Use ArrayView versions of SendRtp and SendRtcp

This CL adds [[deprecated]] to the old signatures, and uses the new
signatures throughout.

Bug: webrtc:14870
Change-Id: Ic9a8198ac0a2f954e1b2e7d05a55dbe04342f958
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314962
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40517}

api/call/transport.h

@@ -48,6 +48,8 @@ class Transport {
   // New style functions. Default implementations are to accomodate
   // subclasses that haven't been converted to new style yet.
   // TODO(bugs.webrtc.org/14870): Deprecate and remove old functions.
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wdeprecated-declarations"
   virtual bool SendRtp(rtc::ArrayView<const uint8_t> packet,
                        const PacketOptions& options) {
     return SendRtp(packet.data(), packet.size(), options);
@@ -55,11 +57,17 @@ class Transport {
   virtual bool SendRtcp(rtc::ArrayView<const uint8_t> packet) {
     return SendRtcp(packet.data(), packet.size());
   }
+#pragma clang diagnostic pop
   // Old style functions.
-  virtual bool SendRtp(const uint8_t* packet,
+  [[deprecated("Use ArrayView version")]] virtual bool
-                       size_t length,
+  SendRtp(const uint8_t* packet, size_t length, const PacketOptions& options) {
-                       const PacketOptions& options) = 0;
+    return SendRtp(rtc::MakeArrayView(packet, length), options);
-  virtual bool SendRtcp(const uint8_t* packet, size_t length) = 0;
+  }
+  [[deprecated("Use ArrayView version")]] virtual bool SendRtcp(
+      const uint8_t* packet,
+      size_t length) {
+    return SendRtcp(rtc::MakeArrayView(packet, length));
+  }
 
  protected:
   virtual ~Transport() {}

call/fake_network_pipe.cc

@@ -356,9 +356,11 @@ void FakeNetworkPipe::DeliverNetworkPacket(NetworkPacket* packet) {
       return;
     }
     if (packet->is_rtcp()) {
-      transport->SendRtcp(packet->data(), packet->data_length());
+      transport->SendRtcp(
+          rtc::MakeArrayView(packet->data(), packet->data_length()));
     } else {
-      transport->SendRtp(packet->data(), packet->data_length(),
+      transport->SendRtp(
+          rtc::MakeArrayView(packet->data(), packet->data_length()),
           packet->packet_options());
     }
   } else if (receiver_) {

media/engine/webrtc_video_engine_unittest.cc

@@ -5589,8 +5589,7 @@ TEST_F(WebRtcVideoChannelTest, TestSetDscpOptions) {
 
   // Packets should also self-identify their dscp in PacketOptions.
   const uint8_t kData[10] = {0};
-  EXPECT_TRUE(ChannelImplAsTransport(send_channel.get())
+  EXPECT_TRUE(ChannelImplAsTransport(send_channel.get())->SendRtcp(kData));
-                  ->SendRtcp(kData, sizeof(kData)));
   EXPECT_EQ(rtc::DSCP_CS1, network_interface->options().dscp);
   send_channel->SetInterface(nullptr);
 

media/engine/webrtc_voice_engine_unittest.cc

@@ -3255,9 +3255,7 @@ TEST_P(WebRtcVoiceEngineTestFake, TestSetDscpOptions) {
 
   // Packets should also self-identify their dscp in PacketOptions.
   const uint8_t kData[10] = {0};
-  EXPECT_TRUE(SendImplFromPointer(channel.get())
+  EXPECT_TRUE(SendImplFromPointer(channel.get())->transport()->SendRtcp(kData));
-                  ->transport()
-                  ->SendRtcp(kData, sizeof(kData)));
   EXPECT_EQ(rtc::DSCP_CS1, network_interface.options().dscp);
   channel->SetInterface(nullptr);
 

modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc

@@ -416,7 +416,7 @@ bool DEPRECATED_RtpSenderEgress::SendPacketToNetwork(
     const PacedPacketInfo& pacing_info) {
   int bytes_sent = -1;
   if (transport_) {
-    bytes_sent = transport_->SendRtp(packet.data(), packet.size(), options)
+    bytes_sent = transport_->SendRtp(packet, options)
                      ? static_cast<int>(packet.size())
                      : -1;
     if (event_log_ && bytes_sent > 0) {

