Add flag enabling more packets to be retransmittable.

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

 /*
  *  Copyright (c) 2012 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 <memory>
 #include <vector>
 
 #include "webrtc/api/video/video_timing.h"
 #include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_cvo.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender_video.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/rtc_base/buffer.h"
 #include "webrtc/rtc_base/rate_limiter.h"
 #include "webrtc/test/field_trial.h"
 #include "webrtc/test/gmock.h"
@@ skipping 16 matching lines @@
 const int kMaxPacketLength = 1500;
 const uint8_t kAudioLevel = 0x5a;
 const uint16_t kTransportSequenceNumber = 0xaabbu;
 const uint8_t kAudioLevelExtensionId = 9;
 const int kAudioPayload = 103;
 const uint64_t kStartTime = 123456789;
 const size_t kMaxPaddingSize = 224u;
 const int kVideoRotationExtensionId = 5;
 const size_t kGenericHeaderLength = 1;
 const uint8_t kPayloadData[] = {47, 11, 32, 93, 89};
+const int64_t kDefaultExpectedRetransmissionTimeMs = 125;
 
 using ::testing::_;
 using ::testing::ElementsAreArray;
 using ::testing::Invoke;
 
 uint64_t ConvertMsToAbsSendTime(int64_t time_ms) {
   return (((time_ms << 18) + 500) / 1000) & 0x00ffffff;
 }
 
 class LoopbackTransportTest : public webrtc::Transport {
@@ skipping 163 matching lines @@
   void SendGenericPayload() {
     const uint32_t kTimestamp = 1234;
     const uint8_t kPayloadType = 127;
     const int64_t kCaptureTimeMs = fake_clock_.TimeInMilliseconds();
     char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
     EXPECT_EQ(0, rtp_sender_->RegisterPayload(payload_name, kPayloadType, 90000,
                                               0, 1500));
 
     EXPECT_TRUE(rtp_sender_->SendOutgoingData(
         kVideoFrameKey, kPayloadType, kTimestamp, kCaptureTimeMs, kPayloadData,
-        sizeof(kPayloadData), nullptr, nullptr, nullptr));
+        sizeof(kPayloadData), nullptr, nullptr, nullptr,
+        kDefaultExpectedRetransmissionTimeMs));
   }
 };
 
 // TODO(pbos): Move tests over from WithoutPacer to RtpSenderTest as this is our
 // default code path.
 class RtpSenderTestWithoutPacer : public RtpSenderTest {
  public:
   void SetUp() override { SetUpRtpSender(false); }
 };
 
+class TestRtpSenderVideo : public RTPSenderVideo {
+ public:
+  TestRtpSenderVideo(Clock* clock,
+                     RTPSender* rtpSender,

[danilchap, 2017/08/29 17:31:19]

rtp_sender

[sprang_webrtc, 2017/08/31 15:54:29]

Done.

+                     FlexfecSender* flexfec_sender)
+      : RTPSenderVideo(clock, rtpSender, flexfec_sender) {}
+  ~TestRtpSenderVideo() override {}
+
+  StorageType GetStorageType(const RTPVideoHeader& header,
+                             int32_t retransmission_settings,
+                             int64_t expected_retransmission_time,
+                             FrameType frame_type) {
+    std::unique_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create(

[danilchap, 2017/08/29 17:31:19]

or
auto packetizer = rtc::WrapUnique(RtpPacketizer::Create(

[sprang_webrtc, 2017/08/31 15:54:29]

Done.

