diff --git a/audio/audio_transport_impl.cc b/audio/audio_transport_impl.cc
index 9f2823b..42a81d5 100644
--- a/audio/audio_transport_impl.cc
+++ b/audio/audio_transport_impl.cc
@@ -107,35 +107,35 @@
 
 int32_t AudioTransportImpl::RecordedDataIsAvailable(
     const void* audio_data,
-    const size_t number_of_frames,
-    const size_t bytes_per_sample,
-    const size_t number_of_channels,
-    const uint32_t sample_rate,
-    const uint32_t audio_delay_milliseconds,
-    const int32_t clock_drift,
-    const uint32_t volume,
-    const bool key_pressed,
+    size_t number_of_frames,
+    size_t bytes_per_sample,
+    size_t number_of_channels,
+    uint32_t sample_rate,
+    uint32_t audio_delay_milliseconds,
+    int32_t clock_drift,
+    uint32_t volume,
+    bool key_pressed,
     uint32_t& new_mic_volume) {  // NOLINT: to avoid changing APIs
   return RecordedDataIsAvailable(
       audio_data, number_of_frames, bytes_per_sample, number_of_channels,
       sample_rate, audio_delay_milliseconds, clock_drift, volume, key_pressed,
-      new_mic_volume, /* estimated_capture_time_ns */ 0);
+      new_mic_volume, /*estimated_capture_time_ns=*/absl::nullopt);
 }
 
 // Not used in Chromium. Process captured audio and distribute to all sending
 // streams, and try to do this at the lowest possible sample rate.
 int32_t AudioTransportImpl::RecordedDataIsAvailable(
     const void* audio_data,
-    const size_t number_of_frames,
-    const size_t bytes_per_sample,
-    const size_t number_of_channels,
-    const uint32_t sample_rate,
-    const uint32_t audio_delay_milliseconds,
-    const int32_t /*clock_drift*/,
-    const uint32_t /*volume*/,
-    const bool key_pressed,
+    size_t number_of_frames,
+    size_t bytes_per_sample,
+    size_t number_of_channels,
+    uint32_t sample_rate,
+    uint32_t audio_delay_milliseconds,
+    int32_t /*clock_drift*/,
+    uint32_t /*volume*/,
+    bool key_pressed,
     uint32_t& /*new_mic_volume*/,
-    const int64_t
+    absl::optional<int64_t>
         estimated_capture_time_ns) {  // NOLINT: to avoid changing APIs
   RTC_DCHECK(audio_data);
   RTC_DCHECK_GE(number_of_channels, 1);
@@ -166,8 +166,11 @@
   ProcessCaptureFrame(audio_delay_milliseconds, key_pressed,
                       swap_stereo_channels, audio_processing_,
                       audio_frame.get());
-  audio_frame->set_absolute_capture_timestamp_ms(estimated_capture_time_ns /
-                                                 1000000);
+
+  if (estimated_capture_time_ns) {
+    audio_frame->set_absolute_capture_timestamp_ms(*estimated_capture_time_ns /
+                                                   1000000);
+  }
 
   RTC_DCHECK_GT(audio_frame->samples_per_channel_, 0);
   if (async_audio_processing_)
diff --git a/audio/audio_transport_impl.h b/audio/audio_transport_impl.h
index ba067de..24b09d2 100644
--- a/audio/audio_transport_impl.h
+++ b/audio/audio_transport_impl.h
@@ -52,17 +52,18 @@
                                   bool keyPressed,
                                   uint32_t& newMicLevel) override;
 
