Merging upstream version 127.0.

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

27 files changed, 324 insertions, 480 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/BUILD.gn b/third_party/libwebrtc/modules/audio_coding/BUILD.gn
index ddd1fd2656..a49df7e7d2 100644
--- a/third_party/libwebrtc/modules/audio_coding/BUILD.gn
+++ b/third_party/libwebrtc/modules/audio_coding/BUILD.gn
@@ -689,8 +689,6 @@ rtc_library("neteq") {
     "neteq/packet_arrival_history.h",
     "neteq/packet_buffer.cc",
     "neteq/packet_buffer.h",
-    "neteq/post_decode_vad.cc",
-    "neteq/post_decode_vad.h",
     "neteq/preemptive_expand.cc",
     "neteq/preemptive_expand.h",
     "neteq/random_vector.cc",
@@ -1655,7 +1653,6 @@ if (rtc_include_tests) {
         "neteq/normal_unittest.cc",
         "neteq/packet_arrival_history_unittest.cc",
         "neteq/packet_buffer_unittest.cc",
-        "neteq/post_decode_vad_unittest.cc",
         "neteq/random_vector_unittest.cc",
         "neteq/red_payload_splitter_unittest.cc",
         "neteq/reorder_optimizer_unittest.cc",
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver.cc b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver.cc
index a5bf88e547..4deabdf7ff 100644
--- a/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver.cc
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver.cc
@@ -50,11 +50,7 @@ std::unique_ptr<NetEq> CreateNetEq(
 
 AcmReceiver::Config::Config(
     rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
-    : clock(*Clock::GetRealTimeClockRaw()), decoder_factory(decoder_factory) {
-  // Post-decode VAD is disabled by default in NetEq, however, Audio
-  // Conference Mixer relies on VAD decisions and fails without them.
-  neteq_config.enable_post_decode_vad = true;
-}
+    : clock(*Clock::GetRealTimeClockRaw()), decoder_factory(decoder_factory) {}
 
 AcmReceiver::Config::Config(const Config&) = default;
 AcmReceiver::Config::~Config() = default;
diff --git a/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
index cda6688157..8b35f4a621 100644
--- a/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
+++ b/third_party/libwebrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
@@ -190,9 +190,6 @@ class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
     const size_t output_channels = info.num_channels;
     const size_t samples_per_ms = rtc::checked_cast<size_t>(
         rtc::CheckedDivExact(output_sample_rate_hz, 1000));
-    const AudioFrame::VADActivity expected_vad_activity =
-        output_sample_rate_hz > 16000 ? AudioFrame::kVadActive
-                                      : AudioFrame::kVadPassive;
 
     // Expect the first output timestamp to be 5*fs/8000 samples before the
     // first inserted timestamp (because of NetEq's look-ahead). (This value is
@@ -217,7 +214,6 @@ class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
         EXPECT_EQ(output_sample_rate_hz, frame.sample_rate_hz_);
         EXPECT_EQ(output_channels, frame.num_channels_);
         EXPECT_EQ(AudioFrame::kNormalSpeech, frame.speech_type_);
-        EXPECT_EQ(expected_vad_activity, frame.vad_activity_);
         EXPECT_FALSE(muted);
       }
     }
@@ -243,61 +239,6 @@ TEST_F(AcmReceiverTestFaxModeOldApi, MAYBE_VerifyAudioFrameOpus) {
 }
 
 #if defined(WEBRTC_ANDROID)
-#define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
-#else
-#define MAYBE_PostdecodingVad PostdecodingVad
-#endif
-TEST_F(AcmReceiverTestOldApi, MAYBE_PostdecodingVad) {
-  EXPECT_TRUE(config_.neteq_config.enable_post_decode_vad);
-  constexpr int payload_type = 34;
-  const SdpAudioFormat codec = {"L16", 16000, 1};
-  const AudioCodecInfo info = SetEncoder(payload_type, codec);
-  receiver_->SetCodecs({{payload_type, codec}});
-  constexpr int kNumPackets = 5;
-  AudioFrame frame;
-  for (int n = 0; n < kNumPackets; ++n) {
-    const int num_10ms_frames = InsertOnePacketOfSilence(info);
-    for (int k = 0; k < num_10ms_frames; ++k) {
-      bool muted;
-      ASSERT_EQ(0, receiver_->GetAudio(info.sample_rate_hz, &frame, &muted));
-    }
-  }
-  EXPECT_EQ(AudioFrame::kVadPassive, frame.vad_activity_);
-}
-
-class AcmReceiverTestPostDecodeVadPassiveOldApi : public AcmReceiverTestOldApi {
- protected:
-  AcmReceiverTestPostDecodeVadPassiveOldApi() {
-    config_.neteq_config.enable_post_decode_vad = false;
-  }
-};
-
-#if defined(WEBRTC_ANDROID)
-#define MAYBE_PostdecodingVad DISABLED_PostdecodingVad
-#else
-#define MAYBE_PostdecodingVad PostdecodingVad
-#endif
-TEST_F(AcmReceiverTestPostDecodeVadPassiveOldApi, MAYBE_PostdecodingVad) {
-  EXPECT_FALSE(config_.neteq_config.enable_post_decode_vad);
-  constexpr int payload_type = 34;
-  const SdpAudioFormat codec = {"L16", 16000, 1};
-  const AudioCodecInfo info = SetEncoder(payload_type, codec);
-  auto const value = encoder_factory_->QueryAudioEncoder(codec);
-  ASSERT_TRUE(value.has_value());
-  receiver_->SetCodecs({{payload_type, codec}});
-  const int kNumPackets = 5;
-  AudioFrame frame;
-  for (int n = 0; n < kNumPackets; ++n) {
-    const int num_10ms_frames = InsertOnePacketOfSilence(info);
-    for (int k = 0; k < num_10ms_frames; ++k) {
-      bool muted;
-      ASSERT_EQ(0, receiver_->GetAudio(info.sample_rate_hz, &frame, &muted));
-    }
-  }
-  EXPECT_EQ(AudioFrame::kVadUnknown, frame.vad_activity_);
-}
-
-#if defined(WEBRTC_ANDROID)
 #define MAYBE_LastAudioCodec DISABLED_LastAudioCodec
 #else
 #define MAYBE_LastAudioCodec LastAudioCodec
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
index 46ac671b30..ff7e919d9b 100644
--- a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
@@ -58,6 +58,11 @@ int AudioDecoderPcmU::PacketDuration(const uint8_t* encoded,
   return static_cast<int>(encoded_len / Channels());
 }
 
+int AudioDecoderPcmU::PacketDurationRedundant(const uint8_t* encoded,
+                                              size_t encoded_len) const {
+  return PacketDuration(encoded, encoded_len);
+}
+
 void AudioDecoderPcmA::Reset() {}
 
 std::vector<AudioDecoder::ParseResult> AudioDecoderPcmA::ParsePayload(
@@ -99,4 +104,9 @@ int AudioDecoderPcmA::PacketDuration(const uint8_t* encoded,
   return static_cast<int>(encoded_len / Channels());
 }
 
+int AudioDecoderPcmA::PacketDurationRedundant(const uint8_t* encoded,
+                                              size_t encoded_len) const {
+  return PacketDuration(encoded, encoded_len);
+}
+
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
index 3fa42cba30..5531d6e7f0 100644
--- a/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.h
@@ -35,6 +35,8 @@ class AudioDecoderPcmU final : public AudioDecoder {
   std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
                                         uint32_t timestamp) override;
   int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int PacketDurationRedundant(const uint8_t* encoded,
+                              size_t encoded_len) const override;
   int SampleRateHz() const override;
   size_t Channels() const override;
 
@@ -62,6 +64,8 @@ class AudioDecoderPcmA final : public AudioDecoder {
   std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
                                         uint32_t timestamp) override;
   int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int PacketDurationRedundant(const uint8_t* encoded,
+                              size_t encoded_len) const override;
   int SampleRateHz() const override;
   size_t Channels() const override;
 
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
index e969ed1189..bca47cea13 100644
--- a/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.cc
@@ -63,6 +63,11 @@ int AudioDecoderG722Impl::PacketDuration(const uint8_t* encoded,
   return static_cast<int>(2 * encoded_len / Channels());
 }
 
