diff --git a/webrtc/modules/audio_coding/neteq/decision_logic.cc b/webrtc/modules/audio_coding/neteq/decision_logic.cc
index 14e0426d7d..39bb4662c7 100644
--- a/webrtc/modules/audio_coding/neteq/decision_logic.cc
+++ b/webrtc/modules/audio_coding/neteq/decision_logic.cc
@@ -128,9 +128,6 @@ Operations DecisionLogic::GetDecision(const SyncBuffer& sync_buffer,
   const size_t cur_size_samples =
       samples_left + packet_buffer_.NumSamplesInBuffer(decoder_database_,
                                                        decoder_frame_length);
-  LOG(LS_VERBOSE) << "Buffers: " << packet_buffer_.NumPacketsInBuffer() <<
-      " packets * " << decoder_frame_length << " samples/packet + " <<
-      samples_left << " samples in sync buffer = " << cur_size_samples;
 
   prev_time_scale_ = prev_time_scale_ &&
       (prev_mode == kModeAccelerateSuccess ||
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index bbd3220c00..5fa2cbd586 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -124,11 +124,6 @@ int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header,
                             rtc::ArrayView<const uint8_t> payload,
                             uint32_t receive_timestamp) {
   CriticalSectionScoped lock(crit_sect_.get());
-  LOG(LS_VERBOSE) << "InsertPacket: ts=" << rtp_header.header.timestamp
-                  << ", sn=" << rtp_header.header.sequenceNumber
-                  << ", pt=" << static_cast<int>(rtp_header.header.payloadType)
-                  << ", ssrc=" << rtp_header.header.ssrc
-                  << ", len=" << payload.size();
   int error =
       InsertPacketInternal(rtp_header, payload, receive_timestamp, false);
   if (error != 0) {
@@ -141,12 +136,6 @@ int NetEqImpl::InsertPacket(const WebRtcRTPHeader& rtp_header,
 int NetEqImpl::InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
                                 uint32_t receive_timestamp) {
   CriticalSectionScoped lock(crit_sect_.get());
-  LOG(LS_VERBOSE) << "InsertPacket-Sync: ts="
-      << rtp_header.header.timestamp <<
-      ", sn=" << rtp_header.header.sequenceNumber <<
-      ", pt=" << static_cast<int>(rtp_header.header.payloadType) <<
-      ", ssrc=" << rtp_header.header.ssrc;
-
   const uint8_t kSyncPayload[] = { 's', 'y', 'n', 'c' };
   int error =
       InsertPacketInternal(rtp_header, kSyncPayload, receive_timestamp, true);
@@ -162,11 +151,8 @@ int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,
                         size_t* samples_per_channel, int* num_channels,
                         NetEqOutputType* type) {
   CriticalSectionScoped lock(crit_sect_.get());
-  LOG(LS_VERBOSE) << "GetAudio";
   int error = GetAudioInternal(max_length, output_audio, samples_per_channel,
                                num_channels);
-  LOG(LS_VERBOSE) << "Produced " << *samples_per_channel <<
-      " samples/channel for " << *num_channels << " channel(s)";
   if (error != 0) {
     error_code_ = error;
     return kFail;
@@ -751,8 +737,6 @@ int NetEqImpl::GetAudioInternal(size_t max_length,
     last_mode_ = kModeError;
     return return_value;
   }
-  LOG(LS_VERBOSE) << "GetDecision returned operation=" << operation <<
-      " and " << packet_list.size() << " packet(s)";
 
   AudioDecoder::SpeechType speech_type;
   int length = 0;
@@ -867,9 +851,6 @@ int NetEqImpl::GetAudioInternal(size_t max_length,
       sync_buffer_->GetNextAudioInterleaved(num_output_samples_per_channel,
                                             output);
   *num_channels = static_cast<int>(sync_buffer_->Channels());
-  LOG(LS_VERBOSE) << "Sync buffer (" << *num_channels << " channel(s)):" <<
-      " insert " << algorithm_buffer_->Size() << " samples, extract " <<
-      samples_from_sync << " samples";
   if (sync_buffer_->FutureLength() < expand_->overlap_length()) {
     // The sync buffer should always contain |overlap_length| samples, but now
     // too many samples have been extracted. Reinstall the |overlap_length|
@@ -1327,7 +1308,6 @@ int NetEqImpl::DecodeCng(AudioDecoder* decoder, int* decoded_length,
             &decoded_buffer_[*decoded_length], speech_type);
     if (length > 0) {
       *decoded_length += length;
-      LOG(LS_VERBOSE) << "Decoded " << length << " CNG samples";
     } else {
       // Error.
