Add original SSRC to RtpPacketToSend and implement RtpPacketSendInfo::From method.
The purpose is to be able to create a RtpPacketSendInfo from Pacing and RtpPacketSendInfo only.
This allow further refactoring where we directly in PacketRouter can notify BWE and early loss detection that a packet will be sent.
RtpPacketSendInfo::From is mostly added to be able to test conversion.
Bug: webrtc:15368
Change-Id: I5ebe2dc91d2eedf2c86e62c3f9738437082a49e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343766
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41961}
diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index 0d35591..c5b526c 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -210,6 +210,7 @@
"source/rtp_header_extension_size.h",
"source/rtp_packet_history.cc",
"source/rtp_packet_history.h",
+ "source/rtp_packet_send_info.cc",
"source/rtp_packetizer_av1.cc",
"source/rtp_packetizer_av1.h",
"source/rtp_rtcp_config.h",
@@ -610,6 +611,7 @@
"source/rtp_header_extension_map_unittest.cc",
"source/rtp_header_extension_size_unittest.cc",
"source/rtp_packet_history_unittest.cc",
+ "source/rtp_packet_send_info_unittest.cc",
"source/rtp_packet_unittest.cc",
"source/rtp_packetizer_av1_unittest.cc",
"source/rtp_rtcp_impl2_unittest.cc",
diff --git a/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
index 249cf83..37410e2 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp_defines.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp_defines.h
@@ -13,6 +13,7 @@
#include <stddef.h>
+#include <cstdint>
#include <memory>
#include <vector>
@@ -185,6 +186,9 @@
};
struct RtpPacketSendInfo {
+ static RtpPacketSendInfo From(const RtpPacketToSend& rtp_packet_to_send,
+ const PacedPacketInfo& pacing_info);
+
uint16_t transport_sequence_number = 0;
absl::optional<uint32_t> media_ssrc;
uint16_t rtp_sequence_number = 0; // Only valid if `media_ssrc` is set.
diff --git a/modules/rtp_rtcp/source/rtp_packet_send_info.cc b/modules/rtp_rtcp/source/rtp_packet_send_info.cc
new file mode 100644
index 0000000..6ec7671
--- /dev/null
+++ b/modules/rtp_rtcp/source/rtp_packet_send_info.cc
@@ -0,0 +1,63 @@
+/*
+ * Copyright (c) 2024 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 <cstdint>
+
+#include "absl/types/optional.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+
+namespace webrtc {
+
+RtpPacketSendInfo RtpPacketSendInfo::From(const RtpPacketToSend& packet,
+ const PacedPacketInfo& pacing_info) {
+ RtpPacketSendInfo packet_info;
+ if (packet.transport_sequence_number()) {
+ packet_info.transport_sequence_number =
+ *packet.transport_sequence_number() & 0xFFFF;
+ } else {
+ absl::optional<uint16_t> packet_id =
+ packet.GetExtension<TransportSequenceNumber>();
+ if (packet_id) {
+ packet_info.transport_sequence_number = *packet_id;
+ }
+ }
+
+ 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 = packet.Ssrc();
+ packet_info.rtp_sequence_number = packet.SequenceNumber();
+ break;
+ case RtpPacketMediaType::kRetransmission:
+ RTC_DCHECK(packet.original_ssrc() &&
+ packet.retransmitted_sequence_number());
+ // For retransmissions, we're want to remove the original media packet
+ // if the retransmit arrives - so populate that in the packet info.
+ packet_info.media_ssrc = packet.original_ssrc().value_or(0);
+ packet_info.rtp_sequence_number =
+ packet.retransmitted_sequence_number().value_or(0);
+ break;
+ case RtpPacketMediaType::kPadding:
+ case RtpPacketMediaType::kForwardErrorCorrection:
+ // We're not interested in feedback about these packets being received
+ // or lost.
