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Check for the existence of a session key when starting to play AP2, and drop the connection if not.

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This commit is contained in:
Mike Brady
2022-10-15 13:31:32 +01:00
parent b0bf6668f3
commit 7004c81a28
1 changed files with 167 additions and 161 deletions
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rtsp.c
+167 -161
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@@ -1501,6 +1501,7 @@ int msg_write_response(rtsp_conn_info *conn, rtsp_message *resp) {
{404, "Not Found"},
{451, "Unavailable"},
{456, "Header Field Not Valid for Resource"},
{470, "Connection Authorization Required"},
{500, "Internal Server Error"},
{501, "Not Implemented"}};
// 451 is really "Unavailable For Legal Reasons"!
@@ -3148,187 +3149,192 @@ void handle_setup_2(rtsp_conn_info *conn, rtsp_message *req, rtsp_message *resp)
plist_t streams_array = plist_new_array(); // to hold the ports and stuff
plist_t stream0dict = plist_new_dict();
// more stuff
// set up a UDP control stream and thread and a UDP or TCP audio stream and thread
// bind a new UDP port and get a socket
conn->local_ap2_control_port = 0; // any port
err = bind_socket_and_port(SOCK_DGRAM, conn->connection_ip_family, conn->self_ip_string,
conn->self_scope_id, &conn->local_ap2_control_port,
&conn->ap2_control_socket);
if (err) {
die("Error %d: could not find a UDP port to use as an ap2_control port", err);
}
debug(2, "Connection %d: UDP control port opened: %u.", conn->connection_number,
conn->local_ap2_control_port);
pthread_create(&conn->rtp_ap2_control_thread, NULL, &rtp_ap2_control_receiver, (void *)conn);
// get the session key
// get the session key -- it must have one
plist_t item = plist_dict_get_item(stream0, "shk"); // session key
uint64_t item_value = 0;
uint64_t item_value = 0; // the length
plist_get_data_val(item, (char **)&conn->session_key, &item_value);
if (item_value != 0) {
// get the DACP-ID and Active Remote for remote control stuff
// more stuff
// set up a UDP control stream and thread and a UDP or TCP audio stream and thread
char *ar = msg_get_header(req, "Active-Remote");
if (ar) {
debug(3, "Connection %d: SETUP AP2 -- Active-Remote string seen: \"%s\".",
conn->connection_number, ar);
// get the active remote
if (conn->dacp_active_remote) // this is in case SETUP was previously called
free(conn->dacp_active_remote);
conn->dacp_active_remote = strdup(ar);
#ifdef CONFIG_METADATA
send_metadata('ssnc', 'acre', ar, strlen(ar), req, 1);
#endif
} else {
debug(1, "Connection %d: SETUP AP2 no Active-Remote information the SETUP Record.",
conn->connection_number);
if (conn->dacp_active_remote) { // this is in case SETUP was previously called
free(conn->dacp_active_remote);
conn->dacp_active_remote = NULL;
}
}
ar = msg_get_header(req, "DACP-ID");
if (ar) {
debug(3, "Connection %d: SETUP AP2 -- DACP-ID string seen: \"%s\".",
conn->connection_number, ar);
if (conn->dacp_id) // this is in case SETUP was previously called
free(conn->dacp_id);
conn->dacp_id = strdup(ar);
#ifdef CONFIG_METADATA
send_metadata('ssnc', 'daid', ar, strlen(ar), req, 1);
#endif
} else {
debug(1, "Connection %d: SETUP AP2 doesn't include DACP-ID string information.",
conn->connection_number);
if (conn->dacp_id) { // this is in case SETUP was previously called
free(conn->dacp_id);
conn->dacp_id = NULL;
}
}
// now, get the type of the stream.
