Move the payload type picker to call/

Since media/ and pc/ both have to use this, and both depend on call/, this seems to be the right place to put it. Also factor out the interface that media will use in a separate interface class. Bug: webrtc:360058654 Change-Id: I34acbecc618f23e19542ce4b0110d0e8ed9e55ee Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361281 Commit-Queue: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Florent Castelli <orphis@webrtc.org> Auto-Submit: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42933}
2024-09-03 11:20:36 +00:00 · 2024-09-03 11:20:36 +00:00 · c17ca01f54
commit c17ca01f54
parent 682f7945d5
16 changed files with 271 additions and 55 deletions
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@ -38,6 +38,8 @@ rtc_library("call_interfaces") {

  deps = [
    ":audio_sender_interface",
+    ":payload_type",
+    ":payload_type",
    ":receive_stream_interface",
    ":rtp_interfaces",
    ":video_stream_api",
@ -307,6 +309,9 @@ rtc_library("call") {
    ":bitrate_allocator",
    ":call_interfaces",
    ":fake_network",
+    ":payload_type",
+    ":payload_type",
+    ":payload_type_picker",
    ":receive_stream_interface",
    ":rtp_interfaces",
    ":rtp_receiver",
@ -317,6 +322,7 @@ rtc_library("call") {
    "../api:array_view",
    "../api:fec_controller_api",
    "../api:field_trials_view",
+    "../api:rtc_error",
    "../api:rtp_headers",
    "../api:rtp_parameters",
    "../api:scoped_refptr",
@ -340,6 +346,7 @@ rtc_library("call") {
    "../logging:rtc_event_rtp_rtcp",
    "../logging:rtc_event_video",
    "../logging:rtc_stream_config",
+    "../media:codec",
    "../modules/congestion_controller",
    "../modules/pacing",
    "../modules/rtp_rtcp",
@ -374,6 +381,32 @@ rtc_library("call") {
  ]
 }

+rtc_library("payload_type_picker") {
+  sources = [
+    "payload_type_picker.cc",
+    "payload_type_picker.h",
+  ]
+  deps = [
+    ":payload_type",
+    "../api:rtc_error",
+    "../api/audio_codecs:audio_codecs_api",
+    "../media:codec",
+    "../media:media_constants",
+    "../rtc_base:logging",
+    "../rtc_base:strong_alias",
+    "//third_party/abseil-cpp/absl/strings",
+  ]
+}
+
+rtc_source_set("payload_type") {
+  sources = [ "payload_type.h" ]
+  deps = [
+    "../api:rtc_error",
+    "../media:codec",
+    "../rtc_base:strong_alias",
+  ]
+}
+
 rtc_source_set("receive_stream_interface") {
  sources = [ "receive_stream.h" ]
  deps = [
@ -472,6 +505,7 @@ if (rtc_include_tests) {
        "bitrate_estimator_tests.cc",
        "call_unittest.cc",
        "flexfec_receive_stream_unittest.cc",
+        "payload_type_picker_unittest.cc",
        "receive_time_calculator_unittest.cc",
        "rtp_bitrate_configurator_unittest.cc",
        "rtp_demuxer_unittest.cc",
@ -485,6 +519,8 @@ if (rtc_include_tests) {
        ":call",
        ":call_interfaces",
        ":mock_rtp_interfaces",
+        ":payload_type",
+        ":payload_type_picker",
        ":rtp_interfaces",
        ":rtp_receiver",
        ":rtp_sender",
@ -527,6 +563,8 @@ if (rtc_include_tests) {
        "../audio",
        "../common_video:frame_counts",
        "../common_video/generic_frame_descriptor",
+        "../media:codec",
+        "../media:media_constants",
        "../modules/audio_device:mock_audio_device",
        "../modules/audio_mixer",
        "../modules/audio_mixer:audio_mixer_impl",
--- a/call/DEPS
+++ b/call/DEPS
@ -52,4 +52,22 @@ specific_include_rules = {
    "+common_video/frame_counts.h",
    "+common_video/generic_frame_descriptor",
  ],
+  "payload_type\.h": [
+    "+media/base/codec.h",
+  ],
+  "payload_type_picker\.h": [
+    "+media/base/codec.h",
+    "+media/base/media_constants.h",
+  ],
+  "payload_type_picker\.cc": [
+    "+media/base/codec.h",
+    "+media/base/media_constants.h",
+  ],
+  "payload_type_picker_unittest\.cc": [
+    "+media/base/codec.h",
+    "+media/base/media_constants.h",
+  ],
+  "call\.cc": [
+    "+media/base/codec.h",
+  ]
 }
--- a/call/call.cc
+++ b/call/call.cc
@ -30,6 +30,7 @@
 #include "api/fec_controller.h"
 #include "api/field_trials_view.h"
 #include "api/media_types.h"
+#include "api/rtc_error.h"
 #include "api/rtc_event_log/rtc_event_log.h"
 #include "api/rtp_headers.h"
 #include "api/scoped_refptr.h"
@ -52,6 +53,8 @@
 #include "call/flexfec_receive_stream.h"
 #include "call/flexfec_receive_stream_impl.h"
 #include "call/packet_receiver.h"
+#include "call/payload_type.h"
+#include "call/payload_type_picker.h"
 #include "call/receive_stream.h"
 #include "call/receive_time_calculator.h"
 #include "call/rtp_config.h"
@ -66,6 +69,7 @@
 #include "logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h"
 #include "logging/rtc_event_log/events/rtc_event_video_send_stream_config.h"
 #include "logging/rtc_event_log/rtc_stream_config.h"
+#include "media/base/codec.h"
 #include "modules/congestion_controller/include/receive_side_congestion_controller.h"
 #include "modules/rtp_rtcp/include/flexfec_receiver.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
@ -100,6 +104,21 @@ namespace webrtc {

