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monitor.c
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monitor.c
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// Listen to multicast group(s), send audio to local sound device via portaudio
// Copyright 2018-2023 Phil Karn, KA9Q
#define _GNU_SOURCE 1
#include <assert.h>
#include <errno.h>
#include <pthread.h>
#include <sys/stat.h>
#include <opus/opus.h>
#include <portaudio.h>
#include <ncurses.h>
#include <locale.h>
#include <signal.h>
#include <getopt.h>
#include <iniparser/iniparser.h>
#if __linux__
#include <bsd/string.h>
#include <alsa/asoundlib.h>
#else
#include <string.h>
#endif
#include <sysexits.h>
#include "conf.h"
#include "config.h"
#include "misc.h"
#include "multicast.h"
#include "iir.h"
#include "morse.h"
// Global constants
#define MAX_MCAST 20 // Maximum number of multicast addresses
#define BUFFERSIZE (1<<17) // about 2.73 sec at 48 kHz - must be power of 2 times page size (4k)!
static float const Latency = 0.02; // chunk size for audio output callback
static float const Tone_period = 0.24; // PL tone integration period
#define NSESSIONS 1500
// Names of config file sections
static char const *Radio = "radio";
static char const *Audio = "audio";
static char const *Repeater = "repeater";
static char const *Display = "display";
// Command line/config file/interactive command parameters
static char const *Tx_on = "set_xcvr txon";
static char const *Tx_off = "set_xcvr txoff";
static int DAC_samprate = 48000; // Actual hardware output rate
static int Update_interval = 100; // Default time in ms between display updates
char const *App_path;
int Verbose; // Verbosity flag
static char const *Config_file;
static bool Quiet; // Disable curses
static bool Quiet_mode; // Toggle screen activity after starting
static float Playout = 100;
static bool Constant_delay;
static bool Start_muted;
static bool Auto_position = true; // first will be in the center
static int64_t Repeater_tail;
static char const *Cwid = "de nocall/r"; // Make this configurable!
static double ID_pitch = 800.0;
static double ID_level = -29.0;
static double ID_speed = 18.0;
static float Gain = 0; // unity gain by default
static bool Notch;
static char *Mcast_address_text[MAX_MCAST]; // Multicast address(es) we're listening to
static char const *Audiodev = ""; // Name of audio device; empty means portaudio's default
static int Position; // auto-position streams
static bool Auto_sort;
// IDs must be at least every 10 minutes per FCC 97.119(a)
static int64_t Mandatory_ID_interval;
// ID early when carrier is about to drop, to avoid stepping on users
static int64_t Quiet_ID_interval;
static int Dit_length;
static int Channels = 2;
static char const *Init;
// Global variables that regularly change
static int64_t Last_xmit_time;
static int64_t Last_id_time;
static float *Output_buffer;
static int Buffer_length; // Bytes left to play out, max BUFFERSIZE
static volatile unsigned int Rptr; // callback thread read pointer, *frames*
static volatile unsigned int Wptr; // For monitoring length of output queue
static volatile bool PTT_state;
static uint64_t Audio_callbacks;
static unsigned long Audio_frames;
static volatile int64_t LastAudioTime;
static int32_t Portaudio_delay;
static pthread_t Repeater_thread;
static pthread_cond_t PTT_cond = PTHREAD_COND_INITIALIZER;
static pthread_mutex_t PTT_mutex = PTHREAD_MUTEX_INITIALIZER;
static int Nfds; // Number of streams
static pthread_mutex_t Sess_mutex = PTHREAD_MUTEX_INITIALIZER;
static PaStream *Pa_Stream; // Portaudio stream handle
static int inDevNum; // Portaudio's audio output device index
static int64_t Start_time;
static pthread_mutex_t Stream_mutex = PTHREAD_MUTEX_INITIALIZER; // Control access to stream start/stop
static PaTime Start_pa_time;
static PaTime Last_callback_time;
static int Invalids;
static int64_t Last_error_time;
static int Nsessions;
static struct session *Sessions[NSESSIONS];
static bool Terminate;
// All the tones from various groups, including special NATO 150 Hz tone
static float PL_tones[] = {
67.0, 69.3, 71.9, 74.4, 77.0, 79.7, 82.5, 85.4, 88.5, 91.5,
94.8, 97.4, 100.0, 103.5, 107.2, 110.9, 114.8, 118.8, 123.0, 127.3,
131.8, 136.5, 141.3, 146.2, 150.0, 151.4, 156.7, 159.8, 162.2, 165.5,
