Merging upstream version 126.0.

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

72 files changed, 1041 insertions, 453 deletions

diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
index bd8d1cc341..998e78e1d3 100644
--- a/third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
@@ -146,7 +146,7 @@ TEST(AudioDecoderFactoryTest, CreateOpus) {
   for (int hz : {8000, 16000, 32000, 48000}) {
     for (int channels : {0, 1, 2, 3}) {
       for (std::string stereo : {"XX", "0", "1", "2"}) {
-        SdpAudioFormat::Parameters params;
+        CodecParameterMap params;
         if (stereo != "XX") {
           params["stereo"] = stereo;
         }
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index a2ebe43bbe..f82ef965db 100644
--- a/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/third_party/libwebrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
@@ -39,7 +39,7 @@ constexpr int kDefaultOpusPacSize = 960;
 constexpr int64_t kInitialTimeUs = 12345678;
 
 AudioEncoderOpusConfig CreateConfigWithParameters(
-    const SdpAudioFormat::Parameters& params) {
+    const CodecParameterMap& params) {
   const SdpAudioFormat format("opus", 48000, 2, params);
   return *AudioEncoderOpus::SdpToConfig(format);
 }
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
index 5a2df24ef6..eddacd3680 100644
--- 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
@@ -106,8 +106,8 @@ NetEqOpusQualityTest::NetEqOpusQualityTest()
 
   // Redefine decoder type if input is stereo.
   if (channels_ > 1) {
-    audio_format_ = SdpAudioFormat("opus", 48000, 2,
-                                   SdpAudioFormat::Parameters{{"stereo", "1"}});
+    audio_format_ =
+        SdpAudioFormat("opus", 48000, 2, CodecParameterMap{{"stereo", "1"}});
   }
   application_ = absl::GetFlag(FLAGS_application);
 }
diff --git a/third_party/libwebrtc/modules/audio_coding/test/TestStereo.cc b/third_party/libwebrtc/modules/audio_coding/test/TestStereo.cc
index 94a1576026..1e65e4a219 100644
--- a/third_party/libwebrtc/modules/audio_coding/test/TestStereo.cc
+++ b/third_party/libwebrtc/modules/audio_coding/test/TestStereo.cc
@@ -469,7 +469,7 @@ void TestStereo::RegisterSendCodec(char side,
                                : sampling_freq_hz;
   const std::string ptime = rtc::ToString(rtc::CheckedDivExact(
       pack_size, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
-  SdpAudioFormat::Parameters params = {{"ptime", ptime}};
+  CodecParameterMap params = {{"ptime", ptime}};
   RTC_CHECK(channels == 1 || channels == 2);
   if (absl::EqualsIgnoreCase(codec_name, "opus")) {
     if (channels == 2) {
diff --git a/third_party/libwebrtc/modules/audio_device/BUILD.gn b/third_party/libwebrtc/modules/audio_device/BUILD.gn
index a135f042db..1672be3f95 100644
--- a/third_party/libwebrtc/modules/audio_device/BUILD.gn
+++ b/third_party/libwebrtc/modules/audio_device/BUILD.gn
@@ -91,6 +91,7 @@ if (!build_with_mozilla) { # See Bug 1820869.
     "../../system_wrappers",
     "../../system_wrappers:metrics",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 }
 
@@ -213,7 +214,6 @@ if (!build_with_chromium) {
       "../../rtc_base:platform_thread",
       "../../rtc_base:random",
       "../../rtc_base:rtc_event",
-      "../../rtc_base:rtc_task_queue",
       "../../rtc_base:safe_conversions",
       "../../rtc_base:timeutils",
       "../../rtc_base/synchronization:mutex",
@@ -461,6 +461,7 @@ rtc_source_set("mock_audio_device") {
     "../../api:make_ref_counted",
     "../../test:test_support",
   ]
+  absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 }
 }
 
diff --git a/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.cc b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.cc
index 86240da196..f483b8dc79 100644
--- a/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.cc
+++ b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.cc
@@ -13,6 +13,7 @@
 #include <cstdint>
 #include <cstring>
 
+#include "api/array_view.h"
 #include "modules/audio_device/audio_device_buffer.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
@@ -107,7 +108,8 @@ void FineAudioBuffer::GetPlayoutData(rtc::ArrayView<int16_t> audio_buffer,
 
 void FineAudioBuffer::DeliverRecordedData(
     rtc::ArrayView<const int16_t> audio_buffer,
-    int record_delay_ms) {
+    int record_delay_ms,
+    absl::optional<int64_t> capture_time_ns) {
   RTC_DCHECK(IsReadyForRecord());
   // Always append new data and grow the buffer when needed.
   record_buffer_.AppendData(audio_buffer.data(), audio_buffer.size());
@@ -118,7 +120,8 @@ void FineAudioBuffer::DeliverRecordedData(
       record_channels_ * record_samples_per_channel_10ms_;
   while (record_buffer_.size() >= num_elements_10ms) {
     audio_device_buffer_->SetRecordedBuffer(record_buffer_.data(),
-                                            record_samples_per_channel_10ms_);
+                                            record_samples_per_channel_10ms_,
+                                            capture_time_ns);
     audio_device_buffer_->SetVQEData(playout_delay_ms_, record_delay_ms);
     audio_device_buffer_->DeliverRecordedData();
     memmove(record_buffer_.data(), record_buffer_.data() + num_elements_10ms,
diff --git a/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.h b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.h
index a6c3042bb2..7af41d3b21 100644
--- a/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.h
+++ b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer.h
@@ -11,6 +11,10 @@
 #ifndef MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
 #define MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
 
+#include <cstddef>
+#include <cstdint>
+
+#include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "rtc_base/buffer.h"
 
@@ -61,7 +65,12 @@ class FineAudioBuffer {
   // 5ms of data and sends a total of 10ms to WebRTC and clears the internal
   // cache. Call #3 restarts the scheme above.
   void DeliverRecordedData(rtc::ArrayView<const int16_t> audio_buffer,
-                           int record_delay_ms);
+                           int record_delay_ms) {
+    DeliverRecordedData(audio_buffer, record_delay_ms, absl::nullopt);
+  }
+  void DeliverRecordedData(rtc::ArrayView<const int16_t> audio_buffer,
+                           int record_delay_ms,
+                           absl::optional<int64_t> capture_time_ns);
 
  private:
   // Device buffer that works with 10ms chunks of data both for playout and
diff --git a/third_party/libwebrtc/modules/audio_device/fine_audio_buffer_unittest.cc b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer_unittest.cc
index 36ea85f7dd..bb9fe63922 100644
--- a/third_party/libwebrtc/modules/audio_device/fine_audio_buffer_unittest.cc
+++ b/third_party/libwebrtc/modules/audio_device/fine_audio_buffer_unittest.cc
@@ -113,7 +113,7 @@ void RunFineBufferTest(int frame_size_in_samples) {
   {
     InSequence s;
     for (int j = 0; j < kNumberOfUpdateBufferCalls - 1; ++j) {
-      EXPECT_CALL(audio_device_buffer, SetRecordedBuffer(_, kSamplesPer10Ms))
+      EXPECT_CALL(audio_device_buffer, SetRecordedBuffer(_, kSamplesPer10Ms, _))
           .WillOnce(VerifyInputBuffer(j, kChannels * kSamplesPer10Ms))
           .RetiresOnSaturation();
     }
diff --git a/third_party/libwebrtc/modules/audio_device/include/test_audio_device.cc b/third_party/libwebrtc/modules/audio_device/include/test_audio_device.cc
index 4c29c98f2c..b3923ae67d 100644
--- a/third_party/libwebrtc/modules/audio_device/include/test_audio_device.cc
+++ b/third_party/libwebrtc/modules/audio_device/include/test_audio_device.cc
@@ -21,6 +21,7 @@
 #include "absl/strings/string_view.h"
 #include "api/array_view.h"
 #include "api/make_ref_counted.h"
+#include "api/task_queue/task_queue_factory.h"
 #include "common_audio/wav_file.h"
 #include "modules/audio_device/audio_device_impl.h"
 #include "modules/audio_device/include/audio_device_default.h"
@@ -33,7 +34,6 @@
 #include "rtc_base/platform_thread.h"
 #include "rtc_base/random.h"
 #include "rtc_base/synchronization/mutex.h"
-#include "rtc_base/task_queue.h"
 #include "rtc_base/task_utils/repeating_task.h"
 #include "rtc_base/thread_annotations.h"
 #include "rtc_base/time_utils.h"
diff --git a/third_party/libwebrtc/modules/audio_device/mock_audio_device_buffer.h b/third_party/libwebrtc/modules/audio_device/mock_audio_device_buffer.h
index b0f54c20ff..0b276185da 100644
--- a/third_party/libwebrtc/modules/audio_device/mock_audio_device_buffer.h
+++ b/third_party/libwebrtc/modules/audio_device/mock_audio_device_buffer.h
@@ -11,6 +11,7 @@
 #ifndef MODULES_AUDIO_DEVICE_MOCK_AUDIO_DEVICE_BUFFER_H_
 #define MODULES_AUDIO_DEVICE_MOCK_AUDIO_DEVICE_BUFFER_H_
 
+#include "absl/types/optional.h"
 #include "modules/audio_device/audio_device_buffer.h"
 #include "test/gmock.h"
 
@@ -24,7 +25,9 @@ class MockAudioDeviceBuffer : public AudioDeviceBuffer {
   MOCK_METHOD(int32_t, GetPlayoutData, (void* audioBuffer), (override));
   MOCK_METHOD(int32_t,
               SetRecordedBuffer,
-              (const void* audioBuffer, size_t nSamples),
+              (const void* audioBuffer,
+               size_t nSamples,
+               absl::optional<int64_t> capture_time_ns),
               (override));
   MOCK_METHOD(void, SetVQEData, (int playDelayMS, int recDelayMS), (override));
   MOCK_METHOD(int32_t, DeliverRecordedData, (), (override));
diff --git a/third_party/libwebrtc/modules/audio_device/test_audio_device_impl.cc b/third_party/libwebrtc/modules/audio_device/test_audio_device_impl.cc
index 627e68b36f..a3742ea581 100644
--- a/third_party/libwebrtc/modules/audio_device/test_audio_device_impl.cc
+++ b/third_party/libwebrtc/modules/audio_device/test_audio_device_impl.cc
@@ -19,7 +19,6 @@
 #include "modules/audio_device/include/test_audio_device.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/synchronization/mutex.h"
-#include "rtc_base/task_queue.h"
 #include "rtc_base/task_utils/repeating_task.h"
 
 namespace webrtc {
@@ -59,11 +58,10 @@ TestAudioDevice::TestAudioDevice(
 }
 
 AudioDeviceGeneric::InitStatus TestAudioDevice::Init() {
-  task_queue_ =
-      std::make_unique<rtc::TaskQueue>(task_queue_factory_->CreateTaskQueue(
-          "TestAudioDeviceModuleImpl", TaskQueueFactory::Priority::NORMAL));
+  task_queue_ = task_queue_factory_->CreateTaskQueue(
+      "TestAudioDeviceModuleImpl", TaskQueueFactory::Priority::NORMAL);
 
-  RepeatingTaskHandle::Start(task_queue_->Get(), [this]() {
+  RepeatingTaskHandle::Start(task_queue_.get(), [this]() {
     ProcessAudio();
     return TimeDelta::Micros(process_interval_us_);
   });
diff --git a/third_party/libwebrtc/modules/audio_device/test_audio_device_impl.h b/third_party/libwebrtc/modules/audio_device/test_audio_device_impl.h
index 36192b7f7f..84b48948ba 100644
--- a/third_party/libwebrtc/modules/audio_device/test_audio_device_impl.h
+++ b/third_party/libwebrtc/modules/audio_device/test_audio_device_impl.h
@@ -14,6 +14,7 @@
 #include <memory>
 #include <vector>
 
+#include "api/task_queue/task_queue_base.h"
 #include "api/task_queue/task_queue_factory.h"
 #include "modules/audio_device/audio_device_buffer.h"
 #include "modules/audio_device/audio_device_generic.h"
@@ -22,7 +23,6 @@
 #include "modules/audio_device/include/test_audio_device.h"
 #include "rtc_base/buffer.h"
 #include "rtc_base/synchronization/mutex.h"
-#include "rtc_base/task_queue.h"
 
 namespace webrtc {
 
@@ -190,7 +190,7 @@ class TestAudioDevice : public AudioDeviceGeneric {
 
   std::vector<int16_t> playout_buffer_ RTC_GUARDED_BY(lock_);
   rtc::BufferT<int16_t> recording_buffer_ RTC_GUARDED_BY(lock_);
-  std::unique_ptr<rtc::TaskQueue> task_queue_;
+  std::unique_ptr<TaskQueueBase, TaskQueueDeleter> task_queue_;
 };
 
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/BUILD.gn
index 6aca7dee46..817a3515b0 100644
--- a/third_party/libwebrtc/modules/audio_processing/BUILD.gn
+++ b/third_party/libwebrtc/modules/audio_processing/BUILD.gn
@@ -34,6 +34,7 @@ rtc_library("api") {
     "../../api/audio:aec3_config",
     "../../api/audio:audio_frame_api",
     "../../api/audio:echo_control",
+    "../../api/task_queue",
     "../../rtc_base:macromagic",
     "../../rtc_base:refcount",
     "../../rtc_base:stringutils",
@@ -43,6 +44,7 @@ rtc_library("api") {
     "agc:gain_control_interface",
   ]
   absl_deps = [
+    "//third_party/abseil-cpp/absl/base:nullability",
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
@@ -185,6 +187,7 @@ rtc_library("audio_processing") {
     "../../api/audio:aec3_config",
     "../../api/audio:audio_frame_api",
     "../../api/audio:echo_control",
+    "../../api/task_queue",
     "../../audio/utility:audio_frame_operations",
     "../../common_audio:common_audio_c",
     "../../common_audio/third_party/ooura:fft_size_256",
@@ -217,6 +220,7 @@ rtc_library("audio_processing") {
     "vad",
   ]
   absl_deps = [
+    "//third_party/abseil-cpp/absl/base:nullability",
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
   ]
@@ -337,9 +341,13 @@ if (rtc_include_tests) {
       ":audio_buffer",
       ":audio_processing",
       ":audio_processing_statistics",
+      "../../api/task_queue",
       "../../test:test_support",
     ]
-    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/base:nullability",
+      "//third_party/abseil-cpp/absl/strings",
+    ]
   }
 
   if (!build_with_chromium) {
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/BUILD.gn b/third_party/libwebrtc/modules/audio_processing/aec_dump/BUILD.gn
index 78bae56835..5193e28dff 100644
--- a/third_party/libwebrtc/modules/audio_processing/aec_dump/BUILD.gn
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/BUILD.gn
@@ -14,10 +14,14 @@ rtc_source_set("aec_dump") {
 
   deps = [
     "..:aec_dump_interface",
+    "../../../api/task_queue",
     "../../../rtc_base/system:file_wrapper",
     "../../../rtc_base/system:rtc_export",
   ]
-  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/base:nullability",
+    "//third_party/abseil-cpp/absl/strings",
+  ]
 }
 
 if (rtc_include_tests) {
@@ -71,11 +75,13 @@ if (rtc_enable_protobuf) {
       "../../../rtc_base:protobuf_utils",
       "../../../rtc_base:race_checker",
       "../../../rtc_base:rtc_event",
-      "../../../rtc_base:rtc_task_queue",
       "../../../rtc_base/system:file_wrapper",
       "../../../system_wrappers",
     ]
-    absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
+    absl_deps = [
+      "//third_party/abseil-cpp/absl/base:nullability",
+      "//third_party/abseil-cpp/absl/strings",
+    ]
 
     deps += [ "../:audioproc_debug_proto" ]
   }
@@ -106,6 +112,10 @@ rtc_library("null_aec_dump_factory") {
   deps = [
     ":aec_dump",
     "..:aec_dump_interface",
+    "../../../api/task_queue",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/base:nullability",
+    "//third_party/abseil-cpp/absl/strings",
   ]
-  absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
 }
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_factory.h b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_factory.h
index 20718c3d7f..0d258a9ebc 100644
--- a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_factory.h
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_factory.h
@@ -13,34 +13,34 @@
 
 #include <memory>
 
+#include "absl/base/nullability.h"
 #include "absl/strings/string_view.h"
+#include "api/task_queue/task_queue_base.h"
 #include "modules/audio_processing/include/aec_dump.h"
 #include "rtc_base/system/file_wrapper.h"
 #include "rtc_base/system/rtc_export.h"
 
-namespace rtc {
-class TaskQueue;
-}  // namespace rtc
-
 namespace webrtc {
 
 class RTC_EXPORT AecDumpFactory {
  public:
-  // The `worker_queue` may not be null and must outlive the created
-  // AecDump instance. `max_log_size_bytes == -1` means the log size
-  // will be unlimited. `handle` may not be null. The AecDump takes
-  // responsibility for `handle` and closes it in the destructor. A
-  // non-null return value indicates that the file has been
+  // The `worker_queue` must outlive the created AecDump instance.
+  // `max_log_size_bytes == -1` means the log size will be unlimited.
+  // The AecDump takes responsibility for `handle` and closes it in the
+  // destructor. A non-null return value indicates that the file has been
   // sucessfully opened.
-  static std::unique_ptr<AecDump> Create(webrtc::FileWrapper file,
-                                         int64_t max_log_size_bytes,
-                                         rtc::TaskQueue* worker_queue);
-  static std::unique_ptr<AecDump> Create(absl::string_view file_name,
-                                         int64_t max_log_size_bytes,
-                                         rtc::TaskQueue* worker_queue);
-  static std::unique_ptr<AecDump> Create(FILE* handle,
-                                         int64_t max_log_size_bytes,
-                                         rtc::TaskQueue* worker_queue);
+  static absl::Nullable<std::unique_ptr<AecDump>> Create(
+      FileWrapper file,
+      int64_t max_log_size_bytes,
+      absl::Nonnull<TaskQueueBase*> worker_queue);
+  static absl::Nullable<std::unique_ptr<AecDump>> Create(
+      absl::string_view file_name,
+      int64_t max_log_size_bytes,
+      absl::Nonnull<TaskQueueBase*> worker_queue);
+  static absl::Nullable<std::unique_ptr<AecDump>> Create(
+      absl::Nonnull<FILE*> handle,
+      int64_t max_log_size_bytes,
+      absl::Nonnull<TaskQueueBase*> worker_queue);
 };
 
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc
index 94c24048e0..8484fcc6e1 100644
--- a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc
@@ -13,11 +13,12 @@
 #include <memory>
 #include <utility>
 
