Files
platform-external-webrtc/webrtc/video_engine/payload_router_unittest.cc
mflodman@webrtc.org 290cb56dca Remove TimeToSendPacket and TimeToSendPadding from the default module.
Thie CL moves the default RTP module logic for TimeToSendPacket and
TimeToSendPadding to PayloadRouter class and asserts on usage of the
default module.

BUG=769
TEST=New unittest.
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/33319004

Cr-Commit-Position: refs/heads/master@{#8383}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8383 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-17 10:15:47 +00:00

306 lines
10 KiB
C++

/*
* Copyright (c) 2015 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 <list>
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/video_engine/payload_router.h"
using ::testing::_;
using ::testing::AnyNumber;
using ::testing::NiceMock;
using ::testing::Return;
namespace webrtc {
class PayloadRouterTest : public ::testing::Test {
protected:
virtual void SetUp() {
payload_router_.reset(new PayloadRouter());
}
scoped_ptr<PayloadRouter> payload_router_;
};
TEST_F(PayloadRouterTest, SendOnOneModule) {
MockRtpRtcp rtp;
std::list<RtpRtcp*> modules(1, &rtp);
payload_router_->SetSendingRtpModules(modules);
uint8_t payload = 'a';
FrameType frame_type = kVideoFrameKey;
int8_t payload_type = 96;
EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
NULL))
.Times(0);
EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
&payload, 1, NULL, NULL));
payload_router_->set_active(true);
EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
NULL))
.Times(1);
EXPECT_TRUE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
&payload, 1, NULL, NULL));
payload_router_->set_active(false);
EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
NULL))
.Times(0);
EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
&payload, 1, NULL, NULL));
payload_router_->set_active(true);
EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
NULL))
.Times(1);
EXPECT_TRUE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
&payload, 1, NULL, NULL));
modules.clear();
payload_router_->SetSendingRtpModules(modules);
EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
NULL))
.Times(0);
EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
&payload, 1, NULL, NULL));
}
TEST_F(PayloadRouterTest, SendSimulcast) {
MockRtpRtcp rtp_1;
MockRtpRtcp rtp_2;
std::list<RtpRtcp*> modules;
modules.push_back(&rtp_1);
modules.push_back(&rtp_2);
payload_router_->SetSendingRtpModules(modules);
uint8_t payload_1 = 'a';
FrameType frame_type_1 = kVideoFrameKey;
int8_t payload_type_1 = 96;
RTPVideoHeader rtp_hdr_1;
rtp_hdr_1.simulcastIdx = 0;
payload_router_->set_active(true);
EXPECT_CALL(rtp_1, SendOutgoingData(frame_type_1, payload_type_1, 0, 0, _, 1,
NULL, &rtp_hdr_1))
.Times(1);
EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
.Times(0);
EXPECT_TRUE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0,
&payload_1, 1, NULL, &rtp_hdr_1));
uint8_t payload_2 = 'b';
FrameType frame_type_2 = kVideoFrameDelta;
int8_t payload_type_2 = 97;
RTPVideoHeader rtp_hdr_2;
rtp_hdr_2.simulcastIdx = 1;
EXPECT_CALL(rtp_2, SendOutgoingData(frame_type_2, payload_type_2, 0, 0, _, 1,
NULL, &rtp_hdr_2))
.Times(1);
EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
.Times(0);
EXPECT_TRUE(payload_router_->RoutePayload(frame_type_2, payload_type_2, 0, 0,
&payload_2, 1, NULL, &rtp_hdr_2));
payload_router_->set_active(false);
EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
.Times(0);
EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
.Times(0);
EXPECT_FALSE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0,
&payload_1, 1, NULL, &rtp_hdr_1));
EXPECT_FALSE(payload_router_->RoutePayload(frame_type_2, payload_type_2, 0, 0,
&payload_2, 1, NULL, &rtp_hdr_2));
}
TEST_F(PayloadRouterTest, MaxPayloadLength) {
// Without any limitations from the modules, verify we get the max payload
// length for IP/UDP/SRTP with a MTU of 150 bytes.
const size_t kDefaultMaxLength = 1500 - 20 - 8 - 12 - 4;
EXPECT_EQ(kDefaultMaxLength, payload_router_->DefaultMaxPayloadLength());
EXPECT_EQ(kDefaultMaxLength, payload_router_->MaxPayloadLength());
MockRtpRtcp rtp_1;
MockRtpRtcp rtp_2;
std::list<RtpRtcp*> modules;
modules.push_back(&rtp_1);
modules.push_back(&rtp_2);
payload_router_->SetSendingRtpModules(modules);
// Modules return a higher length than the default value.
EXPECT_CALL(rtp_1, MaxDataPayloadLength())
.Times(1)
.WillOnce(Return(kDefaultMaxLength + 10));
EXPECT_CALL(rtp_2, MaxDataPayloadLength())
.Times(1)