+int AudioDecoderG722Impl::PacketDurationRedundant(const uint8_t* encoded,
+                                                  size_t encoded_len) const {
+  return PacketDuration(encoded, encoded_len);
+}
+
 int AudioDecoderG722Impl::SampleRateHz() const {
   return 16000;
 }
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
index 5872fad5de..e7083c3fd6 100644
--- a/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/g722/audio_decoder_g722.h
@@ -30,6 +30,8 @@ class AudioDecoderG722Impl final : public AudioDecoder {
   std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
                                         uint32_t timestamp) override;
   int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int PacketDurationRedundant(const uint8_t* encoded,
+                              size_t encoded_len) const override;
   int SampleRateHz() const override;
   size_t Channels() const override;
 
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
index cff9685548..0f53409f48 100644
--- a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
@@ -17,12 +17,15 @@
 #include "api/array_view.h"
 #include "modules/audio_coding/codecs/opus/audio_coder_opus_common.h"
 #include "rtc_base/checks.h"
+#include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
 AudioDecoderOpusImpl::AudioDecoderOpusImpl(size_t num_channels,
                                            int sample_rate_hz)
-    : channels_{num_channels}, sample_rate_hz_{sample_rate_hz} {
+    : channels_(num_channels),
+      sample_rate_hz_(sample_rate_hz),
+      generate_plc_(field_trial::IsEnabled("WebRTC-Audio-OpusGeneratePlc")) {
   RTC_DCHECK(num_channels == 1 || num_channels == 2);
   RTC_DCHECK(sample_rate_hz == 16000 || sample_rate_hz == 48000);
   const int error =
@@ -125,4 +128,22 @@ size_t AudioDecoderOpusImpl::Channels() const {
   return channels_;
 }
 
+void AudioDecoderOpusImpl::GeneratePlc(
+    size_t requested_samples_per_channel,
+    rtc::BufferT<int16_t>* concealment_audio) {
+  if (!generate_plc_) {
+    return;
+  }
+  int plc_size = WebRtcOpus_PlcDuration(dec_state_) * channels_;
+  concealment_audio->AppendData(plc_size, [&](rtc::ArrayView<int16_t> decoded) {
+    int16_t temp_type = 1;
+    int ret =
+        WebRtcOpus_Decode(dec_state_, nullptr, 0, decoded.data(), &temp_type);
+    if (ret < 0) {
+      return 0;
+    }
+    return ret;
+  });
+}
+
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
index e8fd0440bc..2dd62fd4ee 100644
--- a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.h
@@ -40,6 +40,8 @@ class AudioDecoderOpusImpl final : public AudioDecoder {
   bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const override;
   int SampleRateHz() const override;
   size_t Channels() const override;
+  void GeneratePlc(size_t requested_samples_per_channel,
+                   rtc::BufferT<int16_t>* concealment_audio) override;
 
  protected:
   int DecodeInternal(const uint8_t* encoded,
@@ -57,6 +59,7 @@ class AudioDecoderOpusImpl final : public AudioDecoder {
   OpusDecInst* dec_state_;
   const size_t channels_;
   const int sample_rate_hz_;
+  const bool generate_plc_;
 };
 
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
index 7761efe8b3..1e2b5db331 100644
--- a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.cc
@@ -67,4 +67,9 @@ int AudioDecoderPcm16B::PacketDuration(const uint8_t* encoded,
   return static_cast<int>(encoded_len / (2 * Channels()));
 }
 
+int AudioDecoderPcm16B::PacketDurationRedundant(const uint8_t* encoded,
+                                                size_t encoded_len) const {
+  return PacketDuration(encoded, encoded_len);
+}
+
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
index 6f50161d3f..c31cc5d0a2 100644
--- a/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h
@@ -32,6 +32,8 @@ class AudioDecoderPcm16B final : public AudioDecoder {
   std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload,
                                         uint32_t timestamp) override;
   int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
+  int PacketDurationRedundant(const uint8_t* encoded,
+                              size_t encoded_len) const override;
   int SampleRateHz() const override;
   size_t Channels() const override;
 
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.cc b/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.cc
index 2c95d3b390..0c33dba47a 100644
--- a/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.cc
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.cc
@@ -17,7 +17,6 @@
 #include "common_audio/signal_processing/include/signal_processing_library.h"
 #include "modules/audio_coding/neteq/audio_multi_vector.h"
 #include "modules/audio_coding/neteq/cross_correlation.h"
-#include "modules/audio_coding/neteq/post_decode_vad.h"
 
 namespace webrtc {
 namespace {
@@ -44,17 +43,11 @@ void BackgroundNoise::Reset() {
   }
 }
 
-bool BackgroundNoise::Update(const AudioMultiVector& input,
-                             const PostDecodeVad& vad) {
+bool BackgroundNoise::Update(const AudioMultiVector& sync_buffer) {
   bool filter_params_saved = false;
-  if (vad.running() && vad.active_speech()) {
-    // Do not update the background noise parameters if we know that the signal
-    // is active speech.
-    return filter_params_saved;
-  }
 
   int32_t auto_correlation[kMaxLpcOrder + 1];
-  int16_t fiter_output[kMaxLpcOrder + kResidualLength];
+  int16_t filter_output[kMaxLpcOrder + kResidualLength];
   int16_t reflection_coefficients[kMaxLpcOrder];
   int16_t lpc_coefficients[kMaxLpcOrder + 1];
 
@@ -62,14 +55,13 @@ bool BackgroundNoise::Update(const AudioMultiVector& input,
     ChannelParameters& parameters = channel_parameters_[channel_ix];
     int16_t temp_signal_array[kVecLen + kMaxLpcOrder] = {0};
     int16_t* temp_signal = &temp_signal_array[kMaxLpcOrder];
-    RTC_DCHECK_GE(input.Size(), kVecLen);
-    input[channel_ix].CopyTo(kVecLen, input.Size() - kVecLen, temp_signal);
+    RTC_DCHECK_GE(sync_buffer.Size(), kVecLen);
+    sync_buffer[channel_ix].CopyTo(kVecLen, sync_buffer.Size() - kVecLen,
+                                   temp_signal);
     int32_t sample_energy =
         CalculateAutoCorrelation(temp_signal, kVecLen, auto_correlation);
 
-    if ((!vad.running() &&
-         sample_energy < parameters.energy_update_threshold) ||
-        (vad.running() && !vad.active_speech())) {
+    if (sample_energy < parameters.energy_update_threshold) {
       // Generate LPC coefficients.
       if (auto_correlation[0] <= 0) {
         // Center value in auto-correlation is not positive. Do not update.
@@ -95,10 +87,10 @@ bool BackgroundNoise::Update(const AudioMultiVector& input,
 
       // Generate the CNG gain factor by looking at the energy of the residual.
       WebRtcSpl_FilterMAFastQ12(temp_signal + kVecLen - kResidualLength,
-                                fiter_output, lpc_coefficients,
+                                filter_output, lpc_coefficients,
                                 kMaxLpcOrder + 1, kResidualLength);
       int32_t residual_energy = WebRtcSpl_DotProductWithScale(
-          fiter_output, fiter_output, kResidualLength, 0);
+          filter_output, filter_output, kResidualLength, 0);
 
       // Check spectral flatness.
       // Comparing the residual variance with the input signal variance tells
@@ -117,9 +109,8 @@ bool BackgroundNoise::Update(const AudioMultiVector& input,
         filter_params_saved = true;
       }
     } else {
-      // Will only happen if post-decode VAD is disabled and `sample_energy` is
-      // not low enough. Increase the threshold for update so that it increases
-      // by a factor 4 in 4 seconds.
+      // Will only happen if `sample_energy` is not low enough. Increase the
+      // threshold for update so that it increases by a factor 4 in 4 seconds.
       IncrementEnergyThreshold(channel_ix, sample_energy);
     }
   }
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.h b/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.h
index 8e6d5890a0..9ef0131c92 100644
--- a/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.h
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/background_noise.h
@@ -39,9 +39,9 @@ class BackgroundNoise {
   void Reset();
 
   // Updates the parameter estimates based on the signal currently in the
-  // `sync_buffer`, and on the latest decision in `vad` if it is running.
+  // `sync_buffer`.
   // Returns true if the filter parameters are updated.
-  bool Update(const AudioMultiVector& sync_buffer, const PostDecodeVad& vad);
+  bool Update(const AudioMultiVector& sync_buffer);
 
   // Generates background noise given a random vector and writes the output to
   // `buffer`.
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc
index 6648fd8709..f68c05767d 100644
--- a/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic.cc
@@ -14,7 +14,6 @@
 
