gstreamer/gst/rtsp-server/rtsp-media.c
Wim Taymans e4ea72ccdf stream: use the address managed by the stream
Use the address managed by the stream for multicast. This allows us to have 1
multicast address for each stream.
Because the address is now managed by the stream we don't have to pass it around
anymore.
Set the address pool on the streams.
2012-11-15 16:18:29 +01:00

1506 lines
39 KiB
C

/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#include <string.h>
#include <stdlib.h>
#include <gst/app/gstappsrc.h>
#include <gst/app/gstappsink.h>
#include "rtsp-media.h"
#define DEFAULT_SHARED FALSE
#define DEFAULT_REUSABLE FALSE
#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_TCP
//#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP_MCAST
#define DEFAULT_EOS_SHUTDOWN FALSE
#define DEFAULT_BUFFER_SIZE 0x80000
/* define to dump received RTCP packets */
#undef DUMP_STATS
enum
{
PROP_0,
PROP_SHARED,
PROP_REUSABLE,
PROP_PROTOCOLS,
PROP_EOS_SHUTDOWN,
PROP_BUFFER_SIZE,
PROP_LAST
};
enum
{
SIGNAL_NEW_STREAM,
SIGNAL_PREPARED,
SIGNAL_UNPREPARED,
SIGNAL_NEW_STATE,
SIGNAL_LAST
};
GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
#define GST_CAT_DEFAULT rtsp_media_debug
static void gst_rtsp_media_get_property (GObject * object, guint propid,
GValue * value, GParamSpec * pspec);
static void gst_rtsp_media_set_property (GObject * object, guint propid,
const GValue * value, GParamSpec * pspec);
static void gst_rtsp_media_finalize (GObject * obj);
static gpointer do_loop (GstRTSPMediaClass * klass);
static gboolean default_handle_message (GstRTSPMedia * media,
GstMessage * message);
static void finish_unprepare (GstRTSPMedia * media);
static gboolean default_unprepare (GstRTSPMedia * media);
static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
static void
gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
{
GObjectClass *gobject_class;
gobject_class = G_OBJECT_CLASS (klass);
gobject_class->get_property = gst_rtsp_media_get_property;
gobject_class->set_property = gst_rtsp_media_set_property;
gobject_class->finalize = gst_rtsp_media_finalize;
g_object_class_install_property (gobject_class, PROP_SHARED,
g_param_spec_boolean ("shared", "Shared",
"If this media pipeline can be shared", DEFAULT_SHARED,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_REUSABLE,
g_param_spec_boolean ("reusable", "Reusable",
"If this media pipeline can be reused after an unprepare",
DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
g_param_spec_flags ("protocols", "Protocols",
"Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
"Send an EOS event to the pipeline before unpreparing",
DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
g_param_spec_uint ("buffer-size", "Buffer Size",
"The kernel UDP buffer size to use", 0, G_MAXUINT,
DEFAULT_BUFFER_SIZE, 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,
g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
gst_rtsp_media_signals[SIGNAL_PREPARED] =
g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 0, G_TYPE_INT);
klass->context = g_main_context_new ();
klass->loop = g_main_loop_new (klass->context, TRUE);
GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
klass->thread = g_thread_new ("Bus Thread", (GThreadFunc) do_loop, klass);
klass->handle_message = default_handle_message;
klass->unprepare = default_unprepare;
}
static void
gst_rtsp_media_init (GstRTSPMedia * media)
{
media->streams = g_ptr_array_new_with_free_func (g_object_unref);
g_mutex_init (&media->lock);
g_cond_init (&media->cond);
g_rec_mutex_init (&media->state_lock);
media->shared = DEFAULT_SHARED;
media->reusable = DEFAULT_REUSABLE;
media->protocols = DEFAULT_PROTOCOLS;
media->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
media->buffer_size = DEFAULT_BUFFER_SIZE;
}
static void
gst_rtsp_media_finalize (GObject * obj)
{
GstRTSPMedia *media;
media = GST_RTSP_MEDIA (obj);
GST_INFO ("finalize media %p", media);
gst_rtsp_media_unprepare (media);
g_ptr_array_unref (media->streams);
g_list_free_full (media->dynamic, gst_object_unref);
if (media->source) {
g_source_destroy (media->source);
g_source_unref (media->source);
}
if (media->auth)
g_object_unref (media->auth);
if (media->pool)
g_object_unref (media->pool);
g_mutex_clear (&media->lock);
g_cond_clear (&media->cond);
g_rec_mutex_clear (&media->state_lock);
G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
