mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-14 13:21:28 +00:00
9a2828c216
https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1285 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1256>
1386 lines
40 KiB
C
1386 lines
40 KiB
C
/* GStreamer RTMP Library
|
|
* Copyright (C) 2013 David Schleef <ds@schleef.org>
|
|
* 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 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 <string.h>
|
|
#include "rtmpclient.h"
|
|
#include "rtmphandshake.h"
|
|
#include "rtmpmessage.h"
|
|
#include "rtmputils.h"
|
|
|
|
GST_DEBUG_CATEGORY_STATIC (gst_rtmp_client_debug_category);
|
|
#define GST_CAT_DEFAULT gst_rtmp_client_debug_category
|
|
|
|
static void send_connect_done (const gchar * command_name, GPtrArray * args,
|
|
gpointer user_data);
|
|
static void create_stream_done (const gchar * command_name, GPtrArray * args,
|
|
gpointer user_data);
|
|
static void on_publish_or_play_status (const gchar * command_name,
|
|
GPtrArray * args, gpointer user_data);
|
|
|
|
static void
|
|
init_debug (void)
|
|
{
|
|
static volatile gsize done = 0;
|
|
if (g_once_init_enter (&done)) {
|
|
GST_DEBUG_CATEGORY_INIT (gst_rtmp_client_debug_category,
|
|
"rtmpclient", 0, "debug category for the rtmp client");
|
|
GST_DEBUG_REGISTER_FUNCPTR (send_connect_done);
|
|
GST_DEBUG_REGISTER_FUNCPTR (create_stream_done);
|
|
GST_DEBUG_REGISTER_FUNCPTR (on_publish_or_play_status);
|
|
g_once_init_leave (&done, 1);
|
|
}
|
|
}
|
|
|
|
static const gchar *scheme_strings[] = {
|
|
"rtmp",
|
|
"rtmps",
|
|
NULL
|
|
};
|
|
|
|
#define NUM_SCHEMES (G_N_ELEMENTS (scheme_strings) - 1)
|
|
|
|
GType
|
|
gst_rtmp_scheme_get_type (void)
|
|
{
|
|
static volatile gsize scheme_type = 0;
|
|
static const GEnumValue scheme[] = {
|
|
{GST_RTMP_SCHEME_RTMP, "GST_RTMP_SCHEME_RTMP", "rtmp"},
|
|
{GST_RTMP_SCHEME_RTMPS, "GST_RTMP_SCHEME_RTMPS", "rtmps"},
|
|
{0, NULL, NULL},
|
|
};
|
|
|
|
if (g_once_init_enter (&scheme_type)) {
|
|
GType tmp = g_enum_register_static ("GstRtmpScheme", scheme);
|
|
g_once_init_leave (&scheme_type, tmp);
|
|
}
|
|
|
|
return (GType) scheme_type;
|
|
}
|
|
|
|
GstRtmpScheme
|
|
gst_rtmp_scheme_from_string (const gchar * string)
|
|
{
|
|
if (string) {
|
|
gint value;
|
|
|
|
for (value = 0; value < NUM_SCHEMES; value++) {
|
|
if (strcmp (scheme_strings[value], string) == 0) {
|
|
return value;
|
|
}
|
|
}
|
|
}
|
|
|
|
return -1;
|
|
}
|
|
|
|
GstRtmpScheme
|
|
gst_rtmp_scheme_from_uri (const GstUri * uri)
|
|
{
|
|
const gchar *scheme = gst_uri_get_scheme (uri);
|
|
if (!scheme) {
|
|
return GST_RTMP_SCHEME_RTMP;
|
|
}
|
|
|
|
return gst_rtmp_scheme_from_string (scheme);
|
|
}
|
|
|
|
const gchar *
|
|
gst_rtmp_scheme_to_string (GstRtmpScheme scheme)
|
|
{
|
|
if (scheme >= 0 && scheme < NUM_SCHEMES) {
|
|
return scheme_strings[scheme];
|
|
}
|
|
|
|
return "invalid";
|
|
}
|
|
|
|
const gchar *const *
|
|
gst_rtmp_scheme_get_strings (void)
|
|
{
|
|
return scheme_strings;
|
|
}
|
|
|
|
guint
|
|
gst_rtmp_scheme_get_default_port (GstRtmpScheme scheme)
|
|
{
|
|
switch (scheme) {
|
|
case GST_RTMP_SCHEME_RTMP:
|
|
return 1935;
|
|
|
|
case GST_RTMP_SCHEME_RTMPS:
|
|
return 443;
|
|
|
|
default:
|
|
g_return_val_if_reached (0);
|
|
}
|
|
}
|
|
|
|
GType
|
|
gst_rtmp_authmod_get_type (void)
|
|
{
|
|
static volatile gsize authmod_type = 0;
|
|
static const GEnumValue authmod[] = {
|
|
{GST_RTMP_AUTHMOD_NONE, "GST_RTMP_AUTHMOD_NONE", "none"},
|
|
{GST_RTMP_AUTHMOD_AUTO, "GST_RTMP_AUTHMOD_AUTO", "auto"},
|
|
{GST_RTMP_AUTHMOD_ADOBE, "GST_RTMP_AUTHMOD_ADOBE", "adobe"},
|
|
{0, NULL, NULL},
|
|
};
|
|
|
|
if (g_once_init_enter (&authmod_type)) {
|
|
GType tmp = g_enum_register_static ("GstRtmpAuthmod", authmod);
|
|
g_once_init_leave (&authmod_type, tmp);
|
|
}
|
|
|
|
return (GType) authmod_type;
|
|
}
|
|
|
|
static const gchar *
|
|
gst_rtmp_authmod_get_nick (GstRtmpAuthmod value)
|
|
{
|
|
GEnumClass *klass = g_type_class_peek (GST_TYPE_RTMP_AUTHMOD);
|
|
GEnumValue *ev = klass ? g_enum_get_value (klass, value) : NULL;
|
|
return ev ? ev->value_nick : "(unknown)";
|
|
}
|
|
|
|
void
|
|
gst_rtmp_location_copy (GstRtmpLocation * dest, const GstRtmpLocation * src)
|
|
{
|
|
g_return_if_fail (dest);
|
|
g_return_if_fail (src);
|
|
|
|
dest->scheme = src->scheme;
|
|
dest->host = g_strdup (src->host);
|
|
dest->port = src->port;
|
|
dest->application = g_strdup (src->application);
|
|
dest->stream = g_strdup (src->stream);
|
|
dest->username = g_strdup (src->username);
|
|
dest->password = g_strdup (src->password);
|
|
dest->secure_token = g_strdup (src->secure_token);
|
|
dest->authmod = src->authmod;
|
|
dest->timeout = src->timeout;
|
|
dest->tls_flags = src->tls_flags;
|
|
dest->flash_ver = g_strdup (src->flash_ver);
|
|
dest->publish = src->publish;
|
|
}
|
|
|
|
void
|
|
gst_rtmp_location_clear (GstRtmpLocation * location)
|
|
{
|
|
g_return_if_fail (location);
|
|
|
|
g_clear_pointer (&location->host, g_free);
|
|
location->port = 0;
|
|
g_clear_pointer (&location->application, g_free);
