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Don't write pacer exit timestamp without pacer

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And allow populating network2 timestamp if we want to preserve pacer
timestamp.

Bug: webrtc:8853
Change-Id: I895d5ce8a9cca8ceeec3bf08e2eff02bf3b2f5fd
Reviewed-on: https://webrtc-review.googlesource.com/48640
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21937}
This commit is contained in:
Erik Språng
2018-02-07 14:37:37 +01:00
committed by Commit Bot
parent a8b150888c
commit 7b52f102ef
5 changed files with 106 additions and 64 deletions
Download Patch File Download Diff File Expand all files Collapse all files

3
modules/rtp_rtcp/include/rtp_rtcp.h
View File

@ -96,6 +96,9 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
RtpKeepAliveConfig keepalive_config;
RtcpIntervalConfig rtcp_interval_config;
// Update network2 instead of pacer_exit field of video timing extension.
bool populate_network2_timestamp = false;
private:
RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);
};

12
modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -112,20 +112,18 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
rtt_ms_(0) {
if (!configuration.receiver_only) {
rtp_sender_.reset(new RTPSender(
configuration.audio,
configuration.clock,
configuration.outgoing_transport,
configuration.paced_sender,
configuration.audio, configuration.clock,
configuration.outgoing_transport, configuration.paced_sender,
configuration.flexfec_sender,
configuration.transport_sequence_number_allocator,
configuration.transport_feedback_callback,
configuration.send_bitrate_observer,
configuration.send_frame_count_observer,
configuration.send_side_delay_observer,
configuration.event_log,
configuration.send_side_delay_observer, configuration.event_log,
configuration.send_packet_observer,
configuration.retransmission_rate_limiter,
configuration.overhead_observer));
configuration.overhead_observer,
configuration.populate_network2_timestamp));
// Make sure rtcp sender use same timestamp offset as rtp sender.
rtcp_sender_.SetTimestampOffset(rtp_sender_->TimestampOffset());

13
modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -96,7 +96,8 @@ RTPSender::RTPSender(
RtcEventLog* event_log,
SendPacketObserver* send_packet_observer,
RateLimiter* retransmission_rate_limiter,
OverheadObserver* overhead_observer)
OverheadObserver* overhead_observer,
bool populate_network2_timestamp)
: clock_(clock),
// TODO(holmer): Remove this conversion?
clock_delta_ms_(clock_->TimeInMilliseconds() - rtc::TimeMillis()),
@ -139,6 +140,7 @@ RTPSender::RTPSender(
rtp_overhead_bytes_per_packet_(0),
retransmission_rate_limiter_(retransmission_rate_limiter),
overhead_observer_(overhead_observer),
populate_network2_timestamp_(populate_network2_timestamp),
send_side_bwe_with_overhead_(
webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")) {
// This random initialization is not intended to be cryptographic strong.
@ -767,8 +769,13 @@ bool RTPSender::PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet,
packet_to_send->SetExtension<AbsoluteSendTime>(
AbsoluteSendTime::MsTo24Bits(now_ms));
if (packet_to_send->HasExtension<VideoTimingExtension>())
if (packet_to_send->HasExtension<VideoTimingExtension>()) {
if (populate_network2_timestamp_) {
packet_to_send->set_network2_time_ms(now_ms);
} else {
packet_to_send->set_pacer_exit_time_ms(now_ms);
}
}
PacketOptions options;
if (UpdateTransportSequenceNumber(packet_to_send, &options.packet_id)) {
@ -857,8 +864,6 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
if (packet->capture_time_ms() > 0) {
packet->SetExtension<TransmissionOffset>(
kTimestampTicksPerMs * (now_ms - packet->capture_time_ms()));
if (packet->HasExtension<VideoTimingExtension>())
packet->set_pacer_exit_time_ms(now_ms);
}
packet->SetExtension<AbsoluteSendTime>(AbsoluteSendTime::MsTo24Bits(now_ms));

4
modules/rtp_rtcp/source/rtp_sender.h
View File

@ -60,7 +60,8 @@ class RTPSender {
RtcEventLog* event_log,
SendPacketObserver* send_packet_observer,
RateLimiter* nack_rate_limiter,
OverheadObserver* overhead_observer);
OverheadObserver* overhead_observer,
bool populate_network2_timestamp);
~RTPSender();
@ -328,6 +329,7 @@ class RTPSender {
RateLimiter* const retransmission_rate_limiter_;
OverheadObserver* overhead_observer_;
const bool populate_network2_timestamp_;
const bool send_side_bwe_with_overhead_;

