/* GStreamer * Copyright (C) 2014 David Schleef * Copyright (C) 2017 Make.TV, Inc. * Contact: Jan Alexander Steffens (heftig) * * 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 Street, Suite 500, * Boston, MA 02110-1335, USA. */ /** * SECTION:element-gstrtmp2src * * The rtmp2src element receives input streams from an RTMP server. * * * Example launch line * |[ * gst-launch -v rtmp2src ! decodebin ! fakesink * ]| * FIXME Describe what the pipeline does. * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include "gstrtmp2src.h" #include "gstrtmp2locationhandler.h" #include "rtmp/rtmpclient.h" #include "rtmp/rtmpmessage.h" #include #include GST_DEBUG_CATEGORY_STATIC (gst_rtmp2_src_debug_category); #define GST_CAT_DEFAULT gst_rtmp2_src_debug_category /* prototypes */ #define GST_RTMP2_SRC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTMP2_SRC,GstRtmp2Src)) #define GST_IS_RTMP2_SRC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTMP2_SRC)) typedef struct { GstPushSrc parent_instance; /* properties */ GstRtmpLocation location; gboolean async_connect; /* If both self->lock and OBJECT_LOCK are needed, * self->lock must be taken first */ GMutex lock; GCond cond; gboolean running, flushing; GstTask *task; GRecMutex task_lock; GMainLoop *loop; GMainContext *context; GCancellable *cancellable; GstRtmpConnection *connection; guint32 stream_id; GstBuffer *message; gboolean sent_header; GstClockTime last_ts; } GstRtmp2Src; typedef struct { GstPushSrcClass parent_class; } GstRtmp2SrcClass; /* GObject virtual functions */ static void gst_rtmp2_src_set_property (GObject * object, guint property_id, const GValue * value, GParamSpec * pspec); static void gst_rtmp2_src_get_property (GObject * object, guint property_id, GValue * value, GParamSpec * pspec); static void gst_rtmp2_src_finalize (GObject * object); static void gst_rtmp2_src_uri_handler_init (GstURIHandlerInterface * iface); /* GstBaseSrc virtual functions */ static gboolean gst_rtmp2_src_start (GstBaseSrc * src); static gboolean gst_rtmp2_src_stop (GstBaseSrc * src); static gboolean gst_rtmp2_src_unlock (GstBaseSrc * src); static gboolean gst_rtmp2_src_unlock_stop (GstBaseSrc * src); static GstFlowReturn gst_rtmp2_src_create (GstBaseSrc * src, guint64 offset, guint size, GstBuffer ** outbuf); /* Internal API */ static void gst_rtmp2_src_task_func (gpointer user_data); static void client_connect_done (GObject * source, GAsyncResult * result, gpointer user_data); static void start_play_done (GObject * object, GAsyncResult * result, gpointer user_data); static void connect_task_done (GObject * object, GAsyncResult * result, gpointer user_data); enum { PROP_0, PROP_LOCATION, PROP_SCHEME, PROP_HOST, PROP_PORT, PROP_APPLICATION, PROP_STREAM, PROP_SECURE_TOKEN, PROP_USERNAME, PROP_PASSWORD, PROP_AUTHMOD, PROP_TIMEOUT, PROP_TLS_VALIDATION_FLAGS, PROP_ASYNC_CONNECT, }; /* pad templates */ static GstStaticPadTemplate gst_rtmp2_src_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("video/x-flv") ); /* class initialization */ G_DEFINE_TYPE_WITH_CODE (GstRtmp2Src, gst_rtmp2_src, GST_TYPE_PUSH_SRC, G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtmp2_src_uri_handler_init); G_IMPLEMENT_INTERFACE (GST_TYPE_RTMP_LOCATION_HANDLER, NULL)); static void gst_rtmp2_src_class_init (GstRtmp2SrcClass * klass) { GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GstBaseSrcClass *base_src_class = GST_BASE_SRC_CLASS (klass); gst_element_class_add_static_pad_template (GST_ELEMENT_CLASS (klass), &gst_rtmp2_src_src_template); gst_element_class_set_static_metadata (GST_ELEMENT_CLASS (klass), "RTMP source element", "Source", "Source element for RTMP streams", "Make.TV, Inc. "); gobject_class->set_property = gst_rtmp2_src_set_property; gobject_class->get_property = gst_rtmp2_src_get_property; gobject_class->finalize = gst_rtmp2_src_finalize; base_src_class->start = GST_DEBUG_FUNCPTR (gst_rtmp2_src_start); base_src_class->stop = GST_DEBUG_FUNCPTR (gst_rtmp2_src_stop); base_src_class->unlock = GST_DEBUG_FUNCPTR (gst_rtmp2_src_unlock); base_src_class->unlock_stop = GST_DEBUG_FUNCPTR (gst_rtmp2_src_unlock_stop); base_src_class->create = GST_DEBUG_FUNCPTR (gst_rtmp2_src_create); g_object_class_override_property (gobject_class, PROP_LOCATION, "location"); g_object_class_override_property (gobject_class, PROP_SCHEME, "scheme"); g_object_class_override_property (gobject_class, PROP_HOST, "host"); g_object_class_override_property (gobject_class, PROP_PORT, "port"); g_object_class_override_property (gobject_class, PROP_APPLICATION, "application"); g_object_class_override_property (gobject_class, PROP_STREAM, "stream"); g_object_class_override_property (gobject_class, PROP_SECURE_TOKEN, "secure-token"); g_object_class_override_property (gobject_class, PROP_USERNAME, "username"); g_object_class_override_property (gobject_class, PROP_PASSWORD, "password"); g_object_class_override_property (gobject_class, PROP_AUTHMOD, "authmod"); g_object_class_override_property (gobject_class, PROP_TIMEOUT, "timeout"); g_object_class_override_property (gobject_class, PROP_TLS_VALIDATION_FLAGS, "tls-validation-flags"); g_object_class_install_property (gobject_class, PROP_ASYNC_CONNECT, g_param_spec_boolean ("async-connect", "Async connect", "Connect on READY, otherwise on first push", TRUE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); GST_DEBUG_CATEGORY_INIT (gst_rtmp2_src_debug_category, "rtmp2src", 0, "debug category for rtmp2src element"); } static void gst_rtmp2_src_init (GstRtmp2Src * self) { self->async_connect = TRUE; g_mutex_init (&self->lock); g_cond_init (&self->cond); self->task = gst_task_new (gst_rtmp2_src_task_func, self, NULL); g_rec_mutex_init (&self->task_lock); gst_task_set_lock (self->task, &self->task_lock); } static void gst_rtmp2_src_uri_handler_init (GstURIHandlerInterface * iface) { gst_rtmp_location_handler_implement_uri_handler (iface, GST_URI_SRC); } static void gst_rtmp2_src_set_property (GObject * object, guint property_id, const GValue * value, GParamSpec * pspec) { GstRtmp2Src *self = GST_RTMP2_SRC (object); switch (property_id) { case PROP_LOCATION: gst_rtmp_location_handler_set_uri (GST_RTMP_LOCATION_HANDLER (self), g_value_get_string (value)); break; case PROP_SCHEME: GST_OBJECT_LOCK (self); self->location.scheme = g_value_get_enum (value); GST_OBJECT_UNLOCK (self); break; case PROP_HOST: GST_OBJECT_LOCK (self); g_free (self->location.host); self->location.host = g_value_dup_string (value); GST_OBJECT_UNLOCK (self); break; case PROP_PORT: GST_OBJECT_LOCK (self); self->location.port = g_value_get_int (value); GST_OBJECT_UNLOCK (self); break; case PROP_APPLICATION: GST_OBJECT_LOCK (self); g_free (self->location.application); self->location.application = g_value_dup_string (value); GST_OBJECT_UNLOCK (self); break; case PROP_STREAM: GST_OBJECT_LOCK (self); g_free (self->location.stream); self->location.stream = g_value_dup_string (value); GST_OBJECT_UNLOCK (self); break; case PROP_SECURE_TOKEN: GST_OBJECT_LOCK (self); g_free (self->location.secure_token); self->location.secure_token = g_value_dup_string (value); GST_OBJECT_UNLOCK (self); break; case PROP_USERNAME: GST_OBJECT_LOCK (self); g_free (self->location.username); self->location.username = g_value_dup_string (value); GST_OBJECT_UNLOCK (self); break; case PROP_PASSWORD: GST_OBJECT_LOCK (self); g_free (self->location.password); self->location.password = g_value_dup_string (value); GST_OBJECT_UNLOCK (self); break; case PROP_AUTHMOD: GST_OBJECT_LOCK (self); self->location.authmod = g_value_get_enum (value); GST_OBJECT_UNLOCK (self); break; case PROP_TIMEOUT: GST_OBJECT_LOCK (self); self->location.timeout = g_value_get_uint (value); GST_OBJECT_UNLOCK (self); break; case PROP_TLS_VALIDATION_FLAGS: GST_OBJECT_LOCK (self); self->location.tls_flags = g_value_get_flags (value); GST_OBJECT_UNLOCK (self); break; case PROP_ASYNC_CONNECT: GST_OBJECT_LOCK (self); self->async_connect = g_value_get_boolean (value); GST_OBJECT_UNLOCK (self); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec); break; } } static void gst_rtmp2_src_get_property (GObject * object, guint property_id, GValue * value, GParamSpec * pspec) { GstRtmp2Src *self = GST_RTMP2_SRC (object); switch (property_id) { case PROP_LOCATION: GST_OBJECT_LOCK (self); g_value_take_string (value, gst_rtmp_location_get_string (&self->location, TRUE)); GST_OBJECT_UNLOCK (self); break; case PROP_SCHEME: GST_OBJECT_LOCK (self); g_value_set_enum (value, self->location.scheme); GST_OBJECT_UNLOCK (self); break; case PROP_HOST: GST_OBJECT_LOCK (self); g_value_set_string (value, self->location.host); GST_OBJECT_UNLOCK (self); break; case PROP_PORT: GST_OBJECT_LOCK (self); g_value_set_int (value, self->location.port); GST_OBJECT_UNLOCK (self); break; case PROP_APPLICATION: GST_OBJECT_LOCK (self); g_value_set_string (value, self->location.application); GST_OBJECT_UNLOCK (self); break; case PROP_STREAM: GST_OBJECT_LOCK (self); g_value_set_string (value, self->location.stream); GST_OBJECT_UNLOCK (self); break; case PROP_SECURE_TOKEN: GST_OBJECT_LOCK (self); g_value_set_string (value, self->location.secure_token); GST_OBJECT_UNLOCK (self); break; case PROP_USERNAME: GST_OBJECT_LOCK (self); g_value_set_string (value, self->location.username); GST_OBJECT_UNLOCK (self); break; case PROP_PASSWORD: GST_OBJECT_LOCK (self); g_value_set_string (value, self->location.password); GST_OBJECT_UNLOCK (self); break; case PROP_AUTHMOD: GST_OBJECT_LOCK (self); g_value_set_enum (value, self->location.authmod); GST_OBJECT_UNLOCK (self); break; case PROP_TIMEOUT: GST_OBJECT_LOCK (self); g_value_set_uint (value, self->location.timeout); GST_OBJECT_UNLOCK (self); break; case PROP_TLS_VALIDATION_FLAGS: GST_OBJECT_LOCK (self); g_value_set_flags (value, self->location.tls_flags); GST_OBJECT_UNLOCK (self); break; case PROP_ASYNC_CONNECT: GST_OBJECT_LOCK (self); g_value_set_boolean (value, self->async_connect); GST_OBJECT_UNLOCK (self); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec); break; } } static void gst_rtmp2_src_finalize (GObject * object) { GstRtmp2Src *self = GST_RTMP2_SRC (object); gst_buffer_replace (&self->message, NULL); g_clear_object (&self->cancellable); g_clear_object (&self->connection); g_clear_object (&self->task); g_rec_mutex_clear (&self->task_lock); g_mutex_clear (&self->lock); g_cond_clear (&self->cond); gst_rtmp_location_clear (&self->location); G_OBJECT_CLASS (gst_rtmp2_src_parent_class)->finalize (object); } static gboolean gst_rtmp2_src_start (GstBaseSrc * src) { GstRtmp2Src *self = GST_RTMP2_SRC (src); gboolean async; GST_OBJECT_LOCK (self); async = self->async_connect; GST_OBJECT_UNLOCK (self); GST_INFO_OBJECT (self, "Starting (%s)", async ? "async" : "delayed"); g_clear_object (&self->cancellable); self->running = TRUE; self->cancellable = g_cancellable_new (); self->stream_id = 0; self->sent_header = FALSE; self->last_ts = GST_CLOCK_TIME_NONE; if (async) { gst_task_start (self->task); } return TRUE; } static gboolean quit_invoker (gpointer user_data) { g_main_loop_quit (user_data); return G_SOURCE_REMOVE; } static void stop_task (GstRtmp2Src * self) { gst_task_stop (self->task); self->running = FALSE; if (self->cancellable) { GST_DEBUG_OBJECT (self, "Cancelling"); g_cancellable_cancel (self->cancellable); } if (self->loop) { GST_DEBUG_OBJECT (self, "Stopping loop"); g_main_context_invoke_full (self->context, G_PRIORITY_DEFAULT_IDLE, quit_invoker, g_main_loop_ref (self->loop), (GDestroyNotify) g_main_loop_unref); } g_cond_broadcast (&self->cond); } static gboolean gst_rtmp2_src_stop (GstBaseSrc * src) { GstRtmp2Src *self = GST_RTMP2_SRC (src); GST_DEBUG_OBJECT (self, "stop"); g_mutex_lock (&self->lock); stop_task (self); g_mutex_unlock (&self->lock); gst_task_join (self->task); return TRUE; } static gboolean gst_rtmp2_src_unlock (GstBaseSrc * src) { GstRtmp2Src *self = GST_RTMP2_SRC (src); GST_DEBUG_OBJECT (self, "unlock"); g_mutex_lock (&self->lock); self->flushing = TRUE; g_cond_broadcast (&self->cond); g_mutex_unlock (&self->lock); return TRUE; } static gboolean gst_rtmp2_src_unlock_stop (GstBaseSrc * src) { GstRtmp2Src *self = GST_RTMP2_SRC (src); GST_DEBUG_OBJECT (self, "unlock_stop"); g_mutex_lock (&self->lock); self->flushing = FALSE; g_mutex_unlock (&self->lock); return TRUE; } static GstFlowReturn gst_rtmp2_src_create (GstBaseSrc * src, guint64 offset, guint size, GstBuffer ** outbuf) { GstRtmp2Src *self = GST_RTMP2_SRC (src); GstBuffer *message, *buffer; GstRtmpMeta *meta; guint32 timestamp = 0; static const guint8 flv_header_data[] = { 0x46, 0x4c, 0x56, 0x01, 0x01, 0x00, 0x00, 0x00, 0x09, 0x00, 0x00, 0x00, 0x00, }; GST_LOG_OBJECT (self, "create"); g_mutex_lock (&self->lock); if (self->running) { gst_task_start (self->task); } while (!self->message) { if (!self->running) { g_mutex_unlock (&self->lock); return GST_FLOW_EOS; } if (self->flushing) { g_mutex_unlock (&self->lock); return GST_FLOW_FLUSHING; } g_cond_wait (&self->cond, &self->lock); } message = self->message; self->message = NULL; g_cond_signal (&self->cond); g_mutex_unlock (&self->lock); meta = gst_buffer_get_rtmp_meta (message); if (!meta) { GST_ELEMENT_ERROR (self, CORE, FAILED, ("Internal error: No RTMP meta on buffer"), ("No RTMP meta on %" GST_PTR_FORMAT, message)); gst_buffer_unref (message); return GST_FLOW_ERROR; } if (GST_BUFFER_DTS_IS_VALID (message)) { GstClockTime last_ts = self->last_ts, ts = GST_BUFFER_DTS (message); if (GST_CLOCK_TIME_IS_VALID (last_ts) && last_ts > ts) { GST_LOG_OBJECT (self, "Timestamp regression: %" GST_TIME_FORMAT " > %" GST_TIME_FORMAT, GST_TIME_ARGS (last_ts), GST_TIME_ARGS (ts)); } self->last_ts = ts; timestamp = ts / GST_MSECOND; } buffer = gst_buffer_copy_region (message, GST_BUFFER_COPY_MEMORY, 0, -1); { guint8 *tag_header = g_malloc (11); GstMemory *memory = gst_memory_new_wrapped (0, tag_header, 11, 0, 11, tag_header, g_free); GST_WRITE_UINT8 (tag_header, meta->type); GST_WRITE_UINT24_BE (tag_header + 1, meta->size); GST_WRITE_UINT24_BE (tag_header + 4, timestamp); GST_WRITE_UINT8 (tag_header + 7, timestamp >> 24); GST_WRITE_UINT24_BE (tag_header + 8, 0); gst_buffer_prepend_memory (buffer, memory); } { guint8 *tag_footer = g_malloc (4); GstMemory *memory = gst_memory_new_wrapped (0, tag_footer, 4, 0, 4, tag_footer, g_free); GST_WRITE_UINT32_BE (tag_footer, meta->size + 11); gst_buffer_append_memory (buffer, memory); } if (!self->sent_header) { GstMemory *memory = gst_memory_new_wrapped (GST_MEMORY_FLAG_READONLY, (guint8 *) flv_header_data, sizeof flv_header_data, 0, sizeof flv_header_data, NULL, NULL); gst_buffer_prepend_memory (buffer, memory); self->sent_header = TRUE; } *outbuf = buffer; gst_buffer_unref (message); return GST_FLOW_OK; } /* Mainloop task */ static void gst_rtmp2_src_task_func (gpointer user_data) { GstRtmp2Src *self = GST_RTMP2_SRC (user_data); GMainContext *context; GMainLoop *loop; GTask *connector; GST_DEBUG_OBJECT (self, "gst_rtmp2_src_task starting"); g_mutex_lock (&self->lock); context = self->context = g_main_context_new (); g_main_context_push_thread_default (context); loop = self->loop = g_main_loop_new (context, TRUE); connector = g_task_new (self, self->cancellable, connect_task_done, NULL); GST_OBJECT_LOCK (self); gst_rtmp_client_connect_async (&self->location, self->cancellable, client_connect_done, connector); GST_OBJECT_UNLOCK (self); /* Run loop */ g_mutex_unlock (&self->lock); g_main_loop_run (loop); g_mutex_lock (&self->lock); g_clear_pointer (&self->loop, g_main_loop_unref); g_clear_pointer (&self->connection, gst_rtmp_connection_close_and_unref); g_cond_broadcast (&self->cond); /* Run loop cleanup */ g_mutex_unlock (&self->lock); while (g_main_context_pending (context)) { GST_DEBUG_OBJECT (self, "iterating main context to clean up"); g_main_context_iteration (context, FALSE); } g_main_context_pop_thread_default (context); g_mutex_lock (&self->lock); g_clear_pointer (&self->context, g_main_context_unref); gst_buffer_replace (&self->message, NULL); g_mutex_unlock (&self->lock); GST_DEBUG_OBJECT (self, "gst_rtmp2_src_task exiting"); } static void client_connect_done (GObject * source, GAsyncResult * result, gpointer user_data) { GTask *task = user_data; GstRtmp2Src *self = g_task_get_source_object (task); GError *error = NULL; GstRtmpConnection *connection; connection = gst_rtmp_client_connect_finish (result, &error); if (!connection) { g_task_return_error (task, error); g_object_unref (task); return; } g_task_set_task_data (task, connection, g_object_unref); if (g_task_return_error_if_cancelled (task)) { g_object_unref (task); return; } GST_OBJECT_LOCK (self); gst_rtmp_client_start_play_async (connection, self->location.stream, g_task_get_cancellable (task), start_play_done, task); GST_OBJECT_UNLOCK (self); } static void start_play_done (GObject * source, GAsyncResult * result, gpointer user_data) { GTask *task = G_TASK (user_data); GstRtmp2Src *self = g_task_get_source_object (task); GstRtmpConnection *connection = g_task_get_task_data (task); GError *error = NULL; if (g_task_return_error_if_cancelled (task)) { g_object_unref (task); return; } if (gst_rtmp_client_start_play_finish (connection, result, &self->stream_id, &error)) { g_task_return_pointer (task, g_object_ref (connection), gst_rtmp_connection_close_and_unref); } else { g_task_return_error (task, error); } g_task_set_task_data (task, NULL, NULL); g_object_unref (task); } static void got_message (GstRtmpConnection * connection, GstBuffer * buffer, gpointer user_data) { GstRtmp2Src *self = GST_RTMP2_SRC (user_data); GstRtmpMeta *meta = gst_buffer_get_rtmp_meta (buffer); guint32 min_size = 1; g_return_if_fail (meta); if (meta->mstream != self->stream_id) { GST_DEBUG_OBJECT (self, "Ignoring %s message with stream %" G_GUINT32_FORMAT " != %" G_GUINT32_FORMAT, gst_rtmp_message_type_get_nick (meta->type), meta->mstream, self->stream_id); return; } switch (meta->type) { case GST_RTMP_MESSAGE_TYPE_VIDEO: min_size = 6; break; case GST_RTMP_MESSAGE_TYPE_AUDIO: min_size = 2; break; case GST_RTMP_MESSAGE_TYPE_DATA_AMF0: break; default: GST_DEBUG_OBJECT (self, "Ignoring %s message, wrong type", gst_rtmp_message_type_get_nick (meta->type)); return; } if (meta->size < min_size) { GST_DEBUG_OBJECT (self, "Ignoring too small %s message (%" G_GUINT32_FORMAT " < %" G_GUINT32_FORMAT ")", gst_rtmp_message_type_get_nick (meta->type), meta->size, min_size); return; } g_mutex_lock (&self->lock); while (self->message) { if (!self->running) { goto out; } g_cond_wait (&self->cond, &self->lock); } self->message = gst_buffer_ref (buffer); g_cond_signal (&self->cond); out: g_mutex_unlock (&self->lock); return; } static void error_callback (GstRtmpConnection * connection, GstRtmp2Src * self) { g_mutex_lock (&self->lock); if (self->cancellable) { g_cancellable_cancel (self->cancellable); } else if (self->loop) { GST_INFO_OBJECT (self, "Connection error"); stop_task (self); } g_mutex_unlock (&self->lock); } static void control_callback (GstRtmpConnection * connection, gint uc_type, guint stream_id, GstRtmp2Src * self) { GST_INFO_OBJECT (self, "stream %u got %s", stream_id, gst_rtmp_user_control_type_get_nick (uc_type)); if (uc_type == GST_RTMP_USER_CONTROL_TYPE_STREAM_EOF && stream_id == 1) { GST_INFO_OBJECT (self, "went EOS"); stop_task (self); } } static void send_connect_error (GstRtmp2Src * self, GError * error) { if (!error) { GST_ERROR_OBJECT (self, "Connect failed with NULL error"); GST_ELEMENT_ERROR (self, RESOURCE, FAILED, ("Failed to connect"), (NULL)); return; } if (g_error_matches (error, G_IO_ERROR, G_IO_ERROR_CANCELLED)) { GST_DEBUG_OBJECT (self, "Connection was cancelled (%s)", GST_STR_NULL (error->message)); return; } GST_ERROR_OBJECT (self, "Failed to connect (%s:%d): %s", g_quark_to_string (error->domain), error->code, GST_STR_NULL (error->message)); if (g_error_matches (error, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED)) { GST_ELEMENT_ERROR (self, RESOURCE, NOT_AUTHORIZED, ("Not authorized to connect"), ("%s", GST_STR_NULL (error->message))); } else if (g_error_matches (error, G_IO_ERROR, G_IO_ERROR_CONNECTION_REFUSED)) { GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, ("Could not connect"), ("%s", GST_STR_NULL (error->message))); } else { GST_ELEMENT_ERROR (self, RESOURCE, FAILED, ("Failed to connect"), ("error %s:%d: %s", g_quark_to_string (error->domain), error->code, GST_STR_NULL (error->message))); } } static void connect_task_done (GObject * object, GAsyncResult * result, gpointer user_data) { GstRtmp2Src *self = GST_RTMP2_SRC (object); GTask *task = G_TASK (result); GError *error = NULL; g_mutex_lock (&self->lock); g_warn_if_fail (g_task_is_valid (task, object)); if (self->cancellable == g_task_get_cancellable (task)) { g_clear_object (&self->cancellable); } self->connection = g_task_propagate_pointer (task, &error); if (self->connection) { gst_rtmp_connection_set_input_handler (self->connection, got_message, g_object_ref (self), g_object_unref); g_signal_connect_object (self->connection, "error", G_CALLBACK (error_callback), self, 0); g_signal_connect_object (self->connection, "stream-control", G_CALLBACK (control_callback), self, 0); } else { send_connect_error (self, error); stop_task (self); g_error_free (error); } g_cond_broadcast (&self->cond); g_mutex_unlock (&self->lock); }