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Add flag enabling more packets to be retransmittable.

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If not indicated otherwise, allow adding a packet to the retransmission
history at least every 1/15s in order to reduce frame dropping.

BUG=webrtc:7694

Review-Url: https://codereview.webrtc.org/2999063002
Cr-Commit-Position: refs/heads/master@{#19665}
This commit is contained in:
sprang
2017-09-04 07:23:56 -07:00
committed by Commit Bot
parent 791a8f611d
commit a8ae6f2aca
17 changed files with 460 additions and 172 deletions
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14
webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@ -31,6 +31,7 @@ namespace {
const int64_t kRtpRtcpMaxIdleTimeProcessMs = 5;
const int64_t kRtpRtcpRttProcessTimeMs = 1000;
const int64_t kRtpRtcpBitrateProcessTimeMs = 10;
const int64_t kDefaultExpectedRetransmissionTimeMs = 125;
} // namespace
RTPExtensionType StringToRtpExtensionType(const std::string& extension) {
@ -430,9 +431,20 @@ bool ModuleRtpRtcpImpl::SendOutgoingData(
if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) {
rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
}
int64_t expected_retransmission_time_ms = rtt_ms();
if (expected_retransmission_time_ms == 0) {
// No rtt available (|kRtpRtcpRttProcessTimeMs| not yet passed?), so try to
// poll avg_rtt_ms directly from rtcp receiver.
if (rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), nullptr,
&expected_retransmission_time_ms, nullptr,
nullptr) == -1) {
expected_retransmission_time_ms = kDefaultExpectedRetransmissionTimeMs;
}
}
return rtp_sender_->SendOutgoingData(
frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
payload_size, fragmentation, rtp_video_header, transport_frame_id_out);
payload_size, fragmentation, rtp_video_header, transport_frame_id_out,
expected_retransmission_time_ms);
}
bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,