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Add NACK rate throttling for audio channels.

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Not really used for audio today (already in place for video), but should
still function anyway.

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

Review URL: https://codereview.webrtc.org/2181383002 .

Cr-Commit-Position: refs/heads/master@{#13571}
This commit is contained in:
Erik Språng
2016-07-29 12:59:36 +02:00
parent f3882571b0
commit 737336d37a
18 changed files with 218 additions and 73 deletions
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6
webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
View File

@ -13,6 +13,7 @@
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/rate_limiter.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
@ -229,11 +230,13 @@ class RtcpSenderTest : public ::testing::Test {
protected:
RtcpSenderTest()
: clock_(1335900000),
receive_statistics_(ReceiveStatistics::Create(&clock_)) {
receive_statistics_(ReceiveStatistics::Create(&clock_)),
retransmission_rate_limiter_(&clock_, 1000) {
RtpRtcp::Configuration configuration;
configuration.audio = false;
configuration.clock = &clock_;
configuration.outgoing_transport = &test_transport_;
configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
@ -265,6 +268,7 @@ class RtcpSenderTest : public ::testing::Test {
std::unique_ptr<ReceiveStatistics> receive_statistics_;
std::unique_ptr<ModuleRtpRtcpImpl> rtp_rtcp_impl_;
std::unique_ptr<RTCPSender> rtcp_sender_;
RateLimiter retransmission_rate_limiter_;
};
TEST_F(RtcpSenderTest, SetRtcpStatus) {