+        header.codec, kMaxPacketLength, 0, &header.codecHeader, frame_type));
+
+    return RTPSenderVideo::GetStorageType(header, retransmission_settings,
+                                          packetizer.get(),
+                                          expected_retransmission_time);
+  }
+};
+
 class RtpSenderVideoTest : public RtpSenderTest {
  protected:
   void SetUp() override {
     // TODO(pbos): Set up to use pacer.
     SetUpRtpSender(false);
     rtp_sender_video_.reset(
-        new RTPSenderVideo(&fake_clock_, rtp_sender_.get(), nullptr));
+        new TestRtpSenderVideo(&fake_clock_, rtp_sender_.get(), nullptr));
   }
-  std::unique_ptr<RTPSenderVideo> rtp_sender_video_;
+  std::unique_ptr<TestRtpSenderVideo> rtp_sender_video_;
 };
 
 TEST_P(RtpSenderTestWithoutPacer, AllocatePacketSetCsrc) {
   // Configure rtp_sender with csrc.
   std::vector<uint32_t> csrcs;
   csrcs.push_back(0x23456789);
   rtp_sender_->SetCsrcs(csrcs);
 
   auto packet = rtp_sender_->AllocatePacket();
 
@@ skipping 583 matching lines @@
 }
 
 TEST_P(RtpSenderTestWithoutPacer, SendGenericVideo) {
   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));
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
   // Send keyframe
-  ASSERT_TRUE(rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
-                                            4321, payload, sizeof(payload),
-                                            nullptr, nullptr, nullptr));
+  ASSERT_TRUE(rtp_sender_->SendOutgoingData(
+      kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
+      nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
 
   auto sent_payload = transport_.last_sent_packet().payload();
   uint8_t generic_header = sent_payload[0];
   EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kKeyFrameBit);
   EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kFirstPacketBit);
   EXPECT_THAT(sent_payload.subview(1), ElementsAreArray(payload));
 
   // Send delta frame
   payload[0] = 13;
   payload[1] = 42;
   payload[4] = 13;
 
   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
       kVideoFrameDelta, payload_type, 1234, 4321, payload, sizeof(payload),
-      nullptr, nullptr, nullptr));
+      nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
 
   sent_payload = transport_.last_sent_packet().payload();
   generic_header = sent_payload[0];
   EXPECT_FALSE(generic_header & RtpFormatVideoGeneric::kKeyFrameBit);
   EXPECT_TRUE(generic_header & RtpFormatVideoGeneric::kFirstPacketBit);
   EXPECT_THAT(sent_payload.subview(1), ElementsAreArray(payload));
 }
 
 TEST_P(RtpSenderTest, SendFlexfecPackets) {
   constexpr int kMediaPayloadType = 127;
@@ skipping 99 matching lines @@
   const uint8_t kPayloadType = 127;
   const int64_t kCaptureTimeMs = fake_clock_.TimeInMilliseconds();
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
   EXPECT_EQ(0, rtp_sender_->RegisterPayload(payload_name, kPayloadType, 90000,
                                             0, 1500));
   RTPVideoHeader video_header;
   memset(&video_header, 0, sizeof(RTPVideoHeader));
   video_header.video_timing.flags = TimingFrameFlags::kTriggeredByTimer;
   EXPECT_TRUE(rtp_sender_->SendOutgoingData(
       kVideoFrameKey, kPayloadType, kTimestamp, kCaptureTimeMs, kPayloadData,
-      sizeof(kPayloadData), nullptr, &video_header, nullptr));
+      sizeof(kPayloadData), nullptr, &video_header, nullptr,
+      kDefaultExpectedRetransmissionTimeMs));
 
   EXPECT_CALL(mock_rtc_event_log_,
               LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
       .Times(1);
   EXPECT_TRUE(rtp_sender_->TimeToSendPacket(kMediaSsrc, kSeqNum,
                                             fake_clock_.TimeInMilliseconds(),
                                             false, PacedPacketInfo()));
   ASSERT_EQ(1, transport_.packets_sent());
   const RtpPacketReceived& media_packet = transport_.sent_packets_[0];
   EXPECT_EQ(kMediaPayloadType, media_packet.PayloadType());
   EXPECT_EQ(kSeqNum, media_packet.SequenceNumber());
   EXPECT_EQ(kMediaSsrc, media_packet.Ssrc());
 