.WillOnce(Return(kDefaultMaxLength + 10));
EXPECT_EQ(kDefaultMaxLength, payload_router_->MaxPayloadLength());
// The modules return a value lower than default.
const size_t kTestMinPayloadLength = 1001;
EXPECT_CALL(rtp_1, MaxDataPayloadLength())
.Times(1)
.WillOnce(Return(kTestMinPayloadLength + 10));
EXPECT_CALL(rtp_2, MaxDataPayloadLength())
.Times(1)
.WillOnce(Return(kTestMinPayloadLength));
EXPECT_EQ(kTestMinPayloadLength, payload_router_->MaxPayloadLength());
}
TEST_F(PayloadRouterTest, TimeToSendPacket) {
MockRtpRtcp rtp_1;
MockRtpRtcp rtp_2;
std::list<RtpRtcp*> modules;
modules.push_back(&rtp_1);
modules.push_back(&rtp_2);
payload_router_->SetSendingRtpModules(modules);
const uint16_t kSsrc1 = 1234;
uint16_t sequence_number = 17;
uint64_t timestamp = 7890;
bool retransmission = false;
// Send on the first module by letting rtp_1 be sending with correct ssrc.
EXPECT_CALL(rtp_1, SendingMedia())
.Times(1)
.WillOnce(Return(true));
EXPECT_CALL(rtp_1, SSRC())
.Times(1)
.WillOnce(Return(kSsrc1));
EXPECT_CALL(rtp_1, TimeToSendPacket(kSsrc1, sequence_number, timestamp,
retransmission))
.Times(1)
.WillOnce(Return(true));
EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _))
.Times(0);
EXPECT_TRUE(payload_router_->TimeToSendPacket(
kSsrc1, sequence_number, timestamp, retransmission));
// Send on the second module by letting rtp_2 be sending, but not rtp_1.
++sequence_number;
timestamp += 30;
retransmission = true;
const uint16_t kSsrc2 = 4567;
EXPECT_CALL(rtp_1, SendingMedia())
.Times(1)
.WillOnce(Return(false));
EXPECT_CALL(rtp_2, SendingMedia())
.Times(1)
.WillOnce(Return(true));
EXPECT_CALL(rtp_2, SSRC())
.Times(1)
.WillOnce(Return(kSsrc2));
EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _, _))
.Times(0);
EXPECT_CALL(rtp_2, TimeToSendPacket(kSsrc2, sequence_number, timestamp,
retransmission))
.Times(1)
.WillOnce(Return(true));
EXPECT_TRUE(payload_router_->TimeToSendPacket(
kSsrc2, sequence_number, timestamp, retransmission));
// No module is sending, hence no packet should be sent.
EXPECT_CALL(rtp_1, SendingMedia())
.Times(1)
.WillOnce(Return(false));
EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _,_))
.Times(0);
EXPECT_CALL(rtp_2, SendingMedia())
.Times(1)
.WillOnce(Return(false));
EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _))
.Times(0);
EXPECT_TRUE(payload_router_->TimeToSendPacket(
kSsrc1, sequence_number, timestamp, retransmission));
// Add a packet with incorrect ssrc and test it's dropped in the router.
EXPECT_CALL(rtp_1, SendingMedia())
.Times(1)
.WillOnce(Return(true));
EXPECT_CALL(rtp_1, SSRC())
.Times(1)
.WillOnce(Return(kSsrc1));
EXPECT_CALL(rtp_2, SendingMedia())
.Times(1)
.WillOnce(Return(true));
EXPECT_CALL(rtp_2, SSRC())
.Times(1)
.WillOnce(Return(kSsrc2));
EXPECT_CALL(rtp_1, TimeToSendPacket(_, _, _,_))
.Times(0);
EXPECT_CALL(rtp_2, TimeToSendPacket(_, _, _, _))
.Times(0);
EXPECT_TRUE(payload_router_->TimeToSendPacket(
kSsrc1 + kSsrc2, sequence_number, timestamp, retransmission));
}
TEST_F(PayloadRouterTest, TimeToSendPadding) {
MockRtpRtcp rtp_1;
MockRtpRtcp rtp_2;
std::list<RtpRtcp*> modules;
modules.push_back(&rtp_1);
modules.push_back(&rtp_2);
payload_router_->SetSendingRtpModules(modules);
// Default configuration, sending padding on the first sending module.
const size_t requested_padding_bytes = 1000;
const size_t sent_padding_bytes = 890;
EXPECT_CALL(rtp_1, SendingMedia())
.Times(1)
.WillOnce(Return(true));
EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes))
.Times(1)
.WillOnce(Return(sent_padding_bytes));
EXPECT_CALL(rtp_2, TimeToSendPadding(_))
.Times(0);
EXPECT_EQ(sent_padding_bytes,
payload_router_->TimeToSendPadding(requested_padding_bytes));
// Let only the second module be sending and verify the padding request is
// routed there.
EXPECT_CALL(rtp_1, SendingMedia())
.Times(1)
.WillOnce(Return(false));
EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes))
.Times(0);
EXPECT_CALL(rtp_2, SendingMedia())
.Times(1)
.WillOnce(Return(true));
EXPECT_CALL(rtp_2, TimeToSendPadding(_))
.Times(1)
.WillOnce(Return(sent_padding_bytes));
EXPECT_EQ(sent_padding_bytes,
payload_router_->TimeToSendPadding(requested_padding_bytes));
// No sending module at all.
EXPECT_CALL(rtp_1, SendingMedia())
.Times(1)
.WillOnce(Return(false));
EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes))
.Times(0);
EXPECT_CALL(rtp_2, SendingMedia())
.Times(1)
.WillOnce(Return(false));
EXPECT_CALL(rtp_2, TimeToSendPadding(_))
.Times(0);
EXPECT_EQ(static_cast<size_t>(0),
payload_router_->TimeToSendPadding(requested_padding_bytes));
}
} // namespace webrtc