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Fix the ramp-up-down-up test which was using ts-offset extension with the abs-send-time estimator.

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BUG=chromium:444023
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/34579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7968 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org 2014-12-19 18:00:21 +00:00
parent 50f7db8a77
commit 8817256373
1 changed files with 4 additions and 4 deletions
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8
webrtc/video/rampup_tests.cc
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@ -212,12 +212,12 @@ LowRateStreamObserver::LowRateStreamObserver(
rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
rtp_rtcp_->SetREMBStatus(true);
rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound);
rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId);
rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
kAbsSendTimeExtensionId);
AbsoluteSendTimeRemoteBitrateEstimatorFactory rbe_factory;
const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 10000;
remote_bitrate_estimator_.reset(
rbe_factory.Create(this, clock, kMimdControl,
rbe_factory.Create(this, clock, kAimdControl,
kRemoteBitrateEstimatorMinBitrateBps));
forward_transport_config_.link_capacity_kbps =
kHighBandwidthLimitBps / 1000;
@ -461,7 +461,7 @@ void RampUpTest::RunRampUpDownUpTest(size_t number_of_streams, bool rtx) {
send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
send_config_.rtp.extensions.push_back(RtpExtension(
RtpExtension::kTOffset, kTransmissionTimeOffsetExtensionId));
RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
send_config_.suspend_below_min_bitrate = true;
if (rtx) {
send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;