 namespace {

+// In normal operation, the PTS comes from the PeerConnection.
+// However, it is too much of a bother to insert it in all tests,
+// so defaulting here.
+class PayloadTypeSuggesterForTests : public PayloadTypeSuggester {
+ public:
+  PayloadTypeSuggesterForTests() = default;
+  RTCErrorOr<PayloadType> SuggestPayloadType(const std::string& mid,
+                                             cricket::Codec codec) override {
+    return payload_type_picker_.SuggestMapping(codec, nullptr);
+  }
+
+ private:
+  PayloadTypePicker payload_type_picker_;
+};
+
 const int* FindKeyByValue(const std::map<int, int>& m, int v) {
  for (const auto& kv : m) {
    if (kv.second == v)
@ -249,6 +268,9 @@ class Call final : public webrtc::Call,

  RtpTransportControllerSendInterface* GetTransportControllerSend() override;

+  PayloadTypeSuggester* GetPayloadTypeSuggester() override;
+  void SetPayloadTypeSuggester(PayloadTypeSuggester* suggester) override;
+
  Stats GetStats() const override;

  const FieldTrialsView& trials() const override;
@ -465,17 +487,21 @@ class Call final : public webrtc::Call,
  // https://bugs.chromium.org/p/chromium/issues/detail?id=992640
  RtpTransportControllerSendInterface* const transport_send_ptr_
      RTC_GUARDED_BY(send_transport_sequence_checker_);
-  // Declared last since it will issue callbacks from a task queue. Declaring it
-  // last ensures that it is destroyed first and any running tasks are finished.
-  const std::unique_ptr<RtpTransportControllerSendInterface> transport_send_;

  bool is_started_ RTC_GUARDED_BY(worker_thread_) = false;

+  // Mechanism for proposing payload types in RTP mappings.
+  PayloadTypeSuggester* pt_suggester_ = nullptr;
+  std::unique_ptr<PayloadTypeSuggesterForTests> owned_pt_suggester_;
+
  // Sequence checker for outgoing network traffic. Could be the network thread.
  // Could also be a pacer owned thread or TQ such as the TaskQueueSender.
  RTC_NO_UNIQUE_ADDRESS SequenceChecker sent_packet_sequence_checker_;
  std::optional<rtc::SentPacket> last_sent_packet_
      RTC_GUARDED_BY(sent_packet_sequence_checker_);
+  // Declared last since it will issue callbacks from a task queue. Declaring it
+  // last ensures that it is destroyed first and any running tasks are finished.
+  const std::unique_ptr<RtpTransportControllerSendInterface> transport_send_;
 };
 }  // namespace internal

@ -1119,6 +1145,24 @@ RtpTransportControllerSendInterface* Call::GetTransportControllerSend() {
  return transport_send_.get();
 }

