diff --git a/api/audio/audio_processing.h b/api/audio/audio_processing.h
index fefef30a42..293c01f442 100644
--- a/api/audio/audio_processing.h
+++ b/api/audio/audio_processing.h
@@ -86,7 +86,6 @@ class EchoDetector;
 // float interfaces use deinterleaved data.
 //
 // Usage example, omitting error checking:
-// rtc::scoped_refptr<AudioProcessing> apm = AudioProcessingBuilder().Create();
 //
 // AudioProcessing::Config config;
 // config.echo_canceller.enabled = true;
@@ -102,7 +101,8 @@ class EchoDetector;
 //
 // config.high_pass_filter.enabled = true;
 //
-// apm->ApplyConfig(config)
+// scoped_refptr<AudioProcessing> apm =
+//     BuiltinAudioProcessingBuilder(config).Build(CreateEnvironment());
 //
 // // Start a voice call...
 //
@@ -780,9 +780,8 @@ class CustomProcessing {
   virtual ~CustomProcessing() {}
 };
 
-// TODO: bugs.webrtc.org/369904700 - Deprecate and remove in favor of the
-// BuiltinAudioProcessingBuilder.
-class RTC_EXPORT AudioProcessingBuilder {
+// Use BuiltinAudioProcessingBuilder instead, see bugs.webrtc.org/369904700
+class RTC_EXPORT [[deprecated]] AudioProcessingBuilder {
  public:
   AudioProcessingBuilder();
   AudioProcessingBuilder(const AudioProcessingBuilder&) = delete;
diff --git a/api/audio/echo_detector_creator.h b/api/audio/echo_detector_creator.h
index 27dd58cbf2..8f260ae0da 100644
--- a/api/audio/echo_detector_creator.h
+++ b/api/audio/echo_detector_creator.h
@@ -17,9 +17,9 @@
 namespace webrtc {
 
 // Returns an instance of the WebRTC implementation of a residual echo detector.
-// It can be provided to the webrtc::AudioProcessingBuilder to obtain the
+// It can be provided to the webrtc::BuiltinAudioProcessingBuilder to obtain the
 // usual residual echo metrics.
-rtc::scoped_refptr<EchoDetector> CreateEchoDetector();
+scoped_refptr<EchoDetector> CreateEchoDetector();
 
 }  // namespace webrtc
 
diff --git a/api/voip/voip_engine.h b/api/voip/voip_engine.h
index d223f6ad6c..cdb74f9ff9 100644
--- a/api/voip/voip_engine.h
+++ b/api/voip/voip_engine.h
@@ -35,7 +35,8 @@ class VoipVolumeControl;
 //   config.audio_device =
 //       AudioDeviceModule::Create(AudioDeviceModule::kPlatformDefaultAudio,
 //                                 config.task_queue_factory.get());
-//   config.audio_processing = AudioProcessingBuilder().Create();
+//   config.audio_processing_builder =
+//       std::make_unique<BuiltinAudioProcessingBuilder>();
 //
 //   auto voip_engine = CreateVoipEngine(std::move(config));
 //
diff --git a/modules/audio_processing/g3doc/audio_processing_module.md b/modules/audio_processing/g3doc/audio_processing_module.md
index a77f62fbaf..2e0a4b0d4e 100644
--- a/modules/audio_processing/g3doc/audio_processing_module.md
+++ b/modules/audio_processing/g3doc/audio_processing_module.md
@@ -10,8 +10,8 @@ microphone signal. These effects are required for VoIP calling and some
 examples include echo cancellation (AEC), noise suppression (NS) and
 automatic gain control (AGC).
 
-The API for APM resides in [`/modules/audio_processing/include`][https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_processing/include].
-APM is created using the [`AudioProcessingBuilder`][https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/audio_processing/include/audio_processing.h]
+The API for APM resides in [`/api/audio/audio_processing.h`][https://webrtc.googlesource.com/src/+/refs/heads/main/api/audio/audio_processing.h].
+APM is created using the [`BuiltinAudioProcessingBuilder`][https://webrtc.googlesource.com/src/+/refs/heads/main/api/audio/builtin_audio_processing_builder.h]
 builder that allows it to be customized and configured.
 
 Some specific aspects of APM include that: