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Removes deprecated InsertPacket/TimeToSendPacket/TimeToSendPadding

Browse Source 
The methods are no longer in use, this CL cleans away references and
updates any tests using them.

Bug: webrtc:10633
Change-Id: I2db301e0a021a2f85a8b9a74e409303baba407da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150520
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28956}
This commit is contained in:
Erik Språng
2019-08-26 08:58:45 +02:00
committed by Commit Bot
parent 6558fa5d64
commit 4208a13e65
19 changed files with 207 additions and 626 deletions
Download Patch File Download Diff File Expand all files Collapse all files

14
audio/channel_send.cc
View File

@ -385,20 +385,6 @@ class RtpPacketSenderProxy : public RtpPacketSender {
rtp_packet_pacer_->EnqueuePacket(std::move(packet)); rtp_packet_pacer_->EnqueuePacket(std::move(packet));
} }
// Implements RtpPacketSender.
void InsertPacket(Priority priority,
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission) override {
rtc::CritScope lock(&crit_);
if (rtp_packet_pacer_) {
rtp_packet_pacer_->InsertPacket(priority, ssrc, sequence_number,
capture_time_ms, bytes, retransmission);
}
}
private: private:
rtc::ThreadChecker thread_checker_; rtc::ThreadChecker thread_checker_;
rtc::CriticalSection crit_; rtc::CriticalSection crit_;

17
modules/pacing/mock/mock_paced_sender.h
View File

@ -11,6 +11,7 @@
#ifndef MODULES_PACING_MOCK_MOCK_PACED_SENDER_H_ #ifndef MODULES_PACING_MOCK_MOCK_PACED_SENDER_H_
#define MODULES_PACING_MOCK_MOCK_PACED_SENDER_H_ #define MODULES_PACING_MOCK_MOCK_PACED_SENDER_H_
#include <memory>
#include <vector> #include <vector>
#include "modules/pacing/paced_sender.h" #include "modules/pacing/paced_sender.h"
@ -23,20 +24,12 @@ class MockPacedSender : public PacedSender {
public: public:
MockPacedSender() MockPacedSender()
: PacedSender(Clock::GetRealTimeClock(), nullptr, nullptr) {} : PacedSender(Clock::GetRealTimeClock(), nullptr, nullptr) {}
MOCK_METHOD6(SendPacket, MOCK_METHOD1(EnqueuePacket, void(std::unique_ptr<RtpPacketToSend> packet));
bool(Priority priority,
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission));
MOCK_METHOD2(CreateProbeCluster, void(DataRate, int)); MOCK_METHOD2(CreateProbeCluster, void(DataRate, int));
MOCK_METHOD1(SetEstimatedBitrate, void(uint32_t));
MOCK_METHOD2(SetPacingRates, void(DataRate, DataRate)); MOCK_METHOD2(SetPacingRates, void(DataRate, DataRate));
MOCK_CONST_METHOD0(QueueInMs, int64_t()); MOCK_CONST_METHOD0(OldestPacketWaitTime, TimeDelta());
MOCK_CONST_METHOD0(QueueInPackets, int()); MOCK_CONST_METHOD0(QueueSizePackets, size_t());
MOCK_CONST_METHOD0(ExpectedQueueTimeMs, int64_t()); MOCK_CONST_METHOD0(ExpectedQueueTime, TimeDelta());
MOCK_METHOD0(GetApplicationLimitedRegionStartTime, absl::optional<int64_t>());
MOCK_METHOD0(Process, void()); MOCK_METHOD0(Process, void());
}; };

11
modules/pacing/paced_sender.cc
View File

@ -83,17 +83,6 @@ void PacedSender::SetPacingRates(DataRate pacing_rate, DataRate padding_rate) {
pacing_controller_.SetPacingRates(pacing_rate, padding_rate); pacing_controller_.SetPacingRates(pacing_rate, padding_rate);
} }
void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission) {
rtc::CritScope cs(&critsect_);
pacing_controller_.InsertPacket(priority, ssrc, sequence_number,
capture_time_ms, bytes, retransmission);
}
void PacedSender::EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) { void PacedSender::EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) {
rtc::CritScope cs(&critsect_); rtc::CritScope cs(&critsect_);
pacing_controller_.EnqueuePacket(std::move(packet)); pacing_controller_.EnqueuePacket(std::move(packet));

8
modules/pacing/paced_sender.h
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@ -65,14 +65,6 @@ class PacedSender : public Module,
// Methods implementing RtpPacketSender. // Methods implementing RtpPacketSender.
// Adds the packet information to the queue and calls TimeToSendPacket
// when it's time to send.
void InsertPacket(RtpPacketSender::Priority priority,
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission) override;
// Adds the packet to the queue and calls PacketRouter::SendPacket() when // Adds the packet to the queue and calls PacketRouter::SendPacket() when
// it's time to send. // it's time to send.
void EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) override; void EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) override;

9
modules/pacing/paced_sender_unittest.cc
View File

@ -42,15 +42,6 @@ namespace test {
// Mock callback implementing the raw api. // Mock callback implementing the raw api.
class MockCallback : public PacketRouter { class MockCallback : public PacketRouter {
public: public:
MOCK_METHOD5(TimeToSendPacket,
RtpPacketSendResult(uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission,
const PacedPacketInfo& pacing_info));
MOCK_METHOD2(TimeToSendPadding,
size_t(size_t bytes, const PacedPacketInfo& pacing_info));
MOCK_METHOD2(SendPacket, MOCK_METHOD2(SendPacket,
void(std::unique_ptr<RtpPacketToSend> packet, void(std::unique_ptr<RtpPacketToSend> packet,
const PacedPacketInfo& cluster_info)); const PacedPacketInfo& cluster_info));

9
modules/pacing/pacing_controller.cc
View File

@ -184,15 +184,6 @@ void PacingController::SetPacingRates(DataRate pacing_rate,
<< " padding_budget_kbps=" << padding_rate.kbps(); << " padding_budget_kbps=" << padding_rate.kbps();
} }
void PacingController::InsertPacket(RtpPacketSender::Priority priority,
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission) {
RTC_NOTREACHED();
}
void PacingController::EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) { void PacingController::EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) {
RTC_DCHECK(pacing_bitrate_ > DataRate::Zero()) RTC_DCHECK(pacing_bitrate_ > DataRate::Zero())
<< "SetPacingRate must be called before InsertPacket."; << "SetPacingRate must be called before InsertPacket.";

8
modules/pacing/pacing_controller.h
View File

@ -76,14 +76,6 @@ class PacingController {
~PacingController(); ~PacingController();
// Adds the packet information to the queue and calls TimeToSendPacket
// when it's time to send.
void InsertPacket(RtpPacketSender::Priority priority,
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission);
// Adds the packet to the queue and calls PacketRouter::SendPacket() when // Adds the packet to the queue and calls PacketRouter::SendPacket() when
// it's time to send. // it's time to send.
void EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet); void EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet);

