Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(578)

Issues Search
    My Issues | Starred     Open | Closed | All

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

Issue 1247293002: Add support for transport wide sequence numbers (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase, again Created 5 years, 4 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
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtp_sender.cc ('k') | webrtc/test/frame_generator_capturer.h » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
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 23300bbff605cb1abbc8833d5a1d3f4170c4251d..ea6fb6adf3be5229cd08008a0a1c1f09b650c876 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -73,7 +73,7 @@ class LoopbackTransportTest : public webrtc::Transport {
: packets_sent_(0),
last_sent_packet_len_(0),
total_bytes_sent_(0),
- last_sent_packet_(NULL) {}
+ last_sent_packet_(nullptr) {}
~LoopbackTransportTest() {
STLDeleteContainerPointers(sent_packets_.begin(), sent_packets_.end());
@@ -114,8 +114,9 @@ class RtpSenderTest : public ::testing::Test {
}
void SetUp() override {
- rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
- &mock_paced_sender_, NULL, NULL, NULL));
+ rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_,
+ nullptr, &mock_paced_sender_, nullptr,
+ nullptr, nullptr, nullptr, nullptr));
rtp_sender_->SetSequenceNumber(kSeqNum);
}
@@ -308,7 +309,7 @@ TEST_F(RtpSenderTest, BuildRTPPacket) {
webrtc::RtpUtility::RtpHeaderParser rtp_parser(packet_, length);
webrtc::RTPHeader rtp_header;
- const bool valid_rtp_header = rtp_parser.Parse(rtp_header, NULL);
+ const bool valid_rtp_header = rtp_parser.Parse(rtp_header, nullptr);
ASSERT_TRUE(valid_rtp_header);
ASSERT_FALSE(rtp_parser.RTCP());
@@ -351,7 +352,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithTransmissionOffsetExtension) {
// Parse without map extension
webrtc::RTPHeader rtp_header2;
- const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, NULL);
+ const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
ASSERT_TRUE(valid_rtp_header2);
VerifyRTPHeaderCommon(rtp_header2);
@@ -415,7 +416,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAbsoluteSendTimeExtension) {
// Parse without map extension
webrtc::RTPHeader rtp_header2;
- const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, NULL);
+ const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
ASSERT_TRUE(valid_rtp_header2);
VerifyRTPHeaderCommon(rtp_header2);
@@ -509,7 +510,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithAudioLevelExtension) {
// Parse without map extension
webrtc::RTPHeader rtp_header2;
- const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, NULL);
+ const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
ASSERT_TRUE(valid_rtp_header2);
VerifyRTPHeaderCommon(rtp_header2);
@@ -571,7 +572,7 @@ TEST_F(RtpSenderTest, BuildRTPPacketWithHeaderExtensions) {
// Parse without map extension
webrtc::RTPHeader rtp_header2;
- const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, NULL);
+ const bool valid_rtp_header2 = rtp_parser.Parse(rtp_header2, nullptr);
ASSERT_TRUE(valid_rtp_header2);
VerifyRTPHeaderCommon(rtp_header2);
@@ -724,7 +725,7 @@ TEST_F(RtpSenderTest, SendPadding) {
// Create and set up parser.
rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(
webrtc::RtpHeaderParser::Create());
- ASSERT_TRUE(rtp_parser.get() != NULL);
+ ASSERT_TRUE(rtp_parser.get() != nullptr);
rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId);
rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
@@ -822,8 +823,9 @@ TEST_F(RtpSenderTest, SendPadding) {
TEST_F(RtpSenderTest, SendRedundantPayloads) {
MockTransport transport;
- rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport, NULL,
- &mock_paced_sender_, NULL, NULL, NULL));
+ rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport, nullptr,
+ &mock_paced_sender_, nullptr, nullptr,
+ nullptr, nullptr, nullptr));
rtp_sender_->SetSequenceNumber(kSeqNum);
rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
// Make all packets go through the pacer.
