Adding upstream version 124.0.1.upstream/124.0.1

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

7 files changed, 1597 insertions, 0 deletions

diff --git a/third_party/libwebrtc/api/test/pclf/BUILD.gn b/third_party/libwebrtc/api/test/pclf/BUILD.gn
new file mode 100644
index 0000000000..372ff51f49
--- /dev/null
+++ b/third_party/libwebrtc/api/test/pclf/BUILD.gn
@@ -0,0 +1,122 @@
+# Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS.  All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+import("../../../webrtc.gni")
+
+rtc_source_set("media_configuration") {
+  visibility = [ "*" ]
+  testonly = true
+  sources = [
+    "media_configuration.cc",
+    "media_configuration.h",
+  ]
+
+  deps = [
+    "../..:array_view",
+    "../..:audio_options_api",
+    "../..:audio_quality_analyzer_api",
+    "../..:callfactory_api",
+    "../..:fec_controller_api",
+    "../..:frame_generator_api",
+    "../..:function_view",
+    "../..:libjingle_peerconnection_api",
+    "../..:media_stream_interface",
+    "../..:packet_socket_factory",
+    "../..:peer_network_dependencies",
+    "../..:rtp_parameters",
+    "../..:simulated_network_api",
+    "../..:stats_observer_interface",
+    "../..:track_id_stream_info_map",
+    "../..:video_quality_analyzer_api",
+    "../../../modules/audio_processing:api",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:network",
+    "../../../rtc_base:rtc_certificate_generator",
+    "../../../rtc_base:ssl",
+    "../../../rtc_base:stringutils",
+    "../../../rtc_base:threading",
+    "../../../test:fileutils",
+    "../../../test:video_test_support",
+    "../../../test/pc/e2e/analyzer/video:video_dumping",
+    "../../audio:audio_mixer_api",
+    "../../rtc_event_log",
+    "../../task_queue",
+    "../../transport:network_control",
+    "../../units:time_delta",
+    "../../video_codecs:video_codecs_api",
+    "../video:video_frame_writer",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/memory",
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+  ]
+}
+
+rtc_library("media_quality_test_params") {
+  visibility = [ "*" ]
+  testonly = true
+  sources = [ "media_quality_test_params.h" ]
+
+  deps = [
+    ":media_configuration",
+    "../..:async_dns_resolver",
+    "../../../api:fec_controller_api",
+    "../../../api:field_trials_view",
+    "../../../api:libjingle_peerconnection_api",
+    "../../../api:packet_socket_factory",
+    "../../../api/audio:audio_mixer_api",
+    "../../../api/rtc_event_log",
+    "../../../api/transport:network_control",
+    "../../../api/video_codecs:video_codecs_api",
+    "../../../modules/audio_processing:api",
+    "../../../p2p:rtc_p2p",
+    "../../../rtc_base:network",
+    "../../../rtc_base:rtc_certificate_generator",
+    "../../../rtc_base:ssl",
+    "../../../rtc_base:threading",
+  ]
+}
+
+rtc_library("peer_configurer") {
+  visibility = [ "*" ]
+  testonly = true
+  sources = [
+    "peer_configurer.cc",
+    "peer_configurer.h",
+  ]
+  deps = [
+    ":media_configuration",
+    ":media_quality_test_params",
+    "../..:async_dns_resolver",
+    "../../../api:create_peer_connection_quality_test_frame_generator",
+    "../../../api:fec_controller_api",
+    "../../../api:field_trials_view",
+    "../../../api:frame_generator_api",
+    "../../../api:ice_transport_interface",
+    "../../../api:libjingle_peerconnection_api",
+    "../../../api:peer_network_dependencies",
+    "../../../api:scoped_refptr",
+    "../../../api/audio:audio_mixer_api",
+    "../../../api/audio_codecs:audio_codecs_api",
+    "../../../api/neteq:neteq_api",
+    "../../../api/rtc_event_log",
+    "../../../api/transport:bitrate_settings",
+    "../../../api/transport:network_control",
+    "../../../api/video_codecs:video_codecs_api",
+    "../../../modules/audio_processing:api",
+    "../../../rtc_base:checks",
+    "../../../rtc_base:rtc_certificate_generator",
+    "../../../rtc_base:ssl",
+  ]
+  absl_deps = [
+    "//third_party/abseil-cpp/absl/strings",
+    "//third_party/abseil-cpp/absl/types:optional",
+    "//third_party/abseil-cpp/absl/types:variant",
+  ]
+}
diff --git a/third_party/libwebrtc/api/test/pclf/DEPS b/third_party/libwebrtc/api/test/pclf/DEPS
new file mode 100644
index 0000000000..60cc0aeeb3
--- /dev/null
+++ b/third_party/libwebrtc/api/test/pclf/DEPS
@@ -0,0 +1,13 @@
+specific_include_rules = {
+  ".*": [
+    "+modules/audio_processing/include/audio_processing.h",
+    "+rtc_base/checks.h",
+    "+rtc_base/network.h",
+    "+rtc_base/rtc_certificate_generator.h",
+    "+rtc_base/ssl_certificate.h",
+    "+rtc_base/thread.h",
+  ],
+  "media_quality_test_params\.h": [
+    "+p2p/base/port_allocator.h",
+  ],
+}
diff --git a/third_party/libwebrtc/api/test/pclf/media_configuration.cc b/third_party/libwebrtc/api/test/pclf/media_configuration.cc
new file mode 100644
index 0000000000..4446e11400
--- /dev/null
+++ b/third_party/libwebrtc/api/test/pclf/media_configuration.cc
@@ -0,0 +1,312 @@
+/*
+ *  Copyright 2022 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/test/pclf/media_configuration.h"
+
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/test/video/video_frame_writer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/strings/string_builder.h"
+#include "test/pc/e2e/analyzer/video/video_dumping.h"
+#include "test/testsupport/file_utils.h"
+#include "test/testsupport/video_frame_writer.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+namespace {
+
+absl::string_view SpecToString(VideoResolution::Spec spec) {
+  switch (spec) {
+    case VideoResolution::Spec::kNone:
+      return "None";
+    case VideoResolution::Spec::kMaxFromSender:
+      return "MaxFromSender";
+  }
+}
+
+void AppendResolution(const VideoResolution& resolution,
+                      rtc::StringBuilder& builder) {
+  builder << "_" << resolution.width() << "x" << resolution.height() << "_"
+          << resolution.fps();
+}
+
+}  // namespace
+
+ScreenShareConfig::ScreenShareConfig(TimeDelta slide_change_interval)
+    : slide_change_interval(slide_change_interval) {
+  RTC_CHECK_GT(slide_change_interval.ms(), 0);
+}
+VideoSimulcastConfig::VideoSimulcastConfig(int simulcast_streams_count)
+    : simulcast_streams_count(simulcast_streams_count) {
+  RTC_CHECK_GT(simulcast_streams_count, 1);
+}
+EmulatedSFUConfig::EmulatedSFUConfig(int target_layer_index)
+    : target_layer_index(target_layer_index) {
+  RTC_CHECK_GE(target_layer_index, 0);
+}
+
+EmulatedSFUConfig::EmulatedSFUConfig(absl::optional<int> target_layer_index,
+                                     absl::optional<int> target_temporal_index)
+    : target_layer_index(target_layer_index),
+      target_temporal_index(target_temporal_index) {
+  RTC_CHECK_GE(target_temporal_index.value_or(0), 0);
+  if (target_temporal_index)
+    RTC_CHECK_GE(*target_temporal_index, 0);
+}
+
+VideoResolution::VideoResolution(size_t width, size_t height, int32_t fps)
+    : width_(width), height_(height), fps_(fps), spec_(Spec::kNone) {}
+VideoResolution::VideoResolution(Spec spec)
+    : width_(0), height_(0), fps_(0), spec_(spec) {}
+
+bool VideoResolution::operator==(const VideoResolution& other) const {
+  if (spec_ != Spec::kNone && spec_ == other.spec_) {
+    // If there is some particular spec set, then it doesn't matter what
+    // values we have in other fields.
