diff --git a/api/BUILD.gn b/api/BUILD.gn
index 0a6c670936..1e4bac22ff 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -206,7 +206,6 @@ rtc_library("libjingle_peerconnection_api") {
     "../modules/audio_processing:audio_processing_statistics",
     "../rtc_base",
     "../rtc_base:checks",
-    "../rtc_base:deprecation",
     "../rtc_base:ip_address",
     "../rtc_base:rtc_base_approved",
     "../rtc_base:socket_address",
diff --git a/api/DEPS b/api/DEPS
index ffa9e98f47..52de6c37c8 100644
--- a/api/DEPS
+++ b/api/DEPS
@@ -45,7 +45,6 @@ specific_include_rules = {
     "+rtc_base/system/rtc_export.h",
     "+rtc_base/system/rtc_export_template.h",
     "+rtc_base/units/unit_base.h",
-    "+rtc_base/deprecation.h",
   ],
 
   "array_view\.h": [
diff --git a/api/audio_codecs/BUILD.gn b/api/audio_codecs/BUILD.gn
index b6292de570..5926f5ec2e 100644
--- a/api/audio_codecs/BUILD.gn
+++ b/api/audio_codecs/BUILD.gn
@@ -33,7 +33,6 @@ rtc_library("audio_codecs_api") {
     "..:bitrate_allocation",
     "..:scoped_refptr",
     "../../rtc_base:checks",
-    "../../rtc_base:deprecation",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base:sanitizer",
     "../../rtc_base/system:rtc_export",
diff --git a/api/audio_codecs/audio_encoder.h b/api/audio_codecs/audio_encoder.h
index fd2d948863..92e42cf107 100644
--- a/api/audio_codecs/audio_encoder.h
+++ b/api/audio_codecs/audio_encoder.h
@@ -16,12 +16,12 @@
 #include <utility>
 #include <vector>
 
+#include "absl/base/attributes.h"
 #include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "api/call/bitrate_allocation.h"
 #include "api/units/time_delta.h"
 #include "rtc_base/buffer.h"
-#include "rtc_base/deprecation.h"
 
 namespace webrtc {
 
@@ -182,12 +182,11 @@ class AudioEncoder {
   // implementation does nothing.
   virtual void SetMaxPlaybackRate(int frequency_hz);
 
-  // This is to be deprecated. Please use |OnReceivedTargetAudioBitrate|
-  // instead.
   // Tells the encoder what average bitrate we'd like it to produce. The
   // encoder is free to adjust or disregard the given bitrate (the default
   // implementation does the latter).
-  RTC_DEPRECATED virtual void SetTargetBitrate(int target_bps);
+  ABSL_DEPRECATED("Use OnReceivedTargetAudioBitrate instead")
+  virtual void SetTargetBitrate(int target_bps);
 
   // Causes this encoder to let go of any other encoders it contains, and
   // returns a pointer to an array where they are stored (which is required to
@@ -210,7 +209,8 @@ class AudioEncoder {
   virtual void OnReceivedUplinkPacketLossFraction(
       float uplink_packet_loss_fraction);
 
-  RTC_DEPRECATED virtual void OnReceivedUplinkRecoverablePacketLossFraction(
+  ABSL_DEPRECATED("")
+  virtual void OnReceivedUplinkRecoverablePacketLossFraction(
       float uplink_recoverable_packet_loss_fraction);
 
   // Provides target audio bitrate to this encoder to allow it to adapt.
diff --git a/api/jsep.h b/api/jsep.h
index dcf821369e..b56cf1d15b 100644
--- a/api/jsep.h
+++ b/api/jsep.h
@@ -28,7 +28,6 @@
 
 #include "absl/types/optional.h"
 #include "api/rtc_error.h"
-#include "rtc_base/deprecation.h"
 #include "rtc_base/ref_count.h"
 #include "rtc_base/system/rtc_export.h"
 
diff --git a/api/jsep_session_description.h b/api/jsep_session_description.h
index e13d85e71c..70ac9398a6 100644
--- a/api/jsep_session_description.h
+++ b/api/jsep_session_description.h
@@ -23,7 +23,6 @@
 #include "api/jsep.h"
 #include "api/jsep_ice_candidate.h"
 #include "rtc_base/constructor_magic.h"
-#include "rtc_base/deprecation.h"
 
 namespace cricket {
 class SessionDescription;
diff --git a/api/rtp_receiver_interface.h b/api/rtp_receiver_interface.h
index 786ea3aceb..8984d71fe8 100644
--- a/api/rtp_receiver_interface.h
+++ b/api/rtp_receiver_interface.h
@@ -26,7 +26,6 @@
 #include "api/rtp_parameters.h"
 #include "api/scoped_refptr.h"
 #include "api/transport/rtp/rtp_source.h"
-#include "rtc_base/deprecation.h"
 #include "rtc_base/ref_count.h"
 #include "rtc_base/system/rtc_export.h"
 
diff --git a/api/rtp_transceiver_interface.h b/api/rtp_transceiver_interface.h
index 9b46846597..4799c4b153 100644
--- a/api/rtp_transceiver_interface.h
+++ b/api/rtp_transceiver_interface.h
@@ -14,6 +14,7 @@
 #include <string>
 #include <vector>
 
+#include "absl/base/attributes.h"
 #include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "api/media_types.h"
@@ -111,8 +112,8 @@ class RTC_EXPORT RtpTransceiverInterface : public rtc::RefCountInterface {
   // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
   // TODO(hta): Deprecate SetDirection without error and rename
   // SetDirectionWithError to SetDirection, remove default implementations.
-  RTC_DEPRECATED virtual void SetDirection(
-      RtpTransceiverDirection new_direction);
+  ABSL_DEPRECATED("Use SetDirectionWithError instead")
+  virtual void SetDirection(RtpTransceiverDirection new_direction);
   virtual RTCError SetDirectionWithError(RtpTransceiverDirection new_direction);
 