+ break;
+ }
+ return packet_info;
+}
+
+} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_packet_send_info_unittest.cc b/modules/rtp_rtcp/source/rtp_packet_send_info_unittest.cc
new file mode 100644
index 0000000..223ce26
--- /dev/null
+++ b/modules/rtp_rtcp/source/rtp_packet_send_info_unittest.cc
@@ -0,0 +1,119 @@
+/*
+ * Copyright (c) 2024 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 <cstdint>
+
+#include "absl/types/optional.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+RtpPacketToSend BuildPacket(RtpPacketMediaType type) {
+ RtpHeaderExtensionMap extension_manager;
+ RtpPacketToSend packet(&extension_manager);
+
+ packet.SetSsrc(1);
+ packet.SetSequenceNumber(89);
+ if (type == RtpPacketMediaType::kRetransmission) {
+ packet.set_original_ssrc(2);
+ packet.set_retransmitted_sequence_number(678);
+ }
+ packet.set_transport_sequence_number(0xFFFFFFFF01);
+ packet.SetTimestamp(123);
+ packet.SetPayloadSize(5);
+ packet.set_packet_type(type);
+ return packet;
+}
+
+void VerifyDefaultProperties(const RtpPacketSendInfo& send_info,
+ const RtpPacketToSend& packet,
+ const PacedPacketInfo& paced_info) {
+ EXPECT_EQ(send_info.length, packet.size());
+ EXPECT_EQ(send_info.rtp_timestamp, packet.Timestamp());
+ EXPECT_EQ(send_info.packet_type, packet.packet_type());
+ EXPECT_EQ(send_info.pacing_info, paced_info);
+ if (packet.transport_sequence_number()) {
+ EXPECT_EQ(send_info.transport_sequence_number,
+ *packet.transport_sequence_number() & 0xFFFF);
+ } else {
+ EXPECT_EQ(send_info.transport_sequence_number,
+ *packet.GetExtension<TransportSequenceNumber>());
+ }
+}
+
+TEST(RtpPacketSendInfoTest, FromConvertsMediaPackets) {
+ RtpPacketToSend packet = BuildPacket(RtpPacketMediaType::kAudio);
+ PacedPacketInfo paced_info;
+ paced_info.probe_cluster_id = 8;
+
+ RtpPacketSendInfo send_info = RtpPacketSendInfo::From(packet, paced_info);
+ EXPECT_EQ(send_info.media_ssrc, packet.Ssrc());
+ VerifyDefaultProperties(send_info, packet, paced_info);
+}
+
+TEST(RtpPacketSendInfoTest, FromConvertsPadding) {
+ RtpPacketToSend packet = BuildPacket(RtpPacketMediaType::kPadding);
+ PacedPacketInfo paced_info;
+ paced_info.probe_cluster_id = 8;
+
+ RtpPacketSendInfo send_info = RtpPacketSendInfo::From(packet, paced_info);
+ EXPECT_EQ(send_info.media_ssrc, absl::nullopt);
+ VerifyDefaultProperties(send_info, packet, paced_info);
+}
+
+TEST(RtpPacketSendInfoTest, FromConvertsFec) {
+ RtpPacketToSend packet =
+ BuildPacket(RtpPacketMediaType::kForwardErrorCorrection);
+ PacedPacketInfo paced_info;
+ paced_info.probe_cluster_id = 8;
+
+ RtpPacketSendInfo send_info = RtpPacketSendInfo::From(packet, paced_info);
+ EXPECT_EQ(send_info.media_ssrc, absl::nullopt);
+ VerifyDefaultProperties(send_info, packet, paced_info);
+}
+
+TEST(RtpPacketSendInfoTest, FromConvertsRetransmission) {
+ RtpPacketToSend packet = BuildPacket(RtpPacketMediaType::kRetransmission);
+ PacedPacketInfo paced_info;
+ paced_info.probe_cluster_id = 8;
+
+ RtpPacketSendInfo send_info = RtpPacketSendInfo::From(packet, paced_info);
+ EXPECT_EQ(send_info.media_ssrc, *packet.original_ssrc());
+ EXPECT_EQ(send_info.rtp_sequence_number,
+ *packet.retransmitted_sequence_number());
+ VerifyDefaultProperties(send_info, packet, paced_info);
+}
+
+TEST(RtpPacketSendInfoTest, FromFallbackToTranportSequenceHeaderExtension) {
+ RtpHeaderExtensionMap extension_manager;
+ extension_manager.Register<TransportSequenceNumber>(/*id=*/1);
+ PacedPacketInfo paced_info;
+ paced_info.probe_cluster_id = 8;
+ RtpPacketToSend packet(&extension_manager);
+ packet.SetSsrc(1);
+ packet.SetSequenceNumber(89);
+ const uint16_t kTransportSequenceNumber = 5555;
+ packet.SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
+ packet.SetTimestamp(123);
+ packet.AllocatePayload(5);
+ packet.set_packet_type(RtpPacketMediaType::kAudio);
+
+ RtpPacketSendInfo send_info = RtpPacketSendInfo::From(packet, paced_info);
+ VerifyDefaultProperties(send_info, packet, paced_info);
+}
+
+} // namespace
+} // namespace webrtc
diff --git a/modules/rtp_rtcp/source/rtp_packet_to_send.h b/modules/rtp_rtcp/source/rtp_packet_to_send.h
index 515fdc2..47747a0 100644
--- a/modules/rtp_rtcp/source/rtp_packet_to_send.h
+++ b/modules/rtp_rtcp/source/rtp_packet_to_send.h
@@ -71,6 +71,11 @@
return retransmitted_sequence_number_;
}
+ // If this is a retransmission, indicates the SSRC of the original
+ // media packet that this packet represents.