167.9, 171.3, 173.8, 177.3, 179.9, 183.5, 186.2, 189.9, 192.8, 196.6,
199.5, 203.5, 206.5, 210.7, 213.8, 218.1, 221.3, 225.7, 229.1, 233.6,
237.1, 241.8, 245.5, 250.3, 254.1
};
#define N_tones (sizeof(PL_tones)/sizeof(PL_tones[0]))
struct session {
struct sockaddr_storage sender;
char const *dest;
pthread_t task; // Thread reading from queue and running decoder
struct packet *queue; // Incoming RTP packets
pthread_mutex_t qmutex; // Mutex protecting packet queue
pthread_cond_t qcond; // Condition variable for arrival of new packet
struct rtp_state rtp_state; // Incoming RTP session state
uint32_t ssrc; // RTP Sending Source ID
int type; // RTP type (10,11,20,111)
uint32_t last_timestamp; // Last timestamp seen
unsigned int wptr; // current write index into output PCM buffer, *frames*
int playout; // Initial playout delay, frames
long long last_active; // GPS time last active
long long last_start; // GPS time at last transition to active from idle
float tot_active; // Total PCM time, ns
float active; // Seconds we've been active (only when queue has stuff)
OpusDecoder *opus; // Opus codec decoder handle, if needed
int frame_size;
int bandwidth; // Audio bandwidth
struct goertzel tone_detector[N_tones];
int tone_samples;
float current_tone; // Detected tone frequency
int samprate;
int channels; // Channels (1 or 2)
float gain; // Gain; 1 = 0 dB
float pan; // Stereo position: 0 = center; -1 = full left; +1 = full right
unsigned long packets; // RTP packets for this session
unsigned long empties; // RTP but no data
unsigned long lates;
unsigned long earlies;
unsigned long resets;
unsigned long reseqs;
bool terminate; // Set to cause thread to terminate voluntarily
bool muted;
bool reset; // Set to force output timing reset on next packet
char id[32];
bool notch_enable; // Enable PL removal notch
struct iir iir_left;
struct iir iir_right;
float notch_tone;
};
static void load_id(void);
static void cleanup(void);
static void *display(void *);
static void reset_session(struct session *sp,uint32_t timestamp);
static struct session *lookup_session(struct sockaddr_storage const *,uint32_t);
static struct session *create_session(void);
static int sort_session_active(void),sort_session_total(void);
static int close_session(struct session **);
static int pa_callback(void const *,void *,unsigned long,PaStreamCallbackTimeInfo const *,PaStreamCallbackFlags,void *);
static void *decode_task(void *x);
static void *sockproc(void *arg);
static void *repeater_ctl(void *arg);
static char const *lookupid(uint32_t ssrc);
static float make_position(int);
static bool kick_output();
static inline int modsub(unsigned int const a, unsigned int const b, int const modulus){
int diff = (int)a - (int)b;
if(diff > modulus)
return diff % modulus; // Unexpectedly large, just do it the slow way
if(diff > modulus/2)
return diff - modulus;
if(diff < -modulus)
return diff % modulus; // Unexpectedly small
if(diff < -modulus/2)
diff += modulus;
return diff;
}
static char Optstring[] = "CI:LR:Sac:f:g:p:qr:u:vnV";
static struct option Options[] = {
{"center", no_argument, NULL, 'C'},
{"input", required_argument, NULL, 'I'},
{"list-audio", no_argument, NULL, 'L'},
{"device", required_argument, NULL, 'R'},
{"autosort", no_argument, NULL, 'S'},
{"channels", required_argument, NULL, 'c'},
{"config", required_argument, NULL, 'f'},
{"gain", required_argument, NULL, 'g'},
{"playout", required_argument, NULL, 'p'},
{"quiet", no_argument, NULL, 'q'},
{"samprate",required_argument,NULL,'r'},
{"update", required_argument, NULL, 'u'},
{"verbose", no_argument, NULL, 'v'},
{"notch", no_argument, NULL, 'n'},
{"version", no_argument, NULL, 'V'},
{NULL, 0, NULL, 0},
};
#ifdef __linux__
// Get rid of those fucking ALSA error messages that clutter the screen
static void alsa_error_handler(const char *file, int line, const char *function, int err, const char *fmt, ...){
return;
}
#endif
int main(int argc,char * const argv[]){
App_path = argv[0];
setlocale(LC_ALL,getenv("LANG"));
tzset();
// Parse command line for config file, read first so it can be overriden by command line args
int c;
while((c = getopt_long(argc,argv,Optstring,Options,NULL)) != -1){
switch(c){
case 'f':
Config_file = optarg;
break;
case 'V':
VERSION();
exit(EX_OK);
default:
break;
}
}
if(Config_file){
