Run IWYU on some files I intend to work on
and files that broke when I fixed the first set.
Bug: webrtc:42226242
Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42429}
diff --git a/api/BUILD.gn b/api/BUILD.gn
index f25db96..b0f95eb 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -393,11 +393,16 @@
"frame_transformer_interface.h",
]
deps = [
+ ":array_view",
":make_ref_counted",
":ref_count",
":scoped_refptr",
+ "../rtc_base:refcount",
+ "../rtc_base/system:rtc_export",
+ "units:timestamp",
"video:encoded_frame",
"video:video_frame_metadata",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/api/frame_transformer_interface.h b/api/frame_transformer_interface.h
index 03f7b48..6f632bc 100644
--- a/api/frame_transformer_interface.h
+++ b/api/frame_transformer_interface.h
@@ -11,13 +11,17 @@
#ifndef API_FRAME_TRANSFORMER_INTERFACE_H_
#define API_FRAME_TRANSFORMER_INTERFACE_H_
+#include <cstdint>
#include <memory>
#include <string>
+#include "absl/types/optional.h"
+#include "api/array_view.h"
#include "api/ref_count.h"
#include "api/scoped_refptr.h"
-#include "api/video/encoded_frame.h"
+#include "api/units/timestamp.h"
#include "api/video/video_frame_metadata.h"
+#include "rtc_base/system/rtc_export.h"
namespace webrtc {
@@ -104,7 +108,7 @@
};
// Objects implement this interface to be notified with the transformed frame.
-class TransformedFrameCallback : public rtc::RefCountInterface {
+class TransformedFrameCallback : public RefCountInterface {
public:
virtual void OnTransformedFrame(
std::unique_ptr<TransformableFrameInterface> frame) = 0;
@@ -121,7 +125,7 @@
// Transforms encoded frames. The transformed frame is sent in a callback using
// the TransformedFrameCallback interface (see above).
-class FrameTransformerInterface : public rtc::RefCountInterface {
+class FrameTransformerInterface : public RefCountInterface {
public:
// Transforms `frame` using the implementing class' processing logic.
virtual void Transform(
diff --git a/api/rtp_receiver_interface.cc b/api/rtp_receiver_interface.cc
index ffa4216..4868f2c 100644
--- a/api/rtp_receiver_interface.cc
+++ b/api/rtp_receiver_interface.cc
@@ -10,6 +10,16 @@
#include "api/rtp_receiver_interface.h"
+#include <string>
+#include <vector>
+
+#include "api/crypto/frame_decryptor_interface.h"
+#include "api/dtls_transport_interface.h"
+#include "api/frame_transformer_interface.h"
+#include "api/media_stream_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/transport/rtp/rtp_source.h"
+
namespace webrtc {
std::vector<std::string> RtpReceiverInterface::stream_ids() const {
diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 688a7ad..fb9e904 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -161,12 +161,16 @@
":audio",
":audio_end_to_end_test",
":channel_receive_unittest",
+ "../api:array_view",
+ "../api:frame_transformer_interface",
"../api:libjingle_peerconnection_api",
+ "../api:make_ref_counted",
"../api:mock_audio_mixer",
"../api:mock_frame_decryptor",
"../api:mock_frame_encryptor",
"../api:mock_frame_transformer",
"../api:mock_transformable_audio_frame",
+ "../api:rtp_headers",
"../api:scoped_refptr",
"../api/audio:audio_frame_api",
"../api/audio:audio_processing_statistics",
@@ -189,6 +193,7 @@
"../call:rtp_sender",
"../common_audio",
"../logging:mocks",
+ "../modules/audio_coding:audio_coding_module_typedefs",
"../modules/audio_device:audio_device_impl", # For TestAudioDeviceModule
"../modules/audio_device:mock_audio_device",
"../modules/audio_mixer:audio_mixer_impl",
@@ -220,6 +225,7 @@
"utility:utility_tests",
"//testing/gtest",
"//third_party/abseil-cpp/absl/memory",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
diff --git a/audio/channel_receive_frame_transformer_delegate.cc b/audio/channel_receive_frame_transformer_delegate.cc
index 814127b..0b8eacf 100644
--- a/audio/channel_receive_frame_transformer_delegate.cc
+++ b/audio/channel_receive_frame_transformer_delegate.cc
@@ -10,9 +10,18 @@
#include "audio/channel_receive_frame_transformer_delegate.h"