 #include <cstdint>
 #include <memory>
-#include <string>
 
 #include "absl/types/optional.h"
 #include "api/neteq/neteq.h"
@@ -22,7 +21,6 @@
 #include "modules/audio_coding/neteq/packet_arrival_history.h"
 #include "modules/audio_coding/neteq/packet_buffer.h"
 #include "rtc_base/checks.h"
-#include "rtc_base/experiments/field_trial_parser.h"
 #include "rtc_base/experiments/struct_parameters_parser.h"
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_conversions.h"
@@ -102,6 +100,7 @@ DecisionLogic::DecisionLogic(
       packet_arrival_history_(packet_arrival_history
                                   ? std::move(packet_arrival_history)
                                   : std::make_unique<PacketArrivalHistory>(
+                                        config.tick_timer,
                                         config_.packet_history_size_ms)),
       tick_timer_(config.tick_timer),
       disallow_time_stretching_(!config.allow_time_stretching),
@@ -221,14 +220,14 @@ absl::optional<int> DecisionLogic::PacketArrived(
     packet_length_samples_ = info.packet_length_samples;
     delay_manager_->SetPacketAudioLength(packet_length_samples_ * 1000 / fs_hz);
   }
-  int64_t time_now_ms = tick_timer_->ticks() * tick_timer_->ms_per_tick();
-  packet_arrival_history_->Insert(info.main_timestamp, time_now_ms);
-  if (packet_arrival_history_->size() < 2) {
+  bool inserted = packet_arrival_history_->Insert(info.main_timestamp,
+                                                  info.packet_length_samples);
+  if (!inserted || packet_arrival_history_->size() < 2) {
     // No meaningful delay estimate unless at least 2 packets have arrived.
     return absl::nullopt;
   }
   int arrival_delay_ms =
-      packet_arrival_history_->GetDelayMs(info.main_timestamp, time_now_ms);
+      packet_arrival_history_->GetDelayMs(info.main_timestamp);
   bool reordered =
       !packet_arrival_history_->IsNewestRtpTimestamp(info.main_timestamp);
   delay_manager_->Update(arrival_delay_ms, reordered);
@@ -464,8 +463,7 @@ int DecisionLogic::GetPlayoutDelayMs(
     NetEqController::NetEqStatus status) const {
   uint32_t playout_timestamp =
       status.target_timestamp - status.sync_buffer_samples;
-  return packet_arrival_history_->GetDelayMs(
-      playout_timestamp, tick_timer_->ticks() * tick_timer_->ms_per_tick());
+  return packet_arrival_history_->GetDelayMs(playout_timestamp);
 }
 
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
index 9e9902af50..4b306f2639 100644
--- a/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
@@ -14,12 +14,10 @@
 
 #include "api/neteq/neteq_controller.h"
 #include "api/neteq/tick_timer.h"
-#include "modules/audio_coding/neteq/buffer_level_filter.h"
 #include "modules/audio_coding/neteq/delay_manager.h"
 #include "modules/audio_coding/neteq/mock/mock_buffer_level_filter.h"
 #include "modules/audio_coding/neteq/mock/mock_delay_manager.h"
 #include "modules/audio_coding/neteq/mock/mock_packet_arrival_history.h"
-#include "test/field_trial.h"
 #include "test/gtest.h"
 
 namespace webrtc {
@@ -64,7 +62,8 @@ class DecisionLogicTest : public ::testing::Test {
     mock_delay_manager_ = delay_manager.get();
     auto buffer_level_filter = std::make_unique<MockBufferLevelFilter>();
     mock_buffer_level_filter_ = buffer_level_filter.get();
-    auto packet_arrival_history = std::make_unique<MockPacketArrivalHistory>();
+    auto packet_arrival_history =
+        std::make_unique<MockPacketArrivalHistory>(&tick_timer_);
     mock_packet_arrival_history_ = packet_arrival_history.get();
     decision_logic_ = std::make_unique<DecisionLogic>(
         config, std::move(delay_manager), std::move(buffer_level_filter),
@@ -82,7 +81,7 @@ class DecisionLogicTest : public ::testing::Test {
 TEST_F(DecisionLogicTest, NormalOperation) {
   EXPECT_CALL(*mock_delay_manager_, TargetDelayMs())
       .WillRepeatedly(Return(100));
-  EXPECT_CALL(*mock_packet_arrival_history_, GetDelayMs(_, _))
+  EXPECT_CALL(*mock_packet_arrival_history_, GetDelayMs(_))
       .WillRepeatedly(Return(100));
   EXPECT_CALL(*mock_packet_arrival_history_, GetMaxDelayMs())
       .WillRepeatedly(Return(0));
@@ -98,7 +97,7 @@ TEST_F(DecisionLogicTest, NormalOperation) {
 TEST_F(DecisionLogicTest, Accelerate) {
   EXPECT_CALL(*mock_delay_manager_, TargetDelayMs())
       .WillRepeatedly(Return(100));
-  EXPECT_CALL(*mock_packet_arrival_history_, GetDelayMs(_, _))
+  EXPECT_CALL(*mock_packet_arrival_history_, GetDelayMs(_))
       .WillRepeatedly(Return(150));
   EXPECT_CALL(*mock_packet_arrival_history_, GetMaxDelayMs())
       .WillRepeatedly(Return(0));
@@ -114,7 +113,7 @@ TEST_F(DecisionLogicTest, Accelerate) {
 TEST_F(DecisionLogicTest, FastAccelerate) {
   EXPECT_CALL(*mock_delay_manager_, TargetDelayMs())
       .WillRepeatedly(Return(100));
-  EXPECT_CALL(*mock_packet_arrival_history_, GetDelayMs(_, _))
+  EXPECT_CALL(*mock_packet_arrival_history_, GetDelayMs(_))
       .WillRepeatedly(Return(500));
   EXPECT_CALL(*mock_packet_arrival_history_, GetMaxDelayMs())
       .WillRepeatedly(Return(0));
@@ -130,7 +129,7 @@ TEST_F(DecisionLogicTest, FastAccelerate) {
 TEST_F(DecisionLogicTest, PreemptiveExpand) {
   EXPECT_CALL(*mock_delay_manager_, TargetDelayMs())
       .WillRepeatedly(Return(100));
-  EXPECT_CALL(*mock_packet_arrival_history_, GetDelayMs(_, _))
+  EXPECT_CALL(*mock_packet_arrival_history_, GetDelayMs(_))
       .WillRepeatedly(Return(50));
   EXPECT_CALL(*mock_packet_arrival_history_, GetMaxDelayMs())
       .WillRepeatedly(Return(0));
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_arrival_history.h b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_arrival_history.h
index 1b2080cd94..d4217cf2f8 100644
--- a/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_arrival_history.h
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/mock/mock_packet_arrival_history.h
@@ -11,6 +11,7 @@
 #ifndef MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_ARRIVAL_HISTORY_H_
 #define MODULES_AUDIO_CODING_NETEQ_MOCK_MOCK_PACKET_ARRIVAL_HISTORY_H_
 
+#include "api/neteq/tick_timer.h"
 #include "modules/audio_coding/neteq/packet_arrival_history.h"
 #include "test/gmock.h"
 
@@ -18,12 +19,10 @@ namespace webrtc {
 
 class MockPacketArrivalHistory : public PacketArrivalHistory {
  public:
-  MockPacketArrivalHistory() : PacketArrivalHistory(0) {}
+  MockPacketArrivalHistory(const TickTimer* tick_timer)
+      : PacketArrivalHistory(tick_timer, 0) {}
 
-  MOCK_METHOD(int,
-              GetDelayMs,
-              (uint32_t rtp_timestamp, int64_t time_ms),
-              (const override));
+  MOCK_METHOD(int, GetDelayMs, (uint32_t rtp_timestamp), (const override));
   MOCK_METHOD(int, GetMaxDelayMs, (), (const override));
 };
 
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc
index e5c8bf6c08..6a76096b49 100644
--- a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -20,6 +20,7 @@
 #include <vector>
 
