/* 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-gstrtmp2sink * * The rtmp2sink element sends audio and video streams to an RTMP * server. * * * Example launch line * |[ * gst-launch -v videotestsrc ! x264enc ! flvmux ! rtmp2sink * location=rtmp://server.example.com/live/myStream * ]| * FIXME Describe what the pipeline does. * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include "gstrtmp2sink.h" #include "gstrtmp2locationhandler.h" #include "rtmp/rtmpclient.h" #include "rtmp/rtmpmessage.h" #include #include #include #include GST_DEBUG_CATEGORY_STATIC (gst_rtmp2_sink_debug_category); #define GST_CAT_DEFAULT gst_rtmp2_sink_debug_category /* prototypes */ #define GST_RTMP2_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTMP2_SINK,GstRtmp2Sink)) #define GST_IS_RTMP2_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTMP2_SINK)) typedef struct { GstBaseSink parent_instance; /* properties */ GstRtmpLocation location; gboolean async_connect; guint peak_kbps; /* stuff */ gboolean running, flushing; GMutex lock; GCond cond; GstTask *task; GRecMutex task_lock; GMainLoop *loop; GMainContext *context; GCancellable *cancellable; GstRtmpConnection *connection; guint32 stream_id; GPtrArray *headers; guint64 last_ts, base_ts; /* timestamp fixup */ } GstRtmp2Sink; typedef struct { GstBaseSinkClass parent_class; } GstRtmp2SinkClass; /* GObject virtual functions */ static void gst_rtmp2_sink_set_property (GObject * object, guint property_id, const GValue * value, GParamSpec * pspec); static void gst_rtmp2_sink_get_property (GObject * object, guint property_id, GValue * value, GParamSpec * pspec); static void gst_rtmp2_sink_finalize (GObject * object); static void gst_rtmp2_sink_uri_handler_init (GstURIHandlerInterface * iface); /* GstBaseSink virtual functions */ static gboolean gst_rtmp2_sink_start (GstBaseSink * sink); static gboolean gst_rtmp2_sink_stop (GstBaseSink * sink); static gboolean gst_rtmp2_sink_unlock (GstBaseSink * sink); static gboolean gst_rtmp2_sink_unlock_stop (GstBaseSink * sink); static GstFlowReturn gst_rtmp2_sink_render (GstBaseSink * sink, GstBuffer * buffer); static gboolean gst_rtmp2_sink_set_caps (GstBaseSink * sink, GstCaps * caps); /* Internal API */ static void gst_rtmp2_sink_task_func (gpointer user_data); static void client_connect_done (GObject * source, GAsyncResult * result, gpointer user_data); static void start_publish_done (GObject * source, GAsyncResult * result, gpointer user_data); static void connect_task_done (GObject * object, GAsyncResult * result, gpointer user_data); static void set_pacing_rate (GstRtmp2Sink * self); 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, PROP_PEAK_KBPS, }; /* pad templates */ static GstStaticPadTemplate gst_rtmp2_sink_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("video/x-flv") ); /* class initialization */ G_DEFINE_TYPE_WITH_CODE (GstRtmp2Sink, gst_rtmp2_sink, GST_TYPE_BASE_SINK, G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtmp2_sink_uri_handler_init); G_IMPLEMENT_INTERFACE (GST_TYPE_RTMP_LOCATION_HANDLER, NULL)); static void gst_rtmp2_sink_class_init (GstRtmp2SinkClass * klass) { GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GstBaseSinkClass *base_sink_class = GST_BASE_SINK_CLASS (klass); gst_element_class_add_static_pad_template (GST_ELEMENT_CLASS (klass), &gst_rtmp2_sink_sink_template); gst_element_class_set_static_metadata (GST_ELEMENT_CLASS (klass), "RTMP sink element", "Sink", "Sink element for RTMP streams", "Make.TV, Inc. "); gobject_class->set_property = gst_rtmp2_sink_set_property; gobject_class->get_property = gst_rtmp2_sink_get_property; gobject_class->finalize = gst_rtmp2_sink_finalize; base_sink_class->start = GST_DEBUG_FUNCPTR (gst_rtmp2_sink_start); base_sink_class->stop = GST_DEBUG_FUNCPTR (gst_rtmp2_sink_stop); base_sink_class->unlock = GST_DEBUG_FUNCPTR (gst_rtmp2_sink_unlock); base_sink_class->unlock_stop = GST_DEBUG_FUNCPTR (gst_rtmp2_sink_unlock_stop); base_sink_class->render = GST_DEBUG_FUNCPTR (gst_rtmp2_sink_render); base_sink_class->set_caps = GST_DEBUG_FUNCPTR (gst_rtmp2_sink_set_caps); 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)); g_object_class_install_property (gobject_class, PROP_PEAK_KBPS, g_param_spec_uint ("peak-kbps", "Peak bitrate", "Bitrate in kbit/sec to pace outgoing packets", 0, G_MAXINT / 125, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | GST_PARAM_MUTABLE_PLAYING)); GST_DEBUG_CATEGORY_INIT (gst_rtmp2_sink_debug_category, "rtmp2sink", 0, "debug category for rtmp2sink element"); } static void gst_rtmp2_sink_init (GstRtmp2Sink * self) { self->location.flash_ver = g_strdup ("FMLE/3.0 (compatible; FMSc/1.0)"); self->async_connect = TRUE; g_mutex_init (&self->lock); g_cond_init (&self->cond); self->task = gst_task_new (gst_rtmp2_sink_task_func, self, NULL); g_rec_mutex_init (&self->task_lock); gst_task_set_lock (self->task, &self->task_lock); self->headers = g_ptr_array_new_with_free_func ((GDestroyNotify) gst_mini_object_unref); } static void gst_rtmp2_sink_uri_handler_init (GstURIHandlerInterface * iface) { gst_rtmp_location_handler_implement_uri_handler (iface, GST_URI_SINK); } static void gst_rtmp2_sink_set_property (GObject * object, guint property_id, const GValue * value, GParamSpec * pspec) { GstRtmp2Sink *self = GST_RTMP2_SINK (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; case PROP_PEAK_KBPS: GST_OBJECT_LOCK (self); self->peak_kbps = g_value_get_uint (value); GST_OBJECT_UNLOCK (self); g_mutex_lock (&self->lock); set_pacing_rate (self); g_mutex_unlock (&self->lock); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec); break; } } static void gst_rtmp2_sink_get_property (GObject * object, guint property_id, GValue * value, GParamSpec * pspec) { GstRtmp2Sink *self = GST_RTMP2_SINK (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; case PROP_PEAK_KBPS: GST_OBJECT_LOCK (self); g_value_set_uint (value, self->peak_kbps); GST_OBJECT_UNLOCK (self); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec); break; } } static void gst_rtmp2_sink_finalize (GObject * object) { GstRtmp2Sink *self = GST_RTMP2_SINK (object); g_clear_pointer (&self->headers, g_ptr_array_unref); 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_sink_parent_class)->finalize (object); } static gboolean gst_rtmp2_sink_start (GstBaseSink * sink) { GstRtmp2Sink *self = GST_RTMP2_SINK (sink); 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->last_ts = 0; self->base_ts = 0; 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 (GstRtmp2Sink * 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_sink_stop (GstBaseSink * sink) { GstRtmp2Sink *self = GST_RTMP2_SINK (sink); 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_sink_unlock (GstBaseSink * sink) { GstRtmp2Sink *self = GST_RTMP2_SINK (sink); 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_sink_unlock_stop (GstBaseSink * sink) { GstRtmp2Sink *self = GST_RTMP2_SINK (sink); GST_DEBUG_OBJECT (self, "unlock_stop"); g_mutex_lock (&self->lock); self->flushing = FALSE; g_mutex_unlock (&self->lock); return TRUE; } static gboolean buffer_to_message (GstRtmp2Sink * self, GstBuffer * buffer, GstBuffer ** outbuf) { GstBuffer *message; gsize payload_offset, payload_size; guint64 timestamp; guint32 cstream; GstRtmpMessageType type; { GstMapInfo info; if (G_UNLIKELY (!gst_buffer_map (buffer, &info, GST_MAP_READ))) { GST_ERROR_OBJECT (self, "map failed: %" GST_PTR_FORMAT, buffer); return FALSE; } /* FIXME: This is ugly and only works behind flvmux. * Implement true RTMP muxing. */ if (G_UNLIKELY (info.size >= 4 && memcmp (info.data, "FLV", 3) == 0)) { /* drop the header, we don't need it */ GST_DEBUG_OBJECT (self, "ignoring FLV header: %" GST_PTR_FORMAT, buffer); gst_buffer_unmap (buffer, &info); *outbuf = NULL; return TRUE; } if (G_UNLIKELY (info.size < 11 + 4)) { GST_ERROR_OBJECT (self, "too small: %" GST_PTR_FORMAT, buffer); gst_buffer_unmap (buffer, &info); return FALSE; } /* payload between 11 byte header and 4 byte size footer */ payload_offset = 11; payload_size = info.size - 11 - 4; type = GST_READ_UINT8 (info.data); timestamp = GST_READ_UINT24_BE (info.data + 4); timestamp |= (guint32) GST_READ_UINT8 (info.data + 7) << 24; /* flvmux timestamps roll over after about 49 days */ if (timestamp + self->base_ts + G_MAXINT32 < self->last_ts) { GST_WARNING_OBJECT (self, "Timestamp regression %" G_GUINT64_FORMAT " -> %" G_GUINT64_FORMAT "; assuming overflow", self->last_ts, timestamp + self->base_ts); self->base_ts += G_MAXUINT32; self->base_ts += 1; } else if (timestamp + self->base_ts > self->last_ts + G_MAXINT32) { GST_WARNING_OBJECT (self, "Timestamp jump %" G_GUINT64_FORMAT " -> %" G_GUINT64_FORMAT "; assuming underflow", self->last_ts, timestamp + self->base_ts); if (self->base_ts > 0) { self->base_ts -= G_MAXUINT32; self->base_ts -= 1; } else { GST_WARNING_OBJECT (self, "Cannot regress further;" " forcing timestamp to zero"); timestamp = 0; } } timestamp += self->base_ts; self->last_ts = timestamp; gst_buffer_unmap (buffer, &info); } switch (type) { case GST_RTMP_MESSAGE_TYPE_DATA_AMF0: cstream = 4; break; case GST_RTMP_MESSAGE_TYPE_AUDIO: cstream = 5; break; case GST_RTMP_MESSAGE_TYPE_VIDEO: cstream = 6; break; default: GST_ERROR_OBJECT (self, "unknown tag type %d", type); return FALSE; } /* May not know stream ID yet; set later */ message = gst_rtmp_message_new (type, cstream, 0); message = gst_buffer_append_region (message, gst_buffer_ref (buffer), payload_offset, payload_size); GST_BUFFER_DTS (message) = timestamp * GST_MSECOND; if (type == GST_RTMP_MESSAGE_TYPE_DATA_AMF0) { /* FIXME: HACK: Attach a setDataFrame header. * This should be done using a command. */ static const guint8 header[] = { 0x02, 0x00, 0x0d, 0x40, 0x73, 0x65, 0x74, 0x44, 0x61, 0x74, 0x61, 0x46, 0x72, 0x61, 0x6d, 0x65 }; GstMemory *memory = gst_memory_new_wrapped (GST_MEMORY_FLAG_READONLY, (guint8 *) header, sizeof header, 0, sizeof header, NULL, NULL); gst_buffer_prepend_memory (message, memory); } *outbuf = message; return TRUE; } static gboolean should_drop_header (GstRtmp2Sink * self, GstBuffer * buffer) { guint len; if (G_LIKELY (!GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_HEADER))) { return FALSE; } g_mutex_lock (&self->lock); len = self->headers->len; g_mutex_unlock (&self->lock); /* Drop header buffers when we have streamheader caps */ return len > 0; } static void send_message (GstRtmp2Sink * self, GstBuffer * message) { GstRtmpMeta *meta = gst_buffer_get_rtmp_meta (message); g_return_if_fail (self->stream_id != 0); meta->mstream = self->stream_id; gst_rtmp_connection_queue_message (self->connection, message); } static void send_streamheader (GstRtmp2Sink * self) { guint i; if (G_LIKELY (self->headers->len == 0)) { return; } GST_DEBUG_OBJECT (self, "Sending %u streamheader messages", self->headers->len); for (i = 0; i < self->headers->len; i++) { send_message (self, g_ptr_array_index (self->headers, i)); } /* Steal pointers: suppress free */ g_ptr_array_set_free_func (self->headers, NULL); g_ptr_array_set_size (self->headers, 0); g_ptr_array_set_free_func (self->headers, (GDestroyNotify) gst_mini_object_unref); } static inline gboolean is_running (GstRtmp2Sink * self) { return G_LIKELY (self->running && !self->flushing); } static GstFlowReturn gst_rtmp2_sink_render (GstBaseSink * sink, GstBuffer * buffer) { GstRtmp2Sink *self = GST_RTMP2_SINK (sink); GstBuffer *message; GstFlowReturn ret; if (G_UNLIKELY (should_drop_header (self, buffer))) { GST_DEBUG_OBJECT (self, "Skipping header %" GST_PTR_FORMAT, buffer); return GST_FLOW_OK; } GST_LOG_OBJECT (self, "render %" GST_PTR_FORMAT, buffer); if (G_UNLIKELY (!buffer_to_message (self, buffer, &message))) { GST_ELEMENT_ERROR (self, STREAM, FAILED, ("Failed to convert FLV to RTMP"), ("Failed to convert %" GST_PTR_FORMAT, message)); return GST_FLOW_ERROR; } if (G_UNLIKELY (!message)) { GST_DEBUG_OBJECT (self, "Skipping %" GST_PTR_FORMAT, buffer); return GST_FLOW_OK; } g_mutex_lock (&self->lock); if (G_UNLIKELY (is_running (self) && self->cancellable && gst_task_get_state (self->task) != GST_TASK_STARTED)) { GST_DEBUG_OBJECT (self, "Starting connect"); gst_task_start (self->task); } while (G_UNLIKELY (is_running (self) && !self->connection)) { GST_DEBUG_OBJECT (self, "Waiting for connection"); g_cond_wait (&self->cond, &self->lock); } while (G_UNLIKELY (is_running (self) && self->connection && gst_rtmp_connection_get_num_queued (self->connection) > 3)) { GST_LOG_OBJECT (self, "Waiting for queue"); g_cond_wait (&self->cond, &self->lock); } if (G_UNLIKELY (!is_running (self))) { gst_buffer_unref (message); ret = GST_FLOW_FLUSHING; } else if (G_UNLIKELY (!self->connection)) { gst_buffer_unref (message); /* send_connect_error has sent an ERROR message */ ret = GST_FLOW_ERROR; } else { send_streamheader (self); send_message (self, message); ret = GST_FLOW_OK; } g_mutex_unlock (&self->lock); return ret; } static gboolean add_streamheader (GstRtmp2Sink * self, const GValue * value) { GstBuffer *buffer, *message; g_return_val_if_fail (value, FALSE); if (!GST_VALUE_HOLDS_BUFFER (value)) { GST_ERROR_OBJECT (self, "'streamheader' item of unexpected type '%s'", G_VALUE_TYPE_NAME (value)); return FALSE; } buffer = gst_value_get_buffer (value); if (!buffer_to_message (self, buffer, &message)) { GST_ERROR_OBJECT (self, "Failed to read streamheader %" GST_PTR_FORMAT, buffer); return FALSE; } if (message) { GST_DEBUG_OBJECT (self, "Adding streamheader %" GST_PTR_FORMAT, buffer); g_ptr_array_add (self->headers, message); } else { GST_DEBUG_OBJECT (self, "Skipping streamheader %" GST_PTR_FORMAT, buffer); } return TRUE; } static gboolean gst_rtmp2_sink_set_caps (GstBaseSink * sink, GstCaps * caps) { GstRtmp2Sink *self = GST_RTMP2_SINK (sink); GstStructure *s; const GValue *streamheader; guint i = 0; GST_DEBUG_OBJECT (self, "setcaps %" GST_PTR_FORMAT, caps); g_ptr_array_set_size (self->headers, 0); s = gst_caps_get_structure (caps, 0); streamheader = gst_structure_get_value (s, "streamheader"); if (!streamheader) { GST_DEBUG_OBJECT (self, "'streamheader' field not present"); } else if (GST_VALUE_HOLDS_BUFFER (streamheader)) { GST_DEBUG_OBJECT (self, "'streamheader' field holds buffer"); if (!add_streamheader (self, streamheader)) { return FALSE; } i = 1; } else if (GST_VALUE_HOLDS_ARRAY (streamheader)) { guint size = gst_value_array_get_size (streamheader); GST_DEBUG_OBJECT (self, "'streamheader' field holds array"); for (; i < size; i++) { const GValue *v = gst_value_array_get_value (streamheader, i); if (!add_streamheader (self, v)) { return FALSE; } } } else { GST_ERROR_OBJECT (self, "'streamheader' field has unexpected type '%s'", G_VALUE_TYPE_NAME (streamheader)); return FALSE; } GST_DEBUG_OBJECT (self, "Collected streamheaders: %u buffers -> %u messages", i, self->headers->len); return TRUE; } /* Mainloop task */ static void gst_rtmp2_sink_task_func (gpointer user_data) { GstRtmp2Sink *self = GST_RTMP2_SINK (user_data); GMainContext *context; GMainLoop *loop; GTask *connector; GST_DEBUG_OBJECT (self, "gst_rtmp2_sink_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); 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); 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); g_ptr_array_set_size (self->headers, 0); g_mutex_unlock (&self->lock); GST_DEBUG_OBJECT (self, "gst_rtmp2_sink_task exiting"); } static void client_connect_done (GObject * source, GAsyncResult * result, gpointer user_data) { GTask *task = user_data; GstRtmp2Sink *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_publish_async (connection, self->location.stream, g_task_get_cancellable (task), start_publish_done, task); GST_OBJECT_UNLOCK (self); } static void start_publish_done (GObject * source, GAsyncResult * result, gpointer user_data) { GTask *task = G_TASK (user_data); GstRtmp2Sink *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_publish_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 put_chunk (GstRtmpConnection * connection, gpointer user_data) { GstRtmp2Sink *self = GST_RTMP2_SINK (user_data); g_mutex_lock (&self->lock); g_cond_signal (&self->cond); g_mutex_unlock (&self->lock); } static void error_callback (GstRtmpConnection * connection, GstRtmp2Sink * self) { g_mutex_lock (&self->lock); if (self->cancellable) { g_cancellable_cancel (self->cancellable); } else if (self->loop) { GST_ELEMENT_ERROR (self, RESOURCE, WRITE, ("Connection error"), (NULL)); stop_task (self); } g_mutex_unlock (&self->lock); } static void send_connect_error (GstRtmp2Sink * 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) { GstRtmp2Sink *self = GST_RTMP2_SINK (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) { set_pacing_rate (self); gst_rtmp_connection_set_output_handler (self->connection, put_chunk, g_object_ref (self), g_object_unref); g_signal_connect_object (self->connection, "error", G_CALLBACK (error_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); } static gboolean socket_set_pacing_rate (GSocket * socket, gint pacing_rate, GError ** error) { #ifdef SO_MAX_PACING_RATE if (!g_socket_set_option (socket, SOL_SOCKET, SO_MAX_PACING_RATE, pacing_rate, error)) { g_prefix_error (error, "setsockopt failed: "); return FALSE; } #else if (pacing_rate != -1) { g_set_error (error, G_IO_ERROR, G_IO_ERROR_NOT_SUPPORTED, "SO_MAX_PACING_RATE is not supported"); return FALSE; } #endif return TRUE; } static void set_pacing_rate (GstRtmp2Sink * self) { GError *error = NULL; gint pacing_rate; if (!self->connection) return; GST_OBJECT_LOCK (self); pacing_rate = self->peak_kbps ? self->peak_kbps * 125 : -1; GST_OBJECT_UNLOCK (self); if (socket_set_pacing_rate (gst_rtmp_connection_get_socket (self->connection), pacing_rate, &error)) GST_INFO_OBJECT (self, "Set pacing rate to %d Bps", pacing_rate); else GST_WARNING_OBJECT (self, "Could not set pacing rate: %s", error->message); g_clear_error (&error); }