   // Now try to send not a timing frame.
   uint16_t flexfec_seq_num;
   EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kLowPriority,
                                                kFlexfecSsrc, _, _, _, false))
       .WillOnce(testing::SaveArg<2>(&flexfec_seq_num));
   EXPECT_CALL(mock_paced_sender_,
               InsertPacket(RtpPacketSender::kLowPriority, kMediaSsrc,
                            kSeqNum + 1, _, _, false));
   video_header.video_timing.flags = TimingFrameFlags::kInvalid;
   EXPECT_TRUE(rtp_sender_->SendOutgoingData(
       kVideoFrameKey, kPayloadType, kTimestamp + 1, kCaptureTimeMs + 1,
-      kPayloadData, sizeof(kPayloadData), nullptr, &video_header, nullptr));
+      kPayloadData, sizeof(kPayloadData), nullptr, &video_header, nullptr,
+      kDefaultExpectedRetransmissionTimeMs));
 
   EXPECT_CALL(mock_rtc_event_log_,
               LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _))
       .Times(2);
   EXPECT_TRUE(rtp_sender_->TimeToSendPacket(kMediaSsrc, kSeqNum + 1,
                                             fake_clock_.TimeInMilliseconds(),
                                             false, PacedPacketInfo()));
   EXPECT_TRUE(rtp_sender_->TimeToSendPacket(kFlexfecSsrc, flexfec_seq_num,
                                             fake_clock_.TimeInMilliseconds(),
                                             false, PacedPacketInfo()));
@@ skipping 124 matching lines @@
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
 
   EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _))
       .Times(::testing::AtLeast(2));
 
-  ASSERT_TRUE(rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234,
-                                            4321, payload, sizeof(payload),
-                                            nullptr, nullptr, nullptr));
+  ASSERT_TRUE(rtp_sender_->SendOutgoingData(
+      kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
+      nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
 
   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_TRUE(rtp_sender_->SendOutgoingData(
       kVideoFrameDelta, payload_type, 1234, 4321, payload, sizeof(payload),
-      nullptr, nullptr, nullptr));
+      nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
 
   EXPECT_EQ(2U, callback.num_calls_);
   EXPECT_EQ(ssrc, callback.ssrc_);
   EXPECT_EQ(1, callback.frame_counts_.key_frames);
   EXPECT_EQ(1, callback.frame_counts_.delta_frames);
 
   rtp_sender_.reset();
 }
 
 TEST_P(RtpSenderTest, BitrateCallbacks) {
@@ skipping 45 matching lines @@
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
 
   // Initial process call so we get a new time window.
   rtp_sender_->ProcessBitrate();
 
   // Send a few frames.
   for (uint32_t i = 0; i < kNumPackets; ++i) {
     ASSERT_TRUE(rtp_sender_->SendOutgoingData(
         kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
-        nullptr, nullptr, nullptr));
+        nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
     fake_clock_.AdvanceTimeMilliseconds(kPacketInterval);
   }
 
   rtp_sender_->ProcessBitrate();
 
   // 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);
@@ skipping 61 matching lines @@
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
   rtp_sender_->SetStorePacketsStatus(true, 1);
   uint32_t ssrc = rtp_sender_->SSRC();
 
   rtp_sender_->RegisterRtpStatisticsCallback(&callback);
 
   // Send a frame.
   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
-                      kVideoFrameKey, payload_type, 1234, 4321, payload,
-                      sizeof(payload), nullptr, nullptr, nullptr));
+      kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
+      nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
   StreamDataCounters expected;
   expected.transmitted.payload_bytes = 6;
   expected.transmitted.header_bytes = 12;
   expected.transmitted.padding_bytes = 0;
   expected.transmitted.packets = 1;
   expected.retransmitted.payload_bytes = 0;
   expected.retransmitted.header_bytes = 0;
   expected.retransmitted.padding_bytes = 0;
   expected.retransmitted.packets = 0;
   expected.fec.packets = 0;
@@ skipping 20 matching lines @@
   callback.Matches(ssrc, expected);
 
