diff options
| author | Jiri Novak <j.novak@netsystem.cz> | 2018-06-14 23:19:01 +0200 |
|---|---|---|
| committer | Anders Broman <a.broman58@gmail.com> | 2018-06-19 15:05:12 +0000 |
| commit | 27a1906c582b9c7dcd17e7db2726a1059e3aaf8a (patch) | |
| tree | 44b713c72d6c5cac6ca9f319cc8c8511d8597834 /ui/qt/rtp_audio_stream.cpp | |
| parent | 434fbe20dd5e2dc58473fea1363a7350486bc33c (diff) | |
RTP: Code clean up
Changes:
- rtpstream_packet renamed to rtpstream_packet_cb to follow *_cb pattern
- variables/types used in iax2_analysis_dialog were created as copy of *rtp* ones, but names were left as *rtp* -> *iax2*
- struct _rtp_stream_info replaced with rtp_stream_info_t
- there was tap-rtp-analysis.h, but no tap-rtp-analysis.c - related content was moved from tap-rtp-common.c
- *rtp_stream* functions renamed to *rtpstream*
- renamed rtp_stream_info_t to rtpstream_info_t to follow *rtpstream* pattern.
- renamed ui/rtp_stream.c rtpstream_draw -> rtpstream_draw_cb
Change-Id: Ib11ff5367cc464ea1b0c73432bc50b0eb9cd203e
Reviewed-on: https://code.wireshark.org/review/28299
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.cpp | 8 |
1 files changed, 4 insertions, 4 deletions
diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp index 6c6ec385b2..2f3f0830af 100644 --- a/ui/qt/rtp_audio_stream.cpp +++ b/ui/qt/rtp_audio_stream.cpp @@ -38,7 +38,7 @@ static spx_int16_t default_audio_sample_rate_ = 8000; static const spx_int16_t visual_sample_rate_ = 1000; -RtpAudioStream::RtpAudioStream(QObject *parent, _rtp_stream_info *rtp_stream) : +RtpAudioStream::RtpAudioStream(QObject *parent, rtpstream_info_t *rtp_stream) : QObject(parent), decoders_hash_(rtp_decoder_hash_table_new()), global_start_rel_time_(0.0), @@ -84,7 +84,7 @@ RtpAudioStream::~RtpAudioStream() speex_resampler_destroy (visual_resampler_); } -bool RtpAudioStream::isMatch(const _rtp_stream_info *rtp_stream) const +bool RtpAudioStream::isMatch(const rtpstream_info_t *rtp_stream) const { if (rtp_stream && addresses_equal(&rtp_stream->src_addr, &src_addr_) @@ -111,7 +111,7 @@ 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 _rtp_stream_info *rtp_stream) +void RtpAudioStream::addRtpStream(const rtpstream_info_t *rtp_stream) { if (!rtp_stream) return; @@ -296,7 +296,7 @@ void RtpAudioStream::decode() silence_timestamps_.append(stop_rel_time_); decoded_bytes_prev = 0; -/* defined start_timestmp to avoid overflow in timestamp. TODO: handle the timestamp correctly */ +/* defined start_timestamp to avoid overflow in timestamp. TODO: handle the timestamp correctly */ /* 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; |