+#include <cstdint>
+#include <memory>
#include <string>
#include <utility>
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/frame_transformer_interface.h"
+#include "api/rtp_headers.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/task_queue_base.h"
#include "rtc_base/buffer.h"
namespace webrtc {
diff --git a/audio/channel_receive_frame_transformer_delegate.h b/audio/channel_receive_frame_transformer_delegate.h
index d572ffe..0502458 100644
--- a/audio/channel_receive_frame_transformer_delegate.h
+++ b/audio/channel_receive_frame_transformer_delegate.h
@@ -15,6 +15,7 @@
#include <string>
#include "api/frame_transformer_interface.h"
+#include "api/rtp_headers.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_base.h"
#include "rtc_base/system/no_unique_address.h"
diff --git a/audio/channel_receive_frame_transformer_delegate_unittest.cc b/audio/channel_receive_frame_transformer_delegate_unittest.cc
index 8b819f1..3ab2b49 100644
--- a/audio/channel_receive_frame_transformer_delegate_unittest.cc
+++ b/audio/channel_receive_frame_transformer_delegate_unittest.cc
@@ -10,12 +10,19 @@
#include "audio/channel_receive_frame_transformer_delegate.h"
+#include <cstdint>
#include <memory>
#include <utility>
+#include "api/array_view.h"
+#include "api/frame_transformer_interface.h"
+#include "api/make_ref_counted.h"
+#include "api/rtp_headers.h"
+#include "api/scoped_refptr.h"
#include "api/test/mock_frame_transformer.h"
#include "api/test/mock_transformable_audio_frame.h"
#include "audio/channel_send_frame_transformer_delegate.h"
+#include "rtc_base/thread.h"
#include "test/gmock.h"
#include "test/gtest.h"
diff --git a/audio/channel_send_frame_transformer_delegate_unittest.cc b/audio/channel_send_frame_transformer_delegate_unittest.cc
index e8b7aef..0ed8677 100644
--- a/audio/channel_send_frame_transformer_delegate_unittest.cc
+++ b/audio/channel_send_frame_transformer_delegate_unittest.cc
@@ -10,13 +10,20 @@
#include "audio/channel_send_frame_transformer_delegate.h"
+#include <cstdint>
#include <memory>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/frame_transformer_interface.h"
+#include "api/make_ref_counted.h"
+#include "api/scoped_refptr.h"
#include "api/test/mock_frame_transformer.h"
#include "api/test/mock_transformable_audio_frame.h"
+#include "modules/audio_coding/include/audio_coding_module_typedefs.h"
#include "rtc_base/task_queue_for_test.h"
#include "test/gmock.h"
#include "test/gtest.h"
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 6244b6b..8b5fd93 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -358,6 +358,7 @@
":media_constants",
"../api:rtp_parameters",
"../api/audio_codecs:audio_codecs_api",
+ "../api/video_codecs:scalability_mode",
"../api/video_codecs:video_codecs_api",
"../rtc_base:checks",
"../rtc_base:logging",
diff --git a/media/base/codec.cc b/media/base/codec.cc
index 2e7cc6a..820cb97 100644
--- a/media/base/codec.cc
+++ b/media/base/codec.cc
@@ -10,11 +10,21 @@
#include "media/base/codec.h"
+#include <algorithm>
+#include <cstddef>
+#include <iterator>
+#include <string>
+#include <vector>
+
#include "absl/algorithm/container.h"
#include "absl/strings/match.h"
+#include "absl/types/optional.h"
#include "api/audio_codecs/audio_format.h"
+#include "api/media_types.h"
+#include "api/rtp_parameters.h"
#include "api/video_codecs/av1_profile.h"
#include "api/video_codecs/h264_profile_level_id.h"
+#include "api/video_codecs/sdp_video_format.h"
#ifdef RTC_ENABLE_H265
#include "api/video_codecs/h265_profile_tier_level.h"
#endif
diff --git a/media/base/codec.h b/media/base/codec.h
index 99a6852..5230779 100644
--- a/media/base/codec.h
+++ b/media/base/codec.h
@@ -11,8 +11,7 @@
#ifndef MEDIA_BASE_CODEC_H_
#define MEDIA_BASE_CODEC_H_
-#include <map>
-#include <set>
+#include <cstddef>
#include <string>
#include <vector>
@@ -21,6 +20,7 @@
#include "absl/types/optional.h"
#include "api/audio_codecs/audio_format.h"
#include "api/rtp_parameters.h"
+#include "api/video_codecs/scalability_mode.h"