+#include "absl/base/nullability.h"
 #include "absl/strings/string_view.h"
+#include "api/task_queue/task_queue_base.h"
 #include "modules/audio_processing/aec_dump/aec_dump_factory.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/event.h"
-#include "rtc_base/task_queue.h"
 
 namespace webrtc {
 
@@ -59,7 +60,7 @@ void CopyFromConfigToEvent(const webrtc::InternalAPMConfig& config,
 
 AecDumpImpl::AecDumpImpl(FileWrapper debug_file,
                          int64_t max_log_size_bytes,
-                         rtc::TaskQueue* worker_queue)
+                         absl::Nonnull<TaskQueueBase*> worker_queue)
     : debug_file_(std::move(debug_file)),
       num_bytes_left_for_log_(max_log_size_bytes),
       worker_queue_(worker_queue) {}
@@ -254,9 +255,10 @@ void AecDumpImpl::PostWriteToFileTask(std::unique_ptr<audioproc::Event> event) {
   });
 }
 
-std::unique_ptr<AecDump> AecDumpFactory::Create(webrtc::FileWrapper file,
-                                                int64_t max_log_size_bytes,
-                                                rtc::TaskQueue* worker_queue) {
+absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
+    FileWrapper file,
+    int64_t max_log_size_bytes,
+    absl::Nonnull<TaskQueueBase*> worker_queue) {
   RTC_DCHECK(worker_queue);
   if (!file.is_open())
     return nullptr;
@@ -265,16 +267,18 @@ std::unique_ptr<AecDump> AecDumpFactory::Create(webrtc::FileWrapper file,
                                        worker_queue);
 }
 
-std::unique_ptr<AecDump> AecDumpFactory::Create(absl::string_view file_name,
-                                                int64_t max_log_size_bytes,
-                                                rtc::TaskQueue* worker_queue) {
+absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
+    absl::string_view file_name,
+    int64_t max_log_size_bytes,
+    absl::Nonnull<TaskQueueBase*> worker_queue) {
   return Create(FileWrapper::OpenWriteOnly(file_name), max_log_size_bytes,
                 worker_queue);
 }
 
-std::unique_ptr<AecDump> AecDumpFactory::Create(FILE* handle,
-                                                int64_t max_log_size_bytes,
-                                                rtc::TaskQueue* worker_queue) {
+absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
+    absl::Nonnull<FILE*> handle,
+    int64_t max_log_size_bytes,
+    absl::Nonnull<TaskQueueBase*> worker_queue) {
   return Create(FileWrapper(handle), max_log_size_bytes, worker_queue);
 }
 
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h
index 429808f9af..d5af31b01e 100644
--- a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_impl.h
@@ -15,11 +15,11 @@
 #include <string>
 #include <vector>
 
+#include "api/task_queue/task_queue_base.h"
 #include "modules/audio_processing/aec_dump/capture_stream_info.h"
 #include "modules/audio_processing/include/aec_dump.h"
 #include "rtc_base/race_checker.h"
 #include "rtc_base/system/file_wrapper.h"
-#include "rtc_base/task_queue.h"
 #include "rtc_base/thread_annotations.h"
 
 // Files generated at build-time by the protobuf compiler.
@@ -39,7 +39,7 @@ class AecDumpImpl : public AecDump {
   // `max_log_size_bytes == -1` means the log size will be unlimited.
   AecDumpImpl(FileWrapper debug_file,
               int64_t max_log_size_bytes,
-              rtc::TaskQueue* worker_queue);
+              absl::Nonnull<TaskQueueBase*> worker_queue);
   AecDumpImpl(const AecDumpImpl&) = delete;
   AecDumpImpl& operator=(const AecDumpImpl&) = delete;
   ~AecDumpImpl() override;
@@ -74,7 +74,7 @@ class AecDumpImpl : public AecDump {
   FileWrapper debug_file_;
   int64_t num_bytes_left_for_log_ = 0;
   rtc::RaceChecker race_checker_;
-  rtc::TaskQueue* worker_queue_;
+  absl::Nonnull<TaskQueueBase*> worker_queue_;
   CaptureStreamInfo capture_stream_info_;
 };
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc
index 62f896fe14..2a8110c4fc 100644
--- a/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc
@@ -28,7 +28,7 @@ TEST(AecDumper, APICallsDoNotCrash) {
 
   {
     std::unique_ptr<webrtc::AecDump> aec_dump =
-        webrtc::AecDumpFactory::Create(filename, -1, &file_writer_queue);
+        webrtc::AecDumpFactory::Create(filename, -1, file_writer_queue.Get());
 
     constexpr int kNumChannels = 1;
     constexpr int kNumSamplesPerChannel = 160;
@@ -63,7 +63,7 @@ TEST(AecDumper, WriteToFile) {
 
   {
     std::unique_ptr<webrtc::AecDump> aec_dump =
-        webrtc::AecDumpFactory::Create(filename, -1, &file_writer_queue);
+        webrtc::AecDumpFactory::Create(filename, -1, file_writer_queue.Get());
 
     constexpr int kNumChannels = 1;
     constexpr int kNumSamplesPerChannel = 160;
diff --git a/third_party/libwebrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc b/third_party/libwebrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc
index 9bd9745069..63929afac4 100644
--- a/third_party/libwebrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc
+++ b/third_party/libwebrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc
@@ -8,27 +8,32 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include "absl/base/nullability.h"
 #include "absl/strings/string_view.h"
+#include "api/task_queue/task_queue_base.h"
 #include "modules/audio_processing/aec_dump/aec_dump_factory.h"
 #include "modules/audio_processing/include/aec_dump.h"
 
 namespace webrtc {
 
-std::unique_ptr<AecDump> AecDumpFactory::Create(webrtc::FileWrapper file,
-                                                int64_t max_log_size_bytes,
-                                                rtc::TaskQueue* worker_queue) {
+absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
+    FileWrapper file,
+    int64_t max_log_size_bytes,
+    absl::Nonnull<TaskQueueBase*> worker_queue) {
   return nullptr;
 }
 
-std::unique_ptr<AecDump> AecDumpFactory::Create(absl::string_view file_name,
-                                                int64_t max_log_size_bytes,
-                                                rtc::TaskQueue* worker_queue) {
+absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
+    absl::string_view file_name,
+    int64_t max_log_size_bytes,
+    absl::Nonnull<TaskQueueBase*> worker_queue) {
   return nullptr;
 }
 
-std::unique_ptr<AecDump> AecDumpFactory::Create(FILE* handle,
-                                                int64_t max_log_size_bytes,
-                                                rtc::TaskQueue* worker_queue) {
+absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
+    absl::Nonnull<FILE*> handle,
+    int64_t max_log_size_bytes,
+    absl::Nonnull<TaskQueueBase*> worker_queue) {
   return nullptr;
 }
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc
index c304453388..4ac074526c 100644
--- a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc
+++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.cc
@@ -18,11 +18,13 @@
 #include <type_traits>
 #include <utility>
 
+#include "absl/base/nullability.h"
 #include "absl/strings/match.h"
 #include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "api/audio/audio_frame.h"
+#include "api/task_queue/task_queue_base.h"
 #include "common_audio/audio_converter.h"
 #include "common_audio/include/audio_util.h"
 #include "modules/audio_processing/aec_dump/aec_dump_factory.h"
@@ -2083,9 +2085,10 @@ void AudioProcessingImpl::UpdateRecommendedInputVolumeLocked() {
   capture_.recommended_input_volume = capture_.applied_input_volume;
 }
 
-bool AudioProcessingImpl::CreateAndAttachAecDump(absl::string_view file_name,
-                                                 int64_t max_log_size_bytes,
-                                                 rtc::TaskQueue* worker_queue) {
+bool AudioProcessingImpl::CreateAndAttachAecDump(
+    absl::string_view file_name,
+    int64_t max_log_size_bytes,
+    absl::Nonnull<TaskQueueBase*> worker_queue) {
   std::unique_ptr<AecDump> aec_dump =
       AecDumpFactory::Create(file_name, max_log_size_bytes, worker_queue);
   if (!aec_dump) {
@@ -2096,9 +2099,10 @@ bool AudioProcessingImpl::CreateAndAttachAecDump(absl::string_view file_name,
   return true;
 }
 
-bool AudioProcessingImpl::CreateAndAttachAecDump(FILE* handle,
-                                                 int64_t max_log_size_bytes,
-                                                 rtc::TaskQueue* worker_queue) {
+bool AudioProcessingImpl::CreateAndAttachAecDump(
+    FILE* handle,
+    int64_t max_log_size_bytes,
+    absl::Nonnull<TaskQueueBase*> worker_queue) {
   std::unique_ptr<AecDump> aec_dump =
       AecDumpFactory::Create(handle, max_log_size_bytes, worker_queue);
   if (!aec_dump) {
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.h b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.h
index 1e058b5a32..2c0ab198db 100644
--- a/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.h
+++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_impl.h
@@ -19,10 +19,12 @@
 #include <string>
 #include <vector>
 
+#include "absl/base/nullability.h"
 #include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
 #include "api/array_view.h"
 #include "api/function_view.h"
+#include "api/task_queue/task_queue_base.h"
 #include "modules/audio_processing/aec3/echo_canceller3.h"
 #include "modules/audio_processing/agc/agc_manager_direct.h"
 #include "modules/audio_processing/agc/gain_control.h"
@@ -71,12 +73,14 @@ class AudioProcessingImpl : public AudioProcessing {
   int Initialize() override;
   int Initialize(const ProcessingConfig& processing_config) override;
   void ApplyConfig(const AudioProcessing::Config& config) override;
-  bool CreateAndAttachAecDump(absl::string_view file_name,
-                              int64_t max_log_size_bytes,
-                              rtc::TaskQueue* worker_queue) override;
-  bool CreateAndAttachAecDump(FILE* handle,
-                              int64_t max_log_size_bytes,
-                              rtc::TaskQueue* worker_queue) override;
+  bool CreateAndAttachAecDump(
+      absl::string_view file_name,
+      int64_t max_log_size_bytes,
+      absl::Nonnull<TaskQueueBase*> worker_queue) override;
+  bool CreateAndAttachAecDump(
+      FILE* handle,
+      int64_t max_log_size_bytes,
+      absl::Nonnull<TaskQueueBase*> worker_queue) override;
   // TODO(webrtc:5298) Deprecated variant.
   void AttachAecDump(std::unique_ptr<AecDump> aec_dump) override;
   void DetachAecDump() override;
diff --git a/third_party/libwebrtc/modules/audio_processing/audio_processing_unittest.cc b/third_party/libwebrtc/modules/audio_processing/audio_processing_unittest.cc
index c2bedb2da4..2d3684e9b5 100644
--- a/third_party/libwebrtc/modules/audio_processing/audio_processing_unittest.cc
+++ b/third_party/libwebrtc/modules/audio_processing/audio_processing_unittest.cc
@@ -1485,8 +1485,8 @@ void ApmTest::ProcessDebugDump(absl::string_view in_filename,
       if (first_init) {
         // AttachAecDump() writes an additional init message. Don't start
         // recording until after the first init to avoid the extra message.
-        auto aec_dump =
-            AecDumpFactory::Create(out_filename, max_size_bytes, &worker_queue);
+        auto aec_dump = AecDumpFactory::Create(out_filename, max_size_bytes,
+                                               worker_queue.Get());
         EXPECT_TRUE(aec_dump);
         apm_->AttachAecDump(std::move(aec_dump));
         first_init = false;
@@ -1632,7 +1632,7 @@ TEST_F(ApmTest, DebugDump) {
   const std::string filename =
       test::TempFilename(test::OutputPath(), "debug_aec");
   {
-    auto aec_dump = AecDumpFactory::Create("", -1, &worker_queue);
+    auto aec_dump = AecDumpFactory::Create("", -1, worker_queue.Get());
     EXPECT_FALSE(aec_dump);
   }
 
@@ -1640,7 +1640,7 @@ TEST_F(ApmTest, DebugDump) {
   // Stopping without having started should be OK.
   apm_->DetachAecDump();
 
-  auto aec_dump = AecDumpFactory::Create(filename, -1, &worker_queue);
+  auto aec_dump = AecDumpFactory::Create(filename, -1, worker_queue.Get());
   EXPECT_TRUE(aec_dump);
   apm_->AttachAecDump(std::move(aec_dump));
   EXPECT_EQ(apm_->kNoError,
@@ -1683,7 +1683,7 @@ TEST_F(ApmTest, DebugDumpFromFileHandle) {
   // Stopping without having started should be OK.
   apm_->DetachAecDump();
 
-  auto aec_dump = AecDumpFactory::Create(std::move(f), -1, &worker_queue);
+  auto aec_dump = AecDumpFactory::Create(std::move(f), -1, worker_queue.Get());
   EXPECT_TRUE(aec_dump);
   apm_->AttachAecDump(std::move(aec_dump));
   EXPECT_EQ(apm_->kNoError,
diff --git a/third_party/libwebrtc/modules/audio_processing/include/audio_processing.h b/third_party/libwebrtc/modules/audio_processing/include/audio_processing.h
index e3223513af..dd484be4f1 100644
--- a/third_party/libwebrtc/modules/audio_processing/include/audio_processing.h
+++ b/third_party/libwebrtc/modules/audio_processing/include/audio_processing.h
@@ -23,6 +23,7 @@
 
 #include <vector>
 
+#include "absl/base/nullability.h"
 #include "absl/strings/string_view.h"
 #include "absl/types/optional.h"
 #include "api/array_view.h"
@@ -30,15 +31,12 @@
 #include "api/audio/echo_control.h"
 #include "api/ref_count.h"
 #include "api/scoped_refptr.h"
+#include "api/task_queue/task_queue_base.h"
 #include "modules/audio_processing/include/audio_processing_statistics.h"
 #include "rtc_base/arraysize.h"
 #include "rtc_base/system/file_wrapper.h"
 #include "rtc_base/system/rtc_export.h"
 
-namespace rtc {
-class TaskQueue;
-}  // namespace rtc
-
 namespace webrtc {
 
 class AecDump;
@@ -632,12 +630,14 @@ class RTC_EXPORT AudioProcessing : public RefCountInterface {
   // return value of true indicates that the file has been
   // sucessfully opened, while a value of false indicates that
   // opening the file failed.
-  virtual bool CreateAndAttachAecDump(absl::string_view file_name,
-                                      int64_t max_log_size_bytes,
-                                      rtc::TaskQueue* worker_queue) = 0;
-  virtual bool CreateAndAttachAecDump(FILE* handle,
-                                      int64_t max_log_size_bytes,
-                                      rtc::TaskQueue* worker_queue) = 0;
+  virtual bool CreateAndAttachAecDump(
+      absl::string_view file_name,
+      int64_t max_log_size_bytes,
+      absl::Nonnull<TaskQueueBase*> worker_queue) = 0;
+  virtual bool CreateAndAttachAecDump(
+      absl::Nonnull<FILE*> handle,
+      int64_t max_log_size_bytes,
+      absl::Nonnull<TaskQueueBase*> worker_queue) = 0;
 
   // TODO(webrtc:5298) Deprecated variant.
   // Attaches provided webrtc::AecDump for recording debugging
diff --git a/third_party/libwebrtc/modules/audio_processing/include/mock_audio_processing.h b/third_party/libwebrtc/modules/audio_processing/include/mock_audio_processing.h
index 2ea1a865c3..dfe7d84e07 100644
--- a/third_party/libwebrtc/modules/audio_processing/include/mock_audio_processing.h
+++ b/third_party/libwebrtc/modules/audio_processing/include/mock_audio_processing.h
@@ -13,7 +13,9 @@
 
 #include <memory>
 
+#include "absl/base/nullability.h"
 #include "absl/strings/string_view.h"
+#include "api/task_queue/task_queue_base.h"
 #include "modules/audio_processing/include/aec_dump.h"
 #include "modules/audio_processing/include/audio_processing.h"
 #include "modules/audio_processing/include/audio_processing_statistics.h"
@@ -155,13 +157,13 @@ class MockAudioProcessing : public AudioProcessing {
               CreateAndAttachAecDump,
               (absl::string_view file_name,
                int64_t max_log_size_bytes,
-               rtc::TaskQueue* worker_queue),
+               absl::Nonnull<TaskQueueBase*> worker_queue),
               (override));
   MOCK_METHOD(bool,
               CreateAndAttachAecDump,
               (FILE * handle,
                int64_t max_log_size_bytes,
-               rtc::TaskQueue* worker_queue),
+               absl::Nonnull<TaskQueueBase*> worker_queue),
               (override));
   MOCK_METHOD(void, AttachAecDump, (std::unique_ptr<AecDump>), (override));
   MOCK_METHOD(void, DetachAecDump, (), (override));
diff --git a/third_party/libwebrtc/modules/audio_processing/test/audio_processing_simulator.cc b/third_party/libwebrtc/modules/audio_processing/test/audio_processing_simulator.cc
index 7bd6da0133..500005f26a 100644
--- a/third_party/libwebrtc/modules/audio_processing/test/audio_processing_simulator.cc
+++ b/third_party/libwebrtc/modules/audio_processing/test/audio_processing_simulator.cc
@@ -622,7 +622,7 @@ void AudioProcessingSimulator::ConfigureAudioProcessor() {
 
   if (settings_.aec_dump_output_filename) {
     ap_->AttachAecDump(AecDumpFactory::Create(
-        *settings_.aec_dump_output_filename, -1, &worker_queue_));
+        *settings_.aec_dump_output_filename, -1, worker_queue_.Get()));
   }
 }
 
diff --git a/third_party/libwebrtc/modules/audio_processing/test/debug_dump_test.cc b/third_party/libwebrtc/modules/audio_processing/test/debug_dump_test.cc
index cded5de217..0d3eefa94a 100644
--- a/third_party/libwebrtc/modules/audio_processing/test/debug_dump_test.cc
+++ b/third_party/libwebrtc/modules/audio_processing/test/debug_dump_test.cc
@@ -197,7 +197,7 @@ void DebugDumpGenerator::SetOutputChannels(int channels) {
 
 void DebugDumpGenerator::StartRecording() {
   apm_->AttachAecDump(
-      AecDumpFactory::Create(dump_file_name_.c_str(), -1, &worker_queue_));
+      AecDumpFactory::Create(dump_file_name_.c_str(), -1, worker_queue_.Get()));
 }
 
 void DebugDumpGenerator::Process(size_t num_blocks) {
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
index 16bd4153a6..e56aa4a09c 100644
--- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
@@ -26,6 +26,7 @@
 #include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
 #include "rtc_base/checks.h"
 #include "test/field_trial.h"
+#include "test/gtest.h"
 
 namespace webrtc {
 constexpr size_t kMtu = 1200;
@@ -52,6 +53,7 @@ RtpStream::RtpStream(int fps, int bitrate_bps)
 // previous frame, no frame will be generated. The frame is split into
 // packets.
 int64_t RtpStream::GenerateFrame(int64_t time_now_us,
+                                 int64_t* next_sequence_number,
                                  std::vector<PacketResult>* packets) {
   if (time_now_us < next_rtp_time_) {
     return next_rtp_time_;
@@ -66,6 +68,7 @@ int64_t RtpStream::GenerateFrame(int64_t time_now_us,
     packet.sent_packet.send_time =
         Timestamp::Micros(time_now_us + kSendSideOffsetUs);
     packet.sent_packet.size = DataSize::Bytes(payload_size);
+    packet.sent_packet.sequence_number = (*next_sequence_number)++;
     packets->push_back(packet);
   }
   next_rtp_time_ = time_now_us + (1000000 + fps_ / 2) / fps_;
@@ -131,14 +134,15 @@ void StreamGenerator::SetBitrateBps(int bitrate_bps) {
 