}
static void
gst_rtsp_media_get_property (GObject * object, guint propid,
GValue * value, GParamSpec * pspec)
{
GstRTSPMedia *media = GST_RTSP_MEDIA (object);
switch (propid) {
case PROP_SHARED:
g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
break;
case PROP_REUSABLE:
g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
break;
case PROP_PROTOCOLS:
g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
break;
case PROP_EOS_SHUTDOWN:
g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
break;
case PROP_BUFFER_SIZE:
g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
static void
gst_rtsp_media_set_property (GObject * object, guint propid,
const GValue * value, GParamSpec * pspec)
{
GstRTSPMedia *media = GST_RTSP_MEDIA (object);
switch (propid) {
case PROP_SHARED:
gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
break;
case PROP_REUSABLE:
gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
break;
case PROP_PROTOCOLS:
gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
break;
case PROP_EOS_SHUTDOWN:
gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
break;
case PROP_BUFFER_SIZE:
gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
}
}
static gpointer
do_loop (GstRTSPMediaClass * klass)
{
GST_INFO ("enter mainloop");
g_main_loop_run (klass->loop);
GST_INFO ("exit mainloop");
return NULL;
}
/* must be called with state lock */
static void
collect_media_stats (GstRTSPMedia * media)
{
gint64 position, duration;
media->range.unit = GST_RTSP_RANGE_NPT;
GST_INFO ("collect media stats");
if (media->is_live) {
media->range.min.type = GST_RTSP_TIME_NOW;
media->range.min.seconds = -1;
media->range.max.type = GST_RTSP_TIME_END;
media->range.max.seconds = -1;
} else {
/* get the position */
if (!gst_element_query_position (media->pipeline, GST_FORMAT_TIME,
&position)) {
GST_INFO ("position query failed");
position = 0;
}
/* get the duration */
if (!gst_element_query_duration (media->pipeline, GST_FORMAT_TIME,
&duration)) {
GST_INFO ("duration query failed");
duration = -1;
}
GST_INFO ("stats: position %" GST_TIME_FORMAT ", duration %"
GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (duration));
if (position == -1) {
media->range.min.type = GST_RTSP_TIME_NOW;
media->range.min.seconds = -1;
} else {
media->range.min.type = GST_RTSP_TIME_SECONDS;
media->range.min.seconds = ((gdouble) position) / GST_SECOND;
}
if (duration == -1) {
media->range.max.type = GST_RTSP_TIME_END;
media->range.max.seconds = -1;
} else {
media->range.max.type = GST_RTSP_TIME_SECONDS;
media->range.max.seconds = ((gdouble) duration) / GST_SECOND;
}
}
}
/**
* gst_rtsp_media_new:
*
* Create a new #GstRTSPMedia instance. The #GstRTSPMedia object contains the
* element to produce RTP data for one or more related (audio/video/..)
* streams.
*
* Returns: a new #GstRTSPMedia object.
*/
GstRTSPMedia *
gst_rtsp_media_new (void)
{
GstRTSPMedia *result;
result = g_object_new (GST_TYPE_RTSP_MEDIA, NULL);
return result;
}
/**
* gst_rtsp_media_set_shared:
* @media: a #GstRTSPMedia
* @shared: the new value
*
* Set or unset if the pipeline for @media can be shared will multiple clients.
* When @shared is %TRUE, client requests for this media will share the media
* pipeline.
*/
void
gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
{
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
g_mutex_lock (&media->lock);
media->shared = shared;
g_mutex_unlock (&media->lock);
}
/**
* gst_rtsp_media_is_shared:
* @media: a #GstRTSPMedia
*
* Check if the pipeline for @media can be shared between multiple clients.
*
* Returns: %TRUE if the media can be shared between clients.
*/
gboolean
gst_rtsp_media_is_shared (GstRTSPMedia * media)
{
gboolean res;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
g_mutex_lock (&media->lock);
res = media->shared;
g_mutex_unlock (&media->lock);
return res;
}
/**
* gst_rtsp_media_set_reusable:
* @media: a #GstRTSPMedia
* @reusable: the new value
*
* Set or unset if the pipeline for @media can be reused after the pipeline has
* been unprepared.
*/
void
gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
{
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
g_mutex_lock (&media->lock);
media->reusable = reusable;
g_mutex_unlock (&media->lock);
}
/**
* gst_rtsp_media_is_reusable:
* @media: a #GstRTSPMedia
*
* Check if the pipeline for @media can be reused after an unprepare.
