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}
diff --git a/call/BUILD.gn b/call/BUILD.gn
index 1094b48..a74e8ac 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -38,6 +38,8 @@
deps = [
":audio_sender_interface",
+ ":payload_type",
+ ":payload_type",
":receive_stream_interface",
":rtp_interfaces",
":video_stream_api",
@@ -307,6 +309,9 @@
":bitrate_allocator",
":call_interfaces",
":fake_network",
+ ":payload_type",
+ ":payload_type",
+ ":payload_type_picker",
":receive_stream_interface",
":rtp_interfaces",
":rtp_receiver",
@@ -317,6 +322,7 @@
"../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 @@
"../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("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 @@
"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 @@
":call",
":call_interfaces",
":mock_rtp_interfaces",
+ ":payload_type",
+ ":payload_type_picker",
":rtp_interfaces",
":rtp_receiver",
":rtp_sender",
@@ -527,6 +563,8 @@
"../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",
diff --git a/call/DEPS b/call/DEPS
index daff3e9..1fe352d 100644
--- a/call/DEPS
+++ b/call/DEPS
@@ -52,4 +52,22 @@
"+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",
+ ]
}
diff --git a/call/call.cc b/call/call.cc
index 0b1ca6f..cf94cb8 100644
--- 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 {
+// 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 @@
RtpTransportControllerSendInterface* GetTransportControllerSend() override;
+ PayloadTypeSuggester* GetPayloadTypeSuggester() override;
+ void SetPayloadTypeSuggester(PayloadTypeSuggester* suggester) override;
+
Stats GetStats() const override;
const FieldTrialsView& trials() const override;
@@ -465,17 +487,21 @@
// 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 @@
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_);
diff --git a/call/call.h b/call/call.h
index 6c6bc46..313e8a8 100644
--- 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 @@
// 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;
diff --git a/call/payload_type.h b/call/payload_type.h
new file mode 100644
index 0000000..428096b
--- /dev/null
+++ 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_
diff --git a/pc/payload_type_picker.cc b/call/payload_type_picker.cc
similarity index 97%
rename from pc/payload_type_picker.cc
rename to call/payload_type_picker.cc
index 8720b20..4c5936b 100644
--- a/pc/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 {
diff --git a/pc/payload_type_picker.h b/call/payload_type_picker.h
similarity index 82%
rename from pc/payload_type_picker.h
rename to call/payload_type_picker.h
index 4726a31..0b2141c 100644
--- a/pc/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 @@
} // namespace webrtc
-#endif // PC_PAYLOAD_TYPE_PICKER_H_
+#endif // CALL_PAYLOAD_TYPE_PICKER_H_
diff --git a/pc/payload_type_picker_unittest.cc b/call/payload_type_picker_unittest.cc
similarity index 97%
rename from pc/payload_type_picker_unittest.cc
rename to call/payload_type_picker_unittest.cc
index 71a3383..e1704cf 100644
--- a/pc/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 {
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 584ef91..32af232 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -231,9 +231,15 @@
":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",
diff --git a/media/DEPS b/media/DEPS
index 7fbbc0f..bcdbcac 100644
--- a/media/DEPS
+++ b/media/DEPS
@@ -34,4 +34,8 @@
".*fake_webrtc_call\.cc": [
"+video/config",
],
+ # temporary
+ ".*webrtc_voice_engine\.cc": [
+ "+pc/payload_type_picker.h",
+ ],
}
diff --git a/media/base/media_engine.h b/media/base/media_engine.h
index b13d7df..780bac0 100644
--- 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 @@
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 @@
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.
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index c4aa4de..5e11f04 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -145,7 +145,10 @@
}
rtc_source_set("dtls_transport") {
- visibility = [ ":*" ]
+ visibility = [
+ ":*",
+ "../test/*",
+ ]
sources = [
"dtls_transport.cc",
"dtls_transport.h",
@@ -210,7 +213,6 @@
deps = [
":dtls_srtp_transport",
":dtls_transport",
- ":payload_type_picker",
":rtcp_mux_filter",
":rtp_transport",
":rtp_transport_internal",
@@ -226,6 +228,7 @@
"../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 @@
"../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_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_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 @@
":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 @@
"../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 @@
"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 @@
":libjingle_peerconnection",
":media_protocol_names",
":media_session",
- ":payload_type_picker",
":pc_test_utils",
":rtc_pc",
":rtcp_mux_filter",
@@ -2017,6 +2010,7 @@
"../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",
diff --git a/pc/jsep_transport.h b/pc/jsep_transport.h
index 005e606..56840a3 100644
--- 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"
diff --git a/pc/jsep_transport_controller.h b/pc/jsep_transport_controller.h
index d18a375..2846dab 100644
--- 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 @@
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 @@
// 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
diff --git a/test/peer_scenario/BUILD.gn b/test/peer_scenario/BUILD.gn
index dda6345..bf17818 100644
--- a/test/peer_scenario/BUILD.gn
+++ b/test/peer_scenario/BUILD.gn
@@ -27,17 +27,23 @@
"..: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 @@
"../../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 @@
"../../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",
diff --git a/test/peer_scenario/scenario_connection.cc b/test/peer_scenario/scenario_connection.cc
index db3a290..856a092 100644
--- 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,