   // The current_direction attribute indicates the current direction negotiated
@@ -140,7 +141,7 @@ class RTC_EXPORT RtpTransceiverInterface : public rtc::RefCountInterface {
   // This is an internal function, and is exposed for historical reasons.
   // https://w3c.github.io/webrtc-pc/#dfn-stop-the-rtcrtptransceiver
   virtual void StopInternal();
-  RTC_DEPRECATED virtual void Stop();
+  ABSL_DEPRECATED("Use StopStandard instead") virtual void Stop();
 
   // The SetCodecPreferences method overrides the default codec preferences used
   // by WebRTC for this transceiver.
diff --git a/api/transport/BUILD.gn b/api/transport/BUILD.gn
index 7d23adc37a..30955273b0 100644
--- a/api/transport/BUILD.gn
+++ b/api/transport/BUILD.gn
@@ -33,7 +33,6 @@ rtc_library("network_control") {
 
   deps = [
     ":webrtc_key_value_config",
-    "../../rtc_base:deprecation",
     "../rtc_event_log",
     "../units:data_rate",
     "../units:data_size",
@@ -89,8 +88,8 @@ rtc_library("goog_cc") {
     ":webrtc_key_value_config",
     "..:network_state_predictor_api",
     "../../modules/congestion_controller/goog_cc",
-    "../../rtc_base:deprecation",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/base:core_headers" ]
 }
 
 rtc_source_set("sctp_transport_factory_interface") {
diff --git a/api/transport/goog_cc_factory.h b/api/transport/goog_cc_factory.h
index b14d6dcd78..e12755d745 100644
--- a/api/transport/goog_cc_factory.h
+++ b/api/transport/goog_cc_factory.h
@@ -12,9 +12,9 @@
 #define API_TRANSPORT_GOOG_CC_FACTORY_H_
 #include <memory>
 
+#include "absl/base/attributes.h"
 #include "api/network_state_predictor.h"
 #include "api/transport/network_control.h"
-#include "rtc_base/deprecation.h"
 
 namespace webrtc {
 class RtcEventLog;
@@ -31,8 +31,8 @@ class GoogCcNetworkControllerFactory
     : public NetworkControllerFactoryInterface {
  public:
   GoogCcNetworkControllerFactory() = default;
-  explicit RTC_DEPRECATED GoogCcNetworkControllerFactory(
-      RtcEventLog* event_log);
+  ABSL_DEPRECATED("")
+  explicit GoogCcNetworkControllerFactory(RtcEventLog* event_log);
   explicit GoogCcNetworkControllerFactory(
       NetworkStatePredictorFactoryInterface* network_state_predictor_factory);
 
@@ -49,7 +49,8 @@ class GoogCcNetworkControllerFactory
 // Deprecated, use GoogCcFactoryConfig to enable feedback only mode instead.
 // Factory to create packet feedback only GoogCC, this can be used for
 // connections providing packet receive time feedback but no other reports.
-class RTC_DEPRECATED GoogCcFeedbackNetworkControllerFactory
+class ABSL_DEPRECATED("use GoogCcFactoryConfig instead")
+    GoogCcFeedbackNetworkControllerFactory
     : public GoogCcNetworkControllerFactory {
  public:
   explicit GoogCcFeedbackNetworkControllerFactory(RtcEventLog* event_log);
diff --git a/api/transport/network_types.h b/api/transport/network_types.h
index 10fc0beedf..38a8917f1e 100644
--- a/api/transport/network_types.h
+++ b/api/transport/network_types.h
@@ -19,7 +19,6 @@
 #include "api/units/data_size.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
-#include "rtc_base/deprecation.h"
 
 namespace webrtc {
 
diff --git a/api/video/BUILD.gn b/api/video/BUILD.gn
index d50a334635..b697b86283 100644
--- a/api/video/BUILD.gn
+++ b/api/video/BUILD.gn
@@ -142,7 +142,6 @@ rtc_library("encoded_image") {
     "..:rtp_packet_info",
     "..:scoped_refptr",
     "../../rtc_base:checks",
-    "../../rtc_base:deprecation",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base/system:rtc_export",
   ]
diff --git a/api/video/encoded_image.h b/api/video/encoded_image.h
index 650766ab64..60e6625991 100644
--- a/api/video/encoded_image.h
+++ b/api/video/encoded_image.h
@@ -16,6 +16,7 @@
 #include <map>
 #include <utility>
 
+#include "absl/base/attributes.h"
 #include "absl/types/optional.h"
 #include "api/rtp_packet_infos.h"
 #include "api/scoped_refptr.h"
@@ -26,7 +27,6 @@
 #include "api/video/video_rotation.h"
 #include "api/video/video_timing.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/deprecation.h"
 #include "rtc_base/ref_count.h"
 #include "rtc_base/system/rtc_export.h"
 
@@ -73,7 +73,8 @@ class RTC_EXPORT EncodedImage {
   EncodedImage();
   EncodedImage(EncodedImage&&);
   EncodedImage(const EncodedImage&);
-  RTC_DEPRECATED EncodedImage(uint8_t* buffer, size_t length, size_t capacity);
+  ABSL_DEPRECATED("")
+  EncodedImage(uint8_t* buffer, size_t length, size_t capacity);
 