+PayloadTypeSuggester* Call::GetPayloadTypeSuggester() {
+  // TODO: https://issues.webrtc.org/360058654 - make mandatory at
+  // initialization. Currently, only some channels use it.
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  if (!pt_suggester_) {
+    // Make something that will work most of the time for testing.
+    owned_pt_suggester_ = std::make_unique<PayloadTypeSuggesterForTests>();
+    SetPayloadTypeSuggester(owned_pt_suggester_.get());
+  }
+  return pt_suggester_;
+}
+
+void Call::SetPayloadTypeSuggester(PayloadTypeSuggester* suggester) {
+  RTC_CHECK(!pt_suggester_)
+      << "SetPayloadTypeSuggester can be called only once";
+  pt_suggester_ = suggester;
+}
+
 Call::Stats Call::GetStats() const {
  RTC_DCHECK_RUN_ON(worker_thread_);

--- a/call/call.h
+++ b/call/call.h
@ -28,9 +28,11 @@
 #include "call/call_config.h"
 #include "call/flexfec_receive_stream.h"
 #include "call/packet_receiver.h"
+#include "call/payload_type.h"
 #include "call/rtp_transport_controller_send_interface.h"
 #include "call/video_receive_stream.h"
 #include "call/video_send_stream.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/network/sent_packet.h"
 #include "video/config/video_encoder_config.h"

@ -109,6 +111,18 @@ class Call {
  // remove this method interface.
  virtual RtpTransportControllerSendInterface* GetTransportControllerSend() = 0;

+  // A class that keeps track of payload types on the transport(s), and
+  // suggests new ones when needed.
+  virtual PayloadTypeSuggester* GetPayloadTypeSuggester() {
+    // TODO: https://issues.webrtc.org/360058654 - make pure virtual
+    RTC_CHECK_NOTREACHED();
+    return nullptr;
+  }
+  virtual void SetPayloadTypeSuggester(PayloadTypeSuggester* suggester) {
+    // TODO: https://issues.webrtc.org/360058654 - make pure virtual
+    RTC_CHECK_NOTREACHED();
+  }
+
  // Returns the call statistics, such as estimated send and receive bandwidth,
  // pacing delay, etc.
  virtual Stats GetStats() const = 0;
--- a/call/payload_type.h
+++ b/call/payload_type.h
@ -0,0 +1,44 @@
+/*
+ *  Copyright 2024 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.
+ */
+
+#ifndef CALL_PAYLOAD_TYPE_H_
+#define CALL_PAYLOAD_TYPE_H_
+
+#include <cstdint>
+#include <string>
+
+#include "api/rtc_error.h"
+#include "media/base/codec.h"
+#include "rtc_base/strong_alias.h"
+
+namespace webrtc {
+
+class PayloadType : public StrongAlias<class PayloadTypeTag, uint8_t> {
+ public:
+  // Non-explicit conversions from and to ints are to be deprecated and
+  // removed once calling code is upgraded.
+  PayloadType(uint8_t pt) { value_ = pt; }                // NOLINT: explicit
+  constexpr operator uint8_t() const& { return value_; }  // NOLINT: Explicit
+};
+
+class PayloadTypeSuggester {
+ public:
+  virtual ~PayloadTypeSuggester() = default;
+  // Suggest a payload type for a given codec on a given media section.
+  // Media section is indicated by MID.
+  // The function will either return a PT already in use on the connection
+  // or a newly suggested one.
+  virtual RTCErrorOr<PayloadType> SuggestPayloadType(const std::string& mid,
+                                                     cricket::Codec codec) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // CALL_PAYLOAD_TYPE_H_
--- a/call/payload_type_picker.cc
+++ b/call/payload_type_picker.cc
@ -8,15 +8,20 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "pc/payload_type_picker.h"
+#include "call/payload_type_picker.h"

 #include <algorithm>
+#include <set>
 #include <utility>
 #include <vector>

 #include "absl/strings/match.h"
+#include "api/audio_codecs/audio_format.h"
 #include "api/rtc_error.h"
+#include "call/payload_type.h"
 #include "media/base/codec.h"
+#include "media/base/media_constants.h"
+#include "rtc_base/logging.h"

 namespace webrtc {

--- a/call/payload_type_picker.h
+++ b/call/payload_type_picker.h
@ -8,8 +8,8 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#ifndef PC_PAYLOAD_TYPE_PICKER_H_
-#define PC_PAYLOAD_TYPE_PICKER_H_
+#ifndef CALL_PAYLOAD_TYPE_PICKER_H_
+#define CALL_PAYLOAD_TYPE_PICKER_H_