*
* Returns: %TRUE if the media can be reused
*/
gboolean
gst_rtsp_media_is_reusable (GstRTSPMedia * media)
{
gboolean res;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
g_mutex_lock (&media->lock);
res = media->reusable;
g_mutex_unlock (&media->lock);
return res;
}
/**
* gst_rtsp_media_set_protocols:
* @media: a #GstRTSPMedia
* @protocols: the new flags
*
* Configure the allowed lower transport for @media.
*/
void
gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
{
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
g_mutex_lock (&media->lock);
media->protocols = protocols;
g_mutex_unlock (&media->lock);
}
/**
* gst_rtsp_media_get_protocols:
* @media: a #GstRTSPMedia
*
* Get the allowed protocols of @media.
*
* Returns: a #GstRTSPLowerTrans
*/
GstRTSPLowerTrans
gst_rtsp_media_get_protocols (GstRTSPMedia * media)
{
GstRTSPLowerTrans res;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
GST_RTSP_LOWER_TRANS_UNKNOWN);
g_mutex_lock (&media->lock);
res = media->protocols;
g_mutex_unlock (&media->lock);
return res;
}
/**
* gst_rtsp_media_set_eos_shutdown:
* @media: a #GstRTSPMedia
* @eos_shutdown: the new value
*
* Set or unset if an EOS event will be sent to the pipeline for @media before
* it is unprepared.
*/
void
gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
{
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
g_mutex_lock (&media->lock);
media->eos_shutdown = eos_shutdown;
g_mutex_unlock (&media->lock);
}
/**
* gst_rtsp_media_is_eos_shutdown:
* @media: a #GstRTSPMedia
*
* Check if the pipeline for @media will send an EOS down the pipeline before
* unpreparing.
*
* Returns: %TRUE if the media will send EOS before unpreparing.
*/
gboolean
gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
{
gboolean res;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
g_mutex_lock (&media->lock);
res = media->eos_shutdown;
g_mutex_unlock (&media->lock);
return res;
}
/**
* gst_rtsp_media_set_buffer_size:
* @media: a #GstRTSPMedia
* @size: the new value
*
* Set the kernel UDP buffer size.
*/
void
gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
{
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
GST_LOG_OBJECT (media, "set buffer size %u", size);
g_mutex_lock (&media->lock);
media->buffer_size = size;
g_mutex_unlock (&media->lock);
}
/**
* gst_rtsp_media_get_buffer_size:
* @media: a #GstRTSPMedia
*
* Get the kernel UDP buffer size.
*
* Returns: the kernel UDP buffer size.
*/
guint
gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
{
guint res;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
g_mutex_unlock (&media->lock);
res = media->buffer_size;
g_mutex_unlock (&media->lock);
return res;
}
/**
* gst_rtsp_media_set_auth:
* @media: a #GstRTSPMedia
* @auth: a #GstRTSPAuth
*
* configure @auth to be used as the authentication manager of @media.
*/
void
gst_rtsp_media_set_auth (GstRTSPMedia * media, GstRTSPAuth * auth)
{
GstRTSPAuth *old;
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
GST_LOG_OBJECT (media, "set auth %p", auth);
g_mutex_lock (&media->lock);
if ((old = media->auth) != auth)
media->auth = auth ? g_object_ref (auth) : NULL;
else
old = NULL;
g_mutex_unlock (&media->lock);
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_media_get_auth:
* @media: a #GstRTSPMedia
*
* Get the #GstRTSPAuth used as the authentication manager of @media.
*
* Returns: (transfer full): the #GstRTSPAuth of @media. g_object_unref() after
* usage.
*/
GstRTSPAuth *
gst_rtsp_media_get_auth (GstRTSPMedia * media)
{
GstRTSPAuth *result;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
g_mutex_lock (&media->lock);
if ((result = media->auth))
g_object_ref (result);
g_mutex_unlock (&media->lock);
return result;
}
/**
* gst_rtsp_media_set_address_pool:
* @media: a #GstRTSPMedia
* @pool: a #GstRTSPAddressPool
*
* configure @pool to be used as the address pool of @media.
*/
void
gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
GstRTSPAddressPool * pool)
{
GstRTSPAddressPool *old;
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
GST_LOG_OBJECT (media, "set address pool %p", pool);
g_mutex_lock (&media->lock);
if ((old = media->pool) != pool)
media->pool = pool ? g_object_ref (pool) : NULL;
else
old = NULL;
g_ptr_array_foreach (media->streams, (GFunc) gst_rtsp_stream_set_address_pool,
pool);
g_mutex_unlock (&media->lock);
if (old)
g_object_unref (old);
}
/**
* gst_rtsp_media_get_address_pool:
* @media: a #GstRTSPMedia
*
* Get the #GstRTSPAddressPool used as the address pool of @media.
