diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
index b89091f362..2ae75485e0 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -245,12 +245,27 @@ void AcmReceiver::FlushBuffers() {
   neteq_->FlushBuffers();
 }
 
-void AcmReceiver::RemoveAllCodecs() {
+// 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;
   rtc::CritScope lock(&crit_sect_);
-  neteq_->RemoveAllPayloadTypes();
-  decoders_.clear();
+  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.
   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 ee3f81da59..5b864c2b70 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.
   //
-  void RemoveAllCodecs();
+  int 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 73f03a47ad..4784a8721f 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
@@ -948,8 +948,10 @@ 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_)
-    receiver_.RemoveAllCodecs();
+  if (receiver_initialized_) {
+    if (receiver_.RemoveAllCodecs() < 0)
+      return -1;
+  }
   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 f5fbad3446..b189e4bb56 100644
--- a/webrtc/modules/audio_coding/neteq/decoder_database.cc
+++ b/webrtc/modules/audio_coding/neteq/decoder_database.cc
@@ -169,12 +169,6 @@ 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 38498f5143..6d277c95c9 100644
--- a/webrtc/modules/audio_coding/neteq/decoder_database.h
+++ b/webrtc/modules/audio_coding/neteq/decoder_database.h
@@ -136,9 +136,6 @@ 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 eb08151a5d..1f6205f1e4 100644
--- a/webrtc/modules/audio_coding/neteq/include/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/include/neteq.h
@@ -183,9 +183,6 @@ 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 5e2116409d..71eea2114c 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -291,11 +291,6 @@ 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 7903ba665c..8795ef5210 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -123,8 +123,6 @@ 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;