62
modules/pacing/packet_router.cc
View File

@ -54,7 +54,7 @@ void PacketRouter::AddSendRtpModule(RtpRtcp* rtp_module, bool remb_candidate) {
rtp_module) == rtp_send_modules_.end()); rtp_module) == rtp_send_modules_.end());
// Put modules which can use regular payload packets (over rtx) instead of // Put modules which can use regular payload packets (over rtx) instead of
// padding first as it's less of a waste // padding first as it's less of a waste
if ((rtp_module->RtxSendStatus() & kRtxRedundantPayloads) > 0) { if (rtp_module->SupportsRtxPayloadPadding()) {
rtp_send_modules_.push_front(rtp_module); rtp_send_modules_.push_front(rtp_module);
} else { } else {
rtp_send_modules_.push_back(rtp_module); rtp_send_modules_.push_back(rtp_module);
@ -102,29 +102,6 @@ void PacketRouter::RemoveReceiveRtpModule(
rtcp_feedback_senders_.erase(it); rtcp_feedback_senders_.erase(it);
} }
RtpPacketSendResult PacketRouter::TimeToSendPacket(
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_timestamp,
bool retransmission,
const PacedPacketInfo& pacing_info) {
rtc::CritScope cs(&modules_crit_);
RtpRtcp* rtp_module = FindRtpModule(ssrc);
if (rtp_module == nullptr || !rtp_module->SendingMedia()) {
return RtpPacketSendResult::kPacketNotFound;
}
RtpPacketSendResult result = rtp_module->TimeToSendPacket(
ssrc, sequence_number, capture_timestamp, retransmission, pacing_info);
if (result == RtpPacketSendResult::kSuccess &&
rtp_module->SupportsRtxPayloadPadding()) {
// This is now the last module to send media, and has the desired
// properties needed for payload based padding. Cache it for later use.
last_send_module_ = rtp_module;
}
return result;
}
RtpRtcp* PacketRouter::FindRtpModule(uint32_t ssrc) { RtpRtcp* PacketRouter::FindRtpModule(uint32_t ssrc) {
auto it = rtp_module_cache_map_.find(ssrc); auto it = rtp_module_cache_map_.find(ssrc);
if (it != rtp_module_cache_map_.end()) { if (it != rtp_module_cache_map_.end()) {
@ -150,7 +127,7 @@ void PacketRouter::SendPacket(std::unique_ptr<RtpPacketToSend> packet,
// With the new pacer code path, transport sequence numbers are only set here, // With the new pacer code path, transport sequence numbers are only set here,
// on the pacer thread. Therefore we don't need atomics/synchronization. // on the pacer thread. Therefore we don't need atomics/synchronization.
if (packet->IsExtensionReserved<TransportSequenceNumber>()) { if (packet->IsExtensionReserved<TransportSequenceNumber>()) {
packet->SetExtension<TransportSequenceNumber>(++transport_seq_); packet->SetExtension<TransportSequenceNumber>(AllocateSequenceNumber());
} }
auto it = rtp_module_cache_map_.find(packet->Ssrc()); auto it = rtp_module_cache_map_.find(packet->Ssrc());
@ -175,41 +152,6 @@ void PacketRouter::SendPacket(std::unique_ptr<RtpPacketToSend> packet,
<< packet->SequenceNumber(); << packet->SequenceNumber();
} }
size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send,
const PacedPacketInfo& pacing_info) {
size_t total_bytes_sent = 0;
rtc::CritScope cs(&modules_crit_);
// First try on the last rtp module to have sent media. This increases the
// the chance that any payload based padding will be useful as it will be
// somewhat distributed over modules according the packet rate, even if it
// will be more skewed towards the highest bitrate stream. At the very least
// this prevents sending payload padding on a disabled stream where it's
// guaranteed not to be useful.
if (last_send_module_ != nullptr &&
last_send_module_->SupportsRtxPayloadPadding()) {
RTC_DCHECK(std::find(rtp_send_modules_.begin(), rtp_send_modules_.end(),
last_send_module_) != rtp_send_modules_.end());
total_bytes_sent += last_send_module_->TimeToSendPadding(
bytes_to_send - total_bytes_sent, pacing_info);
if (total_bytes_sent >= bytes_to_send) {
return total_bytes_sent;
}
}
// Rtp modules are ordered by which stream can most benefit from padding.
// Don't require RTX payload padding in the general case.
for (RtpRtcp* module : rtp_send_modules_) {
if (module->SupportsPadding()) {
size_t bytes_sent = module->TimeToSendPadding(
bytes_to_send - total_bytes_sent, pacing_info);
total_bytes_sent += bytes_sent;
if (total_bytes_sent >= bytes_to_send)
break;
}
}
return total_bytes_sent;
}
std::vector<std::unique_ptr<RtpPacketToSend>> PacketRouter::GeneratePadding( std::vector<std::unique_ptr<RtpPacketToSend>> PacketRouter::GeneratePadding(
size_t target_size_bytes) { size_t target_size_bytes) {
rtc::CritScope cs(&modules_crit_); rtc::CritScope cs(&modules_crit_);

10
modules/pacing/packet_router.h
View File

@ -53,19 +53,9 @@ class PacketRouter : public TransportSequenceNumberAllocator,
bool remb_candidate); bool remb_candidate);
void RemoveReceiveRtpModule(RtcpFeedbackSenderInterface* rtcp_sender); void RemoveReceiveRtpModule(RtcpFeedbackSenderInterface* rtcp_sender);
virtual RtpPacketSendResult TimeToSendPacket(
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_timestamp,
bool retransmission,
const PacedPacketInfo& packet_info);
virtual void SendPacket(std::unique_ptr<RtpPacketToSend> packet, virtual void SendPacket(std::unique_ptr<RtpPacketToSend> packet,
const PacedPacketInfo& cluster_info); const PacedPacketInfo& cluster_info);
virtual size_t TimeToSendPadding(size_t bytes,
const PacedPacketInfo& packet_info);
virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding( virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
size_t target_size_bytes); size_t target_size_bytes);