#include "api/video_codecs/sdp_video_format.h"
#include "media/base/media_constants.h"
#include "rtc_base/system/rtc_export.h"
diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index 0c878a5..58317c1 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -308,6 +308,7 @@
"../../api/video:video_frame_type",
"../../api/video:video_layers_allocation",
"../../api/video:video_rtp_headers",
+ "../../api/video_codecs:video_codecs_api",
"../../call:rtp_interfaces",
"../../call:video_stream_api",
"../../common_video",
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
index ae80f76..8c24160 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
+++ b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.cc
@@ -10,15 +10,31 @@
#include "modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h"
+#include <cstddef>
+#include <cstdint>
+#include <memory>
#include <string>
#include <utility>
#include <vector>
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/frame_transformer_interface.h"
+#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/task_queue_factory.h"
-#include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h"
+#include "api/transport/rtp/dependency_descriptor.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "api/video/encoded_image.h"
+#include "api/video/video_codec_type.h"
+#include "api/video/video_frame_metadata.h"
+#include "api/video/video_frame_type.h"
+#include "api/video/video_layers_allocation.h"
+#include "api/video_codecs/video_codec.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "rtc_base/checks.h"
-#include "rtc_base/logging.h"
+#include "rtc_base/synchronization/mutex.h"
namespace webrtc {
namespace {
diff --git a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h
index 243f22c..a32c708 100644
--- a/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h
+++ b/modules/rtp_rtcp/source/rtp_sender_video_frame_transformer_delegate.h
@@ -21,7 +21,9 @@
#include "api/task_queue/task_queue_factory.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
+#include "api/video/encoded_image.h"
#include "api/video/video_layers_allocation.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
#include "rtc_base/synchronization/mutex.h"
namespace webrtc {
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index d1b6b29..25dd280 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -451,7 +451,15 @@
sources = [ "rtp_receiver_proxy.h" ]
deps = [
":proxy",
+ "../api:dtls_transport_interface",
+ "../api:frame_transformer_interface",
"../api:libjingle_peerconnection_api",
+ "../api:media_stream_interface",
+ "../api:rtp_parameters",
+ "../api:scoped_refptr",
+ "../api/crypto:frame_decryptor_interface",
+ "../api/transport/rtp:rtp_source",
+ "//third_party/abseil-cpp/absl/types:optional",
]
}
@@ -988,6 +996,7 @@
"../api:dtls_transport_interface",
"../api:field_trials_view",
"../api:libjingle_peerconnection_api",
+ "../api:make_ref_counted",
"../api:media_stream_interface",
"../api:rtc_error",
"../api:rtp_parameters",
@@ -998,8 +1007,10 @@
"../api/crypto:options",
"../api/video:builtin_video_bitrate_allocator_factory",
"../api/video:video_bitrate_allocator_factory",
+ "../api/video:video_codec_constants",
"../media:codec",
"../media:media_channel",
+ "../media:media_engine",
"../media:rid_description",
"../media:stream_params",
"../p2p:connection",
@@ -1517,6 +1528,7 @@
"../api/task_queue",
"../api/task_queue:pending_task_safety_flag",
"../api/video:video_bitrate_allocator_factory",
+ "../api/video_codecs:scalability_mode",
"../media:codec",
"../media:media_channel",
"../media:media_channel_impl",
@@ -2399,6 +2411,7 @@
"../test:run_loop",
"../test:scoped_key_value_config",
"../test/pc/sctp:fake_sctp_transport",
+ "//testing/gtest",
"//third_party/abseil-cpp/absl/algorithm:container",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings",
diff --git a/pc/rtp_receiver.cc b/pc/rtp_receiver.cc
index 9431aab..ce221a2 100644
--- a/pc/rtp_receiver.cc
+++ b/pc/rtp_receiver.cc