 #include "api/audio_codecs/audio_decoder.h"
+#include "api/neteq/neteq_controller.h"
 #include "api/neteq/tick_timer.h"
 #include "common_audio/signal_processing/include/signal_processing_library.h"
 #include "modules/audio_coding/codecs/cng/webrtc_cng.h"
@@ -36,7 +37,6 @@
 #include "modules/audio_coding/neteq/normal.h"
 #include "modules/audio_coding/neteq/packet.h"
 #include "modules/audio_coding/neteq/packet_buffer.h"
-#include "modules/audio_coding/neteq/post_decode_vad.h"
 #include "modules/audio_coding/neteq/preemptive_expand.h"
 #include "modules/audio_coding/neteq/red_payload_splitter.h"
 #include "modules/audio_coding/neteq/statistics_calculator.h"
@@ -50,6 +50,7 @@
 #include "rtc_base/strings/audio_format_to_string.h"
 #include "rtc_base/trace_event.h"
 #include "system_wrappers/include/clock.h"
+#include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 namespace {
@@ -70,49 +71,26 @@ std::unique_ptr<NetEqController> CreateNetEqController(
   return controller_factory.CreateNetEqController(config);
 }
 
-void SetAudioFrameActivityAndType(bool vad_enabled,
-                                  NetEqImpl::OutputType type,
-                                  AudioFrame::VADActivity last_vad_activity,
-                                  AudioFrame* audio_frame) {
+AudioFrame::SpeechType ToSpeechType(NetEqImpl::OutputType type) {
   switch (type) {
     case NetEqImpl::OutputType::kNormalSpeech: {
-      audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
-      audio_frame->vad_activity_ = AudioFrame::kVadActive;
-      break;
-    }
-    case NetEqImpl::OutputType::kVadPassive: {
-      // This should only be reached if the VAD is enabled.
-      RTC_DCHECK(vad_enabled);
-      audio_frame->speech_type_ = AudioFrame::kNormalSpeech;
-      audio_frame->vad_activity_ = AudioFrame::kVadPassive;
-      break;
+      return AudioFrame::kNormalSpeech;
     }
     case NetEqImpl::OutputType::kCNG: {
-      audio_frame->speech_type_ = AudioFrame::kCNG;
-      audio_frame->vad_activity_ = AudioFrame::kVadPassive;
-      break;
+      return AudioFrame::kCNG;
     }
     case NetEqImpl::OutputType::kPLC: {
-      audio_frame->speech_type_ = AudioFrame::kPLC;
-      audio_frame->vad_activity_ = last_vad_activity;
-      break;
+      return AudioFrame::kPLC;
     }
     case NetEqImpl::OutputType::kPLCCNG: {
-      audio_frame->speech_type_ = AudioFrame::kPLCCNG;
-      audio_frame->vad_activity_ = AudioFrame::kVadPassive;
-      break;
+      return AudioFrame::kPLCCNG;
     }
     case NetEqImpl::OutputType::kCodecPLC: {
-      audio_frame->speech_type_ = AudioFrame::kCodecPLC;
-      audio_frame->vad_activity_ = last_vad_activity;
-      break;
+      return AudioFrame::kCodecPLC;
     }
     default:
       RTC_DCHECK_NOTREACHED();
-  }
-  if (!vad_enabled) {
-    // Always set kVadUnknown when receive VAD is inactive.
-    audio_frame->vad_activity_ = AudioFrame::kVadUnknown;
+      return AudioFrame::kUndefined;
   }
 }
 
@@ -169,11 +147,12 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
       packet_buffer_(std::move(deps.packet_buffer)),
       red_payload_splitter_(std::move(deps.red_payload_splitter)),
       timestamp_scaler_(std::move(deps.timestamp_scaler)),
-      vad_(new PostDecodeVad()),
       expand_factory_(std::move(deps.expand_factory)),
       accelerate_factory_(std::move(deps.accelerate_factory)),
       preemptive_expand_factory_(std::move(deps.preemptive_expand_factory)),
       stats_(std::move(deps.stats)),
+      enable_fec_delay_adaptation_(
+          !field_trial::IsDisabled("WebRTC-Audio-NetEqFecDelayAdaptation")),
       controller_(std::move(deps.neteq_controller)),
       last_mode_(Mode::kNormal),
       decoded_buffer_length_(kMaxFrameSize),
@@ -211,10 +190,6 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
   if (create_components) {
     SetSampleRateAndChannels(fs, 1);  // Default is 1 channel.
   }
-  RTC_DCHECK(!vad_->enabled());
-  if (config.enable_post_decode_vad) {
-    vad_->Enable();
-  }
 }
 
 NetEqImpl::~NetEqImpl() = default;
@@ -252,9 +227,7 @@ int NetEqImpl::GetAudio(AudioFrame* audio_frame,
       audio_frame->sample_rate_hz_,
       rtc::dchecked_cast<int>(audio_frame->samples_per_channel_ * 100));
   RTC_DCHECK_EQ(*muted, audio_frame->muted());
-  SetAudioFrameActivityAndType(vad_->enabled(), LastOutputType(),
-                               last_vad_activity_, audio_frame);
-  last_vad_activity_ = audio_frame->vad_activity_;
+  audio_frame->speech_type_ = ToSpeechType(LastOutputType());
   last_output_sample_rate_hz_ = audio_frame->sample_rate_hz_;
   RTC_DCHECK(last_output_sample_rate_hz_ == 8000 ||
              last_output_sample_rate_hz_ == 16000 ||
@@ -398,18 +371,6 @@ NetEqOperationsAndState NetEqImpl::GetOperationsAndState() const {
   return result;
 }
 
-void NetEqImpl::EnableVad() {
-  MutexLock lock(&mutex_);
-  RTC_DCHECK(vad_.get());
-  vad_->Enable();
-}
-
-void NetEqImpl::DisableVad() {
-  MutexLock lock(&mutex_);
-  RTC_DCHECK(vad_.get());
-  vad_->Disable();
-}
-
 absl::optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() const {
   MutexLock lock(&mutex_);
   if (first_packet_ || last_mode_ == Mode::kRfc3389Cng ||
@@ -695,6 +656,7 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
       packet_buffer_->Flush();
       buffer_flush_occured = true;
     }
+    NetEqController::PacketArrivedInfo info = ToPacketArrivedInfo(packet);
     int return_val = packet_buffer_->InsertPacket(std::move(packet));
     if (return_val == PacketBuffer::kFlushed) {
       buffer_flush_occured = true;
@@ -702,6 +664,15 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
       // An error occurred.
       return kOtherError;
     }
+    if (enable_fec_delay_adaptation_) {
+      info.buffer_flush = buffer_flush_occured;
+      const bool should_update_stats = !new_codec_ && !buffer_flush_occured;
+      auto relative_delay =
+          controller_->PacketArrived(fs_hz_, should_update_stats, info);
+      if (relative_delay) {
+        stats_->RelativePacketArrivalDelay(relative_delay.value());
+      }
+    }
   }
 
   if (buffer_flush_occured) {
@@ -752,24 +723,26 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
     }
   }
 
-  const DecoderDatabase::DecoderInfo* dec_info =
-      decoder_database_->GetDecoderInfo(main_payload_type);
-  RTC_DCHECK(dec_info);  // Already checked that the payload type is known.
-
-  NetEqController::PacketArrivedInfo info;
-  info.is_cng_or_dtmf = dec_info->IsComfortNoise() || dec_info->IsDtmf();
-  info.packet_length_samples =
-      number_of_primary_packets * decoder_frame_length_;
-  info.main_timestamp = main_timestamp;
-  info.main_sequence_number = main_sequence_number;
-  info.is_dtx = is_dtx;
-  info.buffer_flush = buffer_flush_occured;
-
-  const bool should_update_stats = !new_codec_;
-  auto relative_delay =
-      controller_->PacketArrived(fs_hz_, should_update_stats, info);
-  if (relative_delay) {
-    stats_->RelativePacketArrivalDelay(relative_delay.value());
+  if (!enable_fec_delay_adaptation_) {
+    const DecoderDatabase::DecoderInfo* dec_info =
+        decoder_database_->GetDecoderInfo(main_payload_type);
+    RTC_DCHECK(dec_info);  // Already checked that the payload type is known.
+
+    NetEqController::PacketArrivedInfo info;
+    info.is_cng_or_dtmf = dec_info->IsComfortNoise() || dec_info->IsDtmf();
+    info.packet_length_samples =
+        number_of_primary_packets * decoder_frame_length_;
+    info.main_timestamp = main_timestamp;
+    info.main_sequence_number = main_sequence_number;
+    info.is_dtx = is_dtx;
+    info.buffer_flush = buffer_flush_occured;
+
+    const bool should_update_stats = !new_codec_;
+    auto relative_delay =
+        controller_->PacketArrived(fs_hz_, should_update_stats, info);
+    if (relative_delay) {
+      stats_->RelativePacketArrivalDelay(relative_delay.value());
+    }
   }
   return 0;
 }
@@ -858,11 +831,8 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame,
     last_decoded_type_ = speech_type;
   }
 
