From dd54e000eab0b8f4cf72cbb4a9b374d03fb1ba58 Mon Sep 17 00:00:00 2001 From: Wim Taymans Date: Wed, 20 Aug 2008 17:30:19 +0000 Subject: [PATCH] gst/rtsp/: Add google RTSP extension, it can only handle udp and responds with unsupported if we do anything else. Fi... Original commit message from CVS: * gst/rtsp/Makefile.am: * gst/rtsp/gstrtsp.c: (plugin_init): * gst/rtsp/gstrtspgoogle.c: (gst_rtsp_google_before_send), (gst_rtsp_google_after_send), (gst_rtsp_google_get_transports), (_do_init), (gst_rtsp_google_base_init), (gst_rtsp_google_class_init), (gst_rtsp_google_init), (gst_rtsp_google_finalize), (gst_rtsp_google_change_state), (gst_rtsp_google_extension_init): * gst/rtsp/gstrtspgoogle.h: Add google RTSP extension, it can only handle udp and responds with unsupported if we do anything else. Fixes #546465. * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_connection_send), (gst_rtspsrc_connection_receive), (gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_create_transports_string), (gst_rtspsrc_setup_streams), (gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_pause): Make transport setup code a bit better using GString. Add some more debug. Check for closed connections before doing anything on them. --- ChangeLog | 23 +++++ gst/rtsp/Makefile.am | 5 +- gst/rtsp/gstrtsp.c | 5 + gst/rtsp/gstrtspgoogle.c | 211 +++++++++++++++++++++++++++++++++++++++ gst/rtsp/gstrtspgoogle.h | 50 ++++++++++ gst/rtsp/gstrtspsrc.c | 74 +++++++++----- 6 files changed, 340 insertions(+), 28 deletions(-) create mode 100644 gst/rtsp/gstrtspgoogle.c create mode 100644 gst/rtsp/gstrtspgoogle.h diff --git a/ChangeLog b/ChangeLog index cbf9ebe87d..feac330715 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,3 +1,26 @@ +2008-08-20 Wim Taymans + + * gst/rtsp/Makefile.am: + * gst/rtsp/gstrtsp.c: (plugin_init): + * gst/rtsp/gstrtspgoogle.c: (gst_rtsp_google_before_send), + (gst_rtsp_google_after_send), (gst_rtsp_google_get_transports), + (_do_init), (gst_rtsp_google_base_init), + (gst_rtsp_google_class_init), (gst_rtsp_google_init), + (gst_rtsp_google_finalize), (gst_rtsp_google_change_state), + (gst_rtsp_google_extension_init): + * gst/rtsp/gstrtspgoogle.h: + Add google RTSP extension, it can only handle udp and responds with + unsupported if we do anything else. Fixes #546465. + + * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_connection_send), + (gst_rtspsrc_connection_receive), (gst_rtspsrc_loop_send_cmd), + (gst_rtspsrc_create_transports_string), + (gst_rtspsrc_setup_streams), (gst_rtspsrc_open), + (gst_rtspsrc_close), (gst_rtspsrc_pause): + Make transport setup code a bit better using GString. + Add some more debug. + Check for closed connections before doing anything on them. + 2008-08-20 Sebastian Dröge * ext/pulse/pulsesrc.c: (gst_pulsesrc_class_init), diff --git a/gst/rtsp/Makefile.am b/gst/rtsp/Makefile.am index 0e65715981..b2e6b84bca 100644 --- a/gst/rtsp/Makefile.am +++ b/gst/rtsp/Makefile.am @@ -1,7 +1,7 @@ plugin_LTLIBRARIES = libgstrtsp.la libgstrtsp_la_SOURCES = gstrtsp.c gstrtspsrc.c \ - gstrtpdec.c gstrtspext.c + gstrtpdec.c gstrtspext.c gstrtspgoogle.c libgstrtsp_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) libgstrtsp_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) \ @@ -14,4 +14,5 @@ libgstrtsp_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) noinst_HEADERS = gstrtspsrc.h \ gstrtsp.h \ gstrtpdec.h \ - gstrtspext.h + gstrtspext.h \ + gstrtspgoogle.h diff --git a/gst/rtsp/gstrtsp.c b/gst/rtsp/gstrtsp.c index 045c03a606..476ebaa5f7 100644 --- a/gst/rtsp/gstrtsp.c +++ b/gst/rtsp/gstrtsp.c @@ -47,6 +47,7 @@ #include "gstrtpdec.h" #include "gstrtspsrc.h" +#include "gstrtspgoogle.h" static gboolean plugin_init (GstPlugin * plugin) @@ -57,6 +58,10 @@ plugin_init (GstPlugin * plugin) if (!gst_element_register (plugin, "rtpdec", GST_RANK_NONE, GST_TYPE_RTP_DEC)) return FALSE; + if (!gst_element_register (plugin, "rtspgoogle", GST_RANK_SECONDARY, + GST_TYPE_RTSP_GOOGLE)) { + return FALSE; + } return TRUE; } diff --git a/gst/rtsp/gstrtspgoogle.c b/gst/rtsp/gstrtspgoogle.c new file mode 100644 index 0000000000..854d69633d --- /dev/null +++ b/gst/rtsp/gstrtspgoogle.c @@ -0,0 +1,211 @@ +/* GStreamer + * Copyright (C) <2008> 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., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ +/* Element-Checklist-Version: 5 */ + +/** + * SECTION:element-rtspgoogle + * + * A Google RTSP extension + * + * Last reviewed on 2008-08-20 (0.10.10) + */ + +#include + +#include + +#include "gstrtspgoogle.h" + +GST_DEBUG_CATEGORY_STATIC (rtspgoogle_debug); +#define GST_CAT_DEFAULT (rtspgoogle_debug) + +/* elementfactory information */ +static const GstElementDetails rtspgoogle_details = +GST_ELEMENT_DETAILS ("Google RTSP Extension", + "Network/Extension/Protocol", + "Extends RTSP so that it can handle Google setup", + "Wim Taymans "); + +#define SERVER_PREFIX "Google RTSP" + +static GstRTSPResult +gst_rtsp_google_before_send (GstRTSPExtension * ext, GstRTSPMessage * request) +{ + GstRTSPGoogle *ctx = (GstRTSPGoogle *) ext; + + GST_DEBUG_OBJECT (ext, "before send"); + + switch (request->type_data.request.method) { + case GST_RTSP_OPTIONS: + { + /* activate ourselves with the first request */ + ctx->active = TRUE; + break; + } + default: + break; + } + return GST_RTSP_OK; +} + +static GstRTSPResult +gst_rtsp_google_after_send (GstRTSPExtension * ext, GstRTSPMessage * req, + GstRTSPMessage * resp) +{ + GstRTSPGoogle *ctx = (GstRTSPGoogle *) ext; + + GST_DEBUG_OBJECT (ext, "after send"); + + switch (req->type_data.request.method) { + case GST_RTSP_OPTIONS: + { + gchar *server = NULL; + + gst_rtsp_message_get_header (resp, GST_RTSP_HDR_SERVER, &server, 0); + if (server && g_str_has_prefix (server, SERVER_PREFIX)) + ctx->active = TRUE; + else + ctx->active = FALSE; + break; + } + default: + break; + } + return GST_RTSP_OK; +} + +static GstRTSPResult +gst_rtsp_google_get_transports (GstRTSPExtension * ext, + GstRTSPLowerTrans protocols, gchar ** transport) +{ + GstRTSPResult res; + GstRTSPGoogle *ctx = (GstRTSPGoogle *) ext; + GString *result; + + if (!ctx->active) + return GST_RTSP_OK; + + /* always only suggest UDP */ + if (protocols & GST_RTSP_LOWER_TRANS_UDP) { + result = g_string_new (""); + + GST_DEBUG_OBJECT (ext, "adding UDP unicast"); + + g_string_append (result, "RTP/AVP"); + g_string_append (result, ";unicast;client_port=%%u1-%%u2"); + + *transport = g_string_free (result, FALSE); + + res = GST_RTSP_OK; + } else { + res = GST_RTSP_ERROR; + } + + return res; +} + +static void gst_rtsp_google_finalize (GObject * object); + +static GstStateChangeReturn gst_rtsp_google_change_state (GstElement * element, + GstStateChange transition); + +static void gst_rtsp_google_extension_init (gpointer g_iface, + gpointer iface_data); + +static void +_do_init (GType rtspgoogle_type) +{ + static const GInterfaceInfo rtspextension_info = { + gst_rtsp_google_extension_init, + NULL, + NULL + }; + + g_type_add_interface_static (rtspgoogle_type, GST_TYPE_RTSP_EXTENSION, + &rtspextension_info); +} + +GST_BOILERPLATE_FULL (GstRTSPGoogle, gst_rtsp_google, GstElement, + GST_TYPE_ELEMENT, _do_init); + +static void +gst_rtsp_google_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + + gst_element_class_set_details (element_class, &rtspgoogle_details); +} + +static void +gst_rtsp_google_class_init (GstRTSPGoogleClass * g_class) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + GstRTSPGoogleClass *klass; + + klass = (GstRTSPGoogleClass *) g_class; + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + + gobject_class->finalize = gst_rtsp_google_finalize; + + gstelement_class->change_state = + GST_DEBUG_FUNCPTR (gst_rtsp_google_change_state); + + GST_DEBUG_CATEGORY_INIT (rtspgoogle_debug, "rtspgoogle", 0, + "Google RTSP extension"); +} + +static void +gst_rtsp_google_init (GstRTSPGoogle * rtspgoogle, GstRTSPGoogleClass * klass) +{ +} + +static void +gst_rtsp_google_finalize (GObject * object) +{ + GstRTSPGoogle *rtspgoogle; + + rtspgoogle = GST_RTSP_GOOGLE (object); + + G_OBJECT_CLASS (parent_class)->finalize (object); +} + +static GstStateChangeReturn +gst_rtsp_google_change_state (GstElement * element, GstStateChange transition) +{ + GstStateChangeReturn ret; + GstRTSPGoogle *rtspgoogle; + + rtspgoogle = GST_RTSP_GOOGLE (element); + + ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); + + return ret; +} + +static void +gst_rtsp_google_extension_init (gpointer g_iface, gpointer iface_data) +{ + GstRTSPExtensionInterface *iface = (GstRTSPExtensionInterface *) g_iface; + + iface->before_send = gst_rtsp_google_before_send; + iface->after_send = gst_rtsp_google_after_send; + iface->get_transports = gst_rtsp_google_get_transports; +} diff --git a/gst/rtsp/gstrtspgoogle.h b/gst/rtsp/gstrtspgoogle.h new file mode 100644 index 0000000000..4d5aa50c26 --- /dev/null +++ b/gst/rtsp/gstrtspgoogle.h @@ -0,0 +1,50 @@ +/* GStreamer + * Copyright (C) <2008> 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., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __GST_RTSP_GOOGLE_H__ +#define __GST_RTSP_GOOGLE_H__ + +#include + +G_BEGIN_DECLS + +#define GST_TYPE_RTSP_GOOGLE (gst_rtsp_google_get_type()) +#define GST_IS_RTSP_GOOGLE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTSP_GOOGLE)) +#define GST_IS_RTSP_GOOGLE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTSP_GOOGLE)) +#define GST_RTSP_GOOGLE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTSP_GOOGLE, GstRTSPGoogle)) +#define GST_RTSP_GOOGLE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTSP_GOOGLE, GstRTSPGoogleClass)) + +typedef struct _GstRTSPGoogle GstRTSPGoogle; +typedef struct _GstRTSPGoogleClass GstRTSPGoogleClass; + +struct _GstRTSPGoogle { + GstElement element; + + gboolean active; +}; + +struct _GstRTSPGoogleClass { + GstElementClass parent_class; +}; + +GType gst_rtsp_google_get_type(void); + +G_END_DECLS + +#endif /* __GST_RTSP_GOOGLE_H__ */ diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index 048786b569..f393697a01 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -1205,7 +1205,10 @@ gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPMessage * message, GstRTSPResult ret; GST_RTSP_CONN_LOCK (src); - ret = gst_rtsp_connection_send (src->connection, message, timeout); + if (src->connection) + ret = gst_rtsp_connection_send (src->connection, message, timeout); + else + ret = GST_RTSP_ERROR; GST_RTSP_CONN_UNLOCK (src); return ret; @@ -1218,7 +1221,10 @@ gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPMessage * message, GstRTSPResult ret; GST_RTSP_CONN_LOCK (src); - ret = gst_rtsp_connection_receive (src->connection, message, timeout); + if (src->connection) + ret = gst_rtsp_connection_receive (src->connection, message, timeout); + else + ret = GST_RTSP_ERROR; GST_RTSP_CONN_UNLOCK (src); return ret; @@ -2919,10 +2925,12 @@ gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gboolean flush) src->loop_cmd = cmd; if (flush) { GST_DEBUG_OBJECT (src, "start connection flush"); - gst_rtsp_connection_flush (src->connection, TRUE); + if (src->connection) + gst_rtsp_connection_flush (src->connection, TRUE); } else { GST_DEBUG_OBJECT (src, "stop connection flush"); - gst_rtsp_connection_flush (src->connection, FALSE); + if (src->connection) + gst_rtsp_connection_flush (src->connection, FALSE); } GST_OBJECT_UNLOCK (src); } @@ -3577,8 +3585,9 @@ static