       LOG(LS_WARNING) << "Failed to decode CNG";
@@ -1367,34 +1347,17 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, const Operations& operation,
     int decode_length;
     if (packet->sync_packet) {
       // Decode to silence with the same frame size as the last decode.
-      LOG(LS_VERBOSE) << "Decoding sync-packet: " <<
-          " ts=" << packet->header.timestamp <<
-          ", sn=" << packet->header.sequenceNumber <<
-          ", pt=" << static_cast<int>(packet->header.payloadType) <<
-          ", ssrc=" << packet->header.ssrc <<
-          ", len=" << packet->payload_length;
       memset(&decoded_buffer_[*decoded_length], 0,
              decoder_frame_length_ * decoder->Channels() *
                  sizeof(decoded_buffer_[0]));
       decode_length = rtc::checked_cast<int>(decoder_frame_length_);
     } else if (!packet->primary) {
       // This is a redundant payload; call the special decoder method.
-      LOG(LS_VERBOSE) << "Decoding packet (redundant):" <<
-          " ts=" << packet->header.timestamp <<
-          ", sn=" << packet->header.sequenceNumber <<
-          ", pt=" << static_cast<int>(packet->header.payloadType) <<
-          ", ssrc=" << packet->header.ssrc <<
-          ", len=" << packet->payload_length;
       decode_length = decoder->DecodeRedundant(
           packet->payload, packet->payload_length, fs_hz_,
           (decoded_buffer_length_ - *decoded_length) * sizeof(int16_t),
           &decoded_buffer_[*decoded_length], speech_type);
     } else {
-      LOG(LS_VERBOSE) << "Decoding packet: ts=" << packet->header.timestamp <<
-          ", sn=" << packet->header.sequenceNumber <<
-          ", pt=" << static_cast<int>(packet->header.payloadType) <<
-          ", ssrc=" << packet->header.ssrc <<
-          ", len=" << packet->payload_length;
       decode_length =
           decoder->Decode(
               packet->payload, packet->payload_length, fs_hz_,
@@ -1410,9 +1373,6 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, const Operations& operation,
       // Update |decoder_frame_length_| with number of samples per channel.
       decoder_frame_length_ =
           static_cast<size_t>(decode_length) / decoder->Channels();
-      LOG(LS_VERBOSE) << "Decoded " << decode_length << " samples ("
-                      << decoder->Channels() << " channel(s) -> "
-                      << decoder_frame_length_ << " samples per channel)";
     } else if (decode_length < 0) {
       // Error.
       LOG(LS_WARNING) << "Decode " << decode_length << " " << payload_length;
diff --git a/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc b/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
index eb69ac7889..9f161e498e 100644
--- a/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
+++ b/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
@@ -87,8 +87,6 @@ uint32_t TimestampScaler::ToInternal(uint32_t external_timestamp,
     assert(denominator_ > 0);  // Should not be possible.
     external_ref_ = external_timestamp;
     internal_ref_ += (external_diff * numerator_) / denominator_;
-    LOG(LS_VERBOSE) << "Converting timestamp: " << external_timestamp <<
-        " -> " << internal_ref_;
     return internal_ref_;
   } else {
     // No scaling.