dictionary *Configtable = iniparser_load(Config_file);
if(Configtable == NULL){
fprintf(stdout,"Can't load config file %s\n",Config_file);
exit(EX_NOINPUT);
}
DAC_samprate = config_getint(Configtable,Audio,"samprate",DAC_samprate);
Channels = config_getint(Configtable,Audio,"channels",Channels);
char const *audiodev = config_getstring(Configtable,"audio","device",NULL);
if(audiodev)
Audiodev = strdup(audiodev);
// Add validity checking
Gain = config_getfloat(Configtable,Audio,"gain",Gain);
Cwid = strdup(config_getstring(Configtable,Repeater,"id","NOCALL"));
// 600 sec is 10 minutes, max ID interval per FCC 97.119(a)
int const period = config_getint(Configtable,Repeater,"period",600);
int pperiod = config_getint(Configtable,Repeater,"pperiod",period/2);
if(pperiod > period)
pperiod = period;
Mandatory_ID_interval = period * BILLION;
Quiet_ID_interval = pperiod * BILLION;
ID_pitch = config_getfloat(Configtable,Repeater,"pitch",ID_pitch);
ID_level = config_getfloat(Configtable,Repeater,"level",ID_level);
Notch = config_getboolean(Configtable,Audio,"notch",Notch);
Quiet = config_getboolean(Configtable,Display,"quiet",Quiet);
if(config_getboolean(Configtable,Audio,"center",false))
Auto_position = false;
Auto_sort = config_getboolean(Configtable,Display,"autosort",Auto_sort);
Update_interval = config_getint(Configtable,Display,"update",Update_interval);
Playout = config_getfloat(Configtable,Audio,"playout",Playout);
Repeater_tail = config_getfloat(Configtable,Repeater,"tail",Repeater_tail);
Verbose = config_getboolean(Configtable,Display,"verbose",Verbose);
char const *txon = config_getstring(Configtable,Radio,"txon",NULL);
char const *txoff = config_getstring(Configtable,Radio,"txoff",NULL);
if(txon)
Tx_on = strdup(txon);
if(txoff)
Tx_off = strdup(txoff);
char const *init = config_getstring(Configtable,Radio,"init",NULL);
if(init)
Init = strdup(init);
char const *input = config_getstring(Configtable,Audio,"input",NULL);
if(input)
Mcast_address_text[Nfds++] = strdup(input);
iniparser_freedict(Configtable);
}
// Rescan args to override config file
bool list_audio = false;
optind = 0; // reset getopt()
while((c = getopt_long(argc,argv,Optstring,Options,NULL)) != -1){
switch(c){
case 'c':
Channels = strtol(optarg,NULL,0);
break;
case 'f':
break; // Ignore this time
case 'g':
Gain = strtof(optarg,NULL);
break;
case 'n':
Notch = true;
break;
case 'p':
Playout = strtof(optarg,NULL);
break;
case 'q': // No ncurses
Quiet = true;
break;
case 'r':
DAC_samprate = strtol(optarg,NULL,0);
break;
case 'u':
Update_interval = strtol(optarg,NULL,0);
break;
case 'v':
Verbose = true;
break;
case 'I':
if(Nfds == MAX_MCAST){
fprintf(stderr,"Too many multicast addresses; max %d\n",MAX_MCAST);
} else
Mcast_address_text[Nfds++] = optarg;
break;
case 'L':
list_audio = true;
break;
case 'R':
Audiodev = optarg;
break;
case 'S':
Auto_sort = true;
break;
default:
fprintf(stderr,"Usage: %s -L\n",App_path);
fprintf(stderr," %s [-a] [-c channels] [-f config_file] [-g gain] [-p playout] [-q] [-r samprate] [-u update] [-v]\
[-I mcast_address] [-R audiodev] [-S] [mcast_address ...]\n",App_path);
exit(EX_USAGE);
}
}
if(list_audio){
// On stdout, not stderr, so we can toss ALSA's noisy error messages
PaError r = Pa_Initialize();
if(r != paNoError){
fprintf(stderr,"Portaudio error: %s\n",Pa_GetErrorText(r));
return r;
}
printf("Audio devices:\n");
int numDevices = Pa_GetDeviceCount();
for(int inDevNum=0; inDevNum < numDevices; inDevNum++){
const PaDeviceInfo *deviceInfo = Pa_GetDeviceInfo(inDevNum);
printf("%d: %s\n",inDevNum,deviceInfo->name);
}
Pa_Terminate();
exit(EX_OK);
}
if(Channels != 1 && Channels != 2){
fprintf(stderr,"Channels = %d invalid; defaulting to 2\n",Channels);
Channels = 2;
}
if(Auto_position && Channels != 2){
fprintf(stderr,"Auto_position requires 2 channels\n");
Auto_position = false;
}
// Also accept groups without -I option
for(int i=optind; i < argc; i++){
if(Nfds == MAX_MCAST){
fprintf(stderr,"Too many multicast addresses; max %d\n",MAX_MCAST);
} else
Mcast_address_text[Nfds++] = argv[i];
}
if(Nfds == 0){
fprintf(stderr,"At least one input group required, exiting\n");
exit(EX_USAGE);
}
if(Init != NULL)
(void) - system(Init);
if(Cwid != NULL){
// Operating as a repeater controller; initialize
// Make these settable parameters
// -29 dB is -15 + (-14).
