Delete Rtx-related methods from RTPPayloadRegistry.

webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc

Index: webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
index 32c9d5b9e80819c118535733110f5ff40846bc13..1a68726b4a6720fa1db145165abdc68696ac75bd 100644
--- a/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
@@ -15,13 +15,13 @@
 #include <set>
 
 #include "webrtc/api/call/transport.h"
+#include "webrtc/call/rtp_stream_receiver_controller.h"
+#include "webrtc/call/rtx_receive_stream.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "webrtc/rtc_base/rate_limiter.h"
 #include "webrtc/test/gtest.h"
 
@@ -29,6 +29,7 @@ namespace webrtc {
 
 const int kVideoNackListSize = 30;
 const uint32_t kTestSsrc = 3456;
+const uint32_t kTestRtxSsrc = kTestSsrc + 1;
 const uint16_t kTestSequenceNumber = 2345;
 const uint32_t kTestNumberOfPackets = 1350;
 const int kTestNumberOfRtxPackets = 149;
@@ -37,33 +38,17 @@ const int kPayloadType = 123;
 const int kRtxPayloadType = 98;
 const int64_t kMaxRttMs = 1000;
 
-class VerifyingRtxReceiver : public RtpData {
+class VerifyingMediaStream : public RtpPacketSinkInterface {
  public:
-  VerifyingRtxReceiver() {}
+  VerifyingMediaStream() {}
 
-  int32_t OnReceivedPayloadData(
-      const uint8_t* data,
-      size_t size,
-      const webrtc::WebRtcRTPHeader* rtp_header) override {
+  void OnRtpPacket(const RtpPacketReceived& packet) override {
     if (!sequence_numbers_.empty())
-      EXPECT_EQ(kTestSsrc, rtp_header->header.ssrc);
-    sequence_numbers_.push_back(rtp_header->header.sequenceNumber);
-    return 0;
-  }
-  std::list<uint16_t> sequence_numbers_;
-};
+      EXPECT_EQ(kTestSsrc, packet.Ssrc());
 
-class TestRtpFeedback : public NullRtpFeedback {
- public:
-  explicit TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
-  virtual ~TestRtpFeedback() {}
-
-  void OnIncomingSSRCChanged(uint32_t ssrc) override {
-    rtp_rtcp_->SetRemoteSSRC(ssrc);
+    sequence_numbers_.push_back(packet.SequenceNumber());
   }
-
- private:
-  RtpRtcp* rtp_rtcp_;
+  std::list<uint16_t> sequence_numbers_;
 };
 
 class RtxLoopBackTransport : public webrtc::Transport {
@@ -75,16 +60,10 @@ class RtxLoopBackTransport : public webrtc::Transport {
         consecutive_drop_end_(0),
         rtx_ssrc_(rtx_ssrc),
         count_rtx_ssrc_(0),
-        rtp_payload_registry_(NULL),
-        rtp_receiver_(NULL),
         module_(NULL) {}
 
-  void SetSendModule(RtpRtcp* rtpRtcpModule,
-                     RTPPayloadRegistry* rtp_payload_registry,
-                     RtpReceiver* receiver) {
+  void SetSendModule(RtpRtcp* rtpRtcpModule) {
     module_ = rtpRtcpModule;
-    rtp_payload_registry_ = rtp_payload_registry;
-    rtp_receiver_ = receiver;
   }
 
   void DropEveryNthPacket(int n) { packet_loss_ = n; }
@@ -99,24 +78,15 @@ class RtxLoopBackTransport : public webrtc::Transport {
                size_t len,
                const PacketOptions& options) override {
     count_++;
-    const unsigned char* ptr = static_cast<const unsigned char*>(data);
-    uint32_t ssrc = (ptr[8] << 24) + (ptr[9] << 16) + (ptr[10] << 8) + ptr[11];
-    if (ssrc == rtx_ssrc_)
-      count_rtx_ssrc_++;
-    uint16_t sequence_number = (ptr[2] << 8) + ptr[3];
-    size_t packet_length = len;
-    uint8_t restored_packet[1500];
-    RTPHeader header;
-    std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
-    if (!parser->Parse(ptr, len, &header)) {
+    RtpPacketReceived packet;
+    if (!packet.Parse(data, len))
       return false;
-    }
-
-    if (!rtp_payload_registry_->IsRtx(header)) {
-      // Don't store retransmitted packets since we compare it to the list
-      // created by the receiver.
+    if (packet.Ssrc() == rtx_ssrc_) {
+      count_rtx_ssrc_++;
+    } else {
+      // For non-RTX packets only.
       expected_sequence_numbers_.insert(expected_sequence_numbers_.end(),
-                                        sequence_number);
+                                        packet.SequenceNumber());
     }
     if (packet_loss_ > 0) {
       if ((count_ % packet_loss_) == 0) {
@@ -126,28 +96,7 @@ class RtxLoopBackTransport : public webrtc::Transport {
                count_ < consecutive_drop_end_) {
       return true;
     }
-    if (rtp_payload_registry_->IsRtx(header)) {
-      // Remove the RTX header and parse the original RTP header.
-      EXPECT_TRUE(rtp_payload_registry_->RestoreOriginalPacket(
-          restored_packet, ptr, &packet_length, rtp_receiver_->SSRC(), header));
-      if (!parser->Parse(restored_packet, packet_length, &header)) {
-        return false;
-      }
-      ptr = restored_packet;
-    } else {
-      rtp_payload_registry_->SetIncomingPayloadType(header);
-    }
-
-    PayloadUnion payload_specific;
-    if (!rtp_payload_registry_->GetPayloadSpecifics(header.payloadType,
-                                                    &payload_specific)) {
-      return false;
-    }
-    if (!rtp_receiver_->IncomingRtpPacket(header, ptr + header.headerLength,
-                                          packet_length - header.headerLength,
-                                          payload_specific, true)) {
-      return false;
-    }
+    EXPECT_TRUE(stream_receiver_controller_.OnRtpPacket(packet));
     return true;
   }
 