130
modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -164,14 +164,14 @@ class RtpSenderTest : public ::testing::TestWithParam<bool> {
field_trials_(GetParam() ? "WebRTC-SendSideBwe-WithOverhead/Enabled/"
: "") {}
void SetUp() override { SetUpRtpSender(true); }
void SetUp() override { SetUpRtpSender(true, false); }
void SetUpRtpSender(bool pacer) {
void SetUpRtpSender(bool pacer, bool populate_network2) {
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport_, pacer ? &mock_paced_sender_ : nullptr,
nullptr, &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
&mock_rtc_event_log_, &send_packet_observer_,
&retransmission_rate_limiter_, nullptr));
&retransmission_rate_limiter_, nullptr, populate_network2));
rtp_sender_->SetSendPayloadType(kPayload);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetTimestampOffset(0);
@ -255,7 +255,7 @@ class RtpSenderTest : public ::testing::TestWithParam<bool> {
// default code path.
class RtpSenderTestWithoutPacer : public RtpSenderTest {
public:
void SetUp() override { SetUpRtpSender(false); }
void SetUp() override { SetUpRtpSender(false, false); }
};
class TestRtpSenderVideo : public RTPSenderVideo {
@ -279,7 +279,7 @@ class RtpSenderVideoTest : public RtpSenderTest {
protected:
void SetUp() override {
// TODO(pbos): Set up to use pacer.
SetUpRtpSender(false);
SetUpRtpSender(false, false);
rtp_sender_video_.reset(
new TestRtpSenderVideo(&fake_clock_, rtp_sender_.get(), nullptr));
}
@ -385,7 +385,7 @@ TEST_P(RtpSenderTestWithoutPacer,
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
&feedback_observer_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
nullptr, &retransmission_rate_limiter_, &mock_overhead_observer));
nullptr, &retransmission_rate_limiter_, &mock_overhead_observer, false));
rtp_sender_->SetSSRC(kSsrc);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
@ -412,7 +412,7 @@ TEST_P(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
&feedback_observer_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
&send_packet_observer_, &retransmission_rate_limiter_, nullptr));
&send_packet_observer_, &retransmission_rate_limiter_, nullptr, false));
rtp_sender_->SetSSRC(kSsrc);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
@ -460,7 +460,7 @@ TEST_P(RtpSenderTest, SendsPacketsWithTransportSequenceNumber) {
false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr,
&seq_num_allocator_, &feedback_observer_, nullptr, nullptr, nullptr,
&mock_rtc_event_log_, &send_packet_observer_,
&retransmission_rate_limiter_, nullptr));
&retransmission_rate_limiter_, nullptr, false));
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetSSRC(kSsrc);
rtp_sender_->SetStorePacketsStatus(true, 10);
@ -491,7 +491,7 @@ TEST_P(RtpSenderTest, SendsPacketsWithTransportSequenceNumber) {
EXPECT_EQ(transport_.last_packet_id_, transport_seq_no);
}
TEST_P(RtpSenderTestWithoutPacer, WritesTimestampToTimingExtension) {
TEST_P(RtpSenderTest, WritesPacerExitToTimingExtension) {
rtp_sender_->SetStorePacketsStatus(true, 10);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionVideoTiming, kVideoTimingExtensionId));
@ -507,11 +507,17 @@ TEST_P(RtpSenderTestWithoutPacer, WritesTimestampToTimingExtension) {
size_t packet_size = packet->size();
const int kStoredTimeInMs = 100;
fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
{
EXPECT_CALL(
mock_paced_sender_,
InsertPacket(RtpPacketSender::kNormalPriority, kSsrc, _, _, _, _));
EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
kAllowRetransmission,
RtpPacketSender::kNormalPriority));
}
fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
rtp_sender_->TimeToSendPacket(kSsrc, kSeqNum, capture_time_ms, false,
PacedPacketInfo());
EXPECT_EQ(1, transport_.packets_sent());
EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
@ -519,17 +525,45 @@ TEST_P(RtpSenderTestWithoutPacer, WritesTimestampToTimingExtension) {
EXPECT_TRUE(transport_.last_sent_packet().GetExtension<VideoTimingExtension>(
&video_timing));
EXPECT_EQ(kStoredTimeInMs, video_timing.pacer_exit_delta_ms);
}
TEST_P(RtpSenderTest, WritesNetwork2ToTimingExtension) {
SetUpRtpSender(true, true);
rtp_sender_->SetStorePacketsStatus(true, 10);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionVideoTiming, kVideoTimingExtensionId));