item = plist_dict_get_item(stream0, "type");
item_value = 0;
plist_get_uint_val(item, &item_value);
switch (item_value) {
case 96: {
debug(1, "Connection %d. AP2 Realtime Audio Stream.", conn->connection_number);
debug_log_rtsp_message(2, "Realtime Audio Stream SETUP incoming message", req);
// get_play_lock(conn);
conn->airplay_stream_type = realtime_stream;
// bind a new UDP port and get a socket
conn->local_realtime_audio_port = 0; // any port
conn->local_ap2_control_port = 0; // any port
err = bind_socket_and_port(SOCK_DGRAM, conn->connection_ip_family, conn->self_ip_string,
conn->self_scope_id, &conn->local_realtime_audio_port,
&conn->realtime_audio_socket);
conn->self_scope_id, &conn->local_ap2_control_port,
&conn->ap2_control_socket);
if (err) {
die("Error %d: could not find a UDP port to use as a realtime_audio port", err);
die("Error %d: could not find a UDP port to use as an ap2_control port", err);
}
debug(2, "Connection %d: UDP realtime audio port opened: %u.", conn->connection_number,
conn->local_realtime_audio_port);
debug(2, "Connection %d: UDP control port opened: %u.", conn->connection_number,
conn->local_ap2_control_port);
pthread_create(&conn->rtp_realtime_audio_thread, NULL, &rtp_realtime_audio_receiver,
(void *)conn);
pthread_create(&conn->rtp_ap2_control_thread, NULL, &rtp_ap2_control_receiver, (void *)conn);
plist_dict_set_item(stream0dict, "type", plist_new_uint(96));
plist_dict_set_item(stream0dict, "dataPort",
plist_new_uint(conn->local_realtime_audio_port));
// get the DACP-ID and Active Remote for remote control stuff
conn->stream.type = ast_apple_lossless;
debug(3, "An ALAC stream has been detected.");
// Set reasonable connection defaults
conn->stream.fmtp[0] = 96;
conn->stream.fmtp[1] = 352;
conn->stream.fmtp[2] = 0;
conn->stream.fmtp[3] = 16;
conn->stream.fmtp[4] = 40;
conn->stream.fmtp[5] = 10;
conn->stream.fmtp[6] = 14;
conn->stream.fmtp[7] = 2;
conn->stream.fmtp[8] = 255;
conn->stream.fmtp[9] = 0;
conn->stream.fmtp[10] = 0;
conn->stream.fmtp[11] = 44100;
// set the parameters of the player (as distinct from the parameters of the decoder --
// that's done later).
conn->max_frames_per_packet = conn->stream.fmtp[1]; // number of audio frames per packet.
conn->input_rate = conn->stream.fmtp[11];
conn->input_num_channels = conn->stream.fmtp[7];
conn->input_bit_depth = conn->stream.fmtp[3];
conn->input_bytes_per_frame = conn->input_num_channels * ((conn->input_bit_depth + 7) / 8);
debug(2, "Realtime Stream Play");
activity_monitor_signify_activity(1);
player_prepare_to_play(conn);
player_play(conn);
conn->rtp_running = 1; // hack!
} break;
case 103: {
debug(1, "Connection %d. AP2 Buffered Audio Stream.", conn->connection_number);
debug_log_rtsp_message(2, "Buffered Audio Stream SETUP incoming message", req);
// get_play_lock(conn);
conn->airplay_stream_type = buffered_stream;
// get needed stuff
// bind a new TCP port and get a socket
conn->local_buffered_audio_port = 0; // any port
err = bind_socket_and_port(SOCK_STREAM, conn->connection_ip_family, conn->self_ip_string,
conn->self_scope_id, &conn->local_buffered_audio_port,
&conn->buffered_audio_socket);
if (err) {
die("SETUP on Connection %d: Error %d: could not find a TCP port to use as a "
"buffered_audio port",
conn->connection_number, err);
char *ar = msg_get_header(req, "Active-Remote");
if (ar) {
debug(3, "Connection %d: SETUP AP2 -- Active-Remote string seen: \"%s\".",
conn->connection_number, ar);
// get the active remote
if (conn->dacp_active_remote) // this is in case SETUP was previously called
free(conn->dacp_active_remote);
conn->dacp_active_remote = strdup(ar);
#ifdef CONFIG_METADATA
send_metadata('ssnc', 'acre', ar, strlen(ar), req, 1);
#endif
} else {
debug(1, "Connection %d: SETUP AP2 no Active-Remote information the SETUP Record.",
conn->connection_number);
if (conn->dacp_active_remote) { // this is in case SETUP was previously called
free(conn->dacp_active_remote);
conn->dacp_active_remote = NULL;
}
}
debug(2, "Connection %d: TCP Buffered Audio port opened: %u.", conn->connection_number,
conn->local_buffered_audio_port);
ar = msg_get_header(req, "DACP-ID");
if (ar) {
debug(3, "Connection %d: SETUP AP2 -- DACP-ID string seen: \"%s\".",
conn->connection_number, ar);
if (conn->dacp_id) // this is in case SETUP was previously called
free(conn->dacp_id);
conn->dacp_id = strdup(ar);
#ifdef CONFIG_METADATA
send_metadata('ssnc', 'daid', ar, strlen(ar), req, 1);
#endif
} else {
debug(1, "Connection %d: SETUP AP2 doesn't include DACP-ID string information.",
conn->connection_number);
if (conn->dacp_id) { // this is in case SETUP was previously called
free(conn->dacp_id);
conn->dacp_id = NULL;
}
}
// hack.