+ void set_original_ssrc(uint32_t ssrc) { original_ssrc_ = ssrc; }
+ absl::optional<uint32_t> original_ssrc() const { return original_ssrc_; }
+
void set_allow_retransmission(bool allow_retransmission) {
allow_retransmission_ = allow_retransmission;
}
@@ -149,6 +154,7 @@
webrtc::Timestamp capture_time_ = webrtc::Timestamp::Zero();
absl::optional<RtpPacketMediaType> packet_type_;
absl::optional<OriginalType> original_packet_type_;
+ absl::optional<uint32_t> original_ssrc_;
absl::optional<int64_t> transport_sequence_number_;
bool allow_retransmission_ = false;
absl::optional<uint16_t> retransmitted_sequence_number_;
diff --git a/modules/rtp_rtcp/source/rtp_sender.cc b/modules/rtp_rtcp/source/rtp_sender.cc
index 07356b8..1c2d746 100644
--- a/modules/rtp_rtcp/source/rtp_sender.cc
+++ b/modules/rtp_rtcp/source/rtp_sender.cc
@@ -291,6 +291,7 @@
if (retransmit_packet) {
retransmit_packet->set_retransmitted_sequence_number(
stored_packet.SequenceNumber());
+ retransmit_packet->set_original_ssrc(stored_packet.Ssrc());
}
return retransmit_packet;
});
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.cc b/modules/rtp_rtcp/source/rtp_sender_egress.cc
index 58ac0f8..705e9d5 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.cc
@@ -269,8 +269,9 @@
} else if (packet->transport_sequence_number()) {
options.packet_id = *packet->transport_sequence_number();
}
- if (options.packet_id >= 0) {
- AddPacketToTransportFeedback(options.packet_id, *packet, pacing_info);
+ if (options.packet_id >= 0 && transport_feedback_observer_) {
+ transport_feedback_observer_->OnAddPacket(
+ RtpPacketSendInfo::From(*packet, pacing_info));
}
if (packet->packet_type() != RtpPacketMediaType::kPadding &&
@@ -415,42 +416,6 @@
return false;
}
-void RtpSenderEgress::AddPacketToTransportFeedback(
- uint16_t packet_id,
- const RtpPacketToSend& packet,
- const PacedPacketInfo& pacing_info) {
- if (transport_feedback_observer_) {
- RtpPacketSendInfo packet_info;
- packet_info.transport_sequence_number = packet_id;
- 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 retransmit 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;
- }
-
- transport_feedback_observer_->OnAddPacket(packet_info);
- }
-}
-
bool RtpSenderEgress::SendPacketToNetwork(const RtpPacketToSend& packet,
const PacketOptions& options,
const PacedPacketInfo& pacing_info) {
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress.h b/modules/rtp_rtcp/source/rtp_sender_egress.h
index 42f3c96..7f038f6 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress.h
+++ b/modules/rtp_rtcp/source/rtp_sender_egress.h
@@ -110,9 +110,6 @@
};
void CompleteSendPacket(const Packet& compound_packet, bool last_in_batch);
bool HasCorrectSsrc(const RtpPacketToSend& packet) const;
- void AddPacketToTransportFeedback(uint16_t packet_id,
- const RtpPacketToSend& packet,
- const PacedPacketInfo& pacing_info);
// Sends packet on to `transport_`, leaving the RTP module.