-  RTC_DCHECK(vad_.get());
   bool sid_frame_available =
       (operation == Operation::kRfc3389Cng && !packet_list.empty());
-  vad_->Update(decoded_buffer_.get(), static_cast<size_t>(length), speech_type,
-               sid_frame_available, fs_hz_);
 
   // This is the criterion that we did decode some data through the speech
   // decoder, and the operation resulted in comfort noise.
@@ -1012,7 +982,7 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame,
       (last_mode_ == Mode::kPreemptiveExpandFail) ||
       (last_mode_ == Mode::kRfc3389Cng) ||
       (last_mode_ == Mode::kCodecInternalCng)) {
-    background_noise_->Update(*sync_buffer_, *vad_.get());
+    background_noise_->Update(*sync_buffer_);
   }
 
   if (operation == Operation::kDtmf) {
@@ -2088,10 +2058,6 @@ void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
   if (cng_decoder)
     cng_decoder->Reset();
 
-  // Reinit post-decode VAD with new sample rate.
-  RTC_DCHECK(vad_.get());  // Cannot be NULL here.
-  vad_->Init();
-
   // Delete algorithm buffer and create a new one.
   algorithm_buffer_.reset(new AudioMultiVector(channels));
 
@@ -2132,7 +2098,6 @@ void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
 }
 
 NetEqImpl::OutputType NetEqImpl::LastOutputType() {
-  RTC_DCHECK(vad_.get());
   RTC_DCHECK(expand_.get());
   if (last_mode_ == Mode::kCodecInternalCng ||
       last_mode_ == Mode::kRfc3389Cng) {
@@ -2142,12 +2107,27 @@ NetEqImpl::OutputType NetEqImpl::LastOutputType() {
     return OutputType::kPLCCNG;
   } else if (last_mode_ == Mode::kExpand) {
     return OutputType::kPLC;
-  } else if (vad_->running() && !vad_->active_speech()) {
-    return OutputType::kVadPassive;
   } else if (last_mode_ == Mode::kCodecPlc) {
     return OutputType::kCodecPLC;
   } else {
     return OutputType::kNormalSpeech;
   }
 }
+
+NetEqController::PacketArrivedInfo NetEqImpl::ToPacketArrivedInfo(
+    const Packet& packet) const {
+  const DecoderDatabase::DecoderInfo* dec_info =
+      decoder_database_->GetDecoderInfo(packet.payload_type);
+
+  NetEqController::PacketArrivedInfo info;
+  info.is_cng_or_dtmf =
+      dec_info && (dec_info->IsComfortNoise() || dec_info->IsDtmf());
+  info.packet_length_samples =
+      packet.frame ? packet.frame->Duration() : decoder_frame_length_;
+  info.main_timestamp = packet.timestamp;
+  info.main_sequence_number = packet.sequence_number;
+  info.is_dtx = packet.frame && packet.frame->IsDtxPacket();
+  return info;
+}
+
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.h b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.h
index f8f2b06410..eed7645e7d 100644
--- a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -48,7 +48,6 @@ class Merge;
 class NackTracker;
 class Normal;
 class RedPayloadSplitter;
-class PostDecodeVad;
 class PreemptiveExpand;
 class RandomVector;
 class SyncBuffer;
@@ -171,13 +170,6 @@ class NetEqImpl : public webrtc::NetEq {
 
   NetEqOperationsAndState GetOperationsAndState() const override;
 
-  // Enables post-decode VAD. When enabled, GetAudio() will return
-  // kOutputVADPassive when the signal contains no speech.
-  void EnableVad() override;
-
-  // Disables post-decode VAD.
-  void DisableVad() override;
-
   absl::optional<uint32_t> GetPlayoutTimestamp() const override;
 
   int last_output_sample_rate_hz() const override;
@@ -342,6 +334,9 @@ class NetEqImpl : public webrtc::NetEq {
   NetEqNetworkStatistics CurrentNetworkStatisticsInternal() const
       RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
 
+  NetEqController::PacketArrivedInfo ToPacketArrivedInfo(
+      const Packet& packet) const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
+
   Clock* const clock_;
 
   mutable Mutex mutex_;
@@ -356,13 +351,13 @@ class NetEqImpl : public webrtc::NetEq {
       RTC_GUARDED_BY(mutex_);
   const std::unique_ptr<TimestampScaler> timestamp_scaler_
       RTC_GUARDED_BY(mutex_);
-  const std::unique_ptr<PostDecodeVad> vad_ RTC_GUARDED_BY(mutex_);
   const std::unique_ptr<ExpandFactory> expand_factory_ RTC_GUARDED_BY(mutex_);
   const std::unique_ptr<AccelerateFactory> accelerate_factory_
       RTC_GUARDED_BY(mutex_);
   const std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
       RTC_GUARDED_BY(mutex_);
   const std::unique_ptr<StatisticsCalculator> stats_ RTC_GUARDED_BY(mutex_);
+  const bool enable_fec_delay_adaptation_ RTC_GUARDED_BY(mutex_);
 
   std::unique_ptr<BackgroundNoise> background_noise_ RTC_GUARDED_BY(mutex_);
   std::unique_ptr<NetEqController> controller_ RTC_GUARDED_BY(mutex_);
@@ -397,8 +392,6 @@ class NetEqImpl : public webrtc::NetEq {
   std::unique_ptr<NackTracker> nack_ RTC_GUARDED_BY(mutex_);
   bool nack_enabled_ RTC_GUARDED_BY(mutex_);
   const bool enable_muted_state_ RTC_GUARDED_BY(mutex_);
-  AudioFrame::VADActivity last_vad_activity_ RTC_GUARDED_BY(mutex_) =
-      AudioFrame::kVadPassive;
   std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
       RTC_GUARDED_BY(mutex_);
   std::vector<RtpPacketInfo> last_decoded_packet_infos_ RTC_GUARDED_BY(mutex_);
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc
index aec7e580ec..7104b7a6dc 100644
--- a/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/neteq_unittest.cc
@@ -76,12 +76,13 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
       webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
 
   const std::string output_checksum =
-      "2efdbea92c3fb2383c59f89d881efec9f94001d0|"
-      "a6831b946b59913852ae3e53f99fa8f209bb23cd";
+      "434bdc4ec08546510ee903d001c8be1a01c44e24|"
+      "4336be0091e2faad7a194c16ee0a05e727325727|"
+      "cefd2de4adfa8f6a9b66a3639ad63c2f6779d0cd";
 
   const std::string network_stats_checksum =
-      "dfaf4399fd60293405290476ccf1c05c807c71a0|"
-      "076662525572dba753b11578330bd491923f7f5e";
+      "5f2c8e3dff9cff55dd7a9f4167939de001566d95|"
+      "80ab17c17da030d4f2dfbf314ac44aacdadd7f0c";
 
   DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
                    absl::GetFlag(FLAGS_gen_ref));
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.cc b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.cc
index 2077383f76..a36c8a2b06 100644
--- a/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.cc
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.cc
@@ -11,95 +11,122 @@
 #include "modules/audio_coding/neteq/packet_arrival_history.h"
 
 #include <algorithm>
+#include <cstdint>
 
 #include "api/neteq/tick_timer.h"
+#include "rtc_base/checks.h"
 
 namespace webrtc {
 
-PacketArrivalHistory::PacketArrivalHistory(int window_size_ms)
-    : window_size_ms_(window_size_ms) {}
+PacketArrivalHistory::PacketArrivalHistory(const TickTimer* tick_timer,
+                                           int window_size_ms)
+    : tick_timer_(tick_timer), window_size_ms_(window_size_ms) {}
 