+    return true;
+  }
+  return width_ == other.width_ && height_ == other.height_ &&
+         fps_ == other.fps_ && spec_ == other.spec_;
+}
+bool VideoResolution::operator!=(const VideoResolution& other) const {
+  return !(*this == other);
+}
+
+bool VideoResolution::IsRegular() const {
+  return spec_ == Spec::kNone;
+}
+std::string VideoResolution::ToString() const {
+  rtc::StringBuilder out;
+  out << "{ width=" << width_ << ", height=" << height_ << ", fps=" << fps_
+      << ", spec=" << SpecToString(spec_) << " }";
+  return out.Release();
+}
+
+VideoDumpOptions::VideoDumpOptions(
+    absl::string_view output_directory,
+    int sampling_modulo,
+    bool export_frame_ids,
+    std::function<std::unique_ptr<test::VideoFrameWriter>(
+        absl::string_view file_name_prefix,
+        const VideoResolution& resolution)> video_frame_writer_factory)
+    : output_directory_(output_directory),
+      sampling_modulo_(sampling_modulo),
+      export_frame_ids_(export_frame_ids),
+      video_frame_writer_factory_(video_frame_writer_factory) {
+  RTC_CHECK_GT(sampling_modulo, 0);
+}
+
+VideoDumpOptions::VideoDumpOptions(absl::string_view output_directory,
+                                   bool export_frame_ids)
+    : VideoDumpOptions(output_directory,
+                       kDefaultSamplingModulo,
+                       export_frame_ids) {}
+
+std::unique_ptr<test::VideoFrameWriter>
+VideoDumpOptions::CreateInputDumpVideoFrameWriter(
+    absl::string_view stream_label,
+    const VideoResolution& resolution) const {
+  std::unique_ptr<test::VideoFrameWriter> writer = video_frame_writer_factory_(
+      GetInputDumpFileName(stream_label, resolution), resolution);
+  absl::optional<std::string> frame_ids_file =
+      GetInputFrameIdsDumpFileName(stream_label, resolution);
+  if (frame_ids_file.has_value()) {
+    writer = CreateVideoFrameWithIdsWriter(std::move(writer), *frame_ids_file);
+  }
+  return writer;
+}
+
+std::unique_ptr<test::VideoFrameWriter>
+VideoDumpOptions::CreateOutputDumpVideoFrameWriter(
+    absl::string_view stream_label,
+    absl::string_view receiver,
+    const VideoResolution& resolution) const {
+  std::unique_ptr<test::VideoFrameWriter> writer = video_frame_writer_factory_(
+      GetOutputDumpFileName(stream_label, receiver, resolution), resolution);
+  absl::optional<std::string> frame_ids_file =
+      GetOutputFrameIdsDumpFileName(stream_label, receiver, resolution);
+  if (frame_ids_file.has_value()) {
+    writer = CreateVideoFrameWithIdsWriter(std::move(writer), *frame_ids_file);
+  }
+  return writer;
+}
+
+std::unique_ptr<test::VideoFrameWriter>
+VideoDumpOptions::Y4mVideoFrameWriterFactory(
+    absl::string_view file_name_prefix,
+    const VideoResolution& resolution) {
+  return std::make_unique<test::Y4mVideoFrameWriterImpl>(
+      std::string(file_name_prefix) + ".y4m", resolution.width(),
+      resolution.height(), resolution.fps());
+}
+
+std::string VideoDumpOptions::GetInputDumpFileName(
+    absl::string_view stream_label,
+    const VideoResolution& resolution) const {
+  rtc::StringBuilder file_name;
+  file_name << stream_label;
+  AppendResolution(resolution, file_name);
+  return test::JoinFilename(output_directory_, file_name.Release());
+}
+
+absl::optional<std::string> VideoDumpOptions::GetInputFrameIdsDumpFileName(
+    absl::string_view stream_label,
+    const VideoResolution& resolution) const {
+  if (!export_frame_ids_) {
+    return absl::nullopt;
+  }
+  return GetInputDumpFileName(stream_label, resolution) + ".frame_ids.txt";
+}
+
+std::string VideoDumpOptions::GetOutputDumpFileName(
+    absl::string_view stream_label,
+    absl::string_view receiver,
+    const VideoResolution& resolution) const {
+  rtc::StringBuilder file_name;
+  file_name << stream_label << "_" << receiver;
+  AppendResolution(resolution, file_name);
+  return test::JoinFilename(output_directory_, file_name.Release());
+}
+
+absl::optional<std::string> VideoDumpOptions::GetOutputFrameIdsDumpFileName(
+    absl::string_view stream_label,
+    absl::string_view receiver,
+    const VideoResolution& resolution) const {
+  if (!export_frame_ids_) {
+    return absl::nullopt;
+  }
+  return GetOutputDumpFileName(stream_label, receiver, resolution) +
+         ".frame_ids.txt";
+}
+
+std::string VideoDumpOptions::ToString() const {
+  rtc::StringBuilder out;
+  out << "{ output_directory_=" << output_directory_
+      << ", sampling_modulo_=" << sampling_modulo_
+      << ", export_frame_ids_=" << export_frame_ids_ << " }";
+  return out.Release();
+}
+
+VideoConfig::VideoConfig(const VideoResolution& resolution)
+    : width(resolution.width()),
+      height(resolution.height()),
+      fps(resolution.fps()) {
+  RTC_CHECK(resolution.IsRegular());
+}
+VideoConfig::VideoConfig(size_t width, size_t height, int32_t fps)
+    : width(width), height(height), fps(fps) {}
+VideoConfig::VideoConfig(absl::string_view stream_label,
+                         size_t width,
+                         size_t height,
+                         int32_t fps)
+    : width(width), height(height), fps(fps), stream_label(stream_label) {}
+
+AudioConfig::AudioConfig(absl::string_view stream_label)
+    : stream_label(stream_label) {}
+
+VideoCodecConfig::VideoCodecConfig(absl::string_view name)
+    : name(name), required_params() {}
+
+VideoCodecConfig::VideoCodecConfig(
+    absl::string_view name,
+    std::map<std::string, std::string> required_params)
+    : name(name), required_params(std::move(required_params)) {}
+
+absl::optional<VideoResolution> VideoSubscription::GetMaxResolution(
+    rtc::ArrayView<const VideoConfig> video_configs) {
+  std::vector<VideoResolution> resolutions;
+  for (const auto& video_config : video_configs) {
+    resolutions.push_back(video_config.GetResolution());
+  }
+  return GetMaxResolution(resolutions);
+}
+
+absl::optional<VideoResolution> VideoSubscription::GetMaxResolution(
+    rtc::ArrayView<const VideoResolution> resolutions) {
+  if (resolutions.empty()) {
+    return absl::nullopt;
+  }
+
+  VideoResolution max_resolution;
+  for (const VideoResolution& resolution : resolutions) {
+    if (max_resolution.width() < resolution.width()) {
+      max_resolution.set_width(resolution.width());
+    }
+    if (max_resolution.height() < resolution.height()) {
+      max_resolution.set_height(resolution.height());
+    }
+    if (max_resolution.fps() < resolution.fps()) {
+      max_resolution.set_fps(resolution.fps());
+    }
+  }
+  return max_resolution;
+}
+
+bool VideoSubscription::operator==(const VideoSubscription& other) const {
+  return default_resolution_ == other.default_resolution_ &&
+         peers_resolution_ == other.peers_resolution_;
+}
+bool VideoSubscription::operator!=(const VideoSubscription& other) const {
+  return !(*this == other);
+}
+
+VideoSubscription& VideoSubscription::SubscribeToPeer(
+    absl::string_view peer_name,
+    VideoResolution resolution) {
+  peers_resolution_[std::string(peer_name)] = resolution;
+  return *this;
+}
+
+VideoSubscription& VideoSubscription::SubscribeToAllPeers(
+    VideoResolution resolution) {
+  default_resolution_ = resolution;
+  return *this;
+}
+
+absl::optional<VideoResolution> VideoSubscription::GetResolutionForPeer(
+    absl::string_view peer_name) const {
+  auto it = peers_resolution_.find(std::string(peer_name));
+  if (it == peers_resolution_.end()) {
+    return default_resolution_;
+  }
+  return it->second;
+}
+
+std::vector<std::string> VideoSubscription::GetSubscribedPeers() const {
+  std::vector<std::string> subscribed_streams;