int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
auto packet = rtp_sender_->AllocatePacket();
packet->SetPayloadType(kPayload);
packet->SetMarker(true);
packet->SetTimestamp(kTimestamp);
packet->set_capture_time_ms(capture_time_ms);
const uint16_t kPacerExitMs = 1234u;
const VideoSendTiming kVideoTiming = {0u, 0u, 0u, kPacerExitMs, 0u, 0u, true};
packet->SetExtension<VideoTimingExtension>(kVideoTiming);
EXPECT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
size_t packet_size = packet->size();
const int kStoredTimeInMs = 100;
{
EXPECT_CALL(
mock_paced_sender_,
InsertPacket(RtpPacketSender::kNormalPriority, kSsrc, _, _, _, _));
EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
kAllowRetransmission,
RtpPacketSender::kNormalPriority));
}
fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
rtp_sender_->TimeToSendPacket(kSsrc, kSeqNum, capture_time_ms, false,
PacedPacketInfo());
EXPECT_EQ(2, transport_.packets_sent());
EXPECT_EQ(1, transport_.packets_sent());
EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
VideoSendTiming video_timing;
EXPECT_TRUE(transport_.last_sent_packet().GetExtension<VideoTimingExtension>(
&video_timing));
EXPECT_EQ(kStoredTimeInMs * 2, video_timing.pacer_exit_delta_ms);
EXPECT_EQ(kStoredTimeInMs, video_timing.network2_timestamp_delta_ms);
EXPECT_EQ(kPacerExitMs, video_timing.pacer_exit_delta_ms);
}
TEST_P(RtpSenderTest, TrafficSmoothingWithExtensions) {
@ -785,7 +819,7 @@ TEST_P(RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator) {
false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr,
nullptr /* TransportSequenceNumberAllocator */, nullptr, nullptr, nullptr,
nullptr, nullptr, &send_packet_observer_, &retransmission_rate_limiter_,
nullptr));
nullptr, false));
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetSSRC(kSsrc);
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
@ -811,7 +845,7 @@ TEST_P(RtpSenderTest, SendRedundantPayloads) {
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport, &mock_paced_sender_, nullptr, nullptr,
nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr,
&retransmission_rate_limiter_, nullptr));
&retransmission_rate_limiter_, nullptr, false));
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetSSRC(kSsrc);
rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
@ -925,7 +959,7 @@ TEST_P(RtpSenderTest, SendFlexfecPackets) {
false, &fake_clock_, &transport_, &mock_paced_sender_, &flexfec_sender,
&seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
&mock_rtc_event_log_, &send_packet_observer_,
&retransmission_rate_limiter_, nullptr));
&retransmission_rate_limiter_, nullptr, false));
rtp_sender_->SetSSRC(kMediaSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetSendPayloadType(kMediaPayloadType);
@ -985,7 +1019,7 @@ TEST_P(RtpSenderTest, NoFlexfecForTimingFrames) {
false, &fake_clock_, &transport_, &mock_paced_sender_, &flexfec_sender,
&seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
&mock_rtc_event_log_, &send_packet_observer_,
&retransmission_rate_limiter_, nullptr));
&retransmission_rate_limiter_, nullptr, false));
rtp_sender_->SetSSRC(kMediaSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetSendPayloadType(kMediaPayloadType);
@ -1081,11 +1115,11 @@ TEST_P(RtpSenderTestWithoutPacer, SendFlexfecPackets) {
nullptr /* rtp_state */, &fake_clock_);
// Reset |rtp_sender_| to use FlexFEC.
rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
&flexfec_sender, &seq_num_allocator_, nullptr,
nullptr, nullptr, nullptr,
rtp_sender_.reset(
new RTPSender(false, &fake_clock_, &transport_, nullptr, &flexfec_sender,
&seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
&mock_rtc_event_log_, &send_packet_observer_,
&retransmission_rate_limiter_, nullptr));
&retransmission_rate_limiter_, nullptr, false));
rtp_sender_->SetSSRC(kMediaSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetSendPayloadType(kMediaPayloadType);
@ -1126,7 +1160,7 @@ TEST_P(RtpSenderTest, FecOverheadRate) {
false, &fake_clock_, &transport_, &mock_paced_sender_, &flexfec_sender,
&seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
&mock_rtc_event_log_, &send_packet_observer_,
&retransmission_rate_limiter_, nullptr));
&retransmission_rate_limiter_, nullptr, false));