GstRTSPResult gst_rtspsrc_create_transports_string (GstRTSPSrc * src, GstRTSPLowerTrans protocols, gchar ** transports) { - gchar *result; GstRTSPResult res; + GString *result; + gboolean add_udp_str; *transports = NULL; @@ -3594,42 +3603,42 @@ gst_rtspsrc_create_transports_string (GstRTSPSrc * src, if (*transports != NULL) return GST_RTSP_OK; - /* the default RTSP transports */ - result = g_strdup (""); - if (protocols & GST_RTSP_LOWER_TRANS_UDP) { - gchar *new; + /* it's the default but some servers need it */ + add_udp_str = TRUE; + /* the default RTSP transports */ + result = g_string_new (""); + if (protocols & GST_RTSP_LOWER_TRANS_UDP) { GST_DEBUG_OBJECT (src, "adding UDP unicast"); - new = - g_strconcat (result, "RTP/AVP/UDP;unicast;client_port=%%u1-%%u2", NULL); - g_free (result); - result = new; + g_string_append (result, "RTP/AVP"); + if (add_udp_str) + g_string_append (result, "/UDP"); + g_string_append (result, ";unicast;client_port=%%u1-%%u2"); } if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) { - gchar *new; - GST_DEBUG_OBJECT (src, "adding UDP multicast"); /* we don't have to allocate any UDP ports yet, if the selected transport * turns out to be multicast we can create them and join the multicast * group indicated in the transport reply */ - new = g_strconcat (result, result[0] ? "," : "", - "RTP/AVP/UDP;multicast", NULL); - g_free (result); - result = new; + if (result->len > 0) + g_string_append (result, ","); + g_string_append (result, "RTP/AVP"); + if (add_udp_str) + g_string_append (result, "/UDP"); + g_string_append (result, ";multicast"); } if (protocols & GST_RTSP_LOWER_TRANS_TCP) { - gchar *new; - GST_DEBUG_OBJECT (src, "adding TCP"); - new = g_strconcat (result, result[0] ? "," : "", - "RTP/AVP/TCP;unicast;interleaved=%%i1-%%i2", NULL); - g_free (result); - result = new; + if (result->len > 0) + g_string_append (result, ","); + g_string_append (result, "RTP/AVP/TCP;unicast;interleaved=%%i1-%%i2"); } - *transports = result; + *transports = g_string_free (result, FALSE); + + GST_DEBUG_OBJECT (src, "prepared transports %s", GST_STR_NULL (*transports)); return GST_RTSP_OK; @@ -3791,12 +3800,16 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src) if (res < 0) goto setup_transport_failed; + GST_DEBUG_OBJECT (src, "replace ports in %s", GST_STR_NULL (transports)); + /* replace placeholders with real values, this function will optionally * allocate UDP ports and other info needed to execute the setup request */ res = gst_rtspsrc_prepare_transports (stream, &transports); if (res < 0) goto setup_transport_failed; + GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports)); + /* create SETUP request */ res = gst_rtsp_message_init_request (&request, GST_RTSP_SETUP, @@ -4209,6 +4222,7 @@ setup_failed: cleanup_error: { if (src->connection) { + GST_DEBUG_OBJECT (src, "free connection"); gst_rtsp_connection_free (src->connection); src->connection = NULL; } @@ -4266,6 +4280,9 @@ gst_rtspsrc_close (GstRTSPSrc * src) src->task = NULL; } + if (!src->connection) + goto done; + GST_DEBUG_OBJECT (src, "stop connection flush"); gst_rtsp_connection_flush (src->connection, FALSE); @@ -4297,6 +4314,7 @@ gst_rtspsrc_close (GstRTSPSrc * src) gst_rtsp_connection_free (src->connection); src->connection = NULL; +done: /* cleanup */ gst_rtspsrc_cleanup (src); @@ -4594,6 +4612,9 @@ gst_rtspsrc_pause (GstRTSPSrc * src) GST_DEBUG_OBJECT (src, "connection is idle now"); GST_RTSP_CONN_UNLOCK (src); + if (!src->connection) + goto no_connection; + GST_DEBUG_OBJECT (src, "stop connection flush"); gst_rtsp_connection_flush (src->connection, FALSE); @@ -4610,6 +4631,7 @@ gst_rtspsrc_pause (GstRTSPSrc * src) gst_rtsp_message_unset (&request); gst_rtsp_message_unset (&response); +no_connection: src->state = GST_RTSP_STATE_READY; done: