Revert "Correctly handle retransmissions/padding in early loss detection."
This reverts commit e9ae4729e03f60dbe3b1828dd9009b401097cd3f.
Reason for revert: Internal test failure
Original change's description:
> Correctly handle retransmissions/padding in early loss detection.
>
> This CL makes sure we don't cull packets from the history based on
> incorrect ack mapping, just like it's predecessor:
> https://webrtc-review.googlesource.com/c/src/+/218000
>
> It also changes the logic to make sure retransmits counts towards
> history pruning - and properly ignores padding/fec.
>
> Bug: webrtc:12713
> Change-Id: I7835d10e483687e960a9cce41d4e2f1a6c3189b4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221863
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34293}
TBR=danilchap@webrtc.org,terelius@webrtc.org,sprang@webrtc.org,philipel@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: Iaca6dc7739d953e97add5f5d516139b4819e43ee
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12713
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222601
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34294}
diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index e20ba32..110e28b 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -932,45 +932,43 @@
// Map from SSRC to all acked packets for that RTP module.
std::map<uint32_t, std::vector<uint16_t>> acked_packets_per_ssrc;
for (const StreamPacketInfo& packet : packet_feedback_vector) {
- if (packet.received && packet.ssrc) {
- acked_packets_per_ssrc[*packet.ssrc].push_back(
- packet.rtp_sequence_number);
+ if (packet.received) {
+ acked_packets_per_ssrc[packet.ssrc].push_back(packet.rtp_sequence_number);
}
}
- // Map from SSRC to vector of RTP sequence numbers that are indicated as
- // lost by feedback, without being trailed by any received packets.
- std::map<uint32_t, std::vector<uint16_t>> early_loss_detected_per_ssrc;
+ // Map from SSRC to vector of RTP sequence numbers that are indicated as
+ // lost by feedback, without being trailed by any received packets.
+ std::map<uint32_t, std::vector<uint16_t>> early_loss_detected_per_ssrc;
- for (const StreamPacketInfo& packet : packet_feedback_vector) {
- // Only include new media packets, not retransmissions/padding/fec.
- if (!packet.received && packet.ssrc && !packet.is_retransmission) {
- // Last known lost packet, might not be detectable as lost by remote
- // jitter buffer.
- early_loss_detected_per_ssrc[*packet.ssrc].push_back(
- packet.rtp_sequence_number);
- } else {
- // Packet received, so any loss prior to this is already detectable.
- early_loss_detected_per_ssrc.erase(*packet.ssrc);
+ for (const StreamPacketInfo& packet : packet_feedback_vector) {
+ if (!packet.received) {
+ // Last known lost packet, might not be detectable as lost by remote
+ // jitter buffer.
+ early_loss_detected_per_ssrc[packet.ssrc].push_back(
+ packet.rtp_sequence_number);
+ } else {
+ // Packet received, so any loss prior to this is already detectable.
+ early_loss_detected_per_ssrc.erase(packet.ssrc);
+ }
}
- }
- for (const auto& kv : early_loss_detected_per_ssrc) {
- const uint32_t ssrc = kv.first;
- auto it = ssrc_to_rtp_module_.find(ssrc);
- RTC_CHECK(it != ssrc_to_rtp_module_.end());
- RTPSender* rtp_sender = it->second->RtpSender();
- for (uint16_t sequence_number : kv.second) {
- rtp_sender->ReSendPacket(sequence_number);
+ for (const auto& kv : early_loss_detected_per_ssrc) {
+ const uint32_t ssrc = kv.first;
+ auto it = ssrc_to_rtp_module_.find(ssrc);
+ RTC_DCHECK(it != ssrc_to_rtp_module_.end());
+ RTPSender* rtp_sender = it->second->RtpSender();
+ for (uint16_t sequence_number : kv.second) {
+ rtp_sender->ReSendPacket(sequence_number);
+ }
}
- }
for (const auto& kv : acked_packets_per_ssrc) {
const uint32_t ssrc = kv.first;
auto it = ssrc_to_rtp_module_.find(ssrc);
if (it == ssrc_to_rtp_module_.end()) {
- // No media, likely FEC or padding. Ignore since there's no RTP history to
- // clean up anyway.
+ // Packets not for a media SSRC, so likely RTX or FEC. If so, ignore
+ // since there's no RTP history to clean up anyway.
continue;
}
rtc::ArrayView<const uint16_t> rtp_sequence_numbers(kv.second);
diff --git a/call/rtp_video_sender_unittest.cc b/call/rtp_video_sender_unittest.cc
index 334d97c..85934cc 100644
--- a/call/rtp_video_sender_unittest.cc
+++ b/call/rtp_video_sender_unittest.cc
@@ -462,13 +462,11 @@
lost_packet_feedback.rtp_sequence_number = rtp_sequence_numbers[0];
lost_packet_feedback.ssrc = kSsrc1;
lost_packet_feedback.received = false;
- lost_packet_feedback.is_retransmission = false;
StreamFeedbackObserver::StreamPacketInfo received_packet_feedback;
received_packet_feedback.rtp_sequence_number = rtp_sequence_numbers[1];
received_packet_feedback.ssrc = kSsrc1;
received_packet_feedback.received = true;
- lost_packet_feedback.is_retransmission = false;
test.router()->OnPacketFeedbackVector(
{lost_packet_feedback, received_packet_feedback});
@@ -640,13 +638,11 @@
first_packet_feedback.rtp_sequence_number = frame1_rtp_sequence_number;
first_packet_feedback.ssrc = kSsrc1;
first_packet_feedback.received = false;
- first_packet_feedback.is_retransmission = false;
StreamFeedbackObserver::StreamPacketInfo second_packet_feedback;
second_packet_feedback.rtp_sequence_number = frame2_rtp_sequence_number;
second_packet_feedback.ssrc = kSsrc2;
second_packet_feedback.received = true;
- first_packet_feedback.is_retransmission = false;
test.router()->OnPacketFeedbackVector(
{first_packet_feedback, second_packet_feedback});
diff --git a/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc b/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
index 933abd9..138cdb6 100644
--- a/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
+++ b/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
@@ -27,9 +27,9 @@
using ::testing::Invoke;
namespace webrtc {
+namespace webrtc_cc {
namespace {
-constexpr uint32_t kSsrc = 8492;
const PacedPacketInfo kPacingInfo0(0, 5, 2000);
const PacedPacketInfo kPacingInfo1(1, 8, 4000);
const PacedPacketInfo kPacingInfo2(2, 14, 7000);
@@ -77,6 +77,10 @@
return res;
}
+} // namespace
+
+namespace test {
+
class MockStreamFeedbackObserver : public webrtc::StreamFeedbackObserver {
public:
MOCK_METHOD(void,
@@ -85,8 +89,6 @@
(override));
};
-} // namespace
-
class TransportFeedbackAdapterTest : public ::testing::Test {
public:
TransportFeedbackAdapterTest() : clock_(0) {}
@@ -106,7 +108,7 @@
void OnSentPacket(const PacketResult& packet_feedback) {
RtpPacketSendInfo packet_info;
- packet_info.media_ssrc = kSsrc;
+ packet_info.ssrc = kSsrc;
packet_info.transport_sequence_number =
packet_feedback.sent_packet.sequence_number;
packet_info.rtp_sequence_number = 0;
@@ -120,6 +122,8 @@
packet_feedback.sent_packet.send_time.ms(), rtc::PacketInfo()));
}
+ static constexpr uint32_t kSsrc = 8492;
+
SimulatedClock clock_;
std::unique_ptr<TransportFeedbackAdapter> adapter_;
};
@@ -389,7 +393,7 @@
// Add a packet and then mark it as sent.
RtpPacketSendInfo packet_info;
- packet_info.media_ssrc = kSsrc;
+ packet_info.ssrc = kSsrc;
packet_info.transport_sequence_number = packet.sent_packet.sequence_number;
packet_info.length = packet.sent_packet.size.bytes();
packet_info.pacing_info = packet.sent_packet.pacing_info;
@@ -408,4 +412,6 @@
EXPECT_FALSE(duplicate_packet.has_value());
}
+} // namespace test
+} // namespace webrtc_cc
} // namespace webrtc
diff --git a/modules/congestion_controller/rtp/transport_feedback_demuxer.cc b/modules/congestion_controller/rtp/transport_feedback_demuxer.cc
index 6ab3ad8..c958a1c 100644
--- a/modules/congestion_controller/rtp/transport_feedback_demuxer.cc
+++ b/modules/congestion_controller/rtp/transport_feedback_demuxer.cc
@@ -38,16 +38,15 @@
void TransportFeedbackDemuxer::AddPacket(const RtpPacketSendInfo& packet_info) {
MutexLock lock(&lock_);
-
- StreamFeedbackObserver::StreamPacketInfo info;
- info.ssrc = packet_info.media_ssrc;
- info.rtp_sequence_number = packet_info.rtp_sequence_number;
- info.received = false;
- info.is_retransmission =
- packet_info.packet_type == RtpPacketMediaType::kRetransmission;
- history_.insert(
- {seq_num_unwrapper_.Unwrap(packet_info.transport_sequence_number), info});
-
+ if (packet_info.ssrc != 0) {
+ StreamFeedbackObserver::StreamPacketInfo info;
+ info.ssrc = packet_info.ssrc;
+ info.rtp_sequence_number = packet_info.rtp_sequence_number;
+ info.received = false;
+ history_.insert(
+ {seq_num_unwrapper_.Unwrap(packet_info.transport_sequence_number),
+ info});
+ }
while (history_.size() > kMaxPacketsInHistory) {
history_.erase(history_.begin());
}
diff --git a/modules/congestion_controller/rtp/transport_feedback_demuxer_unittest.cc b/modules/congestion_controller/rtp/transport_feedback_demuxer_unittest.cc
index 482f58d..6514a4e 100644
--- a/modules/congestion_controller/rtp/transport_feedback_demuxer_unittest.cc
+++ b/modules/congestion_controller/rtp/transport_feedback_demuxer_unittest.cc
@@ -16,11 +16,7 @@
namespace webrtc {
namespace {
-using ::testing::AllOf;
-using ::testing::ElementsAre;
-using ::testing::Field;
-using PacketInfo = StreamFeedbackObserver::StreamPacketInfo;
-
+using ::testing::_;
static constexpr uint32_t kSsrc = 8492;
class MockStreamFeedbackObserver : public webrtc::StreamFeedbackObserver {
@@ -32,65 +28,41 @@
};
RtpPacketSendInfo CreatePacket(uint32_t ssrc,
- uint16_t rtp_sequence_number,
- int64_t transport_sequence_number,
- bool is_retransmission) {
+ int16_t rtp_sequence_number,
+ int64_t transport_sequence_number) {
RtpPacketSendInfo res;
- res.media_ssrc = ssrc;
+ res.ssrc = ssrc;
res.transport_sequence_number = transport_sequence_number;
res.rtp_sequence_number = rtp_sequence_number;
- res.packet_type = is_retransmission ? RtpPacketMediaType::kRetransmission
- : RtpPacketMediaType::kVideo;
return res;
}
} // namespace
-
TEST(TransportFeedbackDemuxerTest, ObserverSanity) {
TransportFeedbackDemuxer demuxer;
MockStreamFeedbackObserver mock;
demuxer.RegisterStreamFeedbackObserver({kSsrc}, &mock);
- const uint16_t kRtpStartSeq = 55;
- const int64_t kTransportStartSeq = 1;
- demuxer.AddPacket(CreatePacket(kSsrc, kRtpStartSeq, kTransportStartSeq,
- /*is_retransmit=*/false));
- demuxer.AddPacket(CreatePacket(kSsrc, kRtpStartSeq + 1,
- kTransportStartSeq + 1,
- /*is_retransmit=*/false));
- demuxer.AddPacket(CreatePacket(
- kSsrc, kRtpStartSeq + 2, kTransportStartSeq + 2, /*is_retransmit=*/true));
+ demuxer.AddPacket(CreatePacket(kSsrc, 55, 1));
+ demuxer.AddPacket(CreatePacket(kSsrc, 56, 2));
+ demuxer.AddPacket(CreatePacket(kSsrc, 57, 3));
rtcp::TransportFeedback feedback;
- feedback.SetBase(kTransportStartSeq, 1000);
- ASSERT_TRUE(feedback.AddReceivedPacket(kTransportStartSeq, 1000));
- // Drop middle packet.
- ASSERT_TRUE(feedback.AddReceivedPacket(kTransportStartSeq + 2, 3000));
+ feedback.SetBase(1, 1000);
+ ASSERT_TRUE(feedback.AddReceivedPacket(1, 1000));
+ ASSERT_TRUE(feedback.AddReceivedPacket(2, 2000));
+ ASSERT_TRUE(feedback.AddReceivedPacket(3, 3000));
- EXPECT_CALL(
- mock, OnPacketFeedbackVector(ElementsAre(
- AllOf(Field(&PacketInfo::received, true),
- Field(&PacketInfo::ssrc, kSsrc),
- Field(&PacketInfo::rtp_sequence_number, kRtpStartSeq),
- Field(&PacketInfo::is_retransmission, false)),
- AllOf(Field(&PacketInfo::received, false),
- Field(&PacketInfo::ssrc, kSsrc),
- Field(&PacketInfo::rtp_sequence_number, kRtpStartSeq + 1),
- Field(&PacketInfo::is_retransmission, false)),
- AllOf(Field(&PacketInfo::received, true),
- Field(&PacketInfo::ssrc, kSsrc),
- Field(&PacketInfo::rtp_sequence_number, kRtpStartSeq + 2),
- Field(&PacketInfo::is_retransmission, true)))));
+ EXPECT_CALL(mock, OnPacketFeedbackVector(_)).Times(1);
demuxer.OnTransportFeedback(feedback);
demuxer.DeRegisterStreamFeedbackObserver(&mock);
- demuxer.AddPacket(
- CreatePacket(kSsrc, kRtpStartSeq + 3, kTransportStartSeq + 3, false));
+ demuxer.AddPacket(CreatePacket(kSsrc, 58, 4));
rtcp::TransportFeedback second_feedback;
- second_feedback.SetBase(kTransportStartSeq + 3, 4000);
- ASSERT_TRUE(second_feedback.AddReceivedPacket(kTransportStartSeq + 3, 4000));
+ second_feedback.SetBase(4, 4000);
+ ASSERT_TRUE(second_feedback.AddReceivedPacket(4, 4000));
- EXPECT_CALL(mock, OnPacketFeedbackVector).Times(0);
+ EXPECT_CALL(mock, OnPacketFeedbackVector(_)).Times(0);
demuxer.OnTransportFeedback(second_feedback);
}
} // namespace webrtc
diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index ca12adf..59c0f29 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -228,10 +228,8 @@
RtpPacketSendInfo() = default;
uint16_t transport_sequence_number = 0;
- // TODO(bugs.webrtc.org/12713): Remove once downstream usage is gone.
uint32_t ssrc = 0;
- absl::optional<uint32_t> media_ssrc;
- uint16_t rtp_sequence_number = 0; // Only valid if |ssrc| is set.
+ uint16_t rtp_sequence_number = 0;
uint32_t rtp_timestamp = 0;
size_t length = 0;
absl::optional<RtpPacketMediaType> packet_type;
@@ -269,13 +267,9 @@
class StreamFeedbackObserver {
public:
struct StreamPacketInfo {
- bool received;
-
- // |rtp_sequence_number| and |is_retransmission| are only valid if |ssrc|
- // is populated.
- absl::optional<uint32_t> ssrc;
+ uint32_t ssrc;
uint16_t rtp_sequence_number;
- bool is_retransmission;
+ bool received;
};
virtual ~StreamFeedbackObserver() = default;
diff --git a/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc b/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc
index 33a1b8e..3f7d22c 100644
--- a/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc
+++ b/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc
@@ -313,7 +313,7 @@
}
RtpPacketSendInfo packet_info;
- packet_info.media_ssrc = ssrc_;
+ packet_info.ssrc = ssrc_;
packet_info.transport_sequence_number = packet_id;
packet_info.rtp_sequence_number = packet.SequenceNumber();
packet_info.length = packet_size;
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.cc b/modules/rtp_rtcp/source/rtp_sender_egress.cc
index 126b89c..55dd9ff 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.cc
@@ -142,9 +142,6 @@
RTC_DCHECK(packet->packet_type().has_value());
RTC_DCHECK(HasCorrectSsrc(*packet));
- if (packet->packet_type() == RtpPacketMediaType::kRetransmission) {
- RTC_DCHECK(packet->retransmitted_sequence_number().has_value());
- }
const uint32_t packet_ssrc = packet->Ssrc();
const int64_t now_ms = clock_->TimeInMilliseconds();
@@ -412,34 +409,13 @@
}
RtpPacketSendInfo packet_info;
+ packet_info.ssrc = ssrc_;
packet_info.transport_sequence_number = packet_id;
+ packet_info.rtp_sequence_number = packet.SequenceNumber();
packet_info.rtp_timestamp = packet.Timestamp();
packet_info.length = packet_size;
packet_info.pacing_info = pacing_info;
packet_info.packet_type = packet.packet_type();
-
- switch (*packet_info.packet_type) {
- case RtpPacketMediaType::kAudio:
- case RtpPacketMediaType::kVideo:
- packet_info.media_ssrc = ssrc_;
- packet_info.rtp_sequence_number = packet.SequenceNumber();
- break;
- case RtpPacketMediaType::kRetransmission:
- // For retransmissions, we're want to remove the original media packet
- // if the rentrasmit arrives - so populate that in the packet info.