bool SendPacketToNetwork(const RtpPacketToSend& packet,
diff --git a/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
index a3cf612..908df95 100644
--- a/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_egress_unittest.cc
@@ -10,6 +10,7 @@
#include "modules/rtp_rtcp/source/rtp_sender_egress.h"
+#include <cstdint>
#include <string>
#include "absl/types/optional.h"
@@ -201,7 +202,7 @@
Field(&RtpPacketSendInfo::pacing_info, PacedPacketInfo()))));
std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
- packet->SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
+ packet->set_transport_sequence_number(kTransportSequenceNumber);
packet->AllocatePayload(kPayloadSize);
std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
@@ -277,6 +278,7 @@
header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
TransportSequenceNumber::Uri());
std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
+ packet->set_transport_sequence_number(1);
sender->SendPacket(std::move(packet), PacedPacketInfo());
EXPECT_TRUE(transport_.last_packet()->options.included_in_feedback);
}
@@ -289,6 +291,7 @@
header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
TransportSequenceNumber::Uri());
std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
+ packet->set_transport_sequence_number(1);
sender->SendPacket(std::move(packet), PacedPacketInfo());
EXPECT_TRUE(transport_.last_packet()->options.included_in_allocation);
}
@@ -397,7 +400,7 @@
send_packet_observer_,
OnSendPacket(Eq(kTransportSequenceNumber), clock_->CurrentTime(), kSsrc));
std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
- packet->SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
+ packet->set_transport_sequence_number(kTransportSequenceNumber);
sender->SendPacket(std::move(packet), PacedPacketInfo());
}
@@ -417,9 +420,10 @@
const uint16_t kTransportSequenceNumber = 1;
EXPECT_CALL(send_packet_observer_, OnSendPacket).Times(0);
std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
- packet->SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
+ packet->set_transport_sequence_number(kTransportSequenceNumber);
packet->set_packet_type(RtpPacketMediaType::kRetransmission);
packet->set_retransmitted_sequence_number(packet->SequenceNumber());
+ packet->set_original_ssrc(packet->Ssrc());
sender->SendPacket(std::move(packet), PacedPacketInfo());
}
@@ -794,14 +798,16 @@
TEST_F(RtpSenderEgressTest, SendPacketSetsPacketOptions) {
const uint16_t kPacketId = 42;
+ const uint16_t kSequenceNumber = 456;
std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
header_extensions_.RegisterByUri(kTransportSequenceNumberExtensionId,
TransportSequenceNumber::Uri());
std::unique_ptr<RtpPacketToSend> packet = BuildRtpPacket();
- packet->SetExtension<TransportSequenceNumber>(kPacketId);
+ uint32_t ssrc = packet->Ssrc();
+ packet->SetSequenceNumber(kSequenceNumber);
+ packet->set_transport_sequence_number(kPacketId);
EXPECT_CALL(send_packet_observer_, OnSendPacket);
- auto packet_sequence_number = packet->SequenceNumber();
sender->SendPacket(std::move(packet), PacedPacketInfo());
PacketOptions packet_options = transport_.last_packet()->options;
@@ -813,9 +819,10 @@
// Send another packet as retransmission, verify options are populated.
std::unique_ptr<RtpPacketToSend> retransmission = BuildRtpPacket();
- retransmission->SetExtension<TransportSequenceNumber>(kPacketId + 1);
+ retransmission->set_transport_sequence_number(kPacketId + 1);
retransmission->set_packet_type(RtpPacketMediaType::kRetransmission);
- retransmission->set_retransmitted_sequence_number(packet_sequence_number);
+ retransmission->set_retransmitted_sequence_number(kSequenceNumber);
+ retransmission->set_original_ssrc(ssrc);
sender->SendPacket(std::move(retransmission), PacedPacketInfo());
EXPECT_TRUE(transport_.last_packet()->options.is_retransmit);
}
@@ -879,20 +886,21 @@
std::unique_ptr<RtpPacketToSend> video_packet = BuildRtpPacket();
video_packet->set_packet_type(RtpPacketMediaType::kVideo);
video_packet->SetPayloadSize(kPayloadSize);
- video_packet->SetExtension<TransportSequenceNumber>(1);
+ video_packet->set_transport_sequence_number(1);
std::unique_ptr<RtpPacketToSend> rtx_packet = BuildRtpPacket();