@@ -160,9 +109,8 @@ class RtxLoopBackTransport : public webrtc::Transport {
   int consecutive_drop_end_;
   uint32_t rtx_ssrc_;
   int count_rtx_ssrc_;
-  RTPPayloadRegistry* rtp_payload_registry_;
-  RtpReceiver* rtp_receiver_;
   RtpRtcp* module_;
+  RtpStreamReceiverController stream_receiver_controller_;
   std::set<uint16_t> expected_sequence_numbers_;
 };
 
@@ -170,8 +118,10 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
  protected:
   RtpRtcpRtxNackTest()
       : rtp_rtcp_module_(nullptr),
-        transport_(kTestSsrc + 1),
-        receiver_(),
+        transport_(kTestRtxSsrc),
+        rtx_stream_(&media_stream_,
+                    rtx_associated_payload_types_,
+                    kTestSsrc),
         payload_data_length(sizeof(payload_data)),
         fake_clock(123456),
         retransmission_rate_limiter_(&fake_clock, kMaxRttMs) {}
@@ -187,11 +137,6 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
     configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
     rtp_rtcp_module_ = RtpRtcp::CreateRtpRtcp(configuration);
 
-    rtp_feedback_.reset(new TestRtpFeedback(rtp_rtcp_module_));
-
-    rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
-        &fake_clock, &receiver_, rtp_feedback_.get(), &rtp_payload_registry_));
-
     rtp_rtcp_module_->SetSSRC(kTestSsrc);
     rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound);
     rtp_rtcp_module_->SetStorePacketsStatus(true, 600);
@@ -199,18 +144,16 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
     rtp_rtcp_module_->SetSequenceNumber(kTestSequenceNumber);
     rtp_rtcp_module_->SetStartTimestamp(111111);
 
-    transport_.SetSendModule(rtp_rtcp_module_, &rtp_payload_registry_,
-                             rtp_receiver_.get());
-
-    VideoCodec video_codec;
-    memset(&video_codec, 0, sizeof(video_codec));
-    video_codec.plType = kPayloadType;
-    memcpy(video_codec.plName, "I420", 5);
+    // Used for NACK processing.
+    // TODO(nisse): Unclear on which side? It's confusing to use a
+    // single rtp_rtcp module for both send and receive side.
+    rtp_rtcp_module_->SetRemoteSSRC(kTestSsrc);
 
-    EXPECT_EQ(0, rtp_rtcp_module_->RegisterSendPayload(video_codec));
+    rtp_rtcp_module_->RegisterVideoSendPayload(kPayloadType, "video");
     rtp_rtcp_module_->SetRtxSendPayloadType(kRtxPayloadType, kPayloadType);
-    EXPECT_EQ(0, rtp_payload_registry_.RegisterReceivePayload(video_codec));
-    rtp_payload_registry_.SetRtxPayloadType(kRtxPayloadType, kPayloadType);
+    transport_.SetSendModule(rtp_rtcp_module_);
+    media_receiver_ = transport_.stream_receiver_controller_.CreateReceiver(
+        kTestSsrc, &media_stream_);
 
     for (size_t n = 0; n < payload_data_length; n++) {
       payload_data[n] = n % 10;
@@ -218,14 +161,14 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
   }
 
   int BuildNackList(uint16_t* nack_list) {
-    receiver_.sequence_numbers_.sort();
+    media_stream_.sequence_numbers_.sort();
     std::list<uint16_t> missing_sequence_numbers;
-    std::list<uint16_t>::iterator it = receiver_.sequence_numbers_.begin();
+    std::list<uint16_t>::iterator it = media_stream_.sequence_numbers_.begin();
 