// -15 dBFS is the target level of the FM demodulator
// -14 dB is 1 kHz ID deviation divided by 5 kHz peak deviation
Dit_length = init_morse(ID_speed,ID_pitch,ID_level,DAC_samprate);
}
#ifdef __linux__
// Get rid of those fucking ALSA error messages that clutter the screen
snd_lib_error_set_handler(alsa_error_handler);
#endif
PaError r = Pa_Initialize();
if(r != paNoError){
fprintf(stderr,"Portaudio error: %s\n",Pa_GetErrorText(r));
return r;
}
atexit(cleanup); // Make sure Pa_Terminate() gets called
load_id();
char *nextp = NULL;
int d;
int numDevices = Pa_GetDeviceCount();
if(Audiodev == NULL || strlen(Audiodev) == 0){
// not specified; use default
inDevNum = Pa_GetDefaultOutputDevice();
} else if(d = strtol(Audiodev,&nextp,0),nextp != Audiodev && *nextp == '\0'){
if(d >= numDevices){
fprintf(stderr,"%d is out of range, use %s -L for a list\n",d,App_path);
exit(EX_USAGE);
}
inDevNum = d;
} else {
for(inDevNum=0; inDevNum < numDevices; inDevNum++){
const PaDeviceInfo *deviceInfo = Pa_GetDeviceInfo(inDevNum);
if(strcmp(deviceInfo->name,Audiodev) == 0)
break;
}
}
if(inDevNum == paNoDevice){
fprintf(stderr,"Portaudio: no available devices, exiting\n");
exit(EX_IOERR);
}
// Create portaudio stream.
// Runs continuously, playing silence until audio arrives.
// This allows multiple streams to be played on hosts that only support one
Output_buffer = mirror_alloc(BUFFERSIZE * Channels * sizeof(*Output_buffer)); // Must be power of 2 times page size
memset(Output_buffer,0,BUFFERSIZE * Channels * sizeof(*Output_buffer)); // Does mmap clear its initial memory? Not sure
PaStreamParameters outputParameters;
memset(&outputParameters,0,sizeof(outputParameters));
outputParameters.channelCount = Channels;
outputParameters.device = inDevNum;
outputParameters.sampleFormat = paFloat32;
outputParameters.suggestedLatency = Latency; // 0 doesn't seem to be a good value on OSX, lots of underruns and stutters
r = Pa_OpenStream(&Pa_Stream,
NULL,
&outputParameters,
DAC_samprate,
paFramesPerBufferUnspecified, // seems to be 31 on OSX
//SAMPPCALLBACK,
0,
pa_callback,
NULL);
if(r != paNoError){
fprintf(stderr,"Portaudio error: %s, exiting\n",Pa_GetErrorText(r));
exit(EX_IOERR);
}
if(Repeater_tail != 0)
pthread_create(&Repeater_thread,NULL,repeater_ctl,NULL); // Repeater mode active
// Spawn one thread per address
// All have to succeed in resolving their targets or we'll exit
// This allows a restart when started automatically from systemd before avahi is fully running
pthread_t sockthreads[Nfds];
for(int i=0; i<Nfds; i++)
pthread_create(&sockthreads[i],NULL,sockproc,Mcast_address_text[i]);
Last_error_time = gps_time_ns();
// Become the display thread
if(!Quiet){
display(NULL);
} else {
while(!Terminate)
sleep(1);
}
exit(EX_OK); // calls cleanup() to clean up Portaudio and ncurses. Can't happen...
