MyFavMirrors
/
owntone-server
mirror of https://github.com/owntone/owntone-server.git
1
0
Fork 0
Code Issues Packages Projects Releases Wiki Activity
owntone-server/src/outputs/cast.c

2468 lines
69 KiB
C
Raw Permalink Blame History

/*
* Copyright (C) 2015-2019 Espen Jürgensen <espenjurgensen@gmail.com>
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <errno.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <arpa/inet.h>
#include <net/if.h>
#include <netinet/in.h>
#include <ifaddrs.h>
#include <unistd.h>
#include <fcntl.h>
#ifdef HAVE_ENDIAN_H
# include <endian.h>
#elif defined(HAVE_SYS_ENDIAN_H)
# include <sys/endian.h>
#elif defined(HAVE_LIBKERN_OSBYTEORDER_H)
#include <libkern/OSByteOrder.h>
#define htobe32(x) OSSwapHostToBigInt32(x)
#define be32toh(x) OSSwapBigToHostInt32(x)
#endif
#include <gnutls/gnutls.h>
#include <event2/event.h>
#include <json.h>
#include "conffile.h"
#include "misc.h"
#include "mdns.h"
#include "transcode.h"
#include "logger.h"
#include "player.h"
#include "rtp_common.h"
#include "outputs.h"
#include "db.h"
#include "artwork.h"
#ifdef HAVE_PROTOBUF_OLD
#include "cast_channel.v0.pb-c.h"
#else
#include "cast_channel.pb-c.h"
#endif
// This implementation of Chromecast uses the same mirroring app that Chromium
// uses. This app supports RTP-streaming, which has the advantage of quick
// startup and similarity with Airplay. However, I have not found much
// documentation for it, so the reference is Chromium itself. Here's how to
// start a Chromium mirroring session with verbose logging:
//
// 1) chromium --user-data-dir=~/chromium --enable-logging --v=1 ~/Music/test.mp3
// 2) right-click, select Cast, select device
// 3) log will now be in ~/chromium/chrome_debug.log
// Number of bytes to request from TLS connection
#define MAX_BUF 4096
// CA file location (not very portable...?)
#define CAFILE "/etc/ssl/certs/ca-certificates.crt"
// Seconds without a heartbeat from the Chromecast before we close the session
//#define HEARTBEAT_TIMEOUT 30
// Seconds to wait for a reply before making the callback requested by caller
#define REPLY_TIMEOUT 5
// ID of the audio mirroring app used by Chrome (Google Home)
#define CAST_APP_ID "85CDB22F"
// Old mirroring app (Chromecast)
#define CAST_APP_ID_OLD "0F5096E8"
// Namespaces
#define NS_CONNECTION "urn:x-cast:com.google.cast.tp.connection"
#define NS_RECEIVER "urn:x-cast:com.google.cast.receiver"
#define NS_HEARTBEAT "urn:x-cast:com.google.cast.tp.heartbeat"
#define NS_MEDIA "urn:x-cast:com.google.cast.media"
#define NS_WEBRTC "urn:x-cast:com.google.cast.webrtc"
#define USE_TRANSPORT_ID (1 << 1)
#define USE_REQUEST_ID (1 << 2)
#define USE_REQUEST_ID_ONLY (1 << 3)
#define CALLBACK_REGISTER_SIZE 32
// Chromium will send OPUS encoded 10 ms packets (48kHz), about 120 bytes. We
// use a 20 ms packet, so 50 pkts/sec, because that's the default for ffmpeg.
// A 20 ms audio packet at 48000 kHz makes this number 48000 * (20 / 1000)
#define CAST_SAMPLES_PER_PACKET 960
#define CAST_QUALITY_SAMPLE_RATE_DEFAULT 48000
#define CAST_QUALITY_BITS_PER_SAMPLE_DEFAULT 16
#define CAST_QUALITY_CHANNELS_DEFAULT 2
// This is an arbitrary value which just needs to be kept in sync with the config
#define CAST_CONFIG_MAX_VOLUME 11
// This makes the rtp session buffer 6 seconds of audio (6 sec * 50 pkts/sec),
// which can be used for delayed transmission (and retransmission)
#define CAST_PACKET_BUFFER_SIZE 300
// Max number of RTP packets for one artwork image
#define CAST_PACKET_ARTWORK_SIZE 200
// Max (absolute) value the user is allowed to set offset_ms in the config file
#define CAST_OFFSET_MAX 1000
// This is just my measurement of how much extra delay is required to start at
// the same time as Airplay. The value was found experimentally.
#define CAST_DEVICE_START_DELAY_MS 100
// See cast_packet_header_make()
#define CAST_HEADER_SIZE 11
// These limits are from components/mirroring/service/session.cc
#define CAST_SSRC_AUDIO_MIN 1
#define CAST_SSRC_AUDIO_MAX 500000
#define CAST_SSRC_VIDEO_MIN 500001
#define CAST_SSRC_VIDEO_MAX 1000000
#define CAST_RTP_PAYLOADTYPE_AUDIO 127
#define CAST_RTP_PAYLOADTYPE_VIDEO 96
/* Notes
* OFFER/ANSWER <-webrtc
* RTCP/RTP
* XR custom receiver report
* Control and data on same UDP connection
* OPUS encoded
*/
//#define DEBUG_CHROMECAST 1
struct cast_session;
struct cast_msg_payload;
static struct encode_ctx *cast_encode_ctx;
static struct evbuffer *cast_encoded_data;
typedef void (*cast_reply_cb)(struct cast_session *cs, struct cast_msg_payload *payload);
// Session is starting up
#define CAST_STATE_F_STARTUP (1 << 13)
// The receiver app is ready
#define CAST_STATE_F_APP_READY (1 << 14)
// Media is playing in the receiver app
#define CAST_STATE_F_STREAMING (1 << 15)
// Beware, the order of this enum has meaning
enum cast_state
{
// Something bad happened during a session
CAST_STATE_FAILED = 0,
// No session allocated
CAST_STATE_NONE = 1,
// Session allocated, but no connection
CAST_STATE_DISCONNECTED = CAST_STATE_F_STARTUP | 0x01,
// TCP connect, TLS handshake, CONNECT and GET_STATUS request
CAST_STATE_CONNECTED = CAST_STATE_F_STARTUP | 0x02,
// Receiver app has been launched
CAST_STATE_APP_LAUNCHED = CAST_STATE_F_STARTUP | 0x03,
// CONNECT, GET_STATUS and OFFER made to receiver app
CAST_STATE_APP_READY = CAST_STATE_F_APP_READY,
// Buffering packets (playback not started yet)
CAST_STATE_BUFFERING = CAST_STATE_F_APP_READY | 0x01,
// Streaming (playback started)
CAST_STATE_STREAMING = CAST_STATE_F_APP_READY | CAST_STATE_F_STREAMING,
};
struct cast_master_session
{
struct evbuffer *evbuf;
int evbuf_samples;
struct rtp_session *rtp_session;
struct media_quality quality;
uint8_t *rawbuf;
size_t rawbuf_size;
int samples_per_packet;
struct rtp_session *rtp_artwork;
};
struct cast_session
{
uint64_t device_id;
int callback_id;
struct cast_master_session *master_session;
// Current state
enum cast_state state;
// Used to register a target state if we are transitioning from one to another
enum cast_state wanted_state;
// Connection fd and session, and listener event
int server_fd;
gnutls_session_t tls_session;
struct event *ev;
char *devname;
char *address;
int family;
unsigned short port;
// ChromeCast uses a float between 0 - 1
float volume;
uint32_t ssrc_id;
// For initial buffering (delay playback to achieve some sort of sync).
struct timespec start_pts;
struct timespec offset_ts;
uint16_t seqnum_next;
uint16_t ack_last;
// Outgoing request which have the USE_REQUEST_ID flag get a new id, and a
// callback is registered. The callback is called when an incoming message
// from the peer with that request id arrives. If nothing arrives within
// REPLY_TIMEOUT we make the callback with a NULL payload pointer.
unsigned int request_id;
cast_reply_cb callback_register[CALLBACK_REGISTER_SIZE];
struct event *reply_timeout;
// This is used to work around a bug where no response is given by the device.
// For certain requests, we will then retry, e.g. by checking status. We
// register our retry so that we on only retry once.
int retry;
// Session info from the Chromecast
char *transport_id;
char *session_id;
unsigned int media_session_id;
int udp_fd;
unsigned short udp_port;
struct event *rtcp_ev;
struct cast_session *next;
};
enum cast_msg_types
{
UNKNOWN,
PING,
PONG,
CONNECT,
CLOSE,
GET_STATUS,
RECEIVER_STATUS,
LAUNCH,
LAUNCH_OLD,
LAUNCH_ERROR,
STOP,
MEDIA_CONNECT,
MEDIA_CLOSE,
OFFER,
ANSWER,
MEDIA_GET_STATUS,
MEDIA_STATUS,
SET_VOLUME,
PRESENTATION,
GET_CAPABILITIES,
CAPABILITIES_RESPONSE,
};
struct cast_msg_basic
{
enum cast_msg_types type;
char *tag; // Used for looking up incoming message type
char *namespace;
char *payload;
int flags;
};
struct cast_msg_payload
{
enum cast_msg_types type;
unsigned int request_id;
const char *app_id;
const char *session_id;
const char *transport_id;
const char *player_state;
const char *result;
unsigned int media_session_id;
unsigned short udp_port;
};
struct cast_rtcp_packet_feedback
{
uint8_t frame_id_last;
uint8_t num_lost_fields;
struct cast_rtcp_lost_fields
{
uint8_t frame_id;
uint16_t packet_id;
uint8_t bitmask;
} lost_fields[32]; // From observation we normally get just 1 or 2 elements, so 32 should be plenty
uint16_t target_delay_ms;
uint8_t count;
uint8_t recv_fields;
};
struct cast_metadata
{
struct evbuffer *artwork;
};
// Array of the cast messages that we use. Must be in sync with cast_msg_types.