-void PacketArrivalHistory::Insert(uint32_t rtp_timestamp,
-                                  int64_t arrival_time_ms) {
-  RTC_DCHECK(sample_rate_khz_ > 0);
-  int64_t unwrapped_rtp_timestamp = timestamp_unwrapper_.Unwrap(rtp_timestamp);
-  if (!newest_rtp_timestamp_ ||
-      unwrapped_rtp_timestamp > *newest_rtp_timestamp_) {
-    newest_rtp_timestamp_ = unwrapped_rtp_timestamp;
+bool PacketArrivalHistory::Insert(uint32_t rtp_timestamp,
+                                  int packet_length_samples) {
+  int64_t arrival_timestamp =
+      tick_timer_->ticks() * tick_timer_->ms_per_tick() * sample_rate_khz_;
+  PacketArrival packet(timestamp_unwrapper_.Unwrap(rtp_timestamp),
+                       arrival_timestamp, packet_length_samples);
+  if (IsObsolete(packet)) {
+    return false;
   }
-  history_.emplace_back(unwrapped_rtp_timestamp / sample_rate_khz_,
-                        arrival_time_ms);
-  MaybeUpdateCachedArrivals(history_.back());
-  while (history_.front().rtp_timestamp_ms + window_size_ms_ <
-         unwrapped_rtp_timestamp / sample_rate_khz_) {
-    if (&history_.front() == min_packet_arrival_) {
-      min_packet_arrival_ = nullptr;
-    }
-    if (&history_.front() == max_packet_arrival_) {
-      max_packet_arrival_ = nullptr;
-    }
-    history_.pop_front();
+  if (Contains(packet)) {
+    return false;
+  }
+  history_.emplace(packet.rtp_timestamp, packet);
+  if (packet != history_.rbegin()->second) {
+    // Packet was reordered.
+    return true;
   }
-  if (!min_packet_arrival_ || !max_packet_arrival_) {
-    for (const PacketArrival& packet : history_) {
-      MaybeUpdateCachedArrivals(packet);
+  // Remove old packets.
+  while (IsObsolete(history_.begin()->second)) {
+    if (history_.begin()->second == min_packet_arrivals_.front()) {
+      min_packet_arrivals_.pop_front();
     }
+    if (history_.begin()->second == max_packet_arrivals_.front()) {
+      max_packet_arrivals_.pop_front();
+    }
+    history_.erase(history_.begin());
   }
-}
-
-void PacketArrivalHistory::MaybeUpdateCachedArrivals(
-    const PacketArrival& packet_arrival) {
-  if (!min_packet_arrival_ || packet_arrival <= *min_packet_arrival_) {
-    min_packet_arrival_ = &packet_arrival;
+  // Ensure ordering constraints.
+  while (!min_packet_arrivals_.empty() &&
+         packet <= min_packet_arrivals_.back()) {
+    min_packet_arrivals_.pop_back();
   }
-  if (!max_packet_arrival_ || packet_arrival >= *max_packet_arrival_) {
-    max_packet_arrival_ = &packet_arrival;
+  while (!max_packet_arrivals_.empty() &&
+         packet >= max_packet_arrivals_.back()) {
+    max_packet_arrivals_.pop_back();
   }
+  min_packet_arrivals_.push_back(packet);
+  max_packet_arrivals_.push_back(packet);
+  return true;
 }
 
 void PacketArrivalHistory::Reset() {
   history_.clear();
-  min_packet_arrival_ = nullptr;
-  max_packet_arrival_ = nullptr;
+  min_packet_arrivals_.clear();
+  max_packet_arrivals_.clear();
   timestamp_unwrapper_.Reset();
-  newest_rtp_timestamp_ = absl::nullopt;
 }
 
-int PacketArrivalHistory::GetDelayMs(uint32_t rtp_timestamp,
-                                     int64_t time_ms) const {
-  RTC_DCHECK(sample_rate_khz_ > 0);
-  int64_t unwrapped_rtp_timestamp_ms =
-      timestamp_unwrapper_.PeekUnwrap(rtp_timestamp) / sample_rate_khz_;
-  PacketArrival packet(unwrapped_rtp_timestamp_ms, time_ms);
+int PacketArrivalHistory::GetDelayMs(uint32_t rtp_timestamp) const {
+  int64_t unwrapped_rtp_timestamp =
+      timestamp_unwrapper_.PeekUnwrap(rtp_timestamp);
+  int64_t current_timestamp =
+      tick_timer_->ticks() * tick_timer_->ms_per_tick() * sample_rate_khz_;
+  PacketArrival packet(unwrapped_rtp_timestamp, current_timestamp,
+                       /*duration_ms=*/0);
   return GetPacketArrivalDelayMs(packet);
 }
 
 int PacketArrivalHistory::GetMaxDelayMs() const {
-  if (!max_packet_arrival_) {
+  if (max_packet_arrivals_.empty()) {
     return 0;
   }
-  return GetPacketArrivalDelayMs(*max_packet_arrival_);
+  return GetPacketArrivalDelayMs(max_packet_arrivals_.front());
 }
 
 bool PacketArrivalHistory::IsNewestRtpTimestamp(uint32_t rtp_timestamp) const {
-  if (!newest_rtp_timestamp_) {
-    return false;
+  if (history_.empty()) {
+    return true;
   }
   int64_t unwrapped_rtp_timestamp =
       timestamp_unwrapper_.PeekUnwrap(rtp_timestamp);
-  return unwrapped_rtp_timestamp == *newest_rtp_timestamp_;
+  return unwrapped_rtp_timestamp == history_.rbegin()->second.rtp_timestamp;
 }
 
 int PacketArrivalHistory::GetPacketArrivalDelayMs(
     const PacketArrival& packet_arrival) const {
-  if (!min_packet_arrival_) {
+  if (min_packet_arrivals_.empty()) {
     return 0;
   }
-  return std::max(static_cast<int>(packet_arrival.arrival_time_ms -
-                                   min_packet_arrival_->arrival_time_ms -
-                                   (packet_arrival.rtp_timestamp_ms -
-                                    min_packet_arrival_->rtp_timestamp_ms)),
-                  0);
+  RTC_DCHECK_NE(sample_rate_khz_, 0);
+  // TODO(jakobi): Timestamps are first converted to millis for bit-exactness.
+  return std::max<int>(
+      packet_arrival.arrival_timestamp / sample_rate_khz_ -
+          min_packet_arrivals_.front().arrival_timestamp / sample_rate_khz_ -
+          (packet_arrival.rtp_timestamp / sample_rate_khz_ -
+           min_packet_arrivals_.front().rtp_timestamp / sample_rate_khz_),
+      0);
+}
+
+bool PacketArrivalHistory::IsObsolete(
+    const PacketArrival& packet_arrival) const {
+  if (history_.empty()) {
+    return false;
+  }
+  return packet_arrival.rtp_timestamp + window_size_ms_ * sample_rate_khz_ <
+         history_.rbegin()->second.rtp_timestamp;
+}
+
+bool PacketArrivalHistory::Contains(const PacketArrival& packet_arrival) const {
+  auto it = history_.upper_bound(packet_arrival.rtp_timestamp);
+  if (it == history_.begin()) {
+    return false;
+  }
+  --it;
+  return it->second.contains(packet_arrival);
 }
 
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h
index 722caf5688..3fa1ea1fa9 100644
--- a/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.h
@@ -11,10 +11,11 @@
 #ifndef MODULES_AUDIO_CODING_NETEQ_PACKET_ARRIVAL_HISTORY_H_
 #define MODULES_AUDIO_CODING_NETEQ_PACKET_ARRIVAL_HISTORY_H_
 
+#include <cstddef>
 #include <cstdint>
 #include <deque>
+#include <map>
 
-#include "absl/types/optional.h"
 #include "api/neteq/tick_timer.h"
 #include "rtc_base/numerics/sequence_number_unwrapper.h"
 
@@ -25,19 +26,22 @@ namespace webrtc {
 // pruned.
 class PacketArrivalHistory {
  public:
-  explicit PacketArrivalHistory(int window_size_ms);
+  explicit PacketArrivalHistory(const TickTimer* tick_timer,
+                                int window_size_ms);
   virtual ~PacketArrivalHistory() = default;
 
-  // Insert packet with `rtp_timestamp` and `arrival_time_ms` into the history.
-  void Insert(uint32_t rtp_timestamp, int64_t arrival_time_ms);
+  // Insert packet with `rtp_timestamp` into the history. Returns true if the
+  // packet was inserted, false if the timestamp is too old or if the timestamp
+  // already exists.
+  bool Insert(uint32_t rtp_timestamp, int packet_length_samples);
 