   // Send ULPFEC.
   rtp_sender_->SetUlpfecConfig(kRedPayloadType, kUlpfecPayloadType);
   FecProtectionParams fec_params;
   fec_params.fec_mask_type = kFecMaskRandom;
   fec_params.fec_rate = 1;
   fec_params.max_fec_frames = 1;
   rtp_sender_->SetFecParameters(fec_params, fec_params);
   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
-                      kVideoFrameDelta, payload_type, 1234, 4321, payload,
-                      sizeof(payload), nullptr, nullptr, nullptr));
+      kVideoFrameDelta, payload_type, 1234, 4321, payload, sizeof(payload),
+      nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
   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(nullptr);
 }
 
 TEST_P(RtpSenderAudioTest, SendAudio) {
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 48000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
-                      kAudioFrameCN, payload_type, 1234, 4321, payload,
-                      sizeof(payload), nullptr, nullptr, nullptr));
+      kAudioFrameCN, payload_type, 1234, 4321, payload, sizeof(payload),
+      nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
 
   auto sent_payload = transport_.last_sent_packet().payload();
   EXPECT_THAT(sent_payload, ElementsAreArray(payload));
 }
 
 TEST_P(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
   EXPECT_EQ(0, rtp_sender_->SetAudioLevel(kAudioLevel));
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAudioLevel,
                                                        kAudioLevelExtensionId));
 
   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "PAYLOAD_NAME";
   const uint8_t payload_type = 127;
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 48000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
-                      kAudioFrameCN, payload_type, 1234, 4321, payload,
-                      sizeof(payload), nullptr, nullptr, nullptr));
+      kAudioFrameCN, payload_type, 1234, 4321, payload, sizeof(payload),
+      nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
 
   auto sent_payload = transport_.last_sent_packet().payload();
   EXPECT_THAT(sent_payload, ElementsAreArray(payload));
   // Verify AudioLevel extension.
   bool voice_activity;
   uint8_t audio_level;
   EXPECT_TRUE(transport_.last_sent_packet().GetExtension<AudioLevel>(
       &voice_activity, &audio_level));
   EXPECT_EQ(kAudioLevel, audio_level);
   EXPECT_FALSE(voice_activity);
@@ skipping 16 matching lines @@
   // Registering the payload again for audio stream with different payload name.
   const char* kPayloadName = "payload_name";
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(kPayloadName, kPayloadType,
                                             kPayloadFrequency, 1, 0));
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   // DTMF event key=9, duration=500 and attenuationdB=10
   rtp_sender_->SendTelephoneEvent(9, 500, 10);
   // During start, it takes the starting timestamp as last sent timestamp.
   // 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_TRUE(rtp_sender_->SendOutgoingData(kEmptyFrame, kPayloadType,
-                                            capture_time_ms, 0, nullptr, 0,
-                                            nullptr, nullptr, nullptr));
+  ASSERT_TRUE(rtp_sender_->SendOutgoingData(
+      kEmptyFrame, kPayloadType, capture_time_ms, 0, nullptr, 0, nullptr,
+      nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
   // DTMF Sample Length is (Frequency/1000) * Duration.
   // So in this case, it is (8000/1000) * 500 = 4000.
   // Sending it as two packets.
   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
-                      kEmptyFrame, kPayloadType, capture_time_ms + 2000, 0,
-                      nullptr, 0, nullptr, nullptr, nullptr));
+      kEmptyFrame, kPayloadType, capture_time_ms + 2000, 0, nullptr, 0, nullptr,
+      nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
 
   // Marker Bit should be set to 1 for first packet.
   EXPECT_TRUE(transport_.last_sent_packet().Marker());
 
   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
-                      kEmptyFrame, kPayloadType, capture_time_ms + 4000, 0,
-                      nullptr, 0, nullptr, nullptr, nullptr));
+      kEmptyFrame, kPayloadType, capture_time_ms + 4000, 0, nullptr, 0, nullptr,
+      nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
   // Marker Bit should be set to 0 for rest of the packets.
   EXPECT_FALSE(transport_.last_sent_packet().Marker());
 }
 