@@ -13,9 +13,12 @@
#include <stddef.h>
#include <atomic>
+#include <string>
#include <utility>
#include <vector>
+#include "api/media_stream_interface.h"
+#include "api/scoped_refptr.h"
#include "pc/media_stream.h"
#include "pc/media_stream_proxy.h"
#include "rtc_base/thread.h"
diff --git a/pc/rtp_receiver.h b/pc/rtp_receiver.h
index 16ab011..53451a3 100644
--- a/pc/rtp_receiver.h
+++ b/pc/rtp_receiver.h
@@ -21,20 +21,11 @@
#include <vector>
#include "absl/types/optional.h"
-#include "api/crypto/frame_decryptor_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/media_stream_interface.h"
-#include "api/media_types.h"
-#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
#include "api/scoped_refptr.h"
-#include "api/video/video_frame.h"
-#include "api/video/video_sink_interface.h"
-#include "api/video/video_source_interface.h"
#include "media/base/media_channel.h"
-#include "media/base/video_broadcaster.h"
-#include "pc/video_track_source.h"
-#include "rtc_base/thread.h"
namespace webrtc {
diff --git a/pc/rtp_receiver_proxy.h b/pc/rtp_receiver_proxy.h
index db677d8..0cdf6ff 100644
--- a/pc/rtp_receiver_proxy.h
+++ b/pc/rtp_receiver_proxy.h
@@ -14,7 +14,16 @@
#include <string>
#include <vector>
+#include "absl/types/optional.h"
+#include "api/crypto/frame_decryptor_interface.h"
+#include "api/dtls_transport_interface.h"
+#include "api/frame_transformer_interface.h"
+#include "api/media_stream_interface.h"
+#include "api/media_types.h"
+#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/transport/rtp/rtp_source.h"
#include "pc/proxy.h"
namespace webrtc {
diff --git a/pc/rtp_sender_receiver_unittest.cc b/pc/rtp_sender_receiver_unittest.cc
index c6c342e..b73edf5 100644
--- a/pc/rtp_sender_receiver_unittest.cc
+++ b/pc/rtp_sender_receiver_unittest.cc
@@ -18,14 +18,15 @@
#include <vector>
#include "absl/algorithm/container.h"
-#include "absl/memory/memory.h"
#include "absl/types/optional.h"
+#include "api/audio_codecs/audio_codec_pair_id.h"
#include "api/audio_options.h"
#include "api/crypto/crypto_options.h"
#include "api/crypto/frame_decryptor_interface.h"
#include "api/crypto/frame_encryptor_interface.h"
#include "api/dtmf_sender_interface.h"
#include "api/environment/environment_factory.h"
+#include "api/make_ref_counted.h"
#include "api/media_stream_interface.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log/rtc_event_log.h"
@@ -51,7 +52,6 @@
#include "p2p/base/p2p_constants.h"
#include "pc/audio_rtp_receiver.h"
#include "pc/audio_track.h"
-#include "pc/channel.h"
#include "pc/dtls_srtp_transport.h"
#include "pc/local_audio_source.h"
#include "pc/media_stream.h"
@@ -63,6 +63,7 @@
#include "rtc_base/checks.h"
#include "rtc_base/gunit.h"
#include "rtc_base/thread.h"
+#include "rtc_base/unique_id_generator.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/run_loop.h"
diff --git a/pc/rtp_transceiver.cc b/pc/rtp_transceiver.cc
index e585667..f0a09ce 100644
--- a/pc/rtp_transceiver.cc
+++ b/pc/rtp_transceiver.cc
@@ -12,25 +12,50 @@
#include <stdint.h>
+#include <algorithm>
+#include <cstddef>
+#include <functional>
#include <iterator>
+#include <memory>
#include <set>
#include <string>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
-#include "absl/memory/memory.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.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/jsep.h"
+#include "api/media_types.h"
+#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
+#include "api/rtp_receiver_interface.h"
+#include "api/rtp_sender_interface.h"
+#include "api/rtp_transceiver_direction.h"
+#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/task_queue/task_queue_base.h"
+#include "api/video/video_bitrate_allocator_factory.h"
+#include "api/video_codecs/scalability_mode.h"
#include "media/base/codec.h"
#include "media/base/media_channel.h"
-#include "media/base/media_channel_impl.h"
-#include "media/base/media_constants.h"
+#include "media/base/media_config.h"
#include "media/base/media_engine.h"
#include "pc/channel.h"