 // TODO(holmer): Break out the channel simulation part from this class to make
 // it possible to simulate different types of channels.
-int64_t StreamGenerator::GenerateFrame(std::vector<PacketResult>* packets,
-                                       int64_t time_now_us) {
+int64_t StreamGenerator::GenerateFrame(int64_t time_now_us,
+                                       int64_t* next_sequence_number,
+                                       std::vector<PacketResult>* packets) {
   RTC_CHECK(packets != NULL);
   RTC_CHECK(packets->empty());
   RTC_CHECK_GT(capacity_, 0);
   auto it =
       std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
-  (*it)->GenerateFrame(time_now_us, packets);
+  (*it)->GenerateFrame(time_now_us, next_sequence_number, packets);
   for (PacketResult& packet : *packets) {
     int capacity_bpus = capacity_ / 1000;
     int64_t required_network_time_us =
@@ -233,8 +237,8 @@ bool DelayBasedBweTest::GenerateAndProcessFrame(uint32_t ssrc,
   stream_generator_->SetBitrateBps(bitrate_bps);
   std::vector<PacketResult> packets;
 
-  int64_t next_time_us =
-      stream_generator_->GenerateFrame(&packets, clock_.TimeInMicroseconds());
+  int64_t next_time_us = stream_generator_->GenerateFrame(
+      clock_.TimeInMicroseconds(), &next_sequence_number_, &packets);
   if (packets.empty())
     return false;
 
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h
index 89eb1a353f..634e6a4d82 100644
--- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h
@@ -15,12 +15,11 @@
 #include <stdint.h>
 
 #include <memory>
-#include <string>
 #include <vector>
 
-#include "absl/strings/string_view.h"
 #include "api/transport/field_trial_based_config.h"
 #include "api/transport/network_types.h"
+#include "api/units/timestamp.h"
 #include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
 #include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
 #include "system_wrappers/include/clock.h"
@@ -61,6 +60,7 @@ class RtpStream {
   // previous frame, no frame will be generated. The frame is split into
   // packets.
   int64_t GenerateFrame(int64_t time_now_us,
+                        int64_t* next_sequence_number,
                         std::vector<PacketResult>* packets);
 
   // The send-side time when the next frame can be generated.
@@ -102,8 +102,9 @@ class StreamGenerator {
 
   // TODO(holmer): Break out the channel simulation part from this class to make
   // it possible to simulate different types of channels.
-  int64_t GenerateFrame(std::vector<PacketResult>* packets,
-                        int64_t time_now_us);
+  int64_t GenerateFrame(int64_t time_now_us,
+                        int64_t* next_sequence_number,
+                        std::vector<PacketResult>* packets);
 
  private:
   // Capacity of the simulated channel in bits per second.
diff --git a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc
index 22693d67e9..211d86c95d 100644
--- a/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc
+++ b/third_party/libwebrtc/modules/congestion_controller/goog_cc/send_side_bandwidth_estimation.cc
@@ -281,7 +281,8 @@ void SendSideBandwidthEstimation::OnRouteChange() {
   uma_update_state_ = kNoUpdate;
   uma_rtt_state_ = kNoUpdate;
   last_rtc_event_log_ = Timestamp::MinusInfinity();
-  if (loss_based_bandwidth_estimator_v2_->UseInStartPhase()) {
+  if (LossBasedBandwidthEstimatorV2Enabled() &&
+      loss_based_bandwidth_estimator_v2_->UseInStartPhase()) {
     loss_based_bandwidth_estimator_v2_.reset(
         new LossBasedBweV2(key_value_config_));
   }
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/BUILD.gn b/third_party/libwebrtc/modules/congestion_controller/rtp/BUILD.gn
index cd13332b7f..8ec755e1aa 100644
--- a/third_party/libwebrtc/modules/congestion_controller/rtp/BUILD.gn
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/BUILD.gn
@@ -34,7 +34,6 @@ rtc_library("control_handler") {
     "../../../rtc_base:safe_conversions",
     "../../../rtc_base:safe_minmax",
     "../../../rtc_base/system:no_unique_address",
-    "../../../system_wrappers:field_trial",
     "../../pacing",
   ]
   absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.cc b/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.cc
index da6451c97e..357a0f0798 100644
--- a/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.cc
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.cc
@@ -18,21 +18,8 @@
 #include "rtc_base/logging.h"
 #include "rtc_base/numerics/safe_conversions.h"
 #include "rtc_base/numerics/safe_minmax.h"
-#include "system_wrappers/include/field_trial.h"
 
 namespace webrtc {
-namespace {
-
-// By default, pacer emergency stops encoder when buffer reaches a high level.
-bool IsPacerEmergencyStopDisabled() {
-  return field_trial::IsEnabled("WebRTC-DisablePacerEmergencyStop");
-}
-
-}  // namespace
-CongestionControlHandler::CongestionControlHandler()
-    : disable_pacer_emergency_stop_(IsPacerEmergencyStopDisabled()) {}
-
-CongestionControlHandler::~CongestionControlHandler() {}
 
 void CongestionControlHandler::SetTargetRate(
     TargetTransferRate new_target_rate) {
@@ -60,9 +47,8 @@ absl::optional<TargetTransferRate> CongestionControlHandler::GetUpdate() {
   bool pause_encoding = false;
   if (!network_available_) {
     pause_encoding = true;
-  } else if (!disable_pacer_emergency_stop_ &&
-             pacer_expected_queue_ms_ >
-                 PacingController::kMaxExpectedQueueLength.ms()) {
+  } else if (pacer_expected_queue_ms_ >
+             PacingController::kMaxExpectedQueueLength.ms()) {
     pause_encoding = true;
   }
   if (pause_encoding)
diff --git a/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.h b/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.h
index d8e7263a02..649d2f911e 100644
--- a/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.h
+++ b/third_party/libwebrtc/modules/congestion_controller/rtp/control_handler.h
@@ -28,12 +28,13 @@ namespace webrtc {
 // destruction unless members are properly ordered.
 class CongestionControlHandler {
  public:
-  CongestionControlHandler();
-  ~CongestionControlHandler();
+  CongestionControlHandler() = default;
 
   CongestionControlHandler(const CongestionControlHandler&) = delete;
   CongestionControlHandler& operator=(const CongestionControlHandler&) = delete;
 
+  ~CongestionControlHandler() = default;
+
   void SetTargetRate(TargetTransferRate new_target_rate);
   void SetNetworkAvailability(bool network_available);
   void SetPacerQueue(TimeDelta expected_queue_time);
@@ -45,7 +46,6 @@ class CongestionControlHandler {
   bool network_available_ = true;
   bool encoder_paused_in_last_report_ = false;
 
-  const bool disable_pacer_emergency_stop_;
   int64_t pacer_expected_queue_ms_ = 0;
 
   RTC_NO_UNIQUE_ADDRESS SequenceChecker sequenced_checker_;
diff --git a/third_party/libwebrtc/modules/pacing/BUILD.gn b/third_party/libwebrtc/modules/pacing/BUILD.gn
index ea80c8c819..87498817b6 100644
--- a/third_party/libwebrtc/modules/pacing/BUILD.gn
+++ b/third_party/libwebrtc/modules/pacing/BUILD.gn
@@ -63,6 +63,7 @@ rtc_library("pacing") {
   ]
   absl_deps = [
     "//third_party/abseil-cpp/absl/cleanup",
+    "//third_party/abseil-cpp/absl/container:inlined_vector",
     "//third_party/abseil-cpp/absl/memory",
     "//third_party/abseil-cpp/absl/strings",
     "//third_party/abseil-cpp/absl/types:optional",
diff --git a/third_party/libwebrtc/modules/pacing/pacing_controller.cc b/third_party/libwebrtc/modules/pacing/pacing_controller.cc
index 5b81207d56..41f97a37fb 100644
--- a/third_party/libwebrtc/modules/pacing/pacing_controller.cc
+++ b/third_party/libwebrtc/modules/pacing/pacing_controller.cc
@@ -19,11 +19,11 @@
 #include "absl/strings/match.h"
 #include "api/units/data_size.h"
 #include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
 #include "modules/pacing/bitrate_prober.h"
-#include "modules/pacing/interval_budget.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
-#include "rtc_base/time_utils.h"
 #include "system_wrappers/include/clock.h"
 
 namespace webrtc {
@@ -44,8 +44,6 @@ bool IsEnabled(const FieldTrialsView& field_trials, absl::string_view key) {
 
 }  // namespace
 
-const TimeDelta PacingController::kMaxExpectedQueueLength =
-    TimeDelta::Millis(2000);
 const TimeDelta PacingController::kPausedProcessInterval =
     kCongestedPacketInterval;
 const TimeDelta PacingController::kMinSleepTime = TimeDelta::Millis(1);
@@ -57,11 +55,13 @@ const TimeDelta PacingController::kMaxEarlyProbeProcessing =
 
 PacingController::PacingController(Clock* clock,
                                    PacketSender* packet_sender,
-                                   const FieldTrialsView& field_trials)
+                                   const FieldTrialsView& field_trials,
+                                   Configuration configuration)
     : clock_(clock),
       packet_sender_(packet_sender),
       field_trials_(field_trials),
       drain_large_queues_(
+          configuration.drain_large_queues &&
           !IsDisabled(field_trials_, "WebRTC-Pacer-DrainQueue")),
       send_padding_if_silent_(
           IsEnabled(field_trials_, "WebRTC-Pacer-PadInSilence")),
@@ -71,9 +71,10 @@ PacingController::PacingController(Clock* clock,
       fast_retransmissions_(
           IsEnabled(field_trials_, "WebRTC-Pacer-FastRetransmissions")),
       keyframe_flushing_(
+          configuration.keyframe_flushing ||
           IsEnabled(field_trials_, "WebRTC-Pacer-KeyframeFlushing")),
       transport_overhead_per_packet_(DataSize::Zero()),
-      send_burst_interval_(kDefaultBurstInterval),
+      send_burst_interval_(configuration.send_burst_interval),
       last_timestamp_(clock_->CurrentTime()),
       paused_(false),
       media_debt_(DataSize::Zero()),
@@ -86,9 +87,11 @@ PacingController::PacingController(Clock* clock,
       last_process_time_(clock->CurrentTime()),
       last_send_time_(last_process_time_),
       seen_first_packet_(false),
-      packet_queue_(/*creation_time=*/last_process_time_),
+      packet_queue_(/*creation_time=*/last_process_time_,
+                    configuration.prioritize_audio_retransmission,
+                    configuration.packet_queue_ttl),
       congested_(false),
-      queue_time_limit_(kMaxExpectedQueueLength),
+      queue_time_limit_(configuration.queue_time_limit),
       account_for_audio_(false),
       include_overhead_(false),
       circuit_breaker_threshold_(1 << 16) {
@@ -710,8 +713,7 @@ Timestamp PacingController::NextUnpacedSendTime() const {
   }
   if (fast_retransmissions_) {
     Timestamp leading_retransmission_send_time =
-        packet_queue_.LeadingPacketEnqueueTime(
-            RtpPacketMediaType::kRetransmission);
+        packet_queue_.LeadingPacketEnqueueTimeForRetransmission();
     if (leading_retransmission_send_time.IsFinite()) {
       return leading_retransmission_send_time;
     }
diff --git a/third_party/libwebrtc/modules/pacing/pacing_controller.h b/third_party/libwebrtc/modules/pacing/pacing_controller.h
index 04e0a820f9..fe6ee737a9 100644
--- a/third_party/libwebrtc/modules/pacing/pacing_controller.h
+++ b/third_party/libwebrtc/modules/pacing/pacing_controller.h
@@ -67,11 +67,6 @@ class PacingController {
     }
   };
 
-  // Expected max pacer delay. If ExpectedQueueTime() is higher than
-  // this value, the packet producers should wait (eg drop frames rather than
-  // encoding them). Bitrate sent may temporarily exceed target set by
-  // UpdateBitrate() so that this limit will be upheld.
-  static const TimeDelta kMaxExpectedQueueLength;
   // If no media or paused, wake up at least every `kPausedProcessIntervalMs` in
   // order to send a keep-alive packet so we don't get stuck in a bad state due
   // to lack of feedback.
@@ -93,14 +88,45 @@ class PacingController {
   // the send burst interval.
   // Ex: max send burst interval = 63Kb / 10Mbit/s = 50ms.
   static constexpr DataSize kMaxBurstSize = DataSize::Bytes(63 * 1000);
-  // The pacer is allowed to send enqued packets in bursts and can build up a
-  // packet "debt" that correspond to approximately the send rate during
-  // the burst interval.
+
+  // Configuration default values.
   static constexpr TimeDelta kDefaultBurstInterval = TimeDelta::Millis(40);
+  static constexpr TimeDelta kMaxExpectedQueueLength = TimeDelta::Millis(2000);
+
+  struct Configuration {
+    // If the pacer queue grows longer than the configured max queue limit,
+    // pacer sends at the minimum rate needed to keep the max queue limit and
+    // ignore the current bandwidth estimate.
+    bool drain_large_queues = true;
+    // Expected max pacer delay. If ExpectedQueueTime() is higher than
+    // this value, the packet producers should wait (eg drop frames rather than
+    // encoding them). Bitrate sent may temporarily exceed target set by
+    // SetPacingRates() so that this limit will be upheld if
+    // `drain_large_queues` is set.
+    TimeDelta queue_time_limit = kMaxExpectedQueueLength;
+    // If the first packet of a keyframe is enqueued on a RTP stream, pacer
+    // skips forward to that packet and drops other enqueued packets on that
+    // stream, unless a keyframe is already being paced.
+    bool keyframe_flushing = false;
+    // Audio retransmission is prioritized before video retransmission packets.
+    bool prioritize_audio_retransmission = false;
+    // Configure separate timeouts per priority. After a timeout, a packet of
+    // that sort will not be paced and instead dropped.
+    // Note: to set TTL on audio retransmission,
+    // `prioritize_audio_retransmission` must be true.
+    PacketQueueTTL packet_queue_ttl;
+    // The pacer is allowed to send enqueued packets in bursts and can build up
+    // a packet "debt" that correspond to approximately the send rate during the
+    // burst interval.
+    TimeDelta send_burst_interval = kDefaultBurstInterval;
+  };
+
+  static Configuration DefaultConfiguration() { return Configuration{}; }
 
   PacingController(Clock* clock,
                    PacketSender* packet_sender,
-                   const FieldTrialsView& field_trials);
+                   const FieldTrialsView& field_trials,
+                   Configuration configuration = DefaultConfiguration());
 
   ~PacingController();
 
diff --git a/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc b/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc
index 9e6ede6dc0..2c3a71b369 100644
--- a/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc
+++ b/third_party/libwebrtc/modules/pacing/pacing_controller_unittest.cc
@@ -2348,5 +2348,43 @@ TEST_F(PacingControllerTest, FlushesPacketsOnKeyFrames) {
   pacer->ProcessPackets();
 }
 
+TEST_F(PacingControllerTest, CanControlQueueSizeUsingTtl) {
+  const uint32_t kSsrc = 12345;
+  const uint32_t kAudioSsrc = 2345;
+  uint16_t sequence_number = 1234;
+
+  PacingController::Configuration config;
+  config.drain_large_queues = false;
+  config.packet_queue_ttl.video = TimeDelta::Millis(500);
+  auto pacer =
+      std::make_unique<PacingController>(&clock_, &callback_, trials_, config);
+  pacer->SetPacingRates(DataRate::BitsPerSec(100'000), DataRate::Zero());
+
+  Timestamp send_time = Timestamp::Zero();
+  for (int i = 0; i < 100; ++i) {
+    // Enqueue a new audio and video frame every 33ms.
+    if (clock_.CurrentTime() - send_time > TimeDelta::Millis(33)) {
+      for (int j = 0; j < 3; ++j) {
+        auto packet = BuildPacket(RtpPacketMediaType::kVideo, kSsrc,
+                                  /*sequence_number=*/++sequence_number,
+                                  /*capture_time_ms=*/2,
+                                  /*size_bytes=*/1000);
+        pacer->EnqueuePacket(std::move(packet));
+      }
+      auto packet = BuildPacket(RtpPacketMediaType::kAudio, kAudioSsrc,
+                                /*sequence_number=*/++sequence_number,
+                                /*capture_time_ms=*/2,
+                                /*size_bytes=*/100);
+      pacer->EnqueuePacket(std::move(packet));
+      send_time = clock_.CurrentTime();
+    }
+
+    EXPECT_LE(clock_.CurrentTime() - pacer->OldestPacketEnqueueTime(),
+              TimeDelta::Millis(500));
+    clock_.AdvanceTime(pacer->NextSendTime() - clock_.CurrentTime());
+    pacer->ProcessPackets();
+  }
+}
+
 }  // namespace
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.cc b/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.cc
index ea211ea683..2d0d829648 100644
--- a/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.cc
+++ b/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.cc
@@ -10,41 +10,70 @@
 
 #include "modules/pacing/prioritized_packet_queue.h"
 
+#include <algorithm>
+#include <array>
 #include <utility>
 
+#include "absl/container/inlined_vector.h"
+#include "absl/types/optional.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
 #include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
 
 namespace webrtc {
 namespace {
 
 constexpr int kAudioPrioLevel = 0;
 
-int GetPriorityForType(RtpPacketMediaType type) {
+int GetPriorityForType(
+    RtpPacketMediaType type,
+    absl::optional<RtpPacketToSend::OriginalType> original_type) {
   // Lower number takes priority over higher.
   switch (type) {
     case RtpPacketMediaType::kAudio:
       // Audio is always prioritized over other packet types.
       return kAudioPrioLevel;
     case RtpPacketMediaType::kRetransmission:
-      // Send retransmissions before new media.
+      // Send retransmissions before new media. If original_type is set, audio
+      // retransmission is prioritized more than video retransmission.
+      if (original_type == RtpPacketToSend::OriginalType::kVideo) {
+        return kAudioPrioLevel + 2;
+      }
       return kAudioPrioLevel + 1;
     case RtpPacketMediaType::kVideo:
     case RtpPacketMediaType::kForwardErrorCorrection:
       // Video has "normal" priority, in the old speak.
       // Send redundancy concurrently to video. If it is delayed it might have a
       // lower chance of being useful.
-      return kAudioPrioLevel + 2;
+      return kAudioPrioLevel + 3;
     case RtpPacketMediaType::kPadding:
       // Packets that are in themselves likely useless, only sent to keep the
       // BWE high.
-      return kAudioPrioLevel + 3;
+      return kAudioPrioLevel + 4;
   }
   RTC_CHECK_NOTREACHED();
 }
 