+  subscribed_streams.reserve(peers_resolution_.size());
+  for (const auto& entry : peers_resolution_) {
+    subscribed_streams.push_back(entry.first);
+  }
+  return subscribed_streams;
+}
+
+std::string VideoSubscription::ToString() const {
+  rtc::StringBuilder out;
+  out << "{ default_resolution_=[";
+  if (default_resolution_.has_value()) {
+    out << default_resolution_->ToString();
+  } else {
+    out << "undefined";
+  }
+  out << "], {";
+  for (const auto& [peer_name, resolution] : peers_resolution_) {
+    out << "[" << peer_name << ": " << resolution.ToString() << "], ";
+  }
+  out << "} }";
+  return out.Release();
+}
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/test/pclf/media_configuration.h b/third_party/libwebrtc/api/test/pclf/media_configuration.h
new file mode 100644
index 0000000000..5bcb308c83
--- /dev/null
+++ b/third_party/libwebrtc/api/test/pclf/media_configuration.h
@@ -0,0 +1,478 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef API_TEST_PCLF_MEDIA_CONFIGURATION_H_
+#define API_TEST_PCLF_MEDIA_CONFIGURATION_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <functional>
+#include <map>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/memory/memory.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/async_resolver_factory.h"
+#include "api/audio/audio_mixer.h"
+#include "api/audio_options.h"
+#include "api/call/call_factory_interface.h"
+#include "api/fec_controller.h"
+#include "api/function_view.h"
+#include "api/media_stream_interface.h"
+#include "api/peer_connection_interface.h"
+#include "api/rtc_event_log/rtc_event_log_factory_interface.h"
+#include "api/rtp_parameters.h"
+#include "api/task_queue/task_queue_factory.h"
+#include "api/test/audio_quality_analyzer_interface.h"
+#include "api/test/frame_generator_interface.h"
+#include "api/test/peer_network_dependencies.h"
+#include "api/test/simulated_network.h"
+#include "api/test/stats_observer_interface.h"
+#include "api/test/track_id_stream_info_map.h"
+#include "api/test/video/video_frame_writer.h"
+#include "api/test/video_quality_analyzer_interface.h"
+#include "api/transport/network_control.h"
+#include "api/units/time_delta.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/network.h"
+#include "rtc_base/rtc_certificate_generator.h"
+#include "rtc_base/ssl_certificate.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+constexpr size_t kDefaultSlidesWidth = 1850;
+constexpr size_t kDefaultSlidesHeight = 1110;
+
+// The index of required capturing device in OS provided list of video
+// devices. On Linux and Windows the list will be obtained via
+// webrtc::VideoCaptureModule::DeviceInfo, on Mac OS via
+// [RTCCameraVideoCapturer captureDevices].
+enum class CapturingDeviceIndex : size_t {};
+
+// Contains parameters for screen share scrolling.
+//
+// If scrolling is enabled, then it will be done by putting sliding window
+// on source video and moving this window from top left corner to the
+// bottom right corner of the picture.
+//
+// In such case source dimensions must be greater or equal to the sliding
+// window dimensions. So `source_width` and `source_height` are the dimensions
+// of the source frame, while `VideoConfig::width` and `VideoConfig::height`
+// are the dimensions of the sliding window.
+//
+// Because `source_width` and `source_height` are dimensions of the source
+// frame, they have to be width and height of videos from
+// `ScreenShareConfig::slides_yuv_file_names`.
+//
+// Because scrolling have to be done on single slide it also requires, that
+// `duration` must be less or equal to
+// `ScreenShareConfig::slide_change_interval`.
+struct ScrollingParams {
+  // Duration of scrolling.
+  TimeDelta duration;
+  // Width of source slides video.
+  size_t source_width = kDefaultSlidesWidth;
+  // Height of source slides video.
+  size_t source_height = kDefaultSlidesHeight;
+};
+
+// Contains screen share video stream properties.
+struct ScreenShareConfig {
+  explicit ScreenShareConfig(TimeDelta slide_change_interval);
+
+  // Shows how long one slide should be presented on the screen during
+  // slide generation.
+  TimeDelta slide_change_interval;
+  // If true, slides will be generated programmatically. No scrolling params
+  // will be applied in such case.
+  bool generate_slides = false;
+  // If present scrolling will be applied. Please read extra requirement on
+  // `slides_yuv_file_names` for scrolling.
+  absl::optional<ScrollingParams> scrolling_params;
+  // Contains list of yuv files with slides.
+  //
+  // If empty, default set of slides will be used. In such case
+  // `VideoConfig::width` must be equal to `kDefaultSlidesWidth` and
+  // `VideoConfig::height` must be equal to `kDefaultSlidesHeight` or if
+  // `scrolling_params` are specified, then `ScrollingParams::source_width`
+  // must be equal to `kDefaultSlidesWidth` and
+  // `ScrollingParams::source_height` must be equal to `kDefaultSlidesHeight`.
+  std::vector<std::string> slides_yuv_file_names;
+};
+
+// Config for Vp8 simulcast or non-standard Vp9 SVC testing.
+//
+// To configure standard SVC setting, use `scalability_mode` in the
+// `encoding_params` array.
+// This configures Vp9 SVC by requesting simulcast layers, the request is
+// internally converted to a request for SVC layers.
+//
+// SVC support is limited:
+// During SVC testing there is no SFU, so framework will try to emulate SFU
+// behavior in regular p2p call. Because of it there are such limitations:
+//  * if `target_spatial_index` is not equal to the highest spatial layer
+//    then no packet/frame drops are allowed.
+//
+//    If there will be any drops, that will affect requested layer, then
+//    WebRTC SVC implementation will continue decoding only the highest
+//    available layer and won't restore lower layers, so analyzer won't
+//    receive required data which will cause wrong results or test failures.
+struct VideoSimulcastConfig {
+  explicit VideoSimulcastConfig(int simulcast_streams_count);
+
+  // Specified amount of simulcast streams/SVC layers, depending on which
+  // encoder is used.
+  int simulcast_streams_count;
+};
+
+// Configuration for the emulated Selective Forward Unit (SFU)
+//
+// The framework can optionally filter out frames that are decoded
+// using an emulated SFU.
+// When using simulcast or SVC, it's not always desirable to receive
+// all frames. In a real world call, a SFU will only forward a subset
+// of the frames.
+// The emulated SFU is not able to change its configuration dynamically,
+// if adaptation happens during the call, layers may be dropped and the
+// analyzer won't receive the required data which will cause wrong results or
+// test failures.
+struct EmulatedSFUConfig {
+  EmulatedSFUConfig() = default;
+  explicit EmulatedSFUConfig(int target_layer_index);
+  EmulatedSFUConfig(absl::optional<int> target_layer_index,
+                    absl::optional<int> target_temporal_index);
+
+  // Specifies simulcast or spatial index of the video stream to analyze.
+  // There are 2 cases:
+  // 1. simulcast encoding is used:
+  //    in such case `target_layer_index` will specify the index of
+  //    simulcast stream, that should be analyzed. Other streams will be
+  //    dropped.
+  // 2. SVC encoding is used:
+  //    in such case `target_layer_index` will specify the top interesting
+  //    spatial layer and all layers below, including target one will be
+  //    processed. All layers above target one will be dropped.
+  // If not specified then all streams will be received and analyzed.