- packet_info.media_ssrc = ssrc_;
- packet_info.rtp_sequence_number =
- *packet.retransmitted_sequence_number();
- break;
- case RtpPacketMediaType::kPadding:
- case RtpPacketMediaType::kForwardErrorCorrection:
- // We're not interested in feedback about these packets being received
- // or lost.
- break;
- }
- // TODO(bugs.webrtc.org/12713): Remove once downstream usage is gone.
- packet_info.ssrc = packet_info.media_ssrc.value_or(0);
-
transport_feedback_observer_->OnAddPacket(packet_info);
}
}
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
index 4f3990c..663638f 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
@@ -221,7 +221,7 @@
EXPECT_CALL(
feedback_observer_,
OnAddPacket(AllOf(
- Field(&RtpPacketSendInfo::media_ssrc, kSsrc),
+ Field(&RtpPacketSendInfo::ssrc, kSsrc),
Field(&RtpPacketSendInfo::transport_sequence_number,
kTransportSequenceNumber),
Field(&RtpPacketSendInfo::rtp_sequence_number, kStartSequenceNumber),
@@ -246,8 +246,6 @@
std::unique_ptr<RtpPacketToSend> retransmission = BuildRtpPacket();
retransmission->set_packet_type(RtpPacketMediaType::kRetransmission);
- retransmission->set_retransmitted_sequence_number(
- media_packet->SequenceNumber());
sender->SendPacket(retransmission.get(), PacedPacketInfo());
EXPECT_TRUE(transport_.last_packet()->options.is_retransmit);
}
@@ -409,7 +407,6 @@
std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
packet->SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
packet->set_packet_type(RtpPacketMediaType::kRetransmission);
- packet->set_retransmitted_sequence_number(packet->SequenceNumber());
sender->SendPacket(packet.get(), PacedPacketInfo());
}
@@ -468,7 +465,6 @@
// Mark all packets as retransmissions - will cause total and retransmission
// rates to be equal.
packet->set_packet_type(RtpPacketMediaType::kRetransmission);
- packet->set_retransmitted_sequence_number(packet->SequenceNumber());
total_data_sent += DataSize::Bytes(packet->size());
EXPECT_CALL(observer, Notify(_, _, kSsrc))
@@ -524,8 +520,6 @@
std::unique_ptr<RtpPacketToSend> retransmission = BuildRtpPacket();
retransmission->set_allow_retransmission(true);
retransmission->set_packet_type(RtpPacketMediaType::kRetransmission);
- retransmission->set_retransmitted_sequence_number(
- retransmission->SequenceNumber());
sender->SendPacket(retransmission.get(), PacedPacketInfo());
EXPECT_FALSE(packet_history_.GetPacketState(retransmission->SequenceNumber())
.has_value());
@@ -606,8 +600,6 @@
// and retransmitted packet statistics.