*
* Returns: (transfer full): the #GstRTSPAddressPool of @media. g_object_unref() after
* usage.
*/
GstRTSPAddressPool *
gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
{
GstRTSPAddressPool *result;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
g_mutex_lock (&media->lock);
if ((result = media->pool))
g_object_ref (result);
g_mutex_unlock (&media->lock);
return result;
}
/**
* gst_rtsp_media_collect_streams:
* @media: a #GstRTSPMedia
*
* Find all payloader elements, they should be named pay%d in the
* element of @media, and create #GstRTSPStreams for them.
*
* Collect all dynamic elements, named dynpay%d, and add them to
* the list of dynamic elements.
*/
void
gst_rtsp_media_collect_streams (GstRTSPMedia * media)
{
GstElement *element, *elem;
GstPad *pad;
gint i;
gboolean have_elem;
g_return_if_fail (GST_IS_RTSP_MEDIA (media));
element = media->element;
have_elem = TRUE;
for (i = 0; have_elem; i++) {
gchar *name;
have_elem = FALSE;
name = g_strdup_printf ("pay%d", i);
if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
GST_INFO ("found stream %d with payloader %p", i, elem);
/* take the pad of the payloader */
pad = gst_element_get_static_pad (elem, "src");
/* create the stream */
gst_rtsp_media_create_stream (media, elem, pad);
g_object_unref (pad);
gst_object_unref (elem);
have_elem = TRUE;
}
g_free (name);
name = g_strdup_printf ("dynpay%d", i);
if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
/* a stream that will dynamically create pads to provide RTP packets */
GST_INFO ("found dynamic element %d, %p", i, elem);
g_mutex_lock (&media->lock);
media->dynamic = g_list_prepend (media->dynamic, elem);
g_mutex_unlock (&media->lock);
have_elem = TRUE;
}
g_free (name);
}
}
/**
* gst_rtsp_media_create_stream:
* @media: a #GstRTSPMedia
* @payloader: a #GstElement
* @srcpad: a source #GstPad
*
* Create a new stream in @media that provides RTP data on @srcpad.
* @srcpad 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.
*/
GstRTSPStream *
gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
GstPad * pad)
{
GstRTSPStream *stream;
GstPad *srcpad;
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);
g_mutex_lock (&media->lock);
idx = media->streams->len;
name = g_strdup_printf ("src_%u", idx);
srcpad = gst_ghost_pad_new (name, pad);
gst_pad_set_active (srcpad, TRUE);
gst_element_add_pad (media->element, srcpad);
g_free (name);
stream = gst_rtsp_stream_new (idx, payloader, srcpad);
if (media->pool)
gst_rtsp_stream_set_address_pool (stream, media->pool);
g_ptr_array_add (media->streams, stream);
g_mutex_unlock (&media->lock);
g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
NULL);
return stream;
}
/**
* gst_rtsp_media_n_streams:
* @media: a #GstRTSPMedia
*
* Get the number of streams in this media.
*
* Returns: The number of streams.
*/
guint
gst_rtsp_media_n_streams (GstRTSPMedia * media)
{
guint res;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
g_mutex_lock (&media->lock);
res = media->streams->len;
g_mutex_unlock (&media->lock);
return res;
}
/**
* gst_rtsp_media_get_stream:
* @media: a #GstRTSPMedia
* @idx: the stream index
*
* Retrieve the stream with index @idx from @media.
*
* Returns: (transfer none): the #GstRTSPStream at index @idx or %NULL when a stream with
* that index did not exist.
*/
GstRTSPStream *
gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
{
GstRTSPStream *res;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
g_mutex_lock (&media->lock);
if (idx < media->streams->len)
res = g_ptr_array_index (media->streams, idx);
else
res = NULL;
g_mutex_unlock (&media->lock);
return res;
}
/**
* gst_rtsp_media_get_range_string:
* @media: a #GstRTSPMedia
* @play: for the PLAY request
*
* Get the current range as a string.
*
* Returns: The range as a string, g_free() after usage.
*/
gchar *
gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play)
{
gchar *result;
GstRTSPTimeRange range;
g_mutex_lock (&media->lock);
/* make copy */
range = media->range;
if (!play && media->n_active > 0) {
range.min.type = GST_RTSP_TIME_NOW;
range.min.seconds = -1;
}
g_mutex_unlock (&media->lock);
result = gst_rtsp_range_to_string (&range);
return result;
}
/**
* gst_rtsp_media_seek:
* @media: a #GstRTSPMedia
* @range: a #GstRTSPTimeRange
*
* Seek the pipeline to @range.