 TEST_P(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
   const char* kPayloadName = "GENERIC";
   const uint8_t kPayloadType = 127;
   rtp_sender_->SetSSRC(1234);
   rtp_sender_->SetRtxSsrc(4321);
   rtp_sender_->SetRtxPayloadType(kPayloadType - 1, kPayloadType);
   rtp_sender_->SetRtxStatus(kRtxRetransmitted | kRtxRedundantPayloads);
 
   ASSERT_EQ(0, rtp_sender_->RegisterPayload(kPayloadName, kPayloadType, 90000,
                                             0, 1500));
   uint8_t payload[] = {47, 11, 32, 93, 89};
 
   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
-                      kVideoFrameKey, kPayloadType, 1234, 4321, payload,
-                      sizeof(payload), nullptr, nullptr, nullptr));
+      kVideoFrameKey, kPayloadType, 1234, 4321, payload, sizeof(payload),
+      nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
 
   // Will send 2 full-size padding packets.
   rtp_sender_->TimeToSendPadding(1, PacedPacketInfo());
   rtp_sender_->TimeToSendPadding(1, PacedPacketInfo());
 
   StreamDataCounters rtp_stats;
   StreamDataCounters rtx_stats;
   rtp_sender_->GetDataCounters(&rtp_stats, &rtx_stats);
 
   // Payload + 1-byte generic header.
@@ skipping 53 matching lines @@
 
 TEST_P(RtpSenderVideoTest, KeyFrameHasCVO) {
   uint8_t kFrame[kMaxPacketLength];
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionVideoRotation, kVideoRotationExtensionId));
 
   RTPVideoHeader hdr = {0};
   hdr.rotation = kVideoRotation_0;
   rtp_sender_video_->SendVideo(kRtpVideoGeneric, kVideoFrameKey, kPayload,
                                kTimestamp, 0, kFrame, sizeof(kFrame), nullptr,
-                               &hdr);
+                               &hdr, kDefaultExpectedRetransmissionTimeMs);
 
   VideoRotation rotation;
   EXPECT_TRUE(
       transport_.last_sent_packet().GetExtension<VideoOrientation>(&rotation));
   EXPECT_EQ(kVideoRotation_0, rotation);
 }
 
 TEST_P(RtpSenderVideoTest, TimingFrameHasPacketizationTimstampSet) {
   uint8_t kFrame[kMaxPacketLength];
   const int64_t kPacketizationTimeMs = 100;
   const int64_t kEncodeStartDeltaMs = 10;
   const int64_t kEncodeFinishDeltaMs = 50;
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionVideoTiming, kVideoTimingExtensionId));
 
   const int64_t kCaptureTimestamp = fake_clock_.TimeInMilliseconds();
 
   RTPVideoHeader hdr = {0};
   hdr.video_timing.flags = TimingFrameFlags::kTriggeredByTimer;
   hdr.video_timing.encode_start_delta_ms = kEncodeStartDeltaMs;
   hdr.video_timing.encode_finish_delta_ms = kEncodeFinishDeltaMs;
 
   fake_clock_.AdvanceTimeMilliseconds(kPacketizationTimeMs);
   rtp_sender_video_->SendVideo(kRtpVideoGeneric, kVideoFrameKey, kPayload,
                                kTimestamp, kCaptureTimestamp, kFrame,
-                               sizeof(kFrame), nullptr, &hdr);
+                               sizeof(kFrame), nullptr, &hdr,
+                               kDefaultExpectedRetransmissionTimeMs);
   VideoSendTiming timing;
   EXPECT_TRUE(transport_.last_sent_packet().GetExtension<VideoTimingExtension>(
       &timing));
   EXPECT_EQ(kPacketizationTimeMs, timing.packetization_finish_delta_ms);
   EXPECT_EQ(kEncodeStartDeltaMs, timing.encode_start_delta_ms);
   EXPECT_EQ(kEncodeFinishDeltaMs, timing.encode_finish_delta_ms);
 }
 
