webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc - Issue 2131913003: Revert of Refactor NACK bitrate allocation

Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

Issue 2131913003: Revert of Refactor NACK bitrate allocation (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Created 4 years, 5 months ago

Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.

Jump to:

View side-by-side diff with in-line comments

Download patch

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

index 99cef009e733a38b9727ddeeb0305951005751dd..f0b6411af2711ffc3d2d39c446774268e5ecebec 100644

--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

@@ -15,7 +15,6 @@

#include "testing/gmock/include/gmock/gmock.h"

#include "testing/gtest/include/gtest/gtest.h"

#include "webrtc/base/buffer.h"

-#include "webrtc/base/rate_limiter.h"

#include "webrtc/call/mock/mock_rtc_event_log.h"

#include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"

#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"

@@ -134,11 +133,11 @@

: fake_clock_(kStartTime),

mock_rtc_event_log_(),

mock_paced_sender_(),

- retransmission_rate_limiter_(&fake_clock_, 1000),

rtp_sender_(),

payload_(kPayload),

transport_(),

- kMarkerBit(true) {}

+ kMarkerBit(true) {

+ }

void SetUp() override { SetUpRtpSender(true); }

@@ -146,8 +145,7 @@

rtp_sender_.reset(new RTPSender(

false, &fake_clock_, &transport_, pacer ? &mock_paced_sender_ : nullptr,

&seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,

- &mock_rtc_event_log_, &send_packet_observer_,

- &retransmission_rate_limiter_));

+ &mock_rtc_event_log_, &send_packet_observer_));

rtp_sender_->SetSequenceNumber(kSeqNum);

}

@@ -156,7 +154,6 @@

MockRtpPacketSender mock_paced_sender_;

MockTransportSequenceNumberAllocator seq_num_allocator_;

MockSendPacketObserver send_packet_observer_;

- RateLimiter retransmission_rate_limiter_;

std::unique_ptr<RTPSender> rtp_sender_;

int payload_;

LoopbackTransportTest transport_;

@@ -746,6 +743,7 @@

EXPECT_EQ(

0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,

kAbsoluteSendTimeExtensionId));

+ rtp_sender_->SetTargetBitrate(300000);

int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();

int rtp_length_int = rtp_sender_->BuildRTPheader(

packet_, kPayload, kMarkerBit, kTimestamp, capture_time_ms);

@@ -799,6 +797,7 @@

EXPECT_EQ(

0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,

kAbsoluteSendTimeExtensionId));

+ rtp_sender_->SetTargetBitrate(300000);

int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();

int rtp_length_int = rtp_sender_->BuildRTPheader(

packet_, kPayload, kMarkerBit, kTimestamp, capture_time_ms);

@@ -880,6 +879,7 @@

kAbsoluteSendTimeExtensionId);

webrtc::RTPHeader rtp_header;

+ rtp_sender_->SetTargetBitrate(300000);

int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();

int rtp_length_int = rtp_sender_->BuildRTPheader(

packet_, kPayload, kMarkerBit, timestamp, capture_time_ms);

@@ -1011,7 +1011,7 @@

rtp_sender_.reset(new RTPSender(

false, &fake_clock_, &transport_, &mock_paced_sender_,

nullptr /* TransportSequenceNumberAllocator */, nullptr, nullptr, nullptr,

- nullptr, nullptr, &send_packet_observer_, nullptr));

+ nullptr, nullptr, &send_packet_observer_));

rtp_sender_->SetSequenceNumber(kSeqNum);

rtp_sender_->SetStorePacketsStatus(true, 10);

@@ -1029,7 +1029,7 @@

MockTransport transport;

rtp_sender_.reset(new RTPSender(

false, &fake_clock_, &transport, &mock_paced_sender_, nullptr, nullptr,

- nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr, nullptr));

+ nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr));

rtp_sender_->SetSequenceNumber(kSeqNum);

rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);

@@ -1054,6 +1054,7 @@

kTransmissionTimeOffsetExtensionId);

rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,

kAbsoluteSendTimeExtensionId);

+ rtp_sender_->SetTargetBitrate(300000);

const size_t kNumPayloadSizes = 10;

const size_t kPayloadSizes[kNumPayloadSizes] = {500, 550, 600, 650, 700,

750, 800, 850, 900, 950};

@@ -1175,7 +1176,7 @@

rtp_sender_.reset(new RTPSender(

false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr, nullptr,

- nullptr, &callback, nullptr, nullptr, nullptr, nullptr));

+ nullptr, &callback, nullptr, nullptr, nullptr));

char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";

const uint8_t payload_type = 127;

@@ -1212,39 +1213,30 @@

TEST_F(RtpSenderTest, BitrateCallbacks) {

class TestCallback : public BitrateStatisticsObserver {

public:

- TestCallback()