*
* Returns: %TRUE on success.
*/
gboolean
gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
{
GstSeekFlags flags;
gboolean res;
gint64 start, stop;
GstSeekType start_type, stop_type;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
g_return_val_if_fail (range != NULL, FALSE);
g_rec_mutex_lock (&media->state_lock);
if (!media->seekable)
goto not_seekable;
if (range->unit != GST_RTSP_RANGE_NPT)
goto not_supported;
/* depends on the current playing state of the pipeline. We might need to
* queue this until we get EOS. */
flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_KEY_UNIT;
start_type = stop_type = GST_SEEK_TYPE_NONE;
switch (range->min.type) {
case GST_RTSP_TIME_NOW:
start = -1;
break;
case GST_RTSP_TIME_SECONDS:
/* only seek when something changed */
if (media->range.min.seconds == range->min.seconds) {
start = -1;
} else {
start = range->min.seconds * GST_SECOND;
start_type = GST_SEEK_TYPE_SET;
}
break;
case GST_RTSP_TIME_END:
default:
goto weird_type;
}
switch (range->max.type) {
case GST_RTSP_TIME_SECONDS:
/* only seek when something changed */
if (media->range.max.seconds == range->max.seconds) {
stop = -1;
} else {
stop = range->max.seconds * GST_SECOND;
stop_type = GST_SEEK_TYPE_SET;
}
break;
case GST_RTSP_TIME_END:
stop = -1;
stop_type = GST_SEEK_TYPE_SET;
break;
case GST_RTSP_TIME_NOW:
default:
goto weird_type;
}
if (start != -1 || stop != -1) {
GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
res = gst_element_seek (media->pipeline, 1.0, GST_FORMAT_TIME,
flags, start_type, start, stop_type, stop);
/* and block for the seek to complete */
GST_INFO ("done seeking %d", res);
gst_element_get_state (media->pipeline, NULL, NULL, -1);
GST_INFO ("prerolled again");
collect_media_stats (media);
} else {
GST_INFO ("no seek needed");
res = TRUE;
}
g_rec_mutex_unlock (&media->state_lock);
return res;
/* ERRORS */
not_seekable:
{
g_rec_mutex_unlock (&media->state_lock);
GST_INFO ("pipeline is not seekable");
return TRUE;
}
not_supported:
{
g_rec_mutex_unlock (&media->state_lock);
GST_WARNING ("seek unit %d not supported", range->unit);
return FALSE;
}
weird_type:
{
g_rec_mutex_unlock (&media->state_lock);
GST_WARNING ("weird range type %d not supported", range->min.type);
return FALSE;
}
}
static void
gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
{
g_mutex_lock (&media->lock);
/* never overwrite the error status */
if (media->status != GST_RTSP_MEDIA_STATUS_ERROR)
media->status = status;
GST_DEBUG ("setting new status to %d", status);
g_cond_broadcast (&media->cond);
g_mutex_unlock (&media->lock);
}
static GstRTSPMediaStatus
gst_rtsp_media_get_status (GstRTSPMedia * media)
{
GstRTSPMediaStatus result;
gint64 end_time;
g_mutex_lock (&media->lock);
end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
/* while we are preparing, wait */
while (media->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
GST_DEBUG ("waiting for status change");
if (!g_cond_wait_until (&media->cond, &media->lock, end_time)) {
GST_DEBUG ("timeout, assuming error status");
media->status = GST_RTSP_MEDIA_STATUS_ERROR;
}
}
/* could be success or error */
result = media->status;
GST_DEBUG ("got status %d", result);
g_mutex_unlock (&media->lock);
return result;
}
/* called with state-lock */
static gboolean
default_handle_message (GstRTSPMedia * media, GstMessage * message)
{
GstMessageType type;
type = GST_MESSAGE_TYPE (message);
switch (type) {
case GST_MESSAGE_STATE_CHANGED:
break;
case GST_MESSAGE_BUFFERING:
{
gint percent;
gst_message_parse_buffering (message, &percent);
/* no state management needed for live pipelines */
if (media->is_live)
break;
if (percent == 100) {
/* a 100% message means buffering is done */
media->buffering = FALSE;
/* if the desired state is playing, go back */
if (media->target_state == GST_STATE_PLAYING) {
