Makes RtpSender indicate allocation and feedback status on packets.

Streams that are part of transport feedback are assumed to be part of
allocation. A SetAsPartOfAllocation method is introduced to be used by
media streams that are part of bitrate allocation but not included in
feedback.

This is part of a series of CLs that allows GoogCC to track sent bitrate
that is included in bitrate allocation but without transport feedback.

Bug: webrtc:9796
Change-Id: If7ac1ad3e6f3c28b2ada2aef1607a42689d899b2
Reviewed-on: https://webrtc-review.googlesource.com/c/104881
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25079}

modules/rtp_rtcp/include/rtp_rtcp.h

@@ -213,6 +213,10 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
   // Returns current media sending status.
   virtual bool SendingMedia() const = 0;
 
+  // Indicate that the packets sent by this module should be counted towards the
+  // bitrate estimate since the stream participates in the bitrate allocation.
+  virtual void SetAsPartOfAllocation(bool part_of_allocation) = 0;
+
   // Returns current bitrate in Kbit/s.
   virtual void BitrateSent(uint32_t* total_rate,
                            uint32_t* video_rate,

modules/rtp_rtcp/mocks/mock_rtp_rtcp.h

@@ -83,6 +83,7 @@ class MockRtpRtcp : public RtpRtcp {
   MOCK_CONST_METHOD0(Sending, bool());
   MOCK_METHOD1(SetSendingMediaStatus, void(bool sending));
   MOCK_CONST_METHOD0(SendingMedia, bool());
+  MOCK_METHOD1(SetAsPartOfAllocation, void(bool));
   MOCK_CONST_METHOD4(BitrateSent,
                      void(uint32_t* total_rate,
                           uint32_t* video_rate,

modules/rtp_rtcp/source/rtp_rtcp_impl.cc

@@ -392,6 +392,11 @@ bool ModuleRtpRtcpImpl::SendingMedia() const {
   return rtp_sender_ ? rtp_sender_->SendingMedia() : false;
 }
 
+void ModuleRtpRtcpImpl::SetAsPartOfAllocation(bool part_of_allocation) {
+  RTC_CHECK(rtp_sender_);
+  rtp_sender_->SetAsPartOfAllocation(part_of_allocation);
+}
+
 bool ModuleRtpRtcpImpl::SendOutgoingData(
     FrameType frame_type,
     int8_t payload_type,

modules/rtp_rtcp/source/rtp_rtcp_impl.h

@@ -115,6 +115,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
 
   bool SendingMedia() const override;
 
+  void SetAsPartOfAllocation(bool part_of_allocation) override;
+
   // Used by the codec module to deliver a video or audio frame for
   // packetization.
   bool SendOutgoingData(FrameType frame_type,

modules/rtp_rtcp/source/rtp_sender.cc

@@ -133,6 +133,7 @@ RTPSender::RTPSender(
       last_capture_time_ms_sent_(0),
       transport_(transport),
       sending_media_(true),  // Default to sending media.
+      force_part_of_allocation_(false),
       max_packet_size_(IP_PACKET_SIZE - 28),  // Default is IP-v4/UDP.
       last_payload_type_(-1),
       payload_type_map_(),
@@ -624,6 +625,9 @@ size_t RTPSender::SendPadData(size_t bytes,
       rtc::CritScope lock(&send_critsect_);
       has_transport_seq_num =
           UpdateTransportSequenceNumber(&padding_packet, &options.packet_id);
+      options.included_in_allocation =
+          has_transport_seq_num || force_part_of_allocation_;
+      options.included_in_feedback = has_transport_seq_num;
     }
     padding_packet.SetPadding(padding_bytes_in_packet, &random_);
     if (has_transport_seq_num) {
@@ -846,6 +850,9 @@ bool RTPSender::PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet,
     rtc::CritScope lock(&send_critsect_);
     has_transport_seq_num =
         UpdateTransportSequenceNumber(packet_to_send, &options.packet_id);
+    options.included_in_allocation =
+        has_transport_seq_num || force_part_of_allocation_;
+    options.included_in_feedback = has_transport_seq_num;
   }
   if (has_transport_seq_num) {
     AddPacketToTransportFeedback(options.packet_id, *packet_to_send,
@@ -986,6 +993,9 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
     rtc::CritScope lock(&send_critsect_);
     has_transport_seq_num =
         UpdateTransportSequenceNumber(packet.get(), &options.packet_id);
+    options.included_in_allocation =
+        has_transport_seq_num || force_part_of_allocation_;
+    options.included_in_feedback = has_transport_seq_num;
   }
   if (has_transport_seq_num) {
     AddPacketToTransportFeedback(options.packet_id, *packet.get(),
@@ -1224,6 +1234,11 @@ bool RTPSender::SendingMedia() const {
   return sending_media_;
 }
 
