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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) {

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

@ -15,6 +15,7 @@
#include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
#include "webrtc/base/rate_limiter.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@ -77,7 +78,8 @@ class RTPCallback : public NullRtpFeedback {
class RtpRtcpAudioTest : public ::testing::Test {
protected:
RtpRtcpAudioTest() : fake_clock(123456) {
RtpRtcpAudioTest()
: fake_clock(123456), retransmission_rate_limiter_(&fake_clock, 1000) {
test_CSRC[0] = 1234;
test_CSRC[2] = 2345;
test_ssrc = 3456;
@ -106,6 +108,7 @@ class RtpRtcpAudioTest : public ::testing::Test {
configuration.clock = &fake_clock;
configuration.receive_statistics = receive_statistics1_.get();
configuration.outgoing_transport = transport1;
configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
module1 = RtpRtcp::CreateRtpRtcp(configuration);
rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
@ -152,6 +155,7 @@ class RtpRtcpAudioTest : public ::testing::Test {
uint16_t test_sequence_number;
uint32_t test_CSRC[webrtc::kRtpCsrcSize];
SimulatedClock fake_clock;
RateLimiter retransmission_rate_limiter_;
};
TEST_F(RtpRtcpAudioTest, Basic) {

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

@ -14,6 +14,7 @@
#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/include/receive_statistics.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
@ -66,7 +67,8 @@ class TestRtpFeedback : public NullRtpFeedback {
class RtpRtcpRtcpTest : public ::testing::Test {
protected:
RtpRtcpRtcpTest() : fake_clock(123456) {
RtpRtcpRtcpTest()
: fake_clock(123456), retransmission_rate_limiter_(&fake_clock, 1000) {
test_csrcs.push_back(1234);
test_csrcs.push_back(2345);
test_ssrc = 3456;
@ -91,6 +93,7 @@ class RtpRtcpRtcpTest : public ::testing::Test {
configuration.receive_statistics = receive_statistics1_.get();
configuration.outgoing_transport = transport1;
configuration.intra_frame_callback = myRTCPFeedback1;
configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_payload_registry1_.reset(new RTPPayloadRegistry(
RTPPayloadStrategy::CreateStrategy(true)));
@ -197,6 +200,7 @@ class RtpRtcpRtcpTest : public ::testing::Test {
uint16_t test_sequence_number;
std::vector<uint32_t> test_csrcs;
SimulatedClock fake_clock;
RateLimiter retransmission_rate_limiter_;
};
TEST_F(RtpRtcpRtcpTest, RTCP_PLI_RPSI) {

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

@ -15,6 +15,7 @@
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/rate_limiter.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
@ -38,7 +39,8 @@ class RtpRtcpVideoTest : public ::testing::Test {
test_ssrc_(3456),
test_timestamp_(4567),
test_sequence_number_(2345),
fake_clock(123456) {}
fake_clock(123456),
retransmission_rate_limiter_(&fake_clock, 1000) {}
~RtpRtcpVideoTest() {}
virtual void SetUp() {
@ -49,6 +51,7 @@ class RtpRtcpVideoTest : public ::testing::Test {
configuration.audio = false;
configuration.clock = &fake_clock;
configuration.outgoing_transport = transport_;
configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
video_module_ = RtpRtcp::CreateRtpRtcp(configuration);
rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
@ -139,6 +142,7 @@ class RtpRtcpVideoTest : public ::testing::Test {
uint8_t video_frame_[65000];
size_t payload_data_length_;
SimulatedClock fake_clock;
RateLimiter retransmission_rate_limiter_;
};
TEST_F(RtpRtcpVideoTest, BasicVideo) {