   ~EncodedImage();
 
diff --git a/api/video_codecs/video_encoder.h b/api/video_codecs/video_encoder.h
index a030362ab7..caf069718b 100644
--- a/api/video_codecs/video_encoder.h
+++ b/api/video_codecs/video_encoder.h
@@ -364,7 +364,7 @@ class RTC_EXPORT VideoEncoder {
   // TODO(bugs.webrtc.org/10720): After updating downstream projects and posting
   // an announcement to discuss-webrtc, remove the three-parameters variant
   // and make the two-parameters variant pure-virtual.
-  /* RTC_DEPRECATED */ virtual int32_t InitEncode(
+  /* ABSL_DEPRECATED("bugs.webrtc.org/10720") */ virtual int32_t InitEncode(
       const VideoCodec* codec_settings,
       int32_t number_of_cores,
       size_t max_payload_size);
diff --git a/audio/utility/BUILD.gn b/audio/utility/BUILD.gn
index 54ca04698d..933553d81b 100644
--- a/audio/utility/BUILD.gn
+++ b/audio/utility/BUILD.gn
@@ -26,10 +26,10 @@ rtc_library("audio_frame_operations") {
     "../../api/audio:audio_frame_api",
     "../../common_audio",
     "../../rtc_base:checks",
-    "../../rtc_base:deprecation",
     "../../rtc_base:rtc_base_approved",
     "../../system_wrappers:field_trial",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/base:core_headers" ]
 }
 
 if (rtc_include_tests) {
diff --git a/audio/utility/audio_frame_operations.h b/audio/utility/audio_frame_operations.h
index 65c310c489..2f1540bcf5 100644
--- a/audio/utility/audio_frame_operations.h
+++ b/audio/utility/audio_frame_operations.h
@@ -14,8 +14,8 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include "absl/base/attributes.h"
 #include "api/audio/audio_frame.h"
-#include "rtc_base/deprecation.h"
 
 namespace webrtc {
 
@@ -36,12 +36,14 @@ class AudioFrameOperations {
   // |frame.num_channels_| will be updated. This version checks for sufficient
   // buffer size and that |num_channels_| is mono. Use UpmixChannels
   // instead. TODO(bugs.webrtc.org/8649): remove.
-  RTC_DEPRECATED static int MonoToStereo(AudioFrame* frame);
+  ABSL_DEPRECATED("bugs.webrtc.org/8649")
+  static int MonoToStereo(AudioFrame* frame);
 
   // |frame.num_channels_| will be updated. This version checks that
   // |num_channels_| is stereo. Use DownmixChannels
   // instead. TODO(bugs.webrtc.org/8649): remove.
-  RTC_DEPRECATED static int StereoToMono(AudioFrame* frame);
+  ABSL_DEPRECATED("bugs.webrtc.org/8649")
+  static int StereoToMono(AudioFrame* frame);
 
   // Downmixes 4 channels |src_audio| to stereo |dst_audio|. This is an in-place
   // operation, meaning |src_audio| and |dst_audio| may point to the same
diff --git a/logging/BUILD.gn b/logging/BUILD.gn
index 25f275ae0e..ff98a2e7a0 100644
--- a/logging/BUILD.gn
+++ b/logging/BUILD.gn
@@ -342,7 +342,6 @@ if (rtc_enable_protobuf) {
       "../modules/rtp_rtcp",
       "../modules/rtp_rtcp:rtp_rtcp_format",
       "../rtc_base:checks",
-      "../rtc_base:deprecation",
       "../rtc_base:ignore_wundef",
       "../rtc_base:protobuf_utils",
       "../rtc_base:rtc_base_approved",
diff --git a/logging/rtc_event_log/rtc_event_log_parser.h b/logging/rtc_event_log/rtc_event_log_parser.h
index 841bbf7d05..05ff94a9a8 100644
--- a/logging/rtc_event_log/rtc_event_log_parser.h
+++ b/logging/rtc_event_log/rtc_event_log_parser.h
@@ -19,6 +19,7 @@
 #include <utility>  // pair
 #include <vector>
 
+#include "absl/base/attributes.h"
 #include "api/rtc_event_log/rtc_event_log.h"
 #include "call/video_receive_stream.h"
 #include "call/video_send_stream.h"
@@ -296,7 +297,7 @@ class ParsedRtcEventLog {
       return error_ + " failed at " + file_ + " line " + std::to_string(line_);
     }
 
-    RTC_DEPRECATED operator bool() const { return ok(); }
+    ABSL_DEPRECATED("Use ok() instead") operator bool() const { return ok(); }
 
    private:
     ParseStatus() : error_(), file_(), line_(0) {}
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 0ae8a6125a..76f06e0156 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -242,7 +242,6 @@ rtc_library("rtc_internal_video_codecs") {
     "../modules/video_coding/codecs/av1:libaom_av1_decoder",
     "../modules/video_coding/codecs/av1:libaom_av1_encoder",
     "../rtc_base:checks",
-    "../rtc_base:deprecation",
     "../rtc_base:rtc_base_approved",
     "../rtc_base/system:rtc_export",
     "../test:fake_video_codecs",
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index c98db4ea1a..c817905de5 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -17,7 +17,6 @@ visibility = [ ":*" ]
 rtc_source_set("audio_coding_module_typedefs") {
   visibility += [ "*" ]
   sources = [ "include/audio_coding_module_typedefs.h" ]
-  deps = [ "../../rtc_base:deprecation" ]
 }
 
 rtc_library("audio_coding") {
@@ -52,7 +51,6 @@ rtc_library("audio_coding") {
     "../../common_audio:common_audio_c",
     "../../rtc_base:audio_format_to_string",
     "../../rtc_base:checks",
-    "../../rtc_base:deprecation",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base/synchronization:mutex",
     "../../system_wrappers",
diff --git a/modules/audio_coding/include/audio_coding_module_typedefs.h b/modules/audio_coding/include/audio_coding_module_typedefs.h
index 07aa8c956f..a7210dadcb 100644
--- a/modules/audio_coding/include/audio_coding_module_typedefs.h
+++ b/modules/audio_coding/include/audio_coding_module_typedefs.h
@@ -13,8 +13,6 @@
 
