Change SIGUSR2 to turn on and off reserving the output device.

audio_alsa.c

@@ -181,15 +181,16 @@ static void deinit(void) {
  }
 }
 
-void open_alsa_device(void) { 
+int open_alsa_device(void) { 
 
  int ret, dir = 0;
  unsigned int my_sample_rate = desired_sample_rate;
  snd_pcm_uframes_t frames = 441*10;
  snd_pcm_uframes_t buffer_size = frames*4;
  ret = snd_pcm_open(&alsa_handle, alsa_out_dev, SND_PCM_STREAM_PLAYBACK, 0);
  if (ret < 0)
- die("Alsa initialization failed: unable to open pcm device: %s.", snd_strerror(ret));
+ return(ret);
+ // die("Alsa initialization failed: unable to open pcm device: %s.", snd_strerror(ret));
 
  snd_pcm_hw_params_alloca(&alsa_params);
  snd_pcm_hw_params_any(alsa_handle, alsa_params);
@@ -206,6 +207,7 @@ void open_alsa_device(void) {
  if (my_sample_rate!=desired_sample_rate) {
  die("Can't set the D/A converter to %d -- set to %d instead./n",desired_sample_rate,my_sample_rate);
  }
+ return(0);
 }
 
 void close_alsa_device(void) {
@@ -220,7 +222,6 @@ static void start(int sample_rate) {
  if (sample_rate != 44100)
  die("Unexpected sample rate %d -- only 44,100 supported!",sample_rate);
  desired_sample_rate = sample_rate; // must be a variable
- // open_alsa_device();
 }
 
 static uint32_t delay() {
@@ -259,25 +260,28 @@ static uint32_t delay() {
 }
 
 static void play(short buf[], int samples) {
+ int ret = 0;
  if (alsa_handle==NULL)
- open_alsa_device();
- pthread_mutex_lock(&alsa_mutex);
- snd_pcm_sframes_t current_delay = 0;
- int err,ignore;
- if ((snd_pcm_state(alsa_handle)==SND_PCM_STATE_PREPARED) || (snd_pcm_state(alsa_handle)==SND_PCM_STATE_RUNNING)) {
- err = snd_pcm_writei(alsa_handle, (char*)buf, samples);
- if (err < 0) {
- ignore = snd_pcm_recover(alsa_handle, err, 0);
- debug(1,"Error %d writing %d samples in play() %s.",err,samples, snd_strerror(err));
- }
- } else {
- debug(1,"Error -- ALSA device in incorrect state (%d) for play.",snd_pcm_state(alsa_handle));
- if (err = snd_pcm_prepare(alsa_handle)) {
- ignore = snd_pcm_recover(alsa_handle, err, 0);
- debug(1,"Error preparing after play error: %s.", snd_strerror(err));
+ ret = open_alsa_device();
+ if (ret==0) {
+ pthread_mutex_lock(&alsa_mutex);
+ snd_pcm_sframes_t current_delay = 0;
+ int err,ignore;
+ if ((snd_pcm_state(alsa_handle)==SND_PCM_STATE_PREPARED) || (snd_pcm_state(alsa_handle)==SND_PCM_STATE_RUNNING)) {
+ err = snd_pcm_writei(alsa_handle, (char*)buf, samples);
+ if (err < 0) {
+ ignore = snd_pcm_recover(alsa_handle, err, 0);
+ debug(1,"Error %d writing %d samples in play() %s.",err,samples, snd_strerror(err));
+ }
+ } else {
+ debug(1,"Error -- ALSA device in incorrect state (%d) for play.",snd_pcm_state(alsa_handle));
+ if (err = snd_pcm_prepare(alsa_handle)) {
+ ignore = snd_pcm_recover(alsa_handle, err, 0);
+ debug(1,"Error preparing after play error: %s.", snd_strerror(err));
+ }
  }
+ pthread_mutex_unlock(&alsa_mutex);
  }
- pthread_mutex_unlock(&alsa_mutex);
 }
 
 static void flush(void) {

common.c

@@ -69,11 +69,23 @@
 #include "common.h"
 #include <libdaemon/dlog.h>
 
+//true if Shairport Sync is supposed to be sending output to the output device, false otherwise
+
+static volatile int requested_connection_state_to_output = 1;
 
 shairport_cfg config;
 
 int debuglev = 0;
 
+int get_requested_connection_state_to_output() {
+ return requested_connection_state_to_output;
+}
+
+void set_requested_connection_state_to_output(int v) {
+ requested_connection_state_to_output = v;
+}
+
+
 void die(char *format, ...) {
  char s[1024];
  s[0]=0;

common.h

@@ -63,6 +63,12 @@ typedef struct {
  char *errfile;
 } shairport_cfg;
 
