diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index ba85fc5403..02a82f4c78 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -1235,7 +1235,7 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
   if (submodules_.echo_control_mobile) {
     // Ensure that the stream delay was set before the call to the
     // AECM ProcessCaptureAudio function.
-    if (!was_stream_delay_set()) {
+    if (!capture_.was_stream_delay_set) {
       return AudioProcessing::kStreamParameterNotSetError;
     }
 
@@ -1252,7 +1252,7 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
     if (submodules_.echo_controller) {
       data_dumper_->DumpRaw("stream_delay", stream_delay_ms());
 
-      if (was_stream_delay_set()) {
+      if (capture_.was_stream_delay_set) {
         submodules_.echo_controller->SetAudioBufferDelay(stream_delay_ms());
       }
 
@@ -1553,7 +1553,6 @@ int AudioProcessingImpl::set_stream_delay_ms(int delay) {
   rtc::CritScope cs(&crit_capture_);
   Error retval = kNoError;
   capture_.was_stream_delay_set = true;
-  delay += capture_.delay_offset_ms;
 
   if (delay < 0) {
     delay = 0;
@@ -1600,26 +1599,11 @@ int AudioProcessingImpl::stream_delay_ms() const {
   return capture_nonlocked_.stream_delay_ms;
 }
 
-bool AudioProcessingImpl::was_stream_delay_set() const {
-  // Used as callback from submodules, hence locking is not allowed.
-  return capture_.was_stream_delay_set;
-}
-
 void AudioProcessingImpl::set_stream_key_pressed(bool key_pressed) {
   rtc::CritScope cs(&crit_capture_);
   capture_.key_pressed = key_pressed;
 }
 
-void AudioProcessingImpl::set_delay_offset_ms(int offset) {
-  rtc::CritScope cs(&crit_capture_);
-  capture_.delay_offset_ms = offset;
-}
-
-int AudioProcessingImpl::delay_offset_ms() const {
-  rtc::CritScope cs(&crit_capture_);
-  return capture_.delay_offset_ms;
-}
-
 void AudioProcessingImpl::set_stream_analog_level(int level) {
   rtc::CritScope cs_capture(&crit_capture_);
 
@@ -2107,8 +2091,7 @@ void AudioProcessingImpl::RecordAudioProcessingState() {
 }
 
 AudioProcessingImpl::ApmCaptureState::ApmCaptureState()
-    : delay_offset_ms(0),
-      was_stream_delay_set(false),
+    : was_stream_delay_set(false),
       output_will_be_muted(false),
       key_pressed(false),
       capture_processing_format(kSampleRate16kHz),
diff --git a/modules/audio_processing/audio_processing_impl.h b/modules/audio_processing/audio_processing_impl.h
index af5a0f63c4..c74d7ca89e 100644
--- a/modules/audio_processing/audio_processing_impl.h
+++ b/modules/audio_processing/audio_processing_impl.h
@@ -89,8 +89,6 @@ class AudioProcessingImpl : public AudioProcessing {
       rtc::ArrayView<std::array<float, 160>> linear_output) const override;
   void set_output_will_be_muted(bool muted) override;
   int set_stream_delay_ms(int delay) override;
-  void set_delay_offset_ms(int offset) override;
-  int delay_offset_ms() const override;
   void set_stream_key_pressed(bool key_pressed) override;
   void set_stream_analog_level(int level) override;
   int recommended_stream_analog_level() const override;
@@ -115,8 +113,6 @@ class AudioProcessingImpl : public AudioProcessing {
   size_t num_output_channels() const override;
   size_t num_reverse_channels() const override;
   int stream_delay_ms() const override;
-  bool was_stream_delay_set() const override
-      RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
 