struct cast_msg_basic cast_msg[] =
{
{
.type = UNKNOWN,
.namespace = "",
.payload = "",
},
{
.type = PING,
.tag = "PING",
.namespace = NS_HEARTBEAT,
.payload = "{'type':'PING'}",
},
{
.type = PONG,
.tag = "PONG",
.namespace = NS_HEARTBEAT,
.payload = "{'type':'PONG'}",
},
{
.type = CONNECT,
.namespace = NS_CONNECTION,
.payload = "{'type':'CONNECT'}",
// msg.payload_utf8 = "{\"origin\":{},\"userAgent\":\"owntone\",\"type\":\"CONNECT\",\"senderInfo\":{\"browserVersion\":\"44.0.2403.30\",\"version\":\"15.605.1.3\",\"connectionType\":1,\"platform\":4,\"sdkType\":2,\"systemVersion\":\"Macintosh; Intel Mac OS X10_10_3\"}}";
},
{
.type = CLOSE,
.tag = "CLOSE",
.namespace = NS_CONNECTION,
.payload = "{'type':'CLOSE'}",
},
{
.type = GET_STATUS,
.namespace = NS_RECEIVER,
.payload = "{'type':'GET_STATUS','requestId':%u}",
.flags = USE_REQUEST_ID_ONLY,
},
{
.type = RECEIVER_STATUS,
.tag = "RECEIVER_STATUS",
},
{
.type = LAUNCH,
.namespace = NS_RECEIVER,
.payload = "{'type':'LAUNCH','requestId':%u,'appId':'" CAST_APP_ID "'}",
.flags = USE_REQUEST_ID_ONLY,
},
{
.type = LAUNCH_OLD,
.namespace = NS_RECEIVER,
.payload = "{'type':'LAUNCH','requestId':%u,'appId':'" CAST_APP_ID_OLD "'}",
.flags = USE_REQUEST_ID_ONLY,
},
{
.type = LAUNCH_ERROR,
.tag = "LAUNCH_ERROR",
},
{
.type = STOP,
.namespace = NS_RECEIVER,
.payload = "{'type':'STOP','sessionId':'%s','requestId':%u}",
.flags = USE_REQUEST_ID,
},
{
.type = MEDIA_CONNECT,
.namespace = NS_CONNECTION,
.payload = "{'type':'CONNECT'}",
.flags = USE_TRANSPORT_ID,
},
{
.type = MEDIA_CLOSE,
.namespace = NS_CONNECTION,
.payload = "{'type':'CLOSE'}",
.flags = USE_TRANSPORT_ID,
},
{
.type = OFFER,
.namespace = NS_WEBRTC,
// codecName can be aac or opus, ssrc should be random
// We don't set 'aesKey' and 'aesIvMask'
// sampleRate seems to be ignored
// TODO calculate bitrate, result should be 102000, ref. Chromium
// storeTime unknown meaning - perhaps size of buffer?
// targetDelay - should be RTP delay in ms, but doesn't seem to change anything?
// vp8 timebase - see rfc7741
.payload = "{'type':'OFFER','seqNum':%u,'offer':{'castMode':'mirroring','supportedStreams':[{'index':0,'type':'audio_source','codecName':'opus','rtpProfile':'cast','rtpPayloadType':" NTOSTR(CAST_RTP_PAYLOADTYPE_AUDIO) ",'ssrc':%" PRIu32 ",'storeTime':400,'targetDelay':400,'bitRate':128000,'sampleRate':" NTOSTR(CAST_QUALITY_SAMPLE_RATE_DEFAULT) ",'timeBase':'1/" NTOSTR(CAST_QUALITY_SAMPLE_RATE_DEFAULT) "','channels':" NTOSTR(CAST_QUALITY_CHANNELS_DEFAULT) ",'receiverRtcpEventLog':false},{'codecName':'vp8','index':1,'maxBitRate':5000000,'maxFrameRate':'30000/1000','receiverRtcpEventLog':false,'renderMode':'video','resolutions':[{'height':900,'width':1600}],'rtpPayloadType':" NTOSTR(CAST_RTP_PAYLOADTYPE_VIDEO) ",'rtpProfile':'cast','ssrc':999999,'targetDelay':400,'timeBase':'1/90000','type':'video_source'}]}}",
.flags = USE_TRANSPORT_ID | USE_REQUEST_ID,
},
{
.type = ANSWER,
.tag = "ANSWER",
},
{
.type = MEDIA_GET_STATUS,
.namespace = NS_MEDIA,
.payload = "{'type':'GET_STATUS','requestId':%u}",
.flags = USE_TRANSPORT_ID | USE_REQUEST_ID_ONLY,
},
{
.type = MEDIA_STATUS,
.tag = "MEDIA_STATUS",
},
{
.type = SET_VOLUME,
.namespace = NS_RECEIVER,
.payload = "{'type':'SET_VOLUME','volume':{'level':%.2f,'muted':0},'requestId':%u}",
.flags = USE_REQUEST_ID,
},
{
.type = PRESENTATION,
.namespace = NS_WEBRTC,
.payload = "{'type':'PRESENTATION','sessionId':'%s','seqNum':%u,'title':'" PACKAGE_NAME "','icons':[{'url':'http://www.gyfgafguf.dk/images/fugl.jpg'}] }",
.flags = USE_TRANSPORT_ID | USE_REQUEST_ID,
},
{
// This message is useful for diagnostics, since it will return the
// codecs that the device supports, but doesn't work for all devices
.type = GET_CAPABILITIES,
.namespace = NS_WEBRTC,
.payload = "{'type':'GET_CAPABILITIES','seqNum':%u}",
.flags = USE_TRANSPORT_ID | USE_REQUEST_ID_ONLY,
},
{
.type = CAPABILITIES_RESPONSE,
.tag = "CAPABILITIES_RESPONSE",
},
{
.type = 0,
},
};
/* From player.c */
extern struct event_base *evbase_player;
/* Globals */
static gnutls_certificate_credentials_t tls_credentials;
static struct cast_session *cast_sessions;
static struct cast_master_session *cast_master_session;
//static struct timeval heartbeat_timeout = { HEARTBEAT_TIMEOUT, 0 };
static struct timeval reply_timeout = { REPLY_TIMEOUT, 0 };
static struct media_quality cast_quality_default = { CAST_QUALITY_SAMPLE_RATE_DEFAULT, CAST_QUALITY_BITS_PER_SAMPLE_DEFAULT, CAST_QUALITY_CHANNELS_DEFAULT, 0 };
/* ------------------------------- MISC HELPERS ----------------------------- */
static void
cast_disconnect(int fd)
{
/* no more receptions */
shutdown(fd, SHUT_RDWR);
close(fd);
}
/*static void
cast_metadata_free(struct cast_metadata *cmd)
{
if (!cmd)
return;
if (cmd->artwork)
evbuffer_free(cmd->artwork);
free(cmd);
}
*/
static char *
squote_to_dquote(char *buf)
{
char *ptr;
for (ptr = buf; *ptr != '\0'; ptr++)
if (*ptr == '\'')
*ptr = '"';
return buf;
}
/* ----------------------------- SESSION CLEANUP ---------------------------- */
static void
master_session_free(struct cast_master_session *cms)
{
if (!cms)
return;
outputs_quality_unsubscribe(&cms->rtp_session->quality);
rtp_session_free(cms->rtp_session);
rtp_session_free(cms->rtp_artwork);
evbuffer_free(cms->evbuf);
free(cms->rawbuf);
free(cms);
}
static void
master_session_cleanup(struct cast_master_session *cms)
{
struct cast_session *cs;
// First check if any other session is using the master session
for (cs = cast_sessions; cs; cs=cs->next)
{
if (cs->master_session == cms)
return;
}
if (cms == cast_master_session)
cast_master_session = NULL;
master_session_free(cms);
}
static void
cast_session_free(struct cast_session *cs)
{
if (!cs)
return;
master_session_cleanup(cs->master_session);
event_free(cs->reply_timeout);
event_free(cs->ev);
if (cs->server_fd >= 0)
cast_disconnect(cs->server_fd);
if (cs->rtcp_ev)
event_free(cs->rtcp_ev);
if (cs->udp_fd >= 0)
cast_disconnect(cs->udp_fd);
gnutls_deinit(cs->tls_session);
free(cs->address);
free(cs->devname);
free(cs->session_id);
free(cs->transport_id);
free(cs);
}
static void
cast_session_cleanup(struct cast_session *cs)
{
struct cast_session *s;
if (cs == cast_sessions)
cast_sessions = cast_sessions->next;
else
{
for (s = cast_sessions; s && (s->next != cs); s = s->next)
; /* EMPTY */
if (!s)
DPRINTF(E_WARN, L_CAST, "WARNING: struct cast_session not found in list; BUG!\n");
else
s->next = cs->next;
}
outputs_device_session_remove(cs->device_id);
cast_session_free(cs);
}
// Forward
static void
cast_session_shutdown(struct cast_session *cs, enum cast_state wanted_state);
/* --------------------------- CAST MESSAGE HANDLING ------------------------ */
static int
cast_msg_send(struct cast_session *cs, enum cast_msg_types type, cast_reply_cb reply_cb)
{
Extensions__CoreApi__CastChannel__CastMessage msg = EXTENSIONS__CORE_API__CAST_CHANNEL__CAST_MESSAGE__INIT;
char msg_buf[MAX_BUF];
uint8_t buf[MAX_BUF];
uint32_t be;
size_t len;
int ret;
#ifdef DEBUG_CHROMECAST
DPRINTF(E_DBG, L_CAST, "Preparing to send message type %d to '%s'\n", type, cs->devname);
#endif
msg.source_id = "sender-0";
msg.namespace_ = cast_msg[type].namespace;
if ((cast_msg[type].flags & USE_TRANSPORT_ID) && !cs->transport_id)
{
DPRINTF(E_LOG, L_CAST, "Error, didn't get transportId for message (type %d) to '%s'\n", type, cs->devname);
return -1;
}
if (cast_msg[type].flags & USE_TRANSPORT_ID)
msg.destination_id = cs->transport_id;
else
msg.destination_id = "receiver-0";
if (cast_msg[type].flags & (USE_REQUEST_ID | USE_REQUEST_ID_ONLY))
{
cs->request_id++;
if (reply_cb)
{
cs->callback_register[cs->request_id % CALLBACK_REGISTER_SIZE] = reply_cb;
event_add(cs->reply_timeout, &reply_timeout);
}
}
// Special handling of some message types
if (cast_msg[type].flags & USE_REQUEST_ID_ONLY)
snprintf(msg_buf, sizeof(msg_buf), cast_msg[type].payload, cs->request_id);