-  // The delay for `rtp_timestamp` at `time_ms` is calculated as
-  // `(time_ms - p.arrival_time_ms) - (rtp_timestamp - p.rtp_timestamp)`
-  // where `p` is chosen as the packet arrival in the history that maximizes the
-  // delay.
-  virtual int GetDelayMs(uint32_t rtp_timestamp, int64_t time_ms) const;
+  // The delay for `rtp_timestamp` at time `now` is calculated as
+  // `(now - p.arrival_timestamp) - (rtp_timestamp - p.rtp_timestamp)` where `p`
+  // is chosen as the packet arrival in the history that maximizes the delay.
+  virtual int GetDelayMs(uint32_t rtp_timestamp) const;
 
-  // Get the maximum packet arrival delay observed in the history.
+  // Get the maximum packet arrival delay observed in the history, excluding
+  // reordered packets.
   virtual int GetMaxDelayMs() const;
 
   bool IsNewestRtpTimestamp(uint32_t rtp_timestamp) const;
@@ -52,30 +56,53 @@ class PacketArrivalHistory {
 
  private:
   struct PacketArrival {
-    PacketArrival(int64_t rtp_timestamp_ms, int64_t arrival_time_ms)
-        : rtp_timestamp_ms(rtp_timestamp_ms),
-          arrival_time_ms(arrival_time_ms) {}
-    int64_t rtp_timestamp_ms;
-    int64_t arrival_time_ms;
+    PacketArrival(int64_t rtp_timestamp,
+                  int64_t arrival_timestamp,
+                  int length_samples)
+        : rtp_timestamp(rtp_timestamp),
+          arrival_timestamp(arrival_timestamp),
+          length_samples(length_samples) {}
+    PacketArrival() = default;
+    int64_t rtp_timestamp;
+    int64_t arrival_timestamp;
+    int length_samples;
+    bool operator==(const PacketArrival& other) const {
+      return rtp_timestamp == other.rtp_timestamp &&
+             arrival_timestamp == other.arrival_timestamp &&
+             length_samples == other.length_samples;
+    }
+    bool operator!=(const PacketArrival& other) const {
+      return !(*this == other);
+    }
     bool operator<=(const PacketArrival& other) const {
-      return arrival_time_ms - rtp_timestamp_ms <=
-             other.arrival_time_ms - other.rtp_timestamp_ms;
+      return arrival_timestamp - rtp_timestamp <=
+             other.arrival_timestamp - other.rtp_timestamp;
     }
     bool operator>=(const PacketArrival& other) const {
-      return arrival_time_ms - rtp_timestamp_ms >=
-             other.arrival_time_ms - other.rtp_timestamp_ms;
+      return arrival_timestamp - rtp_timestamp >=
+             other.arrival_timestamp - other.rtp_timestamp;
+    }
+    bool contains(const PacketArrival& other) const {
+      return rtp_timestamp <= other.rtp_timestamp &&
+             rtp_timestamp + length_samples >=
+                 other.rtp_timestamp + other.length_samples;
     }
   };
-  std::deque<PacketArrival> history_;
   int GetPacketArrivalDelayMs(const PacketArrival& packet_arrival) const;
-  // Updates `min_packet_arrival_` and `max_packet_arrival_`.
-  void MaybeUpdateCachedArrivals(const PacketArrival& packet);
-  const PacketArrival* min_packet_arrival_ = nullptr;
-  const PacketArrival* max_packet_arrival_ = nullptr;
+  // Checks if the packet is older than the window size.
+  bool IsObsolete(const PacketArrival& packet_arrival) const;
+  // Check if the packet exists or fully overlaps with a packet in the history.
+  bool Contains(const PacketArrival& packet_arrival) const;
+  const TickTimer* tick_timer_;
   const int window_size_ms_;
-  RtpTimestampUnwrapper timestamp_unwrapper_;
-  absl::optional<int64_t> newest_rtp_timestamp_;
   int sample_rate_khz_ = 0;
+  RtpTimestampUnwrapper timestamp_unwrapper_;
+  // Packet history ordered by rtp timestamp.
+  std::map<int64_t, PacketArrival> history_;
+  // Tracks min/max packet arrivals in `history_` in ascending/descending order.
+  // Reordered packets are excluded.
+  std::deque<PacketArrival> min_packet_arrivals_;
+  std::deque<PacketArrival> max_packet_arrivals_;
 };
 
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history_unittest.cc
index 539a318fe1..dd95fec0f7 100644
--- a/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history_unittest.cc
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history_unittest.cc
@@ -21,32 +21,36 @@ namespace {
 constexpr int kFs = 8000;
 constexpr int kFsKhz = kFs / 1000;
 constexpr int kFrameSizeMs = 20;
+constexpr int kFrameSizeSamples = kFrameSizeMs * kFsKhz;
 constexpr int kWindowSizeMs = 1000;
 
 class PacketArrivalHistoryTest : public testing::Test {
  public:
-  PacketArrivalHistoryTest() : history_(kWindowSizeMs) {
+  PacketArrivalHistoryTest() : history_(&tick_timer_, kWindowSizeMs) {
     history_.set_sample_rate(kFs);
   }
-  void IncrementTime(int delta_ms) { time_ms_ += delta_ms; }
+  void IncrementTime(int delta_ms) {
+    tick_timer_.Increment(delta_ms / tick_timer_.ms_per_tick());
+  }
   int InsertPacketAndGetDelay(int timestamp_delta_ms) {
     uint32_t timestamp = timestamp_ + timestamp_delta_ms * kFsKhz;
     if (timestamp_delta_ms > 0) {
       timestamp_ = timestamp;
     }
-    history_.Insert(timestamp, time_ms_);
+    EXPECT_TRUE(history_.Insert(timestamp, kFrameSizeSamples));
     EXPECT_EQ(history_.IsNewestRtpTimestamp(timestamp),
               timestamp_delta_ms >= 0);
-    return history_.GetDelayMs(timestamp, time_ms_);
+    return history_.GetDelayMs(timestamp);
   }
 
  protected:
-  int64_t time_ms_ = 0;
+  TickTimer tick_timer_;
   PacketArrivalHistory history_;
   uint32_t timestamp_ = 0x12345678;
 };
 
 TEST_F(PacketArrivalHistoryTest, RelativeArrivalDelay) {
+  // Insert first packet.
   EXPECT_EQ(InsertPacketAndGetDelay(0), 0);
 
   IncrementTime(kFrameSizeMs);
@@ -56,7 +60,7 @@ TEST_F(PacketArrivalHistoryTest, RelativeArrivalDelay) {
   EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 20);
 
   // Reordered packet.
-  EXPECT_EQ(InsertPacketAndGetDelay(-2 * kFrameSizeMs), 60);
+  EXPECT_EQ(InsertPacketAndGetDelay(-3 * kFrameSizeMs), 80);
 
   IncrementTime(2 * kFrameSizeMs);
   EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 40);
@@ -68,7 +72,7 @@ TEST_F(PacketArrivalHistoryTest, RelativeArrivalDelay) {
   EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 20);
 
   // Earlier packet is now more delayed due to the new reference packet.
-  EXPECT_EQ(history_.GetMaxDelayMs(), 100);
+  EXPECT_EQ(history_.GetMaxDelayMs(), 80);
 }
 
 TEST_F(PacketArrivalHistoryTest, ReorderedPackets) {
@@ -86,7 +90,7 @@ TEST_F(PacketArrivalHistoryTest, ReorderedPackets) {
   IncrementTime(4 * kFrameSizeMs);
   EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 60);
 
-  EXPECT_EQ(history_.GetMaxDelayMs(), 80);
+  EXPECT_EQ(history_.GetMaxDelayMs(), 60);
 }
 
 TEST_F(PacketArrivalHistoryTest, MaxHistorySize) {
@@ -117,7 +121,7 @@ TEST_F(PacketArrivalHistoryTest, TimestampWraparound) {
   // Insert another in-order packet after the wraparound.
   EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 0);
 
-  EXPECT_EQ(history_.GetMaxDelayMs(), 3 * kFrameSizeMs);
+  EXPECT_EQ(history_.GetMaxDelayMs(), kFrameSizeMs);
 }
 
 TEST_F(PacketArrivalHistoryTest, TimestampWraparoundBackwards) {
@@ -134,7 +138,33 @@ TEST_F(PacketArrivalHistoryTest, TimestampWraparoundBackwards) {
   // Insert another in-order packet after the wraparound.
   EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 0);
 
