Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

12 files changed, 1619 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/delay_tool/parse_delay_file.m b/third_party/libwebrtc/modules/audio_coding/neteq/test/delay_tool/parse_delay_file.m
new file mode 100644
index 0000000000..031d8a39ee
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/delay_tool/parse_delay_file.m
@@ -0,0 +1,201 @@
+%
+%  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+%
+%  Use of this source code is governed by a BSD-style license
+%  that can be found in the LICENSE file in the root of the source
+%  tree. An additional intellectual property rights grant can be found
+%  in the file PATENTS.  All contributing project authors may
+%  be found in the AUTHORS file in the root of the source tree.
+%
+
+function outStruct = parse_delay_file(file)
+
+fid = fopen(file, 'rb');
+if fid == -1
+    error('Cannot open file %s', file);
+end
+
+textline = fgetl(fid);
+if ~strncmp(textline, '#!NetEQ_Delay_Logging', 21)
+    error('Wrong file format');
+end
+
+ver = sscanf(textline, '#!NetEQ_Delay_Logging%d.%d');
+if ~all(ver == [2; 0])
+    error('Wrong version of delay logging function')
+end
+
+
+start_pos = ftell(fid);
+fseek(fid, -12, 'eof');
+textline = fgetl(fid);
+if ~strncmp(textline, 'End of file', 21)
+    error('File ending is not correct. Seems like the simulation ended abnormally.');
+end
+
+fseek(fid,-12-4, 'eof');
+Npackets = fread(fid, 1, 'int32');
+fseek(fid, start_pos, 'bof');
+
+rtpts = zeros(Npackets, 1);
+seqno = zeros(Npackets, 1);
+pt = zeros(Npackets, 1);
+plen = zeros(Npackets, 1);
+recin_t = nan*ones(Npackets, 1);
+decode_t = nan*ones(Npackets, 1);
+playout_delay = zeros(Npackets, 1);
+optbuf = zeros(Npackets, 1);
+
+fs_ix = 1;
+clock = 0;
+ts_ix = 1;
+ended = 0;
+late_packets = 0;
+fs_now = 8000;
+last_decode_k = 0;
+tot_expand = 0;
+tot_accelerate = 0;
+tot_preemptive = 0;
+
+while not(ended)
+    signal = fread(fid, 1, '*int32');
+    
+    switch signal
+        case 3 % NETEQ_DELAY_LOGGING_SIGNAL_CLOCK
+            clock = fread(fid, 1, '*float32');
+            
+            % keep on reading batches of M until the signal is no longer "3"
+            % read int32 + float32 in one go
+            % this is to save execution time
+            temp = [3; 0];
+            M = 120;
+            while all(temp(1,:) == 3)
+                fp = ftell(fid);
+                temp = fread(fid, [2 M], '*int32');
+            end
+            
+            % back up to last clock event
+            fseek(fid, fp - ftell(fid) + ...
+                (find(temp(1,:) ~= 3, 1 ) - 2) * 2 * 4 + 4, 'cof');
+            % read the last clock value
+            clock = fread(fid, 1, '*float32');
+            
+        case 1 % NETEQ_DELAY_LOGGING_SIGNAL_RECIN
+            temp_ts = fread(fid, 1, 'uint32');
+            
+            if late_packets > 0
+                temp_ix = ts_ix - 1;
+                while (temp_ix >= 1) && (rtpts(temp_ix) ~= temp_ts)
+                    % TODO(hlundin): use matlab vector search instead?
+                    temp_ix = temp_ix - 1;
+                end
+                
+                if temp_ix >= 1
+                    % the ts was found in the vector
+                    late_packets = late_packets - 1;
+                else
+                    temp_ix = ts_ix;
+                    ts_ix = ts_ix + 1;
+                end
+            else
+                temp_ix = ts_ix;
+                ts_ix = ts_ix + 1;
+            end
+            
+            rtpts(temp_ix) = temp_ts;
+            seqno(temp_ix) = fread(fid, 1, 'uint16');
+            pt(temp_ix) = fread(fid, 1, 'int32');
+            plen(temp_ix) = fread(fid, 1, 'int16');
+            recin_t(temp_ix) = clock;
+            
+        case 2 % NETEQ_DELAY_LOGGING_SIGNAL_FLUSH
+            % do nothing
+            
+        case 4 % NETEQ_DELAY_LOGGING_SIGNAL_EOF
+            ended = 1;
+            
+        case 5 % NETEQ_DELAY_LOGGING_SIGNAL_DECODE
+            last_decode_ts = fread(fid, 1, 'uint32');
+            temp_delay = fread(fid, 1, 'uint16');
+            
+            k = find(rtpts(1:(ts_ix - 1))==last_decode_ts,1,'last');
+            if ~isempty(k)
+                decode_t(k) = clock;
+                playout_delay(k) = temp_delay + ...
+                    5 *  fs_now / 8000; % add overlap length
+                last_decode_k = k;
+            end
+            
+        case 6 % NETEQ_DELAY_LOGGING_SIGNAL_CHANGE_FS
+            fsvec(fs_ix) = fread(fid, 1, 'uint16');
+            fschange_ts(fs_ix) = last_decode_ts;
+            fs_now = fsvec(fs_ix);
+            fs_ix = fs_ix + 1;
+            
+        case 7 % NETEQ_DELAY_LOGGING_SIGNAL_MERGE_INFO
+            playout_delay(last_decode_k) = playout_delay(last_decode_k) ...
+                + fread(fid, 1, 'int32');
+            
+        case 8 % NETEQ_DELAY_LOGGING_SIGNAL_EXPAND_INFO
+            temp = fread(fid, 1, 'int32');
+            if last_decode_k ~= 0
+                tot_expand = tot_expand + temp / (fs_now / 1000);
+            end                
+            
+        case 9 % NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO
+            temp = fread(fid, 1, 'int32');
+            if last_decode_k ~= 0
+                tot_accelerate = tot_accelerate + temp / (fs_now / 1000);
+            end                
+
+        case 10 % NETEQ_DELAY_LOGGING_SIGNAL_PREEMPTIVE_INFO
+            temp = fread(fid, 1, 'int32');
+            if last_decode_k ~= 0
+                tot_preemptive = tot_preemptive + temp / (fs_now / 1000);
+            end                
+            
+        case 11 % NETEQ_DELAY_LOGGING_SIGNAL_OPTBUF
+            optbuf(last_decode_k) = fread(fid, 1, 'int32');
+            
+        case 12 % NETEQ_DELAY_LOGGING_SIGNAL_DECODE_ONE_DESC
+            last_decode_ts = fread(fid, 1, 'uint32');
+            k = ts_ix - 1;
+            
+            while (k >= 1) && (rtpts(k) ~= last_decode_ts)
+                % TODO(hlundin): use matlab vector search instead?
+                k = k - 1;
+            end
+            
+            if k < 1
+                % packet not received yet
+                k = ts_ix;
+                rtpts(ts_ix) = last_decode_ts;
+                late_packets = late_packets + 1;
+            end
+            
+            decode_t(k) = clock;
+            playout_delay(k) = fread(fid, 1, 'uint16') + ...
+                5 *  fs_now / 8000; % add overlap length
+            last_decode_k = k;
+             
+    end
+    
+end
+
+
+fclose(fid);
+
+outStruct = struct(...
+    'ts', rtpts, ...
+    'sn', seqno, ...
+    'pt', pt,...
+    'plen', plen,...
+    'arrival', recin_t,...
+    'decode', decode_t,...
+    'fs', fsvec(:),...
+    'fschange_ts', fschange_ts(:),...
+    'playout_delay', playout_delay,...
