Qt: Initial RTP playback.

Note the "initial". This is woefully incomplete.  See the "to do" lists
below and in the code.

This differs a bit from the GTK+ version in that you specify one or more
streams to be decoded.

Instead of showing waveforms in individual widgets, add them all to a
single QCustomPlot. This conserves screen real estate and lets us more
easily take advantage of the QCP API. It also looks better IMHO.

Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We
probably won't use the widgets until we make 5.0 our minimum Qt
version and plain old QtMultimedia lets us support Qt 4 more easily
(in theory at least).

Add resampling code from libspeex. I initially used this to resample
each packet to match the preferred rate of our output device, but this
resulted in poorer audio quality than expected. Leave it in and use to
create visual samples for QCP and to match rates any time the rate
changes. The latter is currently untested.

Add some debugging macros.

Note that both the RTP player and RTP analysis dialogs decode audio data
using different code.

Note that voip_calls_packet and voip_calls_init_tap appear to be dead
code.

To do:

- Add silence frames where needed.
- Implement the jitter buffer.
- Implement the playback timing controls.
- Tapping / scanning streams might be too slow.

Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4
Bug: 9007
Reviewed-on: https://code.wireshark.org/review/10458
Petri-Dish: Gerald Combs <gerald@wireshark.org>
Reviewed-by: Gerald Combs <gerald@wireshark.org>

1 files changed, 14 insertions, 5 deletions

diff --git a/ui/rtp_stream.h b/ui/rtp_stream.h
index d5c85b9f96..2e747ed481 100644
--- a/ui/rtp_stream.h
+++ b/ui/rtp_stream.h
@@ -60,7 +60,7 @@ typedef struct _rtp_stream_info {
     gboolean        end_stream; /**< Used to track streams across payload types */
     int             rtp_event;
 
-    guint16         call_num; /**< Used to match call_num in voip_calls_info_t */
+    int             call_num; /**< Used to match call_num in voip_calls_info_t */
     guint32         setup_frame_number; /**< frame number of setup message */
     /* Start and stop packets needed for .num and .abs_ts */
     frame_data     *start_fd;
@@ -71,14 +71,13 @@ typedef struct _rtp_stream_info {
     gboolean        tag_vlan_error;
     gboolean        tag_diffserv_error;
 
-    gboolean        decode; /**< Decode this stream */
-    GList          *rtp_packet_list; /**< List of RTP rtp_packet_t */
+    gboolean        decode; /**< Decode this stream. GTK+ only? */
+    GList          *rtp_packet_list; /**< List of RTP rtp_packet_t. GTK+ only */
 
     tap_rtp_stat_t  rtp_stats;  /**< here goes the RTP statistics info */
-    gboolean        problem;    /**< if the streams had wrong sequence numbers or wrong timerstamps */
+    gboolean        problem;    /**< if the streams had wrong sequence numbers or wrong timestamps */
 } rtp_stream_info_t;
 
-
 /** tapping modes */
 typedef enum
 {
@@ -111,6 +110,16 @@ struct _rtpstream_tapinfo {
     gboolean           is_registered; /**< if the tap listener is currently registered or not */
 };
 
+#if 0
+#define RTP_STREAM_DEBUG(...) { \
+    char *RTP_STREAM_DEBUG_MSG = g_strdup_printf(__VA_ARGS__); \
+    g_warning("rtp_stream: %s:%d %s", G_STRFUNC, __LINE__, RTP_STREAM_DEBUG_MSG); \
+    g_free(RTP_STREAM_DEBUG_MSG); \
+}
+#else
+#define RTP_STREAM_DEBUG()
+#endif
+
 /****************************************************************************/
 /* INTERFACE */