- : BitrateStatisticsObserver(),

- num_calls_(0),

- ssrc_(0),

- total_bitrate_(0),

- retransmit_bitrate_(0) {}

+ TestCallback() : BitrateStatisticsObserver(), num_calls_(0), ssrc_(0) {}

virtual ~TestCallback() {}

- void Notify(uint32_t total_bitrate,

- uint32_t retransmit_bitrate,

+ void Notify(const BitrateStatistics& total_stats,

+ const BitrateStatistics& retransmit_stats,

uint32_t ssrc) override {

++num_calls_;

ssrc_ = ssrc;

- total_bitrate_ = total_bitrate;

- retransmit_bitrate_ = retransmit_bitrate;

+ total_stats_ = total_stats;

+ retransmit_stats_ = retransmit_stats;

}

uint32_t num_calls_;

uint32_t ssrc_;

- uint32_t total_bitrate_;

- uint32_t retransmit_bitrate_;

+ BitrateStatistics total_stats_;

+ BitrateStatistics retransmit_stats_;

} callback;

rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,

nullptr, nullptr, &callback, nullptr, nullptr,

- nullptr, nullptr, nullptr));

- // Simulate kNumPackets sent with kPacketInterval ms intervals, with the

- // number of packets selected so that we fill (but don't overflow) the one

- // second averaging window.

- const uint32_t kWindowSizeMs = 1000;

+ nullptr, nullptr));

+ // Simulate kNumPackets sent with kPacketInterval ms intervals.

+ const uint32_t kNumPackets = 15;

const uint32_t kPacketInterval = 20;

- const uint32_t kNumPackets =

- (kWindowSizeMs - kPacketInterval) / kPacketInterval;

// Overhead = 12 bytes RTP header + 1 byte generic header.

const uint32_t kPacketOverhead = 13;

@@ -1258,6 +1250,7 @@

// Initial process call so we get a new time window.

rtp_sender_->ProcessBitrate();

+ uint64_t start_time = fake_clock_.CurrentNtpInMilliseconds();

// Send a few frames.

for (uint32_t i = 0; i < kNumPackets; ++i) {

@@ -1269,18 +1262,17 @@

rtp_sender_->ProcessBitrate();

+ const uint32_t expected_packet_rate = 1000 / kPacketInterval;

// We get one call for every stats updated, thus two calls since both the

// stream stats and the retransmit stats are updated once.

EXPECT_EQ(2u, callback.num_calls_);

EXPECT_EQ(ssrc, callback.ssrc_);

- const uint32_t kTotalPacketSize = kPacketOverhead + sizeof(payload);

- // Bitrate measured over delta between last and first timestamp, plus one.

- const uint32_t kExpectedWindowMs = kNumPackets * kPacketInterval + 1;

- const uint32_t kExpectedBitsAccumulated = kTotalPacketSize * kNumPackets * 8;

- const uint32_t kExpectedRateBps =

- (kExpectedBitsAccumulated * 1000 + (kExpectedWindowMs / 2)) /

- kExpectedWindowMs;

- EXPECT_EQ(kExpectedRateBps, callback.total_bitrate_);

+ EXPECT_EQ(start_time + (kNumPackets * kPacketInterval),

+ callback.total_stats_.timestamp_ms);

+ EXPECT_EQ(expected_packet_rate, callback.total_stats_.packet_rate);

+ EXPECT_EQ((kPacketOverhead + sizeof(payload)) * 8 * expected_packet_rate,

+ callback.total_stats_.bitrate_bps);

rtp_sender_.reset();

}

@@ -1293,7 +1285,7 @@

payload_ = kAudioPayload;

rtp_sender_.reset(new RTPSender(true, &fake_clock_, &transport_, nullptr,

nullptr, nullptr, nullptr, nullptr, nullptr,

- nullptr, nullptr, nullptr));

+ nullptr, nullptr));

rtp_sender_->SetSequenceNumber(kSeqNum);

}

};

@@ -1561,9 +1553,9 @@

const int32_t kPacketSize = 1400;

const int32_t kNumPackets = 30;

- retransmission_rate_limiter_.SetMaxRate(kPacketSize * kNumPackets * 8);

rtp_sender_->SetStorePacketsStatus(true, kNumPackets);

+ // Set bitrate (in kbps) to fit kNumPackets á kPacketSize bytes in one second.

+ rtp_sender_->SetTargetBitrate(kNumPackets * kPacketSize * 8);

const uint16_t kStartSequenceNumber = rtp_sender_->SequenceNumber();

std::list<uint16_t> sequence_numbers;

for (int32_t i = 0; i < kNumPackets; ++i) {

@@ -1581,9 +1573,6 @@

rtp_sender_->OnReceivedNACK(sequence_numbers, 0);

EXPECT_EQ(kNumPackets * 2, transport_.packets_sent_);

- // Must be at least 5ms in between retransmission attempts.

- fake_clock_.AdvanceTimeMilliseconds(5);

// Resending should not work, bandwidth exceeded.

rtp_sender_->OnReceivedNACK(sequence_numbers, 0);

EXPECT_EQ(kNumPackets * 2, transport_.packets_sent_);

« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtp_sender.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_sender_video.h » ('j') | no next file with comments »