+  // When set, it instructs the framework to create an emulated Selective
+  // Forwarding Unit (SFU) that will propagate only the requested layers.
+  absl::optional<int> target_layer_index;
+  // Specifies the index of the maximum temporal unit to keep.
+  // If not specified then all temporal layers will be received and analyzed.
+  // When set, it instructs the framework to create an emulated Selective
+  // Forwarding Unit (SFU) that will propagate only up to the requested layer.
+  absl::optional<int> target_temporal_index;
+};
+
+class VideoResolution {
+ public:
+  // Determines special resolutions, which can't be expressed in terms of
+  // width, height and fps.
+  enum class Spec {
+    // No extra spec set. It describes a regular resolution described by
+    // width, height and fps.
+    kNone,
+    // Describes resolution which contains max value among all sender's
+    // video streams in each dimension (width, height, fps).
+    kMaxFromSender
+  };
+
+  VideoResolution(size_t width, size_t height, int32_t fps);
+  explicit VideoResolution(Spec spec = Spec::kNone);
+
+  bool operator==(const VideoResolution& other) const;
+  bool operator!=(const VideoResolution& other) const;
+
+  size_t width() const { return width_; }
+  void set_width(size_t width) { width_ = width; }
+  size_t height() const { return height_; }
+  void set_height(size_t height) { height_ = height; }
+  int32_t fps() const { return fps_; }
+  void set_fps(int32_t fps) { fps_ = fps; }
+
+  // Returns if it is a regular resolution or not. The resolution is regular
+  // if it's spec is `Spec::kNone`.
+  bool IsRegular() const;
+
+  std::string ToString() const;
+
+ private:
+  size_t width_ = 0;
+  size_t height_ = 0;
+  int32_t fps_ = 0;
+  Spec spec_ = Spec::kNone;
+};
+
+class VideoDumpOptions {
+ public:
+  static constexpr int kDefaultSamplingModulo = 1;
+
+  // output_directory - the output directory where stream will be dumped. The
+  // output files' names will be constructed as
+  // <stream_name>_<receiver_name>_<resolution>.<extension> for output dumps
+  // and <stream_name>_<resolution>.<extension> for input dumps.
+  // By default <extension> is "y4m". Resolution is in the format
+  // <width>x<height>_<fps>.
+  // sampling_modulo - the module for the video frames to be dumped. Modulo
+  // equals X means every Xth frame will be written to the dump file. The
+  // value must be greater than 0. (Default: 1)
+  // export_frame_ids - specifies if frame ids should be exported together
+  // with content of the stream. If true, an output file with the same name as
+  // video dump and suffix ".frame_ids.txt" will be created. It will contain
+  // the frame ids in the same order as original frames in the output
+  // file with stream content. File will contain one frame id per line.
+  // (Default: false)
+  // `video_frame_writer_factory` - factory function to create a video frame
+  // writer for input and output video files. (Default: Y4M video writer
+  // factory).
+  explicit VideoDumpOptions(
+      absl::string_view output_directory,
+      int sampling_modulo = kDefaultSamplingModulo,
+      bool export_frame_ids = false,
+      std::function<std::unique_ptr<test::VideoFrameWriter>(
+          absl::string_view file_name_prefix,
+          const VideoResolution& resolution)> video_frame_writer_factory =
+          Y4mVideoFrameWriterFactory);
+  VideoDumpOptions(absl::string_view output_directory, bool export_frame_ids);
+
+  VideoDumpOptions(const VideoDumpOptions&) = default;
+  VideoDumpOptions& operator=(const VideoDumpOptions&) = default;
+  VideoDumpOptions(VideoDumpOptions&&) = default;
+  VideoDumpOptions& operator=(VideoDumpOptions&&) = default;
+
+  std::string output_directory() const { return output_directory_; }
+  int sampling_modulo() const { return sampling_modulo_; }
+  bool export_frame_ids() const { return export_frame_ids_; }
+
+  std::unique_ptr<test::VideoFrameWriter> CreateInputDumpVideoFrameWriter(
+      absl::string_view stream_label,
+      const VideoResolution& resolution) const;
+
+  std::unique_ptr<test::VideoFrameWriter> CreateOutputDumpVideoFrameWriter(
+      absl::string_view stream_label,
+      absl::string_view receiver,
+      const VideoResolution& resolution) const;
+
+  std::string ToString() const;
+
+ private:
+  static std::unique_ptr<test::VideoFrameWriter> Y4mVideoFrameWriterFactory(
+      absl::string_view file_name_prefix,
+      const VideoResolution& resolution);
+  std::string GetInputDumpFileName(absl::string_view stream_label,
+                                   const VideoResolution& resolution) const;
+  // Returns file name for input frame ids dump if `export_frame_ids()` is
+  // true, absl::nullopt otherwise.
+  absl::optional<std::string> GetInputFrameIdsDumpFileName(
+      absl::string_view stream_label,
+      const VideoResolution& resolution) const;
+  std::string GetOutputDumpFileName(absl::string_view stream_label,
+                                    absl::string_view receiver,
+                                    const VideoResolution& resolution) const;
+  // Returns file name for output frame ids dump if `export_frame_ids()` is
+  // true, absl::nullopt otherwise.
+  absl::optional<std::string> GetOutputFrameIdsDumpFileName(
+      absl::string_view stream_label,
+      absl::string_view receiver,
+      const VideoResolution& resolution) const;
+
+  std::string output_directory_;
+  int sampling_modulo_ = 1;
+  bool export_frame_ids_ = false;
+  std::function<std::unique_ptr<test::VideoFrameWriter>(
+      absl::string_view file_name_prefix,
+      const VideoResolution& resolution)>
+      video_frame_writer_factory_;
+};
+
+// Contains properties of single video stream.
+struct VideoConfig {
+  explicit VideoConfig(const VideoResolution& resolution);
+  VideoConfig(size_t width, size_t height, int32_t fps);
+  VideoConfig(absl::string_view stream_label,
+              size_t width,
+              size_t height,
+              int32_t fps);
+
+  // Video stream width.
+  size_t width;
+  // Video stream height.
+  size_t height;
+  int32_t fps;
+  VideoResolution GetResolution() const {
+    return VideoResolution(width, height, fps);
+  }
+
+  // Have to be unique among all specified configs for all peers in the call.
+  // Will be auto generated if omitted.
+  absl::optional<std::string> stream_label;
+  // Will be set for current video track. If equals to kText or kDetailed -
+  // screencast in on.
+  absl::optional<VideoTrackInterface::ContentHint> content_hint;
+  // If presented video will be transfered in simulcast/SVC mode depending on
+  // which encoder is used.
+  //
+  // Simulcast is supported only from 1st added peer. For VP8 simulcast only
+  // without RTX is supported so it will be automatically disabled for all
+  // simulcast tracks. For VP9 simulcast enables VP9 SVC mode and support RTX,
+  // but only on non-lossy networks. See more in documentation to
+  // VideoSimulcastConfig.
+  absl::optional<VideoSimulcastConfig> simulcast_config;
+  // Configuration for the emulated Selective Forward Unit (SFU).
+  absl::optional<EmulatedSFUConfig> emulated_sfu_config;
+  // Encoding parameters for both singlecast and per simulcast layer.
+  // If singlecast is used, if not empty, a single value can be provided.
+  // If simulcast is used, if not empty, `encoding_params` size have to be
+  // equal to `simulcast_config.simulcast_streams_count`. Will be used to set
+  // transceiver send encoding params for each layer.
+  // RtpEncodingParameters::rid may be changed by fixture implementation to
+  // ensure signaling correctness.
+  std::vector<RtpEncodingParameters> encoding_params;
+  // Count of temporal layers for video stream. This value will be set into
+  // each RtpEncodingParameters of RtpParameters of corresponding
+  // RtpSenderInterface for this video stream.