}
static void *sockproc(void *arg){
char const *mcast_address_text = (char *)arg;
{
char name[100];
snprintf(name,sizeof(name),"mon %s",mcast_address_text);
pthread_setname(name);
}
int input_fd;
{
char iface[1024];
struct sockaddr sock;
resolve_mcast(mcast_address_text,&sock,DEFAULT_RTP_PORT,iface,sizeof(iface));
input_fd = listen_mcast(&sock,iface);
}
if(input_fd == -1)
pthread_exit(NULL);
struct packet *pkt = NULL;
realtime();
// Main loop begins here
while(!Terminate){
// Need a new packet buffer?
if(!pkt)
pkt = malloc(sizeof(*pkt));
// Zero these out to catch any uninitialized derefs
pkt->next = NULL;
pkt->data = NULL;
pkt->len = 0;
// Needs a timeout to poll Terminate
struct sockaddr_storage sender;
socklen_t socksize = sizeof(sender);
int size = recvfrom(input_fd,&pkt->content,sizeof(pkt->content),0,(struct sockaddr *)&sender,&socksize);
if(size == -1){
if(errno != EINTR){ // Happens routinely, e.g., when window resized
perror("recvfrom");
usleep(1000);
}
continue; // Reuse current buffer
}
if(size <= RTP_MIN_SIZE)
continue; // Must be big enough for RTP header and at least some data
// Convert RTP header to host format
uint8_t const *dp = ntoh_rtp(&pkt->rtp,pkt->content);
pkt->data = dp;
pkt->len = size - (dp - pkt->content);
if(pkt->rtp.pad){
pkt->len -= dp[pkt->len-1];
pkt->rtp.pad = 0;
}
if(pkt->len <= 0)
continue; // Used to be an assert, but would be triggered by bogus packets
// Find appropriate session; create new one if necessary
pthread_mutex_lock(&Sess_mutex); // Protect Nsessions
struct session *sp = lookup_session(&sender,pkt->rtp.ssrc);
pthread_mutex_unlock(&Sess_mutex);
if(!sp){
// Not found
pthread_mutex_lock(&Sess_mutex); // Protect Nsessions
sp = create_session();
pthread_mutex_unlock(&Sess_mutex); // Protect Nsessions
if(!sp){
fprintf(stderr,"No room!!\n");
continue;
}
pthread_cond_init(&sp->qcond,NULL);
pthread_mutex_init(&sp->qmutex,NULL);
sp->ssrc = pkt->rtp.ssrc;
char const *id = lookupid(pkt->rtp.ssrc);
if(id)
strlcpy(sp->id,id,sizeof(sp->id));
if(Auto_position)
sp->pan = make_position(Position++);
else
sp->pan = 0; // center by default
sp->gain = powf(10.,0.05 * Gain); // Start with global default
sp->notch_enable = Notch;
sp->muted = Start_muted;
sp->dest = mcast_address_text;
sp->last_timestamp = pkt->rtp.timestamp;
sp->rtp_state.seq = pkt->rtp.seq;
sp->reset = true;
sp->type = pkt->rtp.type;
if(sp->type < 0 || sp->type > 127)
continue; // Invalid payload type?
sp->samprate = PT_table[sp->type].samprate;
if(PT_table[sp->type].encoding == OPUS)
sp->samprate = DAC_samprate;
for(int j=0; j < N_tones; j++)
init_goertzel(&sp->tone_detector[j],PL_tones[j]/(float)sp->samprate);
pthread_mutex_init(&sp->qmutex,NULL);
pthread_cond_init(&sp->qcond,NULL);
if(pthread_create(&sp->task,NULL,decode_task,sp) == -1){
perror("pthread_create");
close_session(&sp);
continue;
}
}
// Copy sender, in case the port number changed
memcpy(&sp->sender,&sender,sizeof(sender));
// Insert onto queue sorted by sequence number, wake up thread
struct packet *q_prev = NULL;
struct packet *qe = NULL;
pthread_mutex_lock(&sp->qmutex);
for(qe = sp->queue; qe && pkt->rtp.seq >= qe->rtp.seq; q_prev = qe,qe = qe->next)
;
if(qe)
sp->reseqs++; // Not the last on the list
pkt->next = qe;
if(q_prev)
q_prev->next = pkt;
else
sp->queue = pkt; // Front of list
pkt = NULL; // force new packet to be allocated
long long t = gps_time_ns();
if(t - sp->last_active > BILLION){
// Transition from idle to active
sp->last_start = t;
}
sp->last_active = t;
// wake up decoder thread
pthread_cond_signal(&sp->qcond);
pthread_mutex_unlock(&sp->qmutex);
}
return NULL;
}
static void decode_task_cleanup(void *arg){
struct session *sp = (struct session *)arg;
assert(sp);
pthread_mutex_destroy(&sp->qmutex);
pthread_cond_destroy(&sp->qcond);
if(sp->opus){
opus_decoder_destroy(sp->opus);
sp->opus = NULL;
}
struct packet *pkt_next;
for(struct packet *pkt = sp->queue; pkt; pkt = pkt_next){
pkt_next = pkt->next;
FREE(pkt);
}
}
// Thread to decode incoming RTP packets for each session
static void *decode_task(void *arg){
struct session *sp = (struct session *)arg;
assert(sp);
{
char name[100];
snprintf(name,sizeof(name),"dec %u",sp->ssrc);
pthread_setname(name);
}
pthread_cleanup_push(decode_task_cleanup,arg);
int consec_lates = 0;
int consec_earlies = 0;
float *bounce = NULL;
// Main loop; run until asked to quit
while(!sp->terminate && !Terminate){
struct packet *pkt = NULL;
// Wait for packet to appear on queue
pthread_mutex_lock(&sp->qmutex);
while(!sp->queue){
int64_t const increment = 100000000; // 100 ms
// pthread_cond_timedwait requires UTC clock time! Undefined behavior around a leap second...
