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Remove const in vie_rtp_rtcp, where there is conflict with

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mock defines in fakewebrtcvideoengine.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4399004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5176 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
sprang@webrtc.org 2013-11-26 09:30:04 +00:00
parent d89b52af80
commit 7f959980f8
1 changed files with 12 additions and 12 deletions
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24
webrtc/video_engine/include/vie_rtp_rtcp.h
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@ -314,16 +314,16 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
virtual int RegisterSendChannelRtcpStatisticsCallback(
const int video_channel, RtcpStatisticsCallback* callback) = 0;
int video_channel, RtcpStatisticsCallback* callback) = 0;
virtual int DeregisterSendChannelRtcpStatisticsCallback(
const int video_channel, RtcpStatisticsCallback* callback) = 0;
int video_channel, RtcpStatisticsCallback* callback) = 0;
virtual int RegisterReceiveChannelRtcpStatisticsCallback(
const int video_channel, RtcpStatisticsCallback* callback) = 0;
int video_channel, RtcpStatisticsCallback* callback) = 0;
virtual int DeregisterReceiveChannelRtcpStatisticsCallback(
const int video_channel, RtcpStatisticsCallback* callback) = 0;
int video_channel, RtcpStatisticsCallback* callback) = 0;
// The function gets statistics from the sent and received RTP streams.
virtual int GetRtpStatistics(const int video_channel,
@ -348,16 +348,16 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
}
virtual int RegisterSendChannelRtpStatisticsCallback(
const int video_channel, StreamDataCountersCallback* callback) = 0;
int video_channel, StreamDataCountersCallback* callback) = 0;
virtual int DeregisterSendChannelRtpStatisticsCallback(
const int video_channel, StreamDataCountersCallback* callback) = 0;
int video_channel, StreamDataCountersCallback* callback) = 0;
virtual int RegisterReceiveChannelRtpStatisticsCallback(
const int video_channel, StreamDataCountersCallback* callback) = 0;
int video_channel, StreamDataCountersCallback* callback) = 0;
virtual int DeregisterReceiveChannelRtpStatisticsCallback(
const int video_channel, StreamDataCountersCallback* callback) = 0;
int video_channel, StreamDataCountersCallback* callback) = 0;
// The function gets bandwidth usage statistics from the sent RTP streams in
// bits/s.
@ -369,11 +369,11 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
// (De)Register an observer, called whenever the send bitrate is updated
virtual int RegisterSendBitrateObserver(
const int video_channel,
int video_channel,
BitrateStatisticsObserver* observer) = 0;
virtual int DeregisterSendBitrateObserver(
const int video_channel,
int video_channel,
BitrateStatisticsObserver* observer) = 0;
// This function gets the send-side estimated bandwidth available for video,
@ -418,11 +418,11 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
// Registers and instance of a user implementation of ViEFrameCountObserver
virtual int RegisterSendFrameCountObserver(
const int video_channel, FrameCountObserver* observer) = 0;
int video_channel, FrameCountObserver* observer) = 0;
// Removes a registered instance of a ViEFrameCountObserver
virtual int DeregisterSendFrameCountObserver(
const int video_channel, FrameCountObserver* observer) = 0;
int video_channel, FrameCountObserver* observer) = 0;
protected:
virtual ~ViERTP_RTCP() {}