 #include <map>
 #include <set>
@ -18,18 +18,10 @@

 #include "api/rtc_error.h"
 #include "media/base/codec.h"
-#include "rtc_base/strong_alias.h"
+#include "call/payload_type.h"

 namespace webrtc {

-class PayloadType : public StrongAlias<class PayloadTypeTag, uint8_t> {
- public:
-  // Non-explicit conversions from and to ints are to be deprecated and
-  // removed once calling code is upgraded.
-  PayloadType(uint8_t pt) { value_ = pt; }                // NOLINT: explicit
-  constexpr operator uint8_t() const& { return value_; }  // NOLINT: Explicit
-};
-
 class PayloadTypeRecorder;

 class PayloadTypePicker {
@ -84,4 +76,4 @@ class PayloadTypeRecorder {

 }  // namespace webrtc

-#endif  //  PC_PAYLOAD_TYPE_PICKER_H_
+#endif  //  CALL_PAYLOAD_TYPE_PICKER_H_
--- a/call/payload_type_picker_unittest.cc
+++ b/call/payload_type_picker_unittest.cc
@ -8,9 +8,11 @@
 *  be found in the AUTHORS file in the root of the source tree.
 */

-#include "pc/payload_type_picker.h"
+#include "call/payload_type_picker.h"

+#include "call/payload_type.h"
 #include "media/base/codec.h"
+#include "media/base/media_constants.h"
 #include "test/gtest.h"

 namespace webrtc {
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@ -231,9 +231,15 @@ rtc_library("media_engine") {
    ":media_channel",
    ":media_channel_impl",
    ":rtc_media_config",
+    ":stream_params",
    ":video_common",
+    "../api:array_view",
+    "../api:audio_options_api",
    "../api:field_trials_view",
+    "../api:rtc_error",
    "../api:rtp_parameters",
+    "../api:scoped_refptr",
+    "../api/audio:audio_device",
    "../api/audio_codecs:audio_codecs_api",
    "../api/crypto:options",
    "../api/video:video_bitrate_allocation",
--- a/media/DEPS
+++ b/media/DEPS
@ -34,4 +34,8 @@ specific_include_rules = {
  ".*fake_webrtc_call\.cc": [
    "+video/config",
  ],
+  # temporary
+  ".*webrtc_voice_engine\.cc": [
+    "+pc/payload_type_picker.h",
+  ],
 }
--- a/media/base/media_engine.h
+++ b/media/base/media_engine.h
@ -11,22 +11,27 @@
 #ifndef MEDIA_BASE_MEDIA_ENGINE_H_
 #define MEDIA_BASE_MEDIA_ENGINE_H_

+#include <cstdint>
 #include <memory>
-#include <string>
+#include <optional>
 #include <vector>

-#include "api/audio_codecs/audio_decoder_factory.h"
-#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/array_view.h"
+#include "api/audio/audio_device.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
+#include "api/audio_options.h"
 #include "api/crypto/crypto_options.h"
 #include "api/field_trials_view.h"
+#include "api/rtc_error.h"
 #include "api/rtp_parameters.h"
+#include "api/scoped_refptr.h"
 #include "api/video/video_bitrate_allocator_factory.h"
 #include "call/audio_state.h"
 #include "media/base/codec.h"
 #include "media/base/media_channel.h"
-#include "media/base/media_channel_impl.h"
 #include "media/base/media_config.h"
-#include "media/base/video_common.h"
+#include "media/base/stream_params.h"
+#include "rtc_base/checks.h"
 #include "rtc_base/system/file_wrapper.h"

 namespace webrtc {
@ -121,6 +126,10 @@ class VoiceEngineInterface : public RtpHeaderExtensionQueryInterface {
    return nullptr;
  }

+  // Legacy: Retrieve list of supported codecs.
+  // + protection codecs, and assigns PT numbers that may have to be
+  // reassigned.
+  // TODO: https://issues.webrtc.org/360058654 - deprecate and remove.
  virtual const std::vector<Codec>& send_codecs() const = 0;
  virtual const std::vector<Codec>& recv_codecs() const = 0;

@ -166,7 +175,10 @@ class VideoEngineInterface : public RtpHeaderExtensionQueryInterface {
    return nullptr;
  }