+    'tot_expand', tot_expand,...
+    'tot_accelerate', tot_accelerate,...
+    'tot_preemptive', tot_preemptive,...
+    'optbuf', optbuf);
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/delay_tool/plot_neteq_delay.m b/third_party/libwebrtc/modules/audio_coding/neteq/test/delay_tool/plot_neteq_delay.m
new file mode 100644
index 0000000000..86d533fbeb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/delay_tool/plot_neteq_delay.m
@@ -0,0 +1,197 @@
+%
+%  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+%
+%  Use of this source code is governed by a BSD-style license
+%  that can be found in the LICENSE file in the root of the source
+%  tree. An additional intellectual property rights grant can be found
+%  in the file PATENTS.  All contributing project authors may
+%  be found in the AUTHORS file in the root of the source tree.
+%
+
+function [delay_struct, delayvalues] = plot_neteq_delay(delayfile, varargin)
+
+% InfoStruct = plot_neteq_delay(delayfile)
+% InfoStruct = plot_neteq_delay(delayfile, 'skipdelay', skip_seconds)
+%
+% Henrik Lundin, 2006-11-17
+% Henrik Lundin, 2011-05-17
+%
+
+try
+    s = parse_delay_file(delayfile);
+catch
+    error(lasterr);
+end
+
+delayskip=0;
+noplot=0;
+arg_ptr=1;
+delaypoints=[];
+
+s.sn=unwrap_seqno(s.sn);
+
+while arg_ptr+1 <= nargin
+    switch lower(varargin{arg_ptr})
+    case {'skipdelay', 'delayskip'}
+        % skip a number of seconds in the beginning when calculating delays
+        delayskip = varargin{arg_ptr+1};
+        arg_ptr = arg_ptr + 2;
+    case 'noplot'
+        noplot=1;
+        arg_ptr = arg_ptr + 1;
+    case {'get_delay', 'getdelay'}
+        % return a vector of delay values for the points in the given vector
+        delaypoints = varargin{arg_ptr+1};
+        arg_ptr = arg_ptr + 2;
+    otherwise
+        warning('Unknown switch %s\n', varargin{arg_ptr});
+        arg_ptr = arg_ptr + 1;
+    end
+end
+
+% find lost frames that were covered by one-descriptor decoding
+one_desc_ix=find(isnan(s.arrival));
+for k=1:length(one_desc_ix)
+    ix=find(s.ts==max(s.ts(s.ts(one_desc_ix(k))>s.ts)));
+    s.sn(one_desc_ix(k))=s.sn(ix)+1;
+    s.pt(one_desc_ix(k))=s.pt(ix);
+    s.arrival(one_desc_ix(k))=s.arrival(ix)+s.decode(one_desc_ix(k))-s.decode(ix);
+end
+
+% remove duplicate received frames that were never decoded (RED codec)
+if length(unique(s.ts(isfinite(s.ts)))) < length(s.ts(isfinite(s.ts)))
+    ix=find(isfinite(s.decode));
+    s.sn=s.sn(ix);
+    s.ts=s.ts(ix);
+    s.arrival=s.arrival(ix);
+    s.playout_delay=s.playout_delay(ix);
+    s.pt=s.pt(ix);
+    s.optbuf=s.optbuf(ix);
+    plen=plen(ix);
+    s.decode=s.decode(ix);
+end
+
+% find non-unique sequence numbers
+[~,un_ix]=unique(s.sn);
+nonun_ix=setdiff(1:length(s.sn),un_ix);
+if ~isempty(nonun_ix)
+    warning('RTP sequence numbers are in error');
+end
+            
+% sort vectors
+[s.sn,sort_ix]=sort(s.sn);
+s.ts=s.ts(sort_ix);
+s.arrival=s.arrival(sort_ix);
+s.decode=s.decode(sort_ix);
+s.playout_delay=s.playout_delay(sort_ix);
+s.pt=s.pt(sort_ix);
+
+send_t=s.ts-s.ts(1);
+if length(s.fs)<1
+    warning('No info about sample rate found in file. Using default 8000.');
+    s.fs(1)=8000;
+    s.fschange_ts(1)=min(s.ts);
+elseif s.fschange_ts(1)>min(s.ts)
+    s.fschange_ts(1)=min(s.ts);
+end
+
+end_ix=length(send_t);
+for k=length(s.fs):-1:1
+    start_ix=find(s.ts==s.fschange_ts(k));
+    send_t(start_ix:end_ix)=send_t(start_ix:end_ix)/s.fs(k)*1000;
+    s.playout_delay(start_ix:end_ix)=s.playout_delay(start_ix:end_ix)/s.fs(k)*1000;
+    s.optbuf(start_ix:end_ix)=s.optbuf(start_ix:end_ix)/s.fs(k)*1000;
+    end_ix=start_ix-1;
+end
+
+tot_time=max(send_t)-min(send_t);
+
+seq_ix=s.sn-min(s.sn)+1;
+send_t=send_t+max(min(s.arrival-send_t),0);
+
+plot_send_t=nan*ones(max(seq_ix),1);
+plot_send_t(seq_ix)=send_t;
+plot_nw_delay=nan*ones(max(seq_ix),1);
+plot_nw_delay(seq_ix)=s.arrival-send_t;
+
+cng_ix=find(s.pt~=13); % find those packets that are not CNG/SID
+    
+if noplot==0
+    h=plot(plot_send_t/1000,plot_nw_delay);
+    set(h,'color',0.75*[1 1 1]);
+    hold on
+    if any(s.optbuf~=0)
+        peak_ix=find(s.optbuf(cng_ix)<0); % peak mode is labeled with negative values
+        no_peak_ix=find(s.optbuf(cng_ix)>0); %setdiff(1:length(cng_ix),peak_ix);
+        h1=plot(send_t(cng_ix(peak_ix))/1000,...
+            s.arrival(cng_ix(peak_ix))+abs(s.optbuf(cng_ix(peak_ix)))-send_t(cng_ix(peak_ix)),...
+            'r.');
+        h2=plot(send_t(cng_ix(no_peak_ix))/1000,...
+            s.arrival(cng_ix(no_peak_ix))+abs(s.optbuf(cng_ix(no_peak_ix)))-send_t(cng_ix(no_peak_ix)),...
+            'g.');
+        set([h1, h2],'markersize',1)
+    end
+    %h=plot(send_t(seq_ix)/1000,s.decode+s.playout_delay-send_t(seq_ix));
+    h=plot(send_t(cng_ix)/1000,s.decode(cng_ix)+s.playout_delay(cng_ix)-send_t(cng_ix));
+    set(h,'linew',1.5);
+    hold off
+    ax1=axis;
+    axis tight
+    ax2=axis;
+    axis([ax2(1:3) ax1(4)])
+end
+
+
+% calculate delays and other parameters
+
+delayskip_ix = find(send_t-send_t(1)>=delayskip*1000, 1 );
+
+use_ix = intersect(cng_ix,... % use those that are not CNG/SID frames...
+    intersect(find(isfinite(s.decode)),... % ... that did arrive ...
+    (delayskip_ix:length(s.decode))')); % ... and are sent after delayskip seconds
+
+mean_delay = mean(s.decode(use_ix)+s.playout_delay(use_ix)-send_t(use_ix));
+neteq_delay = mean(s.decode(use_ix)+s.playout_delay(use_ix)-s.arrival(use_ix));
+
+Npack=max(s.sn(delayskip_ix:end))-min(s.sn(delayskip_ix:end))+1;
+nw_lossrate=(Npack-length(s.sn(delayskip_ix:end)))/Npack;
+neteq_lossrate=(length(s.sn(delayskip_ix:end))-length(use_ix))/Npack;
+
+delay_struct=struct('mean_delay',mean_delay,'neteq_delay',neteq_delay,...