struct timespec ts;
ns2ts(&ts,utc_time_ns() + increment);
int r = pthread_cond_timedwait(&sp->qcond,&sp->qmutex,&ts); // Wait 100 ms max so we pick up terminates
if(r != 0){
if(r == EINVAL)
Invalids++;
pthread_mutex_unlock(&sp->qmutex);
goto endloop;// restart loop, checking terminate flags
}
}
// Peek at first packet on queue; is it in sequence?
if(sp->queue->rtp.seq != sp->rtp_state.seq){
// No. If we've got plenty in the playout buffer, sleep to allow some packet resequencing in the input thread.
// Strictly speaking, we will resequence ourselves below with the RTP timestamp. But that works properly only with stateless
// formats like PCM. Opus is stateful, so it's better to resequence input packets (using the RTP sequence #) when possible.
float queue = (float)modsub(sp->wptr,Rptr,BUFFERSIZE) / DAC_samprate;
if(queue > Latency + 0.1){ // 100 ms for scheduling latency?
pthread_mutex_unlock(&sp->qmutex);
struct timespec ss;
ns2ts(&ss,(int64_t)(1e9 * (queue - (Latency + 0.1))));
nanosleep(&ss,NULL);
goto endloop;
}
// else the playout queue is close to draining, accept out of sequence packet anyway
}
pkt = sp->queue;
sp->queue = pkt->next;
pkt->next = NULL;
pthread_mutex_unlock(&sp->qmutex);
sp->packets++; // Count all packets, regardless of type
if(sp->type != pkt->rtp.type) // Handle transitions both ways
sp->type = pkt->rtp.type;
if((int16_t)(pkt->rtp.seq - sp->rtp_state.seq) > 0){ // Doesn't really handle resequencing
if(!pkt->rtp.marker){
sp->rtp_state.drops++; // Avoid spurious drops when session is recreated after silence
Last_error_time = gps_time_ns();
}
if(sp->opus)
opus_decoder_ctl(sp->opus,OPUS_RESET_STATE); // Reset decoder
}
sp->rtp_state.seq = pkt->rtp.seq + 1;
if(!sp->muted && pkt->rtp.marker){
// beginning of talk spurt, resync
reset_session(sp,pkt->rtp.timestamp); // Updates sp->wptr
}
int upsample = 1;
// decode Opus or PCM into bounce buffer
if(PT_table[sp->type].encoding == OPUS){
// Execute Opus decoder even when muted to keep its state updated
if(!sp->opus){
int error;
// Decode Opus to the selected sample rate
sp->opus = opus_decoder_create(DAC_samprate,Channels,&error);
if(error != OPUS_OK)
fprintf(stderr,"opus_decoder_create error %d\n",error);
assert(sp->opus);
}
sp->channels = Channels;
sp->samprate = DAC_samprate;
// Opus RTP timestamps always referenced to 48 kHz
int const r0 = opus_packet_get_nb_samples(pkt->data,pkt->len,48000);
if(r0 == OPUS_INVALID_PACKET || r0 == OPUS_BAD_ARG)
goto endloop;
int const r1 = opus_packet_get_nb_samples(pkt->data,pkt->len,DAC_samprate);
if(r1 == OPUS_INVALID_PACKET || r1 == OPUS_BAD_ARG)
goto endloop;
assert(r1 >= 0);
sp->frame_size = r1;
int const r2 = opus_packet_get_bandwidth(pkt->data);
if(r2 == OPUS_INVALID_PACKET || r2 == OPUS_BAD_ARG)
goto endloop;
switch(r2){
case OPUS_BANDWIDTH_NARROWBAND:
sp->bandwidth = 4;
break;
case OPUS_BANDWIDTH_MEDIUMBAND:
sp->bandwidth = 6;
break;
case OPUS_BANDWIDTH_WIDEBAND:
sp->bandwidth = 8;
break;
case OPUS_BANDWIDTH_SUPERWIDEBAND:
sp->bandwidth = 12;
break;
default:
case OPUS_BANDWIDTH_FULLBAND:
sp->bandwidth = 20;
break;
}
size_t const bounce_size = sizeof(*bounce) * sp->frame_size * sp->channels;
assert(bounce == NULL); // detect possible memory leaks
bounce = malloc(bounce_size);
int const samples = opus_decode_float(sp->opus,pkt->data,pkt->len,bounce,bounce_size,0);
if(samples != sp->frame_size)
fprintf(stderr,"samples %d frame-size %d\n",samples,sp->frame_size);
} else { // PCM
// Test for invalidity
int const samprate = samprate_from_pt(sp->type);
if(samprate == 0)
goto endloop;
sp->samprate = samprate;
upsample = DAC_samprate / sp->samprate; // Upsample lower PCM samprates to output rate (should be cleaner; what about decimation?)
