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authorJiri Novak <j.novak@netsystem.cz>2018-06-21 14:24:17 +0200
committerAnders Broman <a.broman58@gmail.com>2018-06-23 10:03:54 +0000
commit9f8c332c591cad546f41c10d28135ccaba53e206 (patch)
tree3e2d950d8cabea30bcc318d5ac2f3c535c6ac82d /ui/qt/rtp_audio_stream.cpp
parentdb4fb82555e9b2acafc56caeaef2888faea82c01 (diff)
RTP: code cleanup 3
*rtp_stream* -> rtpstream to follow common name Change-Id: I381bc1cdb8206c5cfe67e94dd7fb1a5cb25f9c16 Reviewed-on: https://code.wireshark.org/review/28394 Petri-Dish: Anders Broman <a.broman58@gmail.com> Tested-by: Petri Dish Buildbot Reviewed-by: Anders Broman <a.broman58@gmail.com>
Diffstat (limited to 'ui/qt/rtp_audio_stream.cpp')
-rw-r--r--ui/qt/rtp_audio_stream.cpp20
1 files changed, 10 insertions, 10 deletions
diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp
index a54e61aaef..5b422cc2da 100644
--- a/ui/qt/rtp_audio_stream.cpp
+++ b/ui/qt/rtp_audio_stream.cpp
@@ -33,12 +33,12 @@
#include <QVariant>
// To do:
-// - Only allow one rtp_stream_info_t per RtpAudioStream?
+// - Only allow one rtpstream_info_t per RtpAudioStream?
static spx_int16_t default_audio_sample_rate_ = 8000;
static const spx_int16_t visual_sample_rate_ = 1000;
-RtpAudioStream::RtpAudioStream(QObject *parent, rtpstream_info_t *rtp_stream) :
+RtpAudioStream::RtpAudioStream(QObject *parent, rtpstream_info_t *rtpstream) :
QObject(parent),
decoders_hash_(rtp_decoder_hash_table_new()),
global_start_rel_time_(0.0),
@@ -53,7 +53,7 @@ RtpAudioStream::RtpAudioStream(QObject *parent, rtpstream_info_t *rtp_stream) :
jitter_buffer_size_(50),
timing_mode_(RtpAudioStream::JitterBuffer)
{
- rtpstream_id_copy(&rtp_stream->id, &id_);
+ rtpstream_id_copy(&rtpstream->id, &id_);
// We keep visual samples in memory. Make fewer of them.
visual_resampler_ = speex_resampler_init(1, default_audio_sample_rate_,
@@ -81,10 +81,10 @@ RtpAudioStream::~RtpAudioStream()
rtpstream_id_free(&id_);
}
-bool RtpAudioStream::isMatch(const rtpstream_info_t *rtp_stream) const
+bool RtpAudioStream::isMatch(const rtpstream_info_t *rtpstream) const
{
- if (rtp_stream
- && rtpstream_id_equal(&id_, &(rtp_stream->id), RTPSTREAM_ID_EQUAL_SSRC))
+ if (rtpstream
+ && rtpstream_id_equal(&id_, &(rtpstream->id), RTPSTREAM_ID_EQUAL_SSRC))
return true;
return false;
}
@@ -100,12 +100,12 @@ bool RtpAudioStream::isMatch(const _packet_info *pinfo, const _rtp_info *rtp_inf
// XXX We add multiple RTP streams here because that's what the GTK+ UI does.
// Should we make these distinct, with their own waveforms? It seems like
// that would simplify a lot of things.
-void RtpAudioStream::addRtpStream(const rtpstream_info_t *rtp_stream)
+void RtpAudioStream::addRtpStream(const rtpstream_info_t *rtpstream)
{
- if (!rtp_stream) return;
+ if (!rtpstream) return;
- // RTP_STREAM_DEBUG("added %d:%u packets", g_list_length(rtp_stream->rtp_packet_list), rtp_stream->packet_count);
- rtp_streams_ << rtp_stream;
+ // RTP_STREAM_DEBUG("added %d:%u packets", g_list_length(rtpstream->rtp_packet_list), rtpstream->packet_count);
+ rtpstreams_ << rtpstream;
}
void RtpAudioStream::addRtpPacket(const struct _packet_info *pinfo, const struct _rtp_info *rtp_info)