conn->max_frames_per_packet = 352; // number of audio frames per packet.
conn->input_rate = 44100; // we are stuck with this for the moment.
conn->input_num_channels = 2;
conn->input_bit_depth = 16;
conn->input_bytes_per_frame = conn->input_num_channels * ((conn->input_bit_depth + 7) / 8);
activity_monitor_signify_activity(1);
player_prepare_to_play(
conn); // get capabilities of DAC before creating the buffered audio thread
// now, get the type of the stream.
item = plist_dict_get_item(stream0, "type");
item_value = 0;
plist_get_uint_val(item, &item_value);
pthread_create(&conn->rtp_buffered_audio_thread, NULL, &rtp_buffered_audio_processor,
(void *)conn);
switch (item_value) {
case 96: {
debug(1, "Connection %d. AP2 Realtime Audio Stream.", conn->connection_number);
debug_log_rtsp_message(2, "Realtime Audio Stream SETUP incoming message", req);
// get_play_lock(conn);
conn->airplay_stream_type = realtime_stream;
// bind a new UDP port and get a socket
conn->local_realtime_audio_port = 0; // any port
err = bind_socket_and_port(SOCK_DGRAM, conn->connection_ip_family, conn->self_ip_string,
conn->self_scope_id, &conn->local_realtime_audio_port,
&conn->realtime_audio_socket);
if (err) {
die("Error %d: could not find a UDP port to use as a realtime_audio port", err);
}
debug(2, "Connection %d: UDP realtime audio port opened: %u.", conn->connection_number,
conn->local_realtime_audio_port);
plist_dict_set_item(stream0dict, "type", plist_new_uint(103));
plist_dict_set_item(stream0dict, "dataPort",
plist_new_uint(conn->local_buffered_audio_port));
plist_dict_set_item(stream0dict, "audioBufferSize",
plist_new_uint(conn->ap2_audio_buffer_size));
pthread_create(&conn->rtp_realtime_audio_thread, NULL, &rtp_realtime_audio_receiver,
(void *)conn);
// this should be cancelled by an activity_monitor_signify_activity(1)
// call in the SETRATEANCHORI handler, which should come up right away
activity_monitor_signify_activity(0);
player_play(conn);
plist_dict_set_item(stream0dict, "type", plist_new_uint(96));
plist_dict_set_item(stream0dict, "dataPort",
plist_new_uint(conn->local_realtime_audio_port));
conn->rtp_running = 1; // hack!
} break;
default:
debug(1, "SETUP on Connection %d: Unhandled stream type %" PRIu64 ".",
conn->connection_number, item_value);
debug_log_rtsp_message(1, "Unhandled stream type incoming message", req);
conn->stream.type = ast_apple_lossless;
debug(3, "An ALAC stream has been detected.");
// Set reasonable connection defaults
conn->stream.fmtp[0] = 96;
conn->stream.fmtp[1] = 352;
conn->stream.fmtp[2] = 0;
conn->stream.fmtp[3] = 16;
conn->stream.fmtp[4] = 40;
conn->stream.fmtp[5] = 10;
conn->stream.fmtp[6] = 14;
conn->stream.fmtp[7] = 2;
conn->stream.fmtp[8] = 255;
conn->stream.fmtp[9] = 0;
conn->stream.fmtp[10] = 0;
conn->stream.fmtp[11] = 44100;
// set the parameters of the player (as distinct from the parameters of the decoder --
// that's done later).
conn->max_frames_per_packet = conn->stream.fmtp[1]; // number of audio frames per packet.
conn->input_rate = conn->stream.fmtp[11];
conn->input_num_channels = conn->stream.fmtp[7];
conn->input_bit_depth = conn->stream.fmtp[3];
conn->input_bytes_per_frame = conn->input_num_channels * ((conn->input_bit_depth + 7) / 8);
debug(2, "Realtime Stream Play");
activity_monitor_signify_activity(1);
player_prepare_to_play(conn);
player_play(conn);
conn->rtp_running = 1; // hack!