+    'nw_lossrate',nw_lossrate,'neteq_lossrate',neteq_lossrate,...
+    'tot_expand',round(s.tot_expand),'tot_accelerate',round(s.tot_accelerate),...
+    'tot_preemptive',round(s.tot_preemptive),'tot_time',tot_time,...
+    'filename',delayfile,'units','ms','fs',unique(s.fs));
+    
+if not(isempty(delaypoints))
+    delayvalues=interp1(send_t(cng_ix),...
+        s.decode(cng_ix)+s.playout_delay(cng_ix)-send_t(cng_ix),...
+        delaypoints,'nearest',NaN);
+else
+    delayvalues=[];
+end
+
+
+
+% SUBFUNCTIONS %
+
+function y=unwrap_seqno(x)
+
+jumps=find(abs((diff(x)-1))>65000);
+
+while ~isempty(jumps)
+    n=jumps(1);
+    if x(n+1)-x(n) < 0
+        % negative jump
+        x(n+1:end)=x(n+1:end)+65536;
+    else
+        % positive jump
+        x(n+1:end)=x(n+1:end)-65536;
+    end
+    
+    jumps=find(abs((diff(x(n+1:end))-1))>65000);
+end
+
+y=x;
+
+return;
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.cc
new file mode 100644
index 0000000000..e6c1809fb6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.cc
@@ -0,0 +1,423 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/test/neteq_decoding_test.h"
+
+#include "absl/strings/string_view.h"
+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "api/rtp_headers.h"
+#include "modules/audio_coding/neteq/default_neteq_factory.h"
+#include "modules/audio_coding/neteq/test/result_sink.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/testsupport/file_utils.h"
+
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
+#else
+#include "modules/audio_coding/neteq/neteq_unittest.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+#endif
+
+namespace webrtc {
+
+namespace {
+
+void LoadDecoders(webrtc::NetEq* neteq) {
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(0, SdpAudioFormat("pcmu", 8000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(8, SdpAudioFormat("pcma", 8000, 1)));
+#ifdef WEBRTC_CODEC_ILBC
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(102, SdpAudioFormat("ilbc", 8000, 1)));
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(103, SdpAudioFormat("isac", 16000, 1)));
+#endif
+#ifdef WEBRTC_CODEC_ISAC
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(104, SdpAudioFormat("isac", 32000, 1)));
+#endif
+#ifdef WEBRTC_CODEC_OPUS
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(
+                111, SdpAudioFormat("opus", 48000, 2, {{"stereo", "0"}})));
+#endif
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(93, SdpAudioFormat("L16", 8000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(94, SdpAudioFormat("L16", 16000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(95, SdpAudioFormat("L16", 32000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(13, SdpAudioFormat("cn", 8000, 1)));
+  ASSERT_EQ(true,
+            neteq->RegisterPayloadType(98, SdpAudioFormat("cn", 16000, 1)));
+}
+
+}  // namespace
+
+const int NetEqDecodingTest::kTimeStepMs;
+const size_t NetEqDecodingTest::kBlockSize8kHz;
+const size_t NetEqDecodingTest::kBlockSize16kHz;
+const size_t NetEqDecodingTest::kBlockSize32kHz;
+const int NetEqDecodingTest::kInitSampleRateHz;
+
+NetEqDecodingTest::NetEqDecodingTest()
+    : clock_(0),
+      config_(),
+      output_sample_rate_(kInitSampleRateHz),
+      algorithmic_delay_ms_(0) {
+  config_.sample_rate_hz = kInitSampleRateHz;
+}
+
+void NetEqDecodingTest::SetUp() {
+  auto decoder_factory = CreateBuiltinAudioDecoderFactory();
+  neteq_ = DefaultNetEqFactory().CreateNetEq(config_, decoder_factory, &clock_);
+  NetEqNetworkStatistics stat;
+  ASSERT_EQ(0, neteq_->NetworkStatistics(&stat));
+  algorithmic_delay_ms_ = stat.current_buffer_size_ms;
+  ASSERT_TRUE(neteq_);
+  LoadDecoders(neteq_.get());
+}
+
+void NetEqDecodingTest::TearDown() {}
+
+void NetEqDecodingTest::OpenInputFile(absl::string_view rtp_file) {
+  rtp_source_.reset(test::RtpFileSource::Create(rtp_file));
+}
+
+void NetEqDecodingTest::Process() {
+  // Check if time to receive.
+  while (packet_ && clock_.TimeInMilliseconds() >= packet_->time_ms()) {
+    if (packet_->payload_length_bytes() > 0) {
+#ifndef WEBRTC_CODEC_ISAC
+      // Ignore payload type 104 (iSAC-swb) if ISAC is not supported.
+      if (packet_->header().payloadType != 104)
+#endif
+        ASSERT_EQ(
+            0, neteq_->InsertPacket(
+                   packet_->header(),
+                   rtc::ArrayView<const uint8_t>(
+                       packet_->payload(), packet_->payload_length_bytes())));
+    }
+    // Get next packet.
+    packet_ = rtp_source_->NextPacket();
+  }
+
+  // Get audio from NetEq.
+  bool muted;
+  ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+  ASSERT_FALSE(muted);
+  ASSERT_TRUE((out_frame_.samples_per_channel_ == kBlockSize8kHz) ||
+              (out_frame_.samples_per_channel_ == kBlockSize16kHz) ||
+              (out_frame_.samples_per_channel_ == kBlockSize32kHz) ||
+              (out_frame_.samples_per_channel_ == kBlockSize48kHz));
+  output_sample_rate_ = out_frame_.sample_rate_hz_;
+  EXPECT_EQ(output_sample_rate_, neteq_->last_output_sample_rate_hz());
+
+  // Increase time.
+  clock_.AdvanceTimeMilliseconds(kTimeStepMs);
+}
+
+void NetEqDecodingTest::DecodeAndCompare(
+    absl::string_view rtp_file,
+    absl::string_view output_checksum,
+    absl::string_view network_stats_checksum,
+    bool gen_ref) {
+  OpenInputFile(rtp_file);
+
+  std::string ref_out_file =
+      gen_ref ? webrtc::test::OutputPath() + "neteq_universal_ref.pcm" : "";
+  ResultSink output(ref_out_file);
+
+  std::string stat_out_file =
+      gen_ref ? webrtc::test::OutputPath() + "neteq_network_stats.dat" : "";
+  ResultSink network_stats(stat_out_file);
+
+  packet_ = rtp_source_->NextPacket();
+  int i = 0;
+  uint64_t last_concealed_samples = 0;
+  uint64_t last_total_samples_received = 0;
+  while (packet_) {
+    rtc::StringBuilder ss;
+    ss << "Lap number " << i++ << " in DecodeAndCompare while loop";
+    SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
+    ASSERT_NO_FATAL_FAILURE(Process());
+    ASSERT_NO_FATAL_FAILURE(
+        output.AddResult(out_frame_.data(), out_frame_.samples_per_channel_));
+
+    // Query the network statistics API once per second
+    if (clock_.TimeInMilliseconds() % 1000 == 0) {
+      // Process NetworkStatistics.
+      NetEqNetworkStatistics current_network_stats;
+      ASSERT_EQ(0, neteq_->NetworkStatistics(&current_network_stats));
+      ASSERT_NO_FATAL_FAILURE(network_stats.AddResult(current_network_stats));
+
+      // Verify that liftime stats and network stats report similar loss
+      // concealment rates.