else if (type == STOP)
snprintf(msg_buf, sizeof(msg_buf), cast_msg[type].payload, cs->session_id, cs->request_id);
else if (type == OFFER)
snprintf(msg_buf, sizeof(msg_buf), cast_msg[type].payload, cs->request_id, cs->ssrc_id);
else if (type == PRESENTATION)
snprintf(msg_buf, sizeof(msg_buf), cast_msg[type].payload, cs->session_id, cs->request_id);
else if (type == SET_VOLUME)
snprintf(msg_buf, sizeof(msg_buf), cast_msg[type].payload, cs->volume, cs->request_id);
else
snprintf(msg_buf, sizeof(msg_buf), "%s", cast_msg[type].payload);
squote_to_dquote(msg_buf);
msg.payload_utf8 = msg_buf;
len = extensions__core_api__cast_channel__cast_message__get_packed_size(&msg);
if (len <= 0 || len >= sizeof(buf) - 4)
{
DPRINTF(E_LOG, L_CAST, "Could not send message (type %d), invalid length: %zu\n", type, len);
return -1;
}
// The message must be prefixed with Big-Endian 32 bit length
be = htobe32(len);
memcpy(buf, &be, 4);
// Now add the packed message and send it
extensions__core_api__cast_channel__cast_message__pack(&msg, buf + 4);
ret = gnutls_record_send(cs->tls_session, buf, len + 4);
if (ret < 0)
{
DPRINTF(E_LOG, L_CAST, "Could not send message, TLS error\n");
return -1;
}
else if (ret != len + 4)
{
DPRINTF(E_LOG, L_CAST, "BUG! Message partially sent, and we are not able to send the rest\n");
return -1;
}
if (type != PONG)
DPRINTF(E_DBG, L_CAST, "TX %zu %s %s %s %s\n", len, msg.source_id, msg.destination_id, msg.namespace_, msg.payload_utf8);
return 0;
}
static void *
cast_msg_parse(struct cast_msg_payload *payload, char *s)
{
json_object *haystack;
json_object *somehay;
json_object *needle;
const char *val;
int i;
haystack = json_tokener_parse(s);
if (!haystack)
{
DPRINTF(E_LOG, L_CAST, "JSON parser returned an error\n");
return NULL;
}
payload->type = UNKNOWN;
if (json_object_object_get_ex(haystack, "type", &needle))
{
val = json_object_get_string(needle);
for (i = 1; cast_msg[i].type; i++)
{
if (cast_msg[i].tag && (strcmp(val, cast_msg[i].tag) == 0))
{
payload->type = cast_msg[i].type;
break;
}
}
}
if (json_object_object_get_ex(haystack, "requestId", &needle))
payload->request_id = json_object_get_int(needle);
else if (json_object_object_get_ex(haystack, "seqNum", &needle))
payload->request_id = json_object_get_int(needle);
if (json_object_object_get_ex(haystack, "answer", &somehay) &&
json_object_object_get_ex(somehay, "udpPort", &needle) &&
json_object_get_type(needle) == json_type_int )
payload->udp_port = json_object_get_int(needle);
if (json_object_object_get_ex(haystack, "result", &needle) &&
json_object_get_type(needle) == json_type_string )
payload->result = json_object_get_string(needle);
// Might be done now
if ((payload->type != RECEIVER_STATUS) && (payload->type != MEDIA_STATUS))
return haystack;
// Isn't this marvelous
if ( json_object_object_get_ex(haystack, "status", &needle) &&
(json_object_get_type(needle) == json_type_array) &&
(somehay = json_object_array_get_idx(needle, 0)) )
{
if ( json_object_object_get_ex(somehay, "mediaSessionId", &needle) &&
(json_object_get_type(needle) == json_type_int) )
payload->media_session_id = json_object_get_int(needle);
if ( json_object_object_get_ex(somehay, "playerState", &needle) &&
(json_object_get_type(needle) == json_type_string) )
payload->player_state = json_object_get_string(needle);
}
if ( json_object_object_get_ex(haystack, "status", &somehay) &&
json_object_object_get_ex(somehay, "applications", &needle) &&
(json_object_get_type(needle) == json_type_array) &&
(somehay = json_object_array_get_idx(needle, 0)) )
{
if ( json_object_object_get_ex(somehay, "appId", &needle) &&
(json_object_get_type(needle) == json_type_string) )
payload->app_id = json_object_get_string(needle);
if ( json_object_object_get_ex(somehay, "sessionId", &needle) &&
(json_object_get_type(needle) == json_type_string) )
payload->session_id = json_object_get_string(needle);
if ( json_object_object_get_ex(somehay, "transportId", &needle) &&
(json_object_get_type(needle) == json_type_string) )
payload->transport_id = json_object_get_string(needle);
}
return haystack;
}
static void
cast_msg_parse_free(void *haystack)
{
#ifdef HAVE_JSON_C_OLD
json_object_put((json_object *)haystack);
#else
if (json_object_put((json_object *)haystack) != 1)
DPRINTF(E_LOG, L_CAST, "Memleak: JSON parser did not free object\n");
#endif
}
static void
cast_msg_process(struct cast_session *cs, const uint8_t *data, size_t len)
{
Extensions__CoreApi__CastChannel__CastMessage *reply;
cast_reply_cb reply_cb;
struct cast_msg_payload payload = { 0 };
void *hdl;
int unknown_session_id;
int i;
#ifdef DEBUG_CHROMECAST
char *b64 = b64_encode(data, len);
if (b64)
{
DPRINTF(E_DBG, L_CAST, "Reply dump (len %zu): %s\n", len, b64);
free(b64);
}
#endif
reply = extensions__core_api__cast_channel__cast_message__unpack(NULL, len, data);
if (!reply)
{
DPRINTF(E_LOG, L_CAST, "Could not unpack message!\n");
return;
}
hdl = cast_msg_parse(&payload, reply->payload_utf8);
if (!hdl)
{
DPRINTF(E_DBG, L_CAST, "Could not parse message: %s\n", reply->payload_utf8);
goto out_free_unpacked;
}
if (payload.type == PING)
{
cast_msg_send(cs, PONG, NULL);
goto out_free_parsed;
}
DPRINTF(E_DBG, L_CAST, "RX %zu %s %s %s %s\n", len, reply->source_id, reply->destination_id, reply->namespace_, reply->payload_utf8);
if (payload.type == UNKNOWN)
goto out_free_parsed;
i = payload.request_id % CALLBACK_REGISTER_SIZE;
if (payload.request_id && cs->callback_register[i])
{
reply_cb = cs->callback_register[i];
cs->callback_register[i] = NULL;
// Cancel the timeout if no pending callbacks
for (i = 0; (i < CALLBACK_REGISTER_SIZE) && (!cs->callback_register[i]); i++);
if (i == CALLBACK_REGISTER_SIZE)
evtimer_del(cs->reply_timeout);
reply_cb(cs, &payload);
goto out_free_parsed;
}
// TODO Should we read volume and playerstate changes from the Chromecast?
if (payload.type == RECEIVER_STATUS && (cs->state & CAST_STATE_F_APP_READY))
{
unknown_session_id = payload.session_id && (strcmp(payload.session_id, cs->session_id) != 0);
if (unknown_session_id)
{
DPRINTF(E_LOG, L_CAST, "Our session '%s' on '%s' was lost to session '%s'\n", cs->session_id, cs->devname, payload.session_id);
// Downgrade state, we don't have the receiver app any more
cs->state = CAST_STATE_CONNECTED;
cast_session_shutdown(cs, CAST_STATE_FAILED);
goto out_free_parsed;
}
}
if (payload.type == CLOSE && (cs->state & CAST_STATE_F_APP_READY))
{
// Downgrade state, we can't write any more
cs->state = CAST_STATE_CONNECTED;
cast_session_shutdown(cs, CAST_STATE_FAILED);
goto out_free_parsed;
}
if (payload.type == MEDIA_STATUS && (cs->state & CAST_STATE_F_STREAMING))
{
if (payload.player_state && (strcmp(payload.player_state, "PAUSED") == 0))
{
DPRINTF(E_WARN, L_CAST, "Something paused our session on '%s'\n", cs->devname);
/* cs->state = CAST_STATE_APP_READY;
// Kill the session, the player will need to restart it
cast_session_shutdown(cs, CAST_STATE_NONE);
goto out_free_parsed;
*/ }
}
out_free_parsed:
cast_msg_parse_free(hdl);
out_free_unpacked:
extensions__core_api__cast_channel__cast_message__free_unpacked(reply, NULL);
}
/* ------------------ PREPARING AND SENDING CAST RTP PACKETS ---------------- */
// Makes a Cast RTP packet (source: Chromium's media/cast/net/rtp/rtp_packetizer.cc)
//
// A Cast RTP packet is made of:
// RTP header (12 bytes)
// Cast header (7 bytes)
// Extension data (4 bytes)
// Packet data
//
// The Cast header + extension (optional?) consists of:
// 0 1 2 3
// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// |k|r| n_ext | frame_id | packet id |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | max_packet_id | ref_frame_id | ext_type |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
// | ext_size | new_playout_delay_ms |
// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
//
// k: Is the frame a key frame?
