Files
platform-external-webrtc/webrtc/voice_engine/network_predictor.cc
minyue@webrtc.org 74aaf29a0f Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module.
The filter is an exponential filter borrowed from video coding module.

The method is written in a new class called PacketLossProtector (not sure if the name is nice), which can be used in the future for more sophisticated logic.

BUG=
R=henrika@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6709 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-16 21:28:26 +00:00

38 lines
1.2 KiB
C++

/*
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/voice_engine/network_predictor.h"
namespace webrtc {
namespace voe {
NetworkPredictor::NetworkPredictor(Clock* clock)
: clock_(clock),
last_loss_rate_update_time_ms_(clock_->TimeInMilliseconds()),
loss_rate_filter_(new rtc::ExpFilter(0.9999f)) {
}
void NetworkPredictor::UpdatePacketLossRate(uint8_t loss_rate) {
int64_t now_ms = clock_->TimeInMilliseconds();
// Update the recursive average filter.
loss_rate_filter_->Apply(
static_cast<float>(now_ms - last_loss_rate_update_time_ms_),
static_cast<float>(loss_rate));
last_loss_rate_update_time_ms_ = now_ms;
}
uint8_t NetworkPredictor::GetLossRate() {
float value = loss_rate_filter_->filtered();
return (value == rtc::ExpFilter::kValueUndefined) ? 0 :
static_cast<uint8_t>(value + 0.5);
}
} // namespace voe
} // namespace webrtc