std::unique_ptr<RtpPacketToSend> retransmission_packet = BuildRtpPacket();
retransmission_packet->set_packet_type(RtpPacketMediaType::kRetransmission);
- retransmission_packet->set_retransmitted_sequence_number(
- retransmission_packet->SequenceNumber());
media_packet->SetPayloadSize(7);
expected_transmitted_counter.packets += 1;
expected_transmitted_counter.payload_bytes +=
@@ -718,7 +710,6 @@
rtx_packet->set_packet_type(RtpPacketMediaType::kRetransmission);
rtx_packet->SetSsrc(kRtxSsrc);
rtx_packet->SetPayloadSize(7);
- rtx_packet->set_retransmitted_sequence_number(media_packet->SequenceNumber());
sender->SendPacket(rtx_packet.get(), PacedPacketInfo());
time_controller_.AdvanceTime(TimeDelta::Zero());
@@ -794,7 +785,6 @@
std::unique_ptr<RtpPacketToSend> retransmission = BuildRtpPacket();
retransmission->SetExtension<TransportSequenceNumber>(kPacketId + 1);
retransmission->set_packet_type(RtpPacketMediaType::kRetransmission);
- retransmission->set_retransmitted_sequence_number(packet->SequenceNumber());
sender->SendPacket(retransmission.get(), PacedPacketInfo());
EXPECT_TRUE(transport_.last_packet()->options.is_retransmit);
}
@@ -825,7 +815,6 @@
std::unique_ptr<RtpPacketToSend> rtx_packet = BuildRtpPacket();
rtx_packet->SetSsrc(kRtxSsrc);
rtx_packet->set_packet_type(RtpPacketMediaType::kRetransmission);
- rtx_packet->set_retransmitted_sequence_number(video_packet->SequenceNumber());
rtx_packet->SetPayloadSize(kPayloadSize);
rtx_packet->SetExtension<TransportSequenceNumber>(2);
@@ -865,115 +854,6 @@
EXPECT_EQ(rtx_stats.retransmitted.packets, 1u);
}
-TEST_P(RtpSenderEgressTest, TransportFeedbackObserverWithRetransmission) {
- const uint16_t kTransportSequenceNumber = 17;
- header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
- TransportSequenceNumber::kUri);
- std::unique_ptr<RtpPacketToSend> retransmission = BuildRtpPacket();
- retransmission->set_packet_type(RtpPacketMediaType::kRetransmission);
- retransmission->SetExtension<TransportSequenceNumber>(
- kTransportSequenceNumber);
- uint16_t retransmitted_seq = retransmission->SequenceNumber() - 2;
- retransmission->set_retransmitted_sequence_number(retransmitted_seq);
-
- std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
- EXPECT_CALL(
- feedback_observer_,
- OnAddPacket(AllOf(
- Field(&RtpPacketSendInfo::media_ssrc, kSsrc),
- Field(&RtpPacketSendInfo::rtp_sequence_number, retransmitted_seq),
- Field(&RtpPacketSendInfo::transport_sequence_number,
- kTransportSequenceNumber))));
- sender->SendPacket(retransmission.get(), PacedPacketInfo());
-}
-
-TEST_P(RtpSenderEgressTest, TransportFeedbackObserverWithRtxRetransmission) {
- const uint16_t kTransportSequenceNumber = 17;
- header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
- TransportSequenceNumber::kUri);
-
- std::unique_ptr<RtpPacketToSend> rtx_retransmission = BuildRtpPacket();
- rtx_retransmission->SetSsrc(kRtxSsrc);
- rtx_retransmission->SetExtension<TransportSequenceNumber>(
- kTransportSequenceNumber);
- rtx_retransmission->set_packet_type(RtpPacketMediaType::kRetransmission);
- uint16_t rtx_retransmitted_seq = rtx_retransmission->SequenceNumber() - 2;
- rtx_retransmission->set_retransmitted_sequence_number(rtx_retransmitted_seq);
-
- std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
- EXPECT_CALL(
- feedback_observer_,
- OnAddPacket(AllOf(
- Field(&RtpPacketSendInfo::media_ssrc, kSsrc),
- Field(&RtpPacketSendInfo::rtp_sequence_number, rtx_retransmitted_seq),
- Field(&RtpPacketSendInfo::transport_sequence_number,
- kTransportSequenceNumber))));
- sender->SendPacket(rtx_retransmission.get(), PacedPacketInfo());
-}
-
-TEST_P(RtpSenderEgressTest, TransportFeedbackObserverPadding) {
- const uint16_t kTransportSequenceNumber = 17;
- header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
- TransportSequenceNumber::kUri);