GST_INFO ("Buffering done, setting pipeline to PLAYING");
gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
} else {
GST_INFO ("Buffering done");
}
} else {
/* buffering busy */
if (media->buffering == FALSE) {
if (media->target_state == GST_STATE_PLAYING) {
/* we were not buffering but PLAYING, PAUSE the pipeline. */
GST_INFO ("Buffering, setting pipeline to PAUSED ...");
gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
} else {
GST_INFO ("Buffering ...");
}
}
media->buffering = TRUE;
}
break;
}
case GST_MESSAGE_LATENCY:
{
gst_bin_recalculate_latency (GST_BIN_CAST (media->pipeline));
break;
}
case GST_MESSAGE_ERROR:
{
GError *gerror;
gchar *debug;
gst_message_parse_error (message, &gerror, &debug);
GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
g_error_free (gerror);
g_free (debug);
gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
break;
}
case GST_MESSAGE_WARNING:
{
GError *gerror;
gchar *debug;
gst_message_parse_warning (message, &gerror, &debug);
GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
g_error_free (gerror);
g_free (debug);
break;
}
case GST_MESSAGE_ELEMENT:
break;
case GST_MESSAGE_STREAM_STATUS:
break;
case GST_MESSAGE_ASYNC_DONE:
if (!media->adding) {
/* when we are dynamically adding pads, the addition of the udpsrc will
* temporarily produce ASYNC_DONE messages. We have to ignore them and
* wait for the final ASYNC_DONE after everything prerolled */
GST_INFO ("%p: got ASYNC_DONE", media);
collect_media_stats (media);
gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
} else {
GST_INFO ("%p: ignoring ASYNC_DONE", media);
}
break;
case GST_MESSAGE_EOS:
GST_INFO ("%p: got EOS", media);
if (media->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
GST_DEBUG ("shutting down after EOS");
finish_unprepare (media);
g_object_unref (media);
}
break;
default:
GST_INFO ("%p: got message type %s", media,
gst_message_type_get_name (type));
break;
}
return TRUE;
}
static gboolean
bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
{
GstRTSPMediaClass *klass;
gboolean ret;
klass = GST_RTSP_MEDIA_GET_CLASS (media);
g_rec_mutex_lock (&media->state_lock);
if (klass->handle_message)
ret = klass->handle_message (media, message);
else
ret = FALSE;
g_rec_mutex_unlock (&media->state_lock);
return ret;
}
/* called from streaming threads */
static void
pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
{
GstRTSPStream *stream;
/* FIXME, element is likely not a payloader, find the payloader here */
stream = gst_rtsp_media_create_stream (media, element, pad);
GST_INFO ("pad added %s:%s, stream %d", GST_DEBUG_PAD_NAME (pad),
stream->idx);
g_rec_mutex_lock (&media->state_lock);
/* we will be adding elements below that will cause ASYNC_DONE to be
* posted in the bus. We want to ignore those messages until the
* pipeline really prerolled. */
media->adding = TRUE;
/* join the element in the PAUSED state because this callback is
* called from the streaming thread and it is PAUSED */
gst_rtsp_stream_join_bin (stream, GST_BIN (media->pipeline),
media->rtpbin, GST_STATE_PAUSED);
media->adding = FALSE;
g_rec_mutex_unlock (&media->state_lock);
}
static void
no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
{
GstElement *fakesink;
g_mutex_lock (&media->lock);
GST_INFO ("no more pads");
if ((fakesink = media->fakesink)) {
gst_object_ref (fakesink);
media->fakesink = NULL;
g_mutex_unlock (&media->lock);
gst_bin_remove (GST_BIN (media->pipeline), fakesink);
gst_element_set_state (fakesink, GST_STATE_NULL);
gst_object_unref (fakesink);
GST_INFO ("removed fakesink");
}
}
/**
* gst_rtsp_media_prepare:
* @media: a #GstRTSPMedia
*
* Prepare @media for streaming. This function will create the pipeline and
* other objects to manage the streaming.
*
* It will preroll the pipeline and collect vital information about the streams
* such as the duration.
*
* Returns: %TRUE on success.
