diff options
Diffstat (limited to 'ui/qt/rtp_audio_stream.cpp')
| -rw-r--r-- | ui/qt/rtp_audio_stream.cpp | 20 |
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) |