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Add API for transmission smotthening.

Browse Source 
BUG=818
TEST=Only API tests added now.

Review URL: https://webrtc-codereview.appspot.com/787009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2756 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mflodman@webrtc.org 2012-09-12 13:47:06 +00:00
parent 633a6fa815
commit 5a7507f26a
6 changed files with 53 additions and 1 deletions
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8
src/video_engine/include/vie_rtp_rtcp.h
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@ -219,6 +219,14 @@ class WEBRTC_DLLEXPORT ViERTP_RTCP {
bool enable,
int id) = 0;
// 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.
// NOTE: This is still experimental functionality.
// TODO(mflodman) Remove this note when BUG=818 is closed.
virtual int SetTransmissionSmoothingStatus(int video_channel,
bool enable) = 0;
// This function returns our locally created statistics of the received RTP
// stream.
virtual int GetReceivedRTCPStatistics(

13
src/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
View File

@ -648,7 +648,18 @@ void ViEAutoTest::ViERtpRtcpAPITest()
EXPECT_EQ(0, ViE.rtp_rtcp->SetReceiveTimestampOffsetStatus(
tbChannel.videoChannel, false, 3));
// Transmission smoothening.
const int invalid_channel_id = 17;
EXPECT_EQ(-1, ViE.rtp_rtcp->SetTransmissionSmoothingStatus(
invalid_channel_id, true));
EXPECT_EQ(0, ViE.rtp_rtcp->SetTransmissionSmoothingStatus(
tbChannel.videoChannel, true));
EXPECT_EQ(0, ViE.rtp_rtcp->SetTransmissionSmoothingStatus(
tbChannel.videoChannel, true));
EXPECT_EQ(0, ViE.rtp_rtcp->SetTransmissionSmoothingStatus(
tbChannel.videoChannel, false));
EXPECT_EQ(0, ViE.rtp_rtcp->SetTransmissionSmoothingStatus(
tbChannel.videoChannel, false));
//***************************************************************
// Testing finished. Tear down Video Engine

12
src/video_engine/vie_channel.cc
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@ -230,6 +230,7 @@ WebRtc_Word32 ViEChannel::SetSendCodec(const VideoCodec& video_codec,
rtp_rtcp_->SetSendingStatus(false);
}
NACKMethod nack_method = rtp_rtcp_->NACK();
bool transmission_smoothening = rtp_rtcp_->TransmissionSmoothingStatus();
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
@ -290,6 +291,7 @@ WebRtc_Word32 ViEChannel::SetSendCodec(const VideoCodec& video_codec,
if (mtu_ != 0) {
rtp_rtcp->SetMaxTransferUnit(mtu_);
}
rtp_rtcp->SetTransmissionSmoothingStatus(transmission_smoothening);
if (restart_rtp) {
rtp_rtcp->SetSendingStatus(true);
}
@ -685,6 +687,7 @@ bool ViEChannel::EnableRemb(bool enable) {
}
int ViEChannel::SetSendTimestampOffsetStatus(bool enable, int id) {
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
int error = 0;
if (enable) {
// Enable the extension, but disable possible old id to avoid errors.
@ -724,6 +727,15 @@ int ViEChannel::SetReceiveTimestampOffsetStatus(bool enable, int id) {
}
}
void ViEChannel::SetTransmissionSmoothingStatus(bool enable) {
CriticalSectionScoped cs(rtp_rtcp_cs_.get());
rtp_rtcp_->SetTransmissionSmoothingStatus(enable);
for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
it != simulcast_rtp_rtcp_.end(); ++it) {
(*it)->SetTransmissionSmoothingStatus(enable);
}
}
WebRtc_Word32 ViEChannel::EnableTMMBR(const bool enable) {
WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
"%s: %d", __FUNCTION__, enable);

1
src/video_engine/vie_channel.h
View File

@ -109,6 +109,7 @@ class ViEChannel
bool EnableRemb(bool enable);
int SetSendTimestampOffsetStatus(bool enable, int id);
int SetReceiveTimestampOffsetStatus(bool enable, int id);
void SetTransmissionSmoothingStatus(bool enable);
WebRtc_Word32 EnableTMMBR(const bool enable);
WebRtc_Word32 EnableKeyFrameRequestCallback(const bool enable);

19
src/video_engine/vie_rtp_rtcp_impl.cc
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@ -660,6 +660,25 @@ int ViERTP_RTCPImpl::SetReceiveTimestampOffsetStatus(int video_channel,
return 0;
}
int ViERTP_RTCPImpl::SetTransmissionSmoothingStatus(int video_channel,
bool enable) {
WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
ViEId(shared_data_->instance_id(), video_channel),
"%s(channel: %d, enble: %d)", __FUNCTION__, 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->SetTransmissionSmoothingStatus(enable);
return 0;
}
int ViERTP_RTCPImpl::GetReceivedRTCPStatistics(const int video_channel,
uint16_t& fraction_lost,
unsigned int& cumulative_lost,

1
src/video_engine/vie_rtp_rtcp_impl.h
View File

@ -74,6 +74,7 @@ class ViERTP_RTCPImpl
virtual int SetReceiveTimestampOffsetStatus(int video_channel,
bool enable,
int id);
virtual int SetTransmissionSmoothingStatus(int video_channel, bool enable);
virtual int GetReceivedRTCPStatistics(const int video_channel,
uint16_t& fraction_lost,
unsigned int& cumulative_lost,