diff --git a/common_audio/audio_util_unittest.cc b/common_audio/audio_util_unittest.cc
index fc446d1ff8..497ff7329f 100644
--- a/common_audio/audio_util_unittest.cc
+++ b/common_audio/audio_util_unittest.cc
@@ -201,57 +201,5 @@ TEST(AudioUtilTest, DownmixInterleavedToMono) {
   }
 }
 
-TEST(AudioUtilTest, DownmixToMonoTest) {
-  {
-    const size_t kNumFrames = 4;
-    const int kNumChannels = 1;
-    const float input_data[kNumChannels][kNumFrames] = {{1.f, 2.f, -1.f, -3.f}};
-    const float* input[kNumChannels];
-    for (int i = 0; i < kNumChannels; ++i) {
-      input[i] = input_data[i];
-    }
-
-    float downmixed[kNumFrames];
-
-    DownmixToMono<float, float>(input, kNumFrames, kNumChannels, downmixed);
-
-    EXPECT_THAT(downmixed, ElementsAreArray(input_data[0]));
-  }
-  {
-    const size_t kNumFrames = 3;
-    const int kNumChannels = 2;
-    const float input_data[kNumChannels][kNumFrames] = {{1.f, 2.f, -1.f},
-                                                        {3.f, 0.f, 1.f}};
-    const float* input[kNumChannels];
-    for (int i = 0; i < kNumChannels; ++i) {
-      input[i] = input_data[i];
-    }
-
-    float downmixed[kNumFrames];
-    const float expected[kNumFrames] = {2.f, 1.f, 0.f};
-
-    DownmixToMono<float, float>(input, kNumFrames, kNumChannels, downmixed);
-
-    EXPECT_THAT(downmixed, ElementsAreArray(expected));
-  }
-  {
-    const size_t kNumFrames = 3;
-    const int kNumChannels = 3;
-    const int16_t input_data[kNumChannels][kNumFrames] = {
-        {30000, -5, -30000}, {30000, -10, -30999}, {24001, -20, -30000}};
-    const int16_t* input[kNumChannels];
-    for (int i = 0; i < kNumChannels; ++i) {
-      input[i] = input_data[i];
-    }
-
-    int16_t downmixed[kNumFrames];
-    const int16_t expected[kNumFrames] = {28000, -11, -30333};
-
-    DownmixToMono<int16_t, int32_t>(input, kNumFrames, kNumChannels, downmixed);
-
-    EXPECT_THAT(downmixed, ElementsAreArray(expected));
-  }
-}
-
 }  // namespace
 }  // namespace webrtc
diff --git a/common_audio/include/audio_util.h b/common_audio/include/audio_util.h
index 07ec2ecbcf..12e65259a4 100644
--- a/common_audio/include/audio_util.h
+++ b/common_audio/include/audio_util.h
@@ -149,26 +149,9 @@ void Interleave(const DeinterleavedView<const T>& deinterleaved,
   }
 }
 
-// TODO: b/335805780 - Accept DeInterleavedView and MonoView.
-// Possibly just delete this method if it isn't used.
-template <typename T, typename Intermediate>
-void DownmixToMono(const T* const* input_channels,
-                   size_t num_frames,
-                   int num_channels,
-                   T* out) {
-  for (size_t i = 0; i < num_frames; ++i) {
-    Intermediate value = input_channels[0][i];
-    for (int j = 1; j < num_channels; ++j) {
-      value += input_channels[j][i];
-    }
-    out[i] = value / num_channels;
-  }
-}
-
 // Downmixes an interleaved multichannel signal to a single channel by averaging
 // all channels.
 // TODO: b/335805780 - Accept InterleavedView and DeinterleavedView.
-// Possibly just delete this method if it isn't used.
 template <typename T, typename Intermediate>
 void DownmixInterleavedToMonoImpl(const T* interleaved,
                                   size_t num_frames,
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index c0ac681ca9..e6e5d69d40 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -223,12 +223,6 @@ void AcmReceiver::FlushBuffers() {
   neteq_->FlushBuffers();
 }
 
-void AcmReceiver::RemoveAllCodecs() {
-  MutexLock lock(&mutex_);
-  neteq_->RemoveAllPayloadTypes();
-  last_decoder_ = absl::nullopt;
-}
-
 absl::optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() {
   return neteq_->GetPlayoutTimestamp();
 }
diff --git a/modules/audio_coding/acm2/acm_receiver.h b/modules/audio_coding/acm2/acm_receiver.h
index d631403873..8b57523cb7 100644
--- a/modules/audio_coding/acm2/acm_receiver.h
+++ b/modules/audio_coding/acm2/acm_receiver.h
@@ -168,11 +168,6 @@ class AcmReceiver {
   //
   void FlushBuffers();
 
-  //
-  // Remove all registered codecs.
-  //
-  void RemoveAllCodecs();
-
   // Returns the RTP timestamp for the last sample delivered by GetAudio().
   // The return value will be empty if no valid timestamp is available.
   absl::optional<uint32_t> GetPlayoutTimestamp();