546
modules/pacing/packet_router_unittest.cc
View File

@ -42,7 +42,6 @@ using ::testing::Le;
using ::testing::NiceMock; using ::testing::NiceMock;
using ::testing::Property; using ::testing::Property;
using ::testing::Return; using ::testing::Return;
using ::testing::ReturnPointee;
using ::testing::SaveArg; using ::testing::SaveArg;
constexpr int kProbeMinProbes = 5; constexpr int kProbeMinProbes = 5;
@ -50,217 +49,54 @@ constexpr int kProbeMinBytes = 1000;
} // namespace } // namespace
TEST(PacketRouterTest, Sanity_NoModuleRegistered_TimeToSendPacket) { class PacketRouterTest : public ::testing::Test {
PacketRouter packet_router; public:
PacketRouterTest() {
const int kTransportSequenceNumberExtensionId = 1;
extension_manager.Register(kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId);
}
constexpr uint16_t ssrc = 1234; protected:
constexpr uint16_t sequence_number = 17; std::unique_ptr<RtpPacketToSend> BuildRtpPacket(uint32_t ssrc) {
constexpr uint64_t timestamp = 7890; std::unique_ptr<RtpPacketToSend> packet =
constexpr bool retransmission = false; absl::make_unique<RtpPacketToSend>(&extension_manager);
const PacedPacketInfo paced_info(1, kProbeMinProbes, kProbeMinBytes); packet->SetSsrc(ssrc);
return packet;
}
EXPECT_EQ(RtpPacketSendResult::kPacketNotFound, PacketRouter packet_router_;
packet_router.TimeToSendPacket(ssrc, sequence_number, timestamp, RtpHeaderExtensionMap extension_manager;
retransmission, paced_info)); };
}
TEST(PacketRouterTest, Sanity_NoModuleRegistered_TimeToSendPadding) {
PacketRouter packet_router;
TEST_F(PacketRouterTest, Sanity_NoModuleRegistered_GeneratePadding) {
constexpr size_t bytes = 300; constexpr size_t bytes = 300;
const PacedPacketInfo paced_info(1, kProbeMinProbes, kProbeMinBytes); const PacedPacketInfo paced_info(1, kProbeMinProbes, kProbeMinBytes);
EXPECT_EQ(packet_router.TimeToSendPadding(bytes, paced_info), 0u); EXPECT_TRUE(packet_router_.GeneratePadding(bytes).empty());
} }
TEST(PacketRouterTest, Sanity_NoModuleRegistered_OnReceiveBitrateChanged) { TEST_F(PacketRouterTest, Sanity_NoModuleRegistered_OnReceiveBitrateChanged) {
PacketRouter packet_router;
const std::vector<uint32_t> ssrcs = {1, 2, 3}; const std::vector<uint32_t> ssrcs = {1, 2, 3};
constexpr uint32_t bitrate_bps = 10000; constexpr uint32_t bitrate_bps = 10000;
packet_router.OnReceiveBitrateChanged(ssrcs, bitrate_bps); packet_router_.OnReceiveBitrateChanged(ssrcs, bitrate_bps);
} }
TEST(PacketRouterTest, Sanity_NoModuleRegistered_SendRemb) { TEST_F(PacketRouterTest, Sanity_NoModuleRegistered_SendRemb) {
PacketRouter packet_router;
const std::vector<uint32_t> ssrcs = {1, 2, 3}; const std::vector<uint32_t> ssrcs = {1, 2, 3};
constexpr uint32_t bitrate_bps = 10000; constexpr uint32_t bitrate_bps = 10000;
EXPECT_FALSE(packet_router.SendRemb(bitrate_bps, ssrcs)); EXPECT_FALSE(packet_router_.SendRemb(bitrate_bps, ssrcs));
} }
TEST(PacketRouterTest, Sanity_NoModuleRegistered_SendTransportFeedback) { TEST_F(PacketRouterTest, Sanity_NoModuleRegistered_SendTransportFeedback) {
PacketRouter packet_router;
rtcp::TransportFeedback feedback; rtcp::TransportFeedback feedback;
EXPECT_FALSE(packet_router.SendTransportFeedback(&feedback)); EXPECT_FALSE(packet_router_.SendTransportFeedback(&feedback));
} }
TEST(PacketRouterTest, TimeToSendPacket) { TEST_F(PacketRouterTest, GeneratePaddingPicksCorrectModule) {
PacketRouter packet_router;
NiceMock<MockRtpRtcp> rtp_1;
NiceMock<MockRtpRtcp> rtp_2;
packet_router.AddSendRtpModule(&rtp_1, false);
packet_router.AddSendRtpModule(&rtp_2, false);
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.
ON_CALL(rtp_1, SendingMedia).WillByDefault(Return(true));
ON_CALL(rtp_1, SSRC).WillByDefault(Return(kSsrc1));
EXPECT_CALL(rtp_1, TimeToSendPacket(
kSsrc1, sequence_number, timestamp, retransmission,
Field(&PacedPacketInfo::probe_cluster_id, 1)))
.Times(1)
.WillOnce(Return(RtpPacketSendResult::kSuccess));
EXPECT_CALL(rtp_2, TimeToSendPacket).Times(0);
EXPECT_EQ(RtpPacketSendResult::kSuccess,
packet_router.TimeToSendPacket(
kSsrc1, sequence_number, timestamp, retransmission,
PacedPacketInfo(1, kProbeMinProbes, kProbeMinBytes)));
// 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;
ON_CALL(rtp_1, SendingMedia).WillByDefault(Return(false));
ON_CALL(rtp_2, SendingMedia).WillByDefault(Return(true));
ON_CALL(rtp_2, SSRC).WillByDefault(Return(kSsrc2));
EXPECT_CALL(rtp_1, TimeToSendPacket).Times(0);
EXPECT_CALL(rtp_2, TimeToSendPacket(
kSsrc2, sequence_number, timestamp, retransmission,
Field(&PacedPacketInfo::probe_cluster_id, 2)))
.Times(1)
.WillOnce(Return(RtpPacketSendResult::kSuccess));
EXPECT_EQ(RtpPacketSendResult::kSuccess,
packet_router.TimeToSendPacket(
kSsrc2, sequence_number, timestamp, retransmission,
PacedPacketInfo(2, kProbeMinProbes, kProbeMinBytes)));
// No module is sending, hence no packet should be sent.
ON_CALL(rtp_1, SendingMedia).WillByDefault(Return(false));
ON_CALL(rtp_2, SendingMedia).WillByDefault(Return(false));
EXPECT_CALL(rtp_1, TimeToSendPacket).Times(0);
EXPECT_CALL(rtp_2, TimeToSendPacket).Times(0);
EXPECT_EQ(RtpPacketSendResult::kPacketNotFound,
packet_router.TimeToSendPacket(
kSsrc1, sequence_number, timestamp, retransmission,
PacedPacketInfo(1, kProbeMinProbes, kProbeMinBytes)));
// Add a packet with incorrect ssrc and test it's dropped in the router.
ON_CALL(rtp_1, SendingMedia).WillByDefault(Return(true));
ON_CALL(rtp_1, SSRC).WillByDefault(Return(kSsrc1));
ON_CALL(rtp_2, SendingMedia).WillByDefault(Return(true));
ON_CALL(rtp_2, SSRC).WillByDefault(Return(kSsrc2));
EXPECT_CALL(rtp_1, TimeToSendPacket).Times(0);
EXPECT_CALL(rtp_2, TimeToSendPacket).Times(0);
EXPECT_EQ(RtpPacketSendResult::kPacketNotFound,
packet_router.TimeToSendPacket(
kSsrc1 + kSsrc2, sequence_number, timestamp, retransmission,
PacedPacketInfo(1, kProbeMinProbes, kProbeMinBytes)));
packet_router.RemoveSendRtpModule(&rtp_1);
// rtp_1 has been removed, try sending a packet on that ssrc and make sure
// it is dropped as expected by not expecting any calls to rtp_1.
ON_CALL(rtp_2, SendingMedia).WillByDefault(Return(true));
ON_CALL(rtp_2, SSRC).WillByDefault(Return(kSsrc2));
EXPECT_CALL(rtp_2, TimeToSendPacket).Times(0);
EXPECT_EQ(RtpPacketSendResult::kPacketNotFound,
packet_router.TimeToSendPacket(
kSsrc1, sequence_number, timestamp, retransmission,
PacedPacketInfo(PacedPacketInfo::kNotAProbe, kProbeMinBytes,
kProbeMinBytes)));
packet_router.RemoveSendRtpModule(&rtp_2);
}
TEST(PacketRouterTest, TimeToSendPadding) {
PacketRouter packet_router;
const uint16_t kSsrc1 = 1234;
const uint16_t kSsrc2 = 4567;
NiceMock<MockRtpRtcp> rtp_1;
EXPECT_CALL(rtp_1, RtxSendStatus()).WillOnce(Return(kRtxOff));
EXPECT_CALL(rtp_1, SSRC()).WillRepeatedly(Return(kSsrc1));
NiceMock<MockRtpRtcp> rtp_2;
// rtp_2 will be prioritized for padding.
EXPECT_CALL(rtp_2, RtxSendStatus()).WillOnce(Return(kRtxRedundantPayloads));
EXPECT_CALL(rtp_2, SSRC()).WillRepeatedly(Return(kSsrc2));
packet_router.AddSendRtpModule(&rtp_1, false);
packet_router.AddSendRtpModule(&rtp_2, false);