+  absl::optional<int> temporal_layers_count;
+  // If specified defines how input should be dumped. It is actually one of
+  // the test's output file, which contains copy of what was captured during
+  // the test for this video stream on sender side. It is useful when
+  // generator is used as input.
+  absl::optional<VideoDumpOptions> input_dump_options;
+  // If specified defines how output should be dumped on the receiver side for
+  // this stream. The produced files contain what was rendered for this video
+  // stream on receiver side per each receiver.
+  absl::optional<VideoDumpOptions> output_dump_options;
+  // If set to true uses fixed frame rate while dumping output video to the
+  // file. Requested `VideoSubscription::fps()` will be used as frame rate.
+  bool output_dump_use_fixed_framerate = false;
+  // If true will display input and output video on the user's screen.
+  bool show_on_screen = false;
+  // If specified, determines a sync group to which this video stream belongs.
+  // According to bugs.webrtc.org/4762 WebRTC supports synchronization only
+  // for pair of single audio and single video stream.
+  absl::optional<std::string> sync_group;
+  // If specified, it will be set into RtpParameters of corresponding
+  // RtpSenderInterface for this video stream.
+  // Note that this setting takes precedence over `content_hint`.
+  absl::optional<DegradationPreference> degradation_preference;
+};
+
+// Contains properties for audio in the call.
+struct AudioConfig {
+  AudioConfig() = default;
+  explicit AudioConfig(absl::string_view stream_label);
+
+  // Have to be unique among all specified configs for all peers in the call.
+  // Will be auto generated if omitted.
+  absl::optional<std::string> stream_label;
+  // If no file is specified an audio will be generated.
+  absl::optional<std::string> input_file_name;
+  // If specified the input stream will be also copied to specified file.
+  absl::optional<std::string> input_dump_file_name;
+  // If specified the output stream will be copied to specified file.
+  absl::optional<std::string> output_dump_file_name;
+
+  // Audio options to use.
+  cricket::AudioOptions audio_options;
+  // Sampling frequency of input audio data (from file or generated).
+  int sampling_frequency_in_hz = 48000;
+  // If specified, determines a sync group to which this audio stream belongs.
+  // According to bugs.webrtc.org/4762 WebRTC supports synchronization only
+  // for pair of single audio and single video stream.
+  absl::optional<std::string> sync_group;
+};
+
+struct VideoCodecConfig {
+  explicit VideoCodecConfig(absl::string_view name);
+  VideoCodecConfig(absl::string_view name,
+                   std::map<std::string, std::string> required_params);
+  // Next two fields are used to specify concrete video codec, that should be
+  // used in the test. Video code will be negotiated in SDP during offer/
+  // answer exchange.
+  // Video codec name. You can find valid names in
+  // media/base/media_constants.h
+  std::string name;
+  // Map of parameters, that have to be specified on SDP codec. Each parameter
+  // is described by key and value. Codec parameters will match the specified
+  // map if and only if for each key from `required_params` there will be
+  // a parameter with name equal to this key and parameter value will be equal
+  // to the value from `required_params` for this key.
+  // If empty then only name will be used to match the codec.
+  std::map<std::string, std::string> required_params;
+};
+
+// Subscription to the remote video streams. It declares which remote stream
+// peer should receive and in which resolution (width x height x fps).
+class VideoSubscription {
+ public:
+  // Returns the resolution constructed as maximum from all resolution
+  // dimensions: width, height and fps.
+  static absl::optional<VideoResolution> GetMaxResolution(
+      rtc::ArrayView<const VideoConfig> video_configs);
+  static absl::optional<VideoResolution> GetMaxResolution(
+      rtc::ArrayView<const VideoResolution> resolutions);
+
+  bool operator==(const VideoSubscription& other) const;
+  bool operator!=(const VideoSubscription& other) const;
+
+  // Subscribes receiver to all streams sent by the specified peer with
+  // specified resolution. It will override any resolution that was used in
+  // `SubscribeToAll` independently from methods call order.
+  VideoSubscription& SubscribeToPeer(
+      absl::string_view peer_name,
+      VideoResolution resolution =
+          VideoResolution(VideoResolution::Spec::kMaxFromSender));
+
+  // Subscribes receiver to the all sent streams with specified resolution.
+  // If any stream was subscribed to with `SubscribeTo` method that will
+  // override resolution passed to this function independently from methods
+  // call order.
+  VideoSubscription& SubscribeToAllPeers(
+      VideoResolution resolution =
+          VideoResolution(VideoResolution::Spec::kMaxFromSender));
+
+  // Returns resolution for specific sender. If no specific resolution was
+  // set for this sender, then will return resolution used for all streams.
+  // If subscription doesn't subscribe to all streams, `absl::nullopt` will be
+  // returned.
+  absl::optional<VideoResolution> GetResolutionForPeer(
+      absl::string_view peer_name) const;
+
+  // Returns a maybe empty list of senders for which peer explicitly
+  // subscribed to with specific resolution.
+  std::vector<std::string> GetSubscribedPeers() const;
+
+  std::string ToString() const;
+
+ private:
+  absl::optional<VideoResolution> default_resolution_ = absl::nullopt;
+  std::map<std::string, VideoResolution> peers_resolution_;
+};
+
+// Contains configuration for echo emulator.
+struct EchoEmulationConfig {
+  // Delay which represents the echo path delay, i.e. how soon rendered signal
+  // should reach capturer.
+  TimeDelta echo_delay = TimeDelta::Millis(50);
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // API_TEST_PCLF_MEDIA_CONFIGURATION_H_
diff --git a/third_party/libwebrtc/api/test/pclf/media_quality_test_params.h b/third_party/libwebrtc/api/test/pclf/media_quality_test_params.h
new file mode 100644
index 0000000000..b2ccdf18c5
--- /dev/null
+++ b/third_party/libwebrtc/api/test/pclf/media_quality_test_params.h
@@ -0,0 +1,188 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef API_TEST_PCLF_MEDIA_QUALITY_TEST_PARAMS_H_
+#define API_TEST_PCLF_MEDIA_QUALITY_TEST_PARAMS_H_
+
+#include <cstddef>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "api/async_dns_resolver.h"
+#include "api/audio/audio_mixer.h"
+#include "api/fec_controller.h"
+#include "api/field_trials_view.h"
+#include "api/rtc_event_log/rtc_event_log_factory_interface.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/transport/network_control.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "p2p/base/port_allocator.h"
+#include "rtc_base/network.h"
+#include "rtc_base/rtc_certificate_generator.h"
+#include "rtc_base/ssl_certificate.h"
+#include "rtc_base/thread.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// Contains most part from PeerConnectionFactoryDependencies. Also all fields
+// are optional and defaults will be provided by fixture implementation if
+// any will be omitted.
+//
+// Separate class was introduced to clarify which components can be
+// overridden. For example worker and signaling threads will be provided by
+// fixture implementation. The same is applicable to the media engine. So user
+// can override only some parts of media engine like video encoder/decoder
+// factories.
+struct PeerConnectionFactoryComponents {
+  std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory;
+  std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory;
+  std::unique_ptr<NetworkControllerFactoryInterface> network_controller_factory;
+  std::unique_ptr<NetEqFactory> neteq_factory;
+
+  // Will be passed to MediaEngineInterface, that will be used in
+  // PeerConnectionFactory.
+  std::unique_ptr<VideoEncoderFactory> video_encoder_factory;
+  std::unique_ptr<VideoDecoderFactory> video_decoder_factory;
+  rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory;
+  rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory;
+
+  std::unique_ptr<FieldTrialsView> trials;
+
+  rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing;
+  rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer;
+};
+
+// Contains most parts from PeerConnectionDependencies. Also all fields are
+// optional and defaults will be provided by fixture implementation if any
+// will be omitted.
+//
+// Separate class was introduced to clarify which components can be
+// overridden. For example observer, which is required to
+// PeerConnectionDependencies, will be provided by fixture implementation,
+// so client can't inject its own. Also only network manager can be overridden
+// inside port allocator.