 }  // namespace
 
+absl::InlinedVector<TimeDelta, PrioritizedPacketQueue::kNumPriorityLevels>
+PrioritizedPacketQueue::ToTtlPerPrio(PacketQueueTTL packet_queue_ttl) {
+  absl::InlinedVector<TimeDelta, PrioritizedPacketQueue::kNumPriorityLevels>
+      ttl_per_prio(kNumPriorityLevels, TimeDelta::PlusInfinity());
+  ttl_per_prio[GetPriorityForType(RtpPacketMediaType::kRetransmission,
+                                  RtpPacketToSend::OriginalType::kAudio)] =
+      packet_queue_ttl.audio_retransmission;
+  ttl_per_prio[GetPriorityForType(RtpPacketMediaType::kRetransmission,
+                                  RtpPacketToSend::OriginalType::kVideo)] =
+      packet_queue_ttl.video_retransmission;
+  ttl_per_prio[GetPriorityForType(RtpPacketMediaType::kVideo, absl::nullopt)] =
+      packet_queue_ttl.video;
+  return ttl_per_prio;
+}
+
 DataSize PrioritizedPacketQueue::QueuedPacket::PacketSize() const {
   return DataSize::Bytes(packet->payload_size() + packet->padding_size());
 }
@@ -109,8 +138,13 @@ PrioritizedPacketQueue::StreamQueue::DequeueAll() {
   return packets_by_prio;
 }
 
-PrioritizedPacketQueue::PrioritizedPacketQueue(Timestamp creation_time)
-    : queue_time_sum_(TimeDelta::Zero()),
+PrioritizedPacketQueue::PrioritizedPacketQueue(
+    Timestamp creation_time,
+    bool prioritize_audio_retransmission,
+    PacketQueueTTL packet_queue_ttl)
+    : prioritize_audio_retransmission_(prioritize_audio_retransmission),
+      time_to_live_per_prio_(ToTtlPerPrio(packet_queue_ttl)),
+      queue_time_sum_(TimeDelta::Zero()),
       pause_time_sum_(TimeDelta::Zero()),
       size_packets_(0),
       size_packets_per_media_type_({}),
@@ -133,7 +167,11 @@ void PrioritizedPacketQueue::Push(Timestamp enqueue_time,
       enqueue_times_.insert(enqueue_times_.end(), enqueue_time);
   RTC_DCHECK(packet->packet_type().has_value());
   RtpPacketMediaType packet_type = packet->packet_type().value();
-  int prio_level = GetPriorityForType(packet_type);
+  int prio_level =
+      GetPriorityForType(packet_type, prioritize_audio_retransmission_
+                                          ? packet->original_packet_type()
+                                          : absl::nullopt);
+  PurgeOldPacketsAtPriorityLevel(prio_level, enqueue_time);
   RTC_DCHECK_GE(prio_level, 0);
   RTC_DCHECK_LT(prio_level, kNumPriorityLevels);
   QueuedPacket queued_packed = {.packet = std::move(packet),
@@ -214,7 +252,8 @@ PrioritizedPacketQueue::SizeInPacketsPerRtpPacketMediaType() const {
 
 Timestamp PrioritizedPacketQueue::LeadingPacketEnqueueTime(
     RtpPacketMediaType type) const {
-  const int priority_level = GetPriorityForType(type);
+  RTC_DCHECK(type != RtpPacketMediaType::kRetransmission);
+  const int priority_level = GetPriorityForType(type, absl::nullopt);
   if (streams_by_prio_[priority_level].empty()) {
     return Timestamp::MinusInfinity();
   }
@@ -222,6 +261,39 @@ Timestamp PrioritizedPacketQueue::LeadingPacketEnqueueTime(
       priority_level);
 }
 
+Timestamp PrioritizedPacketQueue::LeadingPacketEnqueueTimeForRetransmission()
+    const {
+  if (!prioritize_audio_retransmission_) {
+    const int priority_level =
+        GetPriorityForType(RtpPacketMediaType::kRetransmission, absl::nullopt);
+    if (streams_by_prio_[priority_level].empty()) {
+      return Timestamp::PlusInfinity();
+    }
+    return streams_by_prio_[priority_level].front()->LeadingPacketEnqueueTime(
+        priority_level);
+  }
+  const int audio_priority_level =
+      GetPriorityForType(RtpPacketMediaType::kRetransmission,
+                         RtpPacketToSend::OriginalType::kAudio);
+  const int video_priority_level =
+      GetPriorityForType(RtpPacketMediaType::kRetransmission,
+                         RtpPacketToSend::OriginalType::kVideo);
+
+  Timestamp next_audio =
+      streams_by_prio_[audio_priority_level].empty()
+          ? Timestamp::PlusInfinity()
+          : streams_by_prio_[audio_priority_level]
+                .front()
+                ->LeadingPacketEnqueueTime(audio_priority_level);
+  Timestamp next_video =
+      streams_by_prio_[video_priority_level].empty()
+          ? Timestamp::PlusInfinity()
+          : streams_by_prio_[video_priority_level]
+                .front()
+                ->LeadingPacketEnqueueTime(video_priority_level);
+  return std::min(next_audio, next_video);
+}
+
 Timestamp PrioritizedPacketQueue::OldestEnqueueTime() const {
   return enqueue_times_.empty() ? Timestamp::MinusInfinity()
                                 : enqueue_times_.front();
@@ -283,9 +355,6 @@ void PrioritizedPacketQueue::RemovePacketsForSsrc(uint32_t ssrc) {
         // Update the global top prio level if neccessary.
         RTC_DCHECK(streams_by_prio_[i].front() == &queue);
         streams_by_prio_[i].pop_front();
-        if (i == top_active_prio_level_) {
-          MaybeUpdateTopPrioLevel();
-        }
       } else {
         // More than stream had packets at this prio level, filter this one out.
         std::deque<StreamQueue*> filtered_queue;
@@ -298,6 +367,7 @@ void PrioritizedPacketQueue::RemovePacketsForSsrc(uint32_t ssrc) {
       }
     }
   }
+  MaybeUpdateTopPrioLevel();
 }
 
 bool PrioritizedPacketQueue::HasKeyframePackets(uint32_t ssrc) const {
@@ -340,18 +410,53 @@ void PrioritizedPacketQueue::DequeuePacketInternal(QueuedPacket& packet) {
 }
 
 void PrioritizedPacketQueue::MaybeUpdateTopPrioLevel() {
-  if (streams_by_prio_[top_active_prio_level_].empty()) {
-    // No stream queues have packets at this prio level, find top priority
-    // that is not empty.
-    if (size_packets_ == 0) {
-      top_active_prio_level_ = -1;
+  if (top_active_prio_level_ != -1 &&
+      !streams_by_prio_[top_active_prio_level_].empty()) {
+    return;
+  }
+  // No stream queues have packets at top_active_prio_level_, find top priority
+  // that is not empty.
+  for (int i = 0; i < kNumPriorityLevels; ++i) {
+    PurgeOldPacketsAtPriorityLevel(i, last_update_time_);
+    if (!streams_by_prio_[i].empty()) {
+      top_active_prio_level_ = i;
+      break;
+    }
+  }
+  if (size_packets_ == 0) {
+    // There are no packets left to send. Last packet may have been purged. Prio
+    // will change when a new packet is pushed.
+    top_active_prio_level_ = -1;
+  }
+}
+
+void PrioritizedPacketQueue::PurgeOldPacketsAtPriorityLevel(int prio_level,
+                                                            Timestamp now) {
+  RTC_DCHECK(prio_level >= 0 && prio_level < kNumPriorityLevels);
+  TimeDelta time_to_live = time_to_live_per_prio_[prio_level];
+  if (time_to_live.IsInfinite()) {
+    return;
+  }
+
+  std::deque<StreamQueue*>& queues = streams_by_prio_[prio_level];
+  auto iter = queues.begin();
+  while (iter != queues.end()) {
+    StreamQueue* queue_ptr = *iter;
+    while (queue_ptr->HasPacketsAtPrio(prio_level) &&
+           (now - queue_ptr->LeadingPacketEnqueueTime(prio_level)) >
+               time_to_live) {
+      QueuedPacket packet = queue_ptr->DequeuePacket(prio_level);
+      RTC_LOG(LS_INFO) << "Dropping old packet on SSRC: "
+                       << packet.packet->Ssrc()
+                       << " seq:" << packet.packet->SequenceNumber()
+                       << " time in queue:" << (now - packet.enqueue_time).ms()
+                       << " ms";
+      DequeuePacketInternal(packet);
+    }
+    if (!queue_ptr->HasPacketsAtPrio(prio_level)) {
+      iter = queues.erase(iter);
     } else {
-      for (int i = 0; i < kNumPriorityLevels; ++i) {
-        if (!streams_by_prio_[i].empty()) {
-          top_active_prio_level_ = i;
-          break;
-        }
-      }
+      ++iter;
     }
   }
 }
diff --git a/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.h b/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.h
index 935c530027..179ef104fe 100644
--- a/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.h
+++ b/third_party/libwebrtc/modules/pacing/prioritized_packet_queue.h
@@ -18,8 +18,8 @@
 #include <list>
 #include <memory>
 #include <unordered_map>
-#include <vector>
 
+#include "absl/container/inlined_vector.h"
 #include "api/units/data_size.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
@@ -27,9 +27,19 @@
 
 namespace webrtc {
 
+// Describes how long time a packet may stay in the queue before being dropped.
+struct PacketQueueTTL {
+  TimeDelta audio_retransmission = TimeDelta::PlusInfinity();
+  TimeDelta video_retransmission = TimeDelta::PlusInfinity();
+  TimeDelta video = TimeDelta::PlusInfinity();
+};
+
 class PrioritizedPacketQueue {
  public:
-  explicit PrioritizedPacketQueue(Timestamp creation_time);
+  explicit PrioritizedPacketQueue(
+      Timestamp creation_time,
+      bool prioritize_audio_retransmission = false,
+      PacketQueueTTL packet_queue_ttl = PacketQueueTTL());
   PrioritizedPacketQueue(const PrioritizedPacketQueue&) = delete;
   PrioritizedPacketQueue& operator=(const PrioritizedPacketQueue&) = delete;
 
@@ -63,6 +73,7 @@ class PrioritizedPacketQueue {
   // method, for the given packet type. If queue has no packets, of that type,
   // returns Timestamp::MinusInfinity().
   Timestamp LeadingPacketEnqueueTime(RtpPacketMediaType type) const;
+  Timestamp LeadingPacketEnqueueTimeForRetransmission() const;
 
   // Enqueue time of the oldest packet in the queue,
   // Timestamp::MinusInfinity() if queue is empty.
@@ -90,7 +101,7 @@ class PrioritizedPacketQueue {
   bool HasKeyframePackets(uint32_t ssrc) const;
 
  private:
-  static constexpr int kNumPriorityLevels = 4;
+  static constexpr int kNumPriorityLevels = 5;
 
   class QueuedPacket {
    public:
@@ -139,6 +150,15 @@ class PrioritizedPacketQueue {
   // if so move it to the lowest non-empty index.
   void MaybeUpdateTopPrioLevel();
 
+  void PurgeOldPacketsAtPriorityLevel(int prio_level, Timestamp now);
+
+  static absl::InlinedVector<TimeDelta, kNumPriorityLevels> ToTtlPerPrio(
+      PacketQueueTTL);
+
+  const bool prioritize_audio_retransmission_;
+  const absl::InlinedVector<TimeDelta, kNumPriorityLevels>
+      time_to_live_per_prio_;
+
   // Cumulative sum, over all packets, of time spent in the queue.
   TimeDelta queue_time_sum_;
   // Cumulative sum of time the queue has spent in a paused state.
diff --git a/third_party/libwebrtc/modules/pacing/prioritized_packet_queue_unittest.cc b/third_party/libwebrtc/modules/pacing/prioritized_packet_queue_unittest.cc
index 9ed19642c7..76c31036b3 100644
--- a/third_party/libwebrtc/modules/pacing/prioritized_packet_queue_unittest.cc
+++ b/third_party/libwebrtc/modules/pacing/prioritized_packet_queue_unittest.cc
@@ -10,6 +10,7 @@
 
 #include "modules/pacing/prioritized_packet_queue.h"
 
+#include <memory>
 #include <utility>
 
 #include "api/units/time_delta.h"
@@ -26,18 +27,39 @@ constexpr uint32_t kDefaultSsrc = 123;
 constexpr int kDefaultPayloadSize = 789;
 
 std::unique_ptr<RtpPacketToSend> CreatePacket(RtpPacketMediaType type,
-                                              uint16_t sequence_number,
+                                              uint16_t seq,
                                               uint32_t ssrc = kDefaultSsrc,
                                               bool is_key_frame = false) {
   auto packet = std::make_unique<RtpPacketToSend>(/*extensions=*/nullptr);
   packet->set_packet_type(type);
   packet->SetSsrc(ssrc);
-  packet->SetSequenceNumber(sequence_number);
+  packet->SetSequenceNumber(seq);
   packet->SetPayloadSize(kDefaultPayloadSize);
   packet->set_is_key_frame(is_key_frame);
   return packet;
 }
 
+std::unique_ptr<RtpPacketToSend> CreateRetransmissionPacket(
+    RtpPacketMediaType original_type,
+    uint16_t seq,
+    uint32_t ssrc = kDefaultSsrc) {
+  auto packet = std::make_unique<RtpPacketToSend>(/*extensions=*/nullptr);
+  packet->set_packet_type(original_type);
+  packet->set_packet_type(RtpPacketMediaType::kRetransmission);
+  RTC_DCHECK(packet->packet_type() == RtpPacketMediaType::kRetransmission);
+  if (original_type == RtpPacketMediaType::kVideo) {
+    RTC_DCHECK(packet->original_packet_type() ==
+               RtpPacketToSend::OriginalType::kVideo);
+  } else {
+    RTC_DCHECK(packet->original_packet_type() ==
+               RtpPacketToSend::OriginalType::kAudio);
+  }
+  packet->SetSsrc(ssrc);
+  packet->SetSequenceNumber(seq);
+  packet->SetPayloadSize(kDefaultPayloadSize);
+  return packet;
+}
+
 }  // namespace
 
 TEST(PrioritizedPacketQueue, ReturnsPacketsInPrioritizedOrder) {
@@ -49,18 +71,42 @@ TEST(PrioritizedPacketQueue, ReturnsPacketsInPrioritizedOrder) {
   queue.Push(now, CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/2));
   queue.Push(now, CreatePacket(RtpPacketMediaType::kForwardErrorCorrection,
                                /*seq=*/3));
-  queue.Push(now, CreatePacket(RtpPacketMediaType::kRetransmission, /*seq=*/4));
-  queue.Push(now, CreatePacket(RtpPacketMediaType::kAudio, /*seq=*/5));
+  queue.Push(now,
+             CreateRetransmissionPacket(RtpPacketMediaType::kVideo, /*seq=*/4));
+  queue.Push(now,
+             CreateRetransmissionPacket(RtpPacketMediaType::kAudio, /*seq=*/5));
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kAudio, /*seq=*/6));
 
   // Packets should be returned in high to low order.
-  EXPECT_EQ(queue.Pop()->SequenceNumber(), 5);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 6);
+  // Audio and video retransmission has same prio, but video was enqueued first.
   EXPECT_EQ(queue.Pop()->SequenceNumber(), 4);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 5);
   // Video and FEC prioritized equally - but video was enqueued first.
   EXPECT_EQ(queue.Pop()->SequenceNumber(), 2);
   EXPECT_EQ(queue.Pop()->SequenceNumber(), 3);
   EXPECT_EQ(queue.Pop()->SequenceNumber(), 1);
 }
 
+TEST(PrioritizedPacketQueue,
+     PrioritizeAudioRetransmissionBeforeVideoRetransmissionIfConfigured) {
+  Timestamp now = Timestamp::Zero();
+  PrioritizedPacketQueue queue(now, /*prioritize_audio_retransmission=*/true);
+
+  // Add packets in low to high packet order.
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/3));
+  queue.Push(now,
+             CreateRetransmissionPacket(RtpPacketMediaType::kVideo, /*seq=*/4));
+  queue.Push(now,
+             CreateRetransmissionPacket(RtpPacketMediaType::kAudio, /*seq=*/5));
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kAudio, /*seq=*/6));
+
+  // Packets should be returned in high to low order.
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 6);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 5);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 4);
+}
+
 TEST(PrioritizedPacketQueue, ReturnsEqualPrioPacketsInRoundRobinOrder) {
   Timestamp now = Timestamp::Zero();
   PrioritizedPacketQueue queue(now);
@@ -251,6 +297,26 @@ TEST(PrioritizedPacketQueue, ReportsLeadingPacketEnqueueTime) {
             Timestamp::MinusInfinity());
 }
 
+TEST(PrioritizedPacketQueue, ReportsLeadingPacketEnqueueTimeForRetransmission) {
+  PrioritizedPacketQueue queue(/*creation_time=*/Timestamp::Zero(),
+                               /*prioritize_audio_retransmission=*/true);
+  EXPECT_EQ(queue.LeadingPacketEnqueueTimeForRetransmission(),
+            Timestamp::PlusInfinity());
+
+  queue.Push(Timestamp::Millis(10),
+             CreateRetransmissionPacket(RtpPacketMediaType::kVideo, /*seq=*/1));
+  queue.Push(Timestamp::Millis(11),
+             CreateRetransmissionPacket(RtpPacketMediaType::kAudio, /*seq=*/2));
+  EXPECT_EQ(queue.LeadingPacketEnqueueTimeForRetransmission(),
+            Timestamp::Millis(10));
+  queue.Pop();  // Pop audio retransmission since it has higher prio.
+  EXPECT_EQ(queue.LeadingPacketEnqueueTimeForRetransmission(),
+            Timestamp::Millis(10));
+  queue.Pop();  // Pop video retransmission.
+  EXPECT_EQ(queue.LeadingPacketEnqueueTimeForRetransmission(),
+            Timestamp::PlusInfinity());
+}
+
 TEST(PrioritizedPacketQueue,
      PushAndPopUpdatesSizeInPacketsPerRtpPacketMediaType) {
   Timestamp now = Timestamp::Zero();
@@ -272,7 +338,7 @@ TEST(PrioritizedPacketQueue,
                 RtpPacketMediaType::kVideo)],
             1);
 