// Default configuration, sending padding on all modules sending media,
// ordered by priority (based on rtx mode).
const size_t requested_padding_bytes = 1000;
const size_t sent_padding_bytes = 890;
EXPECT_CALL(rtp_2, SupportsPadding).Times(1).WillOnce(Return(true));
EXPECT_CALL(rtp_2,
TimeToSendPadding(requested_padding_bytes,
Field(&PacedPacketInfo::probe_cluster_id, 111)))
.Times(1)
.WillOnce(Return(sent_padding_bytes));
EXPECT_CALL(rtp_1, SupportsPadding).WillOnce(Return(true));
EXPECT_CALL(rtp_1,
TimeToSendPadding(requested_padding_bytes - sent_padding_bytes,
Field(&PacedPacketInfo::probe_cluster_id, 111)))
.Times(1)
.WillOnce(Return(requested_padding_bytes - sent_padding_bytes));
EXPECT_EQ(requested_padding_bytes,
packet_router.TimeToSendPadding(
requested_padding_bytes,
PacedPacketInfo(111, kProbeMinBytes, kProbeMinBytes)));
// Let only the lower priority module be sending and verify the padding
// request is routed there.
EXPECT_CALL(rtp_2, SupportsPadding).WillOnce(Return(false));
EXPECT_CALL(rtp_2, TimeToSendPadding(requested_padding_bytes, _)).Times(0);
EXPECT_CALL(rtp_1, SupportsPadding).WillOnce(Return(true));
EXPECT_CALL(rtp_1, TimeToSendPadding(_, _))
.Times(1)
.WillOnce(Return(sent_padding_bytes));
EXPECT_EQ(sent_padding_bytes,
packet_router.TimeToSendPadding(
requested_padding_bytes,
PacedPacketInfo(PacedPacketInfo::kNotAProbe, kProbeMinBytes,
kProbeMinBytes)));
// No sending module at all.
EXPECT_CALL(rtp_1, SupportsPadding).WillOnce(Return(false));
EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes, _)).Times(0);
EXPECT_CALL(rtp_2, SupportsPadding).WillOnce(Return(false));
EXPECT_CALL(rtp_2, TimeToSendPadding(_, _)).Times(0);
EXPECT_EQ(0u, packet_router.TimeToSendPadding(
requested_padding_bytes,
PacedPacketInfo(PacedPacketInfo::kNotAProbe, kProbeMinBytes,
kProbeMinBytes)));
packet_router.RemoveSendRtpModule(&rtp_1);
// rtp_1 has been removed, try sending padding and make sure rtp_1 isn't asked
// to send by not expecting any calls. Instead verify rtp_2 is called.
EXPECT_CALL(rtp_2, SupportsPadding).WillOnce(Return(true));
EXPECT_CALL(rtp_2, TimeToSendPadding(requested_padding_bytes, _)).Times(1);
EXPECT_EQ(0u, packet_router.TimeToSendPadding(
requested_padding_bytes,
PacedPacketInfo(PacedPacketInfo::kNotAProbe, kProbeMinBytes,
kProbeMinBytes)));
packet_router.RemoveSendRtpModule(&rtp_2);
}
TEST(PacketRouterTest, GeneratePaddingPicksCorrectModule) {
PacketRouter packet_router;
// Two RTP modules. The first (prioritized due to rtx) isn't sending media so // Two RTP modules. The first (prioritized due to rtx) isn't sending media so
// should not be called. // should not be called.
const uint16_t kSsrc1 = 1234; const uint16_t kSsrc1 = 1234;
@ -276,8 +112,8 @@ TEST(PacketRouterTest, GeneratePaddingPicksCorrectModule) {
ON_CALL(rtp_2, SSRC()).WillByDefault(Return(kSsrc2)); ON_CALL(rtp_2, SSRC()).WillByDefault(Return(kSsrc2));
ON_CALL(rtp_2, SupportsPadding).WillByDefault(Return(true)); ON_CALL(rtp_2, SupportsPadding).WillByDefault(Return(true));
packet_router.AddSendRtpModule(&rtp_1, false); packet_router_.AddSendRtpModule(&rtp_1, false);
packet_router.AddSendRtpModule(&rtp_2, false); packet_router_.AddSendRtpModule(&rtp_2, false);
const size_t kPaddingSize = 123; const size_t kPaddingSize = 123;
const size_t kExpectedPaddingPackets = 1; const size_t kExpectedPaddingPackets = 1;
@ -287,187 +123,244 @@ TEST(PacketRouterTest, GeneratePaddingPicksCorrectModule) {
return std::vector<std::unique_ptr<RtpPacketToSend>>( return std::vector<std::unique_ptr<RtpPacketToSend>>(
kExpectedPaddingPackets); kExpectedPaddingPackets);
}); });
auto generated_padding = packet_router.GeneratePadding(kPaddingSize); auto generated_padding = packet_router_.GeneratePadding(kPaddingSize);
EXPECT_EQ(generated_padding.size(), kExpectedPaddingPackets); EXPECT_EQ(generated_padding.size(), kExpectedPaddingPackets);
packet_router.RemoveSendRtpModule(&rtp_1); packet_router_.RemoveSendRtpModule(&rtp_1);
packet_router.RemoveSendRtpModule(&rtp_2); packet_router_.RemoveSendRtpModule(&rtp_2);
} }
TEST(PacketRouterTest, PadsOnLastActiveMediaStream) { TEST_F(PacketRouterTest, PadsOnLastActiveMediaStream) {
PacketRouter packet_router;
const uint16_t kSsrc1 = 1234; const uint16_t kSsrc1 = 1234;
const uint16_t kSsrc2 = 4567; const uint16_t kSsrc2 = 4567;
const uint16_t kSsrc3 = 8901; const uint16_t kSsrc3 = 8901;
// First two rtp modules send media and have rtx. // First two rtp modules send media and have rtx.
NiceMock<MockRtpRtcp> rtp_1; NiceMock<MockRtpRtcp> rtp_1;
EXPECT_CALL(rtp_1, RtxSendStatus())
.WillRepeatedly(Return(kRtxRedundantPayloads));
EXPECT_CALL(rtp_1, SSRC()).WillRepeatedly(Return(kSsrc1)); EXPECT_CALL(rtp_1, SSRC()).WillRepeatedly(Return(kSsrc1));
EXPECT_CALL(rtp_1, SendingMedia()).WillRepeatedly(Return(true));
EXPECT_CALL(rtp_1, SupportsPadding).WillRepeatedly(Return(true)); EXPECT_CALL(rtp_1, SupportsPadding).WillRepeatedly(Return(true));
EXPECT_CALL(rtp_1, SupportsRtxPayloadPadding).WillRepeatedly(Return(true)); EXPECT_CALL(rtp_1, SupportsRtxPayloadPadding).WillRepeatedly(Return(true));
EXPECT_CALL(rtp_1, TrySendPacket).WillRepeatedly(Return(false));
EXPECT_CALL(
rtp_1,
TrySendPacket(
::testing::Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc1)), _))
.WillRepeatedly(Return(true));
NiceMock<MockRtpRtcp> rtp_2; NiceMock<MockRtpRtcp> rtp_2;
EXPECT_CALL(rtp_2, RtxSendStatus())
.WillRepeatedly(Return(kRtxRedundantPayloads));
EXPECT_CALL(rtp_2, SSRC()).WillRepeatedly(Return(kSsrc2)); EXPECT_CALL(rtp_2, SSRC()).WillRepeatedly(Return(kSsrc2));
EXPECT_CALL(rtp_2, SendingMedia()).WillRepeatedly(Return(true));
EXPECT_CALL(rtp_2, SupportsPadding).WillRepeatedly(Return(true)); EXPECT_CALL(rtp_2, SupportsPadding).WillRepeatedly(Return(true));
EXPECT_CALL(rtp_2, SupportsRtxPayloadPadding).WillRepeatedly(Return(true)); EXPECT_CALL(rtp_2, SupportsRtxPayloadPadding).WillRepeatedly(Return(true));
EXPECT_CALL(rtp_2, TrySendPacket).WillRepeatedly(Return(false));
EXPECT_CALL(
rtp_2,
TrySendPacket(
::testing::Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc2)), _))
.WillRepeatedly(Return(true));
// Third module is sending media, but does not support rtx. // Third module is sending media, but does not support rtx.
NiceMock<MockRtpRtcp> rtp_3; NiceMock<MockRtpRtcp> rtp_3;
EXPECT_CALL(rtp_3, RtxSendStatus()).WillRepeatedly(Return(kRtxOff));
EXPECT_CALL(rtp_3, SSRC()).WillRepeatedly(Return(kSsrc3)); EXPECT_CALL(rtp_3, SSRC()).WillRepeatedly(Return(kSsrc3));
EXPECT_CALL(rtp_3, SendingMedia()).WillRepeatedly(Return(true));
EXPECT_CALL(rtp_3, SupportsPadding).WillRepeatedly(Return(true)); EXPECT_CALL(rtp_3, SupportsPadding).WillRepeatedly(Return(true));
EXPECT_CALL(rtp_3, SupportsRtxPayloadPadding).WillRepeatedly(Return(true)); EXPECT_CALL(rtp_3, SupportsRtxPayloadPadding).WillRepeatedly(Return(false));
EXPECT_CALL(rtp_3, TrySendPacket).WillRepeatedly(Return(false));
EXPECT_CALL(
rtp_3,
TrySendPacket(
::testing::Pointee(Property(&RtpPacketToSend::Ssrc, kSsrc3)), _))
.WillRepeatedly(Return(true));
packet_router.AddSendRtpModule(&rtp_1, false); packet_router_.AddSendRtpModule(&rtp_1, false);