- std::unique_ptr<RtpPacketToSend> padding = BuildRtpPacket();
- padding->SetPadding(224);
- padding->set_packet_type(RtpPacketMediaType::kPadding);
- padding->SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
-
- std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
- EXPECT_CALL(
- feedback_observer_,
- OnAddPacket(AllOf(Field(&RtpPacketSendInfo::media_ssrc, absl::nullopt),
- Field(&RtpPacketSendInfo::transport_sequence_number,
- kTransportSequenceNumber))));
- sender->SendPacket(padding.get(), PacedPacketInfo());
-}
-
-TEST_P(RtpSenderEgressTest, TransportFeedbackObserverRtxPadding) {
- const uint16_t kTransportSequenceNumber = 17;
- header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
- TransportSequenceNumber::kUri);
-
- std::unique_ptr<RtpPacketToSend> rtx_padding = BuildRtpPacket();
- rtx_padding->SetPadding(224);
- rtx_padding->SetSsrc(kRtxSsrc);
- rtx_padding->set_packet_type(RtpPacketMediaType::kPadding);
- rtx_padding->SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
-
- std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
- EXPECT_CALL(
- feedback_observer_,
- OnAddPacket(AllOf(Field(&RtpPacketSendInfo::media_ssrc, absl::nullopt),
- Field(&RtpPacketSendInfo::transport_sequence_number,
- kTransportSequenceNumber))));
- sender->SendPacket(rtx_padding.get(), PacedPacketInfo());
-}
-
-TEST_P(RtpSenderEgressTest, TransportFeedbackObserverFec) {
- const uint16_t kTransportSequenceNumber = 17;
- header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
- TransportSequenceNumber::kUri);
-
- std::unique_ptr<RtpPacketToSend> fec_packet = BuildRtpPacket();
- fec_packet->SetSsrc(kFlexFecSsrc);
- fec_packet->set_packet_type(RtpPacketMediaType::kForwardErrorCorrection);
- fec_packet->SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
-
- const rtc::ArrayView<const RtpExtensionSize> kNoRtpHeaderExtensionSizes;
- FlexfecSender flexfec(kFlexfectPayloadType, kFlexFecSsrc, kSsrc, /*mid=*/"",
- /*header_extensions=*/{}, kNoRtpHeaderExtensionSizes,
- /*rtp_state=*/nullptr, time_controller_.GetClock());
- RtpRtcpInterface::Configuration config = DefaultConfig();
- config.fec_generator = &flexfec;
- auto sender = std::make_unique<RtpSenderEgress>(config, &packet_history_);
- EXPECT_CALL(
- feedback_observer_,
- OnAddPacket(AllOf(Field(&RtpPacketSendInfo::media_ssrc, absl::nullopt),
- Field(&RtpPacketSendInfo::transport_sequence_number,
- kTransportSequenceNumber))));
- sender->SendPacket(fec_packet.get(), PacedPacketInfo());
-}
-
INSTANTIATE_TEST_SUITE_P(WithAndWithoutOverhead,
RtpSenderEgressTest,
::testing::Values(TestConfig(false),
diff --git a/rtc_tools/rtc_event_log_visualizer/analyzer.cc b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
index 0f727f2..7690d7d 100644
--- a/rtc_tools/rtc_event_log_visualizer/analyzer.cc
+++ b/rtc_tools/rtc_event_log_visualizer/analyzer.cc
@@ -1267,7 +1267,7 @@
const RtpPacketType& rtp_packet = *rtp_iterator->second;
if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
RtpPacketSendInfo packet_info;
- packet_info.media_ssrc = rtp_packet.rtp.header.ssrc;
+ packet_info.ssrc = rtp_packet.rtp.header.ssrc;
packet_info.transport_sequence_number =
rtp_packet.rtp.header.extension.transportSequenceNumber;
packet_info.rtp_sequence_number = rtp_packet.rtp.header.sequenceNumber;
diff --git a/rtc_tools/rtc_event_log_visualizer/log_simulation.cc b/rtc_tools/rtc_event_log_visualizer/log_simulation.cc
index c0b418d..4dbaf0b 100644
--- a/rtc_tools/rtc_event_log_visualizer/log_simulation.cc
+++ b/rtc_tools/rtc_event_log_visualizer/log_simulation.cc
@@ -84,7 +84,7 @@
}
RtpPacketSendInfo packet_info;
- packet_info.media_ssrc = packet.ssrc;
+ packet_info.ssrc = packet.ssrc;
packet_info.transport_sequence_number = packet.transport_seq_no;
packet_info.rtp_sequence_number = packet.stream_seq_no;
packet_info.length = packet.size;