-  queue.Push(now, CreatePacket(RtpPacketMediaType::kRetransmission, 3));
+  queue.Push(now, CreateRetransmissionPacket(RtpPacketMediaType::kVideo, 3));
   EXPECT_EQ(queue.SizeInPacketsPerRtpPacketMediaType()[static_cast<size_t>(
                 RtpPacketMediaType::kRetransmission)],
             1);
@@ -326,6 +392,8 @@ TEST(PrioritizedPacketQueue, ClearsPackets) {
   // Remove all of them.
   queue.RemovePacketsForSsrc(kSsrc);
   EXPECT_TRUE(queue.Empty());
+  queue.RemovePacketsForSsrc(kSsrc);
+  EXPECT_TRUE(queue.Empty());
 }
 
 TEST(PrioritizedPacketQueue, ClearPacketsAffectsOnlySpecifiedSsrc) {
@@ -338,16 +406,16 @@ TEST(PrioritizedPacketQueue, ClearPacketsAffectsOnlySpecifiedSsrc) {
   // ensuring they are first in line.
   queue.Push(
       now, CreatePacket(RtpPacketMediaType::kAudio, /*seq=*/1, kRemovingSsrc));
-  queue.Push(now, CreatePacket(RtpPacketMediaType::kRetransmission, /*seq=*/2,
-                               kRemovingSsrc));
+  queue.Push(now, CreateRetransmissionPacket(RtpPacketMediaType::kVideo,
+                                             /*seq=*/2, kRemovingSsrc));
 
   // Add a video packet and a retransmission for the SSRC that will remain.
   // The retransmission packets now both have pointers to their respective qeues
   // from the same prio level.
   queue.Push(now,
              CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/3, kStayingSsrc));
-  queue.Push(now, CreatePacket(RtpPacketMediaType::kRetransmission, /*seq=*/4,
-                               kStayingSsrc));
+  queue.Push(now, CreateRetransmissionPacket(RtpPacketMediaType::kVideo,
+                                             /*seq=*/4, kStayingSsrc));
 
   EXPECT_EQ(queue.SizeInPackets(), 4);
 
@@ -413,4 +481,87 @@ TEST(PrioritizedPacketQueue, ReportsKeyframePackets) {
   EXPECT_FALSE(queue.HasKeyframePackets(kVideoSsrc2));
 }
 
+TEST(PrioritizedPacketQueue, PacketsDroppedIfNotPulledWithinTttl) {
+  Timestamp now = Timestamp::Zero();
+  PacketQueueTTL ttls;
+  ttls.audio_retransmission = TimeDelta::Millis(200);
+  PrioritizedPacketQueue queue(now, /*prioritize_audio_retransmission=*/true,
+                               ttls);
+
+  queue.Push(now,
+             CreateRetransmissionPacket(RtpPacketMediaType::kAudio, /*seq=*/1));
+  now += ttls.audio_retransmission + TimeDelta::Millis(1);
+  EXPECT_EQ(queue.SizeInPackets(), 1);
+  queue.Push(now,
+             CreateRetransmissionPacket(RtpPacketMediaType::kAudio, /*seq=*/2));
+  EXPECT_EQ(queue.SizeInPackets(), 1);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 2);
+}
+
+TEST(PrioritizedPacketQueue, DontSendPacketsAfterTttl) {
+  Timestamp now = Timestamp::Zero();
+  PacketQueueTTL ttls;
+  ttls.audio_retransmission = TimeDelta::Millis(200);
+  PrioritizedPacketQueue queue(now, /*prioritize_audio_retransmission=*/true,
+                               ttls);
+
+  queue.Push(now,
+             CreateRetransmissionPacket(RtpPacketMediaType::kAudio, /*seq=*/1));
+  now += ttls.audio_retransmission + TimeDelta::Millis(1);
+  EXPECT_EQ(queue.SizeInPackets(), 1);
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/2));
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kAudio, /*seq=*/3));
+  // Expect the old packet to have been removed since it was not popped in time.
+  EXPECT_EQ(queue.SizeInPackets(), 3);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 3);
+  EXPECT_EQ(queue.SizeInPackets(), 1);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 2);
+  EXPECT_EQ(queue.SizeInPackets(), 0);
+}
+
+TEST(PrioritizedPacketQueue, SendsNewVideoPacketAfterPurgingLastOldRtxPacket) {
+  Timestamp now = Timestamp::Zero();
+  PacketQueueTTL ttls;
+  ttls.video_retransmission = TimeDelta::Millis(400);
+  PrioritizedPacketQueue queue(now, /*prioritize_audio_retransmission=*/true,
+                               ttls);
+
+  queue.Push(now,
+             CreateRetransmissionPacket(RtpPacketMediaType::kVideo, /*seq=*/1));
+  now += ttls.video_retransmission + TimeDelta::Millis(1);
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kAudio, /*seq=*/2));
+  EXPECT_EQ(queue.SizeInPackets(), 2);
+  // Expect the audio packet to be send and the video retransmission packet to
+  // be dropped since it is old.
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 2);
+  EXPECT_EQ(queue.SizeInPackets(), 0);
+
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/3));
+  EXPECT_EQ(queue.SizeInPackets(), 1);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 3);
+  EXPECT_EQ(queue.SizeInPackets(), 0);
+}
+
+TEST(PrioritizedPacketQueue,
+     SendsPacketsAfterTttlIfPrioHigherThanPushedPackets) {
+  Timestamp now = Timestamp::Zero();
+  PacketQueueTTL ttls;
+  ttls.audio_retransmission = TimeDelta::Millis(200);
+  PrioritizedPacketQueue queue(now, /*prioritize_audio_retransmission=*/true,
+                               ttls);
+
+  queue.Push(now,
+             CreateRetransmissionPacket(RtpPacketMediaType::kAudio, /*seq=*/1));
+  now += ttls.audio_retransmission + TimeDelta::Millis(1);
+  EXPECT_EQ(queue.SizeInPackets(), 1);
+  queue.Push(now, CreatePacket(RtpPacketMediaType::kVideo, /*seq=*/2));
+
+  // This test just show that TTL is not enforced strictly. If a new audio
+  // packet had been queued before a packet was popped, the audio retransmission
+  // packet would have been dropped.
+  EXPECT_EQ(queue.SizeInPackets(), 2);
+  EXPECT_EQ(queue.Pop()->SequenceNumber(), 1);
+  EXPECT_EQ(queue.SizeInPackets(), 1);
+}
+
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
index e100995b8e..6953ec8400 100644
--- a/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
+++ b/third_party/libwebrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
@@ -260,8 +260,11 @@ void RemoteEstimatorProxy::SendFeedbackOnRequest(
       feedback_request.include_timestamps, first_sequence_number,
       sequence_number + 1, /*is_periodic_update=*/false);
 
-  // This is called when a packet has just been added.
-  RTC_DCHECK(feedback_packet != nullptr);
+  // Even though this is called when a packet has just been added,
+  // no feedback may be produced when that new packet is too old.
+  if (feedback_packet == nullptr) {
+    return;
+  }
 
   // Clear up to the first packet that is included in this feedback packet.
   packet_arrival_times_.EraseTo(first_sequence_number);
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
index 0e5e40f502..1dc56bb96f 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
@@ -18,6 +18,7 @@
 
 #include "api/units/time_delta.h"
 #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
 #include "modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
@@ -159,16 +160,15 @@ void StreamStatisticianImpl::UpdateJitter(const RtpPacketReceived& packet,
   int32_t time_diff_samples =
       receive_diff_rtp - (packet.Timestamp() - last_received_timestamp_);
 
-  time_diff_samples = std::abs(time_diff_samples);
-
   ReviseFrequencyAndJitter(packet.payload_type_frequency());
 
   // lib_jingle sometimes deliver crazy jumps in TS for the same stream.
   // If this happens, don't update jitter value. Use 5 secs video frequency
   // as the threshold.
-  if (time_diff_samples < 450000) {
+  if (time_diff_samples < 5 * kVideoPayloadTypeFrequency &&
+      time_diff_samples > -5 * kVideoPayloadTypeFrequency) {
     // Note we calculate in Q4 to avoid using float.
-    int32_t jitter_diff_q4 = (time_diff_samples << 4) - jitter_q4_;
+    int32_t jitter_diff_q4 = (std::abs(time_diff_samples) << 4) - jitter_q4_;
     jitter_q4_ += ((jitter_diff_q4 + 8) >> 4);
   }
 }
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
index a2558545f0..8b31912f0f 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
@@ -898,5 +898,22 @@ TEST(ReviseJitterTest,
   EXPECT_EQ(GetJitter(*statistics), 172U);
 }
 
+TEST(ReviseJitterTest, TwoPacketsWithMaximumRtpTimestampDifference) {
+  SimulatedClock clock(0);
+  std::unique_ptr<ReceiveStatistics> statistics =
+      ReceiveStatistics::Create(&clock);
+  RtpPacketReceived packet1 = MakeRtpPacket(/*payload_type_frequency=*/90'000,
+                                            /*timestamp=*/0x01234567);
+  RtpPacketReceived packet2 =
+      MakeNextRtpPacket(packet1,
+                        /*payload_type_frequency=*/90'000,
+                        /*timestamp=*/0x81234567);
+  statistics->OnRtpPacket(packet1);
+  statistics->OnRtpPacket(packet2);
+
+  // Expect large jump in RTP timestamp is ignored for jitter calculation.
+  EXPECT_EQ(GetJitter(*statistics), 0U);
+}
+
 }  // namespace
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
index bda6ad9a52..756136866d 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
@@ -1190,11 +1190,11 @@ std::vector<rtcp::TmmbItem> RTCPReceiver::TmmbrReceived() {
   MutexLock lock(&rtcp_receiver_lock_);
   std::vector<rtcp::TmmbItem> candidates;
 
-  Timestamp timeout = clock_->CurrentTime() - kTmmbrTimeoutInterval;
+  Timestamp now = clock_->CurrentTime();
 
   for (auto& kv : tmmbr_infos_) {
     for (auto it = kv.second.tmmbr.begin(); it != kv.second.tmmbr.end();) {
-      if (it->second.last_updated < timeout) {
+      if (now - it->second.last_updated > kTmmbrTimeoutInterval) {
         // Erase timeout entries.
         it = kv.second.tmmbr.erase(it);
       } else {
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.cc
index fd42b798d4..27b0420926 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.cc
@@ -52,4 +52,17 @@ bool RtpDependencyDescriptorExtension::Write(
   return writer.Write();
 }
 
+bool RtpDependencyDescriptorExtensionMandatory::Parse(
+    rtc::ArrayView<const uint8_t> data,
+    DependencyDescriptorMandatory* descriptor) {
+  if (data.size() < 3) {
+    return false;
+  }
+  descriptor->set_first_packet_in_frame(data[0] & 0b1000'0000);
+  descriptor->set_last_packet_in_frame(data[0] & 0b0100'0000);
+  descriptor->set_template_id(data[0] & 0b0011'1111);
+  descriptor->set_frame_number((uint16_t{data[1]} << 8) | data[2]);
+  return true;
+}
+
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h
index 8d6e4b8d37..a3e415917c 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h
@@ -54,6 +54,16 @@ class RtpDependencyDescriptorExtension {
   static constexpr std::bitset<32> kAllChainsAreActive = ~uint32_t{0};
 };
 
+// Trait to only read the mandatory part of the descriptor.
+class RtpDependencyDescriptorExtensionMandatory {
+ public:
+  static constexpr webrtc::RTPExtensionType kId =
+      webrtc::RtpDependencyDescriptorExtension::kId;
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data,
+                    DependencyDescriptorMandatory* descriptor);
+};
+
 }  // namespace webrtc
 
 #endif  // MODULES_RTP_RTCP_SOURCE_RTP_DEPENDENCY_DESCRIPTOR_EXTENSION_H_
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
index d066bafb90..9c1dc4edb8 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
@@ -19,6 +19,7 @@
 #include <utility>
 #include <vector>
 
+#include "absl/algorithm/container.h"
 #include "absl/types/optional.h"
 #include "absl/types/variant.h"
 #include "common_video/h264/h264_common.h"
@@ -52,11 +53,14 @@ RtpPacketizerH264::RtpPacketizerH264(rtc::ArrayView<const uint8_t> payload,
     input_fragments_.push_back(
         payload.subview(nalu.payload_start_offset, nalu.payload_size));
   }
-
-  if (!GeneratePackets(packetization_mode)) {
-    // If failed to generate all the packets, discard already generated
-    // packets in case the caller would ignore return value and still try to
-    // call NextPacket().
+  bool has_empty_fragments = absl::c_any_of(
+      input_fragments_, [](const rtc::ArrayView<const uint8_t> fragment) {
+        return fragment.empty();
+      });
+  if (has_empty_fragments || !GeneratePackets(packetization_mode)) {
+    // If empty fragments were found or we failed to generate all the packets,
+    // discard already generated packets in case the caller would ignore the
+    // return value and still try to call NextPacket().
     num_packets_left_ = 0;
     while (!packets_.empty()) {
       packets_.pop();
@@ -73,6 +77,7 @@ size_t RtpPacketizerH264::NumPackets() const {
 bool RtpPacketizerH264::GeneratePackets(
     H264PacketizationMode packetization_mode) {
   for (size_t i = 0; i < input_fragments_.size();) {
+    RTC_DCHECK(!input_fragments_[i].empty());
     switch (packetization_mode) {
       case H264PacketizationMode::SingleNalUnit:
         if (!PacketizeSingleNalu(i))
@@ -173,11 +178,11 @@ size_t RtpPacketizerH264::PacketizeStapA(size_t fragment_index) {
       return fragment_size;
     }
   };
-
   while (payload_size_left >= payload_size_needed()) {
     RTC_CHECK_GT(fragment.size(), 0);
-    packets_.push(PacketUnit(fragment, aggregated_fragments == 0, false, true,
-                             fragment[0]));
+
+    packets_.push(PacketUnit(fragment, /*first=*/aggregated_fragments == 0,
+                             /*last=*/false, /*aggregated=*/true, fragment[0]));
     payload_size_left -= fragment.size();
     payload_size_left -= fragment_headers_length;
 
@@ -218,9 +223,9 @@ bool RtpPacketizerH264::PacketizeSingleNalu(size_t fragment_index) {
                       << limits_.max_payload_len;
     return false;
   }
-  RTC_CHECK_GT(fragment.size(), 0u);
-  packets_.push(PacketUnit(fragment, true /* first */, true /* last */,
-                           false /* aggregated */, fragment[0]));
+  RTC_CHECK(!fragment.empty());
+  packets_.push(PacketUnit(fragment, /*first=*/true, /*last=*/true,
+                           /*aggregated=*/false, fragment[0]));
   ++num_packets_left_;
   return true;
 }
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
index 18311c6e8c..3920c4acd5 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
@@ -116,6 +116,7 @@ std::vector<RtpPacketToSend> FetchAllPackets(RtpPacketizerH264* packetizer) {
   RtpPacketToSend packet(kNoExtensions);
   while (packetizer->NextPacket(&packet)) {
     result.push_back(packet);
+    packet.Clear();
   }
   EXPECT_THAT(result, SizeIs(num_packets));
   return result;
@@ -527,5 +528,190 @@ TEST(RtpPacketizerH264Test, RejectsOverlongDataInPacketizationMode0) {
 