+//true if Shairport Sync is supposed to be sending output to the output device, false otherwise
+
+int get_requested_connection_state_to_output();
+
+void set_requested_connection_state_to_output(int v);
+
 int debuglev;
 void die(char *format, ...);
 void warn(char *format, ...);

player.c

@@ -82,6 +82,8 @@ static aes_context dctx;
 static pthread_t player_thread;
 static int please_stop;
 
+static int connection_state_to_output; // if true, then play incoming stuff; if false drop everything
+
 static alac_file *decoder_info;
 
 // debug variables
@@ -296,72 +298,75 @@ static void free_buffer(void) {
 void player_put_packet(seq_t seqno,uint32_t timestamp, uint8_t *data, int len) {
 
  packet_count++;
+ time_of_last_audio_packet = get_absolute_time_in_fp();
 
  pthread_mutex_lock(&ab_mutex);
 
- if ((flush_rtp_timestamp!=0x7fffffff) && ((timestamp==flush_rtp_timestamp) || seq32_order(timestamp,flush_rtp_timestamp))) {
- debug(2,"Dropping flushed packet in player_put_packet, seqno %u, timestamp %u, flushing to timestamp: %u.",seqno,timestamp,flush_rtp_timestamp);
- } else {
- if ((flush_rtp_timestamp!=0x7fffffff) && (!seq32_order(timestamp,flush_rtp_timestamp))) // if we have gone past the flush boundary time
- flush_rtp_timestamp=0x7fffffff;
-
- abuf_t *abuf = 0;
-
- if (!ab_synced) {
- debug(2, "syncing to seqno %u.", seqno);
- ab_write = seqno;
- ab_read = seqno;
- ab_synced = 1;
- }
- if (ab_write == seqno) { // expected packet
- abuf = audio_buffer + BUFIDX(seqno);
- ab_write = SUCCESSOR(seqno);
- } else if (seq_order(ab_write, seqno)) { // newer than expected
- //if (ORDINATE(seqno)>(BUFFER_FRAMES*7)/8)
- // debug(1,"An interval of %u frames has opened, with ab_read: %u, ab_write: %u and seqno: %u.",seq_diff(ab_read,seqno),ab_read,ab_write,seqno);
- int32_t gap = seq_diff(ab_write,PREDECESSOR(seqno))+1;
- if (gap<=0)
- debug(1,"Unexpected gap size: %d.",gap);
- int i;
- for (i=0;i<gap;i++) {
- abuf = audio_buffer + BUFIDX(seq_sum(ab_write,i));
- abuf->ready = 0; // to be sure, to be sure
- abuf->timestamp = 0;
- abuf->sequence_number = 0;
- }
- // debug(1,"N %d s %u.",seq_diff(ab_write,PREDECESSOR(seqno))+1,ab_write);
- abuf = audio_buffer + BUFIDX(seqno);
- rtp_request_resend(ab_write,gap);
- resend_requests++;
- ab_write = SUCCESSOR(seqno);
- } else if (seq_order(ab_read, seqno)) { // late but not yet played
- late_packets++;
- abuf = audio_buffer + BUFIDX(seqno);
- } else { // too late.
- too_late_packets++;
- /*
- if (!late_packet_message_sent) {
- debug(1, "too-late packet received: %u; ab_read: %u; ab_write: %u.", seqno, ab_read, ab_write);
- late_packet_message_sent=1;
- }
- */
- }
- // pthread_mutex_unlock(&ab_mutex);
-
- if (abuf) {
- alac_decode(abuf->data, data, len);
- abuf->ready = 1;
- abuf->timestamp = timestamp;
- abuf->sequence_number = seqno;
- }
+ if (connection_state_to_output) { // if we are supposed to be processing these packets
+
+ if ((flush_rtp_timestamp!=0x7fffffff) && ((timestamp==flush_rtp_timestamp) || seq32_order(timestamp,flush_rtp_timestamp))) {
+ debug(2,"Dropping flushed packet in player_put_packet, seqno %u, timestamp %u, flushing to timestamp: %u.",seqno,timestamp,flush_rtp_timestamp);
+ } else {
+ if ((flush_rtp_timestamp!=0x7fffffff) && (!seq32_order(timestamp,flush_rtp_timestamp))) // if we have gone past the flush boundary time
+ flush_rtp_timestamp=0x7fffffff;
+
+ abuf_t *abuf = 0;
+
+ if (!ab_synced) {
+ debug(2, "syncing to seqno %u.", seqno);
+ ab_write = seqno;
+ ab_read = seqno;
+ ab_synced = 1;
+ }
+ if (ab_write == seqno) { // expected packet
+ abuf = audio_buffer + BUFIDX(seqno);
+ ab_write = SUCCESSOR(seqno);
+ } else if (seq_order(ab_write, seqno)) { // newer than expected
+ //if (ORDINATE(seqno)>(BUFFER_FRAMES*7)/8)
+ // debug(1,"An interval of %u frames has opened, with ab_read: %u, ab_write: %u and seqno: %u.",seq_diff(ab_read,seqno),ab_read,ab_write,seqno);
+ int32_t gap = seq_diff(ab_write,PREDECESSOR(seqno))+1;
+ if (gap<=0)
+ debug(1,"Unexpected gap size: %d.",gap);
+ int i;
+ for (i=0;i<gap;i++) {
+ abuf = audio_buffer + BUFIDX(seq_sum(ab_write,i));
+ abuf->ready = 0; // to be sure, to be sure
+ abuf->timestamp = 0;
+ abuf->sequence_number = 0;
+ }
+ // debug(1,"N %d s %u.",seq_diff(ab_write,PREDECESSOR(seqno))+1,ab_write);
+ abuf = audio_buffer + BUFIDX(seqno);
+ rtp_request_resend(ab_write,gap);
+ resend_requests++;
+ ab_write = SUCCESSOR(seqno);
+ } else if (seq_order(ab_read, seqno)) { // late but not yet played
+ late_packets++;
+ abuf = audio_buffer + BUFIDX(seqno);
+ } else { // too late.
+ too_late_packets++;
+ /*
+ if (!late_packet_message_sent) {
+ debug(1, "too-late packet received: %u; ab_read: %u; ab_write: %u.", seqno, ab_read, ab_write);
+ late_packet_message_sent=1;
+ }
+ */
+ }
+ // pthread_mutex_unlock(&ab_mutex);
+
+ if (abuf) {
+ alac_decode(abuf->data, data, len);
+ abuf->ready = 1;
+ abuf->timestamp = timestamp;
+ abuf->sequence_number = seqno;
+ }
 
