diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index 2ae75485e0..b89091f362 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -245,27 +245,12 @@ void AcmReceiver::FlushBuffers() {
   neteq_->FlushBuffers();
 }
 
-// If failed in removing one of the codecs, this method continues to remove as
-// many as it can.
-int AcmReceiver::RemoveAllCodecs() {
-  int ret_val = 0;
+void AcmReceiver::RemoveAllCodecs() {
   rtc::CritScope lock(&crit_sect_);
-  for (auto it = decoders_.begin(); it != decoders_.end(); ) {
-    auto cur = it;
-    ++it;  // it will be valid even if we erase cur
-    if (neteq_->RemovePayloadType(cur->second.payload_type) == 0) {
-      decoders_.erase(cur);
-    } else {
-      LOG_F(LS_ERROR) << "Cannot remove payload "
-                      << static_cast<int>(cur->second.payload_type);
-      ret_val = -1;
-    }
-  }
-
-  // No codec is registered, invalidate last audio decoder.
+  neteq_->RemoveAllPayloadTypes();
+  decoders_.clear();
   last_audio_decoder_ = rtc::Optional<CodecInst>();
   last_packet_sample_rate_hz_ = rtc::Optional<int>();
-  return ret_val;
 }
 
 int AcmReceiver::RemoveCodec(uint8_t payload_type) {
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h
index 5b864c2b70..ee3f81da59 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -186,7 +186,7 @@ class AcmReceiver {
   //
   // Remove all registered codecs.
   //
-  int RemoveAllCodecs();
+  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.
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
index 4784a8721f..73f03a47ad 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
@@ -948,10 +948,8 @@ int AudioCodingModuleImpl::InitializeReceiverSafe() {
   // If the receiver is already initialized then we want to destroy any
   // existing decoders. After a call to this function, we should have a clean
   // start-up.
-  if (receiver_initialized_) {
-    if (receiver_.RemoveAllCodecs() < 0)
-      return -1;
-  }
+  if (receiver_initialized_)
+    receiver_.RemoveAllCodecs();
   receiver_.ResetInitialDelay();
   receiver_.SetMinimumDelay(0);
   receiver_.SetMaximumDelay(0);
diff --git a/webrtc/modules/audio_coding/neteq/decoder_database.cc b/webrtc/modules/audio_coding/neteq/decoder_database.cc
index b189e4bb56..f5fbad3446 100644
--- a/webrtc/modules/audio_coding/neteq/decoder_database.cc
+++ b/webrtc/modules/audio_coding/neteq/decoder_database.cc
@@ -169,6 +169,12 @@ int DecoderDatabase::Remove(uint8_t rtp_payload_type) {
   return kOK;
 }
 
+void DecoderDatabase::RemoveAll() {
+  decoders_.clear();
+  active_decoder_type_ = -1;      // No active decoder.
+  active_cng_decoder_type_ = -1;  // No active CNG decoder.
+}
+
 const DecoderDatabase::DecoderInfo* DecoderDatabase::GetDecoderInfo(
     uint8_t rtp_payload_type) const {
   DecoderMap::const_iterator it = decoders_.find(rtp_payload_type);
diff --git a/webrtc/modules/audio_coding/neteq/decoder_database.h b/webrtc/modules/audio_coding/neteq/decoder_database.h
index 6d277c95c9..38498f5143 100644
--- a/webrtc/modules/audio_coding/neteq/decoder_database.h
+++ b/webrtc/modules/audio_coding/neteq/decoder_database.h
@@ -136,6 +136,9 @@ class DecoderDatabase {
   // Returns kDecoderNotFound or kOK depending on the outcome of the operation.
   virtual int Remove(uint8_t rtp_payload_type);
 
+  // Remove all entries.
+  void RemoveAll();
+
   // Returns a pointer to the DecoderInfo struct for |rtp_payload_type|. If
   // no decoder is registered with that |rtp_payload_type|, NULL is returned.
   virtual const DecoderInfo* GetDecoderInfo(uint8_t rtp_payload_type) const;
diff --git a/webrtc/modules/audio_coding/neteq/include/neteq.h b/webrtc/modules/audio_coding/neteq/include/neteq.h
index 1f6205f1e4..eb08151a5d 100644
--- a/webrtc/modules/audio_coding/neteq/include/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/include/neteq.h
@@ -183,6 +183,9 @@ class NetEq {
   // -1 on failure.
   virtual int RemovePayloadType(uint8_t rtp_payload_type) = 0;
 
+  // Removes all payload types from the codec database.
+  virtual void RemoveAllPayloadTypes() = 0;
+
   // Sets a minimum delay in millisecond for packet buffer. The minimum is
   // maintained unless a higher latency is dictated by channel condition.
   // Returns true if the minimum is successfully applied, otherwise false is
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 71eea2114c..5e2116409d 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -291,6 +291,11 @@ int NetEqImpl::RemovePayloadType(uint8_t rtp_payload_type) {
   return kFail;
 }
 
+void NetEqImpl::RemoveAllPayloadTypes() {
+  rtc::CritScope lock(&crit_sect_);
+  decoder_database_->RemoveAll();
+}
+
 bool NetEqImpl::SetMinimumDelay(int delay_ms) {
   rtc::CritScope lock(&crit_sect_);
   if (delay_ms >= 0 && delay_ms < 10000) {
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
index 8795ef5210..7903ba665c 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -123,6 +123,8 @@ class NetEqImpl : public webrtc::NetEq {
   // -1 on failure.
   int RemovePayloadType(uint8_t rtp_payload_type) override;
 
+  void RemoveAllPayloadTypes() override;
+
   bool SetMinimumDelay(int delay_ms) override;
 
   bool SetMaximumDelay(int delay_ms) override;