+struct PeerConnectionComponents {
+  PeerConnectionComponents(rtc::NetworkManager* network_manager,
+                           rtc::PacketSocketFactory* packet_socket_factory)
+      : network_manager(network_manager),
+        packet_socket_factory(packet_socket_factory) {
+    RTC_CHECK(network_manager);
+  }
+
+  rtc::NetworkManager* const network_manager;
+  rtc::PacketSocketFactory* const packet_socket_factory;
+  std::unique_ptr<webrtc::AsyncDnsResolverFactoryInterface>
+      async_dns_resolver_factory;
+  std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator;
+  std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier;
+  std::unique_ptr<IceTransportFactory> ice_transport_factory;
+};
+
+// Contains all components, that can be overridden in peer connection. Also
+// has a network thread, that will be used to communicate with another peers.
+struct InjectableComponents {
+  InjectableComponents(rtc::Thread* network_thread,
+                       rtc::NetworkManager* network_manager,
+                       rtc::PacketSocketFactory* packet_socket_factory)
+      : network_thread(network_thread),
+        worker_thread(nullptr),
+        pcf_dependencies(std::make_unique<PeerConnectionFactoryComponents>()),
+        pc_dependencies(
+            std::make_unique<PeerConnectionComponents>(network_manager,
+                                                       packet_socket_factory)) {
+    RTC_CHECK(network_thread);
+  }
+
+  rtc::Thread* const network_thread;
+  rtc::Thread* worker_thread;
+
+  std::unique_ptr<PeerConnectionFactoryComponents> pcf_dependencies;
+  std::unique_ptr<PeerConnectionComponents> pc_dependencies;
+};
+
+// Contains information about call media streams (up to 1 audio stream and
+// unlimited amount of video streams) and rtc configuration, that will be used
+// to set up peer connection.
+struct Params {
+  // Peer name. If empty - default one will be set by the fixture.
+  absl::optional<std::string> name;
+  // If `audio_config` is set audio stream will be configured
+  absl::optional<AudioConfig> audio_config;
+  // Flags to set on `cricket::PortAllocator`. These flags will be added
+  // to the default ones that are presented on the port allocator.
+  uint32_t port_allocator_extra_flags = cricket::kDefaultPortAllocatorFlags;
+  // If `rtc_event_log_path` is set, an RTCEventLog will be saved in that
+  // location and it will be available for further analysis.
+  absl::optional<std::string> rtc_event_log_path;
+  // If `aec_dump_path` is set, an AEC dump will be saved in that location and
+  // it will be available for further analysis.
+  absl::optional<std::string> aec_dump_path;
+
+  bool use_ulp_fec = false;
+  bool use_flex_fec = false;
+  // Specifies how much video encoder target bitrate should be different than
+  // target bitrate, provided by WebRTC stack. Must be greater then 0. Can be
+  // used to emulate overshooting of video encoders. This multiplier will
+  // be applied for all video encoder on both sides for all layers. Bitrate
+  // estimated by WebRTC stack will be multiplied by this multiplier and then
+  // provided into VideoEncoder::SetRates(...).
+  double video_encoder_bitrate_multiplier = 1.0;
+
+  PeerConnectionInterface::RTCConfiguration rtc_configuration;
+  PeerConnectionInterface::RTCOfferAnswerOptions rtc_offer_answer_options;
+  BitrateSettings bitrate_settings;
+  std::vector<VideoCodecConfig> video_codecs;
+
+  // A list of RTP header extensions which will be enforced on all video streams
+  // added to this peer.
+  std::vector<std::string> extra_video_rtp_header_extensions;
+  // A list of RTP header extensions which will be enforced on all audio streams
+  // added to this peer.
+  std::vector<std::string> extra_audio_rtp_header_extensions;
+};
+
+// Contains parameters that maybe changed by test writer during the test call.
+struct ConfigurableParams {
+  // If `video_configs` is empty - no video should be added to the test call.
+  std::vector<VideoConfig> video_configs;
+
+  VideoSubscription video_subscription =
+      VideoSubscription().SubscribeToAllPeers();
+};
+
+// Contains parameters, that describe how long framework should run quality
+// test.
+struct RunParams {
+  explicit RunParams(TimeDelta run_duration) : run_duration(run_duration) {}
+
+  // Specifies how long the test should be run. This time shows how long
+  // the media should flow after connection was established and before
+  // it will be shut downed.
+  TimeDelta run_duration;
+
+  // If set to true peers will be able to use Flex FEC, otherwise they won't
+  // be able to negotiate it even if it's enabled on per peer level.
+  bool enable_flex_fec_support = false;
+  // If true will set conference mode in SDP media section for all video
+  // tracks for all peers.
+  bool use_conference_mode = false;
+  // If specified echo emulation will be done, by mixing the render audio into
+  // the capture signal. In such case input signal will be reduced by half to
+  // avoid saturation or compression in the echo path simulation.
+  absl::optional<EchoEmulationConfig> echo_emulation_config;
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // API_TEST_PCLF_MEDIA_QUALITY_TEST_PARAMS_H_
diff --git a/third_party/libwebrtc/api/test/pclf/peer_configurer.cc b/third_party/libwebrtc/api/test/pclf/peer_configurer.cc
new file mode 100644
index 0000000000..5e385452b1
--- /dev/null
+++ b/third_party/libwebrtc/api/test/pclf/peer_configurer.cc
@@ -0,0 +1,276 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/test/pclf/peer_configurer.h"
+
+#include <cstdint>
+#include <memory>
+#include <string>
+#include <utility>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/async_dns_resolver.h"
+#include "api/audio/audio_mixer.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/fec_controller.h"
+#include "api/field_trials_view.h"
+#include "api/ice_transport_interface.h"
+#include "api/neteq/neteq_factory.h"
+#include "api/peer_connection_interface.h"
+#include "api/rtc_event_log/rtc_event_log_factory_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/test/create_peer_connection_quality_test_frame_generator.h"
+#include "api/test/frame_generator_interface.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/pclf/media_quality_test_params.h"
+#include "api/test/peer_network_dependencies.h"
+#include "api/transport/bitrate_settings.h"
+#include "api/transport/network_control.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/rtc_certificate_generator.h"
+#include "rtc_base/ssl_certificate.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+PeerConfigurer::PeerConfigurer(
+    const PeerNetworkDependencies& network_dependencies)
+    : components_(std::make_unique<InjectableComponents>(
+          network_dependencies.network_thread,
+          network_dependencies.network_manager,
+          network_dependencies.packet_socket_factory)),
+      params_(std::make_unique<Params>()),
+      configurable_params_(std::make_unique<ConfigurableParams>()) {}
+
+PeerConfigurer* PeerConfigurer::SetName(absl::string_view name) {
+  params_->name = std::string(name);
+  return this;
+}
+
+PeerConfigurer* PeerConfigurer::SetEventLogFactory(
+    std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory) {
+  components_->pcf_dependencies->event_log_factory =
+      std::move(event_log_factory);
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetFecControllerFactory(
+    std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory) {
+  components_->pcf_dependencies->fec_controller_factory =
+      std::move(fec_controller_factory);
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetNetworkControllerFactory(
+    std::unique_ptr<NetworkControllerFactoryInterface>
+        network_controller_factory) {
+  components_->pcf_dependencies->network_controller_factory =
+      std::move(network_controller_factory);
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetVideoEncoderFactory(
+    std::unique_ptr<VideoEncoderFactory> video_encoder_factory) {
+  components_->pcf_dependencies->video_encoder_factory =
+      std::move(video_encoder_factory);
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetVideoDecoderFactory(
+    std::unique_ptr<VideoDecoderFactory> video_decoder_factory) {
+  components_->pcf_dependencies->video_decoder_factory =
+      std::move(video_decoder_factory);
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetAudioEncoderFactory(