*/
gboolean
gst_rtsp_media_prepare (GstRTSPMedia * media)
{
GstStateChangeReturn ret;
GstRTSPMediaStatus status;
guint i;
GstRTSPMediaClass *klass;
GstBus *bus;
GList *walk;
g_rec_mutex_lock (&media->state_lock);
if (media->status == GST_RTSP_MEDIA_STATUS_PREPARED)
goto was_prepared;
if (media->status == GST_RTSP_MEDIA_STATUS_PREPARING)
goto wait_status;
if (media->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
goto not_unprepared;
if (!media->reusable && media->reused)
goto is_reused;
media->rtpbin = gst_element_factory_make ("rtpbin", NULL);
if (media->rtpbin == NULL)
goto no_rtpbin;
GST_INFO ("preparing media %p", media);
/* reset some variables */
media->is_live = FALSE;
media->seekable = FALSE;
media->buffering = FALSE;
/* we're preparing now */
media->status = GST_RTSP_MEDIA_STATUS_PREPARING;
bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (media->pipeline));
/* add the pipeline bus to our custom mainloop */
media->source = gst_bus_create_watch (bus);
gst_object_unref (bus);
g_source_set_callback (media->source, (GSourceFunc) bus_message, media, NULL);
klass = GST_RTSP_MEDIA_GET_CLASS (media);
media->id = g_source_attach (media->source, klass->context);
/* add stuff to the bin */
gst_bin_add (GST_BIN (media->pipeline), media->rtpbin);
/* link streams we already have, other streams might appear when we have
* dynamic elements */
for (i = 0; i < media->streams->len; i++) {
GstRTSPStream *stream;
stream = g_ptr_array_index (media->streams, i);
gst_rtsp_stream_join_bin (stream, GST_BIN (media->pipeline),
media->rtpbin, GST_STATE_NULL);
}
for (walk = media->dynamic; walk; walk = g_list_next (walk)) {
GstElement *elem = walk->data;
GST_INFO ("adding callbacks for dynamic element %p", elem);
g_signal_connect (elem, "pad-added", (GCallback) pad_added_cb, media);
g_signal_connect (elem, "no-more-pads", (GCallback) no_more_pads_cb, media);
/* we add a fakesink here in order to make the state change async. We remove
* the fakesink again in the no-more-pads callback. */
media->fakesink = gst_element_factory_make ("fakesink", "fakesink");
gst_bin_add (GST_BIN (media->pipeline), media->fakesink);
}
GST_INFO ("setting pipeline to PAUSED for media %p", media);
/* first go to PAUSED */
ret = gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
media->target_state = GST_STATE_PAUSED;
switch (ret) {
case GST_STATE_CHANGE_SUCCESS:
GST_INFO ("SUCCESS state change for media %p", media);
media->seekable = TRUE;
break;
case GST_STATE_CHANGE_ASYNC:
GST_INFO ("ASYNC state change for media %p", media);
media->seekable = TRUE;
break;
case GST_STATE_CHANGE_NO_PREROLL:
/* we need to go to PLAYING */
GST_INFO ("NO_PREROLL state change: live media %p", media);
/* FIXME we disable seeking for live streams for now. We should perform a
* seeking query in preroll instead */
media->seekable = FALSE;
media->is_live = TRUE;
ret = gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
if (ret == GST_STATE_CHANGE_FAILURE)
goto state_failed;
break;
case GST_STATE_CHANGE_FAILURE:
goto state_failed;
}
wait_status:
g_rec_mutex_unlock (&media->state_lock);
/* now wait for all pads to be prerolled, FIXME, we should somehow be
* able to do this async so that we don't block the server thread. */
status = gst_rtsp_media_get_status (media);
if (status == GST_RTSP_MEDIA_STATUS_ERROR)
goto state_failed;
g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
GST_INFO ("object %p is prerolled", media);
return TRUE;
/* OK */
was_prepared:
{
GST_LOG ("media %p was prepared", media);
g_rec_mutex_unlock (&media->state_lock);
return TRUE;
}
/* ERRORS */
not_unprepared:
{
GST_WARNING ("media %p was not unprepared", media);
g_rec_mutex_unlock (&media->state_lock);
return FALSE;
}
is_reused:
{
g_rec_mutex_unlock (&media->state_lock);
GST_WARNING ("can not reuse media %p", media);
return FALSE;
}
no_rtpbin:
{
g_rec_mutex_unlock (&media->state_lock);
GST_WARNING ("no rtpbin element");
g_warning ("failed to create element 'rtpbin', check your installation");
return FALSE;
}
state_failed:
{
GST_WARNING ("failed to preroll pipeline");
gst_rtsp_media_unprepare (media);
g_rec_mutex_unlock (&media->state_lock);
return FALSE;
}
}
/* must be called with state-lock */
static void
finish_unprepare (GstRTSPMedia * media)
{
gint i;
GST_DEBUG ("shutting down");
gst_element_set_state (media->pipeline, GST_STATE_NULL);
for (i = 0; i < media->streams->len; i++) {
GstRTSPStream *stream;
GST_INFO ("Removing elements of stream %d from pipeline", i);
stream = g_ptr_array_index (media->streams, i);
gst_rtsp_stream_leave_bin (stream, GST_BIN (media->pipeline),
media->rtpbin);
}
g_ptr_array_set_size (media->streams, 0);
gst_bin_remove (GST_BIN (media->pipeline), media->rtpbin);
media->rtpbin = NULL;
gst_object_unref (media->pipeline);
media->pipeline = NULL;
media->reused = TRUE;
media->status = GST_RTSP_MEDIA_STATUS_UNPREPARED;
/* when the media is not reusable, this will effectively unref the media and
* recreate it */
g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
}
/* called with state-lock */
static gboolean
default_unprepare (GstRTSPMedia * media)
{
if (media->eos_shutdown) {
GST_DEBUG ("sending EOS for shutdown");
/* ref so that we don't disappear */
g_object_ref (media);
gst_element_send_event (media->pipeline, gst_event_new_eos ());
/* we need to go to playing again for the EOS to propagate, normally in this
* state, nothing is receiving data from us anymore so this is ok. */
gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
media->status = GST_RTSP_MEDIA_STATUS_UNPREPARING;
} else {
finish_unprepare (media);
}
return TRUE;
}
/**
* gst_rtsp_media_unprepare:
* @media: a #GstRTSPMedia
*
* Unprepare @media. After this call, the media should be prepared again before
* it can be used again. If the media is set to be non-reusable, a new instance
* must be created.