} break;
case 103: {
debug(1, "Connection %d. AP2 Buffered Audio Stream.", conn->connection_number);
debug_log_rtsp_message(2, "Buffered Audio Stream SETUP incoming message", req);
// get_play_lock(conn);
conn->airplay_stream_type = buffered_stream;
// get needed stuff
// bind a new TCP port and get a socket
conn->local_buffered_audio_port = 0; // any port
err = bind_socket_and_port(SOCK_STREAM, conn->connection_ip_family, conn->self_ip_string,
conn->self_scope_id, &conn->local_buffered_audio_port,
&conn->buffered_audio_socket);
if (err) {
die("SETUP on Connection %d: Error %d: could not find a TCP port to use as a "
"buffered_audio port",
conn->connection_number, err);
}
debug(2, "Connection %d: TCP Buffered Audio port opened: %u.", conn->connection_number,
conn->local_buffered_audio_port);
// hack.
conn->max_frames_per_packet = 352; // number of audio frames per packet.
conn->input_rate = 44100; // we are stuck with this for the moment.
conn->input_num_channels = 2;
conn->input_bit_depth = 16;
conn->input_bytes_per_frame = conn->input_num_channels * ((conn->input_bit_depth + 7) / 8);
activity_monitor_signify_activity(1);
player_prepare_to_play(
conn); // get capabilities of DAC before creating the buffered audio thread
pthread_create(&conn->rtp_buffered_audio_thread, NULL, &rtp_buffered_audio_processor,
(void *)conn);
plist_dict_set_item(stream0dict, "type", plist_new_uint(103));
plist_dict_set_item(stream0dict, "dataPort",
plist_new_uint(conn->local_buffered_audio_port));
plist_dict_set_item(stream0dict, "audioBufferSize",
plist_new_uint(conn->ap2_audio_buffer_size));
// this should be cancelled by an activity_monitor_signify_activity(1)
// call in the SETRATEANCHORI handler, which should come up right away
activity_monitor_signify_activity(0);
player_play(conn);
conn->rtp_running = 1; // hack!
} break;
default:
debug(1, "SETUP on Connection %d: Unhandled stream type %" PRIu64 ".",
conn->connection_number, item_value);
debug_log_rtsp_message(1, "Unhandled stream type incoming message", req);
}
plist_dict_set_item(stream0dict, "controlPort", plist_new_uint(conn->local_ap2_control_port));
plist_array_append_item(streams_array, stream0dict);
plist_dict_set_item(setupResponsePlist, "streams", streams_array);
resp->respcode = 200;
} else {
warn("this stream can not be played because a session key is missing.");
}
plist_dict_set_item(stream0dict, "controlPort", plist_new_uint(conn->local_ap2_control_port));
plist_array_append_item(streams_array, stream0dict);
plist_dict_set_item(setupResponsePlist, "streams", streams_array);
resp->respcode = 200;
} else if (conn->airplay_stream_category == remote_control_stream) {
debug(2, "Connection %d (RC): SETUP: Remote Control Stream received from %s.",
conn->connection_number, conn->client_ip_string);
@@ -4303,7 +4309,7 @@ static void handle_get_parameter(__attribute__((unused)) rtsp_conn_info *conn, r
if ((req->content) && (req->contentlength == strlen("volume\r\n")) &&
strstr(req->content, "volume") == req->content) {
debug(2, "Connection %d: Current volume (%.6f) requested", conn->connection_number,
debug(1, "Connection %d: Current volume (%.6f) requested", conn->connection_number,
config.airplay_volume);
char *p = malloc(128); // will be automatically deallocated with the response is deleted
if (p) {