+      auto lifetime_stats = neteq_->GetLifetimeStatistics();
+      const uint64_t delta_concealed_samples =
+          lifetime_stats.concealed_samples - last_concealed_samples;
+      last_concealed_samples = lifetime_stats.concealed_samples;
+      const uint64_t delta_total_samples_received =
+          lifetime_stats.total_samples_received - last_total_samples_received;
+      last_total_samples_received = lifetime_stats.total_samples_received;
+      // The tolerance is 1% but expressed in Q14.
+      EXPECT_NEAR(
+          (delta_concealed_samples << 14) / delta_total_samples_received,
+          current_network_stats.expand_rate, (2 << 14) / 100.0);
+    }
+  }
+
+  SCOPED_TRACE("Check output audio.");
+  output.VerifyChecksum(output_checksum);
+  SCOPED_TRACE("Check network stats.");
+  network_stats.VerifyChecksum(network_stats_checksum);
+}
+
+void NetEqDecodingTest::PopulateRtpInfo(int frame_index,
+                                        int timestamp,
+                                        RTPHeader* rtp_info) {
+  rtp_info->sequenceNumber = frame_index;
+  rtp_info->timestamp = timestamp;
+  rtp_info->ssrc = 0x1234;     // Just an arbitrary SSRC.
+  rtp_info->payloadType = 94;  // PCM16b WB codec.
+  rtp_info->markerBit = false;
+}
+
+void NetEqDecodingTest::PopulateCng(int frame_index,
+                                    int timestamp,
+                                    RTPHeader* rtp_info,
+                                    uint8_t* payload,
+                                    size_t* payload_len) {
+  rtp_info->sequenceNumber = frame_index;
+  rtp_info->timestamp = timestamp;
+  rtp_info->ssrc = 0x1234;     // Just an arbitrary SSRC.
+  rtp_info->payloadType = 98;  // WB CNG.
+  rtp_info->markerBit = false;
+  payload[0] = 64;   // Noise level -64 dBov, quite arbitrarily chosen.
+  *payload_len = 1;  // Only noise level, no spectral parameters.
+}
+
+void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
+                                 uint32_t start_timestamp,
+                                 const std::set<uint16_t>& drop_seq_numbers,
+                                 bool expect_seq_no_wrap,
+                                 bool expect_timestamp_wrap) {
+  uint16_t seq_no = start_seq_no;
+  uint32_t timestamp = start_timestamp;
+  const int kBlocksPerFrame = 3;  // Number of 10 ms blocks per frame.
+  const int kFrameSizeMs = kBlocksPerFrame * kTimeStepMs;
+  const int kSamples = kBlockSize16kHz * kBlocksPerFrame;
+  const size_t kPayloadBytes = kSamples * sizeof(int16_t);
+  double next_input_time_ms = 0.0;
+
+  // Insert speech for 2 seconds.
+  const int kSpeechDurationMs = 2000;
+  uint16_t last_seq_no;
+  uint32_t last_timestamp;
+  bool timestamp_wrapped = false;
+  bool seq_no_wrapped = false;
+  for (double t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) {
+    // Each turn in this for loop is 10 ms.
+    while (next_input_time_ms <= t_ms) {
+      // Insert one 30 ms speech frame.
+      uint8_t payload[kPayloadBytes] = {0};
+      RTPHeader rtp_info;
+      PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+      if (drop_seq_numbers.find(seq_no) == drop_seq_numbers.end()) {
+        // This sequence number was not in the set to drop. Insert it.
+        ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+      }
+      NetEqNetworkStatistics network_stats;
+      ASSERT_EQ(0, neteq_->NetworkStatistics(&network_stats));
+
+      EXPECT_LE(network_stats.preferred_buffer_size_ms, 80);
+      EXPECT_LE(network_stats.current_buffer_size_ms,
+                80 + algorithmic_delay_ms_);
+      last_seq_no = seq_no;
+      last_timestamp = timestamp;
+
+      ++seq_no;
+      timestamp += kSamples;
+      next_input_time_ms += static_cast<double>(kFrameSizeMs);
+
+      seq_no_wrapped |= seq_no < last_seq_no;
+      timestamp_wrapped |= timestamp < last_timestamp;
+    }
+    // Pull out data once.
+    AudioFrame output;
+    bool muted;
+    ASSERT_EQ(0, neteq_->GetAudio(&output, &muted));
+    ASSERT_EQ(kBlockSize16kHz, output.samples_per_channel_);
+    ASSERT_EQ(1u, output.num_channels_);
+
+    // Expect delay (in samples) to be less than 2 packets.
+    absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
+    ASSERT_TRUE(playout_timestamp);
+    EXPECT_LE(timestamp - *playout_timestamp,
+              static_cast<uint32_t>(kSamples * 2));
+  }
+  // Make sure we have actually tested wrap-around.
+  ASSERT_EQ(expect_seq_no_wrap, seq_no_wrapped);
+  ASSERT_EQ(expect_timestamp_wrap, timestamp_wrapped);
+}
+
+void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
+                                              double network_freeze_ms,
+                                              bool pull_audio_during_freeze,
+                                              int delay_tolerance_ms,
+                                              int max_time_to_speech_ms) {
+  uint16_t seq_no = 0;
+  uint32_t timestamp = 0;
+  const int kFrameSizeMs = 30;
+  const size_t kSamples = kFrameSizeMs * 16;
+  const size_t kPayloadBytes = kSamples * 2;
+  double next_input_time_ms = 0.0;
+  double t_ms;
+  bool muted;
+
+  // Insert speech for 5 seconds.
+  const int kSpeechDurationMs = 5000;
+  for (t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) {
+    // Each turn in this for loop is 10 ms.
+    while (next_input_time_ms <= t_ms) {
+      // Insert one 30 ms speech frame.
+      uint8_t payload[kPayloadBytes] = {0};
+      RTPHeader rtp_info;
+      PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+      ++seq_no;
+      timestamp += kSamples;
+      next_input_time_ms += static_cast<double>(kFrameSizeMs) * drift_factor;
+    }
+    // Pull out data once.
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  }
+
+  EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
+  absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
+  ASSERT_TRUE(playout_timestamp);
+  int32_t delay_before = timestamp - *playout_timestamp;
+
+  // Insert CNG for 1 minute (= 60000 ms).
+  const int kCngPeriodMs = 100;
+  const int kCngPeriodSamples = kCngPeriodMs * 16;  // Period in 16 kHz samples.
+  const int kCngDurationMs = 60000;
+  for (; t_ms < kSpeechDurationMs + kCngDurationMs; t_ms += 10) {
+    // Each turn in this for loop is 10 ms.
+    while (next_input_time_ms <= t_ms) {
+      // Insert one CNG frame each 100 ms.
+      uint8_t payload[kPayloadBytes];
+      size_t payload_len;
+      RTPHeader rtp_info;
+      PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+                                                      payload, payload_len)));
+      ++seq_no;
+      timestamp += kCngPeriodSamples;
+      next_input_time_ms += static_cast<double>(kCngPeriodMs) * drift_factor;
+    }
+    // Pull out data once.
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+  }
+
+  EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+
+  if (network_freeze_ms > 0) {
+    // First keep pulling audio for `network_freeze_ms` without inserting
+    // any data, then insert CNG data corresponding to `network_freeze_ms`
+    // without pulling any output audio.
+    const double loop_end_time = t_ms + network_freeze_ms;
+    for (; t_ms < loop_end_time; t_ms += 10) {
+      // Pull out data once.
+      ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+      ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+      EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+    }
+    bool pull_once = pull_audio_during_freeze;
+    // If `pull_once` is true, GetAudio will be called once half-way through
+    // the network recovery period.