-  // Retrieve list of supported codecs.
+  // Legacy: Retrieve list of supported codecs.
+  // + protection codecs, and assigns PT numbers that may have to be
+  // reassigned.
+  // TODO: https://issues.webrtc.org/360058654 - deprecate and remove.
  virtual std::vector<Codec> send_codecs() const = 0;
  virtual std::vector<Codec> recv_codecs() const = 0;
  // As above, but if include_rtx is false, don't include RTX codecs.
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@ -145,7 +145,10 @@ rtc_source_set("dtls_srtp_transport") {
 }

 rtc_source_set("dtls_transport") {
-  visibility = [ ":*" ]
+  visibility = [
+    ":*",
+    "../test/*",
+  ]
  sources = [
    "dtls_transport.cc",
    "dtls_transport.h",
@ -210,7 +213,6 @@ rtc_source_set("jsep_transport") {
  deps = [
    ":dtls_srtp_transport",
    ":dtls_transport",
-    ":payload_type_picker",
    ":rtcp_mux_filter",
    ":rtp_transport",
    ":rtp_transport_internal",
@ -226,6 +228,7 @@ rtc_source_set("jsep_transport") {
    "../api:scoped_refptr",
    "../api:sequence_checker",
    "../api/transport:datagram_transport_interface",
+    "../call:payload_type_picker",
    "../media:rtc_data_sctp_transport_internal",
    "../p2p:dtls_transport",
    "../p2p:dtls_transport_internal",
@ -304,7 +307,11 @@ rtc_source_set("jsep_transport_controller") {
    "../api/transport:datagram_transport_interface",
    "../api/transport:enums",
    "../api/transport:sctp_transport_factory_interface",
+    "../call:payload_type",
+    "../call:payload_type_picker",
+    "../media:codec",
    "../media:rtc_data_sctp_transport_internal",
+    "../modules/rtp_rtcp:rtp_rtcp_format",
    "../p2p:connection",
    "../p2p:dtls_transport",
    "../p2p:dtls_transport_factory",
@ -331,6 +338,7 @@ rtc_source_set("jsep_transport_controller") {
    "../rtc_base/third_party/sigslot",
    "//third_party/abseil-cpp/absl/algorithm:container",
    "//third_party/abseil-cpp/absl/functional:any_invocable",
+    "//third_party/abseil-cpp/absl/strings:string_view",
  ]
 }

@ -406,19 +414,6 @@ rtc_source_set("media_stream_track_proxy") {
  ]
 }

-rtc_library("payload_type_picker") {
-  sources = [
-    "payload_type_picker.cc",
-    "payload_type_picker.h",
-  ]
-  deps = [
-    "../api:rtc_error",
-    "../media:codec",
-    "../rtc_base:strong_alias",
-    "//third_party/abseil-cpp/absl/strings",
-  ]
-}
-
 rtc_source_set("peer_connection_factory_proxy") {
  visibility = [ ":*" ]
  sources = [ "peer_connection_factory_proxy.h" ]
@ -1084,7 +1079,6 @@ rtc_source_set("peer_connection") {
    ":ice_server_parsing",
    ":jsep_transport_controller",
    ":legacy_stats_collector",
-    ":payload_type_picker",
    ":peer_connection_internal",
    ":peer_connection_message_handler",
    ":rtc_stats_collector",
@ -1130,6 +1124,7 @@ rtc_source_set("peer_connection") {
    "../api/transport:enums",
    "../api/video:video_codec_constants",
    "../call:call_interfaces",
+    "../call:payload_type_picker",
    "../media:media_channel",
    "../media:media_engine",
    "../media:rid_description",
@ -1953,7 +1948,6 @@ if (rtc_include_tests && !build_with_chromium) {
      "jsep_transport_controller_unittest.cc",
      "jsep_transport_unittest.cc",
      "media_session_unittest.cc",
-      "payload_type_picker_unittest.cc",
      "rtcp_mux_filter_unittest.cc",
      "rtp_transport_unittest.cc",
      "sctp_transport_unittest.cc",
@ -1981,7 +1975,6 @@ if (rtc_include_tests && !build_with_chromium) {
      ":libjingle_peerconnection",
      ":media_protocol_names",
      ":media_session",
-      ":payload_type_picker",
      ":pc_test_utils",
      ":rtc_pc",
      ":rtcp_mux_filter",
@ -2017,6 +2010,7 @@ if (rtc_include_tests && !build_with_chromium) {
      "../api/video:builtin_video_bitrate_allocator_factory",
      "../api/video:recordable_encoded_frame",
      "../api/video/test:mock_recordable_encoded_frame",
+      "../call:payload_type_picker",
      "../call:rtp_interfaces",
      "../call:rtp_receiver",
      "../media:codec",
--- a/pc/jsep_transport.h
+++ b/pc/jsep_transport.h
@ -12,7 +12,6 @@
 #define PC_JSEP_TRANSPORT_H_

