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/* GStreamer RTMP Library
* Copyright (C) 2013 David Schleef <ds@schleef.org>
*
* 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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gio/gio.h>
#include <stdlib.h>
#include <string.h>
#include "rtmpclient.h"
#include "rtmputils.h"
#define GETTEXT_PACKAGE NULL
GstRtmpClient *client;
GstRtmpConnection *connection;
gboolean verbose;
gboolean dump;
static GOptionEntry entries[] = {
{"verbose", 'v', 0, G_OPTION_ARG_NONE, &verbose, "Be verbose", NULL},
{"dump", 'd', 0, G_OPTION_ARG_NONE, &dump, "Dump packets", NULL},
{NULL}
};
static void connect_done (GObject * source, GAsyncResult * result,
gpointer user_data);
static void cmd_connect_done (GstRtmpConnection * connection,
GstRtmpChunk * chunk, const char *command_name, int transaction_id,
GstAmfNode * command_object, GstAmfNode * optional_args,
gpointer user_data);
static void send_connect (void);
static void got_chunk (GstRtmpConnection * connection, GstRtmpChunk * chunk,
gpointer user_data);
static void send_create_stream (void);
static void create_stream_done (GstRtmpConnection * connection,
GstRtmpChunk * chunk, const char *command_name, int transaction_id,
GstAmfNode * command_object, GstAmfNode * optional_args,
gpointer user_data);
static void send_play (void);
static void play_done (GstRtmpConnection * connection, GstRtmpChunk * chunk,
const char *command_name, int transaction_id, GstAmfNode * command_object,
GstAmfNode * optional_args, gpointer user_data);
int
main (int argc, char *argv[])
{
GError *error = NULL;
GOptionContext *context;
GMainLoop *main_loop;
GCancellable *cancellable;
context = g_option_context_new ("- FIXME");
g_option_context_add_main_entries (context, entries, GETTEXT_PACKAGE);
g_option_context_add_group (context, gst_init_get_option_group ());
if (!g_option_context_parse (context, &argc, &argv, &error)) {
g_print ("option parsing failed: %s\n", error->message);
exit (1);
}
g_option_context_free (context);
client = gst_rtmp_client_new ();
gst_rtmp_client_set_server_address (client,
"ec2-54-185-55-241.us-west-2.compute.amazonaws.com");
cancellable = g_cancellable_new ();
main_loop = g_main_loop_new (NULL, TRUE);
gst_rtmp_client_connect_async (client, cancellable, connect_done, client);
g_main_loop_run (main_loop);
exit (0);
}
static void
connect_done (GObject * source, GAsyncResult * result, gpointer user_data)
{
GstRtmpClient *client = user_data;
GError *error = NULL;
gboolean ret;
ret = gst_rtmp_client_connect_finish (client, result, &error);
if (!ret) {
GST_ERROR ("error: %s", error->message);
g_error_free (error);
return;
}
connection = gst_rtmp_client_get_connection (client);
g_signal_connect (connection, "got-chunk", G_CALLBACK (got_chunk), NULL);
send_connect ();
}
static void
got_chunk (GstRtmpConnection * connection, GstRtmpChunk * chunk,
gpointer user_data)
{
gst_rtmp_dump_chunk (chunk, FALSE, TRUE, TRUE);
}
static void
send_connect (void)
{
GstAmfNode *node;
node = gst_amf_node_new (GST_AMF_TYPE_OBJECT);
gst_amf_object_set_string (node, "app", "live");
gst_amf_object_set_string (node, "type", "nonprivate");
gst_amf_object_set_string (node, "tcUrl", "rtmp://localhost:1935/live");
// "fpad": False,
// "capabilities": 15,
// "audioCodecs": 3191,
// "videoCodecs": 252,
// "videoFunction": 1,
gst_rtmp_connection_send_command (connection, 3, "connect", 1, node, NULL,
cmd_connect_done, NULL);
}
static void
cmd_connect_done (GstRtmpConnection * connection, GstRtmpChunk * chunk,
const char *command_name, int transaction_id, GstAmfNode * command_object,
GstAmfNode * optional_args, gpointer user_data)
{
gboolean ret;
ret = FALSE;
if (optional_args) {
const GstAmfNode *n;
n = gst_amf_node_get_object (optional_args, "code");
if (n) {
const char *s;
s = gst_amf_node_get_string (n);
if (strcmp (s, "NetConnection.Connect.Success") == 0) {
ret = TRUE;
}
}
}
if (ret) {
GST_ERROR ("success");
send_create_stream ();
}
}
static void
send_create_stream (void)
{
GstAmfNode *node;
node = gst_amf_node_new (GST_AMF_TYPE_NULL);
gst_rtmp_connection_send_command (connection, 3, "createStream", 2, node,
NULL, create_stream_done, NULL);
}
static void
create_stream_done (GstRtmpConnection * connection, GstRtmpChunk * chunk,
const char *command_name, int transaction_id, GstAmfNode * command_object,
GstAmfNode * optional_args, gpointer user_data)
{
gboolean ret;
int stream_id;
ret = FALSE;
if (optional_args) {
stream_id = gst_amf_node_get_number (optional_args);
ret = TRUE;
}
if (ret) {
GST_ERROR ("createStream success, stream_id=%d", stream_id);
send_play ();
}
}
static void
send_play (void)
{
GstAmfNode *n1;
GstAmfNode *n2;
GstAmfNode *n3;
n1 = gst_amf_node_new (GST_AMF_TYPE_NULL);
n2 = gst_amf_node_new (GST_AMF_TYPE_STRING);
gst_amf_node_set_string (n2, "myStream");
n3 = gst_amf_node_new (GST_AMF_TYPE_NUMBER);
gst_amf_node_set_number (n3, 0);
gst_rtmp_connection_send_command2 (connection, 8, 1, "play", 3, n1, n2, n3,
NULL, play_done, NULL);
}
static void
play_done (GstRtmpConnection * connection, GstRtmpChunk * chunk,
const char *command_name, int transaction_id, GstAmfNode * command_object,
GstAmfNode * optional_args, gpointer user_data)
{
gboolean ret;
int stream_id;
ret = FALSE;
if (optional_args) {
stream_id = gst_amf_node_get_number (optional_args);
ret = TRUE;
}
if (ret) {
GST_ERROR ("play success, stream_id=%d", stream_id);
}
}
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