+    double pull_time_ms = (t_ms + next_input_time_ms) / 2;
+    while (next_input_time_ms <= t_ms) {
+      if (pull_once && next_input_time_ms >= pull_time_ms) {
+        pull_once = false;
+        // Pull out data once.
+        ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+        ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+        EXPECT_EQ(AudioFrame::kCNG, out_frame_.speech_type_);
+        t_ms += 10;
+      }
+      // Insert one CNG frame each 100 ms.
+      uint8_t payload[kPayloadBytes];
+      size_t payload_len;
+      RTPHeader rtp_info;
+      PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, rtc::ArrayView<const uint8_t>(
+                                                      payload, payload_len)));
+      ++seq_no;
+      timestamp += kCngPeriodSamples;
+      next_input_time_ms += kCngPeriodMs * drift_factor;
+    }
+  }
+
+  // Insert speech again until output type is speech.
+  double speech_restart_time_ms = t_ms;
+  while (out_frame_.speech_type_ != AudioFrame::kNormalSpeech) {
+    // Each turn in this for loop is 10 ms.
+    while (next_input_time_ms <= t_ms) {
+      // Insert one 30 ms speech frame.
+      uint8_t payload[kPayloadBytes] = {0};
+      RTPHeader rtp_info;
+      PopulateRtpInfo(seq_no, timestamp, &rtp_info);
+      ASSERT_EQ(0, neteq_->InsertPacket(rtp_info, payload));
+      ++seq_no;
+      timestamp += kSamples;
+      next_input_time_ms += kFrameSizeMs * drift_factor;
+    }
+    // Pull out data once.
+    ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
+    ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
+    // Increase clock.
+    t_ms += 10;
+  }
+
+  // Check that the speech starts again within reasonable time.
+  double time_until_speech_returns_ms = t_ms - speech_restart_time_ms;
+  EXPECT_LT(time_until_speech_returns_ms, max_time_to_speech_ms);
+  playout_timestamp = neteq_->GetPlayoutTimestamp();
+  ASSERT_TRUE(playout_timestamp);
+  int32_t delay_after = timestamp - *playout_timestamp;
+  // Compare delay before and after, and make sure it differs less than 20 ms.
+  EXPECT_LE(delay_after, delay_before + delay_tolerance_ms * 16);
+  EXPECT_GE(delay_after, delay_before - delay_tolerance_ms * 16);
+}
+
+void NetEqDecodingTestTwoInstances::SetUp() {
+  NetEqDecodingTest::SetUp();
+  config2_ = config_;
+}
+
+void NetEqDecodingTestTwoInstances::CreateSecondInstance() {
+  auto decoder_factory = CreateBuiltinAudioDecoderFactory();
+  neteq2_ =
+      DefaultNetEqFactory().CreateNetEq(config2_, decoder_factory, &clock_);
+  ASSERT_TRUE(neteq2_);
+  LoadDecoders(neteq2_.get());
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.h b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.h
new file mode 100644
index 0000000000..456c397fdd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.h
@@ -0,0 +1,96 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_
+#define MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_
+
+#include <memory>
+#include <set>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/audio/audio_frame.h"
+#include "api/neteq/neteq.h"
+#include "api/rtp_headers.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class NetEqDecodingTest : public ::testing::Test {
+ protected:
+  // NetEQ must be polled for data once every 10 ms.
+  // Thus, none of the constants below can be changed.
+  static constexpr int kTimeStepMs = 10;
+  static constexpr size_t kBlockSize8kHz = kTimeStepMs * 8;
+  static constexpr size_t kBlockSize16kHz = kTimeStepMs * 16;
+  static constexpr size_t kBlockSize32kHz = kTimeStepMs * 32;
+  static constexpr size_t kBlockSize48kHz = kTimeStepMs * 48;
+  static constexpr int kInitSampleRateHz = 8000;
+
+  NetEqDecodingTest();
+  virtual void SetUp();
+  virtual void TearDown();
+  void OpenInputFile(absl::string_view rtp_file);
+  void Process();
+
+  void DecodeAndCompare(absl::string_view rtp_file,
+                        absl::string_view output_checksum,
+                        absl::string_view network_stats_checksum,
+                        bool gen_ref);
+
+  static void PopulateRtpInfo(int frame_index,
+                              int timestamp,
+                              RTPHeader* rtp_info);
+  static void PopulateCng(int frame_index,
+                          int timestamp,
+                          RTPHeader* rtp_info,
+                          uint8_t* payload,
+                          size_t* payload_len);
+
+  void WrapTest(uint16_t start_seq_no,
+                uint32_t start_timestamp,
+                const std::set<uint16_t>& drop_seq_numbers,
+                bool expect_seq_no_wrap,
+                bool expect_timestamp_wrap);
+
+  void LongCngWithClockDrift(double drift_factor,
+                             double network_freeze_ms,
+                             bool pull_audio_during_freeze,
+                             int delay_tolerance_ms,
+                             int max_time_to_speech_ms);
+
+  SimulatedClock clock_;
+  std::unique_ptr<NetEq> neteq_;
+  NetEq::Config config_;
+  std::unique_ptr<test::RtpFileSource> rtp_source_;
+  std::unique_ptr<test::Packet> packet_;
+  AudioFrame out_frame_;
+  int output_sample_rate_;
+  int algorithmic_delay_ms_;
+};
+
+class NetEqDecodingTestTwoInstances : public NetEqDecodingTest {
+ public:
+  NetEqDecodingTestTwoInstances() : NetEqDecodingTest() {}
+
+  void SetUp() override;
+
+  void CreateSecondInstance();
+
+ protected:
+  std::unique_ptr<NetEq> neteq2_;
+  NetEq::Config config2_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
new file mode 100644
index 0000000000..1004141f16
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "absl/flags/flag.h"
+#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/testsupport/file_utils.h"
+
+ABSL_FLAG(int, frame_size_ms, 20, "Codec frame size (milliseconds).");
+
+using ::testing::InitGoogleTest;
+
+namespace webrtc {
+namespace test {
+namespace {
+static const int kInputSampleRateKhz = 8;
+static const int kOutputSampleRateKhz = 8;
+}  // namespace
+
+class NetEqIlbcQualityTest : public NetEqQualityTest {
+ protected:
+  NetEqIlbcQualityTest()
+      : NetEqQualityTest(absl::GetFlag(FLAGS_frame_size_ms),
+                         kInputSampleRateKhz,
+                         kOutputSampleRateKhz,
+                         SdpAudioFormat("ilbc", 8000, 1)) {
+    // Flag validation
+    RTC_CHECK(absl::GetFlag(FLAGS_frame_size_ms) == 20 ||
+              absl::GetFlag(FLAGS_frame_size_ms) == 30 ||
+              absl::GetFlag(FLAGS_frame_size_ms) == 40 ||
+              absl::GetFlag(FLAGS_frame_size_ms) == 60)
+        << "Invalid frame size, should be 20, 30, 40, or 60 ms.";
+  }
+
+  void SetUp() override {
+    ASSERT_EQ(1u, channels_) << "iLBC supports only mono audio.";
+    AudioEncoderIlbcConfig config;
+    config.frame_size_ms = absl::GetFlag(FLAGS_frame_size_ms);
+    encoder_.reset(new AudioEncoderIlbcImpl(config, 102));
+    NetEqQualityTest::SetUp();
+  }
+
+  int EncodeBlock(int16_t* in_data,
+                  size_t block_size_samples,
+                  rtc::Buffer* payload,
+                  size_t max_bytes) override {
+    const size_t kFrameSizeSamples = 80;  // Samples per 10 ms.
