
+ }
+}
+
static void
do_set_seqnum (GstRTSPStream * stream)
{
activate = deactivate = FALSE;
- GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
- media);
+ GST_INFO ("going to state %s media %p, target state %s",
+ gst_element_state_get_name (state), media,
+ gst_element_state_get_name (priv->target_state));
switch (state) {
case GST_STATE_NULL:
+ /* we're going from PLAYING or PAUSED to READY or NULL, deactivate */
+ if (priv->target_state >= GST_STATE_PAUSED)
+ deactivate = TRUE;
+ break;
case GST_STATE_PAUSED:
/* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
if (priv->target_state == GST_STATE_PLAYING)
}
old_active = priv->n_active;
+ GST_DEBUG ("%d transports, activate %d, deactivate %d", transports->len,
+ activate, deactivate);
for (i = 0; i < transports->len; i++) {
GstRTSPStreamTransport *trans;
return FALSE;
}
}
+
+/**
+ * gst_rtsp_media_set_transport_mode:
+ * @media: a #GstRTSPMedia
+ * @mode: the new value
+ *
+ * Sets if the media pipeline can work in PLAY or RECORD mode
+ */
+void
+gst_rtsp_media_set_transport_mode (GstRTSPMedia * media,
+ GstRTSPTransportMode mode)
+{
+ GstRTSPMediaPrivate *priv;
+
+ g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ priv->transport_mode = mode;
+ g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_transport_mode:
+ * @media: a #GstRTSPMedia
+ *
+ * Check if the pipeline for @media can be used for PLAY or RECORD methods.
+ *
+ * Returns: The transport mode.
+ */
+GstRTSPTransportMode
+gst_rtsp_media_get_transport_mode (GstRTSPMedia * media)
+{
+ GstRTSPMediaPrivate *priv;
+ GstRTSPTransportMode res;
+
+ g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+ priv = media->priv;
+
+ g_mutex_lock (&priv->lock);
+ res = priv->transport_mode;
+ g_mutex_unlock (&priv->lock);
+
+ return res;
+}
projects / platform / upstream / gstreamer.git / blobdiff