*
* Returns: %TRUE on success.
*/
gboolean
gst_rtsp_media_unprepare (GstRTSPMedia * media)
{
gboolean success;
g_rec_mutex_lock (&media->state_lock);
if (media->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
goto was_unprepared;
GST_INFO ("unprepare media %p", media);
media->target_state = GST_STATE_NULL;
success = TRUE;
if (media->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
GstRTSPMediaClass *klass;
klass = GST_RTSP_MEDIA_GET_CLASS (media);
if (klass->unprepare)
success = klass->unprepare (media);
} else {
finish_unprepare (media);
}
g_rec_mutex_unlock (&media->state_lock);
return success;
was_unprepared:
{
g_rec_mutex_unlock (&media->state_lock);
GST_INFO ("media %p was already unprepared", media);
return TRUE;
}
}
/**
* gst_rtsp_media_set_state:
* @media: a #GstRTSPMedia
* @state: the target state of the media
* @transports: a #GPtrArray of #GstRTSPStreamTransport pointers
*
* Set the state of @media to @state and for the transports in @transports.
*
* Returns: %TRUE on success.
*/
gboolean
gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
GPtrArray * transports)
{
gint i;
gboolean add, remove, do_state;
gint old_active;
g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
g_return_val_if_fail (transports != NULL, FALSE);
g_rec_mutex_lock (&media->state_lock);
/* NULL and READY are the same */
if (state == GST_STATE_READY)
state = GST_STATE_NULL;
add = remove = FALSE;
GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
media);
switch (state) {
case GST_STATE_NULL:
case GST_STATE_PAUSED:
/* we're going from PLAYING to PAUSED, READY or NULL, remove */
if (media->target_state == GST_STATE_PLAYING)
remove = TRUE;
break;
case GST_STATE_PLAYING:
/* we're going to PLAYING, add */
add = TRUE;
break;
default:
break;
}
old_active = media->n_active;
for (i = 0; i < transports->len; i++) {
GstRTSPStreamTransport *trans;
/* we need a non-NULL entry in the array */
trans = g_ptr_array_index (transports, i);
if (trans == NULL)
continue;
/* we need a transport */
if (!trans->transport)
continue;
if (add) {
if (gst_rtsp_stream_add_transport (trans->stream, trans))
media->n_active++;
} else if (remove) {
if (gst_rtsp_stream_remove_transport (trans->stream, trans))
media->n_active--;
}
}
/* we just added the first media, do the playing state change */
if (old_active == 0 && add)
do_state = TRUE;
/* if we have no more active media, do the downward state changes */
else if (media->n_active == 0)
do_state = TRUE;
else
do_state = FALSE;
GST_INFO ("state %d active %d media %p do_state %d", state, media->n_active,
media, do_state);
if (media->target_state != state) {
if (do_state) {
if (state == GST_STATE_NULL) {
gst_rtsp_media_unprepare (media);
} else {
GST_INFO ("state %s media %p", gst_element_state_get_name (state),
media);
media->target_state = state;
gst_element_set_state (media->pipeline, state);
}
}
g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
NULL);
}
/* remember where we are */
if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
old_active != media->n_active))
collect_media_stats (media);
g_rec_mutex_unlock (&media->state_lock);
return TRUE;
}