 TEST_P(RtpSenderVideoTest, DeltaFrameHasCVOWhenChanged) {
   uint8_t kFrame[kMaxPacketLength];
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionVideoRotation, kVideoRotationExtensionId));
 
   RTPVideoHeader hdr = {0};
   hdr.rotation = kVideoRotation_90;
-  EXPECT_TRUE(rtp_sender_video_->SendVideo(kRtpVideoGeneric, kVideoFrameKey,
-                                           kPayload, kTimestamp, 0, kFrame,
-                                           sizeof(kFrame), nullptr, &hdr));
+  EXPECT_TRUE(rtp_sender_video_->SendVideo(
+      kRtpVideoGeneric, kVideoFrameKey, kPayload, kTimestamp, 0, kFrame,
+      sizeof(kFrame), nullptr, &hdr, kDefaultExpectedRetransmissionTimeMs));
 
   hdr.rotation = kVideoRotation_0;
-  EXPECT_TRUE(rtp_sender_video_->SendVideo(kRtpVideoGeneric, kVideoFrameDelta,
-                                           kPayload, kTimestamp + 1, 0, kFrame,
-                                           sizeof(kFrame), nullptr, &hdr));
+  EXPECT_TRUE(rtp_sender_video_->SendVideo(
+      kRtpVideoGeneric, kVideoFrameDelta, kPayload, kTimestamp + 1, 0, kFrame,
+      sizeof(kFrame), nullptr, &hdr, kDefaultExpectedRetransmissionTimeMs));
 
   VideoRotation rotation;
   EXPECT_TRUE(
       transport_.last_sent_packet().GetExtension<VideoOrientation>(&rotation));
   EXPECT_EQ(kVideoRotation_0, rotation);
 }
 
 TEST_P(RtpSenderVideoTest, DeltaFrameHasCVOWhenNonZero) {
   uint8_t kFrame[kMaxPacketLength];
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionVideoRotation, kVideoRotationExtensionId));
 
   RTPVideoHeader hdr = {0};
   hdr.rotation = kVideoRotation_90;
-  EXPECT_TRUE(rtp_sender_video_->SendVideo(kRtpVideoGeneric, kVideoFrameKey,
-                                           kPayload, kTimestamp, 0, kFrame,
-                                           sizeof(kFrame), nullptr, &hdr));
+  EXPECT_TRUE(rtp_sender_video_->SendVideo(
+      kRtpVideoGeneric, kVideoFrameKey, kPayload, kTimestamp, 0, kFrame,
+      sizeof(kFrame), nullptr, &hdr, kDefaultExpectedRetransmissionTimeMs));
 
-  EXPECT_TRUE(rtp_sender_video_->SendVideo(kRtpVideoGeneric, kVideoFrameDelta,
-                                           kPayload, kTimestamp + 1, 0, kFrame,
-                                           sizeof(kFrame), nullptr, &hdr));
+  EXPECT_TRUE(rtp_sender_video_->SendVideo(
+      kRtpVideoGeneric, kVideoFrameDelta, kPayload, kTimestamp + 1, 0, kFrame,
+      sizeof(kFrame), nullptr, &hdr, kDefaultExpectedRetransmissionTimeMs));
 
   VideoRotation rotation;
   EXPECT_TRUE(
       transport_.last_sent_packet().GetExtension<VideoOrientation>(&rotation));
   EXPECT_EQ(kVideoRotation_90, rotation);
 }
 