rtp_sender_->SetSSRC(kMediaSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetSendPayloadType(kMediaPayloadType);
@ -1176,10 +1210,10 @@ TEST_P(RtpSenderTest, FrameCountCallbacks) {
FrameCounts frame_counts_;
} callback;
rtp_sender_.reset(
new RTPSender(false, &fake_clock_, &transport_, &mock_paced_sender_,
nullptr, nullptr, nullptr, nullptr, &callback, nullptr,
nullptr, nullptr, &retransmission_rate_limiter_, nullptr));
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr, nullptr,
nullptr, nullptr, &callback, nullptr, nullptr, nullptr,
&retransmission_rate_limiter_, nullptr, false));
rtp_sender_->SetSSRC(kSsrc);
char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
const uint8_t payload_type = 127;
@ -1238,10 +1272,10 @@ TEST_P(RtpSenderTest, BitrateCallbacks) {
uint32_t total_bitrate_;
uint32_t retransmit_bitrate_;
} callback;
rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
nullptr, nullptr, nullptr, &callback, nullptr,
nullptr, nullptr, nullptr,
&retransmission_rate_limiter_, nullptr));
rtp_sender_.reset(
new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr,
nullptr, &callback, nullptr, nullptr, nullptr, nullptr,
&retransmission_rate_limiter_, nullptr, false));
rtp_sender_->SetSSRC(kSsrc);
// Simulate kNumPackets sent with kPacketInterval ms intervals, with the
@ -1297,10 +1331,10 @@ class RtpSenderAudioTest : public RtpSenderTest {
void SetUp() override {
payload_ = kAudioPayload;
rtp_sender_.reset(new RTPSender(true, &fake_clock_, &transport_, nullptr,
nullptr, nullptr, nullptr, nullptr, nullptr,
nullptr, nullptr, nullptr,
&retransmission_rate_limiter_, nullptr));
rtp_sender_.reset(
new RTPSender(true, &fake_clock_, &transport_, nullptr, nullptr,
nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
nullptr, &retransmission_rate_limiter_, nullptr, false));
rtp_sender_->SetSSRC(kSsrc);
rtp_sender_->SetSequenceNumber(kSeqNum);
}
@ -1897,10 +1931,10 @@ TEST_P(RtpSenderVideoTest, ConditionalRetransmitLimit) {
TEST_P(RtpSenderTest, OnOverheadChanged) {
MockOverheadObserver mock_overhead_observer;
rtp_sender_.reset(
new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr,
nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
&retransmission_rate_limiter_, &mock_overhead_observer));
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
nullptr, nullptr, nullptr, nullptr, nullptr,
&retransmission_rate_limiter_, &mock_overhead_observer, false));
rtp_sender_->SetSSRC(kSsrc);
// RTP overhead is 12B.
@ -1918,10 +1952,10 @@ TEST_P(RtpSenderTest, OnOverheadChanged) {
TEST_P(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) {
MockOverheadObserver mock_overhead_observer;
rtp_sender_.reset(
new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr,
nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
&retransmission_rate_limiter_, &mock_overhead_observer));
rtp_sender_.reset(new RTPSender(
false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
nullptr, nullptr, nullptr, nullptr, nullptr,
&retransmission_rate_limiter_, &mock_overhead_observer, false));
rtp_sender_->SetSSRC(kSsrc);
EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1);
@ -1935,7 +1969,7 @@ TEST_P(RtpSenderTest, SendAudioPadding) {
rtp_sender_.reset(new RTPSender(
kEnableAudio, &fake_clock_, &transport, &mock_paced_sender_, nullptr,
nullptr, nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
nullptr, &retransmission_rate_limiter_, nullptr));
nullptr, &retransmission_rate_limiter_, nullptr, false));
rtp_sender_->SetSendPayloadType(kPayload);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetTimestampOffset(0);
@ -1958,10 +1992,10 @@ TEST_P(RtpSenderTest, SendAudioPadding) {
TEST_P(RtpSenderTest, SendsKeepAlive) {
MockTransport transport;
rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport, nullptr,
nullptr, nullptr, nullptr, nullptr, nullptr,
nullptr, &mock_rtc_event_log_, nullptr,
&retransmission_rate_limiter_, nullptr));
rtp_sender_.reset(
new RTPSender(false, &fake_clock_, &transport, nullptr, nullptr, nullptr,
nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
nullptr, &retransmission_rate_limiter_, nullptr, false));
rtp_sender_->SetSendPayloadType(kPayload);
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetTimestampOffset(0);