module-rtp: use sess.latency.msec also for sender

Use the sess.latency.msec also for the sender and use it to control the
NODE_LATENCY. Make it a float to be in line with the other time values.
Set is to a default of ptime, which was what it used to be.

This makes it possible to set the ptime to a smaller value than the
sess.latency.msec so that we send out multiple packets per quantum.
This will result in some bursty output for now but with a timer that can
be improved later.

Update the docs a little, mention the new rtp.ptime and rtp.frametime.

src/modules/module-rtp-sink.c

@@ -52,9 +52,14 @@
  * - `net.mtu = <int>`: MTU to use, default 1280
  * - `net.ttl = <int>`: TTL to use, default 1
  * - `net.loop = <bool>`: loopback multicast, default false
- * - `sess.min-ptime = <int>`: minimum packet time in milliseconds, default 2
+ * - `sess.min-ptime = <float>`: minimum packet time in milliseconds, default 2
- * - `sess.max-ptime = <int>`: maximum packet time in milliseconds, default 20
+ * - `sess.max-ptime = <float>`: maximum packet time in milliseconds, default 20
  * - `sess.name = <str>`: a session name
+ * - `rtp.ptime = <float>`: size of the packets in milliseconds, default up to MTU but
+ *       between sess.min-ptime and sess.max-ptime
+ * - `rtp.framecount = <int>`: number of samples per packet, default up to MTU but
+ *       between sess.min-ptime and sess.max-ptime
+ * - `sess.latency.msec = <float>`: target node latency in milliseconds, default as rtp.ptime
  * - `sess.ts-offset = <int>`: an offset to apply to the timestamp, default -1 = random offset
  * - `sess.ts-refclk = <string>`: the name of a reference clock
  * - `sess.media = <string>`: the media type audio|midi|opus, default audio

src/modules/module-rtp-source.c

@@ -56,7 +56,7 @@
  *                you want to receive packets from or 0.0.0.0 to receive from any source address.
  * - `source.port = <int>`: the source port
  * - `node.always-process = <bool>`: true to receive even when not running
- * - `sess.latency.msec = <str>`: target network latency in milliseconds, default 100
+ * - `sess.latency.msec = <float>`: target network latency in milliseconds, default 100
  * - `sess.ignore-ssrc = <bool>`: ignore SSRC, default false
  * - `sess.media = <string>`: the media type audio|midi|opus, default audio
  * - `stream.props = {}`: properties to be passed to the stream

src/modules/module-rtp/stream.c

@@ -284,7 +284,7 @@ struct rtp_stream *rtp_stream_new(struct pw_core *core,
 	float min_ptime, max_ptime;
 	const struct spa_pod *params[1];
 	enum pw_stream_flags flags;
-	int latency_msec;
+	float latency_msec;
 	int res;
 
 	impl = calloc(1, sizeof(*impl));
@@ -450,16 +450,23 @@ struct rtp_stream *rtp_stream_new(struct pw_core *core,
 			pw_properties_setf(props, "rtp.framecount", "%u", impl->psamples);
 		}
 	}
-	latency_msec = pw_properties_get_uint32(props,
+	/* For senders, the default latency is ptime and for a receiver it is
-			"sess.latency.msec", DEFAULT_SESS_LATENCY);
+	 * DEFAULT_SESS_LATENCY. Setting the sess.latency.msec for a sender to
+	 * something smaller/bigger will influence the quantum and the amount
+	 * of packets we send in one cycle */
+	str = pw_properties_get(props, "sess.latency.msec");
+	if (!spa_atof(str, &latency_msec)) {
+		latency_msec = direction == PW_DIRECTION_INPUT ?
+			samples_to_msec(impl, impl->psamples) :
+			DEFAULT_SESS_LATENCY;
+	}
 	impl->target_buffer = msec_to_samples(impl, latency_msec);
 	impl->max_error = msec_to_samples(impl, ERROR_MSEC);
 
 	pw_properties_setf(props, PW_KEY_NODE_RATE, "1/%d", impl->rate);
 	if (direction == PW_DIRECTION_INPUT) {
-		// TODO: make sess.latency.msec work for sender streams
 		pw_properties_setf(props, PW_KEY_NODE_LATENCY, "%d/%d",
-				impl->psamples, impl->rate);
+				impl->target_buffer, impl->rate);
 	} else {
 		pw_properties_setf(props, PW_KEY_NODE_LATENCY, "%d/%d",
 				impl->target_buffer / 2, impl->rate);

src/modules/module-rtp/stream.h

@@ -16,12 +16,12 @@ struct rtp_stream;
 #define DEFAULT_CHANNELS	2
 #define DEFAULT_POSITION	"[ FL FR ]"
 
-#define ERROR_MSEC			2
+#define ERROR_MSEC		2.0f
-#define DEFAULT_SESS_LATENCY		100
+#define DEFAULT_SESS_LATENCY	100.0f
 
 #define DEFAULT_MTU		1280
-#define DEFAULT_MIN_PTIME	2
+#define DEFAULT_MIN_PTIME	2.0f
-#define DEFAULT_MAX_PTIME	20
+#define DEFAULT_MAX_PTIME	20.0f
 
 struct rtp_stream_events {
 #define RTP_VERSION_STREAM_EVENTS        0