   EXPECT_THAT(packets, IsEmpty());
 }
+
+TEST(RtpPacketizerH264Test, DoesNotPacketizeWithEmptyNalUnit) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = kMaxPayloadSize;
+
+  uint8_t empty_nal_input[] = {0x00, 0x00, 0x01, /* empty NAL unit data */
+                               0x00, 0x00, 0x01, 0x01};
+  RtpPacketizerH264 packetizer(empty_nal_input, limits,
+                               H264PacketizationMode::NonInterleaved);
+  EXPECT_EQ(packetizer.NumPackets(), 0u);
+}
+
+TEST(RtpPacketizerH264Test, MultipleStapA) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = kMaxPayloadSize;
+  limits.first_packet_reduction_len = 0;
+  limits.last_packet_reduction_len = 0;
+  limits.single_packet_reduction_len = 0;
+  // A lot of small NAL units that will result in two STAP-A being generated.
+  // Input data must exceed the size of a single RTP packet.
+  uint8_t long_input[] = {
+      0x19, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01,
+      0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04,
+      0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01,
+      0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00,
+      0x00, 0x01, 0x00, 0xaf, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01,
+      0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00,
+      0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01,
+      0x04, 0x00, 0x19, 0x00, 0x00, 0x01, 0xf9, 0x01, 0x00, 0x00, 0x00, 0x01,
+      0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0xf7, 0x01,
+      0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0x0b, 0x01, 0x04, 0x00,
+      0x19, 0x00, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00,
+      0x00, 0x01, 0x01, 0x02, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00, 0x00,
+      0x01, 0x01, 0x02, 0x00, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00,
+      0x19, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01,
+      0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04,
+      0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x19,
+      0x7a, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01,
+      0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04,
+      0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0x04, 0x00, 0x00, 0x01, 0x00,
+      0xaf, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01,
+      0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04,
+      0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00,
+      0x01, 0x00, 0xaf, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00,
+      0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00,
+      0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00,
+      0x00, 0x00, 0x11, 0xd4, 0x00, 0x19, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00,
+      0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01,
+      0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x00, 0x00, 0x01,
+      0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01,
+      0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00,
+      0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00, 0x01, 0x00, 0xaf,
+      0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04, 0x00,
+      0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x10, 0x00, 0x00,
+      0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00,
+      0x00, 0x01, 0xf9, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x00,
+      0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff,
+      0x08, 0xfe, 0xfb, 0xff, 0xff, 0xf4, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0x04, 0x00,
+      0x00, 0x01, 0x00, 0xaf, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01,
+      0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00,
+      0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00,
+      0x00, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00,
+      0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
+      0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x11, 0xd4, 0x00, 0x19, 0x00, 0x00,
+      0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x01,
+      0x04, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01,
+      0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00,
+      0xaf, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x04,
+      0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e,
+      0x00, 0x00, 0x00, 0x00, 0x11, 0xd4, 0x00, 0x19, 0x00, 0x00, 0x01, 0x04,
+      0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x01, 0x04, 0x00,
+      0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x00,
+      0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00,
+      0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
+      0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00, 0x00,
+      0x01, 0xf9, 0x01, 0x00, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x01, 0x00, 0xf7, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01,
+      0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00,
+      0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00, 0x00, 0x01, 0x01,
+      0x02, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00, 0x00, 0x01, 0x01, 0x02,
+      0x00, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x01, 0x00, 0xf7, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01,
+      0x00, 0xf7, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x26, 0x00, 0x00, 0x01, 0x00,
+      0xaf, 0x01, 0x04, 0x00, 0x19, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00,
+      0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00, 0x00, 0x01, 0x00, 0x01,
+      0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00,
+      0xaf, 0x01, 0x04, 0x00, 0x19, 0x00, 0x00, 0x01, 0xf9, 0x01, 0x00, 0x00,
+      0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00,
+      0xf7, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0x0b, 0x2c,
+      0x01, 0x04, 0x00, 0x19, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00,
+      0x00, 0x01, 0x01, 0x02, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00, 0x00,
+      0x01, 0x01, 0x02, 0x00, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00,
+      0x19, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01,
+      0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04,
+      0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x01, 0x00, 0x00, 0x19, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01,
+      0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04,
+      0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf,
+      0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04, 0x00,
+      0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x10, 0x00, 0x00,
+      0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00,
+      0x00, 0x01, 0xf9, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x00,
+      0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff,
+      0x08, 0xfe, 0xfb, 0xff, 0xff, 0xf4, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0x04, 0x00,
+      0x00, 0x01, 0x00, 0xaf, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01,
+      0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00,
+      0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00,
+      0x00, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00,
+      0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
+      0x04, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01,
+      0x8e, 0x00, 0x04, 0x00, 0x00, 0x01, 0x00, 0x04, 0x00, 0x00, 0x01, 0x00,
+      0xaf, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x00, 0x00, 0x11, 0xd4, 0x00,
+      0x19, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01,
+      0x8e, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01,
+      0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00,
+      0x00, 0x01, 0x00, 0xaf, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01,
+      0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00,
+      0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01,
+      0x04, 0x00, 0x19, 0x00, 0x00, 0x01, 0xf9, 0x01, 0x00, 0x00, 0x00, 0x01,
+      0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0xf7, 0x01,
+      0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00,
+      0x19, 0x00, 0x00, 0x01, 0x01, 0x02, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19,
+      0x00, 0x00, 0x01, 0x01, 0x02, 0x00, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0xf7, 0x01, 0x04, 0x00, 0x00,
+      0x01, 0x01, 0x00, 0x00, 0x01, 0x00, 0xf7, 0x01, 0x04, 0x00, 0x00, 0x01,
+      0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04,
+      0x26, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00, 0x00,
+      0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x01, 0x04, 0x00, 0x19,
+      0x00, 0x00, 0x01, 0x00, 0x01, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
+      0x04, 0x00, 0x00, 0xaf, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01,
+      0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00,
+      0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00,
+      0x00, 0x00, 0x00, 0x11, 0xd4, 0x00, 0x19, 0x00, 0x00, 0x01, 0x04, 0x00,
+      0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x01, 0x04, 0x00, 0x00,
+      0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x00, 0xaf, 0x00, 0x00,
+      0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01,
+      0x01, 0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01,
+      0x00, 0xaf, 0x01, 0x04, 0x00, 0x19, 0x00, 0x00, 0x01, 0xf9, 0x01, 0x00,
+      0x00, 0x00, 0x01, 0x8e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
+      0x00, 0xf7, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e,
+      0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x01, 0x8e,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x04, 0x00, 0x00, 0x01, 0x04, 0x00,
+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x01};
+  RtpPacketizerH264 packetizer(long_input, limits,
+                               H264PacketizationMode::NonInterleaved);
+  EXPECT_EQ(packetizer.NumPackets(), 2u);
+  EXPECT_THAT(FetchAllPackets(&packetizer), SizeIs(2));
+}
+
 }  // namespace
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
index ae5f4e50a4..34b3fd9d2f 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
@@ -63,7 +63,9 @@ RtpPacketizerVp8::RtpPacketizerVp8(rtc::ArrayView<const uint8_t> payload,
                                    const RTPVideoHeaderVP8& hdr_info)
     : hdr_(BuildHeader(hdr_info)), remaining_payload_(payload) {
   limits.max_payload_len -= hdr_.size();
-  payload_sizes_ = SplitAboutEqually(payload.size(), limits);
+  if (!payload.empty()) {
+    payload_sizes_ = SplitAboutEqually(payload.size(), limits);
+  }
   current_packet_ = payload_sizes_.begin();
 }
 
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
index 7934ff8ea9..cab7fc4422 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
@@ -21,6 +21,18 @@ namespace {
 
 constexpr RtpPacketizer::PayloadSizeLimits kNoSizeLimits;
 
+TEST(RtpPacketizerVp8Test, EmptyPayload) {
+  RTPVideoHeaderVP8 hdr_info;
+  hdr_info.InitRTPVideoHeaderVP8();
+  hdr_info.pictureId = 200;
+  RtpFormatVp8TestHelper helper(&hdr_info, /*payload_len=*/30);
+
+  RtpPacketizer::PayloadSizeLimits limits;
+  limits.max_payload_len = 12;  // Small enough to produce 4 packets.
+  RtpPacketizerVp8 packetizer({}, limits, hdr_info);
+  EXPECT_EQ(packetizer.NumPackets(), 0u);
+}
+
 TEST(RtpPacketizerVp8Test, ResultPacketsAreAlmostEqualSize) {
   RTPVideoHeaderVP8 hdr_info;
   hdr_info.InitRTPVideoHeaderVP8();
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc
index 9ad4aa97c3..5e037859ce 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc
@@ -321,8 +321,9 @@ RtpPacketizerVp9::RtpPacketizerVp9(rtc::ArrayView<const uint8_t> payload,
   limits.max_payload_len -= header_size_;
   limits.first_packet_reduction_len += first_packet_extra_header_size_;
   limits.single_packet_reduction_len += first_packet_extra_header_size_;
-
-  payload_sizes_ = SplitAboutEqually(payload.size(), limits);
+  if (!payload.empty()) {
+    payload_sizes_ = SplitAboutEqually(payload.size(), limits);
+  }
   current_packet_ = payload_sizes_.begin();
 }
 
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
index e18b8a803f..948bcf3bdb 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
@@ -186,6 +186,11 @@ class RtpPacketizerVp9Test : public ::testing::Test {
   }
 };
 
+TEST_F(RtpPacketizerVp9Test, EmptyPayload) {
+  RTPVideoHeader video_header;
+  VideoRtpDepacketizerVp9::ParseRtpPayload({}, &video_header);
+}
+
 TEST_F(RtpPacketizerVp9Test, TestEqualSizedMode_OnePacket) {
   const size_t kFrameSize = 25;
   const size_t kPacketSize = 26;
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet.h
index e91ec6368b..c002e51de6 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet.h
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet.h
@@ -11,6 +11,7 @@
 #define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_H_
 
 #include <string>
+#include <utility>
 #include <vector>
 
 #include "absl/types/optional.h"
@@ -127,7 +128,7 @@ class RtpPacket {
   bool IsRegistered() const;
 
   template <typename Extension, typename FirstValue, typename... Values>
-  bool GetExtension(FirstValue, Values...) const;
+  bool GetExtension(FirstValue&&, Values&&...) const;
 
   template <typename Extension>
   absl::optional<typename Extension::value_type> GetExtension() const;
@@ -231,11 +232,12 @@ bool RtpPacket::IsRegistered() const {
 }
 
 template <typename Extension, typename FirstValue, typename... Values>
-bool RtpPacket::GetExtension(FirstValue first, Values... values) const {
+bool RtpPacket::GetExtension(FirstValue&& first, Values&&... values) const {
   auto raw = FindExtension(Extension::kId);
   if (raw.empty())
     return false;
-  return Extension::Parse(raw, first, values...);
+  return Extension::Parse(raw, std::forward<FirstValue>(first),
+                          std::forward<Values>(values)...);
 }
 
 template <typename Extension>
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc
index b55e74aaf0..691a243c5f 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc
@@ -12,6 +12,8 @@
 
 #include <cstdint>
 
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
 namespace webrtc {
 
 RtpPacketToSend::RtpPacketToSend(const ExtensionManager* extensions)
@@ -28,4 +30,13 @@ RtpPacketToSend& RtpPacketToSend::operator=(RtpPacketToSend&& packet) = default;
 
 RtpPacketToSend::~RtpPacketToSend() = default;
 
+void RtpPacketToSend::set_packet_type(RtpPacketMediaType type) {
+  if (packet_type_ == RtpPacketMediaType::kAudio) {
+    original_packet_type_ = OriginalType::kAudio;
+  } else if (packet_type_ == RtpPacketMediaType::kVideo) {
+    original_packet_type_ = OriginalType::kVideo;
+  }
+  packet_type_ = type;
+}
+
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
index 438ca354ed..64f9ee1ab1 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
@@ -49,11 +49,18 @@ class RtpPacketToSend : public RtpPacket {
   webrtc::Timestamp capture_time() const { return capture_time_; }
   void set_capture_time(webrtc::Timestamp time) { capture_time_ = time; }
 
-  void set_packet_type(RtpPacketMediaType type) { packet_type_ = type; }
+  void set_packet_type(RtpPacketMediaType type);
+
   absl::optional<RtpPacketMediaType> packet_type() const {
     return packet_type_;
   }
 
+  enum class OriginalType { kAudio, kVideo };
+  // Original type does not change if packet type is changed to kRetransmission.
+  absl::optional<OriginalType> original_packet_type() const {
+    return original_packet_type_;
+  }
+
   // If this is a retransmission, indicates the sequence number of the original
   // media packet that this packet represents. If RTX is used this will likely
   // be different from SequenceNumber().
@@ -133,6 +140,7 @@ class RtpPacketToSend : public RtpPacket {
  private:
   webrtc::Timestamp capture_time_ = webrtc::Timestamp::Zero();
   absl::optional<RtpPacketMediaType> packet_type_;
+  absl::optional<OriginalType> original_packet_type_;
   bool allow_retransmission_ = false;
   absl::optional<uint16_t> retransmitted_sequence_number_;
   rtc::scoped_refptr<rtc::RefCountedBase> additional_data_;
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
index b3a9452df9..44f1a9e742 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
@@ -933,6 +933,41 @@ TEST(RtpPacketTest, GetUncopyableExtension) {
   EXPECT_TRUE(rtp_packet.GetExtension<UncopyableExtension>(&value2));
 }
 
+struct ParseByReferenceExtension {
+  static constexpr RTPExtensionType kId = kRtpExtensionDependencyDescriptor;
+  static constexpr absl::string_view Uri() { return "uri"; }
+
+  static size_t ValueSize(uint8_t value1, uint8_t value2) { return 2; }
+  static bool Write(rtc::ArrayView<uint8_t> data,
+                    uint8_t value1,
+                    uint8_t value2) {
+    data[0] = value1;
+    data[1] = value2;
+    return true;
+  }
+  static bool Parse(rtc::ArrayView<const uint8_t> data,
+                    uint8_t& value1,
+                    uint8_t& value2) {
+    value1 = data[0];
+    value2 = data[1];
+    return true;
+  }
+};
+
+TEST(RtpPacketTest, GetExtensionByReference) {
+  RtpHeaderExtensionMap extensions;
+  extensions.Register<ParseByReferenceExtension>(1);
+  RtpPacket rtp_packet(&extensions);
+  rtp_packet.SetExtension<ParseByReferenceExtension>(13, 42);
+
+  uint8_t value1 = 1;
+  uint8_t value2 = 1;
+  EXPECT_TRUE(
+      rtp_packet.GetExtension<ParseByReferenceExtension>(value1, value2));
+  EXPECT_EQ(int{value1}, 13);
+  EXPECT_EQ(int{value2}, 42);
+}
+
 TEST(RtpPacketTest, CreateAndParseTimingFrameExtension) {
   // Create a packet with video frame timing extension populated.
   RtpPacketToSend::ExtensionManager send_extensions;
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc
index 2151a59295..83a2be24ea 100644
--- a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_packetizer_av1_unittest.cc
@@ -99,6 +99,12 @@ Av1Frame ReassembleFrame(rtc::ArrayView<const RtpPayload> rtp_payloads) {
   return Av1Frame(VideoRtpDepacketizerAv1().AssembleFrame(payloads));
 }
 
+TEST(RtpPacketizerAv1Test, EmptyPayload) {
+  RtpPacketizer::PayloadSizeLimits limits;
+  RtpPacketizerAv1 packetizer({}, limits, VideoFrameType::kVideoFrameKey, true);
+  EXPECT_EQ(packetizer.NumPackets(), 0u);
+}
+
 TEST(RtpPacketizerAv1Test, PacketizeOneObuWithoutSizeAndExtension) {
   auto kFrame = BuildAv1Frame({Av1Obu(kAv1ObuTypeFrame)
                                    .WithoutSize()
diff --git a/third_party/libwebrtc/modules/video_capture/BUILD.gn b/third_party/libwebrtc/modules/video_capture/BUILD.gn
index 45a0272eee..3132e452ba 100644
--- a/third_party/libwebrtc/modules/video_capture/BUILD.gn
+++ b/third_party/libwebrtc/modules/video_capture/BUILD.gn
@@ -173,6 +173,7 @@ if (!build_with_chromium || is_linux || is_chromeos) {
         "../../api/video:video_frame",
         "../../api/video:video_rtp_headers",
         "../../common_video",
+        "../../rtc_base:gunit_helpers",
         "../../rtc_base:timeutils",
         "../../rtc_base/synchronization:mutex",
         "../../system_wrappers",
diff --git a/third_party/libwebrtc/modules/video_capture/test/video_capture_unittest.cc b/third_party/libwebrtc/modules/video_capture/test/video_capture_unittest.cc
index c8af222b57..dec8de70cb 100644
--- a/third_party/libwebrtc/modules/video_capture/test/video_capture_unittest.cc
+++ b/third_party/libwebrtc/modules/video_capture/test/video_capture_unittest.cc
@@ -22,35 +22,16 @@
 #include "api/video/video_frame.h"
 #include "common_video/libyuv/include/webrtc_libyuv.h"
 #include "modules/video_capture/video_capture_factory.h"
+#include "rtc_base/gunit.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/time_utils.h"
-#include "system_wrappers/include/sleep.h"
 #include "test/frame_utils.h"
 #include "test/gtest.h"
 
-using webrtc::SleepMs;
 using webrtc::VideoCaptureCapability;
 using webrtc::VideoCaptureFactory;
 using webrtc::VideoCaptureModule;
 
-#define WAIT_(ex, timeout, res)                           \
-  do {                                                    \
-    res = (ex);                                           \
-    int64_t start = rtc::TimeMillis();                    \
-    while (!res && rtc::TimeMillis() < start + timeout) { \
-      SleepMs(5);                                         \
-      res = (ex);                                         \
-    }                                                     \
-  } while (0)
-
-#define EXPECT_TRUE_WAIT(ex, timeout) \
-  do {                                \
-    bool res;                         \
-    WAIT_(ex, timeout, res);          \
-    if (!res)                         \
-      EXPECT_TRUE(ex);                \
-  } while (0)
-
 static const int kTimeOut = 5000;
 #ifdef WEBRTC_MAC
 static const int kTestHeight = 288;
diff --git a/third_party/libwebrtc/modules/video_coding/BUILD.gn b/third_party/libwebrtc/modules/video_coding/BUILD.gn
index d9e614ff81..0457b818c3 100644
--- a/third_party/libwebrtc/modules/video_coding/BUILD.gn
+++ b/third_party/libwebrtc/modules/video_coding/BUILD.gn
@@ -218,6 +218,7 @@ rtc_library("video_coding") {
     "../../api:rtp_packet_info",
     "../../api:scoped_refptr",
     "../../api:sequence_checker",
+    "../../api/environment",
     "../../api/task_queue",
     "../../api/units:data_rate",
     "../../api/units:data_size",
@@ -258,7 +259,6 @@ rtc_library("video_coding") {
     "../../rtc_base/task_utils:repeating_task",
     "../../rtc_base/third_party/base64",
     "../../system_wrappers",
-    "../../system_wrappers:field_trial",
     "../../system_wrappers:metrics",
     "../../video/config:encoder_config",
     "../rtp_rtcp",
@@ -1217,6 +1217,7 @@ if (rtc_include_tests) {
       "../../api:scoped_refptr",
       "../../api:simulcast_test_fixture_api",
       "../../api:videocodec_test_fixture_api",
+      "../../api/environment:environment_factory",
       "../../api/task_queue",
       "../../api/task_queue:default_task_queue_factory",
       "../../api/test/video:function_video_factory",
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/av1/dav1d_decoder.cc b/third_party/libwebrtc/modules/video_coding/codecs/av1/dav1d_decoder.cc
index 6a787ff935..d658e401e8 100644
--- a/third_party/libwebrtc/modules/video_coding/codecs/av1/dav1d_decoder.cc
+++ b/third_party/libwebrtc/modules/video_coding/codecs/av1/dav1d_decoder.cc
@@ -87,6 +87,8 @@ bool Dav1dDecoder::Configure(const Settings& settings) {
   s.n_threads = std::max(2, settings.number_of_cores());
   s.max_frame_delay = 1;   // For low latency decoding.
   s.all_layers = 0;        // Don't output a frame for every spatial layer.
+  // Limit max frame size to avoid OOM'ing fuzzers. crbug.com/325284120.
+  s.frame_size_limit = 16384 * 16384;
   s.operating_point = 31;  // Decode all operating points.
 
   return dav1d_open(&context_, &s) == 0;
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/av1/libaom_av1_unittest.cc b/third_party/libwebrtc/modules/video_coding/codecs/av1/libaom_av1_unittest.cc
index 766b7660e4..d486c1d062 100644
--- a/third_party/libwebrtc/modules/video_coding/codecs/av1/libaom_av1_unittest.cc
+++ b/third_party/libwebrtc/modules/video_coding/codecs/av1/libaom_av1_unittest.cc
@@ -350,7 +350,8 @@ INSTANTIATE_TEST_SUITE_P(
            SvcTestParam{"L3T1", /*num_frames_to_generate=*/3},
            SvcTestParam{"L3T3", /*num_frames_to_generate=*/8},
            SvcTestParam{"S2T1", /*num_frames_to_generate=*/3},
-           SvcTestParam{"S3T3", /*num_frames_to_generate=*/8},
+           // TODO: bugs.webrtc.org/15715 - Re-enable once AV1 is fixed.
+           // SvcTestParam{"S3T3", /*num_frames_to_generate=*/8},
            SvcTestParam{"L2T2", /*num_frames_to_generate=*/4},
            SvcTestParam{"L2T2_KEY", /*num_frames_to_generate=*/4},
            SvcTestParam{"L2T2_KEY_SHIFT",
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
index a9e9926c4f..c6446c25ce 100644
--- a/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
+++ b/third_party/libwebrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
@@ -80,7 +80,11 @@ int H264DecoderImpl::AVGetBuffer2(AVCodecContext* context,
       kPixelFormatsSupported.begin(), kPixelFormatsSupported.end(),
       [context](AVPixelFormat format) { return context->pix_fmt == format; });
 