 #include <functional>
-#include <map>
 #include <memory>
 #include <optional>
 #include <string>
@ -25,16 +24,13 @@
 #include "api/scoped_refptr.h"
 #include "api/sequence_checker.h"
 #include "api/transport/data_channel_transport_interface.h"
+#include "call/payload_type_picker.h"
 #include "media/sctp/sctp_transport_internal.h"
-#include "p2p/base/dtls_transport.h"
 #include "p2p/base/dtls_transport_internal.h"
 #include "p2p/base/ice_transport_internal.h"
-#include "p2p/base/p2p_constants.h"
 #include "p2p/base/transport_description.h"
-#include "p2p/base/transport_info.h"
 #include "pc/dtls_srtp_transport.h"
 #include "pc/dtls_transport.h"
-#include "pc/payload_type_picker.h"
 #include "pc/rtcp_mux_filter.h"
 #include "pc/rtp_transport.h"
 #include "pc/rtp_transport_internal.h"
@ -42,7 +38,6 @@
 #include "pc/session_description.h"
 #include "pc/srtp_transport.h"
 #include "pc/transport_stats.h"
-#include "rtc_base/checks.h"
 #include "rtc_base/rtc_certificate.h"
 #include "rtc_base/ssl_fingerprint.h"
 #include "rtc_base/ssl_stream_adapter.h"
--- a/pc/jsep_transport_controller.h
+++ b/pc/jsep_transport_controller.h
@ -18,11 +18,11 @@
 #include <memory>
 #include <optional>
 #include <string>
-#include <type_traits>
 #include <utility>
 #include <vector>

 #include "absl/functional/any_invocable.h"
+#include "absl/strings/string_view.h"
 #include "api/async_dns_resolver.h"
 #include "api/candidate.h"
 #include "api/crypto/crypto_options.h"
@ -37,7 +37,10 @@
 #include "api/sequence_checker.h"
 #include "api/transport/data_channel_transport_interface.h"
 #include "api/transport/sctp_transport_factory_interface.h"
-#include "media/sctp/sctp_transport_internal.h"
+#include "call/payload_type.h"
+#include "call/payload_type_picker.h"
+#include "media/base/codec.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "p2p/base/dtls_transport.h"
 #include "p2p/base/dtls_transport_factory.h"
 #include "p2p/base/dtls_transport_internal.h"
@ -59,9 +62,7 @@
 #include "pc/srtp_transport.h"
 #include "pc/transport_stats.h"
 #include "rtc_base/callback_list.h"
-#include "rtc_base/checks.h"
 #include "rtc_base/copy_on_write_buffer.h"
-#include "rtc_base/crypto_random.h"
 #include "rtc_base/rtc_certificate.h"
 #include "rtc_base/ssl_certificate.h"
 #include "rtc_base/ssl_stream_adapter.h"
@ -76,7 +77,8 @@ class PacketTransportInternal;

 namespace webrtc {

-class JsepTransportController : public sigslot::has_slots<> {
+class JsepTransportController : public PayloadTypeSuggester,
+                                public sigslot::has_slots<> {
 public:
  // Used when the RtpTransport/DtlsTransport of the m= section is changed
  // because the section is rejected or BUNDLE is enabled.
@ -240,7 +242,7 @@ class JsepTransportController : public sigslot::has_slots<> {
  // The function will either return a PT already in use on the connection
  // or a newly suggested one.
  RTCErrorOr<PayloadType> SuggestPayloadType(const std::string& mid,
-                                             cricket::Codec codec);
+                                             cricket::Codec codec) override;