+    size_t encoded_samples = 0;
+    uint32_t dummy_timestamp = 0;
+    AudioEncoder::EncodedInfo info;
+    do {
+      info = encoder_->Encode(dummy_timestamp,
+                              rtc::ArrayView<const int16_t>(
+                                  in_data + encoded_samples, kFrameSizeSamples),
+                              payload);
+      encoded_samples += kFrameSizeSamples;
+    } while (info.encoded_bytes == 0);
+    return rtc::checked_cast<int>(info.encoded_bytes);
+  }
+
+ private:
+  std::unique_ptr<AudioEncoderIlbcImpl> encoder_;
+};
+
+TEST_F(NetEqIlbcQualityTest, Test) {
+  Simulate();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
new file mode 100644
index 0000000000..5a2df24ef6
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
@@ -0,0 +1,183 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "absl/flags/flag.h"
+#include "modules/audio_coding/codecs/opus/opus_inst.h"
+#include "modules/audio_coding/codecs/opus/opus_interface.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+
+ABSL_FLAG(int, bit_rate_kbps, 32, "Target bit rate (kbps).");
+
+ABSL_FLAG(int,
+          complexity,
+          10,
+          "Complexity: 0 ~ 10 -- defined as in Opus"
+          "specification.");
+
+ABSL_FLAG(int, maxplaybackrate, 48000, "Maximum playback rate (Hz).");
+
+ABSL_FLAG(int, application, 0, "Application mode: 0 -- VOIP, 1 -- Audio.");
+
+ABSL_FLAG(int, reported_loss_rate, 10, "Reported percentile of packet loss.");
+
+ABSL_FLAG(bool, fec, false, "Enable FEC for encoding (-nofec to disable).");
+
+ABSL_FLAG(bool, dtx, false, "Enable DTX for encoding (-nodtx to disable).");
+
+ABSL_FLAG(int, sub_packets, 1, "Number of sub packets to repacketize.");
+
+using ::testing::InitGoogleTest;
+
+namespace webrtc {
+namespace test {
+namespace {
+
+static const int kOpusBlockDurationMs = 20;
+static const int kOpusSamplingKhz = 48;
+}  // namespace
+
+class NetEqOpusQualityTest : public NetEqQualityTest {
+ protected:
+  NetEqOpusQualityTest();
+  void SetUp() override;
+  void TearDown() override;
+  int EncodeBlock(int16_t* in_data,
+                  size_t block_size_samples,
+                  rtc::Buffer* payload,
+                  size_t max_bytes) override;
+
+ private:
+  WebRtcOpusEncInst* opus_encoder_;
+  OpusRepacketizer* repacketizer_;
+  size_t sub_block_size_samples_;
+  int bit_rate_kbps_;
+  bool fec_;
+  bool dtx_;
+  int complexity_;
+  int maxplaybackrate_;
+  int target_loss_rate_;
+  int sub_packets_;
+  int application_;
+};
+
+NetEqOpusQualityTest::NetEqOpusQualityTest()
+    : NetEqQualityTest(kOpusBlockDurationMs * absl::GetFlag(FLAGS_sub_packets),
+                       kOpusSamplingKhz,
+                       kOpusSamplingKhz,
+                       SdpAudioFormat("opus", 48000, 2)),
+      opus_encoder_(NULL),
+      repacketizer_(NULL),
+      sub_block_size_samples_(
+          static_cast<size_t>(kOpusBlockDurationMs * kOpusSamplingKhz)),
+      bit_rate_kbps_(absl::GetFlag(FLAGS_bit_rate_kbps)),
+      fec_(absl::GetFlag(FLAGS_fec)),
+      dtx_(absl::GetFlag(FLAGS_dtx)),
+      complexity_(absl::GetFlag(FLAGS_complexity)),
+      maxplaybackrate_(absl::GetFlag(FLAGS_maxplaybackrate)),
+      target_loss_rate_(absl::GetFlag(FLAGS_reported_loss_rate)),
+      sub_packets_(absl::GetFlag(FLAGS_sub_packets)) {
+  // Flag validation
+  RTC_CHECK(absl::GetFlag(FLAGS_bit_rate_kbps) >= 6 &&
+            absl::GetFlag(FLAGS_bit_rate_kbps) <= 510)
+      << "Invalid bit rate, should be between 6 and 510 kbps.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_complexity) >= -1 &&
+            absl::GetFlag(FLAGS_complexity) <= 10)
+      << "Invalid complexity setting, should be between 0 and 10.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_application) == 0 ||
+            absl::GetFlag(FLAGS_application) == 1)
+      << "Invalid application mode, should be 0 or 1.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_reported_loss_rate) >= 0 &&
+            absl::GetFlag(FLAGS_reported_loss_rate) <= 100)
+      << "Invalid packet loss percentile, should be between 0 and 100.";
+
+  RTC_CHECK(absl::GetFlag(FLAGS_sub_packets) >= 1 &&
+            absl::GetFlag(FLAGS_sub_packets) <= 3)
+      << "Invalid number of sub packets, should be between 1 and 3.";
+
+  // Redefine decoder type if input is stereo.
+  if (channels_ > 1) {
+    audio_format_ = SdpAudioFormat("opus", 48000, 2,
+                                   SdpAudioFormat::Parameters{{"stereo", "1"}});
+  }
+  application_ = absl::GetFlag(FLAGS_application);
+}
+
+void NetEqOpusQualityTest::SetUp() {
+  // Create encoder memory.
+  WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_, 48000);
+  ASSERT_TRUE(opus_encoder_);
+
+  // Create repacketizer.
+  repacketizer_ = opus_repacketizer_create();
+  ASSERT_TRUE(repacketizer_);
+
+  // Set bitrate.
+  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, bit_rate_kbps_ * 1000));
+  if (fec_) {
+    EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
+  }
+  if (dtx_) {
+    EXPECT_EQ(0, WebRtcOpus_EnableDtx(opus_encoder_));
+  }
+  EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity_));
+  EXPECT_EQ(0, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, maxplaybackrate_));
+  EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_, target_loss_rate_));
+  NetEqQualityTest::SetUp();
+}
+
+void NetEqOpusQualityTest::TearDown() {
+  // Free memory.
+  EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
+  opus_repacketizer_destroy(repacketizer_);
+  NetEqQualityTest::TearDown();
+}
+
+int NetEqOpusQualityTest::EncodeBlock(int16_t* in_data,
+                                      size_t block_size_samples,
+                                      rtc::Buffer* payload,
+                                      size_t max_bytes) {
+  EXPECT_EQ(block_size_samples, sub_block_size_samples_ * sub_packets_);
+  int16_t* pointer = in_data;
+  int value;
+  opus_repacketizer_init(repacketizer_);
+  for (int idx = 0; idx < sub_packets_; idx++) {
+    payload->AppendData(max_bytes, [&](rtc::ArrayView<uint8_t> payload) {
+      value = WebRtcOpus_Encode(opus_encoder_, pointer, sub_block_size_samples_,
+                                max_bytes, payload.data());
+
+      Log() << "Encoded a frame with Opus mode "
+            << (value == 0 ? 0 : payload[0] >> 3) << std::endl;
+
+      return (value >= 0) ? static_cast<size_t>(value) : 0;
+    });
+
+    if (OPUS_OK !=
+        opus_repacketizer_cat(repacketizer_, payload->data(), value)) {
+      opus_repacketizer_init(repacketizer_);
+      // If the repacketization fails, we discard this frame.