packet_router.AddSendRtpModule(&rtp_2, false); packet_router_.AddSendRtpModule(&rtp_2, false);
packet_router.AddSendRtpModule(&rtp_3, false); packet_router_.AddSendRtpModule(&rtp_3, false);
const size_t kPaddingBytes = 100; const size_t kPaddingBytes = 100;
// Initially, padding will be sent on last added rtp module that sends media // Initially, padding will be sent on last added rtp module that sends media
// and supports rtx. // and supports rtx.
EXPECT_CALL(rtp_2, TimeToSendPadding(kPaddingBytes, _)) EXPECT_CALL(rtp_2, GeneratePadding(kPaddingBytes))
.Times(1) .Times(1)
.WillOnce(Return(kPaddingBytes)); .WillOnce([](size_t target_size_bytes) {
packet_router.TimeToSendPadding(kPaddingBytes, PacedPacketInfo()); return std::vector<std::unique_ptr<RtpPacketToSend>>();
});
packet_router_.GeneratePadding(kPaddingBytes);
// Send media on first module. Padding should be sent on that module. // Send media on first module. Padding should be sent on that module.
EXPECT_CALL(rtp_1, TimeToSendPacket(kSsrc1, _, _, _, _)); packet_router_.SendPacket(BuildRtpPacket(kSsrc1), PacedPacketInfo());
packet_router.TimeToSendPacket(kSsrc1, 0, 0, false, PacedPacketInfo());
EXPECT_CALL(rtp_1, TimeToSendPadding(kPaddingBytes, _)) EXPECT_CALL(rtp_1, GeneratePadding(kPaddingBytes))
.Times(1) .Times(1)
.WillOnce(Return(kPaddingBytes)); .WillOnce([](size_t target_size_bytes) {
packet_router.TimeToSendPadding(kPaddingBytes, PacedPacketInfo()); return std::vector<std::unique_ptr<RtpPacketToSend>>();
});
packet_router_.GeneratePadding(kPaddingBytes);
// Send media on second module. Padding should be sent there. // Send media on second module. Padding should be sent there.
EXPECT_CALL(rtp_2, TimeToSendPacket(kSsrc2, _, _, _, _)); packet_router_.SendPacket(BuildRtpPacket(kSsrc2), PacedPacketInfo());
packet_router.TimeToSendPacket(kSsrc2, 0, 0, false, PacedPacketInfo());
EXPECT_CALL(rtp_2, TimeToSendPadding(kPaddingBytes, _)) EXPECT_CALL(rtp_2, GeneratePadding(kPaddingBytes))
.Times(1) .Times(1)
.WillOnce(Return(kPaddingBytes)); .WillOnce([](size_t target_size_bytes) {
packet_router.TimeToSendPadding(kPaddingBytes, PacedPacketInfo()); return std::vector<std::unique_ptr<RtpPacketToSend>>();
});
packet_router_.GeneratePadding(kPaddingBytes);
// Remove second module, padding should now fall back to first module. // Remove second module, padding should now fall back to first module.
packet_router.RemoveSendRtpModule(&rtp_2); packet_router_.RemoveSendRtpModule(&rtp_2);
EXPECT_CALL(rtp_1, TimeToSendPadding(kPaddingBytes, _)) EXPECT_CALL(rtp_1, GeneratePadding(kPaddingBytes))
.Times(1) .Times(1)
.WillOnce(Return(kPaddingBytes)); .WillOnce([](size_t target_size_bytes) {
packet_router.TimeToSendPadding(kPaddingBytes, PacedPacketInfo()); return std::vector<std::unique_ptr<RtpPacketToSend>>();
});
packet_router_.GeneratePadding(kPaddingBytes);
// Remove first module too, leaving only the one without rtx. // Remove first module too, leaving only the one without rtx.
packet_router.RemoveSendRtpModule(&rtp_1); packet_router_.RemoveSendRtpModule(&rtp_1);
EXPECT_CALL(rtp_3, TimeToSendPadding(kPaddingBytes, _)) EXPECT_CALL(rtp_3, GeneratePadding(kPaddingBytes))
.Times(1) .Times(1)
.WillOnce(Return(kPaddingBytes)); .WillOnce([](size_t target_size_bytes) {
packet_router.TimeToSendPadding(kPaddingBytes, PacedPacketInfo()); return std::vector<std::unique_ptr<RtpPacketToSend>>();
});
packet_router_.GeneratePadding(kPaddingBytes);
packet_router.RemoveSendRtpModule(&rtp_3); packet_router_.RemoveSendRtpModule(&rtp_3);
} }
TEST(PacketRouterTest, SenderOnlyFunctionsRespectSendingMedia) { TEST_F(PacketRouterTest, AllocateSequenceNumbers) {
PacketRouter packet_router;
NiceMock<MockRtpRtcp> rtp;
packet_router.AddSendRtpModule(&rtp, false);
static const uint16_t kSsrc = 1234;
EXPECT_CALL(rtp, SSRC()).WillRepeatedly(Return(kSsrc));
EXPECT_CALL(rtp, SendingMedia()).WillRepeatedly(Return(false));
// Verify that TimeToSendPacket does not end up in a receiver.
EXPECT_CALL(rtp, TimeToSendPacket(_, _, _, _, _)).Times(0);
EXPECT_EQ(RtpPacketSendResult::kPacketNotFound,
packet_router.TimeToSendPacket(
kSsrc, 1, 1, false,
PacedPacketInfo(PacedPacketInfo::kNotAProbe, kProbeMinBytes,
kProbeMinBytes)));
// Verify that TimeToSendPadding does not end up in a receiver.
EXPECT_CALL(rtp, TimeToSendPadding(_, _)).Times(0);
EXPECT_EQ(0u, packet_router.TimeToSendPadding(
200, PacedPacketInfo(PacedPacketInfo::kNotAProbe,
kProbeMinBytes, kProbeMinBytes)));
packet_router.RemoveSendRtpModule(&rtp);
}
TEST(PacketRouterTest, AllocateSequenceNumbers) {
PacketRouter packet_router;
const uint16_t kStartSeq = 0xFFF0; const uint16_t kStartSeq = 0xFFF0;
const size_t kNumPackets = 32; const size_t kNumPackets = 32;
packet_router.SetTransportWideSequenceNumber(kStartSeq - 1); packet_router_.SetTransportWideSequenceNumber(kStartSeq - 1);
for (size_t i = 0; i < kNumPackets; ++i) { for (size_t i = 0; i < kNumPackets; ++i) {
uint16_t seq = packet_router.AllocateSequenceNumber(); uint16_t seq = packet_router_.AllocateSequenceNumber();
uint32_t expected_unwrapped_seq = static_cast<uint32_t>(kStartSeq) + i; uint32_t expected_unwrapped_seq = static_cast<uint32_t>(kStartSeq) + i;
EXPECT_EQ(static_cast<uint16_t>(expected_unwrapped_seq & 0xFFFF), seq); EXPECT_EQ(static_cast<uint16_t>(expected_unwrapped_seq & 0xFFFF), seq);
} }
} }
TEST(PacketRouterTest, SendTransportFeedback) { TEST_F(PacketRouterTest, SendTransportFeedback) {
PacketRouter packet_router;
NiceMock<MockRtpRtcp> rtp_1; NiceMock<MockRtpRtcp> rtp_1;
NiceMock<MockRtpRtcp> rtp_2; NiceMock<MockRtpRtcp> rtp_2;
packet_router.AddSendRtpModule(&rtp_1, false); packet_router_.AddSendRtpModule(&rtp_1, false);
packet_router.AddReceiveRtpModule(&rtp_2, false); packet_router_.AddReceiveRtpModule(&rtp_2, false);
rtcp::TransportFeedback feedback; rtcp::TransportFeedback feedback;
EXPECT_CALL(rtp_1, SendFeedbackPacket(_)).Times(1).WillOnce(Return(true)); EXPECT_CALL(rtp_1, SendFeedbackPacket(_)).Times(1).WillOnce(Return(true));
packet_router.SendTransportFeedback(&feedback); packet_router_.SendTransportFeedback(&feedback);
packet_router.RemoveSendRtpModule(&rtp_1); packet_router_.RemoveSendRtpModule(&rtp_1);
EXPECT_CALL(rtp_2, SendFeedbackPacket(_)).Times(1).WillOnce(Return(true)); EXPECT_CALL(rtp_2, SendFeedbackPacket(_)).Times(1).WillOnce(Return(true));
packet_router.SendTransportFeedback(&feedback); packet_router_.SendTransportFeedback(&feedback);
packet_router.RemoveReceiveRtpModule(&rtp_2); packet_router_.RemoveReceiveRtpModule(&rtp_2);
}
TEST_F(PacketRouterTest, SendPacketWithoutTransportSequenceNumbers) {
NiceMock<MockRtpRtcp> rtp_1;
packet_router_.AddSendRtpModule(&rtp_1, false);
const uint16_t kSsrc1 = 1234;
ON_CALL(rtp_1, SendingMedia).WillByDefault(Return(true));
ON_CALL(rtp_1, SSRC).WillByDefault(Return(kSsrc1));
// Send a packet without TransportSequenceNumber extension registered,
// packets sent should not have the extension set.
RtpHeaderExtensionMap extension_manager;
auto packet = absl::make_unique<RtpPacketToSend>(&extension_manager);