 // Make sure rotation is parsed correctly when the Camera (C) and Flip (F) bits
 // are set in the CVO byte.
 TEST_P(RtpSenderVideoTest, SendVideoWithCameraAndFlipCVO) {
   // Test extracting rotation when Camera (C) and Flip (F) bits are zero.
   EXPECT_EQ(kVideoRotation_0, ConvertCVOByteToVideoRotation(0));
   EXPECT_EQ(kVideoRotation_90, ConvertCVOByteToVideoRotation(1));
   EXPECT_EQ(kVideoRotation_180, ConvertCVOByteToVideoRotation(2));
   EXPECT_EQ(kVideoRotation_270, ConvertCVOByteToVideoRotation(3));
   // Test extracting rotation when Camera (C) and Flip (F) bits are set.
   const int flip_bit = 1 << 2;
   const int camera_bit = 1 << 3;
   EXPECT_EQ(kVideoRotation_0,
             ConvertCVOByteToVideoRotation(flip_bit | camera_bit | 0));
   EXPECT_EQ(kVideoRotation_90,
             ConvertCVOByteToVideoRotation(flip_bit | camera_bit | 1));
   EXPECT_EQ(kVideoRotation_180,
             ConvertCVOByteToVideoRotation(flip_bit | camera_bit | 2));
   EXPECT_EQ(kVideoRotation_270,
             ConvertCVOByteToVideoRotation(flip_bit | camera_bit | 3));
 }
 
+TEST_P(RtpSenderVideoTest, ConditionalRetransmit) {
+  const int64_t kWindowSize = 2500;

[danilchap, 2017/08/29 17:31:19]

kWindowSizeMs?

[sprang_webrtc, 2017/08/31 15:54:29]

Done.

+  const int64_t kFrameIntervalMs = 33;
+  const int64_t kRttMs = (kFrameIntervalMs * 3) / 2;
+  const int32_t kSettings =

[danilchap, 2017/08/29 17:31:20]

may be uint8_t instead of int32_t

[sprang_webrtc, 2017/08/31 15:54:29]

Done.

+      kRetransmitBaseLayer | kConditionallyRetransmitHigherLayers;
+
+  // Insert VP8 frames for all temporal layers, but stop before the final index.
+  RTPVideoHeader header;
+  header.codec = kRtpVideoVp8;
+  constexpr int kPatternLength = 4;
+  // Fill averaging window to prevent rounding errors.
+  constexpr int kNumRepetitions =
+      (kWindowSize + (kFrameIntervalMs / 2)) / kFrameIntervalMs;
+  for (int i = 0; i < kPatternLength * kNumRepetitions; ++i) {
+    switch (i % kPatternLength) {
+      case 0:
+        header.codecHeader.VP8.temporalIdx = 0;
+        break;
+      case 2:
+        header.codecHeader.VP8.temporalIdx = 1;
+        break;
+      case 1:
+      case 3:
+        header.codecHeader.VP8.temporalIdx = 2;
+        break;
+      default:
+        ADD_FAILURE() << "Bad temporal idx.";

[danilchap, 2017/08/29 17:31:20]

with kPatternLength = 4 and i >= 0 this look impossible.

I wonder if if a small array can better show your intent:
constexpr int kPattern[] = {0, 2, 1, 2};
for(int i = 0; i < arraysize(kPattern) * kNumRepetition; ++i) {
  header.codecHeader.VP8.temporalIdx = kPattern[i % arraysize(kPattern)];
  ...
}

[sprang_webrtc, 2017/08/31 15:54:29]

Yes, nice suggestion!