rtx_packet->SetSsrc(kRtxSsrc);
rtx_packet->set_packet_type(RtpPacketMediaType::kRetransmission);
+ rtx_packet->set_original_ssrc(video_packet->Ssrc());
rtx_packet->set_retransmitted_sequence_number(video_packet->SequenceNumber());
rtx_packet->SetPayloadSize(kPayloadSize);
- rtx_packet->SetExtension<TransportSequenceNumber>(2);
+ rtx_packet->set_transport_sequence_number(2);
std::unique_ptr<RtpPacketToSend> fec_packet = BuildRtpPacket();
fec_packet->SetSsrc(kFlexFecSsrc);
fec_packet->set_packet_type(RtpPacketMediaType::kForwardErrorCorrection);
fec_packet->SetPayloadSize(kPayloadSize);
- fec_packet->SetExtension<TransportSequenceNumber>(3);
+ fec_packet->set_transport_sequence_number(3);
const int64_t kDiffMs = 25;
time_controller_.AdvanceTime(TimeDelta::Millis(kDiffMs));
@@ -924,8 +932,8 @@
TransportSequenceNumber::Uri());
std::unique_ptr<RtpPacketToSend> retransmission = BuildRtpPacket();
retransmission->set_packet_type(RtpPacketMediaType::kRetransmission);
- retransmission->SetExtension<TransportSequenceNumber>(
- kTransportSequenceNumber);
+ retransmission->set_transport_sequence_number(kTransportSequenceNumber);
+ retransmission->set_original_ssrc(kSsrc);
uint16_t retransmitted_seq = retransmission->SequenceNumber() - 2;
retransmission->set_retransmitted_sequence_number(retransmitted_seq);
@@ -947,8 +955,8 @@
std::unique_ptr<RtpPacketToSend> rtx_retransmission = BuildRtpPacket();
rtx_retransmission->SetSsrc(kRtxSsrc);
- rtx_retransmission->SetExtension<TransportSequenceNumber>(
- kTransportSequenceNumber);
+ rtx_retransmission->set_transport_sequence_number(kTransportSequenceNumber);
+ rtx_retransmission->set_original_ssrc(kSsrc);
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);
@@ -971,7 +979,7 @@
std::unique_ptr<RtpPacketToSend> padding = BuildRtpPacket();
padding->SetPadding(224);
padding->set_packet_type(RtpPacketMediaType::kPadding);
- padding->SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
+ padding->set_transport_sequence_number(kTransportSequenceNumber);
std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
EXPECT_CALL(
@@ -991,7 +999,7 @@
rtx_padding->SetPadding(224);
rtx_padding->SetSsrc(kRtxSsrc);
rtx_padding->set_packet_type(RtpPacketMediaType::kPadding);
- rtx_padding->SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
+ rtx_padding->set_transport_sequence_number(kTransportSequenceNumber);
std::unique_ptr<RtpSenderEgress> sender = CreateRtpSenderEgress();
EXPECT_CALL(
@@ -1010,7 +1018,7 @@
std::unique_ptr<RtpPacketToSend> fec_packet = BuildRtpPacket();
fec_packet->SetSsrc(kFlexFecSsrc);
fec_packet->set_packet_type(RtpPacketMediaType::kForwardErrorCorrection);
- fec_packet->SetExtension<TransportSequenceNumber>(kTransportSequenceNumber);
+ fec_packet->set_transport_sequence_number(kTransportSequenceNumber);
const rtc::ArrayView<const RtpExtensionSize> kNoRtpHeaderExtensionSizes;
FlexfecSender flexfec(kFlexfectPayloadType, kFlexFecSsrc, kSsrc, /*mid=*/"",
diff --git a/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index 7c012be..269d003 100644
--- a/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -1017,7 +1017,10 @@
// Build a media packet and put in the packet history.
std::unique_ptr<RtpPacketToSend> packet =
BuildRtpPacket(kPayload, true, 0, clock_->CurrentTime());
- const uint16_t media_sequence_number = packet->SequenceNumber();
+ const uint32_t kMediaSsrc = 567;
+ const uint16_t kMediaSequenceNumber = 123;
+ packet->SetSsrc(kMediaSsrc);
+ packet->SetSequenceNumber(kMediaSequenceNumber);
packet->set_allow_retransmission(true);
packet_history_->PutRtpPacket(std::move(packet), clock_->CurrentTime());
@@ -1028,9 +1031,10 @@
EnqueuePackets(ElementsAre(AllOf(
Pointee(Property(&RtpPacketToSend::packet_type,
RtpPacketMediaType::kRetransmission)),
+ Pointee(Property(&RtpPacketToSend::original_ssrc, kMediaSsrc)),
Pointee(Property(&RtpPacketToSend::retransmitted_sequence_number,
- Eq(media_sequence_number)))))));
- EXPECT_THAT(rtp_sender_->ReSendPacket(media_sequence_number), Gt(0));
+ Eq(kMediaSequenceNumber)))))));
+ EXPECT_THAT(rtp_sender_->ReSendPacket(kMediaSequenceNumber), Gt(0));
}
TEST_F(RtpSenderTest, GeneratedPaddingHasBweExtensions) {