|
|
g_clear_pointer (&location->stream, g_free);
|
|
g_clear_pointer (&location->username, g_free);
|
|
g_clear_pointer (&location->password, g_free);
|
|
g_clear_pointer (&location->secure_token, g_free);
|
|
g_clear_pointer (&location->flash_ver, g_free);
|
|
location->publish = FALSE;
|
|
}
|
|
|
|
gchar *
|
|
gst_rtmp_location_get_string (const GstRtmpLocation * location,
|
|
gboolean with_stream)
|
|
{
|
|
GstUri *uri;
|
|
gchar *base, *string;
|
|
const gchar *scheme_string;
|
|
guint default_port;
|
|
|
|
g_return_val_if_fail (location, NULL);
|
|
|
|
scheme_string = gst_rtmp_scheme_to_string (location->scheme);
|
|
default_port = gst_rtmp_scheme_get_default_port (location->scheme);
|
|
|
|
uri = gst_uri_new (scheme_string, NULL, location->host,
|
|
location->port == default_port ? GST_URI_NO_PORT : location->port, "/",
|
|
NULL, NULL);
|
|
base = gst_uri_to_string (uri);
|
|
|
|
string = g_strconcat (base, location->application, with_stream ? "/" : NULL,
|
|
location->stream, NULL);
|
|
|
|
g_free (base);
|
|
gst_uri_unref (uri);
|
|
|
|
return string;
|
|
}
|
|
|
|
/* Flag values for the audioCodecs property,
|
|
* rtmp_specification_1.0.pdf page 32 */
|
|
enum
|
|
{
|
|
SUPPORT_SND_NONE = 0x001, /* Raw sound, no compression */
|
|
SUPPORT_SND_ADPCM = 0x002, /* ADPCM compression */
|
|
SUPPORT_SND_MP3 = 0x004, /* mp3 compression */
|
|
SUPPORT_SND_INTEL = 0x008, /* Not used */
|
|
SUPPORT_SND_UNUSED = 0x010, /* Not used */
|
|
SUPPORT_SND_NELLY8 = 0x020, /* NellyMoser at 8-kHz compression */
|
|
SUPPORT_SND_NELLY = 0x040, /* NellyMoser compression
|
|
* (5, 11, 22, and 44 kHz) */
|
|
SUPPORT_SND_G711A = 0x080, /* G711A sound compression
|
|
* (Flash Media Server only) */
|
|
SUPPORT_SND_G711U = 0x100, /* G711U sound compression
|
|
* (Flash Media Server only) */
|
|
SUPPORT_SND_NELLY16 = 0x200, /* NellyMoser at 16-kHz compression */
|
|
SUPPORT_SND_AAC = 0x400, /* Advanced audio coding (AAC) codec */
|
|
SUPPORT_SND_SPEEX = 0x800, /* Speex Audio */
|
|
SUPPORT_SND_ALL = 0xFFF, /* All RTMP-supported audio codecs */
|
|
};
|
|
|
|
/* audioCodecs value sent by libavformat. All "used" codecs. */
|
|
#define GST_RTMP_AUDIOCODECS \
|
|
(SUPPORT_SND_ALL & ~SUPPORT_SND_INTEL & ~SUPPORT_SND_UNUSED)
|
|
G_STATIC_ASSERT (GST_RTMP_AUDIOCODECS == 4071); /* libavformat's magic number */
|
|
|
|
/* Flag values for the videoCodecs property,
|
|
* rtmp_specification_1.0.pdf page 32 */
|
|
enum
|
|
{
|
|
SUPPORT_VID_UNUSED = 0x01, /* Obsolete value */
|
|
SUPPORT_VID_JPEG = 0x02, /* Obsolete value */
|
|
SUPPORT_VID_SORENSON = 0x04, /* Sorenson Flash video */
|
|
SUPPORT_VID_HOMEBREW = 0x08, /* V1 screen sharing */
|
|
SUPPORT_VID_VP6 = 0x10, /* On2 video (Flash 8+) */
|
|
SUPPORT_VID_VP6ALPHA = 0x20, /* On2 video with alpha channel */
|
|
SUPPORT_VID_HOMEBREWV = 0x40, /* Screen sharing version 2 (Flash 8+) */
|
|
SUPPORT_VID_H264 = 0x80, /* H264 video */
|
|
SUPPORT_VID_ALL = 0xFF, /* All RTMP-supported video codecs */
|
|
};
|
|
|
|
/* videoCodecs value sent by libavformat. All non-obsolete codecs. */
|
|
#define GST_RTMP_VIDEOCODECS \
|
|
(SUPPORT_VID_ALL & ~SUPPORT_VID_UNUSED & ~SUPPORT_VID_JPEG)
|
|
G_STATIC_ASSERT (GST_RTMP_VIDEOCODECS == 252); /* libavformat's magic number */
|
|
|
|
/* Flag values for the videoFunction property,
|
|
* rtmp_specification_1.0.pdf page 32 */
|
|
enum
|
|
{
|
|
/* Indicates that the client can perform frame-accurate seeks. */
|
|
SUPPORT_VID_CLIENT_SEEK = 1,
|
|
};
|
|
|
|
/* videoFunction value sent by libavformat */
|
|
#define GST_RTMP_VIDEOFUNCTION (SUPPORT_VID_CLIENT_SEEK)
|
|
G_STATIC_ASSERT (GST_RTMP_VIDEOFUNCTION == 1); /* libavformat's magic number */
|
|
|
|
static void socket_connect (GTask * task);
|
|
static void socket_connect_done (GObject * source, GAsyncResult * result,
|
|
gpointer user_data);
|
|
static void handshake_done (GObject * source, GAsyncResult * result,
|
|
gpointer user_data);
|
|
static void send_connect (GTask * task);
|
|
static void send_stop (GstRtmpConnection * connection, const gchar * stream,
|
|
const GstRtmpStopCommands stop_commands);
|
|
static void send_secure_token_response (GTask * task,
|
|
GstRtmpConnection * connection, const gchar * challenge);
|
|
static void connection_error (GstRtmpConnection * connection,
|
|
gpointer user_data);
|
|
|
|
#define DEFAULT_TIMEOUT 5
|
|
|
|
typedef struct
|
|
{
|
|
GstRtmpLocation location;
|
|
gchar *auth_query;
|
|
GstRtmpConnection *connection;
|
|
gulong error_handler_id;
|
|
} ConnectTaskData;
|
|
|
|
static ConnectTaskData *
|
|
connect_task_data_new (const GstRtmpLocation * location)
|
|
{
|
|
ConnectTaskData *data = g_slice_new0 (ConnectTaskData);
|
|
gst_rtmp_location_copy (&data->location, location);
|
|
return data;
|
|
}
|
|
|
|
static void
|
|
connect_task_data_free (gpointer ptr)
|
|
{
|
|
ConnectTaskData *data = ptr;
|
|
gst_rtmp_location_clear (&data->location);
|
|
g_clear_pointer (&data->auth_query, g_free);
|
|
if (data->error_handler_id) {
|
|
g_signal_handler_disconnect (data->connection, data->error_handler_id);