 #include <map>
 
-#include "rtc_base/deprecation.h"
-
 namespace webrtc {
 
 ///////////////////////////////////////////////////////////////////////////
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
index 05f883948d..5d6a1d82fc 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -51,7 +51,6 @@ rtc_source_set("audio_device_api") {
     "../../api:scoped_refptr",
     "../../api/task_queue",
     "../../rtc_base:checks",
-    "../../rtc_base:deprecation",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base:stringutils",
   ]
@@ -171,7 +170,6 @@ rtc_library("audio_device_impl") {
     "../../common_audio:common_audio_c",
     "../../rtc_base",
     "../../rtc_base:checks",
-    "../../rtc_base:deprecation",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base:rtc_task_queue",
     "../../rtc_base/synchronization:mutex",
@@ -183,6 +181,7 @@ rtc_library("audio_device_impl") {
     "../../system_wrappers:metrics",
     "../utility",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/base:core_headers" ]
   if (rtc_include_internal_audio_device && is_ios) {
     deps += [ "../../sdk:audio_device" ]
   }
diff --git a/modules/audio_device/include/audio_device_data_observer.h b/modules/audio_device/include/audio_device_data_observer.h
index e1c2035d67..b59cafcb5d 100644
--- a/modules/audio_device/include/audio_device_data_observer.h
+++ b/modules/audio_device/include/audio_device_data_observer.h
@@ -14,6 +14,7 @@
 #include <stddef.h>
 #include <stdint.h>
 
+#include "absl/base/attributes.h"
 #include "api/scoped_refptr.h"
 #include "api/task_queue/task_queue_factory.h"
 #include "modules/audio_device/include/audio_device.h"
@@ -48,7 +49,7 @@ rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
 
 // Creates an ADMWrapper around an ADM instance that registers
 // the provided AudioDeviceDataObserver.
-RTC_DEPRECATED
+ABSL_DEPRECATED("")
 rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
     rtc::scoped_refptr<AudioDeviceModule> impl,
     AudioDeviceDataObserver* observer);
@@ -60,7 +61,7 @@ rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
     std::unique_ptr<AudioDeviceDataObserver> observer);
 
 // Creates an ADM instance with AudioDeviceDataObserver registered.
-RTC_DEPRECATED
+ABSL_DEPRECATED("")
 rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceWithDataObserver(
     const AudioDeviceModule::AudioLayer audio_layer,
     TaskQueueFactory* task_queue_factory,
diff --git a/modules/audio_device/include/audio_device_defines.h b/modules/audio_device/include/audio_device_defines.h
index d5d4d7372e..01129a47a9 100644
--- a/modules/audio_device/include/audio_device_defines.h
+++ b/modules/audio_device/include/audio_device_defines.h
@@ -16,7 +16,6 @@
 #include <string>
 
 #include "rtc_base/checks.h"
-#include "rtc_base/deprecation.h"
 #include "rtc_base/strings/string_builder.h"
 
 namespace webrtc {
diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index 6dc1add778..147748fc4e 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -43,7 +43,6 @@ rtc_library("api") {
     "../../api/audio:aec3_config",
     "../../api/audio:audio_frame_api",
     "../../api/audio:echo_control",
-    "../../rtc_base:deprecation",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base/system:arch",
     "../../rtc_base/system:file_wrapper",
@@ -119,8 +118,8 @@ rtc_source_set("aec_dump_interface") {
   deps = [
     ":api",
     ":audio_frame_view",
-    "../../rtc_base:deprecation",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/base:core_headers" ]
 }
 
 rtc_library("audio_processing") {
@@ -177,7 +176,6 @@ rtc_library("audio_processing") {
     "../../common_audio:common_audio_c",
     "../../common_audio/third_party/ooura:fft_size_256",
     "../../rtc_base:checks",
-    "../../rtc_base:deprecation",
     "../../rtc_base:gtest_prod",
     "../../rtc_base:ignore_wundef",
     "../../rtc_base:refcount",
diff --git a/modules/audio_processing/include/aec_dump.h b/modules/audio_processing/include/aec_dump.h
index ed5acb0943..a7769d9973 100644
--- a/modules/audio_processing/include/aec_dump.h
+++ b/modules/audio_processing/include/aec_dump.h
@@ -15,9 +15,9 @@
 
 #include <string>
 
+#include "absl/base/attributes.h"
 #include "modules/audio_processing/include/audio_frame_view.h"
 #include "modules/audio_processing/include/audio_processing.h"
-#include "rtc_base/deprecation.h"
 
 namespace webrtc {
 
@@ -76,7 +76,8 @@ class AecDump {
   // Logs Event::Type INIT message.
   virtual void WriteInitMessage(const ProcessingConfig& api_format,
                                 int64_t time_now_ms) = 0;
-  RTC_DEPRECATED void WriteInitMessage(const ProcessingConfig& api_format) {
+  ABSL_DEPRECATED("")
+  void WriteInitMessage(const ProcessingConfig& api_format) {
     WriteInitMessage(api_format, 0);
   }
 
diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h
index 67b9fc1a41..2312ffd536 100644
--- a/modules/audio_processing/include/audio_processing.h
+++ b/modules/audio_processing/include/audio_processing.h
@@ -32,7 +32,6 @@
 #include "modules/audio_processing/include/config.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/constructor_magic.h"
-#include "rtc_base/deprecation.h"
 #include "rtc_base/ref_count.h"
 #include "rtc_base/system/file_wrapper.h"
 #include "rtc_base/system/rtc_export.h"
diff --git a/modules/rtp_rtcp/BUILD.gn b/modules/rtp_rtcp/BUILD.gn
index 4233ef4081..9e5ec4a72d 100644
--- a/modules/rtp_rtcp/BUILD.gn
+++ b/modules/rtp_rtcp/BUILD.gn
@@ -116,7 +116,6 @@ rtc_library("rtp_rtcp_format") {
     "../../api/video:video_rtp_headers",
     "../../common_video",
     "../../rtc_base:checks",
-    "../../rtc_base:deprecation",
     "../../rtc_base:divide_round",
     "../../rtc_base:rtc_base_approved",
     "../../system_wrappers",
@@ -284,7 +283,6 @@ rtc_library("rtp_rtcp") {
     "../../logging:rtc_event_rtp_rtcp",
     "../../modules/audio_coding:audio_coding_module_typedefs",
     "../../rtc_base:checks",
-    "../../rtc_base:deprecation",
     "../../rtc_base:divide_round",
     "../../rtc_base:gtest_prod",
     "../../rtc_base:rate_limiter",
@@ -335,7 +333,6 @@ rtc_source_set("rtp_rtcp_legacy") {
     "../../api/video:video_bitrate_allocation",
     "../../logging:rtc_event_rtp_rtcp",
     "../../rtc_base:checks",
-    "../../rtc_base:deprecation",
     "../../rtc_base:gtest_prod",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base/synchronization:mutex",
diff --git a/modules/rtp_rtcp/include/receive_statistics.h b/modules/rtp_rtcp/include/receive_statistics.h
index 062c90bdbe..2cc3feab3a 100644
--- a/modules/rtp_rtcp/include/receive_statistics.h
+++ b/modules/rtp_rtcp/include/receive_statistics.h
@@ -20,7 +20,6 @@
 #include "modules/rtp_rtcp/include/rtcp_statistics.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
-#include "rtc_base/deprecation.h"
 
 namespace webrtc {
 
diff --git a/modules/rtp_rtcp/include/rtp_header_extension_map.h b/modules/rtp_rtcp/include/rtp_header_extension_map.h
index ff2d34d60d..72e5541d37 100644
--- a/modules/rtp_rtcp/include/rtp_header_extension_map.h
+++ b/modules/rtp_rtcp/include/rtp_header_extension_map.h
@@ -19,7 +19,6 @@
 #include "api/rtp_parameters.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/deprecation.h"
 
 namespace webrtc {
 
diff --git a/modules/rtp_rtcp/include/rtp_rtcp.h b/modules/rtp_rtcp/include/rtp_rtcp.h
index df37bddb7d..727fc6e649 100644
--- a/modules/rtp_rtcp/include/rtp_rtcp.h
+++ b/modules/rtp_rtcp/include/rtp_rtcp.h
@@ -13,9 +13,9 @@
 
 #include <memory>
 
+#include "absl/base/attributes.h"
 #include "modules/include/module.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
-#include "rtc_base/deprecation.h"
 
 namespace webrtc {
 
@@ -23,8 +23,8 @@ namespace webrtc {
 class RtpRtcp : public Module, public RtpRtcpInterface {
  public:
   // Instantiates a deprecated version of the RtpRtcp module.
-  static std::unique_ptr<RtpRtcp> RTC_DEPRECATED
-  Create(const Configuration& configuration) {
+  static std::unique_ptr<RtpRtcp> ABSL_DEPRECATED("")
+      Create(const Configuration& configuration) {
     return DEPRECATED_Create(configuration);
   }
 
diff --git a/modules/rtp_rtcp/source/rtp_packet_received.h b/modules/rtp_rtcp/source/rtp_packet_received.h
index c895313755..e1e1309594 100644
--- a/modules/rtp_rtcp/source/rtp_packet_received.h
+++ b/modules/rtp_rtcp/source/rtp_packet_received.h
@@ -19,7 +19,6 @@
 #include "api/rtp_headers.h"
 #include "api/scoped_refptr.h"
 #include "modules/rtp_rtcp/source/rtp_packet.h"
-#include "rtc_base/deprecation.h"
 
 namespace webrtc {
 // Class to hold rtp packet with metadata for receiver side.
diff --git a/modules/rtp_rtcp/source/rtp_sender.h b/modules/rtp_rtcp/source/rtp_sender.h
index 642c647955..8c579429cc 100644
--- a/modules/rtp_rtcp/source/rtp_sender.h
+++ b/modules/rtp_rtcp/source/rtp_sender.h
@@ -29,7 +29,6 @@
 #include "modules/rtp_rtcp/source/rtp_packet_history.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
-#include "rtc_base/deprecation.h"
 #include "rtc_base/random.h"
 #include "rtc_base/rate_statistics.h"
 #include "rtc_base/synchronization/mutex.h"
diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
index a70adad8b8..ebf281faeb 100644
--- a/modules/video_coding/BUILD.gn
+++ b/modules/video_coding/BUILD.gn
@@ -186,7 +186,6 @@ rtc_library("video_coding") {
     "../../common_video",
     "../../rtc_base",
     "../../rtc_base:checks",
-    "../../rtc_base:deprecation",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base:rtc_numerics",
     "../../rtc_base:rtc_task_queue",
@@ -236,7 +235,6 @@ rtc_library("video_codec_interface") {
     "../../api/video_codecs:video_codecs_api",
     "../../common_video",
     "../../common_video/generic_frame_descriptor",
-    "../../rtc_base:deprecation",
     "../../rtc_base/system:rtc_export",
   ]
   absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
@@ -487,7 +485,6 @@ rtc_library("webrtc_vp8") {
     "../../api/video_codecs:vp8_temporal_layers_factory",
     "../../common_video",
     "../../rtc_base:checks",
-    "../../rtc_base:deprecation",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base:rtc_numerics",
     "../../rtc_base/experiments:cpu_speed_experiment",
diff --git a/modules/video_coding/codecs/vp8/include/vp8.h b/modules/video_coding/codecs/vp8/include/vp8.h
index 44efbeeb3b..d05c3a68d1 100644
--- a/modules/video_coding/codecs/vp8/include/vp8.h
+++ b/modules/video_coding/codecs/vp8/include/vp8.h
@@ -14,10 +14,10 @@
 #include <memory>
 #include <vector>
 
