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Reformatted rtp_sender: made lint clean.

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TESTED=rtp_rtcp_unittests
BUG=

Review URL: https://webrtc-codereview.appspot.com/1062004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3412 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
phoglund@webrtc.org
2013-01-25 10:53:38 +00:00
parent 3e47a0a611
commit 43da54a458
4 changed files with 652 additions and 752 deletions
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18
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@@ -123,11 +123,10 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
simulcast_(false),
key_frame_req_method_(kKeyFrameReqFirRtp),
remote_bitrate_(configuration.remote_bitrate_estimator),
rtt_observer_(configuration.rtt_observer)
#ifdef MATLAB
, plot1_(NULL)
, plot1_(NULL),
#endif
{
rtt_observer_(configuration.rtt_observer) {
RTPReceiverStrategy* rtp_receiver_strategy;
if (configuration.audio) {
// If audio, we need to be able to handle telephone events too, so stash
@@ -1645,7 +1644,7 @@ bool ModuleRtpRtcpImpl::SendTelephoneEventActive(
id_,
"SendTelephoneEventActive()");
return rtp_sender_.SendTelephoneEventActive(telephone_event);
return rtp_sender_.SendTelephoneEventActive(&telephone_event);
}
// Set audio packet size, used to determine when it's time to send a DTMF
@@ -1689,7 +1688,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::GetRTPAudioLevelIndicationStatus(
kTraceRtpRtcp,
id_,
"GetRTPAudioLevelIndicationStatus()");
return rtp_sender_.AudioLevelIndicationStatus(enable, id);
return rtp_sender_.AudioLevelIndicationStatus(&enable, &id);
}
WebRtc_Word32 ModuleRtpRtcpImpl::SetAudioLevel(
@@ -1719,7 +1718,7 @@ WebRtc_Word32 ModuleRtpRtcpImpl::SendREDPayloadType(
WebRtc_Word8& payload_type) const {
WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id_, "SendREDPayloadType()");
return rtp_sender_.RED(payload_type);
return rtp_sender_.RED(&payload_type);
}
RtpVideoCodecTypes ModuleRtpRtcpImpl::ReceivedVideoCodec() const {
@@ -1882,9 +1881,9 @@ WebRtc_Word32 ModuleRtpRtcpImpl::GenericFECStatus(
it++;
}
}
WebRtc_Word32 ret_val = rtp_sender_.GenericFECStatus(enable,
payload_type_red,
payload_type_fec);
WebRtc_Word32 ret_val = rtp_sender_.GenericFECStatus(&enable,
&payload_type_red,
&payload_type_fec);
if (child_enabled) {
// Returns true if enabled for any child module.
enable = child_enabled;
@@ -2083,5 +2082,4 @@ int64_t ModuleRtpRtcpImpl::RtcpReportInterval() {
else
return RTCP_INTERVAL_VIDEO_MS;
}
} // Namespace webrtc