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authorMartin Blanchard <tchaik@gmx.com>2016-11-06 12:54:02 -0600
committerTanu Kaskinen <tanuk@iki.fi>2017-01-19 03:00:45 +0200
commit31e2bc2fcf828e399e62e1525e1f4fc959475106 (patch)
tree7de94900fc7536a072460205e2e4e7c143082aaa /src
parent604bf450dc200ee15a8cb1158aa8130caa5b9e2e (diff)
raop: Better playback resume handling
When playback stops, a FLUSH command is send to the server and the sink goes to IDLE. If playback resumes quickly, sink goes back to RUNNING (without being SUSPENDED) and the sink should just start streaming again. This patch implements this behaviour.
Diffstat (limited to 'src')
-rw-r--r--src/modules/raop/module-raop-sink.c8
-rw-r--r--src/modules/raop/raop_client.c26
-rw-r--r--src/modules/raop/raop_client.h1
3 files changed, 30 insertions, 5 deletions
diff --git a/src/modules/raop/module-raop-sink.c b/src/modules/raop/module-raop-sink.c
index a8a466d8..e251f2fc 100644
--- a/src/modules/raop/module-raop-sink.c
+++ b/src/modules/raop/module-raop-sink.c
@@ -332,10 +332,14 @@ static int udp_sink_process_msg(pa_msgobject *o, int code, void *data, int64_t o
pa_smoother_resume(u->smoother, pa_rtclock_now(), true);
- if (!pa_raop_client_udp_can_stream(u->raop)) {
- /* Connecting will trigger a RECORD */
+ if (!pa_raop_client_udp_is_alive(u->raop)) {
+ /* Connecting will trigger a RECORD and start steaming */
pa_raop_client_connect(u->raop);
+ } else if (!pa_raop_client_udp_can_stream(u->raop)) {
+ /* RECORD alredy sent, simply start streaming */
+ pa_raop_client_udp_stream(u->raop);
}
+
udp_start_wakeup_clock(u);
break;
diff --git a/src/modules/raop/raop_client.c b/src/modules/raop/raop_client.c
index d17cc44a..16a59cca 100644
--- a/src/modules/raop/raop_client.c
+++ b/src/modules/raop/raop_client.c
@@ -952,8 +952,6 @@ static void udp_rtsp_cb(pa_rtsp_client *rtsp, pa_rtsp_state state, pa_headerlist
pa_log_debug("RAOP: RECORD");
- c->is_recording = true;
-
alt = pa_xstrdup(pa_headerlist_gets(headers, "Audio-Latency"));
/* Generate a random synchronization source identifier from this session. */
pa_random(&rand, sizeof(rand));
@@ -965,6 +963,8 @@ static void udp_rtsp_cb(pa_rtsp_client *rtsp, pa_rtsp_state state, pa_headerlist
c->udp_first_packet = true;
c->udp_sync_count = 0;
+ c->is_recording = true;
+
c->udp_record_callback(c->udp_setup_userdata);
pa_xfree(alt);
@@ -1168,11 +1168,12 @@ int pa_raop_client_connect(pa_raop_client *c) {
int pa_raop_client_flush(pa_raop_client *c) {
int rv = 0;
+
pa_assert(c);
if (c->rtsp != NULL) {
rv = pa_rtsp_flush(c->rtsp, c->seq, c->rtptime);
- c->udp_sync_count = -1;
+ c->udp_sync_count = 0;
}
return rv;
@@ -1211,6 +1212,25 @@ int pa_raop_client_udp_can_stream(pa_raop_client *c) {
return rv;
}
+int pa_raop_client_udp_stream(pa_raop_client *c) {
+ int rv = 0;
+
+ pa_assert(c);
+
+ if (c->rtsp != NULL && c->udp_stream_fd > 0) {
+ if (!c->is_recording) {
+ c->udp_first_packet = true;
+ c->udp_sync_count = 0;
+
+ c->is_recording = true;
+ }
+
+ rv = 1;
+ }
+
+ return rv;
+}
+
int pa_raop_client_udp_handle_timing_packet(pa_raop_client *c, const uint8_t packet[], ssize_t size) {
const uint32_t * data = NULL;
uint8_t payload = 0;
diff --git a/src/modules/raop/raop_client.h b/src/modules/raop/raop_client.h
index 6ab6d325..d49c1462 100644
--- a/src/modules/raop/raop_client.h
+++ b/src/modules/raop/raop_client.h
@@ -43,6 +43,7 @@ int pa_raop_client_teardown(pa_raop_client *c);
int pa_raop_client_udp_is_alive(pa_raop_client *c);
int pa_raop_client_udp_can_stream(pa_raop_client *c);
+int pa_raop_client_udp_stream(pa_raop_client *c);
void pa_raop_client_set_encryption(pa_raop_client *c, int encryption);
pa_volume_t pa_raop_client_adjust_volume(pa_raop_client *c, pa_volume_t volume);