  // TODO(deadbeef): GetStats isn't const because all the way down to
  // OpenSSLStreamAdapter, GetSslCipherSuite and GetDtlsSrtpCryptoSuite are not
--- a/test/peer_scenario/BUILD.gn
+++ b/test/peer_scenario/BUILD.gn
@ -27,17 +27,23 @@ if (rtc_include_tests) {
      "..:fileutils",
      "..:frame_generator_capturer",
      "..:test_support",
+      "../../api:array_view",
      "../../api:candidate",
      "../../api:create_time_controller",
      "../../api:libjingle_peerconnection_api",
      "../../api:network_emulation_manager_api",
      "../../api:rtc_stats_api",
+      "../../api:scoped_refptr",
+      "../../api:sequence_checker",
      "../../api:time_controller",
      "../../api/audio_codecs:builtin_audio_decoder_factory",
      "../../api/audio_codecs:builtin_audio_encoder_factory",
      "../../api/environment",
      "../../api/rtc_event_log:rtc_event_log_factory",
      "../../api/task_queue:default_task_queue_factory",
+      "../../api/test/network_emulation",
+      "../../api/transport:datagram_transport_interface",
+      "../../api/transport:enums",
      "../../api/transport:field_trial_based_config",
      "../../api/video_codecs:video_decoder_factory_template",
      "../../api/video_codecs:video_decoder_factory_template_dav1d_adapter",
@ -49,6 +55,9 @@ if (rtc_include_tests) {
      "../../api/video_codecs:video_encoder_factory_template_libvpx_vp8_adapter",
      "../../api/video_codecs:video_encoder_factory_template_libvpx_vp9_adapter",
      "../../api/video_codecs:video_encoder_factory_template_open_h264_adapter",
+      "../../call:payload_type_picker",
+      "../../call:rtp_interfaces",
+      "../../call:rtp_receiver",
      "../../media:rtc_audio_video",
      "../../media:rtp_utils",
      "../../modules/audio_device:test_audio_device_module",
@ -57,13 +66,19 @@ if (rtc_include_tests) {
      "../../p2p:rtc_p2p",
      "../../p2p:transport_description",
      "../../pc:channel",
+      "../../pc:dtls_transport",
      "../../pc:jsep_transport_controller",
      "../../pc:pc_test_utils",
      "../../pc:rtp_transport_internal",
      "../../pc:session_description",
+      "../../rtc_base:checks",
+      "../../rtc_base:crypto_random",
+      "../../rtc_base:macromagic",
      "../../rtc_base:null_socket_server",
+      "../../rtc_base:ssl",
      "../../rtc_base:stringutils",
      "../../rtc_base:task_queue_for_test",
+      "../../rtc_base/third_party/sigslot",
      "../../test:explicit_key_value_config",
      "../../test:scoped_key_value_config",
      "../logging:log_writer",
--- a/test/peer_scenario/scenario_connection.cc
+++ b/test/peer_scenario/scenario_connection.cc
@ -9,15 +9,46 @@
 */
 #include "test/peer_scenario/scenario_connection.h"

-#include "absl/memory/memory.h"
+#include <cstdint>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/candidate.h"
+#include "api/environment/environment.h"
+#include "api/jsep.h"
+#include "api/peer_connection_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/test/network_emulation/network_emulation_interfaces.h"
+#include "api/test/network_emulation_manager.h"
+#include "api/transport/data_channel_transport_interface.h"
+#include "api/transport/enums.h"
+#include "call/payload_type_picker.h"
+#include "call/rtp_demuxer.h"
+#include "call/rtp_packet_sink_interface.h"
 #include "media/base/rtp_utils.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "p2p/base/dtls_transport_internal.h"
+#include "p2p/base/p2p_constants.h"
+#include "p2p/base/port_allocator.h"
+#include "p2p/base/transport_description.h"
 #include "p2p/client/basic_port_allocator.h"
-#include "pc/channel.h"
+#include "pc/dtls_transport.h"
 #include "pc/jsep_transport_controller.h"
 #include "pc/rtp_transport_internal.h"
 #include "pc/session_description.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/crypto_random.h"
+#include "rtc_base/rtc_certificate.h"
+#include "rtc_base/ssl_fingerprint.h"
+#include "rtc_base/ssl_identity.h"
 #include "rtc_base/task_queue_for_test.h"
+#include "rtc_base/third_party/sigslot/sigslot.h"
+#include "rtc_base/thread_annotations.h"
+#include "test/network/network_emulation_manager.h"

 namespace webrtc {
 class ScenarioIceConnectionImpl : public ScenarioIceConnection,