// r: Is there a reference frame id?
// n_ext: Number of Cast extensions (Chromium uses 1: Adaptive Latency)
// ext_type: 0x04 Adaptive Latency extension
// ext_size: 0x02 -> 2 bytes
// new_playout_delay_ms: ??
// OPUS encodes the rawbuf payload
static int
payload_encode(struct evbuffer *evbuf, uint8_t *rawbuf, size_t rawbuf_size, int nsamples, struct media_quality *quality)
{
transcode_frame *frame;
int len;
frame = transcode_frame_new(rawbuf, rawbuf_size, nsamples, quality);
if (!frame)
{
DPRINTF(E_LOG, L_CAST, "Could not convert raw PCM to frame (bufsize=%zu)\n", rawbuf_size);
return -1;
}
len = transcode_encode(evbuf, cast_encode_ctx, frame, 0);
transcode_frame_free(frame);
if (len < 0)
{
DPRINTF(E_LOG, L_CAST, "Could not Opus encode frame\n");
return -1;
}
return len;
}
static int
packet_prepare(struct rtp_packet *pkt, struct evbuffer *evbuf)
{
// Cast header
memset(pkt->payload, 0, CAST_HEADER_SIZE);
pkt->payload[0] = 0xc1; // k = 1, r = 1 and one extension
// frame_id - this is the value that is returned when the packet is ack'ed
// Chromecasts possibly expect this to start at zero, sinze when we start
// non-zero we get ack's all the way from zero to our value. We don't start at
// zero because we can't do that for devices that join anyway.
pkt->payload[1] = (char)pkt->seqnum;
// packet_id and max_packet_id don't seem to be used, so leave them at 0
pkt->payload[6] = (char)pkt->seqnum;
pkt->payload[7] = 0x04; // kCastRtpExtensionAdaptiveLatency has id (1 << 2)
pkt->payload[8] = 0x02; // Extension will use two bytes
// leave extension values at 0, but Chromium sets them to:
// (frame.new_playout_delay_ms >> 8) and frame.new_playout_delay_ms (normal byte values are 0x03 0x20)
// Copy payload
return evbuffer_remove(evbuf, pkt->payload + CAST_HEADER_SIZE, pkt->payload_len - CAST_HEADER_SIZE);
}
static int
packet_make(struct cast_master_session *cms)
{
struct rtp_packet *pkt;
int len;
int ret;
// Encode payload into cast_encoded_data
len = payload_encode(cast_encoded_data, cms->rawbuf, cms->rawbuf_size, cms->samples_per_packet, &cms->quality);
if (len < 0)
return -1;
// For audio it is always a complete frame, so marker bit is 1 (like Chromium does)
pkt = rtp_packet_next(cms->rtp_session, CAST_HEADER_SIZE + len, cms->samples_per_packet, CAST_RTP_PAYLOADTYPE_AUDIO, 1);
// Creates Cast header + adds payload
ret = packet_prepare(pkt, cast_encoded_data);
if (ret < 0)
return -1;
// Commits packet to retransmit buffer, and prepares the session for the next packet
rtp_packet_commit(cms->rtp_session, pkt);
return 0;
}
static inline int
packets_make(struct cast_master_session *cms, struct output_data *odata)
{
int ret;
int npkts;
// TODO avoid this copy
evbuffer_add(cms->evbuf, odata->buffer, odata->bufsize);
cms->evbuf_samples += odata->samples;
// Make as many packets as we have data for (one packet requires rawbuf_size bytes)
npkts = 0;
while (evbuffer_get_length(cms->evbuf) >= cms->rawbuf_size)
{
evbuffer_remove(cms->evbuf, cms->rawbuf, cms->rawbuf_size);
cms->evbuf_samples -= cms->samples_per_packet;
ret = packet_make(cms);
if (ret == 0)
npkts++;
}
return npkts;
}
static int
packet_send(struct cast_session *cs, uint16_t seqnum)
{
struct rtp_session *rtp_session = cs->master_session->rtp_session;
struct rtp_packet *pkt;
int ret;
pkt = rtp_packet_get(rtp_session, seqnum);
if (!pkt)
{
DPRINTF(E_WARN, L_CAST, "Packet to '%s' is missing in our buffer\n", cs->devname);
return 0; // Don't fail session over a missing packet (or should we?)
}
ret = send(cs->udp_fd, pkt->data, pkt->data_len, 0);
if (ret < 0)
{
DPRINTF(E_LOG, L_CAST, "Send error for '%s': %s\n", cs->devname, strerror(errno));
return -1;
}
else if (ret != pkt->data_len)
{
DPRINTF(E_WARN, L_CAST, "Partial send (%d) for '%s'\n", ret, cs->devname);
}
/*
DPRINTF(E_DBG, L_CAST, "Sent RTP PACKET seqnum %u, have until %u, payload 0x%x, pktbuf_s %zu to '%s'\n",
seqnum,
cs->master_session->rtp_session->seqnum,
pkt->header[1],
cs->master_session->rtp_session->pktbuf_len,
cs->devname);
*/
return 0;
}
static int
packet_send_next(struct cast_session *cs)
{
int ret;
if (cs->seqnum_next == cs->master_session->rtp_session->seqnum)
return 0; // Nothing to send right now
ret = packet_send(cs, cs->seqnum_next);
if (ret < 0)
return ret;
cs->seqnum_next++;
return 0;
}
/* TODO This does not currently work - need to investigate what sync the devices support
static void
packets_sync_send(struct cast_master_session *cms, struct timespec pts)
{
struct rtp_packet *sync_pkt;
struct cast_session *cs;
struct rtcp_timestamp cur_stamp;
struct timespec ts;
bool is_sync_time;
// Check if it is time send a sync packet to sessions that are already running
is_sync_time = rtp_sync_is_time(cms->rtp_session);
// (See raop.c for more comments on sync packets)
cur_stamp.ts.tv_sec = pts.tv_sec;
cur_stamp.ts.tv_nsec = pts.tv_nsec;
clock_gettime(CLOCK_MONOTONIC, &ts);
cur_stamp.pos = cms->rtp_session->pos + cms->evbuf_samples - cms->output_buffer_samples;
for (cs = cast_sessions; cs; cs = cs->next)
{
if (cs->master_session != cms)
continue;
// A device has joined and should get an init sync packet
if (cs->state == CAST_STATE_APP_READY)
{
sync_pkt = rtp_sync_packet_next(cms->rtp_session, &cur_stamp, 0x80);
packet_send(cs, sync_pkt);
DPRINTF(E_DBG, L_CAST, "Start sync packet sent to '%s': cur_pos=%" PRIu32 ", cur_ts=%lu:%lu, now=%lu:%lu, rtptime=%" PRIu32 ",\n",
cs->devname, cur_stamp.pos, cur_stamp.ts.tv_sec, cur_stamp.ts.tv_nsec, ts.tv_sec, ts.tv_nsec, cms->rtp_session->pos);
}
else if (is_sync_time && cs->state == CAST_STATE_STREAMING)
{
sync_pkt = rtp_sync_packet_next(cms->rtp_session, &cur_stamp, 0x80);
packet_send(cs, sync_pkt);
}
}
}
*/
/* -------------------------------- CALLBACKS ------------------------------- */
/* Maps our internal state to the generic output state and then makes a callback
* to the player to tell that state
*/
static void
cast_status(struct cast_session *cs)
{
enum output_device_state state;
switch (cs->state)
{
case CAST_STATE_FAILED:
state = OUTPUT_STATE_FAILED;
break;
case CAST_STATE_NONE:
state = OUTPUT_STATE_STOPPED;
break;
case CAST_STATE_DISCONNECTED ... CAST_STATE_APP_LAUNCHED:
state = OUTPUT_STATE_STARTUP;
break;
case CAST_STATE_APP_READY ... CAST_STATE_BUFFERING:
state = OUTPUT_STATE_CONNECTED;
break;
case CAST_STATE_STREAMING:
state = OUTPUT_STATE_STREAMING;
break;
default:
DPRINTF(E_LOG, L_CAST, "Bug! Unhandled state in cast_status()\n");
state = OUTPUT_STATE_FAILED;
}
outputs_cb(cs->callback_id, cs->device_id, state);
cs->callback_id = -1;
}
/* Process CAST feedback content, which looks like this:
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| "CAST" |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| last frame id | lost fields | target delay ms |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
x lost fields
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| frame id | packet id | bitmask |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| "CST2" |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|feedback count | recv fields |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+
x recv fields
+-+-+-+-+-+-+-+-+
| bitmask |
+-+-+-+-+-+-+-+-+
*/
// Let's say short_id is 0xbd and last seqnum was 0x22 0xbf then we can guess
// that the expansion of the short id is 0x22 0xbd. However, the guessing also
// most work for 0xff and last seqnum 0x23 0x01, where the correct answer would
// be 0x22 0xff, and of course also for wrap around, e.g. 0x00 0x00. So if the
// result is higher than last seqnum we decrease the high order byte.