+    rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory) {
+  components_->pcf_dependencies->audio_encoder_factory = audio_encoder_factory;
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetAudioDecoderFactory(
+    rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory) {
+  components_->pcf_dependencies->audio_decoder_factory = audio_decoder_factory;
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetAsyncDnsResolverFactory(
+    std::unique_ptr<webrtc::AsyncDnsResolverFactoryInterface>
+        async_dns_resolver_factory) {
+  components_->pc_dependencies->async_dns_resolver_factory =
+      std::move(async_dns_resolver_factory);
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetRTCCertificateGenerator(
+    std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator) {
+  components_->pc_dependencies->cert_generator = std::move(cert_generator);
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetSSLCertificateVerifier(
+    std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier) {
+  components_->pc_dependencies->tls_cert_verifier =
+      std::move(tls_cert_verifier);
+  return this;
+}
+
+PeerConfigurer* PeerConfigurer::AddVideoConfig(VideoConfig config) {
+  video_sources_.push_back(
+      CreateSquareFrameGenerator(config, /*type=*/absl::nullopt));
+  configurable_params_->video_configs.push_back(std::move(config));
+  return this;
+}
+PeerConfigurer* PeerConfigurer::AddVideoConfig(
+    VideoConfig config,
+    std::unique_ptr<test::FrameGeneratorInterface> generator) {
+  configurable_params_->video_configs.push_back(std::move(config));
+  video_sources_.push_back(std::move(generator));
+  return this;
+}
+PeerConfigurer* PeerConfigurer::AddVideoConfig(VideoConfig config,
+                                               CapturingDeviceIndex index) {
+  configurable_params_->video_configs.push_back(std::move(config));
+  video_sources_.push_back(index);
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetVideoSubscription(
+    VideoSubscription subscription) {
+  configurable_params_->video_subscription = std::move(subscription);
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetVideoCodecs(
+    std::vector<VideoCodecConfig> video_codecs) {
+  params_->video_codecs = std::move(video_codecs);
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetExtraVideoRtpHeaderExtensions(
+    std::vector<std::string> extensions) {
+  params_->extra_video_rtp_header_extensions = std::move(extensions);
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetAudioConfig(AudioConfig config) {
+  params_->audio_config = std::move(config);
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetExtraAudioRtpHeaderExtensions(
+    std::vector<std::string> extensions) {
+  params_->extra_audio_rtp_header_extensions = std::move(extensions);
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetUseUlpFEC(bool value) {
+  params_->use_ulp_fec = value;
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetUseFlexFEC(bool value) {
+  params_->use_flex_fec = value;
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetVideoEncoderBitrateMultiplier(
+    double multiplier) {
+  params_->video_encoder_bitrate_multiplier = multiplier;
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetNetEqFactory(
+    std::unique_ptr<NetEqFactory> neteq_factory) {
+  components_->pcf_dependencies->neteq_factory = std::move(neteq_factory);
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetAudioProcessing(
+    rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing) {
+  components_->pcf_dependencies->audio_processing = audio_processing;
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetAudioMixer(
+    rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) {
+  components_->pcf_dependencies->audio_mixer = audio_mixer;
+  return this;
+}
+
+PeerConfigurer* PeerConfigurer::SetUseNetworkThreadAsWorkerThread() {
+  components_->worker_thread = components_->network_thread;
+  return this;
+}
+
+PeerConfigurer* PeerConfigurer::SetRtcEventLogPath(absl::string_view path) {
+  params_->rtc_event_log_path = std::string(path);
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetAecDumpPath(absl::string_view path) {
+  params_->aec_dump_path = std::string(path);
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetRTCConfiguration(
+    PeerConnectionInterface::RTCConfiguration configuration) {
+  params_->rtc_configuration = std::move(configuration);
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetRTCOfferAnswerOptions(
+    PeerConnectionInterface::RTCOfferAnswerOptions options) {
+  params_->rtc_offer_answer_options = std::move(options);
+  return this;
+}
+PeerConfigurer* PeerConfigurer::SetBitrateSettings(
+    BitrateSettings bitrate_settings) {
+  params_->bitrate_settings = bitrate_settings;
+  return this;
+}
+
+PeerConfigurer* PeerConfigurer::SetIceTransportFactory(
+    std::unique_ptr<IceTransportFactory> factory) {
+  components_->pc_dependencies->ice_transport_factory = std::move(factory);
+  return this;
+}
+
+PeerConfigurer* PeerConfigurer::SetFieldTrials(
+    std::unique_ptr<FieldTrialsView> field_trials) {
+  components_->pcf_dependencies->trials = std::move(field_trials);
+  return this;
+}
+
+PeerConfigurer* PeerConfigurer::SetPortAllocatorExtraFlags(
+    uint32_t extra_flags) {
+  params_->port_allocator_extra_flags = extra_flags;
+  return this;
+}
+std::unique_ptr<InjectableComponents> PeerConfigurer::ReleaseComponents() {
+  RTC_CHECK(components_);
+  auto components = std::move(components_);
+  components_ = nullptr;
+  return components;
+}
+
+// Returns Params and transfer ownership to the caller.
+// Can be called once.
+std::unique_ptr<Params> PeerConfigurer::ReleaseParams() {
+  RTC_CHECK(params_);
+  auto params = std::move(params_);
+  params_ = nullptr;
+  return params;
+}
+
+// Returns ConfigurableParams and transfer ownership to the caller.
+// Can be called once.
+std::unique_ptr<ConfigurableParams>
+PeerConfigurer::ReleaseConfigurableParams() {
+  RTC_CHECK(configurable_params_);
+  auto configurable_params = std::move(configurable_params_);
+  configurable_params_ = nullptr;
+  return configurable_params;
+}
+
+// Returns video sources and transfer frame generators ownership to the
+// caller. Can be called once.
+std::vector<PeerConfigurer::VideoSource> PeerConfigurer::ReleaseVideoSources() {
+  auto video_sources = std::move(video_sources_);
+  video_sources_.clear();
+  return video_sources;
+}
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
diff --git a/third_party/libwebrtc/api/test/pclf/peer_configurer.h b/third_party/libwebrtc/api/test/pclf/peer_configurer.h
new file mode 100644
index 0000000000..c0faf8573a
--- /dev/null
+++ b/third_party/libwebrtc/api/test/pclf/peer_configurer.h
@@ -0,0 +1,208 @@
+/*
+ *  Copyright (c) 2022 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef API_TEST_PCLF_PEER_CONFIGURER_H_
+#define API_TEST_PCLF_PEER_CONFIGURER_H_
+
+#include <cstdint>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/variant.h"
+#include "api/async_dns_resolver.h"
+#include "api/audio/audio_mixer.h"
+#include "api/audio_codecs/audio_decoder_factory.h"
+#include "api/audio_codecs/audio_encoder_factory.h"
+#include "api/fec_controller.h"
+#include "api/field_trials_view.h"
+#include "api/ice_transport_interface.h"
+#include "api/neteq/neteq_factory.h"
+#include "api/peer_connection_interface.h"
+#include "api/rtc_event_log/rtc_event_log_factory_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/test/frame_generator_interface.h"
+#include "api/test/pclf/media_configuration.h"
+#include "api/test/pclf/media_quality_test_params.h"
+#include "api/test/peer_network_dependencies.h"
+#include "api/transport/bitrate_settings.h"
+#include "api/transport/network_control.h"
+#include "api/video_codecs/video_decoder_factory.h"
+#include "api/video_codecs/video_encoder_factory.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "rtc_base/rtc_certificate_generator.h"
+#include "rtc_base/ssl_certificate.h"
+
+namespace webrtc {
+namespace webrtc_pc_e2e {
+
+// This class is used to fully configure one peer inside a call.
+class PeerConfigurer {
+ public:
+  using VideoSource =
+      absl::variant<std::unique_ptr<test::FrameGeneratorInterface>,
+                    CapturingDeviceIndex>;
+
+  explicit PeerConfigurer(const PeerNetworkDependencies& network_dependencies);
+
+  // Sets peer name that will be used to report metrics related to this peer.
