{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"!
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);
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) {
projects / thirdparty / shairport-sync.git / commitdiff