// See media/cast/common/expanded_value_base.h for Chromium's C++ method.
static inline uint16_t
frame_id_expand(uint8_t short_id, uint16_t seqnum_last)
{
uint16_t short_max = UINT8_MAX;
uint16_t retval = (seqnum_last & ~short_max) | short_id;
if (retval > seqnum_last)
retval -= short_max + 1;
// DPRINTF(E_DBG, L_CAST, "RTCP EXPAND ACK is %" PRIu16 " = %02x, seqnum %02x %02x\n", retval, short_id, seqnum_last >> 8, seqnum_last & 0xff);
return retval;
}
static int
feedback_packet_parse(struct cast_rtcp_packet_feedback *feedback, uint8_t *data, size_t len)
{
size_t require_len;
int i;
memset(feedback, 0, sizeof(struct cast_rtcp_packet_feedback));
// Check that we have enough data to read the header and calc length
require_len = 8;
if (len < require_len)
return -1;
if (memcmp(data, "CAST", 4) != 0)
return -1;
// This is normally the last seqnum received truncated to 8 bit, but when we
// start the stream we will get a series of ACKs going from data[4] = 0 ->
// last seqnum. However, we also get the actual seqnum of the last frame
// received by the peer via CST2's frame_id below. Not sure what the logic
// behind all that is...
feedback->frame_id_last = data[4];
feedback->num_lost_fields = data[5];
memcpy(&feedback->target_delay_ms, data + 6, 2);
feedback->target_delay_ms = be16toh(feedback->target_delay_ms);
// Check len again, now we can calculate required size for next step
require_len += 4 * feedback->num_lost_fields;
if (len < require_len)
return -1;
for (i = 0; i < feedback->num_lost_fields && i < ARRAY_SIZE(feedback->lost_fields); i++)
{
feedback->lost_fields[i].frame_id = data[8 + (4 * i)];
memcpy(&feedback->lost_fields[i].packet_id, data + 9 + (4 * i), 2);
feedback->lost_fields[i].packet_id = be16toh(feedback->lost_fields[i].packet_id);
feedback->lost_fields[i].bitmask = data[11 + (4 * i)];
}
/* Reading the CST2 data is disabled because we don't know what to use the data for right now
uint8_t *cst2_data;
uint16_t starting_frame_id;
uint16_t frame_id;
uint8_t bitmask;
// Check len again, now check if we have enough data to read the CST2 header
require_len += 6;
if (len < require_len)
return -1;
cst2_data = data + 8 + 4 * feedback->num_lost_fields;
if (memcmp(cst2_data, "CST2", 4) != 0)
return -1;
feedback->count = cst2_data[4];
feedback->recv_fields = cst2_data[5];
require_len += feedback->recv_fields;
if (len < require_len)
return -1;
starting_frame_id = cs->ack_last + 2;
for (i = 0; i < feedback->recv_fields; i++)
{
frame_id = starting_frame_id;
for (bitmask = cst2_data[6 + i]; bitmask; bitmask >>= 1)
{
// Here the peer seems to be telling us what the latest frame it has
// received is after the ack'ed frame + 2 (?). Chromium stores these
// in an array, but not sure what the use actually is.
if (bitmask & 1)
DPRINTF(E_SPAM, L_CAST, "RTCP later frame ID is %" PRIu16 "\n", frame_id);
frame_id++;
}
starting_frame_id += 8;
}
// TODO what is the final byte?
*/
return 0;
}
// Process an extended report RTCP packet type (PT=207)
static void
xr_packet_process(struct cast_session *cs, uint8_t *data, size_t len)
{
struct rtcp_packet xrpkt;
struct cast_rtcp_packet_feedback feedback;
uint16_t seqnum;
int ret;
int i;
// The CAST payload is an RTCP packet with packet type 206
ret = rtcp_packet_parse(&xrpkt, data, len);
if (ret < 0)
return;
if (xrpkt.packet_type == RTCP_PACKET_PSFB && xrpkt.psfb.message_type == 15)
{
ret = feedback_packet_parse(&feedback, xrpkt.psfb.fci, xrpkt.psfb.fci_len);
if (ret < 0)
return;
// Retransmission
for (i = 0; i < feedback.num_lost_fields; i++)
{
seqnum = frame_id_expand(feedback.lost_fields[i].frame_id, cs->seqnum_next - 1);
DPRINTF(E_DBG, L_CAST, "Retransmission to '%s' of lost RTCP frame_id %" PRIu16", packet_id %" PRIu16 ", bitmask %02x\n",
cs->devname, seqnum, feedback.lost_fields[i].packet_id, feedback.lost_fields[i].bitmask);
packet_send(cs, seqnum);
}
// Expand the 8 bit value into a seqnum by comparing with last sent seqnum
cs->ack_last = frame_id_expand(feedback.frame_id_last, cs->seqnum_next - 1);
if (cs->ack_last + 1 == cs->seqnum_next)
{
packet_send_next(cs); // Last packet was ack'ed, let's send a new packet
}
}
/* else if (xrpkt.packet_type == CAST_RTCP_PT_FEEDBACK && xrpkt.ic == 1)
picturelost_packet_process(&xrpkt);
else if (xrpkt.packet_type == CAST_RTCP_PT_RECVREPORT)
recvreport_packet_process(&xrpkt);*/
}
static void
cast_rtcp_cb(int fd, short what, void *arg)
{
struct cast_session *cs = arg;
struct rtcp_packet pkt;
ssize_t got;
int ret;
uint8_t buf[512];
got = recv(fd, buf, sizeof(buf), 0);
if (got == sizeof(buf))
return; // Longer than expected, give up
ret = rtcp_packet_parse(&pkt, buf, got);
if (ret < 0)
return;
if (pkt.packet_type == RTCP_PACKET_XR)
{
xr_packet_process(cs, pkt.payload, pkt.payload_len);
}
/* else if (pkt.packet_type == RTCP_PACKET_APP)
app_packet_process(cs, &pkt);
*/
}
/* cast_cb_stop*: Callback chain for shutting down a session */
static void
cast_cb_stop(struct cast_session *cs, struct cast_msg_payload *payload)
{
if (!payload)
DPRINTF(E_LOG, L_CAST, "No RECEIVER_STATUS reply to our STOP - will continue anyway\n");
else if (payload->type != RECEIVER_STATUS)
DPRINTF(E_LOG, L_CAST, "No RECEIVER_STATUS reply to our STOP (got type: %d) - will continue anyway\n", payload->type);
cs->state = CAST_STATE_CONNECTED;
if (cs->state == cs->wanted_state)
cast_status(cs);
else
cast_session_shutdown(cs, cs->wanted_state);
}
/* cast_cb_startup*: Callback chain for starting a session */
static void
cast_cb_startup_volume(struct cast_session *cs, struct cast_msg_payload *payload)
{
/* Session startup and setup is done, tell our user */
DPRINTF(E_DBG, L_CAST, "Session ready\n");
cast_status(cs);
}
static void
cast_cb_startup_offer(struct cast_session *cs, struct cast_msg_payload *payload)
{
int ret;
if (!payload)
{
DPRINTF(E_LOG, L_CAST, "No reply from '%s' to our OFFER request\n", cs->devname);
goto error;
}
else if (payload->type != ANSWER)
{
DPRINTF(E_LOG, L_CAST, "The device '%s' did not give us an ANSWER to our OFFER\n", cs->devname);
goto error;
}
else if (!payload->udp_port || strcmp(payload->result, "ok") != 0)
{
DPRINTF(E_LOG, L_CAST, "Missing UDP port (or unexpected result '%s') in ANSWER - aborting\n", payload->result);
goto error;
}
DPRINTF(E_INFO, L_CAST, "UDP port in ANSWER is %d\n", payload->udp_port);
cs->udp_port = payload->udp_port;
cs->udp_fd = net_connect(cs->address, cs->udp_port, SOCK_DGRAM, "Chromecast data");
if (cs->udp_fd < 0)
goto error;
cs->rtcp_ev = event_new(evbase_player, cs->udp_fd, EV_READ | EV_PERSIST, cast_rtcp_cb, cs);
if (!cs->rtcp_ev)
{
DPRINTF(E_LOG, L_CAST, "Out of memory for UDP read event\n");
goto error;
}
event_add(cs->rtcp_ev, NULL);
ret = cast_msg_send(cs, SET_VOLUME, cast_cb_startup_volume);
if (ret < 0)
goto error;
cs->state = CAST_STATE_APP_READY;
return;
error:
cast_session_shutdown(cs, CAST_STATE_FAILED);
}
#ifdef DEBUG_CHROMECAST
// Not all Chromecast devices support this request, so only use for debug
static void
cast_cb_startup_get_capabilities(struct cast_session *cs, struct cast_msg_payload *payload)
{
int ret;
if (!payload)
{
DPRINTF(E_LOG, L_CAST, "No reply to our GET_CAPABILITIES - aborting\n");
goto error;
}
else if (payload->type != CAPABILITIES_RESPONSE)
{
DPRINTF(E_LOG, L_CAST, "No CAPABILITIES_RESPONSE reply to our GET_CAPABILITIES (got type: %d) - aborting\n", payload->type);
goto error;
}
ret = cast_msg_send(cs, OFFER, cast_cb_startup_offer);
if (ret < 0)
goto error;
return;
error:
cast_session_shutdown(cs, CAST_STATE_FAILED);
}
#endif
static void
cast_cb_startup_media(struct cast_session *cs, struct cast_msg_payload *payload)
{
int ret;
if (!payload)
{
DPRINTF(E_LOG, L_CAST, "No MEDIA_STATUS reply to our GET_STATUS - aborting\n");
goto error;
}
else if (payload->type != MEDIA_STATUS)
{
DPRINTF(E_LOG, L_CAST, "No MEDIA_STATUS reply to our GET_STATUS (got type: %d) - aborting\n", payload->type);
goto error;
}
#ifdef DEBUG_CHROMECAST
ret = cast_msg_send(cs, GET_CAPABILITIES, cast_cb_startup_get_capabilities);
#else
ret = cast_msg_send(cs, OFFER, cast_cb_startup_offer);
#endif
if (ret < 0)
goto error;
return;
error:
cast_session_shutdown(cs, CAST_STATE_FAILED);
}
static void
cast_cb_startup_launch(struct cast_session *cs, struct cast_msg_payload *payload)
{
int ret;
// Sometimes the response to a LAUNCH is just a broadcast RECEIVER_STATUS
// without our requestId. That won't be registered by our response handler,
// and we get an empty callback due to timeout. In this case we send a
// GET_STATUS to see if we are good to go anyway.