sp->bandwidth = sp->samprate / 2000; // in kHz allowing for Nyquist
sp->channels = channels_from_pt(sp->type); // channels in packet (not portaudio output buffer)
if(sp->samprate <= 0 || sp->channels <= 0 || sp->channels > 2)
goto endloop;
sp->frame_size = pkt->len / (sizeof(int16_t) * sp->channels); // mono/stereo samples in frame
if(sp->frame_size <= 0)
goto endloop;
int16_t const * const data_ints = (int16_t *)&pkt->data[0];
assert(bounce == NULL);
bounce = malloc(sizeof(*bounce) * sp->frame_size * sp->channels);
for(int i=0; i < sp->channels * sp->frame_size; i++)
bounce[i] = SCALE16 * (int16_t)ntohs(data_ints[i]);
}
// Run PL tone decoders
// Disable if display isn't active and autonotching is off
// Fed audio that might be discontinuous or out of sequence, but it's a pain to fix
if(sp->notch_enable) {
for(int i=0; i < sp->frame_size; i++){
float s;
if(sp->channels == 2)
s = 0.5 * (bounce[2*i] + bounce[2*i+1]); // Mono sum
else // sp->channels == 1
s = bounce[i];
for(int j = 0; j < N_tones; j++)
update_goertzel(&sp->tone_detector[j],s);
}
sp->tone_samples += sp->frame_size;
if(sp->tone_samples >= Tone_period * sp->samprate){
sp->tone_samples = 0;
int pl_tone_index = -1;
float strongest_tone_energy = 0;
float total_energy = 0;
for(int j=0; j < N_tones; j++){
float energy = cnrmf(output_goertzel(&sp->tone_detector[j]));
total_energy += energy;
reset_goertzel(&sp->tone_detector[j]);
if(energy > strongest_tone_energy){
strongest_tone_energy = energy;
pl_tone_index = j;
}
}
if(2*strongest_tone_energy > total_energy && pl_tone_index >= 0){
// Tone must be > -3dB relative to total of all tones
sp->current_tone = PL_tones[pl_tone_index];
} else
sp->current_tone = 0;
} // End of tone observation period
if(sp->current_tone != 0 && sp->notch_tone != sp->current_tone){
// New or changed tone
sp->notch_tone = sp->current_tone;
setIIRnotch(&sp->iir_right,sp->current_tone/sp->samprate);
setIIRnotch(&sp->iir_left,sp->current_tone/sp->samprate);
}
} // sp->notch_enable
// Count samples and frames and advance write pointer even when muted
sp->tot_active += (float)sp->frame_size / sp->samprate;
sp->active += (float)sp->frame_size / sp->samprate;
if(sp->muted)
goto endloop; // No more to do with this frame
kick_output(); // Ensure Rptr is current
// Sequence number processing and write pointer updating
if(modsub(sp->wptr,Rptr,BUFFERSIZE) < 0){
sp->lates++;
if(++consec_lates < 3 || Constant_delay)
goto endloop; // Drop packet as late
// 3 or more consecutive lates triggers a reset
sp->reset = true;
}
consec_lates = 0;
if(modsub(sp->wptr,Rptr,BUFFERSIZE) > BUFFERSIZE/4){
sp->earlies++;
if(++consec_earlies < 3)
goto endloop; // Drop if just a few
sp->reset = true; // should this happen if Constant_delay is set?