+      return 0;
+    }
+    pointer += sub_block_size_samples_ * channels_;
+  }
+  value = opus_repacketizer_out(repacketizer_, payload->data(),
+                                static_cast<opus_int32>(max_bytes));
+  EXPECT_GE(value, 0);
+  return value;
+}
+
+TEST_F(NetEqOpusQualityTest, Test) {
+  Simulate();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
new file mode 100644
index 0000000000..c3e160cb66
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
@@ -0,0 +1,81 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "absl/flags/flag.h"
+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/testsupport/file_utils.h"
+
+ABSL_FLAG(int, frame_size_ms, 20, "Codec frame size (milliseconds).");
+
+using ::testing::InitGoogleTest;
+
+namespace webrtc {
+namespace test {
+namespace {
+static const int kInputSampleRateKhz = 48;
+static const int kOutputSampleRateKhz = 48;
+}  // namespace
+
+class NetEqPcm16bQualityTest : public NetEqQualityTest {
+ protected:
+  NetEqPcm16bQualityTest()
+      : NetEqQualityTest(absl::GetFlag(FLAGS_frame_size_ms),
+                         kInputSampleRateKhz,
+                         kOutputSampleRateKhz,
+                         SdpAudioFormat("l16", 48000, 1)) {
+    // Flag validation
+    RTC_CHECK(absl::GetFlag(FLAGS_frame_size_ms) >= 10 &&
+              absl::GetFlag(FLAGS_frame_size_ms) <= 60 &&
+              (absl::GetFlag(FLAGS_frame_size_ms) % 10) == 0)
+        << "Invalid frame size, should be 10, 20, ..., 60 ms.";
+  }
+
+  void SetUp() override {
+    AudioEncoderPcm16B::Config config;
+    config.frame_size_ms = absl::GetFlag(FLAGS_frame_size_ms);
+    config.sample_rate_hz = 48000;
+    config.num_channels = channels_;
+    encoder_.reset(new AudioEncoderPcm16B(config));
+    NetEqQualityTest::SetUp();
+  }
+
+  int EncodeBlock(int16_t* in_data,
+                  size_t block_size_samples,
+                  rtc::Buffer* payload,
+                  size_t max_bytes) override {
+    const size_t kFrameSizeSamples = 480;  // Samples per 10 ms.
+    size_t encoded_samples = 0;
+    uint32_t dummy_timestamp = 0;
+    AudioEncoder::EncodedInfo info;
+    do {
+      info = encoder_->Encode(dummy_timestamp,
+                              rtc::ArrayView<const int16_t>(
+                                  in_data + encoded_samples, kFrameSizeSamples),
+                              payload);
+      encoded_samples += kFrameSizeSamples;
+    } while (info.encoded_bytes == 0);
+    return rtc::checked_cast<int>(info.encoded_bytes);
+  }
+
+ private:
+  std::unique_ptr<AudioEncoderPcm16B> encoder_;
+};
+
+TEST_F(NetEqPcm16bQualityTest, Test) {
+  Simulate();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
new file mode 100644
index 0000000000..d22170c623
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
@@ -0,0 +1,80 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "absl/flags/flag.h"
+#include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "test/testsupport/file_utils.h"
+
+ABSL_FLAG(int, frame_size_ms, 20, "Codec frame size (milliseconds).");
+
+using ::testing::InitGoogleTest;
+
+namespace webrtc {
+namespace test {
+namespace {
+static const int kInputSampleRateKhz = 8;
+static const int kOutputSampleRateKhz = 8;
+}  // namespace
+
+class NetEqPcmuQualityTest : public NetEqQualityTest {
+ protected:
+  NetEqPcmuQualityTest()
+      : NetEqQualityTest(absl::GetFlag(FLAGS_frame_size_ms),
+                         kInputSampleRateKhz,
+                         kOutputSampleRateKhz,
+                         SdpAudioFormat("pcmu", 8000, 1)) {
+    // Flag validation
+    RTC_CHECK(absl::GetFlag(FLAGS_frame_size_ms) >= 10 &&
+              absl::GetFlag(FLAGS_frame_size_ms) <= 60 &&
+              (absl::GetFlag(FLAGS_frame_size_ms) % 10) == 0)
+        << "Invalid frame size, should be 10, 20, ..., 60 ms.";
+  }
+
+  void SetUp() override {
+    ASSERT_EQ(1u, channels_) << "PCMu supports only mono audio.";
+    AudioEncoderPcmU::Config config;
+    config.frame_size_ms = absl::GetFlag(FLAGS_frame_size_ms);
+    encoder_.reset(new AudioEncoderPcmU(config));
+    NetEqQualityTest::SetUp();
+  }
+
+  int EncodeBlock(int16_t* in_data,
+                  size_t block_size_samples,
+                  rtc::Buffer* payload,
+                  size_t max_bytes) override {
+    const size_t kFrameSizeSamples = 80;  // Samples per 10 ms.
+    size_t encoded_samples = 0;
+    uint32_t dummy_timestamp = 0;
+    AudioEncoder::EncodedInfo info;
+    do {
+      info = encoder_->Encode(dummy_timestamp,
+                              rtc::ArrayView<const int16_t>(
+                                  in_data + encoded_samples, kFrameSizeSamples),
+                              payload);
+      encoded_samples += kFrameSizeSamples;
+    } while (info.encoded_bytes == 0);
+    return rtc::checked_cast<int>(info.encoded_bytes);
+  }
+
+ private:
+  std::unique_ptr<AudioEncoderPcmU> encoder_;
+};
+
+TEST_F(NetEqPcmuQualityTest, Test) {
+  Simulate();
+}
+
+}  // namespace test
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_performance_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_performance_unittest.cc
new file mode 100644
index 0000000000..961f74ab66
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_performance_unittest.cc
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/test/metrics/global_metrics_logger_and_exporter.h"
+#include "api/test/metrics/metric.h"
+#include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
+#include "system_wrappers/include/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+namespace {
+
+using ::webrtc::test::GetGlobalMetricsLogger;
+using ::webrtc::test::ImprovementDirection;
+using ::webrtc::test::Unit;
+
+// Runs a test with 10% packet losses and 10% clock drift, to exercise
+// both loss concealment and time-stretching code.
+TEST(NetEqPerformanceTest, 10_Pl_10_Drift) {
+  const int kSimulationTimeMs = 10000000;
+  const int kQuickSimulationTimeMs = 100000;
+  const int kLossPeriod = 10;  // Drop every 10th packet.
+  const double kDriftFactor = 0.1;
+  int64_t runtime = test::NetEqPerformanceTest::Run(
+      field_trial::IsEnabled("WebRTC-QuickPerfTest") ? kQuickSimulationTimeMs
+                                                     : kSimulationTimeMs,
+      kLossPeriod, kDriftFactor);
+  ASSERT_GT(runtime, 0);
+  GetGlobalMetricsLogger()->LogSingleValueMetric(
+      "neteq_performance", "10_pl_10_drift", runtime, Unit::kMilliseconds,
+      ImprovementDirection::kNeitherIsBetter);
+}
+
+// Runs a test with neither packet losses nor clock drift, to put
+// emphasis on the "good-weather" code path, which is presumably much
+// more lightweight.
+TEST(NetEqPerformanceTest, 0_Pl_0_Drift) {
+  const int kSimulationTimeMs = 10000000;
+  const int kQuickSimulationTimeMs = 100000;
+  const int kLossPeriod = 0;        // No losses.
+  const double kDriftFactor = 0.0;  // No clock drift.