@@ -845,7 +847,7 @@ TEST_F(RtpSenderTest, SendRedundantPayloads) {
// Create and set up parser.
rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(
webrtc::RtpHeaderParser::Create());
- ASSERT_TRUE(rtp_parser.get() != NULL);
+ ASSERT_TRUE(rtp_parser.get() != nullptr);
rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset,
kTransmissionTimeOffsetExtensionId);
rtp_parser->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
@@ -891,9 +893,9 @@ TEST_F(RtpSenderTest, SendGenericVideo) {
uint8_t payload[] = {47, 11, 32, 93, 89};
// Send keyframe
- ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
- 4321, payload, sizeof(payload),
- NULL));
+ ASSERT_EQ(
+ 0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,
+ payload, sizeof(payload), nullptr));
RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
@@ -919,7 +921,7 @@ TEST_F(RtpSenderTest, SendGenericVideo) {
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
1234, 4321, payload,
- sizeof(payload), NULL));
+ sizeof(payload), nullptr));
RtpUtility::RtpHeaderParser rtp_parser2(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
@@ -955,8 +957,9 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
FrameCounts frame_counts_;
} callback;
- rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
- &mock_paced_sender_, NULL, &callback, NULL));
+ rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, nullptr,
+ &mock_paced_sender_, nullptr, nullptr,
+ nullptr, &callback, nullptr));
char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
const uint8_t payload_type = 127;
@@ -966,18 +969,18 @@ TEST_F(RtpSenderTest, FrameCountCallbacks) {
rtp_sender_->SetStorePacketsStatus(true, 1);
uint32_t ssrc = rtp_sender_->SSRC();
- ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
- 4321, payload, sizeof(payload),
- NULL));
+ ASSERT_EQ(
+ 0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,
+ payload, sizeof(payload), nullptr));
EXPECT_EQ(1U, callback.num_calls_);
EXPECT_EQ(ssrc, callback.ssrc_);
EXPECT_EQ(1, callback.frame_counts_.key_frames);
EXPECT_EQ(0, callback.frame_counts_.delta_frames);
- ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta,
- payload_type, 1234, 4321, payload,
- sizeof(payload), NULL));
+ ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
+ 1234, 4321, payload,
+ sizeof(payload), nullptr));
EXPECT_EQ(2U, callback.num_calls_);
EXPECT_EQ(ssrc, callback.ssrc_);
@@ -1007,8 +1010,9 @@ TEST_F(RtpSenderTest, BitrateCallbacks) {
BitrateStatistics total_stats_;
BitrateStatistics retransmit_stats_;
} callback;
- rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, NULL,
- &mock_paced_sender_, &callback, NULL, NULL));
+ rtp_sender_.reset(new RTPSender(0, false, &fake_clock_, &transport_, nullptr,
+ &mock_paced_sender_, nullptr, nullptr,
+ &callback, nullptr, nullptr));
// Simulate kNumPackets sent with kPacketInterval ms intervals.
const uint32_t kNumPackets = 15;
@@ -1065,8 +1069,9 @@ class RtpSenderAudioTest : public RtpSenderTest {
void SetUp() override {
payload_ = kAudioPayload;
- rtp_sender_.reset(new RTPSender(0, true, &fake_clock_, &transport_, NULL,
- &mock_paced_sender_, NULL, NULL, NULL));
+ rtp_sender_.reset(new RTPSender(0, true, &fake_clock_, &transport_, nullptr,
+ &mock_paced_sender_, nullptr, nullptr,
+ nullptr, nullptr, nullptr));
rtp_sender_->SetSequenceNumber(kSeqNum);
}
};
@@ -1117,9 +1122,9 @@ TEST_F(RtpSenderTest, StreamDataCountersCallbacks) {
rtp_sender_->RegisterRtpStatisticsCallback(&callback);
// Send a frame.
- ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
- 4321, payload, sizeof(payload),
- NULL));
+ ASSERT_EQ(
+ 0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,
+ payload, sizeof(payload), nullptr));
StreamDataCounters expected;
expected.transmitted.payload_bytes = 6;
expected.transmitted.header_bytes = 12;
@@ -1162,14 +1167,14 @@ TEST_F(RtpSenderTest, StreamDataCountersCallbacks) {
rtp_sender_->SetFecParameters(&fec_params, &fec_params);
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kVideoFrameDelta, payload_type,
1234, 4321, payload,
- sizeof(payload), NULL));
+ sizeof(payload), nullptr));
expected.transmitted.payload_bytes = 40;
expected.transmitted.header_bytes = 60;
expected.transmitted.packets = 5;
expected.fec.packets = 1;
callback.Matches(ssrc, expected);
- rtp_sender_->RegisterRtpStatisticsCallback(NULL);
+ rtp_sender_->RegisterRtpStatisticsCallback(nullptr);
}
TEST_F(RtpSenderAudioTest, SendAudio) {
@@ -1179,9 +1184,9 @@ TEST_F(RtpSenderAudioTest, SendAudio) {
0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
- ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234,
- 4321, payload, sizeof(payload),
- NULL));
+ ASSERT_EQ(
+ 0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234, 4321,
+ payload, sizeof(payload), nullptr));
RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
@@ -1208,9 +1213,9 @@ TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
0, 1500));
uint8_t payload[] = {47, 11, 32, 93, 89};
- ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234,
- 4321, payload, sizeof(payload),
- NULL));
+ ASSERT_EQ(
+ 0, rtp_sender_->SendOutgoingData(kAudioFrameCN, payload_type, 1234, 4321,
+ payload, sizeof(payload), nullptr));
RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
transport_.last_sent_packet_len_);
@@ -1259,19 +1264,17 @@ TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
// The duration is calculated as the difference of current and last sent
// timestamp. So for first call it will skip since the duration is zero.
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kFrameEmpty, payload_type,
- capture_time_ms,
- 0, NULL, 0,
- NULL));
+ capture_time_ms, 0, nullptr, 0,
+ nullptr));
// DTMF Sample Length is (Frequency/1000) * Duration.
// So in this case, it is (8000/1000) * 500 = 4000.
// Sending it as two packets.
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kFrameEmpty, payload_type,
- capture_time_ms+2000,
- 0, NULL, 0,
- NULL));
+ capture_time_ms + 2000, 0, nullptr,
+ 0, nullptr));
rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_parser(
webrtc::RtpHeaderParser::Create());
- ASSERT_TRUE(rtp_parser.get() != NULL);
+ ASSERT_TRUE(rtp_parser.get() != nullptr);
webrtc::RTPHeader rtp_header;
ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_,
transport_.last_sent_packet_len_,
@@ -1280,9 +1283,8 @@ TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
EXPECT_TRUE(rtp_header.markerBit);
ASSERT_EQ(0, rtp_sender_->SendOutgoingData(kFrameEmpty, payload_type,
- capture_time_ms+4000,
- 0, NULL, 0,
- NULL));
+ capture_time_ms + 4000, 0, nullptr,
+ 0, nullptr));
ASSERT_TRUE(rtp_parser->Parse(transport_.last_sent_packet_,
transport_.last_sent_packet_len_,
&rtp_header));
@@ -1357,7 +1359,7 @@ TEST_F(RtpSenderVideoTest, SendVideoWithCVO) {
rtp_sender_->RtpHeaderExtensionTotalLength());
rtp_sender_video_->SendVideo(kRtpVideoGeneric, kVideoFrameKey, kPayload,
- kTimestamp, 0, packet_, sizeof(packet_), NULL,
+ kTimestamp, 0, packet_, sizeof(packet_), nullptr,
&hdr);
RtpHeaderExtensionMap map;
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtp_sender.cc ('k') | webrtc/test/frame_generator_capturer.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698