modules/rtp_rtcp/source/rtcp_sender.cc

@@ -242,7 +242,7 @@ int32_t RTCPSender::SendLossNotification(const FeedbackState& feedback_state,
                                          bool buffering_allowed) {
   int32_t error_code = -1;
   auto callback = [&](rtc::ArrayView<const uint8_t> packet) {
-    transport_->SendRtcp(packet.data(), packet.size());
+    transport_->SendRtcp(packet);
     error_code = 0;
     if (event_log_) {
       event_log_->Log(std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
@@ -659,7 +659,7 @@ int32_t RTCPSender::SendRTCP(const FeedbackState& feedback_state,
                              const uint16_t* nack_list) {
   int32_t error_code = -1;
   auto callback = [&](rtc::ArrayView<const uint8_t> packet) {
-    if (transport_->SendRtcp(packet.data(), packet.size())) {
+    if (transport_->SendRtcp(packet)) {
       error_code = 0;
       if (event_log_) {
         event_log_->Log(std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
@@ -938,7 +938,7 @@ void RTCPSender::SendCombinedRtcpPacket(
   }
   RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
   auto callback = [&](rtc::ArrayView<const uint8_t> packet) {
-    if (transport_->SendRtcp(packet.data(), packet.size())) {
+    if (transport_->SendRtcp(packet)) {
       if (event_log_)
         event_log_->Log(std::make_unique<RtcEventRtcpPacketOutgoing>(packet));
     }

modules/rtp_rtcp/source/rtp_sender_egress.cc

@@ -522,7 +522,7 @@ bool RtpSenderEgress::SendPacketToNetwork(const RtpPacketToSend& packet,
   RTC_DCHECK_RUN_ON(worker_queue_);
   int bytes_sent = -1;
   if (transport_) {
-    bytes_sent = transport_->SendRtp(packet.data(), packet.size(), options)
+    bytes_sent = transport_->SendRtp(packet, options)
                      ? static_cast<int>(packet.size())
                      : -1;
     if (event_log_ && bytes_sent > 0) {

video/end_to_end_tests/retransmission_tests.cc

@@ -166,7 +166,7 @@ TEST_F(RetransmissionEndToEndTest, ReceivesNackAndRetransmitsAudio) {
         nack.SetPacketIds(nack_list, 1);
         rtc::Buffer buffer = nack.Build();
 
-        EXPECT_TRUE(receive_transport_->SendRtcp(buffer.data(), buffer.size()));
+        EXPECT_TRUE(receive_transport_->SendRtcp(buffer));
       }
 
       return SEND_PACKET;

video/end_to_end_tests/rtp_rtcp_tests.cc

@@ -336,7 +336,7 @@ void RtpRtcpEndToEndTest::TestRtpStatePreservation(
         rtcp::RapidResyncRequest force_send_sr_back_request;
         rtc::Buffer packet = force_send_sr_back_request.Build();
         static_cast<webrtc::Transport*>(receive_transport_.get())
-            ->SendRtcp(packet.data(), packet.size());
+            ->SendRtcp(packet);
       }
       CreateFrameGeneratorCapturer(30, 1280, 720);
       StartVideoSources();

video/transport_adapter.cc

@@ -22,20 +22,19 @@ TransportAdapter::TransportAdapter(Transport* transport)
 
 TransportAdapter::~TransportAdapter() = default;
 
-bool TransportAdapter::SendRtp(const uint8_t* packet,
+bool TransportAdapter::SendRtp(rtc::ArrayView<const uint8_t> packet,
-                               size_t length,
                                const PacketOptions& options) {
   if (!enabled_.load())
     return false;
 
-  return transport_->SendRtp(packet, length, options);
+  return transport_->SendRtp(packet, options);
 }
 
-bool TransportAdapter::SendRtcp(const uint8_t* packet, size_t length) {
+bool TransportAdapter::SendRtcp(rtc::ArrayView<const uint8_t> packet) {
   if (!enabled_.load())
     return false;
 
-  return transport_->SendRtcp(packet, length);
+  return transport_->SendRtcp(packet);
 }
 
 void TransportAdapter::Enable() {

video/transport_adapter.h

@@ -25,10 +25,9 @@ class TransportAdapter : public Transport {
   explicit TransportAdapter(Transport* transport);
   ~TransportAdapter() override;
 
-  bool SendRtp(const uint8_t* packet,
+  bool SendRtp(rtc::ArrayView<const uint8_t> packet,
-               size_t length,
                const PacketOptions& options) override;
-  bool SendRtcp(const uint8_t* packet, size_t length) override;
+  bool SendRtcp(rtc::ArrayView<const uint8_t> packet) override;
 
   void Enable();
   void Disable();