-  int32_t RecordedDataIsAvailable(const void* audioSamples,
-                                  size_t nSamples,
-                                  size_t nBytesPerSample,
-                                  size_t nChannels,
-                                  uint32_t samplesPerSec,
-                                  uint32_t totalDelayMS,
-                                  int32_t clockDrift,
-                                  uint32_t currentMicLevel,
-                                  bool keyPressed,
-                                  uint32_t& newMicLevel,
-                                  int64_t estimated_capture_time_ns) override;
+  int32_t RecordedDataIsAvailable(
+      const void* audioSamples,
+      size_t nSamples,
+      size_t nBytesPerSample,
+      size_t nChannels,
+      uint32_t samplesPerSec,
+      uint32_t totalDelayMS,
+      int32_t clockDrift,
+      uint32_t currentMicLevel,
+      bool keyPressed,
+      uint32_t& newMicLevel,
+      absl::optional<int64_t> estimated_capture_time_ns) override;
 
   int32_t NeedMorePlayData(size_t nSamples,
                            size_t nBytesPerSample,
diff --git a/modules/audio_device/audio_device_buffer.cc b/modules/audio_device/audio_device_buffer.cc
index 6232a93..b1be445 100644
--- a/modules/audio_device/audio_device_buffer.cc
+++ b/modules/audio_device/audio_device_buffer.cc
@@ -55,7 +55,6 @@
       typing_status_(false),
       play_delay_ms_(0),
       rec_delay_ms_(0),
-      capture_timestamp_ns_(0),
       num_stat_reports_(0),
       last_timer_task_time_(0),
       rec_stat_count_(0),
@@ -231,12 +230,13 @@
 
 int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer,
                                              size_t samples_per_channel) {
-  return SetRecordedBuffer(audio_buffer, samples_per_channel, 0);
+  return SetRecordedBuffer(audio_buffer, samples_per_channel, absl::nullopt);
 }
 
-int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audio_buffer,
-                                             size_t samples_per_channel,
-                                             int64_t capture_timestamp_ns) {
+int32_t AudioDeviceBuffer::SetRecordedBuffer(
+    const void* audio_buffer,
+    size_t samples_per_channel,
+    absl::optional<int64_t> capture_timestamp_ns) {
   // Copy the complete input buffer to the local buffer.
   const size_t old_size = rec_buffer_.size();
   rec_buffer_.SetData(static_cast<const int16_t*>(audio_buffer),
@@ -247,17 +247,13 @@
     RTC_LOG(LS_INFO) << "Size of recording buffer: " << rec_buffer_.size();
   }
 
-  // If the timestamp is less then or equal to zero, it's not valid and are
-  // ignored. If we do antimestamp alignment on them they might accidentally
-  // become greater then zero, and will be handled as if they were a correct
-  // timestamp.
-  capture_timestamp_ns_ =
-      (capture_timestamp_ns > 0)
-          ? rtc::kNumNanosecsPerMicrosec *
-                timestamp_aligner_.TranslateTimestamp(
-                    capture_timestamp_ns_ / rtc::kNumNanosecsPerMicrosec,
-                    rtc::TimeMicros())
-          : capture_timestamp_ns;
+  if (capture_timestamp_ns) {
+    capture_timestamp_ns_ =
+        rtc::kNumNanosecsPerMicrosec *
+        timestamp_aligner_.TranslateTimestamp(
+            *capture_timestamp_ns / rtc::kNumNanosecsPerMicrosec,
+            rtc::TimeMicros());
+  }
   // Derive a new level value twice per second and check if it is non-zero.
   int16_t max_abs = 0;
   RTC_DCHECK_LT(rec_stat_count_, 50);
diff --git a/modules/audio_device/audio_device_buffer.h b/modules/audio_device/audio_device_buffer.h
index 9a6a88a..eb681a7 100644
--- a/modules/audio_device/audio_device_buffer.h
+++ b/modules/audio_device/audio_device_buffer.h
@@ -102,9 +102,10 @@
   virtual int32_t SetRecordedBuffer(const void* audio_buffer,
                                     size_t samples_per_channel);
 
-  virtual int32_t SetRecordedBuffer(const void* audio_buffer,
-                                    size_t samples_per_channel,
-                                    int64_t capture_timestamp_ns);
+  virtual int32_t SetRecordedBuffer(
+      const void* audio_buffer,
+      size_t samples_per_channel,
+      absl::optional<int64_t> capture_timestamp_ns);
   virtual void SetVQEData(int play_delay_ms, int rec_delay_ms);
   virtual int32_t DeliverRecordedData();
   uint32_t NewMicLevel() const;
@@ -194,7 +195,7 @@
   int rec_delay_ms_;
 