|
|
}
|
|
g_clear_object (&data->connection);
|
|
g_slice_free (ConnectTaskData, data);
|
|
}
|
|
|
|
static GRegex *auth_regex = NULL;
|
|
|
|
void
|
|
gst_rtmp_client_connect_async (const GstRtmpLocation * location,
|
|
GCancellable * cancellable, GAsyncReadyCallback callback,
|
|
gpointer user_data)
|
|
{
|
|
GTask *task;
|
|
|
|
init_debug ();
|
|
|
|
if (g_once_init_enter (&auth_regex)) {
|
|
GRegex *re = g_regex_new ("\\[ *AccessManager.Reject *\\] *: *"
|
|
"\\[ *authmod=(?<authmod>.*?) *\\] *: *"
|
|
"(?<query>\\?.*)\\Z", G_REGEX_DOTALL, 0, NULL);
|
|
g_once_init_leave (&auth_regex, re);
|
|
}
|
|
|
|
task = g_task_new (NULL, cancellable, callback, user_data);
|
|
|
|
g_task_set_task_data (task, connect_task_data_new (location),
|
|
connect_task_data_free);
|
|
|
|
socket_connect (task);
|
|
}
|
|
|
|
static void
|
|
socket_connect (GTask * task)
|
|
{
|
|
ConnectTaskData *data = g_task_get_task_data (task);
|
|
GSocketConnectable *addr;
|
|
GSocketClient *socket_client;
|
|
|
|
if (data->location.timeout < 0) {
|
|
data->location.timeout = DEFAULT_TIMEOUT;
|
|
}
|
|
|
|
if (data->error_handler_id) {
|
|
g_signal_handler_disconnect (data->connection, data->error_handler_id);
|
|
data->error_handler_id = 0;
|
|
}
|
|
|
|
if (data->connection) {
|
|
gst_rtmp_connection_close (data->connection);
|
|
g_clear_object (&data->connection);
|
|
}
|
|
|
|
if (!data->location.host) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_NOT_INITIALIZED,
|
|
"Host is not set");
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
if (!data->location.port) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_NOT_INITIALIZED,
|
|
"Port is not set");
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
socket_client = g_socket_client_new ();
|
|
g_socket_client_set_timeout (socket_client, data->location.timeout);
|
|
|
|
switch (data->location.scheme) {
|
|
case GST_RTMP_SCHEME_RTMP:
|
|
break;
|
|
|
|
case GST_RTMP_SCHEME_RTMPS:
|
|
GST_DEBUG ("Configuring TLS, validation flags 0x%02x",
|
|
data->location.tls_flags);
|
|
g_socket_client_set_tls (socket_client, TRUE);
|
|
g_socket_client_set_tls_validation_flags (socket_client,
|
|
data->location.tls_flags);
|
|
break;
|
|
|
|
default:
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_NOT_SUPPORTED,
|
|
"Invalid scheme ID %d", data->location.scheme);
|
|
g_object_unref (socket_client);
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
addr = g_network_address_new (data->location.host, data->location.port);
|
|
|
|
GST_DEBUG ("Starting socket connection");
|
|
|
|
g_socket_client_connect_async (socket_client, addr,
|
|
g_task_get_cancellable (task), socket_connect_done, task);
|
|
g_object_unref (addr);
|
|
g_object_unref (socket_client);
|
|
}
|
|
|
|
static void
|
|
socket_connect_done (GObject * source, GAsyncResult * result,
|
|
gpointer user_data)
|
|
{
|
|
GSocketClient *socket_client = G_SOCKET_CLIENT (source);
|
|
GSocketConnection *socket_connection;
|
|
GTask *task = user_data;
|
|
GError *error = NULL;
|
|
|
|
socket_connection =
|
|
g_socket_client_connect_finish (socket_client, result, &error);
|
|
|
|
if (g_task_return_error_if_cancelled (task)) {
|
|
GST_DEBUG ("Socket connection was cancelled");
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
if (socket_connection == NULL) {
|
|
GST_ERROR ("Socket connection error");
|
|
g_task_return_error (task, error);
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
GST_DEBUG ("Socket connection established");
|
|
|
|
gst_rtmp_client_handshake (G_IO_STREAM (socket_connection), FALSE,
|
|
g_task_get_cancellable (task), handshake_done, task);
|
|
g_object_unref (socket_connection);
|
|
}
|
|
|
|
|
|
static void
|
|
handshake_done (GObject * source, GAsyncResult * result, gpointer user_data)
|
|
{
|
|
GIOStream *stream = G_IO_STREAM (source);
|
|
GSocketConnection *socket_connection = G_SOCKET_CONNECTION (stream);
|
|
GTask *task = user_data;
|
|
ConnectTaskData *data = g_task_get_task_data (task);
|
|
GError *error = NULL;
|
|
gboolean res;
|
|
|
|
res = gst_rtmp_client_handshake_finish (stream, result, &error);
|
|
if (!res) {
|
|
g_io_stream_close_async (stream, G_PRIORITY_DEFAULT, NULL, NULL, NULL);
|
|
g_task_return_error (task, error);
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
data->connection = gst_rtmp_connection_new (socket_connection);
|
|
data->error_handler_id = g_signal_connect (data->connection,
|
|
"error", G_CALLBACK (connection_error), task);
|
|
|
|
send_connect (task);
|
|
}
|
|
|
|
static void
|
|
connection_error (GstRtmpConnection * connection, gpointer user_data)
|
|
{
|
|
GTask *task = user_data;
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED,
|
|
"error during connection attempt");
|
|
}
|
|
|
|
static gchar *
|
|
do_adobe_auth (const gchar * username, const gchar * password,
|
|
const gchar * salt, const gchar * opaque, const gchar * challenge)
|
|
{
|
|
guint8 hash[16]; /* MD5 digest */
|
|
gsize hashlen = sizeof hash;
|
|
