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Add transport sequence number on the non-pacer path of the rtp sender.

Browse Source 
BUG=4173
R=sprang@webrtc.org

Review URL: https://codereview.webrtc.org/1635093002 .

Cr-Commit-Position: refs/heads/master@{#11395}
This commit is contained in:
Stefan Holmer
2016-01-27 12:58:51 +01:00
parent 1c3909899d
commit f5dca48dc0
2 changed files with 75 additions and 8 deletions
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28
webrtc/modules/rtp_rtcp/source/rtp_sender.cc
View File

@ -642,11 +642,15 @@ size_t RTPSender::SendPadData(size_t bytes,
payload_type = payload_type_;
over_rtx = false;
} else {
// Without abs-send-time a media packet must be sent before padding so
// that the timestamps used for estimation are correct.
if (!media_has_been_sent_ && !rtp_header_extension_map_.IsRegistered(
kRtpExtensionAbsoluteSendTime))
// Without abs-send-time or transport sequence number a media packet
// must be sent before padding so that the timestamps used for
// estimation are correct.
if (!media_has_been_sent_ &&
!(rtp_header_extension_map_.IsRegistered(
kRtpExtensionAbsoluteSendTime) ||
using_transport_seq)) {
return 0;
}
// Only change change the timestamp of padding packets sent over RTX.
// Padding only packets over RTP has to be sent as part of a media
// frame (and therefore the same timestamp).
@ -1079,7 +1083,21 @@ int32_t RTPSender::SendToNetwork(uint8_t* buffer,
UpdateDelayStatistics(capture_time_ms, now_ms);
}
bool sent = SendPacketToNetwork(buffer, length, PacketOptions());
// TODO(sprang): Potentially too much overhead in IsRegistered()?
bool using_transport_seq = rtp_header_extension_map_.IsRegistered(
kRtpExtensionTransportSequenceNumber) &&
transport_sequence_number_allocator_;
PacketOptions options;
if (using_transport_seq) {
options.packet_id =
UpdateTransportSequenceNumber(buffer, length, rtp_header);
if (transport_feedback_observer_) {
transport_feedback_observer_->AddPacket(options.packet_id, length, true);
}
}
bool sent = SendPacketToNetwork(buffer, length, options);
// Mark the packet as sent in the history even if send failed. Dropping a
// packet here should be treated as any other packet drop so we should be

55
webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
View File

@ -76,7 +76,8 @@ class LoopbackTransportTest : public webrtc::Transport {
: packets_sent_(0),
last_sent_packet_len_(0),
total_bytes_sent_(0),
last_sent_packet_(nullptr) {}
last_sent_packet_(nullptr),
last_packet_id_(-1) {}
~LoopbackTransportTest() {
STLDeleteContainerPointers(sent_packets_.begin(), sent_packets_.end());
@ -89,6 +90,7 @@ class LoopbackTransportTest : public webrtc::Transport {
new rtc::Buffer(reinterpret_cast<const uint8_t*>(data), len);
last_sent_packet_ = buffer->data();
last_sent_packet_len_ = len;
last_packet_id_ = options.packet_id;
total_bytes_sent_ += len;
sent_packets_.push_back(buffer);
return true;
@ -98,6 +100,7 @@ class LoopbackTransportTest : public webrtc::Transport {
size_t last_sent_packet_len_;
size_t total_bytes_sent_;
uint8_t* last_sent_packet_;
int last_packet_id_;
std::vector<rtc::Buffer*> sent_packets_;
};
@ -117,6 +120,12 @@ class MockRtpPacketSender : public RtpPacketSender {
bool retransmission));
};
class MockTransportSequenceNumberAllocator
: public TransportSequenceNumberAllocator {
public:
MOCK_METHOD0(AllocateSequenceNumber, uint16_t());
};
class RtpSenderTest : public ::testing::Test {
protected:
RtpSenderTest()
@ -134,14 +143,15 @@ class RtpSenderTest : public ::testing::Test {
void SetUpRtpSender(bool pacer) {
rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
pacer ? &mock_paced_sender_ : nullptr,
nullptr, nullptr, nullptr, nullptr, nullptr,
&mock_rtc_event_log_));
&seq_num_allocator_, nullptr, nullptr,
nullptr, nullptr, &mock_rtc_event_log_));
rtp_sender_->SetSequenceNumber(kSeqNum);
}
SimulatedClock fake_clock_;
MockRtcEventLog mock_rtc_event_log_;
MockRtpPacketSender mock_paced_sender_;
MockTransportSequenceNumberAllocator seq_num_allocator_;
rtc::scoped_ptr<RTPSender> rtp_sender_;
int payload_;
LoopbackTransportTest transport_;
@ -464,6 +474,45 @@ TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithAbsoluteSendTimeExtension) {
EXPECT_EQ(0u, rtp_header2.extension.absoluteSendTime);
}
TEST_F(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
// Ignore rtc event calls.
EXPECT_CALL(mock_rtc_event_log_,
LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _));
EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId));
char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
const uint8_t payload_type = 127;
ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
0, 1500));
// Create a dummy payload of 5 bytes.
uint8_t payload[] = {47, 11, 32, 93, 89};
const uint16_t kTransportSequenceNumber = 17;
EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
.WillOnce(testing::Return(kTransportSequenceNumber));
const uint32_t kTimestamp = 1234;
const int64_t kCaptureTimeMs = 4321;
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(
kVideoFrameKey, payload_type, kTimestamp, kCaptureTimeMs,
payload, sizeof(payload), nullptr));
RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
webrtc::RTPHeader rtp_header;
RtpHeaderExtensionMap map;
map.Register(kRtpExtensionTransportSequenceNumber,
kTransportSequenceNumberExtensionId);
EXPECT_TRUE(rtp_parser.Parse(&rtp_header, &map));
EXPECT_TRUE(rtp_header.extension.hasTransportSequenceNumber);
EXPECT_EQ(kTransportSequenceNumber,
rtp_header.extension.transportSequenceNumber);
EXPECT_EQ(transport_.last_packet_id_,
rtp_header.extension.transportSequenceNumber);
}
// Test CVO header extension is only set when marker bit is true.
TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithVideoRotation_MarkerBit) {
rtp_sender_->SetVideoRotation(kRotation);