+#include "absl/base/attributes.h"
 #include "api/video_codecs/video_encoder.h"
 #include "api/video_codecs/vp8_frame_buffer_controller.h"
 #include "modules/video_coding/include/video_codec_interface.h"
-#include "rtc_base/deprecation.h"
 
 namespace webrtc {
 
@@ -40,7 +40,8 @@ class VP8Encoder {
   static std::unique_ptr<VideoEncoder> Create();
   static std::unique_ptr<VideoEncoder> Create(Settings settings);
 
-  RTC_DEPRECATED static std::unique_ptr<VideoEncoder> Create(
+  ABSL_DEPRECATED("")
+  static std::unique_ptr<VideoEncoder> Create(
       std::unique_ptr<Vp8FrameBufferControllerFactory>
           frame_buffer_controller_factory);
 };
diff --git a/modules/video_coding/deprecated/BUILD.gn b/modules/video_coding/deprecated/BUILD.gn
index fd3a5fa5fc..487c0267d5 100644
--- a/modules/video_coding/deprecated/BUILD.gn
+++ b/modules/video_coding/deprecated/BUILD.gn
@@ -21,7 +21,6 @@ rtc_library("nack_module") {
     "../../../api/units:timestamp",
     "../../../rtc_base:checks",
     "../../../rtc_base:criticalsection",
-    "../../../rtc_base:deprecation",
     "../../../rtc_base:logging",
     "../../../rtc_base:macromagic",
     "../../../rtc_base:rtc_numerics",
@@ -31,4 +30,5 @@ rtc_library("nack_module") {
     "../../../system_wrappers:field_trial",
     "../../utility",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/base:core_headers" ]
 }
diff --git a/modules/video_coding/deprecated/nack_module.h b/modules/video_coding/deprecated/nack_module.h
index f9580ae80c..0ad009b597 100644
--- a/modules/video_coding/deprecated/nack_module.h
+++ b/modules/video_coding/deprecated/nack_module.h
@@ -17,11 +17,11 @@
 #include <set>
 #include <vector>
 
+#include "absl/base/attributes.h"
 #include "api/units/time_delta.h"
 #include "modules/include/module.h"
 #include "modules/include/module_common_types.h"
 #include "modules/video_coding/histogram.h"
-#include "rtc_base/deprecation.h"
 #include "rtc_base/numerics/sequence_number_util.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/thread_annotations.h"
@@ -125,7 +125,7 @@ class DEPRECATED_NackModule : public Module {
   const absl::optional<BackoffSettings> backoff_settings_;
 };
 
-using NackModule = RTC_DEPRECATED DEPRECATED_NackModule;
+using NackModule = ABSL_DEPRECATED("") DEPRECATED_NackModule;
 
 }  // namespace webrtc
 
diff --git a/modules/video_coding/include/video_codec_interface.h b/modules/video_coding/include/video_codec_interface.h
index b786be1693..4737dde90f 100644
--- a/modules/video_coding/include/video_codec_interface.h
+++ b/modules/video_coding/include/video_codec_interface.h
@@ -13,6 +13,7 @@
 
 #include <vector>
 
+#include "absl/base/attributes.h"
 #include "absl/types/optional.h"
 #include "api/video/video_frame.h"
 #include "api/video_codecs/video_decoder.h"
@@ -21,7 +22,6 @@
 #include "modules/video_coding/codecs/h264/include/h264_globals.h"
 #include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
 #include "modules/video_coding/include/video_error_codes.h"
-#include "rtc_base/deprecation.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace webrtc {
@@ -79,7 +79,7 @@ struct CodecSpecificInfoVP9 {
   uint8_t num_ref_pics;
   uint8_t p_diff[kMaxVp9RefPics];
 
-  RTC_DEPRECATED bool end_of_picture;
+  ABSL_DEPRECATED("") bool end_of_picture;
 };
 static_assert(std::is_pod<CodecSpecificInfoVP9>::value, "");
 
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 8a6fa3c816..2899dafb05 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -117,7 +117,6 @@ rtc_library("rtc_pc_base") {
     "../rtc_base",
     "../rtc_base:callback_list",
     "../rtc_base:checks",
-    "../rtc_base:deprecation",
     "../rtc_base:rtc_task_queue",
     "../rtc_base:socket",
     "../rtc_base:socket_address",
@@ -289,7 +288,6 @@ rtc_library("peerconnection") {
     "../rtc_base",
     "../rtc_base:callback_list",
     "../rtc_base:checks",
-    "../rtc_base:deprecation",
     "../rtc_base:ip_address",
     "../rtc_base:network_constants",
     "../rtc_base:rtc_base_approved",
diff --git a/pc/peer_connection.h b/pc/peer_connection.h
index fc0832468a..d81f3c9918 100644
--- a/pc/peer_connection.h
+++ b/pc/peer_connection.h
@@ -88,7 +88,6 @@
 #include "pc/usage_pattern.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/copy_on_write_buffer.h"
-#include "rtc_base/deprecation.h"
 #include "rtc_base/network/sent_packet.h"
 #include "rtc_base/rtc_certificate.h"
 #include "rtc_base/ssl_certificate.h"
diff --git a/pc/session_description.h b/pc/session_description.h
index 07270ee7e6..dacf1a2174 100644
--- a/pc/session_description.h
+++ b/pc/session_description.h
@@ -37,7 +37,6 @@
 #include "pc/media_protocol_names.h"
 #include "pc/simulcast_description.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/deprecation.h"
 #include "rtc_base/socket_address.h"
 #include "rtc_base/system/rtc_export.h"
 