gchar *challenge2, *auth_query;
|
|
GChecksum *md5;
|
|
|
|
g_return_val_if_fail (username, NULL);
|
|
g_return_val_if_fail (password, NULL);
|
|
g_return_val_if_fail (salt, NULL);
|
|
|
|
md5 = g_checksum_new (G_CHECKSUM_MD5);
|
|
g_checksum_update (md5, (guchar *) username, -1);
|
|
g_checksum_update (md5, (guchar *) salt, -1);
|
|
g_checksum_update (md5, (guchar *) password, -1);
|
|
|
|
g_checksum_get_digest (md5, hash, &hashlen);
|
|
g_warn_if_fail (hashlen == sizeof hash);
|
|
|
|
{
|
|
gchar *hashstr = g_base64_encode ((guchar *) hash, sizeof hash);
|
|
g_checksum_reset (md5);
|
|
g_checksum_update (md5, (guchar *) hashstr, -1);
|
|
g_free (hashstr);
|
|
}
|
|
|
|
if (opaque)
|
|
g_checksum_update (md5, (guchar *) opaque, -1);
|
|
else if (challenge)
|
|
g_checksum_update (md5, (guchar *) challenge, -1);
|
|
|
|
challenge2 = g_strdup_printf ("%08x", g_random_int ());
|
|
g_checksum_update (md5, (guchar *) challenge2, -1);
|
|
|
|
g_checksum_get_digest (md5, hash, &hashlen);
|
|
g_warn_if_fail (hashlen == sizeof hash);
|
|
|
|
{
|
|
gchar *hashstr = g_base64_encode ((guchar *) hash, sizeof hash);
|
|
|
|
if (opaque) {
|
|
auth_query =
|
|
g_strdup_printf
|
|
("authmod=%s&user=%s&challenge=%s&response=%s&opaque=%s", "adobe",
|
|
username, challenge2, hashstr, opaque);
|
|
} else {
|
|
auth_query =
|
|
g_strdup_printf ("authmod=%s&user=%s&challenge=%s&response=%s",
|
|
"adobe", username, challenge2, hashstr);
|
|
}
|
|
g_free (hashstr);
|
|
}
|
|
|
|
g_checksum_free (md5);
|
|
g_free (challenge2);
|
|
|
|
return auth_query;
|
|
}
|
|
|
|
static void
|
|
send_connect (GTask * task)
|
|
{
|
|
ConnectTaskData *data = g_task_get_task_data (task);
|
|
GstAmfNode *node;
|
|
const gchar *app, *flash_ver;
|
|
gchar *uri, *appstr = NULL, *uristr = NULL;
|
|
gboolean publish;
|
|
|
|
node = gst_amf_node_new_object ();
|
|
app = data->location.application;
|
|
flash_ver = data->location.flash_ver;
|
|
publish = data->location.publish;
|
|
uri = gst_rtmp_location_get_string (&data->location, FALSE);
|
|
|
|
if (!app) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_NOT_INITIALIZED,
|
|
"Application is not set");
|
|
g_object_unref (task);
|
|
goto out;
|
|
}
|
|
|
|
if (!flash_ver) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_NOT_INITIALIZED,
|
|
"Flash version is not set");
|
|
g_object_unref (task);
|
|
goto out;
|
|
}
|
|
|
|
if (data->auth_query) {
|
|
const gchar *query = data->auth_query;
|
|
appstr = g_strdup_printf ("%s?%s", app, query);
|
|
uristr = g_strdup_printf ("%s?%s", uri, query);
|
|
} else if (data->location.authmod == GST_RTMP_AUTHMOD_ADOBE) {
|
|
const gchar *user = data->location.username;
|
|
const gchar *authmod = "adobe";
|
|
|
|
if (!user) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED,
|
|
"no username for adobe authentication");
|
|
g_object_unref (task);
|
|
goto out;
|
|
}
|
|
|
|
if (!data->location.password) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED,
|
|
"no password for adobe authentication");
|
|
g_object_unref (task);
|
|
goto out;
|
|
}
|
|
|
|
appstr = g_strdup_printf ("%s?authmod=%s&user=%s", app, authmod, user);
|
|
uristr = g_strdup_printf ("%s?authmod=%s&user=%s", uri, authmod, user);
|
|
} else {
|
|
appstr = g_strdup (app);
|
|
uristr = g_strdup (uri);
|
|
}
|
|
|
|
/* Arguments for the connect command.
|
|
* Most of these are described in rtmp_specification_1.0.pdf page 30 */
|
|
|
|
/* "The server application name the client is connected to." */
|
|
gst_amf_node_append_field_take_string (node, "app", appstr, -1);
|
|
|
|
if (publish) {
|
|
/* Undocumented. Sent by both libavformat and librtmp. */
|
|
gst_amf_node_append_field_string (node, "type", "nonprivate", -1);
|
|
}
|
|
|
|
/* "Flash Player version. It is the same string as returned by the
|
|
* ApplicationScript getversion () function." */
|
|
gst_amf_node_append_field_string (node, "flashVer", flash_ver, -1);
|
|
|
|
/* "URL of the source SWF file making the connection."
|
|
* XXX: libavformat sends "swfUrl" here, if provided. */
|
|
|
|
/* "URL of the Server. It has the following format.
|
|
* protocol://servername:port/appName/appInstance" */
|
|
gst_amf_node_append_field_take_string (node, "tcUrl", uristr, -1);
|
|
|
|
if (!publish) {
|
|
/* "True if proxy is being used." */
|
|
gst_amf_node_append_field_boolean (node, "fpad", FALSE);
|
|
|
|
/* Undocumented. Sent by libavformat. */
|
|
gst_amf_node_append_field_number (node, "capabilities",
|
|
15 /* libavformat's magic number */ );
|
|
|
|
/* "Indicates what audio codecs the client supports." */
|
|
gst_amf_node_append_field_number (node, "audioCodecs",
|
|
GST_RTMP_AUDIOCODECS);
|
|
|
|
/* "Indicates what video codecs are supported." */
|
|
gst_amf_node_append_field_number (node, "videoCodecs",
|
|
GST_RTMP_VIDEOCODECS);
|
|
|
|
/* "Indicates what special video functions are supported." */
|
|
gst_amf_node_append_field_number (node, "videoFunction",
|
|
GST_RTMP_VIDEOFUNCTION);
|
|
|
|
/* "URL of the web page from where the SWF file was loaded."