-  EXPECT_EQ(history_.GetMaxDelayMs(), 3 * kFrameSizeMs);
+  EXPECT_EQ(history_.GetMaxDelayMs(), kFrameSizeMs);
+}
+
+TEST_F(PacketArrivalHistoryTest, OldPacketShouldNotBeInserted) {
+  // Insert first packet as reference.
+  EXPECT_EQ(InsertPacketAndGetDelay(0), 0);
+  // Insert packet with timestamp older than the window size compared to the
+  // first packet.
+  EXPECT_FALSE(history_.Insert(timestamp_ - kWindowSizeMs * kFsKhz - 1,
+                               kFrameSizeSamples));
+}
+
+TEST_F(PacketArrivalHistoryTest, DuplicatePacketShouldNotBeInserted) {
+  // Insert first packet as reference.
+  uint32_t first_timestamp = timestamp_;
+  EXPECT_EQ(InsertPacketAndGetDelay(0), 0);
+  EXPECT_EQ(InsertPacketAndGetDelay(kFrameSizeMs), 0);
+  // Same timestamp as the first packet.
+  EXPECT_FALSE(history_.Insert(first_timestamp, kFrameSizeSamples));
+}
+
+TEST_F(PacketArrivalHistoryTest, OverlappingPacketShouldNotBeInserted) {
+  // Insert first packet as reference.
+  EXPECT_EQ(InsertPacketAndGetDelay(0), 0);
+  // 10 ms overlap with the previous packet.
+  EXPECT_FALSE(history_.Insert(timestamp_ + kFrameSizeSamples / 2,
+                               kFrameSizeSamples / 2));
 }
 
 }  // namespace
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.cc b/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.cc
deleted file mode 100644
index 9999d6764b..0000000000
--- a/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.cc
+++ /dev/null
@@ -1,90 +0,0 @@
-/*
- *  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 "modules/audio_coding/neteq/post_decode_vad.h"
-
-namespace webrtc {
-
-PostDecodeVad::~PostDecodeVad() {
-  if (vad_instance_)
-    WebRtcVad_Free(vad_instance_);
-}
-
-void PostDecodeVad::Enable() {
-  if (!vad_instance_) {
-    // Create the instance.
-    vad_instance_ = WebRtcVad_Create();
-    if (vad_instance_ == nullptr) {
-      // Failed to create instance.
-      Disable();
-      return;
-    }
-  }
-  Init();
-  enabled_ = true;
-}
-
-void PostDecodeVad::Disable() {
-  enabled_ = false;
-  running_ = false;
-}
-
-void PostDecodeVad::Init() {
-  running_ = false;
-  if (vad_instance_) {
-    WebRtcVad_Init(vad_instance_);
-    WebRtcVad_set_mode(vad_instance_, kVadMode);
-    running_ = true;
-  }
-}
-
-void PostDecodeVad::Update(int16_t* signal,
-                           size_t length,
-                           AudioDecoder::SpeechType speech_type,
-                           bool sid_frame,
-                           int fs_hz) {
-  if (!vad_instance_ || !enabled_) {
-    return;
-  }
-
-  if (speech_type == AudioDecoder::kComfortNoise || sid_frame ||
-      fs_hz > 16000) {
-    // TODO(hlundin): Remove restriction on fs_hz.
-    running_ = false;
-    active_speech_ = true;
-    sid_interval_counter_ = 0;
-  } else if (!running_) {
-    ++sid_interval_counter_;
-  }
-
-  if (sid_interval_counter_ >= kVadAutoEnable) {
-    Init();
-  }
-
-  if (length > 0 && running_) {
-    size_t vad_sample_index = 0;
-    active_speech_ = false;
-    // Loop through frame sizes 30, 20, and 10 ms.
-    for (int vad_frame_size_ms = 30; vad_frame_size_ms >= 10;
-         vad_frame_size_ms -= 10) {
-      size_t vad_frame_size_samples =
-          static_cast<size_t>(vad_frame_size_ms * fs_hz / 1000);
-      while (length - vad_sample_index >= vad_frame_size_samples) {
-        int vad_return =
-            WebRtcVad_Process(vad_instance_, fs_hz, &signal[vad_sample_index],
-                              vad_frame_size_samples);
-        active_speech_ |= (vad_return == 1);
-        vad_sample_index += vad_frame_size_samples;
-      }
-    }
-  }
-}
-
-}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.h b/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.h
deleted file mode 100644
index 3bd91b9edb..0000000000
--- a/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.h
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- *  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.
- */
-
-#ifndef MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_
-#define MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_
-
-#include <stddef.h>
-#include <stdint.h>
-
-#include "api/audio_codecs/audio_decoder.h"
-#include "common_audio/vad/include/webrtc_vad.h"
-
-namespace webrtc {
-
-class PostDecodeVad {
- public:
-  PostDecodeVad()
-      : enabled_(false),
-        running_(false),
-        active_speech_(true),
-        sid_interval_counter_(0),
-        vad_instance_(NULL) {}
-
-  virtual ~PostDecodeVad();
-
-  PostDecodeVad(const PostDecodeVad&) = delete;
-  PostDecodeVad& operator=(const PostDecodeVad&) = delete;
-
-  // Enables post-decode VAD.
-  void Enable();
-
-  // Disables post-decode VAD.
-  void Disable();
-
-  // Initializes post-decode VAD.
-  void Init();
-
-  // Updates post-decode VAD with the audio data in `signal` having `length`
-  // samples. The data is of type `speech_type`, at the sample rate `fs_hz`.
-  void Update(int16_t* signal,
-              size_t length,
-              AudioDecoder::SpeechType speech_type,
-              bool sid_frame,
-              int fs_hz);
-
-  // Accessors.
-  bool enabled() const { return enabled_; }
-  bool running() const { return running_; }
-  bool active_speech() const { return active_speech_; }
-
- private:
-  static const int kVadMode = 0;  // Sets aggressiveness to "Normal".
-  // Number of Update() calls without CNG/SID before re-enabling VAD.
-  static const int kVadAutoEnable = 3000;
-
-  bool enabled_;
-  bool running_;
-  bool active_speech_;
-  int sid_interval_counter_;
-  ::VadInst* vad_instance_;
-};
-
-}  // namespace webrtc
-#endif  // MODULES_AUDIO_CODING_NETEQ_POST_DECODE_VAD_H_
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad_unittest.cc b/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad_unittest.cc
deleted file mode 100644
index da3e4e864e..0000000000
--- a/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad_unittest.cc
+++ /dev/null
@@ -1,25 +0,0 @@
-/*
- *  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.
- */
-
-// Unit tests for PostDecodeVad class.
-
-#include "modules/audio_coding/neteq/post_decode_vad.h"
-
-#include "test/gtest.h"
-
-namespace webrtc {
-
-TEST(PostDecodeVad, CreateAndDestroy) {
-  PostDecodeVad vad;
-}
-
-// TODO(hlundin): Write more tests.
-
-}  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
index 081bd9631f..f1a46cd2df 100644
--- a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
@@ -107,7 +107,7 @@ void NetEqReplacementInput::ReplacePacket() {
       next_hdr->timestamp - packet_->header.timestamp;
   const bool opus_dtx = packet_->payload.size() <= 2;
   if (next_hdr->sequenceNumber == packet_->header.sequenceNumber + 1 &&
-      timestamp_diff <= 120 * 48 && !opus_dtx) {
+      timestamp_diff <= 120 * 48 && timestamp_diff > 0 && !opus_dtx) {
     // Packets are in order and the timestamp diff is less than 5760 samples.
     // Accept the timestamp diff as a valid frame size.
     input_frame_size_timestamps = timestamp_diff;
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq_gn/moz.build b/third_party/libwebrtc/modules/audio_coding/neteq_gn/moz.build
index 834a8d1265..9b2996fa22 100644
--- a/third_party/libwebrtc/modules/audio_coding/neteq_gn/moz.build
+++ b/third_party/libwebrtc/modules/audio_coding/neteq_gn/moz.build
@@ -58,7 +58,6 @@ UNIFIED_SOURCES += [
     "/third_party/libwebrtc/modules/audio_coding/neteq/packet.cc",
     "/third_party/libwebrtc/modules/audio_coding/neteq/packet_arrival_history.cc",
     "/third_party/libwebrtc/modules/audio_coding/neteq/packet_buffer.cc",
-    "/third_party/libwebrtc/modules/audio_coding/neteq/post_decode_vad.cc",
     "/third_party/libwebrtc/modules/audio_coding/neteq/preemptive_expand.cc",
     "/third_party/libwebrtc/modules/audio_coding/neteq/random_vector.cc",
     "/third_party/libwebrtc/modules/audio_coding/neteq/red_payload_splitter.cc",