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 49c73a1b47..28ff7fd53c 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -307,10 +307,6 @@ rtc_library("logging") {
       frameworks = [ "Foundation.framework" ]
     }
 
-    # logging.h needs the deprecation header while downstream projects are
-    # removing code that depends on logging implementation details.
-    deps += [ ":deprecation" ]
-
     if (is_android) {
       libs += [ "log" ]
     }
@@ -457,10 +453,6 @@ rtc_source_set("type_traits") {
   sources = [ "type_traits.h" ]
 }
 
-rtc_source_set("deprecation") {
-  sources = [ "deprecation.h" ]
-}
-
 rtc_library("rtc_task_queue") {
   visibility = [ "*" ]
   sources = [
@@ -909,7 +901,6 @@ rtc_library("rtc_base") {
     ":async_resolver_interface",
     ":async_socket",
     ":checks",
-    ":deprecation",
     ":ip_address",
     ":network_constants",
     ":null_socket_server",
diff --git a/rtc_base/deprecation.h b/rtc_base/deprecation.h
deleted file mode 100644
index f285ab04bb..0000000000
--- a/rtc_base/deprecation.h
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- *  Copyright (c) 2015 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 RTC_BASE_DEPRECATION_H_
-#define RTC_BASE_DEPRECATION_H_
-
-// Annotate the declarations of deprecated functions with this to cause a
-// compiler warning when they're used. Like so:
-//
-//   RTC_DEPRECATED std::pony PonyPlz(const std::pony_spec& ps);
-//
-// NOTE 1: The annotation goes on the declaration in the .h file, not the
-// definition in the .cc file!
-//
-// NOTE 2: In order to keep unit testing the deprecated function without
-// getting warnings, do something like this:
-//
-//   std::pony DEPRECATED_PonyPlz(const std::pony_spec& ps);
-//   RTC_DEPRECATED inline std::pony PonyPlz(const std::pony_spec& ps) {
-//     return DEPRECATED_PonyPlz(ps);
-//   }
-//
-// In other words, rename the existing function, and provide an inline wrapper
-// using the original name that calls it. That way, callers who are willing to
-// call it using the DEPRECATED_-prefixed name don't get the warning.
-//
-// TODO(kwiberg): Remove this when we can use [[deprecated]] from C++14.
-#if defined(_MSC_VER)
-// Note: Deprecation warnings seem to fail to trigger on Windows
-// (https://bugs.chromium.org/p/webrtc/issues/detail?id=5368).
-#define RTC_DEPRECATED __declspec(deprecated)
-#elif defined(__GNUC__)
-#define RTC_DEPRECATED __attribute__((__deprecated__))
-#else
-#define RTC_DEPRECATED
-#endif
-
-#endif  // RTC_BASE_DEPRECATION_H_
diff --git a/rtc_base/logging.h b/rtc_base/logging.h
index d2607c28b7..e21c30e21a 100644
--- a/rtc_base/logging.h
+++ b/rtc_base/logging.h
@@ -51,10 +51,10 @@
 #include <string>
 #include <utility>
 
+#include "absl/base/attributes.h"
 #include "absl/meta/type_traits.h"
 #include "absl/strings/string_view.h"
 #include "rtc_base/constructor_magic.h"
-#include "rtc_base/deprecation.h"
 #include "rtc_base/strings/string_builder.h"
 #include "rtc_base/system/inline.h"
 
@@ -434,7 +434,7 @@ class LogMessage {
   // DEPRECATED - DO NOT USE - PLEASE USE THE MACROS INSTEAD OF THE CLASS.
   // Android code should use the 'const char*' version since tags are static
   // and we want to avoid allocating a std::string copy per log line.
-  RTC_DEPRECATED
+  ABSL_DEPRECATED("Use RTC_LOG macros instead of accessing this class directly")
   LogMessage(const char* file,
              int line,
              LoggingSeverity sev,
@@ -508,7 +508,7 @@ class LogMessage {
   // DEPRECATED - DO NOT USE - PLEASE USE THE MACROS INSTEAD OF THE CLASS.
   // Android code should use the 'const char*' version since tags are static
   // and we want to avoid allocating a std::string copy per log line.
-  RTC_DEPRECATED
+  ABSL_DEPRECATED("Use RTC_LOG macros instead of accessing this class directly")
   LogMessage(const char* file,
              int line,
              LoggingSeverity sev,
diff --git a/rtc_base/ssl_identity.h b/rtc_base/ssl_identity.h
index d078b045a7..a9167ef5eb 100644
--- a/rtc_base/ssl_identity.h
+++ b/rtc_base/ssl_identity.h
@@ -18,7 +18,6 @@
 #include <memory>
 #include <string>
 
-#include "rtc_base/deprecation.h"
 #include "rtc_base/system/rtc_export.h"
 
 namespace rtc {
diff --git a/rtc_base/ssl_stream_adapter.h b/rtc_base/ssl_stream_adapter.h
index 7bff726510..977768a657 100644
--- a/rtc_base/ssl_stream_adapter.h
+++ b/rtc_base/ssl_stream_adapter.h
@@ -18,7 +18,6 @@
 #include <vector>
 
