diff --git a/audio/audio_state.cc b/audio/audio_state.cc
index 24a7ce8..06a686331 100644
--- a/audio/audio_state.cc
+++ b/audio/audio_state.cc
@@ -28,8 +28,7 @@
 AudioState::AudioState(const AudioState::Config& config)
     : config_(config),
       audio_transport_(config_.audio_mixer,
-                       config_.audio_processing.get(),
-                       config_.audio_device_module.get()) {
+                       config_.audio_processing.get()) {
   process_thread_checker_.DetachFromThread();
   RTC_DCHECK(config_.audio_mixer);
   RTC_DCHECK(config_.audio_device_module);
diff --git a/audio/audio_transport_impl.cc b/audio/audio_transport_impl.cc
index 586fc46..30ffc6d 100644
--- a/audio/audio_transport_impl.cc
+++ b/audio/audio_transport_impl.cc
@@ -43,8 +43,7 @@
   audio_frame->num_channels_ = std::min(input_num_channels, send_num_channels);
 }
 
-void ProcessCaptureFrame(int analog_level,
-                         uint32_t delay_ms,
+void ProcessCaptureFrame(uint32_t delay_ms,
                          bool key_pressed,
                          bool swap_stereo_channels,
                          AudioProcessing* audio_processing,
@@ -53,13 +52,9 @@
   RTC_DCHECK(audio_frame);
   RTC_DCHECK(
       !audio_processing->echo_cancellation()->is_drift_compensation_enabled());
-  GainControl* agc = audio_processing->gain_control();
-  int error = agc->set_stream_analog_level(analog_level);
-  RTC_DCHECK_EQ(0, error) <<
-      "set_stream_analog_level failed: analog_level = " << analog_level;
   audio_processing->set_stream_delay_ms(delay_ms);
   audio_processing->set_stream_key_pressed(key_pressed);
-  error = audio_processing->ProcessStream(audio_frame);
+  int error = audio_processing->ProcessStream(audio_frame);
   RTC_DCHECK_EQ(0, error) << "ProcessStream() error: " << error;
   if (swap_stereo_channels) {
     AudioFrameOperations::SwapStereoChannels(audio_frame);
@@ -87,14 +82,11 @@
 }  // namespace
 
 AudioTransportImpl::AudioTransportImpl(AudioMixer* mixer,
-                                       AudioProcessing* audio_processing,
-                                       AudioDeviceModule* audio_device_module)
+                                       AudioProcessing* audio_processing)
     : audio_processing_(audio_processing),
-      audio_device_module_(audio_device_module),
       mixer_(mixer) {
   RTC_DCHECK(mixer);
   RTC_DCHECK(audio_processing);
-  RTC_DCHECK(audio_device_module);
 }
 
 AudioTransportImpl::~AudioTransportImpl() {}
@@ -109,7 +101,7 @@
     const uint32_t sample_rate,
     const uint32_t audio_delay_milliseconds,
     const int32_t /*clock_drift*/,
-    const uint32_t volume,
+    const uint32_t /*volume*/,
     const bool key_pressed,
     uint32_t& /*new_mic_volume*/) {  // NOLINT: to avoid changing APIs
   RTC_DCHECK(audio_data);
@@ -122,35 +114,6 @@
   RTC_DCHECK_LE(bytes_per_sample * number_of_frames * number_of_channels,
                 AudioFrame::kMaxDataSizeBytes);
 
-  // TODO(solenberg): Remove volume handling since it is now always 0.
-  uint16_t voe_mic_level = 0;
-  {
-    constexpr uint32_t kMaxVolumeLevel = 255;
-    uint32_t max_volume = 0;
-
-    // Check for zero to skip this calculation; the consumer may use this to
-    // indicate no volume is available.
-    if (volume != 0) {
-      // Scale from ADM to VoE level range
-      if (audio_device_module_->MaxMicrophoneVolume(&max_volume) == 0) {
-        if (max_volume != 0) {
-          voe_mic_level = static_cast<uint16_t>(
-              (volume * kMaxVolumeLevel + static_cast<int>(max_volume / 2)) /
-              max_volume);
-        }
-      }
-      // We learned that on certain systems (e.g Linux) the voe_mic_level
-      // can be greater than the maxVolumeLevel therefore
-      // we are going to cap the voe_mic_level to the maxVolumeLevel
-      // and change the maxVolume to volume if it turns out that
-      // the voe_mic_level is indeed greater than the maxVolumeLevel.
-      if (voe_mic_level > kMaxVolumeLevel) {
-        voe_mic_level = kMaxVolumeLevel;
-        max_volume = volume;
-      }
-    }
-  }
-
   int send_sample_rate_hz = 0;
   size_t send_num_channels = 0;
   bool swap_stereo_channels = false;
@@ -168,7 +131,7 @@
   voe::RemixAndResample(static_cast<const int16_t*>(audio_data),
                         number_of_frames, number_of_channels, sample_rate,
                         &capture_resampler_, audio_frame.get());
-  ProcessCaptureFrame(voe_mic_level, audio_delay_milliseconds, key_pressed,
+  ProcessCaptureFrame(audio_delay_milliseconds, key_pressed,
                       swap_stereo_channels, audio_processing_,
                       audio_frame.get());
 
diff --git a/audio/audio_transport_impl.h b/audio/audio_transport_impl.h
index e7de7e9..8a316a5 100644
--- a/audio/audio_transport_impl.h
+++ b/audio/audio_transport_impl.h
@@ -31,8 +31,7 @@
 class AudioTransportImpl : public AudioTransport {
  public:
   AudioTransportImpl(AudioMixer* mixer,
-                     AudioProcessing* audio_processing,
-                     AudioDeviceModule* audio_device_module);
+                     AudioProcessing* audio_processing);
   ~AudioTransportImpl() override;
 
   int32_t RecordedDataIsAvailable(const void* audioSamples,
@@ -82,7 +81,6 @@
   size_t send_num_channels_ RTC_GUARDED_BY(capture_lock_) = 1;
   bool typing_noise_detected_ RTC_GUARDED_BY(capture_lock_) = false;
   bool swap_stereo_channels_ RTC_GUARDED_BY(capture_lock_) = false;
-  AudioDeviceModule* audio_device_module_ = nullptr;
   PushResampler<int16_t> capture_resampler_;
   voe::AudioLevel audio_level_;
   TypingDetection typing_detection_;