}
consec_earlies = 0;
if(sp->reset)
reset_session(sp,pkt->rtp.timestamp); // Resets sp->wptr and last_timestamp
else {
// Normal packet, relative adjustment to write pointer
// Can difference in timestamps be negative? Cast it anyway
// Opus always counts timestamps at 48 kHz so this breaks when DAC_samprate is not 48 kHz
// For opus, sp->wptr += (int32_t)(pkt->rtp.timestamp - sp->last_timestamp) * DAC_samprate / 48000;
sp->wptr += (int32_t)(pkt->rtp.timestamp - sp->last_timestamp) * upsample;
sp->wptr &= (BUFFERSIZE-1);
sp->last_timestamp = pkt->rtp.timestamp;
}
if(Channels == 2){
/* Compute gains and delays for stereo imaging
Extreme gain differences can make the source sound like it's inside an ear
This can be uncomfortable in good headphones with extreme panning
-6dB for each channel in the center
when full to one side or the other, that channel is +6 dB and the other is -inf dB */
float const left_gain = sp->gain * (1 - sp->pan)/2;
float const right_gain = sp->gain * (1 + sp->pan)/2;
/* Delay less favored channel 0 - 1.5 ms max (determined
empirically) This is really what drives source localization
in humans. The effect is so dramatic even with equal levels
you have to remove one earphone to convince yourself that the
levels really are the same! */
int const left_delay = (sp->pan > 0) ? round(sp->pan * .0015 * DAC_samprate) : 0; // Delay left channel
int const right_delay = (sp->pan < 0) ? round(-sp->pan * .0015 * DAC_samprate) : 0; // Delay right channel
assert(left_delay >= 0 && right_delay >= 0);
// Mix bounce buffer into output buffer read by portaudio callback
// Simplified by mirror buffer wrap
int left_index = 2 * (sp->wptr + left_delay);
int right_index = 2 * (sp->wptr + right_delay) + 1;
for(int i=0; i < sp->frame_size; i++){
float left,right;
if(sp->channels == 1){
// Mono input, put on both channels
left = bounce[i];
if(sp->notch_enable && sp->notch_tone > 0)
left = applyIIRnotch(&sp->iir_left,left);
right = left;
} else {
// stereo input
left = bounce[2*i];
right = bounce[2*i+1];
if(sp->notch_enable && sp->notch_tone > 0){
left = applyIIRnotch(&sp->iir_left,left);
right = applyIIRnotch(&sp->iir_right,right);
}
}
// Not the cleanest way to upsample the sample rate, but it works
for(int j=0; j < upsample; j++){
Output_buffer[left_index] += left * left_gain;
Output_buffer[right_index] += right * right_gain;
left_index += 2;
right_index += 2;
}
if(modsub(right_index/2,Wptr,BUFFERSIZE) > 0)
Wptr = right_index / 2; // samples to frames; For verbose mode
}
} else { // Channels == 1, no panning
int64_t index = sp->wptr;
for(int i=0; i < sp->frame_size; i++){
float s;
if(sp->channels == 1){
s = bounce[i];
} else {
// Downmix to mono
s = 0.5 * (bounce[2*i] + bounce[2*i+1]);
}
if(sp->notch_enable && sp->notch_tone > 0)
s = applyIIRnotch(&sp->iir_left,s);
// Not the cleanest way to upsample the sample rate, but it works
for(int j=0; j < upsample; j++){
Output_buffer[index++] += s * sp->gain;
}
if(modsub(index,Wptr,BUFFERSIZE) > 0)
Wptr = index; // For verbose mode
}
} // Channels == 1
endloop:;
FREE(bounce);
FREE(pkt);
} // !sp->terminate
pthread_cleanup_pop(1);
return NULL;
}
// Use ncurses to display streams
static void *display(void *arg){
pthread_setname("display");
if(initscr() == NULL){
fprintf(stderr,"initscr() failed, disabling control/display thread\n");
pthread_exit(NULL);
}
keypad(stdscr,TRUE);
timeout(Update_interval);
cbreak();
noecho();
int first_session = 0;
int sessions_per_screen = 0;
int current = -1; // No current session
bool help = false;
while(!Terminate){
assert(first_session >= 0);
assert(first_session == 0 || first_session < Nsessions);
assert(current >= -1);
assert(current == -1 || current < Nsessions); // in case Nsessions is 0
move(0,0);
clrtobot();
addstr("KA9Q Multicast Audio Monitor:");
for(int i=0;i<Nfds;i++)
printw(" %s",Mcast_address_text[i]);
addstr("\n");
if(help){
char path [PATH_MAX];
dist_path(path,sizeof(path),"monitor-help.txt");
FILE *fp = fopen(path,"r");
if(fp != NULL){
size_t size = 1024;
char *line = malloc(size);
while(getline(&line,&size,fp) != -1)
addstr(line);
FREE(line);
fclose(fp);
fp = NULL;
}
}
if(Start_muted){
int y;
int x __attribute__ ((unused));
getyx(stdscr,y,x);
mvaddstr(y,0,"**Starting new sessions muted**");
}