 #include "absl/memory/memory.h"
-#include "rtc_base/deprecation.h"
 #include "rtc_base/ssl_certificate.h"
 #include "rtc_base/ssl_identity.h"
 #include "rtc_base/stream.h"
diff --git a/rtc_tools/BUILD.gn b/rtc_tools/BUILD.gn
index 41902bb204..f6352d8742 100644
--- a/rtc_tools/BUILD.gn
+++ b/rtc_tools/BUILD.gn
@@ -365,7 +365,6 @@ if (!build_with_chromium) {
         ":chart_proto",
         "../api:function_view",
         "../api:network_state_predictor_api",
-        "../rtc_base:deprecation",
         "../rtc_base:ignore_wundef",
 
         # TODO(kwiberg): Remove this dependency.
@@ -396,6 +395,7 @@ if (!build_with_chromium) {
       ]
       absl_deps = [
         "//third_party/abseil-cpp/absl/algorithm:container",
+        "//third_party/abseil-cpp/absl/base:core_headers",
         "//third_party/abseil-cpp/absl/strings",
         "//third_party/abseil-cpp/absl/types:optional",
       ]
diff --git a/rtc_tools/rtc_event_log_visualizer/plot_base.h b/rtc_tools/rtc_event_log_visualizer/plot_base.h
index 06a206f031..a26146b5e5 100644
--- a/rtc_tools/rtc_event_log_visualizer/plot_base.h
+++ b/rtc_tools/rtc_event_log_visualizer/plot_base.h
@@ -15,7 +15,7 @@
 #include <utility>
 #include <vector>
 
-#include "rtc_base/deprecation.h"
+#include "absl/base/attributes.h"
 #include "rtc_base/ignore_wundef.h"
 
 RTC_PUSH_IGNORING_WUNDEF()
@@ -101,8 +101,8 @@ class Plot {
  public:
   virtual ~Plot() {}
 
-  // Deprecated. Use PrintPythonCode() or ExportProtobuf() instead.
-  RTC_DEPRECATED virtual void Draw() {}
+  ABSL_DEPRECATED("Use PrintPythonCode() or ExportProtobuf() instead.")
+  virtual void Draw() {}
 
   // Sets the lower x-axis limit to min_value (if left_margin == 0).
   // Sets the upper x-axis limit to max_value (if right_margin == 0).
@@ -189,8 +189,8 @@ class PlotCollection {
  public:
   virtual ~PlotCollection() {}
 
-  // Deprecated. Use PrintPythonCode() or ExportProtobuf() instead.
-  RTC_DEPRECATED virtual void Draw() {}
+  ABSL_DEPRECATED("Use PrintPythonCode() or ExportProtobuf() instead.")
+  virtual void Draw() {}
 
   virtual Plot* AppendNewPlot();
 
diff --git a/rtc_tools/rtc_event_log_visualizer/plot_protobuf.h b/rtc_tools/rtc_event_log_visualizer/plot_protobuf.h
index 0773b58d20..fbe68853a3 100644
--- a/rtc_tools/rtc_event_log_visualizer/plot_protobuf.h
+++ b/rtc_tools/rtc_event_log_visualizer/plot_protobuf.h
@@ -10,6 +10,7 @@
 #ifndef RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_PLOT_PROTOBUF_H_
 #define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_PLOT_PROTOBUF_H_
 
+#include "absl/base/attributes.h"
 #include "rtc_base/ignore_wundef.h"
 RTC_PUSH_IGNORING_WUNDEF()
 #include "rtc_tools/rtc_event_log_visualizer/proto/chart.pb.h"
@@ -25,10 +26,10 @@ class ProtobufPlot final : public Plot {
   void Draw() override;
 };
 
-class ProtobufPlotCollection final : public PlotCollection {
+class ABSL_DEPRECATED("Use PlotCollection and ExportProtobuf() instead")
+    ProtobufPlotCollection final : public PlotCollection {
  public:
-  // This class is deprecated. Use PlotCollection and ExportProtobuf() instead.
-  RTC_DEPRECATED ProtobufPlotCollection();
+  ProtobufPlotCollection();
   ~ProtobufPlotCollection() override;
   void Draw() override;
   Plot* AppendNewPlot() override;
diff --git a/rtc_tools/rtc_event_log_visualizer/plot_python.h b/rtc_tools/rtc_event_log_visualizer/plot_python.h
index 998ed7b221..6acc436d71 100644
--- a/rtc_tools/rtc_event_log_visualizer/plot_python.h
+++ b/rtc_tools/rtc_event_log_visualizer/plot_python.h
@@ -10,6 +10,7 @@
 #ifndef RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_PLOT_PYTHON_H_
 #define RTC_TOOLS_RTC_EVENT_LOG_VISUALIZER_PLOT_PYTHON_H_
 
+#include "absl/base/attributes.h"
 #include "rtc_tools/rtc_event_log_visualizer/plot_base.h"
 
 namespace webrtc {
@@ -21,10 +22,10 @@ class PythonPlot final : public Plot {
   void Draw() override;
 };
 
-class PythonPlotCollection final : public PlotCollection {
+class ABSL_DEPRECATED("Use PlotCollection and PrintPythonCode() instead.")
+    PythonPlotCollection final : public PlotCollection {
  public:
-  // This class is deprecated. Use PlotCollection and PrintPythonCode() instead.
-  RTC_DEPRECATED explicit PythonPlotCollection(bool shared_xaxis = false);
+  explicit PythonPlotCollection(bool shared_xaxis = false);
   ~PythonPlotCollection() override;
   void Draw() override;
   Plot* AppendNewPlot() override;