mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-22 12:00:11 +00:00
b6c577c70c
librtmp allows for attaching arbitrary AMF objects to the end of the connect packet, and this is commonly used for authenticating with servers. Add a new property, extra-connect-args, that mimics librtmp's behavior. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7054>
143 lines
4.4 KiB
C
143 lines
4.4 KiB
C
/* GStreamer RTMP Library
|
|
* Copyright (C) 2017 Make.TV, Inc. <info@make.tv>
|
|
* Contact: Jan Alexander Steffens (heftig) <jsteffens@make.tv>
|
|
*
|
|
* 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.
|
|
*/
|
|
|
|
#ifndef _GST_RTMP_CLIENT_H_
|
|
#define _GST_RTMP_CLIENT_H_
|
|
|
|
#include "rtmpconnection.h"
|
|
|
|
G_BEGIN_DECLS
|
|
|
|
#define GST_TYPE_RTMP_SCHEME (gst_rtmp_scheme_get_type ())
|
|
|
|
typedef enum
|
|
{
|
|
GST_RTMP_SCHEME_RTMP = 0,
|
|
GST_RTMP_SCHEME_RTMPS,
|
|
} GstRtmpScheme;
|
|
|
|
GType gst_rtmp_scheme_get_type (void);
|
|
|
|
GstRtmpScheme gst_rtmp_scheme_from_string (const gchar * string);
|
|
GstRtmpScheme gst_rtmp_scheme_from_uri (const GstUri * uri);
|
|
const gchar * gst_rtmp_scheme_to_string (GstRtmpScheme scheme);
|
|
const gchar * const * gst_rtmp_scheme_get_strings (void);
|
|
guint gst_rtmp_scheme_get_default_port (GstRtmpScheme scheme);
|
|
|
|
|
|
|
|
#define GST_TYPE_RTMP_AUTHMOD (gst_rtmp_authmod_get_type ())
|
|
|
|
typedef enum
|
|
{
|
|
GST_RTMP_AUTHMOD_NONE = 0,
|
|
GST_RTMP_AUTHMOD_AUTO,
|
|
GST_RTMP_AUTHMOD_ADOBE,
|
|
} GstRtmpAuthmod;
|
|
|
|
GType gst_rtmp_authmod_get_type (void);
|
|
|
|
|
|
|
|
#define GST_TYPE_RTMP_STOP_COMMANDS (gst_rtmp_stop_commands_get_type ())
|
|
#define GST_RTMP_DEFAULT_STOP_COMMANDS (GST_RTMP_STOP_COMMANDS_FCUNPUBLISH | \
|
|
GST_RTMP_STOP_COMMANDS_DELETE_STREAM) /* FCUnpublish + deleteStream */
|
|
|
|
/**
|
|
* GstRtmpStopCommands:
|
|
* @GST_RTMP_STOP_COMMANDS_NONE: Don't send any commands
|
|
* @GST_RTMP_STOP_COMMANDS_FCUNPUBLISH: Send FCUnpublish command
|
|
* @GST_RTMP_STOP_COMMANDS_CLOSE_STREAM: Send closeStream command
|
|
* @GST_RTMP_STOP_COMMANDS_DELETE_STREAM: Send deleteStream command
|
|
*
|
|
* Since: 1.18
|
|
*/
|
|
typedef enum
|
|
{
|
|
GST_RTMP_STOP_COMMANDS_NONE = 0,
|
|
GST_RTMP_STOP_COMMANDS_FCUNPUBLISH = (1 << 0),
|
|
GST_RTMP_STOP_COMMANDS_CLOSE_STREAM = (1 << 1),
|
|
GST_RTMP_STOP_COMMANDS_DELETE_STREAM = (1 << 2)
|
|
} GstRtmpStopCommands;
|
|
|
|
GType gst_rtmp_stop_commands_get_type (void);
|
|
|
|
|
|
|
|
typedef struct _GstRtmpLocation
|
|
{
|
|
GstRtmpScheme scheme;
|
|
gchar *host;
|
|
guint port;
|
|
gchar *application;
|
|
gchar *stream;
|
|
gchar *username;
|
|
gchar *password;
|
|
gchar *secure_token;
|
|
gchar *extra_connect_args;
|
|
GstRtmpAuthmod authmod;
|
|
gint timeout;
|
|
GTlsCertificateFlags tls_flags;
|
|
gchar *flash_ver;
|
|
gboolean publish;
|
|
} GstRtmpLocation;
|
|
|
|
void gst_rtmp_location_copy (GstRtmpLocation * dest,
|
|
const GstRtmpLocation * src);
|
|
void gst_rtmp_location_clear (GstRtmpLocation * uri);
|
|
gchar *gst_rtmp_location_get_string (const GstRtmpLocation * location,
|
|
gboolean with_stream);
|
|
|
|
|
|
|
|
void gst_rtmp_client_connect_async (const GstRtmpLocation * location,
|
|
GCancellable * cancellable, GAsyncReadyCallback callback,
|
|
gpointer user_data);
|
|
GstRtmpConnection *gst_rtmp_client_connect_finish (GAsyncResult * result,
|
|
GError ** error);
|
|
void gst_rtmp_client_start_publish_async (GstRtmpConnection * connection,
|
|
const gchar * stream, GCancellable * cancellable,
|
|
GAsyncReadyCallback callback, gpointer user_data);
|
|
gboolean gst_rtmp_client_start_publish_finish (GstRtmpConnection * connection,
|
|
GAsyncResult * result, guint * stream_id, GError ** error);
|
|
|
|
void gst_rtmp_client_start_play_async (GstRtmpConnection * connection,
|
|
const gchar * stream, GCancellable * cancellable,
|
|
GAsyncReadyCallback callback, gpointer user_data);
|
|
gboolean gst_rtmp_client_start_play_finish (GstRtmpConnection * connection,
|
|
GAsyncResult * result, guint * stream_id, GError ** error);
|
|
|
|
void gst_rtmp_client_stop_publish (GstRtmpConnection * connection,
|
|
const gchar * stream, const GstRtmpStopCommands stop_commands);
|
|
|
|
typedef enum
|
|
{
|
|
GST_RTMP_CONN_PARSING_ERROR_INVALID_TYPE,
|
|
GST_RTMP_CONN_PARSING_ERROR_INVALID_VALUE,
|
|
GST_RTMP_CONN_PARSING_ERROR_FAILED_PARSING_DOUBLE,
|
|
GST_RTMP_CONN_PARSING_ERROR_UNSUPPORTED,
|
|
} GstRtmpConnParsingError;
|
|
|
|
GQuark gst_rtmp_conn_parsing_error_quark (void);
|
|
|
|
#define GST_RTMP_CONN_PARSING_ERROR gst_rtmp_conn_parsing_error_quark ()
|
|
|
|
G_END_DECLS
|
|
#endif
|