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Added API for enabling/disabling RTCP Receiver Reference Time extension.

Browse Source 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3419005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5147 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
asapersson@webrtc.org 2013-11-21 08:57:04 +00:00
parent 54a05518e2
commit 8d02f5dc71
12 changed files with 59 additions and 0 deletions
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2
webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
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@ -510,6 +510,8 @@ class RtpRtcp : public Module {
*/
virtual void SetRtcpXrRrtrStatus(bool enable) = 0;
virtual bool RtcpXrRrtrStatus() const = 0;
/*
* (REMB) Receiver Estimated Max Bitrate
*/

2
webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
View File

@ -170,6 +170,8 @@ class MockRtpRtcp : public RtpRtcp {
int32_t(const RTCPVoIPMetric* VoIPMetric));
MOCK_METHOD1(SetRtcpXrRrtrStatus,
void(bool enable));
MOCK_CONST_METHOD0(RtcpXrRrtrStatus,
bool());
MOCK_CONST_METHOD0(REMB,
bool());
MOCK_METHOD1(SetREMBStatus,

5
webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
View File

@ -2182,6 +2182,11 @@ void RTCPSender::SendRtcpXrReceiverReferenceTime(bool enable) {
xrSendReceiverReferenceTimeEnabled_ = enable;
}
bool RTCPSender::RtcpXrReceiverReferenceTime() const {
CriticalSectionScoped lock(_criticalSectionRTCPSender);
return xrSendReceiverReferenceTimeEnabled_;
}
// called under critsect _criticalSectionRTCPSender
int32_t RTCPSender::WriteAllReportBlocksToBuffer(
uint8_t* rtcpbuffer,

2
webrtc/modules/rtp_rtcp/source/rtcp_sender.h
View File

@ -171,6 +171,8 @@ public:
void SendRtcpXrReceiverReferenceTime(bool enable);
bool RtcpXrReceiverReferenceTime() const;
int32_t SetCSRCs(const uint32_t arrOfCSRC[kRtpCsrcSize],
const uint8_t arrLength);

4
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -963,6 +963,10 @@ void ModuleRtpRtcpImpl::SetRtcpXrRrtrStatus(bool enable) {
return rtcp_sender_.SendRtcpXrReceiverReferenceTime(enable);
}
bool ModuleRtpRtcpImpl::RtcpXrRrtrStatus() const {
return rtcp_sender_.RtcpXrReceiverReferenceTime();
}
int32_t ModuleRtpRtcpImpl::DataCountersRTP(
uint32_t* bytes_sent,
uint32_t* packets_sent) const {

2
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
View File

@ -259,6 +259,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp {
// (XR) Receiver reference time report.
virtual void SetRtcpXrRrtrStatus(bool enable) OVERRIDE;
virtual bool RtcpXrRrtrStatus() const OVERRIDE;
// Audio part.
// Set audio packet size, used to determine when it's time to send a DTMF

6
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
View File

@ -124,6 +124,12 @@ TEST_F(RtpRtcpImplTest, Rtt) {
rtp_rtcp_impl_->RTT(kSsrc + 1, &rtt, &avg_rtt, &min_rtt, &max_rtt));
}
TEST_F(RtpRtcpImplTest, SetRtcpXrRrtrStatus) {
EXPECT_FALSE(rtp_rtcp_impl_->RtcpXrRrtrStatus());
rtp_rtcp_impl_->SetRtcpXrRrtrStatus(true);
EXPECT_TRUE(rtp_rtcp_impl_->RtcpXrRrtrStatus());
}
TEST_F(RtpRtcpImplTest, RttForReceiverOnly) {
rtp_rtcp_impl_->SetRtcpXrRrtrStatus(true);
EXPECT_EQ(0, rtp_rtcp_impl_->SetSendingStatus(false));

5
webrtc/video_engine/include/vie_rtp_rtcp.h
View File

@ -254,6 +254,11 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
bool enable,
int id) = 0;
// Enables/disables RTCP Receiver Reference Time Report Block extension/
// DLRR Report Block extension (RFC 3611).
// TODO(asapersson): Remove default implementation.
virtual int SetRtcpXrRrtrStatus(int video_channel, bool enable) { return -1; }
// Enables transmission smoothening, i.e. packets belonging to the same frame
// will be sent over a longer period of time instead of sending them
// back-to-back.

10
webrtc/video_engine/vie_channel.cc
View File

@ -406,6 +406,7 @@ int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec,
rtp_rtcp->DeregisterSendRtpHeaderExtension(
kRtpExtensionAbsoluteSendTime);
}
rtp_rtcp->SetRtcpXrRrtrStatus(rtp_rtcp_->RtcpXrRrtrStatus());
}
// |RegisterSimulcastRtpRtcpModules| resets all old weak pointers and old
// modules can be deleted after this step.
@ -922,6 +923,15 @@ bool ViEChannel::GetReceiveAbsoluteSendTimeStatus() const {
return receive_absolute_send_time_enabled_;
}
void ViEChannel::SetRtcpXrRrtrStatus(bool enable) {
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
rtp_rtcp_->SetRtcpXrRrtrStatus(enable);
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
it != simulcast_rtp_rtcp_.end(); it++) {
(*it)->SetRtcpXrRrtrStatus(enable);
}
}
void ViEChannel::SetTransmissionSmoothingStatus(bool enable) {
assert(paced_sender_ && "No paced sender registered.");
paced_sender_->SetStatus(enable);

1
webrtc/video_engine/vie_channel.h
View File

@ -125,6 +125,7 @@ class ViEChannel
int SetSendAbsoluteSendTimeStatus(bool enable, int id);
int SetReceiveAbsoluteSendTimeStatus(bool enable, int id);
bool GetReceiveAbsoluteSendTimeStatus() const;
void SetRtcpXrRrtrStatus(bool enable);
void SetTransmissionSmoothingStatus(bool enable);
int32_t EnableTMMBR(const bool enable);
int32_t EnableKeyFrameRequestCallback(const bool enable);

19
webrtc/video_engine/vie_rtp_rtcp_impl.cc
View File

@ -804,6 +804,25 @@ int ViERTP_RTCPImpl::SetReceiveAbsoluteSendTimeStatus(int video_channel,
return 0;
}
int ViERTP_RTCPImpl::SetRtcpXrRrtrStatus(int video_channel, bool enable) {
WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
ViEId(shared_data_->instance_id(), video_channel),
"ViERTP_RTCPImpl::SetRtcpXrRrtrStatus(%d, %d)",
video_channel, enable);
ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
ViEChannel* vie_channel = cs.Channel(video_channel);
if (!vie_channel) {
WEBRTC_TRACE(kTraceError, kTraceVideo,
ViEId(shared_data_->instance_id(), video_channel),
"%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
return -1;
}
vie_channel->SetRtcpXrRrtrStatus(enable);
return 0;
}
int ViERTP_RTCPImpl::SetTransmissionSmoothingStatus(int video_channel,
bool enable) {
WEBRTC_TRACE(kTraceApiCall, kTraceVideo,

1
webrtc/video_engine/vie_rtp_rtcp_impl.h
View File

@ -88,6 +88,7 @@ class ViERTP_RTCPImpl
virtual int SetReceiveAbsoluteSendTimeStatus(int video_channel,
bool enable,
int id);
virtual int SetRtcpXrRrtrStatus(int video_channel, bool enable);
virtual int SetTransmissionSmoothingStatus(int video_channel, bool enable);
virtual int GetReceiveChannelRtcpStatistics(const int video_channel,
RtcpStatistics& basic_stats,