|
|
* XXX: libavformat sends "pageUrl" here, if provided. */
|
|
}
|
|
|
|
gst_rtmp_connection_send_command (data->connection, send_connect_done,
|
|
task, 0, "connect", node, NULL);
|
|
|
|
out:
|
|
gst_amf_node_free (node);
|
|
g_free (uri);
|
|
}
|
|
|
|
static void
|
|
send_connect_done (const gchar * command_name, GPtrArray * args,
|
|
gpointer user_data)
|
|
{
|
|
GTask *task = G_TASK (user_data);
|
|
ConnectTaskData *data = g_task_get_task_data (task);
|
|
const GstAmfNode *node, *optional_args;
|
|
const gchar *code;
|
|
|
|
if (g_task_return_error_if_cancelled (task)) {
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
if (!args) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED,
|
|
"connect failed: %s", command_name);
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
if (args->len < 2) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED,
|
|
"connect failed; not enough return arguments");
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
optional_args = g_ptr_array_index (args, 1);
|
|
|
|
node = gst_amf_node_get_field (optional_args, "code");
|
|
if (!node) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED,
|
|
"result code missing from connect cmd result");
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
code = gst_amf_node_peek_string (node, NULL);
|
|
GST_INFO ("connect result: %s", GST_STR_NULL (code));
|
|
|
|
if (g_str_equal (code, "NetConnection.Connect.Success")) {
|
|
node = gst_amf_node_get_field (optional_args, "secureToken");
|
|
send_secure_token_response (task, data->connection,
|
|
node ? gst_amf_node_peek_string (node, NULL) : NULL);
|
|
return;
|
|
}
|
|
|
|
if (g_str_equal (code, "NetConnection.Connect.Rejected")) {
|
|
GstRtmpAuthmod authmod = data->location.authmod;
|
|
GMatchInfo *match_info;
|
|
const gchar *desc;
|
|
GstUri *query;
|
|
|
|
node = gst_amf_node_get_field (optional_args, "description");
|
|
if (!node) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED,
|
|
"Connect rejected; no description");
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
desc = gst_amf_node_peek_string (node, NULL);
|
|
GST_DEBUG ("connect result desc: %s", GST_STR_NULL (desc));
|
|
|
|
if (authmod == GST_RTMP_AUTHMOD_AUTO && strstr (desc, "code=403 need auth")) {
|
|
if (strstr (desc, "authmod=adobe")) {
|
|
GST_INFO ("Reconnecting with authmod=adobe");
|
|
data->location.authmod = GST_RTMP_AUTHMOD_ADOBE;
|
|
socket_connect (task);
|
|
return;
|
|
}
|
|
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED,
|
|
"unhandled authentication mode: %s", desc);
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
if (!g_regex_match (auth_regex, desc, 0, &match_info)) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED,
|
|
"failed to parse auth rejection: %s", desc);
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
{
|
|
gchar *authmod_str = g_match_info_fetch_named (match_info, "authmod");
|
|
gchar *query_str = g_match_info_fetch_named (match_info, "query");
|
|
gboolean matches;
|
|
|
|
GST_INFO ("regex parsed auth: authmod=%s, query=%s",
|
|
GST_STR_NULL (authmod_str), GST_STR_NULL (query_str));
|
|
g_match_info_free (match_info);
|
|
|
|
switch (authmod) {
|
|
case GST_RTMP_AUTHMOD_ADOBE:
|
|
matches = g_str_equal (authmod_str, "adobe");
|
|
break;
|
|
|
|
default:
|
|
matches = FALSE;
|
|
break;
|
|
}
|
|
|
|
if (!matches) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED,
|
|
"server uses wrong authentication mode '%s'; expected %s",
|
|
GST_STR_NULL (authmod_str), gst_rtmp_authmod_get_nick (authmod));
|
|
g_object_unref (task);
|
|
g_free (authmod_str);
|
|
g_free (query_str);
|
|
return;
|
|
}
|
|
g_free (authmod_str);
|
|
|
|
query = gst_uri_from_string (query_str);
|
|
if (!query) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED,
|
|
"failed to parse authentication query '%s'",
|
|
GST_STR_NULL (query_str));
|
|
g_object_unref (task);
|
|
g_free (query_str);
|
|
return;
|
|
}
|
|
g_free (query_str);
|
|
}
|
|
|
|
{
|
|
const gchar *reason = gst_uri_get_query_value (query, "reason");
|
|
|
|
if (g_str_equal (reason, "authfailed")) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED,
|
|
"authentication failed! wrong credentials?");
|
|
g_object_unref (task);
|
|
gst_uri_unref (query);
|
|
return;
|
|
}
|
|
|
|
if (!g_str_equal (reason, "needauth")) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED,
|
|
"unhandled rejection reason '%s'", reason ? reason : "");
|
|
g_object_unref (task);
|
|
gst_uri_unref (query);
|
|
return;
|
|
}
|
|
}
|
|
|
|
g_warn_if_fail (!data->auth_query);
|
|
data->auth_query = do_adobe_auth (data->location.username,
|
|
data->location.password, gst_uri_get_query_value (query, "salt"),
|
|
gst_uri_get_query_value (query, "opaque"),
|
|
gst_uri_get_query_value (query, "challenge"));
|
|
|
|
gst_uri_unref (query);
|
|
|
|
if (!data->auth_query) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED,
|
|
"couldn't generate adobe style authentication query");
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
socket_connect (task);
|
|
return;
|
|
}
|
|
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED,
|
|
"unhandled connect result code: %s", GST_STR_NULL (code));
|
|
g_object_unref (task);
|
|
}
|
|
|
|
/* prep key: pack 1st 16 chars into 4 LittleEndian ints */
|
|
static void
|