+    }
+    rtp_sender_video_->GetStorageType(
+        header, kSettings, kRttMs, i == 0 ? kVideoFrameKey : kVideoFrameDelta);
+    fake_clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+  }
+
+  // Since we're at the start of the pattern, the next expected frame in TL0 is
+  // right now. We will wait at most one expected retransmission time before
+  // acknowledging that it did not arrive, which means this frame and the next
+  // will not be retransmitted.
+  header.codecHeader.VP8.temporalIdx = 1;
+  EXPECT_EQ(StorageType::kDontRetransmit,
+            rtp_sender_video_->GetStorageType(header, kSettings, kRttMs,
+                                              kVideoFrameDelta));
+  fake_clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+  EXPECT_EQ(StorageType::kDontRetransmit,
+            rtp_sender_video_->GetStorageType(header, kSettings, kRttMs,
+                                              kVideoFrameDelta));
+  fake_clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+
+  // The TL0 frame did not arrive. So allow retransmission.
+  EXPECT_EQ(StorageType::kAllowRetransmission,
+            rtp_sender_video_->GetStorageType(header, kSettings, kRttMs,
+                                              kVideoFrameDelta));
+  fake_clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+
+  // Insert a frame for TL2. We just had frame in TL1, so the next one there is
+  // in three frames away. TL0 is still too far in the past. So, allow
+  // retransmission.
+  header.codecHeader.VP8.temporalIdx = 2;
+  EXPECT_EQ(StorageType::kAllowRetransmission,
+            rtp_sender_video_->GetStorageType(header, kSettings, kRttMs,
+                                              kVideoFrameDelta));
+  fake_clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+
+  // Another TL2, next in TL1 is two frames away. Allow again.
+  EXPECT_EQ(StorageType::kAllowRetransmission,
+            rtp_sender_video_->GetStorageType(header, kSettings, kRttMs,
+                                              kVideoFrameDelta));
+  fake_clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+
+  // Yet another TL2, next in TL1 is now only one frame away, so don't store
+  // for retransmission.
+  EXPECT_EQ(StorageType::kDontRetransmit,
+            rtp_sender_video_->GetStorageType(header, kSettings, kRttMs,
+                                              kVideoFrameDelta));
+}
+
+TEST_P(RtpSenderVideoTest, ConditionalRetransmitLimit) {
+  const int64_t kWindowSize = 2500;
+  const int64_t kFrameIntervalMs = 200;
+  const int64_t kRttMs = (kFrameIntervalMs * 3) / 2;
+  const int32_t kSettings =
+      kRetransmitBaseLayer | kConditionallyRetransmitHigherLayers;
+
+  // Insert VP8 frames for all temporal layers, but stop before the final index.
+  RTPVideoHeader header;
+  header.codec = kRtpVideoVp8;
+  constexpr int kPatternLength = 4;

[danilchap, 2017/08/29 17:31:20]

and here 
kPattern = {0, 1, 1, 1};?

[sprang_webrtc, 2017/08/31 15:54:28]

Done.

+  // Fill averaging window to prevent rounding errors.
+  constexpr int kNumRepetitions =
+      (kWindowSize + (kFrameIntervalMs / 2)) / kFrameIntervalMs;
+  for (int i = 0; i < kPatternLength * kNumRepetitions; ++i) {
+    // Put every fourth frame in tl0.
+    if (i % kPatternLength == 0) {
+      header.codecHeader.VP8.temporalIdx = 0;
+    } else {
+      header.codecHeader.VP8.temporalIdx = 1;
+    }
+
+    rtp_sender_video_->GetStorageType(
+        header, kSettings, kRttMs, i == 0 ? kVideoFrameKey : kVideoFrameDelta);
+    fake_clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
+  }
+
+  // Since we're at the start of the pattern, the next expected frame will be
+  // right now in TL0. Put it in TL1 instead. Regular rules would dictate that
+  // we don't store for retransmission because we expect a frame in a lower
+  // layer, but that last frame in TL1 was a long time ago in absolute terms,
+  // so allow retransmission anyway.
+  header.codecHeader.VP8.temporalIdx = 1;
+  EXPECT_EQ(StorageType::kAllowRetransmission,
+            rtp_sender_video_->GetStorageType(header, kSettings, kRttMs,
+                                              kVideoFrameDelta));
+}
+
 TEST_P(RtpSenderTest, OnOverheadChanged) {
   MockOverheadObserver mock_overhead_observer;
   rtp_sender_.reset(
       new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr,
                     nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
                     &retransmission_rate_limiter_, &mock_overhead_observer));
   rtp_sender_->SetSSRC(kSsrc);
 
   // RTP overhead is 12B.
   EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(12)).Times(1);
@@ skipping 88 matching lines @@
 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead,
                         RtpSenderTestWithoutPacer,
                         ::testing::Bool());
 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead,
                         RtpSenderVideoTest,
                         ::testing::Bool());
 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead,
                         RtpSenderAudioTest,
                         ::testing::Bool());
 }  // namespace webrtc