-    while (it != receiver_.sequence_numbers_.end()) {
+    while (it != media_stream_.sequence_numbers_.end()) {
       uint16_t sequence_number_1 = *it;
       ++it;
-      if (it != receiver_.sequence_numbers_.end()) {
+      if (it != media_stream_.sequence_numbers_.end()) {
         uint16_t sequence_number_2 = *it;
         // Add all missing sequence numbers to list
         for (uint16_t i = sequence_number_1 + 1; i < sequence_number_2; ++i) {
@@ -243,8 +186,8 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
 
   bool ExpectedPacketsReceived() {
     std::list<uint16_t> received_sorted;
-    std::copy(receiver_.sequence_numbers_.begin(),
-              receiver_.sequence_numbers_.end(),
+    std::copy(media_stream_.sequence_numbers_.begin(),
+              media_stream_.sequence_numbers_.end(),
               std::back_inserter(received_sorted));
     received_sorted.sort();
     return received_sorted.size() ==
@@ -254,9 +197,10 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
   }
 
   void RunRtxTest(RtxMode rtx_method, int loss) {
-    rtp_payload_registry_.SetRtxSsrc(kTestSsrc + 1);
+    rtx_receiver_ = transport_.stream_receiver_controller_.CreateReceiver(
+        kTestRtxSsrc, &rtx_stream_);
     rtp_rtcp_module_->SetRtxSendStatus(rtx_method);
-    rtp_rtcp_module_->SetRtxSsrc(kTestSsrc + 1);
+    rtp_rtcp_module_->SetRtxSsrc(kTestRtxSsrc);
     transport_.DropEveryNthPacket(loss);
     uint32_t timestamp = 3000;
     uint16_t nack_list[kVideoNackListSize];
@@ -274,22 +218,24 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
       // Prepare next frame.
       timestamp += 3000;
     }
-    receiver_.sequence_numbers_.sort();
+    media_stream_.sequence_numbers_.sort();
   }
 
   void TearDown() override { delete rtp_rtcp_module_; }
 
   std::unique_ptr<ReceiveStatistics> receive_statistics_;
-  RTPPayloadRegistry rtp_payload_registry_;
-  std::unique_ptr<RtpReceiver> rtp_receiver_;
   RtpRtcp* rtp_rtcp_module_;
-  std::unique_ptr<TestRtpFeedback> rtp_feedback_;
   RtxLoopBackTransport transport_;
-  VerifyingRtxReceiver receiver_;
+  const std::map<int, int> rtx_associated_payload_types_ =
+      {{kRtxPayloadType, kPayloadType}};
+  VerifyingMediaStream media_stream_;
+  RtxReceiveStream rtx_stream_;
   uint8_t payload_data[65000];
   size_t payload_data_length;
   SimulatedClock fake_clock;
   RateLimiter retransmission_rate_limiter_;
+  std::unique_ptr<RtpStreamReceiverInterface> media_receiver_;
+  std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_;
 };
 
 TEST_F(RtpRtcpRtxNackTest, LongNackList) {
@@ -316,26 +262,26 @@ TEST_F(RtpRtcpRtxNackTest, LongNackList) {
     rtp_rtcp_module_->Process();
   }
   EXPECT_FALSE(transport_.expected_sequence_numbers_.empty());
-  EXPECT_FALSE(receiver_.sequence_numbers_.empty());
-  size_t last_receive_count = receiver_.sequence_numbers_.size();
+  EXPECT_FALSE(media_stream_.sequence_numbers_.empty());
+  size_t last_receive_count = media_stream_.sequence_numbers_.size();
   int length = BuildNackList(nack_list);
   for (int i = 0; i < kNumRequiredRtcp - 1; ++i) {
     rtp_rtcp_module_->SendNACK(nack_list, length);
-    EXPECT_GT(receiver_.sequence_numbers_.size(), last_receive_count);
-    last_receive_count = receiver_.sequence_numbers_.size();
+    EXPECT_GT(media_stream_.sequence_numbers_.size(), last_receive_count);
+    last_receive_count = media_stream_.sequence_numbers_.size();
     EXPECT_FALSE(ExpectedPacketsReceived());
   }
   rtp_rtcp_module_->SendNACK(nack_list, length);
-  EXPECT_GT(receiver_.sequence_numbers_.size(), last_receive_count);
+  EXPECT_GT(media_stream_.sequence_numbers_.size(), last_receive_count);
   EXPECT_TRUE(ExpectedPacketsReceived());
 }
 
 TEST_F(RtpRtcpRtxNackTest, RtxNack) {
   RunRtxTest(kRtxRetransmitted, 10);
-  EXPECT_EQ(kTestSequenceNumber, *(receiver_.sequence_numbers_.begin()));
+  EXPECT_EQ(kTestSequenceNumber, *(media_stream_.sequence_numbers_.begin()));
   EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
-            *(receiver_.sequence_numbers_.rbegin()));
-  EXPECT_EQ(kTestNumberOfPackets, receiver_.sequence_numbers_.size());
+            *(media_stream_.sequence_numbers_.rbegin()));
+  EXPECT_EQ(kTestNumberOfPackets, media_stream_.sequence_numbers_.size());
   EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_);
   EXPECT_TRUE(ExpectedPacketsReceived());
 }