From: Mike Brady <4265913+mikebrady@users.noreply.github.com> Date: Sat, 15 Oct 2022 12:31:32 +0000 (+0100) Subject: Check for the existence of a session key when starting to play AP2, and drop the... X-Git-Tag: 4.1-rc3~1^2~6 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=7004c81a28b42cd33a5cdc9c3f4e2c1d852c63c5;p=thirdparty%2Fshairport-sync.git Check for the existence of a session key when starting to play AP2, and drop the connection if not. --- diff --git a/rtsp.c b/rtsp.c index 62da82e5..6fb60824 100644 --- a/rtsp.c +++ b/rtsp.c @@ -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 - - 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; - } - } + // more stuff + // set up a UDP control stream and thread and a UDP or TCP audio stream and 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); - - 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); - - pthread_create(&conn->rtp_realtime_audio_thread, NULL, &rtp_realtime_audio_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)); - - 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: 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 DACP-ID and Active Remote for remote control stuff + + 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); - - // 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 + 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; + } + } - pthread_create(&conn->rtp_buffered_audio_thread, NULL, &rtp_buffered_audio_processor, - (void *)conn); + // 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 + 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); + + pthread_create(&conn->rtp_realtime_audio_thread, NULL, &rtp_realtime_audio_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)); + + 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 - 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)); + // 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); + } - // 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); - } + 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_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; + 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."); + } } 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) {