+void RTPSender::SetAsPartOfAllocation(bool part_of_allocation) {
+  rtc::CritScope lock(&send_critsect_);
+  force_part_of_allocation_ = part_of_allocation;
+}
+
 void RTPSender::SetTimestampOffset(uint32_t timestamp) {
   rtc::CritScope lock(&send_critsect_);
   timestamp_offset_ = timestamp;

modules/rtp_rtcp/source/rtp_sender.h

@@ -86,6 +86,8 @@ class RTPSender {
   void SetSendingMediaStatus(bool enabled);
   bool SendingMedia() const;
 
+  void SetAsPartOfAllocation(bool part_of_allocation);
+
   void GetDataCounters(StreamDataCounters* rtp_stats,
                        StreamDataCounters* rtx_stats) const;
 
@@ -277,7 +279,7 @@ class RTPSender {
 
   Transport* transport_;
   bool sending_media_ RTC_GUARDED_BY(send_critsect_);
-
+  bool force_part_of_allocation_ RTC_GUARDED_BY(send_critsect_);
   size_t max_packet_size_;
 
   int8_t last_payload_type_ RTC_GUARDED_BY(send_critsect_);

modules/rtp_rtcp/source/rtp_sender_unittest.cc

@@ -479,6 +479,7 @@ TEST_P(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
   ASSERT_TRUE(packet.GetExtension<TransportSequenceNumber>(&transport_seq_no));
   EXPECT_EQ(kTransportSequenceNumber, transport_seq_no);
   EXPECT_EQ(transport_.last_options_.packet_id, transport_seq_no);
+  EXPECT_TRUE(transport_.last_options_.included_in_allocation);
 }
 
 TEST_P(RtpSenderTestWithoutPacer, PacketOptionsNoRetransmission) {
@@ -493,8 +494,45 @@ TEST_P(RtpSenderTestWithoutPacer, PacketOptionsNoRetransmission) {
   EXPECT_FALSE(transport_.last_options_.is_retransmit);
 }
 
-TEST_P(RtpSenderTestWithoutPacer, NoAllocationIfNotRegistered) {
+TEST_P(RtpSenderTestWithoutPacer,
+       SetsIncludedInFeedbackWhenTransportSequenceNumberExtensionIsRegistered) {
+  SetUpRtpSender(false, false);
+  rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
+                                          kTransportSequenceNumberExtensionId);
+  EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
+      .WillOnce(testing::Return(kTransportSequenceNumber));
+  EXPECT_CALL(send_packet_observer_, OnSendPacket).Times(1);
   SendGenericPayload();
+  EXPECT_TRUE(transport_.last_options_.included_in_feedback);
+}
+
+TEST_P(
+    RtpSenderTestWithoutPacer,
+    SetsIncludedInAllocationWhenTransportSequenceNumberExtensionIsRegistered) {
+  SetUpRtpSender(false, false);
+  rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
+                                          kTransportSequenceNumberExtensionId);
+  EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
+      .WillOnce(testing::Return(kTransportSequenceNumber));
+  EXPECT_CALL(send_packet_observer_, OnSendPacket).Times(1);
+  SendGenericPayload();
+  EXPECT_TRUE(transport_.last_options_.included_in_allocation);
+}
+
+TEST_P(RtpSenderTestWithoutPacer,
+       SetsIncludedInAllocationWhenForcedAsPartOfAllocation) {
+  SetUpRtpSender(false, false);
+  rtp_sender_->SetAsPartOfAllocation(true);
+  SendGenericPayload();
+  EXPECT_FALSE(transport_.last_options_.included_in_feedback);
+  EXPECT_TRUE(transport_.last_options_.included_in_allocation);
+}
+
+TEST_P(RtpSenderTestWithoutPacer, DoesnSetIncludedInAllocationByDefault) {
+  SetUpRtpSender(false, false);
+  SendGenericPayload();
+  EXPECT_FALSE(transport_.last_options_.included_in_feedback);
+  EXPECT_FALSE(transport_.last_options_.included_in_allocation);
 }
 
 TEST_P(RtpSenderTestWithoutPacer, OnSendSideDelayUpdated) {