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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
Download Patch File Download Diff File Expand all files Collapse all files

6
webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
View File

@ -14,6 +14,7 @@
#include <memory>
#include <vector>
#include "webrtc/base/rate_limiter.h"
#include "webrtc/test/null_transport.h"
namespace webrtc {
@ -80,7 +81,8 @@ int32_t TestRtpReceiver::OnReceivedPayloadData(
class RtpRtcpAPITest : public ::testing::Test {
protected:
RtpRtcpAPITest() : fake_clock_(123456) {
RtpRtcpAPITest()
: fake_clock_(123456), retransmission_rate_limiter_(&fake_clock_, 1000) {
test_csrcs_.push_back(1234);
test_csrcs_.push_back(2345);
test_ssrc_ = 3456;
@ -94,6 +96,7 @@ class RtpRtcpAPITest : public ::testing::Test {
configuration.audio = true;
configuration.clock = &fake_clock_;
configuration.outgoing_transport = &null_transport_;
configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
module_.reset(RtpRtcp::CreateRtpRtcp(configuration));
rtp_payload_registry_.reset(new RTPPayloadRegistry(
RTPPayloadStrategy::CreateStrategy(true)));
@ -110,6 +113,7 @@ class RtpRtcpAPITest : public ::testing::Test {
std::vector<uint32_t> test_csrcs_;
SimulatedClock fake_clock_;
test::NullTransport null_transport_;
RateLimiter retransmission_rate_limiter_;
};
TEST_F(RtpRtcpAPITest, Basic) {