if (!payload && !cs->retry)
{
DPRINTF(E_LOG, L_CAST, "No RECEIVER_STATUS reply to our LAUNCH - trying GET_STATUS instead\n");
cs->retry++;
ret = cast_msg_send(cs, GET_STATUS, cast_cb_startup_launch);
if (ret != 0)
goto error;
return;
}
if (!payload)
{
DPRINTF(E_LOG, L_CAST, "No RECEIVER_STATUS reply to our LAUNCH - aborting\n");
goto error;
}
if (payload->type == LAUNCH_ERROR && !cs->retry)
{
DPRINTF(E_WARN, L_CAST, "Device '%s' could not launch app id '%s', trying '%s' instead\n", cs->devname, CAST_APP_ID, CAST_APP_ID_OLD);
cs->retry++;
ret = cast_msg_send(cs, LAUNCH_OLD, cast_cb_startup_launch);
if (ret < 0)
goto error;
return;
}
else if (payload->type == LAUNCH_ERROR)
{
DPRINTF(E_LOG, L_CAST, "Device '%s' could not launch app id '%s' nor '%s' - aborting\n", cs->devname, CAST_APP_ID, CAST_APP_ID_OLD);
goto error;
}
if (payload->type != RECEIVER_STATUS)
{
DPRINTF(E_LOG, L_CAST, "No RECEIVER_STATUS reply to our LAUNCH (got type: %d) - aborting\n", payload->type);
goto error;
}
if (!payload->transport_id || !payload->session_id)
{
DPRINTF(E_LOG, L_CAST, "Missing session id or transport id in RECEIVER_STATUS - aborting\n");
goto error;
}
if (cs->session_id || cs->transport_id)
DPRINTF(E_LOG, L_CAST, "Bug! Memleaking...\n");
cs->session_id = strdup(payload->session_id);
cs->transport_id = strdup(payload->transport_id);
cs->retry = 0;
ret = cast_msg_send(cs, MEDIA_CONNECT, NULL);
if (ret == 0)
ret = cast_msg_send(cs, MEDIA_GET_STATUS, cast_cb_startup_media);
if (ret < 0)
goto error;
cs->state = CAST_STATE_APP_LAUNCHED;
return;
error:
cast_session_shutdown(cs, CAST_STATE_FAILED);
}
static void
cast_cb_startup_connect(struct cast_session *cs, struct cast_msg_payload *payload)
{
int ret;
if (!payload)
{
DPRINTF(E_LOG, L_CAST, "No RECEIVER_STATUS reply to our GET_STATUS - aborting\n");
goto error;
}
else if (payload->type != RECEIVER_STATUS)
{
DPRINTF(E_LOG, L_CAST, "No RECEIVER_STATUS reply to our GET_STATUS (got type: %d) - aborting\n", payload->type);
goto error;
}
ret = cast_msg_send(cs, LAUNCH, cast_cb_startup_launch);
if (ret < 0)
goto error;
cs->state = CAST_STATE_CONNECTED;
return;
error:
cast_session_shutdown(cs, CAST_STATE_FAILED);
}
/* cast_cb_probe: Callback from cast_device_probe */
static void
cast_cb_probe(struct cast_session *cs, struct cast_msg_payload *payload)
{
if (!payload)
{
DPRINTF(E_LOG, L_CAST, "No RECEIVER_STATUS reply to our GET_STATUS - aborting\n");
goto error;
}
else if (payload->type != RECEIVER_STATUS)
{
DPRINTF(E_LOG, L_CAST, "No RECEIVER_STATUS reply to our GET_STATUS (got type: %d) - aborting\n", payload->type);
goto error;
}
cs->state = CAST_STATE_CONNECTED;
cast_status(cs);
cast_session_shutdown(cs, CAST_STATE_NONE);
return;
error:
cast_session_shutdown(cs, CAST_STATE_FAILED);
}
static void
cast_cb_volume(struct cast_session *cs, struct cast_msg_payload *payload)
{
cast_status(cs);
}
/*
static void
cast_cb_presentation(struct cast_session *cs, struct cast_msg_payload *payload)
{
if (!payload)
DPRINTF(E_LOG, L_CAST, "No reply to PRESENTATION request from '%s' - will continue\n", cs->devname);
else if (payload->type != MEDIA_STATUS)
DPRINTF(E_LOG, L_CAST, "Unexpected reply to PRESENTATION request from '%s' - will continue\n", cs->devname);
}
*/
/* The core of this module. Libevent makes a callback to this function whenever
* there is new data to be read on the fd from the ChromeCast. If everything is
* good then the data will be passed to cast_msg_process() that will then
* parse and make callbacks, if relevant.
*/
static void
cast_listen_cb(int fd, short what, void *arg)
{
struct cast_session *cs;
uint8_t buffer[MAX_BUF + 1]; // Not sure about the +1, but is copied from gnutls examples
uint32_t be;
size_t len;
int received;
int ret;
for (cs = cast_sessions; cs; cs = cs->next)
{
if (cs == (struct cast_session *)arg)
break;
}
if (!cs)
{
DPRINTF(E_INFO, L_CAST, "Callback on dead session, ignoring\n");
return;
}
if (what == EV_TIMEOUT)
{
DPRINTF(E_LOG, L_CAST, "No heartbeat from '%s', shutting down\n", cs->devname);
goto fail;
}
#ifdef DEBUG_CHROMECAST
DPRINTF(E_DBG, L_CAST, "New data from '%s'\n", cs->devname);
#endif
// We first read the 4 byte header and then the actual message. The header
// will be the length of the message.
ret = gnutls_record_recv(cs->tls_session, buffer, 4);
if (ret != 4)
goto no_read;
memcpy(&be, buffer, 4);
len = be32toh(be);
if ((len == 0) || (len > MAX_BUF))
{
DPRINTF(E_LOG, L_CAST, "Bad length of incoming message, aborting (len=%zu, size=%d)\n", len, MAX_BUF);
goto fail;
}
received = 0;
while (received < len)
{
ret = gnutls_record_recv(cs->tls_session, buffer + received, len - received);
if (ret <= 0)
goto no_read;
received += ret;
#ifdef DEBUG_CHROMECAST
DPRINTF(E_DBG, L_CAST, "Received %d bytes out of expected %zu bytes\n", received, len);
#endif
}
ret = gnutls_record_check_pending(cs->tls_session);
// Process the message - note that this may result in cs being invalidated
cast_msg_process(cs, buffer, len);
// In the event there was more data waiting for us we go again
if (ret > 0)
{
DPRINTF(E_INFO, L_CAST, "More data pending from device (%d bytes)\n", ret);
cast_listen_cb(fd, what, arg);
}
return;
no_read:
if ((ret != GNUTLS_E_INTERRUPTED) && (ret != GNUTLS_E_AGAIN))
{
DPRINTF(E_LOG, L_CAST, "Session error: %s\n", gnutls_strerror(ret));
goto fail;
}
DPRINTF(E_DBG, L_CAST, "Return value from tls is %d (GNUTLS_E_AGAIN is %d)\n", ret, GNUTLS_E_AGAIN);
return;
fail:
// Downgrade state to make cast_session_shutdown perform an exit which is
// quick and won't require a reponse from the device
cs->state = CAST_STATE_CONNECTED;
cast_session_shutdown(cs, CAST_STATE_FAILED);
}
static void
cast_reply_timeout_cb(int fd, short what, void *arg)
{
struct cast_session *cs;
int i;
cs = (struct cast_session *)arg;
i = cs->request_id % CALLBACK_REGISTER_SIZE;
DPRINTF(E_LOG, L_CAST, "Request %d timed out, will run empty callback\n", i);
if (cs->callback_register[i])
{
cs->callback_register[i](cs, NULL);
cs->callback_register[i] = NULL;
}
}
static void
cast_device_cb(const char *name, const char *type, const char *domain, const char *hostname, int family, const char *address, int port, struct keyval *txt)
{
struct output_device *device;
const char *friendly_name;
cfg_t *devcfg;
uint32_t id;
id = djb_hash(name, strlen(name));
friendly_name = keyval_get(txt, "fn");
if (friendly_name)
name = friendly_name;
DPRINTF(E_DBG, L_CAST, "Event for Chromecast device '%s' (port %d, id %" PRIu32 ")\n", name, port, id);
devcfg = cfg_gettsec(cfg, "chromecast", name);
if (devcfg && cfg_getbool(devcfg, "exclude"))
{
DPRINTF(E_LOG, L_CAST, "Excluding Chromecast device '%s' as set in config\n", name);
return;
}
if (devcfg && cfg_getstr(devcfg, "nickname"))
{
name = cfg_getstr(devcfg, "nickname");
