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Moving the pacer and the pacer thread to ChannelGroup.

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This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out.

BUG=4323
R=mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45549004

Cr-Commit-Position: refs/heads/master@{#8864}
This commit is contained in:
Stefan Holmer
2015-03-26 11:11:06 +01:00
parent 5225dd8180
commit e590416722
47 changed files with 928 additions and 687 deletions
Download Patch File Download Diff File Expand all files Collapse all files

143
webrtc/video_engine/payload_router_unittest.cc
View File

@ -172,147 +172,6 @@ TEST_F(PayloadRouterTest, MaxPayloadLength) {
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));
}
TEST_F(PayloadRouterTest, SetTargetSendBitrates) {
MockRtpRtcp rtp_1;
MockRtpRtcp rtp_2;
@ -346,5 +205,5 @@ TEST_F(PayloadRouterTest, SetTargetSendBitrates) {
EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
.Times(1);
payload_router_->SetTargetSendBitrates(bitrates);
}
}
} // namespace webrtc