Datagram Transport Integration

- Implement datagram transport adaptor, which wraps datagram transport in DtlsTransportInternal. Datagram adaptor owns both ICE and Datagram Transports.
- Implement setup of datagram transport based on RTCConfiguration flag use_datagram_transport. This is very similar to MediaTransport setup with the exception that we create DTLS datagram adaptor.
- Propagate maximum datagram size to video encoder via MediaTransportConfig.

TODO: Currently this CL can only be tested in downstream projects. Once we add fake datagram transport, we will be able to implement unit tests similar to loopback media transport.

Bug: webrtc:9719
Change-Id: I4fa4a5725598dfee5da4f0f374269a7e289d48ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138100
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28047}

api/media_transport_config.cc

@@ -10,11 +10,30 @@
 
 #include "api/media_transport_config.h"
 
+#include "rtc_base/checks.h"
+#include "rtc_base/string_utils.h"
+#include "rtc_base/strings/string_builder.h"
+
 namespace webrtc {
 
-std::string MediaTransportConfig::DebugString() const {
-  return (media_transport != nullptr ? "{media_transport: (Transport)}"
-                                     : "{media_transport: null}");
+MediaTransportConfig::MediaTransportConfig(
+    MediaTransportInterface* media_transport)
+    : media_transport(media_transport) {
+  RTC_DCHECK(media_transport != nullptr);
+}
+
+MediaTransportConfig::MediaTransportConfig(size_t rtp_max_packet_size)
+    : rtp_max_packet_size(rtp_max_packet_size) {
+  RTC_DCHECK_GT(rtp_max_packet_size, 0);
+}
+
+std::string MediaTransportConfig::DebugString()
+    const {  // TODO(sukhanov): Add rtp_max_packet_size (requires fixing
+             // audio_send/receive_stream_unittest.cc).
+  rtc::StringBuilder result;
+  result << "{media_transport: "
+         << (media_transport != nullptr ? "(Transport)" : "null") << "}";
+  return result.Release();
 }
 
 }  // namespace webrtc

api/media_transport_config.h

@@ -13,28 +13,33 @@
 #include <string>
 #include <utility>
 
+#include "absl/types/optional.h"
+
 namespace webrtc {
 
 class MediaTransportInterface;
 
-// MediaTransportConfig contains meida transport (if provided) and passed from
-// PeerConnection to call obeject and media layers that require access to media
-// transport. In the future we can add other transport (for example, datagram
-// transport) and related configuration.
+// Media transport config is made available to both transport and audio / video
+// layers, but access to individual interfaces should not be open without
+// necessity.
 struct MediaTransportConfig {
   // Default constructor for no-media transport scenarios.
   MediaTransportConfig() = default;
 
-  // TODO(sukhanov): Consider adding RtpTransport* to MediaTransportConfig,
-  // because it's almost always passes along with media_transport.
-  // Does not own media_transport.
-  explicit MediaTransportConfig(MediaTransportInterface* media_transport)
-      : media_transport(media_transport) {}
+  // Constructor for media transport scenarios.
+  // Note that |media_transport| may not be nullptr.
+  explicit MediaTransportConfig(MediaTransportInterface* media_transport);
+
+  // Constructor for datagram transport scenarios.
+  explicit MediaTransportConfig(size_t rtp_max_packet_size);
 
   std::string DebugString() const;
 
   // If provided, all media is sent through media_transport.
   MediaTransportInterface* media_transport = nullptr;
+
+  // If provided, limits RTP packet size (excludes ICE, IP or network overhead).
+  absl::optional<size_t> rtp_max_packet_size;
 };
 
 }  // namespace webrtc