packet->SetSsrc(kSsrc1);
EXPECT_CALL(
rtp_1,
TrySendPacket(
Property(&RtpPacketToSend::HasExtension<TransportSequenceNumber>,
false),
_))
.WillOnce(Return(true));
packet_router_.SendPacket(std::move(packet), PacedPacketInfo());
packet_router_.RemoveSendRtpModule(&rtp_1);
}
TEST_F(PacketRouterTest, SendPacketAssignsTransportSequenceNumbers) {
NiceMock<MockRtpRtcp> rtp_1;
NiceMock<MockRtpRtcp> rtp_2;
packet_router_.AddSendRtpModule(&rtp_1, false);
packet_router_.AddSendRtpModule(&rtp_2, false);
const uint16_t kSsrc1 = 1234;
const uint16_t kSsrc2 = 2345;
ON_CALL(rtp_1, SSRC).WillByDefault(Return(kSsrc1));
ON_CALL(rtp_2, SSRC).WillByDefault(Return(kSsrc2));
// Transport sequence numbers start at 1, for historical reasons.
uint16_t transport_sequence_number = 1;
auto packet = BuildRtpPacket(kSsrc1);
EXPECT_TRUE(packet->ReserveExtension<TransportSequenceNumber>());
EXPECT_CALL(
rtp_1,
TrySendPacket(
Property(&RtpPacketToSend::GetExtension<TransportSequenceNumber>,
transport_sequence_number),
_))
.WillOnce(Return(true));
packet_router_.SendPacket(std::move(packet), PacedPacketInfo());
++transport_sequence_number;
packet = BuildRtpPacket(kSsrc2);
EXPECT_TRUE(packet->ReserveExtension<TransportSequenceNumber>());
// There will be a failed attempt to send on kSsrc1 before trying
// the correct RTP module.
EXPECT_CALL(rtp_1, TrySendPacket).WillOnce(Return(false));
EXPECT_CALL(
rtp_2,
TrySendPacket(
Property(&RtpPacketToSend::GetExtension<TransportSequenceNumber>,
transport_sequence_number),
_))
.WillOnce(Return(true));
packet_router_.SendPacket(std::move(packet), PacedPacketInfo());
packet_router_.RemoveSendRtpModule(&rtp_1);
packet_router_.RemoveSendRtpModule(&rtp_2);
} }
#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
TEST(PacketRouterTest, DoubleRegistrationOfSendModuleDisallowed) { TEST_F(PacketRouterTest, DoubleRegistrationOfSendModuleDisallowed) {
PacketRouter packet_router;
NiceMock<MockRtpRtcp> module; NiceMock<MockRtpRtcp> module;
constexpr bool remb_candidate = false; // Value irrelevant. constexpr bool remb_candidate = false; // Value irrelevant.
packet_router.AddSendRtpModule(&module, remb_candidate); packet_router_.AddSendRtpModule(&module, remb_candidate);
EXPECT_DEATH(packet_router.AddSendRtpModule(&module, remb_candidate), ""); EXPECT_DEATH(packet_router_.AddSendRtpModule(&module, remb_candidate), "");
// Test tear-down // Test tear-down
packet_router.RemoveSendRtpModule(&module); packet_router_.RemoveSendRtpModule(&module);
} }
TEST(PacketRouterTest, DoubleRegistrationOfReceiveModuleDisallowed) { TEST_F(PacketRouterTest, DoubleRegistrationOfReceiveModuleDisallowed) {
PacketRouter packet_router;
NiceMock<MockRtpRtcp> module; NiceMock<MockRtpRtcp> module;
constexpr bool remb_candidate = false; // Value irrelevant. constexpr bool remb_candidate = false; // Value irrelevant.
packet_router.AddReceiveRtpModule(&module, remb_candidate); packet_router_.AddReceiveRtpModule(&module, remb_candidate);
EXPECT_DEATH(packet_router.AddReceiveRtpModule(&module, remb_candidate), ""); EXPECT_DEATH(packet_router_.AddReceiveRtpModule(&module, remb_candidate), "");
// Test tear-down // Test tear-down
packet_router.RemoveReceiveRtpModule(&module); packet_router_.RemoveReceiveRtpModule(&module);
} }
TEST(PacketRouterTest, RemovalOfNeverAddedSendModuleDisallowed) { TEST_F(PacketRouterTest, RemovalOfNeverAddedSendModuleDisallowed) {
PacketRouter packet_router;
NiceMock<MockRtpRtcp> module; NiceMock<MockRtpRtcp> module;
EXPECT_DEATH(packet_router.RemoveSendRtpModule(&module), ""); EXPECT_DEATH(packet_router_.RemoveSendRtpModule(&module), "");
} }
TEST(PacketRouterTest, RemovalOfNeverAddedReceiveModuleDisallowed) { TEST_F(PacketRouterTest, RemovalOfNeverAddedReceiveModuleDisallowed) {
PacketRouter packet_router;
NiceMock<MockRtpRtcp> module; NiceMock<MockRtpRtcp> module;
EXPECT_DEATH(packet_router.RemoveReceiveRtpModule(&module), ""); EXPECT_DEATH(packet_router_.RemoveReceiveRtpModule(&module), "");
} }
#endif // RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) #endif // RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
@ -931,87 +824,4 @@ TEST(PacketRouterRembTest, ReceiveModuleTakesOverWhenLastSendModuleRemoved) {
packet_router.RemoveReceiveRtpModule(&receive_module); packet_router.RemoveReceiveRtpModule(&receive_module);
} }
TEST(PacketRouterTest, SendPacketWithoutTransportSequenceNumbers) {
PacketRouter packet_router;
NiceMock<MockRtpRtcp> rtp_1;
packet_router.AddSendRtpModule(&rtp_1, false);
const uint16_t kSsrc1 = 1234;
ON_CALL(rtp_1, SendingMedia).WillByDefault(Return(true));
ON_CALL(rtp_1, SSRC).WillByDefault(Return(kSsrc1));
// Send a packet without TransportSequenceNumber extension registered,
// packets sent should not have the extension set.
RtpHeaderExtensionMap extension_manager;
auto packet = absl::make_unique<RtpPacketToSend>(&extension_manager);
packet->SetSsrc(kSsrc1);
EXPECT_CALL(
rtp_1,
TrySendPacket(
Property(&RtpPacketToSend::HasExtension<TransportSequenceNumber>,
false),
_))
.WillOnce(Return(true));
packet_router.SendPacket(std::move(packet), PacedPacketInfo());
packet_router.RemoveSendRtpModule(&rtp_1);
}
TEST(PacketRouterTest, SendPacketAssignsTransportSequenceNumbers) {
PacketRouter packet_router;
NiceMock<MockRtpRtcp> rtp_1;
NiceMock<MockRtpRtcp> rtp_2;
packet_router.AddSendRtpModule(&rtp_1, false);
packet_router.AddSendRtpModule(&rtp_2, false);
const uint16_t kSsrc1 = 1234;
const uint16_t kSsrc2 = 2345;
ON_CALL(rtp_1, SendingMedia).WillByDefault(Return(true));
ON_CALL(rtp_1, SSRC).WillByDefault(Return(kSsrc1));
ON_CALL(rtp_2, SendingMedia).WillByDefault(Return(true));
ON_CALL(rtp_2, SSRC).WillByDefault(Return(kSsrc2));
RtpHeaderExtensionMap extension_manager;
const int kTransportSequenceNumberExtensionId = 1;
extension_manager.Register(kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId);
// Transport sequence numbers start at 1, for historical reasons.
uint16_t transport_sequence_number = 1;
auto packet = absl::make_unique<RtpPacketToSend>(&extension_manager);
EXPECT_TRUE(packet->ReserveExtension<TransportSequenceNumber>());
packet->SetSsrc(kSsrc1);
EXPECT_CALL(
rtp_1,
TrySendPacket(
Property(&RtpPacketToSend::GetExtension<TransportSequenceNumber>,
transport_sequence_number),
_))
.WillOnce(Return(true));
packet_router.SendPacket(std::move(packet), PacedPacketInfo());
++transport_sequence_number;
packet = absl::make_unique<RtpPacketToSend>(&extension_manager);
EXPECT_TRUE(packet->ReserveExtension<TransportSequenceNumber>());
packet->SetSsrc(kSsrc2);
// There will be a failed attempt to send on kSsrc1 before trying
// the correct RTP module.
EXPECT_CALL(rtp_1, TrySendPacket).WillOnce(Return(false));
EXPECT_CALL(
rtp_2,
TrySendPacket(
Property(&RtpPacketToSend::GetExtension<TransportSequenceNumber>,
transport_sequence_number),
_))
.WillOnce(Return(true));
packet_router.SendPacket(std::move(packet), PacedPacketInfo());
packet_router.RemoveSendRtpModule(&rtp_1);
packet_router.RemoveSendRtpModule(&rtp_2);
}
} // namespace webrtc } // namespace webrtc

