diff --git a/api/audio/audio_frame.cc b/api/audio/audio_frame.cc
index d9212a2..47459ac 100644
--- a/api/audio/audio_frame.cc
+++ b/api/audio/audio_frame.cc
@@ -41,6 +41,7 @@
   vad_activity_ = kVadUnknown;
   profile_timestamp_ms_ = 0;
   packet_infos_ = RtpPacketInfos();
+  absolute_capture_timestamp_ms_ = absl::nullopt;
 }
 
 void AudioFrame::UpdateFrame(uint32_t timestamp,
@@ -86,6 +87,7 @@
   vad_activity_ = src.vad_activity_;
   num_channels_ = src.num_channels_;
   channel_layout_ = src.channel_layout_;
+  absolute_capture_timestamp_ms_ = src.absolute_capture_timestamp_ms();
 
   const size_t length = samples_per_channel_ * num_channels_;
   RTC_CHECK_LE(length, kMaxDataSizeSamples);
diff --git a/api/audio/audio_frame.h b/api/audio/audio_frame.h
index cda8c26..06b0b28 100644
--- a/api/audio/audio_frame.h
+++ b/api/audio/audio_frame.h
@@ -104,6 +104,15 @@
   ChannelLayout channel_layout() const { return channel_layout_; }
   int sample_rate_hz() const { return sample_rate_hz_; }
 
+  void set_absolute_capture_timestamp_ms(
+      int64_t absolute_capture_time_stamp_ms) {
+    absolute_capture_timestamp_ms_ = absolute_capture_time_stamp_ms;
+  }
+
+  absl::optional<int64_t> absolute_capture_timestamp_ms() const {
+    return absolute_capture_timestamp_ms_;
+  }
+
   // RTP timestamp of the first sample in the AudioFrame.
   uint32_t timestamp_ = 0;
   // Time since the first frame in milliseconds.
@@ -121,8 +130,8 @@
   // Monotonically increasing timestamp intended for profiling of audio frames.
   // Typically used for measuring elapsed time between two different points in
   // the audio path. No lock is used to save resources and we are thread safe
-  // by design. Also, absl::optional is not used since it will cause a "complex
-  // class/struct needs an explicit out-of-line destructor" build error.
+  // by design.
+  // TODO(nisse@webrtc.org): consider using absl::optional.
   int64_t profile_timestamp_ms_ = 0;
 
   // Information about packets used to assemble this audio frame. This is needed
@@ -150,6 +159,12 @@
   int16_t data_[kMaxDataSizeSamples];
   bool muted_ = true;
 
+  // Absolute capture timestamp when this audio frame was originally captured.
+  // This is only valid for audio frames captured on this machine. The absolute
+  // capture timestamp of a received frame is found in |packet_infos_|.
+  // This timestamp MUST be based on the same clock as rtc::TimeMillis().
+  absl::optional<int64_t> absolute_capture_timestamp_ms_;
+
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
 };
 
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index 2fe2563..b4b2b4a 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -880,8 +880,12 @@
         audio_frame->timestamp_, static_cast<const int16_t*>(audio_data),
         number_of_frames, sample_rate, audio_frame->speech_type_,
         audio_frame->vad_activity_, number_of_channels);
-    // TODO(bugs.webrtc.org/10739): pass absolute_capture_timestamp_ms to
-    // stream_.
+    // TODO(bugs.webrtc.org/10739): add dcheck that
+    // |absolute_capture_timestamp_ms| always receives a value.
+    if (absolute_capture_timestamp_ms) {
+      audio_frame->set_absolute_capture_timestamp_ms(
+          *absolute_capture_timestamp_ms);
+    }
     stream_->SendAudioData(std::move(audio_frame));
   }
 
diff --git a/modules/audio_coding/acm2/audio_coding_module.cc b/modules/audio_coding/acm2/audio_coding_module.cc
index f3dd5b1..e28be18 100644
--- a/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/modules/audio_coding/acm2/audio_coding_module.cc
@@ -109,7 +109,6 @@
     // If a re-mix is required (up or down), this buffer will store a re-mixed
     // version of the input.
     std::vector<int16_t> buffer;
-    int64_t absolute_capture_timestamp_ms;
   };
 
   InputData input_data_ RTC_GUARDED_BY(acm_crit_sect_);
@@ -132,7 +131,11 @@
 
   int Add10MsDataInternal(const AudioFrame& audio_frame, InputData* input_data)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
-  int Encode(const InputData& input_data)
+
+  // TODO(bugs.webrtc.org/10739): change |absolute_capture_timestamp_ms| to
+  // int64_t when it always receives a valid value.
+  int Encode(const InputData& input_data,
+             absl::optional<int64_t> absolute_capture_timestamp_ms)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
 
   int InitializeReceiverSafe() RTC_EXCLUSIVE_LOCKS_REQUIRED(acm_crit_sect_);
@@ -231,7 +234,11 @@
 
 AudioCodingModuleImpl::~AudioCodingModuleImpl() = default;
 
-int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
+int32_t AudioCodingModuleImpl::Encode(
+    const InputData& input_data,
+    absl::optional<int64_t> absolute_capture_timestamp_ms) {
+  // TODO(bugs.webrtc.org/10739): add dcheck that
+  // |audio_frame.absolute_capture_timestamp_ms()| always has a value.
   AudioEncoder::EncodedInfo encoded_info;
   uint8_t previous_pltype;
 
@@ -304,7 +311,7 @@
       packetization_callback_->SendData(
           frame_type, encoded_info.payload_type, encoded_info.encoded_timestamp,
           encode_buffer_.data(), encode_buffer_.size(),
-          input_data.absolute_capture_timestamp_ms);
+          absolute_capture_timestamp_ms.value_or(-1));
     }
 
     if (vad_callback_) {
@@ -339,7 +346,11 @@
 int AudioCodingModuleImpl::Add10MsData(const AudioFrame& audio_frame) {
   rtc::CritScope lock(&acm_crit_sect_);
   int r = Add10MsDataInternal(audio_frame, &input_data_);
-  return r < 0 ? r : Encode(input_data_);
+  // TODO(bugs.webrtc.org/10739): add dcheck that
+  // |audio_frame.absolute_capture_timestamp_ms()| always has a value.
+  return r < 0
+             ? r
+             : Encode(input_data_, audio_frame.absolute_capture_timestamp_ms());
 }
 
 int AudioCodingModuleImpl::Add10MsDataInternal(const AudioFrame& audio_frame,
@@ -394,9 +405,6 @@
   input_data->input_timestamp = ptr_frame->timestamp_;
   input_data->length_per_channel = ptr_frame->samples_per_channel_;
   input_data->audio_channel = current_num_channels;
-  // TODO(bugs.webrtc.org/10739): Assign from a corresponding field in
-  // audio_frame when it is added in AudioFrame.
-  input_data->absolute_capture_timestamp_ms = 0;
 
   if (!same_num_channels) {
     // Remixes the input frame to the output data and in the process resize the