From: David Woodhouse <dwmw@amazon.co.uk>
Date: Tue, 7 May 2019 15:40:17 +0000 (+0100)
Subject: Attempt to drop only whole frames, not arbitary gaps.
X-Git-Tag: v1.2~3
X-Git-Url: https://www.infradead.org/git/?a=commitdiff_plain;h=b814927fbc4dd276b38a4f501e4b0031002c6040;p=pidgin-chime.git

Attempt to drop only whole frames, not arbitary gaps.

Perhaps this stops the strange effects we get when we lose outbound
audio frames on a busy system and something goes out of sync.
---

diff --git a/chime/chime-call-audio.c b/chime/chime-call-audio.c
index ff463ab..52c7469 100644
--- a/chime/chime-call-audio.c
+++ b/chime/chime-call-audio.c
@@ -147,28 +147,6 @@ static void do_send_rt_packet(ChimeCallAudio *audio, GstBuffer *buffer)
 		chime_debug("RX timeout, reconnect audio\n");
 		audio->timeout_source = g_timeout_add(0, audio_reconnect, audio);
 	}
-	audio->audio_msg.seq = (audio->audio_msg.seq + 1) & 0xffff;
-
-	if (audio->last_server_time_offset) {
-		gint64 t = audio->last_server_time_offset + now;
-		if (audio->echo_server_time) {
-			audio->audio_msg.has_echo_time = 1;
-			audio->audio_msg.echo_time = t;
-			audio->echo_server_time = FALSE;
-		}
-		audio->audio_msg.has_server_time = TRUE;
-		audio->audio_msg.server_time = t;
-	} else
-		audio->audio_msg.has_echo_time = 0;
-
-	audio->audio_msg.has_total_frames_lost = TRUE;
-	audio->audio_msg.total_frames_lost = 0;
-
-	audio->audio_msg.has_ntp_time = TRUE;
-	audio->audio_msg.ntp_time = g_get_real_time();
-
-	audio->audio_msg.has_audio = TRUE;
-
 	if (buffer && GST_BUFFER_DURATION_IS_VALID(buffer) &&
 	    GST_BUFFER_DTS_IS_VALID(buffer) && gst_rtp_buffer_map(buffer, GST_MAP_READ, &rtp)) {
 		GstClockTime dts, pts, dur;
@@ -179,12 +157,24 @@ static void do_send_rt_packet(ChimeCallAudio *audio, GstBuffer *buffer)
 
 		nr_samples = GST_BUFFER_DURATION(buffer) / NS_PER_SAMPLE;
 		chime_debug("buf dts %ld pts %ld dur %ld samples %d\n", dts, pts, dur, nr_samples);
+		if (audio->next_dts) {
+			int frames_missed;
 
-		if (audio->next_dts && dts > audio->next_dts) {
-			/* We skipped some. */
-			audio->audio_msg.sample_time += (dts - audio->next_dts) / NS_PER_SAMPLE;
+			if (dts < audio->next_dts) {
+				chime_debug("Out of order frame %ld < %ld\n", dts, audio->next_dts);
+				goto drop;
+			}
+			frames_missed = (dts - audio->next_dts) / dur;
+			if (frames_missed) {
+				chime_debug("Missed %d frames\n", frames_missed);
+				audio->audio_msg.sample_time += frames_missed * nr_samples;
+				audio->next_dts += frames_missed * dur;
+			}
+			audio->next_dts += dur;
+		} else {
+			audio->next_dts = dts + dur;
 		}
-		audio->next_dts = dts + dur;
+
 		if (audio->state == CHIME_AUDIO_STATE_AUDIO) {
 //			printf ("State %d, send audio\n", audio->state);
 			audio->audio_msg.audio.len = gst_rtp_buffer_get_payload_len(&rtp);
@@ -201,6 +191,28 @@ static void do_send_rt_packet(ChimeCallAudio *audio, GstBuffer *buffer)
 		nr_samples = 320;
 		audio->audio_msg.audio.len = 0;
 	}
+	audio->audio_msg.seq = (audio->audio_msg.seq + 1) & 0xffff;
+
+	if (audio->last_server_time_offset) {
+		gint64 t = audio->last_server_time_offset + now;
+		if (audio->echo_server_time) {
+			audio->audio_msg.has_echo_time = 1;
+			audio->audio_msg.echo_time = t;
+			audio->echo_server_time = FALSE;
+		}
+		audio->audio_msg.has_server_time = TRUE;
+		audio->audio_msg.server_time = t;
+	} else
+		audio->audio_msg.has_echo_time = 0;
+
+	audio->audio_msg.has_total_frames_lost = TRUE;
+	audio->audio_msg.total_frames_lost = 0;
+
+	audio->audio_msg.has_ntp_time = TRUE;
+	audio->audio_msg.ntp_time = g_get_real_time();
+
+	audio->audio_msg.has_audio = TRUE;
+
 	audio->last_send_local_time = now;
 	chime_call_transport_send_packet(audio, XRP_RT_MESSAGE, &audio->rt_msg.base);
 	if (audio->audio_msg.audio.data) {
@@ -208,6 +220,7 @@ static void do_send_rt_packet(ChimeCallAudio *audio, GstBuffer *buffer)
 		gst_rtp_buffer_unmap(&rtp);
 	}
 	audio->audio_msg.sample_time += nr_samples;
+ drop:
 	g_mutex_unlock(&audio->rt_lock);
 }