}
device = calloc(1, sizeof(struct output_device));
if (!device)
{
DPRINTF(E_LOG, L_CAST, "Out of memory for new Chromecast device\n");
return;
}
device->id = id;
device->name = strdup(name);
device->type = OUTPUT_TYPE_CAST;
device->type_name = outputs_name(device->type);
if (port < 0)
{
/* Device stopped advertising */
switch (family)
{
case AF_INET:
device->v4_port = 1;
break;
case AF_INET6:
device->v6_port = 1;
break;
}
player_device_remove(device);
return;
}
// Max volume
device->max_volume = devcfg ? cfg_getint(devcfg, "max_volume") : CAST_CONFIG_MAX_VOLUME;
if ((device->max_volume < 1) || (device->max_volume > CAST_CONFIG_MAX_VOLUME))
{
DPRINTF(E_LOG, L_CAST, "Config has bad max_volume (%d) for device '%s', using default instead\n", device->max_volume, name);
device->max_volume = CAST_CONFIG_MAX_VOLUME;
}
DPRINTF(E_INFO, L_CAST, "Adding Chromecast device '%s'\n", name);
device->advertised = 1;
switch (family)
{
case AF_INET:
device->v4_address = strdup(address);
device->v4_port = port;
break;
case AF_INET6:
device->v6_address = strdup(address);
device->v6_port = port;
break;
}
player_device_add(device);
}
/* --------------------- SESSION CONSTRUCTION AND SHUTDOWN ------------------ */
static struct cast_master_session *
master_session_make(struct media_quality *quality)
{
struct cast_master_session *cms;
int ret;
// First check if we already have a master session, then just use that
if (cast_master_session)
return cast_master_session;
// Let's create a master session
ret = outputs_quality_subscribe(quality);
if (ret < 0)
{
DPRINTF(E_LOG, L_CAST, "Could not subscribe to required audio quality (%d/%d/%d)\n", quality->sample_rate, quality->bits_per_sample, quality->channels);
return NULL;
}
CHECK_NULL(L_CAST, cms = calloc(1, sizeof(struct cast_master_session)));
CHECK_NULL(L_CAST, cms->rtp_session = rtp_session_new(quality, CAST_PACKET_BUFFER_SIZE, 0));
// Change the SSRC to be in the interval [CAST_SSRC_AUDIO_MIN, CAST_SSRC_AUDIO_MAX]
cms->rtp_session->ssrc_id = ((cms->rtp_session->ssrc_id + CAST_SSRC_AUDIO_MIN) % CAST_SSRC_AUDIO_MAX) + CAST_SSRC_AUDIO_MIN;
cms->rtp_session->seqnum = 0; // TODO test
cms->quality = *quality;
cms->samples_per_packet = CAST_SAMPLES_PER_PACKET;
cms->rawbuf_size = STOB(cms->samples_per_packet, quality->bits_per_sample, quality->channels);
CHECK_NULL(L_CAST, cms->rawbuf = malloc(cms->rawbuf_size));
CHECK_NULL(L_CAST, cms->evbuf = evbuffer_new());
CHECK_NULL(L_CAST, cms->rtp_artwork = rtp_session_new(NULL, CAST_PACKET_ARTWORK_SIZE, 0));
// Change the SSRC to be in the interval [CAST_SSRC_VIDEO_MIN, CAST_SSRC_VIDEO_MAX]
cms->rtp_artwork->ssrc_id = ((cms->rtp_artwork->ssrc_id + CAST_SSRC_VIDEO_MIN) % CAST_SSRC_VIDEO_MAX) + CAST_SSRC_VIDEO_MIN;
cast_master_session = cms;
return cms;
}
static struct cast_session *
cast_session_make(struct output_device *device, int family, int callback_id)
{
struct cast_session *cs;
cfg_t *chromecast;
const char *proto;
const char *err;
char *address;
unsigned short port;
int offset_ms;
int flags;
int ret;
switch (family)
{
case AF_INET:
/* We always have the v4 services, so no need to check */
if (!device->v4_address)
return NULL;
address = device->v4_address;
port = device->v4_port;
break;
case AF_INET6:
if (!device->v6_address)
return NULL;
address = device->v6_address;
port = device->v6_port;
break;
default:
return NULL;
}
CHECK_NULL(L_CAST, cs = calloc(1, sizeof(struct cast_session)));
cs->state = CAST_STATE_DISCONNECTED;
cs->device_id = device->id;
cs->callback_id = callback_id;
cs->master_session = master_session_make(&cast_quality_default);
if (!cs->master_session)
{
DPRINTF(E_LOG, L_CAST, "Could not attach a master session for device '%s'\n", device->name);
goto out_free_session;
}
cs->ssrc_id = cs->master_session->rtp_session->ssrc_id;
/* Init TLS session, use default priorities and put the x509 credentials to the current session */
if ( ((ret = gnutls_init(&cs->tls_session, GNUTLS_CLIENT)) != GNUTLS_E_SUCCESS) ||
((ret = gnutls_priority_set_direct(cs->tls_session, "PERFORMANCE", &err)) != GNUTLS_E_SUCCESS) ||
((ret = gnutls_credentials_set(cs->tls_session, GNUTLS_CRD_CERTIFICATE, tls_credentials)) != GNUTLS_E_SUCCESS) )
{
DPRINTF(E_LOG, L_CAST, "Could not initialize GNUTLS session: %s\n", gnutls_strerror(ret));
goto out_free_master_session;
}
cs->server_fd = net_connect(address, port, SOCK_STREAM, "Chomecast control");
if (cs->server_fd < 0)
{
DPRINTF(E_LOG, L_CAST, "Could not connect to %s\n", device->name);
goto out_deinit_gnutls;
}
chromecast = cfg_gettsec(cfg, "chromecast", device->name);
offset_ms = chromecast ? cfg_getint(chromecast, "offset_ms") : 0;
if (abs(offset_ms) > CAST_OFFSET_MAX)
{
DPRINTF(E_LOG, L_CAST, "Ignoring invalid configuration of Chromecast offset (%d ms)\n", offset_ms);
offset_ms = 0;
}
offset_ms += OUTPUTS_BUFFER_DURATION * 1000 + CAST_DEVICE_START_DELAY_MS;
cs->offset_ts.tv_sec = (offset_ms / 1000);
cs->offset_ts.tv_nsec = (offset_ms % 1000) * 1000000UL;
DPRINTF(E_DBG, L_CAST, "Offset is set to %lu:%09lu\n", cs->offset_ts.tv_sec, cs->offset_ts.tv_nsec);
cs->ev = event_new(evbase_player, cs->server_fd, EV_READ | EV_PERSIST, cast_listen_cb, cs);
if (!cs->ev)
{
DPRINTF(E_LOG, L_CAST, "Out of memory for listener event\n");
goto out_close_connection;
}
cs->reply_timeout = evtimer_new(evbase_player, cast_reply_timeout_cb, cs);
if (!cs->reply_timeout)
{
DPRINTF(E_LOG, L_CAST, "Out of memory for reply_timeout\n");
goto out_close_connection;
}
gnutls_transport_set_int(cs->tls_session, cs->server_fd);
ret = gnutls_handshake(cs->tls_session);
if (ret != GNUTLS_E_SUCCESS)
{
DPRINTF(E_LOG, L_CAST, "Could not attach TLS to TCP connection: %s\n", gnutls_strerror(ret));
goto out_free_ev;
}
flags = fcntl(cs->server_fd, F_GETFL, 0);
fcntl(cs->server_fd, F_SETFL, flags | O_NONBLOCK);
event_add(cs->ev, NULL); // &heartbeat_timeout
cs->devname = strdup(device->name);
cs->address = strdup(address);
cs->family = family;
cs->udp_fd = -1;
cs->volume = 0.01 * device->volume;
cs->next = cast_sessions;
cast_sessions = cs;
// cs is now the official device session
outputs_device_session_add(device->id, cs);
proto = gnutls_protocol_get_name(gnutls_protocol_get_version(cs->tls_session));
DPRINTF(E_INFO, L_CAST, "Connection to '%s' established using %s\n", cs->devname, proto);
return cs;
out_free_ev:
event_free(cs->reply_timeout);
event_free(cs->ev);
out_close_connection:
cast_disconnect(cs->server_fd);
out_deinit_gnutls:
gnutls_deinit(cs->tls_session);
out_free_master_session:
master_session_cleanup(cs->master_session);
out_free_session:
free(cs);
return NULL;
}
// Attempts to "nicely" bring down a session to wanted_state, and then issues
// the callback. If wanted_state is CAST_STATE_NONE/FAILED then the session is purged.