|
rtmp_tea_decode_prep_key (const gchar * key, guint32 out[4])
|
|
{
|
|
gchar copy[17];
|
|
|
|
g_return_if_fail (key);
|
|
g_return_if_fail (out);
|
|
|
|
/* ensure we can read 16 bytes */
|
|
strncpy (copy, key, 16);
|
|
/* placate GCC 8 -Wstringop-truncation */
|
|
copy[16] = 0;
|
|
|
|
out[0] = GST_READ_UINT32_LE (copy);
|
|
out[1] = GST_READ_UINT32_LE (copy + 4);
|
|
out[2] = GST_READ_UINT32_LE (copy + 8);
|
|
out[3] = GST_READ_UINT32_LE (copy + 12);
|
|
}
|
|
|
|
/* prep text: hex2bin, each 8 digits -> 4 chars -> 1 uint32 */
|
|
static GArray *
|
|
rtmp_tea_decode_prep_text (const gchar * text)
|
|
{
|
|
GArray *arr;
|
|
gsize len, i;
|
|
|
|
g_return_val_if_fail (text, NULL);
|
|
|
|
len = strlen (text);
|
|
arr = g_array_sized_new (TRUE, TRUE, 4, (len + 7) / 8);
|
|
|
|
for (i = 0; i < len; i += 8) {
|
|
gchar copy[9];
|
|
guchar chars[4];
|
|
gsize j;
|
|
guint32 val;
|
|
|
|
/* ensure we can read 8 bytes */
|
|
strncpy (copy, text + i, 8);
|
|
/* placate GCC 8 -Wstringop-truncation */
|
|
copy[8] = 0;
|
|
|
|
for (j = 0; j < 4; j++) {
|
|
gint hi, lo;
|
|
|
|
hi = g_ascii_xdigit_value (copy[2 * j]);
|
|
lo = g_ascii_xdigit_value (copy[2 * j + 1]);
|
|
|
|
chars[j] = (hi > 0 ? hi << 4 : 0) + (lo > 0 ? lo : 0);
|
|
}
|
|
|
|
val = GST_READ_UINT32_LE (chars);
|
|
g_array_append_val (arr, val);
|
|
}
|
|
|
|
return arr;
|
|
}
|
|
|
|
/* return text from uint32s to chars */
|
|
static gchar *
|
|
rtmp_tea_decode_return_text (GArray * arr)
|
|
{
|
|
#if G_BYTE_ORDER != G_LITTLE_ENDIAN
|
|
gsize i;
|
|
|
|
g_return_val_if_fail (arr, NULL);
|
|
|
|
for (i = 0; i < arr->len; i++) {
|
|
guint32 *val = &g_array_index (arr, guint32, i);
|
|
*val = GUINT32_TO_LE (*val);
|
|
}
|
|
#endif
|
|
|
|
/* array is alredy zero-terminated */
|
|
return g_array_free (arr, FALSE);
|
|
}
|
|
|
|
/* http://www.movable-type.co.uk/scripts/tea-block.html */
|
|
static void
|
|
rtmp_tea_decode_btea (GArray * text, guint32 key[4])
|
|
{
|
|
guint32 *v, n, *k;
|
|
guint32 z, y, sum = 0, e, DELTA = 0x9e3779b9;
|
|
guint32 p, q;
|
|
|
|
g_return_if_fail (text);
|
|
g_return_if_fail (text->len > 0);
|
|
g_return_if_fail (key);
|
|
|
|
v = (guint32 *) text->data;
|
|
n = text->len;
|
|
k = key;
|
|
z = v[n - 1];
|
|
y = v[0];
|
|
q = 6 + 52 / n;
|
|
sum = q * DELTA;
|
|
|
|
#define MX ((z>>5^y<<2) + (y>>3^z<<4)) ^ ((sum^y) + (k[(p&3)^e]^z));
|
|
|
|
while (sum != 0) {
|
|
e = sum >> 2 & 3;
|
|
for (p = n - 1; p > 0; p--)
|
|
z = v[p - 1], y = v[p] -= MX;
|
|
z = v[n - 1];
|
|
y = v[0] -= MX;
|
|
sum -= DELTA;
|
|
}
|
|
|
|
#undef MX
|
|
}
|
|
|
|
/* taken from librtmp */
|
|
static gchar *
|
|
rtmp_tea_decode (const gchar * bin_key, const gchar * hex_text)
|
|
{
|
|
guint32 key[4];
|
|
GArray *text;
|
|
|
|
rtmp_tea_decode_prep_key (bin_key, key);
|
|
text = rtmp_tea_decode_prep_text (hex_text);
|
|
rtmp_tea_decode_btea (text, key);
|
|
return rtmp_tea_decode_return_text (text);
|
|
}
|
|
|
|
static void
|
|
send_secure_token_response (GTask * task, GstRtmpConnection * connection,
|
|
const gchar * challenge)
|
|
{
|
|
ConnectTaskData *data = g_task_get_task_data (task);
|
|
if (challenge) {
|
|
GstAmfNode *node1;
|
|
GstAmfNode *node2;
|
|
gchar *response;
|
|
|
|
if (!data->location.secure_token || !data->location.secure_token[0]) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED,
|
|
"server requires secure token authentication");
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
response = rtmp_tea_decode (data->location.secure_token, challenge);
|
|
|
|
GST_DEBUG ("response: %s", response);
|
|
|
|
node1 = gst_amf_node_new_null ();
|
|
node2 = gst_amf_node_new_take_string (response, -1);
|
|
gst_rtmp_connection_send_command (connection, NULL, NULL, 0,
|
|
"secureTokenResponse", node1, node2, NULL);
|
|
gst_amf_node_free (node1);
|
|
gst_amf_node_free (node2);
|
|
}
|
|
|
|
g_signal_handler_disconnect (connection, data->error_handler_id);
|
|
data->error_handler_id = 0;
|
|
|
|
g_task_return_pointer (task, g_object_ref (connection),
|
|
gst_rtmp_connection_close_and_unref);
|
|
g_object_unref (task);
|
|
}
|
|
|
|
GstRtmpConnection *
|
|
gst_rtmp_client_connect_finish (GAsyncResult * result, GError ** error)
|
|
{
|
|
GTask *task = G_TASK (result);
|
|
return g_task_propagate_pointer (task, error);
|
|
}
|
|
|
|
static void send_create_stream (GTask * task);
|
|
static void send_publish_or_play (GTask * task);
|
|
|
|
typedef struct
|
|
{
|
|
GstRtmpConnection *connection;
|
|
gulong error_handler_id;
|
|
gchar *stream;
|
|
gboolean publish;
|
|
guint32 id;
|
|
} StreamTaskData;
|
|
|
|
static StreamTaskData *
|
|
stream_task_data_new (GstRtmpConnection * connection, const gchar * stream,
|
|
gboolean publish)
|
|
{
|
|
StreamTaskData *data = g_slice_new0 (StreamTaskData);
|
|
data->connection = g_object_ref (connection);
|
|
data->stream = g_strdup (stream);
|
|
data->publish = publish;
|
|
return data;
|
|
}
|
|
|
|
static void
|
|
stream_task_data_free (gpointer ptr)
|
|
{
|
|
StreamTaskData *data = ptr;
|
|
g_clear_pointer (&data->stream, g_free);
|
|
if (data->error_handler_id) {
|
|
g_signal_handler_disconnect (data->connection, data->error_handler_id);
|
|
}
|
|
g_clear_object (&data->connection);
|
|
g_slice_free (StreamTaskData, data);
|
|
}
|
|
|
|
static void
|
|
start_stream (GstRtmpConnection * connection, const gchar * stream,
