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gstreamer/gst/rtmp2/rtmp/rtmpclient.c

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/* 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 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 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 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)";
}
GType
gst_rtmp_stop_commands_get_type (void)
{
static gsize stop_commands_type = 0;
static const GFlagsValue stop_commands[] = {
{GST_RTMP_STOP_COMMANDS_NONE, "No command", "none"},
{GST_RTMP_STOP_COMMANDS_FCUNPUBLISH, "FCUnpublish", "fcunpublish"},
{GST_RTMP_STOP_COMMANDS_CLOSE_STREAM, "closeStream", "closestream"},
{GST_RTMP_STOP_COMMANDS_DELETE_STREAM, "deleteStream", "deletestream"},
{0, NULL, NULL},
};
if (g_once_init_enter (&stop_commands_type)) {
GType tmp = g_flags_register_static ("GstRtmpStopCommands", stop_commands);
g_once_init_leave (&stop_commands_type, tmp);
}
return (GType) stop_commands_type;
}
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,
g_task_get_cancellable (task));
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;
if (!g_task_had_error (task))
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.PublishNotify") == 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_COMMANDS_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_COMMANDS_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_COMMANDS_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);
}
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