   AudioProcessingStats GetStatistics(bool has_remote_tracks) override {
     return GetStatistics();
@@ -394,7 +390,6 @@ class AudioProcessingImpl : public AudioProcessing {
   struct ApmCaptureState {
     ApmCaptureState();
     ~ApmCaptureState();
-    int delay_offset_ms;
     bool was_stream_delay_set;
     bool output_will_be_muted;
     bool key_pressed;
diff --git a/modules/audio_processing/audio_processing_impl_locking_unittest.cc b/modules/audio_processing/audio_processing_impl_locking_unittest.cc
index d9a8741bf8..50747a732d 100644
--- a/modules/audio_processing/audio_processing_impl_locking_unittest.cc
+++ b/modules/audio_processing/audio_processing_impl_locking_unittest.cc
@@ -816,15 +816,10 @@ void CaptureProcessor::ApplyRuntimeSettingScheme() {
         ASSERT_EQ(AudioProcessing::Error::kNoError,
                   apm_->set_stream_delay_ms(30));
         apm_->set_stream_key_pressed(true);
-        apm_->set_delay_offset_ms(15);
-        EXPECT_EQ(apm_->delay_offset_ms(), 15);
       } else {
         ASSERT_EQ(AudioProcessing::Error::kNoError,
                   apm_->set_stream_delay_ms(50));
         apm_->set_stream_key_pressed(false);
-        apm_->set_delay_offset_ms(20);
-        EXPECT_EQ(apm_->delay_offset_ms(), 20);
-        apm_->delay_offset_ms();
       }
       break;
     default:
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index f83efe66f0..2f557e5c70 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -732,30 +732,6 @@ TEST_F(ApmTest, StreamParametersFloat) {
   StreamParametersTest(kFloatFormat);
 }
 
-TEST_F(ApmTest, DefaultDelayOffsetIsZero) {
-  EXPECT_EQ(0, apm_->delay_offset_ms());
-  EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(50));
-  EXPECT_EQ(50, apm_->stream_delay_ms());
-}
-
-TEST_F(ApmTest, DelayOffsetWithLimitsIsSetProperly) {
-  // High limit of 500 ms.
-  apm_->set_delay_offset_ms(100);
-  EXPECT_EQ(100, apm_->delay_offset_ms());
-  EXPECT_EQ(apm_->kBadStreamParameterWarning, apm_->set_stream_delay_ms(450));
-  EXPECT_EQ(500, apm_->stream_delay_ms());
-  EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
-  EXPECT_EQ(200, apm_->stream_delay_ms());
-
-  // Low limit of 0 ms.
-  apm_->set_delay_offset_ms(-50);
-  EXPECT_EQ(-50, apm_->delay_offset_ms());
-  EXPECT_EQ(apm_->kBadStreamParameterWarning, apm_->set_stream_delay_ms(20));
-  EXPECT_EQ(0, apm_->stream_delay_ms());
-  EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
-  EXPECT_EQ(50, apm_->stream_delay_ms());
-}
-
 void ApmTest::TestChangingChannelsInt16Interface(
     size_t num_channels,
     AudioProcessing::Error expected_return) {
diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h
index a41dfa116c..f5d37b9d8c 100644
--- a/modules/audio_processing/include/audio_processing.h
+++ b/modules/audio_processing/include/audio_processing.h
@@ -611,20 +611,11 @@ class RTC_EXPORT AudioProcessing : public rtc::RefCountInterface {
   //     ProcessStream().
   virtual int set_stream_delay_ms(int delay) = 0;
   virtual int stream_delay_ms() const = 0;
-  virtual bool was_stream_delay_set() const = 0;
 
   // Call to signal that a key press occurred (true) or did not occur (false)
   // with this chunk of audio.
   virtual void set_stream_key_pressed(bool key_pressed) = 0;
 
-  // Sets a delay |offset| in ms to add to the values passed in through
-  // set_stream_delay_ms(). May be positive or negative.
-  //
-  // Note that this could cause an otherwise valid value passed to
-  // set_stream_delay_ms() to return an error.
-  virtual void set_delay_offset_ms(int offset) = 0;
-  virtual int delay_offset_ms() const = 0;
-
   // Attaches provided webrtc::AecDump for recording debugging
   // information. Log file and maximum file size logic is supposed to
   // be handled by implementing instance of AecDump. Calling this