18
modules/rtp_rtcp/include/rtp_packet_sender.h
View File

@ -18,28 +18,10 @@
namespace webrtc { namespace webrtc {
// TODO(bugs.webrtc.org/10633): Remove Priority and InsertPacket when old pacer
// code path is gone.
class RtpPacketSender { class RtpPacketSender {
public: public:
virtual ~RtpPacketSender() = default; virtual ~RtpPacketSender() = default;
// These are part of the legacy PacedSender interface and will be removed.
enum Priority {
kHighPriority = 0, // Pass through; will be sent immediately.
kNormalPriority = 2, // Put in back of the line.
kLowPriority = 3, // Put in back of the low priority line.
};
// Adds the packet information to the queue and call TimeToSendPacket when
// it's time to send.
virtual void InsertPacket(Priority priority,
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission) = 0;
// Insert packet into queue, for eventual transmission. Based on the type of // Insert packet into queue, for eventual transmission. Based on the type of
// the packet, it will be prioritized and scheduled relative to other packets // the packet, it will be prioritized and scheduled relative to other packets
// and the current target send rate. // and the current target send rate.

10
modules/rtp_rtcp/include/rtp_rtcp.h
View File

@ -282,22 +282,12 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
int payload_type, int payload_type,
bool force_sender_report) = 0; bool force_sender_report) = 0;
virtual RtpPacketSendResult TimeToSendPacket(
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission,
const PacedPacketInfo& pacing_info) = 0;
// Try to send the provided packet. Returns true iff packet matches any of // Try to send the provided packet. Returns true iff packet matches any of
// the SSRCs for this module (media/rtx/fec etc) and was forwarded to the // the SSRCs for this module (media/rtx/fec etc) and was forwarded to the
// transport. // transport.
virtual bool TrySendPacket(RtpPacketToSend* packet, virtual bool TrySendPacket(RtpPacketToSend* packet,
const PacedPacketInfo& pacing_info) = 0; const PacedPacketInfo& pacing_info) = 0;
virtual size_t TimeToSendPadding(size_t bytes,
const PacedPacketInfo& pacing_info) = 0;
virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding( virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
size_t target_size_bytes) = 0; size_t target_size_bytes) = 0;

8
modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
View File

@ -85,17 +85,9 @@ class MockRtpRtcp : public RtpRtcp {
MOCK_CONST_METHOD1(EstimatedReceiveBandwidth, MOCK_CONST_METHOD1(EstimatedReceiveBandwidth,
int(uint32_t* available_bandwidth)); int(uint32_t* available_bandwidth));
MOCK_METHOD4(OnSendingRtpFrame, bool(uint32_t, int64_t, int, bool)); MOCK_METHOD4(OnSendingRtpFrame, bool(uint32_t, int64_t, int, bool));
MOCK_METHOD5(TimeToSendPacket,
RtpPacketSendResult(uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission,
const PacedPacketInfo& pacing_info));
MOCK_METHOD2(TrySendPacket, MOCK_METHOD2(TrySendPacket,
bool(RtpPacketToSend* packet, bool(RtpPacketToSend* packet,
const PacedPacketInfo& pacing_info)); const PacedPacketInfo& pacing_info));
MOCK_METHOD2(TimeToSendPadding,
size_t(size_t bytes, const PacedPacketInfo& pacing_info));
MOCK_METHOD1( MOCK_METHOD1(
GeneratePadding, GeneratePadding,
std::vector<std::unique_ptr<RtpPacketToSend>>(size_t target_size_bytes)); std::vector<std::unique_ptr<RtpPacketToSend>>(size_t target_size_bytes));

2
modules/rtp_rtcp/source/rtp_packet_history.cc
View File

@ -314,7 +314,7 @@ std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetPayloadPaddingPacket(
StoredPacket* best_packet = *best_packet_it; StoredPacket* best_packet = *best_packet_it;
if (best_packet->pending_transmission_) { if (best_packet->pending_transmission_) {
// Because PacedSender releases it's lock when it calls // Because PacedSender releases it's lock when it calls
// TimeToSendPadding() there is the potential for a race where a new // GeneratePadding() there is the potential for a race where a new
// packet ends up here instead of the regular transmit path. In such a // packet ends up here instead of the regular transmit path. In such a
// case, just return empty and it will be picked up on the next // case, just return empty and it will be picked up on the next
// Process() call. // Process() call.

16
modules/rtp_rtcp/source/rtp_rtcp_impl.cc
View File

@ -354,16 +354,6 @@ bool ModuleRtpRtcpImpl::OnSendingRtpFrame(uint32_t timestamp,
return true; return true;
} }
RtpPacketSendResult ModuleRtpRtcpImpl::TimeToSendPacket(
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission,
const PacedPacketInfo& pacing_info) {
return rtp_sender_->TimeToSendPacket(ssrc, sequence_number, capture_time_ms,
retransmission, pacing_info);
}
bool ModuleRtpRtcpImpl::TrySendPacket(RtpPacketToSend* packet, bool ModuleRtpRtcpImpl::TrySendPacket(RtpPacketToSend* packet,
const PacedPacketInfo& pacing_info) { const PacedPacketInfo& pacing_info) {
return rtp_sender_->TrySendPacket(packet, pacing_info); return rtp_sender_->TrySendPacket(packet, pacing_info);
@ -377,12 +367,6 @@ bool ModuleRtpRtcpImpl::SupportsRtxPayloadPadding() const {
return rtp_sender_->SupportsRtxPayloadPadding(); return rtp_sender_->SupportsRtxPayloadPadding();
} }
size_t ModuleRtpRtcpImpl::TimeToSendPadding(
size_t bytes,
const PacedPacketInfo& pacing_info) {
return rtp_sender_->TimeToSendPadding(bytes, pacing_info);
}
std::vector<std::unique_ptr<RtpPacketToSend>> std::vector<std::unique_ptr<RtpPacketToSend>>
ModuleRtpRtcpImpl::GeneratePadding(size_t target_size_bytes) { ModuleRtpRtcpImpl::GeneratePadding(size_t target_size_bytes) {
return rtp_sender_->GeneratePadding(target_size_bytes); return rtp_sender_->GeneratePadding(target_size_bytes);

11
modules/rtp_rtcp/source/rtp_rtcp_impl.h
View File

@ -135,20 +135,9 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
int payload_type, int payload_type,
bool force_sender_report) override; bool force_sender_report) override;
RtpPacketSendResult TimeToSendPacket(
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission,
const PacedPacketInfo& pacing_info) override;
bool TrySendPacket(RtpPacketToSend* packet, bool TrySendPacket(RtpPacketToSend* packet,
const PacedPacketInfo& pacing_info) override; const PacedPacketInfo& pacing_info) override;
// Returns the number of padding bytes actually sent, which can be more or
// less than |bytes|.
size_t TimeToSendPadding(size_t bytes,
const PacedPacketInfo& pacing_info) override;
std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding( std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
size_t target_size_bytes) override; size_t target_size_bytes) override;

24
modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -471,17 +471,6 @@ void RTPSender::OnReceivedNack(
} }
} }
// Called from pacer when we can send the packet.
RtpPacketSendResult RTPSender::TimeToSendPacket(
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission,
const PacedPacketInfo& pacing_info) {
RTC_NOTREACHED();
return RtpPacketSendResult::kSuccess;
}
// Called from pacer when we can send the packet. // Called from pacer when we can send the packet.
bool RTPSender::TrySendPacket(RtpPacketToSend* packet, bool RTPSender::TrySendPacket(RtpPacketToSend* packet,
const PacedPacketInfo& pacing_info) { const PacedPacketInfo& pacing_info) {
@ -715,19 +704,6 @@ void RTPSender::UpdateRtpStats(const RtpPacketToSend& packet,
rtp_stats_callback_->DataCountersUpdated(*counters, packet.Ssrc()); rtp_stats_callback_->DataCountersUpdated(*counters, packet.Ssrc());
} }
size_t RTPSender::TimeToSendPadding(size_t bytes,
const PacedPacketInfo& pacing_info) {
// TODO(bugs.webrtc.org/10633): Remove when downstream test usage is gone.
size_t padding_bytes_sent = 0;
for (auto& packet : GeneratePadding(bytes)) {
const size_t packet_size = packet->payload_size() + packet->padding_size();
if (TrySendPacket(packet.get(), pacing_info)) {
padding_bytes_sent += packet_size;
}
}
return padding_bytes_sent;
}
std::vector<std::unique_ptr<RtpPacketToSend>> RTPSender::GeneratePadding( std::vector<std::unique_ptr<RtpPacketToSend>> RTPSender::GeneratePadding(
size_t target_size_bytes) { size_t target_size_bytes) {
// This method does not actually send packets, it just generates // This method does not actually send packets, it just generates

6
modules/rtp_rtcp/source/rtp_sender.h
View File

@ -111,16 +111,10 @@ class RTPSender {
// Returns an RtpPacketSendResult indicating success, network unavailable, // Returns an RtpPacketSendResult indicating success, network unavailable,
// or packet not found. // or packet not found.
RtpPacketSendResult TimeToSendPacket(uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission,
const PacedPacketInfo& pacing_info);
bool TrySendPacket(RtpPacketToSend* packet, bool TrySendPacket(RtpPacketToSend* packet,
const PacedPacketInfo& pacing_info); const PacedPacketInfo& pacing_info);
bool SupportsPadding() const; bool SupportsPadding() const;
bool SupportsRtxPayloadPadding() const; bool SupportsRtxPayloadPadding() const;
size_t TimeToSendPadding(size_t bytes, const PacedPacketInfo& pacing_info);
std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding( std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
size_t target_size_bytes); size_t target_size_bytes);

44
modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -165,14 +165,6 @@ class MockRtpPacketPacer : public RtpPacketSender {
MOCK_METHOD1(EnqueuePacket, void(std::unique_ptr<RtpPacketToSend>)); MOCK_METHOD1(EnqueuePacket, void(std::unique_ptr<RtpPacketToSend>));
MOCK_METHOD6(InsertPacket,
void(Priority priority,
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission));
MOCK_METHOD2(CreateProbeCluster, void(int bitrate_bps, int cluster_id)); MOCK_METHOD2(CreateProbeCluster, void(int bitrate_bps, int cluster_id));
MOCK_METHOD0(Pause, void()); MOCK_METHOD0(Pause, void());
@ -296,6 +288,15 @@ class RtpSenderTest : public ::testing::TestWithParam<TestConfig> {
return SendPacket(kCaptureTimeMs, sizeof(kPayloadData)); return SendPacket(kCaptureTimeMs, sizeof(kPayloadData));
} }
size_t GenerateAndSendPadding(size_t target_size_bytes) {
size_t generated_bytes = 0;
for (auto& packet : rtp_sender_->GeneratePadding(target_size_bytes)) {
generated_bytes += packet->payload_size() + packet->padding_size();
rtp_sender_->TrySendPacket(packet.get(), PacedPacketInfo());
}
return generated_bytes;
}
// The following are helpers for configuring the RTPSender. They must be // The following are helpers for configuring the RTPSender. They must be
// called before sending any packets. // called before sending any packets.
@ -399,16 +400,16 @@ TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberMayAllowPaddingOnVideo) {
auto packet = rtp_sender_->AllocatePacket(); auto packet = rtp_sender_->AllocatePacket();
ASSERT_TRUE(packet); ASSERT_TRUE(packet);
ASSERT_FALSE(rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo())); ASSERT_TRUE(rtp_sender_->GeneratePadding(kPaddingSize).empty());
packet->SetMarker(false); packet->SetMarker(false);
ASSERT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get())); ASSERT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
// Packet without marker bit doesn't allow padding on video stream. // Packet without marker bit doesn't allow padding on video stream.
EXPECT_FALSE(rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo())); ASSERT_TRUE(rtp_sender_->GeneratePadding(kPaddingSize).empty());
packet->SetMarker(true); packet->SetMarker(true);
ASSERT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get())); ASSERT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
// Packet with marker bit allows send padding. // Packet with marker bit allows send padding.
EXPECT_TRUE(rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo())); ASSERT_FALSE(rtp_sender_->GeneratePadding(kPaddingSize).empty());
} }
TEST_P(RtpSenderTest, AssignSequenceNumberAllowsPaddingOnAudio) { TEST_P(RtpSenderTest, AssignSequenceNumberAllowsPaddingOnAudio) {
@ -434,15 +435,13 @@ TEST_P(RtpSenderTest, AssignSequenceNumberAllowsPaddingOnAudio) {
const size_t kPaddingSize = 59; const size_t kPaddingSize = 59;
EXPECT_CALL(transport, SendRtp(_, kPaddingSize + kRtpHeaderSize, _)) EXPECT_CALL(transport, SendRtp(_, kPaddingSize + kRtpHeaderSize, _))
.WillOnce(Return(true)); .WillOnce(Return(true));
EXPECT_EQ(kPaddingSize, EXPECT_EQ(kPaddingSize, GenerateAndSendPadding(kPaddingSize));
rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo()));
// Requested padding size is too small, will send a larger one. // Requested padding size is too small, will send a larger one.
const size_t kMinPaddingSize = 50; const size_t kMinPaddingSize = 50;
EXPECT_CALL(transport, SendRtp(_, kMinPaddingSize + kRtpHeaderSize, _)) EXPECT_CALL(transport, SendRtp(_, kMinPaddingSize + kRtpHeaderSize, _))
.WillOnce(Return(true)); .WillOnce(Return(true));
EXPECT_EQ(kMinPaddingSize, rtp_sender_->TimeToSendPadding(kMinPaddingSize - 5, EXPECT_EQ(kMinPaddingSize, GenerateAndSendPadding(kMinPaddingSize - 5));
PacedPacketInfo()));
} }
TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberSetPaddingTimestamps) { TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberSetPaddingTimestamps) {
@ -453,11 +452,11 @@ TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberSetPaddingTimestamps) {
packet->SetTimestamp(kTimestamp); packet->SetTimestamp(kTimestamp);
ASSERT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get())); ASSERT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
ASSERT_TRUE(rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo())); auto padding_packets = rtp_sender_->GeneratePadding(kPaddingSize);
ASSERT_EQ(1u, transport_.sent_packets_.size()); ASSERT_EQ(1u, padding_packets.size());
// Verify padding packet timestamp. // Verify padding packet timestamp.
EXPECT_EQ(kTimestamp, transport_.last_sent_packet().Timestamp()); EXPECT_EQ(kTimestamp, padding_packets[0]->Timestamp());
} }
TEST_P(RtpSenderTestWithoutPacer, TEST_P(RtpSenderTestWithoutPacer,
@ -1003,8 +1002,7 @@ TEST_P(RtpSenderTest, SendPadding) {
const size_t kPaddingBytes = 100; const size_t kPaddingBytes = 100;
const size_t kMaxPaddingLength = 224; // Value taken from rtp_sender.cc. const size_t kMaxPaddingLength = 224; // Value taken from rtp_sender.cc.
// Padding will be forced to full packets. // Padding will be forced to full packets.
EXPECT_EQ(kMaxPaddingLength, EXPECT_EQ(kMaxPaddingLength, GenerateAndSendPadding(kPaddingBytes));
rtp_sender_->TimeToSendPadding(kPaddingBytes, PacedPacketInfo()));
// Process send bucket. Padding should now be sent. // Process send bucket. Padding should now be sent.
EXPECT_EQ(++total_packets_sent, transport_.packets_sent()); EXPECT_EQ(++total_packets_sent, transport_.packets_sent());
@ -1878,7 +1876,7 @@ TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
callback.Matches(ssrc, expected); callback.Matches(ssrc, expected);
// Send padding. // Send padding.
rtp_sender_->TimeToSendPadding(kMaxPaddingSize, PacedPacketInfo()); GenerateAndSendPadding(kMaxPaddingSize);
expected.transmitted.payload_bytes = 12; expected.transmitted.payload_bytes = 12;
expected.transmitted.header_bytes = 36; expected.transmitted.header_bytes = 36;
expected.transmitted.padding_bytes = kMaxPaddingSize; expected.transmitted.padding_bytes = kMaxPaddingSize;
@ -1913,8 +1911,8 @@ TEST_P(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
SendGenericPacket(); SendGenericPacket();
// Will send 2 full-size padding packets. // Will send 2 full-size padding packets.
rtp_sender_->TimeToSendPadding(1, PacedPacketInfo()); GenerateAndSendPadding(1);
rtp_sender_->TimeToSendPadding(1, PacedPacketInfo()); GenerateAndSendPadding(1);
StreamDataCounters rtp_stats; StreamDataCounters rtp_stats;
StreamDataCounters rtx_stats; StreamDataCounters rtx_stats;