|
|
gboolean publish, GCancellable * cancellable,
|
|
GAsyncReadyCallback callback, gpointer user_data)
|
|
{
|
|
GTask *task;
|
|
StreamTaskData *data;
|
|
|
|
init_debug ();
|
|
|
|
task = g_task_new (connection, cancellable, callback, user_data);
|
|
|
|
if (!stream) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_NOT_INITIALIZED,
|
|
"Stream is not set");
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
data = stream_task_data_new (connection, stream, publish);
|
|
g_task_set_task_data (task, data, stream_task_data_free);
|
|
|
|
data->error_handler_id = g_signal_connect (connection,
|
|
"error", G_CALLBACK (connection_error), task);
|
|
|
|
send_create_stream (task);
|
|
}
|
|
|
|
void
|
|
gst_rtmp_client_start_publish_async (GstRtmpConnection * connection,
|
|
const gchar * stream, GCancellable * cancellable,
|
|
GAsyncReadyCallback callback, gpointer user_data)
|
|
{
|
|
start_stream (connection, stream, TRUE, cancellable, callback, user_data);
|
|
}
|
|
|
|
void
|
|
gst_rtmp_client_start_play_async (GstRtmpConnection * connection,
|
|
const gchar * stream, GCancellable * cancellable,
|
|
GAsyncReadyCallback callback, gpointer user_data)
|
|
{
|
|
start_stream (connection, stream, FALSE, cancellable, callback, user_data);
|
|
}
|
|
|
|
static void
|
|
send_set_buffer_length (GstRtmpConnection * connection, guint32 stream,
|
|
guint32 ms)
|
|
{
|
|
GstRtmpUserControl uc = {
|
|
.type = GST_RTMP_USER_CONTROL_TYPE_SET_BUFFER_LENGTH,
|
|
.param = stream,
|
|
.param2 = ms,
|
|
};
|
|
|
|
gst_rtmp_connection_queue_message (connection,
|
|
gst_rtmp_message_new_user_control (&uc));
|
|
}
|
|
|
|
static void
|
|
send_create_stream (GTask * task)
|
|
{
|
|
GstRtmpConnection *connection = g_task_get_source_object (task);
|
|
StreamTaskData *data = g_task_get_task_data (task);
|
|
GstAmfNode *command_object, *stream_name;
|
|
|
|
command_object = gst_amf_node_new_null ();
|
|
stream_name = gst_amf_node_new_string (data->stream, -1);
|
|
|
|
if (data->publish) {
|
|
/* Not part of RTMP documentation */
|
|
GST_DEBUG ("Releasing stream '%s'", data->stream);
|
|
gst_rtmp_connection_send_command (connection, NULL, NULL, 0,
|
|
"releaseStream", command_object, stream_name, NULL);
|
|
gst_rtmp_connection_send_command (connection, NULL, NULL, 0,
|
|
"FCPublish", command_object, stream_name, NULL);
|
|
} else {
|
|
/* Matches librtmp */
|
|
gst_rtmp_connection_request_window_size (connection,
|
|
GST_RTMP_DEFAULT_WINDOW_ACK_SIZE);
|
|
send_set_buffer_length (connection, 0, 300);
|
|
}
|
|
|
|
GST_INFO ("Creating stream '%s'", data->stream);
|
|
gst_rtmp_connection_send_command (connection, create_stream_done, task, 0,
|
|
"createStream", command_object, NULL);
|
|
|
|
gst_amf_node_free (stream_name);
|
|
gst_amf_node_free (command_object);
|
|
}
|
|
|
|
static void
|
|
create_stream_done (const gchar * command_name, GPtrArray * args,
|
|
gpointer user_data)
|
|
{
|
|
GTask *task = G_TASK (user_data);
|
|
StreamTaskData *data = g_task_get_task_data (task);
|
|
GstAmfNode *result;
|
|
|
|
if (g_task_return_error_if_cancelled (task)) {
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
if (!args) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED,
|
|
"createStream failed: %s", command_name);
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
if (args->len < 2) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED,
|
|
"createStream failed; not enough return arguments");
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
result = g_ptr_array_index (args, 1);
|
|
if (gst_amf_node_get_type (result) != GST_AMF_TYPE_NUMBER) {
|
|
GString *error_dump = g_string_new ("");
|
|
|
|
gst_amf_node_dump (result, -1, error_dump);
|
|
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED,
|
|
"createStream failed: %s", error_dump->str);
|
|
g_object_unref (task);
|
|
|
|
g_string_free (error_dump, TRUE);
|
|
return;
|
|
}
|
|
|
|
data->id = gst_amf_node_get_number (result);
|
|
GST_INFO ("createStream success, stream_id=%" G_GUINT32_FORMAT, data->id);
|
|
|
|
if (data->id == 0) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_INVALID_DATA,
|
|
"createStream returned ID 0");
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
send_publish_or_play (task);
|
|
}
|
|
|
|
static void
|
|
send_publish_or_play (GTask * task)
|
|
{
|
|
GstRtmpConnection *connection = g_task_get_source_object (task);
|
|
StreamTaskData *data = g_task_get_task_data (task);
|
|
const gchar *command = data->publish ? "publish" : "play";
|
|
GstAmfNode *command_object, *stream_name, *argument;
|
|
|
|
command_object = gst_amf_node_new_null ();
|
|
stream_name = gst_amf_node_new_string (data->stream, -1);
|
|
|
|
if (data->publish) {
|
|
/* publishing type (live, record, append) */
|
|
argument = gst_amf_node_new_string ("live", -1);
|
|
} else {
|
|
/* "Start" argument: -2 = live or recording, -1 = only live
|
|
0 or positive = only recording, seek to X seconds */
|
|
argument = gst_amf_node_new_number (-2);
|
|
}
|
|
|
|
GST_INFO ("Sending %s for '%s' on stream %" G_GUINT32_FORMAT,
|
|
command, data->stream, data->id);
|
|
gst_rtmp_connection_expect_command (connection, on_publish_or_play_status,
|
|
task, data->id, "onStatus");
|
|
gst_rtmp_connection_send_command (connection, NULL, NULL, data->id,
