diff options
author | Guy Harris <guy@alum.mit.edu> | 2016-12-15 00:29:38 -0800 |
---|---|---|
committer | Anders Broman <a.broman58@gmail.com> | 2016-12-15 11:52:49 +0000 |
commit | c65e5851b2eef0ef99e4cb9388eb780cd6a87aa7 (patch) | |
tree | 520a23ea24e98df2822b2b5b2f1b9975f4e87d12 | |
parent | e6a9877c7177978ff655d6f61d7540fb835ea165 (diff) | |
download | wireshark-c65e5851b2eef0ef99e4cb9388eb780cd6a87aa7.tar.gz |
Make some items that don't need to be size_t guint.
Those sizes are limited by the packet sizes we support, and we only
support a maximum packet size of 2^32.
This squelches some compiler warnings.
Remove some casts that this renders unnecessary.
Change-Id: Id9a7bcf8c2ce30bbed7be6c0e28deb9cf38002e0
Reviewed-on: https://code.wireshark.org/review/19279
Petri-Dish: Guy Harris <guy@alum.mit.edu>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Anders Broman <a.broman58@gmail.com>
-rw-r--r-- | epan/dissectors/packet-rtp.h | 4 | ||||
-rw-r--r-- | ui/gtk/rtp_analysis.c | 2 | ||||
-rw-r--r-- | ui/qt/rtp_audio_stream.cpp | 2 |
3 files changed, 4 insertions, 4 deletions
diff --git a/epan/dissectors/packet-rtp.h b/epan/dissectors/packet-rtp.h index b695c3a6a2..6513016e18 100644 --- a/epan/dissectors/packet-rtp.h +++ b/epan/dissectors/packet-rtp.h @@ -42,8 +42,8 @@ struct _rtp_info { guint32 info_sync_src; guint info_data_len; /* length of raw rtp data as reported */ gboolean info_all_data_present; /* FALSE if data is cut off */ - size_t info_payload_offset; /* start of payload relative to info_data */ - size_t info_payload_len; /* length of payload (incl padding) */ + guint info_payload_offset; /* start of payload relative to info_data */ + guint info_payload_len; /* length of payload (incl padding) */ gboolean info_is_srtp; guint32 info_setup_frame_num; /* the frame num of the packet that set this RTP connection */ const guint8* info_data; /* pointer to raw rtp data */ diff --git a/ui/gtk/rtp_analysis.c b/ui/gtk/rtp_analysis.c index 0089624909..3decd2e41f 100644 --- a/ui/gtk/rtp_analysis.c +++ b/ui/gtk/rtp_analysis.c @@ -783,7 +783,7 @@ rtp_packet_save_payload(tap_rtp_save_info_t *saveinfo, saveinfo->error_type = TAP_RTP_FILE_WRITE_ERROR; return 0; } - saveinfo->count += ((int)rtpinfo->info_payload_len - rtpinfo->info_padding_count); + saveinfo->count += rtpinfo->info_payload_len - rtpinfo->info_padding_count; fflush(saveinfo->fp); saveinfo->saved = TRUE; diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp index fee1ebc391..34ba58649d 100644 --- a/ui/qt/rtp_audio_stream.cpp +++ b/ui/qt/rtp_audio_stream.cpp @@ -139,7 +139,7 @@ void RtpAudioStream::addRtpPacket(const struct _packet_info *pinfo, const struct rtp_packet_t *rtp_packet = g_new0(rtp_packet_t, 1); rtp_packet->info = (struct _rtp_info *) g_memdup(rtp_info, sizeof(struct _rtp_info)); if (rtp_info->info_all_data_present && (rtp_info->info_payload_len != 0)) { - rtp_packet->payload_data = (guint8 *) g_memdup(&(rtp_info->info_data[rtp_info->info_payload_offset]), (guint) rtp_info->info_payload_len); + rtp_packet->payload_data = (guint8 *) g_memdup(&(rtp_info->info_data[rtp_info->info_payload_offset]), rtp_info->info_payload_len); } if (rtp_packets_.size() < 1) { // First packet |