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Remove the non-useful rtx payload padding option

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This CL removes the field trial left in place as a kill-switch in case
there were any regressions related to selecting payload padding based
on the likelihood of being useful instead of matching size.

It also removes the functionality that was only enabled with the
kill-switch active.

The feature has been default-on since June 23rd 2019:
https://webrtc.googlesource.com/src.git/+/214f54365ec210db76218a35ead66c9ce23e068e

Since we have not observed any issues, let's clean this code up.

Bug: webrtc:8975
Change-Id: I7f49fe354227b3f6566a250332e56b6d70fe2f09
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145821
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28616}
This commit is contained in:
Erik Språng
2019-07-16 21:09:14 +02:00
committed by Commit Bot
parent 76c89daac5
commit 21f2fc9c73
6 changed files with 2 additions and 307 deletions
Download Patch File Download Diff File Expand all files Collapse all files

105
modules/rtp_rtcp/source/rtp_sender_unittest.cc
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@ -1229,8 +1229,6 @@ TEST_P(RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator) {
EXPECT_EQ(1, transport_.packets_sent());
}
// TODO(bugs.webrtc.org/8975): Remove this test when non-useful padding is
// removed.
TEST_P(RtpSenderTest, SendRedundantPayloads) {
if (!GetParam().pacer_references_packets) {
// If PacedSender owns the RTP packets, GeneratePadding() family of methods
@ -1238,109 +1236,6 @@ TEST_P(RtpSenderTest, SendRedundantPayloads) {
return;
}
test::ScopedFieldTrials field_trials(
"WebRTC-PayloadPadding-UseMostUsefulPacket/Disabled/");
MockTransport transport;
RtpRtcp::Configuration config;
config.clock = &fake_clock_;
config.outgoing_transport = &transport;
config.paced_sender = &mock_paced_sender_;
config.media_send_ssrc = kSsrc;
config.rtx_send_ssrc = kRtxSsrc;
config.event_log = &mock_rtc_event_log_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_sender_ = absl::make_unique<RTPSender>(config);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
uint16_t seq_num = kSeqNum;
rtp_sender_->SetStorePacketsStatus(true, 10);
int32_t rtp_header_len = kRtpHeaderSize;
EXPECT_EQ(
0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
kAbsoluteSendTimeExtensionId));
rtp_header_len += 4; // 4 bytes extension.
rtp_header_len += 4; // 4 extra bytes common to all extension headers.
rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
const size_t kNumPayloadSizes = 10;
const size_t kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700,
750, 800, 850, 900, 950};
// Expect all packets go through the pacer.
EXPECT_CALL(mock_rtc_event_log_,
LogProxy(SameRtcEventTypeAs(RtcEvent::Type::RtpPacketOutgoing)))
.Times(kNumPayloadSizes);
// Send 10 packets of increasing size.
for (size_t i = 0; i < kNumPayloadSizes; ++i) {
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
EXPECT_CALL(transport, SendRtp(_, _, _)).WillOnce(::testing::Return(true));
if (GetParam().pacer_references_packets) {
EXPECT_CALL(mock_paced_sender_, InsertPacket(_, kSsrc, seq_num, _, _, _));
SendPacket(capture_time_ms, kPayloadSizes[i]);
rtp_sender_->TimeToSendPacket(kSsrc, seq_num,
fake_clock_.TimeInMilliseconds(), false,
PacedPacketInfo());
} else {
EXPECT_CALL(
mock_paced_sender_,
EnqueuePacket(AllOf(
Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc)),
Pointee(Property(&RtpPacketToSend::SequenceNumber, seq_num)))));
auto packet = SendPacket(capture_time_ms, kPayloadSizes[i]);
packet->set_packet_type(RtpPacketToSend::Type::kVideo);
rtp_sender_->TrySendPacket(packet.get(), PacedPacketInfo());
}
++seq_num;
fake_clock_.AdvanceTimeMilliseconds(33);
}
EXPECT_CALL(mock_rtc_event_log_,
LogProxy(SameRtcEventTypeAs(RtcEvent::Type::RtpPacketOutgoing)))
.Times(AtLeast(4));
// The amount of padding to send it too small to send a payload packet.
EXPECT_CALL(transport, SendRtp(_, kMaxPaddingSize + rtp_header_len, _))
.WillOnce(Return(true));
EXPECT_EQ(kMaxPaddingSize,
rtp_sender_->TimeToSendPadding(49, PacedPacketInfo()));
PacketOptions options;
EXPECT_CALL(transport,
SendRtp(_, kPayloadSizes[0] + rtp_header_len + kRtxHeaderSize, _))
.WillOnce(DoAll(SaveArg<2>(&options), Return(true)));
EXPECT_EQ(kPayloadSizes[0],
rtp_sender_->TimeToSendPadding(500, PacedPacketInfo()));
EXPECT_TRUE(options.is_retransmit);
EXPECT_CALL(transport, SendRtp(_,
kPayloadSizes[kNumPayloadSizes - 1] +
rtp_header_len + kRtxHeaderSize,
_))
.WillOnce(Return(true));
options.is_retransmit = false;
EXPECT_CALL(transport, SendRtp(_, kMaxPaddingSize + rtp_header_len, _))
.WillOnce(DoAll(SaveArg<2>(&options), Return(true)));
EXPECT_EQ(kPayloadSizes[kNumPayloadSizes - 1] + kMaxPaddingSize,
rtp_sender_->TimeToSendPadding(999, PacedPacketInfo()));
EXPECT_FALSE(options.is_retransmit);
}
TEST_P(RtpSenderTest, SendRedundantPayloadsUsefulPadding) {
if (!GetParam().pacer_references_packets) {
// If PacedSender owns the RTP packets, GeneratePadding() family of methods
// will be called instead and this test makes no sense.
return;
}
test::ScopedFieldTrials field_trials(
"WebRTC-PayloadPadding-UseMostUsefulPacket/Enabled/");
MockTransport transport;
RtpRtcp::Configuration config;
config.clock = &fake_clock_;