Qt: fix jitter buffer management

Small bugs were introduced when copy/pasting the code from GTK UI:
- arrive_offset is stored in seconds and not milliseconds
- some tests regarding the current playback mode were wrong

Change-Id: I21fb82ba8ff6c8defa7df90c815c040e9e074aaa
Reviewed-on: https://code.wireshark.org/review/13885
Petri-Dish: Pascal Quantin <pascal.quantin@gmail.com>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Pascal Quantin <pascal.quantin@gmail.com>

2 files changed, 5 insertions, 5 deletions

diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp
index 5cfba191e8..65cb0865e7 100644
--- a/ui/qt/rtp_audio_stream.cpp
+++ b/ui/qt/rtp_audio_stream.cpp
@@ -250,14 +250,14 @@ void RtpAudioStream::decode()
 
         double rtp_time = (double)(rtp_packet->info->info_timestamp-start_timestamp)/sample_rate - start_rtp_time;
         double arrive_time;
-        if (timing_mode_ == Uninterrupted) {
+        if (timing_mode_ == RtpTimestamp) {
             arrive_time = rtp_time;
         } else {
-            arrive_time = (double)rtp_packet->arrive_offset/1000 - start_time;
+            arrive_time = rtp_packet->arrive_offset - start_time;
         }
 
         double diff = qAbs(arrive_time - rtp_time);
-        if (diff*1000 > jitter_buffer_size_ && timing_mode_ == Uninterrupted) {
+        if (diff*1000 > jitter_buffer_size_ && timing_mode_ != Uninterrupted) {
             // rtp_player.c:628
 
             jitter_drop_timestamps_.append(stop_rel_time_);
@@ -281,7 +281,7 @@ void RtpAudioStream::decode()
 /* XXX: if timestamps (RTP) are missing/ignored try use packet arrive time only (see also "rtp_time") */
                 start_timestamp = rtp_packet->info->info_timestamp;
                 start_rtp_time = 0;
-                start_time = (double)rtp_packet->arrive_offset/1000;
+                start_time = rtp_packet->arrive_offset;
                 rtp_time_prev = 0;
             }
 
diff --git a/ui/rtp_media.h b/ui/rtp_media.h
index 5432091805..222be304b9 100644
--- a/ui/rtp_media.h
+++ b/ui/rtp_media.h
@@ -50,7 +50,7 @@ typedef gint16 SAMPLE;
 typedef struct _rtp_packet {
     guint32 frame_num;      /* Qt only */
     struct _rtp_info *info;	/* the RTP dissected info */
-    double arrive_offset;	/* arrive offset time since the beginning of the stream in ms */
+    double arrive_offset;	/* arrive offset time since the beginning of the stream as ms in GTK UI and s in Qt UI */
     guint8* payload_data;
 } rtp_packet_t;