- // pthread_mutex_lock(&ab_mutex);
+  // pthread_mutex_lock(&ab_mutex);
 
- time_of_last_audio_packet = get_absolute_time_in_fp();
- }
- int rc = pthread_cond_signal(&flowcontrol);
- if (rc)
- debug(1,"Error signalling flowcontrol.");
+ }
+  int rc = pthread_cond_signal(&flowcontrol);
+  if (rc)
+  debug(1,"Error signalling flowcontrol.");
+ }
  pthread_mutex_unlock(&ab_mutex);
 }
 
@@ -411,7 +416,18 @@ static abuf_t *buffer_get_frame(void) {
  shutdown_requested=1;
  }
  }
-
+ int rco = get_requested_connection_state_to_output();
+
+ if (connection_state_to_output != rco) {
+ connection_state_to_output=rco;
+ // change happening
+ if (connection_state_to_output==0) { //going off
+ pthread_mutex_lock(&flush_mutex);
+ flush_requested=1;
+ pthread_mutex_unlock(&flush_mutex);
+ }
+ }
+
  pthread_mutex_lock(&flush_mutex);
  if (flush_requested==1) {
  if (config.output->flush)
@@ -750,6 +766,8 @@ typedef struct stats { // statistics for running averages
 } stats_t;
 
 static void *player_thread_func(void *arg) {
+ connection_state_to_output = get_requested_connection_state_to_output();
+ debug(1,"States: %d, %d.",connection_state_to_output,get_requested_connection_state_to_output());
 //this is about half a minute
 #define trend_interval 3758
  stats_t statistics[trend_interval];

shairport.c

@@ -95,6 +95,13 @@ static void sig_logrotate(int foo, siginfo_t *bar, void *baz) {
 // log_setup();
 }
 
+static void sig_toggle_audio_output(int foo, siginfo_t *bar, void *baz) {
+ if (get_requested_connection_state_to_output())
+ set_requested_connection_state_to_output(0);
+ else
+ set_requested_connection_state_to_output(1);
+}
+
 static void sig_pause_client(int foo, siginfo_t *bar, void *baz) {
  rtp_request_client_pause();
 }
@@ -306,7 +313,7 @@ void signal_setup(void) {
  sa.sa_sigaction = &sig_logrotate;
  sigaction(SIGHUP, &sa, NULL);
 
- sa.sa_sigaction = &sig_pause_client;
+ sa.sa_sigaction = &sig_toggle_audio_output;
  sigaction(SIGUSR2, &sa, NULL);
 
  sa.sa_sigaction = &sig_child;
@@ -426,6 +433,7 @@ int main(int argc, char **argv) {
  gethostname(hostname, 100);
  config.apname = malloc(20 + 100);
  snprintf(config.apname, 20 + 100, "Shairport Sync on %s", hostname);
+ set_requested_connection_state_to_output(1); // we expect to be able to connect to the output device
 
  // parse arguments into config
  int audio_arg = parse_options(argc, argv);