+  // If not set, some default name will be assigned. All names have to be
+  // unique.
+  PeerConfigurer* SetName(absl::string_view name);
+
+  // The parameters of the following 7 methods will be passed to the
+  // PeerConnectionFactoryInterface implementation that will be created for
+  // this peer.
+  PeerConfigurer* SetEventLogFactory(
+      std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory);
+  PeerConfigurer* SetFecControllerFactory(
+      std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory);
+  PeerConfigurer* SetNetworkControllerFactory(
+      std::unique_ptr<NetworkControllerFactoryInterface>
+          network_controller_factory);
+  PeerConfigurer* SetVideoEncoderFactory(
+      std::unique_ptr<VideoEncoderFactory> video_encoder_factory);
+  PeerConfigurer* SetVideoDecoderFactory(
+      std::unique_ptr<VideoDecoderFactory> video_decoder_factory);
+  PeerConfigurer* SetAudioEncoderFactory(
+      rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory);
+  PeerConfigurer* SetAudioDecoderFactory(
+      rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory);
+  // Set a custom NetEqFactory to be used in the call.
+  PeerConfigurer* SetNetEqFactory(std::unique_ptr<NetEqFactory> neteq_factory);
+  PeerConfigurer* SetAudioProcessing(
+      rtc::scoped_refptr<webrtc::AudioProcessing> audio_processing);
+  PeerConfigurer* SetAudioMixer(
+      rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer);
+
+  // Forces the Peerconnection to use the network thread as the worker thread.
+  // Ie, worker thread and the network thread is the same thread.
+  PeerConfigurer* SetUseNetworkThreadAsWorkerThread();
+
+  // The parameters of the following 4 methods will be passed to the
+  // PeerConnectionInterface implementation that will be created for this
+  // peer.
+  PeerConfigurer* SetAsyncDnsResolverFactory(
+      std::unique_ptr<webrtc::AsyncDnsResolverFactoryInterface>
+          async_dns_resolver_factory);
+  PeerConfigurer* SetRTCCertificateGenerator(
+      std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator);
+  PeerConfigurer* SetSSLCertificateVerifier(
+      std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier);
+  PeerConfigurer* SetIceTransportFactory(
+      std::unique_ptr<IceTransportFactory> factory);
+  // Flags to set on `cricket::PortAllocator`. These flags will be added
+  // to the default ones that are presented on the port allocator.
+  // For possible values check p2p/base/port_allocator.h.
+  PeerConfigurer* SetPortAllocatorExtraFlags(uint32_t extra_flags);
+
+  // Add new video stream to the call that will be sent from this peer.
+  // Default implementation of video frames generator will be used.
+  PeerConfigurer* AddVideoConfig(VideoConfig config);
+  // Add new video stream to the call that will be sent from this peer with
+  // provided own implementation of video frames generator.
+  PeerConfigurer* AddVideoConfig(
+      VideoConfig config,
+      std::unique_ptr<test::FrameGeneratorInterface> generator);
+  // Add new video stream to the call that will be sent from this peer.
+  // Capturing device with specified index will be used to get input video.
+  PeerConfigurer* AddVideoConfig(VideoConfig config,
+                                 CapturingDeviceIndex capturing_device_index);
+  // Sets video subscription for the peer. By default subscription will
+  // include all streams with `VideoSubscription::kSameAsSendStream`
+  // resolution. To this behavior use this method.
+  PeerConfigurer* SetVideoSubscription(VideoSubscription subscription);
+  // Sets the list of video codecs used by the peer during the test. These
+  // codecs will be negotiated in SDP during offer/answer exchange. The order
+  // of these codecs during negotiation will be the same as in `video_codecs`.
+  // Codecs have to be available in codecs list provided by peer connection to
+  // be negotiated. If some of specified codecs won't be found, the test will
+  // crash.
+  PeerConfigurer* SetVideoCodecs(std::vector<VideoCodecConfig> video_codecs);
+  // Sets a list of RTP header extensions which will be enforced on all video
+  // streams added to this peer.
+  PeerConfigurer* SetExtraVideoRtpHeaderExtensions(
+      std::vector<std::string> extensions);
+  // Sets the audio stream for the call from this peer. If this method won't
+  // be invoked, this peer will send no audio.
+  PeerConfigurer* SetAudioConfig(AudioConfig config);
+  // Sets a list of RTP header extensions which will be enforced on all audio
+  // streams added to this peer.
+  PeerConfigurer* SetExtraAudioRtpHeaderExtensions(
+      std::vector<std::string> extensions);
+
+  // Set if ULP FEC should be used or not. False by default.
+  PeerConfigurer* SetUseUlpFEC(bool value);
+  // Set if Flex FEC should be used or not. False by default.
+  // Client also must enable `enable_flex_fec_support` in the `RunParams` to
+  // be able to use this feature.
+  PeerConfigurer* SetUseFlexFEC(bool value);
+  // Specifies how much video encoder target bitrate should be different than
+  // target bitrate, provided by WebRTC stack. Must be greater than 0. Can be
+  // used to emulate overshooting of video encoders. This multiplier will
+  // be applied for all video encoder on both sides for all layers. Bitrate
+  // estimated by WebRTC stack will be multiplied by this multiplier and then
+  // provided into VideoEncoder::SetRates(...). 1.0 by default.
+  PeerConfigurer* SetVideoEncoderBitrateMultiplier(double multiplier);
+
+  // If is set, an RTCEventLog will be saved in that location and it will be
+  // available for further analysis.
+  PeerConfigurer* SetRtcEventLogPath(absl::string_view path);
+  // If is set, an AEC dump will be saved in that location and it will be
+  // available for further analysis.
+  PeerConfigurer* SetAecDumpPath(absl::string_view path);
+  PeerConfigurer* SetRTCConfiguration(
+      PeerConnectionInterface::RTCConfiguration configuration);
+  PeerConfigurer* SetRTCOfferAnswerOptions(
+      PeerConnectionInterface::RTCOfferAnswerOptions options);
+  // Set bitrate parameters on PeerConnection. This constraints will be
+  // applied to all summed RTP streams for this peer.
+  PeerConfigurer* SetBitrateSettings(BitrateSettings bitrate_settings);
+  // Set field trials used for this PeerConnection.
+  PeerConfigurer* SetFieldTrials(std::unique_ptr<FieldTrialsView> field_trials);
+
+  // Returns InjectableComponents and transfer ownership to the caller.
+  // Can be called once.
+  std::unique_ptr<InjectableComponents> ReleaseComponents();
+
+  // Returns Params and transfer ownership to the caller.
+  // Can be called once.
+  std::unique_ptr<Params> ReleaseParams();
+
+  // Returns ConfigurableParams and transfer ownership to the caller.
+  // Can be called once.
+  std::unique_ptr<ConfigurableParams> ReleaseConfigurableParams();
+
+  // Returns video sources and transfer frame generators ownership to the
+  // caller. Can be called once.
+  std::vector<VideoSource> ReleaseVideoSources();
+
+  InjectableComponents* components() { return components_.get(); }
+  Params* params() { return params_.get(); }
+  ConfigurableParams* configurable_params() {
+    return configurable_params_.get();
+  }
+  const Params& params() const { return *params_; }
+  const ConfigurableParams& configurable_params() const {
+    return *configurable_params_;
+  }
+  std::vector<VideoSource>* video_sources() { return &video_sources_; }
+
+ private:
+  std::unique_ptr<InjectableComponents> components_;
+  std::unique_ptr<Params> params_;
+  std::unique_ptr<ConfigurableParams> configurable_params_;
+  std::vector<VideoSource> video_sources_;
+};
+
+}  // namespace webrtc_pc_e2e
+}  // namespace webrtc
+
+#endif  // API_TEST_PCLF_PEER_CONFIGURER_H_