   // Capture timestamp.
-  int64_t capture_timestamp_ns_;
+  absl::optional<int64_t> capture_timestamp_ns_;
 
   // Counts number of times LogStats() has been called.
   size_t num_stat_reports_ RTC_GUARDED_BY(task_queue_);
diff --git a/modules/audio_device/audio_device_data_observer.cc b/modules/audio_device/audio_device_data_observer.cc
index 3775e7c..0524830 100644
--- a/modules/audio_device/audio_device_data_observer.cc
+++ b/modules/audio_device/audio_device_data_observer.cc
@@ -55,24 +55,25 @@
                                   uint32_t currentMicLevel,
                                   bool keyPressed,
                                   uint32_t& newMicLevel) override {
-    return RecordedDataIsAvailable(audioSamples, nSamples, nBytesPerSample,
-                                   nChannels, samples_per_sec, total_delay_ms,
-                                   clockDrift, currentMicLevel, keyPressed,
-                                   newMicLevel, /*capture_timestamp_ns*/ 0);
+    return RecordedDataIsAvailable(
+        audioSamples, nSamples, nBytesPerSample, nChannels, samples_per_sec,
+        total_delay_ms, clockDrift, currentMicLevel, keyPressed, newMicLevel,
+        /*capture_timestamp_ns=*/absl::nullopt);
   }
 
   // AudioTransport methods overrides.
-  int32_t RecordedDataIsAvailable(const void* audioSamples,
-                                  size_t nSamples,
-                                  size_t nBytesPerSample,
-                                  size_t nChannels,
-                                  uint32_t samples_per_sec,
-                                  uint32_t total_delay_ms,
-                                  int32_t clockDrift,
-                                  uint32_t currentMicLevel,
-                                  bool keyPressed,
-                                  uint32_t& newMicLevel,
-                                  int64_t capture_timestamp_ns) override {
+  int32_t RecordedDataIsAvailable(
+      const void* audioSamples,
+      size_t nSamples,
+      size_t nBytesPerSample,
+      size_t nChannels,
+      uint32_t samples_per_sec,
+      uint32_t total_delay_ms,
+      int32_t clockDrift,
+      uint32_t currentMicLevel,
+      bool keyPressed,
+      uint32_t& newMicLevel,
+      absl::optional<int64_t> capture_timestamp_ns) override {
     int32_t res = 0;
     // Capture PCM data of locally captured audio.
     if (observer_) {
diff --git a/modules/audio_device/include/audio_device_defines.h b/modules/audio_device/include/audio_device_defines.h
index 89d33f8..d677d41 100644
--- a/modules/audio_device/include/audio_device_defines.h
+++ b/modules/audio_device/include/audio_device_defines.h
@@ -56,7 +56,7 @@
       uint32_t currentMicLevel,
       bool keyPressed,
       uint32_t& newMicLevel,
-      int64_t estimatedCaptureTimeNS) {  // NOLINT
+      absl::optional<int64_t> estimatedCaptureTimeNS) {  // NOLINT
     // TODO(webrtc:13620) Make the default behaver of the new API to behave as
     // the old API. This can be pure virtual if all uses of the old API is
     // removed.
diff --git a/modules/audio_device/include/mock_audio_transport.h b/modules/audio_device/include/mock_audio_transport.h
index e1be5f4..b886967 100644
--- a/modules/audio_device/include/mock_audio_transport.h
+++ b/modules/audio_device/include/mock_audio_transport.h
@@ -48,7 +48,7 @@
                uint32_t currentMicLevel,
                bool keyPressed,
                uint32_t& newMicLevel,
-               int64_t estimated_capture_time_ns),
+               absl::optional<int64_t> estimated_capture_time_ns),
               (override));
 
   MOCK_METHOD(int32_t,