|
|
command, command_object, stream_name, argument, NULL);
|
|
|
|
if (!data->publish) {
|
|
/* Matches librtmp */
|
|
send_set_buffer_length (connection, data->id, 30000);
|
|
}
|
|
|
|
gst_amf_node_free (command_object);
|
|
gst_amf_node_free (stream_name);
|
|
gst_amf_node_free (argument);
|
|
}
|
|
|
|
static void
|
|
on_publish_or_play_status (const gchar * command_name, GPtrArray * args,
|
|
gpointer user_data)
|
|
{
|
|
GTask *task = G_TASK (user_data);
|
|
GstRtmpConnection *connection = g_task_get_source_object (task);
|
|
StreamTaskData *data = g_task_get_task_data (task);
|
|
const gchar *command = data->publish ? "publish" : "play", *code = NULL;
|
|
GString *info_dump;
|
|
|
|
if (g_task_return_error_if_cancelled (task)) {
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
if (!args) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED,
|
|
"%s failed: %s", command, command_name);
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
if (args->len < 2) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED,
|
|
"%s failed; not enough return arguments", command);
|
|
g_object_unref (task);
|
|
return;
|
|
}
|
|
|
|
{
|
|
const GstAmfNode *info_object, *code_object;
|
|
info_object = g_ptr_array_index (args, 1);
|
|
code_object = gst_amf_node_get_field (info_object, "code");
|
|
|
|
if (code_object) {
|
|
code = gst_amf_node_peek_string (code_object, NULL);
|
|
}
|
|
|
|
info_dump = g_string_new ("");
|
|
gst_amf_node_dump (info_object, -1, info_dump);
|
|
}
|
|
|
|
if (data->publish) {
|
|
if (g_strcmp0 (code, "NetStream.Publish.Start") == 0) {
|
|
GST_INFO ("publish success: %s", info_dump->str);
|
|
g_task_return_boolean (task, TRUE);
|
|
goto out;
|
|
}
|
|
|
|
if (g_strcmp0 (code, "NetStream.Publish.BadName") == 0) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_EXISTS,
|
|
"publish denied: stream already exists: %s", info_dump->str);
|
|
goto out;
|
|
}
|
|
|
|
if (g_strcmp0 (code, "NetStream.Publish.Denied") == 0) {
|
|
g_task_return_new_error (task, G_IO_ERROR,
|
|
G_IO_ERROR_PERMISSION_DENIED, "publish denied: %s", info_dump->str);
|
|
goto out;
|
|
}
|
|
} else {
|
|
if (g_strcmp0 (code, "NetStream.Play.Start") == 0 ||
|
|
g_strcmp0 (code, "NetStream.Play.Reset") == 0) {
|
|
GST_INFO ("play success: %s", info_dump->str);
|
|
g_task_return_boolean (task, TRUE);
|
|
goto out;
|
|
}
|
|
|
|
if (g_strcmp0 (code, "NetStream.Play.StreamNotFound") == 0) {
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_NOT_FOUND,
|
|
"play denied: stream not found: %s", info_dump->str);
|
|
goto out;
|
|
}
|
|
}
|
|
|
|
g_task_return_new_error (task, G_IO_ERROR, G_IO_ERROR_FAILED,
|
|
"unhandled %s result: %s", command, info_dump->str);
|
|
|
|
out:
|
|
g_string_free (info_dump, TRUE);
|
|
|
|
g_signal_handler_disconnect (connection, data->error_handler_id);
|
|
data->error_handler_id = 0;
|
|
|
|
g_object_unref (task);
|
|
}
|
|
|
|
static gboolean
|
|
start_stream_finish (GstRtmpConnection * connection,
|
|
GAsyncResult * result, guint32 * stream_id, GError ** error)
|
|
{
|
|
GTask *task;
|
|
StreamTaskData *data;
|
|
|
|
g_return_val_if_fail (g_task_is_valid (result, connection), FALSE);
|
|
|
|
task = G_TASK (result);
|
|
|
|
if (!g_task_propagate_boolean (G_TASK (result), error)) {
|
|
return FALSE;
|
|
}
|
|
|
|
data = g_task_get_task_data (task);
|
|
|
|
if (stream_id) {
|
|
*stream_id = data->id;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
gboolean
|
|
gst_rtmp_client_start_publish_finish (GstRtmpConnection * connection,
|
|
GAsyncResult * result, guint32 * stream_id, GError ** error)
|
|
{
|
|
return start_stream_finish (connection, result, stream_id, error);
|
|
}
|
|
|
|
gboolean
|
|
gst_rtmp_client_start_play_finish (GstRtmpConnection * connection,
|
|
GAsyncResult * result, guint32 * stream_id, GError ** error)
|
|
{
|
|
return start_stream_finish (connection, result, stream_id, error);
|
|
}
|
|
|
|
void
|
|
gst_rtmp_client_stop_publish (GstRtmpConnection * connection,
|
|
const gchar * stream, const GstRtmpStopCommands stop_commands)
|
|
{
|
|
send_stop (connection, stream, stop_commands);
|
|
}
|
|
|
|
static void
|
|
send_stop (GstRtmpConnection * connection, const gchar * stream,
|
|
const GstRtmpStopCommands stop_commands)
|
|
{
|
|
GstAmfNode *command_object, *stream_name;
|
|
|
|
command_object = gst_amf_node_new_null ();
|
|
stream_name = gst_amf_node_new_string (stream, -1);
|
|
|
|
if (stop_commands & GST_RTMP_STOP_COMMAND_FCUNPUBLISH) {
|
|
GST_DEBUG ("Sending stop command 'FCUnpublish' for stream '%s'", stream);
|
|
gst_rtmp_connection_send_command (connection, NULL, NULL, 0,
|
|
"FCUnpublish", command_object, stream_name, NULL);
|
|
}
|
|
if (stop_commands & GST_RTMP_STOP_COMMAND_CLOSE_STREAM) {
|
|
GST_DEBUG ("Sending stop command 'closeStream' for stream '%s'", stream);
|
|
gst_rtmp_connection_send_command (connection, NULL, NULL, 0,
|
|
"closeStream", command_object, stream_name, NULL);
|
|
}
|
|
if (stop_commands & GST_RTMP_STOP_COMMAND_DELETE_STREAM) {
|
|
GST_DEBUG ("Sending stop command 'deleteStream' for stream '%s'", stream);
|
|
gst_rtmp_connection_send_command (connection, NULL, NULL, 0,
|
|
"deleteStream", command_object, stream_name, NULL);
|
|
}
|
|
|
|
gst_amf_node_free (stream_name);
|
|
gst_amf_node_free (command_object);
|
|
}
|