+  int64_t runtime = test::NetEqPerformanceTest::Run(
+      field_trial::IsEnabled("WebRTC-QuickPerfTest") ? kQuickSimulationTimeMs
+                                                     : kSimulationTimeMs,
+      kLossPeriod, kDriftFactor);
+  ASSERT_GT(runtime, 0);
+  GetGlobalMetricsLogger()->LogSingleValueMetric(
+      "neteq_performance", "0_pl_0_drift", runtime, Unit::kMilliseconds,
+      ImprovementDirection::kNeitherIsBetter);
+}
+
+}  // namespace
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc
new file mode 100644
index 0000000000..a72b2009eb
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc
@@ -0,0 +1,58 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <stdio.h>
+
+#include <iostream>
+#include <vector>
+
+#include "absl/flags/flag.h"
+#include "absl/flags/parse.h"
+#include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
+#include "rtc_base/checks.h"
+
+// Define command line flags.
+ABSL_FLAG(int, runtime_ms, 10000, "Simulated runtime in ms.");
+ABSL_FLAG(int, lossrate, 10, "Packet lossrate; drop every N packets.");
+ABSL_FLAG(float, drift, 0.1f, "Clockdrift factor.");
+
+int main(int argc, char* argv[]) {
+  std::vector<char*> args = absl::ParseCommandLine(argc, argv);
+  std::string program_name = args[0];
+  std::string usage =
+      "Tool for measuring the speed of NetEq.\n"
+      "Usage: " +
+      program_name +
+      " [options]\n\n"
+      "  --runtime_ms=N         runtime in ms; default is 10000 ms\n"
+      "  --lossrate=N           drop every N packets; default is 10\n"
+      "  --drift=F              clockdrift factor between 0.0 and 1.0; "
+      "default is 0.1\n";
+  if (args.size() != 1) {
+    printf("%s", usage.c_str());
+    return 1;
+  }
+  RTC_CHECK_GT(absl::GetFlag(FLAGS_runtime_ms), 0);
+  RTC_CHECK_GE(absl::GetFlag(FLAGS_lossrate), 0);
+  RTC_CHECK(absl::GetFlag(FLAGS_drift) >= 0.0 &&
+            absl::GetFlag(FLAGS_drift) < 1.0);
+
+  int64_t result = webrtc::test::NetEqPerformanceTest::Run(
+      absl::GetFlag(FLAGS_runtime_ms), absl::GetFlag(FLAGS_lossrate),
+      absl::GetFlag(FLAGS_drift));
+  if (result <= 0) {
+    std::cout << "There was an error" << std::endl;
+    return -1;
+  }
+
+  std::cout << "Simulation done" << std::endl;
+  std::cout << "Runtime = " << result << " ms" << std::endl;
+  return 0;
+}
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.cc b/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.cc
new file mode 100644
index 0000000000..f5d50dc859
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.cc
@@ -0,0 +1,109 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/audio_coding/neteq/test/result_sink.h"
+
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "rtc_base/ignore_wundef.h"
+#include "rtc_base/message_digest.h"
+#include "rtc_base/string_encode.h"
+#include "test/gtest.h"
+
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+RTC_PUSH_IGNORING_WUNDEF()
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
+#else
+#include "modules/audio_coding/neteq/neteq_unittest.pb.h"
+#endif
+RTC_POP_IGNORING_WUNDEF()
+#endif
+
+namespace webrtc {
+
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+void Convert(const webrtc::NetEqNetworkStatistics& stats_raw,
+             webrtc::neteq_unittest::NetEqNetworkStatistics* stats) {
+  stats->set_current_buffer_size_ms(stats_raw.current_buffer_size_ms);
+  stats->set_preferred_buffer_size_ms(stats_raw.preferred_buffer_size_ms);
+  stats->set_jitter_peaks_found(stats_raw.jitter_peaks_found);
+  stats->set_expand_rate(stats_raw.expand_rate);
+  stats->set_speech_expand_rate(stats_raw.speech_expand_rate);
+  stats->set_preemptive_rate(stats_raw.preemptive_rate);
+  stats->set_accelerate_rate(stats_raw.accelerate_rate);
+  stats->set_secondary_decoded_rate(stats_raw.secondary_decoded_rate);
+  stats->set_secondary_discarded_rate(stats_raw.secondary_discarded_rate);
+  stats->set_mean_waiting_time_ms(stats_raw.mean_waiting_time_ms);
+  stats->set_median_waiting_time_ms(stats_raw.median_waiting_time_ms);
+  stats->set_min_waiting_time_ms(stats_raw.min_waiting_time_ms);
+  stats->set_max_waiting_time_ms(stats_raw.max_waiting_time_ms);
+}
+
+void AddMessage(FILE* file,
+                rtc::MessageDigest* digest,
+                absl::string_view message) {
+  int32_t size = message.length();
+  if (file)
+    ASSERT_EQ(1u, fwrite(&size, sizeof(size), 1, file));
+  digest->Update(&size, sizeof(size));
+
+  if (file)
+    ASSERT_EQ(static_cast<size_t>(size),
+              fwrite(message.data(), sizeof(char), size, file));
+  digest->Update(message.data(), sizeof(char) * size);
+}
+
+#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
+
+ResultSink::ResultSink(absl::string_view output_file)
+    : output_fp_(nullptr),
+      digest_(rtc::MessageDigestFactory::Create(rtc::DIGEST_SHA_1)) {
+  if (!output_file.empty()) {
+    output_fp_ = fopen(std::string(output_file).c_str(), "wb");
+    EXPECT_TRUE(output_fp_ != NULL);
+  }
+}
+
+ResultSink::~ResultSink() {
+  if (output_fp_)
+    fclose(output_fp_);
+}
+
+void ResultSink::AddResult(const NetEqNetworkStatistics& stats_raw) {
+#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
+  neteq_unittest::NetEqNetworkStatistics stats;
+  Convert(stats_raw, &stats);
+
+  std::string stats_string;
+  ASSERT_TRUE(stats.SerializeToString(&stats_string));
+  AddMessage(output_fp_, digest_.get(), stats_string);
+#else
+  FAIL() << "Writing to reference file requires Proto Buffer.";
+#endif  // WEBRTC_NETEQ_UNITTEST_BITEXACT
+}
+
+void ResultSink::VerifyChecksum(absl::string_view checksum) {
+  std::string buffer;
+  buffer.resize(digest_->Size());
+  digest_->Finish(buffer.data(), buffer.size());
+  const std::string result = rtc::hex_encode(buffer);
+  if (checksum.size() == result.size()) {
+    EXPECT_EQ(checksum, result);
+  } else {
+    // Check result is one the '|'-separated checksums.
+    EXPECT_NE(checksum.find(result), absl::string_view::npos)
+        << result << " should be one of these:\n"
+        << checksum;
+  }
+}
+
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.h b/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.h
new file mode 100644
index 0000000000..c6923d7a7f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/result_sink.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_TEST_RESULT_SINK_H_
+#define MODULES_AUDIO_CODING_NETEQ_TEST_RESULT_SINK_H_
+
+#include <cstdio>
+#include <memory>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/neteq/neteq.h"
+#include "rtc_base/message_digest.h"
+
+namespace webrtc {
+
+class ResultSink {
+ public:
+  explicit ResultSink(absl::string_view output_file);
+  ~ResultSink();
+
+  template <typename T>
+  void AddResult(const T* test_results, size_t length);
+
+  void AddResult(const NetEqNetworkStatistics& stats);
+
+  void VerifyChecksum(absl::string_view ref_check_sum);
+
+ private:
+  FILE* output_fp_;
+  std::unique_ptr<rtc::MessageDigest> digest_;
+};
+
+template <typename T>
+void ResultSink::AddResult(const T* test_results, size_t length) {
+  if (output_fp_) {
+    ASSERT_EQ(length, fwrite(test_results, sizeof(T), length, output_fp_));
+  }
+  digest_->Update(test_results, sizeof(T) * length);
+}
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TEST_RESULT_SINK_H_