-  RTC_CHECK(pixelFormatSupported != kPixelFormatsSupported.end());
+  if (pixelFormatSupported == kPixelFormatsSupported.end()) {
+    RTC_LOG(LS_ERROR) << "Unsupported pixel format: " << context->pix_fmt;
+    decoder->ReportError();
+    return -1;
+  }
 
   // `av_frame->width` and `av_frame->height` are set by FFmpeg. These are the
   // actual image's dimensions and may be different from `context->width` and
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_test.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_test.cc
index 60c2fcbb6e..2ab1106a59 100644
--- a/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_test.cc
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/video_codec_test.cc
@@ -178,7 +178,8 @@ std::string TestOutputPath() {
 }  // namespace
 
 std::unique_ptr<VideoCodecStats> RunEncodeDecodeTest(
-    std::string codec_impl,
+    std::string encoder_impl,
+    std::string decoder_impl,
     const VideoInfo& video_info,
     const std::map<uint32_t, EncodingSettings>& encoding_settings) {
   VideoSourceSettings source_settings{
@@ -190,28 +191,34 @@ std::unique_ptr<VideoCodecStats> RunEncodeDecodeTest(
       encoding_settings.begin()->second.sdp_video_format;
 
   std::unique_ptr<VideoEncoderFactory> encoder_factory =
-      CreateEncoderFactory(codec_impl);
+      CreateEncoderFactory(encoder_impl);
   if (!encoder_factory
            ->QueryCodecSupport(sdp_video_format,
                                /*scalability_mode=*/absl::nullopt)
            .is_supported) {
-    RTC_LOG(LS_WARNING) << "No encoder for video format "
+    RTC_LOG(LS_WARNING) << "No " << encoder_impl << " encoder for video format "
                         << sdp_video_format.ToString();
     return nullptr;
   }
 
   std::unique_ptr<VideoDecoderFactory> decoder_factory =
-      CreateDecoderFactory(codec_impl);
+      CreateDecoderFactory(decoder_impl);
   if (!decoder_factory
            ->QueryCodecSupport(sdp_video_format,
                                /*reference_scaling=*/false)
            .is_supported) {
+    RTC_LOG(LS_WARNING) << "No " << decoder_impl << " decoder for video format "
+                        << sdp_video_format.ToString()
+                        << ". Trying built-in decoder.";
+    // TODO(ssilkin): No H264 support in ffmpeg on ARM. Consider trying HW
+    // decoder.
     decoder_factory = CreateDecoderFactory("builtin");
     if (!decoder_factory
              ->QueryCodecSupport(sdp_video_format,
                                  /*reference_scaling=*/false)
              .is_supported) {
-      RTC_LOG(LS_WARNING) << "No decoder for video format "
+      RTC_LOG(LS_WARNING) << "No " << decoder_impl
+                          << " decoder for video format "
                           << sdp_video_format.ToString();
       return nullptr;
     }
@@ -221,7 +228,7 @@ std::unique_ptr<VideoCodecStats> RunEncodeDecodeTest(
 
   VideoCodecTester::EncoderSettings encoder_settings;
   encoder_settings.pacing_settings.mode =
-      codec_impl == "builtin" ? PacingMode::kNoPacing : PacingMode::kRealTime;
+      encoder_impl == "builtin" ? PacingMode::kNoPacing : PacingMode::kRealTime;
   if (absl::GetFlag(FLAGS_dump_encoder_input)) {
     encoder_settings.encoder_input_base_path = output_path + "_enc_input";
   }
@@ -231,7 +238,7 @@ std::unique_ptr<VideoCodecStats> RunEncodeDecodeTest(
 
   VideoCodecTester::DecoderSettings decoder_settings;
   decoder_settings.pacing_settings.mode =
-      codec_impl == "builtin" ? PacingMode::kNoPacing : PacingMode::kRealTime;
+      decoder_impl == "builtin" ? PacingMode::kNoPacing : PacingMode::kRealTime;
   if (absl::GetFlag(FLAGS_dump_decoder_input)) {
     decoder_settings.decoder_input_base_path = output_path + "_dec_input";
   }
@@ -318,7 +325,7 @@ TEST_P(SpatialQualityTest, SpatialQuality) {
           {bitrate_kbps}, framerate_fps, num_frames);
 
   std::unique_ptr<VideoCodecStats> stats =
-      RunEncodeDecodeTest(codec_impl, video_info, frames_settings);
+      RunEncodeDecodeTest(codec_impl, codec_impl, video_info, frames_settings);
 
   VideoCodecStats::Stream stream;
   if (stats != nullptr) {
@@ -348,15 +355,15 @@ INSTANTIATE_TEST_SUITE_P(
             Values("builtin"),
 #endif
             Values(kRawVideos.at("FourPeople_1280x720_30")),
-            Values(std::make_tuple(320, 180, 30, 32, 28),
-                   std::make_tuple(320, 180, 30, 64, 30),
-                   std::make_tuple(320, 180, 30, 128, 33),
+            Values(std::make_tuple(320, 180, 30, 32, 26),
+                   std::make_tuple(320, 180, 30, 64, 29),
+                   std::make_tuple(320, 180, 30, 128, 32),
                    std::make_tuple(320, 180, 30, 256, 36),
-                   std::make_tuple(640, 360, 30, 128, 31),
+                   std::make_tuple(640, 360, 30, 128, 29),
                    std::make_tuple(640, 360, 30, 256, 33),
                    std::make_tuple(640, 360, 30, 384, 35),
                    std::make_tuple(640, 360, 30, 512, 36),
-                   std::make_tuple(1280, 720, 30, 256, 32),
+                   std::make_tuple(1280, 720, 30, 256, 30),
                    std::make_tuple(1280, 720, 30, 512, 34),
                    std::make_tuple(1280, 720, 30, 1024, 37),
                    std::make_tuple(1280, 720, 30, 2048, 39))),
@@ -538,6 +545,7 @@ TEST(VideoCodecTest, DISABLED_EncodeDecode) {
   // Sync with changes in Stream::LogMetrics (see TODOs there).
   std::unique_ptr<VideoCodecStats> stats = RunEncodeDecodeTest(
       CodecNameToCodecImpl(absl::GetFlag(FLAGS_encoder)),
+      CodecNameToCodecImpl(absl::GetFlag(FLAGS_decoder)),
       kRawVideos.at(absl::GetFlag(FLAGS_video_name)), frames_settings);
   ASSERT_NE(nullptr, stats);
 
diff --git a/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
index eb264e5285..35355d4387 100644
--- a/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
+++ b/third_party/libwebrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
@@ -167,7 +167,7 @@ SdpVideoFormat CreateSdpVideoFormat(
                 H264PacketizationMode::NonInterleaved
             ? "1"
             : "0";
-    SdpVideoFormat::Parameters codec_params = {
+    CodecParameterMap codec_params = {
         {cricket::kH264FmtpProfileLevelId,
          *H264ProfileLevelIdToString(H264ProfileLevelId(
              config.h264_codec_settings.profile, H264Level::kLevel3_1))},
diff --git a/third_party/libwebrtc/modules/video_coding/fec_controller_default.cc b/third_party/libwebrtc/modules/video_coding/fec_controller_default.cc
index f204b01c7c..d548d6580c 100644
--- a/third_party/libwebrtc/modules/video_coding/fec_controller_default.cc
+++ b/third_party/libwebrtc/modules/video_coding/fec_controller_default.cc
@@ -15,30 +15,28 @@
 #include <algorithm>
 #include <string>
 
+#include "api/environment/environment.h"
+#include "api/field_trials_view.h"
 #include "modules/include/module_fec_types.h"
 #include "rtc_base/logging.h"
-#include "system_wrappers/include/field_trial.h"
+#include "system_wrappers/include/clock.h"
 
 namespace webrtc {
 
 const float kProtectionOverheadRateThreshold = 0.5;
 
 FecControllerDefault::FecControllerDefault(
-    Clock* clock,
+    const Environment& env,
     VCMProtectionCallback* protection_callback)
-    : clock_(clock),
+    : env_(env),
       protection_callback_(protection_callback),
       loss_prot_logic_(new media_optimization::VCMLossProtectionLogic(
-          clock_->TimeInMilliseconds())),
+          env_.clock().TimeInMilliseconds())),
       max_payload_size_(1460),
       overhead_threshold_(GetProtectionOverheadRateThreshold()) {}
 
-FecControllerDefault::FecControllerDefault(Clock* clock)
-    : clock_(clock),
-      loss_prot_logic_(new media_optimization::VCMLossProtectionLogic(
-          clock_->TimeInMilliseconds())),
-      max_payload_size_(1460),
-      overhead_threshold_(GetProtectionOverheadRateThreshold()) {}
+FecControllerDefault::FecControllerDefault(const Environment& env)
+    : FecControllerDefault(env, nullptr) {}
 
 FecControllerDefault::~FecControllerDefault(void) {
   loss_prot_logic_->Release();
@@ -61,8 +59,8 @@ void FecControllerDefault::SetEncodingData(size_t width,
 
 float FecControllerDefault::GetProtectionOverheadRateThreshold() {
   float overhead_threshold =
-      strtof(webrtc::field_trial::FindFullName(
-                 "WebRTC-ProtectionOverheadRateThreshold")
+      strtof(env_.field_trials()
+                 .Lookup("WebRTC-ProtectionOverheadRateThreshold")
                  .c_str(),
              nullptr);
   if (overhead_threshold > 0 && overhead_threshold <= 1) {
@@ -107,7 +105,7 @@ uint32_t FecControllerDefault::UpdateFecRates(
     media_optimization::FilterPacketLossMode filter_mode =
         media_optimization::kMaxFilter;
     uint8_t packet_loss_enc = loss_prot_logic_->FilteredLoss(
-        clock_->TimeInMilliseconds(), filter_mode, fraction_lost);
+        env_.clock().TimeInMilliseconds(), filter_mode, fraction_lost);
     // For now use the filtered loss for computing the robustness settings.
     loss_prot_logic_->UpdateFilteredLossPr(packet_loss_enc);
     if (loss_prot_logic_->SelectedType() == media_optimization::kNone) {
@@ -191,11 +189,11 @@ void FecControllerDefault::UpdateWithEncodedData(
       const float min_packets_per_frame =
           encoded_length / static_cast<float>(max_payload_size_);
       if (delta_frame) {
-        loss_prot_logic_->UpdatePacketsPerFrame(min_packets_per_frame,
-                                                clock_->TimeInMilliseconds());
+        loss_prot_logic_->UpdatePacketsPerFrame(
+            min_packets_per_frame, env_.clock().TimeInMilliseconds());
       } else {
         loss_prot_logic_->UpdatePacketsPerFrameKey(
-            min_packets_per_frame, clock_->TimeInMilliseconds());
+            min_packets_per_frame, env_.clock().TimeInMilliseconds());
       }
     }
     if (!delta_frame && encoded_length > 0) {
diff --git a/third_party/libwebrtc/modules/video_coding/fec_controller_default.h b/third_party/libwebrtc/modules/video_coding/fec_controller_default.h
index a97dea011b..f60302b2d8 100644
--- a/third_party/libwebrtc/modules/video_coding/fec_controller_default.h
+++ b/third_party/libwebrtc/modules/video_coding/fec_controller_default.h
@@ -17,24 +17,25 @@
 #include <memory>
 #include <vector>
 
+#include "api/environment/environment.h"
 #include "api/fec_controller.h"
 #include "modules/video_coding/media_opt_util.h"
 #include "rtc_base/synchronization/mutex.h"
 #include "rtc_base/thread_annotations.h"
-#include "system_wrappers/include/clock.h"
 
 namespace webrtc {
 
 class FecControllerDefault : public FecController {
  public:
-  FecControllerDefault(Clock* clock,
+  FecControllerDefault(const Environment& env,
                        VCMProtectionCallback* protection_callback);
-  explicit FecControllerDefault(Clock* clock);
-  ~FecControllerDefault() override;
+  explicit FecControllerDefault(const Environment& env);
 
   FecControllerDefault(const FecControllerDefault&) = delete;
   FecControllerDefault& operator=(const FecControllerDefault&) = delete;
 
+  ~FecControllerDefault() override;
+
   void SetProtectionCallback(
       VCMProtectionCallback* protection_callback) override;
   void SetProtectionMethod(bool enable_fec, bool enable_nack) override;
@@ -54,7 +55,7 @@ class FecControllerDefault : public FecController {
 
  private:
   enum { kBitrateAverageWinMs = 1000 };
-  Clock* const clock_;
+  const Environment env_;
   VCMProtectionCallback* protection_callback_;
   Mutex mutex_;
   std::unique_ptr<media_optimization::VCMLossProtectionLogic> loss_prot_logic_
diff --git a/third_party/libwebrtc/modules/video_coding/fec_controller_unittest.cc b/third_party/libwebrtc/modules/video_coding/fec_controller_unittest.cc
index fa3cc7a93b..c4a0a08fd1 100644
--- a/third_party/libwebrtc/modules/video_coding/fec_controller_unittest.cc
+++ b/third_party/libwebrtc/modules/video_coding/fec_controller_unittest.cc
@@ -14,6 +14,7 @@
 
 #include <vector>
 
+#include "api/environment/environment_factory.h"
 #include "modules/include/module_fec_types.h"
 #include "modules/video_coding/fec_controller_default.h"
 #include "system_wrappers/include/clock.h"
@@ -50,7 +51,8 @@ class ProtectionBitrateCalculatorTest : public ::testing::Test {
   // Note: simulated clock starts at 1 seconds, since parts of webrtc use 0 as
   // a special case (e.g. frame rate in media optimization).
   ProtectionBitrateCalculatorTest()
-      : clock_(1000), fec_controller_(&clock_, &protection_callback_) {}
+      : clock_(1000),
+        fec_controller_(CreateEnvironment(&clock_), &protection_callback_) {}
 
   SimulatedClock clock_;
   ProtectionCallback protection_callback_;
diff --git a/third_party/libwebrtc/modules/video_coding/nack_requester.cc b/third_party/libwebrtc/modules/video_coding/nack_requester.cc
index 008420f4da..b3e928d05e 100644
--- a/third_party/libwebrtc/modules/video_coding/nack_requester.cc
+++ b/third_party/libwebrtc/modules/video_coding/nack_requester.cc
@@ -141,13 +141,12 @@ void NackRequester::ProcessNacks() {
   }
 }
 
-int NackRequester::OnReceivedPacket(uint16_t seq_num, bool is_keyframe) {
+int NackRequester::OnReceivedPacket(uint16_t seq_num) {
   RTC_DCHECK_RUN_ON(worker_thread_);
-  return OnReceivedPacket(seq_num, is_keyframe, false);
+  return OnReceivedPacket(seq_num, false);
 }
 
 int NackRequester::OnReceivedPacket(uint16_t seq_num,
-                                    bool is_keyframe,
                                     bool is_recovered) {
   RTC_DCHECK_RUN_ON(worker_thread_);
   // TODO(philipel): When the packet includes information whether it is
@@ -158,8 +157,6 @@ int NackRequester::OnReceivedPacket(uint16_t seq_num,
 
   if (!initialized_) {
     newest_seq_num_ = seq_num;
-    if (is_keyframe)
-      keyframe_list_.insert(seq_num);
     initialized_ = true;
     return 0;
   }
@@ -182,15 +179,6 @@ int NackRequester::OnReceivedPacket(uint16_t seq_num,
     return nacks_sent_for_packet;
   }
 
-  // Keep track of new keyframes.
-  if (is_keyframe)
-    keyframe_list_.insert(seq_num);
-
-  // And remove old ones so we don't accumulate keyframes.
-  auto it = keyframe_list_.lower_bound(seq_num - kMaxPacketAge);
-  if (it != keyframe_list_.begin())
-    keyframe_list_.erase(keyframe_list_.begin(), it);
-
   if (is_recovered) {
     recovered_list_.insert(seq_num);
 
@@ -225,8 +213,6 @@ void NackRequester::ClearUpTo(uint16_t seq_num) {
   // thread.
   RTC_DCHECK_RUN_ON(worker_thread_);
   nack_list_.erase(nack_list_.begin(), nack_list_.lower_bound(seq_num));
-  keyframe_list_.erase(keyframe_list_.begin(),
-                       keyframe_list_.lower_bound(seq_num));
   recovered_list_.erase(recovered_list_.begin(),
                         recovered_list_.lower_bound(seq_num));
 }
@@ -236,25 +222,6 @@ void NackRequester::UpdateRtt(int64_t rtt_ms) {
   rtt_ = TimeDelta::Millis(rtt_ms);
 }
 
-bool NackRequester::RemovePacketsUntilKeyFrame() {
-  // Called on worker_thread_.
-  while (!keyframe_list_.empty()) {
-    auto it = nack_list_.lower_bound(*keyframe_list_.begin());
-
-    if (it != nack_list_.begin()) {
-      // We have found a keyframe that actually is newer than at least one
-      // packet in the nack list.
-      nack_list_.erase(nack_list_.begin(), it);
-      return true;
-    }
-
-    // If this keyframe is so old it does not remove any packets from the list,
-    // remove it from the list of keyframes and try the next keyframe.
-    keyframe_list_.erase(keyframe_list_.begin());
-  }
-  return false;
-}
-
 void NackRequester::AddPacketsToNack(uint16_t seq_num_start,
                                      uint16_t seq_num_end) {
   // Called on worker_thread_.
@@ -262,22 +229,13 @@ void NackRequester::AddPacketsToNack(uint16_t seq_num_start,
   auto it = nack_list_.lower_bound(seq_num_end - kMaxPacketAge);
   nack_list_.erase(nack_list_.begin(), it);
 
-  // If the nack list is too large, remove packets from the nack list until
-  // the latest first packet of a keyframe. If the list is still too large,
-  // clear it and request a keyframe.
   uint16_t num_new_nacks = ForwardDiff(seq_num_start, seq_num_end);
   if (nack_list_.size() + num_new_nacks > kMaxNackPackets) {
-    while (RemovePacketsUntilKeyFrame() &&
-           nack_list_.size() + num_new_nacks > kMaxNackPackets) {
-    }
-
-    if (nack_list_.size() + num_new_nacks > kMaxNackPackets) {
-      nack_list_.clear();
-      RTC_LOG(LS_WARNING) << "NACK list full, clearing NACK"
-                             " list and requesting keyframe.";
-      keyframe_request_sender_->RequestKeyFrame();
-      return;
-    }
+    nack_list_.clear();
+    RTC_LOG(LS_WARNING) << "NACK list full, clearing NACK"
+                           " list and requesting keyframe.";
+    keyframe_request_sender_->RequestKeyFrame();
+    return;
   }
 
   for (uint16_t seq_num = seq_num_start; seq_num != seq_num_end; ++seq_num) {
diff --git a/third_party/libwebrtc/modules/video_coding/nack_requester.h b/third_party/libwebrtc/modules/video_coding/nack_requester.h
index c860787dcf..b5ef30f01b 100644
--- a/third_party/libwebrtc/modules/video_coding/nack_requester.h
+++ b/third_party/libwebrtc/modules/video_coding/nack_requester.h
@@ -78,8 +78,8 @@ class NackRequester final : public NackRequesterBase {
 
   void ProcessNacks() override;
 
-  int OnReceivedPacket(uint16_t seq_num, bool is_keyframe);
-  int OnReceivedPacket(uint16_t seq_num, bool is_keyframe, bool is_recovered);
+  int OnReceivedPacket(uint16_t seq_num);
+  int OnReceivedPacket(uint16_t seq_num, bool is_recovered);
 
   void ClearUpTo(uint16_t seq_num);
   void UpdateRtt(int64_t rtt_ms);
@@ -108,10 +108,6 @@ class NackRequester final : public NackRequesterBase {
   void AddPacketsToNack(uint16_t seq_num_start, uint16_t seq_num_end)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(worker_thread_);
 
-  // Removes packets from the nack list until the next keyframe. Returns true
-  // if packets were removed.
-  bool RemovePacketsUntilKeyFrame()
-      RTC_EXCLUSIVE_LOCKS_REQUIRED(worker_thread_);
   std::vector<uint16_t> GetNackBatch(NackFilterOptions options)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(worker_thread_);
 
@@ -134,8 +130,6 @@ class NackRequester final : public NackRequesterBase {
   // synchronized access.
   std::map<uint16_t, NackInfo, DescendingSeqNumComp<uint16_t>> nack_list_
       RTC_GUARDED_BY(worker_thread_);
-  std::set<uint16_t, DescendingSeqNumComp<uint16_t>> keyframe_list_
-      RTC_GUARDED_BY(worker_thread_);
   std::set<uint16_t, DescendingSeqNumComp<uint16_t>> recovered_list_
       RTC_GUARDED_BY(worker_thread_);
   video_coding::Histogram reordering_histogram_ RTC_GUARDED_BY(worker_thread_);
diff --git a/third_party/libwebrtc/modules/video_coding/nack_requester_unittest.cc b/third_party/libwebrtc/modules/video_coding/nack_requester_unittest.cc
index 6f11cb6e91..2f24df2aef 100644
--- a/third_party/libwebrtc/modules/video_coding/nack_requester_unittest.cc
+++ b/third_party/libwebrtc/modules/video_coding/nack_requester_unittest.cc
@@ -102,90 +102,25 @@ class TestNackRequester : public ::testing::Test,
 
 TEST_F(TestNackRequester, NackOnePacket) {
   NackRequester& nack_module = CreateNackModule();
-  nack_module.OnReceivedPacket(1, false, false);
-  nack_module.OnReceivedPacket(3, false, false);
+  nack_module.OnReceivedPacket(1);
+  nack_module.OnReceivedPacket(3);
   ASSERT_EQ(1u, sent_nacks_.size());
   EXPECT_EQ(2, sent_nacks_[0]);
 }
 
 TEST_F(TestNackRequester, WrappingSeqNum) {
   NackRequester& nack_module = CreateNackModule();
-  nack_module.OnReceivedPacket(0xfffe, false, false);
-  nack_module.OnReceivedPacket(1, false, false);
+  nack_module.OnReceivedPacket(0xfffe);
+  nack_module.OnReceivedPacket(1);
   ASSERT_EQ(2u, sent_nacks_.size());
   EXPECT_EQ(0xffff, sent_nacks_[0]);
   EXPECT_EQ(0, sent_nacks_[1]);
 }
 
-TEST_F(TestNackRequester, WrappingSeqNumClearToKeyframe) {
-  NackRequester& nack_module = CreateNackModule(TimeDelta::Millis(10));
-  nack_module.OnReceivedPacket(0xfffe, false, false);
-  nack_module.OnReceivedPacket(1, false, false);
-  ASSERT_EQ(2u, sent_nacks_.size());
-  EXPECT_EQ(0xffff, sent_nacks_[0]);
-  EXPECT_EQ(0, sent_nacks_[1]);
-
-  sent_nacks_.clear();
-  nack_module.OnReceivedPacket(2, true, false);
-  ASSERT_EQ(0u, sent_nacks_.size());
-
-  nack_module.OnReceivedPacket(501, true, false);
-  ASSERT_EQ(498u, sent_nacks_.size());
-  for (int seq_num = 3; seq_num < 501; ++seq_num)
-    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 3]);
-
-  sent_nacks_.clear();
-  nack_module.OnReceivedPacket(1001, false, false);
-  EXPECT_EQ(499u, sent_nacks_.size());
-  for (int seq_num = 502; seq_num < 1001; ++seq_num)
-    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 502]);
-
-  sent_nacks_.clear();
-  clock_->AdvanceTimeMilliseconds(100);
-  ASSERT_TRUE(WaitForSendNack());
-  ASSERT_EQ(999u, sent_nacks_.size());
-  EXPECT_EQ(0xffff, sent_nacks_[0]);
-  EXPECT_EQ(0, sent_nacks_[1]);
-  for (int seq_num = 3; seq_num < 501; ++seq_num)
-    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 1]);
-  for (int seq_num = 502; seq_num < 1001; ++seq_num)
-    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 2]);
-
-  // Adding packet 1004 will cause the nack list to reach it's max limit.
-  // It will then clear all nacks up to the next keyframe (seq num 2),
-  // thus removing 0xffff and 0 from the nack list.
-  sent_nacks_.clear();
-  nack_module.OnReceivedPacket(1004, false, false);
-  ASSERT_EQ(2u, sent_nacks_.size());
-  EXPECT_EQ(1002, sent_nacks_[0]);
-  EXPECT_EQ(1003, sent_nacks_[1]);
-
-  sent_nacks_.clear();
-  clock_->AdvanceTimeMilliseconds(100);
-  ASSERT_TRUE(WaitForSendNack());
-  ASSERT_EQ(999u, sent_nacks_.size());
-  for (int seq_num = 3; seq_num < 501; ++seq_num)
-    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 3]);
-  for (int seq_num = 502; seq_num < 1001; ++seq_num)
-    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 4]);
-
-  // Adding packet 1007 will cause the nack module to overflow again, thus
-  // clearing everything up to 501 which is the next keyframe.
-  nack_module.OnReceivedPacket(1007, false, false);
-  sent_nacks_.clear();
-  clock_->AdvanceTimeMilliseconds(100);
-  ASSERT_TRUE(WaitForSendNack());
-  ASSERT_EQ(503u, sent_nacks_.size());
-  for (int seq_num = 502; seq_num < 1001; ++seq_num)
-    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 502]);
-  EXPECT_EQ(1005, sent_nacks_[501]);
-  EXPECT_EQ(1006, sent_nacks_[502]);
-}
-
 TEST_F(TestNackRequester, ResendNack) {
   NackRequester& nack_module = CreateNackModule(TimeDelta::Millis(1));
-  nack_module.OnReceivedPacket(1, false, false);
-  nack_module.OnReceivedPacket(3, false, false);
+  nack_module.OnReceivedPacket(1);
+  nack_module.OnReceivedPacket(3);
   size_t expected_nacks_sent = 1;
   ASSERT_EQ(expected_nacks_sent, sent_nacks_.size());
   EXPECT_EQ(2, sent_nacks_[0]);
@@ -225,8 +160,8 @@ TEST_F(TestNackRequester, ResendNack) {
 
 TEST_F(TestNackRequester, ResendPacketMaxRetries) {
   NackRequester& nack_module = CreateNackModule(TimeDelta::Millis(1));
-  nack_module.OnReceivedPacket(1, false, false);
-  nack_module.OnReceivedPacket(3, false, false);
+  nack_module.OnReceivedPacket(1);
+  nack_module.OnReceivedPacket(3);
   ASSERT_EQ(1u, sent_nacks_.size());
   EXPECT_EQ(2, sent_nacks_[0]);
 
@@ -246,37 +181,22 @@ TEST_F(TestNackRequester, ResendPacketMaxRetries) {
 
 TEST_F(TestNackRequester, TooLargeNackList) {
   NackRequester& nack_module = CreateNackModule();
-  nack_module.OnReceivedPacket(0, false, false);
-  nack_module.OnReceivedPacket(1001, false, false);
+  nack_module.OnReceivedPacket(0);
+  nack_module.OnReceivedPacket(1001);
   EXPECT_EQ(1000u, sent_nacks_.size());
   EXPECT_EQ(0, keyframes_requested_);
-  nack_module.OnReceivedPacket(1003, false, false);
+  nack_module.OnReceivedPacket(1003);
   EXPECT_EQ(1000u, sent_nacks_.size());
   EXPECT_EQ(1, keyframes_requested_);
-  nack_module.OnReceivedPacket(1004, false, false);
-  EXPECT_EQ(1000u, sent_nacks_.size());
-  EXPECT_EQ(1, keyframes_requested_);
-}
-
-TEST_F(TestNackRequester, TooLargeNackListWithKeyFrame) {
-  NackRequester& nack_module = CreateNackModule();
-  nack_module.OnReceivedPacket(0, false, false);
-  nack_module.OnReceivedPacket(1, true, false);
-  nack_module.OnReceivedPacket(1001, false, false);
-  EXPECT_EQ(999u, sent_nacks_.size());
-  EXPECT_EQ(0, keyframes_requested_);
-  nack_module.OnReceivedPacket(1003, false, false);
-  EXPECT_EQ(1000u, sent_nacks_.size());
-  EXPECT_EQ(0, keyframes_requested_);
-  nack_module.OnReceivedPacket(1005, false, false);
+  nack_module.OnReceivedPacket(1004);
   EXPECT_EQ(1000u, sent_nacks_.size());
   EXPECT_EQ(1, keyframes_requested_);
 }
 
 TEST_F(TestNackRequester, ClearUpTo) {
   NackRequester& nack_module = CreateNackModule(TimeDelta::Millis(1));
-  nack_module.OnReceivedPacket(0, false, false);
-  nack_module.OnReceivedPacket(100, false, false);
+  nack_module.OnReceivedPacket(0);
+  nack_module.OnReceivedPacket(100);
   EXPECT_EQ(99u, sent_nacks_.size());
 
   sent_nacks_.clear();
@@ -289,8 +209,8 @@ TEST_F(TestNackRequester, ClearUpTo) {
 
 TEST_F(TestNackRequester, ClearUpToWrap) {
   NackRequester& nack_module = CreateNackModule();
-  nack_module.OnReceivedPacket(0xfff0, false, false);
-  nack_module.OnReceivedPacket(0xf, false, false);
+  nack_module.OnReceivedPacket(0xfff0);
+  nack_module.OnReceivedPacket(0xf);
   EXPECT_EQ(30u, sent_nacks_.size());
 
   sent_nacks_.clear();
@@ -303,13 +223,13 @@ TEST_F(TestNackRequester, ClearUpToWrap) {
 
 TEST_F(TestNackRequester, PacketNackCount) {
   NackRequester& nack_module = CreateNackModule(TimeDelta::Millis(1));
-  EXPECT_EQ(0, nack_module.OnReceivedPacket(0, false, false));
-  EXPECT_EQ(0, nack_module.OnReceivedPacket(2, false, false));
-  EXPECT_EQ(1, nack_module.OnReceivedPacket(1, false, false));
+  EXPECT_EQ(0, nack_module.OnReceivedPacket(0));
+  EXPECT_EQ(0, nack_module.OnReceivedPacket(2));
+  EXPECT_EQ(1, nack_module.OnReceivedPacket(1));
 
   sent_nacks_.clear();
   nack_module.UpdateRtt(100);
-  EXPECT_EQ(0, nack_module.OnReceivedPacket(5, false, false));
+  EXPECT_EQ(0, nack_module.OnReceivedPacket(5));
   clock_->AdvanceTimeMilliseconds(100);
   WaitForSendNack();
   EXPECT_EQ(4u, sent_nacks_.size());
@@ -319,40 +239,24 @@ TEST_F(TestNackRequester, PacketNackCount) {
 
   EXPECT_EQ(6u, sent_nacks_.size());
 
-  EXPECT_EQ(3, nack_module.OnReceivedPacket(3, false, false));
-  EXPECT_EQ(3, nack_module.OnReceivedPacket(4, false, false));
-  EXPECT_EQ(0, nack_module.OnReceivedPacket(4, false, false));
-}
-
-TEST_F(TestNackRequester, NackListFullAndNoOverlapWithKeyframes) {
-  NackRequester& nack_module = CreateNackModule();
-  const int kMaxNackPackets = 1000;
-  const unsigned int kFirstGap = kMaxNackPackets - 20;
-  const unsigned int kSecondGap = 200;
-  uint16_t seq_num = 0;
-  nack_module.OnReceivedPacket(seq_num++, true, false);
-  seq_num += kFirstGap;
-  nack_module.OnReceivedPacket(seq_num++, true, false);
-  EXPECT_EQ(kFirstGap, sent_nacks_.size());
-  sent_nacks_.clear();
-  seq_num += kSecondGap;
-  nack_module.OnReceivedPacket(seq_num, true, false);
-  EXPECT_EQ(kSecondGap, sent_nacks_.size());
+  EXPECT_EQ(3, nack_module.OnReceivedPacket(3));
+  EXPECT_EQ(3, nack_module.OnReceivedPacket(4));
+  EXPECT_EQ(0, nack_module.OnReceivedPacket(4));
 }
 
 TEST_F(TestNackRequester, HandleFecRecoveredPacket) {
   NackRequester& nack_module = CreateNackModule();
-  nack_module.OnReceivedPacket(1, false, false);
-  nack_module.OnReceivedPacket(4, false, true);
+  nack_module.OnReceivedPacket(1);
+  nack_module.OnReceivedPacket(4, /*is_recovered=*/true);
   EXPECT_EQ(0u, sent_nacks_.size());
-  nack_module.OnReceivedPacket(5, false, false);
+  nack_module.OnReceivedPacket(5);
   EXPECT_EQ(2u, sent_nacks_.size());
 }
 
 TEST_F(TestNackRequester, SendNackWithoutDelay) {
   NackRequester& nack_module = CreateNackModule();
-  nack_module.OnReceivedPacket(0, false, false);
-  nack_module.OnReceivedPacket(100, false, false);
+  nack_module.OnReceivedPacket(0);
+  nack_module.OnReceivedPacket(100);
   EXPECT_EQ(99u, sent_nacks_.size());
 }
 
@@ -389,14 +293,14 @@ class TestNackRequesterWithFieldTrial : public ::testing::Test,
 };
 
 TEST_F(TestNackRequesterWithFieldTrial, SendNackWithDelay) {
-  nack_module_.OnReceivedPacket(0, false, false);
-  nack_module_.OnReceivedPacket(100, false, false);
+  nack_module_.OnReceivedPacket(0);
+  nack_module_.OnReceivedPacket(100);
   EXPECT_EQ(0u, sent_nacks_.size());
   clock_->AdvanceTimeMilliseconds(10);
-  nack_module_.OnReceivedPacket(106, false, false);
+  nack_module_.OnReceivedPacket(106);
   EXPECT_EQ(99u, sent_nacks_.size());
   clock_->AdvanceTimeMilliseconds(10);
-  nack_module_.OnReceivedPacket(109, false, false);
+  nack_module_.OnReceivedPacket(109);
   EXPECT_EQ(104u, sent_nacks_.size());
 }
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/qp_parser.cc b/third_party/libwebrtc/modules/video_coding/utility/qp_parser.cc
index 3b9aaa377e..a531b54a7e 100644
--- a/third_party/libwebrtc/modules/video_coding/utility/qp_parser.cc
+++ b/third_party/libwebrtc/modules/video_coding/utility/qp_parser.cc
@@ -37,7 +37,10 @@ absl::optional<uint32_t> QpParser::Parse(VideoCodecType codec_type,
   } else if (codec_type == kVideoCodecH264) {
     return h264_parsers_[spatial_idx].Parse(frame_data, frame_size);
   } else if (codec_type == kVideoCodecH265) {
-    // TODO(bugs.webrtc.org/13485)
+    // H.265 bitstream parser is conditionally built.
+#ifdef RTC_ENABLE_H265
+    return h265_parsers_[spatial_idx].Parse(frame_data, frame_size);
+#endif
   }
 
   return absl::nullopt;
@@ -52,4 +55,15 @@ absl::optional<uint32_t> QpParser::H264QpParser::Parse(
   return bitstream_parser_.GetLastSliceQp();
 }
 
+#ifdef RTC_ENABLE_H265
+absl::optional<uint32_t> QpParser::H265QpParser::Parse(
+    const uint8_t* frame_data,
+    size_t frame_size) {
+  MutexLock lock(&mutex_);
+  bitstream_parser_.ParseBitstream(
+      rtc::ArrayView<const uint8_t>(frame_data, frame_size));
+  return bitstream_parser_.GetLastSliceQp();
+}
+#endif
+
 }  // namespace webrtc
diff --git a/third_party/libwebrtc/modules/video_coding/utility/qp_parser.h b/third_party/libwebrtc/modules/video_coding/utility/qp_parser.h
index f132ff9337..210fe02bc3 100644
--- a/third_party/libwebrtc/modules/video_coding/utility/qp_parser.h
+++ b/third_party/libwebrtc/modules/video_coding/utility/qp_parser.h
@@ -15,6 +15,9 @@
 #include "api/video/video_codec_constants.h"
 #include "api/video/video_codec_type.h"
 #include "common_video/h264/h264_bitstream_parser.h"
+#ifdef RTC_ENABLE_H265
+#include "common_video/h265/h265_bitstream_parser.h"
+#endif
 #include "rtc_base/synchronization/mutex.h"
 
 namespace webrtc {
@@ -38,6 +41,21 @@ class QpParser {
   };
 
   H264QpParser h264_parsers_[kMaxSimulcastStreams];
+
+#ifdef RTC_ENABLE_H265
+  // A thread safe wrapper for H.265 bitstream parser.
+  class H265QpParser {
+   public:
+    absl::optional<uint32_t> Parse(const uint8_t* frame_data,
+                                   size_t frame_size);
+
+   private:
+    Mutex mutex_;
+    H265BitstreamParser bitstream_parser_ RTC_GUARDED_BY(mutex_);
+  };
+
+  H265QpParser h265_parsers_[kMaxSimulcastStreams];
+#endif
 };
 
 }  // namespace webrtc