+#include "pc/channel_interface.h"
+#include "pc/connection_context.h"
#include "pc/rtp_media_utils.h"
+#include "pc/rtp_receiver.h"
+#include "pc/rtp_receiver_proxy.h"
+#include "pc/rtp_sender.h"
+#include "pc/rtp_sender_proxy.h"
+#include "pc/rtp_transport_internal.h"
#include "pc/session_description.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
diff --git a/pc/sdp_offer_answer.cc b/pc/sdp_offer_answer.cc
index 68d89e4..8b348ba 100644
--- a/pc/sdp_offer_answer.cc
+++ b/pc/sdp_offer_answer.cc
@@ -12,56 +12,90 @@
#include <algorithm>
#include <cstddef>
+#include <cstdint>
+#include <functional>
#include <iterator>
#include <map>
#include <memory>
#include <queue>
+#include <set>
#include <string>
#include <utility>
+#include <vector>
#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "absl/strings/match.h"
#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
#include "api/array_view.h"
+#include "api/candidate.h"
#include "api/crypto/crypto_options.h"
-#include "api/dtls_transport_interface.h"
-#include "api/field_trials_view.h"
+#include "api/jsep.h"
+#include "api/jsep_ice_candidate.h"
+#include "api/make_ref_counted.h"
+#include "api/media_stream_interface.h"
+#include "api/media_types.h"
+#include "api/peer_connection_interface.h"
+#include "api/rtc_error.h"
#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
#include "api/rtp_sender_interface.h"
+#include "api/rtp_transceiver_direction.h"
+#include "api/rtp_transceiver_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/sequence_checker.h"
+#include "api/set_local_description_observer_interface.h"
+#include "api/set_remote_description_observer_interface.h"
+#include "api/uma_metrics.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
+#include "api/video/video_codec_constants.h"
#include "media/base/codec.h"
+#include "media/base/media_engine.h"
#include "media/base/rid_description.h"
+#include "media/base/stream_params.h"
#include "p2p/base/ice_transport_internal.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/p2p_transport_channel.h"
-#include "p2p/base/port.h"
+#include "p2p/base/port_allocator.h"
#include "p2p/base/transport_description.h"
#include "p2p/base/transport_description_factory.h"
#include "p2p/base/transport_info.h"
#include "pc/channel_interface.h"
+#include "pc/connection_context.h"
#include "pc/dtls_transport.h"
+#include "pc/jsep_transport_controller.h"
#include "pc/legacy_stats_collector.h"
+#include "pc/media_session.h"
#include "pc/media_stream.h"
+#include "pc/media_stream_observer.h"
#include "pc/media_stream_proxy.h"
#include "pc/peer_connection_internal.h"
#include "pc/peer_connection_message_handler.h"
#include "pc/rtp_media_utils.h"
-#include "pc/rtp_receiver_proxy.h"
+#include "pc/rtp_receiver.h"
#include "pc/rtp_sender.h"
#include "pc/rtp_sender_proxy.h"
+#include "pc/rtp_transceiver.h"
+#include "pc/rtp_transmission_manager.h"
+#include "pc/session_description.h"
#include "pc/simulcast_description.h"
+#include "pc/stream_collection.h"
+#include "pc/transceiver_list.h"
#include "pc/usage_pattern.h"
#include "pc/used_ids.h"
#include "pc/webrtc_session_description_factory.h"
+#include "rtc_base/checks.h"
#include "rtc_base/helpers.h"
#include "rtc_base/logging.h"
+#include "rtc_base/operations_chain.h"
#include "rtc_base/rtc_certificate.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/string_encode.h"
#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/thread.h"
#include "rtc_base/trace_event.h"
+#include "rtc_base/weak_ptr.h"
#include "system_wrappers/include/metrics.h"
using cricket::ContentInfo;
diff --git a/pc/sdp_offer_answer.h b/pc/sdp_offer_answer.h
index 88ddfe0..8b845f4 100644
--- a/pc/sdp_offer_answer.h
+++ b/pc/sdp_offer_answer.h
@@ -55,7 +55,6 @@
#include "pc/stream_collection.h"
#include "pc/transceiver_list.h"
#include "pc/webrtc_session_description_factory.h"
-#include "rtc_base/checks.h"
#include "rtc_base/operations_chain.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/thread.h"