static void
cast_session_shutdown(struct cast_session *cs, enum cast_state wanted_state)
{
int pending;
int ret;
if (cs->state == wanted_state)
{
cast_status(cs);
return;
}
else if (cs->state < wanted_state)
{
DPRINTF(E_LOG, L_CAST, "Bug! Shutdown request got wanted_state (%d) that is higher than current state (%d)\n", wanted_state, cs->state);
return;
}
cs->wanted_state = wanted_state;
pending = 0;
switch (cs->state)
{
case CAST_STATE_STREAMING:
case CAST_STATE_BUFFERING:
case CAST_STATE_APP_READY:
cast_disconnect(cs->udp_fd);
cs->udp_fd = -1;
ret = cast_msg_send(cs, MEDIA_CLOSE, NULL);
cs->state = CAST_STATE_APP_LAUNCHED;
if ((ret < 0) || (wanted_state >= CAST_STATE_APP_LAUNCHED))
break;
/* FALLTHROUGH */
case CAST_STATE_APP_LAUNCHED:
ret = cast_msg_send(cs, STOP, cast_cb_stop);
pending = 1;
break;
case CAST_STATE_CONNECTED:
ret = cast_msg_send(cs, CLOSE, NULL);
if (ret == 0)
gnutls_bye(cs->tls_session, GNUTLS_SHUT_RDWR);
cast_disconnect(cs->server_fd);
cs->server_fd = -1;
cs->state = CAST_STATE_DISCONNECTED;
break;
case CAST_STATE_DISCONNECTED:
ret = 0;
break;
default:
DPRINTF(E_LOG, L_CAST, "Bug! Shutdown doesn't know how to handle current state\n");
ret = -1;
}
// We couldn't talk to the device, tell the user and clean up
if (ret < 0)
{
cs->state = CAST_STATE_FAILED;
cast_status(cs);
cast_session_cleanup(cs);
return;
}
// If pending callbacks then we let them take care of the rest
if (pending)
return;
// Asked to destroy the session
if (wanted_state == CAST_STATE_NONE || wanted_state == CAST_STATE_FAILED)
{
cs->state = wanted_state;
cast_status(cs);
cast_session_cleanup(cs);
return;
}
cast_status(cs);
}
/* ------------------ INTERFACE FUNCTIONS CALLED BY OUTPUTS.C --------------- */
static int
cast_device_start_generic(struct output_device *device, int callback_id, cast_reply_cb reply_cb)
{
struct cast_session *cs;
int ret;
cs = cast_session_make(device, AF_INET6, callback_id);
if (cs)
{
ret = cast_msg_send(cs, CONNECT, NULL);
if (ret == 0)
ret = cast_msg_send(cs, GET_STATUS, reply_cb);
if (ret < 0)
{
DPRINTF(E_WARN, L_CAST, "Could not send CONNECT or GET_STATUS request on IPv6 (start)\n");
cast_session_cleanup(cs);
}
else
return 1;
}
cs = cast_session_make(device, AF_INET, callback_id);
if (!cs)
return -1;
ret = cast_msg_send(cs, CONNECT, NULL);
if (ret == 0)
ret = cast_msg_send(cs, GET_STATUS, reply_cb);
if (ret < 0)
{
DPRINTF(E_LOG, L_CAST, "Could not send CONNECT or GET_STATUS request on IPv4 (start)\n");
cast_session_cleanup(cs);
return -1;
}
return 1;
}
static int
cast_device_start(struct output_device *device, int callback_id)
{
return cast_device_start_generic(device, callback_id, cast_cb_startup_connect);
}
static int
cast_device_probe(struct output_device *device, int callback_id)
{
return cast_device_start_generic(device, callback_id, cast_cb_probe);
}
static int
cast_device_stop(struct output_device *device, int callback_id)
{
struct cast_session *cs = device->session;
cs->callback_id = callback_id;
cast_session_shutdown(cs, CAST_STATE_NONE);
return 1;
}
static int
cast_device_flush(struct output_device *device, int callback_id)
{
struct cast_session *cs = device->session;
cs->callback_id = callback_id;
cs->state = CAST_STATE_APP_READY;
cast_status(cs);
return 1;
}
static void
cast_device_cb_set(struct output_device *device, int callback_id)
{
struct cast_session *cs = device->session;
cs->callback_id = callback_id;
}
static int
cast_device_volume_set(struct output_device *device, int callback_id)
{
struct cast_session *cs = device->session;
int ret;
if (!cs || !(cs->state & CAST_STATE_F_APP_READY))
return 0;
cs->volume = ((float)device->max_volume * (float)device->volume * 1.0) / (100.0 * CAST_CONFIG_MAX_VOLUME);
ret = cast_msg_send(cs, SET_VOLUME, cast_cb_volume);
if (ret < 0)
{
cast_session_shutdown(cs, CAST_STATE_FAILED);
return 0;
}
// Setting it here means it will not be used for the above cast_session_shutdown
cs->callback_id = callback_id;
return 1;
}
static void
cast_write(struct output_buffer *obuf)
{
struct cast_session *cs;
struct cast_session *next;
struct timespec ts;
int i;
int ret;
if (!cast_sessions)
return;
for (i = 0; obuf->data[i].buffer; i++)
{
if (quality_is_equal(&obuf->data[i].quality, &cast_quality_default))
break;
}
if (!obuf->data[i].buffer)
{
DPRINTF(E_LOG, L_CAST, "Bug! Output not delivering required data quality\n");
return;
}
// Converts the raw audio in the output_buffer to Chromecast packets
packets_make(cast_master_session, &obuf->data[i]);
for (cs = cast_sessions; cs; cs = next)
{
next = cs->next;
if (!(cs->state & CAST_STATE_F_APP_READY))
continue;
if (cs->state == CAST_STATE_APP_READY)
{
// Sets that playback will start at time = start_pts with the packet that comes after seqnum_last
cs->start_pts = timespec_add(obuf->pts, cs->offset_ts);
cs->seqnum_next = cast_master_session->rtp_session->seqnum;
cs->state = CAST_STATE_BUFFERING;
clock_gettime(CLOCK_MONOTONIC, &ts);
DPRINTF(E_DBG, L_CAST, "Start time is %lu:%lu, current time is %lu:%lu\n", cs->start_pts.tv_sec, cs->start_pts.tv_nsec, ts.tv_sec, ts.tv_nsec);
}
if (cs->state == CAST_STATE_BUFFERING)
{
clock_gettime(CLOCK_MONOTONIC, &ts);
if (timespec_cmp(cs->start_pts, ts) > 0)
continue; // Keep buffering
cs->state = CAST_STATE_STREAMING;
}
// We send packets to the device ping-pong style, meaning that we send the
// first packet, wait for an ack, then send the next, wait etc. This can
// be broken by "no ping", meaning cast_rtcp_cb() didn't have a packet to
// send, or "no pong", meaning the ack is late or lost. To keep going we
// must send a packet from here, so this condition is an inverse check for
// such a state. The first part will be false if we didn't get an ACK,
// or when sending first packet, and the second will be false if we were
// out of packets.
if (cs->ack_last + 1 == cs->seqnum_next && cs->seqnum_next + 1 != cast_master_session->rtp_session->seqnum)
continue;
ret = packet_send_next(cs);
if (ret < 0)
{
// Downgrade state immediately to avoid further write attempts (session shutdown is async)
cs->state = CAST_STATE_APP_LAUNCHED;
cast_session_shutdown(cs, CAST_STATE_FAILED);
}
}
}
/*
// *** Thread: worker ***
static void *
cast_metadata_prepare(struct output_metadata *metadata)
{
struct db_queue_item *queue_item;
struct cast_metadata *cmd;
int ret;
if (!cast_sessions)
return NULL;
queue_item = db_queue_fetch_byitemid(metadata->item_id);
if (!queue_item)
{
DPRINTF(E_LOG, L_CAST, "Could not fetch queue item\n");
return NULL;
}
CHECK_NULL(L_CAST, cmd = calloc(1, sizeof(struct cast_metadata)));
CHECK_NULL(L_CAST, cmd->artwork = evbuffer_new());
ret = artwork_get_item(cmd->artwork, queue_item->file_id, ART_DEFAULT_WIDTH, ART_DEFAULT_HEIGHT, ART_FMT_VP8);
if (ret < 0)
{
DPRINTF(E_INFO, L_CAST, "Failed to retrieve artwork for file '%s'; no artwork will be sent\n", queue_item->path);
cast_metadata_free(cmd);
return NULL;
}
return cmd;
}
static void
cast_metadata_send(struct output_metadata *metadata)
{
struct cast_metadata *cmd = metadata->priv;
struct cast_session *cs;
struct cast_session *next;
struct rtp_packet *pkt;
size_t artwork_size;
int ret;
artwork_size = evbuffer_get_length(cmd->artwork);
if (artwork_size == 0)
return;
for (cs = cast_sessions; cs; cs = next)
{
next = cs->next;
if (! (cs->state & CAST_STATE_APP_READY))
continue;
// Marker bit is 1 because we send a complete frame
pkt = rtp_packet_next(cs->master_session->rtp_artwork, CAST_HEADER_SIZE + artwork_size, 1, CAST_RTP_PAYLOADTYPE_VIDEO, 1);
if (!pkt)
continue;
ret = packet_prepare(pkt, cmd->artwork);
if (ret < 0)
continue;
packet_send(cs, pkt);
// TODO Handle partial send
rtp_packet_commit(cs->master_session->rtp_artwork, pkt);
}
cast_metadata_free(cmd);
}
*/
static int
cast_init(void)
{
struct decode_ctx *decode_ctx;
int i;
int ret;
// Sanity check
for (i = 1; cast_msg[i].type; i++)
{
if (cast_msg[i].type != i)
{
DPRINTF(E_LOG, L_CAST, "BUG! Cast messages and types are misaligned (type %d!=%d). Could not initialize.\n", cast_msg[i].type, i);
return -1;
}
}
// Setting the cert file seems not to be required
if ( ((ret = gnutls_global_init()) != GNUTLS_E_SUCCESS)
|| ((ret = gnutls_certificate_allocate_credentials(&tls_credentials)) != GNUTLS_E_SUCCESS)
// || ((ret = gnutls_certificate_set_x509_trust_file(tls_credentials, CAFILE, GNUTLS_X509_FMT_PEM)) < 0)
)
{
DPRINTF(E_LOG, L_CAST, "Could not initialize GNUTLS: %s\n", gnutls_strerror(ret));
return -1;
}
decode_ctx = transcode_decode_setup_raw(XCODE_PCM16, &cast_quality_default);
if (!decode_ctx)
{
DPRINTF(E_LOG, L_CAST, "Could not create decoding context\n");
goto out_tls_deinit;
}
cast_encode_ctx = transcode_encode_setup(XCODE_OPUS, &cast_quality_default, decode_ctx, 0, 0);
transcode_decode_cleanup(&decode_ctx);
if (!cast_encode_ctx)
{
DPRINTF(E_LOG, L_CAST, "Will not be able to stream Chromecast, libav does not support Opus encoding\n");
goto out_tls_deinit;
}
ret = mdns_browse("_googlecast._tcp", cast_device_cb, 0);
if (ret < 0)
{
DPRINTF(E_LOG, L_CAST, "Could not add mDNS browser for Chromecast devices\n");
goto out_encode_ctx_free;
}
CHECK_NULL(L_CAST, cast_encoded_data = evbuffer_new());
return 0;
out_encode_ctx_free:
transcode_encode_cleanup(&cast_encode_ctx);
out_tls_deinit:
gnutls_certificate_free_credentials(tls_credentials);
gnutls_global_deinit();
return -1;
}
static void
cast_deinit(void)
{
struct cast_session *cs;
for (cs = cast_sessions; cast_sessions; cs = cast_sessions)
{
cast_sessions = cs->next;
cast_session_free(cs);
}
evbuffer_free(cast_encoded_data);
transcode_encode_cleanup(&cast_encode_ctx);
gnutls_certificate_free_credentials(tls_credentials);
gnutls_global_deinit();
}
struct output_definition output_cast =
{
.name = "Chromecast",
.type = OUTPUT_TYPE_CAST,
.priority = 2,
.disabled = 0,
.init = cast_init,
.deinit = cast_deinit,
.device_start = cast_device_start,
.device_probe = cast_device_probe,
.device_stop = cast_device_stop,
.device_flush = cast_device_flush,
.device_cb_set = cast_device_cb_set,
.device_volume_set = cast_device_volume_set,
.write = cast_write,
// .metadata_prepare = cast_metadata_prepare,
// .metadata_send = cast_metadata_send,
// .metadata_purge = cast_metadata_purge,
};
Reference in New Issue View Git Blame Copy Permalink